diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 4603515..a7ac2d7 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -73,7 +73,7 @@
 static bool gMonotonicTime = false;     // use system monotonic time for timestamps
 static bool gPersistentSurface = false; // use persistent surface
 static enum {
-    FORMAT_MP4, FORMAT_H264, FORMAT_FRAMES, FORMAT_RAW_FRAMES
+    FORMAT_MP4, FORMAT_H264, FORMAT_WEBM, FORMAT_3GPP, FORMAT_FRAMES, FORMAT_RAW_FRAMES
 } gOutputFormat = FORMAT_MP4;           // data format for output
 static AString gCodecName = "";         // codec name override
 static bool gSizeSpecified = false;     // was size explicitly requested?
@@ -669,7 +669,9 @@
     sp<MediaMuxer> muxer = NULL;
     FILE* rawFp = NULL;
     switch (gOutputFormat) {
-        case FORMAT_MP4: {
+        case FORMAT_MP4:
+        case FORMAT_WEBM:
+        case FORMAT_3GPP: {
             // Configure muxer.  We have to wait for the CSD blob from the encoder
             // before we can start it.
             err = unlink(fileName);
@@ -682,7 +684,13 @@
                 fprintf(stderr, "ERROR: couldn't open file\n");
                 abort();
             }
-            muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+            if (gOutputFormat == FORMAT_MP4) {
+                muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_MPEG_4);
+            } else if (gOutputFormat == FORMAT_WEBM) {
+                muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_WEBM);
+            } else {
+                muxer = new MediaMuxer(fd, MediaMuxer::OUTPUT_FORMAT_THREE_GPP);
+            }
             close(fd);
             if (gRotate) {
                 muxer->setOrientationHint(90);  // TODO: does this do anything?
@@ -1002,6 +1010,10 @@
                 gOutputFormat = FORMAT_MP4;
             } else if (strcmp(optarg, "h264") == 0) {
                 gOutputFormat = FORMAT_H264;
+            } else if (strcmp(optarg, "webm") == 0) {
+                gOutputFormat = FORMAT_WEBM;
+            } else if (strcmp(optarg, "3gpp") == 0) {
+                gOutputFormat = FORMAT_3GPP;
             } else if (strcmp(optarg, "frames") == 0) {
                 gOutputFormat = FORMAT_FRAMES;
             } else if (strcmp(optarg, "raw-frames") == 0) {
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index 61fc897..85aab57 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -78,6 +78,7 @@
 static bool gPlaybackAudio;
 static bool gWriteMP4;
 static bool gDisplayHistogram;
+static bool gVerbose = false;
 static bool showProgress = true;
 static String8 gWriteMP4Filename;
 static String8 gComponentNameOverride;
@@ -159,6 +160,11 @@
             break;
         }
 
+        if (gVerbose) {
+            MetaDataBase &meta = mbuf->meta_data();
+            fprintf(stdout, "sample format: %s\n", meta.toString().c_str());
+        }
+
         CHECK_EQ(
                 fwrite((const uint8_t *)mbuf->data() + mbuf->range_offset(),
                        1,
@@ -579,12 +585,12 @@
                 break;
             }
 
+            CHECK(buffer != NULL);
+
             if (buffer->range_length() > 0) {
                 break;
             }
 
-            CHECK(buffer != NULL);
-
             buffer->release();
             buffer = NULL;
         }
@@ -630,6 +636,7 @@
     fprintf(stderr, "       -T allocate buffers from a surface texture\n");
     fprintf(stderr, "       -d(ump) output_filename (raw stream data to a file)\n");
     fprintf(stderr, "       -D(ump) output_filename (decoded PCM data to a file)\n");
+    fprintf(stderr, "       -v be more verbose\n");
 }
 
 static void dumpCodecProfiles(bool queryDecoders) {
@@ -708,7 +715,7 @@
     sp<ALooper> looper;
 
     int res;
-    while ((res = getopt(argc, argv, "haqn:lm:b:ptsrow:kN:xSTd:D:")) >= 0) {
+    while ((res = getopt(argc, argv, "vhaqn:lm:b:ptsrow:kN:xSTd:D:")) >= 0) {
         switch (res) {
             case 'a':
             {
@@ -832,6 +839,12 @@
                 break;
             }
 
+            case 'v':
+            {
+                gVerbose = true;
+                break;
+            }
+
             case '?':
             case 'h':
             default:
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index ca119d5..5f19f74 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -538,6 +538,10 @@
         mTimestampMutator.push(timestamp);
     }
 
+    virtual ExtendedTimestamp getTimestamp() const {
+        return mTimestampMutator.last();
+    }
+
     // Flushes the shared ring buffer if the client had requested it using mStreaming.mFlush.
     // If flush occurs then:
     //   cblk->u.mStreaming.mFront, ServerProxy::mFlush and ServerProxy::mFlushed will be modified
@@ -551,6 +555,9 @@
     // Total count of the number of flushed frames since creation (never reset).
     virtual int64_t     framesFlushed() const { return mFlushed; }
 
+    // Safe frames ready query with no side effects.
+    virtual size_t      framesReadySafe() const = 0;
+
     // Get dynamic buffer size from the shared control block.
     uint32_t            getBufferSizeInFrames() const {
         return android_atomic_acquire_load((int32_t *)&mCblk->mBufferSizeInFrames);
@@ -588,8 +595,7 @@
     // which may include non-contiguous frames
     virtual size_t      framesReady();
 
-    // Safe frames ready query used by dump() - this has no side effects.
-    virtual size_t      framesReadySafe() const;
+    size_t              framesReadySafe() const override; // frames available to read by server.
 
     // Currently AudioFlinger will call framesReady() for a fast track from two threads:
     // FastMixer thread, and normal mixer thread.  This is dangerous, as the proxy is intended
@@ -693,6 +699,8 @@
         return mCblk->u.mStreaming.mRear; // For completeness only; mRear written by server.
     }
 
+    size_t framesReadySafe() const override; // frames available to read by client.
+
 protected:
     virtual ~AudioRecordServerProxy() { }
 };
diff --git a/media/OWNERS b/media/OWNERS
index 1f687a2..1e2d123 100644
--- a/media/OWNERS
+++ b/media/OWNERS
@@ -2,8 +2,10 @@
 dwkang@google.com
 elaurent@google.com
 essick@google.com
+gkasten@google.com
 hkuang@google.com
 hunga@google.com
+jiabin@google.com
 jmtrivi@google.com
 krocard@google.com
 lajos@google.com
diff --git a/media/audioserver/audioserver.rc b/media/audioserver/audioserver.rc
index 1f2e82f..f1e815b 100644
--- a/media/audioserver/audioserver.rc
+++ b/media/audioserver/audioserver.rc
@@ -7,6 +7,7 @@
     ioprio rt 4
     writepid /dev/cpuset/foreground/tasks /dev/stune/foreground/tasks
     onrestart restart vendor.audio-hal-2-0
+    onrestart restart vendor.audio-hal-4-0-msd
     # Keep the original service name for backward compatibility when upgrading
     # O-MR1 devices with framework-only.
     onrestart restart audio-hal-2-0
diff --git a/media/extractors/mp4/ItemTable.cpp b/media/extractors/mp4/ItemTable.cpp
index ca9deab..be442e6 100644
--- a/media/extractors/mp4/ItemTable.cpp
+++ b/media/extractors/mp4/ItemTable.cpp
@@ -1529,12 +1529,16 @@
         if (thumbItemIndex >= 0) {
             const ImageItem &thumbnail = mItemIdToItemMap[thumbItemIndex];
 
-            meta->setInt32(kKeyThumbnailWidth, thumbnail.width);
-            meta->setInt32(kKeyThumbnailHeight, thumbnail.height);
-            meta->setData(kKeyThumbnailHVCC, kTypeHVCC,
-                    thumbnail.hvcc->data(), thumbnail.hvcc->size());
-            ALOGV("image[%u]: thumbnail: size %dx%d, item index %zd",
-                    imageIndex, thumbnail.width, thumbnail.height, thumbItemIndex);
+            if (thumbnail.hvcc != NULL) {
+                meta->setInt32(kKeyThumbnailWidth, thumbnail.width);
+                meta->setInt32(kKeyThumbnailHeight, thumbnail.height);
+                meta->setData(kKeyThumbnailHVCC, kTypeHVCC,
+                        thumbnail.hvcc->data(), thumbnail.hvcc->size());
+                ALOGV("image[%u]: thumbnail: size %dx%d, item index %zd",
+                        imageIndex, thumbnail.width, thumbnail.height, thumbItemIndex);
+            } else {
+                ALOGW("%s: thumbnail data is missing for image[%u]!", __FUNCTION__, imageIndex);
+            }
         } else {
             ALOGW("%s: Referenced thumbnail does not exist!", __FUNCTION__);
         }
diff --git a/media/libaaudio/examples/utils/AAudioArgsParser.h b/media/libaaudio/examples/utils/AAudioArgsParser.h
index 88d7401..0e61589 100644
--- a/media/libaaudio/examples/utils/AAudioArgsParser.h
+++ b/media/libaaudio/examples/utils/AAudioArgsParser.h
@@ -272,7 +272,9 @@
                     if (strlen(arg) > 2) {
                         policy = atoi(&arg[2]);
                     }
-                    AAudio_setMMapPolicy(policy);
+                    if (!AAudio_setMMapPolicy(policy)) {
+                        printf("ERROR: invalid MMAP policy mode %i\n", policy);
+                    }
                 } break;
                 case 'n':
                     setNumberOfBursts(atoi(&arg[2]));
@@ -363,7 +365,7 @@
                 mode = AAUDIO_PERFORMANCE_MODE_POWER_SAVING;
                 break;
             default:
-                printf("ERROR invalid performance mode %c\n", c);
+                printf("ERROR: invalid performance mode %c\n", c);
                 break;
         }
         return mode;
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 3a7a578..4ef765d 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -150,6 +150,11 @@
         allowMMap = false;
     }
 
+    if (!allowMMap && !allowLegacy) {
+        ALOGE("%s() no backend available: neither MMAP nor legacy path are allowed", __func__);
+        return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+    }
+
     result = builder_createStream(getDirection(), sharingMode, allowMMap, &audioStream);
     if (result == AAUDIO_OK) {
         // Open the stream using the parameters from the builder.
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 2df37a8..6146c0e 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -49,6 +49,7 @@
         "libaudiomanager",
         "libmedia_helper",
         "libmediametrics",
+        "libmediautils",
     ],
     export_shared_lib_headers: ["libbinder"],
 
diff --git a/media/libaudioclient/AudioTrackShared.cpp b/media/libaudioclient/AudioTrackShared.cpp
index dced3c4..a018b22 100644
--- a/media/libaudioclient/AudioTrackShared.cpp
+++ b/media/libaudioclient/AudioTrackShared.cpp
@@ -1231,6 +1231,21 @@
     return 0;
 }
 
+__attribute__((no_sanitize("integer")))
+size_t AudioRecordServerProxy::framesReadySafe() const
+{
+    if (mIsShutdown) {
+        return 0;
+    }
+    const int32_t front = android_atomic_acquire_load(&mCblk->u.mStreaming.mFront);
+    const int32_t rear = mCblk->u.mStreaming.mRear;
+    const ssize_t filled = rear - front;
+    if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
+        return 0; // error condition, silently return 0.
+    }
+    return filled;
+}
+
 // ---------------------------------------------------------------------------
 
 }   // namespace android
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 00af7e8..37c62a8 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -24,10 +24,8 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/Parcel.h>
-#include <cutils/multiuser.h>
 #include <media/TimeCheck.h>
-#include <private/android_filesystem_config.h>
-
+#include <mediautils/ServiceUtilities.h>
 #include "IAudioFlinger.h"
 
 namespace android {
@@ -912,7 +910,7 @@
         case SET_MIC_MUTE:
         case SET_LOW_RAM_DEVICE:
         case SYSTEM_READY: {
-            if (multiuser_get_app_id(IPCThreadState::self()->getCallingUid()) >= AID_APP_START) {
+            if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
                 ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
                       __func__, code, IPCThreadState::self()->getCallingPid(),
                       IPCThreadState::self()->getCallingUid());
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index a1236e7..316105c 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -24,11 +24,10 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/Parcel.h>
-#include <cutils/multiuser.h>
 #include <media/AudioEffect.h>
 #include <media/IAudioPolicyService.h>
 #include <media/TimeCheck.h>
-#include <private/android_filesystem_config.h>
+#include <mediautils/ServiceUtilities.h>
 #include <system/audio.h>
 
 namespace android {
@@ -936,7 +935,7 @@
         case STOP_AUDIO_SOURCE:
         case GET_SURROUND_FORMATS:
         case SET_SURROUND_FORMAT_ENABLED: {
-            if (multiuser_get_app_id(IPCThreadState::self()->getCallingUid()) >= AID_APP_START) {
+            if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
                 ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
                       __func__, code, IPCThreadState::self()->getCallingPid(),
                       IPCThreadState::self()->getCallingUid());
diff --git a/media/libaudioclient/include/media/AudioParameter.h b/media/libaudioclient/include/media/AudioParameter.h
index 967d895..24837e3 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libaudioclient/include/media/AudioParameter.h
@@ -64,6 +64,9 @@
     static const char * const keyPresentationId;
     static const char * const keyProgramId;
 
+    //  keyAudioLanguagePreferred: Preferred audio language
+    static const char * const keyAudioLanguagePreferred;
+
     //  keyStreamConnect / Disconnect: value is an int in audio_devices_t
     static const char * const keyStreamConnect;
     static const char * const keyStreamDisconnect;
diff --git a/media/libaudioclient/include/media/AudioTimestamp.h b/media/libaudioclient/include/media/AudioTimestamp.h
index 498de8e..e5925dd 100644
--- a/media/libaudioclient/include/media/AudioTimestamp.h
+++ b/media/libaudioclient/include/media/AudioTimestamp.h
@@ -135,8 +135,23 @@
         return INVALID_OPERATION;
     }
 
+    double getOutputServerLatencyMs(uint32_t sampleRate) const {
+        return getLatencyMs(sampleRate, LOCATION_SERVER, LOCATION_KERNEL);
+    }
+
+    double getLatencyMs(uint32_t sampleRate, Location location1, Location location2) const {
+        if (sampleRate > 0 && mTimeNs[location1] > 0 && mTimeNs[location2] > 0) {
+            const int64_t frameDifference =
+                    mPosition[location1] - mPosition[location2];
+            const int64_t timeDifferenceNs =
+                    mTimeNs[location1] - mTimeNs[location2];
+            return ((double)frameDifference * 1e9 / sampleRate - timeDifferenceNs) * 1e-6;
+        }
+        return 0.;
+    }
+
     // convert fields to a printable string
-    std::string toString() {
+    std::string toString() const {
         std::stringstream ss;
 
         ss << "BOOTTIME offset " << mTimebaseOffset[TIMEBASE_BOOTTIME] << "\n";
diff --git a/media/libaudiohal/2.0/Android.bp b/media/libaudiohal/2.0/Android.bp
deleted file mode 100644
index 574b435..0000000
--- a/media/libaudiohal/2.0/Android.bp
+++ /dev/null
@@ -1,54 +0,0 @@
-cc_library_shared {
-    name: "libaudiohal@2.0",
-
-    srcs: [
-        "DeviceHalLocal.cpp",
-        "DevicesFactoryHalHybrid.cpp",
-        "DevicesFactoryHalLocal.cpp",
-        "StreamHalLocal.cpp",
-
-        "ConversionHelperHidl.cpp",
-        "DeviceHalHidl.cpp",
-        "DevicesFactoryHalHidl.cpp",
-        "EffectBufferHalHidl.cpp",
-        "EffectHalHidl.cpp",
-        "EffectsFactoryHalHidl.cpp",
-        "StreamHalHidl.cpp",
-    ],
-
-    export_include_dirs: ["."],
-
-    cflags: [
-        "-Wall",
-        "-Werror",
-    ],
-    shared_libs: [
-        "libaudiohal_deathhandler",
-        "libaudioutils",
-        "libcutils",
-        "liblog",
-        "libutils",
-        "libhardware",
-        "libbase",
-        "libfmq",
-        "libhwbinder",
-        "libhidlbase",
-        "libhidlmemory",
-        "libhidltransport",
-        "android.hardware.audio@2.0",
-        "android.hardware.audio.common@2.0",
-        "android.hardware.audio.common@2.0-util",
-        "android.hardware.audio.effect@2.0",
-        "android.hidl.allocator@1.0",
-        "android.hidl.memory@1.0",
-        "libmedia_helper",
-        "libmediautils",
-    ],
-    header_libs: [
-        "libaudiohal_headers"
-    ],
-
-    export_shared_lib_headers: [
-        "libfmq",
-    ],
-}
diff --git a/media/libaudiohal/2.0/ConversionHelperHidl.cpp b/media/libaudiohal/2.0/ConversionHelperHidl.cpp
deleted file mode 100644
index f60bf8b..0000000
--- a/media/libaudiohal/2.0/ConversionHelperHidl.cpp
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <string.h>
-
-#define LOG_TAG "HalHidl"
-#include <media/AudioParameter.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-
-using ::android::hardware::audio::V2_0::Result;
-
-namespace android {
-
-// static
-status_t ConversionHelperHidl::keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys) {
-    AudioParameter halKeys(keys);
-    if (halKeys.size() == 0) return BAD_VALUE;
-    hidlKeys->resize(halKeys.size());
-    //FIXME:  keyStreamSupportedChannels and keyStreamSupportedSamplingRates come with a
-    // "keyFormat=<value>" pair. We need to transform it into a single key string so that it is
-    // carried over to the legacy HAL via HIDL.
-    String8 value;
-    bool keepFormatValue = halKeys.size() == 2 &&
-         (halKeys.get(String8(AudioParameter::keyStreamSupportedChannels), value) == NO_ERROR ||
-         halKeys.get(String8(AudioParameter::keyStreamSupportedSamplingRates), value) == NO_ERROR);
-
-    for (size_t i = 0; i < halKeys.size(); ++i) {
-        String8 key;
-        status_t status = halKeys.getAt(i, key);
-        if (status != OK) return status;
-        if (keepFormatValue && key == AudioParameter::keyFormat) {
-            AudioParameter formatParam;
-            halKeys.getAt(i, key, value);
-            formatParam.add(key, value);
-            key = formatParam.toString();
-        }
-        (*hidlKeys)[i] = key.string();
-    }
-    return OK;
-}
-
-// static
-status_t ConversionHelperHidl::parametersFromHal(
-        const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams) {
-    AudioParameter params(kvPairs);
-    if (params.size() == 0) return BAD_VALUE;
-    hidlParams->resize(params.size());
-    for (size_t i = 0; i < params.size(); ++i) {
-        String8 key, value;
-        status_t status = params.getAt(i, key, value);
-        if (status != OK) return status;
-        (*hidlParams)[i].key = key.string();
-        (*hidlParams)[i].value = value.string();
-    }
-    return OK;
-}
-
-// static
-void ConversionHelperHidl::parametersToHal(
-        const hidl_vec<ParameterValue>& parameters, String8 *values) {
-    AudioParameter params;
-    for (size_t i = 0; i < parameters.size(); ++i) {
-        params.add(String8(parameters[i].key.c_str()), String8(parameters[i].value.c_str()));
-    }
-    values->setTo(params.toString());
-}
-
-ConversionHelperHidl::ConversionHelperHidl(const char* className)
-        : mClassName(className) {
-}
-
-// static
-status_t ConversionHelperHidl::analyzeResult(const Result& result) {
-    switch (result) {
-        case Result::OK: return OK;
-        case Result::INVALID_ARGUMENTS: return BAD_VALUE;
-        case Result::INVALID_STATE: return NOT_ENOUGH_DATA;
-        case Result::NOT_INITIALIZED: return NO_INIT;
-        case Result::NOT_SUPPORTED: return INVALID_OPERATION;
-        default: return NO_INIT;
-    }
-}
-
-void ConversionHelperHidl::emitError(const char* funcName, const char* description) {
-    ALOGE("%s %p %s: %s (from rpc)", mClassName, this, funcName, description);
-}
-
-}  // namespace android
diff --git a/media/libaudiohal/2.0/DeviceHalHidl.cpp b/media/libaudiohal/2.0/DeviceHalHidl.cpp
deleted file mode 100644
index 5b99d70..0000000
--- a/media/libaudiohal/2.0/DeviceHalHidl.cpp
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <stdio.h>
-
-#define LOG_TAG "DeviceHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/2.0/IPrimaryDevice.h>
-#include <cutils/native_handle.h>
-#include <hwbinder/IPCThreadState.h>
-#include <utils/Log.h>
-
-#include "DeviceHalHidl.h"
-#include "HidlUtils.h"
-#include "StreamHalHidl.h"
-
-using ::android::hardware::audio::common::V2_0::AudioConfig;
-using ::android::hardware::audio::common::V2_0::AudioDevice;
-using ::android::hardware::audio::common::V2_0::AudioInputFlag;
-using ::android::hardware::audio::common::V2_0::AudioOutputFlag;
-using ::android::hardware::audio::common::V2_0::AudioPatchHandle;
-using ::android::hardware::audio::common::V2_0::AudioPort;
-using ::android::hardware::audio::common::V2_0::AudioPortConfig;
-using ::android::hardware::audio::common::V2_0::AudioMode;
-using ::android::hardware::audio::common::V2_0::AudioSource;
-using ::android::hardware::audio::common::V2_0::HidlUtils;
-using ::android::hardware::audio::V2_0::DeviceAddress;
-using ::android::hardware::audio::V2_0::IPrimaryDevice;
-using ::android::hardware::audio::V2_0::ParameterValue;
-using ::android::hardware::audio::V2_0::Result;
-using ::android::hardware::hidl_string;
-using ::android::hardware::hidl_vec;
-
-namespace android {
-
-namespace {
-
-status_t deviceAddressFromHal(
-        audio_devices_t device, const char* halAddress, DeviceAddress* address) {
-    address->device = AudioDevice(device);
-
-    if (halAddress == nullptr || strnlen(halAddress, AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0) {
-        return OK;
-    }
-    const bool isInput = (device & AUDIO_DEVICE_BIT_IN) != 0;
-    if (isInput) device &= ~AUDIO_DEVICE_BIT_IN;
-    if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_A2DP) != 0)
-            || (isInput && (device & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) != 0)) {
-        int status = sscanf(halAddress,
-                "%hhX:%hhX:%hhX:%hhX:%hhX:%hhX",
-                &address->address.mac[0], &address->address.mac[1], &address->address.mac[2],
-                &address->address.mac[3], &address->address.mac[4], &address->address.mac[5]);
-        return status == 6 ? OK : BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_IP) != 0)
-            || (isInput && (device & AUDIO_DEVICE_IN_IP) != 0)) {
-        int status = sscanf(halAddress,
-                "%hhu.%hhu.%hhu.%hhu",
-                &address->address.ipv4[0], &address->address.ipv4[1],
-                &address->address.ipv4[2], &address->address.ipv4[3]);
-        return status == 4 ? OK : BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_ALL_USB)) != 0
-            || (isInput && (device & AUDIO_DEVICE_IN_ALL_USB)) != 0) {
-        int status = sscanf(halAddress,
-                "card=%d;device=%d",
-                &address->address.alsa.card, &address->address.alsa.device);
-        return status == 2 ? OK : BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_BUS) != 0)
-            || (isInput && (device & AUDIO_DEVICE_IN_BUS) != 0)) {
-        if (halAddress != NULL) {
-            address->busAddress = halAddress;
-            return OK;
-        }
-        return BAD_VALUE;
-    } else if ((!isInput && (device & AUDIO_DEVICE_OUT_REMOTE_SUBMIX)) != 0
-            || (isInput && (device & AUDIO_DEVICE_IN_REMOTE_SUBMIX) != 0)) {
-        if (halAddress != NULL) {
-            address->rSubmixAddress = halAddress;
-            return OK;
-        }
-        return BAD_VALUE;
-    }
-    return OK;
-}
-
-}  // namespace
-
-DeviceHalHidl::DeviceHalHidl(const sp<IDevice>& device)
-        : ConversionHelperHidl("Device"), mDevice(device),
-          mPrimaryDevice(IPrimaryDevice::castFrom(device)) {
-}
-
-DeviceHalHidl::~DeviceHalHidl() {
-    if (mDevice != 0) {
-        mDevice.clear();
-        hardware::IPCThreadState::self()->flushCommands();
-    }
-}
-
-status_t DeviceHalHidl::getSupportedDevices(uint32_t*) {
-    // Obsolete.
-    return INVALID_OPERATION;
-}
-
-status_t DeviceHalHidl::initCheck() {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("initCheck", mDevice->initCheck());
-}
-
-status_t DeviceHalHidl::setVoiceVolume(float volume) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    return processReturn("setVoiceVolume", mPrimaryDevice->setVoiceVolume(volume));
-}
-
-status_t DeviceHalHidl::setMasterVolume(float volume) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    return processReturn("setMasterVolume", mPrimaryDevice->setMasterVolume(volume));
-}
-
-status_t DeviceHalHidl::getMasterVolume(float *volume) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    Result retval;
-    Return<void> ret = mPrimaryDevice->getMasterVolume(
-            [&](Result r, float v) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *volume = v;
-                }
-            });
-    return processReturn("getMasterVolume", ret, retval);
-}
-
-status_t DeviceHalHidl::setMode(audio_mode_t mode) {
-    if (mDevice == 0) return NO_INIT;
-    if (mPrimaryDevice == 0) return INVALID_OPERATION;
-    return processReturn("setMode", mPrimaryDevice->setMode(AudioMode(mode)));
-}
-
-status_t DeviceHalHidl::setMicMute(bool state) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("setMicMute", mDevice->setMicMute(state));
-}
-
-status_t DeviceHalHidl::getMicMute(bool *state) {
-    if (mDevice == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mDevice->getMicMute(
-            [&](Result r, bool mute) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *state = mute;
-                }
-            });
-    return processReturn("getMicMute", ret, retval);
-}
-
-status_t DeviceHalHidl::setMasterMute(bool state) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("setMasterMute", mDevice->setMasterMute(state));
-}
-
-status_t DeviceHalHidl::getMasterMute(bool *state) {
-    if (mDevice == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mDevice->getMasterMute(
-            [&](Result r, bool mute) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *state = mute;
-                }
-            });
-    return processReturn("getMasterMute", ret, retval);
-}
-
-status_t DeviceHalHidl::setParameters(const String8& kvPairs) {
-    if (mDevice == 0) return NO_INIT;
-    hidl_vec<ParameterValue> hidlParams;
-    status_t status = parametersFromHal(kvPairs, &hidlParams);
-    if (status != OK) return status;
-    return processReturn("setParameters", mDevice->setParameters(hidlParams));
-}
-
-status_t DeviceHalHidl::getParameters(const String8& keys, String8 *values) {
-    values->clear();
-    if (mDevice == 0) return NO_INIT;
-    hidl_vec<hidl_string> hidlKeys;
-    status_t status = keysFromHal(keys, &hidlKeys);
-    if (status != OK) return status;
-    Result retval;
-    Return<void> ret = mDevice->getParameters(
-            hidlKeys,
-            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
-                retval = r;
-                if (retval == Result::OK) {
-                    parametersToHal(parameters, values);
-                }
-            });
-    return processReturn("getParameters", ret, retval);
-}
-
-status_t DeviceHalHidl::getInputBufferSize(
-        const struct audio_config *config, size_t *size) {
-    if (mDevice == 0) return NO_INIT;
-    AudioConfig hidlConfig;
-    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
-    Result retval;
-    Return<void> ret = mDevice->getInputBufferSize(
-            hidlConfig,
-            [&](Result r, uint64_t bufferSize) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *size = static_cast<size_t>(bufferSize);
-                }
-            });
-    return processReturn("getInputBufferSize", ret, retval);
-}
-
-status_t DeviceHalHidl::openOutputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        audio_output_flags_t flags,
-        struct audio_config *config,
-        const char *address,
-        sp<StreamOutHalInterface> *outStream) {
-    if (mDevice == 0) return NO_INIT;
-    DeviceAddress hidlDevice;
-    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
-    if (status != OK) return status;
-    AudioConfig hidlConfig;
-    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevice->openOutputStream(
-            handle,
-            hidlDevice,
-            hidlConfig,
-            AudioOutputFlag(flags),
-            [&](Result r, const sp<IStreamOut>& result, const AudioConfig& suggestedConfig) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *outStream = new StreamOutHalHidl(result);
-                }
-                HidlUtils::audioConfigToHal(suggestedConfig, config);
-            });
-    return processReturn("openOutputStream", ret, retval);
-}
-
-status_t DeviceHalHidl::openInputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        struct audio_config *config,
-        audio_input_flags_t flags,
-        const char *address,
-        audio_source_t source,
-        sp<StreamInHalInterface> *inStream) {
-    if (mDevice == 0) return NO_INIT;
-    DeviceAddress hidlDevice;
-    status_t status = deviceAddressFromHal(devices, address, &hidlDevice);
-    if (status != OK) return status;
-    AudioConfig hidlConfig;
-    HidlUtils::audioConfigFromHal(*config, &hidlConfig);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevice->openInputStream(
-            handle,
-            hidlDevice,
-            hidlConfig,
-            AudioInputFlag(flags),
-            AudioSource(source),
-            [&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *inStream = new StreamInHalHidl(result);
-                }
-                HidlUtils::audioConfigToHal(suggestedConfig, config);
-            });
-    return processReturn("openInputStream", ret, retval);
-}
-
-status_t DeviceHalHidl::supportsAudioPatches(bool *supportsPatches) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("supportsAudioPatches", mDevice->supportsAudioPatches(), supportsPatches);
-}
-
-status_t DeviceHalHidl::createAudioPatch(
-        unsigned int num_sources,
-        const struct audio_port_config *sources,
-        unsigned int num_sinks,
-        const struct audio_port_config *sinks,
-        audio_patch_handle_t *patch) {
-    if (mDevice == 0) return NO_INIT;
-    hidl_vec<AudioPortConfig> hidlSources, hidlSinks;
-    HidlUtils::audioPortConfigsFromHal(num_sources, sources, &hidlSources);
-    HidlUtils::audioPortConfigsFromHal(num_sinks, sinks, &hidlSinks);
-    Result retval;
-    Return<void> ret = mDevice->createAudioPatch(
-            hidlSources, hidlSinks,
-            [&](Result r, AudioPatchHandle hidlPatch) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *patch = static_cast<audio_patch_handle_t>(hidlPatch);
-                }
-            });
-    return processReturn("createAudioPatch", ret, retval);
-}
-
-status_t DeviceHalHidl::releaseAudioPatch(audio_patch_handle_t patch) {
-    if (mDevice == 0) return NO_INIT;
-    return processReturn("releaseAudioPatch", mDevice->releaseAudioPatch(patch));
-}
-
-status_t DeviceHalHidl::getAudioPort(struct audio_port *port) {
-    if (mDevice == 0) return NO_INIT;
-    AudioPort hidlPort;
-    HidlUtils::audioPortFromHal(*port, &hidlPort);
-    Result retval;
-    Return<void> ret = mDevice->getAudioPort(
-            hidlPort,
-            [&](Result r, const AudioPort& p) {
-                retval = r;
-                if (retval == Result::OK) {
-                    HidlUtils::audioPortToHal(p, port);
-                }
-            });
-    return processReturn("getAudioPort", ret, retval);
-}
-
-status_t DeviceHalHidl::setAudioPortConfig(const struct audio_port_config *config) {
-    if (mDevice == 0) return NO_INIT;
-    AudioPortConfig hidlConfig;
-    HidlUtils::audioPortConfigFromHal(*config, &hidlConfig);
-    return processReturn("setAudioPortConfig", mDevice->setAudioPortConfig(hidlConfig));
-}
-
-status_t DeviceHalHidl::getMicrophones(
-        std::vector<media::MicrophoneInfo> *microphonesInfo __unused) {
-    if (mDevice == 0) return NO_INIT;
-    return INVALID_OPERATION;
-}
-
-status_t DeviceHalHidl::dump(int fd) {
-    if (mDevice == 0) return NO_INIT;
-    native_handle_t* hidlHandle = native_handle_create(1, 0);
-    hidlHandle->data[0] = fd;
-    Return<void> ret = mDevice->debugDump(hidlHandle);
-    native_handle_delete(hidlHandle);
-    return processReturn("dump", ret);
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/DeviceHalLocal.cpp b/media/libaudiohal/2.0/DeviceHalLocal.cpp
deleted file mode 100644
index ec3bf78..0000000
--- a/media/libaudiohal/2.0/DeviceHalLocal.cpp
+++ /dev/null
@@ -1,204 +0,0 @@
-/*
- * Copyright (C) 2016 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 "DeviceHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <utils/Log.h>
-
-#include "DeviceHalLocal.h"
-#include "StreamHalLocal.h"
-
-namespace android {
-
-DeviceHalLocal::DeviceHalLocal(audio_hw_device_t *dev)
-        : mDev(dev) {
-}
-
-DeviceHalLocal::~DeviceHalLocal() {
-    int status = audio_hw_device_close(mDev);
-    ALOGW_IF(status, "Error closing audio hw device %p: %s", mDev, strerror(-status));
-    mDev = 0;
-}
-
-status_t DeviceHalLocal::getSupportedDevices(uint32_t *devices) {
-    if (mDev->get_supported_devices == NULL) return INVALID_OPERATION;
-    *devices = mDev->get_supported_devices(mDev);
-    return OK;
-}
-
-status_t DeviceHalLocal::initCheck() {
-    return mDev->init_check(mDev);
-}
-
-status_t DeviceHalLocal::setVoiceVolume(float volume) {
-    return mDev->set_voice_volume(mDev, volume);
-}
-
-status_t DeviceHalLocal::setMasterVolume(float volume) {
-    if (mDev->set_master_volume == NULL) return INVALID_OPERATION;
-    return mDev->set_master_volume(mDev, volume);
-}
-
-status_t DeviceHalLocal::getMasterVolume(float *volume) {
-    if (mDev->get_master_volume == NULL) return INVALID_OPERATION;
-    return mDev->get_master_volume(mDev, volume);
-}
-
-status_t DeviceHalLocal::setMode(audio_mode_t mode) {
-    return mDev->set_mode(mDev, mode);
-}
-
-status_t DeviceHalLocal::setMicMute(bool state) {
-    return mDev->set_mic_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::getMicMute(bool *state) {
-    return mDev->get_mic_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::setMasterMute(bool state) {
-    if (mDev->set_master_mute == NULL) return INVALID_OPERATION;
-    return mDev->set_master_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::getMasterMute(bool *state) {
-    if (mDev->get_master_mute == NULL) return INVALID_OPERATION;
-    return mDev->get_master_mute(mDev, state);
-}
-
-status_t DeviceHalLocal::setParameters(const String8& kvPairs) {
-    return mDev->set_parameters(mDev, kvPairs.string());
-}
-
-status_t DeviceHalLocal::getParameters(const String8& keys, String8 *values) {
-    char *halValues = mDev->get_parameters(mDev, keys.string());
-    if (halValues != NULL) {
-        values->setTo(halValues);
-        free(halValues);
-    } else {
-        values->clear();
-    }
-    return OK;
-}
-
-status_t DeviceHalLocal::getInputBufferSize(
-        const struct audio_config *config, size_t *size) {
-    *size = mDev->get_input_buffer_size(mDev, config);
-    return OK;
-}
-
-status_t DeviceHalLocal::openOutputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        audio_output_flags_t flags,
-        struct audio_config *config,
-        const char *address,
-        sp<StreamOutHalInterface> *outStream) {
-    audio_stream_out_t *halStream;
-    ALOGV("open_output_stream handle: %d devices: %x flags: %#x"
-            "srate: %d format %#x channels %x address %s",
-            handle, devices, flags,
-            config->sample_rate, config->format, config->channel_mask,
-            address);
-    int openResut = mDev->open_output_stream(
-            mDev, handle, devices, flags, config, &halStream, address);
-    if (openResut == OK) {
-        *outStream = new StreamOutHalLocal(halStream, this);
-    }
-    ALOGV("open_output_stream status %d stream %p", openResut, halStream);
-    return openResut;
-}
-
-status_t DeviceHalLocal::openInputStream(
-        audio_io_handle_t handle,
-        audio_devices_t devices,
-        struct audio_config *config,
-        audio_input_flags_t flags,
-        const char *address,
-        audio_source_t source,
-        sp<StreamInHalInterface> *inStream) {
-    audio_stream_in_t *halStream;
-    ALOGV("open_input_stream handle: %d devices: %x flags: %#x "
-            "srate: %d format %#x channels %x address %s source %d",
-            handle, devices, flags,
-            config->sample_rate, config->format, config->channel_mask,
-            address, source);
-    int openResult = mDev->open_input_stream(
-            mDev, handle, devices, config, &halStream, flags, address, source);
-    if (openResult == OK) {
-        *inStream = new StreamInHalLocal(halStream, this);
-    }
-    ALOGV("open_input_stream status %d stream %p", openResult, inStream);
-    return openResult;
-}
-
-status_t DeviceHalLocal::supportsAudioPatches(bool *supportsPatches) {
-    *supportsPatches = version() >= AUDIO_DEVICE_API_VERSION_3_0;
-    return OK;
-}
-
-status_t DeviceHalLocal::createAudioPatch(
-        unsigned int num_sources,
-        const struct audio_port_config *sources,
-        unsigned int num_sinks,
-        const struct audio_port_config *sinks,
-        audio_patch_handle_t *patch) {
-    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
-        return mDev->create_audio_patch(
-                mDev, num_sources, sources, num_sinks, sinks, patch);
-    } else {
-        return INVALID_OPERATION;
-    }
-}
-
-status_t DeviceHalLocal::releaseAudioPatch(audio_patch_handle_t patch) {
-    if (version() >= AUDIO_DEVICE_API_VERSION_3_0) {
-        return mDev->release_audio_patch(mDev, patch);
-    } else {
-        return INVALID_OPERATION;
-    }
-}
-
-status_t DeviceHalLocal::getAudioPort(struct audio_port *port) {
-    return mDev->get_audio_port(mDev, port);
-}
-
-status_t DeviceHalLocal::setAudioPortConfig(const struct audio_port_config *config) {
-    if (version() >= AUDIO_DEVICE_API_VERSION_3_0)
-        return mDev->set_audio_port_config(mDev, config);
-    else
-        return INVALID_OPERATION;
-}
-
-status_t DeviceHalLocal::getMicrophones(
-        std::vector<media::MicrophoneInfo> *microphones __unused) {
-    return INVALID_OPERATION;
-}
-
-status_t DeviceHalLocal::dump(int fd) {
-    return mDev->dump(mDev, fd);
-}
-
-void DeviceHalLocal::closeOutputStream(struct audio_stream_out *stream_out) {
-    mDev->close_output_stream(mDev, stream_out);
-}
-
-void DeviceHalLocal::closeInputStream(struct audio_stream_in *stream_in) {
-    mDev->close_input_stream(mDev, stream_in);
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp b/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp
deleted file mode 100644
index 5b33592..0000000
--- a/media/libaudiohal/2.0/DevicesFactoryHalHidl.cpp
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <string.h>
-
-#define LOG_TAG "DevicesFactoryHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/2.0/IDevice.h>
-#include <media/audiohal/hidl/HalDeathHandler.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "DeviceHalHidl.h"
-#include "DevicesFactoryHalHidl.h"
-
-using ::android::hardware::audio::V2_0::IDevice;
-using ::android::hardware::audio::V2_0::Result;
-using ::android::hardware::Return;
-
-namespace android {
-
-DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
-    mDevicesFactory = IDevicesFactory::getService();
-    if (mDevicesFactory != 0) {
-        // It is assumed that DevicesFactory is owned by AudioFlinger
-        // and thus have the same lifespan.
-        mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
-    } else {
-        ALOGE("Failed to obtain IDevicesFactory service, terminating process.");
-        exit(1);
-    }
-    // The MSD factory is optional
-    mDevicesFactoryMsd = IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD);
-    // TODO: Register death handler, and add 'restart' directive to audioserver.rc
-}
-
-DevicesFactoryHalHidl::~DevicesFactoryHalHidl() {
-}
-
-// static
-status_t DevicesFactoryHalHidl::nameFromHal(const char *name, IDevicesFactory::Device *device) {
-    if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
-        *device = IDevicesFactory::Device::PRIMARY;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
-        *device = IDevicesFactory::Device::A2DP;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
-        *device = IDevicesFactory::Device::USB;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
-        *device = IDevicesFactory::Device::R_SUBMIX;
-        return OK;
-    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_STUB) == 0) {
-        *device = IDevicesFactory::Device::STUB;
-        return OK;
-    }
-    ALOGE("Invalid device name %s", name);
-    return BAD_VALUE;
-}
-
-status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    if (mDevicesFactory == 0) return NO_INIT;
-    IDevicesFactory::Device hidlDevice;
-    status_t status = nameFromHal(name, &hidlDevice);
-    if (status != OK) return status;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevicesFactory->openDevice(
-            hidlDevice,
-            [&](Result r, const sp<IDevice>& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *device = new DeviceHalHidl(result);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return BAD_VALUE;
-        else return NO_INIT;
-    }
-    return FAILED_TRANSACTION;
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalLocal.cpp b/media/libaudiohal/2.0/DevicesFactoryHalLocal.cpp
deleted file mode 100644
index 13a9acd..0000000
--- a/media/libaudiohal/2.0/DevicesFactoryHalLocal.cpp
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (C) 2016 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 "DevicesFactoryHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <string.h>
-
-#include <hardware/audio.h>
-#include <utils/Log.h>
-
-#include "DeviceHalLocal.h"
-#include "DevicesFactoryHalLocal.h"
-
-namespace android {
-
-static status_t load_audio_interface(const char *if_name, audio_hw_device_t **dev)
-{
-    const hw_module_t *mod;
-    int rc;
-
-    rc = hw_get_module_by_class(AUDIO_HARDWARE_MODULE_ID, if_name, &mod);
-    if (rc) {
-        ALOGE("%s couldn't load audio hw module %s.%s (%s)", __func__,
-                AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
-        goto out;
-    }
-    rc = audio_hw_device_open(mod, dev);
-    if (rc) {
-        ALOGE("%s couldn't open audio hw device in %s.%s (%s)", __func__,
-                AUDIO_HARDWARE_MODULE_ID, if_name, strerror(-rc));
-        goto out;
-    }
-    if ((*dev)->common.version < AUDIO_DEVICE_API_VERSION_MIN) {
-        ALOGE("%s wrong audio hw device version %04x", __func__, (*dev)->common.version);
-        rc = BAD_VALUE;
-        audio_hw_device_close(*dev);
-        goto out;
-    }
-    return OK;
-
-out:
-    *dev = NULL;
-    return rc;
-}
-
-status_t DevicesFactoryHalLocal::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    audio_hw_device_t *dev;
-    status_t rc = load_audio_interface(name, &dev);
-    if (rc == OK) {
-        *device = new DeviceHalLocal(dev);
-    }
-    return rc;
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/EffectBufferHalHidl.cpp b/media/libaudiohal/2.0/EffectBufferHalHidl.cpp
deleted file mode 100644
index 226a500..0000000
--- a/media/libaudiohal/2.0/EffectBufferHalHidl.cpp
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <atomic>
-
-#define LOG_TAG "EffectBufferHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hidl/allocator/1.0/IAllocator.h>
-#include <hidlmemory/mapping.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "EffectBufferHalHidl.h"
-
-using ::android::hardware::Return;
-using ::android::hidl::allocator::V1_0::IAllocator;
-
-namespace android {
-
-// static
-uint64_t EffectBufferHalHidl::makeUniqueId() {
-    static std::atomic<uint64_t> counter{1};
-    return counter++;
-}
-
-status_t EffectBufferHalHidl::allocate(
-        size_t size, sp<EffectBufferHalInterface>* buffer) {
-    return mirror(nullptr, size, buffer);
-}
-
-status_t EffectBufferHalHidl::mirror(
-        void* external, size_t size, sp<EffectBufferHalInterface>* buffer) {
-    sp<EffectBufferHalInterface> tempBuffer = new EffectBufferHalHidl(size);
-    status_t result = static_cast<EffectBufferHalHidl*>(tempBuffer.get())->init();
-    if (result == OK) {
-        tempBuffer->setExternalData(external);
-        *buffer = tempBuffer;
-    }
-    return result;
-}
-
-EffectBufferHalHidl::EffectBufferHalHidl(size_t size)
-        : mBufferSize(size), mFrameCountChanged(false),
-          mExternalData(nullptr), mAudioBuffer{0, {nullptr}} {
-    mHidlBuffer.id = makeUniqueId();
-    mHidlBuffer.frameCount = 0;
-}
-
-EffectBufferHalHidl::~EffectBufferHalHidl() {
-}
-
-status_t EffectBufferHalHidl::init() {
-    sp<IAllocator> ashmem = IAllocator::getService("ashmem");
-    if (ashmem == 0) {
-        ALOGE("Failed to retrieve ashmem allocator service");
-        return NO_INIT;
-    }
-    status_t retval = NO_MEMORY;
-    Return<void> result = ashmem->allocate(
-            mBufferSize,
-            [&](bool success, const hidl_memory& memory) {
-                if (success) {
-                    mHidlBuffer.data = memory;
-                    retval = OK;
-                }
-            });
-    if (result.isOk() && retval == OK) {
-        mMemory = hardware::mapMemory(mHidlBuffer.data);
-        if (mMemory != 0) {
-            mMemory->update();
-            mAudioBuffer.raw = static_cast<void*>(mMemory->getPointer());
-            memset(mAudioBuffer.raw, 0, mMemory->getSize());
-            mMemory->commit();
-        } else {
-            ALOGE("Failed to map allocated ashmem");
-            retval = NO_MEMORY;
-        }
-    } else {
-        ALOGE("Failed to allocate %d bytes from ashmem", (int)mBufferSize);
-    }
-    return result.isOk() ? retval : FAILED_TRANSACTION;
-}
-
-audio_buffer_t* EffectBufferHalHidl::audioBuffer() {
-    return &mAudioBuffer;
-}
-
-void* EffectBufferHalHidl::externalData() const {
-    return mExternalData;
-}
-
-void EffectBufferHalHidl::setFrameCount(size_t frameCount) {
-    mHidlBuffer.frameCount = frameCount;
-    mAudioBuffer.frameCount = frameCount;
-    mFrameCountChanged = true;
-}
-
-bool EffectBufferHalHidl::checkFrameCountChange() {
-    bool result = mFrameCountChanged;
-    mFrameCountChanged = false;
-    return result;
-}
-
-void EffectBufferHalHidl::setExternalData(void* external) {
-    mExternalData = external;
-}
-
-void EffectBufferHalHidl::update() {
-    update(mBufferSize);
-}
-
-void EffectBufferHalHidl::commit() {
-    commit(mBufferSize);
-}
-
-void EffectBufferHalHidl::update(size_t size) {
-    if (mExternalData == nullptr) return;
-    mMemory->update();
-    if (size > mBufferSize) size = mBufferSize;
-    memcpy(mAudioBuffer.raw, mExternalData, size);
-    mMemory->commit();
-}
-
-void EffectBufferHalHidl::commit(size_t size) {
-    if (mExternalData == nullptr) return;
-    if (size > mBufferSize) size = mBufferSize;
-    memcpy(mExternalData, mAudioBuffer.raw, size);
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/EffectHalHidl.cpp b/media/libaudiohal/2.0/EffectHalHidl.cpp
deleted file mode 100644
index 4fb032c..0000000
--- a/media/libaudiohal/2.0/EffectHalHidl.cpp
+++ /dev/null
@@ -1,338 +0,0 @@
-/*
- * Copyright (C) 2016 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 "EffectHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <hwbinder/IPCThreadState.h>
-#include <media/EffectsFactoryApi.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "EffectBufferHalHidl.h"
-#include "EffectHalHidl.h"
-#include "HidlUtils.h"
-
-using ::android::hardware::audio::effect::V2_0::AudioBuffer;
-using ::android::hardware::audio::effect::V2_0::EffectBufferAccess;
-using ::android::hardware::audio::effect::V2_0::EffectConfigParameters;
-using ::android::hardware::audio::effect::V2_0::MessageQueueFlagBits;
-using ::android::hardware::audio::effect::V2_0::Result;
-using ::android::hardware::audio::common::V2_0::HidlUtils;
-using ::android::hardware::audio::common::V2_0::AudioChannelMask;
-using ::android::hardware::audio::common::V2_0::AudioFormat;
-using ::android::hardware::hidl_vec;
-using ::android::hardware::MQDescriptorSync;
-using ::android::hardware::Return;
-
-namespace android {
-
-EffectHalHidl::EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId)
-        : mEffect(effect), mEffectId(effectId), mBuffersChanged(true), mEfGroup(nullptr) {
-}
-
-EffectHalHidl::~EffectHalHidl() {
-    if (mEffect != 0) {
-        close();
-        mEffect.clear();
-        hardware::IPCThreadState::self()->flushCommands();
-    }
-    if (mEfGroup) {
-        EventFlag::deleteEventFlag(&mEfGroup);
-    }
-}
-
-// static
-void EffectHalHidl::effectDescriptorToHal(
-        const EffectDescriptor& descriptor, effect_descriptor_t* halDescriptor) {
-    HidlUtils::uuidToHal(descriptor.type, &halDescriptor->type);
-    HidlUtils::uuidToHal(descriptor.uuid, &halDescriptor->uuid);
-    halDescriptor->flags = static_cast<uint32_t>(descriptor.flags);
-    halDescriptor->cpuLoad = descriptor.cpuLoad;
-    halDescriptor->memoryUsage = descriptor.memoryUsage;
-    memcpy(halDescriptor->name, descriptor.name.data(), descriptor.name.size());
-    memcpy(halDescriptor->implementor,
-            descriptor.implementor.data(), descriptor.implementor.size());
-}
-
-// TODO(mnaganov): These buffer conversion functions should be shared with Effect wrapper
-// via HidlUtils. Move them there when hardware/interfaces will get un-frozen again.
-
-// static
-void EffectHalHidl::effectBufferConfigFromHal(
-        const buffer_config_t& halConfig, EffectBufferConfig* config) {
-    config->samplingRateHz = halConfig.samplingRate;
-    config->channels = AudioChannelMask(halConfig.channels);
-    config->format = AudioFormat(halConfig.format);
-    config->accessMode = EffectBufferAccess(halConfig.accessMode);
-    config->mask = EffectConfigParameters(halConfig.mask);
-}
-
-// static
-void EffectHalHidl::effectBufferConfigToHal(
-        const EffectBufferConfig& config, buffer_config_t* halConfig) {
-    halConfig->buffer.frameCount = 0;
-    halConfig->buffer.raw = NULL;
-    halConfig->samplingRate = config.samplingRateHz;
-    halConfig->channels = static_cast<uint32_t>(config.channels);
-    halConfig->bufferProvider.cookie = NULL;
-    halConfig->bufferProvider.getBuffer = NULL;
-    halConfig->bufferProvider.releaseBuffer = NULL;
-    halConfig->format = static_cast<uint8_t>(config.format);
-    halConfig->accessMode = static_cast<uint8_t>(config.accessMode);
-    halConfig->mask = static_cast<uint8_t>(config.mask);
-}
-
-// static
-void EffectHalHidl::effectConfigFromHal(const effect_config_t& halConfig, EffectConfig* config) {
-    effectBufferConfigFromHal(halConfig.inputCfg, &config->inputCfg);
-    effectBufferConfigFromHal(halConfig.outputCfg, &config->outputCfg);
-}
-
-// static
-void EffectHalHidl::effectConfigToHal(const EffectConfig& config, effect_config_t* halConfig) {
-    effectBufferConfigToHal(config.inputCfg, &halConfig->inputCfg);
-    effectBufferConfigToHal(config.outputCfg, &halConfig->outputCfg);
-}
-
-// static
-status_t EffectHalHidl::analyzeResult(const Result& result) {
-    switch (result) {
-        case Result::OK: return OK;
-        case Result::INVALID_ARGUMENTS: return BAD_VALUE;
-        case Result::INVALID_STATE: return NOT_ENOUGH_DATA;
-        case Result::NOT_INITIALIZED: return NO_INIT;
-        case Result::NOT_SUPPORTED: return INVALID_OPERATION;
-        case Result::RESULT_TOO_BIG: return NO_MEMORY;
-        default: return NO_INIT;
-    }
-}
-
-status_t EffectHalHidl::setInBuffer(const sp<EffectBufferHalInterface>& buffer) {
-    if (!mBuffersChanged) {
-        if (buffer.get() == nullptr || mInBuffer.get() == nullptr) {
-            mBuffersChanged = buffer.get() != mInBuffer.get();
-        } else {
-            mBuffersChanged = buffer->audioBuffer() != mInBuffer->audioBuffer();
-        }
-    }
-    mInBuffer = buffer;
-    return OK;
-}
-
-status_t EffectHalHidl::setOutBuffer(const sp<EffectBufferHalInterface>& buffer) {
-    if (!mBuffersChanged) {
-        if (buffer.get() == nullptr || mOutBuffer.get() == nullptr) {
-            mBuffersChanged = buffer.get() != mOutBuffer.get();
-        } else {
-            mBuffersChanged = buffer->audioBuffer() != mOutBuffer->audioBuffer();
-        }
-    }
-    mOutBuffer = buffer;
-    return OK;
-}
-
-status_t EffectHalHidl::process() {
-    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS));
-}
-
-status_t EffectHalHidl::processReverse() {
-    return processImpl(static_cast<uint32_t>(MessageQueueFlagBits::REQUEST_PROCESS_REVERSE));
-}
-
-status_t EffectHalHidl::prepareForProcessing() {
-    std::unique_ptr<StatusMQ> tempStatusMQ;
-    Result retval;
-    Return<void> ret = mEffect->prepareForProcessing(
-            [&](Result r, const MQDescriptorSync<Result>& statusMQ) {
-                retval = r;
-                if (retval == Result::OK) {
-                    tempStatusMQ.reset(new StatusMQ(statusMQ));
-                    if (tempStatusMQ->isValid() && tempStatusMQ->getEventFlagWord()) {
-                        EventFlag::createEventFlag(tempStatusMQ->getEventFlagWord(), &mEfGroup);
-                    }
-                }
-            });
-    if (!ret.isOk() || retval != Result::OK) {
-        return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
-    }
-    if (!tempStatusMQ || !tempStatusMQ->isValid() || !mEfGroup) {
-        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for effects");
-        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
-                "Status message queue for effects is invalid");
-        ALOGE_IF(!mEfGroup, "Event flag creation for effects failed");
-        return NO_INIT;
-    }
-    mStatusMQ = std::move(tempStatusMQ);
-    return OK;
-}
-
-bool EffectHalHidl::needToResetBuffers() {
-    if (mBuffersChanged) return true;
-    bool inBufferFrameCountUpdated = mInBuffer->checkFrameCountChange();
-    bool outBufferFrameCountUpdated = mOutBuffer->checkFrameCountChange();
-    return inBufferFrameCountUpdated || outBufferFrameCountUpdated;
-}
-
-status_t EffectHalHidl::processImpl(uint32_t mqFlag) {
-    if (mEffect == 0 || mInBuffer == 0 || mOutBuffer == 0) return NO_INIT;
-    status_t status;
-    if (!mStatusMQ && (status = prepareForProcessing()) != OK) {
-        return status;
-    }
-    if (needToResetBuffers() && (status = setProcessBuffers()) != OK) {
-        return status;
-    }
-    // The data is already in the buffers, just need to flush it and wake up the server side.
-    std::atomic_thread_fence(std::memory_order_release);
-    mEfGroup->wake(mqFlag);
-    uint32_t efState = 0;
-retry:
-    status_t ret = mEfGroup->wait(
-            static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING), &efState);
-    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::DONE_PROCESSING)) {
-        Result retval = Result::NOT_INITIALIZED;
-        mStatusMQ->read(&retval);
-        if (retval == Result::OK || retval == Result::INVALID_STATE) {
-            // Sync back the changed contents of the buffer.
-            std::atomic_thread_fence(std::memory_order_acquire);
-        }
-        return analyzeResult(retval);
-    }
-    if (ret == -EAGAIN || ret == -EINTR) {
-        // Spurious wakeup. This normally retries no more than once.
-        goto retry;
-    }
-    return ret;
-}
-
-status_t EffectHalHidl::setProcessBuffers() {
-    Return<Result> ret = mEffect->setProcessBuffers(
-            static_cast<EffectBufferHalHidl*>(mInBuffer.get())->hidlBuffer(),
-            static_cast<EffectBufferHalHidl*>(mOutBuffer.get())->hidlBuffer());
-    if (ret.isOk() && ret == Result::OK) {
-        mBuffersChanged = false;
-        return OK;
-    }
-    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
-        uint32_t *replySize, void *pReplyData) {
-    if (mEffect == 0) return NO_INIT;
-
-    // Special cases.
-    if (cmdCode == EFFECT_CMD_SET_CONFIG || cmdCode == EFFECT_CMD_SET_CONFIG_REVERSE) {
-        return setConfigImpl(cmdCode, cmdSize, pCmdData, replySize, pReplyData);
-    } else if (cmdCode == EFFECT_CMD_GET_CONFIG || cmdCode == EFFECT_CMD_GET_CONFIG_REVERSE) {
-        return getConfigImpl(cmdCode, replySize, pReplyData);
-    }
-
-    // Common case.
-    hidl_vec<uint8_t> hidlData;
-    if (pCmdData != nullptr && cmdSize > 0) {
-        hidlData.setToExternal(reinterpret_cast<uint8_t*>(pCmdData), cmdSize);
-    }
-    status_t status;
-    uint32_t replySizeStub = 0;
-    if (replySize == nullptr || pReplyData == nullptr) replySize = &replySizeStub;
-    Return<void> ret = mEffect->command(cmdCode, hidlData, *replySize,
-            [&](int32_t s, const hidl_vec<uint8_t>& result) {
-                status = s;
-                if (status == 0) {
-                    if (*replySize > result.size()) *replySize = result.size();
-                    if (pReplyData != nullptr && *replySize > 0) {
-                        memcpy(pReplyData, &result[0], *replySize);
-                    }
-                }
-            });
-    return ret.isOk() ? status : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::getDescriptor(effect_descriptor_t *pDescriptor) {
-    if (mEffect == 0) return NO_INIT;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffect->getDescriptor(
-            [&](Result r, const EffectDescriptor& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    effectDescriptorToHal(result, pDescriptor);
-                }
-            });
-    return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::close() {
-    if (mEffect == 0) return NO_INIT;
-    Return<Result> ret = mEffect->close();
-    return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
-}
-
-status_t EffectHalHidl::getConfigImpl(
-        uint32_t cmdCode, uint32_t *replySize, void *pReplyData) {
-    if (replySize == NULL || *replySize != sizeof(effect_config_t) || pReplyData == NULL) {
-        return BAD_VALUE;
-    }
-    status_t result = FAILED_TRANSACTION;
-    Return<void> ret;
-    if (cmdCode == EFFECT_CMD_GET_CONFIG) {
-        ret = mEffect->getConfig([&] (Result r, const EffectConfig &hidlConfig) {
-            result = analyzeResult(r);
-            if (r == Result::OK) {
-                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
-            }
-        });
-    } else {
-        ret = mEffect->getConfigReverse([&] (Result r, const EffectConfig &hidlConfig) {
-            result = analyzeResult(r);
-            if (r == Result::OK) {
-                effectConfigToHal(hidlConfig, static_cast<effect_config_t*>(pReplyData));
-            }
-        });
-    }
-    if (!ret.isOk()) {
-        result = FAILED_TRANSACTION;
-    }
-    return result;
-}
-
-status_t EffectHalHidl::setConfigImpl(
-        uint32_t cmdCode, uint32_t cmdSize, void *pCmdData, uint32_t *replySize, void *pReplyData) {
-    if (pCmdData == NULL || cmdSize != sizeof(effect_config_t) ||
-            replySize == NULL || *replySize != sizeof(int32_t) || pReplyData == NULL) {
-        return BAD_VALUE;
-    }
-    const effect_config_t *halConfig = static_cast<effect_config_t*>(pCmdData);
-    if (halConfig->inputCfg.bufferProvider.getBuffer != NULL ||
-            halConfig->inputCfg.bufferProvider.releaseBuffer != NULL ||
-            halConfig->outputCfg.bufferProvider.getBuffer != NULL ||
-            halConfig->outputCfg.bufferProvider.releaseBuffer != NULL) {
-        ALOGE("Buffer provider callbacks are not supported");
-    }
-    EffectConfig hidlConfig;
-    effectConfigFromHal(*halConfig, &hidlConfig);
-    Return<Result> ret = cmdCode == EFFECT_CMD_SET_CONFIG ?
-            mEffect->setConfig(hidlConfig, nullptr, nullptr) :
-            mEffect->setConfigReverse(hidlConfig, nullptr, nullptr);
-    status_t result = FAILED_TRANSACTION;
-    if (ret.isOk()) {
-        result = analyzeResult(ret);
-        *static_cast<int32_t*>(pReplyData) = result;
-    }
-    return result;
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/EffectsFactoryHalHidl.cpp b/media/libaudiohal/2.0/EffectsFactoryHalHidl.cpp
deleted file mode 100644
index 0d40e6d..0000000
--- a/media/libaudiohal/2.0/EffectsFactoryHalHidl.cpp
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright (C) 2016 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 "EffectsFactoryHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <cutils/native_handle.h>
-
-#include "ConversionHelperHidl.h"
-#include "EffectBufferHalHidl.h"
-#include "EffectHalHidl.h"
-#include "EffectsFactoryHalHidl.h"
-#include "HidlUtils.h"
-
-using ::android::hardware::audio::common::V2_0::HidlUtils;
-using ::android::hardware::audio::common::V2_0::Uuid;
-using ::android::hardware::audio::effect::V2_0::IEffect;
-using ::android::hardware::audio::effect::V2_0::Result;
-using ::android::hardware::Return;
-
-namespace android {
-
-EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
-    mEffectsFactory = IEffectsFactory::getService();
-    if (mEffectsFactory == 0) {
-        ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
-        exit(1);
-    }
-}
-
-EffectsFactoryHalHidl::~EffectsFactoryHalHidl() {
-}
-
-status_t EffectsFactoryHalHidl::queryAllDescriptors() {
-    if (mEffectsFactory == 0) return NO_INIT;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffectsFactory->getAllDescriptors(
-            [&](Result r, const hidl_vec<EffectDescriptor>& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    mLastDescriptors = result;
-                }
-            });
-    if (ret.isOk()) {
-        return retval == Result::OK ? OK : NO_INIT;
-    }
-    mLastDescriptors.resize(0);
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::queryNumberEffects(uint32_t *pNumEffects) {
-    status_t queryResult = queryAllDescriptors();
-    if (queryResult == OK) {
-        *pNumEffects = mLastDescriptors.size();
-    }
-    return queryResult;
-}
-
-status_t EffectsFactoryHalHidl::getDescriptor(
-        uint32_t index, effect_descriptor_t *pDescriptor) {
-    // TODO: We need somehow to track the changes on the server side
-    // or figure out how to convert everybody to query all the descriptors at once.
-    // TODO: check for nullptr
-    if (mLastDescriptors.size() == 0) {
-        status_t queryResult = queryAllDescriptors();
-        if (queryResult != OK) return queryResult;
-    }
-    if (index >= mLastDescriptors.size()) return NAME_NOT_FOUND;
-    EffectHalHidl::effectDescriptorToHal(mLastDescriptors[index], pDescriptor);
-    return OK;
-}
-
-status_t EffectsFactoryHalHidl::getDescriptor(
-        const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
-    // TODO: check for nullptr
-    if (mEffectsFactory == 0) return NO_INIT;
-    Uuid hidlUuid;
-    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffectsFactory->getDescriptor(hidlUuid,
-            [&](Result r, const EffectDescriptor& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    EffectHalHidl::effectDescriptorToHal(result, pDescriptor);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
-        else return NO_INIT;
-    }
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::createEffect(
-        const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
-        sp<EffectHalInterface> *effect) {
-    if (mEffectsFactory == 0) return NO_INIT;
-    Uuid hidlUuid;
-    HidlUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mEffectsFactory->createEffect(
-            hidlUuid, sessionId, ioId,
-            [&](Result r, const sp<IEffect>& result, uint64_t effectId) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *effect = new EffectHalHidl(result, effectId);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return NAME_NOT_FOUND;
-        else return NO_INIT;
-    }
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::dumpEffects(int fd) {
-    if (mEffectsFactory == 0) return NO_INIT;
-    native_handle_t* hidlHandle = native_handle_create(1, 0);
-    hidlHandle->data[0] = fd;
-    Return<void> ret = mEffectsFactory->debugDump(hidlHandle);
-    native_handle_delete(hidlHandle);
-    return processReturn(__FUNCTION__, ret);
-}
-
-status_t EffectsFactoryHalHidl::allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) {
-    return EffectBufferHalHidl::allocate(size, buffer);
-}
-
-status_t EffectsFactoryHalHidl::mirrorBuffer(void* external, size_t size,
-                          sp<EffectBufferHalInterface>* buffer) {
-    return EffectBufferHalHidl::mirror(external, size, buffer);
-}
-
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/StreamHalHidl.cpp b/media/libaudiohal/2.0/StreamHalHidl.cpp
deleted file mode 100644
index 9869cd2..0000000
--- a/media/libaudiohal/2.0/StreamHalHidl.cpp
+++ /dev/null
@@ -1,768 +0,0 @@
-/*
- * Copyright (C) 2016 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 "StreamHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/2.0/IStreamOutCallback.h>
-#include <hwbinder/IPCThreadState.h>
-#include <mediautils/SchedulingPolicyService.h>
-#include <utils/Log.h>
-
-#include "DeviceHalHidl.h"
-#include "EffectHalHidl.h"
-#include "StreamHalHidl.h"
-
-using ::android::hardware::audio::common::V2_0::AudioChannelMask;
-using ::android::hardware::audio::common::V2_0::AudioFormat;
-using ::android::hardware::audio::common::V2_0::ThreadInfo;
-using ::android::hardware::audio::V2_0::AudioDrain;
-using ::android::hardware::audio::V2_0::IStreamOutCallback;
-using ::android::hardware::audio::V2_0::MessageQueueFlagBits;
-using ::android::hardware::audio::V2_0::MmapBufferInfo;
-using ::android::hardware::audio::V2_0::MmapPosition;
-using ::android::hardware::audio::V2_0::ParameterValue;
-using ::android::hardware::audio::V2_0::Result;
-using ::android::hardware::audio::V2_0::TimeSpec;
-using ::android::hardware::MQDescriptorSync;
-using ::android::hardware::Return;
-using ::android::hardware::Void;
-using ReadCommand = ::android::hardware::audio::V2_0::IStreamIn::ReadCommand;
-
-namespace android {
-
-StreamHalHidl::StreamHalHidl(IStream *stream)
-        : ConversionHelperHidl("Stream"),
-          mStream(stream),
-          mHalThreadPriority(HAL_THREAD_PRIORITY_DEFAULT),
-          mCachedBufferSize(0){
-
-    // Instrument audio signal power logging.
-    // Note: This assumes channel mask, format, and sample rate do not change after creation.
-    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
-        // Obtain audio properties (see StreamHalHidl::getAudioProperties() below).
-        Return<void> ret = mStream->getAudioProperties(
-                [&](uint32_t sr, AudioChannelMask m, AudioFormat f) {
-                mStreamPowerLog.init(sr,
-                        static_cast<audio_channel_mask_t>(m),
-                        static_cast<audio_format_t>(f));
-            });
-    }
-}
-
-StreamHalHidl::~StreamHalHidl() {
-    mStream = nullptr;
-}
-
-status_t StreamHalHidl::getSampleRate(uint32_t *rate) {
-    if (!mStream) return NO_INIT;
-    return processReturn("getSampleRate", mStream->getSampleRate(), rate);
-}
-
-status_t StreamHalHidl::getBufferSize(size_t *size) {
-    if (!mStream) return NO_INIT;
-    status_t status = processReturn("getBufferSize", mStream->getBufferSize(), size);
-    if (status == OK) {
-        mCachedBufferSize = *size;
-    }
-    return status;
-}
-
-status_t StreamHalHidl::getChannelMask(audio_channel_mask_t *mask) {
-    if (!mStream) return NO_INIT;
-    return processReturn("getChannelMask", mStream->getChannelMask(), mask);
-}
-
-status_t StreamHalHidl::getFormat(audio_format_t *format) {
-    if (!mStream) return NO_INIT;
-    return processReturn("getFormat", mStream->getFormat(), format);
-}
-
-status_t StreamHalHidl::getAudioProperties(
-        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
-    if (!mStream) return NO_INIT;
-    Return<void> ret = mStream->getAudioProperties(
-            [&](uint32_t sr, AudioChannelMask m, AudioFormat f) {
-                *sampleRate = sr;
-                *mask = static_cast<audio_channel_mask_t>(m);
-                *format = static_cast<audio_format_t>(f);
-            });
-    return processReturn("getAudioProperties", ret);
-}
-
-status_t StreamHalHidl::setParameters(const String8& kvPairs) {
-    if (!mStream) return NO_INIT;
-    hidl_vec<ParameterValue> hidlParams;
-    status_t status = parametersFromHal(kvPairs, &hidlParams);
-    if (status != OK) return status;
-    return processReturn("setParameters", mStream->setParameters(hidlParams));
-}
-
-status_t StreamHalHidl::getParameters(const String8& keys, String8 *values) {
-    values->clear();
-    if (!mStream) return NO_INIT;
-    hidl_vec<hidl_string> hidlKeys;
-    status_t status = keysFromHal(keys, &hidlKeys);
-    if (status != OK) return status;
-    Result retval;
-    Return<void> ret = mStream->getParameters(
-            hidlKeys,
-            [&](Result r, const hidl_vec<ParameterValue>& parameters) {
-                retval = r;
-                if (retval == Result::OK) {
-                    parametersToHal(parameters, values);
-                }
-            });
-    return processReturn("getParameters", ret, retval);
-}
-
-status_t StreamHalHidl::addEffect(sp<EffectHalInterface> effect) {
-    if (!mStream) return NO_INIT;
-    return processReturn("addEffect", mStream->addEffect(
-                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
-}
-
-status_t StreamHalHidl::removeEffect(sp<EffectHalInterface> effect) {
-    if (!mStream) return NO_INIT;
-    return processReturn("removeEffect", mStream->removeEffect(
-                    static_cast<EffectHalHidl*>(effect.get())->effectId()));
-}
-
-status_t StreamHalHidl::standby() {
-    if (!mStream) return NO_INIT;
-    return processReturn("standby", mStream->standby());
-}
-
-status_t StreamHalHidl::dump(int fd) {
-    if (!mStream) return NO_INIT;
-    native_handle_t* hidlHandle = native_handle_create(1, 0);
-    hidlHandle->data[0] = fd;
-    Return<void> ret = mStream->debugDump(hidlHandle);
-    native_handle_delete(hidlHandle);
-    mStreamPowerLog.dump(fd);
-    return processReturn("dump", ret);
-}
-
-status_t StreamHalHidl::start() {
-    if (!mStream) return NO_INIT;
-    return processReturn("start", mStream->start());
-}
-
-status_t StreamHalHidl::stop() {
-    if (!mStream) return NO_INIT;
-    return processReturn("stop", mStream->stop());
-}
-
-status_t StreamHalHidl::createMmapBuffer(int32_t minSizeFrames,
-                                  struct audio_mmap_buffer_info *info) {
-    Result retval;
-    Return<void> ret = mStream->createMmapBuffer(
-            minSizeFrames,
-            [&](Result r, const MmapBufferInfo& hidlInfo) {
-                retval = r;
-                if (retval == Result::OK) {
-                    const native_handle *handle = hidlInfo.sharedMemory.handle();
-                    if (handle->numFds > 0) {
-                        info->shared_memory_fd = handle->data[0];
-                        info->buffer_size_frames = hidlInfo.bufferSizeFrames;
-                        info->burst_size_frames = hidlInfo.burstSizeFrames;
-                        // info->shared_memory_address is not needed in HIDL context
-                        info->shared_memory_address = NULL;
-                    } else {
-                        retval = Result::NOT_INITIALIZED;
-                    }
-                }
-            });
-    return processReturn("createMmapBuffer", ret, retval);
-}
-
-status_t StreamHalHidl::getMmapPosition(struct audio_mmap_position *position) {
-    Result retval;
-    Return<void> ret = mStream->getMmapPosition(
-            [&](Result r, const MmapPosition& hidlPosition) {
-                retval = r;
-                if (retval == Result::OK) {
-                    position->time_nanoseconds = hidlPosition.timeNanoseconds;
-                    position->position_frames = hidlPosition.positionFrames;
-                }
-            });
-    return processReturn("getMmapPosition", ret, retval);
-}
-
-status_t StreamHalHidl::setHalThreadPriority(int priority) {
-    mHalThreadPriority = priority;
-    return OK;
-}
-
-status_t StreamHalHidl::getCachedBufferSize(size_t *size) {
-    if (mCachedBufferSize != 0) {
-        *size = mCachedBufferSize;
-        return OK;
-    }
-    return getBufferSize(size);
-}
-
-bool StreamHalHidl::requestHalThreadPriority(pid_t threadPid, pid_t threadId) {
-    if (mHalThreadPriority == HAL_THREAD_PRIORITY_DEFAULT) {
-        return true;
-    }
-    int err = requestPriority(
-            threadPid, threadId,
-            mHalThreadPriority, false /*isForApp*/, true /*asynchronous*/);
-    ALOGE_IF(err, "failed to set priority %d for pid %d tid %d; error %d",
-            mHalThreadPriority, threadPid, threadId, err);
-    // Audio will still work, but latency will be higher and sometimes unacceptable.
-    return err == 0;
-}
-
-namespace {
-
-/* Notes on callback ownership.
-
-This is how (Hw)Binder ownership model looks like. The server implementation
-is owned by Binder framework (via sp<>). Proxies are owned by clients.
-When the last proxy disappears, Binder framework releases the server impl.
-
-Thus, it is not needed to keep any references to StreamOutCallback (this is
-the server impl) -- it will live as long as HAL server holds a strong ref to
-IStreamOutCallback proxy. We clear that reference by calling 'clearCallback'
-from the destructor of StreamOutHalHidl.
-
-The callback only keeps a weak reference to the stream. The stream is owned
-by AudioFlinger.
-
-*/
-
-struct StreamOutCallback : public IStreamOutCallback {
-    StreamOutCallback(const wp<StreamOutHalHidl>& stream) : mStream(stream) {}
-
-    // IStreamOutCallback implementation
-    Return<void> onWriteReady()  override {
-        sp<StreamOutHalHidl> stream = mStream.promote();
-        if (stream != 0) {
-            stream->onWriteReady();
-        }
-        return Void();
-    }
-
-    Return<void> onDrainReady()  override {
-        sp<StreamOutHalHidl> stream = mStream.promote();
-        if (stream != 0) {
-            stream->onDrainReady();
-        }
-        return Void();
-    }
-
-    Return<void> onError()  override {
-        sp<StreamOutHalHidl> stream = mStream.promote();
-        if (stream != 0) {
-            stream->onError();
-        }
-        return Void();
-    }
-
-  private:
-    wp<StreamOutHalHidl> mStream;
-};
-
-}  // namespace
-
-StreamOutHalHidl::StreamOutHalHidl(const sp<IStreamOut>& stream)
-        : StreamHalHidl(stream.get()), mStream(stream), mWriterClient(0), mEfGroup(nullptr) {
-}
-
-StreamOutHalHidl::~StreamOutHalHidl() {
-    if (mStream != 0) {
-        if (mCallback.unsafe_get()) {
-            processReturn("clearCallback", mStream->clearCallback());
-        }
-        processReturn("close", mStream->close());
-        mStream.clear();
-    }
-    mCallback.clear();
-    hardware::IPCThreadState::self()->flushCommands();
-    if (mEfGroup) {
-        EventFlag::deleteEventFlag(&mEfGroup);
-    }
-}
-
-status_t StreamOutHalHidl::getFrameSize(size_t *size) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("getFrameSize", mStream->getFrameSize(), size);
-}
-
-status_t StreamOutHalHidl::getLatency(uint32_t *latency) {
-    if (mStream == 0) return NO_INIT;
-    if (mWriterClient == gettid() && mCommandMQ) {
-        return callWriterThread(
-                WriteCommand::GET_LATENCY, "getLatency", nullptr, 0,
-                [&](const WriteStatus& writeStatus) {
-                    *latency = writeStatus.reply.latencyMs;
-                });
-    } else {
-        return processReturn("getLatency", mStream->getLatency(), latency);
-    }
-}
-
-status_t StreamOutHalHidl::setVolume(float left, float right) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("setVolume", mStream->setVolume(left, right));
-}
-
-status_t StreamOutHalHidl::write(const void *buffer, size_t bytes, size_t *written) {
-    if (mStream == 0) return NO_INIT;
-    *written = 0;
-
-    if (bytes == 0 && !mDataMQ) {
-        // Can't determine the size for the MQ buffer. Wait for a non-empty write request.
-        ALOGW_IF(mCallback.unsafe_get(), "First call to async write with 0 bytes");
-        return OK;
-    }
-
-    status_t status;
-    if (!mDataMQ) {
-        // In case if playback starts close to the end of a compressed track, the bytes
-        // that need to be written is less than the actual buffer size. Need to use
-        // full buffer size for the MQ since otherwise after seeking back to the middle
-        // data will be truncated.
-        size_t bufferSize;
-        if ((status = getCachedBufferSize(&bufferSize)) != OK) {
-            return status;
-        }
-        if (bytes > bufferSize) bufferSize = bytes;
-        if ((status = prepareForWriting(bufferSize)) != OK) {
-            return status;
-        }
-    }
-
-    status = callWriterThread(
-            WriteCommand::WRITE, "write", static_cast<const uint8_t*>(buffer), bytes,
-            [&] (const WriteStatus& writeStatus) {
-                *written = writeStatus.reply.written;
-                // Diagnostics of the cause of b/35813113.
-                ALOGE_IF(*written > bytes,
-                        "hal reports more bytes written than asked for: %lld > %lld",
-                        (long long)*written, (long long)bytes);
-            });
-    mStreamPowerLog.log(buffer, *written);
-    return status;
-}
-
-status_t StreamOutHalHidl::callWriterThread(
-        WriteCommand cmd, const char* cmdName,
-        const uint8_t* data, size_t dataSize, StreamOutHalHidl::WriterCallback callback) {
-    if (!mCommandMQ->write(&cmd)) {
-        ALOGE("command message queue write failed for \"%s\"", cmdName);
-        return -EAGAIN;
-    }
-    if (data != nullptr) {
-        size_t availableToWrite = mDataMQ->availableToWrite();
-        if (dataSize > availableToWrite) {
-            ALOGW("truncating write data from %lld to %lld due to insufficient data queue space",
-                    (long long)dataSize, (long long)availableToWrite);
-            dataSize = availableToWrite;
-        }
-        if (!mDataMQ->write(data, dataSize)) {
-            ALOGE("data message queue write failed for \"%s\"", cmdName);
-        }
-    }
-    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY));
-
-    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
-    uint32_t efState = 0;
-retry:
-    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL), &efState);
-    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL)) {
-        WriteStatus writeStatus;
-        writeStatus.retval = Result::NOT_INITIALIZED;
-        if (!mStatusMQ->read(&writeStatus)) {
-            ALOGE("status message read failed for \"%s\"", cmdName);
-        }
-        if (writeStatus.retval == Result::OK) {
-            ret = OK;
-            callback(writeStatus);
-        } else {
-            ret = processReturn(cmdName, writeStatus.retval);
-        }
-        return ret;
-    }
-    if (ret == -EAGAIN || ret == -EINTR) {
-        // Spurious wakeup. This normally retries no more than once.
-        goto retry;
-    }
-    return ret;
-}
-
-status_t StreamOutHalHidl::prepareForWriting(size_t bufferSize) {
-    std::unique_ptr<CommandMQ> tempCommandMQ;
-    std::unique_ptr<DataMQ> tempDataMQ;
-    std::unique_ptr<StatusMQ> tempStatusMQ;
-    Result retval;
-    pid_t halThreadPid, halThreadTid;
-    Return<void> ret = mStream->prepareForWriting(
-            1, bufferSize,
-            [&](Result r,
-                    const CommandMQ::Descriptor& commandMQ,
-                    const DataMQ::Descriptor& dataMQ,
-                    const StatusMQ::Descriptor& statusMQ,
-                    const ThreadInfo& halThreadInfo) {
-                retval = r;
-                if (retval == Result::OK) {
-                    tempCommandMQ.reset(new CommandMQ(commandMQ));
-                    tempDataMQ.reset(new DataMQ(dataMQ));
-                    tempStatusMQ.reset(new StatusMQ(statusMQ));
-                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
-                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
-                    }
-                    halThreadPid = halThreadInfo.pid;
-                    halThreadTid = halThreadInfo.tid;
-                }
-            });
-    if (!ret.isOk() || retval != Result::OK) {
-        return processReturn("prepareForWriting", ret, retval);
-    }
-    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
-            !tempDataMQ || !tempDataMQ->isValid() ||
-            !tempStatusMQ || !tempStatusMQ->isValid() ||
-            !mEfGroup) {
-        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
-        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
-                "Command message queue for writing is invalid");
-        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for writing");
-        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for writing is invalid");
-        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for writing");
-        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
-                "Status message queue for writing is invalid");
-        ALOGE_IF(!mEfGroup, "Event flag creation for writing failed");
-        return NO_INIT;
-    }
-    requestHalThreadPriority(halThreadPid, halThreadTid);
-
-    mCommandMQ = std::move(tempCommandMQ);
-    mDataMQ = std::move(tempDataMQ);
-    mStatusMQ = std::move(tempStatusMQ);
-    mWriterClient = gettid();
-    return OK;
-}
-
-status_t StreamOutHalHidl::getRenderPosition(uint32_t *dspFrames) {
-    if (mStream == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mStream->getRenderPosition(
-            [&](Result r, uint32_t d) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *dspFrames = d;
-                }
-            });
-    return processReturn("getRenderPosition", ret, retval);
-}
-
-status_t StreamOutHalHidl::getNextWriteTimestamp(int64_t *timestamp) {
-    if (mStream == 0) return NO_INIT;
-    Result retval;
-    Return<void> ret = mStream->getNextWriteTimestamp(
-            [&](Result r, int64_t t) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *timestamp = t;
-                }
-            });
-    return processReturn("getRenderPosition", ret, retval);
-}
-
-status_t StreamOutHalHidl::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
-    if (mStream == 0) return NO_INIT;
-    status_t status = processReturn(
-            "setCallback", mStream->setCallback(new StreamOutCallback(this)));
-    if (status == OK) {
-        mCallback = callback;
-    }
-    return status;
-}
-
-status_t StreamOutHalHidl::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
-    if (mStream == 0) return NO_INIT;
-    Return<void> ret = mStream->supportsPauseAndResume(
-            [&](bool p, bool r) {
-                *supportsPause = p;
-                *supportsResume = r;
-            });
-    return processReturn("supportsPauseAndResume", ret);
-}
-
-status_t StreamOutHalHidl::pause() {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("pause", mStream->pause());
-}
-
-status_t StreamOutHalHidl::resume() {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("pause", mStream->resume());
-}
-
-status_t StreamOutHalHidl::supportsDrain(bool *supportsDrain) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("supportsDrain", mStream->supportsDrain(), supportsDrain);
-}
-
-status_t StreamOutHalHidl::drain(bool earlyNotify) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn(
-            "drain", mStream->drain(earlyNotify ? AudioDrain::EARLY_NOTIFY : AudioDrain::ALL));
-}
-
-status_t StreamOutHalHidl::flush() {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("pause", mStream->flush());
-}
-
-status_t StreamOutHalHidl::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
-    if (mStream == 0) return NO_INIT;
-    if (mWriterClient == gettid() && mCommandMQ) {
-        return callWriterThread(
-                WriteCommand::GET_PRESENTATION_POSITION, "getPresentationPosition", nullptr, 0,
-                [&](const WriteStatus& writeStatus) {
-                    *frames = writeStatus.reply.presentationPosition.frames;
-                    timestamp->tv_sec = writeStatus.reply.presentationPosition.timeStamp.tvSec;
-                    timestamp->tv_nsec = writeStatus.reply.presentationPosition.timeStamp.tvNSec;
-                });
-    } else {
-        Result retval;
-        Return<void> ret = mStream->getPresentationPosition(
-                [&](Result r, uint64_t hidlFrames, const TimeSpec& hidlTimeStamp) {
-                    retval = r;
-                    if (retval == Result::OK) {
-                        *frames = hidlFrames;
-                        timestamp->tv_sec = hidlTimeStamp.tvSec;
-                        timestamp->tv_nsec = hidlTimeStamp.tvNSec;
-                    }
-                });
-        return processReturn("getPresentationPosition", ret, retval);
-    }
-}
-
-status_t StreamOutHalHidl::updateSourceMetadata(const SourceMetadata& /* sourceMetadata */) {
-    // Audio HAL V2.0 does not support propagating source metadata
-    return INVALID_OPERATION;
-}
-
-void StreamOutHalHidl::onWriteReady() {
-    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
-    if (callback == 0) return;
-    ALOGV("asyncCallback onWriteReady");
-    callback->onWriteReady();
-}
-
-void StreamOutHalHidl::onDrainReady() {
-    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
-    if (callback == 0) return;
-    ALOGV("asyncCallback onDrainReady");
-    callback->onDrainReady();
-}
-
-void StreamOutHalHidl::onError() {
-    sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
-    if (callback == 0) return;
-    ALOGV("asyncCallback onError");
-    callback->onError();
-}
-
-
-StreamInHalHidl::StreamInHalHidl(const sp<IStreamIn>& stream)
-        : StreamHalHidl(stream.get()), mStream(stream), mReaderClient(0), mEfGroup(nullptr) {
-}
-
-StreamInHalHidl::~StreamInHalHidl() {
-    if (mStream != 0) {
-        processReturn("close", mStream->close());
-        mStream.clear();
-        hardware::IPCThreadState::self()->flushCommands();
-    }
-    if (mEfGroup) {
-        EventFlag::deleteEventFlag(&mEfGroup);
-    }
-}
-
-status_t StreamInHalHidl::getFrameSize(size_t *size) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("getFrameSize", mStream->getFrameSize(), size);
-}
-
-status_t StreamInHalHidl::setGain(float gain) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("setGain", mStream->setGain(gain));
-}
-
-status_t StreamInHalHidl::read(void *buffer, size_t bytes, size_t *read) {
-    if (mStream == 0) return NO_INIT;
-    *read = 0;
-
-    if (bytes == 0 && !mDataMQ) {
-        // Can't determine the size for the MQ buffer. Wait for a non-empty read request.
-        return OK;
-    }
-
-    status_t status;
-    if (!mDataMQ && (status = prepareForReading(bytes)) != OK) {
-        return status;
-    }
-
-    ReadParameters params;
-    params.command = ReadCommand::READ;
-    params.params.read = bytes;
-    status = callReaderThread(params, "read",
-            [&](const ReadStatus& readStatus) {
-                const size_t availToRead = mDataMQ->availableToRead();
-                if (!mDataMQ->read(static_cast<uint8_t*>(buffer), std::min(bytes, availToRead))) {
-                    ALOGE("data message queue read failed for \"read\"");
-                }
-                ALOGW_IF(availToRead != readStatus.reply.read,
-                        "HAL read report inconsistent: mq = %d, status = %d",
-                        (int32_t)availToRead, (int32_t)readStatus.reply.read);
-                *read = readStatus.reply.read;
-            });
-    mStreamPowerLog.log(buffer, *read);
-    return status;
-}
-
-status_t StreamInHalHidl::callReaderThread(
-        const ReadParameters& params, const char* cmdName,
-        StreamInHalHidl::ReaderCallback callback) {
-    if (!mCommandMQ->write(&params)) {
-        ALOGW("command message queue write failed");
-        return -EAGAIN;
-    }
-    mEfGroup->wake(static_cast<uint32_t>(MessageQueueFlagBits::NOT_FULL));
-
-    // TODO: Remove manual event flag handling once blocking MQ is implemented. b/33815422
-    uint32_t efState = 0;
-retry:
-    status_t ret = mEfGroup->wait(static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY), &efState);
-    if (efState & static_cast<uint32_t>(MessageQueueFlagBits::NOT_EMPTY)) {
-        ReadStatus readStatus;
-        readStatus.retval = Result::NOT_INITIALIZED;
-        if (!mStatusMQ->read(&readStatus)) {
-            ALOGE("status message read failed for \"%s\"", cmdName);
-        }
-         if (readStatus.retval == Result::OK) {
-            ret = OK;
-            callback(readStatus);
-        } else {
-            ret = processReturn(cmdName, readStatus.retval);
-        }
-        return ret;
-    }
-    if (ret == -EAGAIN || ret == -EINTR) {
-        // Spurious wakeup. This normally retries no more than once.
-        goto retry;
-    }
-    return ret;
-}
-
-status_t StreamInHalHidl::prepareForReading(size_t bufferSize) {
-    std::unique_ptr<CommandMQ> tempCommandMQ;
-    std::unique_ptr<DataMQ> tempDataMQ;
-    std::unique_ptr<StatusMQ> tempStatusMQ;
-    Result retval;
-    pid_t halThreadPid, halThreadTid;
-    Return<void> ret = mStream->prepareForReading(
-            1, bufferSize,
-            [&](Result r,
-                    const CommandMQ::Descriptor& commandMQ,
-                    const DataMQ::Descriptor& dataMQ,
-                    const StatusMQ::Descriptor& statusMQ,
-                    const ThreadInfo& halThreadInfo) {
-                retval = r;
-                if (retval == Result::OK) {
-                    tempCommandMQ.reset(new CommandMQ(commandMQ));
-                    tempDataMQ.reset(new DataMQ(dataMQ));
-                    tempStatusMQ.reset(new StatusMQ(statusMQ));
-                    if (tempDataMQ->isValid() && tempDataMQ->getEventFlagWord()) {
-                        EventFlag::createEventFlag(tempDataMQ->getEventFlagWord(), &mEfGroup);
-                    }
-                    halThreadPid = halThreadInfo.pid;
-                    halThreadTid = halThreadInfo.tid;
-                }
-            });
-    if (!ret.isOk() || retval != Result::OK) {
-        return processReturn("prepareForReading", ret, retval);
-    }
-    if (!tempCommandMQ || !tempCommandMQ->isValid() ||
-            !tempDataMQ || !tempDataMQ->isValid() ||
-            !tempStatusMQ || !tempStatusMQ->isValid() ||
-            !mEfGroup) {
-        ALOGE_IF(!tempCommandMQ, "Failed to obtain command message queue for writing");
-        ALOGE_IF(tempCommandMQ && !tempCommandMQ->isValid(),
-                "Command message queue for writing is invalid");
-        ALOGE_IF(!tempDataMQ, "Failed to obtain data message queue for reading");
-        ALOGE_IF(tempDataMQ && !tempDataMQ->isValid(), "Data message queue for reading is invalid");
-        ALOGE_IF(!tempStatusMQ, "Failed to obtain status message queue for reading");
-        ALOGE_IF(tempStatusMQ && !tempStatusMQ->isValid(),
-                "Status message queue for reading is invalid");
-        ALOGE_IF(!mEfGroup, "Event flag creation for reading failed");
-        return NO_INIT;
-    }
-    requestHalThreadPriority(halThreadPid, halThreadTid);
-
-    mCommandMQ = std::move(tempCommandMQ);
-    mDataMQ = std::move(tempDataMQ);
-    mStatusMQ = std::move(tempStatusMQ);
-    mReaderClient = gettid();
-    return OK;
-}
-
-status_t StreamInHalHidl::getInputFramesLost(uint32_t *framesLost) {
-    if (mStream == 0) return NO_INIT;
-    return processReturn("getInputFramesLost", mStream->getInputFramesLost(), framesLost);
-}
-
-status_t StreamInHalHidl::getCapturePosition(int64_t *frames, int64_t *time) {
-    if (mStream == 0) return NO_INIT;
-    if (mReaderClient == gettid() && mCommandMQ) {
-        ReadParameters params;
-        params.command = ReadCommand::GET_CAPTURE_POSITION;
-        return callReaderThread(params, "getCapturePosition",
-                [&](const ReadStatus& readStatus) {
-                    *frames = readStatus.reply.capturePosition.frames;
-                    *time = readStatus.reply.capturePosition.time;
-                });
-    } else {
-        Result retval;
-        Return<void> ret = mStream->getCapturePosition(
-                [&](Result r, uint64_t hidlFrames, uint64_t hidlTime) {
-                    retval = r;
-                    if (retval == Result::OK) {
-                        *frames = hidlFrames;
-                        *time = hidlTime;
-                    }
-                });
-        return processReturn("getCapturePosition", ret, retval);
-    }
-}
-
-status_t StreamInHalHidl::getActiveMicrophones(
-        std::vector<media::MicrophoneInfo> *microphones __unused) {
-    if (mStream == 0) return NO_INIT;
-    return INVALID_OPERATION;
-}
-
-status_t StreamInHalHidl::updateSinkMetadata(const SinkMetadata& /* sinkMetadata */) {
-    // Audio HAL V2.0 does not support propagating sink metadata
-    return INVALID_OPERATION;
-}
-
-} // namespace android
diff --git a/media/libaudiohal/2.0/StreamHalLocal.cpp b/media/libaudiohal/2.0/StreamHalLocal.cpp
deleted file mode 100644
index 98107e5..0000000
--- a/media/libaudiohal/2.0/StreamHalLocal.cpp
+++ /dev/null
@@ -1,347 +0,0 @@
-/*
- * Copyright (C) 2016 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 "StreamHalLocal"
-//#define LOG_NDEBUG 0
-
-#include <hardware/audio.h>
-#include <utils/Log.h>
-
-#include "DeviceHalLocal.h"
-#include "StreamHalLocal.h"
-
-namespace android {
-
-StreamHalLocal::StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device)
-        : mDevice(device),
-          mStream(stream) {
-    // Instrument audio signal power logging.
-    // Note: This assumes channel mask, format, and sample rate do not change after creation.
-    if (mStream != nullptr && mStreamPowerLog.isUserDebugOrEngBuild()) {
-        mStreamPowerLog.init(mStream->get_sample_rate(mStream),
-                mStream->get_channels(mStream),
-                mStream->get_format(mStream));
-    }
-}
-
-StreamHalLocal::~StreamHalLocal() {
-    mStream = 0;
-    mDevice.clear();
-}
-
-status_t StreamHalLocal::getSampleRate(uint32_t *rate) {
-    *rate = mStream->get_sample_rate(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getBufferSize(size_t *size) {
-    *size = mStream->get_buffer_size(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getChannelMask(audio_channel_mask_t *mask) {
-    *mask = mStream->get_channels(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getFormat(audio_format_t *format) {
-    *format = mStream->get_format(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::getAudioProperties(
-        uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format) {
-    *sampleRate = mStream->get_sample_rate(mStream);
-    *mask = mStream->get_channels(mStream);
-    *format = mStream->get_format(mStream);
-    return OK;
-}
-
-status_t StreamHalLocal::setParameters(const String8& kvPairs) {
-    return mStream->set_parameters(mStream, kvPairs.string());
-}
-
-status_t StreamHalLocal::getParameters(const String8& keys, String8 *values) {
-    char *halValues = mStream->get_parameters(mStream, keys.string());
-    if (halValues != NULL) {
-        values->setTo(halValues);
-        free(halValues);
-    } else {
-        values->clear();
-    }
-    return OK;
-}
-
-status_t StreamHalLocal::addEffect(sp<EffectHalInterface>) {
-    LOG_ALWAYS_FATAL("Local streams can not have effects");
-    return INVALID_OPERATION;
-}
-
-status_t StreamHalLocal::removeEffect(sp<EffectHalInterface>) {
-    LOG_ALWAYS_FATAL("Local streams can not have effects");
-    return INVALID_OPERATION;
-}
-
-status_t StreamHalLocal::standby() {
-    return mStream->standby(mStream);
-}
-
-status_t StreamHalLocal::dump(int fd) {
-    status_t status = mStream->dump(mStream, fd);
-    mStreamPowerLog.dump(fd);
-    return status;
-}
-
-status_t StreamHalLocal::setHalThreadPriority(int) {
-    // Don't need to do anything as local hal is executed by audioflinger directly
-    // on the same thread.
-    return OK;
-}
-
-StreamOutHalLocal::StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device)
-        : StreamHalLocal(&stream->common, device), mStream(stream) {
-}
-
-StreamOutHalLocal::~StreamOutHalLocal() {
-    mCallback.clear();
-    mDevice->closeOutputStream(mStream);
-    mStream = 0;
-}
-
-status_t StreamOutHalLocal::getFrameSize(size_t *size) {
-    *size = audio_stream_out_frame_size(mStream);
-    return OK;
-}
-
-status_t StreamOutHalLocal::getLatency(uint32_t *latency) {
-    *latency = mStream->get_latency(mStream);
-    return OK;
-}
-
-status_t StreamOutHalLocal::setVolume(float left, float right) {
-    if (mStream->set_volume == NULL) return INVALID_OPERATION;
-    return mStream->set_volume(mStream, left, right);
-}
-
-status_t StreamOutHalLocal::write(const void *buffer, size_t bytes, size_t *written) {
-    ssize_t writeResult = mStream->write(mStream, buffer, bytes);
-    if (writeResult > 0) {
-        *written = writeResult;
-        mStreamPowerLog.log(buffer, *written);
-        return OK;
-    } else {
-        *written = 0;
-        return writeResult;
-    }
-}
-
-status_t StreamOutHalLocal::getRenderPosition(uint32_t *dspFrames) {
-    return mStream->get_render_position(mStream, dspFrames);
-}
-
-status_t StreamOutHalLocal::getNextWriteTimestamp(int64_t *timestamp) {
-    if (mStream->get_next_write_timestamp == NULL) return INVALID_OPERATION;
-    return mStream->get_next_write_timestamp(mStream, timestamp);
-}
-
-status_t StreamOutHalLocal::setCallback(wp<StreamOutHalInterfaceCallback> callback) {
-    if (mStream->set_callback == NULL) return INVALID_OPERATION;
-    status_t result = mStream->set_callback(mStream, StreamOutHalLocal::asyncCallback, this);
-    if (result == OK) {
-        mCallback = callback;
-    }
-    return result;
-}
-
-// static
-int StreamOutHalLocal::asyncCallback(stream_callback_event_t event, void*, void *cookie) {
-    // We act as if we gave a wp<StreamOutHalLocal> to HAL. This way we should handle
-    // correctly the case when the callback is invoked while StreamOutHalLocal's destructor is
-    // already running, because the destructor is invoked after the refcount has been atomically
-    // decremented.
-    wp<StreamOutHalLocal> weakSelf(static_cast<StreamOutHalLocal*>(cookie));
-    sp<StreamOutHalLocal> self = weakSelf.promote();
-    if (self == 0) return 0;
-    sp<StreamOutHalInterfaceCallback> callback = self->mCallback.promote();
-    if (callback == 0) return 0;
-    ALOGV("asyncCallback() event %d", event);
-    switch (event) {
-        case STREAM_CBK_EVENT_WRITE_READY:
-            callback->onWriteReady();
-            break;
-        case STREAM_CBK_EVENT_DRAIN_READY:
-            callback->onDrainReady();
-            break;
-        case STREAM_CBK_EVENT_ERROR:
-            callback->onError();
-            break;
-        default:
-            ALOGW("asyncCallback() unknown event %d", event);
-            break;
-    }
-    return 0;
-}
-
-status_t StreamOutHalLocal::supportsPauseAndResume(bool *supportsPause, bool *supportsResume) {
-    *supportsPause = mStream->pause != NULL;
-    *supportsResume = mStream->resume != NULL;
-    return OK;
-}
-
-status_t StreamOutHalLocal::pause() {
-    if (mStream->pause == NULL) return INVALID_OPERATION;
-    return mStream->pause(mStream);
-}
-
-status_t StreamOutHalLocal::resume() {
-    if (mStream->resume == NULL) return INVALID_OPERATION;
-    return mStream->resume(mStream);
-}
-
-status_t StreamOutHalLocal::supportsDrain(bool *supportsDrain) {
-    *supportsDrain = mStream->drain != NULL;
-    return OK;
-}
-
-status_t StreamOutHalLocal::drain(bool earlyNotify) {
-    if (mStream->drain == NULL) return INVALID_OPERATION;
-    return mStream->drain(mStream, earlyNotify ? AUDIO_DRAIN_EARLY_NOTIFY : AUDIO_DRAIN_ALL);
-}
-
-status_t StreamOutHalLocal::flush() {
-    if (mStream->flush == NULL) return INVALID_OPERATION;
-    return mStream->flush(mStream);
-}
-
-status_t StreamOutHalLocal::getPresentationPosition(uint64_t *frames, struct timespec *timestamp) {
-    if (mStream->get_presentation_position == NULL) return INVALID_OPERATION;
-    return mStream->get_presentation_position(mStream, frames, timestamp);
-}
-
-status_t StreamOutHalLocal::updateSourceMetadata(const SourceMetadata& sourceMetadata) {
-    if (mStream->update_source_metadata == nullptr) {
-        return INVALID_OPERATION;
-    }
-    const source_metadata_t metadata {
-        .track_count = sourceMetadata.tracks.size(),
-        // const cast is fine as it is in a const structure
-        .tracks = const_cast<playback_track_metadata*>(sourceMetadata.tracks.data()),
-    };
-    mStream->update_source_metadata(mStream, &metadata);
-    return OK;
-}
-
-status_t StreamOutHalLocal::start() {
-    if (mStream->start == NULL) return INVALID_OPERATION;
-    return mStream->start(mStream);
-}
-
-status_t StreamOutHalLocal::stop() {
-    if (mStream->stop == NULL) return INVALID_OPERATION;
-    return mStream->stop(mStream);
-}
-
-status_t StreamOutHalLocal::createMmapBuffer(int32_t minSizeFrames,
-                                  struct audio_mmap_buffer_info *info) {
-    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
-    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
-}
-
-status_t StreamOutHalLocal::getMmapPosition(struct audio_mmap_position *position) {
-    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
-    return mStream->get_mmap_position(mStream, position);
-}
-
-StreamInHalLocal::StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device)
-        : StreamHalLocal(&stream->common, device), mStream(stream) {
-}
-
-StreamInHalLocal::~StreamInHalLocal() {
-    mDevice->closeInputStream(mStream);
-    mStream = 0;
-}
-
-status_t StreamInHalLocal::getFrameSize(size_t *size) {
-    *size = audio_stream_in_frame_size(mStream);
-    return OK;
-}
-
-status_t StreamInHalLocal::setGain(float gain) {
-    return mStream->set_gain(mStream, gain);
-}
-
-status_t StreamInHalLocal::read(void *buffer, size_t bytes, size_t *read) {
-    ssize_t readResult = mStream->read(mStream, buffer, bytes);
-    if (readResult > 0) {
-        *read = readResult;
-        mStreamPowerLog.log( buffer, *read);
-        return OK;
-    } else {
-        *read = 0;
-        return readResult;
-    }
-}
-
-status_t StreamInHalLocal::getInputFramesLost(uint32_t *framesLost) {
-    *framesLost = mStream->get_input_frames_lost(mStream);
-    return OK;
-}
-
-status_t StreamInHalLocal::getCapturePosition(int64_t *frames, int64_t *time) {
-    if (mStream->get_capture_position == NULL) return INVALID_OPERATION;
-    return mStream->get_capture_position(mStream, frames, time);
-}
-
-status_t StreamInHalLocal::updateSinkMetadata(const SinkMetadata& sinkMetadata) {
-    if (mStream->update_sink_metadata == nullptr) {
-        return INVALID_OPERATION;
-    }
-    const sink_metadata_t metadata {
-        .track_count = sinkMetadata.tracks.size(),
-        // const cast is fine as it is in a const structure
-        .tracks = const_cast<record_track_metadata*>(sinkMetadata.tracks.data()),
-    };
-    mStream->update_sink_metadata(mStream, &metadata);
-    return OK;
-}
-
-status_t StreamInHalLocal::start() {
-    if (mStream->start == NULL) return INVALID_OPERATION;
-    return mStream->start(mStream);
-}
-
-status_t StreamInHalLocal::stop() {
-    if (mStream->stop == NULL) return INVALID_OPERATION;
-    return mStream->stop(mStream);
-}
-
-status_t StreamInHalLocal::createMmapBuffer(int32_t minSizeFrames,
-                                  struct audio_mmap_buffer_info *info) {
-    if (mStream->create_mmap_buffer == NULL) return INVALID_OPERATION;
-    return mStream->create_mmap_buffer(mStream, minSizeFrames, info);
-}
-
-status_t StreamInHalLocal::getMmapPosition(struct audio_mmap_position *position) {
-    if (mStream->get_mmap_position == NULL) return INVALID_OPERATION;
-    return mStream->get_mmap_position(mStream, position);
-}
-
-status_t StreamInHalLocal::getActiveMicrophones(
-        std::vector<media::MicrophoneInfo> *microphones __unused) {
-    return INVALID_OPERATION;
-}
-
-} // namespace android
diff --git a/media/libaudiohal/4.0/ConversionHelperHidl.h b/media/libaudiohal/4.0/ConversionHelperHidl.h
deleted file mode 100644
index 8823a8d..0000000
--- a/media/libaudiohal/4.0/ConversionHelperHidl.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_4_0_H
-#define ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_4_0_H
-
-#include <android/hardware/audio/4.0/types.h>
-#include <hidl/HidlSupport.h>
-#include <system/audio.h>
-#include <utils/String8.h>
-
-using ::android::hardware::audio::V4_0::ParameterValue;
-using ::android::hardware::audio::V4_0::MicrophoneInfo;
-using ::android::hardware::Return;
-using ::android::hardware::hidl_string;
-using ::android::hardware::hidl_vec;
-
-namespace android {
-namespace V4_0 {
-
-class ConversionHelperHidl {
-  protected:
-    static status_t keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys);
-    static status_t parametersFromHal(const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams);
-    static void parametersToHal(const hidl_vec<ParameterValue>& parameters, String8 *values);
-    static void microphoneInfoToHal(const MicrophoneInfo& src,
-                                    audio_microphone_characteristic_t *pDst);
-
-    ConversionHelperHidl(const char* className);
-
-    template<typename R, typename T>
-    status_t processReturn(const char* funcName, const Return<R>& ret, T *retval) {
-        if (ret.isOk()) {
-            // This way it also works for enum class to unscoped enum conversion.
-            *retval = static_cast<T>(static_cast<R>(ret));
-            return OK;
-        }
-        return processReturn(funcName, ret);
-    }
-
-    template<typename T>
-    status_t processReturn(const char* funcName, const Return<T>& ret) {
-        if (!ret.isOk()) {
-            emitError(funcName, ret.description().c_str());
-        }
-        return ret.isOk() ? OK : FAILED_TRANSACTION;
-    }
-
-    status_t processReturn(const char* funcName, const Return<hardware::audio::V4_0::Result>& ret) {
-        if (!ret.isOk()) {
-            emitError(funcName, ret.description().c_str());
-        }
-        return ret.isOk() ? analyzeResult(ret) : FAILED_TRANSACTION;
-    }
-
-    template<typename T>
-    status_t processReturn(
-            const char* funcName, const Return<T>& ret, hardware::audio::V4_0::Result retval) {
-        if (!ret.isOk()) {
-            emitError(funcName, ret.description().c_str());
-        }
-        return ret.isOk() ? analyzeResult(retval) : FAILED_TRANSACTION;
-    }
-
-  private:
-    const char* mClassName;
-
-    static status_t analyzeResult(const hardware::audio::V4_0::Result& result);
-
-    void emitError(const char* funcName, const char* description);
-};
-
-}  // namespace V4_0
-}  // namespace android
-
-#endif // ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_4_0_H
diff --git a/media/libaudiohal/4.0/DeviceHalHidl.h b/media/libaudiohal/4.0/DeviceHalHidl.h
deleted file mode 100644
index 0bd2175..0000000
--- a/media/libaudiohal/4.0/DeviceHalHidl.h
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICE_HAL_HIDL_4_0_H
-#define ANDROID_HARDWARE_DEVICE_HAL_HIDL_4_0_H
-
-#include <android/hardware/audio/4.0/IDevice.h>
-#include <android/hardware/audio/4.0/IPrimaryDevice.h>
-#include <media/audiohal/DeviceHalInterface.h>
-
-#include "ConversionHelperHidl.h"
-
-using ::android::hardware::audio::V4_0::IDevice;
-using ::android::hardware::audio::V4_0::IPrimaryDevice;
-using ::android::hardware::Return;
-
-namespace android {
-namespace V4_0 {
-
-class DeviceHalHidl : public DeviceHalInterface, public ConversionHelperHidl
-{
-  public:
-    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
-    virtual status_t getSupportedDevices(uint32_t *devices);
-
-    // Check to see if the audio hardware interface has been initialized.
-    virtual status_t initCheck();
-
-    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
-    virtual status_t setVoiceVolume(float volume);
-
-    // Set the audio volume for all audio activities other than voice call.
-    virtual status_t setMasterVolume(float volume);
-
-    // Get the current master volume value for the HAL.
-    virtual status_t getMasterVolume(float *volume);
-
-    // Called when the audio mode changes.
-    virtual status_t setMode(audio_mode_t mode);
-
-    // Muting control.
-    virtual status_t setMicMute(bool state);
-    virtual status_t getMicMute(bool *state);
-    virtual status_t setMasterMute(bool state);
-    virtual status_t getMasterMute(bool *state);
-
-    // Set global audio parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get global audio parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Returns audio input buffer size according to parameters passed.
-    virtual status_t getInputBufferSize(const struct audio_config *config,
-            size_t *size);
-
-    // Creates and opens the audio hardware output stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openOutputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            audio_output_flags_t flags,
-            struct audio_config *config,
-            const char *address,
-            sp<StreamOutHalInterface> *outStream);
-
-    // Creates and opens the audio hardware input stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openInputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            struct audio_config *config,
-            audio_input_flags_t flags,
-            const char *address,
-            audio_source_t source,
-            sp<StreamInHalInterface> *inStream);
-
-    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
-    virtual status_t supportsAudioPatches(bool *supportsPatches);
-
-    // Creates an audio patch between several source and sink ports.
-    virtual status_t createAudioPatch(
-            unsigned int num_sources,
-            const struct audio_port_config *sources,
-            unsigned int num_sinks,
-            const struct audio_port_config *sinks,
-            audio_patch_handle_t *patch);
-
-    // Releases an audio patch.
-    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
-
-    // Fills the list of supported attributes for a given audio port.
-    virtual status_t getAudioPort(struct audio_port *port);
-
-    // Set audio port configuration.
-    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
-
-    // List microphones
-    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
-
-    virtual status_t dump(int fd);
-
-  private:
-    friend class DevicesFactoryHalHidl;
-    sp<IDevice> mDevice;
-    sp<IPrimaryDevice> mPrimaryDevice;  // Null if it's not a primary device.
-
-    // Can not be constructed directly by clients.
-    explicit DeviceHalHidl(const sp<IDevice>& device);
-
-    // The destructor automatically closes the device.
-    virtual ~DeviceHalHidl();
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICE_HAL_HIDL_4_0_H
diff --git a/media/libaudiohal/4.0/DeviceHalLocal.h b/media/libaudiohal/4.0/DeviceHalLocal.h
deleted file mode 100644
index 08341a4..0000000
--- a/media/libaudiohal/4.0/DeviceHalLocal.h
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICE_HAL_LOCAL_4_0_H
-#define ANDROID_HARDWARE_DEVICE_HAL_LOCAL_4_0_H
-
-#include <hardware/audio.h>
-#include <media/audiohal/DeviceHalInterface.h>
-
-namespace android {
-namespace V4_0 {
-
-class DeviceHalLocal : public DeviceHalInterface
-{
-  public:
-    // Sets the value of 'devices' to a bitmask of 1 or more values of audio_devices_t.
-    virtual status_t getSupportedDevices(uint32_t *devices);
-
-    // Check to see if the audio hardware interface has been initialized.
-    virtual status_t initCheck();
-
-    // Set the audio volume of a voice call. Range is between 0.0 and 1.0.
-    virtual status_t setVoiceVolume(float volume);
-
-    // Set the audio volume for all audio activities other than voice call.
-    virtual status_t setMasterVolume(float volume);
-
-    // Get the current master volume value for the HAL.
-    virtual status_t getMasterVolume(float *volume);
-
-    // Called when the audio mode changes.
-    virtual status_t setMode(audio_mode_t mode);
-
-    // Muting control.
-    virtual status_t setMicMute(bool state);
-    virtual status_t getMicMute(bool *state);
-    virtual status_t setMasterMute(bool state);
-    virtual status_t getMasterMute(bool *state);
-
-    // Set global audio parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get global audio parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Returns audio input buffer size according to parameters passed.
-    virtual status_t getInputBufferSize(const struct audio_config *config,
-            size_t *size);
-
-    // Creates and opens the audio hardware output stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openOutputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            audio_output_flags_t flags,
-            struct audio_config *config,
-            const char *address,
-            sp<StreamOutHalInterface> *outStream);
-
-    // Creates and opens the audio hardware input stream. The stream is closed
-    // by releasing all references to the returned object.
-    virtual status_t openInputStream(
-            audio_io_handle_t handle,
-            audio_devices_t devices,
-            struct audio_config *config,
-            audio_input_flags_t flags,
-            const char *address,
-            audio_source_t source,
-            sp<StreamInHalInterface> *inStream);
-
-    // Returns whether createAudioPatch and releaseAudioPatch operations are supported.
-    virtual status_t supportsAudioPatches(bool *supportsPatches);
-
-    // Creates an audio patch between several source and sink ports.
-    virtual status_t createAudioPatch(
-            unsigned int num_sources,
-            const struct audio_port_config *sources,
-            unsigned int num_sinks,
-            const struct audio_port_config *sinks,
-            audio_patch_handle_t *patch);
-
-    // Releases an audio patch.
-    virtual status_t releaseAudioPatch(audio_patch_handle_t patch);
-
-    // Fills the list of supported attributes for a given audio port.
-    virtual status_t getAudioPort(struct audio_port *port);
-
-    // Set audio port configuration.
-    virtual status_t setAudioPortConfig(const struct audio_port_config *config);
-
-    // List microphones
-    virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
-
-    virtual status_t dump(int fd);
-
-    void closeOutputStream(struct audio_stream_out *stream_out);
-    void closeInputStream(struct audio_stream_in *stream_in);
-
-  private:
-    audio_hw_device_t *mDev;
-
-    friend class DevicesFactoryHalLocal;
-
-    // Can not be constructed directly by clients.
-    explicit DeviceHalLocal(audio_hw_device_t *dev);
-
-    // The destructor automatically closes the device.
-    virtual ~DeviceHalLocal();
-
-    uint32_t version() const { return mDev->common.version; }
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICE_HAL_LOCAL_4_0_H
diff --git a/media/libaudiohal/4.0/DevicesFactoryHalHidl.cpp b/media/libaudiohal/4.0/DevicesFactoryHalHidl.cpp
deleted file mode 100644
index c83194e..0000000
--- a/media/libaudiohal/4.0/DevicesFactoryHalHidl.cpp
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <string.h>
-
-#define LOG_TAG "DevicesFactoryHalHidl"
-//#define LOG_NDEBUG 0
-
-#include <android/hardware/audio/4.0/IDevice.h>
-#include <media/audiohal/hidl/HalDeathHandler.h>
-#include <utils/Log.h>
-
-#include "ConversionHelperHidl.h"
-#include "DeviceHalHidl.h"
-#include "DevicesFactoryHalHidl.h"
-
-using ::android::hardware::audio::V4_0::IDevice;
-using ::android::hardware::audio::V4_0::Result;
-using ::android::hardware::Return;
-
-namespace android {
-namespace V4_0 {
-
-DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
-    mDevicesFactory = IDevicesFactory::getService();
-    if (mDevicesFactory != 0) {
-        // It is assumed that DevicesFactory is owned by AudioFlinger
-        // and thus have the same lifespan.
-        mDevicesFactory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
-    } else {
-        ALOGE("Failed to obtain IDevicesFactory service, terminating process.");
-        exit(1);
-    }
-    // The MSD factory is optional
-    mDevicesFactoryMsd = IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD);
-    // TODO: Register death handler, and add 'restart' directive to audioserver.rc
-}
-
-DevicesFactoryHalHidl::~DevicesFactoryHalHidl() {
-}
-
-status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    if (mDevicesFactory == 0) return NO_INIT;
-    Result retval = Result::NOT_INITIALIZED;
-    Return<void> ret = mDevicesFactory->openDevice(
-            name,
-            [&](Result r, const sp<IDevice>& result) {
-                retval = r;
-                if (retval == Result::OK) {
-                    *device = new DeviceHalHidl(result);
-                }
-            });
-    if (ret.isOk()) {
-        if (retval == Result::OK) return OK;
-        else if (retval == Result::INVALID_ARGUMENTS) return BAD_VALUE;
-        else return NO_INIT;
-    }
-    return FAILED_TRANSACTION;
-}
-
-} // namespace V4_0
-} // namespace android
diff --git a/media/libaudiohal/4.0/DevicesFactoryHalHidl.h b/media/libaudiohal/4.0/DevicesFactoryHalHidl.h
deleted file mode 100644
index 114889b..0000000
--- a/media/libaudiohal/4.0/DevicesFactoryHalHidl.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_4_0_H
-#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_4_0_H
-
-#include <android/hardware/audio/4.0/IDevicesFactory.h>
-#include <media/audiohal/DevicesFactoryHalInterface.h>
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-
-#include "DeviceHalHidl.h"
-
-using ::android::hardware::audio::V4_0::IDevicesFactory;
-
-namespace android {
-namespace V4_0 {
-
-class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
-{
-  public:
-    // Opens a device with the specified name. To close the device, it is
-    // necessary to release references to the returned object.
-    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
-
-  private:
-    friend class DevicesFactoryHalHybrid;
-
-    sp<IDevicesFactory> mDevicesFactory;
-    sp<IDevicesFactory> mDevicesFactoryMsd;
-
-    // Can not be constructed directly by clients.
-    DevicesFactoryHalHidl();
-
-    virtual ~DevicesFactoryHalHidl();
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_4_0_H
diff --git a/media/libaudiohal/4.0/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/4.0/DevicesFactoryHalHybrid.cpp
deleted file mode 100644
index 7ff1ec7d..0000000
--- a/media/libaudiohal/4.0/DevicesFactoryHalHybrid.cpp
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (C) 2017 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 "DevicesFactoryHalHybrid"
-//#define LOG_NDEBUG 0
-
-#include <libaudiohal/4.0/DevicesFactoryHalHybrid.h>
-#include "DevicesFactoryHalLocal.h"
-#include "DevicesFactoryHalHidl.h"
-
-namespace android {
-namespace V4_0 {
-
-DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
-        : mLocalFactory(new DevicesFactoryHalLocal()),
-          mHidlFactory(new DevicesFactoryHalHidl()) {
-}
-
-DevicesFactoryHalHybrid::~DevicesFactoryHalHybrid() {
-}
-
-status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
-    if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
-        strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
-        return mHidlFactory->openDevice(name, device);
-    }
-    return mLocalFactory->openDevice(name, device);
-}
-
-} // namespace V4_0
-} // namespace android
diff --git a/media/libaudiohal/4.0/DevicesFactoryHalLocal.h b/media/libaudiohal/4.0/DevicesFactoryHalLocal.h
deleted file mode 100644
index bc1c521..0000000
--- a/media/libaudiohal/4.0/DevicesFactoryHalLocal.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_4_0_H
-#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_4_0_H
-
-#include <media/audiohal/DevicesFactoryHalInterface.h>
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-
-#include "DeviceHalLocal.h"
-
-namespace android {
-namespace V4_0 {
-
-class DevicesFactoryHalLocal : public DevicesFactoryHalInterface
-{
-  public:
-    // Opens a device with the specified name. To close the device, it is
-    // necessary to release references to the returned object.
-    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
-
-  private:
-    friend class DevicesFactoryHalHybrid;
-
-    // Can not be constructed directly by clients.
-    DevicesFactoryHalLocal() {}
-
-    virtual ~DevicesFactoryHalLocal() {}
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_4_0_H
diff --git a/media/libaudiohal/4.0/EffectBufferHalHidl.h b/media/libaudiohal/4.0/EffectBufferHalHidl.h
deleted file mode 100644
index 6d578c6..0000000
--- a/media/libaudiohal/4.0/EffectBufferHalHidl.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_4_0_H
-#define ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_4_0_H
-
-#include <android/hardware/audio/effect/4.0/types.h>
-#include <android/hidl/memory/1.0/IMemory.h>
-#include <hidl/HidlSupport.h>
-#include <media/audiohal/EffectBufferHalInterface.h>
-#include <system/audio_effect.h>
-
-using android::hardware::audio::effect::V4_0::AudioBuffer;
-using android::hardware::hidl_memory;
-using android::hidl::memory::V1_0::IMemory;
-
-namespace android {
-namespace V4_0 {
-
-class EffectBufferHalHidl : public EffectBufferHalInterface
-{
-  public:
-    static status_t allocate(size_t size, sp<EffectBufferHalInterface>* buffer);
-    static status_t mirror(void* external, size_t size, sp<EffectBufferHalInterface>* buffer);
-
-    virtual audio_buffer_t* audioBuffer();
-    virtual void* externalData() const;
-
-    virtual size_t getSize() const override { return mBufferSize; }
-
-    virtual void setExternalData(void* external);
-    virtual void setFrameCount(size_t frameCount);
-    virtual bool checkFrameCountChange();
-
-    virtual void update();
-    virtual void commit();
-    virtual void update(size_t size);
-    virtual void commit(size_t size);
-
-    const AudioBuffer& hidlBuffer() const { return mHidlBuffer; }
-
-  private:
-    friend class EffectBufferHalInterface;
-
-    static uint64_t makeUniqueId();
-
-    const size_t mBufferSize;
-    bool mFrameCountChanged;
-    void* mExternalData;
-    AudioBuffer mHidlBuffer;
-    sp<IMemory> mMemory;
-    audio_buffer_t mAudioBuffer;
-
-    // Can not be constructed directly by clients.
-    explicit EffectBufferHalHidl(size_t size);
-
-    virtual ~EffectBufferHalHidl();
-
-    status_t init();
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_4_0_H
diff --git a/media/libaudiohal/4.0/EffectHalHidl.h b/media/libaudiohal/4.0/EffectHalHidl.h
deleted file mode 100644
index 5a4dab1..0000000
--- a/media/libaudiohal/4.0/EffectHalHidl.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECT_HAL_HIDL_4_0_H
-#define ANDROID_HARDWARE_EFFECT_HAL_HIDL_4_0_H
-
-#include <android/hardware/audio/effect/4.0/IEffect.h>
-#include <media/audiohal/EffectHalInterface.h>
-#include <fmq/EventFlag.h>
-#include <fmq/MessageQueue.h>
-#include <system/audio_effect.h>
-
-using ::android::hardware::audio::effect::V4_0::EffectBufferConfig;
-using ::android::hardware::audio::effect::V4_0::EffectConfig;
-using ::android::hardware::audio::effect::V4_0::EffectDescriptor;
-using ::android::hardware::audio::effect::V4_0::IEffect;
-using ::android::hardware::EventFlag;
-using ::android::hardware::MessageQueue;
-
-namespace android {
-namespace V4_0 {
-
-class EffectHalHidl : public EffectHalInterface
-{
-  public:
-    // Set the input buffer.
-    virtual status_t setInBuffer(const sp<EffectBufferHalInterface>& buffer);
-
-    // Set the output buffer.
-    virtual status_t setOutBuffer(const sp<EffectBufferHalInterface>& buffer);
-
-    // Effect process function.
-    virtual status_t process();
-
-    // Process reverse stream function. This function is used to pass
-    // a reference stream to the effect engine.
-    virtual status_t processReverse();
-
-    // Send a command and receive a response to/from effect engine.
-    virtual status_t command(uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
-            uint32_t *replySize, void *pReplyData);
-
-    // Returns the effect descriptor.
-    virtual status_t getDescriptor(effect_descriptor_t *pDescriptor);
-
-    // Free resources on the remote side.
-    virtual status_t close();
-
-    // Whether it's a local implementation.
-    virtual bool isLocal() const { return false; }
-
-    uint64_t effectId() const { return mEffectId; }
-
-    static void effectDescriptorToHal(
-            const EffectDescriptor& descriptor, effect_descriptor_t* halDescriptor);
-
-  private:
-    friend class EffectsFactoryHalHidl;
-    typedef MessageQueue<
-        hardware::audio::effect::V4_0::Result, hardware::kSynchronizedReadWrite> StatusMQ;
-
-    sp<IEffect> mEffect;
-    const uint64_t mEffectId;
-    sp<EffectBufferHalInterface> mInBuffer;
-    sp<EffectBufferHalInterface> mOutBuffer;
-    bool mBuffersChanged;
-    std::unique_ptr<StatusMQ> mStatusMQ;
-    EventFlag* mEfGroup;
-
-    static status_t analyzeResult(const hardware::audio::effect::V4_0::Result& result);
-    static void effectBufferConfigFromHal(
-            const buffer_config_t& halConfig, EffectBufferConfig* config);
-    static void effectBufferConfigToHal(
-            const EffectBufferConfig& config, buffer_config_t* halConfig);
-    static void effectConfigFromHal(const effect_config_t& halConfig, EffectConfig* config);
-    static void effectConfigToHal(const EffectConfig& config, effect_config_t* halConfig);
-
-    // Can not be constructed directly by clients.
-    EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId);
-
-    // The destructor automatically releases the effect.
-    virtual ~EffectHalHidl();
-
-    status_t getConfigImpl(uint32_t cmdCode, uint32_t *replySize, void *pReplyData);
-    status_t prepareForProcessing();
-    bool needToResetBuffers();
-    status_t processImpl(uint32_t mqFlag);
-    status_t setConfigImpl(
-            uint32_t cmdCode, uint32_t cmdSize, void *pCmdData,
-            uint32_t *replySize, void *pReplyData);
-    status_t setProcessBuffers();
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECT_HAL_HIDL_4_0_H
diff --git a/media/libaudiohal/4.0/StreamHalHidl.h b/media/libaudiohal/4.0/StreamHalHidl.h
deleted file mode 100644
index 2dda0f8..0000000
--- a/media/libaudiohal/4.0/StreamHalHidl.h
+++ /dev/null
@@ -1,250 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_STREAM_HAL_HIDL_4_0_H
-#define ANDROID_HARDWARE_STREAM_HAL_HIDL_4_0_H
-
-#include <atomic>
-
-#include <android/hardware/audio/4.0/IStream.h>
-#include <android/hardware/audio/4.0/IStreamIn.h>
-#include <android/hardware/audio/4.0/IStreamOut.h>
-#include <fmq/EventFlag.h>
-#include <fmq/MessageQueue.h>
-#include <media/audiohal/StreamHalInterface.h>
-
-#include "ConversionHelperHidl.h"
-#include "StreamPowerLog.h"
-
-using ::android::hardware::audio::V4_0::IStream;
-using ::android::hardware::audio::V4_0::IStreamIn;
-using ::android::hardware::audio::V4_0::IStreamOut;
-using ::android::hardware::EventFlag;
-using ::android::hardware::MessageQueue;
-using ::android::hardware::Return;
-using ReadParameters = ::android::hardware::audio::V4_0::IStreamIn::ReadParameters;
-using ReadStatus = ::android::hardware::audio::V4_0::IStreamIn::ReadStatus;
-using WriteCommand = ::android::hardware::audio::V4_0::IStreamOut::WriteCommand;
-using WriteStatus = ::android::hardware::audio::V4_0::IStreamOut::WriteStatus;
-
-namespace android {
-namespace V4_0 {
-
-class DeviceHalHidl;
-
-class StreamHalHidl : public virtual StreamHalInterface, public ConversionHelperHidl
-{
-  public:
-    // Return the sampling rate in Hz - eg. 44100.
-    virtual status_t getSampleRate(uint32_t *rate);
-
-    // Return size of input/output buffer in bytes for this stream - eg. 4800.
-    virtual status_t getBufferSize(size_t *size);
-
-    // Return the channel mask.
-    virtual status_t getChannelMask(audio_channel_mask_t *mask);
-
-    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
-    virtual status_t getFormat(audio_format_t *format);
-
-    // Convenience method.
-    virtual status_t getAudioProperties(
-            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
-
-    // Set audio stream parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get audio stream parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Add or remove the effect on the stream.
-    virtual status_t addEffect(sp<EffectHalInterface> effect);
-    virtual status_t removeEffect(sp<EffectHalInterface> effect);
-
-    // Put the audio hardware input/output into standby mode.
-    virtual status_t standby();
-
-    virtual status_t dump(int fd);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start();
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop();
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position);
-
-    // Set the priority of the thread that interacts with the HAL
-    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
-    virtual status_t setHalThreadPriority(int priority);
-
-  protected:
-    // Subclasses can not be constructed directly by clients.
-    explicit StreamHalHidl(IStream *stream);
-
-    // The destructor automatically closes the stream.
-    virtual ~StreamHalHidl();
-
-    status_t getCachedBufferSize(size_t *size);
-
-    bool requestHalThreadPriority(pid_t threadPid, pid_t threadId);
-
-    // mStreamPowerLog is used for audio signal power logging.
-    StreamPowerLog mStreamPowerLog;
-
-  private:
-    const int HAL_THREAD_PRIORITY_DEFAULT = -1;
-    IStream *mStream;
-    int mHalThreadPriority;
-    size_t mCachedBufferSize;
-};
-
-class StreamOutHalHidl : public StreamOutHalInterface, public StreamHalHidl {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Return the audio hardware driver estimated latency in milliseconds.
-    virtual status_t getLatency(uint32_t *latency);
-
-    // Use this method in situations where audio mixing is done in the hardware.
-    virtual status_t setVolume(float left, float right);
-
-    // Write audio buffer to driver.
-    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
-
-    // Return the number of audio frames written by the audio dsp to DAC since
-    // the output has exited standby.
-    virtual status_t getRenderPosition(uint32_t *dspFrames);
-
-    // Get the local time at which the next write to the audio driver will be presented.
-    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
-
-    // Set the callback for notifying completion of non-blocking write and drain.
-    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
-
-    // Returns whether pause and resume operations are supported.
-    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t pause();
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t resume();
-
-    // Returns whether drain operation is supported.
-    virtual status_t supportsDrain(bool *supportsDrain);
-
-    // Requests notification when data buffered by the driver/hardware has been played.
-    virtual status_t drain(bool earlyNotify);
-
-    // Notifies to the audio driver to flush the queued data.
-    virtual status_t flush();
-
-    // Return a recent count of the number of audio frames presented to an external observer.
-    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
-
-    // Called when the metadata of the stream's source has been changed.
-    status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
-
-    // Methods used by StreamOutCallback (HIDL).
-    void onWriteReady();
-    void onDrainReady();
-    void onError();
-
-  private:
-    friend class DeviceHalHidl;
-    typedef MessageQueue<WriteCommand, hardware::kSynchronizedReadWrite> CommandMQ;
-    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
-    typedef MessageQueue<WriteStatus, hardware::kSynchronizedReadWrite> StatusMQ;
-
-    wp<StreamOutHalInterfaceCallback> mCallback;
-    sp<IStreamOut> mStream;
-    std::unique_ptr<CommandMQ> mCommandMQ;
-    std::unique_ptr<DataMQ> mDataMQ;
-    std::unique_ptr<StatusMQ> mStatusMQ;
-    std::atomic<pid_t> mWriterClient;
-    EventFlag* mEfGroup;
-
-    // Can not be constructed directly by clients.
-    StreamOutHalHidl(const sp<IStreamOut>& stream);
-
-    virtual ~StreamOutHalHidl();
-
-    using WriterCallback = std::function<void(const WriteStatus& writeStatus)>;
-    status_t callWriterThread(
-            WriteCommand cmd, const char* cmdName,
-            const uint8_t* data, size_t dataSize, WriterCallback callback);
-    status_t prepareForWriting(size_t bufferSize);
-};
-
-class StreamInHalHidl : public StreamInHalInterface, public StreamHalHidl {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Set the input gain for the audio driver.
-    virtual status_t setGain(float gain);
-
-    // Read audio buffer in from driver.
-    virtual status_t read(void *buffer, size_t bytes, size_t *read);
-
-    // Return the amount of input frames lost in the audio driver.
-    virtual status_t getInputFramesLost(uint32_t *framesLost);
-
-    // Return a recent count of the number of audio frames received and
-    // the clock time associated with that frame count.
-    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
-
-    // Get active microphones
-    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
-
-    // Called when the metadata of the stream's sink has been changed.
-    status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
-
-  private:
-    friend class DeviceHalHidl;
-    typedef MessageQueue<ReadParameters, hardware::kSynchronizedReadWrite> CommandMQ;
-    typedef MessageQueue<uint8_t, hardware::kSynchronizedReadWrite> DataMQ;
-    typedef MessageQueue<ReadStatus, hardware::kSynchronizedReadWrite> StatusMQ;
-
-    sp<IStreamIn> mStream;
-    std::unique_ptr<CommandMQ> mCommandMQ;
-    std::unique_ptr<DataMQ> mDataMQ;
-    std::unique_ptr<StatusMQ> mStatusMQ;
-    std::atomic<pid_t> mReaderClient;
-    EventFlag* mEfGroup;
-
-    // Can not be constructed directly by clients.
-    StreamInHalHidl(const sp<IStreamIn>& stream);
-
-    virtual ~StreamInHalHidl();
-
-    using ReaderCallback = std::function<void(const ReadStatus& readStatus)>;
-    status_t callReaderThread(
-            const ReadParameters& params, const char* cmdName, ReaderCallback callback);
-    status_t prepareForReading(size_t bufferSize);
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_STREAM_HAL_HIDL_4_0_H
diff --git a/media/libaudiohal/4.0/StreamHalLocal.h b/media/libaudiohal/4.0/StreamHalLocal.h
deleted file mode 100644
index 7237509..0000000
--- a/media/libaudiohal/4.0/StreamHalLocal.h
+++ /dev/null
@@ -1,221 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_STREAM_HAL_LOCAL_4_0_H
-#define ANDROID_HARDWARE_STREAM_HAL_LOCAL_4_0_H
-
-#include <media/audiohal/StreamHalInterface.h>
-#include "StreamPowerLog.h"
-
-namespace android {
-namespace V4_0 {
-
-class DeviceHalLocal;
-
-class StreamHalLocal : public virtual StreamHalInterface
-{
-  public:
-    // Return the sampling rate in Hz - eg. 44100.
-    virtual status_t getSampleRate(uint32_t *rate);
-
-    // Return size of input/output buffer in bytes for this stream - eg. 4800.
-    virtual status_t getBufferSize(size_t *size);
-
-    // Return the channel mask.
-    virtual status_t getChannelMask(audio_channel_mask_t *mask);
-
-    // Return the audio format - e.g. AUDIO_FORMAT_PCM_16_BIT.
-    virtual status_t getFormat(audio_format_t *format);
-
-    // Convenience method.
-    virtual status_t getAudioProperties(
-            uint32_t *sampleRate, audio_channel_mask_t *mask, audio_format_t *format);
-
-    // Set audio stream parameters.
-    virtual status_t setParameters(const String8& kvPairs);
-
-    // Get audio stream parameters.
-    virtual status_t getParameters(const String8& keys, String8 *values);
-
-    // Add or remove the effect on the stream.
-    virtual status_t addEffect(sp<EffectHalInterface> effect);
-    virtual status_t removeEffect(sp<EffectHalInterface> effect);
-
-    // Put the audio hardware input/output into standby mode.
-    virtual status_t standby();
-
-    virtual status_t dump(int fd);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start() = 0;
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop() = 0;
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info) = 0;
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position) = 0;
-
-    // Set the priority of the thread that interacts with the HAL
-    // (must match the priority of the audioflinger's thread that calls 'read' / 'write')
-    virtual status_t setHalThreadPriority(int priority);
-
-  protected:
-    // Subclasses can not be constructed directly by clients.
-    StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device);
-
-    // The destructor automatically closes the stream.
-    virtual ~StreamHalLocal();
-
-    sp<DeviceHalLocal> mDevice;
-
-    // mStreamPowerLog is used for audio signal power logging.
-    StreamPowerLog mStreamPowerLog;
-
-  private:
-    audio_stream_t *mStream;
-};
-
-class StreamOutHalLocal : public StreamOutHalInterface, public StreamHalLocal {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Return the audio hardware driver estimated latency in milliseconds.
-    virtual status_t getLatency(uint32_t *latency);
-
-    // Use this method in situations where audio mixing is done in the hardware.
-    virtual status_t setVolume(float left, float right);
-
-    // Write audio buffer to driver.
-    virtual status_t write(const void *buffer, size_t bytes, size_t *written);
-
-    // Return the number of audio frames written by the audio dsp to DAC since
-    // the output has exited standby.
-    virtual status_t getRenderPosition(uint32_t *dspFrames);
-
-    // Get the local time at which the next write to the audio driver will be presented.
-    virtual status_t getNextWriteTimestamp(int64_t *timestamp);
-
-    // Set the callback for notifying completion of non-blocking write and drain.
-    virtual status_t setCallback(wp<StreamOutHalInterfaceCallback> callback);
-
-    // Returns whether pause and resume operations are supported.
-    virtual status_t supportsPauseAndResume(bool *supportsPause, bool *supportsResume);
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t pause();
-
-    // Notifies to the audio driver to resume playback following a pause.
-    virtual status_t resume();
-
-    // Returns whether drain operation is supported.
-    virtual status_t supportsDrain(bool *supportsDrain);
-
-    // Requests notification when data buffered by the driver/hardware has been played.
-    virtual status_t drain(bool earlyNotify);
-
-    // Notifies to the audio driver to flush the queued data.
-    virtual status_t flush();
-
-    // Return a recent count of the number of audio frames presented to an external observer.
-    virtual status_t getPresentationPosition(uint64_t *frames, struct timespec *timestamp);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start();
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop();
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position);
-
-    // Called when the metadata of the stream's source has been changed.
-    status_t updateSourceMetadata(const SourceMetadata& sourceMetadata) override;
-
-  private:
-    audio_stream_out_t *mStream;
-    wp<StreamOutHalInterfaceCallback> mCallback;
-
-    friend class DeviceHalLocal;
-
-    // Can not be constructed directly by clients.
-    StreamOutHalLocal(audio_stream_out_t *stream, sp<DeviceHalLocal> device);
-
-    virtual ~StreamOutHalLocal();
-
-    static int asyncCallback(stream_callback_event_t event, void *param, void *cookie);
-};
-
-class StreamInHalLocal : public StreamInHalInterface, public StreamHalLocal {
-  public:
-    // Return the frame size (number of bytes per sample) of a stream.
-    virtual status_t getFrameSize(size_t *size);
-
-    // Set the input gain for the audio driver.
-    virtual status_t setGain(float gain);
-
-    // Read audio buffer in from driver.
-    virtual status_t read(void *buffer, size_t bytes, size_t *read);
-
-    // Return the amount of input frames lost in the audio driver.
-    virtual status_t getInputFramesLost(uint32_t *framesLost);
-
-    // Return a recent count of the number of audio frames received and
-    // the clock time associated with that frame count.
-    virtual status_t getCapturePosition(int64_t *frames, int64_t *time);
-
-    // Start a stream operating in mmap mode.
-    virtual status_t start();
-
-    // Stop a stream operating in mmap mode.
-    virtual status_t stop();
-
-    // Retrieve information on the data buffer in mmap mode.
-    virtual status_t createMmapBuffer(int32_t minSizeFrames,
-                                      struct audio_mmap_buffer_info *info);
-
-    // Get current read/write position in the mmap buffer
-    virtual status_t getMmapPosition(struct audio_mmap_position *position);
-
-    // Get active microphones
-    virtual status_t getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones);
-
-    // Called when the metadata of the stream's sink has been changed.
-    status_t updateSinkMetadata(const SinkMetadata& sinkMetadata) override;
-
-  private:
-    audio_stream_in_t *mStream;
-
-    friend class DeviceHalLocal;
-
-    // Can not be constructed directly by clients.
-    StreamInHalLocal(audio_stream_in_t *stream, sp<DeviceHalLocal> device);
-
-    virtual ~StreamInHalLocal();
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_STREAM_HAL_LOCAL_4_0_H
diff --git a/media/libaudiohal/4.0/StreamPowerLog.h b/media/libaudiohal/4.0/StreamPowerLog.h
deleted file mode 100644
index 57b7201..0000000
--- a/media/libaudiohal/4.0/StreamPowerLog.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_STREAM_POWER_LOG_4_0_H
-#define ANDROID_HARDWARE_STREAM_POWER_LOG_4_0_H
-
-#include <audio_utils/clock.h>
-#include <audio_utils/PowerLog.h>
-#include <cutils/properties.h>
-#include <system/audio.h>
-
-namespace android {
-namespace V4_0 {
-
-class StreamPowerLog {
-public:
-    StreamPowerLog() :
-        mIsUserDebugOrEngBuild(is_userdebug_or_eng_build()),
-        mPowerLog(nullptr),
-        mFrameSize(0) {
-        // use init() to set up the power log.
-    }
-
-    ~StreamPowerLog() {
-        power_log_destroy(mPowerLog); // OK for null mPowerLog
-        mPowerLog = nullptr;
-    }
-
-    // A one-time initialization (do not call twice) before using StreamPowerLog.
-    void init(uint32_t sampleRate, audio_channel_mask_t channelMask, audio_format_t format) {
-        if (mPowerLog == nullptr) {
-            // Note: A way to get channel count for both input and output channel masks
-            // but does not check validity of the channel mask.
-            const uint32_t channelCount = popcount(audio_channel_mask_get_bits(channelMask));
-            mFrameSize = channelCount * audio_bytes_per_sample(format);
-            if (mFrameSize > 0) {
-                const size_t kPowerLogFramesPerEntry =
-                        (long long)sampleRate * kPowerLogSamplingIntervalMs / 1000;
-                mPowerLog = power_log_create(
-                        sampleRate,
-                        channelCount,
-                        format,
-                        kPowerLogEntries,
-                        kPowerLogFramesPerEntry);
-            }
-        }
-        // mPowerLog may be NULL (not the right build, format not accepted, etc.).
-    }
-
-    // Dump the power log to fd.
-    void dump(int fd) const {
-        // OK for null mPowerLog
-        (void)power_log_dump(
-                mPowerLog, fd, "      " /* prefix */, kPowerLogLines, 0 /* limit_ns */);
-    }
-
-    // Log the audio data contained in buffer.
-    void log(const void *buffer, size_t sizeInBytes) const {
-        if (mPowerLog != nullptr) { // mFrameSize is always nonzero if mPowerLog exists.
-            power_log_log(
-                    mPowerLog, buffer, sizeInBytes / mFrameSize, audio_utils_get_real_time_ns());
-        }
-    }
-
-    bool isUserDebugOrEngBuild() const {
-        return mIsUserDebugOrEngBuild;
-    }
-
-private:
-
-    static inline bool is_userdebug_or_eng_build() {
-        char value[PROPERTY_VALUE_MAX];
-        (void)property_get("ro.build.type", value, "unknown"); // ignore actual length
-        return strcmp(value, "userdebug") == 0 || strcmp(value, "eng") == 0;
-    }
-
-    // Audio signal power log configuration.
-    static const size_t kPowerLogLines = 40;
-    static const size_t kPowerLogSamplingIntervalMs = 50;
-    static const size_t kPowerLogEntries = (1 /* minutes */ * 60 /* seconds */ * 1000 /* msec */
-            / kPowerLogSamplingIntervalMs);
-
-    const bool mIsUserDebugOrEngBuild;
-    power_log_t *mPowerLog;
-    size_t mFrameSize;
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_STREAM_POWER_LOG_4_0_H
diff --git a/media/libaudiohal/4.0/include/libaudiohal/4.0/DevicesFactoryHalHybrid.h b/media/libaudiohal/4.0/include/libaudiohal/4.0/DevicesFactoryHalHybrid.h
deleted file mode 100644
index abf6de0..0000000
--- a/media/libaudiohal/4.0/include/libaudiohal/4.0/DevicesFactoryHalHybrid.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (C) 2017 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_4_0_H
-#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_4_0_H
-
-#include <media/audiohal/DevicesFactoryHalInterface.h>
-#include <utils/Errors.h>
-#include <utils/RefBase.h>
-
-namespace android {
-namespace V4_0 {
-
-class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
-{
-  public:
-    // Opens a device with the specified name. To close the device, it is
-    // necessary to release references to the returned object.
-    virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
-
-  private:
-    friend class DevicesFactoryHalInterface;
-
-    // Can not be constructed directly by clients.
-    DevicesFactoryHalHybrid();
-
-    virtual ~DevicesFactoryHalHybrid();
-
-    sp<DevicesFactoryHalInterface> mLocalFactory;
-    sp<DevicesFactoryHalInterface> mHidlFactory;
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_4_0_H
diff --git a/media/libaudiohal/4.0/include/libaudiohal/4.0/EffectsFactoryHalHidl.h b/media/libaudiohal/4.0/include/libaudiohal/4.0/EffectsFactoryHalHidl.h
deleted file mode 100644
index 680b7a1..0000000
--- a/media/libaudiohal/4.0/include/libaudiohal/4.0/EffectsFactoryHalHidl.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_4_0_H
-#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_4_0_H
-
-#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/4.0/types.h>
-#include <media/audiohal/EffectsFactoryHalInterface.h>
-
-#include "ConversionHelperHidl.h"
-
-namespace android {
-namespace V4_0 {
-
-using ::android::hardware::audio::effect::V4_0::EffectDescriptor;
-using ::android::hardware::audio::effect::V4_0::IEffectsFactory;
-using ::android::hardware::hidl_vec;
-
-class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
-{
-  public:
-    // Returns the number of different effects in all loaded libraries.
-    virtual status_t queryNumberEffects(uint32_t *pNumEffects);
-
-    // Returns a descriptor of the next available effect.
-    virtual status_t getDescriptor(uint32_t index,
-            effect_descriptor_t *pDescriptor);
-
-    virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
-            effect_descriptor_t *pDescriptor);
-
-    // Creates an effect engine of the specified type.
-    // To release the effect engine, it is necessary to release references
-    // to the returned effect object.
-    virtual status_t createEffect(const effect_uuid_t *pEffectUuid,
-            int32_t sessionId, int32_t ioId,
-            sp<EffectHalInterface> *effect);
-
-    virtual status_t dumpEffects(int fd);
-
-    status_t allocateBuffer(size_t size, sp<EffectBufferHalInterface>* buffer) override;
-    status_t mirrorBuffer(void* external, size_t size,
-                          sp<EffectBufferHalInterface>* buffer) override;
-
-  private:
-    friend class EffectsFactoryHalInterface;
-
-    sp<IEffectsFactory> mEffectsFactory;
-    hidl_vec<EffectDescriptor> mLastDescriptors;
-
-    // Can not be constructed directly by clients.
-    EffectsFactoryHalHidl();
-    virtual ~EffectsFactoryHalHidl();
-
-    status_t queryAllDescriptors();
-};
-
-} // namespace V4_0
-} // namespace android
-
-#endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_4_0_H
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 3a5df27..0ff0d4a 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -8,6 +8,7 @@
 
     cflags: [
         "-Wall",
+        "-Wextra",
         "-Werror",
     ],
 
diff --git a/media/libaudiohal/DevicesFactoryHalInterface.cpp b/media/libaudiohal/DevicesFactoryHalInterface.cpp
index 4c8eaf6..e631ace 100644
--- a/media/libaudiohal/DevicesFactoryHalInterface.cpp
+++ b/media/libaudiohal/DevicesFactoryHalInterface.cpp
@@ -17,18 +17,17 @@
 #include <android/hardware/audio/2.0/IDevicesFactory.h>
 #include <android/hardware/audio/4.0/IDevicesFactory.h>
 
-#include <DevicesFactoryHalHybrid.h>
-#include <libaudiohal/4.0/DevicesFactoryHalHybrid.h>
+#include <libaudiohal/FactoryHalHidl.h>
 
 namespace android {
 
 // static
 sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
     if (hardware::audio::V4_0::IDevicesFactory::getService() != nullptr) {
-        return new V4_0::DevicesFactoryHalHybrid();
+        return V4_0::createDevicesFactoryHal();
     }
     if (hardware::audio::V2_0::IDevicesFactory::getService() != nullptr) {
-        return new DevicesFactoryHalHybrid();
+        return V2_0::createDevicesFactoryHal();
     }
     return nullptr;
 }
diff --git a/media/libaudiohal/EffectsFactoryHalInterface.cpp b/media/libaudiohal/EffectsFactoryHalInterface.cpp
index ead1fa2..f7734a8 100644
--- a/media/libaudiohal/EffectsFactoryHalInterface.cpp
+++ b/media/libaudiohal/EffectsFactoryHalInterface.cpp
@@ -17,19 +17,17 @@
 #include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
 #include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
 
-#include <EffectsFactoryHalHidl.h>
-#include <libaudiohal/4.0/EffectsFactoryHalHidl.h>
-
+#include <libaudiohal/FactoryHalHidl.h>
 
 namespace android {
 
 // static
 sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
     if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
-        return new V4_0::EffectsFactoryHalHidl();
+        return V4_0::createEffectsFactoryHal();
     }
     if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
-        return new EffectsFactoryHalHidl();
+        return V2_0::createEffectsFactoryHal();
     }
     return nullptr;
 }
diff --git a/media/libaudiohal/4.0/Android.bp b/media/libaudiohal/impl/Android.bp
similarity index 68%
rename from media/libaudiohal/4.0/Android.bp
rename to media/libaudiohal/impl/Android.bp
index 833defa..3827336 100644
--- a/media/libaudiohal/4.0/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -1,5 +1,5 @@
-cc_library_shared {
-    name: "libaudiohal@4.0",
+cc_defaults {
+    name: "libaudiohal_default",
 
     srcs: [
         "DeviceHalLocal.cpp",
@@ -24,28 +24,30 @@
         "-Werror",
     ],
     shared_libs: [
+        "android.hardware.audio.common-util",
+        "android.hardware.audio.common@2.0",
+        "android.hardware.audio.common@4.0",
+        "android.hardware.audio.effect@2.0",
+        "android.hardware.audio.effect@4.0",
+        "android.hardware.audio@2.0",
+        "android.hardware.audio@4.0",
+        "android.hidl.allocator@1.0",
+        "android.hidl.memory@1.0",
         "libaudiohal_deathhandler",
         "libaudioutils",
+        "libbase",
         "libbinder",
         "libcutils",
-        "liblog",
-        "libutils",
-        "libhardware",
-        "libbase",
         "libfmq",
-        "libhwbinder",
+        "libhardware",
         "libhidlbase",
         "libhidlmemory",
         "libhidltransport",
-        "android.hardware.audio@4.0",
-        "android.hardware.audio.common-util",
-        "android.hardware.audio.common@4.0",
-        "android.hardware.audio.common@4.0-util",
-        "android.hardware.audio.effect@4.0",
-        "android.hidl.allocator@1.0",
-        "android.hidl.memory@1.0",
+        "libhwbinder",
+        "liblog",
         "libmedia_helper",
         "libmediautils",
+        "libutils",
     ],
     header_libs: [
         "android.hardware.audio.common.util@all-versions",
@@ -56,3 +58,29 @@
         "libfmq",
     ],
 }
+
+cc_library_shared {
+    name: "libaudiohal@2.0",
+    defaults: ["libaudiohal_default"],
+    shared_libs: [
+        "android.hardware.audio.common@2.0-util",
+    ],
+    cflags: [
+        "-DMAJOR_VERSION=2",
+        "-DMINOR_VERSION=0",
+        "-include VersionMacro.h",
+    ]
+}
+
+cc_library_shared {
+    name: "libaudiohal@4.0",
+    defaults: ["libaudiohal_default"],
+    shared_libs: [
+        "android.hardware.audio.common@4.0-util",
+    ],
+    cflags: [
+        "-DMAJOR_VERSION=4",
+        "-DMINOR_VERSION=0",
+        "-include VersionMacro.h",
+    ]
+}
diff --git a/media/libaudiohal/4.0/ConversionHelperHidl.cpp b/media/libaudiohal/impl/ConversionHelperHidl.cpp
similarity index 90%
rename from media/libaudiohal/4.0/ConversionHelperHidl.cpp
rename to media/libaudiohal/impl/ConversionHelperHidl.cpp
index fe27504..5d12fad 100644
--- a/media/libaudiohal/4.0/ConversionHelperHidl.cpp
+++ b/media/libaudiohal/impl/ConversionHelperHidl.cpp
@@ -22,15 +22,18 @@
 
 #include "ConversionHelperHidl.h"
 
-using ::android::hardware::audio::V4_0::AudioMicrophoneChannelMapping;
-using ::android::hardware::audio::V4_0::AudioMicrophoneDirectionality;
-using ::android::hardware::audio::V4_0::AudioMicrophoneLocation;
-using ::android::hardware::audio::V4_0::DeviceAddress;
-using ::android::hardware::audio::V4_0::MicrophoneInfo;
-using ::android::hardware::audio::V4_0::Result;
+using ::android::hardware::audio::CPP_VERSION::Result;
+
+#if MAJOR_VERSION == 4
+using ::android::hardware::audio::CPP_VERSION::AudioMicrophoneChannelMapping;
+using ::android::hardware::audio::CPP_VERSION::AudioMicrophoneDirectionality;
+using ::android::hardware::audio::CPP_VERSION::AudioMicrophoneLocation;
+using ::android::hardware::audio::CPP_VERSION::DeviceAddress;
+using ::android::hardware::audio::CPP_VERSION::MicrophoneInfo;
+#endif
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 // static
 status_t ConversionHelperHidl::keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys) {
@@ -106,8 +109,9 @@
     ALOGE("%s %p %s: %s (from rpc)", mClassName, this, funcName, description);
 }
 
+#if MAJOR_VERSION == 4
 // TODO: Use the same implementation in the hal when it moves to a util library.
-std::string deviceAddressToHal(const DeviceAddress& address) {
+static std::string deviceAddressToHal(const DeviceAddress& address) {
     // HAL assumes that the address is NUL-terminated.
     char halAddress[AUDIO_DEVICE_MAX_ADDRESS_LEN];
     memset(halAddress, 0, sizeof(halAddress));
@@ -141,7 +145,7 @@
 
 //local conversion helpers
 
-audio_microphone_channel_mapping_t  channelMappingToHal(AudioMicrophoneChannelMapping mapping) {
+static audio_microphone_channel_mapping_t  channelMappingToHal(AudioMicrophoneChannelMapping mapping) {
     switch (mapping) {
         case AudioMicrophoneChannelMapping::UNUSED:
             return AUDIO_MICROPHONE_CHANNEL_MAPPING_UNUSED;
@@ -154,7 +158,7 @@
     }
 }
 
-audio_microphone_location_t locationToHal(AudioMicrophoneLocation location) {
+static audio_microphone_location_t locationToHal(AudioMicrophoneLocation location) {
     switch (location) {
         case AudioMicrophoneLocation::UNKNOWN:
             return AUDIO_MICROPHONE_LOCATION_UNKNOWN;
@@ -168,7 +172,7 @@
             LOG_ALWAYS_FATAL("Unknown locationToHal conversion %d", location);
     }
 }
-audio_microphone_directionality_t directionalityToHal(AudioMicrophoneDirectionality dir) {
+static audio_microphone_directionality_t directionalityToHal(AudioMicrophoneDirectionality dir) {
     switch (dir) {
         case AudioMicrophoneDirectionality::UNKNOWN:
             return AUDIO_MICROPHONE_DIRECTIONALITY_UNKNOWN;
@@ -187,9 +191,8 @@
     }
 }
 
-// static
-void ConversionHelperHidl::microphoneInfoToHal(const MicrophoneInfo& src,
-                                                     audio_microphone_characteristic_t *pDst) {
+void microphoneInfoToHal(const MicrophoneInfo& src,
+                         audio_microphone_characteristic_t *pDst) {
     if (pDst != NULL) {
         snprintf(pDst->device_id, sizeof(pDst->device_id),
                  "%s", src.deviceId.c_str());
@@ -232,6 +235,7 @@
         pDst->orientation.z = src.orientation.z;
     }
 }
+#endif
 
-}  // namespace V4_0
+}  // namespace CPP_VERSION
 }  // namespace android
diff --git a/media/libaudiohal/2.0/ConversionHelperHidl.h b/media/libaudiohal/impl/ConversionHelperHidl.h
similarity index 78%
rename from media/libaudiohal/2.0/ConversionHelperHidl.h
rename to media/libaudiohal/impl/ConversionHelperHidl.h
index c356f37..1a9319f 100644
--- a/media/libaudiohal/2.0/ConversionHelperHidl.h
+++ b/media/libaudiohal/impl/ConversionHelperHidl.h
@@ -18,15 +18,20 @@
 #define ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_H
 
 #include <android/hardware/audio/2.0/types.h>
+#include <android/hardware/audio/4.0/types.h>
 #include <hidl/HidlSupport.h>
+#include <system/audio.h>
 #include <utils/String8.h>
 
-using ::android::hardware::audio::V2_0::ParameterValue;
+using ::android::hardware::audio::CPP_VERSION::ParameterValue;
+using CoreResult = ::android::hardware::audio::CPP_VERSION::Result;
+
 using ::android::hardware::Return;
 using ::android::hardware::hidl_string;
 using ::android::hardware::hidl_vec;
 
 namespace android {
+namespace CPP_VERSION {
 
 class ConversionHelperHidl {
   protected:
@@ -54,7 +59,7 @@
         return ret.isOk() ? OK : FAILED_TRANSACTION;
     }
 
-    status_t processReturn(const char* funcName, const Return<hardware::audio::V2_0::Result>& ret) {
+    status_t processReturn(const char* funcName, const Return<CoreResult>& ret) {
         if (!ret.isOk()) {
             emitError(funcName, ret.description().c_str());
         }
@@ -63,7 +68,7 @@
 
     template<typename T>
     status_t processReturn(
-            const char* funcName, const Return<T>& ret, hardware::audio::V2_0::Result retval) {
+            const char* funcName, const Return<T>& ret, CoreResult retval) {
         if (!ret.isOk()) {
             emitError(funcName, ret.description().c_str());
         }
@@ -73,11 +78,18 @@
   private:
     const char* mClassName;
 
-    static status_t analyzeResult(const hardware::audio::V2_0::Result& result);
+    static status_t analyzeResult(const CoreResult& result);
 
     void emitError(const char* funcName, const char* description);
 };
 
+#if MAJOR_VERSION == 4
+using ::android::hardware::audio::CPP_VERSION::MicrophoneInfo;
+void microphoneInfoToHal(const MicrophoneInfo& src,
+                         audio_microphone_characteristic_t *pDst);
+#endif
+
+}  // namespace CPP_VERSION
 }  // namespace android
 
 #endif // ANDROID_HARDWARE_CONVERSION_HELPER_HIDL_H
diff --git a/media/libaudiohal/4.0/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
similarity index 88%
rename from media/libaudiohal/4.0/DeviceHalHidl.cpp
rename to media/libaudiohal/impl/DeviceHalHidl.cpp
index 6facca9..723e2eb 100644
--- a/media/libaudiohal/4.0/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -19,6 +19,7 @@
 #define LOG_TAG "DeviceHalHidl"
 //#define LOG_NDEBUG 0
 
+#include <android/hardware/audio/2.0/IPrimaryDevice.h>
 #include <android/hardware/audio/4.0/IPrimaryDevice.h>
 #include <cutils/native_handle.h>
 #include <hwbinder/IPCThreadState.h>
@@ -31,27 +32,30 @@
 #include "StreamHalHidl.h"
 #include "VersionUtils.h"
 
-using ::android::hardware::audio::common::V4_0::AudioConfig;
-using ::android::hardware::audio::common::V4_0::AudioDevice;
-using ::android::hardware::audio::common::V4_0::AudioInputFlag;
-using ::android::hardware::audio::common::V4_0::AudioOutputFlag;
-using ::android::hardware::audio::common::V4_0::AudioPatchHandle;
-using ::android::hardware::audio::common::V4_0::AudioPort;
-using ::android::hardware::audio::common::V4_0::AudioPortConfig;
-using ::android::hardware::audio::common::V4_0::AudioMode;
-using ::android::hardware::audio::common::V4_0::AudioSource;
-using ::android::hardware::audio::common::V4_0::HidlUtils;
+using ::android::hardware::audio::common::CPP_VERSION::AudioConfig;
+using ::android::hardware::audio::common::CPP_VERSION::AudioDevice;
+using ::android::hardware::audio::common::CPP_VERSION::AudioInputFlag;
+using ::android::hardware::audio::common::CPP_VERSION::AudioOutputFlag;
+using ::android::hardware::audio::common::CPP_VERSION::AudioPatchHandle;
+using ::android::hardware::audio::common::CPP_VERSION::AudioPort;
+using ::android::hardware::audio::common::CPP_VERSION::AudioPortConfig;
+using ::android::hardware::audio::common::CPP_VERSION::AudioMode;
+using ::android::hardware::audio::common::CPP_VERSION::AudioSource;
+using ::android::hardware::audio::common::CPP_VERSION::HidlUtils;
 using ::android::hardware::audio::common::utils::mkEnumConverter;
-using ::android::hardware::audio::V4_0::DeviceAddress;
-using ::android::hardware::audio::V4_0::IPrimaryDevice;
-using ::android::hardware::audio::V4_0::ParameterValue;
-using ::android::hardware::audio::V4_0::Result;
-using ::android::hardware::audio::V4_0::SinkMetadata;
+using ::android::hardware::audio::CPP_VERSION::DeviceAddress;
+using ::android::hardware::audio::CPP_VERSION::IPrimaryDevice;
+using ::android::hardware::audio::CPP_VERSION::ParameterValue;
+using ::android::hardware::audio::CPP_VERSION::Result;
 using ::android::hardware::hidl_string;
 using ::android::hardware::hidl_vec;
 
+#if MAJOR_VERSION == 4
+using ::android::hardware::audio::CPP_VERSION::SinkMetadata;
+#endif
+
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 namespace {
 
@@ -259,7 +263,9 @@
             hidlDevice,
             hidlConfig,
             mkEnumConverter<AudioOutputFlag>(flags),
+#if MAJOR_VERSION == 4
             {} /* metadata */,
+#endif
             [&](Result r, const sp<IStreamOut>& result, const AudioConfig& suggestedConfig) {
                 retval = r;
                 if (retval == Result::OK) {
@@ -285,15 +291,19 @@
     AudioConfig hidlConfig;
     HidlUtils::audioConfigFromHal(*config, &hidlConfig);
     Result retval = Result::NOT_INITIALIZED;
+#if MAJOR_VERSION == 2
+    auto sourceMetadata = AudioSource(source);
+#elif MAJOR_VERSION == 4
     // TODO: correctly propagate the tracks sources and volume
     //       for now, only send the main source at 1dbfs
-    SinkMetadata metadata = {{{AudioSource(source), 1}}};
+    SinkMetadata sourceMetadata = {{{AudioSource(source), 1}}};
+#endif
     Return<void> ret = mDevice->openInputStream(
             handle,
             hidlDevice,
             hidlConfig,
-            flags,
-            metadata,
+            mkEnumConverter<AudioInputFlag>(flags),
+            sourceMetadata,
             [&](Result r, const sp<IStreamIn>& result, const AudioConfig& suggestedConfig) {
                 retval = r;
                 if (retval == Result::OK) {
@@ -359,6 +369,13 @@
     return processReturn("setAudioPortConfig", mDevice->setAudioPortConfig(hidlConfig));
 }
 
+#if MAJOR_VERSION == 2
+status_t DeviceHalHidl::getMicrophones(
+        std::vector<media::MicrophoneInfo> *microphonesInfo __unused) {
+    if (mDevice == 0) return NO_INIT;
+    return INVALID_OPERATION;
+}
+#elif MAJOR_VERSION == 4
 status_t DeviceHalHidl::getMicrophones(std::vector<media::MicrophoneInfo> *microphonesInfo) {
     if (mDevice == 0) return NO_INIT;
     Result retval;
@@ -375,6 +392,7 @@
     });
     return processReturn("getMicrophones", ret, retval);
 }
+#endif
 
 status_t DeviceHalHidl::dump(int fd) {
     if (mDevice == 0) return NO_INIT;
@@ -385,5 +403,5 @@
     return processReturn("dump", ret);
 }
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
similarity index 94%
rename from media/libaudiohal/2.0/DeviceHalHidl.h
rename to media/libaudiohal/impl/DeviceHalHidl.h
index 3c1cb594..fb5e7e7 100644
--- a/media/libaudiohal/2.0/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -18,16 +18,19 @@
 #define ANDROID_HARDWARE_DEVICE_HAL_HIDL_H
 
 #include <android/hardware/audio/2.0/IDevice.h>
+#include <android/hardware/audio/4.0/IDevice.h>
 #include <android/hardware/audio/2.0/IPrimaryDevice.h>
+#include <android/hardware/audio/4.0/IPrimaryDevice.h>
 #include <media/audiohal/DeviceHalInterface.h>
 
 #include "ConversionHelperHidl.h"
 
-using ::android::hardware::audio::V2_0::IDevice;
-using ::android::hardware::audio::V2_0::IPrimaryDevice;
+using ::android::hardware::audio::CPP_VERSION::IDevice;
+using ::android::hardware::audio::CPP_VERSION::IPrimaryDevice;
 using ::android::hardware::Return;
 
 namespace android {
+namespace CPP_VERSION {
 
 class DeviceHalHidl : public DeviceHalInterface, public ConversionHelperHidl
 {
@@ -124,6 +127,7 @@
     virtual ~DeviceHalHidl();
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_DEVICE_HAL_HIDL_H
diff --git a/media/libaudiohal/4.0/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
similarity index 96%
rename from media/libaudiohal/4.0/DeviceHalLocal.cpp
rename to media/libaudiohal/impl/DeviceHalLocal.cpp
index a245dd9..14e26f5 100644
--- a/media/libaudiohal/4.0/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -23,7 +23,7 @@
 #include "StreamHalLocal.h"
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 DeviceHalLocal::DeviceHalLocal(audio_hw_device_t *dev)
         : mDev(dev) {
@@ -185,6 +185,12 @@
         return INVALID_OPERATION;
 }
 
+#if MAJOR_VERSION == 2
+status_t DeviceHalLocal::getMicrophones(
+        std::vector<media::MicrophoneInfo> *microphones __unused) {
+    return INVALID_OPERATION;
+}
+#elif MAJOR_VERSION == 4
 status_t DeviceHalLocal::getMicrophones(std::vector<media::MicrophoneInfo> *microphones) {
     if (mDev->get_microphones == NULL) return INVALID_OPERATION;
     size_t actual_mics = AUDIO_MICROPHONE_MAX_COUNT;
@@ -196,6 +202,7 @@
     }
     return status;
 }
+#endif
 
 status_t DeviceHalLocal::dump(int fd) {
     return mDev->dump(mDev, fd);
@@ -209,5 +216,5 @@
     mDev->close_input_stream(mDev, stream_in);
 }
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
similarity index 98%
rename from media/libaudiohal/2.0/DeviceHalLocal.h
rename to media/libaudiohal/impl/DeviceHalLocal.h
index aec201a..18bd879 100644
--- a/media/libaudiohal/2.0/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -21,6 +21,7 @@
 #include <media/audiohal/DeviceHalInterface.h>
 
 namespace android {
+namespace CPP_VERSION {
 
 class DeviceHalLocal : public DeviceHalInterface
 {
@@ -122,6 +123,7 @@
     uint32_t version() const { return mDev->common.version; }
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_DEVICE_HAL_LOCAL_H
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
new file mode 100644
index 0000000..28001da
--- /dev/null
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
@@ -0,0 +1,116 @@
+/*
+ * Copyright (C) 2016 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <string.h>
+#include <vector>
+
+#define LOG_TAG "DevicesFactoryHalHidl"
+//#define LOG_NDEBUG 0
+
+#include <android/hardware/audio/2.0/IDevice.h>
+#include <android/hardware/audio/4.0/IDevice.h>
+#include <media/audiohal/hidl/HalDeathHandler.h>
+#include <utils/Log.h>
+
+#include "ConversionHelperHidl.h"
+#include "DeviceHalHidl.h"
+#include "DevicesFactoryHalHidl.h"
+
+using ::android::hardware::audio::CPP_VERSION::IDevice;
+using ::android::hardware::audio::CPP_VERSION::Result;
+using ::android::hardware::Return;
+
+namespace android {
+namespace CPP_VERSION {
+
+DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
+    sp<IDevicesFactory> defaultFactory{IDevicesFactory::getService()};
+    if (!defaultFactory) {
+        ALOGE("Failed to obtain IDevicesFactory/default service, terminating process.");
+        exit(1);
+    }
+    mDeviceFactories.push_back(defaultFactory);
+    if (MAJOR_VERSION >= 4) {
+        // The MSD factory is optional and only available starting at HAL 4.0
+        sp<IDevicesFactory> msdFactory{IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD)};
+        if (msdFactory) {
+            mDeviceFactories.push_back(msdFactory);
+        }
+    }
+    for (const auto& factory : mDeviceFactories) {
+        // It is assumed that the DevicesFactoryHalInterface instance is owned
+        // by AudioFlinger and thus have the same lifespan.
+        factory->linkToDeath(HalDeathHandler::getInstance(), 0 /*cookie*/);
+    }
+}
+
+
+#if MAJOR_VERSION == 2
+static IDevicesFactory::Device idFromHal(const char *name, status_t* status) {
+    *status = OK;
+    if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_PRIMARY) == 0) {
+        return IDevicesFactory::Device::PRIMARY;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_A2DP) == 0) {
+        return IDevicesFactory::Device::A2DP;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_USB) == 0) {
+        return IDevicesFactory::Device::USB;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_REMOTE_SUBMIX) == 0) {
+        return IDevicesFactory::Device::R_SUBMIX;
+    } else if(strcmp(name, AUDIO_HARDWARE_MODULE_ID_STUB) == 0) {
+        return IDevicesFactory::Device::STUB;
+    }
+    ALOGE("Invalid device name %s", name);
+    *status = BAD_VALUE;
+    return {};
+}
+#elif MAJOR_VERSION == 4
+static const char* idFromHal(const char *name, status_t* status) {
+    *status = OK;
+    return name;
+}
+#endif
+
+status_t DevicesFactoryHalHidl::openDevice(const char *name, sp<DeviceHalInterface> *device) {
+    if (mDeviceFactories.empty()) return NO_INIT;
+    status_t status;
+    auto hidlId = idFromHal(name, &status);
+    if (status != OK) return status;
+    Result retval = Result::NOT_INITIALIZED;
+    for (const auto& factory : mDeviceFactories) {
+        Return<void> ret = factory->openDevice(
+                hidlId,
+                [&](Result r, const sp<IDevice>& result) {
+                    retval = r;
+                    if (retval == Result::OK) {
+                        *device = new DeviceHalHidl(result);
+                    }
+                });
+        if (!ret.isOk()) return FAILED_TRANSACTION;
+        switch (retval) {
+            // Device was found and was initialized successfully.
+            case Result::OK: return OK;
+            // Device was found but failed to initalize.
+            case Result::NOT_INITIALIZED: return NO_INIT;
+            // Otherwise continue iterating.
+            default: ;
+        }
+    }
+    ALOGW("The specified device name is not recognized: \"%s\"", name);
+    return BAD_VALUE;
+}
+
+} // namespace CPP_VERSION
+} // namespace android
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHidl.h b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
similarity index 84%
rename from media/libaudiohal/2.0/DevicesFactoryHalHidl.h
rename to media/libaudiohal/impl/DevicesFactoryHalHidl.h
index 0748849..a4282b0 100644
--- a/media/libaudiohal/2.0/DevicesFactoryHalHidl.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
@@ -18,15 +18,17 @@
 #define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H
 
 #include <android/hardware/audio/2.0/IDevicesFactory.h>
+#include <android/hardware/audio/4.0/IDevicesFactory.h>
 #include <media/audiohal/DevicesFactoryHalInterface.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
 
 #include "DeviceHalHidl.h"
 
-using ::android::hardware::audio::V2_0::IDevicesFactory;
+using ::android::hardware::audio::CPP_VERSION::IDevicesFactory;
 
 namespace android {
+namespace CPP_VERSION {
 
 class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
 {
@@ -38,17 +40,15 @@
   private:
     friend class DevicesFactoryHalHybrid;
 
-    sp<IDevicesFactory> mDevicesFactory;
-    sp<IDevicesFactory> mDevicesFactoryMsd;
-
-    static status_t nameFromHal(const char *name, IDevicesFactory::Device *device);
+    std::vector<sp<IDevicesFactory>> mDeviceFactories;
 
     // Can not be constructed directly by clients.
     DevicesFactoryHalHidl();
 
-    virtual ~DevicesFactoryHalHidl();
+    virtual ~DevicesFactoryHalHidl() = default;
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HIDL_H
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
similarity index 95%
rename from media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp
rename to media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
index 1c4be74..f337a8b 100644
--- a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
@@ -22,15 +22,13 @@
 #include "DevicesFactoryHalHidl.h"
 
 namespace android {
+namespace CPP_VERSION {
 
 DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
         : mLocalFactory(new DevicesFactoryHalLocal()),
           mHidlFactory(new DevicesFactoryHalHidl()) {
 }
 
-DevicesFactoryHalHybrid::~DevicesFactoryHalHybrid() {
-}
-
 status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
     if (mHidlFactory != 0 && strcmp(AUDIO_HARDWARE_MODULE_ID_A2DP, name) != 0 &&
         strcmp(AUDIO_HARDWARE_MODULE_ID_HEARING_AID, name) != 0) {
@@ -39,4 +37,5 @@
     return mLocalFactory->openDevice(name, device);
 }
 
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.h b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
similarity index 89%
rename from media/libaudiohal/2.0/DevicesFactoryHalHybrid.h
rename to media/libaudiohal/impl/DevicesFactoryHalHybrid.h
index abd57d6..5ac0d0d 100644
--- a/media/libaudiohal/2.0/DevicesFactoryHalHybrid.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
@@ -22,26 +22,27 @@
 #include <utils/RefBase.h>
 
 namespace android {
+namespace CPP_VERSION {
 
 class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
 {
   public:
+    DevicesFactoryHalHybrid();
+
     // Opens a device with the specified name. To close the device, it is
     // necessary to release references to the returned object.
     virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
 
   private:
-    friend class DevicesFactoryHalInterface;
-
-    // Can not be constructed directly by clients.
-    DevicesFactoryHalHybrid();
-
-    virtual ~DevicesFactoryHalHybrid();
-
     sp<DevicesFactoryHalInterface> mLocalFactory;
     sp<DevicesFactoryHalInterface> mHidlFactory;
 };
 
+sp<DevicesFactoryHalInterface> createDevicesFactoryHal() {
+    return new DevicesFactoryHalHybrid();
+}
+
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
diff --git a/media/libaudiohal/4.0/DevicesFactoryHalLocal.cpp b/media/libaudiohal/impl/DevicesFactoryHalLocal.cpp
similarity index 97%
rename from media/libaudiohal/4.0/DevicesFactoryHalLocal.cpp
rename to media/libaudiohal/impl/DevicesFactoryHalLocal.cpp
index e54edd4..af67ff5 100644
--- a/media/libaudiohal/4.0/DevicesFactoryHalLocal.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalLocal.cpp
@@ -26,7 +26,7 @@
 #include "DevicesFactoryHalLocal.h"
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 static status_t load_audio_interface(const char *if_name, audio_hw_device_t **dev)
 {
@@ -67,5 +67,5 @@
     return rc;
 }
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/DevicesFactoryHalLocal.h b/media/libaudiohal/impl/DevicesFactoryHalLocal.h
similarity index 96%
rename from media/libaudiohal/2.0/DevicesFactoryHalLocal.h
rename to media/libaudiohal/impl/DevicesFactoryHalLocal.h
index b9d18ab..5d108dd 100644
--- a/media/libaudiohal/2.0/DevicesFactoryHalLocal.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalLocal.h
@@ -24,6 +24,7 @@
 #include "DeviceHalLocal.h"
 
 namespace android {
+namespace CPP_VERSION {
 
 class DevicesFactoryHalLocal : public DevicesFactoryHalInterface
 {
@@ -41,6 +42,7 @@
     virtual ~DevicesFactoryHalLocal() {}
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_DEVICES_FACTORY_HAL_LOCAL_H
diff --git a/media/libaudiohal/4.0/EffectBufferHalHidl.cpp b/media/libaudiohal/impl/EffectBufferHalHidl.cpp
similarity index 98%
rename from media/libaudiohal/4.0/EffectBufferHalHidl.cpp
rename to media/libaudiohal/impl/EffectBufferHalHidl.cpp
index 957c89f..6ef4e8a 100644
--- a/media/libaudiohal/4.0/EffectBufferHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectBufferHalHidl.cpp
@@ -30,7 +30,7 @@
 using ::android::hidl::allocator::V1_0::IAllocator;
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 // static
 uint64_t EffectBufferHalHidl::makeUniqueId() {
@@ -142,5 +142,5 @@
     memcpy(mExternalData, mAudioBuffer.raw, size);
 }
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/EffectBufferHalHidl.h b/media/libaudiohal/impl/EffectBufferHalHidl.h
similarity index 92%
rename from media/libaudiohal/2.0/EffectBufferHalHidl.h
rename to media/libaudiohal/impl/EffectBufferHalHidl.h
index 31e0087..029d71a 100644
--- a/media/libaudiohal/2.0/EffectBufferHalHidl.h
+++ b/media/libaudiohal/impl/EffectBufferHalHidl.h
@@ -18,16 +18,18 @@
 #define ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
 
 #include <android/hardware/audio/effect/2.0/types.h>
+#include <android/hardware/audio/effect/4.0/types.h>
 #include <android/hidl/memory/1.0/IMemory.h>
 #include <hidl/HidlSupport.h>
 #include <media/audiohal/EffectBufferHalInterface.h>
 #include <system/audio_effect.h>
 
-using android::hardware::audio::effect::V2_0::AudioBuffer;
+using android::hardware::audio::effect::CPP_VERSION::AudioBuffer;
 using android::hardware::hidl_memory;
 using android::hidl::memory::V1_0::IMemory;
 
 namespace android {
+namespace CPP_VERSION {
 
 class EffectBufferHalHidl : public EffectBufferHalInterface
 {
@@ -71,6 +73,7 @@
     status_t init();
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_EFFECT_BUFFER_HAL_HIDL_H
diff --git a/media/libaudiohal/4.0/EffectHalHidl.cpp b/media/libaudiohal/impl/EffectHalHidl.cpp
similarity index 95%
rename from media/libaudiohal/4.0/EffectHalHidl.cpp
rename to media/libaudiohal/impl/EffectHalHidl.cpp
index c99c4c8..12649a1 100644
--- a/media/libaudiohal/4.0/EffectHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectHalHidl.cpp
@@ -22,26 +22,25 @@
 #include <media/EffectsFactoryApi.h>
 #include <utils/Log.h>
 
-#include "ConversionHelperHidl.h"
 #include "EffectBufferHalHidl.h"
 #include "EffectHalHidl.h"
 #include "HidlUtils.h"
 
-using ::android::hardware::audio::effect::V4_0::AudioBuffer;
-using ::android::hardware::audio::effect::V4_0::EffectBufferAccess;
-using ::android::hardware::audio::effect::V4_0::EffectConfigParameters;
-using ::android::hardware::audio::effect::V4_0::MessageQueueFlagBits;
-using ::android::hardware::audio::effect::V4_0::Result;
-using ::android::hardware::audio::common::V4_0::HidlUtils;
-using ::android::hardware::audio::common::V4_0::AudioChannelMask;
-using ::android::hardware::audio::common::V4_0::AudioFormat;
+using ::android::hardware::audio::effect::CPP_VERSION::AudioBuffer;
+using ::android::hardware::audio::effect::CPP_VERSION::EffectBufferAccess;
+using ::android::hardware::audio::effect::CPP_VERSION::EffectConfigParameters;
+using ::android::hardware::audio::effect::CPP_VERSION::MessageQueueFlagBits;
+using ::android::hardware::audio::effect::CPP_VERSION::Result;
+using ::android::hardware::audio::common::CPP_VERSION::HidlUtils;
+using ::android::hardware::audio::common::CPP_VERSION::AudioChannelMask;
+using ::android::hardware::audio::common::CPP_VERSION::AudioFormat;
 using ::android::hardware::audio::common::utils::mkEnumConverter;
 using ::android::hardware::hidl_vec;
 using ::android::hardware::MQDescriptorSync;
 using ::android::hardware::Return;
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 EffectHalHidl::EffectHalHidl(const sp<IEffect>& effect, uint64_t effectId)
         : mEffect(effect), mEffectId(effectId), mBuffersChanged(true), mEfGroup(nullptr) {
@@ -338,5 +337,5 @@
     return result;
 }
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/EffectHalHidl.h b/media/libaudiohal/impl/EffectHalHidl.h
similarity index 85%
rename from media/libaudiohal/2.0/EffectHalHidl.h
rename to media/libaudiohal/impl/EffectHalHidl.h
index 6ffdaf1..04f40d3 100644
--- a/media/libaudiohal/2.0/EffectHalHidl.h
+++ b/media/libaudiohal/impl/EffectHalHidl.h
@@ -18,19 +18,22 @@
 #define ANDROID_HARDWARE_EFFECT_HAL_HIDL_H
 
 #include <android/hardware/audio/effect/2.0/IEffect.h>
+#include <android/hardware/audio/effect/4.0/IEffect.h>
 #include <media/audiohal/EffectHalInterface.h>
 #include <fmq/EventFlag.h>
 #include <fmq/MessageQueue.h>
 #include <system/audio_effect.h>
 
-using ::android::hardware::audio::effect::V2_0::EffectBufferConfig;
-using ::android::hardware::audio::effect::V2_0::EffectConfig;
-using ::android::hardware::audio::effect::V2_0::EffectDescriptor;
-using ::android::hardware::audio::effect::V2_0::IEffect;
+using ::android::hardware::audio::effect::CPP_VERSION::EffectBufferConfig;
+using ::android::hardware::audio::effect::CPP_VERSION::EffectConfig;
+using ::android::hardware::audio::effect::CPP_VERSION::EffectDescriptor;
+using ::android::hardware::audio::effect::CPP_VERSION::IEffect;
+using EffectResult = ::android::hardware::audio::effect::CPP_VERSION::Result;
 using ::android::hardware::EventFlag;
 using ::android::hardware::MessageQueue;
 
 namespace android {
+namespace CPP_VERSION {
 
 class EffectHalHidl : public EffectHalInterface
 {
@@ -68,8 +71,7 @@
 
   private:
     friend class EffectsFactoryHalHidl;
-    typedef MessageQueue<
-        hardware::audio::effect::V2_0::Result, hardware::kSynchronizedReadWrite> StatusMQ;
+    typedef MessageQueue<EffectResult, hardware::kSynchronizedReadWrite> StatusMQ;
 
     sp<IEffect> mEffect;
     const uint64_t mEffectId;
@@ -79,7 +81,7 @@
     std::unique_ptr<StatusMQ> mStatusMQ;
     EventFlag* mEfGroup;
 
-    static status_t analyzeResult(const hardware::audio::effect::V2_0::Result& result);
+    static status_t analyzeResult(const EffectResult& result);
     static void effectBufferConfigFromHal(
             const buffer_config_t& halConfig, EffectBufferConfig* config);
     static void effectBufferConfigToHal(
@@ -103,6 +105,7 @@
     status_t setProcessBuffers();
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_EFFECT_HAL_HIDL_H
diff --git a/media/libaudiohal/4.0/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
similarity index 93%
rename from media/libaudiohal/4.0/EffectsFactoryHalHidl.cpp
rename to media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index dfed784..b880433 100644
--- a/media/libaudiohal/4.0/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -18,21 +18,21 @@
 //#define LOG_NDEBUG 0
 
 #include <cutils/native_handle.h>
-#include <libaudiohal/4.0/EffectsFactoryHalHidl.h>
 
+#include "EffectsFactoryHalHidl.h"
 #include "ConversionHelperHidl.h"
 #include "EffectBufferHalHidl.h"
 #include "EffectHalHidl.h"
 #include "HidlUtils.h"
 
-using ::android::hardware::audio::common::V4_0::HidlUtils;
-using ::android::hardware::audio::common::V4_0::Uuid;
-using ::android::hardware::audio::effect::V4_0::IEffect;
-using ::android::hardware::audio::effect::V4_0::Result;
+using ::android::hardware::audio::common::CPP_VERSION::HidlUtils;
+using ::android::hardware::audio::common::CPP_VERSION::Uuid;
+using ::android::hardware::audio::effect::CPP_VERSION::IEffect;
+using ::android::hardware::audio::effect::CPP_VERSION::Result;
 using ::android::hardware::Return;
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
     mEffectsFactory = IEffectsFactory::getService();
@@ -42,9 +42,6 @@
     }
 }
 
-EffectsFactoryHalHidl::~EffectsFactoryHalHidl() {
-}
-
 status_t EffectsFactoryHalHidl::queryAllDescriptors() {
     if (mEffectsFactory == 0) return NO_INIT;
     Result retval = Result::NOT_INITIALIZED;
@@ -148,5 +145,5 @@
 }
 
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
similarity index 84%
rename from media/libaudiohal/2.0/EffectsFactoryHalHidl.h
rename to media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 82b5481..c6fced7 100644
--- a/media/libaudiohal/2.0/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -18,20 +18,25 @@
 #define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
 
 #include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
+#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
 #include <android/hardware/audio/effect/2.0/types.h>
+#include <android/hardware/audio/effect/4.0/types.h>
 #include <media/audiohal/EffectsFactoryHalInterface.h>
 
 #include "ConversionHelperHidl.h"
 
 namespace android {
+namespace CPP_VERSION {
 
-using ::android::hardware::audio::effect::V2_0::EffectDescriptor;
-using ::android::hardware::audio::effect::V2_0::IEffectsFactory;
+using ::android::hardware::audio::effect::CPP_VERSION::EffectDescriptor;
+using ::android::hardware::audio::effect::CPP_VERSION::IEffectsFactory;
 using ::android::hardware::hidl_vec;
 
 class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
 {
   public:
+    EffectsFactoryHalHidl();
+
     // Returns the number of different effects in all loaded libraries.
     virtual status_t queryNumberEffects(uint32_t *pNumEffects);
 
@@ -56,18 +61,17 @@
                           sp<EffectBufferHalInterface>* buffer) override;
 
   private:
-    friend class EffectsFactoryHalInterface;
-
     sp<IEffectsFactory> mEffectsFactory;
     hidl_vec<EffectDescriptor> mLastDescriptors;
 
-    // Can not be constructed directly by clients.
-    EffectsFactoryHalHidl();
-    virtual ~EffectsFactoryHalHidl();
-
     status_t queryAllDescriptors();
 };
 
+sp<EffectsFactoryHalInterface> createEffectsFactoryHal() {
+    return new EffectsFactoryHalHidl();
+}
+
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
diff --git a/media/libaudiohal/4.0/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
similarity index 93%
rename from media/libaudiohal/4.0/StreamHalHidl.cpp
rename to media/libaudiohal/impl/StreamHalHidl.cpp
index 1c2fdb0..b23e018 100644
--- a/media/libaudiohal/4.0/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -17,6 +17,7 @@
 #define LOG_TAG "StreamHalHidl"
 //#define LOG_NDEBUG 0
 
+#include <android/hardware/audio/2.0/IStreamOutCallback.h>
 #include <android/hardware/audio/4.0/IStreamOutCallback.h>
 #include <hwbinder/IPCThreadState.h>
 #include <mediautils/SchedulingPolicyService.h>
@@ -27,30 +28,33 @@
 #include "StreamHalHidl.h"
 #include "VersionUtils.h"
 
-using ::android::hardware::audio::common::V4_0::AudioChannelMask;
-using ::android::hardware::audio::common::V4_0::AudioContentType;
-using ::android::hardware::audio::common::V4_0::AudioFormat;
-using ::android::hardware::audio::common::V4_0::AudioSource;
-using ::android::hardware::audio::common::V4_0::AudioUsage;
-using ::android::hardware::audio::common::V4_0::ThreadInfo;
-using ::android::hardware::audio::V4_0::AudioDrain;
-using ::android::hardware::audio::V4_0::IStreamOutCallback;
-using ::android::hardware::audio::V4_0::MessageQueueFlagBits;
-using ::android::hardware::audio::V4_0::MicrophoneInfo;
-using ::android::hardware::audio::V4_0::MmapBufferInfo;
-using ::android::hardware::audio::V4_0::MmapPosition;
-using ::android::hardware::audio::V4_0::ParameterValue;
-using ::android::hardware::audio::V4_0::PlaybackTrackMetadata;
-using ::android::hardware::audio::V4_0::RecordTrackMetadata;
-using ::android::hardware::audio::V4_0::Result;
-using ::android::hardware::audio::V4_0::TimeSpec;
+using ::android::hardware::audio::common::CPP_VERSION::AudioChannelMask;
+using ::android::hardware::audio::common::CPP_VERSION::AudioFormat;
+using ::android::hardware::audio::common::CPP_VERSION::ThreadInfo;
+using ::android::hardware::audio::CPP_VERSION::AudioDrain;
+using ::android::hardware::audio::CPP_VERSION::IStreamOutCallback;
+using ::android::hardware::audio::CPP_VERSION::MessageQueueFlagBits;
+using ::android::hardware::audio::CPP_VERSION::MmapBufferInfo;
+using ::android::hardware::audio::CPP_VERSION::MmapPosition;
+using ::android::hardware::audio::CPP_VERSION::ParameterValue;
+using ::android::hardware::audio::CPP_VERSION::Result;
+using ::android::hardware::audio::CPP_VERSION::TimeSpec;
 using ::android::hardware::MQDescriptorSync;
 using ::android::hardware::Return;
 using ::android::hardware::Void;
-using ReadCommand = ::android::hardware::audio::V4_0::IStreamIn::ReadCommand;
+using ReadCommand = ::android::hardware::audio::CPP_VERSION::IStreamIn::ReadCommand;
+
+#if MAJOR_VERSION == 4
+using ::android::hardware::audio::common::CPP_VERSION::AudioContentType;
+using ::android::hardware::audio::common::CPP_VERSION::AudioSource;
+using ::android::hardware::audio::common::CPP_VERSION::AudioUsage;
+using ::android::hardware::audio::CPP_VERSION::MicrophoneInfo;
+using ::android::hardware::audio::CPP_VERSION::PlaybackTrackMetadata;
+using ::android::hardware::audio::CPP_VERSION::RecordTrackMetadata;
+#endif
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 StreamHalHidl::StreamHalHidl(IStream *stream)
         : ConversionHelperHidl("Stream"),
@@ -566,6 +570,12 @@
     }
 }
 
+#if MAJOR_VERSION == 2
+status_t StreamOutHalHidl::updateSourceMetadata(const SourceMetadata& /* sourceMetadata */) {
+    // Audio HAL V2.0 does not support propagating source metadata
+    return INVALID_OPERATION;
+}
+#elif MAJOR_VERSION == 4
 /** Transform a standard collection to an HIDL vector. */
 template <class Values, class ElementConverter>
 static auto transformToHidlVec(const Values& values, ElementConverter converter) {
@@ -576,7 +586,7 @@
 }
 
 status_t StreamOutHalHidl::updateSourceMetadata(const SourceMetadata& sourceMetadata) {
-    hardware::audio::V4_0::SourceMetadata halMetadata = {
+    hardware::audio::CPP_VERSION::SourceMetadata halMetadata = {
         .tracks = transformToHidlVec(sourceMetadata.tracks,
               [](const playback_track_metadata& metadata) -> PlaybackTrackMetadata {
                   return {
@@ -587,6 +597,7 @@
               })};
     return processReturn("updateSourceMetadata", mStream->updateSourceMetadata(halMetadata));
 }
+#endif
 
 void StreamOutHalHidl::onWriteReady() {
     sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
@@ -782,7 +793,19 @@
     }
 }
 
+#if MAJOR_VERSION == 2
+status_t StreamInHalHidl::getActiveMicrophones(
+        std::vector<media::MicrophoneInfo> *microphones __unused) {
+    if (mStream == 0) return NO_INIT;
+    return INVALID_OPERATION;
+}
 
+status_t StreamInHalHidl::updateSinkMetadata(const SinkMetadata& /* sinkMetadata */) {
+    // Audio HAL V2.0 does not support propagating sink metadata
+    return INVALID_OPERATION;
+}
+
+#elif MAJOR_VERSION == 4
 status_t StreamInHalHidl::getActiveMicrophones(
         std::vector<media::MicrophoneInfo> *microphonesInfo) {
     if (!mStream) return NO_INIT;
@@ -802,7 +825,7 @@
 }
 
 status_t StreamInHalHidl::updateSinkMetadata(const SinkMetadata& sinkMetadata) {
-    hardware::audio::V4_0::SinkMetadata halMetadata = {
+    hardware::audio::CPP_VERSION::SinkMetadata halMetadata = {
         .tracks = transformToHidlVec(sinkMetadata.tracks,
               [](const record_track_metadata& metadata) -> RecordTrackMetadata {
                   return {
@@ -812,6 +835,7 @@
               })};
     return processReturn("updateSinkMetadata", mStream->updateSinkMetadata(halMetadata));
 }
+#endif
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
similarity index 92%
rename from media/libaudiohal/2.0/StreamHalHidl.h
rename to media/libaudiohal/impl/StreamHalHidl.h
index ebad8ae..95ec7f1 100644
--- a/media/libaudiohal/2.0/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -20,8 +20,11 @@
 #include <atomic>
 
 #include <android/hardware/audio/2.0/IStream.h>
+#include <android/hardware/audio/4.0/IStream.h>
 #include <android/hardware/audio/2.0/IStreamIn.h>
+#include <android/hardware/audio/4.0/IStreamIn.h>
 #include <android/hardware/audio/2.0/IStreamOut.h>
+#include <android/hardware/audio/4.0/IStreamOut.h>
 #include <fmq/EventFlag.h>
 #include <fmq/MessageQueue.h>
 #include <media/audiohal/StreamHalInterface.h>
@@ -29,18 +32,19 @@
 #include "ConversionHelperHidl.h"
 #include "StreamPowerLog.h"
 
-using ::android::hardware::audio::V2_0::IStream;
-using ::android::hardware::audio::V2_0::IStreamIn;
-using ::android::hardware::audio::V2_0::IStreamOut;
+using ::android::hardware::audio::CPP_VERSION::IStream;
+using ::android::hardware::audio::CPP_VERSION::IStreamIn;
+using ::android::hardware::audio::CPP_VERSION::IStreamOut;
 using ::android::hardware::EventFlag;
 using ::android::hardware::MessageQueue;
 using ::android::hardware::Return;
-using ReadParameters = ::android::hardware::audio::V2_0::IStreamIn::ReadParameters;
-using ReadStatus = ::android::hardware::audio::V2_0::IStreamIn::ReadStatus;
-using WriteCommand = ::android::hardware::audio::V2_0::IStreamOut::WriteCommand;
-using WriteStatus = ::android::hardware::audio::V2_0::IStreamOut::WriteStatus;
+using ReadParameters = ::android::hardware::audio::CPP_VERSION::IStreamIn::ReadParameters;
+using ReadStatus = ::android::hardware::audio::CPP_VERSION::IStreamIn::ReadStatus;
+using WriteCommand = ::android::hardware::audio::CPP_VERSION::IStreamOut::WriteCommand;
+using WriteStatus = ::android::hardware::audio::CPP_VERSION::IStreamOut::WriteStatus;
 
 namespace android {
+namespace CPP_VERSION {
 
 class DeviceHalHidl;
 
@@ -243,6 +247,7 @@
     status_t prepareForReading(size_t bufferSize);
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_STREAM_HAL_HIDL_H
diff --git a/media/libaudiohal/4.0/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
similarity index 97%
rename from media/libaudiohal/4.0/StreamHalLocal.cpp
rename to media/libaudiohal/impl/StreamHalLocal.cpp
index e9d96bf..b134f57 100644
--- a/media/libaudiohal/4.0/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -25,7 +25,7 @@
 #include "VersionUtils.h"
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 
 StreamHalLocal::StreamHalLocal(audio_stream_t *stream, sp<DeviceHalLocal> device)
         : mDevice(device),
@@ -341,6 +341,12 @@
     return mStream->get_mmap_position(mStream, position);
 }
 
+#if MAJOR_VERSION == 2
+status_t StreamInHalLocal::getActiveMicrophones(
+        std::vector<media::MicrophoneInfo> *microphones __unused) {
+    return INVALID_OPERATION;
+}
+#elif MAJOR_VERSION == 4
 status_t StreamInHalLocal::getActiveMicrophones(std::vector<media::MicrophoneInfo> *microphones) {
     if (mStream->get_active_microphones == NULL) return INVALID_OPERATION;
     size_t actual_mics = AUDIO_MICROPHONE_MAX_COUNT;
@@ -352,6 +358,7 @@
     }
     return status;
 }
+#endif
 
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
diff --git a/media/libaudiohal/2.0/StreamHalLocal.h b/media/libaudiohal/impl/StreamHalLocal.h
similarity index 99%
rename from media/libaudiohal/2.0/StreamHalLocal.h
rename to media/libaudiohal/impl/StreamHalLocal.h
index cda8d0c..cea4229 100644
--- a/media/libaudiohal/2.0/StreamHalLocal.h
+++ b/media/libaudiohal/impl/StreamHalLocal.h
@@ -21,6 +21,7 @@
 #include "StreamPowerLog.h"
 
 namespace android {
+namespace CPP_VERSION {
 
 class DeviceHalLocal;
 
@@ -214,6 +215,7 @@
     virtual ~StreamInHalLocal();
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_STREAM_HAL_LOCAL_H
diff --git a/media/libaudiohal/2.0/StreamPowerLog.h b/media/libaudiohal/impl/StreamPowerLog.h
similarity index 98%
rename from media/libaudiohal/2.0/StreamPowerLog.h
rename to media/libaudiohal/impl/StreamPowerLog.h
index a78b1aa..5fd3912 100644
--- a/media/libaudiohal/2.0/StreamPowerLog.h
+++ b/media/libaudiohal/impl/StreamPowerLog.h
@@ -23,6 +23,7 @@
 #include <system/audio.h>
 
 namespace android {
+namespace CPP_VERSION {
 
 class StreamPowerLog {
 public:
@@ -97,6 +98,7 @@
     size_t mFrameSize;
 };
 
+} // namespace CPP_VERSION
 } // namespace android
 
 #endif // ANDROID_HARDWARE_STREAM_POWER_LOG_H
diff --git a/media/libaudiohal/impl/VersionMacro.h b/media/libaudiohal/impl/VersionMacro.h
new file mode 100644
index 0000000..98e9c07
--- /dev/null
+++ b/media/libaudiohal/impl/VersionMacro.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_VERSION_MACRO_H
+#define ANDROID_HARDWARE_VERSION_MACRO_H
+
+#if !defined(MAJOR_VERSION) || !defined(MINOR_VERSION)
+#error "MAJOR_VERSION and MINOR_VERSION must be defined"
+#endif
+
+#define CONCAT_3(a,b,c) a##b##c
+#define EXPAND_CONCAT_3(a,b,c) CONCAT_3(a,b,c)
+/** The directory name of the version: <major>.<minor> */
+#define FILE_VERSION EXPAND_CONCAT_3(MAJOR_VERSION,.,MINOR_VERSION)
+
+#define CONCAT_4(a,b,c,d) a##b##c##d
+#define EXPAND_CONCAT_4(a,b,c,d) CONCAT_4(a,b,c,d)
+/** The c++ namespace of the version: V<major>_<minor> */
+#define CPP_VERSION EXPAND_CONCAT_4(V,MAJOR_VERSION,_,MINOR_VERSION)
+
+#endif // ANDROID_HARDWARE_VERSION_MACRO_H
diff --git a/media/libaudiohal/4.0/VersionUtils.h b/media/libaudiohal/impl/VersionUtils.h
similarity index 61%
rename from media/libaudiohal/4.0/VersionUtils.h
rename to media/libaudiohal/impl/VersionUtils.h
index 1246c2e..5004895 100644
--- a/media/libaudiohal/4.0/VersionUtils.h
+++ b/media/libaudiohal/impl/VersionUtils.h
@@ -14,22 +14,36 @@
  * limitations under the License.
  */
 
-#ifndef ANDROID_HARDWARE_VERSION_UTILS_4_0_H
-#define ANDROID_HARDWARE_VERSION_UTILS_4_0_H
+#ifndef ANDROID_HARDWARE_VERSION_UTILS_H
+#define ANDROID_HARDWARE_VERSION_UTILS_H
 
+#include <android/hardware/audio/2.0/types.h>
 #include <android/hardware/audio/4.0/types.h>
 #include <hidl/HidlSupport.h>
 
-using ::android::hardware::audio::V4_0::ParameterValue;
-using ::android::hardware::audio::V4_0::Result;
+using ::android::hardware::audio::CPP_VERSION::ParameterValue;
+using ::android::hardware::audio::CPP_VERSION::Result;
 using ::android::hardware::Return;
 using ::android::hardware::hidl_vec;
 using ::android::hardware::hidl_string;
 
 namespace android {
-namespace V4_0 {
+namespace CPP_VERSION {
 namespace utils {
 
+#if MAJOR_VERSION == 2
+template <class T, class Callback>
+Return<void> getParameters(T& object, hidl_vec<ParameterValue> /*context*/,
+                           hidl_vec<hidl_string> keys, Callback callback) {
+    return object->getParameters(keys, callback);
+}
+
+template <class T>
+Return<Result> setParameters(T& object, hidl_vec<ParameterValue> /*context*/,
+                             hidl_vec<ParameterValue> keys) {
+    return object->setParameters(keys);
+}
+#elif MAJOR_VERSION == 4
 template <class T, class Callback>
 Return<void> getParameters(T& object, hidl_vec<ParameterValue> context,
                            hidl_vec<hidl_string> keys, Callback callback) {
@@ -41,9 +55,10 @@
                              hidl_vec<ParameterValue> keys) {
     return object->setParameters(context, keys);
 }
+#endif
 
 } // namespace utils
-} // namespace V4_0
+} // namespace CPP_VERSION
 } // namespace android
 
-#endif // ANDROID_HARDWARE_VERSION_UTILS_4_0_H
+#endif // ANDROID_HARDWARE_VERSION_UTILS_H
diff --git a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
new file mode 100644
index 0000000..fa0effc
--- /dev/null
+++ b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
+#define ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
+
+/** @file Library entry points to create the HAL factories. */
+
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <utils/StrongPointer.h>
+
+namespace android {
+
+namespace V2_0 {
+sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
+sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
+} // namespace V2_0
+
+namespace V4_0 {
+sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
+sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
+} // namespace V4_0
+
+} // namespace android
+
+#endif // ANDROID_HARDWARE_FACTORY_HAL_HIDL_H
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 1b3a1be..a22819a 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -32,6 +32,9 @@
         "libaudioclient_headers",
         "libaudio_system_headers",
     ],
+    export_header_lib_headers: [
+        "libmedia_headers",
+    ],
     clang: true,
 }
 
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 034f7c2..1c95e27 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/AudioParameter.cpp
@@ -36,6 +36,8 @@
 const char * const AudioParameter::keyHwAvSync = AUDIO_PARAMETER_HW_AV_SYNC;
 const char * const AudioParameter::keyPresentationId = AUDIO_PARAMETER_STREAM_PRESENTATION_ID;
 const char * const AudioParameter::keyProgramId = AUDIO_PARAMETER_STREAM_PROGRAM_ID;
+const char * const AudioParameter::keyAudioLanguagePreferred =
+        AUDIO_PARAMETER_KEY_AUDIO_LANGUAGE_PREFERRED;
 const char * const AudioParameter::keyMonoOutput = AUDIO_PARAMETER_MONO_OUTPUT;
 const char * const AudioParameter::keyStreamHwAvSync = AUDIO_PARAMETER_STREAM_HW_AV_SYNC;
 const char * const AudioParameter::keyStreamConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
diff --git a/media/libmedia/TypeConverter.cpp b/media/libmedia/TypeConverter.cpp
index a3db754..514c795 100644
--- a/media/libmedia/TypeConverter.cpp
+++ b/media/libmedia/TypeConverter.cpp
@@ -132,6 +132,7 @@
     MAKE_STRING_FROM_ENUM(AUDIO_INPUT_FLAG_MMAP_NOIRQ),
     MAKE_STRING_FROM_ENUM(AUDIO_INPUT_FLAG_VOIP_TX),
     MAKE_STRING_FROM_ENUM(AUDIO_INPUT_FLAG_HW_AV_SYNC),
+    MAKE_STRING_FROM_ENUM(AUDIO_INPUT_FLAG_DIRECT),
     TERMINATOR
 };
 
diff --git a/media/libmedia/include/media/PatchBuilder.h b/media/libmedia/include/media/PatchBuilder.h
new file mode 100644
index 0000000..f2722a6
--- /dev/null
+++ b/media/libmedia/include/media/PatchBuilder.h
@@ -0,0 +1,103 @@
+/*
+ * Copyright (C) 2018 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef ANDROID_PATCH_BUILDER_H
+#define ANDROID_PATCH_BUILDER_H
+
+#include <functional>
+#include <utility>
+
+#include <system/audio.h>
+#include <utils/StrongPointer.h>
+
+// This is a header-only utility.
+
+namespace android {
+
+class PatchBuilder {
+  public:
+    using mix_usecase_t = decltype(audio_port_config_mix_ext::usecase);
+
+    PatchBuilder() = default;
+
+    // All existing methods operating on audio patches take a pointer to const.
+    // It's OK to construct a temporary PatchBuilder while preparing a parameter
+    // to such a function because the Builder will be kept alive until the code
+    // execution reaches the function call statement semicolon.
+    const struct audio_patch* patch() const { return &mPatch; }
+
+    template<typename T, typename... S>
+    PatchBuilder& addSink(T&& t, S&&... s) {
+        sinks().add(std::forward<T>(t), std::forward<S>(s)...);
+        return *this;
+    }
+    // Explicit type of the second parameter allows clients to provide the struct inline.
+    template<typename T>
+    PatchBuilder& addSink(T&& t, const mix_usecase_t& update) {
+        sinks().add(std::forward<T>(t), update);
+        return *this;
+    }
+    template<typename T, typename... S>
+    PatchBuilder& addSource(T&& t, S&&... s) {
+        sources().add(std::forward<T>(t), std::forward<S>(s)...);
+        return *this;
+    }
+    // Explicit type of the second parameter allows clients to provide the struct inline.
+    template<typename T>
+    PatchBuilder& addSource(T&& t, const mix_usecase_t& update) {
+        sources().add(std::forward<T>(t), update);
+        return *this;
+    }
+
+  private:
+    struct PortCfgs {
+        PortCfgs(unsigned int *countPtr, struct audio_port_config *portCfgs)
+                : mCountPtr(countPtr), mPortCfgs(portCfgs) {}
+        audio_port_config& add(const audio_port_config& portCfg) {
+            return *advance() = portCfg;
+        }
+        template<typename T>
+        audio_port_config& add(const sp<T>& entity) {
+            audio_port_config* added = advance();
+            entity->toAudioPortConfig(added);
+            return *added;
+        }
+        template<typename T>
+        void add(const sp<T>& entity, const mix_usecase_t& usecaseUpdate) {
+            add(entity).ext.mix.usecase = usecaseUpdate;
+        }
+        template<typename T>
+        void add(const sp<T>& entity,
+                std::function<mix_usecase_t(const mix_usecase_t&)> usecaseUpdater) {
+            mix_usecase_t* usecase = &add(entity).ext.mix.usecase;
+            *usecase = usecaseUpdater(*usecase);
+        }
+        struct audio_port_config* advance() {
+            return &mPortCfgs[(*mCountPtr)++];
+        }
+        unsigned int *mCountPtr;
+        struct audio_port_config *mPortCfgs;
+    };
+
+    PortCfgs sinks() { return PortCfgs(&mPatch.num_sinks, mPatch.sinks); }
+    PortCfgs sources() { return PortCfgs(&mPatch.num_sources, mPatch.sources); }
+
+    struct audio_patch mPatch = {};
+};
+
+}  // namespace android
+
+#endif  // ANDROID_PATCH_BUILDER_H
diff --git a/media/libmedia/include/media/SingleStateQueue.h b/media/libmedia/include/media/SingleStateQueue.h
index d423962..c2761cb 100644
--- a/media/libmedia/include/media/SingleStateQueue.h
+++ b/media/libmedia/include/media/SingleStateQueue.h
@@ -99,6 +99,13 @@
             return mShared->mAck - sequence >= 0;
         }
 
+        // returns the last value written (or the contents of the shared buffer after initialization
+        // if no value was written).
+        T last() const
+        {   // assume no sequence check required - we are the writer.
+            return mShared->mValue;
+        }
+
     private:
         int32_t     mSequence;
         Shared * const mShared;
diff --git a/media/libmediaextractor/MediaBuffer.cpp b/media/libmediaextractor/MediaBuffer.cpp
index 39f8d6e..d197b3f 100644
--- a/media/libmediaextractor/MediaBuffer.cpp
+++ b/media/libmediaextractor/MediaBuffer.cpp
@@ -39,7 +39,7 @@
       mRangeOffset(0),
       mRangeLength(size),
       mOwnsData(false),
-      mMetaData(new MetaData),
+      mMetaData(new MetaDataBase),
       mOriginal(NULL) {
 }
 
@@ -51,7 +51,7 @@
       mRangeOffset(0),
       mRangeLength(size),
       mOwnsData(true),
-      mMetaData(new MetaData),
+      mMetaData(new MetaDataBase),
       mOriginal(NULL) {
     if (size < kSharedMemThreshold
             || std::atomic_load_explicit(&mUseSharedMemory, std::memory_order_seq_cst) == 0) {
@@ -84,7 +84,7 @@
       mRangeLength(mSize),
       mBuffer(buffer),
       mOwnsData(false),
-      mMetaData(new MetaData),
+      mMetaData(new MetaDataBase),
       mOriginal(NULL) {
 }
 
@@ -96,7 +96,7 @@
         return;
     }
 
-    int prevCount = __sync_fetch_and_sub(&mRefCount, 1);
+    int prevCount = mRefCount.fetch_sub(1);
     if (prevCount == 1) {
         if (mObserver == NULL) {
             delete this;
@@ -110,13 +110,13 @@
 
 void MediaBuffer::claim() {
     CHECK(mObserver != NULL);
-    CHECK_EQ(mRefCount, 1);
+    CHECK_EQ(mRefCount.load(std::memory_order_relaxed), 1);
 
-    mRefCount = 0;
+    mRefCount.store(0, std::memory_order_relaxed);
 }
 
 void MediaBuffer::add_ref() {
-    (void) __sync_fetch_and_add(&mRefCount, 1);
+    (void) mRefCount.fetch_add(1);
 }
 
 void *MediaBuffer::data() const {
diff --git a/media/libmediaextractor/include/media/stagefright/MediaBuffer.h b/media/libmediaextractor/include/media/stagefright/MediaBuffer.h
index f944d51..5a25965 100644
--- a/media/libmediaextractor/include/media/stagefright/MediaBuffer.h
+++ b/media/libmediaextractor/include/media/stagefright/MediaBuffer.h
@@ -86,12 +86,14 @@
     virtual MediaBufferBase *clone();
 
     // sum of localRefcount() and remoteRefcount()
+    // Result should be treated as approximate unless the result precludes concurrent accesses.
     virtual int refcount() const {
         return localRefcount() + remoteRefcount();
     }
 
+    // Result should be treated as approximate unless the result precludes concurrent accesses.
     virtual int localRefcount() const {
-        return mRefCount;
+        return mRefCount.load(std::memory_order_relaxed);
     }
 
     virtual int remoteRefcount() const {
@@ -146,7 +148,7 @@
     void claim();
 
     MediaBufferObserver *mObserver;
-    int mRefCount;
+    std::atomic<int> mRefCount;
 
     void *mData;
     size_t mSize, mRangeOffset, mRangeLength;
diff --git a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
index 3905b55..2fb5a2c 100644
--- a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
+++ b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
@@ -150,11 +150,11 @@
 
 // Do not change these values without updating their counterparts in MediaPlayer2.java
 enum mediaplayer2_states {
-    MEDIAPLAYER2_STATE_IDLE         = 1,
-    MEDIAPLAYER2_STATE_PREPARED     = 2,
-    MEDIAPLAYER2_STATE_PLAYING      = 3,
-    MEDIAPLAYER2_STATE_PAUSED       = 4,
-    MEDIAPLAYER2_STATE_ERROR        = 5,
+    MEDIAPLAYER2_STATE_IDLE         = 1001,
+    MEDIAPLAYER2_STATE_PREPARED     = 1002,
+    MEDIAPLAYER2_STATE_PLAYING      = 1003,
+    MEDIAPLAYER2_STATE_PAUSED       = 1004,
+    MEDIAPLAYER2_STATE_ERROR        = 1005,
 };
 
 enum media_player2_internal_states {
diff --git a/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h b/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
index d586192..3af212e 100644
--- a/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
+++ b/media/libmediaplayer2/include/mediaplayer2/mediaplayer2.h
@@ -68,7 +68,7 @@
             status_t        stop();
             status_t        pause();
             bool            isPlaying();
-            mediaplayer2_states getMediaPlayer2State();
+            mediaplayer2_states getState();
             status_t        setPlaybackSettings(const AudioPlaybackRate& rate);
             status_t        getPlaybackSettings(AudioPlaybackRate* rate /* nonnull */);
             status_t        setSyncSettings(const AVSyncSettings& sync, float videoFpsHint);
diff --git a/media/libmediaplayer2/mediaplayer2.cpp b/media/libmediaplayer2/mediaplayer2.cpp
index e5567dc..4fb47b8 100644
--- a/media/libmediaplayer2/mediaplayer2.cpp
+++ b/media/libmediaplayer2/mediaplayer2.cpp
@@ -860,7 +860,7 @@
     return false;
 }
 
-mediaplayer2_states MediaPlayer2::getMediaPlayer2State() {
+mediaplayer2_states MediaPlayer2::getState() {
     Mutex::Autolock _l(mLock);
     if (mCurrentState & MEDIA_PLAYER2_STATE_ERROR) {
         return MEDIAPLAYER2_STATE_ERROR;
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index 6e94517..3370df1 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -501,9 +501,10 @@
         return ERROR_MALFORMED;
     }
 
-    int32_t width, height;
+    int32_t width, height, stride;
     CHECK(outputFormat->findInt32("width", &width));
     CHECK(outputFormat->findInt32("height", &height));
+    CHECK(outputFormat->findInt32("stride", &stride));
 
     int32_t crop_left, crop_top, crop_right, crop_bottom;
     if (!outputFormat->findRect("crop", &crop_left, &crop_top, &crop_right, &crop_bottom)) {
@@ -530,11 +531,10 @@
     if (converter.isValid()) {
         converter.convert(
                 (const uint8_t *)videoFrameBuffer->data(),
-                width, height,
+                width, height, stride,
                 crop_left, crop_top, crop_right, crop_bottom,
                 frame->getFlattenedData(),
-                frame->mWidth,
-                frame->mHeight,
+                frame->mWidth, frame->mHeight, frame->mRowBytes,
                 crop_left, crop_top, crop_right, crop_bottom);
         return OK;
     }
@@ -681,9 +681,10 @@
         return ERROR_MALFORMED;
     }
 
-    int32_t width, height;
+    int32_t width, height, stride;
     CHECK(outputFormat->findInt32("width", &width));
     CHECK(outputFormat->findInt32("height", &height));
+    CHECK(outputFormat->findInt32("stride", &stride));
 
     if (mFrame == NULL) {
         sp<IMemory> frameMem = allocVideoFrame(
@@ -727,11 +728,10 @@
     if (converter.isValid()) {
         converter.convert(
                 (const uint8_t *)videoFrameBuffer->data(),
-                width, height,
+                width, height, stride,
                 crop_left, crop_top, crop_right, crop_bottom,
                 mFrame->getFlattenedData(),
-                mFrame->mWidth,
-                mFrame->mHeight,
+                mFrame->mWidth, mFrame->mHeight, mFrame->mRowBytes,
                 dstLeft, dstTop, dstRight, dstBottom);
         return OK;
     }
diff --git a/media/libstagefright/MediaExtractorFactory.cpp b/media/libstagefright/MediaExtractorFactory.cpp
index f6c61a0..2d4bd39 100644
--- a/media/libstagefright/MediaExtractorFactory.cpp
+++ b/media/libstagefright/MediaExtractorFactory.cpp
@@ -123,7 +123,7 @@
 };
 
 Mutex MediaExtractorFactory::gPluginMutex;
-std::shared_ptr<List<sp<ExtractorPlugin>>> MediaExtractorFactory::gPlugins;
+std::shared_ptr<std::list<sp<ExtractorPlugin>>> MediaExtractorFactory::gPlugins;
 bool MediaExtractorFactory::gPluginsRegistered = false;
 
 // static
@@ -133,7 +133,7 @@
     *confidence = 0.0f;
     *meta = nullptr;
 
-    std::shared_ptr<List<sp<ExtractorPlugin>>> plugins;
+    std::shared_ptr<std::list<sp<ExtractorPlugin>>> plugins;
     {
         Mutex::Autolock autoLock(gPluginMutex);
         if (!gPluginsRegistered) {
@@ -145,6 +145,7 @@
     MediaExtractor::CreatorFunc curCreator = NULL;
     MediaExtractor::CreatorFunc bestCreator = NULL;
     for (auto it = plugins->begin(); it != plugins->end(); ++it) {
+        ALOGV("sniffing %s", (*it)->def.extractor_name);
         float newConfidence;
         void *newMeta = nullptr;
         MediaExtractor::FreeMetaFunc newFreeMeta = nullptr;
@@ -171,7 +172,7 @@
 
 // static
 void MediaExtractorFactory::RegisterExtractor(const sp<ExtractorPlugin> &plugin,
-        List<sp<ExtractorPlugin>> &pluginList) {
+        std::list<sp<ExtractorPlugin>> &pluginList) {
     // sanity check check struct version, uuid, name
     if (plugin->def.def_version == 0
             || plugin->def.def_version > MediaExtractor::EXTRACTORDEF_VERSION) {
@@ -213,7 +214,7 @@
 
 //static
 void MediaExtractorFactory::RegisterExtractorsInApk(
-        const char *apkPath, List<sp<ExtractorPlugin>> &pluginList) {
+        const char *apkPath, std::list<sp<ExtractorPlugin>> &pluginList) {
     ALOGV("search for plugins at %s", apkPath);
     ZipArchiveHandle zipHandle;
     int32_t ret = OpenArchive(apkPath, &zipHandle);
@@ -261,7 +262,7 @@
 
 //static
 void MediaExtractorFactory::RegisterExtractorsInSystem(
-        const char *libDirPath, List<sp<ExtractorPlugin>> &pluginList) {
+        const char *libDirPath, std::list<sp<ExtractorPlugin>> &pluginList) {
     ALOGV("search for plugins at %s", libDirPath);
     DIR *libDir = opendir(libDirPath);
     if (libDir) {
@@ -291,6 +292,10 @@
     }
 }
 
+static bool compareFunc(const sp<ExtractorPlugin>& first, const sp<ExtractorPlugin>& second) {
+    return strcmp(first->def.extractor_name, second->def.extractor_name) < 0;
+}
+
 // static
 void MediaExtractorFactory::UpdateExtractors(const char *newUpdateApkPath) {
     Mutex::Autolock autoLock(gPluginMutex);
@@ -301,7 +306,7 @@
         return;
     }
 
-    std::shared_ptr<List<sp<ExtractorPlugin>>> newList(new List<sp<ExtractorPlugin>>());
+    std::shared_ptr<std::list<sp<ExtractorPlugin>>> newList(new std::list<sp<ExtractorPlugin>>());
 
     RegisterExtractorsInSystem("/system/lib"
 #ifdef __LP64__
@@ -319,6 +324,7 @@
         RegisterExtractorsInApk(newUpdateApkPath, *newList);
     }
 
+    newList->sort(compareFunc);
     gPlugins = newList;
     gPluginsRegistered = true;
 }
diff --git a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
index 813004b..942f850 100644
--- a/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
+++ b/media/libstagefright/codecs/opus/dec/SoftOpus.cpp
@@ -431,7 +431,7 @@
             }
 
             if (mInputBufferCount == 0) {
-                CHECK(mHeader == NULL);
+                delete mHeader;
                 mHeader = new OpusHeader();
                 memset(mHeader, 0, sizeof(*mHeader));
                 if (!ParseOpusHeader(data, size, mHeader)) {
@@ -452,6 +452,9 @@
                 }
 
                 int status = OPUS_INVALID_STATE;
+                if (mDecoder != NULL) {
+                    opus_multistream_decoder_destroy(mDecoder);
+                }
                 mDecoder = opus_multistream_decoder_create(kRate,
                                                            mHeader->channels,
                                                            mHeader->num_streams,
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index 05f4104..c46a40f 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -85,9 +85,15 @@
             || mDstFormat == OMX_COLOR_Format32bitBGRA8888;
 }
 
+/*
+ * If stride is non-zero, client's stride will be used. For planar
+ * or semi-planar YUV formats, stride must be even numbers.
+ * If stride is zero, it will be calculated based on width and bpp
+ * of the format, assuming no padding on the right edge.
+ */
 ColorConverter::BitmapParams::BitmapParams(
         void *bits,
-        size_t width, size_t height,
+        size_t width, size_t height, size_t stride,
         size_t cropLeft, size_t cropTop,
         size_t cropRight, size_t cropBottom,
         OMX_COLOR_FORMATTYPE colorFromat)
@@ -101,6 +107,8 @@
       mCropBottom(cropBottom) {
     switch(mColorFormat) {
     case OMX_COLOR_Format16bitRGB565:
+    case OMX_COLOR_FormatYUV420Planar16:
+    case OMX_COLOR_FormatCbYCrY:
         mBpp = 2;
         mStride = 2 * mWidth;
         break;
@@ -112,13 +120,7 @@
         mStride = 4 * mWidth;
         break;
 
-    case OMX_COLOR_FormatYUV420Planar16:
-        mBpp = 2;
-        mStride = 2 * mWidth;
-        break;
-
     case OMX_COLOR_FormatYUV420Planar:
-    case OMX_COLOR_FormatCbYCrY:
     case OMX_QCOM_COLOR_FormatYVU420SemiPlanar:
     case OMX_COLOR_FormatYUV420SemiPlanar:
     case OMX_TI_COLOR_FormatYUV420PackedSemiPlanar:
@@ -132,6 +134,10 @@
         mStride = mWidth;
         break;
     }
+    // use client's stride if it's specified.
+    if (stride != 0) {
+        mStride = stride;
+    }
 }
 
 size_t ColorConverter::BitmapParams::cropWidth() const {
@@ -144,21 +150,21 @@
 
 status_t ColorConverter::convert(
         const void *srcBits,
-        size_t srcWidth, size_t srcHeight,
+        size_t srcWidth, size_t srcHeight, size_t srcStride,
         size_t srcCropLeft, size_t srcCropTop,
         size_t srcCropRight, size_t srcCropBottom,
         void *dstBits,
-        size_t dstWidth, size_t dstHeight,
+        size_t dstWidth, size_t dstHeight, size_t dstStride,
         size_t dstCropLeft, size_t dstCropTop,
         size_t dstCropRight, size_t dstCropBottom) {
     BitmapParams src(
             const_cast<void *>(srcBits),
-            srcWidth, srcHeight,
+            srcWidth, srcHeight, srcStride,
             srcCropLeft, srcCropTop, srcCropRight, srcCropBottom, mSrcFormat);
 
     BitmapParams dst(
             dstBits,
-            dstWidth, dstHeight,
+            dstWidth, dstHeight, dstStride,
             dstCropLeft, dstCropTop, dstCropRight, dstCropBottom, mDstFormat);
 
     if (!((src.mCropLeft & 1) == 0
@@ -792,15 +798,15 @@
 
     uint8_t *kAdjustedClip = initClip();
 
-    uint16_t *dst_ptr = (uint16_t *)dst.mBits
-        + dst.mCropTop * dst.mWidth + dst.mCropLeft;
+    uint16_t *dst_ptr = (uint16_t *)((uint8_t *)
+            dst.mBits + dst.mCropTop * dst.mStride + dst.mCropLeft * dst.mBpp);
 
     const uint8_t *src_y =
-        (const uint8_t *)src.mBits + src.mCropTop * src.mWidth + src.mCropLeft;
+        (const uint8_t *)src.mBits + src.mCropTop * src.mStride + src.mCropLeft;
 
     const uint8_t *src_u =
-        (const uint8_t *)src_y + src.mWidth * src.mHeight
-        + src.mCropTop * src.mWidth + src.mCropLeft;
+        (const uint8_t *)src.mBits + src.mHeight * src.mStride +
+        src.mCropTop * src.mStride / 2 + src.mCropLeft;
 
     for (size_t y = 0; y < src.cropHeight(); ++y) {
         for (size_t x = 0; x < src.cropWidth(); x += 2) {
@@ -842,13 +848,13 @@
             }
         }
 
-        src_y += src.mWidth;
+        src_y += src.mStride;
 
         if (y & 1) {
-            src_u += src.mWidth;
+            src_u += src.mStride;
         }
 
-        dst_ptr += dst.mWidth;
+        dst_ptr = (uint16_t*)((uint8_t*)dst_ptr + dst.mStride);
     }
 
     return OK;
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index 657a05b..359df3d 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -44,6 +44,7 @@
       mNativeWindow(nativeWindow),
       mWidth(0),
       mHeight(0),
+      mStride(0),
       mCropLeft(0),
       mCropTop(0),
       mCropRight(0),
@@ -67,9 +68,10 @@
     int32_t colorFormatNew;
     CHECK(format->findInt32("color-format", &colorFormatNew));
 
-    int32_t widthNew, heightNew;
-    CHECK(format->findInt32("stride", &widthNew));
+    int32_t widthNew, heightNew, strideNew;
+    CHECK(format->findInt32("width", &widthNew));
     CHECK(format->findInt32("slice-height", &heightNew));
+    CHECK(format->findInt32("stride", &strideNew));
 
     int32_t cropLeftNew, cropTopNew, cropRightNew, cropBottomNew;
     if (!format->findRect(
@@ -106,6 +108,7 @@
     mColorFormat = static_cast<OMX_COLOR_FORMATTYPE>(colorFormatNew);
     mWidth = widthNew;
     mHeight = heightNew;
+    mStride = strideNew;
     mCropLeft = cropLeftNew;
     mCropTop = cropTopNew;
     mCropRight = cropRightNew;
@@ -276,20 +279,15 @@
     if (mConverter) {
         mConverter->convert(
                 data,
-                mWidth, mHeight,
+                mWidth, mHeight, mStride,
                 mCropLeft, mCropTop, mCropRight, mCropBottom,
                 dst,
-                buf->stride, buf->height,
+                buf->stride, buf->height, 0,
                 0, 0, mCropWidth - 1, mCropHeight - 1);
     } else if (mColorFormat == OMX_COLOR_FormatYUV420Planar) {
-        const uint8_t *src_y = (const uint8_t *)data;
-        const uint8_t *src_u =
-                (const uint8_t *)data + mWidth * mHeight;
-        const uint8_t *src_v = src_u + (mWidth / 2 * mHeight / 2);
-
-        src_y +=mCropLeft + mCropTop * mWidth;
-        src_u +=(mCropLeft + mCropTop * mWidth / 2)/2;
-        src_v +=(mCropLeft + mCropTop * mWidth / 2)/2;
+        const uint8_t *src_y = (const uint8_t *)data + mCropTop * mStride + mCropLeft;
+        const uint8_t *src_u = (const uint8_t *)data + mStride * mHeight + mCropTop * mStride / 4;
+        const uint8_t *src_v = (const uint8_t *)src_u + mStride * mHeight / 4;
 
         uint8_t *dst_y = (uint8_t *)dst;
         size_t dst_y_size = buf->stride * buf->height;
@@ -305,7 +303,7 @@
         for (int y = 0; y < mCropHeight; ++y) {
             memcpy(dst_y, src_y, mCropWidth);
 
-            src_y += mWidth;
+            src_y += mStride;
             dst_y += buf->stride;
         }
 
@@ -313,19 +311,15 @@
             memcpy(dst_u, src_u, (mCropWidth + 1) / 2);
             memcpy(dst_v, src_v, (mCropWidth + 1) / 2);
 
-            src_u += mWidth / 2;
-            src_v += mWidth / 2;
+            src_u += mStride / 2;
+            src_v += mStride / 2;
             dst_u += dst_c_stride;
             dst_v += dst_c_stride;
         }
     } else if (mColorFormat == OMX_COLOR_FormatYUV420Planar16) {
-        const uint16_t *src_y = (const uint16_t *)data;
-        const uint16_t *src_u = (const uint16_t *)data + mWidth * mHeight;
-        const uint16_t *src_v = src_u + (mWidth / 2 * mHeight / 2);
-
-        src_y += mCropLeft + mCropTop * mWidth;
-        src_u += (mCropLeft + mCropTop * mWidth / 2) / 2;
-        src_v += (mCropLeft + mCropTop * mWidth / 2) / 2;
+        const uint8_t *src_y = (const uint8_t *)data + mCropTop * mStride + mCropLeft * 2;
+        const uint8_t *src_u = (const uint8_t *)data + mStride * mHeight + mCropTop * mStride / 4;
+        const uint8_t *src_v = (const uint8_t *)src_u + mStride * mHeight / 4;
 
         uint8_t *dst_y = (uint8_t *)dst;
         size_t dst_y_size = buf->stride * buf->height;
@@ -340,21 +334,21 @@
 
         for (int y = 0; y < mCropHeight; ++y) {
             for (int x = 0; x < mCropWidth; ++x) {
-                dst_y[x] = (uint8_t)(src_y[x] >> 2);
+                dst_y[x] = (uint8_t)(((uint16_t *)src_y)[x] >> 2);
             }
 
-            src_y += mWidth;
+            src_y += mStride;
             dst_y += buf->stride;
         }
 
         for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
             for (int x = 0; x < (mCropWidth + 1) / 2; ++x) {
-                dst_u[x] = (uint8_t)(src_u[x] >> 2);
-                dst_v[x] = (uint8_t)(src_v[x] >> 2);
+                dst_u[x] = (uint8_t)(((uint16_t *)src_u)[x] >> 2);
+                dst_v[x] = (uint8_t)(((uint16_t *)src_v)[x] >> 2);
             }
 
-            src_u += mWidth / 2;
-            src_v += mWidth / 2;
+            src_u += mStride / 2;
+            src_v += mStride / 2;
             dst_u += dst_c_stride;
             dst_v += dst_c_stride;
         }
diff --git a/media/libstagefright/include/SoftwareRenderer.h b/media/libstagefright/include/SoftwareRenderer.h
index c286516..64dca4e 100644
--- a/media/libstagefright/include/SoftwareRenderer.h
+++ b/media/libstagefright/include/SoftwareRenderer.h
@@ -51,7 +51,7 @@
     ColorConverter *mConverter;
     YUVMode mYUVMode;
     sp<ANativeWindow> mNativeWindow;
-    int32_t mWidth, mHeight;
+    int32_t mWidth, mHeight, mStride;
     int32_t mCropLeft, mCropTop, mCropRight, mCropBottom;
     int32_t mCropWidth, mCropHeight;
     int32_t mRotationDegrees;
diff --git a/media/libstagefright/include/media/stagefright/ColorConverter.h b/media/libstagefright/include/media/stagefright/ColorConverter.h
index 5b3543d..2b8c7c8 100644
--- a/media/libstagefright/include/media/stagefright/ColorConverter.h
+++ b/media/libstagefright/include/media/stagefright/ColorConverter.h
@@ -37,11 +37,11 @@
 
     status_t convert(
             const void *srcBits,
-            size_t srcWidth, size_t srcHeight,
+            size_t srcWidth, size_t srcHeight, size_t srcStride,
             size_t srcCropLeft, size_t srcCropTop,
             size_t srcCropRight, size_t srcCropBottom,
             void *dstBits,
-            size_t dstWidth, size_t dstHeight,
+            size_t dstWidth, size_t dstHeight, size_t dstStride,
             size_t dstCropLeft, size_t dstCropTop,
             size_t dstCropRight, size_t dstCropBottom);
 
@@ -49,7 +49,7 @@
     struct BitmapParams {
         BitmapParams(
                 void *bits,
-                size_t width, size_t height,
+                size_t width, size_t height, size_t stride,
                 size_t cropLeft, size_t cropTop,
                 size_t cropRight, size_t cropBottom,
                 OMX_COLOR_FORMATTYPE colorFromat);
diff --git a/media/libstagefright/include/media/stagefright/MediaExtractorFactory.h b/media/libstagefright/include/media/stagefright/MediaExtractorFactory.h
index fb9f5bd..d5f4b35 100644
--- a/media/libstagefright/include/media/stagefright/MediaExtractorFactory.h
+++ b/media/libstagefright/include/media/stagefright/MediaExtractorFactory.h
@@ -22,7 +22,6 @@
 
 #include <media/IMediaExtractor.h>
 #include <media/MediaExtractor.h>
-#include <utils/List.h>
 
 namespace android {
 
@@ -40,15 +39,15 @@
 
 private:
     static Mutex gPluginMutex;
-    static std::shared_ptr<List<sp<ExtractorPlugin>>> gPlugins;
+    static std::shared_ptr<std::list<sp<ExtractorPlugin>>> gPlugins;
     static bool gPluginsRegistered;
 
     static void RegisterExtractorsInApk(
-            const char *apkPath, List<sp<ExtractorPlugin>> &pluginList);
+            const char *apkPath, std::list<sp<ExtractorPlugin>> &pluginList);
     static void RegisterExtractorsInSystem(
-            const char *libDirPath, List<sp<ExtractorPlugin>> &pluginList);
+            const char *libDirPath, std::list<sp<ExtractorPlugin>> &pluginList);
     static void RegisterExtractor(
-            const sp<ExtractorPlugin> &plugin, List<sp<ExtractorPlugin>> &pluginList);
+            const sp<ExtractorPlugin> &plugin, std::list<sp<ExtractorPlugin>> &pluginList);
 
     static MediaExtractor::CreatorFunc sniff(DataSourceBase *source,
             float *confidence, void **meta, MediaExtractor::FreeMetaFunc *freeMeta,
diff --git a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
index 935dc34..672a37c 100644
--- a/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
+++ b/media/libstagefright/omx/SoftVideoDecoderOMXComponent.cpp
@@ -154,12 +154,12 @@
     outDef->format.video.nFrameWidth = outputBufferWidth();
     outDef->format.video.nFrameHeight = outputBufferHeight();
     outDef->format.video.eColorFormat = mOutputFormat;
-    outDef->format.video.nStride = outDef->format.video.nFrameWidth;
     outDef->format.video.nSliceHeight = outDef->format.video.nFrameHeight;
 
     int32_t bpp = (mOutputFormat == OMX_COLOR_FormatYUV420Planar16) ? 2 : 1;
+    outDef->format.video.nStride = outDef->format.video.nFrameWidth * bpp;
     outDef->nBufferSize =
-        (outDef->format.video.nStride * outDef->format.video.nSliceHeight * bpp * 3) / 2;
+            (outDef->format.video.nStride * outDef->format.video.nSliceHeight * 3) / 2;
 
     OMX_PARAM_PORTDEFINITIONTYPE *inDef = &editPortInfo(kInputPortIndex)->mDef;
     inDef->format.video.nFrameWidth = mWidth;
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index d6dae5b..de8e46a 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -21,9 +21,11 @@
         "MemoryLeakTrackUtil.cpp",
         "ProcessInfo.cpp",
         "SchedulingPolicyService.cpp",
+        "ServiceUtilities.cpp",
     ],
     shared_libs: [
         "libbinder",
+        "libcutils",
         "liblog",
         "libutils",
         "libmemunreachable",
diff --git a/services/audioflinger/ServiceUtilities.cpp b/media/utils/ServiceUtilities.cpp
similarity index 84%
rename from services/audioflinger/ServiceUtilities.cpp
rename to media/utils/ServiceUtilities.cpp
index aa267ea..0d50be0 100644
--- a/services/audioflinger/ServiceUtilities.cpp
+++ b/media/utils/ServiceUtilities.cpp
@@ -14,12 +14,13 @@
  * limitations under the License.
  */
 
+#define LOG_TAG "ServiceUtilities"
+
 #include <binder/AppOpsManager.h>
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
 #include <binder/PermissionCache.h>
-#include <private/android_filesystem_config.h>
-#include "ServiceUtilities.h"
+#include "mediautils/ServiceUtilities.h"
 
 /* When performing permission checks we do not use permission cache for
  * runtime permissions (protection level dangerous) as they may change at
@@ -32,24 +33,6 @@
 
 static const String16 sAndroidPermissionRecordAudio("android.permission.RECORD_AUDIO");
 
-// Not valid until initialized by AudioFlinger constructor.  It would have to be
-// re-initialized if the process containing AudioFlinger service forks (which it doesn't).
-// This is often used to validate binder interface calls within audioserver
-// (e.g. AudioPolicyManager to AudioFlinger).
-pid_t getpid_cached;
-
-// A trusted calling UID may specify the client UID as part of a binder interface call.
-// otherwise the calling UID must be equal to the client UID.
-bool isTrustedCallingUid(uid_t uid) {
-    switch (uid) {
-    case AID_MEDIA:
-    case AID_AUDIOSERVER:
-        return true;
-    default:
-        return false;
-    }
-}
-
 static String16 resolveCallingPackage(PermissionController& permissionController,
         const String16& opPackageName, uid_t uid) {
     if (opPackageName.size() > 0) {
@@ -71,16 +54,11 @@
     return packages[0];
 }
 
-static inline bool isAudioServerOrRoot(uid_t uid) {
-    // AID_ROOT is OK for command-line tests.  Native unforked audioserver always OK.
-    return uid == AID_ROOT || uid == AID_AUDIOSERVER ;
-}
-
 static bool checkRecordingInternal(const String16& opPackageName, pid_t pid,
         uid_t uid, bool start) {
     // Okay to not track in app ops as audio server is us and if
     // device is rooted security model is considered compromised.
-    if (isAudioServerOrRoot(uid)) return true;
+    if (isAudioServerOrRootUid(uid)) return true;
 
     // We specify a pid and uid here as mediaserver (aka MediaRecorder or StageFrightRecorder)
     // may open a record track on behalf of a client.  Note that pid may be a tid.
@@ -127,7 +105,7 @@
 void finishRecording(const String16& opPackageName, uid_t uid) {
     // Okay to not track in app ops as audio server is us and if
     // device is rooted security model is considered compromised.
-    if (isAudioServerOrRoot(uid)) return;
+    if (isAudioServerOrRootUid(uid)) return;
 
     PermissionController permissionController;
     String16 resolvedOpPackageName = resolveCallingPackage(
@@ -142,7 +120,7 @@
 }
 
 bool captureAudioOutputAllowed(pid_t pid, uid_t uid) {
-    if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+    if (isAudioServerOrRootUid(uid)) return true;
     static const String16 sCaptureAudioOutput("android.permission.CAPTURE_AUDIO_OUTPUT");
     bool ok = PermissionCache::checkPermission(sCaptureAudioOutput, pid, uid);
     if (!ok) ALOGE("Request requires android.permission.CAPTURE_AUDIO_OUTPUT");
@@ -163,7 +141,8 @@
 }
 
 bool settingsAllowed() {
-    if (getpid_cached == IPCThreadState::self()->getCallingPid()) return true;
+    // given this is a permission check, could this be isAudioServerOrRootUid()?
+    if (isAudioServerUid(IPCThreadState::self()->getCallingUid())) return true;
     static const String16 sAudioSettings("android.permission.MODIFY_AUDIO_SETTINGS");
     // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
     bool ok = PermissionCache::checkCallingPermission(sAudioSettings);
@@ -180,7 +159,6 @@
 }
 
 bool dumpAllowed() {
-    // don't optimize for same pid, since mediaserver never dumps itself
     static const String16 sDump("android.permission.DUMP");
     // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
     bool ok = PermissionCache::checkCallingPermission(sDump);
@@ -196,4 +174,29 @@
     return ok;
 }
 
+status_t checkIMemory(const sp<IMemory>& iMemory)
+{
+    if (iMemory == 0) {
+        ALOGE("%s check failed: NULL IMemory pointer", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
+    sp<IMemoryHeap> heap = iMemory->getMemory();
+    if (heap == 0) {
+        ALOGE("%s check failed: NULL heap pointer", __FUNCTION__);
+        return BAD_VALUE;
+    }
+
+    off_t size = lseek(heap->getHeapID(), 0, SEEK_END);
+    lseek(heap->getHeapID(), 0, SEEK_SET);
+
+    if (iMemory->pointer() == NULL || size < (off_t)iMemory->size()) {
+        ALOGE("%s check failed: pointer %p size %zu fd size %u",
+              __FUNCTION__, iMemory->pointer(), iMemory->size(), (uint32_t)size);
+        return BAD_VALUE;
+    }
+
+    return NO_ERROR;
+}
+
 } // namespace android
diff --git a/media/utils/include/mediautils/ServiceUtilities.h b/media/utils/include/mediautils/ServiceUtilities.h
new file mode 100644
index 0000000..0911744
--- /dev/null
+++ b/media/utils/include/mediautils/ServiceUtilities.h
@@ -0,0 +1,74 @@
+/*
+ * Copyright (C) 2012 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <unistd.h>
+
+#include <binder/IMemory.h>
+#include <binder/PermissionController.h>
+#include <cutils/multiuser.h>
+#include <private/android_filesystem_config.h>
+
+namespace android {
+
+// Audio permission utilities
+
+// Used for calls that should originate from system services.
+// We allow that some services might have separate processes to
+// handle multiple users, e.g. u10_system, u10_bluetooth, u10_radio.
+static inline bool isServiceUid(uid_t uid) {
+    return multiuser_get_app_id(uid) < AID_APP_START;
+}
+
+// Used for calls that should originate from audioserver.
+static inline bool isAudioServerUid(uid_t uid) {
+    return uid == AID_AUDIOSERVER;
+}
+
+// Used for some permission checks.
+// AID_ROOT is OK for command-line tests.  Native audioserver always OK.
+static inline bool isAudioServerOrRootUid(uid_t uid) {
+    return uid == AID_AUDIOSERVER || uid == AID_ROOT;
+}
+
+// Used for calls that should come from system server or internal.
+// Note: system server is multiprocess for multiple users.  audioserver is not.
+static inline bool isAudioServerOrSystemServerUid(uid_t uid) {
+    return multiuser_get_app_id(uid) == AID_SYSTEM || uid == AID_AUDIOSERVER;
+}
+
+// Mediaserver may forward the client PID and UID as part of a binder interface call;
+// otherwise the calling UID must be equal to the client UID.
+static inline bool isAudioServerOrMediaServerUid(uid_t uid) {
+    switch (uid) {
+    case AID_MEDIA:
+    case AID_AUDIOSERVER:
+        return true;
+    default:
+        return false;
+    }
+}
+
+bool recordingAllowed(const String16& opPackageName, pid_t pid, uid_t uid);
+bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid);
+void finishRecording(const String16& opPackageName, uid_t uid);
+bool captureAudioOutputAllowed(pid_t pid, uid_t uid);
+bool captureHotwordAllowed(pid_t pid, uid_t uid);
+bool settingsAllowed();
+bool modifyAudioRoutingAllowed();
+bool dumpAllowed();
+bool modifyPhoneStateAllowed(pid_t pid, uid_t uid);
+status_t checkIMemory(const sp<IMemory>& iMemory);
+}
diff --git a/packages/MediaComponents/Android.mk b/packages/MediaComponents/Android.mk
index def9dc5..55a5424 100644
--- a/packages/MediaComponents/Android.mk
+++ b/packages/MediaComponents/Android.mk
@@ -42,7 +42,7 @@
 #
 #LOCAL_MULTILIB := first
 #
-#LOCAL_JAVA_LIBRARIES += android-support-annotations
+#LOCAL_JAVA_LIBRARIES += androidx.annotation_annotation
 #
 ## To embed native libraries in package, uncomment the lines below.
 ##LOCAL_MODULE_TAGS := samples
@@ -60,9 +60,9 @@
 #
 ## TODO: Remove dependency with other support libraries.
 #LOCAL_STATIC_ANDROID_LIBRARIES += \
-#    android-support-v4 \
-#    android-support-v7-appcompat \
-#    android-support-v7-palette
+#    androidx.legacy_legacy-support-v4 \
+#    androidx.appcompat_appcompat \
+#    androidx.palette_palette
 #LOCAL_USE_AAPT2 := true
 #
 #include $(BUILD_PACKAGE)
diff --git a/packages/MediaComponents/res/layout/mr_controller_material_dialog_b.xml b/packages/MediaComponents/res/layout/mr_controller_material_dialog_b.xml
index b304471..f6f7be5 100644
--- a/packages/MediaComponents/res/layout/mr_controller_material_dialog_b.xml
+++ b/packages/MediaComponents/res/layout/mr_controller_material_dialog_b.xml
@@ -169,7 +169,7 @@
             android:layout_height="wrap_content"
             android:fillViewport="true"
             android:scrollIndicators="top|bottom">
-            <android.support.v7.widget.ButtonBarLayout
+            <androidx.appcompat.widget.ButtonBarLayout
                 android:layout_width="match_parent"
                 android:layout_height="wrap_content"
                 android:gravity="bottom"
@@ -184,7 +184,7 @@
                     style="?android:attr/buttonBarNeutralButtonStyle"
                     android:layout_width="wrap_content"
                     android:layout_height="wrap_content"/>
-                <android.support.v4.widget.Space
+                <androidx.legacy.widget.Space
                     android:id="@+id/spacer"
                     android:layout_width="0dp"
                     android:layout_height="0dp"
@@ -200,7 +200,7 @@
                     style="?android:attr/buttonBarPositiveButtonStyle"
                     android:layout_width="wrap_content"
                     android:layout_height="wrap_content"/>
-            </android.support.v7.widget.ButtonBarLayout>
+            </androidx.appcompat.widget.ButtonBarLayout>
         </ScrollView>
     </LinearLayout>
 </FrameLayout>
diff --git a/packages/MediaComponents/res/layout/mr_controller_volume_item.xml b/packages/MediaComponents/res/layout/mr_controller_volume_item.xml
index a89058b..12d85ae 100644
--- a/packages/MediaComponents/res/layout/mr_controller_volume_item.xml
+++ b/packages/MediaComponents/res/layout/mr_controller_volume_item.xml
@@ -40,7 +40,7 @@
                        android:layout_marginBottom="8dp"
                        android:scaleType="fitCenter"
                        android:src="?attr/mediaRouteAudioTrackDrawable" />
-            <android.support.v7.app.MediaRouteVolumeSlider
+            <androidx.mediarouter.app.MediaRouteVolumeSlider
                 android:id="@+id/mr_volume_slider"
                 android:layout_width="fill_parent"
                 android:layout_height="40dp"
diff --git a/packages/MediaComponents/src/com/android/media/MediaBrowser2Impl.java b/packages/MediaComponents/src/com/android/media/MediaBrowser2Impl.java
index c909099..0327beb 100644
--- a/packages/MediaComponents/src/com/android/media/MediaBrowser2Impl.java
+++ b/packages/MediaComponents/src/com/android/media/MediaBrowser2Impl.java
@@ -19,7 +19,6 @@
 import android.content.Context;
 import android.media.MediaBrowser2;
 import android.media.MediaBrowser2.BrowserCallback;
-import android.media.MediaController2;
 import android.media.MediaItem2;
 import android.media.SessionToken2;
 import android.media.update.MediaBrowser2Provider;
diff --git a/packages/MediaComponents/src/com/android/media/MediaController2Impl.java b/packages/MediaComponents/src/com/android/media/MediaController2Impl.java
index 249365a..2883087 100644
--- a/packages/MediaComponents/src/com/android/media/MediaController2Impl.java
+++ b/packages/MediaComponents/src/com/android/media/MediaController2Impl.java
@@ -16,7 +16,6 @@
 
 package com.android.media;
 
-import static android.media.SessionCommand2.COMMAND_CODE_SET_VOLUME;
 import static android.media.SessionCommand2.COMMAND_CODE_PLAYLIST_ADD_ITEM;
 import static android.media.SessionCommand2.COMMAND_CODE_PLAYLIST_REMOVE_ITEM;
 import static android.media.SessionCommand2.COMMAND_CODE_PLAYLIST_REPLACE_ITEM;
@@ -30,6 +29,7 @@
 import static android.media.SessionCommand2.COMMAND_CODE_SESSION_PREPARE_FROM_MEDIA_ID;
 import static android.media.SessionCommand2.COMMAND_CODE_SESSION_PREPARE_FROM_SEARCH;
 import static android.media.SessionCommand2.COMMAND_CODE_SESSION_PREPARE_FROM_URI;
+import static android.media.SessionCommand2.COMMAND_CODE_SET_VOLUME;
 
 import android.app.PendingIntent;
 import android.content.ComponentName;
@@ -44,11 +44,11 @@
 import android.media.MediaMetadata2;
 import android.media.MediaPlaylistAgent.RepeatMode;
 import android.media.MediaPlaylistAgent.ShuffleMode;
-import android.media.SessionCommand2;
 import android.media.MediaSession2.CommandButton;
-import android.media.SessionCommandGroup2;
 import android.media.MediaSessionService2;
 import android.media.Rating2;
+import android.media.SessionCommand2;
+import android.media.SessionCommandGroup2;
 import android.media.SessionToken2;
 import android.media.update.MediaController2Provider;
 import android.net.Uri;
@@ -58,10 +58,11 @@
 import android.os.RemoteException;
 import android.os.ResultReceiver;
 import android.os.UserHandle;
-import android.support.annotation.GuardedBy;
 import android.text.TextUtils;
 import android.util.Log;
 
+import androidx.annotation.GuardedBy;
+
 import java.util.ArrayList;
 import java.util.List;
 import java.util.concurrent.Executor;
diff --git a/packages/MediaComponents/src/com/android/media/MediaController2Stub.java b/packages/MediaComponents/src/com/android/media/MediaController2Stub.java
index 2cfc5df..ece4a00 100644
--- a/packages/MediaComponents/src/com/android/media/MediaController2Stub.java
+++ b/packages/MediaComponents/src/com/android/media/MediaController2Stub.java
@@ -21,8 +21,8 @@
 import android.media.MediaController2;
 import android.media.MediaItem2;
 import android.media.MediaMetadata2;
-import android.media.SessionCommand2;
 import android.media.MediaSession2.CommandButton;
+import android.media.SessionCommand2;
 import android.media.SessionCommandGroup2;
 import android.os.Bundle;
 import android.os.ResultReceiver;
diff --git a/packages/MediaComponents/src/com/android/media/MediaSession2Impl.java b/packages/MediaComponents/src/com/android/media/MediaSession2Impl.java
index 4ec6042..72ecf54 100644
--- a/packages/MediaComponents/src/com/android/media/MediaSession2Impl.java
+++ b/packages/MediaComponents/src/com/android/media/MediaSession2Impl.java
@@ -44,13 +44,13 @@
 import android.media.MediaPlaylistAgent.PlaylistEventCallback;
 import android.media.MediaSession2;
 import android.media.MediaSession2.Builder;
-import android.media.SessionCommand2;
 import android.media.MediaSession2.CommandButton;
-import android.media.SessionCommandGroup2;
 import android.media.MediaSession2.ControllerInfo;
 import android.media.MediaSession2.OnDataSourceMissingHelper;
 import android.media.MediaSession2.SessionCallback;
 import android.media.MediaSessionService2;
+import android.media.SessionCommand2;
+import android.media.SessionCommandGroup2;
 import android.media.SessionToken2;
 import android.media.VolumeProvider2;
 import android.media.session.MediaSessionManager;
@@ -60,10 +60,11 @@
 import android.os.Parcelable;
 import android.os.Process;
 import android.os.ResultReceiver;
-import android.support.annotation.GuardedBy;
 import android.text.TextUtils;
 import android.util.Log;
 
+import androidx.annotation.GuardedBy;
+
 import java.lang.ref.WeakReference;
 import java.lang.reflect.Field;
 import java.util.ArrayList;
diff --git a/packages/MediaComponents/src/com/android/media/MediaSession2Stub.java b/packages/MediaComponents/src/com/android/media/MediaSession2Stub.java
index ec657d7..11ccd9f 100644
--- a/packages/MediaComponents/src/com/android/media/MediaSession2Stub.java
+++ b/packages/MediaComponents/src/com/android/media/MediaSession2Stub.java
@@ -22,11 +22,11 @@
 import android.media.MediaItem2;
 import android.media.MediaLibraryService2.LibraryRoot;
 import android.media.MediaMetadata2;
-import android.media.SessionCommand2;
 import android.media.MediaSession2.CommandButton;
-import android.media.SessionCommandGroup2;
 import android.media.MediaSession2.ControllerInfo;
 import android.media.Rating2;
+import android.media.SessionCommand2;
+import android.media.SessionCommandGroup2;
 import android.media.VolumeProvider2;
 import android.net.Uri;
 import android.os.Binder;
@@ -35,13 +35,14 @@
 import android.os.IBinder;
 import android.os.RemoteException;
 import android.os.ResultReceiver;
-import android.support.annotation.GuardedBy;
-import android.support.annotation.NonNull;
 import android.text.TextUtils;
 import android.util.ArrayMap;
 import android.util.Log;
 import android.util.SparseArray;
 
+import androidx.annotation.GuardedBy;
+import androidx.annotation.NonNull;
+
 import com.android.media.MediaLibraryService2Impl.MediaLibrarySessionImpl;
 import com.android.media.MediaSession2Impl.CommandButtonImpl;
 import com.android.media.MediaSession2Impl.CommandGroupImpl;
diff --git a/packages/MediaComponents/src/com/android/media/MediaSessionService2Impl.java b/packages/MediaComponents/src/com/android/media/MediaSessionService2Impl.java
index c33eb65..d975839 100644
--- a/packages/MediaComponents/src/com/android/media/MediaSessionService2Impl.java
+++ b/packages/MediaComponents/src/com/android/media/MediaSessionService2Impl.java
@@ -20,7 +20,6 @@
 
 import android.app.Notification;
 import android.app.NotificationManager;
-import android.content.Context;
 import android.content.Intent;
 import android.media.MediaPlayerBase;
 import android.media.MediaPlayerBase.PlayerEventCallback;
@@ -31,9 +30,10 @@
 import android.media.SessionToken2.TokenType;
 import android.media.update.MediaSessionService2Provider;
 import android.os.IBinder;
-import android.support.annotation.GuardedBy;
 import android.util.Log;
 
+import androidx.annotation.GuardedBy;
+
 // TODO(jaewan): Need a test for session service itself.
 public class MediaSessionService2Impl implements MediaSessionService2Provider {
 
diff --git a/packages/MediaComponents/src/com/android/media/Rating2Impl.java b/packages/MediaComponents/src/com/android/media/Rating2Impl.java
index d558129..e2b9f0a 100644
--- a/packages/MediaComponents/src/com/android/media/Rating2Impl.java
+++ b/packages/MediaComponents/src/com/android/media/Rating2Impl.java
@@ -18,7 +18,6 @@
 
 import static android.media.Rating2.*;
 
-import android.content.Context;
 import android.media.Rating2;
 import android.media.Rating2.Style;
 import android.media.update.Rating2Provider;
diff --git a/packages/MediaComponents/src/com/android/media/RoutePlayer.java b/packages/MediaComponents/src/com/android/media/RoutePlayer.java
index 9450d34..ebff0e2 100644
--- a/packages/MediaComponents/src/com/android/media/RoutePlayer.java
+++ b/packages/MediaComponents/src/com/android/media/RoutePlayer.java
@@ -23,7 +23,8 @@
 import android.net.Uri;
 import android.os.Build;
 import android.os.Bundle;
-import android.support.annotation.RequiresApi;
+
+import androidx.annotation.RequiresApi;
 
 import com.android.support.mediarouter.media.MediaItemStatus;
 import com.android.support.mediarouter.media.MediaRouter;
@@ -33,8 +34,6 @@
 import com.android.support.mediarouter.media.RemotePlaybackClient.SessionActionCallback;
 import com.android.support.mediarouter.media.RemotePlaybackClient.StatusCallback;
 
-import java.util.Map;
-
 @RequiresApi(api = Build.VERSION_CODES.LOLLIPOP)
 public class RoutePlayer extends MediaSession.Callback {
     public static final long PLAYBACK_ACTIONS = PlaybackState.ACTION_PAUSE
diff --git a/packages/MediaComponents/src/com/android/media/SessionToken2Impl.java b/packages/MediaComponents/src/com/android/media/SessionToken2Impl.java
index a5cf8c4..f792712 100644
--- a/packages/MediaComponents/src/com/android/media/SessionToken2Impl.java
+++ b/packages/MediaComponents/src/com/android/media/SessionToken2Impl.java
@@ -16,9 +16,9 @@
 
 package com.android.media;
 
+import static android.media.SessionToken2.TYPE_LIBRARY_SERVICE;
 import static android.media.SessionToken2.TYPE_SESSION;
 import static android.media.SessionToken2.TYPE_SESSION_SERVICE;
-import static android.media.SessionToken2.TYPE_LIBRARY_SERVICE;
 
 import android.content.Context;
 import android.content.Intent;
diff --git a/packages/MediaComponents/src/com/android/media/subtitle/SubtitleController.java b/packages/MediaComponents/src/com/android/media/subtitle/SubtitleController.java
index a4d55d7..97d3927 100644
--- a/packages/MediaComponents/src/com/android/media/subtitle/SubtitleController.java
+++ b/packages/MediaComponents/src/com/android/media/subtitle/SubtitleController.java
@@ -16,12 +16,8 @@
 
 package com.android.media.subtitle;
 
-import java.util.Locale;
-import java.util.Vector;
-
 import android.content.Context;
 import android.media.MediaFormat;
-import android.media.MediaPlayer2;
 import android.media.MediaPlayer2.TrackInfo;
 import android.os.Handler;
 import android.os.Looper;
@@ -30,6 +26,9 @@
 
 import com.android.media.subtitle.SubtitleTrack.RenderingWidget;
 
+import java.util.Locale;
+import java.util.Vector;
+
 // Note: This is forked from android.media.SubtitleController since P
 /**
  * The subtitle controller provides the architecture to display subtitles for a
diff --git a/packages/MediaComponents/src/com/android/media/update/ApiFactory.java b/packages/MediaComponents/src/com/android/media/update/ApiFactory.java
index d7be549..f75b75e 100644
--- a/packages/MediaComponents/src/com/android/media/update/ApiFactory.java
+++ b/packages/MediaComponents/src/com/android/media/update/ApiFactory.java
@@ -31,13 +31,13 @@
 import android.media.MediaMetadata2;
 import android.media.MediaPlaylistAgent;
 import android.media.MediaSession2;
-import android.media.SessionCommand2;
-import android.media.SessionCommandGroup2;
 import android.media.MediaSession2.ControllerInfo;
 import android.media.MediaSession2.SessionCallback;
 import android.media.MediaSessionService2;
 import android.media.MediaSessionService2.MediaNotification;
 import android.media.Rating2;
+import android.media.SessionCommand2;
+import android.media.SessionCommandGroup2;
 import android.media.SessionToken2;
 import android.media.VolumeProvider2;
 import android.media.update.MediaBrowser2Provider;
@@ -59,11 +59,12 @@
 import android.media.update.VolumeProvider2Provider;
 import android.os.Bundle;
 import android.os.IInterface;
-import android.support.annotation.Nullable;
 import android.util.AttributeSet;
 import android.widget.MediaControlView2;
 import android.widget.VideoView2;
 
+import androidx.annotation.Nullable;
+
 import com.android.media.IMediaController2;
 import com.android.media.MediaBrowser2Impl;
 import com.android.media.MediaController2Impl;
diff --git a/packages/MediaComponents/src/com/android/media/update/ApiHelper.java b/packages/MediaComponents/src/com/android/media/update/ApiHelper.java
index ad8bb48..dc5e5e2 100644
--- a/packages/MediaComponents/src/com/android/media/update/ApiHelper.java
+++ b/packages/MediaComponents/src/com/android/media/update/ApiHelper.java
@@ -18,21 +18,21 @@
 
 import android.annotation.Nullable;
 import android.content.Context;
-import android.content.ContextWrapper;
 import android.content.pm.ApplicationInfo;
 import android.content.pm.PackageManager.NameNotFoundException;
 import android.content.res.Resources;
 import android.content.res.Resources.Theme;
 import android.content.res.XmlResourceParser;
-import android.support.annotation.GuardedBy;
-import android.support.v4.widget.Space;
-import android.support.v7.widget.ButtonBarLayout;
 import android.util.AttributeSet;
 import android.view.ContextThemeWrapper;
 import android.view.LayoutInflater;
 import android.view.View;
 import android.view.ViewGroup;
 
+import androidx.annotation.GuardedBy;
+import androidx.appcompat.widget.ButtonBarLayout;
+import androidx.legacy.widget.Space;
+
 import com.android.support.mediarouter.app.MediaRouteButton;
 import com.android.support.mediarouter.app.MediaRouteExpandCollapseButton;
 import com.android.support.mediarouter.app.MediaRouteVolumeSlider;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteActionProvider.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteActionProvider.java
index d3e8d47..98c0d17 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteActionProvider.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteActionProvider.java
@@ -19,11 +19,12 @@
 import android.annotation.NonNull;
 import android.annotation.Nullable;
 import android.content.Context;
-import android.support.v4.view.ActionProvider;
 import android.util.Log;
 import android.view.View;
 import android.view.ViewGroup;
 
+import androidx.core.view.ActionProvider;
+
 import com.android.support.mediarouter.media.MediaRouteSelector;
 import com.android.support.mediarouter.media.MediaRouter;
 
@@ -48,7 +49,7 @@
  * <h3>Prerequisites</h3>
  * <p>
  * To use the media route action provider, the activity must be a subclass of
- * {@link AppCompatActivity} from the <code>android.support.v7.appcompat</code>
+ * {@link AppCompatActivity} from the <code>androidx.appcompat.appcompat</code>
  * support library.  Refer to support library documentation for details.
  * </p>
  *
@@ -65,7 +66,7 @@
  *     &lt;item android:id="@+id/media_route_menu_item"
  *         android:title="@string/media_route_menu_title"
  *         app:showAsAction="always"
- *         app:actionProviderClass="android.support.v7.app.MediaRouteActionProvider"/>
+ *         app:actionProviderClass="androidx.mediarouter.app.MediaRouteActionProvider"/>
  * &lt;/menu>
  * </pre><p>
  * Then configure the menu and set the route selector for the chooser.
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteButton.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteButton.java
index fde8a63..e82fcb9 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteButton.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteButton.java
@@ -28,14 +28,15 @@
 import android.graphics.drawable.AnimationDrawable;
 import android.graphics.drawable.Drawable;
 import android.os.AsyncTask;
-import android.support.v4.graphics.drawable.DrawableCompat;
-import android.support.v7.widget.TooltipCompat;
 import android.util.AttributeSet;
 import android.util.Log;
 import android.util.SparseArray;
 import android.view.SoundEffectConstants;
 import android.view.View;
 
+import androidx.appcompat.widget.TooltipCompat;
+import androidx.core.graphics.drawable.DrawableCompat;
+
 import com.android.media.update.ApiHelper;
 import com.android.media.update.R;
 import com.android.support.mediarouter.media.MediaRouteSelector;
@@ -70,7 +71,7 @@
  * <h3>Prerequisites</h3>
  * <p>
  * To use the media route button, the activity must be a subclass of
- * {@link FragmentActivity} from the <code>android.support.v4</code>
+ * {@link FragmentActivity} from the <code>androidx.core./code>
  * support library.  Refer to support library documentation for details.
  * </p>
  *
@@ -81,9 +82,9 @@
     private static final String TAG = "MediaRouteButton";
 
     private static final String CHOOSER_FRAGMENT_TAG =
-            "android.support.v7.mediarouter:MediaRouteChooserDialogFragment";
+            "androidx.mediarouter.media.outer:MediaRouteChooserDialogFragment";
     private static final String CONTROLLER_FRAGMENT_TAG =
-            "android.support.v7.mediarouter:MediaRouteControllerDialogFragment";
+            "androidx.mediarouter.media.outer:MediaRouteControllerDialogFragment";
 
     private final MediaRouter mRouter;
     private final MediaRouterCallback mCallback;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteChooserDialog.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteChooserDialog.java
index cac64d9..f24028a 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteChooserDialog.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteChooserDialog.java
@@ -16,13 +16,14 @@
 
 package com.android.support.mediarouter.app;
 
-import static com.android.support.mediarouter.media.MediaRouter.RouteInfo.CONNECTION_STATE_CONNECTED;
-import static com.android.support.mediarouter.media.MediaRouter.RouteInfo.CONNECTION_STATE_CONNECTING;
+import static com.android.support.mediarouter.media.MediaRouter.RouteInfo
+        .CONNECTION_STATE_CONNECTED;
+import static com.android.support.mediarouter.media.MediaRouter.RouteInfo
+        .CONNECTION_STATE_CONNECTING;
 
 import android.annotation.NonNull;
 import android.app.Dialog;
 import android.content.Context;
-import android.content.res.Resources;
 import android.content.res.TypedArray;
 import android.graphics.drawable.Drawable;
 import android.net.Uri;
@@ -30,12 +31,10 @@
 import android.os.Handler;
 import android.os.Message;
 import android.os.SystemClock;
-import android.support.v7.app.AppCompatDialog;
 import android.text.TextUtils;
 import android.util.Log;
 import android.view.ContextThemeWrapper;
 import android.view.Gravity;
-import android.view.LayoutInflater;
 import android.view.View;
 import android.view.ViewGroup;
 import android.widget.AdapterView;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteControllerDialog.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteControllerDialog.java
index 060cfca..f6c1d2f 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteControllerDialog.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteControllerDialog.java
@@ -40,9 +40,6 @@
 import android.support.v4.media.session.MediaControllerCompat;
 import android.support.v4.media.session.MediaSessionCompat;
 import android.support.v4.media.session.PlaybackStateCompat;
-import android.support.v4.util.ObjectsCompat;
-import android.support.v4.view.accessibility.AccessibilityEventCompat;
-import android.support.v7.graphics.Palette;
 import android.text.TextUtils;
 import android.util.Log;
 import android.view.ContextThemeWrapper;
@@ -72,11 +69,15 @@
 import android.widget.SeekBar;
 import android.widget.TextView;
 
+import androidx.core.util.ObjectsCompat;
+import androidx.core.view.accessibility.AccessibilityEventCompat;
+import androidx.palette.graphics.Palette;
+
 import com.android.media.update.ApiHelper;
 import com.android.media.update.R;
+import com.android.support.mediarouter.app.OverlayListView.OverlayObject;
 import com.android.support.mediarouter.media.MediaRouteSelector;
 import com.android.support.mediarouter.media.MediaRouter;
-import com.android.support.mediarouter.app.OverlayListView.OverlayObject;
 
 import java.io.BufferedInputStream;
 import java.io.IOException;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDialogFactory.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDialogFactory.java
index a9eaf39..b5ee63e 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDialogFactory.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDialogFactory.java
@@ -16,7 +16,7 @@
 
 package com.android.support.mediarouter.app;
 
-import android.support.annotation.NonNull;
+import androidx.annotation.NonNull;
 
 /**
  * The media route dialog factory is responsible for creating the media route
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDiscoveryFragment.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDiscoveryFragment.java
index 02ee118..52aecd88 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDiscoveryFragment.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteDiscoveryFragment.java
@@ -17,10 +17,11 @@
 package com.android.support.mediarouter.app;
 
 import android.os.Bundle;
-import android.support.v4.app.Fragment;
 
-import com.android.support.mediarouter.media.MediaRouter;
+import androidx.fragment.app.Fragment;
+
 import com.android.support.mediarouter.media.MediaRouteSelector;
+import com.android.support.mediarouter.media.MediaRouter;
 
 /**
  * Media route discovery fragment.
@@ -34,7 +35,7 @@
  * provide the {@link MediaRouter} callback to register.
  * </p><p>
  * Note that the discovery callback makes the application be connected with all the
- * {@link android.support.v7.media.MediaRouteProviderService media route provider services}
+ * {@link androidx.mediarouter.media.MediaRouteProviderService media route provider services}
  * while it is registered.
  * </p>
  */
@@ -114,7 +115,7 @@
     }
 
     /**
-     * Called to create the {@link android.support.v7.media.MediaRouter.Callback callback}
+     * Called to create the {@link androidx.mediarouter.media.MediaRouter.Callback callback}
      * that will be registered.
      * <p>
      * The default callback does nothing.  The application may override this method to
@@ -129,7 +130,7 @@
 
     /**
      * Called to prepare the callback flags that will be used when the
-     * {@link android.support.v7.media.MediaRouter.Callback callback} is registered.
+     * {@link androidx.mediarouter.media.MediaRouter.Callback callback} is registered.
      * <p>
      * The default implementation returns {@link MediaRouter#CALLBACK_FLAG_REQUEST_DISCOVERY}.
      * </p>
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteExpandCollapseButton.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteExpandCollapseButton.java
index 6a0a95a..dcca6a0 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteExpandCollapseButton.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouteExpandCollapseButton.java
@@ -21,7 +21,6 @@
 import android.graphics.PorterDuff;
 import android.graphics.PorterDuffColorFilter;
 import android.graphics.drawable.AnimationDrawable;
-import android.support.v4.content.ContextCompat;
 import android.util.AttributeSet;
 import android.view.View;
 import android.widget.ImageButton;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouterThemeHelper.java b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouterThemeHelper.java
index 63f042f..b4bf8d1 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouterThemeHelper.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/app/MediaRouterThemeHelper.java
@@ -19,12 +19,13 @@
 import android.content.Context;
 import android.content.res.TypedArray;
 import android.graphics.Color;
-import android.support.annotation.IntDef;
-import android.support.v4.graphics.ColorUtils;
 import android.util.TypedValue;
 import android.view.ContextThemeWrapper;
 import android.view.View;
 
+import androidx.annotation.IntDef;
+import androidx.core.graphics.ColorUtils;
+
 import com.android.media.update.R;
 
 import java.lang.annotation.Retention;
@@ -170,7 +171,7 @@
     private static boolean isLightTheme(Context context) {
         TypedValue value = new TypedValue();
         // TODO(sungsoo): Switch to com.android.internal.R.attr.isLightTheme
-        return context.getTheme().resolveAttribute(android.support.v7.appcompat.R.attr.isLightTheme,
+        return context.getTheme().resolveAttribute(androidx.appcompat.R.attr.isLightTheme,
                 value, true) && value.data != 0;
     }
 
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/jellybean-mr1/MediaRouterJellybeanMr1.java b/packages/MediaComponents/src/com/android/support/mediarouter/jellybean-mr1/MediaRouterJellybeanMr1.java
index f8539bd..5a0bc95 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/jellybean-mr1/MediaRouterJellybeanMr1.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/jellybean-mr1/MediaRouterJellybeanMr1.java
@@ -20,7 +20,6 @@
 import android.hardware.display.DisplayManager;
 import android.os.Build;
 import android.os.Handler;
-import android.support.annotation.RequiresApi;
 import android.util.Log;
 import android.view.Display;
 
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaItemStatus.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaItemStatus.java
index 90ea2d5..92f608b 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaItemStatus.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaItemStatus.java
@@ -19,7 +19,8 @@
 import android.app.PendingIntent;
 import android.os.Bundle;
 import android.os.SystemClock;
-import android.support.v4.util.TimeUtils;
+
+import androidx.core.util.TimeUtils;
 
 /**
  * Describes the playback status of a media item.
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProvider.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProvider.java
index 91a2e1a..7ea328c 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProvider.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProvider.java
@@ -23,7 +23,8 @@
 import android.content.Intent;
 import android.os.Handler;
 import android.os.Message;
-import android.support.v4.util.ObjectsCompat;
+
+import androidx.core.util.ObjectsCompat;
 
 import com.android.support.mediarouter.media.MediaRouter.ControlRequestCallback;
 
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProviderService.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProviderService.java
index 43cde10..a186fee 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProviderService.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteProviderService.java
@@ -29,12 +29,14 @@
         .CLIENT_MSG_RELEASE_ROUTE_CONTROLLER;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_ROUTE_CONTROL_REQUEST;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.CLIENT_MSG_SELECT_ROUTE;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .CLIENT_MSG_SELECT_ROUTE;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_SET_DISCOVERY_REQUEST;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_SET_ROUTE_VOLUME;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.CLIENT_MSG_UNREGISTER;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .CLIENT_MSG_UNREGISTER;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_UNSELECT_ROUTE;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
@@ -51,9 +53,12 @@
         .SERVICE_MSG_GENERIC_FAILURE;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .SERVICE_MSG_GENERIC_SUCCESS;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.SERVICE_MSG_REGISTERED;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.SERVICE_VERSION_CURRENT;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.isValidRemoteMessenger;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .SERVICE_MSG_REGISTERED;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .SERVICE_VERSION_CURRENT;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .isValidRemoteMessenger;
 
 import android.app.Service;
 import android.content.Intent;
@@ -65,11 +70,12 @@
 import android.os.Message;
 import android.os.Messenger;
 import android.os.RemoteException;
-import android.support.annotation.VisibleForTesting;
-import android.support.v4.util.ObjectsCompat;
 import android.util.Log;
 import android.util.SparseArray;
 
+import androidx.annotation.VisibleForTesting;
+import androidx.core.util.ObjectsCompat;
+
 import java.lang.ref.WeakReference;
 import java.util.ArrayList;
 
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteSelector.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteSelector.java
index 5669b19..f20dcc0 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteSelector.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouteSelector.java
@@ -17,8 +17,9 @@
 
 import android.content.IntentFilter;
 import android.os.Bundle;
-import android.support.annotation.NonNull;
-import android.support.annotation.Nullable;
+
+import androidx.annotation.NonNull;
+import androidx.annotation.Nullable;
 
 import java.util.ArrayList;
 import java.util.Arrays;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouter.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouter.java
index db0052e..4b56b19 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouter.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaRouter.java
@@ -33,15 +33,16 @@
 import android.os.Handler;
 import android.os.Looper;
 import android.os.Message;
-import android.support.v4.app.ActivityManagerCompat;
-import android.support.v4.hardware.display.DisplayManagerCompat;
-import android.support.v4.media.VolumeProviderCompat;
 import android.support.v4.media.session.MediaSessionCompat;
-import android.support.v4.util.Pair;
 import android.text.TextUtils;
 import android.util.Log;
 import android.view.Display;
 
+import androidx.core.app.ActivityManagerCompat;
+import androidx.core.hardware.display.DisplayManagerCompat;
+import androidx.core.util.Pair;
+import androidx.media.VolumeProviderCompat;
+
 import com.android.support.mediarouter.media.MediaRouteProvider.ProviderMetadata;
 import com.android.support.mediarouter.media.MediaRouteProvider.RouteController;
 
@@ -81,13 +82,13 @@
     static final boolean DEBUG = Log.isLoggable(TAG, Log.DEBUG);
 
     /**
-     * Passed to {@link android.support.v7.media.MediaRouteProvider.RouteController#onUnselect(int)}
+     * Passed to {@link androidx.mediarouter.media.MediaRouteProvider.RouteController#onUnselect(int)}
      * and {@link Callback#onRouteUnselected(MediaRouter, RouteInfo, int)} when the reason the route
      * was unselected is unknown.
      */
     public static final int UNSELECT_REASON_UNKNOWN = 0;
     /**
-     * Passed to {@link android.support.v7.media.MediaRouteProvider.RouteController#onUnselect(int)}
+     * Passed to {@link androidx.mediarouter.media.MediaRouteProvider.RouteController#onUnselect(int)}
      * and {@link Callback#onRouteUnselected(MediaRouter, RouteInfo, int)} when the user pressed
      * the disconnect button to disconnect and keep playing.
      * <p>
@@ -96,13 +97,13 @@
      */
     public static final int UNSELECT_REASON_DISCONNECTED = 1;
     /**
-     * Passed to {@link android.support.v7.media.MediaRouteProvider.RouteController#onUnselect(int)}
+     * Passed to {@link androidx.mediarouter.media.MediaRouteProvider.RouteController#onUnselect(int)}
      * and {@link Callback#onRouteUnselected(MediaRouter, RouteInfo, int)} when the user pressed
      * the stop casting button.
      */
     public static final int UNSELECT_REASON_STOPPED = 2;
     /**
-     * Passed to {@link android.support.v7.media.MediaRouteProvider.RouteController#onUnselect(int)}
+     * Passed to {@link androidx.mediarouter.media.MediaRouteProvider.RouteController#onUnselect(int)}
      * and {@link Callback#onRouteUnselected(MediaRouter, RouteInfo, int)} when the user selected
      * a different route.
      */
@@ -174,7 +175,7 @@
      * Applications should typically add a callback using this flag in the
      * {@link android.app.Activity activity's} {@link android.app.Activity#onStart onStart}
      * method and remove it in the {@link android.app.Activity#onStop onStop} method.
-     * The {@link android.support.v7.app.MediaRouteDiscoveryFragment} fragment may
+     * The {@link androidx.mediarouter.app.MediaRouteDiscoveryFragment} fragment may
      * also be used for this purpose.
      * </p><p class="note">
      * On {@link ActivityManager#isLowRamDevice low-RAM devices} this flag
@@ -182,7 +183,7 @@
      * {@link #addCallback(MediaRouteSelector, Callback, int) addCallback} for details.
      * </p>
      *
-     * @see android.support.v7.app.MediaRouteDiscoveryFragment
+     * @see androidx.mediarouter.app.MediaRouteDiscoveryFragment
      */
     public static final int CALLBACK_FLAG_REQUEST_DISCOVERY = 1 << 2;
 
@@ -197,7 +198,7 @@
      * {@link #addCallback(MediaRouteSelector, Callback, int) addCallback} for details.
      * </p>
      *
-     * @see android.support.v7.app.MediaRouteDiscoveryFragment
+     * @see androidx.mediarouter.app.MediaRouteDiscoveryFragment
      */
     public static final int CALLBACK_FLAG_FORCE_DISCOVERY = 1 << 3;
 
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaSessionStatus.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaSessionStatus.java
index 3206596..0e7514c 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaSessionStatus.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/MediaSessionStatus.java
@@ -19,7 +19,8 @@
 import android.app.PendingIntent;
 import android.os.Bundle;
 import android.os.SystemClock;
-import android.support.v4.util.TimeUtils;
+
+import androidx.core.util.TimeUtils;
 
 /**
  * Describes the playback status of a media session.
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/RegisteredMediaRouteProvider.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/RegisteredMediaRouteProvider.java
index 98e4e28..eacf1c8 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/RegisteredMediaRouteProvider.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/RegisteredMediaRouteProvider.java
@@ -29,17 +29,20 @@
         .CLIENT_MSG_RELEASE_ROUTE_CONTROLLER;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_ROUTE_CONTROL_REQUEST;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.CLIENT_MSG_SELECT_ROUTE;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .CLIENT_MSG_SELECT_ROUTE;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_SET_DISCOVERY_REQUEST;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_SET_ROUTE_VOLUME;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.CLIENT_MSG_UNREGISTER;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .CLIENT_MSG_UNREGISTER;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_UNSELECT_ROUTE;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .CLIENT_MSG_UPDATE_ROUTE_VOLUME;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.CLIENT_VERSION_CURRENT;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .CLIENT_VERSION_CURRENT;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.SERVICE_DATA_ERROR;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .SERVICE_MSG_CONTROL_REQUEST_FAILED;
@@ -51,9 +54,11 @@
         .SERVICE_MSG_GENERIC_FAILURE;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
         .SERVICE_MSG_GENERIC_SUCCESS;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.SERVICE_MSG_REGISTERED;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .SERVICE_MSG_REGISTERED;
 import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.SERVICE_VERSION_1;
-import static com.android.support.mediarouter.media.MediaRouteProviderProtocol.isValidRemoteMessenger;
+import static com.android.support.mediarouter.media.MediaRouteProviderProtocol
+        .isValidRemoteMessenger;
 
 import android.annotation.NonNull;
 import android.content.ComponentName;
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/RemoteControlClientCompat.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/RemoteControlClientCompat.java
index 826449b..65c5518 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/RemoteControlClientCompat.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/RemoteControlClientCompat.java
@@ -18,7 +18,6 @@
 import android.content.Context;
 import android.media.AudioManager;
 import android.os.Build;
-import android.support.annotation.RequiresApi;
 
 import java.lang.ref.WeakReference;
 
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/RemotePlaybackClient.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/RemotePlaybackClient.java
index f6e1497..e76564e 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/RemotePlaybackClient.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/RemotePlaybackClient.java
@@ -22,9 +22,10 @@
 import android.content.IntentFilter;
 import android.net.Uri;
 import android.os.Bundle;
-import android.support.v4.util.ObjectsCompat;
 import android.util.Log;
 
+import androidx.core.util.ObjectsCompat;
+
 /**
  * A helper class for playing media on remote routes using the remote playback protocol
  * defined by {@link MediaControlIntent}.
@@ -867,11 +868,11 @@
 
     private final class ActionReceiver extends BroadcastReceiver {
         public static final String ACTION_ITEM_STATUS_CHANGED =
-                "android.support.v7.media.actions.ACTION_ITEM_STATUS_CHANGED";
+                "androidx.mediarouter.media.actions.ACTION_ITEM_STATUS_CHANGED";
         public static final String ACTION_SESSION_STATUS_CHANGED =
-                "android.support.v7.media.actions.ACTION_SESSION_STATUS_CHANGED";
+                "androidx.mediarouter.media.actions.ACTION_SESSION_STATUS_CHANGED";
         public static final String ACTION_MESSAGE_RECEIVED =
-                "android.support.v7.media.actions.ACTION_MESSAGE_RECEIVED";
+                "androidx.mediarouter.media.actions.ACTION_MESSAGE_RECEIVED";
 
         ActionReceiver() {
         }
diff --git a/packages/MediaComponents/src/com/android/support/mediarouter/media/SystemMediaRouteProvider.java b/packages/MediaComponents/src/com/android/support/mediarouter/media/SystemMediaRouteProvider.java
index a38491f..53901a4 100644
--- a/packages/MediaComponents/src/com/android/support/mediarouter/media/SystemMediaRouteProvider.java
+++ b/packages/MediaComponents/src/com/android/support/mediarouter/media/SystemMediaRouteProvider.java
@@ -24,7 +24,6 @@
 import android.content.res.Resources;
 import android.media.AudioManager;
 import android.os.Build;
-import android.support.annotation.RequiresApi;
 import android.view.Display;
 
 import com.android.media.update.ApiHelper;
diff --git a/packages/MediaComponents/src/com/android/widget/MediaControlView2Impl.java b/packages/MediaComponents/src/com/android/widget/MediaControlView2Impl.java
index 3aff150..ad85af4 100644
--- a/packages/MediaComponents/src/com/android/widget/MediaControlView2Impl.java
+++ b/packages/MediaComponents/src/com/android/widget/MediaControlView2Impl.java
@@ -20,15 +20,13 @@
 import android.content.res.Resources;
 import android.graphics.Point;
 import android.media.MediaMetadata;
+import android.media.SessionToken2;
 import android.media.session.MediaController;
 import android.media.session.PlaybackState;
-import android.media.SessionToken2;
 import android.media.update.MediaControlView2Provider;
 import android.media.update.ViewGroupProvider;
 import android.os.Bundle;
-import android.support.annotation.Nullable;
 import android.util.AttributeSet;
-import android.util.Log;
 import android.view.Gravity;
 import android.view.MotionEvent;
 import android.view.View;
@@ -36,27 +34,28 @@
 import android.view.WindowManager;
 import android.widget.AdapterView;
 import android.widget.BaseAdapter;
-import android.widget.Button;
 import android.widget.ImageButton;
 import android.widget.ImageView;
 import android.widget.LinearLayout;
 import android.widget.ListView;
 import android.widget.MediaControlView2;
-import android.widget.ProgressBar;
 import android.widget.PopupWindow;
+import android.widget.ProgressBar;
 import android.widget.RelativeLayout;
 import android.widget.SeekBar;
 import android.widget.SeekBar.OnSeekBarChangeListener;
 import android.widget.TextView;
 
+import androidx.annotation.Nullable;
+
 import com.android.media.update.ApiHelper;
 import com.android.media.update.R;
 import com.android.support.mediarouter.app.MediaRouteButton;
-import com.android.support.mediarouter.media.MediaRouter;
 import com.android.support.mediarouter.media.MediaRouteSelector;
+import com.android.support.mediarouter.media.MediaRouter;
 
-import java.util.Arrays;
 import java.util.ArrayList;
+import java.util.Arrays;
 import java.util.Formatter;
 import java.util.List;
 import java.util.Locale;
diff --git a/packages/MediaComponents/src/com/android/widget/SubtitleView.java b/packages/MediaComponents/src/com/android/widget/SubtitleView.java
index 67b2cd1..db0ae33 100644
--- a/packages/MediaComponents/src/com/android/widget/SubtitleView.java
+++ b/packages/MediaComponents/src/com/android/widget/SubtitleView.java
@@ -19,10 +19,11 @@
 import android.content.Context;
 import android.graphics.Canvas;
 import android.os.Looper;
-import android.support.annotation.Nullable;
 import android.util.AttributeSet;
 import android.widget.FrameLayout;
 
+import androidx.annotation.Nullable;
+
 import com.android.media.subtitle.SubtitleController.Anchor;
 import com.android.media.subtitle.SubtitleTrack.RenderingWidget;
 
diff --git a/packages/MediaComponents/src/com/android/widget/VideoSurfaceView.java b/packages/MediaComponents/src/com/android/widget/VideoSurfaceView.java
index fc92e85..c9869c0 100644
--- a/packages/MediaComponents/src/com/android/widget/VideoSurfaceView.java
+++ b/packages/MediaComponents/src/com/android/widget/VideoSurfaceView.java
@@ -16,17 +16,18 @@
 
 package com.android.widget;
 
+import static android.widget.VideoView2.VIEW_TYPE_SURFACEVIEW;
+
 import android.content.Context;
 import android.graphics.Rect;
 import android.media.MediaPlayer2;
-import android.support.annotation.NonNull;
 import android.util.AttributeSet;
 import android.util.Log;
 import android.view.SurfaceHolder;
 import android.view.SurfaceView;
 import android.view.View;
 
-import static android.widget.VideoView2.VIEW_TYPE_SURFACEVIEW;
+import androidx.annotation.NonNull;
 
 class VideoSurfaceView extends SurfaceView implements VideoViewInterface, SurfaceHolder.Callback {
     private static final String TAG = "VideoSurfaceView";
diff --git a/packages/MediaComponents/src/com/android/widget/VideoTextureView.java b/packages/MediaComponents/src/com/android/widget/VideoTextureView.java
index 024a3aa..40fb046 100644
--- a/packages/MediaComponents/src/com/android/widget/VideoTextureView.java
+++ b/packages/MediaComponents/src/com/android/widget/VideoTextureView.java
@@ -16,18 +16,19 @@
 
 package com.android.widget;
 
+import static android.widget.VideoView2.VIEW_TYPE_TEXTUREVIEW;
+
 import android.content.Context;
 import android.graphics.SurfaceTexture;
 import android.media.MediaPlayer2;
-import android.support.annotation.NonNull;
-import android.support.annotation.RequiresApi;
 import android.util.AttributeSet;
 import android.util.Log;
 import android.view.Surface;
 import android.view.TextureView;
 import android.view.View;
 
-import static android.widget.VideoView2.VIEW_TYPE_TEXTUREVIEW;
+import androidx.annotation.NonNull;
+import androidx.annotation.RequiresApi;
 
 @RequiresApi(26)
 class VideoTextureView extends TextureView
diff --git a/packages/MediaComponents/src/com/android/widget/VideoView2Impl.java b/packages/MediaComponents/src/com/android/widget/VideoView2Impl.java
index 97279d6..ffb145a 100644
--- a/packages/MediaComponents/src/com/android/widget/VideoView2Impl.java
+++ b/packages/MediaComponents/src/com/android/widget/VideoView2Impl.java
@@ -28,30 +28,29 @@
 import android.media.AudioFocusRequest;
 import android.media.AudioManager;
 import android.media.DataSourceDesc;
+import android.media.MediaItem2;
 import android.media.MediaMetadata;
+import android.media.MediaMetadata2;
+import android.media.MediaMetadataRetriever;
 import android.media.MediaPlayer2;
 import android.media.MediaPlayer2.MediaPlayer2EventCallback;
 import android.media.MediaPlayer2.OnSubtitleDataListener;
 import android.media.MediaPlayer2Impl;
-import android.media.SubtitleData;
-import android.media.MediaItem2;
-import android.media.MediaMetadata2;
-import android.media.MediaMetadataRetriever;
 import android.media.Metadata;
 import android.media.PlaybackParams;
+import android.media.SessionToken2;
+import android.media.SubtitleData;
 import android.media.TimedText;
 import android.media.session.MediaController;
 import android.media.session.MediaController.PlaybackInfo;
 import android.media.session.MediaSession;
 import android.media.session.PlaybackState;
-import android.media.SessionToken2;
 import android.media.update.VideoView2Provider;
 import android.media.update.ViewGroupProvider;
 import android.net.Uri;
 import android.os.AsyncTask;
 import android.os.Bundle;
 import android.os.ResultReceiver;
-import android.support.annotation.Nullable;
 import android.util.AttributeSet;
 import android.util.DisplayMetrics;
 import android.util.Log;
@@ -66,6 +65,8 @@
 import android.widget.TextView;
 import android.widget.VideoView2;
 
+import androidx.annotation.Nullable;
+
 import com.android.internal.graphics.palette.Palette;
 import com.android.media.RoutePlayer;
 import com.android.media.subtitle.ClosedCaptionRenderer;
@@ -73,10 +74,10 @@
 import com.android.media.subtitle.SubtitleTrack;
 import com.android.media.update.ApiHelper;
 import com.android.media.update.R;
-import com.android.support.mediarouter.media.MediaItemStatus;
 import com.android.support.mediarouter.media.MediaControlIntent;
-import com.android.support.mediarouter.media.MediaRouter;
+import com.android.support.mediarouter.media.MediaItemStatus;
 import com.android.support.mediarouter.media.MediaRouteSelector;
+import com.android.support.mediarouter.media.MediaRouter;
 
 import java.util.ArrayList;
 import java.util.List;
diff --git a/services/audioflinger/Android.mk b/services/audioflinger/Android.mk
index 7419e64..c0aa477 100644
--- a/services/audioflinger/Android.mk
+++ b/services/audioflinger/Android.mk
@@ -2,24 +2,6 @@
 
 include $(CLEAR_VARS)
 
-LOCAL_SRC_FILES := \
-    ServiceUtilities.cpp
-
-# FIXME Move this library to frameworks/native
-LOCAL_MODULE := libserviceutility
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    liblog \
-    libbinder
-
-LOCAL_CFLAGS := -Wall -Werror
-
-include $(BUILD_SHARED_LIBRARY)
-
-include $(CLEAR_VARS)
-
 LOCAL_SRC_FILES:=               \
     AudioFlinger.cpp            \
     Threads.cpp                 \
@@ -31,7 +13,8 @@
     PatchPanel.cpp              \
     StateQueue.cpp              \
     BufLog.cpp                  \
-    TypedLogger.cpp
+    TypedLogger.cpp             \
+    NBAIO_Tee.cpp               \
 
 LOCAL_C_INCLUDES := \
     frameworks/av/services/audiopolicy \
@@ -53,13 +36,13 @@
     libnbaio \
     libnblog \
     libpowermanager \
-    libserviceutility \
     libmediautils \
     libmemunreachable \
     libmedia_helper
 
 LOCAL_STATIC_LIBRARIES := \
     libcpustats \
+    libsndfile \
 
 LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
 
@@ -82,6 +65,7 @@
 LOCAL_CFLAGS += -fvisibility=hidden
 
 LOCAL_CFLAGS += -Werror -Wall
+LOCAL_SANITIZE := integer_overflow
 
 include $(BUILD_SHARED_LIBRARY)
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index bdd39c6..53a4ce9 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -28,7 +28,6 @@
 
 #include <binder/IPCThreadState.h>
 #include <binder/IServiceManager.h>
-#include <cutils/multiuser.h>
 #include <utils/Log.h>
 #include <utils/Trace.h>
 #include <binder/Parcel.h>
@@ -47,7 +46,7 @@
 #include <system/audio.h>
 
 #include "AudioFlinger.h"
-#include "ServiceUtilities.h"
+#include "NBAIO_Tee.h"
 
 #include <media/AudioResamplerPublic.h>
 
@@ -56,7 +55,6 @@
 #include <system/audio_effects/effect_aec.h>
 
 #include <audio_utils/primitives.h>
-#include <audio_utils/string.h>
 
 #include <powermanager/PowerManager.h>
 
@@ -66,6 +64,7 @@
 #include <media/nbaio/PipeReader.h>
 #include <media/AudioParameter.h>
 #include <mediautils/BatteryNotifier.h>
+#include <mediautils/ServiceUtilities.h>
 #include <private/android_filesystem_config.h>
 
 //#define BUFLOG_NDEBUG 0
@@ -100,17 +99,6 @@
 
 uint32_t AudioFlinger::mScreenState;
 
-
-#ifdef TEE_SINK
-bool AudioFlinger::mTeeSinkInputEnabled = false;
-bool AudioFlinger::mTeeSinkOutputEnabled = false;
-bool AudioFlinger::mTeeSinkTrackEnabled = false;
-
-size_t AudioFlinger::mTeeSinkInputFrames = kTeeSinkInputFramesDefault;
-size_t AudioFlinger::mTeeSinkOutputFrames = kTeeSinkOutputFramesDefault;
-size_t AudioFlinger::mTeeSinkTrackFrames = kTeeSinkTrackFramesDefault;
-#endif
-
 // In order to avoid invalidating offloaded tracks each time a Visualizer is turned on and off
 // we define a minimum time during which a global effect is considered enabled.
 static const nsecs_t kMinGlobalEffectEnabletimeNs = seconds(7200);
@@ -160,6 +148,7 @@
       mTotalMemory(0),
       mClientSharedHeapSize(kMinimumClientSharedHeapSizeBytes),
       mGlobalEffectEnableTime(0),
+      mPatchPanel(this),
       mSystemReady(false)
 {
     // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
@@ -168,7 +157,6 @@
         mNextUniqueIds[use] = AUDIO_UNIQUE_ID_USE_MAX;
     }
 
-    getpid_cached = getpid();
     const bool doLog = property_get_bool("ro.test_harness", false);
     if (doLog) {
         mLogMemoryDealer = new MemoryDealer(kLogMemorySize, "LogWriters",
@@ -185,27 +173,6 @@
     mEffectsFactoryHal = EffectsFactoryHalInterface::create();
 
     mMediaLogNotifier->run("MediaLogNotifier");
-
-#ifdef TEE_SINK
-    char value[PROPERTY_VALUE_MAX];
-    (void) property_get("ro.debuggable", value, "0");
-    int debuggable = atoi(value);
-    int teeEnabled = 0;
-    if (debuggable) {
-        (void) property_get("af.tee", value, "0");
-        teeEnabled = atoi(value);
-    }
-    // FIXME symbolic constants here
-    if (teeEnabled & 1) {
-        mTeeSinkInputEnabled = true;
-    }
-    if (teeEnabled & 2) {
-        mTeeSinkOutputEnabled = true;
-    }
-    if (teeEnabled & 4) {
-        mTeeSinkTrackEnabled = true;
-    }
-#endif
 }
 
 void AudioFlinger::onFirstRef()
@@ -226,8 +193,6 @@
         }
     }
 
-    mPatchPanel = new PatchPanel(this);
-
     mMode = AUDIO_MODE_NORMAL;
 
     gAudioFlinger = this;
@@ -534,12 +499,7 @@
             dev->dump(fd);
         }
 
-#ifdef TEE_SINK
-        // dump the serially shared record tee sink
-        if (mRecordTeeSource != 0) {
-            dumpTee(fd, mRecordTeeSource, AUDIO_IO_HANDLE_NONE, 'C');
-        }
-#endif
+        mPatchPanel.dump(fd);
 
         BUFLOG_RESET;
 
@@ -547,6 +507,10 @@
             mLock.unlock();
         }
 
+#ifdef TEE_SINK
+        // NBAIO_Tee dump is safe to call outside of AF lock.
+        NBAIO_Tee::dumpAll(fd, "_DUMP");
+#endif
         // append a copy of media.log here by forwarding fd to it, but don't attempt
         // to lookup the service if it's not running, as it will block for a second
         if (sMediaLogServiceAsBinder != 0) {
@@ -666,7 +630,7 @@
     bool updatePid = (input.clientInfo.clientPid == -1);
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
     uid_t clientUid = input.clientInfo.clientUid;
-    if (!isTrustedCallingUid(callingUid)) {
+    if (!isAudioServerOrMediaServerUid(callingUid)) {
         ALOGW_IF(clientUid != callingUid,
                 "%s uid %d tried to pass itself off as %d",
                 __FUNCTION__, callingUid, clientUid);
@@ -1078,9 +1042,9 @@
         ALOGW("checkStreamType() invalid stream %d", stream);
         return BAD_VALUE;
     }
-    pid_t caller = IPCThreadState::self()->getCallingPid();
-    if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && caller != getpid_cached) {
-        ALOGW("checkStreamType() pid %d cannot use internal stream type %d", caller, stream);
+    const uid_t callerUid = IPCThreadState::self()->getCallingUid();
+    if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT && !isAudioServerUid(callerUid)) {
+        ALOGW("checkStreamType() uid %d cannot use internal stream type %d", callerUid, stream);
         return PERMISSION_DENIED;
     }
 
@@ -1200,9 +1164,8 @@
         String8(AudioParameter::keyStreamSupportedSamplingRates),
     };
 
-    // multiuser friendly app ID check for requests coming from audioserver
-    if (multiuser_get_app_id(callingUid) == AID_AUDIOSERVER) {
-        return;
+    if (isAudioServerUid(callingUid)) {
+        return; // no need to filter if audioserver.
     }
 
     AudioParameter param = AudioParameter(keyValuePairs);
@@ -1636,7 +1599,7 @@
     bool updatePid = (input.clientInfo.clientPid == -1);
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
     uid_t clientUid = input.clientInfo.clientUid;
-    if (!isTrustedCallingUid(callingUid)) {
+    if (!isAudioServerOrMediaServerUid(callingUid)) {
         ALOGW_IF(clientUid != callingUid,
                 "%s uid %d tried to pass itself off as %d",
                 __FUNCTION__, callingUid, clientUid);
@@ -1885,7 +1848,7 @@
 status_t AudioFlinger::setLowRamDevice(bool isLowRamDevice, int64_t totalMemory)
 {
     uid_t uid = IPCThreadState::self()->getCallingUid();
-    if (uid != AID_SYSTEM) {
+    if (!isAudioServerOrSystemServerUid(uid)) {
         return PERMISSION_DENIED;
     }
     Mutex::Autolock _l(mLock);
@@ -1930,6 +1893,28 @@
     return mClientSharedHeapSize;
 }
 
+status_t AudioFlinger::setAudioPortConfig(const struct audio_port_config *config)
+{
+    ALOGV(__func__);
+
+    audio_module_handle_t module;
+    if (config->type == AUDIO_PORT_TYPE_DEVICE) {
+        module = config->ext.device.hw_module;
+    } else {
+        module = config->ext.mix.hw_module;
+    }
+
+    Mutex::Autolock _l(mLock);
+    ssize_t index = mAudioHwDevs.indexOfKey(module);
+    if (index < 0) {
+        ALOGW("%s() bad hw module %d", __func__, module);
+        return BAD_VALUE;
+    }
+
+    AudioHwDevice *audioHwDevice = mAudioHwDevs.valueAt(index);
+    return audioHwDevice->hwDevice()->setAudioPortConfig(config);
+}
+
 audio_hw_sync_t AudioFlinger::getAudioHwSyncForSession(audio_session_t sessionId)
 {
     Mutex::Autolock _l(mLock);
@@ -2274,7 +2259,7 @@
     delete out;
 }
 
-void AudioFlinger::closeOutputInternal_l(const sp<PlaybackThread>& thread)
+void AudioFlinger::closeThreadInternal_l(const sp<PlaybackThread>& thread)
 {
     mPlaybackThreads.removeItem(thread->mId);
     thread->exit();
@@ -2350,6 +2335,9 @@
         return 0;
     }
 
+    // Some flags are specific to framework and must not leak to the HAL.
+    flags = static_cast<audio_input_flags_t>(flags & ~AUDIO_INPUT_FRAMEWORK_FLAGS);
+
     // Audio Policy can request a specific handle for hardware hotword.
     // The goal here is not to re-open an already opened input.
     // It is to use a pre-assigned I/O handle.
@@ -2407,55 +2395,6 @@
                     thread.get());
             return thread;
         } else {
-#ifdef TEE_SINK
-            // Try to re-use most recently used Pipe to archive a copy of input for dumpsys,
-            // or (re-)create if current Pipe is idle and does not match the new format
-            sp<NBAIO_Sink> teeSink;
-            enum {
-                TEE_SINK_NO,    // don't copy input
-                TEE_SINK_NEW,   // copy input using a new pipe
-                TEE_SINK_OLD,   // copy input using an existing pipe
-            } kind;
-            NBAIO_Format format = Format_from_SR_C(halconfig.sample_rate,
-                    audio_channel_count_from_in_mask(halconfig.channel_mask), halconfig.format);
-            if (!mTeeSinkInputEnabled) {
-                kind = TEE_SINK_NO;
-            } else if (!Format_isValid(format)) {
-                kind = TEE_SINK_NO;
-            } else if (mRecordTeeSink == 0) {
-                kind = TEE_SINK_NEW;
-            } else if (mRecordTeeSink->getStrongCount() != 1) {
-                kind = TEE_SINK_NO;
-            } else if (Format_isEqual(format, mRecordTeeSink->format())) {
-                kind = TEE_SINK_OLD;
-            } else {
-                kind = TEE_SINK_NEW;
-            }
-            switch (kind) {
-            case TEE_SINK_NEW: {
-                Pipe *pipe = new Pipe(mTeeSinkInputFrames, format);
-                size_t numCounterOffers = 0;
-                const NBAIO_Format offers[1] = {format};
-                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
-                ALOG_ASSERT(index == 0);
-                PipeReader *pipeReader = new PipeReader(*pipe);
-                numCounterOffers = 0;
-                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
-                ALOG_ASSERT(index == 0);
-                mRecordTeeSink = pipe;
-                mRecordTeeSource = pipeReader;
-                teeSink = pipe;
-                }
-                break;
-            case TEE_SINK_OLD:
-                teeSink = mRecordTeeSink;
-                break;
-            case TEE_SINK_NO:
-            default:
-                break;
-            }
-#endif
-
             // Start record thread
             // RecordThread requires both input and output device indication to forward to audio
             // pre processing modules
@@ -2465,9 +2404,6 @@
                                       primaryOutputDevice_l(),
                                       devices,
                                       mSystemReady
-#ifdef TEE_SINK
-                                      , teeSink
-#endif
                                       );
             mRecordThreads.add(*input, thread);
             ALOGV("openInput_l() created record thread: ID %d thread %p", *input, thread.get());
@@ -2567,7 +2503,7 @@
     delete in;
 }
 
-void AudioFlinger::closeInputInternal_l(const sp<RecordThread>& thread)
+void AudioFlinger::closeThreadInternal_l(const sp<RecordThread>& thread)
 {
     mRecordThreads.removeItem(thread->mId);
     closeInputFinish(thread);
@@ -2605,7 +2541,8 @@
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
     ALOGV("acquiring %d from %d, for %d", audioSession, caller, pid);
-    if (pid != -1 && (caller == getpid_cached)) {
+    const uid_t callerUid = IPCThreadState::self()->getCallingUid();
+    if (pid != -1 && isAudioServerUid(callerUid)) { // check must match releaseAudioSessionId()
         caller = pid;
     }
 
@@ -2639,7 +2576,8 @@
     Mutex::Autolock _l(mLock);
     pid_t caller = IPCThreadState::self()->getCallingPid();
     ALOGV("releasing %d from %d for %d", audioSession, caller, pid);
-    if (pid != -1 && (caller == getpid_cached)) {
+    const uid_t callerUid = IPCThreadState::self()->getCallingUid();
+    if (pid != -1 && isAudioServerUid(callerUid)) { // check must match acquireAudioSessionId()
         caller = pid;
     }
     size_t num = mAudioSessionRefs.size();
@@ -2656,9 +2594,10 @@
             return;
         }
     }
-    // If the caller is mediaserver it is likely that the session being released was acquired
+    // If the caller is audioserver it is likely that the session being released was acquired
     // on behalf of a process not in notification clients and we ignore the warning.
-    ALOGW_IF(caller != getpid_cached, "session id %d not found for pid %d", audioSession, caller);
+    ALOGW_IF(!isAudioServerUid(callerUid),
+            "session id %d not found for pid %d", audioSession, caller);
 }
 
 bool AudioFlinger::isSessionAcquired_l(audio_session_t audioSession)
@@ -2966,7 +2905,7 @@
     effect_descriptor_t desc;
 
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    if (pid == -1 || !isTrustedCallingUid(callingUid)) {
+    if (pid == -1 || !isAudioServerOrMediaServerUid(callingUid)) {
         const pid_t callingPid = IPCThreadState::self()->getCallingPid();
         ALOGW_IF(pid != -1 && pid != callingPid,
                  "%s uid %d pid %d tried to pass itself off as pid %d",
@@ -2989,8 +2928,8 @@
     }
 
     // Session AUDIO_SESSION_OUTPUT_STAGE is reserved for output stage effects
-    // that can only be created by audio policy manager (running in same process)
-    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && getpid_cached != pid) {
+    // that can only be created by audio policy manager
+    if (sessionId == AUDIO_SESSION_OUTPUT_STAGE && !isAudioServerUid(callingUid)) {
         lStatus = PERMISSION_DENIED;
         goto Exit;
     }
@@ -3370,136 +3309,6 @@
 }
 
 
-struct Entry {
-#define TEE_MAX_FILENAME 32 // %Y%m%d%H%M%S_%d.wav = 4+2+2+2+2+2+1+1+4+1 = 21
-    char mFileName[TEE_MAX_FILENAME];
-};
-
-int comparEntry(const void *p1, const void *p2)
-{
-    return strcmp(((const Entry *) p1)->mFileName, ((const Entry *) p2)->mFileName);
-}
-
-#ifdef TEE_SINK
-void AudioFlinger::dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id, char suffix)
-{
-    NBAIO_Source *teeSource = source.get();
-    if (teeSource != NULL) {
-        // .wav rotation
-        // There is a benign race condition if 2 threads call this simultaneously.
-        // They would both traverse the directory, but the result would simply be
-        // failures at unlink() which are ignored.  It's also unlikely since
-        // normally dumpsys is only done by bugreport or from the command line.
-        char teePath[PATH_MAX] = "/data/misc/audioserver";
-        size_t teePathLen = strlen(teePath);
-        DIR *dir = opendir(teePath);
-        teePath[teePathLen++] = '/';
-        if (dir != NULL) {
-#define TEE_MAX_SORT 20 // number of entries to sort
-#define TEE_MAX_KEEP 10 // number of entries to keep
-            struct Entry entries[TEE_MAX_SORT];
-            size_t entryCount = 0;
-            while (entryCount < TEE_MAX_SORT) {
-                errno = 0; // clear errno before readdir() to track potential errors.
-                const struct dirent *result = readdir(dir);
-                if (result == nullptr) {
-                    ALOGW_IF(errno != 0, "tee readdir() failure %s", strerror(errno));
-                    break;
-                }
-                // ignore non .wav file entries
-                const size_t nameLen = strlen(result->d_name);
-                if (nameLen <= 4 || nameLen >= TEE_MAX_FILENAME ||
-                        strcmp(&result->d_name[nameLen - 4], ".wav")) {
-                    continue;
-                }
-                (void)audio_utils_strlcpy(entries[entryCount++].mFileName, result->d_name);
-            }
-            (void) closedir(dir);
-            if (entryCount > TEE_MAX_KEEP) {
-                qsort(entries, entryCount, sizeof(Entry), comparEntry);
-                for (size_t i = 0; i < entryCount - TEE_MAX_KEEP; ++i) {
-                    strcpy(&teePath[teePathLen], entries[i].mFileName);
-                    (void) unlink(teePath);
-                }
-            }
-        } else {
-            if (fd >= 0) {
-                dprintf(fd, "unable to rotate tees in %.*s: %s\n", (int) teePathLen, teePath,
-                        strerror(errno));
-            }
-        }
-        char teeTime[16];
-        struct timeval tv;
-        gettimeofday(&tv, NULL);
-        struct tm tm;
-        localtime_r(&tv.tv_sec, &tm);
-        strftime(teeTime, sizeof(teeTime), "%Y%m%d%H%M%S", &tm);
-        snprintf(&teePath[teePathLen], sizeof(teePath) - teePathLen, "%s_%d_%c.wav", teeTime, id,
-                suffix);
-        // if 2 dumpsys are done within 1 second, and rotation didn't work, then discard 2nd
-        int teeFd = open(teePath, O_WRONLY | O_CREAT | O_EXCL | O_NOFOLLOW, S_IRUSR | S_IWUSR);
-        if (teeFd >= 0) {
-            // FIXME use libsndfile
-            char wavHeader[44];
-            memcpy(wavHeader,
-                "RIFF\0\0\0\0WAVEfmt \20\0\0\0\1\0\2\0\104\254\0\0\0\0\0\0\4\0\20\0data\0\0\0\0",
-                sizeof(wavHeader));
-            NBAIO_Format format = teeSource->format();
-            unsigned channelCount = Format_channelCount(format);
-            uint32_t sampleRate = Format_sampleRate(format);
-            size_t frameSize = Format_frameSize(format);
-            wavHeader[22] = channelCount;       // number of channels
-            wavHeader[24] = sampleRate;         // sample rate
-            wavHeader[25] = sampleRate >> 8;
-            wavHeader[32] = frameSize;          // block alignment
-            wavHeader[33] = frameSize >> 8;
-            write(teeFd, wavHeader, sizeof(wavHeader));
-            size_t total = 0;
-            bool firstRead = true;
-#define TEE_SINK_READ 1024                      // frames per I/O operation
-            void *buffer = malloc(TEE_SINK_READ * frameSize);
-            for (;;) {
-                size_t count = TEE_SINK_READ;
-                ssize_t actual = teeSource->read(buffer, count);
-                bool wasFirstRead = firstRead;
-                firstRead = false;
-                if (actual <= 0) {
-                    if (actual == (ssize_t) OVERRUN && wasFirstRead) {
-                        continue;
-                    }
-                    break;
-                }
-                ALOG_ASSERT(actual <= (ssize_t)count);
-                write(teeFd, buffer, actual * frameSize);
-                total += actual;
-            }
-            free(buffer);
-            lseek(teeFd, (off_t) 4, SEEK_SET);
-            uint32_t temp = 44 + total * frameSize - 8;
-            // FIXME not big-endian safe
-            write(teeFd, &temp, sizeof(temp));
-            lseek(teeFd, (off_t) 40, SEEK_SET);
-            temp =  total * frameSize;
-            // FIXME not big-endian safe
-            write(teeFd, &temp, sizeof(temp));
-            close(teeFd);
-            // TODO Should create file with temporary name and then rename to final if non-empty.
-            if (total > 0) {
-                if (fd >= 0) {
-                    dprintf(fd, "tee copied to %s\n", teePath);
-                }
-            } else {
-                unlink(teePath);
-            }
-        } else {
-            if (fd >= 0) {
-                dprintf(fd, "unable to create tee %s: %s\n", teePath, strerror(errno));
-            }
-        }
-    }
-}
-#endif
-
 // ----------------------------------------------------------------------------
 
 status_t AudioFlinger::onTransact(
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 963a87d..0276cad 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -63,6 +63,7 @@
 #include <media/VolumeShaper.h>
 
 #include <audio_utils/SimpleLog.h>
+#include <audio_utils/TimestampVerifier.h>
 
 #include "FastCapture.h"
 #include "FastMixer.h"
@@ -71,6 +72,7 @@
 #include "AudioStreamOut.h"
 #include "SpdifStreamOut.h"
 #include "AudioHwDevice.h"
+#include "NBAIO_Tee.h"
 
 #include <powermanager/IPowerManager.h>
 
@@ -791,44 +793,16 @@
 
     // for use from destructor
     status_t    closeOutput_nonvirtual(audio_io_handle_t output);
-    void        closeOutputInternal_l(const sp<PlaybackThread>& thread);
+    void        closeThreadInternal_l(const sp<PlaybackThread>& thread);
     status_t    closeInput_nonvirtual(audio_io_handle_t input);
-    void        closeInputInternal_l(const sp<RecordThread>& thread);
+    void        closeThreadInternal_l(const sp<RecordThread>& thread);
     void        setAudioHwSyncForSession_l(PlaybackThread *thread, audio_session_t sessionId);
 
     status_t    checkStreamType(audio_stream_type_t stream) const;
 
     void        filterReservedParameters(String8& keyValuePairs, uid_t callingUid);
 
-#ifdef TEE_SINK
-    // all record threads serially share a common tee sink, which is re-created on format change
-    sp<NBAIO_Sink>   mRecordTeeSink;
-    sp<NBAIO_Source> mRecordTeeSource;
-#endif
-
 public:
-
-#ifdef TEE_SINK
-    // tee sink, if enabled by property, allows dumpsys to write most recent audio to .wav file
-    static void dumpTee(int fd, const sp<NBAIO_Source>& source, audio_io_handle_t id, char suffix);
-
-    // whether tee sink is enabled by property
-    static bool mTeeSinkInputEnabled;
-    static bool mTeeSinkOutputEnabled;
-    static bool mTeeSinkTrackEnabled;
-
-    // runtime configured size of each tee sink pipe, in frames
-    static size_t mTeeSinkInputFrames;
-    static size_t mTeeSinkOutputFrames;
-    static size_t mTeeSinkTrackFrames;
-
-    // compile-time default size of tee sink pipes, in frames
-    // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
-    static const size_t kTeeSinkInputFramesDefault = 0x200000;
-    static const size_t kTeeSinkOutputFramesDefault = 0x200000;
-    static const size_t kTeeSinkTrackFramesDefault = 0x200000;
-#endif
-
     // These methods read variables atomically without mLock,
     // though the variables are updated with mLock.
     bool    isLowRamDevice() const { return mIsLowRamDevice; }
@@ -843,7 +817,8 @@
 
     nsecs_t mGlobalEffectEnableTime;  // when a global effect was last enabled
 
-    sp<PatchPanel> mPatchPanel;
+    // protected by mLock
+    PatchPanel mPatchPanel;
     sp<EffectsFactoryHalInterface> mEffectsFactoryHal;
 
     bool        mSystemReady;
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index 2047dfd..786c4af 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -31,9 +31,9 @@
 #include <media/AudioEffect.h>
 #include <media/audiohal/EffectHalInterface.h>
 #include <media/audiohal/EffectsFactoryHalInterface.h>
+#include <mediautils/ServiceUtilities.h>
 
 #include "AudioFlinger.h"
-#include "ServiceUtilities.h"
 
 // ----------------------------------------------------------------------------
 
@@ -1815,7 +1815,7 @@
     bool locked = mCblk != NULL && AudioFlinger::dumpTryLock(mCblk->lock);
 
     snprintf(buffer, size, "\t\t\t%5d    %5d  %3s    %3s  %5u  %5u\n",
-            (mClient == 0) ? getpid_cached : mClient->pid(),
+            (mClient == 0) ? getpid() : mClient->pid(),
             mPriority,
             mHasControl ? "yes" : "no",
             locked ? "yes" : "no",
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 79bb9fe..ad35264 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -47,7 +47,8 @@
 
 /*static*/ const FastMixerState FastMixer::sInitial;
 
-FastMixer::FastMixer() : FastThread("cycle_ms", "load_us"),
+FastMixer::FastMixer(audio_io_handle_t parentIoHandle)
+    : FastThread("cycle_ms", "load_us"),
     // mFastTrackNames
     // mGenerations
     mOutputSink(NULL),
@@ -66,8 +67,11 @@
     mTotalNativeFramesWritten(0),
     // timestamp
     mNativeFramesWrittenButNotPresented(0),   // the = 0 is to silence the compiler
-    mMasterMono(false)
+    mMasterMono(false),
+    mThreadIoHandle(parentIoHandle)
 {
+    (void)mThreadIoHandle; // prevent unused warning, see C++17 [[maybe_unused]]
+
     // FIXME pass sInitial as parameter to base class constructor, and make it static local
     mPrevious = &sInitial;
     mCurrent = &sInitial;
@@ -220,6 +224,10 @@
         previousTrackMask = 0;
         mFastTracksGen = current->mFastTracksGen - 1;
         dumpState->mFrameCount = frameCount;
+#ifdef TEE_SINK
+        mTee.set(mFormat, NBAIO_Tee::TEE_FLAG_OUTPUT_THREAD);
+        mTee.setId(std::string("_") + std::to_string(mThreadIoHandle) + "_F");
+#endif
     } else {
         previousTrackMask = previous->mTrackMask;
     }
@@ -328,13 +336,15 @@
 {
     // TODO: pass an ID parameter to indicate which time series we want to write to in NBLog.cpp
     // Or: pass both of these into a single call with a boolean
+    const FastMixerState * const current = (const FastMixerState *) mCurrent;
+    FastMixerDumpState * const dumpState = (FastMixerDumpState *) mDumpState;
+
     if (mIsWarm) {
         LOG_HIST_TS();
     } else {
+        dumpState->mTimestampVerifier.discontinuity();
         LOG_AUDIO_STATE();
     }
-    const FastMixerState * const current = (const FastMixerState *) mCurrent;
-    FastMixerDumpState * const dumpState = (FastMixerDumpState *) mDumpState;
     const FastMixerState::Command command = mCommand;
     const size_t frameCount = current->mFrameCount;
 
@@ -446,10 +456,9 @@
                     frameCount * Format_channelCount(mFormat));
         }
         // if non-NULL, then duplicate write() to this non-blocking sink
-        NBAIO_Sink* teeSink;
-        if ((teeSink = current->mTeeSink) != NULL) {
-            (void) teeSink->write(buffer, frameCount);
-        }
+#ifdef TEE_SINK
+        mTee.write(buffer, frameCount);
+#endif
         // FIXME write() is non-blocking and lock-free for a properly implemented NBAIO sink,
         //       but this code should be modified to handle both non-blocking and blocking sinks
         dumpState->mWriteSequence++;
@@ -470,35 +479,47 @@
         mAttemptedWrite = true;
         // FIXME count # of writes blocked excessively, CPU usage, etc. for dump
 
-        ExtendedTimestamp timestamp; // local
-        status_t status = mOutputSink->getTimestamp(timestamp);
-        if (status == NO_ERROR) {
-            const int64_t totalNativeFramesPresented =
-                    timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
-            if (totalNativeFramesPresented <= mTotalNativeFramesWritten) {
-                mNativeFramesWrittenButNotPresented =
-                    mTotalNativeFramesWritten - totalNativeFramesPresented;
-                mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
+        if (mIsWarm) {
+            ExtendedTimestamp timestamp; // local
+            status_t status = mOutputSink->getTimestamp(timestamp);
+            if (status == NO_ERROR) {
+                dumpState->mTimestampVerifier.add(
+                        timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
+                        timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
+                        mSampleRate);
+                const int64_t totalNativeFramesPresented =
                         timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
-                mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
-                        timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+                if (totalNativeFramesPresented <= mTotalNativeFramesWritten) {
+                    mNativeFramesWrittenButNotPresented =
+                        mTotalNativeFramesWritten - totalNativeFramesPresented;
+                    mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
+                            timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
+                    mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
+                            timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+                    // We don't compensate for server - kernel time difference and
+                    // only update latency if we have valid info.
+                    dumpState->mLatencyMs =
+                            (double)mNativeFramesWrittenButNotPresented * 1000 / mSampleRate;
+                } else {
+                    // HAL reported that more frames were presented than were written
+                    mNativeFramesWrittenButNotPresented = 0;
+                    status = INVALID_OPERATION;
+                }
             } else {
-                // HAL reported that more frames were presented than were written
-                mNativeFramesWrittenButNotPresented = 0;
-                status = INVALID_OPERATION;
+                dumpState->mTimestampVerifier.error();
             }
-        }
-        if (status == NO_ERROR) {
-            mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] =
-                    mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
-        } else {
-            // fetch server time if we can't get timestamp
-            mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] =
-                    systemTime(SYSTEM_TIME_MONOTONIC);
-            // clear out kernel cached position as this may get rapidly stale
-            // if we never get a new valid timestamp
-            mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 0;
-            mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = -1;
+            if (status == NO_ERROR) {
+                mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] =
+                        mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+            } else {
+                // fetch server time if we can't get timestamp
+                mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] =
+                        systemTime(SYSTEM_TIME_MONOTONIC);
+                // clear out kernel cached position as this may get rapidly stale
+                // if we never get a new valid timestamp
+                mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = 0;
+                mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = -1;
+            }
         }
     }
 }
diff --git a/services/audioflinger/FastMixer.h b/services/audioflinger/FastMixer.h
index 235d23f..1c86d9a 100644
--- a/services/audioflinger/FastMixer.h
+++ b/services/audioflinger/FastMixer.h
@@ -22,6 +22,7 @@
 #include "StateQueue.h"
 #include "FastMixerState.h"
 #include "FastMixerDumpState.h"
+#include "NBAIO_Tee.h"
 
 namespace android {
 
@@ -32,7 +33,9 @@
 class FastMixer : public FastThread {
 
 public:
-            FastMixer();
+    /** FastMixer constructor takes as param the parent MixerThread's io handle (id)
+        for purposes of identification. */
+    explicit FastMixer(audio_io_handle_t threadIoHandle);
     virtual ~FastMixer();
 
             FastMixerStateQueue* sq();
@@ -87,6 +90,11 @@
     // accessed without lock between multiple threads.
     std::atomic_bool mMasterMono;
     std::atomic_int_fast64_t mBoottimeOffset;
+
+    const audio_io_handle_t mThreadIoHandle; // parent thread id for debugging purposes
+#ifdef TEE_SINK
+    NBAIO_Tee       mTee;
+#endif
 };  // class FastMixer
 
 }   // namespace android
diff --git a/services/audioflinger/FastMixerDumpState.cpp b/services/audioflinger/FastMixerDumpState.cpp
index 2e4fb8c..d60643c 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -68,11 +68,12 @@
     dprintf(fd, "  FastMixer command=%s writeSequence=%u framesWritten=%u\n"
                 "            numTracks=%u writeErrors=%u underruns=%u overruns=%u\n"
                 "            sampleRate=%u frameCount=%zu measuredWarmup=%.3g ms, warmupCycles=%u\n"
-                "            mixPeriod=%.2f ms\n",
+                "            mixPeriod=%.2f ms latency=%.2f ms\n",
                 FastMixerState::commandToString(mCommand), mWriteSequence, mFramesWritten,
                 mNumTracks, mWriteErrors, mUnderruns, mOverruns,
                 mSampleRate, mFrameCount, measuredWarmupMs, mWarmupCycles,
-                mixPeriodSec * 1e3);
+                mixPeriodSec * 1e3, mLatencyMs);
+    dprintf(fd, "  FastMixer Timestamp stats: %s\n", mTimestampVerifier.toString().c_str());
 #ifdef FAST_THREAD_STATISTICS
     // find the interval of valid samples
     uint32_t bounds = mBounds;
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 8ef31d1..9b91cbc 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -18,6 +18,7 @@
 #define ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
 
 #include <stdint.h>
+#include <audio_utils/TimestampVerifier.h>
 #include "Configuration.h"
 #include "FastThreadDumpState.h"
 #include "FastMixerState.h"
@@ -66,6 +67,7 @@
 
     void dump(int fd) const;    // should only be called on a stable copy, not the original
 
+    double   mLatencyMs = 0.;   // measured latency, default of 0 if no valid timestamp read.
     uint32_t mWriteSequence;    // incremented before and after each write()
     uint32_t mFramesWritten;    // total number of frames written successfully
     uint32_t mNumTracks;        // total number of active fast tracks
@@ -74,6 +76,9 @@
     size_t   mFrameCount;
     uint32_t mTrackMask;        // mask of active tracks
     FastTrackDump   mTracks[FastMixerState::kMaxFastTracks];
+
+    // For timestamp statistics.
+    TimestampVerifier<int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
 };
 
 }   // android
diff --git a/services/audioflinger/FastMixerState.cpp b/services/audioflinger/FastMixerState.cpp
index 36d8eef..b98842d 100644
--- a/services/audioflinger/FastMixerState.cpp
+++ b/services/audioflinger/FastMixerState.cpp
@@ -35,7 +35,7 @@
 FastMixerState::FastMixerState() : FastThreadState(),
     // mFastTracks
     mFastTracksGen(0), mTrackMask(0), mOutputSink(NULL), mOutputSinkGen(0),
-    mFrameCount(0), mTeeSink(NULL)
+    mFrameCount(0)
 {
     int ok = pthread_once(&sMaxFastTracksOnce, sMaxFastTracksInit);
     if (ok != 0) {
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index 2be1e91..c7fcbd8 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -77,9 +77,6 @@
         WRITE = 0x10,           // write to output sink
         MIX_WRITE = 0x18;       // mix tracks and write to output sink
 
-    // This might be a one-time configuration rather than per-state
-    NBAIO_Sink* mTeeSink;       // if non-NULL, then duplicate write()s to this non-blocking sink
-
     // never returns NULL; asserts if command is invalid
     static const char *commandToString(Command command);
 
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index 6f546c3..968d5aa 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -28,6 +28,7 @@
                             audio_format_t format,
                             audio_channel_mask_t channelMask,
                             audio_session_t sessionId,
+                            bool isOut,
                             uid_t uid,
                             pid_t pid,
                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
@@ -39,8 +40,9 @@
                               audio_session_t triggerSession);
     virtual void        stop();
     virtual bool        isFastTrack() const { return false; }
+            bool        isDirect() const override { return true; }
 
-     static void        appendDumpHeader(String8& result);
+            void        appendDumpHeader(String8& result);
             void        appendDump(String8& result, bool active);
 
                         // protected by MMapThread::mLock
diff --git a/services/audioflinger/NBAIO_Tee.cpp b/services/audioflinger/NBAIO_Tee.cpp
new file mode 100644
index 0000000..53083d5
--- /dev/null
+++ b/services/audioflinger/NBAIO_Tee.cpp
@@ -0,0 +1,517 @@
+/*
+ * Copyright (C) 2018 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 "NBAIO_Tee"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+
+#include <deque>
+#include <dirent.h>
+#include <future>
+#include <list>
+#include <vector>
+
+#include <audio_utils/format.h>
+#include <audio_utils/sndfile.h>
+#include <media/nbaio/PipeReader.h>
+
+#include "Configuration.h"
+#include "NBAIO_Tee.h"
+
+// Enabled with TEE_SINK in Configuration.h
+#ifdef TEE_SINK
+
+namespace android {
+
+/*
+ Tee filenames generated as follows:
+
+ "aftee_Date_ThreadId_C_reason.wav" RecordThread
+ "aftee_Date_ThreadId_M_reason.wav" MixerThread (Normal)
+ "aftee_Date_ThreadId_F_reason.wav" MixerThread (Fast)
+ "aftee_Date_ThreadId_TrackId_R_reason.wav" RecordTrack
+ "aftee_Date_ThreadId_TrackId_TrackName_T_reason.wav" PlaybackTrack
+
+ where Date = YYYYmmdd_HHMMSS_MSEC
+
+ where Reason = [ DTOR | DUMP | REMOVE ]
+
+ Examples:
+  aftee_20180424_153811_038_13_57_2_T_REMOVE.wav
+  aftee_20180424_153811_218_13_57_2_T_REMOVE.wav
+  aftee_20180424_153811_378_13_57_2_T_REMOVE.wav
+  aftee_20180424_153825_147_62_C_DUMP.wav
+  aftee_20180424_153825_148_62_59_R_DUMP.wav
+  aftee_20180424_153825_149_13_F_DUMP.wav
+  aftee_20180424_153842_125_62_59_R_REMOVE.wav
+  aftee_20180424_153842_168_62_C_DTOR.wav
+*/
+
+static constexpr char DEFAULT_PREFIX[] = "aftee_";
+static constexpr char DEFAULT_DIRECTORY[] = "/data/misc/audioserver";
+static constexpr size_t DEFAULT_THREADPOOL_SIZE = 8;
+
+/** AudioFileHandler manages temporary audio wav files with a least recently created
+    retention policy.
+
+    The temporary filenames are systematically generated. A common filename prefix,
+    storage directory, and concurrency pool are passed in on creating the object.
+
+    Temporary files are created by "create", which returns a filename generated by
+
+    prefix + 14 char date + suffix
+
+    TODO Move to audio_utils.
+    TODO Avoid pointing two AudioFileHandlers to the same directory and prefix
+    as we don't have a prefix specific lock file. */
+
+class AudioFileHandler {
+public:
+
+    AudioFileHandler(const std::string &prefix, const std::string &directory, size_t pool)
+        : mThreadPool(pool)
+        , mPrefix(prefix)
+    {
+        (void)setDirectory(directory);
+    }
+
+    /** returns filename of created audio file, else empty string on failure. */
+    std::string create(
+            std::function<ssize_t /* frames_read */
+                        (void * /* buffer */, size_t /* size_in_frames */)> reader,
+            uint32_t sampleRate,
+            uint32_t channelCount,
+            audio_format_t format,
+            const std::string &suffix);
+
+private:
+    /** sets the current directory. this is currently private to avoid confusion
+        when changing while pending operations are occurring (it's okay, but
+        weakly synchronized). */
+    status_t setDirectory(const std::string &directory);
+
+    /** cleans current directory and returns the directory name done. */
+    status_t clean(std::string *dir = nullptr);
+
+    /** creates an audio file from a reader functor passed in. */
+    status_t createInternal(
+            std::function<ssize_t /* frames_read */
+                        (void * /* buffer */, size_t /* size_in_frames */)> reader,
+            uint32_t sampleRate,
+            uint32_t channelCount,
+            audio_format_t format,
+            const std::string &filename);
+
+    static bool isDirectoryValid(const std::string &directory) {
+        return directory.size() > 0 && directory[0] == '/';
+    }
+
+    std::string generateFilename(const std::string &suffix) const {
+        char fileTime[sizeof("YYYYmmdd_HHMMSS_\0")];
+        struct timeval tv;
+        gettimeofday(&tv, NULL);
+        struct tm tm;
+        localtime_r(&tv.tv_sec, &tm);
+        LOG_ALWAYS_FATAL_IF(strftime(fileTime, sizeof(fileTime), "%Y%m%d_%H%M%S_", &tm) == 0,
+            "incorrect fileTime buffer");
+        char msec[4];
+        (void)snprintf(msec, sizeof(msec), "%03d", (int)(tv.tv_usec / 1000));
+        return mPrefix + fileTime + msec + suffix + ".wav";
+    }
+
+    bool isManagedFilename(const char *name) {
+        constexpr size_t FILENAME_LEN_DATE = 4 + 2 + 2 // %Y%m%d%
+            + 1 + 2 + 2 + 2 // _H%M%S
+            + 1 + 3; //_MSEC
+        const size_t prefixLen = mPrefix.size();
+        const size_t nameLen = strlen(name);
+
+        // reject on size, prefix, and .wav
+        if (nameLen < prefixLen + FILENAME_LEN_DATE + 4 /* .wav */
+             || strncmp(name, mPrefix.c_str(), prefixLen) != 0
+             || strcmp(name + nameLen - 4, ".wav") != 0) {
+            return false;
+        }
+
+        // validate date portion
+        const char *date = name + prefixLen;
+        return std::all_of(date, date + 8, isdigit)
+            && date[8] == '_'
+            && std::all_of(date + 9, date + 15, isdigit)
+            && date[15] == '_'
+            && std::all_of(date + 16, date + 19, isdigit);
+    }
+
+    // yet another ThreadPool implementation.
+    class ThreadPool {
+    public:
+        ThreadPool(size_t size)
+            : mThreadPoolSize(size)
+        { }
+
+        /** launches task "name" with associated function "func".
+            if the threadpool is exhausted, it will launch on calling function */
+        status_t launch(const std::string &name, std::function<status_t()> func);
+
+    private:
+        std::mutex mLock;
+        std::list<std::pair<
+                std::string, std::future<status_t>>> mFutures; // GUARDED_BY(mLock)
+
+        const size_t mThreadPoolSize;
+    } mThreadPool;
+
+    const std::string mPrefix;
+    std::mutex mLock;
+    std::string mDirectory;         // GUARDED_BY(mLock)
+    std::deque<std::string> mFiles; // GUARDED_BY(mLock)  sorted list of files by creation time
+
+    static constexpr size_t FRAMES_PER_READ = 1024;
+    static constexpr size_t MAX_FILES_READ = 1024;
+    static constexpr size_t MAX_FILES_KEEP = 32;
+};
+
+/* static */
+void NBAIO_Tee::NBAIO_TeeImpl::dumpTee(
+        int fd, const NBAIO_SinkSource &sinkSource, const std::string &suffix)
+{
+    // Singleton. Constructed thread-safe on first call, never destroyed.
+    static AudioFileHandler audioFileHandler(
+            DEFAULT_PREFIX, DEFAULT_DIRECTORY, DEFAULT_THREADPOOL_SIZE);
+
+    auto &source = sinkSource.second;
+    if (source.get() == nullptr) {
+        return;
+    }
+
+    const NBAIO_Format format = source->format();
+    bool firstRead = true;
+    std::string filename = audioFileHandler.create(
+            // this functor must not hold references to stack
+            [firstRead, sinkSource] (void *buffer, size_t frames) mutable {
+                    auto &source = sinkSource.second;
+                    ssize_t actualRead = source->read(buffer, frames);
+                    if (actualRead == (ssize_t)OVERRUN && firstRead) {
+                        // recheck once
+                        actualRead = source->read(buffer, frames);
+                    }
+                    firstRead = false;
+                    return actualRead;
+                },
+            Format_sampleRate(format),
+            Format_channelCount(format),
+            format.mFormat,
+            suffix);
+
+    if (fd >= 0 && filename.size() > 0) {
+        dprintf(fd, "tee wrote to %s\n", filename.c_str());
+    }
+}
+
+/* static */
+NBAIO_Tee::NBAIO_TeeImpl::NBAIO_SinkSource NBAIO_Tee::NBAIO_TeeImpl::makeSinkSource(
+        const NBAIO_Format &format, size_t frames, bool *enabled)
+{
+    if (Format_isValid(format) && audio_is_linear_pcm(format.mFormat)) {
+        Pipe *pipe = new Pipe(frames, format);
+        size_t numCounterOffers = 0;
+        const NBAIO_Format offers[1] = {format};
+        ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
+        if (index != 0) {
+            ALOGW("pipe failure to negotiate: %zd", index);
+            goto exit;
+        }
+        PipeReader *pipeReader = new PipeReader(*pipe);
+        numCounterOffers = 0;
+        index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
+        if (index != 0) {
+            ALOGW("pipeReader failure to negotiate: %zd", index);
+            goto exit;
+        }
+        if (enabled != nullptr) *enabled = true;
+        return {pipe, pipeReader};
+    }
+exit:
+    if (enabled != nullptr) *enabled = false;
+    return {nullptr, nullptr};
+}
+
+std::string AudioFileHandler::create(
+        std::function<ssize_t /* frames_read */
+                    (void * /* buffer */, size_t /* size_in_frames */)> reader,
+        uint32_t sampleRate,
+        uint32_t channelCount,
+        audio_format_t format,
+        const std::string &suffix)
+{
+    const std::string filename = generateFilename(suffix);
+
+    if (mThreadPool.launch(std::string("create ") + filename,
+            [=]() { return createInternal(reader, sampleRate, channelCount, format, filename); })
+            == NO_ERROR) {
+        return filename;
+    }
+    return "";
+}
+
+status_t AudioFileHandler::setDirectory(const std::string &directory)
+{
+    if (!isDirectoryValid(directory)) return BAD_VALUE;
+
+    // TODO: consider using std::filesystem in C++17
+    DIR *dir = opendir(directory.c_str());
+
+    if (dir == nullptr) {
+        ALOGW("%s: cannot open directory %s", __func__, directory.c_str());
+        return BAD_VALUE;
+    }
+
+    size_t toRemove = 0;
+    decltype(mFiles) files;
+
+    while (files.size() < MAX_FILES_READ) {
+        errno = 0;
+        const struct dirent *result = readdir(dir);
+        if (result == nullptr) {
+            ALOGW_IF(errno != 0, "%s: readdir failure %s", __func__, strerror(errno));
+            break;
+        }
+        // is it a managed filename?
+        if (!isManagedFilename(result->d_name)) {
+            continue;
+        }
+        files.emplace_back(result->d_name);
+    }
+    (void)closedir(dir);
+
+    // OPTIMIZATION: we don't need to stat each file, the filenames names are
+    // already (roughly) ordered by creation date.  we use std::deque instead
+    // of std::set for faster insertion and sorting times.
+
+    if (files.size() > MAX_FILES_KEEP) {
+        // removed files can use a partition (no need to do a full sort).
+        toRemove = files.size() - MAX_FILES_KEEP;
+        std::nth_element(files.begin(), files.begin() + toRemove - 1, files.end());
+    }
+
+    // kept files must be sorted.
+    std::sort(files.begin() + toRemove, files.end());
+
+    {
+        std::lock_guard<std::mutex> _l(mLock);
+
+        mDirectory = directory;
+        mFiles = std::move(files);
+    }
+
+    if (toRemove > 0) { // launch a clean in background.
+        (void)mThreadPool.launch(
+                std::string("cleaning ") + directory, [this]() { return clean(); });
+    }
+    return NO_ERROR;
+}
+
+status_t AudioFileHandler::clean(std::string *directory)
+{
+    std::vector<std::string> filesToRemove;
+    std::string dir;
+    {
+        std::lock_guard<std::mutex> _l(mLock);
+
+        if (!isDirectoryValid(mDirectory)) return NO_INIT;
+
+        dir = mDirectory;
+        if (mFiles.size() > MAX_FILES_KEEP) {
+            size_t toRemove = mFiles.size() - MAX_FILES_KEEP;
+
+            // use move and erase to efficiently transfer std::string
+            std::move(mFiles.begin(),
+                    mFiles.begin() + toRemove,
+                    std::back_inserter(filesToRemove));
+            mFiles.erase(mFiles.begin(), mFiles.begin() + toRemove);
+        }
+    }
+
+    std::string dirp = dir + "/";
+    // remove files outside of lock for better concurrency.
+    for (const auto &file : filesToRemove) {
+        (void)unlink((dirp + file).c_str());
+    }
+
+    // return the directory if requested.
+    if (directory != nullptr) {
+        *directory = dir;
+    }
+    return NO_ERROR;
+}
+
+status_t AudioFileHandler::ThreadPool::launch(
+        const std::string &name, std::function<status_t()> func)
+{
+    if (mThreadPoolSize > 1) {
+        std::lock_guard<std::mutex> _l(mLock);
+        if (mFutures.size() >= mThreadPoolSize) {
+            for (auto it = mFutures.begin(); it != mFutures.end();) {
+                const std::string &filename = it->first;
+                std::future<status_t> &future = it->second;
+                if (!future.valid() ||
+                        future.wait_for(std::chrono::seconds(0)) == std::future_status::ready) {
+                    ALOGV("%s: future %s ready", __func__, filename.c_str());
+                    it = mFutures.erase(it);
+                } else {
+                    ALOGV("%s: future %s not ready", __func__, filename.c_str());
+                    ++it;
+                }
+            }
+        }
+        if (mFutures.size() < mThreadPoolSize) {
+            ALOGV("%s: deferred calling %s", __func__, name.c_str());
+            mFutures.emplace_back(name, std::async(std::launch::async, func));
+            return NO_ERROR;
+        }
+    }
+    ALOGV("%s: immediate calling %s", __func__, name.c_str());
+    return func();
+}
+
+status_t AudioFileHandler::createInternal(
+        std::function<ssize_t /* frames_read */
+                    (void * /* buffer */, size_t /* size_in_frames */)> reader,
+        uint32_t sampleRate,
+        uint32_t channelCount,
+        audio_format_t format,
+        const std::string &filename)
+{
+    // Attempt to choose the best matching file format.
+    // We can choose any sf_format
+    // but writeFormat must be one of 16, 32, float
+    // due to sf_writef compatibility.
+    int sf_format;
+    audio_format_t writeFormat;
+    switch (format) {
+    case AUDIO_FORMAT_PCM_8_BIT:
+    case AUDIO_FORMAT_PCM_16_BIT:
+        sf_format = SF_FORMAT_PCM_16;
+        writeFormat = AUDIO_FORMAT_PCM_16_BIT;
+        ALOGV("%s: %s using PCM_16 for format %#x", __func__, filename.c_str(), format);
+        break;
+    case AUDIO_FORMAT_PCM_8_24_BIT:
+    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+    case AUDIO_FORMAT_PCM_32_BIT:
+        sf_format = SF_FORMAT_PCM_32;
+        writeFormat = AUDIO_FORMAT_PCM_32_BIT;
+        ALOGV("%s: %s using PCM_32 for format %#x", __func__, filename.c_str(), format);
+        break;
+    case AUDIO_FORMAT_PCM_FLOAT:
+        sf_format = SF_FORMAT_FLOAT;
+        writeFormat = AUDIO_FORMAT_PCM_FLOAT;
+        ALOGV("%s: %s using PCM_FLOAT for format %#x", __func__, filename.c_str(), format);
+        break;
+    default:
+        // TODO:
+        // handle audio_has_proportional_frames() formats.
+        // handle compressed formats as single byte files.
+        return BAD_VALUE;
+    }
+
+    std::string directory;
+    status_t status = clean(&directory);
+    if (status != NO_ERROR) return status;
+    std::string dirPrefix = directory + "/";
+
+    const std::string path = dirPrefix + filename;
+
+    /* const */ SF_INFO info = {
+        .frames = 0,
+        .samplerate = (int)sampleRate,
+        .channels = (int)channelCount,
+        .format = SF_FORMAT_WAV | sf_format,
+    };
+    SNDFILE *sf = sf_open(path.c_str(), SFM_WRITE, &info);
+    if (sf == nullptr) {
+        return INVALID_OPERATION;
+    }
+
+    size_t total = 0;
+    void *buffer = malloc(FRAMES_PER_READ * std::max(
+            channelCount * audio_bytes_per_sample(writeFormat), //output framesize
+            channelCount * audio_bytes_per_sample(format))); // input framesize
+    if (buffer == nullptr) {
+        sf_close(sf);
+        return NO_MEMORY;
+    }
+
+    for (;;) {
+        const ssize_t actualRead = reader(buffer, FRAMES_PER_READ);
+        if (actualRead <= 0) {
+            break;
+        }
+
+        // Convert input format to writeFormat as needed.
+        if (format != writeFormat) {
+            memcpy_by_audio_format(
+                    buffer, writeFormat, buffer, format, actualRead * info.channels);
+        }
+
+        ssize_t reallyWritten;
+        switch (writeFormat) {
+        case AUDIO_FORMAT_PCM_16_BIT:
+            reallyWritten = sf_writef_short(sf, (const int16_t *)buffer, actualRead);
+            break;
+        case AUDIO_FORMAT_PCM_32_BIT:
+            reallyWritten = sf_writef_int(sf, (const int32_t *)buffer, actualRead);
+            break;
+        case AUDIO_FORMAT_PCM_FLOAT:
+            reallyWritten = sf_writef_float(sf, (const float *)buffer, actualRead);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("%s: %s writeFormat: %#x", __func__, filename.c_str(), writeFormat);
+            break;
+        }
+
+        if (reallyWritten < 0) {
+            ALOGW("%s: %s write error: %zd", __func__, filename.c_str(), reallyWritten);
+            break;
+        }
+        total += reallyWritten;
+        if (reallyWritten < actualRead) {
+            ALOGW("%s: %s write short count: %zd < %zd",
+                     __func__, filename.c_str(), reallyWritten, actualRead);
+            break;
+        }
+    }
+    sf_close(sf);
+    free(buffer);
+    if (total == 0) {
+        (void)unlink(path.c_str());
+        return NOT_ENOUGH_DATA;
+    }
+
+    // Success: add our name to managed files.
+    {
+        std::lock_guard<std::mutex> _l(mLock);
+        // weak synchronization - only update mFiles if the directory hasn't changed.
+        if (mDirectory == directory) {
+            mFiles.emplace_back(filename);  // add to the end to preserve sort.
+        }
+    }
+    return NO_ERROR; // return full path
+}
+
+} // namespace android
+
+#endif // TEE_SINK
diff --git a/services/audioflinger/NBAIO_Tee.h b/services/audioflinger/NBAIO_Tee.h
new file mode 100644
index 0000000..fed8cc8
--- /dev/null
+++ b/services/audioflinger/NBAIO_Tee.h
@@ -0,0 +1,326 @@
+/*
+ * Copyright (C) 2018 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.
+ */
+
+// Enabled with TEE_SINK in Configuration.h
+#ifndef ANDROID_NBAIO_TEE_H
+#define ANDROID_NBAIO_TEE_H
+
+#ifdef TEE_SINK
+
+#include <atomic>
+#include <mutex>
+#include <set>
+
+#include <cutils/properties.h>
+#include <media/nbaio/NBAIO.h>
+
+namespace android {
+
+/**
+ * The NBAIO_Tee uses the NBAIO Pipe and PipeReader for nonblocking
+ * data collection, for eventual dump to log files.
+ * See https://source.android.com/devices/audio/debugging for how to
+ * enable by ro.debuggable and af.tee properties.
+ *
+ * The write() into the NBAIO_Tee is therefore nonblocking,
+ * but changing NBAIO_Tee formats with set() cannot be done during a write();
+ * usually the caller already implements this mutual exclusion.
+ *
+ * All other calls except set() vs write() may occur at any time.
+ *
+ * dump() disruption is minimized to the caller since system calls are executed
+ * in an asynchronous thread (when possible).
+ *
+ * Currently the NBAIO_Tee is "hardwired" for AudioFlinger support.
+ *
+ * Some AudioFlinger specific notes:
+ *
+ * 1) Tees capture only linear PCM data.
+ * 2) Tees without any data written are considered empty and do not generate
+ *    any output files.
+ * 2) Once a Tee dumps data, it is considered "emptied" and new data
+ *    needs to be written before another Tee file is generated.
+ * 3) Tee file format is
+ *    WAV integer PCM 16 bit for AUDIO_FORMAT_PCM_8_BIT, AUDIO_FORMAT_PCM_16_BIT.
+ *    WAV integer PCM 32 bit for AUDIO_FORMAT_PCM_8_24_BIT, AUDIO_FORMAT_PCM_24_BIT_PACKED
+ *                               AUDIO_FORMAT_PCM_32_BIT.
+ *    WAV float PCM 32 bit for AUDIO_FORMAT_PCM_FLOAT.
+ *
+ * Input_Thread:
+ * 1) Capture buffer is teed when read from the HAL, before resampling for the AudioRecord
+ *    client.
+ *
+ * Output_Thread:
+ * 1) MixerThreads will tee at the FastMixer output (if it has one) or at the
+ *    NormalMixer output (if no FastMixer).
+ * 2) DuplicatingThreads do not tee any mixed data. Apply a tee on the downstream OutputTrack
+ *    or on the upstream playback Tracks.
+ * 3) DirectThreads and OffloadThreads do not tee any data. The upstream track
+ *    (if linear PCM format) may be teed to discover data.
+ * 4) MmapThreads are not supported.
+ *
+ * Tracks:
+ * 1) RecordTracks and playback Tracks tee as data is being written to or
+ *    read from the shared client-server track buffer by the associated Threads.
+ * 2) The mechanism is on the AudioBufferProvider release() so large static Track
+ *    playback may not show any Tee data depending on when it is released.
+ * 3) When a track becomes inactive, the Thread will trigger a dump.
+ */
+
+class NBAIO_Tee {
+public:
+    /* TEE_FLAG is used in set() and must match the flags for the af.tee property
+       given in https://source.android.com/devices/audio/debugging
+    */
+    enum TEE_FLAG {
+        TEE_FLAG_NONE = 0,
+        TEE_FLAG_INPUT_THREAD = (1 << 0),  // treat as a Tee for input (Capture) Threads
+        TEE_FLAG_OUTPUT_THREAD = (1 << 1), // treat as a Tee for output (Playback) Threads
+        TEE_FLAG_TRACK = (1 << 2),         // treat as a Tee for tracks (Record and Playback)
+    };
+
+    NBAIO_Tee()
+        : mTee(std::make_shared<NBAIO_TeeImpl>())
+    {
+        getRunningTees().add(mTee);
+    }
+
+    ~NBAIO_Tee() {
+        getRunningTees().remove(mTee);
+        dump(-1, "_DTOR"); // log any data remaining in Tee.
+    }
+
+    /**
+     * \brief set is used for deferred configuration of Tee.
+     *
+     *  May be called anytime except concurrently with write().
+     *
+     * \param format NBAIO_Format used to open NBAIO pipes
+     * \param flags (https://source.android.com/devices/audio/debugging)
+     *              - TEE_FLAG_NONE to bypass af.tee property checks (default);
+     *              - TEE_FLAG_INPUT_THREAD to check af.tee if input thread logging set;
+     *              - TEE_FLAG_OUTPUT_THREAD to check af.tee if output thread logging set;
+     *              - TEE_FLAG_TRACK to check af.tee if track logging set.
+     * \param frames number of frames to open the NBAIO pipe (set to 0 to use default).
+     *
+     * \return
+     *         - NO_ERROR on success (or format unchanged)
+     *         - BAD_VALUE if format or flags invalid.
+     *         - PERMISSION_DENIED if flags not allowed by af.tee
+     */
+
+    status_t set(const NBAIO_Format &format,
+            TEE_FLAG flags = TEE_FLAG_NONE, size_t frames = 0) const {
+        return mTee->set(format, flags, frames);
+    }
+
+    status_t set(uint32_t sampleRate, uint32_t channelCount, audio_format_t format,
+            TEE_FLAG flags = TEE_FLAG_NONE, size_t frames = 0) const {
+        return mTee->set(Format_from_SR_C(sampleRate, channelCount, format), flags, frames);
+    }
+
+    /**
+     * \brief write data to the tee.
+     *
+     * This call is lock free (as shared pointer and NBAIO is lock free);
+     * may be called simultaneous to all methods except set().
+     *
+     * \param buffer to write to pipe.
+     * \param frameCount in frames as specified by the format passed to set()
+     */
+
+    void write(const void *buffer, size_t frameCount) const {
+        mTee->write(buffer, frameCount);
+    }
+
+    /** sets Tee id string which identifies the generated file (should be unique). */
+    void setId(const std::string &id) const {
+        mTee->setId(id);
+    }
+
+    /**
+     * \brief dump the audio content written to the Tee.
+     *
+     * \param fd file descriptor to write dumped filename for logging, use -1 to ignore.
+     * \param reason string suffix to append to the generated file.
+     */
+    void dump(int fd, const std::string &reason = "") const {
+        mTee->dump(fd, reason);
+    }
+
+    /**
+     * \brief dump all Tees currently alive.
+     *
+     * \param fd file descriptor to write dumped filename for logging, use -1 to ignore.
+     * \param reason string suffix to append to the generated file.
+     */
+    static void dumpAll(int fd, const std::string &reason = "") {
+        getRunningTees().dump(fd, reason);
+    }
+
+private:
+
+    /** The underlying implementation of the Tee - the lifetime is through
+        a shared pointer so destruction of the NBAIO_Tee container may proceed
+        even though dumping is occurring. */
+    class NBAIO_TeeImpl {
+    public:
+        status_t set(const NBAIO_Format &format, TEE_FLAG flags, size_t frames) {
+            static const int teeConfig = property_get_bool("ro.debuggable", false)
+                   ? property_get_int32("af.tee", 0) : 0;
+
+            // check the type of Tee
+            const TEE_FLAG type = TEE_FLAG(
+                    flags & (TEE_FLAG_INPUT_THREAD | TEE_FLAG_OUTPUT_THREAD | TEE_FLAG_TRACK));
+
+            // parameter flags can't select multiple types.
+            if (__builtin_popcount(type) > 1) {
+                return BAD_VALUE;
+            }
+
+            // if type is set, we check to see if it is permitted by configuration.
+            if (type != 0 && (type & teeConfig) == 0) {
+                return PERMISSION_DENIED;
+            }
+
+            // determine number of frames for Tee
+            if (frames == 0) {
+                // TODO: consider varying frame count based on type.
+                frames = DEFAULT_TEE_FRAMES;
+            }
+
+            // TODO: should we check minimum number of frames?
+
+            // don't do anything if format and frames are the same.
+            if (Format_isEqual(format, mFormat) && frames == mFrames) {
+                return NO_ERROR;
+            }
+
+            bool enabled = false;
+            auto sinksource = makeSinkSource(format, frames, &enabled);
+
+            // enabled is set if makeSinkSource is successful.
+            // Note: as mentioned in NBAIO_Tee::set(), don't call set() while write() is
+            // ongoing.
+            if (enabled) {
+                std::lock_guard<std::mutex> _l(mLock);
+                mFlags = flags;
+                mFormat = format; // could get this from the Sink.
+                mFrames = frames;
+                mSinkSource = std::move(sinksource);
+                mEnabled.store(true);
+                return NO_ERROR;
+            }
+            return BAD_VALUE;
+        }
+
+        void setId(const std::string &id) {
+            std::lock_guard<std::mutex> _l(mLock);
+            mId = id;
+        }
+
+        void dump(int fd, const std::string &reason) {
+            if (!mDataReady.exchange(false)) return;
+            std::string suffix;
+            NBAIO_SinkSource sinkSource;
+            {
+                std::lock_guard<std::mutex> _l(mLock);
+                suffix = mId + reason;
+                sinkSource = mSinkSource;
+            }
+            dumpTee(fd, sinkSource, suffix);
+        }
+
+        void write(const void *buffer, size_t frameCount) {
+            if (!mEnabled.load() || frameCount == 0) return;
+            (void)mSinkSource.first->write(buffer, frameCount);
+            mDataReady.store(true);
+        }
+
+    private:
+        // TRICKY: We need to keep the NBAIO_Sink and NBAIO_Source both alive at the same time
+        // because PipeReader holds a naked reference (not a strong or weak pointer) to Pipe.
+        using NBAIO_SinkSource = std::pair<sp<NBAIO_Sink>, sp<NBAIO_Source>>;
+
+        static void dumpTee(int fd, const NBAIO_SinkSource& sinkSource, const std::string& suffix);
+
+        static NBAIO_SinkSource makeSinkSource(
+                const NBAIO_Format &format, size_t frames, bool *enabled);
+
+        // 0x200000 stereo 16-bit PCM frames = 47.5 seconds at 44.1 kHz, 8 megabytes
+        static constexpr size_t DEFAULT_TEE_FRAMES = 0x200000;
+
+        // atomic status checking
+        std::atomic<bool> mEnabled{false};
+        std::atomic<bool> mDataReady{false};
+
+        // locked dump information
+        mutable std::mutex mLock;
+        std::string mId;                                         // GUARDED_BY(mLock)
+        TEE_FLAG mFlags = TEE_FLAG_NONE;                         // GUARDED_BY(mLock)
+        NBAIO_Format mFormat = Format_Invalid;                   // GUARDED_BY(mLock)
+        size_t mFrames = 0;                                      // GUARDED_BY(mLock)
+        NBAIO_SinkSource mSinkSource;                            // GUARDED_BY(mLock)
+    };
+
+    /** RunningTees tracks current running tees for dump purposes.
+        It is implemented to have minimal locked regions, to be transparent to the caller. */
+    class RunningTees {
+    public:
+        void add(const std::shared_ptr<NBAIO_TeeImpl> &tee) {
+            std::lock_guard<std::mutex> _l(mLock);
+            ALOGW_IF(!mTees.emplace(tee).second,
+                    "%s: %p already exists in mTees", __func__, tee.get());
+        }
+
+        void remove(const std::shared_ptr<NBAIO_TeeImpl> &tee) {
+            std::lock_guard<std::mutex> _l(mLock);
+            ALOGW_IF(mTees.erase(tee) != 1,
+                    "%s: %p doesn't exist in mTees", __func__, tee.get());
+        }
+
+        void dump(int fd, const std::string &reason) {
+            std::vector<std::shared_ptr<NBAIO_TeeImpl>> tees; // safe snapshot of tees
+            {
+                std::lock_guard<std::mutex> _l(mLock);
+                tees.insert(tees.end(), mTees.begin(), mTees.end());
+            }
+            for (const auto &tee : tees) {
+                tee->dump(fd, reason);
+            }
+        }
+
+    private:
+        std::mutex mLock;
+        std::set<std::shared_ptr<NBAIO_TeeImpl>> mTees; // GUARDED_BY(mLock)
+    };
+
+    // singleton
+    static RunningTees &getRunningTees() {
+        static RunningTees runningTees;
+        return runningTees;
+    }
+
+    // The NBAIO TeeImpl may have lifetime longer than NBAIO_Tee if
+    // RunningTees::dump() is being called simultaneous to ~NBAIO_Tee().
+    // This is allowed for maximum concurrency.
+    const std::shared_ptr<NBAIO_TeeImpl> mTee;
+}; // NBAIO_Tee
+
+} // namespace android
+
+#endif // TEE_SINK
+#endif // !ANDROID_NBAIO_TEE_H
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index e5cb8a2..42a5a90 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -24,8 +24,9 @@
 #include <audio_utils/primitives.h>
 
 #include "AudioFlinger.h"
-#include "ServiceUtilities.h"
 #include <media/AudioParameter.h>
+#include <media/PatchBuilder.h>
+#include <mediautils/ServiceUtilities.h>
 
 // ----------------------------------------------------------------------------
 
@@ -49,107 +50,65 @@
                                 struct audio_port *ports)
 {
     Mutex::Autolock _l(mLock);
-    if (mPatchPanel != 0) {
-        return mPatchPanel->listAudioPorts(num_ports, ports);
-    }
-    return NO_INIT;
+    return mPatchPanel.listAudioPorts(num_ports, ports);
 }
 
 /* Get supported attributes for a given audio port */
 status_t AudioFlinger::getAudioPort(struct audio_port *port)
 {
     Mutex::Autolock _l(mLock);
-    if (mPatchPanel != 0) {
-        return mPatchPanel->getAudioPort(port);
-    }
-    return NO_INIT;
+    return mPatchPanel.getAudioPort(port);
 }
 
-
 /* Connect a patch between several source and sink ports */
 status_t AudioFlinger::createAudioPatch(const struct audio_patch *patch,
                                    audio_patch_handle_t *handle)
 {
     Mutex::Autolock _l(mLock);
-    if (mPatchPanel != 0) {
-        return mPatchPanel->createAudioPatch(patch, handle);
-    }
-    return NO_INIT;
+    return mPatchPanel.createAudioPatch(patch, handle);
 }
 
 /* Disconnect a patch */
 status_t AudioFlinger::releaseAudioPatch(audio_patch_handle_t handle)
 {
     Mutex::Autolock _l(mLock);
-    if (mPatchPanel != 0) {
-        return mPatchPanel->releaseAudioPatch(handle);
-    }
-    return NO_INIT;
+    return mPatchPanel.releaseAudioPatch(handle);
 }
 
-
 /* List connected audio ports and they attributes */
 status_t AudioFlinger::listAudioPatches(unsigned int *num_patches,
                                   struct audio_patch *patches)
 {
     Mutex::Autolock _l(mLock);
-    if (mPatchPanel != 0) {
-        return mPatchPanel->listAudioPatches(num_patches, patches);
-    }
-    return NO_INIT;
-}
-
-/* Set audio port configuration */
-status_t AudioFlinger::setAudioPortConfig(const struct audio_port_config *config)
-{
-    Mutex::Autolock _l(mLock);
-    if (mPatchPanel != 0) {
-        return mPatchPanel->setAudioPortConfig(config);
-    }
-    return NO_INIT;
-}
-
-
-AudioFlinger::PatchPanel::PatchPanel(const sp<AudioFlinger>& audioFlinger)
-                                   : mAudioFlinger(audioFlinger)
-{
-}
-
-AudioFlinger::PatchPanel::~PatchPanel()
-{
+    return mPatchPanel.listAudioPatches(num_patches, patches);
 }
 
 /* List connected audio ports and their attributes */
 status_t AudioFlinger::PatchPanel::listAudioPorts(unsigned int *num_ports __unused,
                                 struct audio_port *ports __unused)
 {
-    ALOGV("listAudioPorts");
+    ALOGV(__func__);
     return NO_ERROR;
 }
 
 /* Get supported attributes for a given audio port */
 status_t AudioFlinger::PatchPanel::getAudioPort(struct audio_port *port __unused)
 {
-    ALOGV("getAudioPort");
+    ALOGV(__func__);
     return NO_ERROR;
 }
 
-
 /* Connect a patch between several source and sink ports */
 status_t AudioFlinger::PatchPanel::createAudioPatch(const struct audio_patch *patch,
                                    audio_patch_handle_t *handle)
 {
-    status_t status = NO_ERROR;
-    audio_patch_handle_t halHandle = AUDIO_PATCH_HANDLE_NONE;
-    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
     if (handle == NULL || patch == NULL) {
         return BAD_VALUE;
     }
-    ALOGV("createAudioPatch() num_sources %d num_sinks %d handle %d",
-          patch->num_sources, patch->num_sinks, *handle);
-    if (audioflinger == 0) {
-        return NO_INIT;
-    }
+    ALOGV("%s() num_sources %d num_sinks %d handle %d",
+            __func__, patch->num_sources, patch->num_sinks, *handle);
+    status_t status = NO_ERROR;
+    audio_patch_handle_t halHandle = AUDIO_PATCH_HANDLE_NONE;
 
     if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
             (patch->num_sinks == 0 && patch->num_sources != 2) ||
@@ -163,81 +122,73 @@
     }
 
     if (*handle != AUDIO_PATCH_HANDLE_NONE) {
-        for (size_t index = 0; *handle != 0 && index < mPatches.size(); index++) {
-            if (*handle == mPatches[index]->mHandle) {
-                ALOGV("createAudioPatch() removing patch handle %d", *handle);
-                halHandle = mPatches[index]->mHalHandle;
-                Patch *removedPatch = mPatches[index];
-                // free resources owned by the removed patch if applicable
-                // 1) if a software patch is present, release the playback and capture threads and
-                // tracks created. This will also release the corresponding audio HAL patches
-                if ((removedPatch->mRecordPatchHandle
-                        != AUDIO_PATCH_HANDLE_NONE) ||
-                        (removedPatch->mPlaybackPatchHandle !=
-                                AUDIO_PATCH_HANDLE_NONE)) {
-                    clearPatchConnections(removedPatch);
-                }
-                // 2) if the new patch and old patch source or sink are devices from different
-                // hw modules,  clear the audio HAL patches now because they will not be updated
-                // by call to create_audio_patch() below which will happen on a different HW module
-                if (halHandle != AUDIO_PATCH_HANDLE_NONE) {
-                    audio_module_handle_t hwModule = AUDIO_MODULE_HANDLE_NONE;
-                    if ((removedPatch->mAudioPatch.sources[0].type == AUDIO_PORT_TYPE_DEVICE) &&
-                        ((patch->sources[0].type != AUDIO_PORT_TYPE_DEVICE) ||
-                          (removedPatch->mAudioPatch.sources[0].ext.device.hw_module !=
-                           patch->sources[0].ext.device.hw_module))) {
-                        hwModule = removedPatch->mAudioPatch.sources[0].ext.device.hw_module;
-                    } else if ((patch->num_sinks == 0) ||
-                            ((removedPatch->mAudioPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE) &&
-                             ((patch->sinks[0].type != AUDIO_PORT_TYPE_DEVICE) ||
-                              (removedPatch->mAudioPatch.sinks[0].ext.device.hw_module !=
-                               patch->sinks[0].ext.device.hw_module)))) {
-                        // Note on (patch->num_sinks == 0): this situation should not happen as
-                        // these special patches are only created by the policy manager but just
-                        // in case, systematically clear the HAL patch.
-                        // Note that removedPatch->mAudioPatch.num_sinks cannot be 0 here because
-                        // halHandle would be AUDIO_PATCH_HANDLE_NONE in this case.
-                        hwModule = removedPatch->mAudioPatch.sinks[0].ext.device.hw_module;
-                    }
-                    if (hwModule != AUDIO_MODULE_HANDLE_NONE) {
-                        ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(hwModule);
-                        if (index >= 0) {
-                            sp<DeviceHalInterface> hwDevice =
-                                    audioflinger->mAudioHwDevs.valueAt(index)->hwDevice();
-                            hwDevice->releaseAudioPatch(halHandle);
-                        }
-                    }
-                }
-                mPatches.removeAt(index);
-                delete removedPatch;
-                break;
+        auto iter = mPatches.find(*handle);
+        if (iter != mPatches.end()) {
+            ALOGV("%s() removing patch handle %d", __func__, *handle);
+            Patch &removedPatch = iter->second;
+            // free resources owned by the removed patch if applicable
+            // 1) if a software patch is present, release the playback and capture threads and
+            // tracks created. This will also release the corresponding audio HAL patches
+            if (removedPatch.isSoftware()) {
+                removedPatch.clearConnections(this);
             }
+            // 2) if the new patch and old patch source or sink are devices from different
+            // hw modules,  clear the audio HAL patches now because they will not be updated
+            // by call to create_audio_patch() below which will happen on a different HW module
+            if (removedPatch.mHalHandle != AUDIO_PATCH_HANDLE_NONE) {
+                audio_module_handle_t hwModule = AUDIO_MODULE_HANDLE_NONE;
+                const struct audio_patch &oldPatch = removedPatch.mAudioPatch;
+                if (oldPatch.sources[0].type == AUDIO_PORT_TYPE_DEVICE &&
+                        (patch->sources[0].type != AUDIO_PORT_TYPE_DEVICE ||
+                                oldPatch.sources[0].ext.device.hw_module !=
+                                patch->sources[0].ext.device.hw_module)) {
+                    hwModule = oldPatch.sources[0].ext.device.hw_module;
+                } else if (patch->num_sinks == 0 ||
+                        (oldPatch.sinks[0].type == AUDIO_PORT_TYPE_DEVICE &&
+                                (patch->sinks[0].type != AUDIO_PORT_TYPE_DEVICE ||
+                                        oldPatch.sinks[0].ext.device.hw_module !=
+                                        patch->sinks[0].ext.device.hw_module))) {
+                    // Note on (patch->num_sinks == 0): this situation should not happen as
+                    // these special patches are only created by the policy manager but just
+                    // in case, systematically clear the HAL patch.
+                    // Note that removedPatch.mAudioPatch.num_sinks cannot be 0 here because
+                    // removedPatch.mHalHandle would be AUDIO_PATCH_HANDLE_NONE in this case.
+                    hwModule = oldPatch.sinks[0].ext.device.hw_module;
+                }
+                sp<DeviceHalInterface> hwDevice = findHwDeviceByModule(hwModule);
+                if (hwDevice != 0) {
+                    hwDevice->releaseAudioPatch(removedPatch.mHalHandle);
+                }
+            }
+            mPatches.erase(iter);
         }
     }
 
-    Patch *newPatch = new Patch(patch);
+    Patch newPatch{*patch};
 
     switch (patch->sources[0].type) {
         case AUDIO_PORT_TYPE_DEVICE: {
             audio_module_handle_t srcModule = patch->sources[0].ext.device.hw_module;
-            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(srcModule);
+            ssize_t index = mAudioFlinger.mAudioHwDevs.indexOfKey(srcModule);
             if (index < 0) {
-                ALOGW("createAudioPatch() bad src hw module %d", srcModule);
+                ALOGW("%s() bad src hw module %d", __func__, srcModule);
                 status = BAD_VALUE;
                 goto exit;
             }
-            AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
+            AudioHwDevice *audioHwDevice = mAudioFlinger.mAudioHwDevs.valueAt(index);
             for (unsigned int i = 0; i < patch->num_sinks; i++) {
                 // support only one sink if connection to a mix or across HW modules
                 if ((patch->sinks[i].type == AUDIO_PORT_TYPE_MIX ||
-                        patch->sinks[i].ext.mix.hw_module != srcModule) &&
+                                (patch->sinks[i].type == AUDIO_PORT_TYPE_DEVICE &&
+                                        patch->sinks[i].ext.device.hw_module != srcModule)) &&
                         patch->num_sinks > 1) {
+                    ALOGW("%s() multiple sinks for mix or across modules not supported", __func__);
                     status = INVALID_OPERATION;
                     goto exit;
                 }
                 // reject connection to different sink types
                 if (patch->sinks[i].type != patch->sinks[0].type) {
-                    ALOGW("createAudioPatch() different sink types in same patch not supported");
+                    ALOGW("%s() different sink types in same patch not supported", __func__);
                     status = BAD_VALUE;
                     goto exit;
                 }
@@ -256,38 +207,42 @@
                     if (patch->sources[1].type != AUDIO_PORT_TYPE_MIX ||
                             (patch->num_sinks != 0 && patch->sinks[0].ext.device.hw_module !=
                                     patch->sources[1].ext.mix.hw_module)) {
-                        ALOGW("createAudioPatch() invalid source combination");
+                        ALOGW("%s() invalid source combination", __func__);
                         status = INVALID_OPERATION;
                         goto exit;
                     }
 
                     sp<ThreadBase> thread =
-                            audioflinger->checkPlaybackThread_l(patch->sources[1].ext.mix.handle);
-                    newPatch->mPlaybackThread = (MixerThread *)thread.get();
+                            mAudioFlinger.checkPlaybackThread_l(patch->sources[1].ext.mix.handle);
                     if (thread == 0) {
-                        ALOGW("createAudioPatch() cannot get playback thread");
+                        ALOGW("%s() cannot get playback thread", __func__);
                         status = INVALID_OPERATION;
                         goto exit;
                     }
+                    // existing playback thread is reused, so it is not closed when patch is cleared
+                    newPatch.mPlayback.setThread(
+                            reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
                 } else {
                     audio_config_t config = AUDIO_CONFIG_INITIALIZER;
                     audio_devices_t device = patch->sinks[0].ext.device.type;
                     String8 address = String8(patch->sinks[0].ext.device.address);
                     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-                    sp<ThreadBase> thread = audioflinger->openOutput_l(
+                    audio_output_flags_t flags =
+                            patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
+                            patch->sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
+                    sp<ThreadBase> thread = mAudioFlinger.openOutput_l(
                                                             patch->sinks[0].ext.device.hw_module,
                                                             &output,
                                                             &config,
                                                             device,
                                                             address,
-                                                            AUDIO_OUTPUT_FLAG_NONE);
-                    newPatch->mPlaybackThread = (PlaybackThread *)thread.get();
-                    ALOGV("audioflinger->openOutput_l() returned %p",
-                                          newPatch->mPlaybackThread.get());
-                    if (newPatch->mPlaybackThread == 0) {
+                                                            flags);
+                    ALOGV("mAudioFlinger.openOutput_l() returned %p", thread.get());
+                    if (thread == 0) {
                         status = NO_MEMORY;
                         goto exit;
                     }
+                    newPatch.mPlayback.setThread(reinterpret_cast<PlaybackThread*>(thread.get()));
                 }
                 audio_devices_t device = patch->sources[0].ext.device.type;
                 String8 address = String8(patch->sources[0].ext.device.address);
@@ -297,47 +252,50 @@
                 if (patch->sources[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
                     config.sample_rate = patch->sources[0].sample_rate;
                 } else {
-                    config.sample_rate = newPatch->mPlaybackThread->sampleRate();
+                    config.sample_rate = newPatch.mPlayback.thread()->sampleRate();
                 }
                 if (patch->sources[0].config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
                     config.channel_mask = patch->sources[0].channel_mask;
                 } else {
-                    config.channel_mask =
-                        audio_channel_in_mask_from_count(newPatch->mPlaybackThread->channelCount());
+                    config.channel_mask = audio_channel_in_mask_from_count(
+                            newPatch.mPlayback.thread()->channelCount());
                 }
                 if (patch->sources[0].config_mask & AUDIO_PORT_CONFIG_FORMAT) {
                     config.format = patch->sources[0].format;
                 } else {
-                    config.format = newPatch->mPlaybackThread->format();
+                    config.format = newPatch.mPlayback.thread()->format();
                 }
+                audio_input_flags_t flags =
+                        patch->sources[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
+                        patch->sources[0].flags.input : AUDIO_INPUT_FLAG_NONE;
                 audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
-                sp<ThreadBase> thread = audioflinger->openInput_l(srcModule,
+                sp<ThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
                                                                     &input,
                                                                     &config,
                                                                     device,
                                                                     address,
                                                                     AUDIO_SOURCE_MIC,
-                                                                    AUDIO_INPUT_FLAG_NONE);
-                newPatch->mRecordThread = (RecordThread *)thread.get();
-                ALOGV("audioflinger->openInput_l() returned %p inChannelMask %08x",
-                      newPatch->mRecordThread.get(), config.channel_mask);
-                if (newPatch->mRecordThread == 0) {
+                                                                    flags);
+                ALOGV("mAudioFlinger.openInput_l() returned %p inChannelMask %08x",
+                      thread.get(), config.channel_mask);
+                if (thread == 0) {
                     status = NO_MEMORY;
                     goto exit;
                 }
-                status = createPatchConnections(newPatch, patch);
+                newPatch.mRecord.setThread(reinterpret_cast<RecordThread*>(thread.get()));
+                status = newPatch.createConnections(this);
                 if (status != NO_ERROR) {
                     goto exit;
                 }
             } else {
                 if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
-                    sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
+                    sp<ThreadBase> thread = mAudioFlinger.checkRecordThread_l(
                                                               patch->sinks[0].ext.mix.handle);
                     if (thread == 0) {
-                        thread = audioflinger->checkMmapThread_l(patch->sinks[0].ext.mix.handle);
+                        thread = mAudioFlinger.checkMmapThread_l(patch->sinks[0].ext.mix.handle);
                         if (thread == 0) {
-                            ALOGW("createAudioPatch() bad capture I/O handle %d",
-                                                                  patch->sinks[0].ext.mix.handle);
+                            ALOGW("%s() bad capture I/O handle %d",
+                                    __func__, patch->sinks[0].ext.mix.handle);
                             status = BAD_VALUE;
                             goto exit;
                         }
@@ -356,9 +314,9 @@
         } break;
         case AUDIO_PORT_TYPE_MIX: {
             audio_module_handle_t srcModule =  patch->sources[0].ext.mix.hw_module;
-            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(srcModule);
+            ssize_t index = mAudioFlinger.mAudioHwDevs.indexOfKey(srcModule);
             if (index < 0) {
-                ALOGW("createAudioPatch() bad src hw module %d", srcModule);
+                ALOGW("%s() bad src hw module %d", __func__, srcModule);
                 status = BAD_VALUE;
                 goto exit;
             }
@@ -366,8 +324,8 @@
             audio_devices_t type = AUDIO_DEVICE_NONE;
             for (unsigned int i = 0; i < patch->num_sinks; i++) {
                 if (patch->sinks[i].type != AUDIO_PORT_TYPE_DEVICE) {
-                    ALOGW("createAudioPatch() invalid sink type %d for mix source",
-                          patch->sinks[i].type);
+                    ALOGW("%s() invalid sink type %d for mix source",
+                            __func__, patch->sinks[i].type);
                     status = BAD_VALUE;
                     goto exit;
                 }
@@ -379,21 +337,21 @@
                 type |= patch->sinks[i].ext.device.type;
             }
             sp<ThreadBase> thread =
-                            audioflinger->checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
+                            mAudioFlinger.checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
             if (thread == 0) {
-                thread = audioflinger->checkMmapThread_l(patch->sources[0].ext.mix.handle);
+                thread = mAudioFlinger.checkMmapThread_l(patch->sources[0].ext.mix.handle);
                 if (thread == 0) {
-                    ALOGW("createAudioPatch() bad playback I/O handle %d",
-                              patch->sources[0].ext.mix.handle);
+                    ALOGW("%s() bad playback I/O handle %d",
+                            __func__, patch->sources[0].ext.mix.handle);
                     status = BAD_VALUE;
                     goto exit;
                 }
             }
-            if (thread == audioflinger->primaryPlaybackThread_l()) {
+            if (thread == mAudioFlinger.primaryPlaybackThread_l()) {
                 AudioParameter param = AudioParameter();
                 param.addInt(String8(AudioParameter::keyRouting), (int)type);
 
-                audioflinger->broacastParametersToRecordThreads_l(param.toString());
+                mAudioFlinger.broacastParametersToRecordThreads_l(param.toString());
             }
 
             status = thread->sendCreateAudioPatchConfigEvent(patch, &halHandle);
@@ -403,295 +361,284 @@
             goto exit;
     }
 exit:
-    ALOGV("createAudioPatch() status %d", status);
+    ALOGV("%s() status %d", __func__, status);
     if (status == NO_ERROR) {
-        *handle = (audio_patch_handle_t) audioflinger->nextUniqueId(AUDIO_UNIQUE_ID_USE_PATCH);
-        newPatch->mHandle = *handle;
-        newPatch->mHalHandle = halHandle;
-        mPatches.add(newPatch);
-        ALOGV("createAudioPatch() added new patch handle %d halHandle %d", *handle, halHandle);
+        *handle = (audio_patch_handle_t) mAudioFlinger.nextUniqueId(AUDIO_UNIQUE_ID_USE_PATCH);
+        newPatch.mHalHandle = halHandle;
+        mPatches.insert(std::make_pair(*handle, std::move(newPatch)));
+        ALOGV("%s() added new patch handle %d halHandle %d", __func__, *handle, halHandle);
     } else {
-        clearPatchConnections(newPatch);
-        delete newPatch;
+        newPatch.clearConnections(this);
     }
     return status;
 }
 
-status_t AudioFlinger::PatchPanel::createPatchConnections(Patch *patch,
-                                                          const struct audio_patch *audioPatch)
+AudioFlinger::PatchPanel::Patch::~Patch()
+{
+    ALOGE_IF(isSoftware(), "Software patch connections leaked %d %d",
+            mRecord.handle(), mPlayback.handle());
+}
+
+status_t AudioFlinger::PatchPanel::Patch::createConnections(PatchPanel *panel)
 {
     // create patch from source device to record thread input
-    struct audio_patch subPatch;
-    subPatch.num_sources = 1;
-    subPatch.sources[0] = audioPatch->sources[0];
-    subPatch.num_sinks = 1;
-
-    patch->mRecordThread->getAudioPortConfig(&subPatch.sinks[0]);
-    subPatch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_MIC;
-
-    status_t status = createAudioPatch(&subPatch, &patch->mRecordPatchHandle);
+    status_t status = panel->createAudioPatch(
+            PatchBuilder().addSource(mAudioPatch.sources[0]).
+                addSink(mRecord.thread(), { .source = AUDIO_SOURCE_MIC }).patch(),
+            mRecord.handlePtr());
     if (status != NO_ERROR) {
-        patch->mRecordPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+        *mRecord.handlePtr() = AUDIO_PATCH_HANDLE_NONE;
         return status;
     }
 
     // create patch from playback thread output to sink device
-    if (audioPatch->num_sinks != 0) {
-        patch->mPlaybackThread->getAudioPortConfig(&subPatch.sources[0]);
-        subPatch.sinks[0] = audioPatch->sinks[0];
-        status = createAudioPatch(&subPatch, &patch->mPlaybackPatchHandle);
+    if (mAudioPatch.num_sinks != 0) {
+        status = panel->createAudioPatch(
+                PatchBuilder().addSource(mPlayback.thread()).addSink(mAudioPatch.sinks[0]).patch(),
+                mPlayback.handlePtr());
         if (status != NO_ERROR) {
-            patch->mPlaybackPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+            *mPlayback.handlePtr() = AUDIO_PATCH_HANDLE_NONE;
             return status;
         }
     } else {
-        patch->mPlaybackPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+        *mPlayback.handlePtr() = AUDIO_PATCH_HANDLE_NONE;
     }
 
     // use a pseudo LCM between input and output framecount
-    size_t playbackFrameCount = patch->mPlaybackThread->frameCount();
+    size_t playbackFrameCount = mPlayback.thread()->frameCount();
     int playbackShift = __builtin_ctz(playbackFrameCount);
-    size_t recordFramecount = patch->mRecordThread->frameCount();
+    size_t recordFramecount = mRecord.thread()->frameCount();
     int shift = __builtin_ctz(recordFramecount);
     if (playbackShift < shift) {
         shift = playbackShift;
     }
     size_t frameCount = (playbackFrameCount * recordFramecount) >> shift;
-    ALOGV("createPatchConnections() playframeCount %zu recordFramecount %zu frameCount %zu",
-          playbackFrameCount, recordFramecount, frameCount);
+    ALOGV("%s() playframeCount %zu recordFramecount %zu frameCount %zu",
+            __func__, playbackFrameCount, recordFramecount, frameCount);
 
     // create a special record track to capture from record thread
-    uint32_t channelCount = patch->mPlaybackThread->channelCount();
+    uint32_t channelCount = mPlayback.thread()->channelCount();
     audio_channel_mask_t inChannelMask = audio_channel_in_mask_from_count(channelCount);
-    audio_channel_mask_t outChannelMask = patch->mPlaybackThread->channelMask();
-    uint32_t sampleRate = patch->mPlaybackThread->sampleRate();
-    audio_format_t format = patch->mPlaybackThread->format();
+    audio_channel_mask_t outChannelMask = mPlayback.thread()->channelMask();
+    uint32_t sampleRate = mPlayback.thread()->sampleRate();
+    audio_format_t format = mPlayback.thread()->format();
 
-    patch->mPatchRecord = new RecordThread::PatchRecord(
-                                             patch->mRecordThread.get(),
+    audio_format_t inputFormat = mRecord.thread()->format();
+    if (!audio_is_linear_pcm(inputFormat)) {
+        // The playbackThread format will say PCM for IEC61937 packetized stream.
+        // Use recordThread format.
+        format = inputFormat;
+    }
+    audio_input_flags_t inputFlags = mAudioPatch.sources[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
+            mAudioPatch.sources[0].flags.input : AUDIO_INPUT_FLAG_NONE;
+    sp<RecordThread::PatchRecord> tempRecordTrack = new (std::nothrow) RecordThread::PatchRecord(
+                                             mRecord.thread().get(),
                                              sampleRate,
                                              inChannelMask,
                                              format,
                                              frameCount,
                                              NULL,
                                              (size_t)0 /* bufferSize */,
-                                             AUDIO_INPUT_FLAG_NONE);
-    if (patch->mPatchRecord == 0) {
-        return NO_MEMORY;
-    }
-    status = patch->mPatchRecord->initCheck();
+                                             inputFlags);
+    status = mRecord.checkTrack(tempRecordTrack.get());
     if (status != NO_ERROR) {
         return status;
     }
-    patch->mRecordThread->addPatchRecord(patch->mPatchRecord);
+
+    audio_output_flags_t outputFlags = mAudioPatch.sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
+            mAudioPatch.sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
 
     // create a special playback track to render to playback thread.
     // this track is given the same buffer as the PatchRecord buffer
-    patch->mPatchTrack = new PlaybackThread::PatchTrack(
-                                           patch->mPlaybackThread.get(),
-                                           audioPatch->sources[1].ext.mix.usecase.stream,
+    sp<PlaybackThread::PatchTrack> tempPatchTrack = new (std::nothrow) PlaybackThread::PatchTrack(
+                                           mPlayback.thread().get(),
+                                           mAudioPatch.sources[1].ext.mix.usecase.stream,
                                            sampleRate,
                                            outChannelMask,
                                            format,
                                            frameCount,
-                                           patch->mPatchRecord->buffer(),
-                                           patch->mPatchRecord->bufferSize(),
-                                           AUDIO_OUTPUT_FLAG_NONE);
-    status = patch->mPatchTrack->initCheck();
+                                           tempRecordTrack->buffer(),
+                                           tempRecordTrack->bufferSize(),
+                                           outputFlags);
+    status = mPlayback.checkTrack(tempPatchTrack.get());
     if (status != NO_ERROR) {
         return status;
     }
-    patch->mPlaybackThread->addPatchTrack(patch->mPatchTrack);
 
     // tie playback and record tracks together
-    patch->mPatchRecord->setPeerProxy(patch->mPatchTrack.get());
-    patch->mPatchTrack->setPeerProxy(patch->mPatchRecord.get());
+    mRecord.setTrackAndPeer(tempRecordTrack, tempPatchTrack.get());
+    mPlayback.setTrackAndPeer(tempPatchTrack, tempRecordTrack.get());
 
     // start capture and playback
-    patch->mPatchRecord->start(AudioSystem::SYNC_EVENT_NONE, AUDIO_SESSION_NONE);
-    patch->mPatchTrack->start();
+    mRecord.track()->start(AudioSystem::SYNC_EVENT_NONE, AUDIO_SESSION_NONE);
+    mPlayback.track()->start();
 
     return status;
 }
 
-void AudioFlinger::PatchPanel::clearPatchConnections(Patch *patch)
+void AudioFlinger::PatchPanel::Patch::clearConnections(PatchPanel *panel)
 {
-    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
-    if (audioflinger == 0) {
-        return;
+    ALOGV("%s() mRecord.handle %d mPlayback.handle %d",
+            __func__, mRecord.handle(), mPlayback.handle());
+    mRecord.stopTrack();
+    mPlayback.stopTrack();
+    mRecord.closeConnections(panel);
+    mPlayback.closeConnections(panel);
+}
+
+status_t AudioFlinger::PatchPanel::Patch::getLatencyMs(double *latencyMs) const
+{
+    if (!isSoftware()) return INVALID_OPERATION;
+
+    auto recordTrack = mRecord.const_track();
+    if (recordTrack.get() == nullptr) return INVALID_OPERATION;
+
+    auto playbackTrack = mPlayback.const_track();
+    if (playbackTrack.get() == nullptr) return INVALID_OPERATION;
+
+    // Latency information for tracks may be called without obtaining
+    // the underlying thread lock.
+    //
+    // We use record server latency + playback track latency (generally smaller than the
+    // reverse due to internal biases).
+    //
+    // TODO: is this stable enough? Consider a PatchTrack synchronized version of this.
+    double recordServerLatencyMs;
+    if (recordTrack->getServerLatencyMs(&recordServerLatencyMs) != OK) return INVALID_OPERATION;
+
+    double playbackTrackLatencyMs;
+    if (playbackTrack->getTrackLatencyMs(&playbackTrackLatencyMs) != OK) return INVALID_OPERATION;
+
+    *latencyMs = recordServerLatencyMs + playbackTrackLatencyMs;
+    return OK;
+}
+
+String8 AudioFlinger::PatchPanel::Patch::dump(audio_patch_handle_t myHandle)
+{
+    String8 result;
+
+    // TODO: Consider table dump form for patches, just like tracks.
+    result.appendFormat("Patch %d: thread %p => thread %p",
+            myHandle, mRecord.thread().get(), mPlayback.thread().get());
+
+    // add latency if it exists
+    double latencyMs;
+    if (getLatencyMs(&latencyMs) == OK) {
+        result.appendFormat("  latency: %.2lf", latencyMs);
     }
 
-    ALOGV("clearPatchConnections() patch->mRecordPatchHandle %d patch->mPlaybackPatchHandle %d",
-          patch->mRecordPatchHandle, patch->mPlaybackPatchHandle);
-
-    if (patch->mPatchRecord != 0) {
-        patch->mPatchRecord->stop();
-    }
-    if (patch->mPatchTrack != 0) {
-        patch->mPatchTrack->stop();
-    }
-    if (patch->mRecordPatchHandle != AUDIO_PATCH_HANDLE_NONE) {
-        releaseAudioPatch(patch->mRecordPatchHandle);
-        patch->mRecordPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-    }
-    if (patch->mPlaybackPatchHandle != AUDIO_PATCH_HANDLE_NONE) {
-        releaseAudioPatch(patch->mPlaybackPatchHandle);
-        patch->mPlaybackPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-    }
-    if (patch->mRecordThread != 0) {
-        if (patch->mPatchRecord != 0) {
-            patch->mRecordThread->deletePatchRecord(patch->mPatchRecord);
-        }
-        audioflinger->closeInputInternal_l(patch->mRecordThread);
-    }
-    if (patch->mPlaybackThread != 0) {
-        if (patch->mPatchTrack != 0) {
-            patch->mPlaybackThread->deletePatchTrack(patch->mPatchTrack);
-        }
-        // if num sources == 2 we are reusing an existing playback thread so we do not close it
-        if (patch->mAudioPatch.num_sources != 2) {
-            audioflinger->closeOutputInternal_l(patch->mPlaybackThread);
-        }
-    }
-    if (patch->mRecordThread != 0) {
-        if (patch->mPatchRecord != 0) {
-            patch->mPatchRecord.clear();
-        }
-        patch->mRecordThread.clear();
-    }
-    if (patch->mPlaybackThread != 0) {
-        if (patch->mPatchTrack != 0) {
-            patch->mPatchTrack.clear();
-        }
-        patch->mPlaybackThread.clear();
-    }
-
+    result.append("\n");
+    return result;
 }
 
 /* Disconnect a patch */
 status_t AudioFlinger::PatchPanel::releaseAudioPatch(audio_patch_handle_t handle)
 {
-    ALOGV("releaseAudioPatch handle %d", handle);
+    ALOGV("%s handle %d", __func__, handle);
     status_t status = NO_ERROR;
-    size_t index;
 
-    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
-    if (audioflinger == 0) {
-        return NO_INIT;
-    }
-
-    for (index = 0; index < mPatches.size(); index++) {
-        if (handle == mPatches[index]->mHandle) {
-            break;
-        }
-    }
-    if (index == mPatches.size()) {
+    auto iter = mPatches.find(handle);
+    if (iter == mPatches.end()) {
         return BAD_VALUE;
     }
-    Patch *removedPatch = mPatches[index];
-    mPatches.removeAt(index);
+    Patch &removedPatch = iter->second;
+    const struct audio_patch &patch = removedPatch.mAudioPatch;
 
-    struct audio_patch *patch = &removedPatch->mAudioPatch;
-
-    switch (patch->sources[0].type) {
+    const struct audio_port_config &src = patch.sources[0];
+    switch (src.type) {
         case AUDIO_PORT_TYPE_DEVICE: {
-            audio_module_handle_t srcModule = patch->sources[0].ext.device.hw_module;
-            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(srcModule);
-            if (index < 0) {
-                ALOGW("releaseAudioPatch() bad src hw module %d", srcModule);
+            sp<DeviceHalInterface> hwDevice = findHwDeviceByModule(src.ext.device.hw_module);
+            if (hwDevice == 0) {
+                ALOGW("%s() bad src hw module %d", __func__, src.ext.device.hw_module);
                 status = BAD_VALUE;
                 break;
             }
 
-            if (removedPatch->mRecordPatchHandle != AUDIO_PATCH_HANDLE_NONE ||
-                    removedPatch->mPlaybackPatchHandle != AUDIO_PATCH_HANDLE_NONE) {
-                clearPatchConnections(removedPatch);
+            if (removedPatch.isSoftware()) {
+                removedPatch.clearConnections(this);
                 break;
             }
 
-            if (patch->sinks[0].type == AUDIO_PORT_TYPE_MIX) {
-                sp<ThreadBase> thread = audioflinger->checkRecordThread_l(
-                                                                patch->sinks[0].ext.mix.handle);
+            if (patch.sinks[0].type == AUDIO_PORT_TYPE_MIX) {
+                audio_io_handle_t ioHandle = patch.sinks[0].ext.mix.handle;
+                sp<ThreadBase> thread = mAudioFlinger.checkRecordThread_l(ioHandle);
                 if (thread == 0) {
-                    thread = audioflinger->checkMmapThread_l(patch->sinks[0].ext.mix.handle);
+                    thread = mAudioFlinger.checkMmapThread_l(ioHandle);
                     if (thread == 0) {
-                        ALOGW("releaseAudioPatch() bad capture I/O handle %d",
-                                                                  patch->sinks[0].ext.mix.handle);
+                        ALOGW("%s() bad capture I/O handle %d", __func__, ioHandle);
                         status = BAD_VALUE;
                         break;
                     }
                 }
-                status = thread->sendReleaseAudioPatchConfigEvent(removedPatch->mHalHandle);
+                status = thread->sendReleaseAudioPatchConfigEvent(removedPatch.mHalHandle);
             } else {
-                AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
-                sp<DeviceHalInterface> hwDevice = audioHwDevice->hwDevice();
-                status = hwDevice->releaseAudioPatch(removedPatch->mHalHandle);
+                status = hwDevice->releaseAudioPatch(removedPatch.mHalHandle);
             }
         } break;
         case AUDIO_PORT_TYPE_MIX: {
-            audio_module_handle_t srcModule =  patch->sources[0].ext.mix.hw_module;
-            ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(srcModule);
-            if (index < 0) {
-                ALOGW("releaseAudioPatch() bad src hw module %d", srcModule);
+            if (findHwDeviceByModule(src.ext.mix.hw_module) == 0) {
+                ALOGW("%s() bad src hw module %d", __func__, src.ext.mix.hw_module);
                 status = BAD_VALUE;
                 break;
             }
-            sp<ThreadBase> thread =
-                            audioflinger->checkPlaybackThread_l(patch->sources[0].ext.mix.handle);
+            audio_io_handle_t ioHandle = src.ext.mix.handle;
+            sp<ThreadBase> thread = mAudioFlinger.checkPlaybackThread_l(ioHandle);
             if (thread == 0) {
-                thread = audioflinger->checkMmapThread_l(patch->sources[0].ext.mix.handle);
+                thread = mAudioFlinger.checkMmapThread_l(ioHandle);
                 if (thread == 0) {
-                    ALOGW("releaseAudioPatch() bad playback I/O handle %d",
-                                                                  patch->sources[0].ext.mix.handle);
+                    ALOGW("%s() bad playback I/O handle %d", __func__, ioHandle);
                     status = BAD_VALUE;
                     break;
                 }
             }
-            status = thread->sendReleaseAudioPatchConfigEvent(removedPatch->mHalHandle);
+            status = thread->sendReleaseAudioPatchConfigEvent(removedPatch.mHalHandle);
         } break;
         default:
             status = BAD_VALUE;
-            break;
     }
 
-    delete removedPatch;
+    mPatches.erase(iter);
     return status;
 }
 
-
 /* List connected audio ports and they attributes */
 status_t AudioFlinger::PatchPanel::listAudioPatches(unsigned int *num_patches __unused,
                                   struct audio_patch *patches __unused)
 {
-    ALOGV("listAudioPatches");
+    ALOGV(__func__);
     return NO_ERROR;
 }
 
-/* Set audio port configuration */
-status_t AudioFlinger::PatchPanel::setAudioPortConfig(const struct audio_port_config *config)
+sp<DeviceHalInterface> AudioFlinger::PatchPanel::findHwDeviceByModule(audio_module_handle_t module)
 {
-    ALOGV("setAudioPortConfig");
-
-    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
-    if (audioflinger == 0) {
-        return NO_INIT;
-    }
-
-    audio_module_handle_t module;
-    if (config->type == AUDIO_PORT_TYPE_DEVICE) {
-        module = config->ext.device.hw_module;
-    } else {
-        module = config->ext.mix.hw_module;
-    }
-
-    ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(module);
+    if (module == AUDIO_MODULE_HANDLE_NONE) return nullptr;
+    ssize_t index = mAudioFlinger.mAudioHwDevs.indexOfKey(module);
     if (index < 0) {
-        ALOGW("setAudioPortConfig() bad hw module %d", module);
-        return BAD_VALUE;
+        return nullptr;
     }
+    return mAudioFlinger.mAudioHwDevs.valueAt(index)->hwDevice();
+}
 
-    AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
-    return audioHwDevice->hwDevice()->setAudioPortConfig(config);
+void AudioFlinger::PatchPanel::dump(int fd)
+{
+    // Only dump software patches.
+    bool headerPrinted = false;
+    for (auto& iter : mPatches) {
+        if (iter.second.isSoftware()) {
+            if (!headerPrinted) {
+                String8 header("\nSoftware patches:\n");
+                write(fd, header.string(), header.size());
+                headerPrinted = true;
+            }
+            String8 patchDump("  ");
+            patchDump.append(iter.second.dump(iter.first));
+            write(fd, patchDump.string(), patchDump.size());
+        }
+    }
+    if (headerPrinted) {
+        String8 trailing("\n");
+        write(fd, trailing.string(), trailing.size());
+    }
 }
 
 } // namespace android
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index d37c0d3..5d6bf00 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -19,13 +19,10 @@
     #error This header file should only be included from AudioFlinger.h
 #endif
 
-class PatchPanel : public RefBase {
+// PatchPanel is concealed within AudioFlinger, their lifetimes are the same.
+class PatchPanel {
 public:
-
-    class Patch;
-
-    explicit PatchPanel(const sp<AudioFlinger>& audioFlinger);
-    virtual ~PatchPanel();
+    explicit PatchPanel(AudioFlinger* audioFlinger) : mAudioFlinger(*audioFlinger) {}
 
     /* List connected audio ports and their attributes */
     status_t listAudioPorts(unsigned int *num_ports,
@@ -45,46 +42,104 @@
     status_t listAudioPatches(unsigned int *num_patches,
                                       struct audio_patch *patches);
 
-    /* Set audio port configuration */
-    status_t setAudioPortConfig(const struct audio_port_config *config);
+    void dump(int fd);
 
-    status_t createPatchConnections(Patch *patch,
-                                    const struct audio_patch *audioPatch);
-    void clearPatchConnections(Patch *patch);
+private:
+    template<typename ThreadType, typename TrackType>
+    class Endpoint {
+    public:
+        Endpoint() = default;
+        Endpoint(Endpoint&& other) { *this = std::move(other); }
+        Endpoint& operator=(Endpoint&& other) {
+            ALOGE_IF(mHandle != AUDIO_PATCH_HANDLE_NONE,
+                    "A non empty Patch Endpoint leaked, handle %d", mHandle);
+            *this = other;
+            other.mHandle = AUDIO_PATCH_HANDLE_NONE;
+            return *this;
+        }
+
+        status_t checkTrack(TrackType *trackOrNull) const {
+            if (trackOrNull == nullptr) return NO_MEMORY;
+            return trackOrNull->initCheck();
+        }
+        audio_patch_handle_t handle() const { return mHandle; }
+        sp<ThreadType> thread() { return mThread; }
+        sp<TrackType> track() { return mTrack; }
+        sp<const TrackType> const_track() const { return mTrack; }
+
+        void closeConnections(PatchPanel *panel) {
+            if (mHandle != AUDIO_PATCH_HANDLE_NONE) {
+                panel->releaseAudioPatch(mHandle);
+                mHandle = AUDIO_PATCH_HANDLE_NONE;
+            }
+            if (mThread != 0) {
+                if (mTrack != 0) {
+                    mThread->deletePatchTrack(mTrack);
+                }
+                if (mCloseThread) {
+                    panel->mAudioFlinger.closeThreadInternal_l(mThread);
+                }
+            }
+        }
+        audio_patch_handle_t* handlePtr() { return &mHandle; }
+        void setThread(const sp<ThreadType>& thread, bool closeThread = true) {
+            mThread = thread;
+            mCloseThread = closeThread;
+        }
+        void setTrackAndPeer(const sp<TrackType>& track,
+                             ThreadBase::PatchProxyBufferProvider *peer) {
+            mTrack = track;
+            mThread->addPatchTrack(mTrack);
+            mTrack->setPeerProxy(peer);
+        }
+        void stopTrack() { if (mTrack) mTrack->stop(); }
+
+    private:
+        Endpoint(const Endpoint&) = default;
+        Endpoint& operator=(const Endpoint&) = default;
+
+        sp<ThreadType> mThread;
+        bool mCloseThread = true;
+        audio_patch_handle_t mHandle = AUDIO_PATCH_HANDLE_NONE;
+        sp<TrackType> mTrack;
+    };
 
     class Patch {
     public:
-        explicit Patch(const struct audio_patch *patch) :
-            mAudioPatch(*patch), mHandle(AUDIO_PATCH_HANDLE_NONE),
-            mHalHandle(AUDIO_PATCH_HANDLE_NONE), mRecordPatchHandle(AUDIO_PATCH_HANDLE_NONE),
-            mPlaybackPatchHandle(AUDIO_PATCH_HANDLE_NONE) {}
-        ~Patch() {}
+        explicit Patch(const struct audio_patch &patch) : mAudioPatch(patch) {}
+        ~Patch();
+        Patch(const Patch&) = delete;
+        Patch(Patch&&) = default;
+        Patch& operator=(const Patch&) = delete;
+        Patch& operator=(Patch&&) = default;
 
+        status_t createConnections(PatchPanel *panel);
+        void clearConnections(PatchPanel *panel);
+        bool isSoftware() const {
+            return mRecord.handle() != AUDIO_PATCH_HANDLE_NONE ||
+                    mPlayback.handle() != AUDIO_PATCH_HANDLE_NONE; }
+
+        // returns the latency of the patch (from record to playback).
+        status_t getLatencyMs(double *latencyMs) const;
+
+        String8 dump(audio_patch_handle_t myHandle);
+
+        // Note that audio_patch::id is only unique within a HAL module
         struct audio_patch              mAudioPatch;
-        audio_patch_handle_t            mHandle;
         // handle for audio HAL patch handle present only when the audio HAL version is >= 3.0
-        audio_patch_handle_t            mHalHandle;
+        audio_patch_handle_t            mHalHandle = AUDIO_PATCH_HANDLE_NONE;
         // below members are used by a software audio patch connecting a source device from a
         // given audio HW module to a sink device on an other audio HW module.
-        // playback thread created by createAudioPatch() and released by clearPatchConnections() if
-        // no existing playback thread can be used by the software patch
-        sp<PlaybackThread>              mPlaybackThread;
-        // audio track created by createPatchConnections() and released by clearPatchConnections()
-        sp<PlaybackThread::PatchTrack>  mPatchTrack;
-        // record thread created by createAudioPatch() and released by clearPatchConnections()
-        sp<RecordThread>                mRecordThread;
-        // audio record created by createPatchConnections() and released by clearPatchConnections()
-        sp<RecordThread::PatchRecord>   mPatchRecord;
-        // handle for audio patch connecting source device to record thread input.
-        // created by createPatchConnections() and released by clearPatchConnections()
-        audio_patch_handle_t            mRecordPatchHandle;
-        // handle for audio patch connecting playback thread output to sink device
-        // created by createPatchConnections() and released by clearPatchConnections()
-        audio_patch_handle_t            mPlaybackPatchHandle;
-
+        // the objects are created by createConnections() and released by clearConnections()
+        // playback thread is created if no existing playback thread can be used
+        // connects playback thread output to sink device
+        Endpoint<PlaybackThread, PlaybackThread::PatchTrack> mPlayback;
+        // connects source device to record thread input
+        Endpoint<RecordThread, RecordThread::PatchRecord> mRecord;
     };
 
-private:
-    const wp<AudioFlinger>      mAudioFlinger;
-    SortedVector <Patch *>      mPatches;
+    sp<DeviceHalInterface> findHwDeviceByModule(audio_module_handle_t module);
+
+    AudioFlinger &mAudioFlinger;
+    std::map<audio_patch_handle_t, Patch> mPatches;
 };
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index a78be992..4d5f6b0 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -41,7 +41,7 @@
     virtual             ~Track();
     virtual status_t    initCheck() const;
 
-    static  void        appendDumpHeader(String8& result);
+            void        appendDumpHeader(String8& result);
             void        appendDump(String8& result, bool active);
     virtual status_t    start(AudioSystem::sync_event_t event =
                                     AudioSystem::SYNC_EVENT_NONE,
@@ -56,6 +56,12 @@
                 LOG_ALWAYS_FATAL_IF(mName >= 0 && name >= 0,
                         "%s both old name %d and new name %d are valid", __func__, mName, name);
                 mName = name;
+#ifdef TEE_SINK
+                mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+                        + "_" + std::to_string(mId)
+                        + "_" + std::to_string(mName)
+                        + "_T");
+#endif
             }
 
     virtual uint32_t    sampleRate() const;
@@ -65,10 +71,12 @@
             }
             bool        isOffloaded() const
                                 { return (mFlags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) != 0; }
-            bool        isDirect() const { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
+            bool        isDirect() const override
+                                { return (mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0; }
             bool        isOffloadedOrDirect() const { return (mFlags
                             & (AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD
                                     | AUDIO_OUTPUT_FLAG_DIRECT)) != 0; }
+            bool        isStatic() const { return  mSharedBuffer.get() != nullptr; }
 
             status_t    setParameters(const String8& keyValuePairs);
             status_t    attachAuxEffect(int EffectId);
@@ -87,6 +95,10 @@
 
     virtual bool        isFastTrack() const { return (mFlags & AUDIO_OUTPUT_FLAG_FAST) != 0; }
 
+            double      bufferLatencyMs() const override {
+                            return isStatic() ? 0. : TrackBase::bufferLatencyMs();
+                        }
+
 // implement volume handling.
     media::VolumeShaper::Status applyVolumeShaper(
                                 const sp<media::VolumeShaper::Configuration>& configuration,
@@ -140,7 +152,7 @@
     bool isResumePending();
     void resumeAck();
     void updateTrackFrameInfo(int64_t trackFramesReleased, int64_t sinkFramesWritten,
-            const ExtendedTimestamp &timeStamp);
+            uint32_t halSampleRate, const ExtendedTimestamp &timeStamp);
 
     sp<IMemory> sharedBuffer() const { return mSharedBuffer; }
 
@@ -233,7 +245,7 @@
                                     AudioSystem::SYNC_EVENT_NONE,
                              audio_session_t triggerSession = AUDIO_SESSION_NONE);
     virtual void        stop();
-            bool        write(void* data, uint32_t frames);
+            ssize_t     write(void* data, uint32_t frames);
             bool        bufferQueueEmpty() const { return mBufferQueue.size() == 0; }
             bool        isActive() const { return mActive; }
     const wp<ThreadBase>& thread() const { return mThread; }
@@ -241,6 +253,18 @@
             void        copyMetadataTo(MetadataInserter& backInserter) const override;
     /** Set the metadatas of the upstream tracks. Thread safe. */
             void        setMetadatas(const SourceMetadatas& metadatas);
+    /** returns client timestamp to the upstream duplicating thread. */
+    ExtendedTimestamp   getClientProxyTimestamp() const {
+                            // server - kernel difference is not true latency when drained
+                            // i.e. mServerProxy->isDrained().
+                            ExtendedTimestamp timestamp;
+                            (void) mClientProxy->getTimestamp(&timestamp);
+                            // On success, the timestamp LOCATION_SERVER and LOCATION_KERNEL
+                            // entries will be properly filled. If getTimestamp()
+                            // is unsuccessful, then a default initialized timestamp
+                            // (with mTimeNs[] filled with -1's) is returned.
+                            return timestamp;
+                        }
 
 private:
     status_t            obtainBuffer(AudioBufferProvider::Buffer* buffer,
@@ -257,6 +281,7 @@
     bool                        mActive;
     DuplicatingThread* const    mSourceThread; // for waitTimeMs() in write()
     sp<AudioTrackClientProxy>   mClientProxy;
+
     /** Attributes of the source tracks.
      *
      * This member must be accessed with mTrackMetadatasMutex taken.
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index fc2dbbb..b0c9fda 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -51,7 +51,7 @@
             bool        setOverflow() { bool tmp = mOverflow; mOverflow = true;
                                                 return tmp; }
 
-    static  void        appendDumpHeader(String8& result);
+            void        appendDumpHeader(String8& result);
             void        appendDump(String8& result, bool active);
 
             void        handleSyncStartEvent(const sp<SyncEvent>& event);
@@ -63,6 +63,8 @@
                                              const ExtendedTimestamp &timestamp);
 
     virtual bool        isFastTrack() const { return (mFlags & AUDIO_INPUT_FLAG_FAST) != 0; }
+            bool        isDirect() const override
+                                { return (mFlags & AUDIO_INPUT_FLAG_DIRECT) != 0; }
 
             void        setSilenced(bool silenced) { if (!isPatchTrack()) mSilenced = silenced; }
             bool        isSilenced() const { return mSilenced; }
diff --git a/services/audioflinger/ServiceUtilities.h b/services/audioflinger/ServiceUtilities.h
deleted file mode 100644
index f45ada1..0000000
--- a/services/audioflinger/ServiceUtilities.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#include <unistd.h>
-
-#include <binder/PermissionController.h>
-
-namespace android {
-
-extern pid_t getpid_cached;
-bool isTrustedCallingUid(uid_t uid);
-bool recordingAllowed(const String16& opPackageName, pid_t pid, uid_t uid);
-bool startRecording(const String16& opPackageName, pid_t pid, uid_t uid);
-void finishRecording(const String16& opPackageName, uid_t uid);
-bool captureAudioOutputAllowed(pid_t pid, uid_t uid);
-bool captureHotwordAllowed(pid_t pid, uid_t uid);
-bool settingsAllowed();
-bool modifyAudioRoutingAllowed();
-bool dumpAllowed();
-bool modifyPhoneStateAllowed(pid_t pid, uid_t uid);
-}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index dcad866..70af5c6 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -62,8 +62,8 @@
 #include "AudioFlinger.h"
 #include "FastMixer.h"
 #include "FastCapture.h"
-#include "ServiceUtilities.h"
-#include "mediautils/SchedulingPolicyService.h"
+#include <mediautils/SchedulingPolicyService.h>
+#include <mediautils/ServiceUtilities.h>
 
 #ifdef ADD_BATTERY_DATA
 #include <media/IMediaPlayerService.h>
@@ -200,7 +200,7 @@
 // Initially this heap is used to allocate client buffers for "fast" AudioRecord.
 // Eventually it will be the single buffer that FastCapture writes into via HAL read(),
 // and that all "fast" AudioRecord clients read from.  In either case, the size can be small.
-static const size_t kRecordThreadReadOnlyHeapSize = 0x4000;
+static const size_t kRecordThreadReadOnlyHeapSize = 0xD000;
 
 // ----------------------------------------------------------------------------
 
@@ -769,6 +769,8 @@
             if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_LEFT) s.append("top-back-left, ");
             if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_CENTER) s.append("top-back-center, " );
             if (mask & AUDIO_CHANNEL_OUT_TOP_BACK_RIGHT) s.append("top-back-right, " );
+            if (mask & AUDIO_CHANNEL_OUT_TOP_SIDE_LEFT) s.append("top-side-left, " );
+            if (mask & AUDIO_CHANNEL_OUT_TOP_SIDE_RIGHT) s.append("top-side-right, " );
             if (mask & ~AUDIO_CHANNEL_OUT_ALL) s.append("unknown,  ");
         } else {
             if (mask & AUDIO_CHANNEL_IN_LEFT) s.append("left, ");
@@ -783,6 +785,12 @@
             if (mask & AUDIO_CHANNEL_IN_X_AXIS) s.append("X, ");
             if (mask & AUDIO_CHANNEL_IN_Y_AXIS) s.append("Y, ");
             if (mask & AUDIO_CHANNEL_IN_Z_AXIS) s.append("Z, ");
+            if (mask & AUDIO_CHANNEL_IN_BACK_LEFT) s.append("back-left, ");
+            if (mask & AUDIO_CHANNEL_IN_BACK_RIGHT) s.append("back-right, ");
+            if (mask & AUDIO_CHANNEL_IN_CENTER) s.append("center, ");
+            if (mask & AUDIO_CHANNEL_IN_LOW_FREQUENCY) s.append("low freq, ");
+            if (mask & AUDIO_CHANNEL_IN_TOP_LEFT) s.append("top-left, " );
+            if (mask & AUDIO_CHANNEL_IN_TOP_RIGHT) s.append("top-right, " );
             if (mask & AUDIO_CHANNEL_IN_VOICE_UPLINK) s.append("voice-uplink, ");
             if (mask & AUDIO_CHANNEL_IN_VOICE_DNLINK) s.append("voice-dnlink, ");
             if (mask & ~AUDIO_CHANNEL_IN_ALL) s.append("unknown,  ");
@@ -845,6 +853,14 @@
     dprintf(fd, "  Input device: %#x (%s)\n", mInDevice, devicesToString(mInDevice).c_str());
     dprintf(fd, "  Audio source: %d (%s)\n", mAudioSource, sourceToString(mAudioSource));
 
+    // Dump timestamp statistics for the Thread types that support it.
+    if (mType == RECORD
+            || mType == MIXER
+            || mType == DUPLICATING
+            || (mType == DIRECT && audio_is_linear_pcm(mHALFormat))) {
+        dprintf(fd, "  Timestamp stats: %s\n", mTimestampVerifier.toString().c_str());
+    }
+
     if (locked) {
         mLock.unlock();
     }
@@ -1519,7 +1535,7 @@
     }
 }
 
-void AudioFlinger::ThreadBase::getAudioPortConfig(struct audio_port_config *config)
+void AudioFlinger::ThreadBase::toAudioPortConfig(struct audio_port_config *config)
 {
     config->type = AUDIO_PORT_TYPE_MIX;
     config->ext.mix.handle = mId;
@@ -1571,6 +1587,9 @@
     --mBatteryCounter[track->uid()].second;
     // mLatestActiveTrack is not cleared even if is the same as track.
     mHasChanged = true;
+#ifdef TEE_SINK
+    track->dumpTee(-1 /* fd */, "_REMOVE");
+#endif
     return index;
 }
 
@@ -1773,7 +1792,7 @@
     if (numtracks) {
         dprintf(fd, " of which %zu are active\n", numactive);
         result.append(prefix);
-        Track::appendDumpHeader(result);
+        mTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numtracks; ++i) {
             sp<Track> track = mTracks[i];
             if (track != 0) {
@@ -1793,7 +1812,7 @@
         result.append("  The following tracks are in the active list but"
                 " not in the track list\n");
         result.append(prefix);
-        Track::appendDumpHeader(result);
+        mActiveTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numactive; ++i) {
             sp<Track> track = mActiveTracks[i];
             if (mTracks.indexOf(track) < 0) {
@@ -1880,11 +1899,17 @@
     status_t lStatus;
     audio_output_flags_t outputFlags = mOutput->flags;
     audio_output_flags_t requestedFlags = *flags;
+    uint32_t sampleRate;
+
+    if (sharedBuffer != 0 && checkIMemory(sharedBuffer) != NO_ERROR) {
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
 
     if (*pSampleRate == 0) {
         *pSampleRate = mSampleRate;
     }
-    uint32_t sampleRate = *pSampleRate;
+    sampleRate = *pSampleRate;
 
     // special case for FAST flag considered OK if fast mixer is present
     if (hasFastMixer()) {
@@ -2845,6 +2870,9 @@
         ATRACE_END();
         if (framesWritten > 0) {
             bytesWritten = framesWritten * mFrameSize;
+#ifdef TEE_SINK
+            mTee.write((char *)mSinkBuffer + offset, framesWritten);
+#endif
         } else {
             bytesWritten = framesWritten;
         }
@@ -3185,16 +3213,23 @@
                 logString = NULL;
             }
 
+            // Collect timestamp statistics for the Playback Thread types that support it.
+            if (mType == MIXER
+                    || mType == DUPLICATING
+                    || (mType == DIRECT && audio_is_linear_pcm(mHALFormat))) { // no indentation
             // Gather the framesReleased counters for all active tracks,
             // and associate with the sink frames written out.  We need
             // this to convert the sink timestamp to the track timestamp.
             bool kernelLocationUpdate = false;
-            if (mNormalSink != 0) {
-                // Note: The DuplicatingThread may not have a mNormalSink.
+            ExtendedTimestamp timestamp; // use private copy to fetch
+            if (mStandby) {
+                mTimestampVerifier.discontinuity();
+            } else if (threadloop_getHalTimestamp_l(&timestamp) == OK) {
+                mTimestampVerifier.add(timestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
+                        timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
+                        mSampleRate);
                 // We always fetch the timestamp here because often the downstream
                 // sink will block while writing.
-                ExtendedTimestamp timestamp; // use private copy to fetch
-                (void) mNormalSink->getTimestamp(timestamp);
 
                 // We keep track of the last valid kernel position in case we are in underrun
                 // and the normal mixer period is the same as the fast mixer period, or there
@@ -3223,7 +3258,10 @@
                         + mSuspendedFrames; // add frames discarded when suspended
                 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
                         timestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+            } else {
+                mTimestampVerifier.error();
             }
+
             // mFramesWritten for non-offloaded tracks are contiguous
             // even after standby() is called. This is useful for the track frame
             // to sink frame mapping.
@@ -3251,10 +3289,12 @@
                         t->updateTrackFrameInfo(
                                 t->mAudioTrackServerProxy->framesReleased(),
                                 mFramesWritten,
+                                mSampleRate,
                                 mTimestamp);
                     }
                 }
             }
+            } // if (mType ... ) { // no indentation
 #if 0
             // logFormat example
             if (z % 100 == 0) {
@@ -3772,12 +3812,16 @@
     destroyTrack_l(track);
 }
 
-void AudioFlinger::PlaybackThread::getAudioPortConfig(struct audio_port_config *config)
+void AudioFlinger::PlaybackThread::toAudioPortConfig(struct audio_port_config *config)
 {
-    ThreadBase::getAudioPortConfig(config);
+    ThreadBase::toAudioPortConfig(config);
     config->role = AUDIO_PORT_ROLE_SOURCE;
     config->ext.mix.hw_module = mOutput->audioHwDev->handle();
     config->ext.mix.usecase.stream = AUDIO_STREAM_DEFAULT;
+    if (mOutput && mOutput->flags != AUDIO_OUTPUT_FLAG_NONE) {
+        config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+        config->flags.output = mOutput->flags;
+    }
 }
 
 // ----------------------------------------------------------------------------
@@ -3858,9 +3902,7 @@
 
         // create a MonoPipe to connect our submix to FastMixer
         NBAIO_Format format = mOutputSink->format();
-#ifdef TEE_SINK
-        NBAIO_Format origformat = format;
-#endif
+
         // adjust format to match that of the Fast Mixer
         ALOGV("format changed from %#x to %#x", format.mFormat, fastMixerFormat);
         format.mFormat = fastMixerFormat;
@@ -3872,7 +3914,7 @@
         MonoPipe *monoPipe = new MonoPipe(mNormalFrameCount * 4, format, true /*writeCanBlock*/);
         const NBAIO_Format offers[1] = {format};
         size_t numCounterOffers = 0;
-#if !LOG_NDEBUG || defined(TEE_SINK)
+#if !LOG_NDEBUG
         ssize_t index =
 #else
         (void)
@@ -3883,25 +3925,8 @@
                 (monoPipe->maxFrames() * 7) / 8 : mNormalFrameCount * 2);
         mPipeSink = monoPipe;
 
-#ifdef TEE_SINK
-        if (mTeeSinkOutputEnabled) {
-            // create a Pipe to archive a copy of FastMixer's output for dumpsys
-            Pipe *teeSink = new Pipe(mTeeSinkOutputFrames, origformat);
-            const NBAIO_Format offers2[1] = {origformat};
-            numCounterOffers = 0;
-            index = teeSink->negotiate(offers2, 1, NULL, numCounterOffers);
-            ALOG_ASSERT(index == 0);
-            mTeeSink = teeSink;
-            PipeReader *teeSource = new PipeReader(*teeSink);
-            numCounterOffers = 0;
-            index = teeSource->negotiate(offers2, 1, NULL, numCounterOffers);
-            ALOG_ASSERT(index == 0);
-            mTeeSource = teeSource;
-        }
-#endif
-
         // create fast mixer and configure it initially with just one fast track for our submix
-        mFastMixer = new FastMixer();
+        mFastMixer = new FastMixer(mId);
         FastMixerStateQueue *sq = mFastMixer->sq();
 #ifdef STATE_QUEUE_DUMP
         sq->setObserverDump(&mStateQueueObserverDump);
@@ -3927,9 +3952,6 @@
         state->mColdFutexAddr = &mFastMixerFutex;
         state->mColdGen++;
         state->mDumpState = &mFastMixerDumpState;
-#ifdef TEE_SINK
-        state->mTeeSink = mTeeSink.get();
-#endif
         mFastMixerNBLogWriter = audioFlinger->newWriter_l(kFastMixerLogSize, "FastMixer");
         state->mNBLogWriter = mFastMixerNBLogWriter.get();
         sq->end();
@@ -3938,7 +3960,7 @@
         // start the fast mixer
         mFastMixer->run("FastMixer", PRIORITY_URGENT_AUDIO);
         pid_t tid = mFastMixer->getTid();
-        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer, false /*forApp*/);
+        sendPrioConfigEvent(getpid(), tid, kPriorityFastMixer, false /*forApp*/);
         stream()->setHalThreadPriority(kPriorityFastMixer);
 
 #ifdef AUDIO_WATCHDOG
@@ -3947,9 +3969,14 @@
         mAudioWatchdog->setDump(&mAudioWatchdogDump);
         mAudioWatchdog->run("AudioWatchdog", PRIORITY_URGENT_AUDIO);
         tid = mAudioWatchdog->getTid();
-        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastMixer, false /*forApp*/);
+        sendPrioConfigEvent(getpid(), tid, kPriorityFastMixer, false /*forApp*/);
 #endif
-
+    } else {
+#ifdef TEE_SINK
+        // Only use the MixerThread tee if there is no FastMixer.
+        mTee.set(mOutputSink->format(), NBAIO_Tee::TEE_FLAG_OUTPUT_THREAD);
+        mTee.setId(std::string("_") + std::to_string(mId) + "_M");
+#endif
     }
 
     switch (kUseFastMixer) {
@@ -4265,6 +4292,37 @@
     mMixerBufferValid = false;  // mMixerBuffer has no valid data until appropriate tracks found.
     mEffectBufferValid = false; // mEffectBuffer has no valid data until tracks found.
 
+    // DeferredOperations handles statistics after setting mixerStatus.
+    class DeferredOperations {
+    public:
+        DeferredOperations(mixer_state *mixerStatus)
+            : mMixerStatus(mixerStatus) { }
+
+        // when leaving scope, tally frames properly.
+        ~DeferredOperations() {
+            // Tally underrun frames only if we are actually mixing (MIXER_TRACKS_READY)
+            // because that is when the underrun occurs.
+            // We do not distinguish between FastTracks and NormalTracks here.
+            if (*mMixerStatus == MIXER_TRACKS_READY) {
+                for (const auto &underrun : mUnderrunFrames) {
+                    underrun.first->mAudioTrackServerProxy->tallyUnderrunFrames(
+                            underrun.second);
+                }
+            }
+        }
+
+        // tallyUnderrunFrames() is called to update the track counters
+        // with the number of underrun frames for a particular mixer period.
+        // We defer tallying until we know the final mixer status.
+        void tallyUnderrunFrames(sp<Track> track, size_t underrunFrames) {
+            mUnderrunFrames.emplace_back(track, underrunFrames);
+        }
+
+    private:
+        const mixer_state * const mMixerStatus;
+        std::vector<std::pair<sp<Track>, size_t>> mUnderrunFrames;
+    } deferredOperations(&mixerStatus); // implicit nested scope for variable capture
+
     for (size_t i=0 ; i<count ; i++) {
         const sp<Track> t = mActiveTracks[i];
 
@@ -4299,13 +4357,14 @@
             track->mObservedUnderruns = underruns;
             // don't count underruns that occur while stopping or pausing
             // or stopped which can occur when flush() is called while active
+            size_t underrunFrames = 0;
             if (!(track->isStopping() || track->isPausing() || track->isStopped()) &&
                     recentUnderruns > 0) {
                 // FIXME fast mixer will pull & mix partial buffers, but we count as a full underrun
-                track->mAudioTrackServerProxy->tallyUnderrunFrames(recentUnderruns * mFrameCount);
-            } else {
-                track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
+                underrunFrames = recentUnderruns * mFrameCount;
             }
+            // Immediately account for FastTrack underruns.
+            track->mAudioTrackServerProxy->tallyUnderrunFrames(underrunFrames);
 
             // This is similar to the state machine for normal tracks,
             // with a few modifications for fast tracks.
@@ -4720,13 +4779,13 @@
                 mixerStatus = MIXER_TRACKS_READY;
             }
         } else {
+            size_t underrunFrames = 0;
             if (framesReady < desiredFrames && !track->isStopped() && !track->isPaused()) {
                 ALOGV("track(%p) underrun,  framesReady(%zu) < framesDesired(%zd)",
                         track, framesReady, desiredFrames);
-                track->mAudioTrackServerProxy->tallyUnderrunFrames(desiredFrames);
-            } else {
-                track->mAudioTrackServerProxy->tallyUnderrunFrames(0);
+                underrunFrames = desiredFrames;
             }
+            deferredOperations.tallyUnderrunFrames(track, underrunFrames);
 
             // clear effect chain input buffer if an active track underruns to avoid sending
             // previous audio buffer again to effects
@@ -5025,6 +5084,12 @@
     dprintf(fd, "  Thread throttle time (msecs): %u\n", mThreadThrottleTimeMs);
     dprintf(fd, "  AudioMixer tracks: %s\n", mAudioMixer->trackNames().c_str());
     dprintf(fd, "  Master mono: %s\n", mMasterMono ? "on" : "off");
+    const double latencyMs = mTimestamp.getOutputServerLatencyMs(mSampleRate);
+    if (latencyMs != 0.) {
+        dprintf(fd, "  NormalMixer latency ms: %.2lf\n", latencyMs);
+    } else {
+        dprintf(fd, "  NormalMixer latency ms: unavail\n");
+    }
 
     if (hasFastMixer()) {
         dprintf(fd, "  FastMixer thread %p tid=%d", mFastMixer.get(), mFastMixer->getTid());
@@ -5056,12 +5121,6 @@
     } else {
         dprintf(fd, "  No FastMixer\n");
     }
-
-#ifdef TEE_SINK
-    // Write the tee output to a .wav file
-    dumpTee(fd, mTeeSource, mId, 'M');
-#endif
-
 }
 
 uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
@@ -6062,7 +6121,22 @@
 ssize_t AudioFlinger::DuplicatingThread::threadLoop_write()
 {
     for (size_t i = 0; i < outputTracks.size(); i++) {
-        outputTracks[i]->write(mSinkBuffer, writeFrames);
+        const ssize_t actualWritten = outputTracks[i]->write(mSinkBuffer, writeFrames);
+
+        // Consider the first OutputTrack for timestamp and frame counting.
+
+        // The threadLoop() generally assumes writing a full sink buffer size at a time.
+        // Here, we correct for writeFrames of 0 (a stop) or underruns because
+        // we always claim success.
+        if (i == 0) {
+            const ssize_t correction = mSinkBufferSize / mFrameSize - actualWritten;
+            ALOGD_IF(correction != 0 && writeFrames != 0,
+                    "%s: writeFrames:%u  actualWritten:%zd  correction:%zd  mFramesWritten:%lld",
+                    __func__, writeFrames, actualWritten, correction, (long long)mFramesWritten);
+            mFramesWritten -= correction;
+        }
+
+        // TODO: Report correction for the other output tracks and show in the dump.
     }
     mStandby = false;
     return (ssize_t)mSinkBufferSize;
@@ -6227,9 +6301,6 @@
                                          audio_devices_t outDevice,
                                          audio_devices_t inDevice,
                                          bool systemReady
-#ifdef TEE_SINK
-                                         , const sp<NBAIO_Sink>& teeSink
-#endif
                                          ) :
     ThreadBase(audioFlinger, id, outDevice, inDevice, RECORD, systemReady),
     mInput(input),
@@ -6237,9 +6308,6 @@
     mRsmpInBuffer(NULL),
     // mRsmpInFrames, mRsmpInFramesP2, and mRsmpInFramesOA are set by readInputParameters_l()
     mRsmpInRear(0)
-#ifdef TEE_SINK
-    , mTeeSink(teeSink)
-#endif
     , mReadOnlyHeap(new MemoryDealer(kRecordThreadReadOnlyHeapSize,
             "RecordThreadRO", MemoryHeapBase::READ_ONLY))
     // mFastCapture below
@@ -6354,7 +6422,7 @@
         // start the fast capture
         mFastCapture->run("FastCapture", ANDROID_PRIORITY_URGENT_AUDIO);
         pid_t tid = mFastCapture->getTid();
-        sendPrioConfigEvent(getpid_cached, tid, kPriorityFastCapture, false /*forApp*/);
+        sendPrioConfigEvent(getpid(), tid, kPriorityFastCapture, false /*forApp*/);
         stream()->setHalThreadPriority(kPriorityFastCapture);
 #ifdef AUDIO_WATCHDOG
         // FIXME
@@ -6362,6 +6430,10 @@
 
         mFastTrackAvail = true;
     }
+#ifdef TEE_SINK
+    mTee.set(mInputSource->format(), NBAIO_Tee::TEE_FLAG_INPUT_THREAD);
+    mTee.setId(std::string("_") + std::to_string(mId) + "_C");
+#endif
 failed: ;
 
     // FIXME mNormalSource
@@ -6679,8 +6751,9 @@
         // Update server timestamp with kernel stats
         if (mPipeSource.get() == nullptr /* don't obtain for FastCapture, could block */) {
             int64_t position, time;
-            int ret = mInput->stream->getCapturePosition(&position, &time);
-            if (ret == NO_ERROR) {
+            if (mStandby) {
+                mTimestampVerifier.discontinuity();
+            } else if (mInput->stream->getCapturePosition(&position, &time) == NO_ERROR) {
                 mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL] = position;
                 mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] = time;
                 // Note: In general record buffers should tend to be empty in
@@ -6688,6 +6761,12 @@
                 //
                 // Also, it is not advantageous to call get_presentation_position during the read
                 // as the read obtains a lock, preventing the timestamp call from executing.
+
+                mTimestampVerifier.add(mTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL],
+                        mTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL],
+                        mSampleRate);
+            } else {
+                mTimestampVerifier.error();
             }
         }
         // Use this to track timestamp information
@@ -6704,9 +6783,9 @@
         }
         ALOG_ASSERT(framesRead > 0);
 
-        if (mTeeSink != 0) {
-            (void) mTeeSink->write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
-        }
+#ifdef TEE_SINK
+        (void)mTee.write((uint8_t*)mRsmpInBuffer + rear * mFrameSize, framesRead);
+#endif
         // If destination is non-contiguous, we now correct for reading past end of buffer.
         {
             size_t part1 = mRsmpInFramesP2 - rear;
@@ -6764,9 +6843,33 @@
                 framesOut = min(framesOut,
                         destinationFramesPossible(
                                 framesIn, mSampleRate, activeTrack->mSampleRate));
-                // process frames from the RecordThread buffer provider to the RecordTrack buffer
-                framesOut = activeTrack->mRecordBufferConverter->convert(
-                        activeTrack->mSink.raw, activeTrack->mResamplerBufferProvider, framesOut);
+
+                if (activeTrack->isDirect()) {
+                    // No RecordBufferConverter used for compressed formats. Pass
+                    // straight from RecordThread buffer to RecordTrack buffer.
+                    AudioBufferProvider::Buffer buffer;
+                    buffer.frameCount = framesOut;
+                    status_t status = activeTrack->mResamplerBufferProvider->getNextBuffer(&buffer);
+                    if (status == OK && buffer.frameCount != 0) {
+                        ALOGV_IF(buffer.frameCount != framesOut,
+                                "%s() read less than expected (%zu vs %zu)",
+                                __func__, buffer.frameCount, framesOut);
+                        framesOut = buffer.frameCount;
+                        memcpy(activeTrack->mSink.raw, buffer.raw, buffer.frameCount);
+                        activeTrack->mResamplerBufferProvider->releaseBuffer(&buffer);
+                    } else {
+                        framesOut = 0;
+                        ALOGE("%s() cannot fill request, status: %d, frameCount: %zu",
+                            __func__, status, buffer.frameCount);
+                    }
+                } else {
+                    // process frames from the RecordThread buffer provider to the RecordTrack
+                    // buffer
+                    framesOut = activeTrack->mRecordBufferConverter->convert(
+                            activeTrack->mSink.raw,
+                            activeTrack->mResamplerBufferProvider,
+                            framesOut);
+                }
 
                 if (framesOut > 0 && (overrun == OVERRUN_UNKNOWN)) {
                     overrun = OVERRUN_FALSE;
@@ -7138,8 +7241,10 @@
         // see previously buffered data before it called start(), but with greater risk of overrun.
 
         recordTrack->mResamplerBufferProvider->reset();
-        // clear any converter state as new data will be discontinuous
-        recordTrack->mRecordBufferConverter->reset();
+        if (!recordTrack->isDirect()) {
+            // clear any converter state as new data will be discontinuous
+            recordTrack->mRecordBufferConverter->reset();
+        }
         recordTrack->mState = TrackBase::STARTING_2;
         // signal thread to start
         mWaitWorkCV.broadcast();
@@ -7304,6 +7409,13 @@
         (void)input->stream->dump(fd);
     }
 
+    const double latencyMs = - mTimestamp.getOutputServerLatencyMs(mSampleRate);
+    if (latencyMs != 0.) {
+        dprintf(fd, "  NormalRecord latency ms: %.2lf\n", latencyMs);
+    } else {
+        dprintf(fd, "  NormalRecord latency ms: unavail\n");
+    }
+
     dprintf(fd, "  Fast capture thread: %s\n", hasFastCapture() ? "yes" : "no");
     dprintf(fd, "  Fast track available: %s\n", mFastTrackAvail ? "yes" : "no");
 
@@ -7327,7 +7439,7 @@
     if (numtracks) {
         dprintf(fd, " of which %zu are active\n", numactive);
         result.append(prefix);
-        RecordTrack::appendDumpHeader(result);
+        mTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numtracks ; ++i) {
             sp<RecordTrack> track = mTracks[i];
             if (track != 0) {
@@ -7347,7 +7459,7 @@
         result.append("  The following tracks are in the active list but"
                 " not in the track list\n");
         result.append(prefix);
-        RecordTrack::appendDumpHeader(result);
+        mActiveTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numactive; ++i) {
             sp<RecordTrack> track = mActiveTracks[i];
             if (mTracks.indexOf(track) < 0) {
@@ -7855,24 +7967,28 @@
     return status;
 }
 
-void AudioFlinger::RecordThread::addPatchRecord(const sp<PatchRecord>& record)
+void AudioFlinger::RecordThread::addPatchTrack(const sp<PatchRecord>& record)
 {
     Mutex::Autolock _l(mLock);
     mTracks.add(record);
 }
 
-void AudioFlinger::RecordThread::deletePatchRecord(const sp<PatchRecord>& record)
+void AudioFlinger::RecordThread::deletePatchTrack(const sp<PatchRecord>& record)
 {
     Mutex::Autolock _l(mLock);
     destroyTrack_l(record);
 }
 
-void AudioFlinger::RecordThread::getAudioPortConfig(struct audio_port_config *config)
+void AudioFlinger::RecordThread::toAudioPortConfig(struct audio_port_config *config)
 {
-    ThreadBase::getAudioPortConfig(config);
+    ThreadBase::toAudioPortConfig(config);
     config->role = AUDIO_PORT_ROLE_SINK;
     config->ext.mix.hw_module = mInput->audioHwDev->handle();
     config->ext.mix.usecase.source = mAudioSource;
+    if (mInput && mInput->flags != AUDIO_INPUT_FLAG_NONE) {
+        config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+        config->flags.input = mInput->flags;
+    }
 }
 
 // ----------------------------------------------------------------------------
@@ -8108,7 +8224,7 @@
 
     // Given that MmapThread::mAttr is mutable, should a MmapTrack have attributes ?
     sp<MmapTrack> track = new MmapTrack(this, mAttr, mSampleRate, mFormat, mChannelMask, mSessionId,
-                                        client.clientUid, client.clientPid, portId);
+                                        isOutput(), client.clientUid, client.clientPid, portId);
 
     track->setSilenced_l(silenced);
     mActiveTracks.add(track);
@@ -8454,9 +8570,9 @@
     return status;
 }
 
-void AudioFlinger::MmapThread::getAudioPortConfig(struct audio_port_config *config)
+void AudioFlinger::MmapThread::toAudioPortConfig(struct audio_port_config *config)
 {
-    ThreadBase::getAudioPortConfig(config);
+    ThreadBase::toAudioPortConfig(config);
     if (isOutput()) {
         config->role = AUDIO_PORT_ROLE_SOURCE;
         config->ext.mix.hw_module = mAudioHwDev->handle();
@@ -8634,7 +8750,7 @@
     const char *prefix = "    ";
     if (numtracks) {
         result.append(prefix);
-        MmapTrack::appendDumpHeader(result);
+        mActiveTracks[0]->appendDumpHeader(result);
         for (size_t i = 0; i < numtracks ; ++i) {
             sp<MmapTrack> track = mActiveTracks[i];
             result.append(prefix);
@@ -8840,6 +8956,15 @@
     }
 }
 
+void AudioFlinger::MmapPlaybackThread::toAudioPortConfig(struct audio_port_config *config)
+{
+    MmapThread::toAudioPortConfig(config);
+    if (mOutput && mOutput->flags != AUDIO_OUTPUT_FLAG_NONE) {
+        config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+        config->flags.output = mOutput->flags;
+    }
+}
+
 void AudioFlinger::MmapPlaybackThread::dumpInternals(int fd, const Vector<String16>& args)
 {
     MmapThread::dumpInternals(fd, args);
@@ -8930,4 +9055,13 @@
     }
 }
 
+void AudioFlinger::MmapCaptureThread::toAudioPortConfig(struct audio_port_config *config)
+{
+    MmapThread::toAudioPortConfig(config);
+    if (mInput && mInput->flags != AUDIO_INPUT_FLAG_NONE) {
+        config->config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+        config->flags.input = mInput->flags;
+    }
+}
+
 } // namespace android
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 28d4482..064e291 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -281,7 +281,7 @@
     virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
                                                audio_patch_handle_t *handle) = 0;
     virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle) = 0;
-    virtual     void        getAudioPortConfig(struct audio_port_config *config) = 0;
+    virtual     void        toAudioPortConfig(struct audio_port_config *config) = 0;
 
 
                 // see note at declaration of mStandby, mOutDevice and mInDevice
@@ -434,6 +434,12 @@
     virtual     void        setMasterMono_l(bool mono __unused) { }
     virtual     bool        requireMonoBlend() { return false; }
 
+                            // called within the threadLoop to obtain timestamp from the HAL.
+    virtual     status_t    threadloop_getHalTimestamp_l(
+                                    ExtendedTimestamp *timestamp __unused) const {
+                                return INVALID_OPERATION;
+                            }
+
     friend class AudioFlinger;      // for mEffectChains
 
                 const type_t            mType;
@@ -493,10 +499,16 @@
                 sp<NBLog::Writer>       mNBLogWriter;
                 bool                    mSystemReady;
                 ExtendedTimestamp       mTimestamp;
+                TimestampVerifier< // For timestamp statistics.
+                        int64_t /* frame count */, int64_t /* time ns */> mTimestampVerifier;
+
                 // A condition that must be evaluated by the thread loop has changed and
                 // we must not wait for async write callback in the thread loop before evaluating it
                 bool                    mSignalPending;
 
+#ifdef TEE_SINK
+                NBAIO_Tee               mTee;
+#endif
                 // ActiveTracks is a sorted vector of track type T representing the
                 // active tracks of threadLoop() to be considered by the locked prepare portion.
                 // ActiveTracks should be accessed with the ThreadBase lock held.
@@ -782,7 +794,7 @@
                 void        addPatchTrack(const sp<PatchTrack>& track);
                 void        deletePatchTrack(const sp<PatchTrack>& track);
 
-    virtual     void        getAudioPortConfig(struct audio_port_config *config);
+    virtual     void        toAudioPortConfig(struct audio_port_config *config);
 
                 // Return the asynchronous signal wait time.
     virtual     int64_t     computeWaitTimeNs_l() const { return INT64_MAX; }
@@ -1054,11 +1066,6 @@
     sp<NBAIO_Sink>          mPipeSink;
     // The current sink for the normal mixer to write it's (sub)mix, mOutputSink or mPipeSink
     sp<NBAIO_Sink>          mNormalSink;
-#ifdef TEE_SINK
-    // For dumpsys
-    sp<NBAIO_Sink>          mTeeSink;
-    sp<NBAIO_Source>        mTeeSource;
-#endif
     uint32_t                mScreenState;   // cached copy of gScreenState
     // TODO: add comment and adjust size as needed
     static const size_t     kFastMixerLogSize = 8 * 1024;
@@ -1152,6 +1159,14 @@
                               return mFastMixerDumpState.mTracks[fastIndex].mUnderruns;
                             }
 
+                status_t    threadloop_getHalTimestamp_l(
+                                    ExtendedTimestamp *timestamp) const override {
+                                if (mNormalSink.get() != nullptr) {
+                                    return mNormalSink->getTimestamp(*timestamp);
+                                }
+                                return INVALID_OPERATION;
+                            }
+
 protected:
     virtual     void       setMasterMono_l(bool mono) {
                                mMasterMono.store(mono);
@@ -1316,6 +1331,22 @@
     SortedVector < sp<OutputTrack> >  mOutputTracks;
 public:
     virtual     bool        hasFastMixer() const { return false; }
+                status_t    threadloop_getHalTimestamp_l(
+                                    ExtendedTimestamp *timestamp) const override {
+        if (mOutputTracks.size() > 0) {
+            // forward the first OutputTrack's kernel information for timestamp.
+            const ExtendedTimestamp trackTimestamp =
+                    mOutputTracks[0]->getClientProxyTimestamp();
+            if (trackTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] > 0) {
+                timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL] =
+                        trackTimestamp.mTimeNs[ExtendedTimestamp::LOCATION_KERNEL];
+                timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL] =
+                        trackTimestamp.mPosition[ExtendedTimestamp::LOCATION_KERNEL];
+                return OK;  // discard server timestamp - that's ignored.
+            }
+        }
+        return INVALID_OPERATION;
+    }
 };
 
 // record thread
@@ -1374,9 +1405,6 @@
                     audio_devices_t outDevice,
                     audio_devices_t inDevice,
                     bool systemReady
-#ifdef TEE_SINK
-                    , const sp<NBAIO_Sink>& teeSink
-#endif
                     );
             virtual     ~RecordThread();
 
@@ -1437,8 +1465,8 @@
                                            audio_patch_handle_t *handle);
     virtual status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
 
-            void        addPatchRecord(const sp<PatchRecord>& record);
-            void        deletePatchRecord(const sp<PatchRecord>& record);
+            void        addPatchTrack(const sp<PatchRecord>& record);
+            void        deletePatchTrack(const sp<PatchRecord>& record);
 
             void        readInputParameters_l();
     virtual uint32_t    getInputFramesLost();
@@ -1459,7 +1487,7 @@
 
     virtual size_t      frameCount() const { return mFrameCount; }
             bool        hasFastCapture() const { return mFastCapture != 0; }
-    virtual void        getAudioPortConfig(struct audio_port_config *config);
+    virtual void        toAudioPortConfig(struct audio_port_config *config);
 
     virtual status_t    checkEffectCompatibility_l(const effect_descriptor_t *desc,
                                                    audio_session_t sessionId);
@@ -1506,8 +1534,6 @@
             int32_t                             mRsmpInRear;    // last filled frame + 1
 
             // For dumpsys
-            const sp<NBAIO_Sink>                mTeeSink;
-
             const sp<MemoryDealer>              mReadOnlyHeap;
 
             // one-time initialization, no locks required
@@ -1602,7 +1628,7 @@
     virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
                                                audio_patch_handle_t *handle);
     virtual     status_t    releaseAudioPatch_l(const audio_patch_handle_t handle);
-    virtual     void        getAudioPortConfig(struct audio_port_config *config);
+    virtual     void        toAudioPortConfig(struct audio_port_config *config);
 
     virtual     sp<StreamHalInterface> stream() const { return mHalStream; }
     virtual     status_t    addEffectChain_l(const sp<EffectChain>& chain);
@@ -1686,6 +1712,8 @@
 
                 void        updateMetadata_l() override;
 
+    virtual     void        toAudioPortConfig(struct audio_port_config *config);
+
 protected:
 
                 audio_stream_type_t         mStreamType;
@@ -1714,6 +1742,8 @@
                 void           processVolume_l() override;
                 void           setRecordSilenced(uid_t uid, bool silenced) override;
 
+    virtual     void           toAudioPortConfig(struct audio_port_config *config);
+
 protected:
 
                 AudioStreamIn*  mInput;
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index ccfb69f..95da9d7 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -91,6 +91,7 @@
             void*       buffer() const { return mBuffer; }
             size_t      bufferSize() const { return mBufferSize; }
     virtual bool        isFastTrack() const = 0;
+    virtual bool        isDirect() const = 0;
             bool        isOutputTrack() const { return (mType == TYPE_OUTPUT); }
             bool        isPatchTrack() const { return (mType == TYPE_PATCH); }
             bool        isExternalTrack() const { return !isOutputTrack() && !isPatchTrack(); }
@@ -100,6 +101,92 @@
 
     audio_attributes_t  attributes() const { return mAttr; }
 
+#ifdef TEE_SINK
+           void         dumpTee(int fd, const std::string &reason) const {
+                                mTee.dump(fd, reason);
+                        }
+#endif
+
+            /** returns the buffer contents size converted to time in milliseconds
+             * for PCM Playback or Record streaming tracks. The return value is zero for
+             * PCM static tracks and not defined for non-PCM tracks.
+             *
+             * This may be called without the thread lock.
+             */
+    virtual double      bufferLatencyMs() const {
+                            return mServerProxy->framesReadySafe() * 1000 / sampleRate();
+                        }
+
+            /** returns whether the track supports server latency computation.
+             * This is set in the constructor and constant throughout the track lifetime.
+             */
+
+            bool        isServerLatencySupported() const { return mServerLatencySupported; }
+
+            /** computes the server latency for PCM Playback or Record track
+             * to the device sink/source.  This is the time for the next frame in the track buffer
+             * written or read from the server thread to the device source or sink.
+             *
+             * This may be called without the thread lock, but latencyMs and fromTrack
+             * may be not be synchronized. For example PatchPanel may not obtain the
+             * thread lock before calling.
+             *
+             * \param latencyMs on success is set to the latency in milliseconds of the
+             *        next frame written/read by the server thread to/from the track buffer
+             *        from the device source/sink.
+             * \param fromTrack on success is set to true if latency was computed directly
+             *        from the track timestamp; otherwise set to false if latency was
+             *        estimated from the server timestamp.
+             *        fromTrack may be nullptr or omitted if not required.
+             *
+             * \returns OK or INVALID_OPERATION on failure.
+             */
+            status_t    getServerLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
+                            if (!isServerLatencySupported()) {
+                                return INVALID_OPERATION;
+                            }
+
+                            // if no thread lock is acquired, these atomics are not
+                            // synchronized with each other, considered a benign race.
+
+                            const double serverLatencyMs = mServerLatencyMs.load();
+                            if (serverLatencyMs == 0.) {
+                                return INVALID_OPERATION;
+                            }
+                            if (fromTrack != nullptr) {
+                                *fromTrack = mServerLatencyFromTrack.load();
+                            }
+                            *latencyMs = serverLatencyMs;
+                            return OK;
+                        }
+
+            /** computes the total client latency for PCM Playback or Record tracks
+             * for the next client app access to the device sink/source; i.e. the
+             * server latency plus the buffer latency.
+             *
+             * This may be called without the thread lock, but latencyMs and fromTrack
+             * may be not be synchronized. For example PatchPanel may not obtain the
+             * thread lock before calling.
+             *
+             * \param latencyMs on success is set to the latency in milliseconds of the
+             *        next frame written/read by the client app to/from the track buffer
+             *        from the device sink/source.
+             * \param fromTrack on success is set to true if latency was computed directly
+             *        from the track timestamp; otherwise set to false if latency was
+             *        estimated from the server timestamp.
+             *        fromTrack may be nullptr or omitted if not required.
+             *
+             * \returns OK or INVALID_OPERATION on failure.
+             */
+            status_t    getTrackLatencyMs(double *latencyMs, bool *fromTrack = nullptr) const {
+                            double serverLatencyMs;
+                            status_t status = getServerLatencyMs(&serverLatencyMs, fromTrack);
+                            if (status == OK) {
+                                *latencyMs = serverLatencyMs + bufferLatencyMs();
+                            }
+                            return status;
+                        }
+
 protected:
     DISALLOW_COPY_AND_ASSIGN(TrackBase);
 
@@ -208,13 +295,18 @@
     const bool          mIsOut;
     sp<ServerProxy>     mServerProxy;
     const int           mId;
-    sp<NBAIO_Sink>      mTeeSink;
-    sp<NBAIO_Source>    mTeeSource;
+#ifdef TEE_SINK
+    NBAIO_Tee           mTee;
+#endif
     bool                mTerminated;
     track_type          mType;      // must be one of TYPE_DEFAULT, TYPE_OUTPUT, TYPE_PATCH ...
     audio_io_handle_t   mThreadIoHandle; // I/O handle of the thread the track is attached to
     audio_port_handle_t mPortId; // unique ID for this track used by audio policy
     bool                mIsInvalid; // non-resettable latch, set by invalidate()
+
+    bool                mServerLatencySupported = false;
+    std::atomic<bool>   mServerLatencyFromTrack{}; // latency from track or server timestamp.
+    std::atomic<double> mServerLatencyMs{};        // last latency pushed from server thread.
 };
 
 // PatchProxyBufferProvider interface is implemented by PatchTrack and PatchRecord.
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index a7c4253..22e610e 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -28,11 +28,11 @@
 #include <private/media/AudioTrackShared.h>
 
 #include "AudioFlinger.h"
-#include "ServiceUtilities.h"
 
 #include <media/nbaio/Pipe.h>
 #include <media/nbaio/PipeReader.h>
 #include <media/RecordBufferConverter.h>
+#include <mediautils/ServiceUtilities.h>
 #include <audio_utils/minifloat.h>
 
 // ----------------------------------------------------------------------------
@@ -102,7 +102,7 @@
         mIsInvalid(false)
 {
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    if (!isTrustedCallingUid(callingUid) || clientUid == AUDIO_UID_INVALID) {
+    if (!isAudioServerOrMediaServerUid(callingUid) || clientUid == AUDIO_UID_INVALID) {
         ALOGW_IF(clientUid != AUDIO_UID_INVALID && clientUid != callingUid,
                 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, clientUid);
         clientUid = callingUid;
@@ -210,22 +210,7 @@
         mBufferSize = bufferSize;
 
 #ifdef TEE_SINK
-        if (mTeeSinkTrackEnabled) {
-            NBAIO_Format pipeFormat = Format_from_SR_C(mSampleRate, mChannelCount, mFormat);
-            if (Format_isValid(pipeFormat)) {
-                Pipe *pipe = new Pipe(mTeeSinkTrackFrames, pipeFormat);
-                size_t numCounterOffers = 0;
-                const NBAIO_Format offers[1] = {pipeFormat};
-                ssize_t index = pipe->negotiate(offers, 1, NULL, numCounterOffers);
-                ALOG_ASSERT(index == 0);
-                PipeReader *pipeReader = new PipeReader(*pipe);
-                numCounterOffers = 0;
-                index = pipeReader->negotiate(offers, 1, NULL, numCounterOffers);
-                ALOG_ASSERT(index == 0);
-                mTeeSink = pipe;
-                mTeeSource = pipeReader;
-            }
-        }
+        mTee.set(sampleRate, mChannelCount, format, NBAIO_Tee::TEE_FLAG_TRACK);
 #endif
 
     }
@@ -244,9 +229,6 @@
 
 AudioFlinger::ThreadBase::TrackBase::~TrackBase()
 {
-#ifdef TEE_SINK
-    dumpTee(-1, mTeeSource, mId, 'T');
-#endif
     // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
     mServerProxy.clear();
     if (mCblk != NULL) {
@@ -274,9 +256,7 @@
 void AudioFlinger::ThreadBase::TrackBase::releaseBuffer(AudioBufferProvider::Buffer* buffer)
 {
 #ifdef TEE_SINK
-    if (mTeeSink != 0) {
-        (void) mTeeSink->write(buffer->raw, buffer->frameCount);
-    }
+    mTee.write(buffer->raw, buffer->frameCount);
 #endif
 
     ServerProxy::Buffer buf;
@@ -454,6 +434,14 @@
         thread->mFastTrackAvailMask &= ~(1 << i);
     }
     mName = TRACK_NAME_PENDING;
+
+    mServerLatencySupported = thread->type() == ThreadBase::MIXER
+            || thread->type() == ThreadBase::DUPLICATING;
+#ifdef TEE_SINK
+    mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+            + "_" + std::to_string(mId) +
+            + "_PEND_T");
+#endif
 }
 
 AudioFlinger::PlaybackThread::Track::~Track()
@@ -503,13 +491,15 @@
     }
 }
 
-/*static*/ void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
+void AudioFlinger::PlaybackThread::Track::appendDumpHeader(String8& result)
 {
-    result.append("T Name Active Client Session S  Flags "
-                  "  Format Chn mask  SRate "
-                  "ST  L dB  R dB  VS dB "
-                  "  Server FrmCnt  FrmRdy F Underruns  Flushed "
-                  "Main Buf  Aux Buf\n");
+    result.appendFormat("T Name Active Client Session S  Flags "
+                        "  Format Chn mask  SRate "
+                        "ST Usg CT "
+                        " G db  L dB  R dB  VS dB "
+                        "  Server FrmCnt  FrmRdy F Underruns  Flushed"
+                        "%s\n",
+                        isServerLatencySupported() ? "   Latency" : "");
 }
 
 void AudioFlinger::PlaybackThread::Track::appendDump(String8& result, bool active)
@@ -518,7 +508,7 @@
     switch (mType) {
     case TYPE_DEFAULT:
     case TYPE_OUTPUT:
-        if (mSharedBuffer.get() != nullptr) {
+        if (isStatic()) {
             trackType = 'S'; // static
         } else {
             trackType = ' '; // normal
@@ -594,21 +584,25 @@
                     ? 'e' /* error */ : ' ' /* identical */;
 
     result.appendFormat("%7s %6u %7u %2s 0x%03X "
-                           "%08X %08X %6u "
-                           "%2u %5.2g %5.2g %5.2g%c "
-                           "%08X %6zu%c %6zu %c %9u%c %7u "
-                           "%08zX %08zX\n",
+                        "%08X %08X %6u "
+                        "%2u %3x %2x "
+                        "%5.2g %5.2g %5.2g %5.2g%c "
+                        "%08X %6zu%c %6zu %c %9u%c %7u",
             active ? "yes" : "no",
-            (mClient == 0) ? getpid_cached : mClient->pid(),
+            (mClient == 0) ? getpid() : mClient->pid(),
             mSessionId,
             getTrackStateString(),
             mCblk->mFlags,
 
             mFormat,
             mChannelMask,
-            mAudioTrackServerProxy->getSampleRate(),
+            sampleRate(),
 
             mStreamType,
+            mAttr.usage,
+            mAttr.content_type,
+
+            20.0 * log10(mFinalVolume),
             20.0 * log10(float_from_gain(gain_minifloat_unpack_left(vlr))),
             20.0 * log10(float_from_gain(gain_minifloat_unpack_right(vlr))),
             20.0 * log10(vsVolume.first), // VolumeShaper(s) total volume
@@ -621,11 +615,21 @@
             fillingStatus,
             mAudioTrackServerProxy->getUnderrunFrames(),
             nowInUnderrun,
-            (unsigned)mAudioTrackServerProxy->framesFlushed() % 10000000,
-
-            (size_t)mMainBuffer, // use %zX as %p appends 0x
-            (size_t)mAuxBuffer   // use %zX as %p appends 0x
+            (unsigned)mAudioTrackServerProxy->framesFlushed() % 10000000
             );
+
+    if (isServerLatencySupported()) {
+        double latencyMs;
+        bool fromTrack;
+        if (getTrackLatencyMs(&latencyMs, &fromTrack) == OK) {
+            // Show latency in msec, followed by 't' if from track timestamp (the most accurate)
+            // or 'k' if estimated from kernel because track frames haven't been presented yet.
+            result.appendFormat(" %7.2lf %c", latencyMs, fromTrack ? 't' : 'k');
+        } else {
+            result.appendFormat("%10s", mCblk->mServer != 0 ? "unavail" : "new");
+        }
+    }
+    result.append("\n");
 }
 
 uint32_t AudioFlinger::PlaybackThread::Track::sampleRate() const {
@@ -682,6 +686,13 @@
     mAudioTrackServerProxy->setTimestamp(timestamp);
 
     // We do not set drained here, as FastTrack timestamp may not go to very last frame.
+
+    // Compute latency.
+    // TODO: Consider whether the server latency may be passed in by FastMixer
+    // as a constant for all active FastTracks.
+    const double latencyMs = timestamp.getOutputServerLatencyMs(sampleRate());
+    mServerLatencyFromTrack.store(true);
+    mServerLatencyMs.store(latencyMs);
 }
 
 // Don't call for fast tracks; the framesReady() could result in priority inversion
@@ -985,7 +996,7 @@
         // Signal thread to fetch new volume.
         sp<ThreadBase> thread = mThread.promote();
         if (thread != 0) {
-             Mutex::Autolock _l(thread->mLock);
+            Mutex::Autolock _l(thread->mLock);
             thread->broadcast_l();
         }
     }
@@ -1246,7 +1257,7 @@
 //To be called with thread lock held
 void AudioFlinger::PlaybackThread::Track::updateTrackFrameInfo(
         int64_t trackFramesReleased, int64_t sinkFramesWritten,
-        const ExtendedTimestamp &timeStamp) {
+        uint32_t halSampleRate, const ExtendedTimestamp &timeStamp) {
     //update frame map
     mFrameMap.push(trackFramesReleased, sinkFramesWritten);
 
@@ -1255,6 +1266,7 @@
     // Our timestamps are only updated when the track is on the Thread active list.
     // We need to ensure that tracks are not removed before full drain.
     ExtendedTimestamp local = timeStamp;
+    bool drained = true; // default assume drained, if no server info found
     bool checked = false;
     for (int i = ExtendedTimestamp::LOCATION_MAX - 1;
             i >= ExtendedTimestamp::LOCATION_SERVER; --i) {
@@ -1263,18 +1275,25 @@
             local.mPosition[i] = mFrameMap.findX(local.mPosition[i]);
             // check drain state from the latest stage in the pipeline.
             if (!checked && i <= ExtendedTimestamp::LOCATION_KERNEL) {
-                mAudioTrackServerProxy->setDrained(
-                        local.mPosition[i] >= mAudioTrackServerProxy->framesReleased());
+                drained = local.mPosition[i] >= mAudioTrackServerProxy->framesReleased();
                 checked = true;
             }
         }
     }
-    if (!checked) { // no server info, assume drained.
-        mAudioTrackServerProxy->setDrained(true);
-    }
+
+    mAudioTrackServerProxy->setDrained(drained);
     // Set correction for flushed frames that are not accounted for in released.
     local.mFlushed = mAudioTrackServerProxy->framesFlushed();
     mServerProxy->setTimestamp(local);
+
+    // Compute latency info.
+    const bool useTrackTimestamp = !drained;
+    const double latencyMs = useTrackTimestamp
+            ? local.getOutputServerLatencyMs(sampleRate())
+            : timeStamp.getOutputServerLatencyMs(halSampleRate);
+
+    mServerLatencyFromTrack.store(useTrackTimestamp);
+    mServerLatencyMs.store(latencyMs);
 }
 
 // ----------------------------------------------------------------------------
@@ -1342,7 +1361,7 @@
     mActive = false;
 }
 
-bool AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
+ssize_t AudioFlinger::PlaybackThread::OutputTrack::write(void* data, uint32_t frames)
 {
     Buffer *pInBuffer;
     Buffer inBuffer;
@@ -1431,9 +1450,12 @@
                 mBufferQueue.add(pInBuffer);
                 ALOGV("OutputTrack::write() %p thread %p adding overflow buffer %zu", this,
                         mThread.unsafe_get(), mBufferQueue.size());
+                // audio data is consumed (stored locally); set frameCount to 0.
+                inBuffer.frameCount = 0;
             } else {
                 ALOGW("OutputTrack::write() %p thread %p no more overflow buffers",
                         mThread.unsafe_get(), this);
+                // TODO: return error for this.
             }
         }
     }
@@ -1444,7 +1466,7 @@
         stop();
     }
 
-    return outputBufferFull;
+    return frames - inBuffer.frameCount;  // number of frames consumed.
 }
 
 void AudioFlinger::PlaybackThread::OutputTrack::copyMetadataTo(MetadataInserter& backInserter) const
@@ -1509,7 +1531,7 @@
               audio_attributes_t{} /* currently unused for patch track */,
               sampleRate, format, channelMask, frameCount,
               buffer, bufferSize, nullptr /* sharedBuffer */,
-              AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
+              AUDIO_SESSION_NONE, AID_AUDIOSERVER, flags, TYPE_PATCH),
               mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true))
 {
     uint64_t mixBufferNs = ((uint64_t)2 * playbackThread->frameCount() * 1000000000) /
@@ -1528,7 +1550,7 @@
 }
 
 status_t AudioFlinger::PlaybackThread::PatchTrack::start(AudioSystem::sync_event_t event,
-                                                          audio_session_t triggerSession)
+                                                         audio_session_t triggerSession)
 {
     status_t status = Track::start(event, triggerSession);
     if (status != NO_ERROR) {
@@ -1672,18 +1694,20 @@
         return;
     }
 
-    mRecordBufferConverter = new RecordBufferConverter(
-            thread->mChannelMask, thread->mFormat, thread->mSampleRate,
-            channelMask, format, sampleRate);
-    // Check if the RecordBufferConverter construction was successful.
-    // If not, don't continue with construction.
-    //
-    // NOTE: It would be extremely rare that the record track cannot be created
-    // for the current device, but a pending or future device change would make
-    // the record track configuration valid.
-    if (mRecordBufferConverter->initCheck() != NO_ERROR) {
-        ALOGE("RecordTrack unable to create record buffer converter");
-        return;
+    if (!isDirect()) {
+        mRecordBufferConverter = new RecordBufferConverter(
+                thread->mChannelMask, thread->mFormat, thread->mSampleRate,
+                channelMask, format, sampleRate);
+        // Check if the RecordBufferConverter construction was successful.
+        // If not, don't continue with construction.
+        //
+        // NOTE: It would be extremely rare that the record track cannot be created
+        // for the current device, but a pending or future device change would make
+        // the record track configuration valid.
+        if (mRecordBufferConverter->initCheck() != NO_ERROR) {
+            ALOGE("RecordTrack unable to create record buffer converter");
+            return;
+        }
     }
 
     mServerProxy = new AudioRecordServerProxy(mCblk, mBuffer, frameCount,
@@ -1694,7 +1718,15 @@
     if (flags & AUDIO_INPUT_FLAG_FAST) {
         ALOG_ASSERT(thread->mFastTrackAvail);
         thread->mFastTrackAvail = false;
+    } else {
+        // TODO: only Normal Record has timestamps (Fast Record does not).
+        mServerLatencySupported = true;
     }
+#ifdef TEE_SINK
+    mTee.setId(std::string("_") + std::to_string(mThreadIoHandle)
+            + "_" + std::to_string(mId)
+            + "_R");
+#endif
 }
 
 AudioFlinger::RecordThread::RecordTrack::~RecordTrack()
@@ -1783,19 +1815,22 @@
 }
 
 
-/*static*/ void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
+void AudioFlinger::RecordThread::RecordTrack::appendDumpHeader(String8& result)
 {
-    result.append("Active Client Session S  Flags   Format Chn mask  SRate   Server FrmCnt Sil\n");
+    result.appendFormat("Active Client Session S  Flags  "
+                        " Format Chn mask  SRate Source "
+                        " Server FrmCnt FrmRdy Sil%s\n",
+                        isServerLatencySupported() ? "   Latency" : "");
 }
 
 void AudioFlinger::RecordThread::RecordTrack::appendDump(String8& result, bool active)
 {
     result.appendFormat("%c%5s %6u %7u %2s 0x%03X "
-            "%08X %08X %6u "
-            "%08X %6zu %3c\n",
+            "%08X %08X %6u %6X "
+            "%08X %6zu %6zu %3c",
             isFastTrack() ? 'F' : ' ',
             active ? "yes" : "no",
-            (mClient == 0) ? getpid_cached : mClient->pid(),
+            (mClient == 0) ? getpid() : mClient->pid(),
             mSessionId,
             getTrackStateString(),
             mCblk->mFlags,
@@ -1803,11 +1838,25 @@
             mFormat,
             mChannelMask,
             mSampleRate,
+            mAttr.source,
 
             mCblk->mServer,
             mFrameCount,
+            mServerProxy->framesReadySafe(),
             isSilenced() ? 's' : 'n'
             );
+    if (isServerLatencySupported()) {
+        double latencyMs;
+        bool fromTrack;
+        if (getTrackLatencyMs(&latencyMs, &fromTrack) == OK) {
+            // Show latency in msec, followed by 't' if from track timestamp (the most accurate)
+            // or 'k' if estimated from kernel (usually for debugging).
+            result.appendFormat(" %7.2lf %c", latencyMs, fromTrack ? 't' : 'k');
+        } else {
+            result.appendFormat("%10s", mCblk->mServer != 0 ? "unavail" : "new");
+        }
+    }
+    result.append("\n");
 }
 
 void AudioFlinger::RecordThread::RecordTrack::handleSyncStartEvent(const sp<SyncEvent>& event)
@@ -1850,6 +1899,15 @@
         }
     }
     mServerProxy->setTimestamp(local);
+
+    // Compute latency info.
+    const bool useTrackTimestamp = true; // use track unless debugging.
+    const double latencyMs = - (useTrackTimestamp
+            ? local.getOutputServerLatencyMs(sampleRate())
+            : timestamp.getOutputServerLatencyMs(halSampleRate));
+
+    mServerLatencyFromTrack.store(useTrackTimestamp);
+    mServerLatencyMs.store(latencyMs);
 }
 
 status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
@@ -1875,7 +1933,8 @@
     :   RecordTrack(recordThread, NULL,
                 audio_attributes_t{} /* currently unused for patch track */,
                 sampleRate, format, channelMask, frameCount,
-                buffer, bufferSize, AUDIO_SESSION_NONE, getuid(), flags, TYPE_PATCH),
+                buffer, bufferSize, AUDIO_SESSION_NONE, AID_AUDIOSERVER,
+                flags, TYPE_PATCH),
                 mProxy(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true))
 {
     uint64_t mixBufferNs = ((uint64_t)2 * recordThread->frameCount() * 1000000000) /
@@ -1940,13 +1999,14 @@
         audio_format_t format,
         audio_channel_mask_t channelMask,
         audio_session_t sessionId,
+        bool isOut,
         uid_t uid,
         pid_t pid,
         audio_port_handle_t portId)
     :   TrackBase(thread, NULL, attr, sampleRate, format,
                   channelMask, (size_t)0 /* frameCount */,
                   nullptr /* buffer */, (size_t)0 /* bufferSize */,
-                  sessionId, uid, false /* isOut */,
+                  sessionId, uid, isOut,
                   ALLOC_NONE,
                   TYPE_DEFAULT, portId),
         mPid(pid), mSilenced(false), mSilencedNotified(false)
@@ -1963,7 +2023,7 @@
 }
 
 status_t AudioFlinger::MmapThread::MmapTrack::start(AudioSystem::sync_event_t event __unused,
-                                                        audio_session_t triggerSession __unused)
+                                                    audio_session_t triggerSession __unused)
 {
     return NO_ERROR;
 }
@@ -1994,19 +2054,27 @@
 {
 }
 
-/*static*/ void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result)
+void AudioFlinger::MmapThread::MmapTrack::appendDumpHeader(String8& result)
 {
-    result.append("Client Session   Format Chn mask  SRate\n");
+    result.appendFormat("Client Session   Format Chn mask  SRate Flags %s\n",
+                        isOut() ? "Usg CT": "Source");
 }
 
 void AudioFlinger::MmapThread::MmapTrack::appendDump(String8& result, bool active __unused)
 {
-    result.appendFormat("%6u %7u %08X %08X %6u\n",
+    result.appendFormat("%6u %7u %08X %08X %6u 0x%03X ",
             mPid,
             mSessionId,
             mFormat,
             mChannelMask,
-            mSampleRate);
+            mSampleRate,
+            mAttr.flags);
+    if (isOut()) {
+        result.appendFormat("%3x %2x", mAttr.usage, mAttr.content_type);
+    } else {
+        result.appendFormat("%6x", mAttr.source);
+    }
+    result.append("\n");
 }
 
 } // namespace android
diff --git a/services/audiopolicy/Android.mk b/services/audiopolicy/Android.mk
index d29cae1..b75e957 100644
--- a/services/audiopolicy/Android.mk
+++ b/services/audiopolicy/Android.mk
@@ -13,7 +13,6 @@
     $(call include-path-for, audio-utils) \
     frameworks/av/services/audiopolicy/common/include \
     frameworks/av/services/audiopolicy/engine/interface \
-    frameworks/av/services/audiopolicy/utilities
 
 LOCAL_SHARED_LIBRARIES := \
     libcutils \
@@ -22,10 +21,10 @@
     libbinder \
     libaudioclient \
     libhardware_legacy \
-    libserviceutility \
     libaudiopolicymanager \
     libmedia_helper \
     libmediametrics \
+    libmediautils \
     libeffectsconfig
 
 LOCAL_STATIC_LIBRARIES := \
@@ -74,7 +73,6 @@
 LOCAL_C_INCLUDES += \
     frameworks/av/services/audiopolicy/common/include \
     frameworks/av/services/audiopolicy/engine/interface \
-    frameworks/av/services/audiopolicy/utilities
 
 LOCAL_STATIC_LIBRARIES := \
     libaudiopolicycomponents
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 4812b1f..fe49483 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -324,11 +324,6 @@
     // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
     virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
 
-    // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
-    // over a telephony device during a phone call.
-    virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream) = 0;
-    virtual status_t stopTone() = 0;
-
     // set down link audio volume.
     virtual status_t setVoiceVolume(float volume, int delayMs = 0) = 0;
 
diff --git a/services/audiopolicy/common/managerdefinitions/Android.mk b/services/audiopolicy/common/managerdefinitions/Android.mk
index e69e687..b3611c4 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.mk
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -35,8 +35,7 @@
     $(LOCAL_PATH)/include \
     frameworks/av/services/audiopolicy/common/include \
     frameworks/av/services/audiopolicy \
-    frameworks/av/services/audiopolicy/utilities \
-    system/media/audio_utils/include \
+    $(call include-path-for, audio-utils) \
 
 ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSessionInfoProvider.h b/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
similarity index 67%
rename from services/audiopolicy/common/managerdefinitions/include/AudioSessionInfoProvider.h
rename to services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
index e0037fc..9f3fc0c 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioSessionInfoProvider.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
@@ -19,26 +19,27 @@
 namespace android {
 
 /**
- * Interface for input descriptors to implement so dependent audio sessions can query information
- * about their context
+ * Interface for I/O descriptors to implement so information about their context
+ * can be queried and updated.
  */
-class AudioSessionInfoProvider
+class AudioIODescriptorInterface
 {
 public:
-    virtual ~AudioSessionInfoProvider() {};
+    virtual ~AudioIODescriptorInterface() {};
 
     virtual audio_config_base_t getConfig() const = 0;
 
     virtual audio_patch_handle_t getPatchHandle() const = 0;
 
+    virtual void setPatchHandle(audio_patch_handle_t handle) = 0;
 };
 
-class AudioSessionInfoUpdateListener
+class AudioIODescriptorUpdateListener
 {
 public:
-    virtual ~AudioSessionInfoUpdateListener() {};
+    virtual ~AudioIODescriptorUpdateListener() {};
 
-    virtual void onSessionInfoUpdate() const = 0;;
+    virtual void onIODescriptorUpdate() const = 0;
 };
 
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index b25d6d4..85f3b86 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -16,9 +16,9 @@
 
 #pragma once
 
+#include "AudioIODescriptorInterface.h"
 #include "AudioPort.h"
 #include "AudioSession.h"
-#include "AudioSessionInfoProvider.h"
 #include <utils/Errors.h>
 #include <system/audio.h>
 #include <utils/SortedVector.h>
@@ -31,7 +31,7 @@
 
 // descriptor for audio inputs. Used to maintain current configuration of each opened audio input
 // and keep track of the usage of this input.
-class AudioInputDescriptor: public AudioPortConfig, public AudioSessionInfoProvider
+class AudioInputDescriptor: public AudioPortConfig, public AudioIODescriptorInterface
 {
 public:
     explicit AudioInputDescriptor(const sp<IOProfile>& profile,
@@ -67,11 +67,10 @@
     size_t getAudioSessionCount(bool activeOnly) const;
     audio_source_t getHighestPrioritySource(bool activeOnly) const;
 
-    // implementation of AudioSessionInfoProvider
-    virtual audio_config_base_t getConfig() const;
-    virtual audio_patch_handle_t getPatchHandle() const;
-
-    void setPatchHandle(audio_patch_handle_t handle);
+    // implementation of AudioIODescriptorInterface
+    audio_config_base_t getConfig() const override;
+    audio_patch_handle_t getPatchHandle() const override;
+    void setPatchHandle(audio_patch_handle_t handle) override;
 
     status_t open(const audio_config_t *config,
                   audio_devices_t device,
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 5e5d38b..57d1cfa 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -24,6 +24,7 @@
 #include <utils/Timers.h>
 #include <utils/KeyedVector.h>
 #include <system/audio.h>
+#include "AudioIODescriptorInterface.h"
 #include "AudioSourceDescriptor.h"
 
 namespace android {
@@ -35,7 +36,7 @@
 
 // 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.
-class AudioOutputDescriptor: public AudioPortConfig
+class AudioOutputDescriptor: public AudioPortConfig, public AudioIODescriptorInterface
 {
 public:
     AudioOutputDescriptor(const sp<AudioPort>& port,
@@ -73,8 +74,10 @@
 
     audio_module_handle_t getModuleHandle() const;
 
-    audio_patch_handle_t getPatchHandle() const { return mPatchHandle; };
-    void setPatchHandle(audio_patch_handle_t handle) { mPatchHandle = handle; };
+    // implementation of AudioIODescriptorInterface
+    audio_config_base_t getConfig() const override;
+    audio_patch_handle_t getPatchHandle() const override;
+    void setPatchHandle(audio_patch_handle_t handle) override;
 
     sp<AudioPort>       mPort;
     audio_devices_t mDevice;                   // current device this output is routed to
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 43f6ed6..f861b95 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -38,16 +38,24 @@
     AudioPolicyConfig(HwModuleCollection &hwModules,
                       DeviceVector &availableOutputDevices,
                       DeviceVector &availableInputDevices,
-                      sp<DeviceDescriptor> &defaultOutputDevices,
+                      sp<DeviceDescriptor> &defaultOutputDevice,
                       VolumeCurvesCollection *volumes = nullptr)
         : mHwModules(hwModules),
           mAvailableOutputDevices(availableOutputDevices),
           mAvailableInputDevices(availableInputDevices),
-          mDefaultOutputDevices(defaultOutputDevices),
+          mDefaultOutputDevice(defaultOutputDevice),
           mVolumeCurves(volumes),
           mIsSpeakerDrcEnabled(false)
     {}
 
+    const std::string& getSource() const {
+        return mSource;
+    }
+
+    void setSource(const std::string& file) {
+        mSource = file;
+    }
+
     void setVolumes(const VolumeCurvesCollection &volumes)
     {
         if (mVolumeCurves != nullptr) {
@@ -100,46 +108,52 @@
 
     void setDefaultOutputDevice(const sp<DeviceDescriptor> &defaultDevice)
     {
-        mDefaultOutputDevices = defaultDevice;
+        mDefaultOutputDevice = defaultDevice;
     }
 
-    const sp<DeviceDescriptor> &getDefaultOutputDevice() const { return mDefaultOutputDevices; }
+    const sp<DeviceDescriptor> &getDefaultOutputDevice() const { return mDefaultOutputDevice; }
 
     void setDefault(void)
     {
-        mDefaultOutputDevices = new DeviceDescriptor(AUDIO_DEVICE_OUT_SPEAKER);
-        sp<HwModule> module;
+        mSource = "AudioPolicyConfig::setDefault";
+        mDefaultOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_SPEAKER);
+        mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic());
         sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_BUILTIN_MIC);
-        mAvailableOutputDevices.add(mDefaultOutputDevices);
+        defaultInputDevice->addAudioProfile(AudioProfile::createFullDynamic());
+        sp<AudioProfile> micProfile = new AudioProfile(
+                AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_MONO, 8000);
+        defaultInputDevice->addAudioProfile(micProfile);
+        mAvailableOutputDevices.add(mDefaultOutputDevice);
         mAvailableInputDevices.add(defaultInputDevice);
 
-        module = new HwModule(AUDIO_HARDWARE_MODULE_ID_PRIMARY);
+        sp<HwModule> module = new HwModule(AUDIO_HARDWARE_MODULE_ID_PRIMARY, 2 /*halVersionMajor*/);
+        mHwModules.add(module);
+        mDefaultOutputDevice->attach(module);
+        defaultInputDevice->attach(module);
 
         sp<OutputProfile> outProfile;
         outProfile = new OutputProfile(String8("primary"));
         outProfile->attach(module);
         outProfile->addAudioProfile(
                 new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 44100));
-        outProfile->addSupportedDevice(mDefaultOutputDevices);
+        outProfile->addSupportedDevice(mDefaultOutputDevice);
         outProfile->setFlags(AUDIO_OUTPUT_FLAG_PRIMARY);
         module->addOutputProfile(outProfile);
 
         sp<InputProfile> inProfile;
         inProfile = new InputProfile(String8("primary"));
         inProfile->attach(module);
-        inProfile->addAudioProfile(
-                new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_MONO, 8000));
+        inProfile->addAudioProfile(micProfile);
         inProfile->addSupportedDevice(defaultInputDevice);
         module->addInputProfile(inProfile);
-
-        mHwModules.add(module);
     }
 
 private:
+    std::string mSource;
     HwModuleCollection &mHwModules; /**< Collection of Module, with Profiles, i.e. Mix Ports. */
     DeviceVector &mAvailableOutputDevices;
     DeviceVector &mAvailableInputDevices;
-    sp<DeviceDescriptor> &mDefaultOutputDevices;
+    sp<DeviceDescriptor> &mDefaultOutputDevice;
     VolumeCurvesCollection *mVolumeCurves;
     // TODO: remove when legacy conf file is removed. true on devices that use DRC on the
     // DEVICE_CATEGORY_SPEAKER path to boost soft sounds, used to adjust volume curves accordingly.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index 09a86dd..bd7517f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -153,9 +153,6 @@
 class AudioPortConfig : public virtual RefBase
 {
 public:
-    AudioPortConfig();
-    virtual ~AudioPortConfig() {}
-
     status_t applyAudioPortConfig(const struct audio_port_config *config,
                                   struct audio_port_config *backupConfig = NULL);
     virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
@@ -165,10 +162,11 @@
         return (other != 0) &&
                 (other->getAudioPort()->getModuleHandle() == getAudioPort()->getModuleHandle());
     }
-    uint32_t mSamplingRate;
-    audio_format_t mFormat;
-    audio_channel_mask_t mChannelMask;
-    struct audio_gain_config mGain;
+    unsigned int mSamplingRate = 0u;
+    audio_format_t mFormat = AUDIO_FORMAT_INVALID;
+    audio_channel_mask_t mChannelMask = AUDIO_CHANNEL_NONE;
+    struct audio_gain_config mGain = { .index = -1 };
+    union audio_io_flags mFlags = { AUDIO_INPUT_FLAG_NONE };
 };
 
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
index 8741c66..4226ff2 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
@@ -36,6 +36,8 @@
 class AudioProfile : public virtual RefBase
 {
 public:
+    static sp<AudioProfile> createFullDynamic();
+
     AudioProfile(audio_format_t format,
                  audio_channel_mask_t channelMasks,
                  uint32_t samplingRate) :
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSession.h b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
index dd5247d..53e6ec9 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
@@ -23,13 +23,13 @@
 #include <utils/KeyedVector.h>
 #include <media/AudioPolicy.h>
 #include <media/IAudioPolicyServiceClient.h>
-#include "AudioSessionInfoProvider.h"
+#include "AudioIODescriptorInterface.h"
 
 namespace android {
 
 class AudioPolicyClientInterface;
 
-class AudioSession : public RefBase, public AudioSessionInfoUpdateListener
+class AudioSession : public RefBase, public AudioIODescriptorUpdateListener
 {
 public:
     AudioSession(audio_session_t session,
@@ -63,9 +63,9 @@
     uint32_t changeOpenCount(int delta);
     uint32_t changeActiveCount(int delta);
 
-    void setInfoProvider(AudioSessionInfoProvider *provider);
-    // implementation of AudioSessionInfoUpdateListener
-    virtual void onSessionInfoUpdate() const;
+    void setInfoProvider(AudioIODescriptorInterface *provider);
+    // implementation of AudioIODescriptorUpdateListener
+    virtual void onIODescriptorUpdate() const;
 
 private:
     record_client_info_t mRecordClientInfo;
@@ -77,17 +77,17 @@
     uint32_t  mActiveCount;
     AudioMix* mPolicyMix; // non NULL when used by a dynamic policy
     AudioPolicyClientInterface* mClientInterface;
-    const AudioSessionInfoProvider* mInfoProvider;
+    const AudioIODescriptorInterface* mInfoProvider;
 };
 
 class AudioSessionCollection :
     public DefaultKeyedVector<audio_session_t, sp<AudioSession> >,
-    public AudioSessionInfoUpdateListener
+    public AudioIODescriptorUpdateListener
 {
 public:
     status_t addSession(audio_session_t session,
                              const sp<AudioSession>& audioSession,
-                             AudioSessionInfoProvider *provider);
+                             AudioIODescriptorInterface *provider);
 
     status_t removeSession(audio_session_t session);
 
@@ -99,8 +99,8 @@
     bool isSourceActive(audio_source_t source) const;
     audio_source_t getHighestPrioritySource(bool activeOnly) const;
 
-    // implementation of AudioSessionInfoUpdateListener
-    virtual void onSessionInfoUpdate() const;
+    // implementation of AudioIODescriptorUpdateListener
+    virtual void onIODescriptorUpdate() const;
 
     status_t dump(int fd, int spaces) const;
 };
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 92a4c3e..2325e4f 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -76,11 +76,12 @@
 
     audio_devices_t types() const { return mDeviceTypes; }
 
-    sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8& address) const;
-    DeviceVector getDevicesFromType(audio_devices_t types) const;
+    // If 'address' is empty, a device with a non-empty address may be returned
+    // if there is no device with the specified 'type' and empty address.
+    sp<DeviceDescriptor> getDevice(audio_devices_t type, const String8 &address) const;
+    DeviceVector getDevicesFromTypeMask(audio_devices_t types) const;
     sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
     sp<DeviceDescriptor> getDeviceFromTagName(const String8 &tagName) const;
-    DeviceVector getDevicesFromTypeAddr(audio_devices_t type, const String8& address) const;
 
     audio_devices_t getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index cb9f49e..05cfc31 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -107,7 +107,7 @@
     sp<DeviceDescriptor> getDeviceDescriptor(const audio_devices_t device,
                                              const char *device_address,
                                              const char *device_name,
-                                             bool matchAdress = true) const;
+                                             bool matchAddress = true) const;
 
     status_t dump(int fd) const;
 };
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index 92332fb..f0144db 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -164,7 +164,7 @@
 
 status_t AudioInputDescriptor::addAudioSession(audio_session_t session,
                          const sp<AudioSession>& audioSession) {
-    return mSessions.addSession(session, audioSession, /*AudioSessionInfoProvider*/this);
+    return mSessions.addSession(session, audioSession, /*AudioIODescriptorInterface*/this);
 }
 
 status_t AudioInputDescriptor::removeAudioSession(audio_session_t session) {
@@ -179,7 +179,7 @@
 void AudioInputDescriptor::setPatchHandle(audio_patch_handle_t handle)
 {
     mPatchHandle = handle;
-    mSessions.onSessionInfoUpdate();
+    mSessions.onIODescriptorUpdate();
 }
 
 audio_config_base_t AudioInputDescriptor::getConfig() const
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 294a2a6..3c69de5 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -55,11 +55,28 @@
     }
 }
 
+audio_config_base_t AudioOutputDescriptor::getConfig() const
+{
+    const audio_config_base_t config = { .sample_rate = mSamplingRate, .channel_mask = mChannelMask,
+            .format = mFormat };
+    return config;
+}
+
 audio_module_handle_t AudioOutputDescriptor::getModuleHandle() const
 {
     return mPort.get() != nullptr ? mPort->getModuleHandle() : AUDIO_MODULE_HANDLE_NONE;
 }
 
+audio_patch_handle_t AudioOutputDescriptor::getPatchHandle() const
+{
+    return mPatchHandle;
+}
+
+void AudioOutputDescriptor::setPatchHandle(audio_patch_handle_t handle)
+{
+    mPatchHandle = handle;
+}
+
 audio_port_handle_t AudioOutputDescriptor::getId() const
 {
     return mId;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
index a9fe48d..e78e121 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -34,51 +34,32 @@
 {
 }
 
+static String8 dumpPatchEndpoints(
+        int spaces, const char *prefix, int count, const audio_port_config *cfgs)
+{
+    String8 result;
+    for (int i = 0; i < count; ++i) {
+        const audio_port_config &cfg = cfgs[i];
+        result.appendFormat("%*s  [%s %d] ", spaces, "", prefix, i + 1);
+        if (cfg.type == AUDIO_PORT_TYPE_DEVICE) {
+            std::string device;
+            deviceToString(cfg.ext.device.type, device);
+            result.appendFormat("Device ID %d %s", cfg.id, device.c_str());
+        } else {
+            result.appendFormat("Mix ID %d I/O handle %d", cfg.id, cfg.ext.mix.handle);
+        }
+        result.append("\n");
+    }
+    return result;
+}
+
 status_t AudioPatch::dump(int fd, int spaces, int index) const
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
     String8 result;
-
-    snprintf(buffer, SIZE, "%*sAudio patch %d:\n", spaces, "", index+1);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- handle: %2d\n", spaces, "", mHandle);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- audio flinger handle: %2d\n", spaces, "", mAfPatchHandle);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- owner uid: %2d\n", spaces, "", mUid);
-    result.append(buffer);
-    snprintf(buffer, SIZE, "%*s- %d sources:\n", spaces, "", mPatch.num_sources);
-    result.append(buffer);
-    for (size_t i = 0; i < mPatch.num_sources; i++) {
-        if (mPatch.sources[i].type == AUDIO_PORT_TYPE_DEVICE) {
-            std::string device;
-            deviceToString(mPatch.sources[i].ext.device.type, device);
-            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
-                     mPatch.sources[i].id,
-                     device.c_str());
-        } else {
-            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
-                     mPatch.sources[i].id, mPatch.sources[i].ext.mix.handle);
-        }
-        result.append(buffer);
-    }
-    snprintf(buffer, SIZE, "%*s- %d sinks:\n", spaces, "", mPatch.num_sinks);
-    result.append(buffer);
-    for (size_t i = 0; i < mPatch.num_sinks; i++) {
-        if (mPatch.sinks[i].type == AUDIO_PORT_TYPE_DEVICE) {
-            std::string device;
-            deviceToString(mPatch.sinks[i].ext.device.type, device);
-            snprintf(buffer, SIZE, "%*s- Device ID %d %s\n", spaces + 2, "",
-                     mPatch.sinks[i].id,
-                     device.c_str());
-        } else {
-            snprintf(buffer, SIZE, "%*s- Mix ID %d I/O handle %d\n", spaces + 2, "",
-                     mPatch.sinks[i].id, mPatch.sinks[i].ext.mix.handle);
-        }
-        result.append(buffer);
-    }
-
+    result.appendFormat("%*sPatch %d: owner uid %4d, handle %2d, af handle %2d\n",
+            spaces, "", index + 1, mUid, mHandle, mAfPatchHandle);
+    result.append(dumpPatchEndpoints(spaces, "src ", mPatch.num_sources, mPatch.sources));
+    result.append(dumpPatchEndpoints(spaces, "sink", mPatch.num_sinks, mPatch.sinks));
     write(fd, result.string(), result.size());
     return NO_ERROR;
 }
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index d85562e..3fe37ab 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -386,22 +386,12 @@
 
 // --- AudioPortConfig class implementation
 
-AudioPortConfig::AudioPortConfig()
-{
-    mSamplingRate = 0;
-    mChannelMask = AUDIO_CHANNEL_NONE;
-    mFormat = AUDIO_FORMAT_INVALID;
-    memset(&mGain, 0, sizeof(struct audio_gain_config));
-    mGain.index = -1;
-}
-
 status_t AudioPortConfig::applyAudioPortConfig(const struct audio_port_config *config,
                                                struct audio_port_config *backupConfig)
 {
-    struct audio_port_config localBackupConfig;
+    struct audio_port_config localBackupConfig = { .config_mask = config->config_mask };
     status_t status = NO_ERROR;
 
-    localBackupConfig.config_mask = config->config_mask;
     toAudioPortConfig(&localBackupConfig);
 
     sp<AudioPort> audioport = getAudioPort();
@@ -425,6 +415,9 @@
     if (config->config_mask & AUDIO_PORT_CONFIG_GAIN) {
         mGain = config->gain;
     }
+    if (config->config_mask & AUDIO_PORT_CONFIG_FLAGS) {
+        mFlags = config->flags;
+    }
 
 exit:
     if (status != NO_ERROR) {
@@ -436,33 +429,38 @@
     return status;
 }
 
+namespace {
+
+template<typename T>
+void updateField(
+        const T& portConfigField, T audio_port_config::*port_config_field,
+        struct audio_port_config *dstConfig, const struct audio_port_config *srcConfig,
+        unsigned int configMask, T defaultValue)
+{
+    if (dstConfig->config_mask & configMask) {
+        if ((srcConfig != nullptr) && (srcConfig->config_mask & configMask)) {
+            dstConfig->*port_config_field = srcConfig->*port_config_field;
+        } else {
+            dstConfig->*port_config_field = portConfigField;
+        }
+    } else {
+        dstConfig->*port_config_field = defaultValue;
+    }
+}
+
+} // namespace
+
 void AudioPortConfig::toAudioPortConfig(struct audio_port_config *dstConfig,
                                         const struct audio_port_config *srcConfig) const
 {
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
-        dstConfig->sample_rate = mSamplingRate;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE)) {
-            dstConfig->sample_rate = srcConfig->sample_rate;
-        }
-    } else {
-        dstConfig->sample_rate = 0;
-    }
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
-        dstConfig->channel_mask = mChannelMask;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK)) {
-            dstConfig->channel_mask = srcConfig->channel_mask;
-        }
-    } else {
-        dstConfig->channel_mask = AUDIO_CHANNEL_NONE;
-    }
-    if (dstConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT) {
-        dstConfig->format = mFormat;
-        if ((srcConfig != NULL) && (srcConfig->config_mask & AUDIO_PORT_CONFIG_FORMAT)) {
-            dstConfig->format = srcConfig->format;
-        }
-    } else {
-        dstConfig->format = AUDIO_FORMAT_INVALID;
-    }
+    updateField(mSamplingRate, &audio_port_config::sample_rate,
+            dstConfig, srcConfig, AUDIO_PORT_CONFIG_SAMPLE_RATE, 0u);
+    updateField(mChannelMask, &audio_port_config::channel_mask,
+            dstConfig, srcConfig, AUDIO_PORT_CONFIG_CHANNEL_MASK,
+            (audio_channel_mask_t)AUDIO_CHANNEL_NONE);
+    updateField(mFormat, &audio_port_config::format,
+            dstConfig, srcConfig, AUDIO_PORT_CONFIG_FORMAT, AUDIO_FORMAT_INVALID);
+
     sp<AudioPort> audioport = getAudioPort();
     if ((dstConfig->config_mask & AUDIO_PORT_CONFIG_GAIN) && audioport != NULL) {
         dstConfig->gain = mGain;
@@ -478,6 +476,9 @@
     } else {
         dstConfig->config_mask &= ~AUDIO_PORT_CONFIG_GAIN;
     }
+
+    updateField(mFlags, &audio_port_config::flags,
+            dstConfig, srcConfig, AUDIO_PORT_CONFIG_FLAGS, { AUDIO_INPUT_FLAG_NONE });
 }
 
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
index fd6fc1c..26af9b4 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
@@ -28,6 +28,23 @@
 
 namespace android {
 
+static AudioProfile* createFullDynamicImpl()
+{
+    AudioProfile* dynamicProfile = new AudioProfile(gDynamicFormat,
+            ChannelsVector(), SampleRateVector());
+    dynamicProfile->setDynamicFormat(true);
+    dynamicProfile->setDynamicChannels(true);
+    dynamicProfile->setDynamicRate(true);
+    return dynamicProfile;
+}
+
+// static
+sp<AudioProfile> AudioProfile::createFullDynamic()
+{
+    static sp<AudioProfile> dynamicProfile = createFullDynamicImpl();
+    return dynamicProfile;
+}
+
 status_t AudioProfile::checkExact(uint32_t samplingRate, audio_channel_mask_t channelMask,
                                   audio_format_t format) const
 {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
index 7cda46b..91dee35 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
@@ -86,7 +86,7 @@
         }
 
         // Recording configuration callback:
-        const AudioSessionInfoProvider* provider = mInfoProvider;
+        const AudioIODescriptorInterface* provider = mInfoProvider;
         const audio_config_base_t deviceConfig = (provider != NULL) ? provider->getConfig() :
                 AUDIO_CONFIG_BASE_INITIALIZER;
         const audio_patch_handle_t patchHandle = (provider != NULL) ? provider->getPatchHandle() :
@@ -114,16 +114,16 @@
     return false;
 }
 
-void AudioSession::setInfoProvider(AudioSessionInfoProvider *provider)
+void AudioSession::setInfoProvider(AudioIODescriptorInterface *provider)
 {
     mInfoProvider = provider;
 }
 
-void AudioSession::onSessionInfoUpdate() const
+void AudioSession::onIODescriptorUpdate() const
 {
     if (mActiveCount > 0) {
         // resend the callback after requerying the informations from the info provider
-        const AudioSessionInfoProvider* provider = mInfoProvider;
+        const AudioIODescriptorInterface* provider = mInfoProvider;
         const audio_config_base_t deviceConfig = (provider != NULL) ? provider->getConfig() :
                 AUDIO_CONFIG_BASE_INITIALIZER;
         const audio_patch_handle_t patchHandle = (provider != NULL) ? provider->getPatchHandle() :
@@ -170,7 +170,7 @@
 
 status_t AudioSessionCollection::addSession(audio_session_t session,
                                          const sp<AudioSession>& audioSession,
-                                         AudioSessionInfoProvider *provider)
+                                         AudioIODescriptorInterface *provider)
 {
     ssize_t index = indexOfKey(session);
 
@@ -271,10 +271,10 @@
     return source;
 }
 
-void AudioSessionCollection::onSessionInfoUpdate() const
+void AudioSessionCollection::onIODescriptorUpdate() const
 {
     for (size_t i = 0; i < size(); i++) {
-        valueAt(i)->onSessionInfoUpdate();
+        valueAt(i)->onIODescriptorUpdate();
     }
 }
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
index 1e105f5..19eac26 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ConfigParsingUtils.cpp
@@ -412,6 +412,7 @@
     free(data);
 
     ALOGI("loadAudioPolicyConfig() loaded %s\n", path);
+    config.setSource(path);
 
     return NO_ERROR;
 }
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index 19c2062..d3cc8b9 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -145,8 +145,8 @@
             }
         }
     }
-    ALOGV("DeviceVector::getDevice() for type %08x address %s found %p",
-          type, address.string(), device.get());
+    ALOGV("DeviceVector::%s() for type %08x address \"%s\" found %p",
+            __func__, type, address.string(), device.get());
     return device;
 }
 
@@ -160,7 +160,7 @@
     return nullptr;
 }
 
-DeviceVector DeviceVector::getDevicesFromType(audio_devices_t type) const
+DeviceVector DeviceVector::getDevicesFromTypeMask(audio_devices_t type) const
 {
     DeviceVector devices;
     bool isOutput = audio_is_output_devices(type);
@@ -171,20 +171,8 @@
         if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
             devices.add(itemAt(i));
             type &= ~curType;
-            ALOGV("DeviceVector::getDevicesFromType() for type %x found %p",
-                  itemAt(i)->type(), itemAt(i).get());
-        }
-    }
-    return devices;
-}
-
-DeviceVector DeviceVector::getDevicesFromTypeAddr(
-        audio_devices_t type, const String8& address) const
-{
-    DeviceVector devices;
-    for (const auto& device : *this) {
-        if (device->type() == type && device->mAddress == address) {
-            devices.add(device);
+            ALOGV("DeviceVector::%s() for type %08x found %p",
+                    __func__, itemAt(i)->type(), itemAt(i).get());
         }
     }
     return devices;
@@ -253,7 +241,7 @@
 
 void DeviceDescriptor::toAudioPort(struct audio_port *port) const
 {
-    ALOGV("DeviceDescriptor::toAudioPort() handle %d type %x", mId, mDeviceType);
+    ALOGV("DeviceDescriptor::toAudioPort() handle %d type %08x", mId, mDeviceType);
     AudioPort::toAudioPort(port);
     port->id = mId;
     toAudioPortConfig(&port->active_config);
@@ -305,7 +293,7 @@
 {
     std::string device;
     deviceToString(mDeviceType, device);
-    ALOGI("Device id:%d type:0x%X:%s, addr:%s", mId,  mDeviceType, device.c_str(),
+    ALOGI("Device id:%d type:0x%08X:%s, addr:%s", mId,  mDeviceType, device.c_str(),
           mAddress.string());
 
     AudioPort::log("  ");
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index aef7dbe..dcc0ec8 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -278,9 +278,10 @@
 sp<DeviceDescriptor> HwModuleCollection::getDeviceDescriptor(const audio_devices_t device,
                                                              const char *device_address,
                                                              const char *device_name,
-                                                             bool matchAdress) const
+                                                             bool matchAddress) const
 {
-    String8 address = (device_address == nullptr) ? String8("") : String8(device_address);
+    String8 address = (device_address == nullptr || !matchAddress) ?
+            String8("") : String8(device_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");
@@ -288,15 +289,9 @@
 
     for (const auto& hwModule : *this) {
         DeviceVector declaredDevices = hwModule->getDeclaredDevices();
-        DeviceVector deviceList = declaredDevices.getDevicesFromTypeAddr(device, address);
-        if (!deviceList.isEmpty()) {
-            return deviceList.itemAt(0);
-        }
-        if (!matchAdress) {
-            deviceList = declaredDevices.getDevicesFromType(device);
-            if (!deviceList.isEmpty()) {
-                return deviceList.itemAt(0);
-            }
+        sp<DeviceDescriptor> deviceDesc = declaredDevices.getDevice(device, address);
+        if (deviceDesc) {
+            return deviceDesc;
         }
     }
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index a253113..8008a7c 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -242,12 +242,7 @@
     AudioProfileTraits::Collection profiles;
     deserializeCollection<AudioProfileTraits>(doc, child, profiles, NULL);
     if (profiles.isEmpty()) {
-        sp <AudioProfile> dynamicProfile = new AudioProfile(gDynamicFormat,
-                                                            ChannelsVector(), SampleRateVector());
-        dynamicProfile->setDynamicFormat(true);
-        dynamicProfile->setDynamicChannels(true);
-        dynamicProfile->setDynamicRate(true);
-        profiles.add(dynamicProfile);
+        profiles.add(AudioProfile::createFullDynamic());
     }
     mixPort->setAudioProfiles(profiles);
 
@@ -328,12 +323,7 @@
     AudioProfileTraits::Collection profiles;
     deserializeCollection<AudioProfileTraits>(doc, root, profiles, NULL);
     if (profiles.isEmpty()) {
-        sp <AudioProfile> dynamicProfile = new AudioProfile(gDynamicFormat,
-                                                            ChannelsVector(), SampleRateVector());
-        dynamicProfile->setDynamicFormat(true);
-        dynamicProfile->setDynamicChannels(true);
-        dynamicProfile->setDynamicRate(true);
-        profiles.add(dynamicProfile);
+        profiles.add(AudioProfile::createFullDynamic());
     }
     deviceDesc->setAudioProfiles(profiles);
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp b/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
index d34214b..38ab560 100644
--- a/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
@@ -129,7 +129,7 @@
         if (streamType == route->mStreamType && route->isActiveOrChanged()
                 && route->mDeviceDescriptor != 0) {
             device = route->mDeviceDescriptor->type();
-            if (!availableDevices.getDevicesFromType(device).isEmpty()) {
+            if (!availableDevices.getDevicesFromTypeMask(device).isEmpty()) {
                 break;
             }
         }
diff --git a/services/audiopolicy/config/audio_policy_configuration.xml b/services/audiopolicy/config/audio_policy_configuration.xml
index a75f1cb..9381f1f 100644
--- a/services/audiopolicy/config/audio_policy_configuration.xml
+++ b/services/audiopolicy/config/audio_policy_configuration.xml
@@ -185,6 +185,9 @@
         <!-- Hearing aid Audio HAL -->
         <xi:include href="hearing_aid_audio_policy_configuration.xml"/>
 
+        <!-- MSD Audio HAL (optional) -->
+        <xi:include href="msd_audio_policy_configuration.xml"/>
+
     </modules>
     <!-- End of Modules section -->
 
diff --git a/services/audiopolicy/config/msd_audio_policy_configuration.xml b/services/audiopolicy/config/msd_audio_policy_configuration.xml
new file mode 100644
index 0000000..a84117e
--- /dev/null
+++ b/services/audiopolicy/config/msd_audio_policy_configuration.xml
@@ -0,0 +1,62 @@
+<?xml version="1.0" encoding="UTF-8"?>
+<!-- Copyright (C) 2017-2018 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.
+-->
+<!-- Multi Stream Decoder Audio Policy Configuration file -->
+<module name="msd" halVersion="2.0">
+    <attachedDevices>
+        <item>MS12 Input</item>
+        <item>MS12 Output</item>
+    </attachedDevices>
+    <mixPorts>
+        <mixPort name="ms12 input" role="source">
+            <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                     samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+        </mixPort>
+        <mixPort name="ms12 compressed input" role="source"
+                flags="AUDIO_OUTPUT_FLAG_DIRECT|AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD|AUDIO_OUTPUT_FLAG_NON_BLOCKING">
+            <profile name="" format="AUDIO_FORMAT_AC3"
+                     samplingRates="32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_5POINT1"/>
+            <profile name="" format="AUDIO_FORMAT_E_AC3"
+                     samplingRates="32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+            <profile name="" format="AUDIO_FORMAT_AC4"
+                     samplingRates="32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO,AUDIO_CHANNEL_OUT_5POINT1,AUDIO_CHANNEL_OUT_7POINT1"/>
+        </mixPort>
+        <mixPort name="ms12 output" role="sink">
+            <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                     samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+            <profile name="" format="AUDIO_FORMAT_AC3"
+                     samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+            <profile name="" format="AUDIO_FORMAT_E_AC3"
+                     samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_5POINT1"/>
+        </mixPort>
+   </mixPorts>
+   <devicePorts>
+       <devicePort tagName="MS12 Input" type="AUDIO_DEVICE_OUT_BUS"  role="sink">
+           <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                    samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+       </devicePort>
+       <devicePort tagName="MS12 Output" type="AUDIO_DEVICE_IN_BUS"  role="source">
+           <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                    samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+        </devicePort>
+    </devicePorts>
+    <routes>
+        <route type="mix" sink="MS12 Input" sources="ms12 input,ms12 compressed input"/>
+        <route type="mix" sink="ms12 output" sources="MS12 Output"/>
+    </routes>
+</module>
diff --git a/services/audiopolicy/engineconfigurable/wrapper/Android.mk b/services/audiopolicy/engineconfigurable/wrapper/Android.mk
index 36e0f42..b128a38 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/Android.mk
+++ b/services/audiopolicy/engineconfigurable/wrapper/Android.mk
@@ -10,7 +10,6 @@
     $(LOCAL_PATH)/include \
     frameworks/av/services/audiopolicy/engineconfigurable/include \
     frameworks/av/services/audiopolicy/engineconfigurable/interface \
-    frameworks/av/services/audiopolicy/utilities/convert \
 
 LOCAL_SRC_FILES:= ParameterManagerWrapper.cpp
 
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 3e13e50..941119b 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -408,8 +408,7 @@
 
     case STRATEGY_SONIFICATION:
 
-        // If incall, just select the STRATEGY_PHONE device: The rest of the behavior is handled by
-        // handleIncallSonification().
+        // If incall, just select the STRATEGY_PHONE device
         if (isInCall() || outputs.isStreamActiveLocally(AUDIO_STREAM_VOICE_CALL)) {
             device = getDeviceForStrategyInt(
                     STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
@@ -482,7 +481,7 @@
                 }
             }
             availableOutputDevices =
-                    availableOutputDevices.getDevicesFromType(availableOutputDevicesType);
+                    availableOutputDevices.getDevicesFromTypeMask(availableOutputDevicesType);
             if (outputs.isStreamActive(AUDIO_STREAM_RING) ||
                     outputs.isStreamActive(AUDIO_STREAM_ALARM)) {
                 return getDeviceForStrategyInt(
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index a546c8f..6e66d3e 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -38,6 +38,8 @@
 #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>
 #include <audio_policy_conf.h>
@@ -476,20 +478,15 @@
 
 sp<AudioPatch> AudioPolicyManager::createTelephonyPatch(
         bool isRx, audio_devices_t device, uint32_t delayMs) {
-    struct audio_patch patch;
-    patch.num_sources = 1;
-    patch.num_sinks = 1;
+    PatchBuilder patchBuilder;
 
     sp<DeviceDescriptor> txSourceDeviceDesc;
     if (isRx) {
-        fillAudioPortConfigForDevice(mAvailableOutputDevices, device, &patch.sinks[0]);
-        fillAudioPortConfigForDevice(
-                mAvailableInputDevices, AUDIO_DEVICE_IN_TELEPHONY_RX, &patch.sources[0]);
+        patchBuilder.addSink(findDevice(mAvailableOutputDevices, device)).
+                addSource(findDevice(mAvailableInputDevices, AUDIO_DEVICE_IN_TELEPHONY_RX));
     } else {
-        txSourceDeviceDesc = fillAudioPortConfigForDevice(
-                mAvailableInputDevices, device, &patch.sources[0]);
-        fillAudioPortConfigForDevice(
-                mAvailableOutputDevices, AUDIO_DEVICE_OUT_TELEPHONY_TX, &patch.sinks[0]);
+        patchBuilder.addSource(txSourceDeviceDesc = findDevice(mAvailableInputDevices, device)).
+                addSink(findDevice(mAvailableOutputDevices, AUDIO_DEVICE_OUT_TELEPHONY_TX));
     }
 
     audio_devices_t outputDevice = isRx ? device : AUDIO_DEVICE_OUT_TELEPHONY_TX;
@@ -500,9 +497,7 @@
         sp<AudioOutputDescriptor> outputDesc = mOutputs.valueFor(output);
         ALOG_ASSERT(!outputDesc->isDuplicated(),
                 "%s() %#x device output %d is duplicated", __func__, outputDevice, output);
-        outputDesc->toAudioPortConfig(&patch.sources[1]);
-        patch.sources[1].ext.mix.usecase.stream = AUDIO_STREAM_PATCH;
-        patch.num_sources = 2;
+        patchBuilder.addSource(outputDesc, { .stream = AUDIO_STREAM_PATCH });
     }
 
     if (!isRx) {
@@ -524,26 +519,25 @@
     }
 
     audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-    status_t status = mpClientInterface->createAudioPatch(&patch, &afPatchHandle, delayMs);
+    status_t status = mpClientInterface->createAudioPatch(
+            patchBuilder.patch(), &afPatchHandle, delayMs);
     ALOGW_IF(status != NO_ERROR,
             "%s() error %d creating %s audio patch", __func__, status, isRx ? "RX" : "TX");
     sp<AudioPatch> audioPatch;
     if (status == NO_ERROR) {
-        audioPatch = new AudioPatch(&patch, mUidCached);
+        audioPatch = new AudioPatch(patchBuilder.patch(), mUidCached);
         audioPatch->mAfPatchHandle = afPatchHandle;
         audioPatch->mUid = mUidCached;
     }
     return audioPatch;
 }
 
-sp<DeviceDescriptor> AudioPolicyManager::fillAudioPortConfigForDevice(
-        const DeviceVector& devices, audio_devices_t device, audio_port_config *config) {
-    DeviceVector deviceList = devices.getDevicesFromType(device);
+sp<DeviceDescriptor> AudioPolicyManager::findDevice(
+        const DeviceVector& devices, audio_devices_t device) {
+    DeviceVector deviceList = devices.getDevicesFromTypeMask(device);
     ALOG_ASSERT(!deviceList.isEmpty(),
             "%s() selected device type %#x is not in devices list", __func__, device);
-    sp<DeviceDescriptor> deviceDesc = deviceList.itemAt(0);
-    deviceDesc->toAudioPortConfig(config);
-    return deviceDesc;
+    return deviceList.itemAt(0);
 }
 
 void AudioPolicyManager::setPhoneState(audio_mode_t state)
@@ -557,14 +551,8 @@
         return;
     }
     /// Opens: can these line be executed after the switch of volume curves???
-    // if leaving call state, handle special case of active streams
-    // pertaining to sonification strategy see handleIncallSonification()
     if (isStateInCall(oldState)) {
         ALOGV("setPhoneState() in call state management: new state is %d", state);
-        for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
-            handleIncallSonification((audio_stream_type_t)stream, false, true);
-        }
-
         // force reevaluating accessibility routing when call stops
         mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
     }
@@ -643,14 +631,8 @@
         }
     }
 
-    // if entering in call state, handle special case of active streams
-    // pertaining to sonification strategy see handleIncallSonification()
     if (isStateInCall(state)) {
         ALOGV("setPhoneState() in call state management: new state is %d", state);
-        for (int stream = 0; stream < AUDIO_STREAM_FOR_POLICY_CNT; stream++) {
-            handleIncallSonification((audio_stream_type_t)stream, true, true);
-        }
-
         // force reevaluating accessibility routing when call starts
         mpClientInterface->invalidateStream(AUDIO_STREAM_ACCESSIBILITY);
     }
@@ -871,7 +853,7 @@
         return INVALID_OPERATION;
     }
 
-    DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromType(device);
+    DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
     *selectedDeviceId = outputDevices.size() > 0 ? outputDevices.itemAt(0)->getId()
             : AUDIO_PORT_HANDLE_NONE;
 
@@ -976,7 +958,7 @@
         sp<SwAudioOutputDescriptor> outputDesc =
                 new SwAudioOutputDescriptor(profile, mpClientInterface);
 
-        DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromType(device);
+        DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
         String8 address = outputDevices.size() > 0 ? outputDevices.itemAt(0)->mAddress
                 : String8("");
 
@@ -1285,11 +1267,6 @@
         const uint32_t muteWaitMs =
                 setOutputDevice(outputDesc, device, force, 0, NULL, address, requiresMuteCheck);
 
-        // handle special case for sonification while in call
-        if (isInCall()) {
-            handleIncallSonification(stream, true, false);
-        }
-
         // apply volume rules for current stream and device if necessary
         checkAndSetVolume(stream,
                           mVolumeCurves->getVolumeIndex(stream, outputDesc->device()),
@@ -1384,11 +1361,6 @@
     // always handle stream stop, check which stream type is stopping
     handleEventForBeacon(stream == AUDIO_STREAM_TTS ? STOPPING_BEACON : STOPPING_OUTPUT);
 
-    // handle special case for sonification while in call
-    if (isInCall()) {
-        handleIncallSonification(stream, false, false);
-    }
-
     if (outputDesc->mRefCount[stream] > 0) {
         // decrement usage count of this stream on the output
         outputDesc->changeRefCount(stream, -1);
@@ -1545,7 +1517,7 @@
         if (*portId == AUDIO_PORT_HANDLE_NONE) {
             *portId = AudioPort::getNextUniqueId();
         }
-        inputDevices = mAvailableInputDevices.getDevicesFromType(inputDesc->mDevice);
+        inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(inputDesc->mDevice);
         *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId()
                 : AUDIO_PORT_HANDLE_NONE;
         ALOGI("%s reusing MMAP input %d for session %d", __FUNCTION__, *input, session);
@@ -1612,7 +1584,7 @@
         goto error;
     }
 
-    inputDevices = mAvailableInputDevices.getDevicesFromType(device);
+    inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
     *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId()
             : AUDIO_PORT_HANDLE_NONE;
 
@@ -1780,7 +1752,7 @@
     lConfig.format = profileFormat;
 
     if (address == "") {
-        DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(device);
+        DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
         // the inputs vector must be of size >= 1, but we don't want to crash here
         address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress : String8("");
     }
@@ -2279,11 +2251,10 @@
         sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
         audio_devices_t curDevice = Volume::getDeviceForVolume(desc->device());
         for (int curStream = 0; curStream < AUDIO_STREAM_FOR_POLICY_CNT; curStream++) {
-            if (!streamsMatchForvolume(stream, (audio_stream_type_t)curStream)) {
+            if (!(streamsMatchForvolume(stream, (audio_stream_type_t)curStream) || isInCall())) {
                 continue;
             }
-            if (!(desc->isStreamActive((audio_stream_type_t)curStream) ||
-                    (isInCall() && (curStream == AUDIO_STREAM_VOICE_CALL)))) {
+            if (!(desc->isStreamActive((audio_stream_type_t)curStream) || isInCall())) {
                 continue;
             }
             routing_strategy curStrategy = getStrategy((audio_stream_type_t)curStream);
@@ -2627,42 +2598,24 @@
 
 status_t AudioPolicyManager::dump(int fd)
 {
-    const size_t SIZE = 256;
-    char buffer[SIZE];
     String8 result;
-
-    snprintf(buffer, SIZE, "\nAudioPolicyManager Dump: %p\n", this);
-    result.append(buffer);
-
-    snprintf(buffer, SIZE, " Primary Output: %d\n",
+    result.appendFormat("\nAudioPolicyManager Dump: %p\n", this);
+    result.appendFormat(" Primary Output: %d\n",
              hasPrimaryOutput() ? mPrimaryOutput->mIoHandle : AUDIO_IO_HANDLE_NONE);
-    result.append(buffer);
     std::string stateLiteral;
     AudioModeConverter::toString(mEngine->getPhoneState(), stateLiteral);
-    snprintf(buffer, SIZE, " Phone state: %s\n", stateLiteral.c_str());
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for communications %d\n",
-             mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for media %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for record %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for dock %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for system %d\n", mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for hdmi system audio %d\n",
-            mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Force use for encoded surround output %d\n",
-            mEngine->getForceUse(AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND));
-    result.append(buffer);
-    snprintf(buffer, SIZE, " TTS output %s\n", mTtsOutputAvailable ? "available" : "not available");
-    result.append(buffer);
-    snprintf(buffer, SIZE, " Master mono: %s\n", mMasterMono ? "on" : "off");
-    result.append(buffer);
-
+    result.appendFormat(" Phone state: %s\n", stateLiteral.c_str());
+    const char* forceUses[AUDIO_POLICY_FORCE_USE_CNT] = {
+        "communications", "media", "record", "dock", "system",
+        "HDMI system audio", "encoded surround output", "vibrate ringing" };
+    for (audio_policy_force_use_t i = AUDIO_POLICY_FORCE_FOR_COMMUNICATION;
+         i < AUDIO_POLICY_FORCE_USE_CNT; i = (audio_policy_force_use_t)((int)i + 1)) {
+        result.appendFormat(" Force use for %s: %d\n",
+                forceUses[i], mEngine->getForceUse(i));
+    }
+    result.appendFormat(" TTS output %savailable\n", mTtsOutputAvailable ? "" : "not ");
+    result.appendFormat(" Master mono: %s\n", mMasterMono ? "on" : "off");
+    result.appendFormat(" Config source: %s\n", getConfig().getSource().c_str());
     write(fd, result.string(), result.size());
 
     mAvailableOutputDevices.dump(fd, String8("Available output"));
@@ -3078,28 +3031,8 @@
             }
             // TODO: check from routing capabilities in config file and other conflicting patches
 
-            audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-            if (index >= 0) {
-                afPatchHandle = patchDesc->mAfPatchHandle;
-            }
-
-            status_t status = mpClientInterface->createAudioPatch(&newPatch,
-                                                                  &afPatchHandle,
-                                                                  0);
-            ALOGV("createAudioPatch() patch panel returned %d patchHandle %d",
-                                                                  status, afPatchHandle);
-            if (status == NO_ERROR) {
-                if (index < 0) {
-                    patchDesc = new AudioPatch(&newPatch, uid);
-                    addAudioPatch(patchDesc->mHandle, patchDesc);
-                } else {
-                    patchDesc->mPatch = newPatch;
-                }
-                patchDesc->mAfPatchHandle = afPatchHandle;
-                *handle = patchDesc->mHandle;
-                nextAudioPortGeneration();
-                mpClientInterface->onAudioPatchListUpdate();
-            } else {
+            status_t status = installPatch(__func__, index, handle, &newPatch, 0, uid, &patchDesc);
+            if (status != NO_ERROR) {
                 ALOGW("createAudioPatch() patch panel could not connect device patch, error %d",
                 status);
                 return INVALID_OPERATION;
@@ -3232,10 +3165,10 @@
         return BAD_VALUE;
     }
 
-    struct audio_port_config backupConfig;
+    struct audio_port_config backupConfig = {};
     status_t status = audioPortConfig->applyAudioPortConfig(config, &backupConfig);
     if (status == NO_ERROR) {
-        struct audio_port_config newConfig;
+        struct audio_port_config newConfig = {};
         audioPortConfig->toAudioPortConfig(&newConfig, config);
         status = mpClientInterface->setAudioPortConfig(&newConfig, 0);
     }
@@ -3383,7 +3316,7 @@
     sp<AudioSourceDescriptor> sourceDesc =
             new AudioSourceDescriptor(srcDeviceDesc, attributes, uid);
 
-    struct audio_patch dummyPatch;
+    struct audio_patch dummyPatch = {};
     sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
     sourceDesc->mPatchDesc = patchDesc;
 
@@ -3411,7 +3344,6 @@
             mAvailableOutputDevices.getDevice(sinkDevice, String8(""));
 
     audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-    struct audio_patch *patch = &sourceDesc->mPatchDesc->mPatch;
 
     if (srcDeviceDesc->getAudioPort()->mModule->getHandle() ==
             sinkDeviceDesc->getAudioPort()->mModule->getHandle() &&
@@ -3443,16 +3375,14 @@
         // be connected as well as the stream type for volume control
         // - the sink is defined by whatever output device is currently selected for the output
         // though which this patch is routed.
-        patch->num_sinks = 0;
-        patch->num_sources = 2;
-        srcDeviceDesc->toAudioPortConfig(&patch->sources[0], NULL);
-        outputDesc->toAudioPortConfig(&patch->sources[1], NULL);
-        patch->sources[1].ext.mix.usecase.stream = stream;
-        status = mpClientInterface->createAudioPatch(patch,
+        PatchBuilder patchBuilder;
+        patchBuilder.addSource(srcDeviceDesc).addSource(outputDesc, { .stream = stream });
+        status = mpClientInterface->createAudioPatch(patchBuilder.patch(),
                                                               &afPatchHandle,
                                                               0);
         ALOGV("%s patch panel returned %d patchHandle %d", __FUNCTION__,
                                                               status, afPatchHandle);
+        sourceDesc->mPatchDesc->mPatch = *patchBuilder.patch();
         if (status != NO_ERROR) {
             ALOGW("%s patch panel could not connect device patch, error %d",
                   __FUNCTION__, status);
@@ -3615,7 +3545,7 @@
             }
         }
         // Open an output to query dynamic parameters.
-        DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromType(
+        DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(
                 AUDIO_DEVICE_OUT_HDMI);
         for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
             String8 address = hdmiOutputDevices[i]->mAddress;
@@ -3741,7 +3671,7 @@
 
     sp<SwAudioOutputDescriptor> outputDesc;
     bool profileUpdated = false;
-    DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromType(
+    DeviceVector hdmiOutputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(
             AUDIO_DEVICE_OUT_HDMI);
     for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
         // Simulate reconnection to update enabled surround sound formats.
@@ -3760,7 +3690,7 @@
                                              name.c_str());
         profileUpdated |= (status == NO_ERROR);
     }
-    DeviceVector hdmiInputDevices = mAvailableInputDevices.getDevicesFromType(
+    DeviceVector hdmiInputDevices = mAvailableInputDevices.getDevicesFromTypeMask(
                 AUDIO_DEVICE_IN_HDMI);
     for (size_t i = 0; i < hdmiInputDevices.size(); i++) {
         // Simulate reconnection to update enabled surround sound formats.
@@ -3904,6 +3834,7 @@
                      "%s/%s", kConfigLocationList[i], fileName);
             ret = serializer.deserialize(audioPolicyXmlConfigFile, config);
             if (ret == NO_ERROR) {
+                config.setSource(audioPolicyXmlConfigFile);
                 return ret;
             }
         }
@@ -3915,7 +3846,7 @@
 AudioPolicyManager::AudioPolicyManager(AudioPolicyClientInterface *clientInterface,
                                        bool /*forTesting*/)
     :
-    mUidCached(getuid()),
+    mUidCached(AID_AUDIOSERVER), // no need to call getuid(), there's only one of us running.
     mpClientInterface(clientInterface),
     mLimitRingtoneVolume(false), mLastVoiceVolume(-1.0f),
     mA2dpSuspended(false),
@@ -4026,7 +3957,8 @@
             sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
                                                                                  mpClientInterface);
             const DeviceVector &supportedDevices = outProfile->getSupportedDevices();
-            const DeviceVector &devicesForType = supportedDevices.getDevicesFromType(profileType);
+            const DeviceVector &devicesForType = supportedDevices.getDevicesFromTypeMask(
+                    profileType);
             String8 address = devicesForType.size() > 0 ? devicesForType.itemAt(0)->mAddress
                     : String8("");
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
@@ -4080,7 +4012,7 @@
             sp<AudioInputDescriptor> inputDesc =
                     new AudioInputDescriptor(inProfile, mpClientInterface);
 
-            DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromType(profileType);
+            DeviceVector inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(profileType);
             //   the inputs vector must be of size >= 1, but we don't want to crash here
             String8 address = inputDevices.size() > 0 ? inputDevices.itemAt(0)->mAddress
                     : String8("");
@@ -5249,54 +5181,20 @@
     } else {
         DeviceVector deviceList;
         if ((address == NULL) || (strlen(address) == 0)) {
-            deviceList = mAvailableOutputDevices.getDevicesFromType(device);
+            deviceList = mAvailableOutputDevices.getDevicesFromTypeMask(device);
         } else {
-            deviceList = mAvailableOutputDevices.getDevicesFromTypeAddr(device, String8(address));
+            sp<DeviceDescriptor> deviceDesc = mAvailableOutputDevices.getDevice(
+                    device, String8(address));
+            if (deviceDesc) deviceList.add(deviceDesc);
         }
 
         if (!deviceList.isEmpty()) {
-            struct audio_patch patch;
-            outputDesc->toAudioPortConfig(&patch.sources[0]);
-            patch.num_sources = 1;
-            patch.num_sinks = 0;
+            PatchBuilder patchBuilder;
+            patchBuilder.addSource(outputDesc);
             for (size_t i = 0; i < deviceList.size() && i < AUDIO_PATCH_PORTS_MAX; i++) {
-                deviceList.itemAt(i)->toAudioPortConfig(&patch.sinks[i]);
-                patch.num_sinks++;
+                patchBuilder.addSink(deviceList.itemAt(i));
             }
-            ssize_t index;
-            if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
-                index = mAudioPatches.indexOfKey(*patchHandle);
-            } else {
-                index = mAudioPatches.indexOfKey(outputDesc->getPatchHandle());
-            }
-            sp< AudioPatch> patchDesc;
-            audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-            if (index >= 0) {
-                patchDesc = mAudioPatches.valueAt(index);
-                afPatchHandle = patchDesc->mAfPatchHandle;
-            }
-
-            status_t status = mpClientInterface->createAudioPatch(&patch,
-                                                                   &afPatchHandle,
-                                                                   delayMs);
-            ALOGV("setOutputDevice() createAudioPatch returned %d patchHandle %d"
-                    "num_sources %d num_sinks %d",
-                                       status, afPatchHandle, patch.num_sources, patch.num_sinks);
-            if (status == NO_ERROR) {
-                if (index < 0) {
-                    patchDesc = new AudioPatch(&patch, mUidCached);
-                    addAudioPatch(patchDesc->mHandle, patchDesc);
-                } else {
-                    patchDesc->mPatch = patch;
-                }
-                patchDesc->mAfPatchHandle = afPatchHandle;
-                if (patchHandle) {
-                    *patchHandle = patchDesc->mHandle;
-                }
-                outputDesc->setPatchHandle(patchDesc->mHandle);
-                nextAudioPortGeneration();
-                mpClientInterface->onAudioPatchListUpdate();
-            }
+            installPatch(__func__, patchHandle, outputDesc.get(), patchBuilder.patch(), delayMs);
         }
 
         // inform all input as well
@@ -5354,53 +5252,21 @@
     if ((device != AUDIO_DEVICE_NONE) && ((device != inputDesc->mDevice) || force)) {
         inputDesc->mDevice = device;
 
-        DeviceVector deviceList = mAvailableInputDevices.getDevicesFromType(device);
+        DeviceVector deviceList = mAvailableInputDevices.getDevicesFromTypeMask(device);
         if (!deviceList.isEmpty()) {
-            struct audio_patch patch;
-            inputDesc->toAudioPortConfig(&patch.sinks[0]);
+            PatchBuilder patchBuilder;
+            patchBuilder.addSink(inputDesc,
             // AUDIO_SOURCE_HOTWORD is for internal use only:
             // handled as AUDIO_SOURCE_VOICE_RECOGNITION by the audio HAL
-            if (patch.sinks[0].ext.mix.usecase.source == AUDIO_SOURCE_HOTWORD &&
-                    !inputDesc->isSoundTrigger()) {
-                patch.sinks[0].ext.mix.usecase.source = AUDIO_SOURCE_VOICE_RECOGNITION;
-            }
-            patch.num_sinks = 1;
+                    [inputDesc](const PatchBuilder::mix_usecase_t& usecase) {
+                        auto result = usecase;
+                        if (result.source == AUDIO_SOURCE_HOTWORD && !inputDesc->isSoundTrigger()) {
+                            result.source = AUDIO_SOURCE_VOICE_RECOGNITION;
+                        }
+                        return result; }).
             //only one input device for now
-            deviceList.itemAt(0)->toAudioPortConfig(&patch.sources[0]);
-            patch.num_sources = 1;
-            ssize_t index;
-            if (patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE) {
-                index = mAudioPatches.indexOfKey(*patchHandle);
-            } else {
-                index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
-            }
-            sp< AudioPatch> patchDesc;
-            audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
-            if (index >= 0) {
-                patchDesc = mAudioPatches.valueAt(index);
-                afPatchHandle = patchDesc->mAfPatchHandle;
-            }
-
-            status_t status = mpClientInterface->createAudioPatch(&patch,
-                                                                  &afPatchHandle,
-                                                                  0);
-            ALOGV("setInputDevice() createAudioPatch returned %d patchHandle %d",
-                                                                          status, afPatchHandle);
-            if (status == NO_ERROR) {
-                if (index < 0) {
-                    patchDesc = new AudioPatch(&patch, mUidCached);
-                    addAudioPatch(patchDesc->mHandle, patchDesc);
-                } else {
-                    patchDesc->mPatch = patch;
-                }
-                patchDesc->mAfPatchHandle = afPatchHandle;
-                if (patchHandle) {
-                    *patchHandle = patchDesc->mHandle;
-                }
-                inputDesc->setPatchHandle(patchDesc->mHandle);
-                nextAudioPortGeneration();
-                mpClientInterface->onAudioPatchListUpdate();
-            }
+                    addSource(deviceList.itemAt(0));
+            status = installPatch(__func__, patchHandle, inputDesc.get(), patchBuilder.patch(), 0);
         }
     }
     return status;
@@ -5534,8 +5400,8 @@
         return ringVolumeDB - 4 > volumeDB ? ringVolumeDB - 4 : volumeDB;
     }
 
-    // in-call: always cap earpiece volume by voice volume + some low headroom
-    if ((stream != AUDIO_STREAM_VOICE_CALL) && (device & AUDIO_DEVICE_OUT_EARPIECE) &&
+    // in-call: always cap volume by voice volume + some low headroom
+    if ((stream != AUDIO_STREAM_VOICE_CALL) &&
             (isInCall() || mOutputs.isStreamActiveLocally(AUDIO_STREAM_VOICE_CALL))) {
         switch (stream) {
         case AUDIO_STREAM_SYSTEM:
@@ -5547,9 +5413,9 @@
         case AUDIO_STREAM_DTMF:
         case AUDIO_STREAM_ACCESSIBILITY: {
             int voiceVolumeIndex =
-                mVolumeCurves->getVolumeIndex(AUDIO_STREAM_VOICE_CALL, AUDIO_DEVICE_OUT_EARPIECE);
+                mVolumeCurves->getVolumeIndex(AUDIO_STREAM_VOICE_CALL, device);
             const float maxVoiceVolDb =
-                computeVolume(AUDIO_STREAM_VOICE_CALL, voiceVolumeIndex, AUDIO_DEVICE_OUT_EARPIECE)
+                computeVolume(AUDIO_STREAM_VOICE_CALL, voiceVolumeIndex, device)
                 + IN_CALL_EARPIECE_HEADROOM_DB;
             if (volumeDB > maxVoiceVolDb) {
                 ALOGV("computeVolume() stream %d at vol=%f overriden by stream %d at vol=%f",
@@ -5746,55 +5612,6 @@
     }
 }
 
-void AudioPolicyManager::handleIncallSonification(audio_stream_type_t stream,
-                                                      bool starting, bool stateChange)
-{
-    if(!hasPrimaryOutput()) {
-        return;
-    }
-
-    // if the stream pertains to sonification strategy and we are in call we must
-    // mute the stream if it is low visibility. If it is high visibility, we must play a tone
-    // in the device used for phone strategy and play the tone if the selected device does not
-    // interfere with the device used for phone strategy
-    // if stateChange is true, we are called from setPhoneState() and we must mute or unmute as
-    // many times as there are active tracks on the output
-    const routing_strategy stream_strategy = getStrategy(stream);
-    if ((stream_strategy == STRATEGY_SONIFICATION) ||
-            ((stream_strategy == STRATEGY_SONIFICATION_RESPECTFUL))) {
-        sp<SwAudioOutputDescriptor> outputDesc = mPrimaryOutput;
-        ALOGV("handleIncallSonification() stream %d starting %d device %x stateChange %d",
-                stream, starting, outputDesc->mDevice, stateChange);
-        if (outputDesc->mRefCount[stream]) {
-            int muteCount = 1;
-            if (stateChange) {
-                muteCount = outputDesc->mRefCount[stream];
-            }
-            if (audio_is_low_visibility(stream)) {
-                ALOGV("handleIncallSonification() low visibility, muteCount %d", muteCount);
-                for (int i = 0; i < muteCount; i++) {
-                    setStreamMute(stream, starting, mPrimaryOutput);
-                }
-            } else {
-                ALOGV("handleIncallSonification() high visibility");
-                if (outputDesc->device() &
-                        getDeviceForStrategy(STRATEGY_PHONE, true /*fromCache*/)) {
-                    ALOGV("handleIncallSonification() high visibility muted, muteCount %d", muteCount);
-                    for (int i = 0; i < muteCount; i++) {
-                        setStreamMute(stream, starting, mPrimaryOutput);
-                    }
-                }
-                if (starting) {
-                    mpClientInterface->startTone(AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION,
-                                                 AUDIO_STREAM_VOICE_CALL);
-                } else {
-                    mpClientInterface->stopTone();
-                }
-            }
-        }
-    }
-}
-
 audio_stream_type_t AudioPolicyManager::streamTypefromAttributesInt(const audio_attributes_t *attr)
 {
     // flags to stream type mapping
@@ -6170,4 +5987,58 @@
     }
 }
 
+status_t AudioPolicyManager::installPatch(const char *caller,
+                                          audio_patch_handle_t *patchHandle,
+                                          AudioIODescriptorInterface *ioDescriptor,
+                                          const struct audio_patch *patch,
+                                          int delayMs)
+{
+    ssize_t index = mAudioPatches.indexOfKey(
+            patchHandle && *patchHandle != AUDIO_PATCH_HANDLE_NONE ?
+            *patchHandle : ioDescriptor->getPatchHandle());
+    sp<AudioPatch> patchDesc;
+    status_t status = installPatch(
+            caller, index, patchHandle, patch, delayMs, mUidCached, &patchDesc);
+    if (status == NO_ERROR) {
+        ioDescriptor->setPatchHandle(patchDesc->mHandle);
+    }
+    return status;
+}
+
+status_t AudioPolicyManager::installPatch(const char *caller,
+                                          ssize_t index,
+                                          audio_patch_handle_t *patchHandle,
+                                          const struct audio_patch *patch,
+                                          int delayMs,
+                                          uid_t uid,
+                                          sp<AudioPatch> *patchDescPtr)
+{
+    sp<AudioPatch> patchDesc;
+    audio_patch_handle_t afPatchHandle = AUDIO_PATCH_HANDLE_NONE;
+    if (index >= 0) {
+        patchDesc = mAudioPatches.valueAt(index);
+        afPatchHandle = patchDesc->mAfPatchHandle;
+    }
+
+    status_t status = mpClientInterface->createAudioPatch(patch, &afPatchHandle, delayMs);
+    ALOGV("%s() AF::createAudioPatch returned %d patchHandle %d num_sources %d num_sinks %d",
+            caller, status, afPatchHandle, patch->num_sources, patch->num_sinks);
+    if (status == NO_ERROR) {
+        if (index < 0) {
+            patchDesc = new AudioPatch(patch, uid);
+            addAudioPatch(patchDesc->mHandle, patchDesc);
+        } else {
+            patchDesc->mPatch = *patch;
+        }
+        patchDesc->mAfPatchHandle = afPatchHandle;
+        if (patchHandle) {
+            *patchHandle = patchDesc->mHandle;
+        }
+        nextAudioPortGeneration();
+        mpClientInterface->onAudioPatchListUpdate();
+    }
+    if (patchDescPtr) *patchDescPtr = patchDesc;
+    return status;
+}
+
 } // namespace android
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 48e0472..136e522 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -377,10 +377,6 @@
                            int delayMs = 0,
                            audio_devices_t device = (audio_devices_t)0);
 
-        // handle special cases for sonification strategy while in call: mute streams or replace by
-        // a special tone in the device used for communication
-        void handleIncallSonification(audio_stream_type_t stream, bool starting, bool stateChange);
-
         audio_mode_t getPhoneState();
 
         // true if device is in a telephony or VoIP call
@@ -506,8 +502,8 @@
 
         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> fillAudioPortConfigForDevice(
-                const DeviceVector& devices, audio_devices_t device, audio_port_config *config);
+        sp<DeviceDescriptor> findDevice(
+                const DeviceVector& devices, audio_devices_t device);
 
         // if argument "device" is different from AUDIO_DEVICE_NONE,  startSource() will force
         // the re-evaluation of the output device.
@@ -544,7 +540,7 @@
         static bool streamsMatchForvolume(audio_stream_type_t stream1,
                                           audio_stream_type_t stream2);
 
-        uid_t mUidCached;
+        const uid_t mUidCached;                         // AID_AUDIOSERVER
         AudioPolicyClientInterface *mpClientInterface;  // audio policy client interface
         sp<SwAudioOutputDescriptor> mPrimaryOutput;     // primary output descriptor
         // list of descriptors for outputs currently opened
@@ -684,6 +680,18 @@
             param.addInt(String8(AudioParameter::keyMonoOutput), (int)mMasterMono);
             mpClientInterface->setParameters(output, param.toString());
         }
+        status_t installPatch(const char *caller,
+                audio_patch_handle_t *patchHandle,
+                AudioIODescriptorInterface *ioDescriptor,
+                const struct audio_patch *patch,
+                int delayMs);
+        status_t installPatch(const char *caller,
+                ssize_t index,
+                audio_patch_handle_t *patchHandle,
+                const struct audio_patch *patch,
+                int delayMs,
+                uid_t uid,
+                sp<AudioPatch> *patchDescPtr);
 
         bool soundTriggerSupportsConcurrentCapture();
         bool mSoundTriggerSupportsConcurrentCapture;
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index b064f8c..21fffec 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -155,17 +155,6 @@
     return result;
 }
 
-status_t AudioPolicyService::AudioPolicyClient::startTone(audio_policy_tone_t tone,
-              audio_stream_type_t stream)
-{
-    return mAudioPolicyService->startTone(tone, stream);
-}
-
-status_t AudioPolicyService::AudioPolicyClient::stopTone()
-{
-    return mAudioPolicyService->stopTone();
-}
-
 status_t AudioPolicyService::AudioPolicyClient::setVoiceVolume(float volume, int delay_ms)
 {
     return mAudioPolicyService->setVoiceVolume(volume, delay_ms);
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index c7dfe0f..fdae23b 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -24,6 +24,7 @@
 #include <cutils/misc.h>
 #include <media/AudioEffect.h>
 #include <media/EffectsConfig.h>
+#include <mediautils/ServiceUtilities.h>
 #include <system/audio.h>
 #include <system/audio_effects/audio_effects_conf.h>
 #include <utils/Vector.h>
@@ -31,7 +32,6 @@
 #include <cutils/config_utils.h>
 #include <binder/IPCThreadState.h>
 #include "AudioPolicyEffects.h"
-#include "ServiceUtilities.h"
 
 namespace android {
 
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 7337f04..d2bc40d 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -21,7 +21,7 @@
 #include <media/MediaAnalyticsItem.h>
 
 #include "AudioPolicyService.h"
-#include "ServiceUtilities.h"
+#include <mediautils/ServiceUtilities.h>
 #include "TypeConverter.h"
 
 namespace android {
@@ -183,7 +183,7 @@
     Mutex::Autolock _l(mLock);
 
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    if (!isTrustedCallingUid(callingUid) || uid == (uid_t)-1) {
+    if (!isAudioServerOrMediaServerUid(callingUid) || uid == (uid_t)-1) {
         ALOGW_IF(uid != (uid_t)-1 && uid != callingUid,
                 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, uid);
         uid = callingUid;
@@ -320,7 +320,7 @@
 
     bool updatePid = (pid == -1);
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    if (!isTrustedCallingUid(callingUid)) {
+    if (!isAudioServerOrMediaServerUid(callingUid)) {
         ALOGW_IF(uid != (uid_t)-1 && uid != callingUid,
                 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid, uid);
         uid = callingUid;
@@ -528,10 +528,13 @@
 
             item->setCString(kAudioPolicyRqstSrc,
                              audioSourceString(client->attributes.source).c_str());
-            item->setCString(kAudioPolicyRqstPkg,
-                             std::string(String8(client->opPackageName).string()).c_str());
             item->setInt32(kAudioPolicyRqstSession, client->session);
-
+            if (client->opPackageName.size() != 0) {
+                item->setCString(kAudioPolicyRqstPkg,
+                                 std::string(String8(client->opPackageName).string()).c_str());
+            } else {
+                item->setCString(kAudioPolicyRqstPkg, std::to_string(client->uid).c_str());
+            }
             item->setCString(
                     kAudioPolicyRqstDevice, getDeviceTypeStrForPortId(client->deviceId).c_str());
 
@@ -550,9 +553,14 @@
                         // keeps the last of the clients marked active
                         item->setCString(kAudioPolicyActiveSrc,
                                          audioSourceString(other->attributes.source).c_str());
-                        item->setCString(kAudioPolicyActivePkg,
-                                     std::string(String8(other->opPackageName).string()).c_str());
                         item->setInt32(kAudioPolicyActiveSession, other->session);
+                        if (other->opPackageName.size() != 0) {
+                            item->setCString(kAudioPolicyActivePkg,
+                                 std::string(String8(other->opPackageName).string()).c_str());
+                        } else {
+                            item->setCString(kAudioPolicyRqstPkg,
+                                             std::to_string(other->uid).c_str());
+                        }
                         item->setCString(kAudioPolicyActiveDevice,
                                          getDeviceTypeStrForPortId(other->deviceId).c_str());
                     }
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index f3cddc3..ca3b6b6 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -26,7 +26,6 @@
 #include <sys/time.h>
 #include <binder/IServiceManager.h>
 #include <utils/Log.h>
-#include <cutils/multiuser.h>
 #include <cutils/properties.h>
 #include <binder/IPCThreadState.h>
 #include <binder/ActivityManager.h>
@@ -35,16 +34,14 @@
 #include <utils/String16.h>
 #include <utils/threads.h>
 #include "AudioPolicyService.h"
-#include "ServiceUtilities.h"
 #include <hardware_legacy/power.h>
 #include <media/AudioEffect.h>
 #include <media/AudioParameter.h>
+#include <mediautils/ServiceUtilities.h>
 
 #include <system/audio.h>
 #include <system/audio_policy.h>
 
-#include <private/android_filesystem_config.h>
-
 namespace android {
 
 static const char kDeadlockedString[] = "AudioPolicyService may be deadlocked\n";
@@ -70,8 +67,6 @@
     {
         Mutex::Autolock _l(mLock);
 
-        // start tone playback thread
-        mTonePlaybackThread = new AudioCommandThread(String8("ApmTone"), this);
         // start audio commands thread
         mAudioCommandThread = new AudioCommandThread(String8("ApmAudio"), this);
         // start output activity command thread
@@ -93,7 +88,6 @@
 
 AudioPolicyService::~AudioPolicyService()
 {
-    mTonePlaybackThread->exit();
     mAudioCommandThread->exit();
     mOutputCommandThread->exit();
 
@@ -118,13 +112,17 @@
     Mutex::Autolock _l(mNotificationClientsLock);
 
     uid_t uid = IPCThreadState::self()->getCallingUid();
-    if (mNotificationClients.indexOfKey(uid) < 0) {
+    pid_t pid = IPCThreadState::self()->getCallingPid();
+    int64_t token = ((int64_t)uid<<32) | pid;
+
+    if (mNotificationClients.indexOfKey(token) < 0) {
         sp<NotificationClient> notificationClient = new NotificationClient(this,
                                                                            client,
-                                                                           uid);
-        ALOGV("registerClient() client %p, uid %d", client.get(), uid);
+                                                                           uid,
+                                                                           pid);
+        ALOGV("registerClient() client %p, uid %d pid %d", client.get(), uid, pid);
 
-        mNotificationClients.add(uid, notificationClient);
+        mNotificationClients.add(token, notificationClient);
 
         sp<IBinder> binder = IInterface::asBinder(client);
         binder->linkToDeath(notificationClient);
@@ -136,22 +134,33 @@
     Mutex::Autolock _l(mNotificationClientsLock);
 
     uid_t uid = IPCThreadState::self()->getCallingUid();
-    if (mNotificationClients.indexOfKey(uid) < 0) {
+    pid_t pid = IPCThreadState::self()->getCallingPid();
+    int64_t token = ((int64_t)uid<<32) | pid;
+
+    if (mNotificationClients.indexOfKey(token) < 0) {
         return;
     }
-    mNotificationClients.valueFor(uid)->setAudioPortCallbacksEnabled(enabled);
+    mNotificationClients.valueFor(token)->setAudioPortCallbacksEnabled(enabled);
 }
 
 // removeNotificationClient() is called when the client process dies.
-void AudioPolicyService::removeNotificationClient(uid_t uid)
+void AudioPolicyService::removeNotificationClient(uid_t uid, pid_t pid)
 {
     {
         Mutex::Autolock _l(mNotificationClientsLock);
-        mNotificationClients.removeItem(uid);
+        int64_t token = ((int64_t)uid<<32) | pid;
+        mNotificationClients.removeItem(token);
     }
     {
         Mutex::Autolock _l(mLock);
-        if (mAudioPolicyManager) {
+        bool hasSameUid = false;
+        for (size_t i = 0; i < mNotificationClients.size(); i++) {
+            if (mNotificationClients.valueAt(i)->uid() == uid) {
+                hasSameUid = true;
+                break;
+            }
+        }
+        if (mAudioPolicyManager && !hasSameUid) {
             // called from binder death notification: no need to clear caller identity
             mAudioPolicyManager->releaseResourcesForUid(uid);
         }
@@ -239,8 +248,9 @@
 
 AudioPolicyService::NotificationClient::NotificationClient(const sp<AudioPolicyService>& service,
                                                      const sp<IAudioPolicyServiceClient>& client,
-                                                     uid_t uid)
-    : mService(service), mUid(uid), mAudioPolicyServiceClient(client),
+                                                     uid_t uid,
+                                                     pid_t pid)
+    : mService(service), mUid(uid), mPid(pid), mAudioPolicyServiceClient(client),
       mAudioPortCallbacksEnabled(false)
 {
 }
@@ -254,7 +264,7 @@
     sp<NotificationClient> keep(this);
     sp<AudioPolicyService> service = mService.promote();
     if (service != 0) {
-        service->removeNotificationClient(mUid);
+        service->removeNotificationClient(mUid, mPid);
     }
 }
 
@@ -275,7 +285,7 @@
 void AudioPolicyService::NotificationClient::onDynamicPolicyMixStateUpdate(
         const String8& regId, int32_t state)
 {
-    if (mAudioPolicyServiceClient != 0 && multiuser_get_app_id(mUid) < AID_APP_START) {
+    if (mAudioPolicyServiceClient != 0 && isServiceUid(mUid)) {
         mAudioPolicyServiceClient->onDynamicPolicyMixStateUpdate(regId, state);
     }
 }
@@ -285,7 +295,7 @@
         const audio_config_base_t *clientConfig, const audio_config_base_t *deviceConfig,
         audio_patch_handle_t patchHandle)
 {
-    if (mAudioPolicyServiceClient != 0 && multiuser_get_app_id(mUid) < AID_APP_START) {
+    if (mAudioPolicyServiceClient != 0 && isServiceUid(mUid)) {
         mAudioPolicyServiceClient->onRecordingConfigurationUpdate(event, clientInfo,
                 clientConfig, deviceConfig, patchHandle);
     }
@@ -325,8 +335,6 @@
     result.append(buffer);
     snprintf(buffer, SIZE, "Command Thread: %p\n", mAudioCommandThread.get());
     result.append(buffer);
-    snprintf(buffer, SIZE, "Tones Thread: %p\n", mTonePlaybackThread.get());
-    result.append(buffer);
 
     write(fd, result.string(), result.size());
     return NO_ERROR;
@@ -362,9 +370,6 @@
         if (mAudioCommandThread != 0) {
             mAudioCommandThread->dump(fd);
         }
-        if (mTonePlaybackThread != 0) {
-            mTonePlaybackThread->dump(fd);
-        }
 
         if (mAudioPolicyManager) {
             mAudioPolicyManager->dump(fd);
@@ -577,10 +582,6 @@
     updateUidCache(uid, false, true);
 }
 
-bool AudioPolicyService::UidPolicy::isServiceUid(uid_t uid) const {
-    return multiuser_get_app_id(uid) < AID_APP_START;
-}
-
 void AudioPolicyService::UidPolicy::notifyService(uid_t uid, bool active) {
     sp<AudioPolicyService> service = mService.promote();
     if (service != nullptr) {
@@ -639,7 +640,6 @@
                                                            const wp<AudioPolicyService>& service)
     : Thread(false), mName(name), mService(service)
 {
-    mpToneGenerator = NULL;
 }
 
 
@@ -649,7 +649,6 @@
         release_wake_lock(mName.string());
     }
     mAudioCommands.clear();
-    delete mpToneGenerator;
 }
 
 void AudioPolicyService::AudioCommandThread::onFirstRef()
@@ -674,26 +673,6 @@
                 mLastCommand = command;
 
                 switch (command->mCommand) {
-                case START_TONE: {
-                    mLock.unlock();
-                    ToneData *data = (ToneData *)command->mParam.get();
-                    ALOGV("AudioCommandThread() processing start tone %d on stream %d",
-                            data->mType, data->mStream);
-                    delete mpToneGenerator;
-                    mpToneGenerator = new ToneGenerator(data->mStream, 1.0);
-                    mpToneGenerator->startTone(data->mType);
-                    mLock.lock();
-                    }break;
-                case STOP_TONE: {
-                    mLock.unlock();
-                    ALOGV("AudioCommandThread() processing stop tone");
-                    if (mpToneGenerator != NULL) {
-                        mpToneGenerator->stopTone();
-                        delete mpToneGenerator;
-                        mpToneGenerator = NULL;
-                    }
-                    mLock.lock();
-                    }break;
                 case SET_VOLUME: {
                     VolumeData *data = (VolumeData *)command->mParam.get();
                     ALOGV("AudioCommandThread() processing set volume stream %d, \
@@ -900,27 +879,6 @@
     return NO_ERROR;
 }
 
-void AudioPolicyService::AudioCommandThread::startToneCommand(ToneGenerator::tone_type type,
-        audio_stream_type_t stream)
-{
-    sp<AudioCommand> command = new AudioCommand();
-    command->mCommand = START_TONE;
-    sp<ToneData> data = new ToneData();
-    data->mType = type;
-    data->mStream = stream;
-    command->mParam = data;
-    ALOGV("AudioCommandThread() adding tone start type %d, stream %d", type, stream);
-    sendCommand(command);
-}
-
-void AudioPolicyService::AudioCommandThread::stopToneCommand()
-{
-    sp<AudioCommand> command = new AudioCommand();
-    command->mCommand = STOP_TONE;
-    ALOGV("AudioCommandThread() adding tone stop");
-    sendCommand(command);
-}
-
 status_t AudioPolicyService::AudioCommandThread::volumeCommand(audio_stream_type_t stream,
                                                                float volume,
                                                                audio_io_handle_t output,
@@ -1212,6 +1170,7 @@
                 patch = ((CreateAudioPatchData *)command->mParam.get())->mPatch;
             } else {
                 handle = ((ReleaseAudioPatchData *)command->mParam.get())->mHandle;
+                memset(&patch, 0, sizeof(patch));
             }
             audio_patch_handle_t handle2;
             struct audio_patch patch2;
@@ -1256,8 +1215,6 @@
 
         } break;
 
-        case START_TONE:
-        case STOP_TONE:
         default:
             break;
         }
@@ -1330,27 +1287,6 @@
                                                    output, delayMs);
 }
 
-int AudioPolicyService::startTone(audio_policy_tone_t tone,
-                                  audio_stream_type_t stream)
-{
-    if (tone != AUDIO_POLICY_TONE_IN_CALL_NOTIFICATION) {
-        ALOGE("startTone: illegal tone requested (%d)", tone);
-    }
-    if (stream != AUDIO_STREAM_VOICE_CALL) {
-        ALOGE("startTone: illegal stream (%d) requested for tone %d", stream,
-            tone);
-    }
-    mTonePlaybackThread->startToneCommand(ToneGenerator::TONE_SUP_CALL_WAITING,
-                                          AUDIO_STREAM_VOICE_CALL);
-    return 0;
-}
-
-int AudioPolicyService::stopTone()
-{
-    mTonePlaybackThread->stopToneCommand();
-    return 0;
-}
-
 int AudioPolicyService::setVoiceVolume(float volume, int delayMs)
 {
     return (int)mAudioCommandThread->voiceVolumeCommand(volume, delayMs);
@@ -1406,9 +1342,6 @@
 int aps_set_stream_volume(void *service, audio_stream_type_t stream,
                                      float volume, audio_io_handle_t output,
                                      int delay_ms);
-int aps_start_tone(void *service, audio_policy_tone_t tone,
-                              audio_stream_type_t stream);
-int aps_stop_tone(void *service);
 int aps_set_voice_volume(void *service, float volume, int delay_ms);
 };
 
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 407d7a5..a1366bb 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -38,8 +38,6 @@
 
 namespace android {
 
-using namespace std;
-
 // ----------------------------------------------------------------------------
 
 class AudioPolicyService :
@@ -159,8 +157,6 @@
                                      float volume,
                                      audio_io_handle_t output,
                                      int delayMs = 0);
-    virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream);
-    virtual status_t stopTone();
     virtual status_t setVoiceVolume(float volume, int delayMs = 0);
     virtual bool isOffloadSupported(const audio_offload_info_t &config);
 
@@ -224,7 +220,7 @@
             virtual status_t clientSetAudioPortConfig(const struct audio_port_config *config,
                                                       int delayMs);
 
-            void removeNotificationClient(uid_t uid);
+            void removeNotificationClient(uid_t uid, pid_t pid);
             void onAudioPortListUpdate();
             void doOnAudioPortListUpdate();
             void onAudioPatchListUpdate();
@@ -293,7 +289,6 @@
         void removeOverrideUid(uid_t uid) { updateOverrideUid(uid, false, false); }
 
     private:
-        bool isServiceUid(uid_t uid) const;
         void notifyService(uid_t uid, bool active);
         void updateOverrideUid(uid_t uid, bool active, bool insert);
         void updateUidCache(uid_t uid, bool active, bool insert);
@@ -307,10 +302,7 @@
         std::unordered_map<uid_t, bool> mCachedUids;
     };
 
-    // Thread used for tone playback and to send audio config commands to audio flinger
-    // For tone playback, using a separate thread is necessary to avoid deadlock with mLock because
-    // startTone() and stopTone() are normally called with mLock locked and requesting a tone start
-    // or stop will cause calls to AudioPolicyService and an attempt to lock mLock.
+    // Thread used to send audio config commands to audio flinger
     // For audio config commands, it is necessary because audio flinger requires that the calling
     // process (user) has permission to modify audio settings.
     class AudioCommandThread : public Thread {
@@ -319,8 +311,6 @@
 
         // commands for tone AudioCommand
         enum {
-            START_TONE,
-            STOP_TONE,
             SET_VOLUME,
             SET_PARAMETERS,
             SET_VOICE_VOLUME,
@@ -345,9 +335,6 @@
         virtual     bool        threadLoop();
 
                     void        exit();
-                    void        startToneCommand(ToneGenerator::tone_type type,
-                                                 audio_stream_type_t stream);
-                    void        stopToneCommand();
                     status_t    volumeCommand(audio_stream_type_t stream, float volume,
                                             audio_io_handle_t output, int delayMs = 0);
                     status_t    parametersCommand(audio_io_handle_t ioHandle,
@@ -390,7 +377,7 @@
 
             void dump(char* buffer, size_t size);
 
-            int mCommand;   // START_TONE, STOP_TONE ...
+            int mCommand;   // SET_VOLUME, SET_PARAMETERS...
             nsecs_t mTime;  // time stamp
             Mutex mLock;    // mutex associated to mCond
             Condition mCond; // condition for status return
@@ -406,12 +393,6 @@
             AudioCommandData() {}
         };
 
-        class ToneData : public AudioCommandData {
-        public:
-            ToneGenerator::tone_type mType; // tone type (START_TONE only)
-            audio_stream_type_t mStream;    // stream type (START_TONE only)
-        };
-
         class VolumeData : public AudioCommandData {
         public:
             audio_stream_type_t mStream;
@@ -478,7 +459,6 @@
         Mutex   mLock;
         Condition mWaitWorkCV;
         Vector < sp<AudioCommand> > mAudioCommands; // list of pending commands
-        ToneGenerator *mpToneGenerator;     // the tone generator
         sp<AudioCommand> mLastCommand;      // last processed command (used by dump)
         String8 mName;                      // string used by wake lock fo delayed commands
         wp<AudioPolicyService> mService;
@@ -553,11 +533,6 @@
         // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
         virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys);
 
-        // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
-        // over a telephony device during a phone call.
-        virtual status_t startTone(audio_policy_tone_t tone, audio_stream_type_t stream);
-        virtual status_t stopTone();
-
         // set down link audio volume.
         virtual status_t setVoiceVolume(float volume, int delayMs = 0);
 
@@ -597,7 +572,7 @@
     public:
                             NotificationClient(const sp<AudioPolicyService>& service,
                                                 const sp<IAudioPolicyServiceClient>& client,
-                                                uid_t uid);
+                                                uid_t uid, pid_t pid);
         virtual             ~NotificationClient();
 
                             void      onAudioPortListUpdate();
@@ -610,6 +585,10 @@
                                         audio_patch_handle_t patchHandle);
                             void      setAudioPortCallbacksEnabled(bool enabled);
 
+                            uid_t uid() {
+                                return mUid;
+                            }
+
                 // IBinder::DeathRecipient
                 virtual     void        binderDied(const wp<IBinder>& who);
 
@@ -619,6 +598,7 @@
 
         const wp<AudioPolicyService>        mService;
         const uid_t                         mUid;
+        const pid_t                         mPid;
         const sp<IAudioPolicyServiceClient> mAudioPolicyServiceClient;
               bool                          mAudioPortCallbacksEnabled;
     };
@@ -676,14 +656,13 @@
     // mLock protects AudioPolicyManager methods that can call into audio flinger
     // and possibly back in to audio policy service and acquire mEffectsLock.
     sp<AudioCommandThread> mAudioCommandThread;     // audio commands thread
-    sp<AudioCommandThread> mTonePlaybackThread;     // tone playback thread
     sp<AudioCommandThread> mOutputCommandThread;    // process stop and release output
     struct audio_policy_device *mpAudioPolicyDev;
     struct audio_policy *mpAudioPolicy;
     AudioPolicyInterface *mAudioPolicyManager;
     AudioPolicyClient *mAudioPolicyClient;
 
-    DefaultKeyedVector< uid_t, sp<NotificationClient> >    mNotificationClients;
+    DefaultKeyedVector< int64_t, sp<NotificationClient> >    mNotificationClients;
     Mutex mNotificationClientsLock;  // protects mNotificationClients
     // Manage all effects configured in audio_effects.conf
     sp<AudioPolicyEffects> mAudioPolicyEffects;
diff --git a/services/audiopolicy/tests/Android.mk b/services/audiopolicy/tests/Android.mk
index a43daea..cfa9ab1 100644
--- a/services/audiopolicy/tests/Android.mk
+++ b/services/audiopolicy/tests/Android.mk
@@ -6,7 +6,6 @@
   frameworks/av/services/audiopolicy \
   frameworks/av/services/audiopolicy/common/include \
   frameworks/av/services/audiopolicy/engine/interface \
-  frameworks/av/services/audiopolicy/utilities
 
 LOCAL_SHARED_LIBRARIES := \
   libaudiopolicymanagerdefault \
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index eb8222c..2ff7675 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -60,9 +60,6 @@
                        int /*delayMs*/) override { }
     String8 getParameters(audio_io_handle_t /*ioHandle*/,
                           const String8& /*keys*/) override { return String8(); }
-    status_t startTone(audio_policy_tone_t /*tone*/,
-                       audio_stream_type_t /*stream*/) override { return NO_INIT; }
-    status_t stopTone() override { return NO_INIT; }
     status_t setVoiceVolume(float /*volume*/, int /*delayMs*/) override { return NO_INIT; }
     status_t moveEffects(audio_session_t /*session*/,
                          audio_io_handle_t /*srcOutput*/,
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index a9593b8..56af152 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -16,9 +16,15 @@
 
 #include <memory>
 #include <set>
+#include <sys/wait.h>
+#include <unistd.h>
 
 #include <gtest/gtest.h>
 
+#define LOG_TAG "APM_Test"
+#include <log/log.h>
+#include <media/PatchBuilder.h>
+
 #include "AudioPolicyTestClient.h"
 #include "AudioPolicyTestManager.h"
 
@@ -132,6 +138,36 @@
     // SetUp must finish with no assertions.
 }
 
+TEST_F(AudioPolicyManagerTest, Dump) {
+    int pipefd[2];
+    ASSERT_NE(-1, pipe(pipefd));
+    pid_t cpid = fork();
+    ASSERT_NE(-1, cpid);
+    if (cpid == 0) {
+        // Child process reads from the pipe and logs.
+        close(pipefd[1]);
+        std::string line;
+        char buf;
+        while (read(pipefd[0], &buf, sizeof(buf)) > 0) {
+            if (buf != '\n') {
+                line += buf;
+            } else {
+                ALOGI("%s", line.c_str());
+                line = "";
+            }
+        }
+        if (!line.empty()) ALOGI("%s", line.c_str());
+        close(pipefd[0]);
+        _exit(EXIT_SUCCESS);
+    } else {
+        // Parent does the dump and checks the status code.
+        close(pipefd[0]);
+        ASSERT_EQ(NO_ERROR, mManager->dump(pipefd[1]));
+        close(pipefd[1]);
+        wait(NULL);  // Wait for the child to exit.
+    }
+}
+
 TEST_F(AudioPolicyManagerTest, CreateAudioPatchFailure) {
     audio_patch patch{};
     audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
@@ -166,29 +202,14 @@
 }
 
 TEST_F(AudioPolicyManagerTest, CreateAudioPatchFromMix) {
-    audio_patch patch{};
     audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
     uid_t uid = 42;
     const size_t patchCountBefore = mClient->getActivePatchesCount();
-    patch.num_sources = 1;
-    {
-        auto& src = patch.sources[0];
-        src.role = AUDIO_PORT_ROLE_SOURCE;
-        src.type = AUDIO_PORT_TYPE_MIX;
-        src.id = mManager->getConfig().getAvailableInputDevices()[0]->getId();
-        // Note: these are the parameters of the output device.
-        src.sample_rate = 44100;
-        src.format = AUDIO_FORMAT_PCM_16_BIT;
-        src.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
-    }
-    patch.num_sinks = 1;
-    {
-        auto& sink = patch.sinks[0];
-        sink.role = AUDIO_PORT_ROLE_SINK;
-        sink.type = AUDIO_PORT_TYPE_DEVICE;
-        sink.id = mManager->getConfig().getDefaultOutputDevice()->getId();
-    }
-    ASSERT_EQ(NO_ERROR, mManager->createAudioPatch(&patch, &handle, uid));
+    ASSERT_FALSE(mManager->getConfig().getAvailableInputDevices().isEmpty());
+    PatchBuilder patchBuilder;
+    patchBuilder.addSource(mManager->getConfig().getAvailableInputDevices()[0]).
+            addSink(mManager->getConfig().getDefaultOutputDevice());
+    ASSERT_EQ(NO_ERROR, mManager->createAudioPatch(patchBuilder.patch(), &handle, uid));
     ASSERT_NE(AUDIO_PATCH_HANDLE_NONE, handle);
     ASSERT_EQ(patchCountBefore + 1, mClient->getActivePatchesCount());
 }
diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
index 0c738e7..683e84d 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
@@ -33,7 +33,10 @@
     FrameProcessorBase(device),
     mClient(client),
     mLastFrameNumberOfFaces(0),
-    mLast3AFrameNumber(-1) {
+    mLast3AFrameNumber(-1),
+    mLastAEFrameNumber(-1),
+    mLastAFrameNumber(-1),
+    mLastAWBFrameNumber(-1) {
 
     sp<CameraDeviceBase> d = device.promote();
     mSynthesize3ANotify = !(d->willNotify3A());
@@ -262,24 +265,73 @@
     bool gotAllStates = true;
 
     // TODO: Also use AE mode, AE trigger ID
-    gotAllStates &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AF_MODE,
+    bool gotAFState = updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AF_MODE,
             &pendingState.afMode, frameNumber, cameraId);
 
-    gotAllStates &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AWB_MODE,
+    bool gotAWBState = updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AWB_MODE,
             &pendingState.awbMode, frameNumber, cameraId);
 
-    gotAllStates &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AE_STATE,
+    bool gotAEState = updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AE_STATE,
             &pendingState.aeState, frameNumber, cameraId);
 
-    gotAllStates &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AF_STATE,
+    gotAFState &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AF_STATE,
             &pendingState.afState, frameNumber, cameraId);
 
-    gotAllStates &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AWB_STATE,
+    gotAWBState &= updatePendingState<uint8_t>(metadata, ANDROID_CONTROL_AWB_STATE,
             &pendingState.awbState, frameNumber, cameraId);
 
     pendingState.afTriggerId = frame.mResultExtras.afTriggerId;
     pendingState.aeTriggerId = frame.mResultExtras.precaptureTriggerId;
 
+    if (gotAEState && (frameNumber > mLastAEFrameNumber)) {
+        if (pendingState.aeState != m3aState.aeState ||
+                pendingState.aeTriggerId > m3aState.aeTriggerId) {
+            ALOGV("%s: Camera %d: AE state %d->%d",
+                    __FUNCTION__, cameraId,
+                    m3aState.aeState, pendingState.aeState);
+            client->notifyAutoExposure(pendingState.aeState, pendingState.aeTriggerId);
+
+            m3aState.aeState = pendingState.aeState;
+            m3aState.aeTriggerId = pendingState.aeTriggerId;
+            mLastAEFrameNumber = frameNumber;
+        }
+    }
+
+    if (gotAFState && (frameNumber > mLastAFrameNumber)) {
+        if (pendingState.afState != m3aState.afState ||
+                pendingState.afMode != m3aState.afMode ||
+                pendingState.afTriggerId != m3aState.afTriggerId) {
+            ALOGV("%s: Camera %d: AF state %d->%d. AF mode %d->%d. Trigger %d->%d",
+                    __FUNCTION__, cameraId,
+                    m3aState.afState, pendingState.afState,
+                    m3aState.afMode, pendingState.afMode,
+                    m3aState.afTriggerId, pendingState.afTriggerId);
+            client->notifyAutoFocus(pendingState.afState, pendingState.afTriggerId);
+
+            m3aState.afState = pendingState.afState;
+            m3aState.afMode = pendingState.afMode;
+            m3aState.afTriggerId = pendingState.afTriggerId;
+            mLastAFrameNumber = frameNumber;
+        }
+    }
+
+    if (gotAWBState && (frameNumber > mLastAWBFrameNumber)) {
+        if (pendingState.awbState != m3aState.awbState ||
+                pendingState.awbMode != m3aState.awbMode) {
+            ALOGV("%s: Camera %d: AWB state %d->%d. AWB mode %d->%d",
+                    __FUNCTION__, cameraId,
+                    m3aState.awbState, pendingState.awbState,
+                    m3aState.awbMode, pendingState.awbMode);
+            client->notifyAutoWhitebalance(pendingState.awbState,
+                    pendingState.aeTriggerId);
+
+            m3aState.awbMode = pendingState.awbMode;
+            m3aState.awbState = pendingState.awbState;
+            mLastAWBFrameNumber = frameNumber;
+        }
+    }
+
+    gotAllStates &= gotAEState & gotAFState & gotAWBState;
     if (!gotAllStates) {
         // If not all states are received, put the pending state to mPending3AStates.
         if (index == NAME_NOT_FOUND) {
@@ -290,40 +342,10 @@
         return NOT_ENOUGH_DATA;
     }
 
-    // Once all 3A states are received, notify the client about 3A changes.
-    if (pendingState.aeState != m3aState.aeState ||
-            pendingState.aeTriggerId > m3aState.aeTriggerId) {
-        ALOGV("%s: Camera %d: AE state %d->%d",
-                __FUNCTION__, cameraId,
-                m3aState.aeState, pendingState.aeState);
-        client->notifyAutoExposure(pendingState.aeState, pendingState.aeTriggerId);
-    }
-
-    if (pendingState.afState != m3aState.afState ||
-        pendingState.afMode != m3aState.afMode ||
-        pendingState.afTriggerId != m3aState.afTriggerId) {
-        ALOGV("%s: Camera %d: AF state %d->%d. AF mode %d->%d. Trigger %d->%d",
-                __FUNCTION__, cameraId,
-                m3aState.afState, pendingState.afState,
-                m3aState.afMode, pendingState.afMode,
-                m3aState.afTriggerId, pendingState.afTriggerId);
-        client->notifyAutoFocus(pendingState.afState, pendingState.afTriggerId);
-    }
-    if (pendingState.awbState != m3aState.awbState ||
-        pendingState.awbMode != m3aState.awbMode) {
-        ALOGV("%s: Camera %d: AWB state %d->%d. AWB mode %d->%d",
-                __FUNCTION__, cameraId,
-                m3aState.awbState, pendingState.awbState,
-                m3aState.awbMode, pendingState.awbMode);
-        client->notifyAutoWhitebalance(pendingState.awbState,
-                pendingState.aeTriggerId);
-    }
-
     if (index != NAME_NOT_FOUND) {
         mPending3AStates.removeItemsAt(index);
     }
 
-    m3aState = pendingState;
     mLast3AFrameNumber = frameNumber;
 
     return OK;
diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.h b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
index 62a4e91..8183c12 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.h
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
@@ -104,8 +104,7 @@
     // Track most recent frame number for which 3A notifications were sent for.
     // Used to filter against sending 3A notifications for the same frame
     // several times.
-    int32_t mLast3AFrameNumber;
-
+    int32_t mLast3AFrameNumber, mLastAEFrameNumber, mLastAFrameNumber, mLastAWBFrameNumber;
     // Emit FaceDetection event to java if faces changed
     void callbackFaceDetection(const sp<Camera2Client>& client,
                                const camera_frame_metadata &metadata);
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.cpp b/services/camera/libcameraservice/common/CameraProviderManager.cpp
index 43f1a91..0ce4318 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.cpp
+++ b/services/camera/libcameraservice/common/CameraProviderManager.cpp
@@ -631,7 +631,12 @@
 
     mUniqueCameraIds.insert(id);
     if (isAPI1Compatible) {
-        mUniqueAPI1CompatibleCameraIds.push_back(id);
+        // addDevice can be called more than once for the same camera id if HAL
+        // supports openLegacy.
+        if (std::find(mUniqueAPI1CompatibleCameraIds.begin(), mUniqueAPI1CompatibleCameraIds.end(),
+                id) == mUniqueAPI1CompatibleCameraIds.end()) {
+            mUniqueAPI1CompatibleCameraIds.push_back(id);
+        }
     }
 
     if (parsedId != nullptr) {
diff --git a/services/mediaextractor/Android.mk b/services/mediaextractor/Android.mk
index 37d6cc9..73c9535 100644
--- a/services/mediaextractor/Android.mk
+++ b/services/mediaextractor/Android.mk
@@ -18,6 +18,7 @@
 LOCAL_REQUIRED_MODULES_arm := crash_dump.policy mediaextractor.policy
 LOCAL_REQUIRED_MODULES_arm64 := crash_dump.policy mediaextractor.policy
 LOCAL_REQUIRED_MODULES_x86 := crash_dump.policy mediaextractor.policy
+LOCAL_REQUIRED_MODULES_x86_64 := crash_dump.policy mediaextractor.policy
 
 # extractor libraries
 LOCAL_REQUIRED_MODULES += \
diff --git a/services/mediaextractor/seccomp_policy/mediaextractor-x86_64.policy b/services/mediaextractor/seccomp_policy/mediaextractor-x86_64.policy
old mode 100755
new mode 100644
index 63c7780..6d9ed6f
--- a/services/mediaextractor/seccomp_policy/mediaextractor-x86_64.policy
+++ b/services/mediaextractor/seccomp_policy/mediaextractor-x86_64.policy
@@ -21,6 +21,7 @@
 getuid: 1
 setpriority: 1
 sigaltstack: 1
+fstatfs: 1
 newfstatat: 1
 restart_syscall: 1
 exit: 1
@@ -30,28 +31,21 @@
 sched_setscheduler: 1
 getrlimit: 1
 nanosleep: 1
+getrandom: 1
+
+# for dynamically loading extractors
+getdents64: 1
+readlinkat: 1
+pread64: 1
+mremap: 1
 
 # for FileSource
 readlinkat: 1
 
-# for attaching to debuggerd on process crash
-tgkill: 1
-socket: arg0 == 1
-connect: 1
-fcntl: 1
-rt_sigprocmask: 1
-rt_sigaction: 1
-rt_tgsigqueueinfo: 1
-geteuid: 1
-getgid: 1
-getegid: 1
-getgroups: 1
-getdents64: 1
-pipe2: 1
-ppoll: 1
-
 # Required by AddressSanitizer
 gettid: 1
 sched_yield: 1
 getpid: 1
 gettid: 1
+
+@include /system/etc/seccomp_policy/crash_dump.x86_64.policy
diff --git a/services/medialog/Android.bp b/services/medialog/Android.bp
index 29e6dfc..ca96f62 100644
--- a/services/medialog/Android.bp
+++ b/services/medialog/Android.bp
@@ -9,7 +9,9 @@
     shared_libs: [
         "libaudioutils",
         "libbinder",
+        "libcutils",
         "liblog",
+        "libmediautils",
         "libnbaio",
         "libnblog",
         "libutils",
diff --git a/services/medialog/MediaLogService.cpp b/services/medialog/MediaLogService.cpp
index 1be5544..e58dff7 100644
--- a/services/medialog/MediaLogService.cpp
+++ b/services/medialog/MediaLogService.cpp
@@ -21,7 +21,7 @@
 #include <utils/Log.h>
 #include <binder/PermissionCache.h>
 #include <media/nblog/NBLog.h>
-#include <private/android_filesystem_config.h>
+#include <mediautils/ServiceUtilities.h>
 #include "MediaLogService.h"
 
 namespace android {
@@ -53,7 +53,7 @@
 
 void MediaLogService::registerWriter(const sp<IMemory>& shared, size_t size, const char *name)
 {
-    if (IPCThreadState::self()->getCallingUid() != AID_AUDIOSERVER || shared == 0 ||
+    if (!isAudioServerOrMediaServerUid(IPCThreadState::self()->getCallingUid()) || shared == 0 ||
             size < kMinSize || size > kMaxSize || name == NULL ||
             shared->size() < NBLog::Timeline::sharedSize(size)) {
         return;
@@ -67,7 +67,7 @@
 
 void MediaLogService::unregisterWriter(const sp<IMemory>& shared)
 {
-    if (IPCThreadState::self()->getCallingUid() != AID_AUDIOSERVER || shared == 0) {
+    if (!isAudioServerOrMediaServerUid(IPCThreadState::self()->getCallingUid()) || shared == 0) {
         return;
     }
     Mutex::Autolock _l(mLock);
@@ -95,10 +95,8 @@
 
 status_t MediaLogService::dump(int fd, const Vector<String16>& args __unused)
 {
-    // FIXME merge with similar but not identical code at services/audioflinger/ServiceUtilities.cpp
-    static const String16 sDump("android.permission.DUMP");
-    if (!(IPCThreadState::self()->getCallingUid() == AID_AUDIOSERVER ||
-            PermissionCache::checkCallingPermission(sDump))) {
+    if (!(isAudioServerOrMediaServerUid(IPCThreadState::self()->getCallingUid())
+            || dumpAllowed())) {
         dprintf(fd, "Permission Denial: can't dump media.log from pid=%d, uid=%d\n",
                 IPCThreadState::self()->getCallingPid(),
                 IPCThreadState::self()->getCallingUid());
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index 6a72e5b..94440b1 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -24,6 +24,7 @@
 
 #include <aaudio/AAudio.h>
 #include <mediautils/SchedulingPolicyService.h>
+#include <mediautils/ServiceUtilities.h>
 #include <utils/String16.h>
 
 #include "binding/AAudioServiceMessage.h"
@@ -33,7 +34,6 @@
 #include "AAudioServiceStreamMMAP.h"
 #include "AAudioServiceStreamShared.h"
 #include "binding/IAAudioService.h"
-#include "ServiceUtilities.h"
 
 using namespace android;
 using namespace aaudio;
diff --git a/services/oboeservice/Android.mk b/services/oboeservice/Android.mk
index 584b2ef..3d5f140 100644
--- a/services/oboeservice/Android.mk
+++ b/services/oboeservice/Android.mk
@@ -53,7 +53,6 @@
     libbinder \
     libcutils \
     libmediautils \
-    libserviceutility \
     libutils \
     liblog
 
diff --git a/services/soundtrigger/Android.mk b/services/soundtrigger/Android.mk
index ad3666e..3c7d29d 100644
--- a/services/soundtrigger/Android.mk
+++ b/services/soundtrigger/Android.mk
@@ -34,8 +34,7 @@
     libhardware \
     libsoundtrigger \
     libaudioclient \
-    libserviceutility
-
+    libmediautils \
 
 ifeq ($(USE_LEGACY_LOCAL_AUDIO_HAL),true)
 # libhardware configuration
diff --git a/services/soundtrigger/SoundTriggerHwService.cpp b/services/soundtrigger/SoundTriggerHwService.cpp
index a7d6e83..eb9cd1d 100644
--- a/services/soundtrigger/SoundTriggerHwService.cpp
+++ b/services/soundtrigger/SoundTriggerHwService.cpp
@@ -27,13 +27,13 @@
 #include <cutils/properties.h>
 #include <hardware/hardware.h>
 #include <media/AudioSystem.h>
+#include <mediautils/ServiceUtilities.h>
 #include <utils/Errors.h>
 #include <utils/Log.h>
 #include <binder/IServiceManager.h>
 #include <binder/MemoryBase.h>
 #include <binder/MemoryHeapBase.h>
 #include <system/sound_trigger.h>
-#include <ServiceUtilities.h>
 #include "SoundTriggerHwService.h"
 
 #ifdef SOUND_TRIGGER_USE_STUB_MODULE
@@ -562,10 +562,7 @@
     if (mHalInterface == 0) {
         return NO_INIT;
     }
-    if (modelMemory == 0 || modelMemory->pointer() == NULL) {
-        ALOGE("loadSoundModel() modelMemory is 0 or has NULL pointer()");
-        return BAD_VALUE;
-    }
+
     struct sound_trigger_sound_model *sound_model =
             (struct sound_trigger_sound_model *)modelMemory->pointer();
 
@@ -659,11 +656,6 @@
     if (mHalInterface == 0) {
         return NO_INIT;
     }
-    if (dataMemory == 0 || dataMemory->pointer() == NULL) {
-        ALOGE("startRecognition() dataMemory is 0 or has NULL pointer()");
-        return BAD_VALUE;
-
-    }
 
     struct sound_trigger_recognition_config *config =
             (struct sound_trigger_recognition_config *)dataMemory->pointer();
@@ -966,6 +958,9 @@
                                IPCThreadState::self()->getCallingUid())) {
         return PERMISSION_DENIED;
     }
+    if (checkIMemory(modelMemory) != NO_ERROR) {
+        return BAD_VALUE;
+    }
 
     sp<Module> module = mModule.promote();
     if (module == 0) {
@@ -997,6 +992,9 @@
                                IPCThreadState::self()->getCallingUid())) {
         return PERMISSION_DENIED;
     }
+    if (checkIMemory(dataMemory) != NO_ERROR) {
+        return BAD_VALUE;
+    }
 
     sp<Module> module = mModule.promote();
     if (module == 0) {