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..bddf945 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,
@@ -348,7 +354,10 @@
                     decodeTimesUs.push(delayDecodeUs);
                 }
 
-                if (showProgress && (n++ % 16) == 0) {
+                if (gVerbose) {
+                    MetaDataBase &meta = buffer->meta_data();
+                    fprintf(stdout, "%ld sample format: %s\n", numFrames, meta.toString().c_str());
+                } else if (showProgress && (n++ % 16) == 0) {
                     printf(".");
                     fflush(stdout);
                 }
@@ -579,12 +588,12 @@
                 break;
             }
 
+            CHECK(buffer != NULL);
+
             if (buffer->range_length() > 0) {
                 break;
             }
 
-            CHECK(buffer != NULL);
-
             buffer->release();
             buffer = NULL;
         }
@@ -630,6 +639,7 @@
     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) {
@@ -640,7 +650,8 @@
         MEDIA_MIMETYPE_AUDIO_MPEG, MEDIA_MIMETYPE_AUDIO_G711_MLAW,
         MEDIA_MIMETYPE_AUDIO_G711_ALAW, MEDIA_MIMETYPE_AUDIO_VORBIS,
         MEDIA_MIMETYPE_VIDEO_VP8, MEDIA_MIMETYPE_VIDEO_VP9,
-        MEDIA_MIMETYPE_VIDEO_DOLBY_VISION
+        MEDIA_MIMETYPE_VIDEO_DOLBY_VISION,
+        MEDIA_MIMETYPE_AUDIO_EAC3, MEDIA_MIMETYPE_AUDIO_AC4
     };
 
     const char *codecType = queryDecoders? "decoder" : "encoder";
@@ -708,7 +719,7 @@
     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 +843,12 @@
                 break;
             }
 
+            case 'v':
+            {
+                gVerbose = true;
+                break;
+            }
+
             case '?':
             case 'h':
             default:
diff --git a/include/media/EventLog.h b/include/media/EventLog.h
new file mode 120000
index 0000000..9b2c4bf
--- /dev/null
+++ b/include/media/EventLog.h
@@ -0,0 +1 @@
+../../media/utils/include/mediautils/EventLog.h
\ No newline at end of file
diff --git a/include/media/TimeCheck.h b/include/media/TimeCheck.h
index e3ef134..85e17f9 120000
--- a/include/media/TimeCheck.h
+++ b/include/media/TimeCheck.h
@@ -1 +1 @@
-../../media/libmedia/include/media/TimeCheck.h
\ No newline at end of file
+../../media/utils/include/mediautils/TimeCheck.h
\ No newline at end of file
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/Android.mk b/media/audioserver/Android.mk
index 70c281a..8a97299 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -21,6 +21,9 @@
 	libsoundtriggerservice \
 	libutils
 
+LOCAL_STATIC_LIBRARIES := \
+	libjsoncpp
+
 # TODO oboeservice is the old folder name for aaudioservice. It will be changed.
 LOCAL_C_INCLUDES := \
 	frameworks/av/services/audioflinger \
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/AC4Parser.cpp b/media/extractors/mp4/AC4Parser.cpp
new file mode 100644
index 0000000..167d474
--- /dev/null
+++ b/media/extractors/mp4/AC4Parser.cpp
@@ -0,0 +1,624 @@
+/*
+ * 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_NDEBUG 0
+#define LOG_TAG "AC4Parser"
+
+#include <inttypes.h>
+#include <utils/Log.h>
+#include <utils/misc.h>
+
+#include "AC4Parser.h"
+
+#define BOOLSTR(a)  ((a)?"true":"false")
+#define BYTE_ALIGN mBitReader.skipBits(mBitReader.numBitsLeft() % 8)
+#define CHECK_BITS_LEFT(n) if (mBitReader.numBitsLeft() < n) {return false;}
+
+namespace android {
+
+AC4Parser::AC4Parser() {
+}
+
+AC4DSIParser::AC4DSIParser(ABitReader &br)
+    : mBitReader(br){
+
+    mDSISize = mBitReader.numBitsLeft();
+}
+
+// ETSI TS 103 190-2 V1.1.1 (2015-09) Table 79: channel_mode
+static const char *ChannelModes[] = {
+    "mono",
+    "stereo",
+    "3.0",
+    "5.0",
+    "5.1",
+    "7.0 (3/4/0)",
+    "7.1 (3/4/0.1)",
+    "7.0 (5/2/0)",
+    "7.1 (5/2/0.1)",
+    "7.0 (3/2/2)",
+    "7.1 (3/2/2.1)",
+    "7.0.4",
+    "7.1.4",
+    "9.0.4",
+    "9.1.4",
+    "22.2"
+};
+
+static const char* ContentClassifier[] = {
+    "Complete Main",
+    "Music and Effects",
+    "Visually Impaired",
+    "Hearing Impaired",
+    "Dialog",
+    "Commentary",
+    "Emergency",
+    "Voice Over"
+};
+
+bool AC4DSIParser::parseLanguageTag(uint32_t presentationID, uint32_t substreamID){
+    CHECK_BITS_LEFT(6);
+    uint32_t n_language_tag_bytes = mBitReader.getBits(6);
+    if (n_language_tag_bytes < 2 || n_language_tag_bytes >= 42) {
+        return false;
+    }
+    CHECK_BITS_LEFT(n_language_tag_bytes * 8);
+    char language_tag_bytes[42]; // TS 103 190 part 1 4.3.3.8.7
+    for (uint32_t i = 0; i < n_language_tag_bytes; i++) {
+        language_tag_bytes[i] = (char)mBitReader.getBits(8);
+    }
+    language_tag_bytes[n_language_tag_bytes] = 0;
+    ALOGV("%u.%u: language_tag = %s\n", presentationID, substreamID, language_tag_bytes);
+
+    std::string language(language_tag_bytes, n_language_tag_bytes);
+    mPresentations[presentationID].mLanguage = language;
+
+    return true;
+}
+
+// TS 103 190-1 v1.2.1 E.5 and TS 103 190-2 v1.1.1 E.9
+bool AC4DSIParser::parseSubstreamDSI(uint32_t presentationID, uint32_t substreamID){
+    CHECK_BITS_LEFT(5);
+    uint32_t channel_mode = mBitReader.getBits(5);
+    CHECK_BITS_LEFT(2);
+    uint32_t dsi_sf_multiplier = mBitReader.getBits(2);
+    CHECK_BITS_LEFT(1);
+    bool b_substream_bitrate_indicator = (mBitReader.getBits(1) == 1);
+    ALOGV("%u.%u: channel_mode = %u (%s)\n", presentationID, substreamID, channel_mode,
+    channel_mode < NELEM(ChannelModes) ? ChannelModes[channel_mode] : "reserved");
+    ALOGV("%u.%u: dsi_sf_multiplier = %u\n", presentationID,
+        substreamID, dsi_sf_multiplier);
+    ALOGV("%u.%u: b_substream_bitrate_indicator = %s\n", presentationID,
+        substreamID, BOOLSTR(b_substream_bitrate_indicator));
+
+    if (b_substream_bitrate_indicator) {
+        CHECK_BITS_LEFT(5);
+        uint32_t substream_bitrate_indicator = mBitReader.getBits(5);
+        ALOGV("%u.%u: substream_bitrate_indicator = %u\n", presentationID, substreamID,
+            substream_bitrate_indicator);
+    }
+    if (channel_mode >= 7 && channel_mode <= 10) {
+        CHECK_BITS_LEFT(1);
+        uint32_t add_ch_base = mBitReader.getBits(1);
+        ALOGV("%u.%u: add_ch_base = %u\n", presentationID, substreamID, add_ch_base);
+    }
+    CHECK_BITS_LEFT(1);
+    bool b_content_type = (mBitReader.getBits(1) == 1);
+    ALOGV("%u.%u: b_content_type = %s\n", presentationID, substreamID, BOOLSTR(b_content_type));
+    if (b_content_type) {
+        CHECK_BITS_LEFT(3);
+        uint32_t content_classifier = mBitReader.getBits(3);
+        ALOGV("%u.%u: content_classifier = %u (%s)\n", presentationID, substreamID,
+            content_classifier, ContentClassifier[content_classifier]);
+
+        // For streams based on TS 103 190 part 1 the presentation level channel_mode doesn't
+        // exist and so we use the channel_mode from either the CM or M&E substream
+        // (they are mutually exclusive)
+        if (mPresentations[presentationID].mChannelMode == -1 &&
+            (content_classifier == 0 || content_classifier == 1)) {
+            mPresentations[presentationID].mChannelMode = channel_mode;
+        }
+        mPresentations[presentationID].mContentClassifier = content_classifier;
+        CHECK_BITS_LEFT(1);
+        bool b_language_indicator = (mBitReader.getBits(1) == 1);
+        ALOGV("%u.%u: b_language_indicator = %s\n", presentationID, substreamID,
+            BOOLSTR(b_language_indicator));
+        if (b_language_indicator) {
+            if (!parseLanguageTag(presentationID, substreamID)) {
+                return false;
+            }
+        }
+    }
+
+    return true;
+}
+
+// ETSI TS 103 190-2 v1.1.1 section E.11
+bool AC4DSIParser::parseSubstreamGroupDSI(uint32_t presentationID, uint32_t groupID)
+{
+    CHECK_BITS_LEFT(1);
+    bool b_substreams_present = (mBitReader.getBits(1) == 1);
+    CHECK_BITS_LEFT(1);
+    bool b_hsf_ext = (mBitReader.getBits(1) == 1);
+    CHECK_BITS_LEFT(1);
+    bool b_channel_coded = (mBitReader.getBits(1) == 1);
+    CHECK_BITS_LEFT(8);
+    uint32_t n_substreams = mBitReader.getBits(8);
+    ALOGV("%u.%u: b_substreams_present = %s\n", presentationID, groupID,
+        BOOLSTR(b_substreams_present));
+    ALOGV("%u.%u: b_hsf_ext = %s\n", presentationID, groupID, BOOLSTR(b_hsf_ext));
+    ALOGV("%u.%u: b_channel_coded = %s\n", presentationID, groupID, BOOLSTR(b_channel_coded));
+    ALOGV("%u.%u: n_substreams = %u\n", presentationID, groupID, n_substreams);
+
+    for (uint32_t i = 0; i < n_substreams; i++) {
+        CHECK_BITS_LEFT(2);
+        uint32_t dsi_sf_multiplier = mBitReader.getBits(2);
+        CHECK_BITS_LEFT(1);
+        bool b_substream_bitrate_indicator = (mBitReader.getBits(1) == 1);
+        ALOGV("%u.%u.%u: dsi_sf_multiplier = %u\n", presentationID, groupID, i, dsi_sf_multiplier);
+        ALOGV("%u.%u.%u: b_substream_bitrate_indicator = %s\n", presentationID, groupID, i,
+            BOOLSTR(b_substream_bitrate_indicator));
+
+        if (b_substream_bitrate_indicator) {
+            CHECK_BITS_LEFT(5);
+            uint32_t substream_bitrate_indicator = mBitReader.getBits(5);
+            ALOGV("%u.%u.%u: substream_bitrate_indicator = %u\n", presentationID, groupID, i,
+                substream_bitrate_indicator);
+        }
+        if (b_channel_coded) {
+            CHECK_BITS_LEFT(24);
+            uint32_t dsi_substream_channel_mask = mBitReader.getBits(24);
+            ALOGV("%u.%u.%u: dsi_substream_channel_mask = 0x%06x\n", presentationID, groupID, i,
+                dsi_substream_channel_mask);
+        } else {
+            CHECK_BITS_LEFT(1);
+            bool b_ajoc = (mBitReader.getBits(1) == 1);
+            ALOGV("%u.%u.%u: b_ajoc = %s\n", presentationID, groupID, i, BOOLSTR(b_ajoc));
+            if (b_ajoc) {
+                CHECK_BITS_LEFT(1);
+                bool b_static_dmx = (mBitReader.getBits(1) == 1);
+                ALOGV("%u.%u.%u: b_static_dmx = %s\n", presentationID, groupID, i,
+                    BOOLSTR(b_static_dmx));
+                if (!b_static_dmx) {
+                    CHECK_BITS_LEFT(4);
+                    uint32_t n_dmx_objects_minus1 = mBitReader.getBits(4);
+                    ALOGV("%u.%u.%u: n_dmx_objects_minus1 = %u\n", presentationID, groupID, i,
+                        n_dmx_objects_minus1);
+                }
+                CHECK_BITS_LEFT(6);
+                uint32_t n_umx_objects_minus1 = mBitReader.getBits(6);
+                ALOGV("%u.%u.%u: n_umx_objects_minus1 = %u\n", presentationID, groupID, i,
+                    n_umx_objects_minus1);
+            }
+            CHECK_BITS_LEFT(4);
+            mBitReader.skipBits(4); // objects_assignment_mask
+        }
+    }
+
+    CHECK_BITS_LEFT(1);
+    bool b_content_type = (mBitReader.getBits(1) == 1);
+    ALOGV("%u.%u: b_content_type = %s\n", presentationID, groupID, BOOLSTR(b_content_type));
+    if (b_content_type) {
+        CHECK_BITS_LEFT(3);
+        uint32_t content_classifier = mBitReader.getBits(3);
+        ALOGV("%u.%u: content_classifier = %s (%u)\n", presentationID, groupID,
+            ContentClassifier[content_classifier], content_classifier);
+
+        mPresentations[presentationID].mContentClassifier = content_classifier;
+
+        CHECK_BITS_LEFT(1);
+        bool b_language_indicator = (mBitReader.getBits(1) == 1);
+        ALOGV("%u.%u: b_language_indicator = %s\n", presentationID, groupID,
+            BOOLSTR(b_language_indicator));
+
+        if (b_language_indicator) {
+            if (!parseLanguageTag(presentationID, groupID)) {
+                return false;
+            }
+        }
+    }
+
+    return true;
+}
+
+bool AC4DSIParser::parseBitrateDsi() {
+    CHECK_BITS_LEFT(2 + 32 + 32);
+    mBitReader.skipBits(2); // bit_rate_mode
+    mBitReader.skipBits(32); // bit_rate
+    mBitReader.skipBits(32); // bit_rate_precision
+
+    return true;
+}
+
+// TS 103 190-1 section E.4 (ac4_dsi) and TS 103 190-2 section E.6 (ac4_dsi_v1)
+bool AC4DSIParser::parse() {
+    CHECK_BITS_LEFT(3);
+    uint32_t ac4_dsi_version = mBitReader.getBits(3);
+    if (ac4_dsi_version > 1) {
+        ALOGE("error while parsing ac-4 dsi: only versions 0 and 1 are supported");
+        return false;
+    }
+
+    CHECK_BITS_LEFT(7 + 1 + 4 + 9);
+    uint32_t bitstream_version = mBitReader.getBits(7);
+    mBitReader.skipBits(1); // fs_index
+    mBitReader.skipBits(4); // frame_rate_index
+    uint32_t n_presentations = mBitReader.getBits(9);
+
+    int32_t short_program_id = -1;
+    if (bitstream_version > 1) {
+        if (ac4_dsi_version == 0){
+            ALOGE("invalid ac4 dsi");
+            return false;
+        }
+        CHECK_BITS_LEFT(1);
+        bool b_program_id = (mBitReader.getBits(1) == 1);
+        if (b_program_id) {
+            CHECK_BITS_LEFT(16 + 1);
+            short_program_id = mBitReader.getBits(16);
+            bool b_uuid = (mBitReader.getBits(1) == 1);
+            if (b_uuid) {
+                const uint32_t kAC4UUIDSizeInBytes = 16;
+                char program_uuid[kAC4UUIDSizeInBytes];
+                CHECK_BITS_LEFT(kAC4UUIDSizeInBytes * 8);
+                for (uint32_t i = 0; i < kAC4UUIDSizeInBytes; i++) {
+                    program_uuid[i] = (char)(mBitReader.getBits(8));
+                }
+                ALOGV("UUID = %s", program_uuid);
+            }
+        }
+    }
+
+    if (ac4_dsi_version == 1) {
+        if (!parseBitrateDsi()) {
+            return false;
+        }
+        BYTE_ALIGN;
+    }
+
+    for (uint32_t presentation = 0; presentation < n_presentations; presentation++) {
+        mPresentations[presentation].mProgramID = short_program_id;
+        // known as b_single_substream in ac4_dsi_version 0
+        bool b_single_substream_group = false;
+        uint32_t presentation_config = 0, presentation_version = 0;
+        uint32_t pres_bytes = 0;
+
+        if (ac4_dsi_version == 0) {
+            CHECK_BITS_LEFT(1 + 5 + 5);
+            b_single_substream_group = (mBitReader.getBits(1) == 1);
+            presentation_config = mBitReader.getBits(5);
+            presentation_version = mBitReader.getBits(5);
+        } else if (ac4_dsi_version == 1) {
+            CHECK_BITS_LEFT(8 + 8);
+            presentation_version = mBitReader.getBits(8);
+            pres_bytes = mBitReader.getBits(8);
+            if (pres_bytes == 0xff) {
+                CHECK_BITS_LEFT(16);
+                pres_bytes += mBitReader.getBits(16);
+            }
+            ALOGV("%u: pres_bytes = %u\n", presentation, pres_bytes);
+            if (presentation_version > 1) {
+                CHECK_BITS_LEFT(pres_bytes * 8);
+                mBitReader.skipBits(pres_bytes * 8);
+                continue;
+            }
+            // ac4_presentation_v0_dsi() and ac4_presentation_v1_dsi() both
+            // start with a presentation_config of 5 bits
+            CHECK_BITS_LEFT(5);
+            presentation_config = mBitReader.getBits(5);
+            b_single_substream_group = (presentation_config == 0x1f);
+        }
+
+        static const char *PresentationConfig[] = {
+            "Music&Effects + Dialog",
+            "Main + DE",
+            "Main + Associate",
+            "Music&Effects + Dialog + Associate",
+            "Main + DE + Associate",
+            "Arbitrary substream groups",
+            "EMDF only"
+        };
+        ALOGV("%u: b_single_substream/group = %s\n", presentation,
+            BOOLSTR(b_single_substream_group));
+        ALOGV("%u: presentation_version = %u\n", presentation, presentation_version);
+        ALOGV("%u: presentation_config = %u (%s)\n", presentation, presentation_config,
+            (presentation_config >= NELEM(PresentationConfig) ?
+            "reserved" : PresentationConfig[presentation_config]));
+
+        /* record a marker, less the size of the presentation_config */
+        uint64_t start = (mDSISize - mBitReader.numBitsLeft()) / 8;
+
+        bool b_add_emdf_substreams = false;
+        if (!b_single_substream_group && presentation_config == 6) {
+            b_add_emdf_substreams = true;
+            ALOGV("%u: b_add_emdf_substreams = %s\n", presentation, BOOLSTR(b_add_emdf_substreams));
+        } else {
+            CHECK_BITS_LEFT(3 + 1);
+            uint32_t mdcompat = mBitReader.getBits(3);
+            ALOGV("%u: mdcompat = %d\n", presentation, mdcompat);
+
+            bool b_presentation_group_index = (mBitReader.getBits(1) == 1);
+            ALOGV("%u: b_presentation_group_index = %s\n", presentation,
+                BOOLSTR(b_presentation_group_index));
+            if (b_presentation_group_index) {
+                CHECK_BITS_LEFT(5);
+                mPresentations[presentation].mGroupIndex = mBitReader.getBits(5);
+                ALOGV("%u: presentation_group_index = %d\n", presentation,
+                    mPresentations[presentation].mGroupIndex);
+            }
+            CHECK_BITS_LEFT(2);
+            uint32_t dsi_frame_rate_multiply_info = mBitReader.getBits(2);
+            ALOGV("%u: dsi_frame_rate_multiply_info = %d\n", presentation,
+                dsi_frame_rate_multiply_info);
+            if (ac4_dsi_version == 1 && presentation_version == 1) {
+                CHECK_BITS_LEFT(2);
+                uint32_t dsi_frame_rate_fraction_info = mBitReader.getBits(2);
+                ALOGV("%u: dsi_frame_rate_fraction_info = %d\n", presentation,
+                    dsi_frame_rate_fraction_info);
+            }
+            CHECK_BITS_LEFT(5 + 10);
+            uint32_t presentation_emdf_version = mBitReader.getBits(5);
+            uint32_t presentation_key_id = mBitReader.getBits(10);
+            ALOGV("%u: presentation_emdf_version = %d\n", presentation, presentation_emdf_version);
+            ALOGV("%u: presentation_key_id = %d\n", presentation, presentation_key_id);
+
+            if (ac4_dsi_version == 1) {
+                bool b_presentation_channel_coded = false;
+                if (presentation_version == 0) {
+                    b_presentation_channel_coded = true;
+                } else {
+                    CHECK_BITS_LEFT(1);
+                    b_presentation_channel_coded = (mBitReader.getBits(1) == 1);
+                }
+                ALOGV("%u: b_presentation_channel_coded = %s\n", presentation,
+                    BOOLSTR(b_presentation_channel_coded));
+                if (b_presentation_channel_coded) {
+                    if (presentation_version == 1) {
+                        CHECK_BITS_LEFT(5);
+                        uint32_t dsi_presentation_ch_mode = mBitReader.getBits(5);
+                        mPresentations[presentation].mChannelMode = dsi_presentation_ch_mode;
+                        ALOGV("%u: dsi_presentation_ch_mode = %d (%s)\n", presentation,
+                            dsi_presentation_ch_mode,
+                            dsi_presentation_ch_mode < NELEM(ChannelModes) ?
+                            ChannelModes[dsi_presentation_ch_mode] : "reserved");
+
+                        if (dsi_presentation_ch_mode >= 11 && dsi_presentation_ch_mode <= 14) {
+                            CHECK_BITS_LEFT(1 + 2);
+                            uint32_t pres_b_4_back_channels_present = mBitReader.getBits(1);
+                            uint32_t pres_top_channel_pairs = mBitReader.getBits(2);
+                            ALOGV("%u: pres_b_4_back_channels_present = %s\n", presentation,
+                                BOOLSTR(pres_b_4_back_channels_present));
+                            ALOGV("%u: pres_top_channel_pairs = %d\n", presentation,
+                                pres_top_channel_pairs);
+                        }
+                    }
+                    // presentation_channel_mask in ac4_presentation_v0_dsi()
+                    CHECK_BITS_LEFT(24);
+                    uint32_t presentation_channel_mask_v1 = mBitReader.getBits(24);
+                    ALOGV("%u: presentation_channel_mask_v1 = 0x%06x\n", presentation,
+                        presentation_channel_mask_v1);
+                }
+                if (presentation_version == 1) {
+                    CHECK_BITS_LEFT(1);
+                    bool b_presentation_core_differs = (mBitReader.getBits(1) == 1);
+                    ALOGV("%u: b_presentation_core_differs = %s\n", presentation,
+                        BOOLSTR(b_presentation_core_differs));
+                    if (b_presentation_core_differs) {
+                        CHECK_BITS_LEFT(1);
+                        bool b_presentation_core_channel_coded = (mBitReader.getBits(1) == 1);
+                        if (b_presentation_core_channel_coded) {
+                            CHECK_BITS_LEFT(2);
+                            mBitReader.skipBits(2); // dsi_presentation_channel_mode_core
+                        }
+                    }
+                    CHECK_BITS_LEFT(1);
+                    bool b_presentation_filter = (mBitReader.getBits(1) == 1);
+                    ALOGV("%u: b_presentation_filter = %s\n", presentation,
+                        BOOLSTR(b_presentation_filter));
+                    if (b_presentation_filter) {
+                        CHECK_BITS_LEFT(1 + 8);
+                        bool b_enable_presentation = (mBitReader.getBits(1) == 1);
+                        if (!b_enable_presentation) {
+                            mPresentations[presentation].mEnabled = false;
+                        }
+                        ALOGV("%u: b_enable_presentation = %s\n", presentation,
+                            BOOLSTR(b_enable_presentation));
+                        uint32_t n_filter_bytes = mBitReader.getBits(8);
+                        CHECK_BITS_LEFT(n_filter_bytes * 8);
+                        for (uint32_t i = 0; i < n_filter_bytes; i++) {
+                            mBitReader.skipBits(8); // filter_data
+                        }
+                    }
+                }
+            } /* ac4_dsi_version == 1 */
+
+            if (b_single_substream_group) {
+                if (presentation_version == 0) {
+                    if (!parseSubstreamDSI(presentation, 0)) {
+                        return false;
+                    }
+                } else {
+                    if (!parseSubstreamGroupDSI(presentation, 0)) {
+                        return false;
+                    }
+                }
+            } else {
+                if (ac4_dsi_version == 1) {
+                    CHECK_BITS_LEFT(1);
+                    bool b_multi_pid = (mBitReader.getBits(1) == 1);
+                    ALOGV("%u: b_multi_pid = %s\n", presentation, BOOLSTR(b_multi_pid));
+                } else {
+                    CHECK_BITS_LEFT(1);
+                    bool b_hsf_ext = (mBitReader.getBits(1) == 1);
+                    ALOGV("%u: b_hsf_ext = %s\n", presentation, BOOLSTR(b_hsf_ext));
+                }
+                switch (presentation_config) {
+                case 0:
+                case 1:
+                case 2:
+                    if (presentation_version == 0) {
+                        if (!parseSubstreamDSI(presentation, 0)) {
+                            return false;
+                        }
+                        if (!parseSubstreamDSI(presentation, 1)) {
+                            return false;
+                        }
+                    } else {
+                        if (!parseSubstreamGroupDSI(presentation, 0)) {
+                            return false;
+                        }
+                        if (!parseSubstreamGroupDSI(presentation, 1)) {
+                            return false;
+                        }
+                    }
+                    break;
+                case 3:
+                case 4:
+                    if (presentation_version == 0) {
+                        if (!parseSubstreamDSI(presentation, 0)) {
+                            return false;
+                        }
+                        if (!parseSubstreamDSI(presentation, 1)) {
+                            return false;
+                        }
+                        if (!parseSubstreamDSI(presentation, 2)) {
+                            return false;
+                        }
+                    } else {
+                        if (!parseSubstreamGroupDSI(presentation, 0)) {
+                            return false;
+                        }
+                        if (!parseSubstreamGroupDSI(presentation, 1)) {
+                            return false;
+                        }
+                        if (!parseSubstreamGroupDSI(presentation, 2)) {
+                            return false;
+                        }
+                    }
+                    break;
+                case 5:
+                    if (presentation_version == 0) {
+                        if (!parseSubstreamDSI(presentation, 0)) {
+                            return false;
+                        }
+                    } else {
+                        CHECK_BITS_LEFT(3);
+                        uint32_t n_substream_groups_minus2 = mBitReader.getBits(3);
+                        ALOGV("%u: n_substream_groups_minus2 = %d\n", presentation,
+                            n_substream_groups_minus2);
+                        for (uint32_t sg = 0; sg < n_substream_groups_minus2 + 2; sg++) {
+                            if (!parseSubstreamGroupDSI(presentation, sg)) {
+                                return false;
+                            }
+                        }
+                    }
+                    break;
+                default:
+                    CHECK_BITS_LEFT(7);
+                    uint32_t n_skip_bytes = mBitReader.getBits(7);
+                    CHECK_BITS_LEFT(n_skip_bytes * 8)
+                    for (uint32_t j = 0; j < n_skip_bytes; j++) {
+                        mBitReader.getBits(8);
+                    }
+                    break;
+                }
+                CHECK_BITS_LEFT(1 + 1);
+                bool b_pre_virtualized = (mBitReader.getBits(1) == 1);
+                mPresentations[presentation].mPreVirtualized = b_pre_virtualized;
+                b_add_emdf_substreams = (mBitReader.getBits(1) == 1);
+                ALOGV("%u: b_pre_virtualized = %s\n", presentation, BOOLSTR(b_pre_virtualized));
+                ALOGV("%u: b_add_emdf_substreams = %s\n", presentation,
+                    BOOLSTR(b_add_emdf_substreams));
+            }
+        }
+        if (b_add_emdf_substreams) {
+            CHECK_BITS_LEFT(7);
+            uint32_t n_add_emdf_substreams = mBitReader.getBits(7);
+            for (uint32_t j = 0; j < n_add_emdf_substreams; j++) {
+                CHECK_BITS_LEFT(5 + 10);
+                uint32_t substream_emdf_version = mBitReader.getBits(5);
+                uint32_t substream_key_id = mBitReader.getBits(10);
+                ALOGV("%u: emdf_substream[%d]: version=%d, key_id=%d\n", presentation, j,
+                    substream_emdf_version, substream_key_id);
+            }
+        }
+
+        bool b_presentation_bitrate_info = false;
+        if (presentation_version > 0) {
+            CHECK_BITS_LEFT(1);
+            b_presentation_bitrate_info = (mBitReader.getBits(1) == 1);
+        }
+
+        ALOGV("b_presentation_bitrate_info = %s\n", BOOLSTR(b_presentation_bitrate_info));
+        if (b_presentation_bitrate_info) {
+            if (!parseBitrateDsi()) {
+                return false;
+            }
+        }
+
+        if (presentation_version > 0) {
+            CHECK_BITS_LEFT(1);
+            bool b_alternative = (mBitReader.getBits(1) == 1);
+            ALOGV("b_alternative = %s\n", BOOLSTR(b_alternative));
+            if (b_alternative) {
+                BYTE_ALIGN;
+                CHECK_BITS_LEFT(16);
+                uint32_t name_len = mBitReader.getBits(16);
+                char* presentation_name = new char[name_len+1];
+                CHECK_BITS_LEFT(name_len * 8);
+                for (uint32_t i = 0; i < name_len; i++) {
+                    presentation_name[i] = (char)(mBitReader.getBits(8));
+                }
+                presentation_name[name_len] = '\0';
+                std::string description(presentation_name, name_len);
+                mPresentations[presentation].mDescription = description;
+                CHECK_BITS_LEFT(5);
+                uint32_t n_targets = mBitReader.getBits(5);
+                CHECK_BITS_LEFT(n_targets * (3 + 8));
+                for (uint32_t i = 0; i < n_targets; i++){
+                    mBitReader.skipBits(3); // target_md_compat
+                    mBitReader.skipBits(8); // target_device_category
+                }
+            }
+        }
+
+        BYTE_ALIGN;
+
+        if (ac4_dsi_version == 1) {
+            uint64_t end = (mDSISize - mBitReader.numBitsLeft()) / 8;
+            if (mBitReader.numBitsLeft() % 8 != 0) {
+                end += 1;
+            }
+
+            uint64_t presentation_bytes = end - start;
+            uint64_t skip_bytes = pres_bytes - presentation_bytes;
+            ALOGV("skipping = %" PRIu64 " bytes", skip_bytes);
+            CHECK_BITS_LEFT(skip_bytes * 8);
+            mBitReader.skipBits(skip_bytes * 8);
+        }
+
+        // we should know this or something is probably wrong
+        // with the bitstream (or we don't support it)
+        if (mPresentations[presentation].mChannelMode == -1){
+            ALOGE("could not determing channel mode of presentation %d", presentation);
+            return false;
+        }
+    } /* each presentation */
+
+    return true;
+}
+
+};
diff --git a/media/extractors/mp4/AC4Parser.h b/media/extractors/mp4/AC4Parser.h
new file mode 100644
index 0000000..73b6e31
--- /dev/null
+++ b/media/extractors/mp4/AC4Parser.h
@@ -0,0 +1,109 @@
+/*
+ * 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 AC4_PARSER_H_
+#define AC4_PARSER_H_
+
+#include <cstdint>
+#include <map>
+#include <string>
+
+#include <media/stagefright/foundation/ABitReader.h>
+
+namespace android {
+
+class AC4Parser {
+public:
+    AC4Parser();
+    virtual ~AC4Parser() { }
+
+    virtual bool parse() = 0;
+
+    struct AC4Presentation {
+        int32_t mChannelMode = -1;
+        int32_t mProgramID = -1;
+        int32_t mGroupIndex = -1;
+
+        // TS 103 190-1 v1.2.1 4.3.3.8.1
+        enum ContentClassifiers {
+            kCompleteMain,
+            kMusicAndEffects,
+            kVisuallyImpaired,
+            kHearingImpaired,
+            kDialog,
+            kCommentary,
+            kEmergency,
+            kVoiceOver
+        };
+
+        uint32_t mContentClassifier = kCompleteMain;
+
+        // ETSI TS 103 190-2 V1.1.1 (2015-09) Table 79: channel_mode
+        enum InputChannelMode {
+            kChannelMode_Mono,
+            kChannelMode_Stereo,
+            kChannelMode_3_0,
+            kChannelMode_5_0,
+            kChannelMode_5_1,
+            kChannelMode_7_0_34,
+            kChannelMode_7_1_34,
+            kChannelMode_7_0_52,
+            kChannelMode_7_1_52,
+            kChannelMode_7_0_322,
+            kChannelMode_7_1_322,
+            kChannelMode_7_0_4,
+            kChannelMode_7_1_4,
+            kChannelMode_9_0_4,
+            kChannelMode_9_1_4,
+            kChannelMode_22_2,
+            kChannelMode_Reserved,
+        };
+
+        bool mHasDialogEnhancements = false;
+        bool mPreVirtualized = false;
+        bool mEnabled = true;
+
+        std::string mLanguage;
+        std::string mDescription;
+    };
+    typedef std::map<uint32_t, AC4Presentation> AC4Presentations;
+
+    const AC4Presentations& getPresentations() const { return mPresentations; }
+
+protected:
+    AC4Presentations mPresentations;
+};
+
+class AC4DSIParser: public AC4Parser {
+public:
+    explicit AC4DSIParser(ABitReader &br);
+    virtual ~AC4DSIParser() { }
+
+    bool parse();
+
+private:
+    bool parseSubstreamDSI(uint32_t presentationID, uint32_t substreamID);
+    bool parseSubstreamGroupDSI(uint32_t presentationID, uint32_t groupID);
+    bool parseLanguageTag(uint32_t presentationID, uint32_t substreamID);
+    bool parseBitrateDsi();
+
+    uint64_t mDSISize;
+    ABitReader& mBitReader;
+};
+
+};
+
+#endif  // AC4_PARSER_H_
diff --git a/media/extractors/mp4/Android.bp b/media/extractors/mp4/Android.bp
index fa739e8..40b2c97 100644
--- a/media/extractors/mp4/Android.bp
+++ b/media/extractors/mp4/Android.bp
@@ -2,6 +2,7 @@
     name: "libmp4extractor_defaults",
 
     srcs: [
+        "AC4Parser.cpp",
         "ItemTable.cpp",
         "MPEG4Extractor.cpp",
         "SampleIterator.cpp",
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/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index 7b3b81d..fe9f99c 100644
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -26,6 +26,7 @@
 
 #include <utils/Log.h>
 
+#include "AC4Parser.h"
 #include "MPEG4Extractor.h"
 #include "SampleTable.h"
 #include "ItemTable.h"
@@ -125,6 +126,8 @@
 
     bool mIsAVC;
     bool mIsHEVC;
+    bool mIsAC4;
+
     size_t mNALLengthSize;
 
     bool mStarted;
@@ -310,6 +313,9 @@
         case FOURCC('s', 'a', 'w', 'b'):
             return MEDIA_MIMETYPE_AUDIO_AMR_WB;
 
+        case FOURCC('e', 'c', '-', '3'):
+            return MEDIA_MIMETYPE_AUDIO_EAC3;
+
         case FOURCC('m', 'p', '4', 'v'):
             return MEDIA_MIMETYPE_VIDEO_MPEG4;
 
@@ -324,6 +330,8 @@
         case FOURCC('h', 'v', 'c', '1'):
         case FOURCC('h', 'e', 'v', '1'):
             return MEDIA_MIMETYPE_VIDEO_HEVC;
+        case FOURCC('a', 'c', '-', '4'):
+            return MEDIA_MIMETYPE_AUDIO_AC4;
         default:
             ALOGW("Unknown fourcc: %c%c%c%c",
                    (fourcc >> 24) & 0xff,
@@ -2433,7 +2441,19 @@
         case FOURCC('a', 'c', '-', '3'):
         {
             *offset += chunk_size;
-            return parseAC3SampleEntry(data_offset);
+            return parseAC3SpecificBox(data_offset);
+        }
+
+        case FOURCC('e', 'c', '-', '3'):
+        {
+            *offset += chunk_size;
+            return parseEAC3SpecificBox(data_offset);
+        }
+
+        case FOURCC('a', 'c', '-', '4'):
+        {
+            *offset += chunk_size;
+            return parseAC4SpecificBox(data_offset);
         }
 
         case FOURCC('f', 't', 'y', 'p'):
@@ -2507,36 +2527,260 @@
     return OK;
 }
 
-status_t MPEG4Extractor::parseAC3SampleEntry(off64_t offset) {
+status_t MPEG4Extractor::parseChannelCountSampleRate(
+        off64_t *offset, uint16_t *channelCount, uint16_t *sampleRate) {
     // skip 16 bytes:
     //  + 6-byte reserved,
     //  + 2-byte data reference index,
     //  + 8-byte reserved
-    offset += 16;
-    uint16_t channelCount;
-    if (!mDataSource->getUInt16(offset, &channelCount)) {
+    *offset += 16;
+    if (!mDataSource->getUInt16(*offset, channelCount)) {
+        ALOGE("MPEG4Extractor: error while reading sample entry box: cannot read channel count");
         return ERROR_MALFORMED;
     }
     // skip 8 bytes:
     //  + 2-byte channelCount,
     //  + 2-byte sample size,
     //  + 4-byte reserved
-    offset += 8;
-    uint16_t sampleRate;
-    if (!mDataSource->getUInt16(offset, &sampleRate)) {
-        ALOGE("MPEG4Extractor: error while reading ac-3 block: cannot read sample rate");
+    *offset += 8;
+    if (!mDataSource->getUInt16(*offset, sampleRate)) {
+        ALOGE("MPEG4Extractor: error while reading sample entry box: cannot read sample rate");
         return ERROR_MALFORMED;
     }
-
     // skip 4 bytes:
     //  + 2-byte sampleRate,
     //  + 2-byte reserved
-    offset += 4;
-    return parseAC3SpecificBox(offset, sampleRate);
+    *offset += 4;
+    return OK;
 }
 
-status_t MPEG4Extractor::parseAC3SpecificBox(
-        off64_t offset, uint16_t sampleRate) {
+status_t MPEG4Extractor::parseAC4SpecificBox(off64_t offset) {
+    if (mLastTrack == NULL) {
+        return ERROR_MALFORMED;
+    }
+
+    uint16_t sampleRate, channelCount;
+    status_t status;
+    if ((status = parseChannelCountSampleRate(&offset, &channelCount, &sampleRate)) != OK) {
+        return status;
+    }
+    uint32_t size;
+    // + 4-byte size
+    // + 4-byte type
+    // + 3-byte payload
+    const uint32_t kAC4MinimumBoxSize = 4 + 4 + 3;
+    if (!mDataSource->getUInt32(offset, &size) || size < kAC4MinimumBoxSize) {
+        ALOGE("MPEG4Extractor: error while reading ac-4 block: cannot read specific box size");
+        return ERROR_MALFORMED;
+    }
+
+    // + 4-byte size
+    offset += 4;
+    uint32_t type;
+    if (!mDataSource->getUInt32(offset, &type) || type != FOURCC('d', 'a', 'c', '4')) {
+        ALOGE("MPEG4Extractor: error while reading ac-4 specific block: header not dac4");
+        return ERROR_MALFORMED;
+    }
+
+    // + 4-byte type
+    offset += 4;
+    // at least for AC4 DSI v1 this is big enough
+    const uint32_t kAC4SpecificBoxPayloadSize = 256;
+    uint8_t chunk[kAC4SpecificBoxPayloadSize];
+    ssize_t dsiSize = size - 8; // size of box - size and type fields
+    if (dsiSize >= (ssize_t)kAC4SpecificBoxPayloadSize ||
+        mDataSource->readAt(offset, chunk, dsiSize) != dsiSize) {
+        ALOGE("MPEG4Extractor: error while reading ac-4 specific block: bitstream fields");
+        return ERROR_MALFORMED;
+    }
+    // + size-byte payload
+    offset += dsiSize;
+    ABitReader br(chunk, dsiSize);
+    AC4DSIParser parser(br);
+    if (!parser.parse()){
+        ALOGE("MPEG4Extractor: error while parsing ac-4 specific block");
+        return ERROR_MALFORMED;
+    }
+
+    mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC4);
+    mLastTrack->meta.setInt32(kKeyChannelCount, channelCount);
+    mLastTrack->meta.setInt32(kKeySampleRate, sampleRate);
+    return OK;
+}
+
+status_t MPEG4Extractor::parseEAC3SpecificBox(off64_t offset) {
+    if (mLastTrack == NULL) {
+        return ERROR_MALFORMED;
+    }
+
+    uint16_t sampleRate, channels;
+    status_t status;
+    if ((status = parseChannelCountSampleRate(&offset, &channels, &sampleRate)) != OK) {
+        return status;
+    }
+    uint32_t size;
+    // + 4-byte size
+    // + 4-byte type
+    // + 3-byte payload
+    const uint32_t kEAC3SpecificBoxMinSize = 11;
+    // 13 + 3 + (8 * (2 + 5 + 5 + 3 + 1 + 3 + 4 + (14 * 9 + 1))) bits == 152 bytes theoretical max
+    // calculated from the required bits read below as well as the maximum number of independent
+    // and dependant sub streams you can have
+    const uint32_t kEAC3SpecificBoxMaxSize = 152;
+    if (!mDataSource->getUInt32(offset, &size) ||
+        size < kEAC3SpecificBoxMinSize ||
+        size > kEAC3SpecificBoxMaxSize) {
+        ALOGE("MPEG4Extractor: error while reading eac-3 block: cannot read specific box size");
+        return ERROR_MALFORMED;
+    }
+
+    offset += 4;
+    uint32_t type;
+    if (!mDataSource->getUInt32(offset, &type) || type != FOURCC('d', 'e', 'c', '3')) {
+        ALOGE("MPEG4Extractor: error while reading eac-3 specific block: header not dec3");
+        return ERROR_MALFORMED;
+    }
+
+    offset += 4;
+    uint8_t* chunk = new (std::nothrow) uint8_t[size];
+    if (chunk == NULL) {
+        return ERROR_MALFORMED;
+    }
+
+    if (mDataSource->readAt(offset, chunk, size) != (ssize_t)size) {
+        ALOGE("MPEG4Extractor: error while reading eac-3 specific block: bitstream fields");
+        delete[] chunk;
+        return ERROR_MALFORMED;
+    }
+
+    ABitReader br(chunk, size);
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 3, 4, 4, 5};
+    static const unsigned sampleRateTable[] = {48000, 44100, 32000};
+
+    if (br.numBitsLeft() < 16) {
+        delete[] chunk;
+        return ERROR_MALFORMED;
+    }
+    unsigned data_rate = br.getBits(13);
+    ALOGV("EAC3 data rate = %d", data_rate);
+
+    unsigned num_ind_sub = br.getBits(3) + 1;
+    ALOGV("EAC3 independant substreams = %d", num_ind_sub);
+    if (br.numBitsLeft() < (num_ind_sub * 23)) {
+        delete[] chunk;
+        return ERROR_MALFORMED;
+    }
+
+    unsigned channelCount = 0;
+    for (unsigned i = 0; i < num_ind_sub; i++) {
+        unsigned fscod = br.getBits(2);
+        if (fscod == 3) {
+            ALOGE("Incorrect fscod (3) in EAC3 header");
+            delete[] chunk;
+            return ERROR_MALFORMED;
+        }
+        unsigned boxSampleRate = sampleRateTable[fscod];
+        if (boxSampleRate != sampleRate) {
+            ALOGE("sample rate mismatch: boxSampleRate = %d, sampleRate = %d",
+                boxSampleRate, sampleRate);
+            delete[] chunk;
+            return ERROR_MALFORMED;
+        }
+
+        unsigned bsid = br.getBits(5);
+        if (bsid < 8) {
+            ALOGW("Incorrect bsid in EAC3 header. Possibly AC-3?");
+            delete[] chunk;
+            return ERROR_MALFORMED;
+        }
+
+        // skip
+        br.skipBits(2);
+        unsigned bsmod = br.getBits(3);
+        unsigned acmod = br.getBits(3);
+        unsigned lfeon = br.getBits(1);
+        // we currently only support the first stream
+        if (i == 0)
+            channelCount = channelCountTable[acmod] + lfeon;
+        ALOGV("bsmod = %d, acmod = %d, lfeon = %d", bsmod, acmod, lfeon);
+
+        br.skipBits(3);
+        unsigned num_dep_sub = br.getBits(4);
+        ALOGV("EAC3 dependant substreams = %d", num_dep_sub);
+        if (num_dep_sub != 0) {
+            if (br.numBitsLeft() < 9) {
+                delete[] chunk;
+                return ERROR_MALFORMED;
+            }
+            static const char* chan_loc_tbl[] = { "Lc/Rc","Lrs/Rrs","Cs","Ts","Lsd/Rsd",
+                "Lw/Rw","Lvh/Rvh","Cvh","Lfe2" };
+            unsigned chan_loc = br.getBits(9);
+            unsigned mask = 1;
+            for (unsigned j = 0; j < 9; j++, mask <<= 1) {
+                if ((chan_loc & mask) != 0) {
+                    // we currently only support the first stream
+                    if (i == 0) {
+                        channelCount++;
+                        // these are 2 channels in the mask
+                        if (j == 0 || j == 1 || j == 4 || j == 5 || j == 6) {
+                            channelCount++;
+                        }
+                    }
+                    ALOGV(" %s", chan_loc_tbl[j]);
+                }
+            }
+        } else {
+            if (br.numBitsLeft() == 0) {
+                delete[] chunk;
+                return ERROR_MALFORMED;
+            }
+            br.skipBits(1);
+        }
+    }
+
+    if (br.numBitsLeft() != 0) {
+        if (br.numBitsLeft() < 8) {
+            delete[] chunk;
+            return ERROR_MALFORMED;
+        }
+        unsigned mask = br.getBits(8);
+        for (unsigned i = 0; i < 8; i++) {
+            if (((0x1 << i) && mask) == 0)
+                continue;
+
+            if (br.numBitsLeft() < 8) {
+                delete[] chunk;
+                return ERROR_MALFORMED;
+            }
+            switch (i) {
+                case 0: {
+                    unsigned complexity = br.getBits(8);
+                    ALOGV("Found a JOC stream with complexity = %d", complexity);
+                }break;
+                default: {
+                    br.skipBits(8);
+                }break;
+            }
+        }
+    }
+    mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_EAC3);
+    mLastTrack->meta.setInt32(kKeyChannelCount, channelCount);
+    mLastTrack->meta.setInt32(kKeySampleRate, sampleRate);
+
+    delete[] chunk;
+    return OK;
+}
+
+status_t MPEG4Extractor::parseAC3SpecificBox(off64_t offset) {
+    if (mLastTrack == NULL) {
+        return ERROR_MALFORMED;
+    }
+
+    uint16_t sampleRate, channels;
+    status_t status;
+    if ((status = parseChannelCountSampleRate(&offset, &channels, &sampleRate)) != OK) {
+        return status;
+    }
     uint32_t size;
     // + 4-byte size
     // + 4-byte type
@@ -2591,9 +2835,6 @@
     unsigned lfeon = br.getBits(1);
     unsigned channelCount = channelCountTable[acmod] + lfeon;
 
-    if (mLastTrack == NULL) {
-        return ERROR_MALFORMED;
-    }
     mLastTrack->meta.setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC3);
     mLastTrack->meta.setInt32(kKeyChannelCount, channelCount);
     mLastTrack->meta.setInt32(kKeySampleRate, sampleRate);
@@ -3857,6 +4098,7 @@
       mCurrentSampleInfoOffsets(NULL),
       mIsAVC(false),
       mIsHEVC(false),
+      mIsAC4(false),
       mNALLengthSize(0),
       mStarted(false),
       mGroup(NULL),
@@ -3890,6 +4132,7 @@
     mIsAVC = !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_AVC);
     mIsHEVC = !strcasecmp(mime, MEDIA_MIMETYPE_VIDEO_HEVC) ||
               !strcasecmp(mime, MEDIA_MIMETYPE_IMAGE_ANDROID_HEIC);
+    mIsAC4 = !strcasecmp(mime, MEDIA_MIMETYPE_AUDIO_AC4);
 
     if (mIsAVC) {
         uint32_t type;
@@ -4830,7 +5073,7 @@
         }
     }
 
-    if ((!mIsAVC && !mIsHEVC) || mWantsNALFragments) {
+    if ((!mIsAVC && !mIsHEVC && !mIsAC4) || mWantsNALFragments) {
         if (newBuffer) {
             ssize_t num_bytes_read =
                 mDataSource->readAt(offset, (uint8_t *)mBuffer->data(), size);
@@ -4862,13 +5105,20 @@
             ++mCurrentSampleIndex;
         }
 
-        if (!mIsAVC && !mIsHEVC) {
+        if (!mIsAVC && !mIsHEVC && !mIsAC4) {
             *out = mBuffer;
             mBuffer = NULL;
 
             return OK;
         }
 
+        if (mIsAC4) {
+            mBuffer->release();
+            mBuffer = NULL;
+
+            return ERROR_IO;
+        }
+
         // Each NAL unit is split up into its constituent fragments and
         // each one of them returned in its own buffer.
 
@@ -4907,6 +5157,58 @@
         *out = clone;
 
         return OK;
+    } else if (mIsAC4) {
+        CHECK(mBuffer != NULL);
+        // Make sure there is enough space to write the sync header and the raw frame
+        if (mBuffer->range_length() < (7 + size)) {
+            mBuffer->release();
+            mBuffer = NULL;
+
+            return ERROR_IO;
+        }
+
+        uint8_t *dstData = (uint8_t *)mBuffer->data();
+        size_t dstOffset = 0;
+        // Add AC-4 sync header to MPEG4 encapsulated AC-4 raw frame
+        // AC40 sync word, meaning no CRC at the end of the frame
+        dstData[dstOffset++] = 0xAC;
+        dstData[dstOffset++] = 0x40;
+        dstData[dstOffset++] = 0xFF;
+        dstData[dstOffset++] = 0xFF;
+        dstData[dstOffset++] = (uint8_t)((size >> 16) & 0xFF);
+        dstData[dstOffset++] = (uint8_t)((size >> 8) & 0xFF);
+        dstData[dstOffset++] = (uint8_t)((size >> 0) & 0xFF);
+
+        ssize_t numBytesRead = mDataSource->readAt(offset, dstData + dstOffset, size);
+        if (numBytesRead != (ssize_t)size) {
+            mBuffer->release();
+            mBuffer = NULL;
+
+            return ERROR_IO;
+        }
+
+        mBuffer->set_range(0, dstOffset + size);
+        mBuffer->meta_data().clear();
+        mBuffer->meta_data().setInt64(
+                kKeyTime, ((int64_t)cts * 1000000) / mTimescale);
+        mBuffer->meta_data().setInt64(
+                kKeyDuration, ((int64_t)stts * 1000000) / mTimescale);
+
+        if (targetSampleTimeUs >= 0) {
+            mBuffer->meta_data().setInt64(
+                    kKeyTargetTime, targetSampleTimeUs);
+        }
+
+        if (isSyncSample) {
+            mBuffer->meta_data().setInt32(kKeyIsSyncFrame, 1);
+        }
+
+        ++mCurrentSampleIndex;
+
+        *out = mBuffer;
+        mBuffer = NULL;
+
+        return OK;
     } else {
         // Whole NAL units are returned but each fragment is prefixed by
         // the start code (0x00 00 00 01).
@@ -5081,9 +5383,13 @@
     uint32_t cts = 0;
     bool isSyncSample = false;
     bool newBuffer = false;
-    if (mBuffer == NULL) {
+    if (mBuffer == NULL || mCurrentSampleIndex >= mCurrentSamples.size()) {
         newBuffer = true;
 
+        if (mBuffer != NULL) {
+            mBuffer->release();
+            mBuffer = NULL;
+        }
         if (mCurrentSampleIndex >= mCurrentSamples.size()) {
             // move to next fragment if there is one
             if (mNextMoofOffset <= mCurrentMoofOffset) {
@@ -5361,6 +5667,8 @@
 
         return OK;
     }
+
+    return OK;
 }
 
 MPEG4Extractor::Track *MPEG4Extractor::findTrackByMimePrefix(
diff --git a/media/extractors/mp4/MPEG4Extractor.h b/media/extractors/mp4/MPEG4Extractor.h
index 3ea0963..a4a5684 100644
--- a/media/extractors/mp4/MPEG4Extractor.h
+++ b/media/extractors/mp4/MPEG4Extractor.h
@@ -139,8 +139,11 @@
 
     Track *findTrackByMimePrefix(const char *mimePrefix);
 
-    status_t parseAC3SampleEntry(off64_t offset);
-    status_t parseAC3SpecificBox(off64_t offset, uint16_t sampleRate);
+    status_t parseChannelCountSampleRate(
+            off64_t *offset, uint16_t *channelCount, uint16_t *sampleRate);
+    status_t parseAC3SpecificBox(off64_t offset);
+    status_t parseEAC3SpecificBox(off64_t offset);
+    status_t parseAC4SpecificBox(off64_t offset);
 
     MPEG4Extractor(const MPEG4Extractor &);
     MPEG4Extractor &operator=(const MPEG4Extractor &);
diff --git a/media/extractors/mp4/SampleIterator.cpp b/media/extractors/mp4/SampleIterator.cpp
index 93ee7c6..1a6d306 100644
--- a/media/extractors/mp4/SampleIterator.cpp
+++ b/media/extractors/mp4/SampleIterator.cpp
@@ -328,7 +328,15 @@
         ++mTimeToSampleIndex;
     }
 
-    *time = mTTSSampleTime + mTTSDuration * (sampleIndex - mTTSSampleIndex);
+    // below is equivalent to:
+    // *time = mTTSSampleTime + mTTSDuration * (sampleIndex - mTTSSampleIndex);
+    uint32_t tmp;
+    if (__builtin_sub_overflow(sampleIndex, mTTSSampleIndex, &tmp) ||
+            __builtin_mul_overflow(mTTSDuration, tmp, &tmp) ||
+            __builtin_add_overflow(mTTSSampleTime, tmp, &tmp)) {
+        return ERROR_OUT_OF_RANGE;
+    }
+    *time = tmp;
 
     int32_t offset = mTable->getCompositionTimeOffset(sampleIndex);
     if ((offset < 0 && *time < (offset == INT32_MIN ?
diff --git a/media/libaaudio/examples/input_monitor/jni/Android.mk b/media/libaaudio/examples/input_monitor/jni/Android.mk
deleted file mode 100644
index a0b981c..0000000
--- a/media/libaaudio/examples/input_monitor/jni/Android.mk
+++ /dev/null
@@ -1,35 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src \
-    frameworks/av/media/libaaudio/examples/utils
-
-# NDK recommends using this kind of relative path instead of an absolute path.
-LOCAL_SRC_FILES:= ../src/input_monitor.cpp
-LOCAL_CFLAGS := -Wall -Werror
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := input_monitor
-include $(BUILD_EXECUTABLE)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/examples/utils
-
-LOCAL_SRC_FILES:= ../src/input_monitor_callback.cpp
-LOCAL_CFLAGS := -Wall -Werror
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := input_monitor_callback
-include $(BUILD_EXECUTABLE)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := libaaudio_prebuilt
-LOCAL_SRC_FILES := libaaudio.so
-LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
-include $(PREBUILT_SHARED_LIBRARY)
diff --git a/media/libaaudio/examples/input_monitor/jni/Application.mk b/media/libaaudio/examples/input_monitor/jni/Application.mk
deleted file mode 100644
index e74475c..0000000
--- a/media/libaaudio/examples/input_monitor/jni/Application.mk
+++ /dev/null
@@ -1,3 +0,0 @@
-# TODO remove then when we support other architectures
-APP_ABI := arm64-v8a
-APP_CPPFLAGS += -std=c++11
diff --git a/media/libaaudio/examples/loopback/jni/Android.mk b/media/libaaudio/examples/loopback/jni/Android.mk
deleted file mode 100644
index aebe877..0000000
--- a/media/libaaudio/examples/loopback/jni/Android.mk
+++ /dev/null
@@ -1,16 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/examples/utils
-
-# NDK recommends using this kind of relative path instead of an absolute path.
-LOCAL_SRC_FILES:= ../src/loopback.cpp
-LOCAL_CFLAGS := -Wall -Werror
-LOCAL_STATIC_LIBRARIES := libsndfile
-LOCAL_SHARED_LIBRARIES := libaaudio libaudioutils
-LOCAL_MODULE := aaudio_loopback
-include $(BUILD_EXECUTABLE)
diff --git a/media/libaaudio/examples/loopback/jni/Application.mk b/media/libaaudio/examples/loopback/jni/Application.mk
deleted file mode 100644
index ba44f37..0000000
--- a/media/libaaudio/examples/loopback/jni/Application.mk
+++ /dev/null
@@ -1 +0,0 @@
-APP_CPPFLAGS += -std=c++11
diff --git a/media/libaaudio/examples/loopback/src/loopback.cpp b/media/libaaudio/examples/loopback/src/loopback.cpp
index 91ebf73..84f9c22 100644
--- a/media/libaaudio/examples/loopback/src/loopback.cpp
+++ b/media/libaaudio/examples/loopback/src/loopback.cpp
@@ -338,7 +338,7 @@
     aaudio_sharing_mode_t requestedInputSharingMode  = AAUDIO_SHARING_MODE_SHARED;
     int                   requestedInputChannelCount = NUM_INPUT_CHANNELS;
     aaudio_format_t       requestedInputFormat       = AAUDIO_FORMAT_UNSPECIFIED;
-    int32_t               requestedInputCapacity     = -1;
+    int32_t               requestedInputCapacity     = AAUDIO_UNSPECIFIED;
     aaudio_performance_mode_t inputPerformanceLevel  = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY;
 
     int32_t               outputFramesPerBurst = 0;
@@ -459,15 +459,8 @@
     argParser.setPerformanceMode(inputPerformanceLevel);
     argParser.setChannelCount(requestedInputChannelCount);
     argParser.setSharingMode(requestedInputSharingMode);
-
-    // Make sure the input buffer has plenty of capacity.
-    // Extra capacity on input should not increase latency if we keep it drained.
-    int32_t inputBufferCapacity = requestedInputCapacity;
-    if (inputBufferCapacity < 0) {
-        int32_t outputBufferCapacity = AAudioStream_getBufferCapacityInFrames(outputStream);
-        inputBufferCapacity = 2 * outputBufferCapacity;
-    }
-    argParser.setBufferCapacity(inputBufferCapacity);
+    // Warning! If you change input capacity then you may not get a FAST track on Legacy path.
+    argParser.setBufferCapacity(requestedInputCapacity);
 
     result = recorder.open(argParser);
     if (result != AAUDIO_OK) {
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/examples/write_sine/jni/Android.mk b/media/libaaudio/examples/write_sine/jni/Android.mk
deleted file mode 100644
index 1a1bd43..0000000
--- a/media/libaaudio/examples/write_sine/jni/Android.mk
+++ /dev/null
@@ -1,35 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/src \
-    frameworks/av/media/libaaudio/examples/utils
-
-# NDK recommends using this kind of relative path instead of an absolute path.
-LOCAL_SRC_FILES:= ../src/write_sine.cpp
-LOCAL_CFLAGS := -Wall -Werror
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := write_sine
-include $(BUILD_EXECUTABLE)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE_TAGS := tests
-LOCAL_C_INCLUDES := \
-    $(call include-path-for, audio-utils) \
-    frameworks/av/media/libaaudio/include \
-    frameworks/av/media/libaaudio/examples/utils
-
-LOCAL_SRC_FILES:= ../src/write_sine_callback.cpp
-LOCAL_CFLAGS := -Wall -Werror
-LOCAL_SHARED_LIBRARIES := libaaudio
-LOCAL_MODULE := write_sine_callback
-include $(BUILD_EXECUTABLE)
-
-include $(CLEAR_VARS)
-LOCAL_MODULE := libaaudio_prebuilt
-LOCAL_SRC_FILES := libaaudio.so
-LOCAL_EXPORT_C_INCLUDES := $(LOCAL_PATH)/include
-include $(PREBUILT_SHARED_LIBRARY)
diff --git a/media/libaaudio/examples/write_sine/jni/Application.mk b/media/libaaudio/examples/write_sine/jni/Application.mk
deleted file mode 100644
index ba44f37..0000000
--- a/media/libaaudio/examples/write_sine/jni/Application.mk
+++ /dev/null
@@ -1 +0,0 @@
-APP_CPPFLAGS += -std=c++11
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/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index b1cb0e7..0641b6e 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -430,14 +430,15 @@
 }
 
 status_t AudioEffect::getEffectDescriptor(const effect_uuid_t *uuid,
-        effect_descriptor_t *descriptor) /*const*/
+                                          const effect_uuid_t *type,
+                                          uint32_t preferredTypeFlag,
+                                          effect_descriptor_t *descriptor)
 {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
     if (af == 0) return PERMISSION_DENIED;
-    return af->getEffectDescriptor(uuid, descriptor);
+    return af->getEffectDescriptor(uuid, type, preferredTypeFlag, descriptor);
 }
 
-
 status_t AudioEffect::queryDefaultPreProcessing(audio_session_t audioSession,
                                           effect_descriptor_t *descriptors,
                                           uint32_t *count)
@@ -446,6 +447,55 @@
     if (aps == 0) return PERMISSION_DENIED;
     return aps->queryDefaultPreProcessing(audioSession, descriptors, count);
 }
+
+status_t AudioEffect::newEffectUniqueId(audio_unique_id_t* id)
+{
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return PERMISSION_DENIED;
+    *id = af->newAudioUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT);
+    return NO_ERROR;
+}
+
+status_t AudioEffect::addStreamDefaultEffect(const char *typeStr,
+                                             const String16& opPackageName,
+                                             const char *uuidStr,
+                                             int32_t priority,
+                                             audio_usage_t usage,
+                                             audio_unique_id_t *id)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+
+    if (typeStr == NULL && uuidStr == NULL) return BAD_VALUE;
+
+    // Convert type & uuid from string to effect_uuid_t.
+    effect_uuid_t type;
+    if (typeStr != NULL) {
+        status_t res = stringToGuid(typeStr, &type);
+        if (res != OK) return res;
+    } else {
+        type = *EFFECT_UUID_NULL;
+    }
+
+    effect_uuid_t uuid;
+    if (uuidStr != NULL) {
+        status_t res = stringToGuid(uuidStr, &uuid);
+        if (res != OK) return res;
+    } else {
+        uuid = *EFFECT_UUID_NULL;
+    }
+
+    return aps->addStreamDefaultEffect(&type, opPackageName, &uuid, priority, usage, id);
+}
+
+status_t AudioEffect::removeStreamDefaultEffect(audio_unique_id_t id)
+{
+    const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
+    if (aps == 0) return PERMISSION_DENIED;
+
+    return aps->removeStreamDefaultEffect(id);
+}
+
 // -------------------------------------------------------------------------
 
 status_t AudioEffect::stringToGuid(const char *str, effect_uuid_t *guid)
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index c072901..e260fd8 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -878,31 +878,25 @@
                                  flags, selectedDeviceId, portId);
 }
 
-status_t AudioSystem::startOutput(audio_io_handle_t output,
-                                  audio_stream_type_t stream,
-                                  audio_session_t session)
+status_t AudioSystem::startOutput(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->startOutput(output, stream, session);
+    return aps->startOutput(portId);
 }
 
-status_t AudioSystem::stopOutput(audio_io_handle_t output,
-                                 audio_stream_type_t stream,
-                                 audio_session_t session)
+status_t AudioSystem::stopOutput(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->stopOutput(output, stream, session);
+    return aps->stopOutput(portId);
 }
 
-void AudioSystem::releaseOutput(audio_io_handle_t output,
-                                audio_stream_type_t stream,
-                                audio_session_t session)
+void AudioSystem::releaseOutput(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return;
-    aps->releaseOutput(output, stream, session);
+    aps->releaseOutput(portId);
 }
 
 status_t AudioSystem::getInputForAttr(const audio_attributes_t *attr,
@@ -1244,18 +1238,18 @@
 
 status_t AudioSystem::startAudioSource(const struct audio_port_config *source,
                                        const audio_attributes_t *attributes,
-                                       audio_patch_handle_t *handle)
+                                       audio_port_handle_t *portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->startAudioSource(source, attributes, handle);
+    return aps->startAudioSource(source, attributes, portId);
 }
 
-status_t AudioSystem::stopAudioSource(audio_patch_handle_t handle)
+status_t AudioSystem::stopAudioSource(audio_port_handle_t portId)
 {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
-    return aps->stopAudioSource(handle);
+    return aps->stopAudioSource(portId);
 }
 
 status_t AudioSystem::setMasterMono(bool mono)
diff --git a/media/libaudioclient/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..00678c2 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 {
@@ -600,14 +598,18 @@
     }
 
     virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
-            effect_descriptor_t *pDescriptor) const
+                                         const effect_uuid_t *pType,
+                                         uint32_t preferredTypeFlag,
+                                         effect_descriptor_t *pDescriptor) const
     {
-        if (pUuid == NULL || pDescriptor == NULL) {
+        if (pUuid == NULL || pType == NULL || pDescriptor == NULL) {
             return BAD_VALUE;
         }
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
         data.write(pUuid, sizeof(effect_uuid_t));
+        data.write(pType, sizeof(effect_uuid_t));
+        data.writeUint32(preferredTypeFlag);
         status_t status = remote()->transact(GET_EFFECT_DESCRIPTOR, data, &reply);
         if (status != NO_ERROR) {
             return status;
@@ -636,10 +638,10 @@
         sp<IEffect> effect;
 
         if (pDesc == NULL) {
-            return effect;
             if (status != NULL) {
                 *status = BAD_VALUE;
             }
+            return effect;
         }
 
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
@@ -912,7 +914,7 @@
         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());
@@ -951,7 +953,8 @@
             break;
     }
 
-    TimeCheck check("IAudioFlinger");
+    std::string tag("IAudioFlinger command " + std::to_string(code));
+    TimeCheck check(tag.c_str());
 
     switch (code) {
         case CREATE_TRACK: {
@@ -1278,8 +1281,11 @@
             CHECK_INTERFACE(IAudioFlinger, data, reply);
             effect_uuid_t uuid;
             data.read(&uuid, sizeof(effect_uuid_t));
+            effect_uuid_t type;
+            data.read(&type, sizeof(effect_uuid_t));
+            uint32_t preferredTypeFlag = data.readUint32();
             effect_descriptor_t desc = {};
-            status_t status = getEffectDescriptor(&uuid, &desc);
+            status_t status = getEffectDescriptor(&uuid, &type, preferredTypeFlag, &desc);
             reply->writeInt32(status);
             if (status == NO_ERROR) {
                 reply->write(&desc, sizeof(effect_descriptor_t));
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index a1236e7..abf74f8 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 {
@@ -82,7 +81,9 @@
     GET_MASTER_MONO,
     GET_STREAM_VOLUME_DB,
     GET_SURROUND_FORMATS,
-    SET_SURROUND_FORMAT_ENABLED
+    SET_SURROUND_FORMAT_ENABLED,
+    ADD_STREAM_DEFAULT_EFFECT,
+    REMOVE_STREAM_DEFAULT_EFFECT
 };
 
 #define MAX_ITEMS_PER_LIST 1024
@@ -245,41 +246,29 @@
             return status;
         }
 
-    virtual status_t startOutput(audio_io_handle_t output,
-                                 audio_stream_type_t stream,
-                                 audio_session_t session)
+    virtual status_t startOutput(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(output);
-        data.writeInt32((int32_t) stream);
-        data.writeInt32((int32_t) session);
+        data.writeInt32((int32_t)portId);
         remote()->transact(START_OUTPUT, data, &reply);
         return static_cast <status_t> (reply.readInt32());
     }
 
-    virtual status_t stopOutput(audio_io_handle_t output,
-                                audio_stream_type_t stream,
-                                audio_session_t session)
+    virtual status_t stopOutput(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(output);
-        data.writeInt32((int32_t) stream);
-        data.writeInt32((int32_t) session);
+        data.writeInt32((int32_t)portId);
         remote()->transact(STOP_OUTPUT, data, &reply);
         return static_cast <status_t> (reply.readInt32());
     }
 
-    virtual void releaseOutput(audio_io_handle_t output,
-                               audio_stream_type_t stream,
-                               audio_session_t session)
+    virtual void releaseOutput(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(output);
-        data.writeInt32((int32_t)stream);
-        data.writeInt32((int32_t)session);
+        data.writeInt32((int32_t)portId);
         remote()->transact(RELEASE_OUTPUT, data, &reply);
     }
 
@@ -753,11 +742,11 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle)
+                                      audio_port_handle_t *portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        if (source == NULL || attributes == NULL || handle == NULL) {
+        if (source == NULL || attributes == NULL || portId == NULL) {
             return BAD_VALUE;
         }
         data.write(source, sizeof(struct audio_port_config));
@@ -770,15 +759,15 @@
         if (status != NO_ERROR) {
             return status;
         }
-        *handle = (audio_patch_handle_t)reply.readInt32();
+        *portId = (audio_port_handle_t)reply.readInt32();
         return status;
     }
 
-    virtual status_t stopAudioSource(audio_patch_handle_t handle)
+    virtual status_t stopAudioSource(audio_port_handle_t portId)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
-        data.writeInt32(handle);
+        data.writeInt32(portId);
         status_t status = remote()->transact(STOP_AUDIO_SOURCE, data, &reply);
         if (status != NO_ERROR) {
             return status;
@@ -879,6 +868,42 @@
         }
         return reply.readInt32();
     }
+
+    virtual status_t addStreamDefaultEffect(const effect_uuid_t *type,
+                                            const String16& opPackageName,
+                                            const effect_uuid_t *uuid,
+                                            int32_t priority,
+                                            audio_usage_t usage,
+                                            audio_unique_id_t* id)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.write(type, sizeof(effect_uuid_t));
+        data.writeString16(opPackageName);
+        data.write(uuid, sizeof(effect_uuid_t));
+        data.writeInt32(priority);
+        data.writeInt32((int32_t) usage);
+        status_t status = remote()->transact(ADD_STREAM_DEFAULT_EFFECT, data, &reply);
+        if (status != NO_ERROR) {
+            return status;
+        }
+        status = static_cast <status_t> (reply.readInt32());
+        *id = reply.readInt32();
+        return status;
+    }
+
+    virtual status_t removeStreamDefaultEffect(audio_unique_id_t id)
+    {
+        Parcel data, reply;
+        data.writeInterfaceToken(IAudioPolicyService::getInterfaceDescriptor());
+        data.writeInt32(id);
+        status_t status = remote()->transact(REMOVE_STREAM_DEFAULT_EFFECT, data, &reply);
+        if (status != NO_ERROR) {
+            return status;
+        }
+        return static_cast <status_t> (reply.readInt32());
+    }
+
 };
 
 IMPLEMENT_META_INTERFACE(AudioPolicyService, "android.media.IAudioPolicyService");
@@ -936,7 +961,7 @@
         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());
@@ -948,7 +973,8 @@
             break;
     }
 
-    TimeCheck check("IAudioPolicyService");
+    std::string tag("IAudioPolicyService command " + std::to_string(code));
+    TimeCheck check(tag.c_str());
 
     switch (code) {
         case SET_DEVICE_CONNECTION_STATE: {
@@ -1075,34 +1101,22 @@
 
         case START_OUTPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
-            audio_stream_type_t stream =
-                                static_cast <audio_stream_type_t>(data.readInt32());
-            audio_session_t session = (audio_session_t)data.readInt32();
-            reply->writeInt32(static_cast <uint32_t>(startOutput(output,
-                                                                 stream,
-                                                                 session)));
+            const audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+            reply->writeInt32(static_cast <uint32_t>(startOutput(portId)));
             return NO_ERROR;
         } break;
 
         case STOP_OUTPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
-            audio_stream_type_t stream =
-                                static_cast <audio_stream_type_t>(data.readInt32());
-            audio_session_t session = (audio_session_t)data.readInt32();
-            reply->writeInt32(static_cast <uint32_t>(stopOutput(output,
-                                                                stream,
-                                                                session)));
+            const audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+            reply->writeInt32(static_cast <uint32_t>(stopOutput(portId)));
             return NO_ERROR;
         } break;
 
         case RELEASE_OUTPUT: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_io_handle_t output = static_cast <audio_io_handle_t>(data.readInt32());
-            audio_stream_type_t stream = (audio_stream_type_t)data.readInt32();
-            audio_session_t session = (audio_session_t)data.readInt32();
-            releaseOutput(output, stream, session);
+            const audio_port_handle_t portId = static_cast <audio_port_handle_t>(data.readInt32());
+            releaseOutput(portId);
             return NO_ERROR;
         } break;
 
@@ -1496,17 +1510,17 @@
             audio_attributes_t attributes = {};
             data.read(&attributes, sizeof(audio_attributes_t));
             sanetizeAudioAttributes(&attributes);
-            audio_patch_handle_t handle = AUDIO_PATCH_HANDLE_NONE;
-            status_t status = startAudioSource(&source, &attributes, &handle);
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+            status_t status = startAudioSource(&source, &attributes, &portId);
             reply->writeInt32(status);
-            reply->writeInt32(handle);
+            reply->writeInt32(portId);
             return NO_ERROR;
         } break;
 
         case STOP_AUDIO_SOURCE: {
             CHECK_INTERFACE(IAudioPolicyService, data, reply);
-            audio_patch_handle_t handle = (audio_patch_handle_t) data.readInt32();
-            status_t status = stopAudioSource(handle);
+            audio_port_handle_t portId = (audio_port_handle_t) data.readInt32();
+            status_t status = stopAudioSource(portId);
             reply->writeInt32(status);
             return NO_ERROR;
         } break;
@@ -1585,6 +1599,43 @@
             return NO_ERROR;
         }
 
+        case ADD_STREAM_DEFAULT_EFFECT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            effect_uuid_t type;
+            status_t status = data.read(&type, sizeof(effect_uuid_t));
+            if (status != NO_ERROR) {
+                return status;
+            }
+            String16 opPackageName;
+            status = data.readString16(&opPackageName);
+            if (status != NO_ERROR) {
+                return status;
+            }
+            effect_uuid_t uuid;
+            status = data.read(&uuid, sizeof(effect_uuid_t));
+            if (status != NO_ERROR) {
+                return status;
+            }
+            int32_t priority = data.readInt32();
+            audio_usage_t usage = (audio_usage_t) data.readInt32();
+            audio_unique_id_t id = 0;
+            reply->writeInt32(static_cast <int32_t>(addStreamDefaultEffect(&type,
+                                                                           opPackageName,
+                                                                           &uuid,
+                                                                           priority,
+                                                                           usage,
+                                                                           &id)));
+            reply->writeInt32(id);
+            return NO_ERROR;
+        }
+
+        case REMOVE_STREAM_DEFAULT_EFFECT: {
+            CHECK_INTERFACE(IAudioPolicyService, data, reply);
+            audio_unique_id_t id = static_cast<audio_unique_id_t>(data.readInt32());
+            reply->writeInt32(static_cast <int32_t>(removeStreamDefaultEffect(id)));
+            return NO_ERROR;
+        }
+
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/media/libaudioclient/include/media/AudioEffect.h b/media/libaudioclient/include/media/AudioEffect.h
index bfc068b..c97f783 100644
--- a/media/libaudioclient/include/media/AudioEffect.h
+++ b/media/libaudioclient/include/media/AudioEffect.h
@@ -90,27 +90,34 @@
      */
     static status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor);
 
-
     /*
-     * Returns the descriptor for the specified effect uuid.
+     * Returns a descriptor for the specified effect uuid or type.
+     *
+     * Lookup an effect by uuid, or if that's unspecified (EFFECT_UUID_NULL),
+     * do so by type and preferred flags instead.
      *
      * Parameters:
      *      uuid:       pointer to effect uuid.
+     *      type:       pointer to effect type uuid.
+     *      preferredTypeFlags: if multiple effects of the given type exist,
+     *                  one with a matching type flag will be chosen over one without.
+     *                  Use EFFECT_FLAG_TYPE_MASK to indicate no preference.
      *      descriptor: address where the effect descriptor should be returned.
      *
      * Returned status (from utils/Errors.h) can be:
      *      NO_ERROR        successful operation.
      *      PERMISSION_DENIED could not get AudioFlinger interface
      *      NO_INIT         effect library failed to initialize
-     *      BAD_VALUE       invalid uuid or descriptor pointers
+     *      BAD_VALUE       invalid type or descriptor pointers
      *      NAME_NOT_FOUND  no effect with this uuid found
      *
      * Returned value
      *   *descriptor updated with effect descriptor
      */
     static status_t getEffectDescriptor(const effect_uuid_t *uuid,
-                                        effect_descriptor_t *descriptor) /*const*/;
-
+                                        const effect_uuid_t *type,
+                                        uint32_t preferredTypeFlag,
+                                        effect_descriptor_t *descriptor);
 
     /*
      * Returns a list of descriptors corresponding to the pre processings enabled by default
@@ -144,6 +151,79 @@
                                               uint32_t *count);
 
     /*
+     * Gets a new system-wide unique effect id.
+     *
+     * Parameters:
+     *      id: The address to return the generated id.
+     *
+     * Returned status (from utils/Errors.h) can be:
+     *      NO_ERROR        successful operation.
+     *      PERMISSION_DENIED could not get AudioFlinger interface
+     *                        or caller lacks required permissions.
+     * Returned value
+     *   *id:  The new unique system-wide effect id.
+     */
+    static status_t newEffectUniqueId(audio_unique_id_t* id);
+
+    /*
+     * Static methods for adding/removing system-wide effects.
+     */
+
+    /*
+     * Adds an effect to the list of default output effects for a given stream type.
+     *
+     * If the effect is no longer available when a stream of the given type
+     * is created, the system will continue without adding it.
+     *
+     * Parameters:
+     *   typeStr:  Type uuid of effect to be a default: can be null if uuidStr is specified.
+     *             This may correspond to the OpenSL ES interface implemented by this effect,
+     *             or could be some vendor-defined type.
+     *   opPackageName: The package name used for app op checks.
+     *   uuidStr:  Uuid of effect to be a default: can be null if type is specified.
+     *             This uuid corresponds to a particular implementation of an effect type.
+     *             Note if both uuidStr and typeStr are specified, typeStr is ignored.
+     *   priority: Requested priority for effect control: the priority level corresponds to the
+     *             value of priority parameter: negative values indicate lower priorities, positive
+     *             values higher priorities, 0 being the normal priority.
+     *   usage:    The usage this effect should be a default for. Unrecognized values will be
+     *             treated as AUDIO_USAGE_UNKNOWN.
+     *   id:       Address where the system-wide unique id of the default effect should be returned.
+     *
+     * Returned status (from utils/Errors.h) can be:
+     *      NO_ERROR        successful operation.
+     *      PERMISSION_DENIED could not get AudioFlinger interface
+     *                        or caller lacks required permissions.
+     *      NO_INIT         effect library failed to initialize.
+     *      BAD_VALUE       invalid type uuid or implementation uuid.
+     *      NAME_NOT_FOUND  no effect with this uuid or type found.
+     *
+     * Returned value
+     *   *id:  The system-wide unique id of the added default effect.
+     */
+    static status_t addStreamDefaultEffect(const char* typeStr,
+                                           const String16& opPackageName,
+                                           const char* uuidStr,
+                                           int32_t priority,
+                                           audio_usage_t usage,
+                                           audio_unique_id_t* id);
+
+    /*
+     * Removes an effect from the list of default output effects for a given stream type.
+     *
+     * Parameters:
+     *      id: The system-wide unique id of the effect that should no longer be a default.
+     *
+     * Returned status (from utils/Errors.h) can be:
+     *      NO_ERROR        successful operation.
+     *      PERMISSION_DENIED could not get AudioFlinger interface
+     *                        or caller lacks required permissions.
+     *      NO_INIT         effect library failed to initialize.
+     *      BAD_VALUE       invalid id.
+     */
+    static status_t removeStreamDefaultEffect(audio_unique_id_t id);
+
+    /*
      * Events used by callback function (effect_callback_t).
      */
     enum event_type {
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/AudioPolicyHelper.h b/media/libaudioclient/include/media/AudioPolicyHelper.h
index 73ee0a7..35d2e85 100644
--- a/media/libaudioclient/include/media/AudioPolicyHelper.h
+++ b/media/libaudioclient/include/media/AudioPolicyHelper.h
@@ -19,6 +19,43 @@
 #include <system/audio.h>
 
 static inline
+audio_stream_type_t audio_usage_to_stream_type(const audio_usage_t usage)
+{
+    switch(usage) {
+        case AUDIO_USAGE_MEDIA:
+        case AUDIO_USAGE_GAME:
+        case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
+        case AUDIO_USAGE_ASSISTANT:
+            return AUDIO_STREAM_MUSIC;
+        case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
+            return AUDIO_STREAM_ACCESSIBILITY;
+        case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
+            return AUDIO_STREAM_SYSTEM;
+        case AUDIO_USAGE_VOICE_COMMUNICATION:
+            return AUDIO_STREAM_VOICE_CALL;
+
+        case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
+            return AUDIO_STREAM_DTMF;
+
+        case AUDIO_USAGE_ALARM:
+            return AUDIO_STREAM_ALARM;
+        case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
+            return AUDIO_STREAM_RING;
+
+        case AUDIO_USAGE_NOTIFICATION:
+        case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
+        case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
+        case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
+        case AUDIO_USAGE_NOTIFICATION_EVENT:
+            return AUDIO_STREAM_NOTIFICATION;
+
+        case AUDIO_USAGE_UNKNOWN:
+        default:
+            return AUDIO_STREAM_MUSIC;
+    }
+}
+
+static inline
 audio_stream_type_t audio_attributes_to_stream_type(const audio_attributes_t *attr)
 {
     // flags to stream type mapping
@@ -30,38 +67,7 @@
     }
 
     // usage to stream type mapping
-    switch (attr->usage) {
-    case AUDIO_USAGE_MEDIA:
-    case AUDIO_USAGE_GAME:
-    case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
-    case AUDIO_USAGE_ASSISTANT:
-        return AUDIO_STREAM_MUSIC;
-    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
-        return AUDIO_STREAM_ACCESSIBILITY;
-    case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
-        return AUDIO_STREAM_SYSTEM;
-    case AUDIO_USAGE_VOICE_COMMUNICATION:
-        return AUDIO_STREAM_VOICE_CALL;
-
-    case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
-        return AUDIO_STREAM_DTMF;
-
-    case AUDIO_USAGE_ALARM:
-        return AUDIO_STREAM_ALARM;
-    case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
-        return AUDIO_STREAM_RING;
-
-    case AUDIO_USAGE_NOTIFICATION:
-    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
-    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
-    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
-    case AUDIO_USAGE_NOTIFICATION_EVENT:
-        return AUDIO_STREAM_NOTIFICATION;
-
-    case AUDIO_USAGE_UNKNOWN:
-    default:
-        return AUDIO_STREAM_MUSIC;
-    }
+    return audio_usage_to_stream_type(attr->usage);
 }
 
 static inline
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 4c0f796..adfee8b 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -224,15 +224,9 @@
                                      audio_output_flags_t flags,
                                      audio_port_handle_t *selectedDeviceId,
                                      audio_port_handle_t *portId);
-    static status_t startOutput(audio_io_handle_t output,
-                                audio_stream_type_t stream,
-                                audio_session_t session);
-    static status_t stopOutput(audio_io_handle_t output,
-                               audio_stream_type_t stream,
-                               audio_session_t session);
-    static void releaseOutput(audio_io_handle_t output,
-                              audio_stream_type_t stream,
-                              audio_session_t session);
+    static status_t startOutput(audio_port_handle_t portId);
+    static status_t stopOutput(audio_port_handle_t portId);
+    static void releaseOutput(audio_port_handle_t portId);
 
     // Client must successfully hand off the handle reference to AudioFlinger via createRecord(),
     // or release it with releaseInput().
@@ -328,9 +322,9 @@
     static status_t registerPolicyMixes(const Vector<AudioMix>& mixes, bool registration);
 
     static status_t startAudioSource(const struct audio_port_config *source,
-                                      const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle);
-    static status_t stopAudioSource(audio_patch_handle_t handle);
+                                     const audio_attributes_t *attributes,
+                                     audio_port_handle_t *portId);
+    static status_t stopAudioSource(audio_port_handle_t portId);
 
     static status_t setMasterMono(bool mono);
     static status_t getMasterMono(bool *mono);
diff --git a/media/libaudioclient/include/media/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/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index e6bf72f..31326ab 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -428,7 +428,9 @@
     virtual status_t queryEffect(uint32_t index, effect_descriptor_t *pDescriptor) const = 0;
 
     virtual status_t getEffectDescriptor(const effect_uuid_t *pEffectUUID,
-                                        effect_descriptor_t *pDescriptor) const = 0;
+                                         const effect_uuid_t *pTypeUUID,
+                                         uint32_t preferredTypeFlag,
+                                         effect_descriptor_t *pDescriptor) const = 0;
 
     virtual sp<IEffect> createEffect(
                                     effect_descriptor_t *pDesc,
diff --git a/media/libaudioclient/include/media/IAudioPolicyService.h b/media/libaudioclient/include/media/IAudioPolicyService.h
index c3876af..c2899f8 100644
--- a/media/libaudioclient/include/media/IAudioPolicyService.h
+++ b/media/libaudioclient/include/media/IAudioPolicyService.h
@@ -66,15 +66,9 @@
                                       audio_output_flags_t flags,
                                       audio_port_handle_t *selectedDeviceId,
                                       audio_port_handle_t *portId) = 0;
-    virtual status_t startOutput(audio_io_handle_t output,
-                                 audio_stream_type_t stream,
-                                 audio_session_t session) = 0;
-    virtual status_t stopOutput(audio_io_handle_t output,
-                                audio_stream_type_t stream,
-                                audio_session_t session) = 0;
-    virtual void releaseOutput(audio_io_handle_t output,
-                               audio_stream_type_t stream,
-                               audio_session_t session) = 0;
+    virtual status_t startOutput(audio_port_handle_t portId) = 0;
+    virtual status_t stopOutput(audio_port_handle_t portId) = 0;
+    virtual void releaseOutput(audio_port_handle_t portId) = 0;
     virtual status_t  getInputForAttr(const audio_attributes_t *attr,
                               audio_io_handle_t *input,
                               audio_session_t session,
@@ -115,6 +109,13 @@
     virtual status_t queryDefaultPreProcessing(audio_session_t audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count) = 0;
+    virtual status_t addStreamDefaultEffect(const effect_uuid_t *type,
+                                            const String16& opPackageName,
+                                            const effect_uuid_t *uuid,
+                                            int32_t priority,
+                                            audio_usage_t usage,
+                                            audio_unique_id_t* id) = 0;
+    virtual status_t removeStreamDefaultEffect(audio_unique_id_t id) = 0;
    // Check if offload is possible for given format, stream type, sample rate,
     // bit rate, duration, video and streaming or offload property is enabled
     virtual bool isOffloadSupported(const audio_offload_info_t& info) = 0;
@@ -159,8 +160,8 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle) = 0;
-    virtual status_t stopAudioSource(audio_patch_handle_t handle) = 0;
+                                      audio_port_handle_t *portId) = 0;
+    virtual status_t stopAudioSource(audio_port_handle_t portId) = 0;
 
     virtual status_t setMasterMono(bool mono) = 0;
     virtual status_t getMasterMono(bool *mono) = 0;
diff --git a/media/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 3c1cb59..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 91%
rename from media/libaudiohal/4.0/StreamHalHidl.cpp
rename to media/libaudiohal/impl/StreamHalHidl.cpp
index 1c2fdb0..bfa80e8 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"),
@@ -188,7 +192,17 @@
                     const native_handle *handle = hidlInfo.sharedMemory.handle();
                     if (handle->numFds > 0) {
                         info->shared_memory_fd = handle->data[0];
+#if MAJOR_VERSION == 4
+                        info->flags = audio_mmap_buffer_flag(hidlInfo.flags);
+#endif
                         info->buffer_size_frames = hidlInfo.bufferSizeFrames;
+                        // Negative buffer size frame was a hack in O and P to
+                        // indicate that the buffer is shareable to applications
+                        if (info->buffer_size_frames < 0) {
+                            info->buffer_size_frames *= -1;
+                            info->flags = audio_mmap_buffer_flag(
+                                    info->flags | AUDIO_MMAP_APPLICATION_SHAREABLE);
+                        }
                         info->burst_size_frames = hidlInfo.burstSizeFrames;
                         // info->shared_memory_address is not needed in HIDL context
                         info->shared_memory_address = NULL;
@@ -566,6 +580,12 @@
     }
 }
 
+#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 +596,7 @@
 }
 
 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 +607,7 @@
               })};
     return processReturn("updateSourceMetadata", mStream->updateSourceMetadata(halMetadata));
 }
+#endif
 
 void StreamOutHalHidl::onWriteReady() {
     sp<StreamOutHalInterfaceCallback> callback = mCallback.promote();
@@ -782,7 +803,19 @@
     }
 }
 
+#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 +835,7 @@
 }
 
 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 +845,7 @@
               })};
     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/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index f6f817a..f3ea826 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -875,7 +875,7 @@
                 t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
                         t->mMixerInFormat, t->mMixerFormat);
                 ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                        "Track %d needs downmix + resample", i);
+                        "Track %d needs downmix + resample", name);
             } else {
                 if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
                     t->hook = Track::getTrackHook(
@@ -890,7 +890,7 @@
                     t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
                             t->mMixerInFormat, t->mMixerFormat);
                     ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                            "Track %d needs downmix", i);
+                            "Track %d needs downmix", name);
                 }
             }
         }
diff --git a/media/libeffects/config/include/media/EffectsConfig.h b/media/libeffects/config/include/media/EffectsConfig.h
index 55b946f..fa0415b 100644
--- a/media/libeffects/config/include/media/EffectsConfig.h
+++ b/media/libeffects/config/include/media/EffectsConfig.h
@@ -96,7 +96,7 @@
     /** Parsed config, nullptr if the xml lib could not load the file */
     std::unique_ptr<Config> parsedConfig;
     size_t nbSkippedElement; //< Number of skipped invalid library, effect or processing chain
-    const char* configPath; //< Path to the loaded configuration
+    const std::string configPath; //< Path to the loaded configuration
 };
 
 /** Parses the provided effect configuration.
diff --git a/media/libeffects/config/src/EffectsConfig.cpp b/media/libeffects/config/src/EffectsConfig.cpp
index d79501f..351b1ee 100644
--- a/media/libeffects/config/src/EffectsConfig.cpp
+++ b/media/libeffects/config/src/EffectsConfig.cpp
@@ -250,14 +250,14 @@
     return true;
 }
 
-/** Internal version of the public parse(const char* path) with precondition `path != nullptr`. */
-ParsingResult parseWithPath(const char* path) {
+/** Internal version of the public parse(const char* path) where path always exist. */
+ParsingResult parseWithPath(std::string&& path) {
     XMLDocument doc;
-    doc.LoadFile(path);
+    doc.LoadFile(path.c_str());
     if (doc.Error()) {
-        ALOGE("Failed to parse %s: Tinyxml2 error (%d): %s", path,
+        ALOGE("Failed to parse %s: Tinyxml2 error (%d): %s", path.c_str(),
               doc.ErrorID(), doc.ErrorStr());
-        return {nullptr, 0, path};
+        return {nullptr, 0, std::move(path)};
     }
 
     auto config = std::make_unique<Config>();
@@ -295,7 +295,7 @@
             }
         }
     }
-    return {std::move(config), nbSkippedElements, path};
+    return {std::move(config), nbSkippedElements, std::move(path)};
 }
 
 }; // namespace
@@ -310,14 +310,14 @@
         if (access(defaultPath.c_str(), R_OK) != 0) {
             continue;
         }
-        auto result = parseWithPath(defaultPath.c_str());
+        auto result = parseWithPath(std::move(defaultPath));
         if (result.parsedConfig != nullptr) {
             return result;
         }
     }
 
     ALOGE("Could not parse effect configuration in any of the default locations.");
-    return {nullptr, 0, nullptr};
+    return {nullptr, 0, ""};
 }
 
 } // namespace effectsConfig
diff --git a/media/libeffects/factory/EffectsXmlConfigLoader.cpp b/media/libeffects/factory/EffectsXmlConfigLoader.cpp
index 7a7d431..052a88b 100644
--- a/media/libeffects/factory/EffectsXmlConfigLoader.cpp
+++ b/media/libeffects/factory/EffectsXmlConfigLoader.cpp
@@ -327,7 +327,8 @@
                                            &gSkippedEffects, &gSubEffectList);
 
     ALOGE_IF(result.nbSkippedElement != 0, "%zu errors during loading of configuration: %s",
-             result.nbSkippedElement, result.configPath ?: "No config file found");
+             result.nbSkippedElement,
+             result.configPath.empty() ? "No config file found" : result.configPath.c_str());
 
     return result.nbSkippedElement;
 }
diff --git a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
index 9d29cf1..d61efd3 100644
--- a/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
+++ b/media/libeffects/loudness/EffectLoudnessEnhancer.cpp
@@ -30,6 +30,26 @@
 #include <audio_effects/effect_loudnessenhancer.h>
 #include "dsp/core/dynamic_range_compression.h"
 
+// BUILD_FLOAT targets building a float effect instead of the legacy int16_t effect.
+#define BUILD_FLOAT
+
+#ifdef BUILD_FLOAT
+
+static constexpr audio_format_t kProcessFormat = AUDIO_FORMAT_PCM_FLOAT;
+
+#else
+
+static constexpr audio_format_t kProcessFormat = AUDIO_FORMAT_PCM_16_BIT;
+
+static inline int16_t clamp16(int32_t sample)
+{
+    if ((sample>>15) ^ (sample>>31))
+        sample = 0x7FFF ^ (sample>>31);
+    return sample;
+}
+
+#endif // BUILD_FLOAT
+
 extern "C" {
 
 // effect_handle_t interface implementation for LE effect
@@ -80,13 +100,6 @@
     }
 }
 
-static inline int16_t clamp16(int32_t sample)
-{
-    if ((sample>>15) ^ (sample>>31))
-        sample = 0x7FFF ^ (sample>>31);
-    return sample;
-}
-
 //----------------------------------------------------------------------------
 // LE_setConfig()
 //----------------------------------------------------------------------------
@@ -111,7 +124,7 @@
     if (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO) return -EINVAL;
     if (pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_WRITE &&
             pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_ACCUMULATE) return -EINVAL;
-    if (pConfig->inputCfg.format != AUDIO_FORMAT_PCM_16_BIT) return -EINVAL;
+    if (pConfig->inputCfg.format != kProcessFormat) return -EINVAL;
 
     pContext->mConfig = *pConfig;
 
@@ -159,7 +172,7 @@
 
     pContext->mConfig.inputCfg.accessMode = EFFECT_BUFFER_ACCESS_READ;
     pContext->mConfig.inputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
-    pContext->mConfig.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+    pContext->mConfig.inputCfg.format = kProcessFormat;
     pContext->mConfig.inputCfg.samplingRate = 44100;
     pContext->mConfig.inputCfg.bufferProvider.getBuffer = NULL;
     pContext->mConfig.inputCfg.bufferProvider.releaseBuffer = NULL;
@@ -167,7 +180,7 @@
     pContext->mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
     pContext->mConfig.outputCfg.accessMode = EFFECT_BUFFER_ACCESS_ACCUMULATE;
     pContext->mConfig.outputCfg.channels = AUDIO_CHANNEL_OUT_STEREO;
-    pContext->mConfig.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
+    pContext->mConfig.outputCfg.format = kProcessFormat;
     pContext->mConfig.outputCfg.samplingRate = 44100;
     pContext->mConfig.outputCfg.bufferProvider.getBuffer = NULL;
     pContext->mConfig.outputCfg.bufferProvider.releaseBuffer = NULL;
@@ -284,18 +297,41 @@
 
     //ALOGV("LE about to process %d samples", inBuffer->frameCount);
     uint16_t inIdx;
+#ifdef BUILD_FLOAT
+    constexpr float scale = 1 << 15; // power of 2 is lossless conversion to int16_t range
+    constexpr float inverseScale = 1.f / scale;
+    const float inputAmp = pow(10, pContext->mTargetGainmB/2000.0f) * scale;
+#else
     float inputAmp = pow(10, pContext->mTargetGainmB/2000.0f);
+#endif
     float leftSample, rightSample;
     for (inIdx = 0 ; inIdx < inBuffer->frameCount ; inIdx++) {
         // makeup gain is applied on the input of the compressor
+#ifdef BUILD_FLOAT
+        leftSample  = inputAmp * inBuffer->f32[2*inIdx];
+        rightSample = inputAmp * inBuffer->f32[2*inIdx +1];
+        pContext->mCompressor->Compress(&leftSample, &rightSample);
+        inBuffer->f32[2*inIdx]    = leftSample * inverseScale;
+        inBuffer->f32[2*inIdx +1] = rightSample * inverseScale;
+#else
         leftSample  = inputAmp * (float)inBuffer->s16[2*inIdx];
         rightSample = inputAmp * (float)inBuffer->s16[2*inIdx +1];
         pContext->mCompressor->Compress(&leftSample, &rightSample);
         inBuffer->s16[2*inIdx]    = (int16_t) leftSample;
         inBuffer->s16[2*inIdx +1] = (int16_t) rightSample;
+#endif // BUILD_FLOAT
     }
 
     if (inBuffer->raw != outBuffer->raw) {
+#ifdef BUILD_FLOAT
+        if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
+            for (size_t i = 0; i < outBuffer->frameCount*2; i++) {
+                outBuffer->f32[i] += inBuffer->f32[i];
+            }
+        } else {
+            memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(float));
+        }
+#else
         if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
             for (size_t i = 0; i < outBuffer->frameCount*2; i++) {
                 outBuffer->s16[i] = clamp16(outBuffer->s16[i] + inBuffer->s16[i]);
@@ -303,6 +339,7 @@
         } else {
             memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(int16_t));
         }
+#endif // BUILD_FLOAT
     }
     if (pContext->mState != LOUDNESS_ENHANCER_STATE_ACTIVE) {
         return -ENODATA;
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 1b3a1be..e6d6b3e 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -20,7 +20,7 @@
     vndk: {
         enabled: true,
     },
-    srcs: ["AudioParameter.cpp", "TypeConverter.cpp", "TimeCheck.cpp"],
+    srcs: ["AudioParameter.cpp", "TypeConverter.cpp"],
     cflags: [
         "-Werror",
         "-Wno-error=deprecated-declarations",
@@ -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/Android.bp b/media/libmediaplayer2/Android.bp
index 1fa8789..0fb5abc 100644
--- a/media/libmediaplayer2/Android.bp
+++ b/media/libmediaplayer2/Android.bp
@@ -9,6 +9,7 @@
 
     srcs: [
         "JAudioTrack.cpp",
+        "JavaVMHelper.cpp",
         "MediaPlayer2AudioOutput.cpp",
         "mediaplayer2.cpp",
     ],
@@ -49,6 +50,10 @@
         "media_plugin_headers",
     ],
 
+    include_dirs: [
+        "frameworks/base/core/jni",
+    ],
+
     static_libs: [
         "libmedia_helper",
         "libstagefright_nuplayer2",
diff --git a/media/libmediaplayer2/JAudioTrack.cpp b/media/libmediaplayer2/JAudioTrack.cpp
index ac0cc57..778ae1b 100644
--- a/media/libmediaplayer2/JAudioTrack.cpp
+++ b/media/libmediaplayer2/JAudioTrack.cpp
@@ -21,7 +21,7 @@
 #include "mediaplayer2/JAudioTrack.h"
 
 #include <android_media_AudioErrors.h>
-#include <android_runtime/AndroidRuntime.h>
+#include <mediaplayer2/JavaVMHelper.h>
 
 namespace android {
 
@@ -39,7 +39,7 @@
         const audio_attributes_t* pAttributes,        // AudioAttributes
         float maxRequiredSpeed) {                     // bufferSizeInBytes
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jclass jAudioTrackCls = env->FindClass("android/media/AudioTrack");
     mAudioTrackCls = (jclass) env->NewGlobalRef(jAudioTrackCls);
 
@@ -116,19 +116,19 @@
 }
 
 JAudioTrack::~JAudioTrack() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     env->DeleteGlobalRef(mAudioTrackCls);
 }
 
 size_t JAudioTrack::frameCount() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetBufferSizeInFrames = env->GetMethodID(
             mAudioTrackCls, "getBufferSizeInFrames", "()I");
     return env->CallIntMethod(mAudioTrackObj, jGetBufferSizeInFrames);
 }
 
 size_t JAudioTrack::channelCount() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetChannelCount = env->GetMethodID(mAudioTrackCls, "getChannelCount", "()I");
     return env->CallIntMethod(mAudioTrackObj, jGetChannelCount);
 }
@@ -143,7 +143,7 @@
         return BAD_VALUE;
     }
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetPlaybackHeadPosition = env->GetMethodID(
             mAudioTrackCls, "getPlaybackHeadPosition", "()I");
     *position = env->CallIntMethod(mAudioTrackObj, jGetPlaybackHeadPosition);
@@ -152,7 +152,7 @@
 }
 
 bool JAudioTrack::getTimestamp(AudioTimestamp& timestamp) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
 
     jclass jAudioTimeStampCls = env->FindClass("android/media/AudioTimestamp");
     jobject jAudioTimeStampObj = env->AllocObject(jAudioTimeStampCls);
@@ -189,7 +189,7 @@
 status_t JAudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) {
     // TODO: existing native AudioTrack returns INVALID_OPERATION on offload/direct/fast tracks.
     // Should we do the same thing?
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
 
     jclass jPlaybackParamsCls = env->FindClass("android/media/PlaybackParams");
     jmethodID jPlaybackParamsCtor = env->GetMethodID(jPlaybackParamsCls, "<init>", "()V");
@@ -224,7 +224,7 @@
 }
 
 const AudioPlaybackRate JAudioTrack::getPlaybackRate() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
 
     jmethodID jGetPlaybackParams = env->GetMethodID(
             mAudioTrackCls, "getPlaybackParams", "()Landroid/media/PlaybackParams;");
@@ -266,7 +266,7 @@
         return media::VolumeShaper::Status(BAD_VALUE);
     }
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
 
     jmethodID jCreateVolumeShaper = env->GetMethodID(mAudioTrackCls, "createVolumeShaper",
             "(Landroid/media/VolumeShaper$Configuration;)Landroid/media/VolumeShaper;");
@@ -282,7 +282,7 @@
 }
 
 status_t JAudioTrack::setAuxEffectSendLevel(float level) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jSetAuxEffectSendLevel = env->GetMethodID(
             mAudioTrackCls, "setAuxEffectSendLevel", "(F)I");
     int result = env->CallIntMethod(mAudioTrackObj, jSetAuxEffectSendLevel, level);
@@ -290,14 +290,14 @@
 }
 
 status_t JAudioTrack::attachAuxEffect(int effectId) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jAttachAuxEffect = env->GetMethodID(mAudioTrackCls, "attachAuxEffect", "(I)I");
     int result = env->CallIntMethod(mAudioTrackObj, jAttachAuxEffect, effectId);
     return javaToNativeStatus(result);
 }
 
 status_t JAudioTrack::setVolume(float left, float right) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     // TODO: Java setStereoVolume is deprecated. Do we really need this method?
     jmethodID jSetStereoVolume = env->GetMethodID(mAudioTrackCls, "setStereoVolume", "(FF)I");
     int result = env->CallIntMethod(mAudioTrackObj, jSetStereoVolume, left, right);
@@ -305,14 +305,14 @@
 }
 
 status_t JAudioTrack::setVolume(float volume) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jSetVolume = env->GetMethodID(mAudioTrackCls, "setVolume", "(F)I");
     int result = env->CallIntMethod(mAudioTrackObj, jSetVolume, volume);
     return javaToNativeStatus(result);
 }
 
 status_t JAudioTrack::start() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jPlay = env->GetMethodID(mAudioTrackCls, "play", "()V");
     // TODO: Should we catch the Java IllegalStateException from play()?
     env->CallVoidMethod(mAudioTrackObj, jPlay);
@@ -324,7 +324,7 @@
         return BAD_VALUE;
     }
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jbyteArray jAudioData = env->NewByteArray(size);
     env->SetByteArrayRegion(jAudioData, 0, size, (jbyte *) buffer);
 
@@ -353,7 +353,7 @@
 }
 
 void JAudioTrack::stop() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jStop = env->GetMethodID(mAudioTrackCls, "stop", "()V");
     env->CallVoidMethod(mAudioTrackObj, jStop);
     // TODO: Should we catch IllegalStateException?
@@ -365,20 +365,20 @@
 }
 
 void JAudioTrack::flush() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jFlush = env->GetMethodID(mAudioTrackCls, "flush", "()V");
     env->CallVoidMethod(mAudioTrackObj, jFlush);
 }
 
 void JAudioTrack::pause() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jPause = env->GetMethodID(mAudioTrackCls, "pause", "()V");
     env->CallVoidMethod(mAudioTrackObj, jPause);
     // TODO: Should we catch IllegalStateException?
 }
 
 bool JAudioTrack::isPlaying() const {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetPlayState = env->GetMethodID(mAudioTrackCls, "getPlayState", "()I");
     int currentPlayState = env->CallIntMethod(mAudioTrackObj, jGetPlayState);
 
@@ -393,7 +393,7 @@
 }
 
 uint32_t JAudioTrack::getSampleRate() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetSampleRate = env->GetMethodID(mAudioTrackCls, "getSampleRate", "()I");
     return env->CallIntMethod(mAudioTrackObj, jGetSampleRate);
 }
@@ -403,7 +403,7 @@
         return BAD_VALUE;
     }
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetBufferSizeInFrames = env->GetMethodID(
             mAudioTrackCls, "getBufferSizeInFrames", "()I");
     int bufferSizeInFrames = env->CallIntMethod(mAudioTrackObj, jGetBufferSizeInFrames);
@@ -417,7 +417,7 @@
 }
 
 audio_format_t JAudioTrack::format() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetAudioFormat = env->GetMethodID(mAudioTrackCls, "getAudioFormat", "()I");
     int javaFormat = env->CallIntMethod(mAudioTrackObj, jGetAudioFormat);
     return audioFormatToNative(javaFormat);
@@ -454,7 +454,7 @@
 }
 
 audio_port_handle_t JAudioTrack::getRoutedDeviceId() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetRoutedDevice = env->GetMethodID(mAudioTrackCls, "getRoutedDevice",
             "()Landroid/media/AudioDeviceInfo;");
     jobject jAudioDeviceInfoObj = env->CallObjectMethod(mAudioTrackObj, jGetRoutedDevice);
@@ -469,14 +469,14 @@
 }
 
 audio_session_t JAudioTrack::getAudioSessionId() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jmethodID jGetAudioSessionId = env->GetMethodID(mAudioTrackCls, "getAudioSessionId", "()I");
     jint sessionId = env->CallIntMethod(mAudioTrackObj, jGetAudioSessionId);
     return (audio_session_t) sessionId;
 }
 
 status_t JAudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jclass jMP2ImplCls = env->FindClass("android/media/MediaPlayer2Impl");
     jmethodID jSetAudioOutputDeviceById = env->GetMethodID(
             jMP2ImplCls, "setAudioOutputDeviceById", "(Landroid/media/AudioTrack;I)Z");
@@ -550,7 +550,7 @@
         return NULL;
     }
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
 
     // Referenced "android_media_VolumeShaper.h".
     jfloatArray xarray = nullptr;
@@ -595,7 +595,7 @@
 jobject JAudioTrack::createVolumeShaperOperationObj(
         const sp<media::VolumeShaper::Operation>& operation) {
 
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
 
     jclass jBuilderCls = env->FindClass("android/media/VolumeShaper$Operation$Builder");
     jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
@@ -647,7 +647,7 @@
 }
 
 jobject JAudioTrack::createStreamEventCallback(callback_t cbf, void* user) {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jclass jCallbackCls = env->FindClass("android/media/MediaPlayer2Impl$StreamEventCallback");
     jmethodID jCallbackCtor = env->GetMethodID(jCallbackCls, "<init>", "(JJJ)V");
     jobject jCallbackObj = env->NewObject(jCallbackCls, jCallbackCtor, this, cbf, user);
@@ -655,7 +655,7 @@
 }
 
 jobject JAudioTrack::createCallbackExecutor() {
-    JNIEnv *env = AndroidRuntime::getJNIEnv();
+    JNIEnv *env = JavaVMHelper::getJNIEnv();
     jclass jExecutorsCls = env->FindClass("java/util/concurrent/Executors");
     jmethodID jNewSingleThreadExecutor = env->GetStaticMethodID(jExecutorsCls,
             "newSingleThreadExecutor", "()Ljava/util/concurrent/ExecutorService;");
diff --git a/media/libmediaplayer2/JavaVMHelper.cpp b/media/libmediaplayer2/JavaVMHelper.cpp
new file mode 100644
index 0000000..90aaa7f
--- /dev/null
+++ b/media/libmediaplayer2/JavaVMHelper.cpp
@@ -0,0 +1,48 @@
+/*
+ * Copyright 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 "JavaVMHelper"
+
+#include "mediaplayer2/JavaVMHelper.h"
+
+#include <media/stagefright/foundation/ADebug.h>
+
+#include <stdlib.h>
+
+namespace android {
+
+// static
+std::atomic<JavaVM *> JavaVMHelper::sJavaVM(NULL);
+
+// static
+JNIEnv *JavaVMHelper::getJNIEnv() {
+    JNIEnv *env;
+    JavaVM *vm = sJavaVM.load();
+    CHECK(vm != NULL);
+
+    if (vm->GetEnv((void **)&env, JNI_VERSION_1_4) != JNI_OK) {
+        return NULL;
+    }
+
+    return env;
+}
+
+// static
+void JavaVMHelper::setJavaVM(JavaVM *vm) {
+    sJavaVM.store(vm);
+}
+
+}  // namespace android
diff --git a/media/libmediaplayer2/include/mediaplayer2/JavaVMHelper.h b/media/libmediaplayer2/include/mediaplayer2/JavaVMHelper.h
new file mode 100644
index 0000000..35091b7
--- /dev/null
+++ b/media/libmediaplayer2/include/mediaplayer2/JavaVMHelper.h
@@ -0,0 +1,40 @@
+/*
+ * Copyright 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 JAVA_VM_HELPER_H_
+
+#define JAVA_VM_HELPER_H_
+
+#include "jni.h"
+
+#include <atomic>
+
+namespace android {
+
+struct JavaVMHelper {
+    static JNIEnv *getJNIEnv();
+    static void setJavaVM(JavaVM *vm);
+
+private:
+    // Once a valid JavaVM has been set, it should never be reset or changed.
+    // However, as it may be accessed from multiple threads, access needs to be
+    // synchronized.
+    static std::atomic<JavaVM *> sJavaVM;
+};
+
+}  // namespace android
+
+#endif  // JAVA_VM_HELPER_H_
diff --git a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
index 3905b55..bc84729 100644
--- a/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
+++ b/media/libmediaplayer2/include/mediaplayer2/MediaPlayer2Types.h
@@ -96,17 +96,17 @@
 enum media2_info_type {
     // 0xx
     MEDIA2_INFO_UNKNOWN = 1,
-    // The player was started because it was used as the next player for another
-    // player, which just completed playback
-    MEDIA2_INFO_STARTED_AS_NEXT = 2,
+    // The player just started the playback of this data source.
+    MEDIA2_INFO_DATA_SOURCE_START = 2,
     // The player just pushed the very first video frame for rendering
     MEDIA2_INFO_VIDEO_RENDERING_START = 3,
     // The player just pushed the very first audio frame for rendering
     MEDIA2_INFO_AUDIO_RENDERING_START = 4,
     // The player just completed the playback of this data source
-    MEDIA2_INFO_PLAYBACK_COMPLETE = 5,
-    // The player just completed the playback of the full play list
-    MEDIA2_INFO_PLAYLIST_END = 6,
+    MEDIA2_INFO_DATA_SOURCE_END = 5,
+    // The player just completed the playback of all data sources.
+    // But this is not visible in native code. Just keep this entry for completeness.
+    MEDIA2_INFO_DATA_SOURCE_LIST_END = 6,
 
     //1xx
     // The player just prepared a data source.
@@ -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/libmediaplayer2/nuplayer2/NuPlayer2.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
index 060b698..c649573 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2.cpp
@@ -2474,8 +2474,8 @@
     if (mDriver != NULL) {
         sp<NuPlayer2Driver> driver = mDriver.promote();
         if (driver != NULL) {
-            notifyListener(previousSrcId, MEDIA2_INFO, MEDIA2_INFO_PLAYBACK_COMPLETE, 0);
-            notifyListener(mSrcId, MEDIA2_INFO, MEDIA2_INFO_STARTED_AS_NEXT, 0);
+            notifyListener(previousSrcId, MEDIA2_INFO, MEDIA2_INFO_DATA_SOURCE_END, 0);
+            notifyListener(mSrcId, MEDIA2_INFO, MEDIA2_INFO_DATA_SOURCE_START, 0);
         }
     }
 
diff --git a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
index 645138a..931b86e 100644
--- a/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
+++ b/media/libmediaplayer2/nuplayer2/NuPlayer2Decoder.cpp
@@ -1088,6 +1088,12 @@
                         static_cast<MediaBufferHolder*>(holder.get())->mediaBuffer() : nullptr;
                 }
                 if (mediaBuf != NULL) {
+                    if (mediaBuf->size() > codecBuffer->capacity()) {
+                        handleError(ERROR_BUFFER_TOO_SMALL);
+                        mDequeuedInputBuffers.push_back(bufferIx);
+                        return false;
+                    }
+
                     codecBuffer->setRange(0, mediaBuf->size());
                     memcpy(codecBuffer->data(), mediaBuf->data(), mediaBuf->size());
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 69cd82e..050e4fb 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -1069,6 +1069,12 @@
                         static_cast<MediaBufferHolder*>(holder.get())->mediaBuffer() : nullptr;
                 }
                 if (mediaBuf != NULL) {
+                    if (mediaBuf->size() > codecBuffer->capacity()) {
+                        handleError(ERROR_BUFFER_TOO_SMALL);
+                        mDequeuedInputBuffers.push_back(bufferIx);
+                        return false;
+                    }
+
                     codecBuffer->setRange(0, mediaBuf->size());
                     memcpy(codecBuffer->data(), mediaBuf->data(), mediaBuf->size());
 
diff --git a/media/libnblog/PerformanceAnalysis.cpp b/media/libnblog/PerformanceAnalysis.cpp
index f09e93d..3418dc0 100644
--- a/media/libnblog/PerformanceAnalysis.cpp
+++ b/media/libnblog/PerformanceAnalysis.cpp
@@ -32,6 +32,7 @@
 #include <sys/prctl.h>
 #include <time.h>
 #include <new>
+#include <audio_utils/LogPlot.h>
 #include <audio_utils/roundup.h>
 #include <media/nblog/NBLog.h>
 #include <media/nblog/PerformanceAnalysis.h>
@@ -208,27 +209,6 @@
     return isOutlier;
 }
 
-static int widthOf(int x) {
-    int width = 0;
-    if (x < 0) {
-        width++;
-        x = x == INT_MIN ? INT_MAX : -x;
-    }
-    // assert (x >= 0)
-    do {
-        ++width;
-        x /= 10;
-    } while (x > 0);
-    return width;
-}
-
-// computes the column width required for a specific histogram value
-inline int numberWidth(double number, int leftPadding) {
-    // Added values account for whitespaces needed around numbers, and for the
-    // dot and decimal digit not accounted for by widthOf
-    return std::max(std::max(widthOf(static_cast<int>(number)) + 3, 2), leftPadding + 1);
-}
-
 // rounds value to precision based on log-distance from mean
 __attribute__((no_sanitize("signed-integer-overflow")))
 inline double logRound(double x, double mean) {
@@ -254,7 +234,7 @@
 // of PerformanceAnalysis
 void PerformanceAnalysis::reportPerformance(String8 *body, int author, log_hash_t hash,
                                             int maxHeight) {
-    if (mHists.empty()) {
+    if (mHists.empty() || body == nullptr) {
         return;
     }
 
@@ -273,69 +253,16 @@
         }
     }
 
-    // underscores and spaces length corresponds to maximum width of histogram
-    static const int kLen = 200;
-    std::string underscores(kLen, '_');
-    std::string spaces(kLen, ' ');
-
-    auto it = buckets.begin();
-    double maxDelta = it->first;
-    int maxCount = it->second;
-    // Compute maximum values
-    while (++it != buckets.end()) {
-        if (it->first > maxDelta) {
-            maxDelta = it->first;
-        }
-        if (it->second > maxCount) {
-            maxCount = it->second;
-        }
-    }
-    int height = log2(maxCount) + 1; // maxCount > 0, safe to call log2
-    const int leftPadding = widthOf(1 << height);
-    const int bucketWidth = numberWidth(maxDelta, leftPadding);
-    int scalingFactor = 1;
-    // scale data if it exceeds maximum height
-    if (height > maxHeight) {
-        scalingFactor = (height + maxHeight) / maxHeight;
-        height /= scalingFactor;
-    }
-    body->appendFormat("\n%*s %3.2f %s", leftPadding + 11,
-            "Occurrences in", (elapsedMs / kMsPerSec), "seconds of audio:");
-    body->appendFormat("\n%*s%d, %lld, %lld\n", leftPadding + 11,
+    static const int SIZE = 128;
+    char title[SIZE];
+    snprintf(title, sizeof(title), "\n%s %3.2f %s\n%s%d, %lld, %lld\n",
+            "Occurrences in", (elapsedMs / kMsPerSec), "seconds of audio:",
             "Thread, hash, starting timestamp: ", author,
-            static_cast<long long int>(hash), static_cast<long long int>(startingTs));
-    // write histogram label line with bucket values
-    body->appendFormat("\n%s", " ");
-    body->appendFormat("%*s", leftPadding, " ");
-    for (auto const &x : buckets) {
-        const int colWidth = numberWidth(x.first, leftPadding);
-        body->appendFormat("%*d", colWidth, x.second);
-    }
-    // write histogram ascii art
-    body->appendFormat("\n%s", " ");
-    for (int row = height * scalingFactor; row >= 0; row -= scalingFactor) {
-        const int value = 1 << row;
-        body->appendFormat("%.*s", leftPadding, spaces.c_str());
-        for (auto const &x : buckets) {
-            const int colWidth = numberWidth(x.first, leftPadding);
-            body->appendFormat("%.*s%s", colWidth - 1,
-                               spaces.c_str(), x.second < value ? " " : "|");
-        }
-        body->appendFormat("\n%s", " ");
-    }
-    // print x-axis
-    const int columns = static_cast<int>(buckets.size());
-    body->appendFormat("%*c", leftPadding, ' ');
-    body->appendFormat("%.*s", (columns + 1) * bucketWidth, underscores.c_str());
-    body->appendFormat("\n%s", " ");
+            static_cast<long long>(hash), static_cast<long long>(startingTs));
+    static const char * const kLabel = "ms";
 
-    // write footer with bucket labels
-    body->appendFormat("%*s", leftPadding, " ");
-    for (auto const &x : buckets) {
-        const int colWidth = numberWidth(x.first, leftPadding);
-        body->appendFormat("%*.*f", colWidth, 1, x.first);
-    }
-    body->appendFormat("%.*s%s", bucketWidth, spaces.c_str(), "ms\n");
+    body->appendFormat("%s",
+            audio_utils_plot_histogram(buckets, title, kLabel, maxHeight).c_str());
 
     // Now report glitches
     body->appendFormat("\ntime elapsed between glitches and glitch timestamps:\n");
diff --git a/media/libnblog/include/media/nblog/PerformanceAnalysis.h b/media/libnblog/include/media/nblog/PerformanceAnalysis.h
index ddfe9d6..56e0ea6 100644
--- a/media/libnblog/include/media/nblog/PerformanceAnalysis.h
+++ b/media/libnblog/include/media/nblog/PerformanceAnalysis.h
@@ -25,6 +25,8 @@
 
 namespace android {
 
+class String8;
+
 namespace ReportPerformance {
 
 class PerformanceAnalysis;
diff --git a/media/libnblog/include/media/nblog/ReportPerformance.h b/media/libnblog/include/media/nblog/ReportPerformance.h
index ec0842f..1b11197 100644
--- a/media/libnblog/include/media/nblog/ReportPerformance.h
+++ b/media/libnblog/include/media/nblog/ReportPerformance.h
@@ -23,9 +23,6 @@
 
 namespace android {
 
-// The String8 class is used by reportPerformance function
-class String8;
-
 namespace ReportPerformance {
 
 constexpr int kMsPerSec = 1000;
diff --git a/media/libstagefright/ACodec.cpp b/media/libstagefright/ACodec.cpp
index 7f39d10..3526047 100644
--- a/media/libstagefright/ACodec.cpp
+++ b/media/libstagefright/ACodec.cpp
@@ -5556,6 +5556,11 @@
             break;
         }
 
+        case kWhatCheckIfStuck: {
+            ALOGV("No-op by default");
+            break;
+        }
+
         default:
             return false;
     }
@@ -7873,6 +7878,18 @@
             break;
         }
 
+        case kWhatCheckIfStuck:
+        {
+            int32_t generation = 0;
+            CHECK(msg->findInt32("generation", &generation));
+            if (generation == mCodec->mStateGeneration) {
+                mCodec->signalError(OMX_ErrorUndefined, TIMED_OUT);
+            }
+
+            handled = true;
+            break;
+        }
+
         default:
             handled = BaseState::onMessageReceived(msg);
             break;
@@ -7884,6 +7901,11 @@
 void ACodec::OutputPortSettingsChangedState::stateEntered() {
     ALOGV("[%s] Now handling output port settings change",
          mCodec->mComponentName.c_str());
+
+    // If we haven't transitioned after 3 seconds, we're probably stuck.
+    sp<AMessage> msg = new AMessage(ACodec::kWhatCheckIfStuck, mCodec);
+    msg->setInt32("generation", mCodec->mStateGeneration);
+    msg->post(3000000);
 }
 
 bool ACodec::OutputPortSettingsChangedState::onOMXFrameRendered(
@@ -8146,6 +8168,11 @@
     ALOGV("[%s] Now Flushing", mCodec->mComponentName.c_str());
 
     mFlushComplete[kPortIndexInput] = mFlushComplete[kPortIndexOutput] = false;
+
+    // If we haven't transitioned after 3 seconds, we're probably stuck.
+    sp<AMessage> msg = new AMessage(ACodec::kWhatCheckIfStuck, mCodec);
+    msg->setInt32("generation", mCodec->mStateGeneration);
+    msg->post(3000000);
 }
 
 bool ACodec::FlushingState::onMessageReceived(const sp<AMessage> &msg) {
@@ -8160,6 +8187,7 @@
                 msg->setInt32("generation", mCodec->mStateGeneration);
                 msg->post(3000000);
             }
+            handled = true;
             break;
         }
 
@@ -8180,6 +8208,18 @@
             break;
         }
 
+        case kWhatCheckIfStuck:
+        {
+            int32_t generation = 0;
+            CHECK(msg->findInt32("generation", &generation));
+            if (generation == mCodec->mStateGeneration) {
+                mCodec->signalError(OMX_ErrorUndefined, TIMED_OUT);
+            }
+
+            handled = true;
+            break;
+        }
+
         default:
             handled = BaseState::onMessageReceived(msg);
             break;
diff --git a/media/libstagefright/ACodecBufferChannel.cpp b/media/libstagefright/ACodecBufferChannel.cpp
index 710ae68..266a240 100644
--- a/media/libstagefright/ACodecBufferChannel.cpp
+++ b/media/libstagefright/ACodecBufferChannel.cpp
@@ -129,6 +129,7 @@
         secureHandle = static_cast<native_handle_t *>(secureData->getDestinationPointer());
     }
     ssize_t result = -1;
+    ssize_t codecDataOffset = 0;
     if (mCrypto != NULL) {
         ICrypto::DestinationBuffer destination;
         if (secure) {
@@ -180,9 +181,16 @@
 
         Status status = Status::OK;
         hidl_string detailedError;
+        ScramblingControl sctrl = ScramblingControl::UNSCRAMBLED;
+
+        if (key != NULL) {
+            sctrl = (ScramblingControl)key[0];
+            // Adjust for the PES offset
+            codecDataOffset = key[2] | (key[3] << 8);
+        }
 
         auto returnVoid = mDescrambler->descramble(
-                key != NULL ? (ScramblingControl)key[0] : ScramblingControl::UNSCRAMBLED,
+                sctrl,
                 hidlSubSamples,
                 srcBuffer,
                 0,
@@ -202,6 +210,11 @@
             return UNKNOWN_ERROR;
         }
 
+        if (result < codecDataOffset) {
+            ALOGD("invalid codec data offset: %zd, result %zd", codecDataOffset, result);
+            return BAD_VALUE;
+        }
+
         ALOGV("descramble succeeded, %zd bytes", result);
 
         if (dstBuffer.type == BufferType::SHARED_MEMORY) {
@@ -210,7 +223,7 @@
         }
     }
 
-    it->mCodecBuffer->setRange(0, result);
+    it->mCodecBuffer->setRange(codecDataOffset, result - codecDataOffset);
 
     // Copy metadata from client to codec buffer.
     it->mCodecBuffer->meta()->clear();
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/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 353e407..f91c543 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -2674,7 +2674,7 @@
             CHECK(msg->senderAwaitsResponse(&replyID));
 
             if (mFlags & kFlagIsAsync) {
-                ALOGE("dequeueOutputBuffer can't be used in async mode");
+                ALOGE("dequeueInputBuffer can't be used in async mode");
                 PostReplyWithError(replyID, INVALID_OPERATION);
                 break;
             }
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/MetaDataUtils.cpp b/media/libstagefright/MetaDataUtils.cpp
index 04f6ade..2475e7b 100644
--- a/media/libstagefright/MetaDataUtils.cpp
+++ b/media/libstagefright/MetaDataUtils.cpp
@@ -16,8 +16,10 @@
 
 //#define LOG_NDEBUG 0
 #define LOG_TAG "MetaDataUtils"
+#include <utils/Log.h>
 
 #include <media/stagefright/foundation/avc_utils.h>
+#include <media/stagefright/foundation/ABitReader.h>
 #include <media/stagefright/foundation/ABuffer.h>
 #include <media/stagefright/MediaDefs.h>
 #include <media/stagefright/MetaDataUtils.h>
@@ -25,6 +27,10 @@
 namespace android {
 
 bool MakeAVCCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size) {
+    if (data == nullptr || size == 0) {
+        return false;
+    }
+
     int32_t width;
     int32_t height;
     int32_t sarWidth;
@@ -46,6 +52,44 @@
     return true;
 }
 
+bool MakeAACCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size) {
+    if (data == nullptr || size < 7) {
+        return false;
+    }
+
+    ABitReader bits(data, size);
+
+    // adts_fixed_header
+
+    if (bits.getBits(12) != 0xfffu) {
+        ALOGE("Wrong atds_fixed_header");
+        return false;
+    }
+
+    bits.skipBits(4);  // ID, layer, protection_absent
+
+    unsigned profile = bits.getBits(2);
+    if (profile == 3u) {
+        ALOGE("profile should not be 3");
+        return false;
+    }
+    unsigned sampling_freq_index = bits.getBits(4);
+    bits.getBits(1);  // private_bit
+    unsigned channel_configuration = bits.getBits(3);
+    if (channel_configuration == 0u) {
+        ALOGE("channel_config should not be 0");
+        return false;
+    }
+
+    if (!MakeAACCodecSpecificData(
+            meta, profile, sampling_freq_index, channel_configuration)) {
+        return false;
+    }
+
+    meta.setInt32(kKeyIsADTS, true);
+    return true;
+}
+
 bool MakeAACCodecSpecificData(
         MetaDataBase &meta,
         unsigned profile, unsigned sampling_freq_index,
diff --git a/media/libstagefright/StagefrightPluginLoader.cpp b/media/libstagefright/StagefrightPluginLoader.cpp
index 519e870..dd5903a 100644
--- a/media/libstagefright/StagefrightPluginLoader.cpp
+++ b/media/libstagefright/StagefrightPluginLoader.cpp
@@ -46,7 +46,7 @@
     }
     mCreateInputSurface = (CodecBase::CreateInputSurfaceFunc)dlsym(
             mLibHandle, "CreateInputSurface");
-    if (mCreateBuilder == nullptr) {
+    if (mCreateInputSurface == nullptr) {
         ALOGD("Failed to find symbol: CreateInputSurface (%s)", dlerror());
     }
 }
diff --git a/media/libstagefright/Utils.cpp b/media/libstagefright/Utils.cpp
index cf5e91e..ada37a6 100644
--- a/media/libstagefright/Utils.cpp
+++ b/media/libstagefright/Utils.cpp
@@ -1577,6 +1577,8 @@
     { MEDIA_MIMETYPE_AUDIO_VORBIS,      AUDIO_FORMAT_VORBIS },
     { MEDIA_MIMETYPE_AUDIO_OPUS,        AUDIO_FORMAT_OPUS},
     { MEDIA_MIMETYPE_AUDIO_AC3,         AUDIO_FORMAT_AC3},
+    { MEDIA_MIMETYPE_AUDIO_EAC3,        AUDIO_FORMAT_E_AC3},
+    { MEDIA_MIMETYPE_AUDIO_AC4,         AUDIO_FORMAT_AC4},
     { MEDIA_MIMETYPE_AUDIO_FLAC,        AUDIO_FORMAT_FLAC},
     { 0, AUDIO_FORMAT_INVALID }
 };
@@ -1867,4 +1869,3 @@
 }
 
 }  // namespace android
-
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/foundation/MediaDefs.cpp b/media/libstagefright/foundation/MediaDefs.cpp
index 1695c75..a32cf08 100644
--- a/media/libstagefright/foundation/MediaDefs.cpp
+++ b/media/libstagefright/foundation/MediaDefs.cpp
@@ -50,6 +50,7 @@
 const char *MEDIA_MIMETYPE_AUDIO_MSGSM = "audio/gsm";
 const char *MEDIA_MIMETYPE_AUDIO_AC3 = "audio/ac3";
 const char *MEDIA_MIMETYPE_AUDIO_EAC3 = "audio/eac3";
+const char *MEDIA_MIMETYPE_AUDIO_AC4 = "audio/ac4";
 const char *MEDIA_MIMETYPE_AUDIO_SCRAMBLED = "audio/scrambled";
 
 const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h b/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
index 25be89f..b165bcb 100644
--- a/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
@@ -52,6 +52,7 @@
 extern const char *MEDIA_MIMETYPE_AUDIO_MSGSM;
 extern const char *MEDIA_MIMETYPE_AUDIO_AC3;
 extern const char *MEDIA_MIMETYPE_AUDIO_EAC3;
+extern const char *MEDIA_MIMETYPE_AUDIO_AC4;
 extern const char *MEDIA_MIMETYPE_AUDIO_SCRAMBLED;
 
 extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
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/ACodec.h b/media/libstagefright/include/media/stagefright/ACodec.h
index 97d15a7..1137cf1 100644
--- a/media/libstagefright/include/media/stagefright/ACodec.h
+++ b/media/libstagefright/include/media/stagefright/ACodec.h
@@ -137,6 +137,7 @@
         kWhatOMXDied                 = 'OMXd',
         kWhatReleaseCodecInstance    = 'relC',
         kWhatForceStateTransition    = 'fstt',
+        kWhatCheckIfStuck            = 'Cstk',
     };
 
     enum {
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/include/media/stagefright/MetaDataUtils.h b/media/libstagefright/include/media/stagefright/MetaDataUtils.h
index d5a8080..4a7107d 100644
--- a/media/libstagefright/include/media/stagefright/MetaDataUtils.h
+++ b/media/libstagefright/include/media/stagefright/MetaDataUtils.h
@@ -24,6 +24,7 @@
 
 struct ABuffer;
 bool MakeAVCCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size);
+bool MakeAACCodecSpecificData(MetaDataBase &meta, const uint8_t *data, size_t size);
 bool MakeAACCodecSpecificData(MetaDataBase &meta, unsigned profile, unsigned sampling_freq_index,
         unsigned channel_configuration);
 
diff --git a/media/libstagefright/mpeg2ts/ATSParser.cpp b/media/libstagefright/mpeg2ts/ATSParser.cpp
index 5cc5093..fb498d4 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.cpp
+++ b/media/libstagefright/mpeg2ts/ATSParser.cpp
@@ -119,6 +119,7 @@
 private:
     struct StreamInfo {
         unsigned mType;
+        unsigned mTypeExt;
         unsigned mPID;
         int32_t mCASystemId;
     };
@@ -145,10 +146,12 @@
     Stream(Program *program,
            unsigned elementaryPID,
            unsigned streamType,
+           unsigned streamTypeExt,
            unsigned PCR_PID,
            int32_t CA_system_ID);
 
     unsigned type() const { return mStreamType; }
+    unsigned typeExt() const { return mStreamTypeExt; }
     unsigned pid() const { return mElementaryPID; }
     void setPID(unsigned pid) { mElementaryPID = pid; }
 
@@ -194,6 +197,7 @@
     Program *mProgram;
     unsigned mElementaryPID;
     unsigned mStreamType;
+    unsigned mStreamTypeExt;
     unsigned mPCR_PID;
     int32_t mExpectedContinuityCounter;
 
@@ -447,7 +451,7 @@
         if (descriptor_length > infoLength) {
             break;
         }
-        if (descriptor_tag == 9 && descriptor_length >= 4) {
+        if (descriptor_tag == DESCRIPTOR_CA && descriptor_length >= 4) {
             found = true;
             caDescriptor->mSystemID = br->getBits(16);
             caDescriptor->mPID = br->getBits(16) & 0x1fff;
@@ -513,37 +517,65 @@
     // infoBytesRemaining is the number of bytes that make up the
     // variable length section of ES_infos. It does not include the
     // final CRC.
-    size_t infoBytesRemaining = section_length - 9 - program_info_length - 4;
+    int32_t infoBytesRemaining = section_length - 9 - program_info_length - 4;
 
     while (infoBytesRemaining >= 5) {
-
-        unsigned streamType = br->getBits(8);
-        ALOGV("    stream_type = 0x%02x", streamType);
-
+        StreamInfo info;
+        info.mType = br->getBits(8);
+        ALOGV("    stream_type = 0x%02x", info.mType);
         MY_LOGV("    reserved = %u", br->getBits(3));
 
-        unsigned elementaryPID = br->getBits(13);
-        ALOGV("    elementary_PID = 0x%04x", elementaryPID);
+        info.mPID = br->getBits(13);
+        ALOGV("    elementary_PID = 0x%04x", info.mPID);
 
         MY_LOGV("    reserved = %u", br->getBits(4));
 
         unsigned ES_info_length = br->getBits(12);
         ALOGV("    ES_info_length = %u", ES_info_length);
+        infoBytesRemaining -= 5 + ES_info_length;
 
         CADescriptor streamCA;
-        bool hasStreamCA = findCADescriptor(br, ES_info_length, &streamCA);
+        info.mTypeExt = EXT_DESCRIPTOR_DVB_RESERVED_MAX;
+        bool hasStreamCA = false;
+        while (ES_info_length > 2 && infoBytesRemaining >= 0) {
+            unsigned descriptor_tag = br->getBits(8);
+            ALOGV("      tag = 0x%02x", descriptor_tag);
+
+            unsigned descriptor_length = br->getBits(8);
+            ALOGV("      len = %u", descriptor_length);
+
+            ES_info_length -= 2;
+            if (descriptor_length > ES_info_length) {
+                return ERROR_MALFORMED;
+            }
+            if (descriptor_tag == DESCRIPTOR_CA && descriptor_length >= 4) {
+                hasStreamCA = true;
+                streamCA.mSystemID = br->getBits(16);
+                streamCA.mPID = br->getBits(16) & 0x1fff;
+                ES_info_length -= 4;
+                streamCA.mPrivateData.assign(br->data(), br->data() + descriptor_length - 4);
+            } else if (info.mType == STREAMTYPE_PES_PRIVATE_DATA &&
+                       descriptor_tag == DESCRIPTOR_DVB_EXTENSION && descriptor_length >= 1) {
+                unsigned descTagExt = br->getBits(8);
+                ALOGV("      tag_ext = 0x%02x", descTagExt);
+                if (descTagExt == EXT_DESCRIPTOR_DVB_AC4) {
+                    info.mTypeExt = EXT_DESCRIPTOR_DVB_AC4;
+                }
+                ES_info_length -= descriptor_length;
+                descriptor_length--;
+                br->skipBits(descriptor_length * 8);
+            } else {
+                ES_info_length -= descriptor_length;
+                br->skipBits(descriptor_length * 8);
+            }
+        }
         if (hasStreamCA && !mParser->mCasManager->addStream(
-                mProgramNumber, elementaryPID, streamCA)) {
+                mProgramNumber, info.mPID, streamCA)) {
             return ERROR_MALFORMED;
         }
-        StreamInfo info;
-        info.mType = streamType;
-        info.mPID = elementaryPID;
         info.mCASystemId = hasProgramCA ? programCA.mSystemID :
                            hasStreamCA ? streamCA.mSystemID  : -1;
         infos.push(info);
-
-        infoBytesRemaining -= 5 + ES_info_length;
     }
 
     if (infoBytesRemaining != 0) {
@@ -602,7 +634,7 @@
 
         if (index < 0) {
             sp<Stream> stream = new Stream(
-                    this, info.mPID, info.mType, PCR_PID, info.mCASystemId);
+                    this, info.mPID, info.mType, info.mTypeExt, PCR_PID, info.mCASystemId);
 
             if (mSampleAesKeyItem != NULL) {
                 stream->signalNewSampleAesKey(mSampleAesKeyItem);
@@ -720,11 +752,13 @@
         Program *program,
         unsigned elementaryPID,
         unsigned streamType,
+        unsigned streamTypeExt,
         unsigned PCR_PID,
         int32_t CA_system_ID)
     : mProgram(program),
       mElementaryPID(elementaryPID),
       mStreamType(streamType),
+      mStreamTypeExt(streamTypeExt),
       mPCR_PID(PCR_PID),
       mExpectedContinuityCounter(-1),
       mPayloadStarted(false),
@@ -741,10 +775,12 @@
     ALOGV("new stream PID 0x%02x, type 0x%02x, scrambled %d, SampleEncrypted: %d",
             elementaryPID, streamType, mScrambled, mSampleEncrypted);
 
-    uint32_t flags =
-            (isVideo() && mScrambled) ? ElementaryStreamQueue::kFlag_ScrambledData :
-            (mSampleEncrypted) ? ElementaryStreamQueue::kFlag_SampleEncryptedData :
-            0;
+    uint32_t flags = 0;
+    if (((isVideo() || isAudio()) && mScrambled)) {
+        flags = ElementaryStreamQueue::kFlag_ScrambledData;
+    } else if (mSampleEncrypted) {
+        flags = ElementaryStreamQueue::kFlag_SampleEncryptedData;
+    }
 
     ElementaryStreamQueue::Mode mode = ElementaryStreamQueue::INVALID;
 
@@ -781,6 +817,16 @@
             mode = ElementaryStreamQueue::AC3;
             break;
 
+        case STREAMTYPE_EAC3:
+            mode = ElementaryStreamQueue::EAC3;
+            break;
+
+        case STREAMTYPE_PES_PRIVATE_DATA:
+            if (mStreamTypeExt == EXT_DESCRIPTOR_DVB_AC4) {
+                mode = ElementaryStreamQueue::AC4;
+            }
+            break;
+
         case STREAMTYPE_METADATA:
             mode = ElementaryStreamQueue::METADATA;
             break;
@@ -986,9 +1032,12 @@
         case STREAMTYPE_MPEG2_AUDIO_ADTS:
         case STREAMTYPE_LPCM_AC3:
         case STREAMTYPE_AC3:
+        case STREAMTYPE_EAC3:
         case STREAMTYPE_AAC_ENCRYPTED:
         case STREAMTYPE_AC3_ENCRYPTED:
             return true;
+        case STREAMTYPE_PES_PRIVATE_DATA:
+            return mStreamTypeExt == EXT_DESCRIPTOR_DVB_AC4;
 
         default:
             return false;
@@ -1395,7 +1444,7 @@
     // Perform the 1st pass descrambling if needed
     if (descrambleBytes > 0) {
         memcpy(mDescrambledBuffer->data(), mBuffer->data(), descrambleBytes);
-        mDescrambledBuffer->setRange(0, descrambleBytes);
+        mDescrambledBuffer->setRange(0, mBuffer->size());
 
         hidl_vec<SubSample> subSamples;
         subSamples.resize(descrambleSubSamples);
@@ -1412,10 +1461,9 @@
             }
         }
 
-        uint64_t srcOffset = 0, dstOffset = 0;
-        // If scrambled at PES-level, PES header should be skipped
+        // If scrambled at PES-level, PES header is in the clear
         if (pesScramblingControl != 0) {
-            srcOffset = dstOffset = pesOffset;
+            subSamples[0].numBytesOfClearData = pesOffset;
             subSamples[0].numBytesOfEncryptedData -= pesOffset;
         }
 
@@ -1431,9 +1479,9 @@
                 (ScramblingControl) sctrl,
                 subSamples,
                 mDescramblerSrcBuffer,
-                srcOffset,
+                0 /*srcOffset*/,
                 dstBuffer,
-                dstOffset,
+                0 /*dstOffset*/,
                 [&status, &bytesWritten, &detailedError] (
                         Status _status, uint32_t _bytesWritten,
                         const hidl_string& _detailedError) {
@@ -1450,9 +1498,21 @@
 
         ALOGV("[stream %d] descramble succeeded, %d bytes",
                 mElementaryPID, bytesWritten);
-        memcpy(mBuffer->data(), mDescrambledBuffer->data(), descrambleBytes);
+
+        // Set descrambleBytes to the returned result.
+        // Note that this might be smaller than the total length of input data.
+        // (eg. when we're descrambling the PES header portion of a secure stream,
+        // the plugin might cut it off right after the PES header.)
+        descrambleBytes = bytesWritten;
     }
 
+    // |buffer| points to the buffer from which we'd parse the PES header.
+    // When the output stream is scrambled, it points to mDescrambledBuffer
+    // (unless all packets in this PES are actually clear, in which case,
+    // it points to mBuffer since we never copied into mDescrambledBuffer).
+    // When the output stream is clear, it points to mBuffer, and we'll
+    // copy all descrambled data back to mBuffer.
+    sp<ABuffer> buffer = mBuffer;
     if (mQueue->isScrambled()) {
         // Queue subSample info for scrambled queue
         sp<ABuffer> clearSizesBuffer = new ABuffer(mSubSamples.size() * 4);
@@ -1464,8 +1524,7 @@
         for (auto it = mSubSamples.begin();
                 it != mSubSamples.end(); it++, i++) {
             if ((it->transport_scrambling_mode == 0
-                    && pesScramblingControl == 0)
-                    || i < descrambleSubSamples) {
+                    && pesScramblingControl == 0)) {
                 clearSizePtr[i] = it->subSampleSize;
                 encSizePtr[i] = 0;
             } else {
@@ -1474,14 +1533,29 @@
             }
             isSync |= it->random_access_indicator;
         }
+
+        // If scrambled at PES-level, PES header is in the clear
+        if (pesScramblingControl != 0) {
+            clearSizePtr[0] = pesOffset;
+            encSizePtr[0] -= pesOffset;
+        }
         // Pass the original TS subsample size now. The PES header adjust
         // will be applied when the scrambled AU is dequeued.
+        // Note that if descrambleBytes is 0, it means this PES contains only
+        // all ts packets, leadingClearBytes is entire buffer size.
         mQueue->appendScrambledData(
-                mBuffer->data(), mBuffer->size(), sctrl,
-                isSync, clearSizesBuffer, encSizesBuffer);
+                mBuffer->data(), mBuffer->size(),
+                (descrambleBytes > 0) ? descrambleBytes : mBuffer->size(),
+                sctrl, isSync, clearSizesBuffer, encSizesBuffer);
+
+        if (descrambleBytes > 0) {
+            buffer = mDescrambledBuffer;
+        }
+    } else {
+        memcpy(mBuffer->data(), mDescrambledBuffer->data(), descrambleBytes);
     }
 
-    ABitReader br(mBuffer->data(), mBuffer->size());
+    ABitReader br(buffer->data(), buffer->size());
     status_t err = parsePES(&br, event);
 
     if (err != OK) {
diff --git a/media/libstagefright/mpeg2ts/ATSParser.h b/media/libstagefright/mpeg2ts/ATSParser.h
index 45ca06b..a31dc46 100644
--- a/media/libstagefright/mpeg2ts/ATSParser.h
+++ b/media/libstagefright/mpeg2ts/ATSParser.h
@@ -142,6 +142,7 @@
         STREAMTYPE_MPEG2_VIDEO          = 0x02,
         STREAMTYPE_MPEG1_AUDIO          = 0x03,
         STREAMTYPE_MPEG2_AUDIO          = 0x04,
+        STREAMTYPE_PES_PRIVATE_DATA     = 0x06,
         STREAMTYPE_MPEG2_AUDIO_ADTS     = 0x0f,
         STREAMTYPE_MPEG4_VIDEO          = 0x10,
         STREAMTYPE_METADATA             = 0x15,
@@ -153,6 +154,7 @@
         // Stream type 0x83 is non-standard,
         // it could be LPCM or TrueHD AC3
         STREAMTYPE_LPCM_AC3             = 0x83,
+        STREAMTYPE_EAC3                 = 0x87,
 
         //Sample Encrypted types
         STREAMTYPE_H264_ENCRYPTED       = 0xDB,
@@ -160,6 +162,20 @@
         STREAMTYPE_AC3_ENCRYPTED        = 0xC1,
     };
 
+    enum {
+        // From ISO/IEC 13818-1: 2007 (E), Table 2-29
+        DESCRIPTOR_CA                   = 0x09,
+
+        // DVB BlueBook A038 Table 12
+        DESCRIPTOR_DVB_EXTENSION        = 0x7F,
+    };
+
+    // DVB BlueBook A038 Table 109
+    enum {
+        EXT_DESCRIPTOR_DVB_AC4              = 0x15,
+        EXT_DESCRIPTOR_DVB_RESERVED_MAX     = 0x7F,
+    };
+
 protected:
     virtual ~ATSParser();
 
diff --git a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
index ece0692..9e154a3 100644
--- a/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
+++ b/media/libstagefright/mpeg2ts/AnotherPacketSource.cpp
@@ -226,6 +226,7 @@
         int32_t cryptoMode;
         if (buffer->meta()->findInt32("cryptoMode", &cryptoMode)) {
             int32_t cryptoKey;
+            int32_t pesOffset;
             sp<ABuffer> clearBytesBuffer, encBytesBuffer;
 
             CHECK(buffer->meta()->findInt32("cryptoKey", &cryptoKey));
@@ -233,6 +234,8 @@
                     && clearBytesBuffer != NULL);
             CHECK(buffer->meta()->findBuffer("encBytes", &encBytesBuffer)
                     && encBytesBuffer != NULL);
+            CHECK(buffer->meta()->findInt32("pesOffset", &pesOffset)
+                    && (pesOffset >= 0) && (pesOffset < 65536));
 
             bufmeta.setInt32(kKeyCryptoMode, cryptoMode);
 
@@ -240,6 +243,11 @@
             bufmeta.setData(kKeyCryptoIV, 0, array, 16);
 
             array[0] = (uint8_t) (cryptoKey & 0xff);
+            // array[1] contains PES header flag, which we don't use.
+            // array[2~3] contain the PES offset.
+            array[2] = (uint8_t) (pesOffset & 0xff);
+            array[3] = (uint8_t) ((pesOffset >> 8) & 0xff);
+
             bufmeta.setData(kKeyCryptoKey, 0, array, 16);
 
             bufmeta.setData(kKeyPlainSizes, 0,
diff --git a/media/libstagefright/mpeg2ts/ESQueue.cpp b/media/libstagefright/mpeg2ts/ESQueue.cpp
index 0fa9fcb..fb8b9fd 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.cpp
+++ b/media/libstagefright/mpeg2ts/ESQueue.cpp
@@ -86,6 +86,21 @@
     mCasSessionId = sessionId;
 }
 
+static int32_t readVariableBits(ABitReader &bits, int32_t nbits) {
+    int32_t value = 0;
+    int32_t more_bits = 1;
+
+    while (more_bits) {
+        value += bits.getBits(nbits);
+        more_bits = bits.getBits(1);
+        if (!more_bits)
+            break;
+        value++;
+        value <<= nbits;
+    }
+    return value;
+}
+
 // Parse AC3 header assuming the current ptr is start position of syncframe,
 // update metadata only applicable, and return the payload size
 static unsigned parseAC3SyncFrame(
@@ -195,8 +210,153 @@
     return payloadSize;
 }
 
-static bool IsSeeminglyValidAC3Header(const uint8_t *ptr, size_t size) {
-    return parseAC3SyncFrame(ptr, size, NULL) > 0;
+// Parse EAC3 header assuming the current ptr is start position of syncframe,
+// update metadata only applicable, and return the payload size
+// ATSC A/52:2012 E2.3.1
+static unsigned parseEAC3SyncFrame(
+    const uint8_t *ptr, size_t size, sp<MetaData> *metaData) {
+    static const unsigned channelCountTable[] = {2, 1, 2, 3, 3, 4, 4, 5};
+    static const unsigned samplingRateTable[] = {48000, 44100, 32000};
+    static const unsigned samplingRateTable2[] = {24000, 22050, 16000};
+
+    ABitReader bits(ptr, size);
+    if (bits.numBitsLeft() < 16) {
+        ALOGE("Not enough bits left for further parsing");
+        return 0;
+    }
+    if (bits.getBits(16) != 0x0B77) {
+        ALOGE("No valid sync word in EAC3 header");
+        return 0;
+    }
+
+    // we parse up to bsid so there needs to be at least that many bits
+    if (bits.numBitsLeft() < 2 + 3 + 11 + 2 + 2 + 3 + 1 + 5) {
+        ALOGE("Not enough bits left for further parsing");
+        return 0;
+    }
+
+    unsigned strmtyp = bits.getBits(2);
+    if (strmtyp == 3) {
+        ALOGE("Incorrect strmtyp in EAC3 header");
+        return 0;
+    }
+
+    unsigned substreamid = bits.getBits(3);
+    // only the first independent stream is supported
+    if ((strmtyp == 0 || strmtyp == 2) && substreamid != 0)
+        return 0;
+
+    unsigned frmsiz = bits.getBits(11);
+    unsigned fscod = bits.getBits(2);
+
+    unsigned samplingRate = 0;
+    if (fscod == 0x3) {
+        unsigned fscod2 = bits.getBits(2);
+        if (fscod2 == 3) {
+            ALOGW("Incorrect fscod2 in EAC3 header");
+            return 0;
+        }
+        samplingRate = samplingRateTable2[fscod2];
+    } else {
+        samplingRate = samplingRateTable[fscod];
+        unsigned numblkscod __unused = bits.getBits(2);
+    }
+
+    unsigned acmod = bits.getBits(3);
+    unsigned lfeon = bits.getBits(1);
+    unsigned bsid = bits.getBits(5);
+    if (bsid < 11 || bsid > 16) {
+        ALOGW("Incorrect bsid in EAC3 header. Could be AC-3 or some unknown EAC3 format");
+        return 0;
+    }
+
+    // we currently only support the first independant stream
+    if (metaData != NULL && (strmtyp == 0 || strmtyp == 2)) {
+        unsigned channelCount = channelCountTable[acmod] + lfeon;
+        ALOGV("EAC3 channelCount = %d", channelCount);
+        ALOGV("EAC3 samplingRate = %d", samplingRate);
+        (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_EAC3);
+        (*metaData)->setInt32(kKeyChannelCount, channelCount);
+        (*metaData)->setInt32(kKeySampleRate, samplingRate);
+        (*metaData)->setInt32(kKeyIsSyncFrame, 1);
+    }
+
+    unsigned payloadSize = frmsiz + 1;
+    payloadSize <<= 1;  // convert from 16-bit words to bytes
+
+    return payloadSize;
+}
+
+// Parse AC4 header assuming the current ptr is start position of syncframe
+// and update frameSize and metadata.
+static status_t parseAC4SyncFrame(
+        const uint8_t *ptr, size_t size, unsigned &frameSize, sp<MetaData> *metaData) {
+    // ETSI TS 103 190-2 V1.1.1 (2015-09), Annex C
+    // The sync_word can be either 0xAC40 or 0xAC41.
+    static const int kSyncWordAC40 = 0xAC40;
+    static const int kSyncWordAC41 = 0xAC41;
+
+    size_t headerSize = 0;
+    ABitReader bits(ptr, size);
+    int32_t syncWord = bits.getBits(16);
+    if ((syncWord != kSyncWordAC40) && (syncWord != kSyncWordAC41)) {
+        ALOGE("Invalid syncword in AC4 header");
+        return ERROR_MALFORMED;
+    }
+    headerSize += 2;
+
+    frameSize = bits.getBits(16);
+    headerSize += 2;
+    if (frameSize == 0xFFFF) {
+        frameSize = bits.getBits(24);
+        headerSize += 3;
+    }
+
+    if (frameSize == 0) {
+        ALOGE("Invalid frame size in AC4 header");
+        return ERROR_MALFORMED;
+    }
+    frameSize += headerSize;
+    // If the sync_word is 0xAC41, a crc_word is also transmitted.
+    if (syncWord == kSyncWordAC41) {
+        frameSize += 2; // crc_word
+    }
+    ALOGV("AC4 frameSize = %u", frameSize);
+
+    // ETSI TS 103 190-2 V1.1.1 6.2.1.1
+    uint32_t bitstreamVersion = bits.getBits(2);
+    if (bitstreamVersion == 3) {
+        bitstreamVersion += readVariableBits(bits, 2);
+    }
+
+    bits.skipBits(10); // Sequence Counter
+
+    uint32_t bWaitFrames = bits.getBits(1);
+    if (bWaitFrames) {
+        uint32_t waitFrames = bits.getBits(3);
+        if (waitFrames > 0) {
+            bits.skipBits(2); // br_code;
+        }
+    }
+
+    // ETSI TS 103 190 V1.1.1 Table 82
+    bool fsIndex = bits.getBits(1);
+    uint32_t samplingRate = fsIndex ? 48000 : 44100;
+
+    if (metaData != NULL) {
+        ALOGV("dequeueAccessUnitAC4 Setting mFormat");
+        (*metaData)->setCString(kKeyMIMEType, MEDIA_MIMETYPE_AUDIO_AC4);
+        (*metaData)->setInt32(kKeyIsSyncFrame, 1);
+        // [FIXME] AC4 channel count is defined per presentation. Provide a default channel count
+        // as stereo for the entire stream.
+        (*metaData)->setInt32(kKeyChannelCount, 2);
+        (*metaData)->setInt32(kKeySampleRate, samplingRate);
+    }
+    return OK;
+}
+
+static status_t IsSeeminglyValidAC4Header(const uint8_t *ptr, size_t size, unsigned &frameSize) {
+    return parseAC4SyncFrame(ptr, size, frameSize, NULL);
 }
 
 static bool IsSeeminglyValidADTSHeader(
@@ -279,7 +439,8 @@
         ALOGE("appending data after EOS");
         return ERROR_MALFORMED;
     }
-    if (mBuffer == NULL || mBuffer->size() == 0) {
+
+    if (!isScrambled() && (mBuffer == NULL || mBuffer->size() == 0)) {
         switch (mMode) {
             case H264:
             case MPEG_VIDEO:
@@ -390,12 +551,19 @@
             }
 
             case AC3:
+            case EAC3:
             {
                 uint8_t *ptr = (uint8_t *)data;
 
                 ssize_t startOffset = -1;
                 for (size_t i = 0; i < size; ++i) {
-                    if (IsSeeminglyValidAC3Header(&ptr[i], size - i)) {
+                    unsigned payloadSize = 0;
+                    if (mMode == AC3) {
+                        payloadSize = parseAC3SyncFrame(&ptr[i], size - i, NULL);
+                    } else if (mMode == EAC3) {
+                        payloadSize = parseEAC3SyncFrame(&ptr[i], size - i, NULL);
+                    }
+                    if (payloadSize > 0) {
                         startOffset = i;
                         break;
                     }
@@ -406,7 +574,7 @@
                 }
 
                 if (startOffset > 0) {
-                    ALOGI("found something resembling an AC3 syncword at "
+                    ALOGI("found something resembling an (E)AC3 syncword at "
                           "offset %zd",
                           startOffset);
                 }
@@ -416,6 +584,43 @@
                 break;
             }
 
+            case AC4:
+            {
+                uint8_t *ptr = (uint8_t *)data;
+                unsigned frameSize = 0;
+                ssize_t startOffset = -1;
+
+                // A valid AC4 stream should have minimum of 7 bytes in its buffer.
+                // (Sync header 4 bytes + AC4 toc 3 bytes)
+                if (size < 7) {
+                    return ERROR_MALFORMED;
+                }
+                for (size_t i = 0; i < size; ++i) {
+                    if (IsSeeminglyValidAC4Header(&ptr[i], size - i, frameSize) == OK) {
+                        startOffset = i;
+                        break;
+                    }
+                }
+
+                if (startOffset < 0) {
+                    return ERROR_MALFORMED;
+                }
+
+                if (startOffset > 0) {
+                    ALOGI("found something resembling an AC4 syncword at "
+                          "offset %zd",
+                          startOffset);
+                }
+                if (frameSize != size - startOffset) {
+                    ALOGV("AC4 frame size is %u bytes, while the buffer size is %zd bytes.",
+                          frameSize, size - startOffset);
+                }
+
+                data = &ptr[startOffset];
+                size -= startOffset;
+                break;
+            }
+
             case MPEG_AUDIO:
             {
                 uint8_t *ptr = (uint8_t *)data;
@@ -494,6 +699,7 @@
 
 void ElementaryStreamQueue::appendScrambledData(
         const void *data, size_t size,
+        size_t leadingClearBytes,
         int32_t keyId, bool isSync,
         sp<ABuffer> clearSizes, sp<ABuffer> encSizes) {
     if (!isScrambled()) {
@@ -521,6 +727,7 @@
 
     ScrambledRangeInfo scrambledInfo;
     scrambledInfo.mLength = size;
+    scrambledInfo.mLeadingClearBytes = leadingClearBytes;
     scrambledInfo.mKeyId = keyId;
     scrambledInfo.mIsSync = isSync;
     scrambledInfo.mClearSizes = clearSizes;
@@ -533,7 +740,6 @@
 
 sp<ABuffer> ElementaryStreamQueue::dequeueScrambledAccessUnit() {
     size_t nextScan = mBuffer->size();
-    mBuffer->setRange(0, 0);
     int32_t pesOffset = 0, pesScramblingControl = 0;
     int64_t timeUs = fetchTimestamp(nextScan, &pesOffset, &pesScramblingControl);
     if (timeUs < 0ll) {
@@ -544,6 +750,7 @@
     // return scrambled unit
     int32_t keyId = pesScramblingControl, isSync = 0, scrambledLength = 0;
     sp<ABuffer> clearSizes, encSizes;
+    size_t leadingClearBytes;
     while (mScrambledRangeInfos.size() > mRangeInfos.size()) {
         auto it = mScrambledRangeInfos.begin();
         ALOGV("[stream %d] fetching scrambled range: size=%zu", mMode, it->mLength);
@@ -561,6 +768,7 @@
         clearSizes = it->mClearSizes;
         encSizes = it->mEncSizes;
         isSync = it->mIsSync;
+        leadingClearBytes = it->mLeadingClearBytes;
         mScrambledRangeInfos.erase(it);
     }
     if (scrambledLength == 0) {
@@ -568,26 +776,74 @@
         return NULL;
     }
 
-    // skip the PES header, and copy the rest into scrambled access unit
-    sp<ABuffer> scrambledAccessUnit = ABuffer::CreateAsCopy(
-            mScrambledBuffer->data() + pesOffset,
-            scrambledLength - pesOffset);
-
-    // fix up first sample size after skipping the PES header
-    if (pesOffset > 0) {
-        int32_t &firstClearSize = *(int32_t*)clearSizes->data();
-        int32_t &firstEncSize = *(int32_t*)encSizes->data();
-        // Cut away the PES header
-        if (firstClearSize >= pesOffset) {
-            // This is for TS-level scrambling, we descrambled the first
-            // (or it was clear to begin with)
-            firstClearSize -= pesOffset;
-        } else if (firstEncSize >= pesOffset) {
-            // This can only be PES-level scrambling
-            firstEncSize -= pesOffset;
-        }
+    // Retrieve the leading clear bytes info, and use it to set the clear
+    // range on mBuffer. Note that the leading clear bytes includes the
+    // PES header portion, while mBuffer doesn't.
+    if ((int32_t)leadingClearBytes > pesOffset) {
+        mBuffer->setRange(0, leadingClearBytes - pesOffset);
+    } else {
+        mBuffer->setRange(0, 0);
     }
 
+    // Try to parse formats, and if unavailable set up a dummy format.
+    // Only support the following modes for scrambled content for now.
+    // (will be expanded later).
+    if (mFormat == NULL) {
+        mFormat = new MetaData;
+        switch (mMode) {
+            case H264:
+            {
+                if (!MakeAVCCodecSpecificData(
+                        *mFormat, mBuffer->data(), mBuffer->size())) {
+                    ALOGI("Creating dummy AVC format for scrambled content");
+
+                    mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
+                    mFormat->setInt32(kKeyWidth, 1280);
+                    mFormat->setInt32(kKeyHeight, 720);
+                }
+                break;
+            }
+            case AAC:
+            {
+                if (!MakeAACCodecSpecificData(
+                        *mFormat, mBuffer->data(), mBuffer->size())) {
+                    ALOGI("Creating dummy AAC format for scrambled content");
+
+                    MakeAACCodecSpecificData(*mFormat,
+                            1 /*profile*/, 7 /*sampling_freq_index*/, 1 /*channel_config*/);
+                    mFormat->setInt32(kKeyIsADTS, true);
+                }
+
+                break;
+            }
+            case MPEG_VIDEO:
+            {
+                ALOGI("Creating dummy MPEG format for scrambled content");
+
+                mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG2);
+                mFormat->setInt32(kKeyWidth, 1280);
+                mFormat->setInt32(kKeyHeight, 720);
+                break;
+            }
+            default:
+            {
+                ALOGE("Unknown mode for scrambled content");
+                return NULL;
+            }
+        }
+
+        // for MediaExtractor.CasInfo
+        mFormat->setInt32(kKeyCASystemID, mCASystemId);
+        mFormat->setData(kKeyCASessionID,
+                0, mCasSessionId.data(), mCasSessionId.size());
+    }
+
+    mBuffer->setRange(0, 0);
+
+    // copy into scrambled access unit
+    sp<ABuffer> scrambledAccessUnit = ABuffer::CreateAsCopy(
+            mScrambledBuffer->data(), scrambledLength);
+
     scrambledAccessUnit->meta()->setInt64("timeUs", timeUs);
     if (isSync) {
         scrambledAccessUnit->meta()->setInt32("isSync", 1);
@@ -600,6 +856,7 @@
     scrambledAccessUnit->meta()->setInt32("cryptoKey", keyId);
     scrambledAccessUnit->meta()->setBuffer("clearBytes", clearSizes);
     scrambledAccessUnit->meta()->setBuffer("encBytes", encSizes);
+    scrambledAccessUnit->meta()->setInt32("pesOffset", pesOffset);
 
     memmove(mScrambledBuffer->data(),
             mScrambledBuffer->data() + scrambledLength,
@@ -614,7 +871,11 @@
 }
 
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnit() {
-    if ((mFlags & kFlag_AlignedData) && mMode == H264 && !isScrambled()) {
+    if (isScrambled()) {
+        return dequeueScrambledAccessUnit();
+    }
+
+    if ((mFlags & kFlag_AlignedData) && mMode == H264) {
         if (mRangeInfos.empty()) {
             return NULL;
         }
@@ -648,7 +909,10 @@
         case AAC:
             return dequeueAccessUnitAAC();
         case AC3:
-            return dequeueAccessUnitAC3();
+        case EAC3:
+            return dequeueAccessUnitEAC3();
+        case AC4:
+            return dequeueAccessUnitAC4();
         case MPEG_VIDEO:
             return dequeueAccessUnitMPEGVideo();
         case MPEG4_VIDEO:
@@ -666,34 +930,38 @@
     }
 }
 
-sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC3() {
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitEAC3() {
     unsigned syncStartPos = 0;  // in bytes
     unsigned payloadSize = 0;
     sp<MetaData> format = new MetaData;
 
-    ALOGV("dequeueAccessUnit_AC3[%d]: mBuffer %p(%zu)", mAUIndex, mBuffer->data(), mBuffer->size());
+    ALOGV("dequeueAccessUnitEAC3[%d]: mBuffer %p(%zu)", mAUIndex,
+            mBuffer->data(), mBuffer->size());
 
     while (true) {
         if (syncStartPos + 2 >= mBuffer->size()) {
             return NULL;
         }
 
-        payloadSize = parseAC3SyncFrame(
-                mBuffer->data() + syncStartPos,
-                mBuffer->size() - syncStartPos,
-                &format);
+        uint8_t *ptr = mBuffer->data() + syncStartPos;
+        size_t size = mBuffer->size() - syncStartPos;
+        if (mMode == AC3) {
+            payloadSize = parseAC3SyncFrame(ptr, size, &format);
+        } else if (mMode == EAC3) {
+            payloadSize = parseEAC3SyncFrame(ptr, size, &format);
+        }
         if (payloadSize > 0) {
             break;
         }
 
-        ALOGV("dequeueAccessUnit_AC3[%d]: syncStartPos %u payloadSize %u",
+        ALOGV("dequeueAccessUnitEAC3[%d]: syncStartPos %u payloadSize %u",
                 mAUIndex, syncStartPos, payloadSize);
 
         ++syncStartPos;
     }
 
     if (mBuffer->size() < syncStartPos + payloadSize) {
-        ALOGV("Not enough buffer size for AC3");
+        ALOGV("Not enough buffer size for E/AC3");
         return NULL;
     }
 
@@ -701,7 +969,6 @@
         mFormat = format;
     }
 
-
     int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
     if (timeUs < 0ll) {
         ALOGE("negative timeUs");
@@ -710,7 +977,12 @@
 
     // Not decrypting if key info not available (e.g., scanner/extractor parsing ts files)
     if (mSampleDecryptor != NULL) {
-        mSampleDecryptor->processAC3(mBuffer->data() + syncStartPos, payloadSize);
+        if (mMode == AC3) {
+            mSampleDecryptor->processAC3(mBuffer->data() + syncStartPos, payloadSize);
+        } else if (mMode == EAC3) {
+            ALOGE("EAC3 AU is encrypted and decryption is not supported");
+            return NULL;
+        }
     }
     mAUIndex++;
 
@@ -730,6 +1002,69 @@
     return accessUnit;
 }
 
+sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitAC4() {
+    unsigned syncStartPos = 0;
+    unsigned payloadSize = 0;
+    sp<MetaData> format = new MetaData;
+    ALOGV("dequeueAccessUnit_AC4[%d]: mBuffer %p(%zu)", mAUIndex, mBuffer->data(), mBuffer->size());
+
+    // A valid AC4 stream should have minimum of 7 bytes in its buffer.
+    // (Sync header 4 bytes + AC4 toc 3 bytes)
+    if (mBuffer->size() < 7) {
+        return NULL;
+    }
+
+    while (true) {
+        if (syncStartPos + 2 >= mBuffer->size()) {
+            return NULL;
+        }
+
+        status_t status = parseAC4SyncFrame(
+                    mBuffer->data() + syncStartPos,
+                    mBuffer->size() - syncStartPos,
+                    payloadSize,
+                    &format);
+        if (status == OK) {
+            break;
+        }
+
+        ALOGV("dequeueAccessUnit_AC4[%d]: syncStartPos %u payloadSize %u",
+                mAUIndex, syncStartPos, payloadSize);
+
+        ++syncStartPos;
+    }
+
+    if (mBuffer->size() < syncStartPos + payloadSize) {
+        ALOGV("Not enough buffer size for AC4");
+        return NULL;
+    }
+
+    if (mFormat == NULL) {
+        mFormat = format;
+    }
+
+    int64_t timeUs = fetchTimestamp(syncStartPos + payloadSize);
+    if (timeUs < 0ll) {
+        ALOGE("negative timeUs");
+        return NULL;
+    }
+    mAUIndex++;
+
+    sp<ABuffer> accessUnit = new ABuffer(syncStartPos + payloadSize);
+    memcpy(accessUnit->data(), mBuffer->data(), syncStartPos + payloadSize);
+
+    accessUnit->meta()->setInt64("timeUs", timeUs);
+    accessUnit->meta()->setInt32("isSync", 1);
+
+    memmove(
+            mBuffer->data(),
+            mBuffer->data() + syncStartPos + payloadSize,
+            mBuffer->size() - syncStartPos - payloadSize);
+
+    mBuffer->setRange(0, mBuffer->size() - syncStartPos - payloadSize);
+    return accessUnit;
+}
+
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitPCMAudio() {
     if (mBuffer->size() < 4) {
         return NULL;
@@ -851,25 +1186,11 @@
         bool protection_absent = bits.getBits(1) != 0;
 
         if (mFormat == NULL) {
-            unsigned profile = bits.getBits(2);
-            if (profile == 3u) {
-                ALOGE("profile should not be 3");
-                return NULL;
-            }
-            unsigned sampling_freq_index = bits.getBits(4);
-            bits.getBits(1);  // private_bit
-            unsigned channel_configuration = bits.getBits(3);
-            if (channel_configuration == 0u) {
-                ALOGE("channel_config should not be 0");
-                return NULL;
-            }
-            bits.skipBits(2);  // original_copy, home
-
             mFormat = new MetaData;
-            MakeAACCodecSpecificData(*mFormat,
-                    profile, sampling_freq_index, channel_configuration);
-
-            mFormat->setInt32(kKeyIsADTS, true);
+            if (!MakeAACCodecSpecificData(
+                    *mFormat, mBuffer->data() + offset, mBuffer->size() - offset)) {
+                return NULL;
+            }
 
             int32_t sampleRate;
             int32_t numChannels;
@@ -884,12 +1205,12 @@
 
             ALOGI("found AAC codec config (%d Hz, %d channels)",
                  sampleRate, numChannels);
-        } else {
-            // profile_ObjectType, sampling_frequency_index, private_bits,
-            // channel_configuration, original_copy, home
-            bits.skipBits(12);
         }
 
+        // profile_ObjectType, sampling_frequency_index, private_bits,
+        // channel_configuration, original_copy, home
+        bits.skipBits(12);
+
         // adts_variable_header
 
         // copyright_identification_bit, copyright_identification_start
@@ -1004,27 +1325,6 @@
 }
 
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitH264() {
-    if (isScrambled()) {
-        if (mBuffer == NULL || mBuffer->size() == 0) {
-            return NULL;
-        }
-        if (mFormat == NULL) {
-            mFormat = new MetaData;
-            if (!MakeAVCCodecSpecificData(*mFormat, mBuffer->data(), mBuffer->size())) {
-                ALOGW("Creating dummy AVC format for scrambled content");
-                mFormat = new MetaData;
-                mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_AVC);
-                mFormat->setInt32(kKeyWidth, 1280);
-                mFormat->setInt32(kKeyHeight, 720);
-            }
-            // for MediaExtractor.CasInfo
-            mFormat->setInt32(kKeyCASystemID, mCASystemId);
-            mFormat->setData(kKeyCASessionID, 0,
-                    mCasSessionId.data(), mCasSessionId.size());
-        }
-        return dequeueScrambledAccessUnit();
-    }
-
     const uint8_t *data = mBuffer->data();
 
     size_t size = mBuffer->size();
@@ -1324,25 +1624,6 @@
 }
 
 sp<ABuffer> ElementaryStreamQueue::dequeueAccessUnitMPEGVideo() {
-    if (isScrambled()) {
-        if (mBuffer == NULL || mBuffer->size() == 0) {
-            return NULL;
-        }
-        if (mFormat == NULL) {
-            ALOGI("Creating dummy MPEG format for scrambled content");
-            mFormat = new MetaData;
-            mFormat->setCString(kKeyMIMEType, MEDIA_MIMETYPE_VIDEO_MPEG2);
-            mFormat->setInt32(kKeyWidth, 1280);
-            mFormat->setInt32(kKeyHeight, 720);
-
-            // for MediaExtractor.CasInfo
-            mFormat->setInt32(kKeyCASystemID, mCASystemId);
-            mFormat->setData(kKeyCASessionID, 0,
-                    mCasSessionId.data(), mCasSessionId.size());
-        }
-        return dequeueScrambledAccessUnit();
-    }
-
     const uint8_t *data = mBuffer->data();
     size_t size = mBuffer->size();
 
diff --git a/media/libstagefright/mpeg2ts/ESQueue.h b/media/libstagefright/mpeg2ts/ESQueue.h
index ffcb502..3227f47 100644
--- a/media/libstagefright/mpeg2ts/ESQueue.h
+++ b/media/libstagefright/mpeg2ts/ESQueue.h
@@ -38,6 +38,8 @@
         H264,
         AAC,
         AC3,
+        EAC3,
+        AC4,
         MPEG_AUDIO,
         MPEG_VIDEO,
         MPEG4_VIDEO,
@@ -59,6 +61,7 @@
 
     void appendScrambledData(
             const void *data, size_t size,
+            size_t leadingClearBytes,
             int32_t keyId, bool isSync,
             sp<ABuffer> clearSizes, sp<ABuffer> encSizes);
 
@@ -84,8 +87,8 @@
     };
 
     struct ScrambledRangeInfo {
-        //int64_t mTimestampUs;
         size_t mLength;
+        size_t mLeadingClearBytes;
         int32_t mKeyId;
         int32_t mIsSync;
         sp<ABuffer> mClearSizes;
@@ -115,7 +118,8 @@
 
     sp<ABuffer> dequeueAccessUnitH264();
     sp<ABuffer> dequeueAccessUnitAAC();
-    sp<ABuffer> dequeueAccessUnitAC3();
+    sp<ABuffer> dequeueAccessUnitEAC3();
+    sp<ABuffer> dequeueAccessUnitAC4();
     sp<ABuffer> dequeueAccessUnitMPEGAudio();
     sp<ABuffer> dequeueAccessUnitMPEGVideo();
     sp<ABuffer> dequeueAccessUnitMPEG4Video();
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/ndk/NdkMediaDrm.cpp b/media/ndk/NdkMediaDrm.cpp
index 6d10f1c..5597488 100644
--- a/media/ndk/NdkMediaDrm.cpp
+++ b/media/ndk/NdkMediaDrm.cpp
@@ -309,6 +309,7 @@
     }
     String8 defaultUrl;
     DrmPlugin::KeyRequestType keyRequestType;
+    mObj->mKeyRequest.clear();
     status_t status = mObj->mDrm->getKeyRequest(*iter, mdInit, String8(mimeType),
             mdKeyType, mdOptionalParameters, mObj->mKeyRequest, defaultUrl,
             &keyRequestType);
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index d6dae5b..f5b3f92 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -21,20 +21,31 @@
         "MemoryLeakTrackUtil.cpp",
         "ProcessInfo.cpp",
         "SchedulingPolicyService.cpp",
+        "ServiceUtilities.cpp",
+        "TimeCheck.cpp",
     ],
     shared_libs: [
         "libbinder",
+        "libcutils",
         "liblog",
         "libutils",
         "libmemunreachable",
     ],
 
+    logtags: ["EventLogTags.logtags"],
+
     cflags: [
         "-Wall",
         "-Wextra",
         "-Werror",
     ],
 
+    product_variables: {
+        product_is_iot: {
+            cflags: ["-DTARGET_ANDROID_THINGS"],
+        },
+    },
+
     local_include_dirs: ["include"],
     export_include_dirs: ["include"],
 }
diff --git a/media/utils/EventLogTags.logtags b/media/utils/EventLogTags.logtags
new file mode 100644
index 0000000..67f0ea8
--- /dev/null
+++ b/media/utils/EventLogTags.logtags
@@ -0,0 +1,41 @@
+# The entries in this file map a sparse set of log tag numbers to tag names.
+# This is installed on the device, in /system/etc, and parsed by logcat.
+#
+# Tag numbers are decimal integers, from 0 to 2^31.  (Let's leave the
+# negative values alone for now.)
+#
+# Tag names are one or more ASCII letters and numbers or underscores, i.e.
+# "[A-Z][a-z][0-9]_".  Do not include spaces or punctuation (the former
+# impacts log readability, the latter makes regex searches more annoying).
+#
+# Tag numbers and names are separated by whitespace.  Blank lines and lines
+# starting with '#' are ignored.
+#
+# Optionally, after the tag names can be put a description for the value(s)
+# of the tag. Description are in the format
+#    (<name>|data type[|data unit])
+# Multiple values are separated by commas.
+#
+# The data type is a number from the following values:
+# 1: int
+# 2: long
+# 3: string
+# 4: list
+#
+# The data unit is a number taken from the following list:
+# 1: Number of objects
+# 2: Number of bytes
+# 3: Number of milliseconds
+# 4: Number of allocations
+# 5: Id
+# 6: Percent
+# Default value for data of type int/long is 2 (bytes).
+#
+# See system/core/logcat/event.logtags for the master copy of the tags.
+
+# 61000 - 61199 reserved for audioserver
+
+61000 audioserver_binder_timeout (command|3)
+
+# NOTE - the range 1000000-2000000 is reserved for partners and others who
+# want to define their own log tags without conflicting with the core platform.
diff --git a/services/audioflinger/ServiceUtilities.cpp b/media/utils/ServiceUtilities.cpp
similarity index 76%
rename from services/audioflinger/ServiceUtilities.cpp
rename to media/utils/ServiceUtilities.cpp
index aa267ea..1c54aec 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);
@@ -179,8 +158,28 @@
     return ok;
 }
 
+bool modifyDefaultAudioEffectsAllowed() {
+    static const String16 sModifyDefaultAudioEffectsAllowed(
+            "android.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
+    // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
+    bool ok = PermissionCache::checkCallingPermission(sModifyDefaultAudioEffectsAllowed);
+
+#ifdef TARGET_ANDROID_THINGS
+    if (!ok) {
+        // Use a secondary permission on Android Things to allow a more lenient level of protection.
+        static const String16 sModifyDefaultAudioEffectsAndroidThingsAllowed(
+                "com.google.android.things.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
+        ok = PermissionCache::checkCallingPermission(
+                sModifyDefaultAudioEffectsAndroidThingsAllowed);
+    }
+    if (!ok) ALOGE("com.google.android.things.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
+#else
+    if (!ok) ALOGE("android.permission.MODIFY_DEFAULT_AUDIO_EFFECTS");
+#endif
+    return ok;
+}
+
 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 +195,29 @@
     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/libmedia/TimeCheck.cpp b/media/utils/TimeCheck.cpp
similarity index 91%
rename from media/libmedia/TimeCheck.cpp
rename to media/utils/TimeCheck.cpp
index dab5d4f..59cf4ef 100644
--- a/media/libmedia/TimeCheck.cpp
+++ b/media/utils/TimeCheck.cpp
@@ -15,7 +15,9 @@
  */
 
 
+#include <utils/Log.h>
 #include <media/TimeCheck.h>
+#include <media/EventLog.h>
 
 namespace android {
 
@@ -81,7 +83,10 @@
             status = mCond.waitRelative(mMutex, waitTimeNs);
         }
     }
-    LOG_ALWAYS_FATAL_IF(status != NO_ERROR, "TimeCheck timeout for %s", tag);
+    if (status != NO_ERROR) {
+        LOG_EVENT_STRING(LOGTAG_AUDIO_BINDER_TIMEOUT, tag);
+        LOG_ALWAYS_FATAL("TimeCheck timeout for %s", tag);
+    }
     return true;
 }
 
diff --git a/media/utils/include/mediautils/EventLog.h b/media/utils/include/mediautils/EventLog.h
new file mode 100644
index 0000000..553d3bd
--- /dev/null
+++ b/media/utils/include/mediautils/EventLog.h
@@ -0,0 +1,28 @@
+/*
+ * 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_AUDIO_EVENT_LOG_H_
+#define ANDROID_AUDIO_EVENT_LOG_H_
+
+namespace android {
+
+// keep values in sync with frameworks/av/media/utils/EventLogTags.logtags
+enum {
+  LOGTAG_AUDIO_BINDER_TIMEOUT = 61000,
+};
+
+}  // namespace android
+
+#endif  // ANDROID_AUDIO_EVENT_LOG_H_
diff --git a/media/utils/include/mediautils/ServiceUtilities.h b/media/utils/include/mediautils/ServiceUtilities.h
new file mode 100644
index 0000000..98f54c2
--- /dev/null
+++ b/media/utils/include/mediautils/ServiceUtilities.h
@@ -0,0 +1,75 @@
+/*
+ * 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 modifyDefaultAudioEffectsAllowed();
+bool dumpAllowed();
+bool modifyPhoneStateAllowed(pid_t pid, uid_t uid);
+status_t checkIMemory(const sp<IMemory>& iMemory);
+}
diff --git a/media/libmedia/include/media/TimeCheck.h b/media/utils/include/mediautils/TimeCheck.h
similarity index 100%
rename from media/libmedia/include/media/TimeCheck.h
rename to media/utils/include/mediautils/TimeCheck.h
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..2c26ba4 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 +36,14 @@
     libnbaio \
     libnblog \
     libpowermanager \
-    libserviceutility \
     libmediautils \
     libmemunreachable \
     libmedia_helper
 
 LOCAL_STATIC_LIBRARIES := \
     libcpustats \
+    libjsoncpp \
+    libsndfile \
 
 LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
 
@@ -82,6 +66,7 @@
 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..43566b7 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -20,15 +20,16 @@
 //#define LOG_NDEBUG 0
 
 #include "Configuration.h"
+#include <algorithm>    // std::any_of
 #include <dirent.h>
 #include <math.h>
 #include <signal.h>
+#include <string>
 #include <sys/time.h>
 #include <sys/resource.h>
 
 #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 +48,7 @@
 #include <system/audio.h>
 
 #include "AudioFlinger.h"
-#include "ServiceUtilities.h"
+#include "NBAIO_Tee.h"
 
 #include <media/AudioResamplerPublic.h>
 
@@ -56,7 +57,8 @@
 #include <system/audio_effects/effect_aec.h>
 
 #include <audio_utils/primitives.h>
-#include <audio_utils/string.h>
+
+#include <json/json.h>
 
 #include <powermanager/PowerManager.h>
 
@@ -66,6 +68,7 @@
 #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 +103,6 @@
 
 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 +152,7 @@
       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 +161,6 @@
         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 +177,6 @@
     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 +197,6 @@
         }
     }
 
-    mPatchPanel = new PatchPanel(this);
-
     mMode = AUDIO_MODE_NORMAL;
 
     gAudioFlinger = this;
@@ -341,7 +310,7 @@
         *sessionId = actualSessionId;
     } else {
         if (direction == MmapStreamInterface::DIRECTION_OUTPUT) {
-            AudioSystem::releaseOutput(io, streamType, actualSessionId);
+            AudioSystem::releaseOutput(portId);
         } else {
             AudioSystem::releaseInput(portId);
         }
@@ -469,6 +438,18 @@
     if (!dumpAllowed()) {
         dumpPermissionDenial(fd, args);
     } else {
+        // XXX This is sort of hacky for now.
+        const bool formatJson = std::any_of(args.begin(), args.end(),
+                [](const String16 &arg) { return arg == String16("--json"); });
+        if (formatJson) {
+            Json::Value root = getJsonDump();
+            Json::FastWriter writer;
+            std::string rootStr = writer.write(root);
+            // XXX consider buffering if the string happens to be too long.
+            dprintf(fd, "%s", rootStr.c_str());
+            return NO_ERROR;
+        }
+
         // get state of hardware lock
         bool hardwareLocked = dumpTryLock(mHardwareLock);
         if (!hardwareLocked) {
@@ -478,7 +459,7 @@
             mHardwareLock.unlock();
         }
 
-        bool locked = dumpTryLock(mLock);
+        const bool locked = dumpTryLock(mLock);
 
         // failed to lock - AudioFlinger is probably deadlocked
         if (!locked) {
@@ -534,12 +515,7 @@
             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 +523,10 @@
             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) {
@@ -581,6 +561,32 @@
     return NO_ERROR;
 }
 
+Json::Value AudioFlinger::getJsonDump()
+{
+    Json::Value root(Json::objectValue);
+    const bool locked = dumpTryLock(mLock);
+
+    // failed to lock - AudioFlinger is probably deadlocked
+    if (!locked) {
+        root["deadlock_message"] = kDeadlockedString;
+    }
+    // FIXME risky to access data structures without a lock held?
+
+    Json::Value playbackThreads = Json::arrayValue;
+    // dump playback threads
+    for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
+        playbackThreads.append(mPlaybackThreads.valueAt(i)->getJsonDump());
+    }
+
+    if (locked) {
+        mLock.unlock();
+    }
+
+    root["playback_threads"] = playbackThreads;
+
+    return root;
+}
+
 sp<AudioFlinger::Client> AudioFlinger::registerPid(pid_t pid)
 {
     Mutex::Autolock _cl(mClientLock);
@@ -666,7 +672,7 @@
     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);
@@ -813,7 +819,7 @@
 
 Exit:
     if (lStatus != NO_ERROR && output.outputId != AUDIO_IO_HANDLE_NONE) {
-        AudioSystem::releaseOutput(output.outputId, streamType, sessionId);
+        AudioSystem::releaseOutput(portId);
     }
     *status = lStatus;
     return trackHandle;
@@ -1078,9 +1084,9 @@
         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;
     }
 
@@ -1180,6 +1186,23 @@
     }
 }
 
+// forwardAudioHwSyncToDownstreamPatches_l() must be called with AudioFlinger::mLock held
+void AudioFlinger::forwardParametersToDownstreamPatches_l(
+        audio_io_handle_t upStream, const String8& keyValuePairs,
+        std::function<bool(const sp<PlaybackThread>&)> useThread)
+{
+    std::vector<PatchPanel::SoftwarePatch> swPatches;
+    if (mPatchPanel.getDownstreamSoftwarePatches(upStream, &swPatches) != OK) return;
+    ALOGV_IF(!swPatches.empty(), "%s found %zu downstream patches for stream ID %d",
+            __func__, swPatches.size(), upStream);
+    for (const auto& swPatch : swPatches) {
+        sp<PlaybackThread> downStream = checkPlaybackThread_l(swPatch.getPlaybackThreadHandle());
+        if (downStream != NULL && (useThread == nullptr || useThread(downStream))) {
+            downStream->setParameters(keyValuePairs);
+        }
+    }
+}
+
 // Filter reserved keys from setParameters() before forwarding to audio HAL or acting upon.
 // Some keys are used for audio routing and audio path configuration and should be reserved for use
 // by audio policy and audio flinger for functional, privacy and security reasons.
@@ -1200,9 +1223,8 @@
         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);
@@ -1296,7 +1318,9 @@
         }
     }
     if (thread != 0) {
-        return thread->setParameters(filteredKeyValuePairs);
+        status_t result = thread->setParameters(filteredKeyValuePairs);
+        forwardParametersToDownstreamPatches_l(thread->id(), filteredKeyValuePairs);
+        return result;
     }
     return BAD_VALUE;
 }
@@ -1636,7 +1660,7 @@
     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);
@@ -1854,6 +1878,10 @@
 
         mHardwareStatus = AUDIO_HW_IDLE;
     }
+    if (strcmp(name, AUDIO_HARDWARE_MODULE_ID_MSD) == 0) {
+        // An MSD module is inserted before hardware modules in order to mix encoded streams.
+        flags = static_cast<AudioHwDevice::Flags>(flags | AudioHwDevice::AHWD_IS_INSERT);
+    }
 
     audio_module_handle_t handle = (audio_module_handle_t) nextUniqueId(AUDIO_UNIQUE_ID_USE_MODULE);
     mAudioHwDevs.add(handle, new AudioHwDevice(handle, name, dev, flags));
@@ -1885,7 +1913,7 @@
 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 +1958,28 @@
     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);
@@ -1975,7 +2025,10 @@
         if (sessions & ThreadBase::TRACK_SESSION) {
             AudioParameter param = AudioParameter();
             param.addInt(String8(AudioParameter::keyStreamHwAvSync), value);
-            thread->setParameters(param.toString());
+            String8 keyValuePairs = param.toString();
+            thread->setParameters(keyValuePairs);
+            forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
+                    [](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
             break;
         }
     }
@@ -2021,7 +2074,10 @@
         ALOGV("setAudioHwSyncForSession_l found ID %d for session %d", syncId, sessionId);
         AudioParameter param = AudioParameter();
         param.addInt(String8(AudioParameter::keyStreamHwAvSync), syncId);
-        thread->setParameters(param.toString());
+        String8 keyValuePairs = param.toString();
+        thread->setParameters(keyValuePairs);
+        forwardParametersToDownstreamPatches_l(thread->id(), keyValuePairs,
+                [](const sp<PlaybackThread>& thread) { return thread->usesHwAvSync(); });
     }
 }
 
@@ -2111,6 +2167,7 @@
                       *output, thread.get());
             }
             mPlaybackThreads.add(*output, thread);
+            mPatchPanel.notifyStreamOpened(outHwDev, *output);
             return thread;
         }
     }
@@ -2246,6 +2303,7 @@
         const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
         ioDesc->mIoHandle = output;
         ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
+        mPatchPanel.notifyStreamClosed(output);
     }
     // The thread entity (active unit of execution) is no longer running here,
     // but the ThreadBase container still exists.
@@ -2274,7 +2332,7 @@
     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 +2408,9 @@
         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 +2468,6 @@
                     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 +2477,6 @@
                                       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 +2576,7 @@
     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 +2614,8 @@
     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 +2649,8 @@
     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 +2667,10 @@
             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)
@@ -2938,16 +2950,74 @@
 }
 
 status_t AudioFlinger::getEffectDescriptor(const effect_uuid_t *pUuid,
-        effect_descriptor_t *descriptor) const
+                                           const effect_uuid_t *pTypeUuid,
+                                           uint32_t preferredTypeFlag,
+                                           effect_descriptor_t *descriptor) const
 {
+    if (pUuid == NULL || pTypeUuid == NULL || descriptor == NULL) {
+        return BAD_VALUE;
+    }
+
     Mutex::Autolock _l(mLock);
-    if (mEffectsFactoryHal.get()) {
-        return mEffectsFactoryHal->getDescriptor(pUuid, descriptor);
-    } else {
+
+    if (!mEffectsFactoryHal.get()) {
         return -ENODEV;
     }
-}
 
+    status_t status = NO_ERROR;
+    if (!EffectsFactoryHalInterface::isNullUuid(pUuid)) {
+        // If uuid is specified, request effect descriptor from that.
+        status = mEffectsFactoryHal->getDescriptor(pUuid, descriptor);
+    } else if (!EffectsFactoryHalInterface::isNullUuid(pTypeUuid)) {
+        // If uuid is not specified, look for an available implementation
+        // of the required type instead.
+
+        // Use a temporary descriptor to avoid modifying |descriptor| in the failure case.
+        effect_descriptor_t desc;
+        desc.flags = 0; // prevent compiler warning
+
+        uint32_t numEffects = 0;
+        status = mEffectsFactoryHal->queryNumberEffects(&numEffects);
+        if (status < 0) {
+            ALOGW("getEffectDescriptor() error %d from FactoryHal queryNumberEffects", status);
+            return status;
+        }
+
+        bool found = false;
+        for (uint32_t i = 0; i < numEffects; i++) {
+            status = mEffectsFactoryHal->getDescriptor(i, &desc);
+            if (status < 0) {
+                ALOGW("getEffectDescriptor() error %d from FactoryHal getDescriptor", status);
+                continue;
+            }
+            if (memcmp(&desc.type, pTypeUuid, sizeof(effect_uuid_t)) == 0) {
+                // If matching type found save effect descriptor.
+                found = true;
+                *descriptor = desc;
+
+                // If there's no preferred flag or this descriptor matches the preferred
+                // flag, success! If this descriptor doesn't match the preferred
+                // flag, continue enumeration in case a better matching version of this
+                // effect type is available. Note that this means if no effect with a
+                // correct flag is found, the descriptor returned will correspond to the
+                // last effect that at least had a matching type uuid (if any).
+                if (preferredTypeFlag == EFFECT_FLAG_TYPE_MASK ||
+                    (desc.flags & EFFECT_FLAG_TYPE_MASK) == preferredTypeFlag) {
+                    break;
+                }
+            }
+        }
+
+        if (!found) {
+            status = NAME_NOT_FOUND;
+            ALOGW("getEffectDescriptor(): Effect not found by type.");
+        }
+    } else {
+        status = BAD_VALUE;
+        ALOGE("getEffectDescriptor(): Either uuid or type uuid must be non-null UUIDs.");
+    }
+    return status;
+}
 
 sp<IEffect> AudioFlinger::createEffect(
         effect_descriptor_t *pDesc,
@@ -2966,7 +3036,7 @@
     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 +3059,8 @@
     }
 
     // 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;
     }
@@ -3001,60 +3071,15 @@
     }
 
     {
-        if (!EffectsFactoryHalInterface::isNullUuid(&pDesc->uuid)) {
-            // if uuid is specified, request effect descriptor
-            lStatus = mEffectsFactoryHal->getDescriptor(&pDesc->uuid, &desc);
-            if (lStatus < 0) {
-                ALOGW("createEffect() error %d from EffectGetDescriptor", lStatus);
-                goto Exit;
-            }
-        } else {
-            // if uuid is not specified, look for an available implementation
-            // of the required type in effect factory
-            if (EffectsFactoryHalInterface::isNullUuid(&pDesc->type)) {
-                ALOGW("createEffect() no effect type");
-                lStatus = BAD_VALUE;
-                goto Exit;
-            }
-            uint32_t numEffects = 0;
-            effect_descriptor_t d;
-            d.flags = 0; // prevent compiler warning
-            bool found = false;
-
-            lStatus = mEffectsFactoryHal->queryNumberEffects(&numEffects);
-            if (lStatus < 0) {
-                ALOGW("createEffect() error %d from EffectQueryNumberEffects", lStatus);
-                goto Exit;
-            }
-            for (uint32_t i = 0; i < numEffects; i++) {
-                lStatus = mEffectsFactoryHal->getDescriptor(i, &desc);
-                if (lStatus < 0) {
-                    ALOGW("createEffect() error %d from EffectQueryEffect", lStatus);
-                    continue;
-                }
-                if (memcmp(&desc.type, &pDesc->type, sizeof(effect_uuid_t)) == 0) {
-                    // If matching type found save effect descriptor. If the session is
-                    // 0 and the effect is not auxiliary, continue enumeration in case
-                    // an auxiliary version of this effect type is available
-                    found = true;
-                    d = desc;
-                    if (sessionId != AUDIO_SESSION_OUTPUT_MIX ||
-                            (desc.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
-                        break;
-                    }
-                }
-            }
-            if (!found) {
-                lStatus = BAD_VALUE;
-                ALOGW("createEffect() effect not found");
-                goto Exit;
-            }
-            // For same effect type, chose auxiliary version over insert version if
-            // connect to output mix (Compliance to OpenSL ES)
-            if (sessionId == AUDIO_SESSION_OUTPUT_MIX &&
-                    (d.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_AUXILIARY) {
-                desc = d;
-            }
+        // Get the full effect descriptor from the uuid/type.
+        // If the session is the output mix, prefer an auxiliary effect,
+        // otherwise no preference.
+        uint32_t preferredType = (sessionId == AUDIO_SESSION_OUTPUT_MIX ?
+                                  EFFECT_FLAG_TYPE_AUXILIARY : EFFECT_FLAG_TYPE_MASK);
+        lStatus = getEffectDescriptor(&pDesc->uuid, &pDesc->type, preferredType, &desc);
+        if (lStatus < 0) {
+            ALOGW("createEffect() error %d from getEffectDescriptor", lStatus);
+            goto Exit;
         }
 
         // Do not allow auxiliary effects on a session different from 0 (output mix)
@@ -3370,136 +3395,6 @@
 }
 
 
-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..e9e6e94 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -23,6 +23,8 @@
 #include <mutex>
 #include <deque>
 #include <map>
+#include <memory>
+#include <string>
 #include <vector>
 #include <stdint.h>
 #include <sys/types.h>
@@ -62,7 +64,9 @@
 #include <media/LinearMap.h>
 #include <media/VolumeShaper.h>
 
+#include <audio_utils/clock.h>
 #include <audio_utils/SimpleLog.h>
+#include <audio_utils/TimestampVerifier.h>
 
 #include "FastCapture.h"
 #include "FastMixer.h"
@@ -71,9 +75,11 @@
 #include "AudioStreamOut.h"
 #include "SpdifStreamOut.h"
 #include "AudioHwDevice.h"
+#include "NBAIO_Tee.h"
 
 #include <powermanager/IPowerManager.h>
 
+#include <json/json.h>
 #include <media/nblog/NBLog.h>
 #include <private/media/AudioEffectShared.h>
 #include <private/media/AudioTrackShared.h>
@@ -109,6 +115,7 @@
     static const char* getServiceName() ANDROID_API { return "media.audio_flinger"; }
 
     virtual     status_t    dump(int fd, const Vector<String16>& args);
+                Json::Value getJsonDump();
 
     // IAudioFlinger interface, in binder opcode order
     virtual sp<IAudioTrack> createTrack(const CreateTrackInput& input,
@@ -202,6 +209,8 @@
     virtual status_t queryEffect(uint32_t index, effect_descriptor_t *descriptor) const;
 
     virtual status_t getEffectDescriptor(const effect_uuid_t *pUuid,
+                                         const effect_uuid_t *pTypeUuid,
+                                         uint32_t preferredTypeFlag,
                                          effect_descriptor_t *descriptor) const;
 
     virtual sp<IEffect> createEffect(
@@ -675,6 +684,9 @@
                 bool            updateOrphanEffectChains(const sp<EffectModule>& effect);
 
                 void broacastParametersToRecordThreads_l(const String8& keyValuePairs);
+                void forwardParametersToDownstreamPatches_l(
+                        audio_io_handle_t upStream, const String8& keyValuePairs,
+                        std::function<bool(const sp<PlaybackThread>&)> useThread = nullptr);
 
     // AudioStreamIn is immutable, so their fields are const.
     // For emphasis, we could also make all pointers to them be "const *",
@@ -791,44 +803,16 @@
 
     // 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 +827,8 @@
 
     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/AudioHwDevice.h b/services/audioflinger/AudioHwDevice.h
index eb826c6..d4299b0 100644
--- a/services/audioflinger/AudioHwDevice.h
+++ b/services/audioflinger/AudioHwDevice.h
@@ -35,6 +35,9 @@
     enum Flags {
         AHWD_CAN_SET_MASTER_VOLUME  = 0x1,
         AHWD_CAN_SET_MASTER_MUTE    = 0x2,
+        // Means that this isn't a terminal module, and software patches
+        // are used to transport audio data further.
+        AHWD_IS_INSERT              = 0x4,
     };
 
     AudioHwDevice(audio_module_handle_t handle,
@@ -55,6 +58,10 @@
         return (0 != (mFlags & AHWD_CAN_SET_MASTER_MUTE));
     }
 
+    bool isInsert() const {
+        return (0 != (mFlags & AHWD_IS_INSERT));
+    }
+
     audio_module_handle_t handle() const { return mHandle; }
     const char *moduleName() const { return mModuleName; }
     sp<DeviceHalInterface> hwDevice() const { return mHwDevice; }
diff --git a/services/audioflinger/Configuration.h b/services/audioflinger/Configuration.h
index ede8e3f..34cd821 100644
--- a/services/audioflinger/Configuration.h
+++ b/services/audioflinger/Configuration.h
@@ -27,7 +27,7 @@
 //#define AUDIO_WATCHDOG
 
 // uncomment to display CPU load adjusted for CPU frequency
-//#define CPU_FREQUENCY_STATISTICS
+//define CPU_FREQUENCY_STATISTICS
 
 // uncomment to enable fast threads to take performance samples for later statistical analysis
 #define FAST_THREAD_STATISTICS
diff --git a/services/audioflinger/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/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index d063772..dd84bf2 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -20,6 +20,7 @@
 #define ATRACE_TAG ATRACE_TAG_AUDIO
 
 #include "Configuration.h"
+#include <audio_utils/format.h>
 #include <linux/futex.h>
 #include <sys/syscall.h>
 #include <media/AudioBufferProvider.h>
@@ -161,7 +162,21 @@
     const FastCaptureState * const current = (const FastCaptureState *) mCurrent;
     FastCaptureDumpState * const dumpState = (FastCaptureDumpState *) mDumpState;
     const FastCaptureState::Command command = mCommand;
-    const size_t frameCount = current->mFrameCount;
+    size_t frameCount = current->mFrameCount;
+    AudioBufferProvider* fastPatchRecordBufferProvider = current->mFastPatchRecordBufferProvider;
+    AudioBufferProvider::Buffer patchBuffer;
+
+    if (fastPatchRecordBufferProvider != 0) {
+        patchBuffer.frameCount = ~0;
+        status_t status = fastPatchRecordBufferProvider->getNextBuffer(&patchBuffer);
+        if (status != NO_ERROR) {
+            frameCount = 0;
+        } else if (patchBuffer.frameCount < frameCount) {
+            // TODO: Make sure that it doesn't cause any issues if we just get a small available
+            // buffer from the buffer provider.
+            frameCount = patchBuffer.frameCount;
+        }
+    }
 
     if ((command & FastCaptureState::READ) /*&& isWarm*/) {
         ALOG_ASSERT(mInputSource != NULL);
@@ -176,6 +191,7 @@
             mTotalNativeFramesRead += framesRead;
             dumpState->mFramesRead = mTotalNativeFramesRead;
             mReadBufferState = framesRead;
+            patchBuffer.frameCount = framesRead;
         } else {
             dumpState->mReadErrors++;
             mReadBufferState = 0;
@@ -193,11 +209,18 @@
         }
         if (mReadBufferState > 0) {
             ssize_t framesWritten = mPipeSink->write(mReadBuffer, mReadBufferState);
-            // FIXME This supports at most one fast capture client.
-            //       To handle multiple clients this could be converted to an array,
-            //       or with a lot more work the control block could be shared by all clients.
             audio_track_cblk_t* cblk = current->mCblk;
-            if (cblk != NULL && framesWritten > 0) {
+            if (fastPatchRecordBufferProvider != 0) {
+                // This indicates the fast track is a patch record, update the cblk by
+                // calling releaseBuffer().
+                memcpy_by_audio_format(patchBuffer.raw, current->mFastPatchRecordFormat,
+                        mReadBuffer, mFormat.mFormat, framesWritten * mFormat.mChannelCount);
+                patchBuffer.frameCount = framesWritten;
+                fastPatchRecordBufferProvider->releaseBuffer(&patchBuffer);
+            } else if (cblk != NULL && framesWritten > 0) {
+                // FIXME This supports at most one fast capture client.
+                //       To handle multiple clients this could be converted to an array,
+                //       or with a lot more work the control block could be shared by all clients.
                 int32_t rear = cblk->u.mStreaming.mRear;
                 android_atomic_release_store(framesWritten + rear, &cblk->u.mStreaming.mRear);
                 cblk->mServer += framesWritten;
diff --git a/services/audioflinger/FastCaptureState.h b/services/audioflinger/FastCaptureState.h
index 9bca2d4..d287232 100644
--- a/services/audioflinger/FastCaptureState.h
+++ b/services/audioflinger/FastCaptureState.h
@@ -18,6 +18,7 @@
 #define ANDROID_AUDIO_FAST_CAPTURE_STATE_H
 
 #include <media/nbaio/NBAIO.h>
+#include <media/AudioBufferProvider.h>
 #include "FastThreadState.h"
 #include <private/media/AudioTrackShared.h>
 
@@ -37,6 +38,10 @@
     size_t          mFrameCount;        // number of frames per fast capture buffer
     audio_track_cblk_t* mCblk;          // control block for the single fast client, or NULL
 
+    audio_format_t  mFastPatchRecordFormat = AUDIO_FORMAT_INVALID;
+    AudioBufferProvider* mFastPatchRecordBufferProvider = nullptr;   // a reference to a patch
+                                                                     // record in fast mode
+
     // Extends FastThreadState::Command
     static const Command
         // The following commands also process configuration changes, and can be "or"ed:
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 79bb9fe..a42d6b3 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -32,7 +32,7 @@
 #include <utils/Trace.h>
 #include <system/audio.h>
 #ifdef FAST_THREAD_STATISTICS
-#include <cpustats/CentralTendencyStatistics.h>
+#include <audio_utils/Statistics.h>
 #ifdef CPU_FREQUENCY_STATISTICS
 #include <cpustats/ThreadCpuUsage.h>
 #endif
@@ -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..2abfbfb 100644
--- a/services/audioflinger/FastMixerDumpState.cpp
+++ b/services/audioflinger/FastMixerDumpState.cpp
@@ -19,11 +19,13 @@
 
 #include "Configuration.h"
 #ifdef FAST_THREAD_STATISTICS
-#include <cpustats/CentralTendencyStatistics.h>
+#include <audio_utils/Statistics.h>
 #ifdef CPU_FREQUENCY_STATISTICS
 #include <cpustats/ThreadCpuUsage.h>
 #endif
 #endif
+#include <json/json.h>
+#include <string>
 #include <utils/Debug.h>
 #include <utils/Log.h>
 #include "FastMixerDumpState.h"
@@ -68,21 +70,22 @@
     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;
-    uint32_t newestOpen = bounds & 0xFFFF;
+    const uint32_t bounds = mBounds;
+    const uint32_t newestOpen = bounds & 0xFFFF;
     uint32_t oldestClosed = bounds >> 16;
 
     //uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
     uint32_t n;
     __builtin_sub_overflow(newestOpen, oldestClosed, &n);
-    n = n & 0xFFFF;
+    n &= 0xFFFF;
 
     if (n > mSamplingN) {
         ALOGE("too many samples %u", n);
@@ -90,9 +93,9 @@
     }
     // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
     // and adjusted CPU load in MHz normalized for CPU clock frequency
-    CentralTendencyStatistics wall, loadNs;
+    audio_utils::Statistics<double> wall, loadNs;
 #ifdef CPU_FREQUENCY_STATISTICS
-    CentralTendencyStatistics kHz, loadMHz;
+    audio_utils::Statistics<double> kHz, loadMHz;
     uint32_t previousCpukHz = 0;
 #endif
     // Assuming a normal distribution for cycle times, three standard deviations on either side of
@@ -107,18 +110,18 @@
         if (tail != NULL) {
             tail[j] = wallNs;
         }
-        wall.sample(wallNs);
+        wall.add(wallNs);
         uint32_t sampleLoadNs = mLoadNs[i];
-        loadNs.sample(sampleLoadNs);
+        loadNs.add(sampleLoadNs);
 #ifdef CPU_FREQUENCY_STATISTICS
         uint32_t sampleCpukHz = mCpukHz[i];
         // skip bad kHz samples
         if ((sampleCpukHz & ~0xF) != 0) {
-            kHz.sample(sampleCpukHz >> 4);
+            kHz.add(sampleCpukHz >> 4);
             if (sampleCpukHz == previousCpukHz) {
                 double megacycles = (double) sampleLoadNs * (double) (sampleCpukHz >> 4) * 1e-12;
                 double adjMHz = megacycles / mixPeriodSec;  // _not_ wallNs * 1e9
-                loadMHz.sample(adjMHz);
+                loadMHz.add(adjMHz);
             }
         }
         previousCpukHz = sampleCpukHz;
@@ -126,42 +129,42 @@
     }
     if (n) {
         dprintf(fd, "  Simple moving statistics over last %.1f seconds:\n",
-                    wall.n() * mixPeriodSec);
+                    wall.getN() * mixPeriodSec);
         dprintf(fd, "    wall clock time in ms per mix cycle:\n"
                     "      mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                    wall.mean()*1e-6, wall.minimum()*1e-6, wall.maximum()*1e-6,
-                    wall.stddev()*1e-6);
+                    wall.getMean()*1e-6, wall.getMin()*1e-6, wall.getMax()*1e-6,
+                    wall.getStdDev()*1e-6);
         dprintf(fd, "    raw CPU load in us per mix cycle:\n"
                     "      mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                    loadNs.mean()*1e-3, loadNs.minimum()*1e-3, loadNs.maximum()*1e-3,
-                    loadNs.stddev()*1e-3);
+                    loadNs.getMean()*1e-3, loadNs.getMin()*1e-3, loadNs.getMax()*1e-3,
+                    loadNs.getStdDev()*1e-3);
     } else {
         dprintf(fd, "  No FastMixer statistics available currently\n");
     }
 #ifdef CPU_FREQUENCY_STATISTICS
     dprintf(fd, "  CPU clock frequency in MHz:\n"
                 "    mean=%.0f min=%.0f max=%.0f stddev=%.0f\n",
-                kHz.mean()*1e-3, kHz.minimum()*1e-3, kHz.maximum()*1e-3, kHz.stddev()*1e-3);
+                kHz.getMean()*1e-3, kHz.getMin()*1e-3, kHz.getMax()*1e-3, kHz.getStdDev()*1e-3);
     dprintf(fd, "  adjusted CPU load in MHz (i.e. normalized for CPU clock frequency):\n"
                 "    mean=%.1f min=%.1f max=%.1f stddev=%.1f\n",
-                loadMHz.mean(), loadMHz.minimum(), loadMHz.maximum(), loadMHz.stddev());
+                loadMHz.getMean(), loadMHz.getMin(), loadMHz.getMax(), loadMHz.getStdDev());
 #endif
     if (tail != NULL) {
         qsort(tail, n, sizeof(uint32_t), compare_uint32_t);
         // assume same number of tail samples on each side, left and right
         uint32_t count = n / kTailDenominator;
-        CentralTendencyStatistics left, right;
+        audio_utils::Statistics<double> left, right;
         for (uint32_t i = 0; i < count; ++i) {
-            left.sample(tail[i]);
-            right.sample(tail[n - (i + 1)]);
+            left.add(tail[i]);
+            right.add(tail[n - (i + 1)]);
         }
         dprintf(fd, "  Distribution of mix cycle times in ms for the tails "
                     "(> ~3 stddev outliers):\n"
                     "    left tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n"
                     "    right tail: mean=%.2f min=%.2f max=%.2f stddev=%.2f\n",
-                    left.mean()*1e-6, left.minimum()*1e-6, left.maximum()*1e-6, left.stddev()*1e-6,
-                    right.mean()*1e-6, right.minimum()*1e-6, right.maximum()*1e-6,
-                    right.stddev()*1e-6);
+                    left.getMean()*1e-6, left.getMin()*1e-6, left.getMax()*1e-6, left.getStdDev()*1e-6,
+                    right.getMean()*1e-6, right.getMin()*1e-6, right.getMax()*1e-6,
+                    right.getStdDev()*1e-6);
         delete[] tail;
     }
 #endif
@@ -203,4 +206,49 @@
     }
 }
 
+Json::Value FastMixerDumpState::getJsonDump() const
+{
+    Json::Value root(Json::objectValue);
+    if (mCommand == FastMixerState::INITIAL) {
+        root["status"] = "uninitialized";
+        return root;
+    }
+#ifdef FAST_THREAD_STATISTICS
+    // find the interval of valid samples
+    const uint32_t bounds = mBounds;
+    const uint32_t newestOpen = bounds & 0xFFFF;
+    uint32_t oldestClosed = bounds >> 16;
+
+    //uint32_t n = (newestOpen - oldestClosed) & 0xFFFF;
+    uint32_t n;
+    __builtin_sub_overflow(newestOpen, oldestClosed, &n);
+    n &= 0xFFFF;
+
+    if (n > mSamplingN) {
+        ALOGE("too many samples %u", n);
+        n = mSamplingN;
+    }
+    // statistics for monotonic (wall clock) time, thread raw CPU load in time, CPU clock frequency,
+    // and adjusted CPU load in MHz normalized for CPU clock frequency
+    Json::Value jsonWall(Json::arrayValue);
+    Json::Value jsonLoadNs(Json::arrayValue);
+    // loop over all the samples
+    for (uint32_t j = 0; j < n; ++j) {
+        size_t i = oldestClosed++ & (mSamplingN - 1);
+        uint32_t wallNs = mMonotonicNs[i];
+        jsonWall.append(wallNs);
+        uint32_t sampleLoadNs = mLoadNs[i];
+        jsonLoadNs.append(sampleLoadNs);
+    }
+    if (n) {
+        root["wall_clock_time_ns"] = jsonWall;
+        root["raw_cpu_load_ns"] = jsonLoadNs;
+        root["status"] = "ok";
+    } else {
+        root["status"] = "unavailable";
+    }
+#endif
+    return root;
+}
+
 }   // android
diff --git a/services/audioflinger/FastMixerDumpState.h b/services/audioflinger/FastMixerDumpState.h
index 8ef31d1..69c2e4e 100644
--- a/services/audioflinger/FastMixerDumpState.h
+++ b/services/audioflinger/FastMixerDumpState.h
@@ -18,6 +18,9 @@
 #define ANDROID_AUDIO_FAST_MIXER_DUMP_STATE_H
 
 #include <stdint.h>
+#include <string>
+#include <audio_utils/TimestampVerifier.h>
+#include <json/json.h>
 #include "Configuration.h"
 #include "FastThreadDumpState.h"
 #include "FastMixerState.h"
@@ -64,8 +67,10 @@
     FastMixerDumpState();
     /*virtual*/ ~FastMixerDumpState();
 
-    void dump(int fd) const;    // should only be called on a stable copy, not the original
+    void dump(int fd) const;             // should only be called on a stable copy, not the original
+    Json::Value getJsonDump() 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 +79,9 @@
     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..ada8572 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>
 
 // ----------------------------------------------------------------------------
 
@@ -49,111 +50,77 @@
                                 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 mPatchPanel.listAudioPatches(num_patches, patches);
+}
+
+status_t AudioFlinger::PatchPanel::SoftwarePatch::getLatencyMs_l(double *latencyMs) const
+{
+    const auto& iter = mPatchPanel.mPatches.find(mPatchHandle);
+    if (iter != mPatchPanel.mPatches.end()) {
+        return iter->second.getLatencyMs(latencyMs);
+    } else {
+        return BAD_VALUE;
     }
-    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()
-{
 }
 
 /* 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) ||
-            patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
+    if (!audio_patch_is_valid(patch) || (patch->num_sinks == 0 && patch->num_sources != 2)) {
         return BAD_VALUE;
     }
     // limit number of sources to 1 for now or 2 sources for special cross hw module case.
@@ -163,81 +130,73 @@
     }
 
     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);
+            removeSoftwarePatchFromInsertedModules(*handle);
         }
     }
 
-    Patch *newPatch = new Patch(patch);
+    Patch newPatch{*patch};
+    audio_module_handle_t insertedModule = AUDIO_MODULE_HANDLE_NONE;
 
     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);
-            if (index < 0) {
-                ALOGW("createAudioPatch() bad src hw module %d", srcModule);
+            AudioHwDevice *audioHwDevice = findAudioHwDeviceByModule(srcModule);
+            if (!audioHwDevice) {
                 status = BAD_VALUE;
                 goto exit;
             }
-            AudioHwDevice *audioHwDevice = audioflinger->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 +215,52 @@
                     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 = AUDIO_OUTPUT_FLAG_NONE;
+                    if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+                        config.sample_rate = patch->sinks[0].sample_rate;
+                    }
+                    if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+                        config.channel_mask = patch->sinks[0].channel_mask;
+                    }
+                    if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+                        config.format = patch->sinks[0].format;
+                    }
+                    if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS) {
+                        flags = patch->sinks[0].flags.output;
+                    }
+                    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 +270,53 @@
                 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;
                 }
+                if (audioHwDevice->isInsert()) {
+                    insertedModule = audioHwDevice->handle();
+                }
             } 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 +335,9 @@
         } 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 +345,8 @@
             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 +358,21 @@
                 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 +382,383 @@
             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)));
+        if (insertedModule != AUDIO_MODULE_HANDLE_NONE) {
+            addSoftwarePatchToInsertedModules(insertedModule, *handle);
+        }
+        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();
-    int shift = __builtin_ctz(recordFramecount);
+    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);
+    size_t frameCount = (playbackFrameCount * recordFrameCount) >> shift;
+    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;
+    if (sampleRate == mRecord.thread()->sampleRate() &&
+            inChannelMask == mRecord.thread()->channelMask() &&
+            mRecord.thread()->fastTrackAvailable() &&
+            mRecord.thread()->hasFastCapture()) {
+        // Create a fast track if the record thread has fast capture to get better performance.
+        // Only enable fast mode when there is no resample needed.
+        inputFlags = (audio_input_flags_t) (inputFlags | AUDIO_INPUT_FLAG_FAST);
+    } else {
+        // Fast mode is not available in this case.
+        inputFlags = (audio_input_flags_t) (inputFlags & ~AUDIO_INPUT_FLAG_FAST);
+    }
+    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;
+    audio_stream_type_t streamType = AUDIO_STREAM_PATCH;
+    if (mAudioPatch.num_sources == 2 && mAudioPatch.sources[1].type == AUDIO_PORT_TYPE_MIX) {
+        // "reuse one existing output mix" case
+        streamType = mAudioPatch.sources[1].ext.mix.usecase.stream;
+    }
+    if (mPlayback.thread()->hasFastMixer()) {
+        // Create a fast track if the playback thread has fast mixer to get better performance.
+        outputFlags = (audio_output_flags_t) (outputFlags | AUDIO_OUTPUT_FLAG_FAST);
+    }
 
     // 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(),
+                                           streamType,
                                            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);
+}
 
-    ALOGV("clearPatchConnections() patch->mRecordPatchHandle %d patch->mPlaybackPatchHandle %d",
-          patch->mRecordPatchHandle, patch->mPlaybackPatchHandle);
+status_t AudioFlinger::PatchPanel::Patch::getLatencyMs(double *latencyMs) const
+{
+    if (!isSoftware()) return INVALID_OPERATION;
 
-    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();
-    }
+    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) const
+{
+    // TODO: Consider table dump form for patches, just like tracks.
+    String8 result = String8::format("Patch %d: thread %p => thread %p",
+            myHandle, mRecord.const_thread().get(), mPlayback.const_thread().get());
+
+    // add latency if it exists
+    double latencyMs;
+    if (getLatencyMs(&latencyMs) == OK) {
+        result.appendFormat("  latency: %.2lf", latencyMs);
+    }
+    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);
+    removeSoftwarePatchFromInsertedModules(handle);
     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)
+status_t AudioFlinger::PatchPanel::getDownstreamSoftwarePatches(
+        audio_io_handle_t stream,
+        std::vector<AudioFlinger::PatchPanel::SoftwarePatch> *patches) const
 {
-    ALOGV("setAudioPortConfig");
-
-    sp<AudioFlinger> audioflinger = mAudioFlinger.promote();
-    if (audioflinger == 0) {
-        return NO_INIT;
+    for (const auto& module : mInsertedModules) {
+        if (module.second.streams.count(stream)) {
+            for (const auto& patchHandle : module.second.sw_patches) {
+                const auto& patch_iter = mPatches.find(patchHandle);
+                if (patch_iter != mPatches.end()) {
+                    const Patch &patch = patch_iter->second;
+                    patches->emplace_back(*this, patchHandle,
+                            patch.mPlayback.const_thread()->id(),
+                            patch.mRecord.const_thread()->id());
+                } else {
+                    ALOGE("Stale patch handle in the cache: %d", patchHandle);
+                }
+            }
+            return OK;
+        }
     }
+    // The stream is not associated with any of inserted modules.
+    return BAD_VALUE;
+}
 
-    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;
+void AudioFlinger::PatchPanel::notifyStreamOpened(
+        AudioHwDevice *audioHwDevice, audio_io_handle_t stream)
+{
+    if (audioHwDevice->isInsert()) {
+        mInsertedModules[audioHwDevice->handle()].streams.insert(stream);
     }
+}
 
-    ssize_t index = audioflinger->mAudioHwDevs.indexOfKey(module);
+void AudioFlinger::PatchPanel::notifyStreamClosed(audio_io_handle_t stream)
+{
+    for (auto& module : mInsertedModules) {
+        module.second.streams.erase(stream);
+    }
+}
+
+AudioHwDevice* AudioFlinger::PatchPanel::findAudioHwDeviceByModule(audio_module_handle_t module)
+{
+    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;
+        ALOGW("%s() bad hw module %d", __func__, module);
+        return nullptr;
+    }
+    return mAudioFlinger.mAudioHwDevs.valueAt(index);
+}
+
+sp<DeviceHalInterface> AudioFlinger::PatchPanel::findHwDeviceByModule(audio_module_handle_t module)
+{
+    AudioHwDevice *audioHwDevice = findAudioHwDeviceByModule(module);
+    return audioHwDevice ? audioHwDevice->hwDevice() : nullptr;
+}
+
+void AudioFlinger::PatchPanel::addSoftwarePatchToInsertedModules(
+        audio_module_handle_t module, audio_patch_handle_t handle)
+{
+    mInsertedModules[module].sw_patches.insert(handle);
+}
+
+void AudioFlinger::PatchPanel::removeSoftwarePatchFromInsertedModules(
+        audio_patch_handle_t handle)
+{
+    for (auto& module : mInsertedModules) {
+        module.second.sw_patches.erase(handle);
+    }
+}
+
+void AudioFlinger::PatchPanel::dump(int fd) const
+{
+    String8 patchPanelDump;
+    const char *indent = "  ";
+
+    // Only dump software patches.
+    bool headerPrinted = false;
+    for (const auto& iter : mPatches) {
+        if (iter.second.isSoftware()) {
+            if (!headerPrinted) {
+                patchPanelDump += "\nSoftware patches:\n";
+                headerPrinted = true;
+            }
+            patchPanelDump.appendFormat("%s%s\n", indent, iter.second.dump(iter.first).string());
+        }
     }
 
-    AudioHwDevice *audioHwDevice = audioflinger->mAudioHwDevs.valueAt(index);
-    return audioHwDevice->hwDevice()->setAudioPortConfig(config);
+    headerPrinted = false;
+    for (const auto& module : mInsertedModules) {
+        if (!module.second.streams.empty() || !module.second.sw_patches.empty()) {
+            if (!headerPrinted) {
+                patchPanelDump += "\nTracked inserted modules:\n";
+                headerPrinted = true;
+            }
+            String8 moduleDump = String8::format("Module %d: I/O handles: ", module.first);
+            for (const auto& stream : module.second.streams) {
+                moduleDump.appendFormat("%d ", stream);
+            }
+            moduleDump.append("; SW Patches: ");
+            for (const auto& patch : module.second.sw_patches) {
+                moduleDump.appendFormat("%d ", patch);
+            }
+            patchPanelDump.appendFormat("%s%s\n", indent, moduleDump.string());
+        }
+    }
+
+    if (!patchPanelDump.isEmpty()) {
+        write(fd, patchPanelDump.string(), patchPanelDump.size());
+    }
 }
 
 } // namespace android
diff --git a/services/audioflinger/PatchPanel.h b/services/audioflinger/PatchPanel.h
index d37c0d3..269a398 100644
--- a/services/audioflinger/PatchPanel.h
+++ b/services/audioflinger/PatchPanel.h
@@ -19,13 +19,32 @@
     #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 SoftwarePatch {
+      public:
+        SoftwarePatch(const PatchPanel &patchPanel, audio_patch_handle_t patchHandle,
+                audio_io_handle_t playbackThreadHandle, audio_io_handle_t recordThreadHandle)
+                : mPatchPanel(patchPanel), mPatchHandle(patchHandle),
+                  mPlaybackThreadHandle(playbackThreadHandle),
+                  mRecordThreadHandle(recordThreadHandle) {}
+        SoftwarePatch(const SoftwarePatch&) = default;
+        SoftwarePatch& operator=(const SoftwarePatch&) = default;
 
-    class Patch;
+        // Must be called under AudioFlinger::mLock
+        status_t getLatencyMs_l(double *latencyMs) const;
+        audio_io_handle_t getPlaybackThreadHandle() const { return mPlaybackThreadHandle; };
+        audio_io_handle_t getRecordThreadHandle() const { return mRecordThreadHandle; };
+      private:
+        const PatchPanel &mPatchPanel;
+        const audio_patch_handle_t mPatchHandle;
+        const audio_io_handle_t mPlaybackThreadHandle;
+        const audio_io_handle_t mRecordThreadHandle;
+    };
 
-    explicit PatchPanel(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 +64,144 @@
     status_t listAudioPatches(unsigned int *num_patches,
                                       struct audio_patch *patches);
 
-    /* Set audio port configuration */
-    status_t setAudioPortConfig(const struct audio_port_config *config);
+    // Retrieves all currently estrablished software patches for a stream
+    // opened on an intermediate module.
+    status_t getDownstreamSoftwarePatches(audio_io_handle_t stream,
+            std::vector<SoftwarePatch> *patches) const;
 
-    status_t createPatchConnections(Patch *patch,
-                                    const struct audio_patch *audioPatch);
-    void clearPatchConnections(Patch *patch);
+    // Notifies patch panel about all opened and closed streams.
+    void notifyStreamOpened(AudioHwDevice *audioHwDevice, audio_io_handle_t stream);
+    void notifyStreamClosed(audio_io_handle_t stream);
+
+    void dump(int fd) const;
+
+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 ThreadType> const_thread() const { return mThread; }
+        sp<const TrackType> const_track() const { return mTrack; }
+
+        void closeConnections(PatchPanel *panel) {
+            if (mHandle != AUDIO_PATCH_HANDLE_NONE) {
+                panel->releaseAudioPatch(mHandle);
+                mHandle = AUDIO_PATCH_HANDLE_NONE;
+            }
+            if (mThread != 0) {
+                if (mTrack != 0) {
+                    mThread->deletePatchTrack(mTrack);
+                }
+                if (mCloseThread) {
+                    panel->mAudioFlinger.closeThreadInternal_l(mThread);
+                }
+            }
+        }
+        audio_patch_handle_t* handlePtr() { return &mHandle; }
+        void setThread(const sp<ThreadType>& thread, bool closeThread = true) {
+            mThread = thread;
+            mCloseThread = closeThread;
+        }
+        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) const;
+
+        // 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;
+    AudioHwDevice* findAudioHwDeviceByModule(audio_module_handle_t module);
+    sp<DeviceHalInterface> findHwDeviceByModule(audio_module_handle_t module);
+    void addSoftwarePatchToInsertedModules(
+            audio_module_handle_t module, audio_patch_handle_t handle);
+    void removeSoftwarePatchFromInsertedModules(audio_patch_handle_t handle);
+
+    AudioFlinger &mAudioFlinger;
+    std::map<audio_patch_handle_t, Patch> mPatches;
+
+    // This map allows going from a thread to "downstream" software patches
+    // when a processing module inserted in between. Example:
+    //
+    //  from map value.streams                               map key
+    //  [Mixer thread] --> [Virtual output device] --> [Processing module] ---\
+    //       [Harware module] <-- [Physical output device] <-- [S/W Patch] <--/
+    //                                                 from map value.sw_patches
+    //
+    // This allows the mixer thread to look up the threads of the software patch
+    // for propagating timing info, parameters, etc.
+    //
+    // The current assumptions are:
+    //   1) The processing module acts as a mixer with several outputs which
+    //      represent differently downmixed and / or encoded versions of the same
+    //      mixed stream. There is no 1:1 correspondence between the input streams
+    //      and the software patches, but rather a N:N correspondence between
+    //      a group of streams and a group of patches.
+    //   2) There are only a couple of inserted processing modules in the system,
+    //      so when looking for a stream or patch handle we can iterate over
+    //      all modules.
+    struct ModuleConnections {
+        std::set<audio_io_handle_t> streams;
+        std::set<audio_patch_handle_t> sw_patches;
+    };
+    std::map<audio_module_handle_t, ModuleConnections> mInsertedModules;
 };
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index a78be99..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..cd585f5 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -23,6 +23,8 @@
 #include "Configuration.h"
 #include <math.h>
 #include <fcntl.h>
+#include <memory>
+#include <string>
 #include <linux/futex.h>
 #include <sys/stat.h>
 #include <sys/syscall.h>
@@ -40,6 +42,7 @@
 #include <audio_utils/primitives.h>
 #include <audio_utils/format.h>
 #include <audio_utils/minifloat.h>
+#include <json/json.h>
 #include <system/audio_effects/effect_ns.h>
 #include <system/audio_effects/effect_aec.h>
 #include <system/audio.h>
@@ -62,8 +65,8 @@
 #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>
@@ -71,7 +74,7 @@
 #endif
 
 #ifdef DEBUG_CPU_USAGE
-#include <cpustats/CentralTendencyStatistics.h>
+#include <audio_utils/Statistics.h>
 #include <cpustats/ThreadCpuUsage.h>
 #endif
 
@@ -200,7 +203,7 @@
 // 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;
 
 // ----------------------------------------------------------------------------
 
@@ -333,9 +336,9 @@
 #ifdef DEBUG_CPU_USAGE
 private:
     ThreadCpuUsage mCpuUsage;           // instantaneous thread CPU usage in wall clock ns
-    CentralTendencyStatistics mWcStats; // statistics on thread CPU usage in wall clock ns
+    audio_utils::Statistics<double> mWcStats; // statistics on thread CPU usage in wall clock ns
 
-    CentralTendencyStatistics mHzStats; // statistics on thread CPU usage in cycles
+    audio_utils::Statistics<double> mHzStats; // statistics on thread CPU usage in cycles
 
     int mCpuNum;                        // thread's current CPU number
     int mCpukHz;                        // frequency of thread's current CPU in kHz
@@ -361,7 +364,7 @@
 
     // record sample for wall clock statistics
     if (valid) {
-        mWcStats.sample(wcNs);
+        mWcStats.add(wcNs);
     }
 
     // get the current CPU number
@@ -380,26 +383,26 @@
 
     // if no change in CPU number or frequency, then record sample for cycle statistics
     if (valid && mCpukHz > 0) {
-        double cycles = wcNs * cpukHz * 0.000001;
-        mHzStats.sample(cycles);
+        const double cycles = wcNs * cpukHz * 0.000001;
+        mHzStats.add(cycles);
     }
 
-    unsigned n = mWcStats.n();
+    const unsigned n = mWcStats.getN();
     // mCpuUsage.elapsed() is expensive, so don't call it every loop
     if ((n & 127) == 1) {
-        long long elapsed = mCpuUsage.elapsed();
+        const long long elapsed = mCpuUsage.elapsed();
         if (elapsed >= DEBUG_CPU_USAGE * 1000000000LL) {
-            double perLoop = elapsed / (double) n;
-            double perLoop100 = perLoop * 0.01;
-            double perLoop1k = perLoop * 0.001;
-            double mean = mWcStats.mean();
-            double stddev = mWcStats.stddev();
-            double minimum = mWcStats.minimum();
-            double maximum = mWcStats.maximum();
-            double meanCycles = mHzStats.mean();
-            double stddevCycles = mHzStats.stddev();
-            double minCycles = mHzStats.minimum();
-            double maxCycles = mHzStats.maximum();
+            const double perLoop = elapsed / (double) n;
+            const double perLoop100 = perLoop * 0.01;
+            const double perLoop1k = perLoop * 0.001;
+            const double mean = mWcStats.getMean();
+            const double stddev = mWcStats.getStdDev();
+            const double minimum = mWcStats.getMin();
+            const double maximum = mWcStats.getMax();
+            const double meanCycles = mHzStats.getMean();
+            const double stddevCycles = mHzStats.getStdDev();
+            const double minCycles = mHzStats.getMin();
+            const double maxCycles = mHzStats.getMax();
             mCpuUsage.resetElapsed();
             mWcStats.reset();
             mHzStats.reset();
@@ -769,6 +772,8 @@
             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 +788,12 @@
             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 +856,15 @@
     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
+            || mType == OFFLOAD) {
+        dprintf(fd, "  Timestamp stats: %s\n", mTimestampVerifier.toString().c_str());
+    }
+
     if (locked) {
         mLock.unlock();
     }
@@ -1519,7 +1539,7 @@
     }
 }
 
-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 +1591,9 @@
     --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;
 }
 
@@ -1741,6 +1764,11 @@
     mLocalLog.dump(fd, "   " /* prefix */, 40 /* lines */);
 }
 
+Json::Value AudioFlinger::PlaybackThread::getJsonDump() const
+{
+    return Json::Value(Json::objectValue);
+}
+
 void AudioFlinger::PlaybackThread::dumpTracks(int fd, const Vector<String16>& args __unused)
 {
     String8 result;
@@ -1773,7 +1801,7 @@
     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 +1821,7 @@
         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 +1908,17 @@
     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()) {
@@ -2298,15 +2332,13 @@
         if (track->isExternalTrack()) {
             TrackBase::track_state state = track->mState;
             mLock.unlock();
-            status = AudioSystem::startOutput(mId, track->streamType(),
-                                              track->sessionId());
+            status = AudioSystem::startOutput(track->portId());
             mLock.lock();
             // abort track was stopped/paused while we released the lock
             if (state != track->mState) {
                 if (status == NO_ERROR) {
                     mLock.unlock();
-                    AudioSystem::stopOutput(mId, track->streamType(),
-                                            track->sessionId());
+                    AudioSystem::stopOutput(track->portId());
                     mLock.lock();
                 }
                 return INVALID_OPERATION;
@@ -2457,6 +2489,11 @@
     Mutex::Autolock _l(mLock);
     // reject out of sequence requests
     if ((mDrainSequence & 1) && (sequence == mDrainSequence)) {
+        // Register discontinuity when HW drain is completed because that can cause
+        // the timestamp frame position to reset to 0 for direct and offload threads.
+        // (Out of sequence requests are ignored, since the discontinuity would be handled
+        // elsewhere, e.g. in flush).
+        mTimestampVerifier.discontinuity();
         mDrainSequence &= ~1;
         mWaitWorkCV.signal();
     }
@@ -2781,15 +2818,13 @@
         for (size_t i = 0 ; i < count ; i++) {
             const sp<Track>& track = tracksToRemove.itemAt(i);
             if (track->isExternalTrack()) {
-                AudioSystem::stopOutput(mId, track->streamType(),
-                                        track->sessionId());
+                AudioSystem::stopOutput(track->portId());
 #ifdef ADD_BATTERY_DATA
                 // to track the speaker usage
                 addBatteryData(IMediaPlayerService::kBatteryDataAudioFlingerStop);
 #endif
                 if (track->isTerminated()) {
-                    AudioSystem::releaseOutput(mId, track->streamType(),
-                                               track->sessionId());
+                    AudioSystem::releaseOutput(track->portId());
                 }
             }
         }
@@ -2845,6 +2880,9 @@
         ATRACE_END();
         if (framesWritten > 0) {
             bytesWritten = framesWritten * mFrameSize;
+#ifdef TEE_SINK
+            mTee.write((char *)mSinkBuffer + offset, framesWritten);
+#endif
         } else {
             bytesWritten = framesWritten;
         }
@@ -3162,6 +3200,15 @@
 
     checkSilentMode_l();
 
+    // DIRECT and OFFLOAD threads should reset frame count to zero on stop/flush
+    // TODO: add confirmation checks:
+    // 1) DIRECT threads and linear PCM format really resets to 0?
+    // 2) Is frame count really valid if not linear pcm?
+    // 3) Are all 64 bits of position returned, not just lowest 32 bits?
+    if (mType == OFFLOAD || mType == DIRECT) {
+        mTimestampVerifier.setDiscontinuityMode(mTimestampVerifier.DISCONTINUITY_MODE_ZERO);
+    }
+
     while (!exitPending())
     {
         // Log merge requests are performed during AudioFlinger binder transactions, but
@@ -3185,16 +3232,24 @@
                 logString = NULL;
             }
 
+            // Collect timestamp statistics for the Playback Thread types that support it.
+            if (mType == MIXER
+                    || mType == DUPLICATING
+                    || mType == DIRECT
+                    || mType == OFFLOAD) { // 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 +3278,10 @@
                         + 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 +3309,12 @@
                         t->updateTrackFrameInfo(
                                 t->mAudioTrackServerProxy->framesReleased(),
                                 mFramesWritten,
+                                mSampleRate,
                                 mTimestamp);
                     }
                 }
             }
+            } // if (mType ... ) { // no indentation
 #if 0
             // logFormat example
             if (z % 100 == 0) {
@@ -3295,7 +3355,7 @@
 
                 continue;
             }
-            if ((!mActiveTracks.size() && systemTime() > mStandbyTimeNs) ||
+            if ((mActiveTracks.isEmpty() && systemTime() > mStandbyTimeNs) ||
                                    isSuspended()) {
                 // put audio hardware into standby after short delay
                 if (shouldStandby_l()) {
@@ -3309,7 +3369,7 @@
                     mStandby = true;
                 }
 
-                if (!mActiveTracks.size() && mConfigEvents.isEmpty()) {
+                if (mActiveTracks.isEmpty() && mConfigEvents.isEmpty()) {
                     // we're about to wait, flush the binder command buffer
                     IPCThreadState::self()->flushCommands();
 
@@ -3772,12 +3832,16 @@
     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 +3922,7 @@
 
         // 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 +3934,7 @@
         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 +3945,8 @@
                 (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 +3972,6 @@
         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 +3980,7 @@
         // 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 +3989,14 @@
         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 +4312,37 @@
     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 +4377,14 @@
             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 +4799,13 @@
                 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 +5104,12 @@
     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());
@@ -5033,9 +5118,8 @@
         // while we are dumping it.  It may be inconsistent, but it won't mutate!
         // This is a large object so we place it on the heap.
         // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
-        const FastMixerDumpState *copy = new FastMixerDumpState(mFastMixerDumpState);
+        const std::unique_ptr<FastMixerDumpState> copy(new FastMixerDumpState(mFastMixerDumpState));
         copy->dump(fd);
-        delete copy;
 
 #ifdef STATE_QUEUE_DUMP
         // Similar for state queue
@@ -5056,12 +5140,22 @@
     } else {
         dprintf(fd, "  No FastMixer\n");
     }
+}
 
-#ifdef TEE_SINK
-    // Write the tee output to a .wav file
-    dumpTee(fd, mTeeSource, mId, 'M');
-#endif
-
+Json::Value AudioFlinger::MixerThread::getJsonDump() const
+{
+    Json::Value root;
+    if (hasFastMixer()) {
+        // Make a non-atomic copy of fast mixer dump state so it won't change underneath us
+        // while we are dumping it.  It may be inconsistent, but it won't mutate!
+        // This is a large object so we place it on the heap.
+        // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
+        const std::unique_ptr<FastMixerDumpState> copy(new FastMixerDumpState(mFastMixerDumpState));
+        root["fastmixer_stats"] = copy->getJsonDump();
+    } else {
+        root["fastmixer_stats"] = "no_fastmixer";
+    }
+    return root;
 }
 
 uint32_t AudioFlinger::MixerThread::idleSleepTimeUs() const
@@ -5563,6 +5657,7 @@
     mOutput->flush();
     mHwPaused = false;
     mFlushPending = false;
+    mTimestampVerifier.discontinuity(); // DIRECT and OFFLOADED flush resets frame count.
 }
 
 int64_t AudioFlinger::DirectOutputThread::computeWaitTimeNs_l() const {
@@ -5897,6 +5992,14 @@
                     track->presentationComplete(framesWritten, audioHALFrames);
                     track->reset();
                     tracksToRemove->add(track);
+                    // DIRECT and OFFLOADED stop resets frame counts.
+                    if (!mUseAsyncWrite) {
+                        // If we don't get explicit drain notification we must
+                        // register discontinuity regardless of whether this is
+                        // the previous (!last) or the upcoming (last) track
+                        // to avoid skipping the discontinuity.
+                        mTimestampVerifier.discontinuity();
+                    }
                 }
             } else {
                 // No buffers for this track. Give it a few chances to
@@ -6062,7 +6165,22 @@
 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 +6345,6 @@
                                          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 +6352,6 @@
     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 +6466,7 @@
         // 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 +6474,10 @@
 
         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
@@ -6540,7 +6656,7 @@
             }
 
             // sleep if there are no active tracks to process
-            if (activeTracks.size() == 0) {
+            if (activeTracks.isEmpty()) {
                 if (sleepUs == 0) {
                     sleepUs = kRecordThreadSleepUs;
                 }
@@ -6591,6 +6707,14 @@
                 }
                 didModify = true;
             }
+            AudioBufferProvider* abp = (fastTrack != 0 && fastTrack->isPatchTrack()) ?
+                    reinterpret_cast<AudioBufferProvider*>(fastTrack.get()) : nullptr;
+            if (state->mFastPatchRecordBufferProvider != abp) {
+                state->mFastPatchRecordBufferProvider = abp;
+                state->mFastPatchRecordFormat = fastTrack == 0 ?
+                        AUDIO_FORMAT_INVALID : fastTrack->format();
+                didModify = true;
+            }
             sq->end(didModify);
             if (didModify) {
                 sq->push(block);
@@ -6616,8 +6740,7 @@
 
         // If an NBAIO source is present, use it to read the normal capture's data
         if (mPipeSource != 0) {
-            size_t framesToRead = mBufferSize / mFrameSize;
-            framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2);
+            size_t framesToRead = min(mRsmpInFramesOA - rear, mRsmpInFramesP2 / 2);
 
             // The audio fifo read() returns OVERRUN on overflow, and advances the read pointer
             // to the full buffer point (clearing the overflow condition).  Upon OVERRUN error,
@@ -6679,8 +6802,9 @@
         // 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 +6812,12 @@
                 //
                 // 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 +6834,9 @@
         }
         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 +6894,33 @@
                 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;
@@ -6937,6 +7091,12 @@
         goto Exit;
     }
 
+    if (!audio_is_linear_pcm(mFormat) && (*flags & AUDIO_INPUT_FLAG_DIRECT) == 0) {
+        ALOGE("createRecordTrack_l() on an encoded stream requires AUDIO_INPUT_FLAG_DIRECT");
+        lStatus = BAD_VALUE;
+        goto Exit;
+    }
+
     if (*pSampleRate == 0) {
         *pSampleRate = mSampleRate;
     }
@@ -7138,8 +7298,10 @@
         // 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();
@@ -7295,7 +7457,7 @@
     audio_input_flags_t flags = input != NULL ? input->flags : AUDIO_INPUT_FLAG_NONE;
     dprintf(fd, "  AudioStreamIn: %p flags %#x (%s)\n",
             input, flags, inputFlagsToString(flags).c_str());
-    if (mActiveTracks.size() == 0) {
+    if (mActiveTracks.isEmpty()) {
         dprintf(fd, "  No active record clients\n");
     }
 
@@ -7304,6 +7466,13 @@
         (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");
 
@@ -7311,9 +7480,8 @@
     // while we are dumping it.  It may be inconsistent, but it won't mutate!
     // This is a large object so we place it on the heap.
     // FIXME 25972958: Need an intelligent copy constructor that does not touch unused pages.
-    const FastCaptureDumpState *copy = new FastCaptureDumpState(mFastCaptureDumpState);
+    std::unique_ptr<FastCaptureDumpState> copy(new FastCaptureDumpState(mFastCaptureDumpState));
     copy->dump(fd);
-    delete copy;
 }
 
 void AudioFlinger::RecordThread::dumpTracks(int fd, const Vector<String16>& args __unused)
@@ -7327,7 +7495,7 @@
     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 +7515,7 @@
         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) {
@@ -7642,10 +7810,15 @@
 {
     status_t result = mInput->stream->getAudioProperties(&mSampleRate, &mChannelMask, &mHALFormat);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving audio properties from HAL: %d", result);
-    mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
-    LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d", mChannelCount, FCC_8);
     mFormat = mHALFormat;
-    LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat);
+    mChannelCount = audio_channel_count_from_in_mask(mChannelMask);
+    if (audio_is_linear_pcm(mFormat)) {
+        LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_8, "HAL channel count %d > %d",
+                mChannelCount, FCC_8);
+    } else {
+        // Can have more that FCC_8 channels in encoded streams.
+        ALOGI("HAL format %#x is not linear pcm", mFormat);
+    }
     result = mInput->stream->getFrameSize(&mFrameSize);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
     result = mInput->stream->getBufferSize(&mBufferSize);
@@ -7750,7 +7923,7 @@
 status_t AudioFlinger::RecordThread::addEffectChain_l(const sp<EffectChain>& chain)
 {
     // only one chain per input thread
-    if (mEffectChains.size() != 0) {
+    if (!mEffectChains.isEmpty()) {
         ALOGW("addEffectChain_l() already one chain %p on thread %p", chain.get(), this);
         return INVALID_OPERATION;
     }
@@ -7855,24 +8028,28 @@
     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;
+    }
 }
 
 // ----------------------------------------------------------------------------
@@ -7959,7 +8136,7 @@
     }
     // This will decrement references and may cause the destruction of this thread.
     if (isOutput()) {
-        AudioSystem::releaseOutput(mId, streamType(), mSessionId);
+        AudioSystem::releaseOutput(mPortId);
     } else {
         AudioSystem::releaseInput(mPortId);
     }
@@ -8073,7 +8250,7 @@
 
     bool silenced = false;
     if (isOutput()) {
-        ret = AudioSystem::startOutput(mId, streamType(), mSessionId);
+        ret = AudioSystem::startOutput(portId);
     } else {
         ret = AudioSystem::startInput(portId, &silenced);
     }
@@ -8082,10 +8259,10 @@
     // abort if start is rejected by audio policy manager
     if (ret != NO_ERROR) {
         ALOGE("%s: error start rejected by AudioPolicyManager = %d", __FUNCTION__, ret);
-        if (mActiveTracks.size() != 0) {
+        if (!mActiveTracks.isEmpty()) {
             mLock.unlock();
             if (isOutput()) {
-                AudioSystem::releaseOutput(mId, streamType(), mSessionId);
+                AudioSystem::releaseOutput(portId);
             } else {
                 AudioSystem::releaseInput(portId);
             }
@@ -8108,7 +8285,7 @@
 
     // 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);
@@ -8157,8 +8334,8 @@
 
     mLock.unlock();
     if (isOutput()) {
-        AudioSystem::stopOutput(mId, streamType(), track->sessionId());
-        AudioSystem::releaseOutput(mId, streamType(), track->sessionId());
+        AudioSystem::stopOutput(track->portId());
+        AudioSystem::releaseOutput(track->portId());
     } else {
         AudioSystem::stopInput(track->portId());
         AudioSystem::releaseInput(track->portId());
@@ -8183,7 +8360,7 @@
     if (mHalStream == 0) {
         return NO_INIT;
     }
-    if (mActiveTracks.size() != 0) {
+    if (!mActiveTracks.isEmpty()) {
         return INVALID_OPERATION;
     }
     mHalStream->standby();
@@ -8454,9 +8631,9 @@
     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();
@@ -8621,7 +8798,7 @@
     dprintf(fd, "  Attributes: content type %d usage %d source %d\n",
             mAttr.content_type, mAttr.usage, mAttr.source);
     dprintf(fd, "  Session: %d port Id: %d\n", mSessionId, mPortId);
-    if (mActiveTracks.size() == 0) {
+    if (mActiveTracks.isEmpty()) {
         dprintf(fd, "  No active clients\n");
     }
 }
@@ -8634,7 +8811,7 @@
     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 +9017,15 @@
     }
 }
 
+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 +9116,13 @@
     }
 }
 
+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..2ab0bee 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.
@@ -543,6 +555,9 @@
                     size_t          size() const {
                         return mActiveTracks.size();
                     }
+                    bool            isEmpty() const {
+                        return mActiveTracks.isEmpty();
+                    }
                     ssize_t         indexOf(const sp<T>& item) {
                         return mActiveTracks.indexOf(item);
                     }
@@ -646,6 +661,8 @@
     virtual             ~PlaybackThread();
 
                 void        dump(int fd, const Vector<String16>& args);
+                // returns a string of audio performance related data in JSON format.
+    virtual     Json::Value getJsonDump() const;
 
     // Thread virtuals
     virtual     bool        threadLoop();
@@ -782,7 +799,7 @@
                 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 +1071,6 @@
     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;
@@ -1096,6 +1108,7 @@
     virtual     bool        checkForNewParameter_l(const String8& keyValuePair,
                                                    status_t& status);
     virtual     void        dumpInternals(int fd, const Vector<String16>& args);
+                Json::Value getJsonDump() const override;
 
     virtual     bool        isTrackAllowed_l(
                                     audio_channel_mask_t channelMask, audio_format_t format,
@@ -1152,6 +1165,14 @@
                               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);
@@ -1209,6 +1230,23 @@
     virtual     bool        hasFastMixer() const { return false; }
 
     virtual     int64_t     computeWaitTimeNs_l() const override;
+
+    status_t    threadloop_getHalTimestamp_l(ExtendedTimestamp *timestamp) const override {
+                    // For DIRECT and OFFLOAD threads, query the output sink directly.
+                    if (mOutput != nullptr) {
+                        uint64_t uposition64;
+                        struct timespec time;
+                        if (mOutput->getPresentationPosition(
+                                &uposition64, &time) == OK) {
+                            timestamp->mPosition[ExtendedTimestamp::LOCATION_KERNEL]
+                                    = (int64_t)uposition64;
+                            timestamp->mTimeNs[ExtendedTimestamp::LOCATION_KERNEL]
+                                    = audio_utils_ns_from_timespec(&time);
+                            return NO_ERROR;
+                        }
+                    }
+                    return INVALID_OPERATION;
+                }
 };
 
 class OffloadThread : public DirectOutputThread {
@@ -1316,6 +1354,22 @@
     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 +1428,6 @@
                     audio_devices_t outDevice,
                     audio_devices_t inDevice,
                     bool systemReady
-#ifdef TEE_SINK
-                    , const sp<NBAIO_Sink>& teeSink
-#endif
                     );
             virtual     ~RecordThread();
 
@@ -1437,8 +1488,8 @@
                                            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 +1510,7 @@
 
     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);
@@ -1479,6 +1530,8 @@
 
             void        updateMetadata_l() override;
 
+            bool        fastTrackAvailable() const { return mFastTrackAvail; }
+
 private:
             // Enter standby if not already in standby, and set mStandby flag
             void    standbyIfNotAlreadyInStandby();
@@ -1506,8 +1559,6 @@
             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 +1653,7 @@
     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 +1737,8 @@
 
                 void        updateMetadata_l() override;
 
+    virtual     void        toAudioPortConfig(struct audio_port_config *config);
+
 protected:
 
                 audio_stream_type_t         mStreamType;
@@ -1714,6 +1767,8 @@
                 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..8b9485f 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()
@@ -498,18 +486,20 @@
             wasActive = playbackThread->destroyTrack_l(this);
         }
         if (isExternalTrack() && !wasActive) {
-            AudioSystem::releaseOutput(mThreadIoHandle, mStreamType, mSessionId);
+            AudioSystem::releaseOutput(mPortId);
         }
     }
 }
 
-/*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..d4c49d9 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -124,17 +124,11 @@
                                         audio_port_handle_t *selectedDeviceId,
                                         audio_port_handle_t *portId) = 0;
     // indicates to the audio policy manager that the output starts being used by corresponding stream.
-    virtual status_t startOutput(audio_io_handle_t output,
-                                 audio_stream_type_t stream,
-                                 audio_session_t session) = 0;
+    virtual status_t startOutput(audio_port_handle_t portId) = 0;
     // indicates to the audio policy manager that the output stops being used by corresponding stream.
-    virtual status_t stopOutput(audio_io_handle_t output,
-                                audio_stream_type_t stream,
-                                audio_session_t session) = 0;
+    virtual status_t stopOutput(audio_port_handle_t portId) = 0;
     // releases the output.
-    virtual void releaseOutput(audio_io_handle_t output,
-                               audio_stream_type_t stream,
-                               audio_session_t session) = 0;
+    virtual void releaseOutput(audio_port_handle_t portId) = 0;
 
     // request an input appropriate for record from the supplied device with supplied parameters.
     virtual status_t getInputForAttr(const audio_attributes_t *attr,
@@ -147,16 +141,13 @@
                                      input_type_t *inputType,
                                      audio_port_handle_t *portId) = 0;
     // indicates to the audio policy manager that the input starts being used.
-    virtual status_t startInput(audio_io_handle_t input,
-                                audio_session_t session,
+    virtual status_t startInput(audio_port_handle_t portId,
                                 bool silenced,
                                 concurrency_type__mask_t *concurrency) = 0;
     // indicates to the audio policy manager that the input stops being used.
-    virtual status_t stopInput(audio_io_handle_t input,
-                               audio_session_t session) = 0;
+    virtual status_t stopInput(audio_port_handle_t portId) = 0;
     // releases the input.
-    virtual void releaseInput(audio_io_handle_t input,
-                              audio_session_t session) = 0;
+    virtual void releaseInput(audio_port_handle_t portId) = 0;
 
     //
     // volume control functions
@@ -235,9 +226,9 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle,
+                                      audio_port_handle_t *portId,
                                       uid_t uid) = 0;
-    virtual status_t stopAudioSource(audio_patch_handle_t handle) = 0;
+    virtual status_t stopAudioSource(audio_port_handle_t portId) = 0;
 
     virtual status_t setMasterMono(bool mono) = 0;
     virtual status_t getMasterMono(bool *mono) = 0;
@@ -324,11 +315,6 @@
     // 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..9b8f095 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.mk
+++ b/services/audiopolicy/common/managerdefinitions/Android.mk
@@ -18,10 +18,10 @@
     src/EffectDescriptor.cpp \
     src/SoundTriggerSession.cpp \
     src/SessionRoute.cpp \
-    src/AudioSourceDescriptor.cpp \
     src/VolumeCurve.cpp \
     src/TypeConverter.cpp \
-    src/AudioSession.cpp
+    src/AudioSession.cpp \
+    src/ClientDescriptor.cpp
 
 LOCAL_SHARED_LIBRARIES := \
     libcutils \
@@ -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 69%
rename from services/audiopolicy/common/managerdefinitions/include/AudioSessionInfoProvider.h
rename to services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
index e0037fc..555412e 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioSessionInfoProvider.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioIODescriptorInterface.h
@@ -19,26 +19,19 @@
 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;
 
-};
-
-class AudioSessionInfoUpdateListener
-{
-public:
-    virtual ~AudioSessionInfoUpdateListener() {};
-
-    virtual void onSessionInfoUpdate() const = 0;;
+    virtual void setPatchHandle(audio_patch_handle_t handle) = 0;
 };
 
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index b25d6d4..44662e5 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -16,9 +16,10 @@
 
 #pragma once
 
+#include "AudioIODescriptorInterface.h"
 #include "AudioPort.h"
 #include "AudioSession.h"
-#include "AudioSessionInfoProvider.h"
+#include "ClientDescriptor.h"
 #include <utils/Errors.h>
 #include <system/audio.h>
 #include <utils/SortedVector.h>
@@ -31,7 +32,7 @@
 
 // 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,
@@ -66,12 +67,13 @@
     AudioSessionCollection getAudioSessions(bool activeOnly) const;
     size_t getAudioSessionCount(bool activeOnly) const;
     audio_source_t getHighestPrioritySource(bool activeOnly) const;
+    void changeRefCount(audio_session_t session, int delta);
 
-    // 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,
@@ -80,14 +82,20 @@
                   audio_input_flags_t flags,
                   audio_io_handle_t *input);
     // Called when a stream is about to be started.
-    // Note: called after AudioSession::changeActiveCount(1)
+    // Note: called after changeRefCount(session, 1)
     status_t start();
     // Called after a stream is stopped
-    // Note: called after AudioSession::changeActiveCount(-1)
+    // Note: called after changeRefCount(session, -1)
     void stop();
     void close();
 
-private:
+    RecordClientMap& clients() { return mClients; }
+    RecordClientVector getClientsForSession(audio_session_t session);
+
+ private:
+
+    void updateSessionRecordingConfiguration(int event, const sp<AudioSession>& audioSession);
+
     audio_patch_handle_t          mPatchHandle;
     audio_port_handle_t           mId;
     // audio sessions attached to this input
@@ -100,6 +108,9 @@
     // We also inherit sessions from the preempted input to avoid a 3 way preemption loop etc...
     SortedVector<audio_session_t> mPreemptedSessions;
     AudioPolicyClientInterface *mClientInterface;
+    uint32_t mGlobalRefCount;  // non-session-specific ref count
+
+    RecordClientMap mClients;
 };
 
 class AudioInputCollection :
@@ -123,6 +134,8 @@
 
     audio_devices_t getSupportedDevices(audio_io_handle_t handle) const;
 
+    sp<AudioInputDescriptor> getInputForClient(audio_port_handle_t portId);
+
     status_t dump(int fd) const;
 };
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 5e5d38b..ff0201a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -17,14 +17,14 @@
 #pragma once
 
 #include <sys/types.h>
-
-#include "AudioPort.h"
-#include <RoutingStrategy.h>
 #include <utils/Errors.h>
 #include <utils/Timers.h>
 #include <utils/KeyedVector.h>
 #include <system/audio.h>
-#include "AudioSourceDescriptor.h"
+#include <RoutingStrategy.h>
+#include "AudioIODescriptorInterface.h"
+#include "AudioPort.h"
+#include "ClientDescriptor.h"
 
 namespace android {
 
@@ -35,7 +35,7 @@
 
 // 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,
@@ -73,10 +73,14 @@
 
     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;
+    TrackClientMap& clients() { return mClients; }
+
+    sp<AudioPort> mPort;
     audio_devices_t mDevice;                   // current device this output is routed to
     uint32_t mRefCount[AUDIO_STREAM_CNT]; // number of streams of each type using this output
     nsecs_t mStopTime[AUDIO_STREAM_CNT];
@@ -89,6 +93,7 @@
 protected:
     audio_patch_handle_t mPatchHandle;
     audio_port_handle_t mId;
+    TrackClientMap mClients;
 };
 
 // Audio output driven by a software mixer in audio flinger.
@@ -153,7 +158,7 @@
 class HwAudioOutputDescriptor: public AudioOutputDescriptor
 {
 public:
-    HwAudioOutputDescriptor(const sp<AudioSourceDescriptor>& source,
+    HwAudioOutputDescriptor(const sp<SourceClientDescriptor>& source,
                             AudioPolicyClientInterface *clientInterface);
     virtual ~HwAudioOutputDescriptor() {}
 
@@ -170,7 +175,7 @@
                            const struct audio_port_config *srcConfig = NULL) const;
     virtual void toAudioPort(struct audio_port *port) const;
 
-    const sp<AudioSourceDescriptor> mSource;
+    const sp<SourceClientDescriptor> mSource;
 
 };
 
@@ -224,6 +229,8 @@
 
     audio_devices_t getSupportedDevices(audio_io_handle_t handle) const;
 
+    sp<SwAudioOutputDescriptor> getOutputForClient(audio_port_handle_t portId);
+
     status_t dump(int fd) const;
 };
 
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..a1ee708 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
@@ -16,91 +16,84 @@
 
 #pragma once
 
-#include "policy.h"
-#include <utils/String8.h>
-#include <utils/SortedVector.h>
-#include <utils/RefBase.h>
-#include <utils/Errors.h>
+#include <vector>
+
 #include <system/audio.h>
-#include <cutils/config_utils.h>
+#include <utils/RefBase.h>
+#include <utils/SortedVector.h>
+#include <utils/String8.h>
+
+#include "policy.h"
 
 namespace android {
 
 typedef SortedVector<uint32_t> SampleRateVector;
-typedef SortedVector<audio_channel_mask_t> ChannelsVector;
 typedef Vector<audio_format_t> FormatVector;
 
 template <typename T>
-bool operator == (const SortedVector<T> &left, const SortedVector<T> &right);
+bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+    if (left.size() != right.size()) {
+        return false;
+    }
+    for (size_t index = 0; index < right.size(); index++) {
+        if (left[index] != right[index]) {
+            return false;
+        }
+    }
+    return true;
+}
+
+template <typename T>
+bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+    return !(left == right);
+}
+
+class ChannelsVector : public SortedVector<audio_channel_mask_t>
+{
+public:
+    ChannelsVector() = default;
+    ChannelsVector(const ChannelsVector&) = default;
+    ChannelsVector(const SortedVector<audio_channel_mask_t>& sv) :
+            SortedVector<audio_channel_mask_t>(sv) {}
+    ChannelsVector& operator=(const ChannelsVector&) = default;
+
+    // Applies audio_channel_mask_out_to_in to all elements and returns the result.
+    ChannelsVector asInMask() const;
+    // Applies audio_channel_mask_in_to_out to all elements and returns the result.
+    ChannelsVector asOutMask() const;
+};
 
 class AudioProfile : public virtual RefBase
 {
 public:
-    AudioProfile(audio_format_t format,
-                 audio_channel_mask_t channelMasks,
-                 uint32_t samplingRate) :
-        mName(String8("")),
-        mFormat(format)
-    {
-        mChannelMasks.add(channelMasks);
-        mSamplingRates.add(samplingRate);
-    }
+    static sp<AudioProfile> createFullDynamic();
 
+    AudioProfile(audio_format_t format, audio_channel_mask_t channelMasks, uint32_t samplingRate);
     AudioProfile(audio_format_t format,
                  const ChannelsVector &channelMasks,
-                 const SampleRateVector &samplingRateCollection) :
-        mName(String8("")),
-        mFormat(format),
-        mChannelMasks(channelMasks),
-        mSamplingRates(samplingRateCollection)
-    {}
+                 const SampleRateVector &samplingRateCollection);
 
     audio_format_t getFormat() const { return mFormat; }
-
-    void setChannels(const ChannelsVector &channelMasks)
-    {
-        if (mIsDynamicChannels) {
-            mChannelMasks = channelMasks;
-        }
-    }
     const ChannelsVector &getChannels() const { return mChannelMasks; }
-
-    void setSampleRates(const SampleRateVector &sampleRates)
-    {
-        if (mIsDynamicRate) {
-            mSamplingRates = sampleRates;
-        }
-    }
     const SampleRateVector &getSampleRates() const { return mSamplingRates; }
+    void setChannels(const ChannelsVector &channelMasks);
+    void setSampleRates(const SampleRateVector &sampleRates);
 
+    void clear();
     bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
-
-    void clear()
-    {
-        if (mIsDynamicChannels) {
-            mChannelMasks.clear();
-        }
-        if (mIsDynamicRate) {
-            mSamplingRates.clear();
-        }
-    }
-
-    inline bool supportsChannels(audio_channel_mask_t channels) const
+    bool supportsChannels(audio_channel_mask_t channels) const
     {
         return mChannelMasks.indexOf(channels) >= 0;
     }
-    inline bool supportsRate(uint32_t rate) const
-    {
-        return mSamplingRates.indexOf(rate) >= 0;
-    }
+    bool supportsRate(uint32_t rate) const { return mSamplingRates.indexOf(rate) >= 0; }
 
     status_t checkExact(uint32_t rate, audio_channel_mask_t channels, audio_format_t format) const;
-
     status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
                                         audio_channel_mask_t &updatedChannelMask,
                                         audio_port_type_t portType,
                                         audio_port_role_t portRole) const;
-
     status_t checkCompatibleSamplingRate(uint32_t samplingRate,
                                          uint32_t &updatedSamplingRate) const;
 
@@ -136,213 +129,48 @@
 class AudioProfileVector : public Vector<sp<AudioProfile> >
 {
 public:
-    ssize_t add(const sp<AudioProfile> &profile)
-    {
-        ssize_t index = Vector::add(profile);
-        // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
-        // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
-        // [](const audio_format_t *format1, const audio_format_t *format2) {
-        //     return compareFormats(*format1, *format2);
-        // }
-        sort(compareFormats);
-        return index;
-    }
-
+    ssize_t add(const sp<AudioProfile> &profile);
     // This API is intended to be used by the policy manager once retrieving capabilities
     // for a profile with dynamic format, rate and channels attributes
-    ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd)
-    {
-        // Check valid profile to add:
-        if (!profileToAdd->hasValidFormat()) {
-            return -1;
-        }
-        if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
-            FormatVector formats;
-            formats.add(profileToAdd->getFormat());
-            setFormats(FormatVector(formats));
-            return 0;
-        }
-        if (!profileToAdd->hasValidChannels() && profileToAdd->hasValidRates()) {
-            setSampleRatesFor(profileToAdd->getSampleRates(), profileToAdd->getFormat());
-            return 0;
-        }
-        if (profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
-            setChannelsFor(profileToAdd->getChannels(), profileToAdd->getFormat());
-            return 0;
-        }
-        // Go through the list of profile to avoid duplicates
-        for (size_t profileIndex = 0; profileIndex < size(); profileIndex++) {
-            const sp<AudioProfile> &profile = itemAt(profileIndex);
-            if (profile->isValid() && profile == profileToAdd) {
-                // Nothing to do
-                return profileIndex;
-            }
-        }
-        profileToAdd->setDynamicFormat(true); // set the format as dynamic to allow removal
-        return add(profileToAdd);
-    }
-
-    sp<AudioProfile> getFirstValidProfile() const
-    {
-        for (size_t i = 0; i < size(); i++) {
-            if (itemAt(i)->isValid()) {
-                return itemAt(i);
-            }
-        }
-        return 0;
-    }
-
-    bool hasValidProfile() const { return getFirstValidProfile() != 0; }
+    ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd);
 
     status_t checkExactProfile(uint32_t samplingRate, audio_channel_mask_t channelMask,
                                audio_format_t format) const;
-
     status_t checkCompatibleProfile(uint32_t &samplingRate, audio_channel_mask_t &channelMask,
                                     audio_format_t &format,
                                     audio_port_type_t portType,
                                     audio_port_role_t portRole) const;
+    void clearProfiles();
+    // Assuming that this profile vector contains input profiles,
+    // find the best matching config from 'outputProfiles', according to
+    // the given preferences for audio formats and channel masks.
+    // Note: std::vectors are used because specialized containers for formats
+    //       and channels can be sorted and use their own ordering.
+    status_t findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
+            const std::vector<audio_format_t>& preferredFormats, // order: most pref -> least pref
+            const std::vector<audio_channel_mask_t>& preferredOutputChannels,
+            bool preferHigherSamplingRates,
+            audio_config_base *bestOutputConfig) const;
 
-    FormatVector getSupportedFormats() const
-    {
-        FormatVector supportedFormats;
-        for (size_t i = 0; i < size(); i++) {
-            if (itemAt(i)->hasValidFormat()) {
-                supportedFormats.add(itemAt(i)->getFormat());
-            }
-        }
-        return supportedFormats;
-    }
+    sp<AudioProfile> getFirstValidProfile() const;
+    sp<AudioProfile> getFirstValidProfileFor(audio_format_t format) const;
+    bool hasValidProfile() const { return getFirstValidProfile() != 0; }
 
-    bool hasDynamicProfile() const
-    {
-        for (size_t i = 0; i < size(); i++) {
-            if (itemAt(i)->isDynamic()) {
-                return true;
-            }
-        }
-        return false;
-    }
-
-    bool hasDynamicFormat() const
-    {
-        return getProfileFor(gDynamicFormat) != 0;
-    }
-
-    bool hasDynamicChannelsFor(audio_format_t format) const
-    {
-       for (size_t i = 0; i < size(); i++) {
-           sp<AudioProfile> profile = itemAt(i);
-           if (profile->getFormat() == format && profile->isDynamicChannels()) {
-               return true;
-           }
-       }
-       return false;
-    }
-
-    bool hasDynamicRateFor(audio_format_t format) const
-    {
-        for (size_t i = 0; i < size(); i++) {
-            sp<AudioProfile> profile = itemAt(i);
-            if (profile->getFormat() == format && profile->isDynamicRate()) {
-                return true;
-            }
-        }
-        return false;
-    }
+    FormatVector getSupportedFormats() const;
+    bool hasDynamicChannelsFor(audio_format_t format) const;
+    bool hasDynamicFormat() const { return getProfileFor(gDynamicFormat) != 0; }
+    bool hasDynamicProfile() const;
+    bool hasDynamicRateFor(audio_format_t format) const;
 
     // One audio profile will be added for each format supported by Audio HAL
-    void setFormats(const FormatVector &formats)
-    {
-        // Only allow to change the format of dynamic profile
-        sp<AudioProfile> dynamicFormatProfile = getProfileFor(gDynamicFormat);
-        if (dynamicFormatProfile == 0) {
-            return;
-        }
-        for (size_t i = 0; i < formats.size(); i++) {
-            sp<AudioProfile> profile = new AudioProfile(formats[i],
-                                                        dynamicFormatProfile->getChannels(),
-                                                        dynamicFormatProfile->getSampleRates());
-            profile->setDynamicFormat(true);
-            profile->setDynamicChannels(dynamicFormatProfile->isDynamicChannels());
-            profile->setDynamicRate(dynamicFormatProfile->isDynamicRate());
-            add(profile);
-        }
-    }
+    void setFormats(const FormatVector &formats);
 
-    void clearProfiles()
-    {
-        for (size_t i = size(); i != 0; ) {
-            sp<AudioProfile> profile = itemAt(--i);
-            if (profile->isDynamicFormat() && profile->hasValidFormat()) {
-                removeAt(i);
-                continue;
-            }
-            profile->clear();
-        }
-    }
-
-    void dump(int fd, int spaces) const
-    {
-        const size_t SIZE = 256;
-        char buffer[SIZE];
-
-        snprintf(buffer, SIZE, "%*s- Profiles:\n", spaces, "");
-        write(fd, buffer, strlen(buffer));
-        for (size_t i = 0; i < size(); i++) {
-            snprintf(buffer, SIZE, "%*sProfile %zu:", spaces + 4, "", i);
-            write(fd, buffer, strlen(buffer));
-            itemAt(i)->dump(fd, spaces + 8);
-        }
-    }
+    void dump(int fd, int spaces) const;
 
 private:
-    void setSampleRatesFor(const SampleRateVector &sampleRates, audio_format_t format)
-    {
-        for (size_t i = 0; i < size(); i++) {
-            sp<AudioProfile> profile = itemAt(i);
-            if (profile->getFormat() == format && profile->isDynamicRate()) {
-                if (profile->hasValidRates()) {
-                    // Need to create a new profile with same format
-                    sp<AudioProfile> profileToAdd = new AudioProfile(format, profile->getChannels(),
-                                                                     sampleRates);
-                    profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
-                    add(profileToAdd);
-                } else {
-                    profile->setSampleRates(sampleRates);
-                }
-                return;
-            }
-        }
-    }
-
-    void setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format)
-    {
-        for (size_t i = 0; i < size(); i++) {
-            sp<AudioProfile> profile = itemAt(i);
-            if (profile->getFormat() == format && profile->isDynamicChannels()) {
-                if (profile->hasValidChannels()) {
-                    // Need to create a new profile with same format
-                    sp<AudioProfile> profileToAdd = new AudioProfile(format, channelMasks,
-                                                                     profile->getSampleRates());
-                    profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
-                    add(profileToAdd);
-                } else {
-                    profile->setChannels(channelMasks);
-                }
-                return;
-            }
-        }
-    }
-
-    sp<AudioProfile> getProfileFor(audio_format_t format) const
-    {
-        for (size_t i = 0; i < size(); i++) {
-            if (itemAt(i)->getFormat() == format) {
-                return itemAt(i);
-            }
-        }
-        return 0;
-    }
+    sp<AudioProfile> getProfileFor(audio_format_t format) const;
+    void setSampleRatesFor(const SampleRateVector &sampleRates, audio_format_t format);
+    void setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format);
 
     static int compareFormats(const sp<AudioProfile> *profile1, const sp<AudioProfile> *profile2);
 };
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSession.h b/services/audiopolicy/common/managerdefinitions/include/AudioSession.h
index dd5247d..1636d3a 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:
     AudioSession(audio_session_t session,
@@ -39,9 +39,7 @@
                  audio_channel_mask_t channelMask,
                  audio_input_flags_t flags,
                  uid_t uid,
-                 bool isSoundTrigger,
-                 AudioMix* policyMix,
-                 AudioPolicyClientInterface *clientInterface);
+                 bool isSoundTrigger);
 
     status_t dump(int fd, int spaces, int index) const;
 
@@ -50,6 +48,8 @@
     audio_format_t format() const { return mConfig.format; }
     uint32_t sampleRate() const { return mConfig.sample_rate; }
     audio_channel_mask_t channelMask() const { return mConfig.channel_mask; }
+    audio_config_base config() const { return mConfig; }
+    record_client_info_t recordClientInfo() const { return mRecordClientInfo; }
     audio_input_flags_t flags() const { return mFlags; }
     uid_t uid() const { return mRecordClientInfo.uid; }
     void setUid(uid_t uid) { mRecordClientInfo.uid = uid; }
@@ -63,10 +63,6 @@
     uint32_t changeOpenCount(int delta);
     uint32_t changeActiveCount(int delta);
 
-    void setInfoProvider(AudioSessionInfoProvider *provider);
-    // implementation of AudioSessionInfoUpdateListener
-    virtual void onSessionInfoUpdate() const;
-
 private:
     record_client_info_t mRecordClientInfo;
     const struct audio_config_base mConfig;
@@ -75,19 +71,14 @@
     bool mSilenced;
     uint32_t  mOpenCount;
     uint32_t  mActiveCount;
-    AudioMix* mPolicyMix; // non NULL when used by a dynamic policy
-    AudioPolicyClientInterface* mClientInterface;
-    const AudioSessionInfoProvider* mInfoProvider;
 };
 
 class AudioSessionCollection :
-    public DefaultKeyedVector<audio_session_t, sp<AudioSession> >,
-    public AudioSessionInfoUpdateListener
+    public DefaultKeyedVector<audio_session_t, sp<AudioSession> >
 {
 public:
     status_t addSession(audio_session_t session,
-                             const sp<AudioSession>& audioSession,
-                             AudioSessionInfoProvider *provider);
+                             const sp<AudioSession>& audioSession);
 
     status_t removeSession(audio_session_t session);
 
@@ -99,9 +90,6 @@
     bool isSourceActive(audio_source_t source) const;
     audio_source_t getHighestPrioritySource(bool activeOnly) const;
 
-    // implementation of AudioSessionInfoUpdateListener
-    virtual void onSessionInfoUpdate() const;
-
     status_t dump(int fd, int spaces) const;
 };
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioSourceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioSourceDescriptor.h
deleted file mode 100644
index 0d90f42..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/AudioSourceDescriptor.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-#include <system/audio.h>
-#include <utils/Errors.h>
-#include <utils/KeyedVector.h>
-#include <utils/RefBase.h>
-#include <RoutingStrategy.h>
-#include <AudioPatch.h>
-
-namespace android {
-
-class SwAudioOutputDescriptor;
-class HwAudioOutputDescriptor;
-class DeviceDescriptor;
-
-class AudioSourceDescriptor: public RefBase
-{
-public:
-    AudioSourceDescriptor(const sp<DeviceDescriptor> device, const audio_attributes_t *attributes,
-                          uid_t uid) :
-        mDevice(device), mAttributes(*attributes), mUid(uid) {}
-    virtual ~AudioSourceDescriptor() {}
-
-    audio_patch_handle_t getHandle() const { return mPatchDesc->mHandle; }
-
-    status_t    dump(int fd);
-
-    const sp<DeviceDescriptor> mDevice;
-    const audio_attributes_t mAttributes;
-    uid_t mUid;
-    sp<AudioPatch> mPatchDesc;
-    wp<SwAudioOutputDescriptor> mSwOutput;
-    wp<HwAudioOutputDescriptor> mHwOutput;
-};
-
-class AudioSourceCollection :
-        public DefaultKeyedVector< audio_patch_handle_t, sp<AudioSourceDescriptor> >
-{
-public:
-    status_t dump(int fd) const;
-};
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
new file mode 100644
index 0000000..9efe57f
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -0,0 +1,154 @@
+/*
+ * 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.
+ */
+
+#pragma once
+
+#include <vector>
+#include <map>
+#include <unistd.h>
+#include <sys/types.h>
+
+#include <system/audio.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <utils/RefBase.h>
+#include <utils/String8.h>
+#include "AudioPatch.h"
+
+namespace android {
+
+class DeviceDescriptor;
+class HwAudioOutputDescriptor;
+class SwAudioOutputDescriptor;
+
+class ClientDescriptor: public RefBase
+{
+public:
+    ClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_session_t sessionId,
+                   audio_attributes_t attributes, audio_config_base_t config,
+                   audio_port_handle_t preferredDeviceId) :
+        mPortId(portId), mUid(uid), mSessionId(sessionId), mAttributes(attributes),
+        mConfig(config), mPreferredDeviceId(preferredDeviceId), mActive(false) {}
+    ~ClientDescriptor() override = default;
+
+    status_t dump(int fd, int spaces, int index);
+    virtual status_t dump(String8& dst, int spaces, int index);
+
+    audio_port_handle_t portId() const { return mPortId; }
+    uid_t uid() const { return mUid; }
+    audio_session_t session() const { return mSessionId; };
+    audio_attributes_t attributes() const { return mAttributes; }
+    audio_config_base_t config() const { return mConfig; }
+    audio_port_handle_t preferredDeviceId() const { return mPreferredDeviceId; };
+    void setActive(bool active) { mActive = active; }
+    bool active() const { return mActive; }
+
+private:
+    const audio_port_handle_t mPortId;  // unique Id for this client
+    const uid_t mUid;                     // client UID
+    const audio_session_t mSessionId;       // audio session ID
+    const audio_attributes_t mAttributes; // usage...
+    const audio_config_base_t mConfig;
+    const audio_port_handle_t mPreferredDeviceId;  // selected input device port ID
+          bool mActive;
+
+protected:
+    // FIXME: use until other descriptor classes have a dump to String8 method
+    int mDumpFd;
+};
+
+class TrackClientDescriptor: public ClientDescriptor
+{
+public:
+    TrackClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_session_t sessionId,
+                   audio_attributes_t attributes, audio_config_base_t config,
+                   audio_port_handle_t preferredDeviceId,
+                   audio_stream_type_t stream, audio_output_flags_t flags) :
+        ClientDescriptor(portId, uid, sessionId, attributes, config, preferredDeviceId),
+        mStream(stream), mFlags(flags) {}
+    ~TrackClientDescriptor() override = default;
+
+    using ClientDescriptor::dump;
+    status_t dump(String8& dst, int spaces, int index) override;
+
+    audio_output_flags_t flags() const { return mFlags; }
+    audio_stream_type_t stream() const { return mStream; }
+
+private:
+    const audio_stream_type_t mStream;
+    const audio_output_flags_t mFlags;
+};
+
+class RecordClientDescriptor: public ClientDescriptor
+{
+public:
+    RecordClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_session_t sessionId,
+                        audio_attributes_t attributes, audio_config_base_t config,
+                        audio_port_handle_t preferredDeviceId,
+                        audio_source_t source, audio_input_flags_t flags) :
+        ClientDescriptor(portId, uid, sessionId, attributes, config, preferredDeviceId),
+        mSource(source), mFlags(flags) {}
+    ~RecordClientDescriptor() override = default;
+
+    using ClientDescriptor::dump;
+    status_t dump(String8& dst, int spaces, int index) override;
+
+    audio_source_t source() const { return mSource; }
+    audio_input_flags_t flags() const { return mFlags; }
+
+private:
+    const audio_source_t mSource;
+    const audio_input_flags_t mFlags;
+};
+
+class SourceClientDescriptor: public TrackClientDescriptor
+{
+public:
+    SourceClientDescriptor(audio_port_handle_t portId, uid_t uid, audio_attributes_t attributes,
+                           const sp<AudioPatch>& patchDesc, const sp<DeviceDescriptor>& srcDevice,
+                           audio_stream_type_t stream);
+    ~SourceClientDescriptor() override = default;
+
+    sp<AudioPatch> patchDesc() const { return mPatchDesc; }
+    sp<DeviceDescriptor> srcDevice() const { return mSrcDevice; };
+    wp<SwAudioOutputDescriptor> swOutput() const { return mSwOutput; }
+    void setSwOutput(const sp<SwAudioOutputDescriptor>& swOutput);
+    wp<HwAudioOutputDescriptor> hwOutput() const { return mHwOutput; }
+    void setHwOutput(const sp<HwAudioOutputDescriptor>& hwOutput);
+
+    using ClientDescriptor::dump;
+    status_t dump(String8& dst, int spaces, int index) override;
+
+ private:
+    const sp<AudioPatch> mPatchDesc;
+    const sp<DeviceDescriptor> mSrcDevice;
+    wp<SwAudioOutputDescriptor> mSwOutput;
+    wp<HwAudioOutputDescriptor> mHwOutput;
+};
+
+class SourceClientCollection :
+    public DefaultKeyedVector< audio_port_handle_t, sp<SourceClientDescriptor> >
+{
+public:
+    status_t dump(int fd) const;
+};
+
+typedef std::vector< sp<TrackClientDescriptor> > TrackClientVector;
+typedef std::map< audio_port_handle_t, sp<TrackClientDescriptor> > TrackClientMap;
+typedef std::vector< sp<RecordClientDescriptor> > RecordClientVector;
+typedef std::map< audio_port_handle_t, sp<RecordClientDescriptor> > RecordClientMap;
+
+} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 92a4c3e..c08e752 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -76,13 +76,14 @@
 
     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;
+    DeviceVector getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
+    audio_devices_t getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const;
 
     status_t dump(int fd, const String8 &tag, int spaces = 0, bool verbose = true) const;
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index 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..2770e74 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -32,7 +32,7 @@
     : mIoHandle(0),
       mDevice(AUDIO_DEVICE_NONE), mPolicyMix(NULL),
       mProfile(profile), mPatchHandle(AUDIO_PATCH_HANDLE_NONE), mId(0),
-      mClientInterface(clientInterface)
+      mClientInterface(clientInterface), mGlobalRefCount(0)
 {
     if (profile != NULL) {
         profile->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
@@ -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);
 }
 
 status_t AudioInputDescriptor::removeAudioSession(audio_session_t session) {
@@ -179,7 +179,11 @@
 void AudioInputDescriptor::setPatchHandle(audio_patch_handle_t handle)
 {
     mPatchHandle = handle;
-    mSessions.onSessionInfoUpdate();
+    for (size_t i = 0; i < mSessions.size(); i++) {
+        if (mSessions[i]->activeCount() > 0) {
+            updateSessionRecordingConfiguration(RECORD_CONFIG_EVENT_START, mSessions[i]);
+        }
+    }
 }
 
 audio_config_base_t AudioInputDescriptor::getConfig() const
@@ -266,7 +270,7 @@
         LOG_ALWAYS_FATAL_IF(mProfile->curOpenCount < 1, "%s profile open count %u",
                             __FUNCTION__, mProfile->curOpenCount);
         // do not call stop() here as stop() is supposed to be called after
-        // AudioSession::changeActiveCount(-1) and we don't know how many sessions
+        //  changeRefCount(session, -1) and we don't know how many sessions
         // are still active at this time
         if (isActive()) {
             mProfile->curActiveCount--;
@@ -276,6 +280,78 @@
     }
 }
 
+void AudioInputDescriptor::changeRefCount(audio_session_t session, int delta)
+{
+    sp<AudioSession> audioSession = mSessions.valueFor(session);
+    if (audioSession == 0) {
+        return;
+    }
+    // handle session-independent ref count
+    uint32_t oldGlobalRefCount = mGlobalRefCount;
+    if ((delta + (int)mGlobalRefCount) < 0) {
+        ALOGW("changeRefCount() invalid delta %d globalRefCount %d", delta, mGlobalRefCount);
+        delta = -((int)mGlobalRefCount);
+    }
+    mGlobalRefCount += delta;
+    if ((oldGlobalRefCount == 0) && (mGlobalRefCount > 0)) {
+        if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
+        {
+            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mDeviceAddress,
+                                                            MIX_STATE_MIXING);
+        }
+
+    } else if ((oldGlobalRefCount > 0) && (mGlobalRefCount == 0)) {
+        if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
+        {
+            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mDeviceAddress,
+                                                            MIX_STATE_IDLE);
+        }
+    }
+
+    uint32_t oldActiveCount = audioSession->activeCount();
+    if ((delta + (int)oldActiveCount) < 0) {
+        ALOGW("changeRefCount() invalid delta %d for sesion %d active count %d",
+              delta, session, oldActiveCount);
+        delta = -((int)oldActiveCount);
+    }
+
+    audioSession->changeActiveCount(delta);
+
+    int event = RECORD_CONFIG_EVENT_NONE;
+    if ((oldActiveCount == 0) && (audioSession->activeCount() > 0)) {
+        event = RECORD_CONFIG_EVENT_START;
+    } else if ((oldActiveCount > 0) && (audioSession->activeCount() == 0)) {
+        event = RECORD_CONFIG_EVENT_STOP;
+    }
+    if (event != RECORD_CONFIG_EVENT_NONE) {
+        updateSessionRecordingConfiguration(event, audioSession);
+    }
+
+}
+
+void AudioInputDescriptor::updateSessionRecordingConfiguration(
+    int event, const sp<AudioSession>& audioSession) {
+
+    const audio_config_base_t sessionConfig = audioSession->config();
+    const record_client_info_t recordClientInfo = audioSession->recordClientInfo();
+    const audio_config_base_t config = getConfig();
+    mClientInterface->onRecordingConfigurationUpdate(event,
+                                                     &recordClientInfo, &sessionConfig,
+                                                     &config, mPatchHandle);
+}
+
+RecordClientVector AudioInputDescriptor::getClientsForSession(
+    audio_session_t session)
+{
+    RecordClientVector clients;
+    for (const auto &client : mClients) {
+        if (client.second->session() == session) {
+            clients.push_back(client.second);
+        }
+    }
+    return clients;
+}
+
 status_t AudioInputDescriptor::dump(int fd)
 {
     const size_t SIZE = 256;
@@ -297,6 +373,13 @@
 
     mSessions.dump(fd, 1);
 
+    size_t index = 0;
+    result = " AudioRecord clients:\n";
+    for (const auto& client: mClients) {
+        client.second->dump(result, 2, index++);
+    }
+    result.append(" \n");
+    write(fd, result.string(), result.size());
     return NO_ERROR;
 }
 
@@ -359,6 +442,19 @@
     return devices;
 }
 
+sp<AudioInputDescriptor> AudioInputCollection::getInputForClient(audio_port_handle_t portId)
+{
+    for (size_t i = 0; i < size(); i++) {
+        sp<AudioInputDescriptor> inputDesc = valueAt(i);
+        for (const auto& client : inputDesc->clients()) {
+            if (client.second->portId() == portId) {
+                return inputDesc;
+            }
+        }
+    }
+    return 0;
+}
+
 status_t AudioInputCollection::dump(int fd) const
 {
     const size_t SIZE = 256;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 294a2a6..39fce4d 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;
@@ -207,6 +224,13 @@
                  i, mCurVolume[i], mRefCount[i], mMuteCount[i]);
         result.append(buffer);
     }
+
+    result.append(" AudioTrack clients:\n");
+    size_t index = 0;
+    for (const auto& client : mClients) {
+        client.second->dump(result, 2, index++);
+    }
+    result.append(" \n");
     write(fd, result.string(), result.size());
 
     return NO_ERROR;
@@ -534,9 +558,9 @@
 }
 
 // HwAudioOutputDescriptor implementation
-HwAudioOutputDescriptor::HwAudioOutputDescriptor(const sp<AudioSourceDescriptor>& source,
+HwAudioOutputDescriptor::HwAudioOutputDescriptor(const sp<SourceClientDescriptor>& source,
                                                  AudioPolicyClientInterface *clientInterface)
-    : AudioOutputDescriptor(source->mDevice, clientInterface),
+    : AudioOutputDescriptor(source->srcDevice(), clientInterface),
       mSource(source)
 {
 }
@@ -552,7 +576,7 @@
     snprintf(buffer, SIZE, "Source:\n");
     result.append(buffer);
     write(fd, result.string(), result.size());
-    mSource->dump(fd);
+    mSource->dump(fd, 0, 0);
 
     return NO_ERROR;
 }
@@ -566,13 +590,13 @@
                                                  struct audio_port_config *dstConfig,
                                                  const struct audio_port_config *srcConfig) const
 {
-    mSource->mDevice->toAudioPortConfig(dstConfig, srcConfig);
+    mSource->srcDevice()->toAudioPortConfig(dstConfig, srcConfig);
 }
 
 void HwAudioOutputDescriptor::toAudioPort(
                                                     struct audio_port *port) const
 {
-    mSource->mDevice->toAudioPort(port);
+    mSource->srcDevice()->toAudioPort(port);
 }
 
 
@@ -714,6 +738,18 @@
     return devices;
 }
 
+sp<SwAudioOutputDescriptor> SwAudioOutputCollection::getOutputForClient(audio_port_handle_t portId)
+{
+    for (size_t i = 0; i < size(); i++) {
+        sp<SwAudioOutputDescriptor> outputDesc = valueAt(i);
+        for (const auto& client : outputDesc->clients()) {
+            if (client.second->portId() == portId) {
+                return outputDesc;
+            }
+        }
+    }
+    return 0;
+}
 
 status_t SwAudioOutputCollection::dump(int fd) const
 {
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..d04beec 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
@@ -14,20 +14,112 @@
  * limitations under the License.
  */
 
+#include <algorithm>
+#include <set>
+#include <string>
+
 #define LOG_TAG "APM::AudioProfile"
 //#define LOG_NDEBUG 0
 
-#include "AudioProfile.h"
-#include "AudioPort.h"
-#include "HwModule.h"
-#include "AudioGain.h"
-#include <utils/SortedVector.h>
-#include "TypeConverter.h"
 #include <media/AudioResamplerPublic.h>
-#include <algorithm>
+#include <utils/Errors.h>
+
+#include "AudioGain.h"
+#include "AudioPort.h"
+#include "AudioProfile.h"
+#include "HwModule.h"
+#include "TypeConverter.h"
 
 namespace android {
 
+ChannelsVector ChannelsVector::asInMask() const
+{
+    ChannelsVector inMaskVector;
+    for (const auto& channel : *this) {
+        if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
+            inMaskVector.add(audio_channel_mask_out_to_in(channel));
+        }
+    }
+    return inMaskVector;
+}
+
+ChannelsVector ChannelsVector::asOutMask() const
+{
+    ChannelsVector outMaskVector;
+    for (const auto& channel : *this) {
+        if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
+            outMaskVector.add(audio_channel_mask_in_to_out(channel));
+        }
+    }
+    return outMaskVector;
+}
+
+bool operator == (const AudioProfile &left, const AudioProfile &compareTo)
+{
+    return (left.getFormat() == compareTo.getFormat()) &&
+            (left.getChannels() == compareTo.getChannels()) &&
+            (left.getSampleRates() == compareTo.getSampleRates());
+}
+
+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;
+}
+
+AudioProfile::AudioProfile(audio_format_t format,
+                           audio_channel_mask_t channelMasks,
+                           uint32_t samplingRate) :
+        mName(String8("")),
+        mFormat(format)
+{
+    mChannelMasks.add(channelMasks);
+    mSamplingRates.add(samplingRate);
+}
+
+AudioProfile::AudioProfile(audio_format_t format,
+                           const ChannelsVector &channelMasks,
+                           const SampleRateVector &samplingRateCollection) :
+        mName(String8("")),
+        mFormat(format),
+        mChannelMasks(channelMasks),
+        mSamplingRates(samplingRateCollection) {}
+
+void AudioProfile::setChannels(const ChannelsVector &channelMasks)
+{
+    if (mIsDynamicChannels) {
+        mChannelMasks = channelMasks;
+    }
+}
+
+void AudioProfile::setSampleRates(const SampleRateVector &sampleRates)
+{
+    if (mIsDynamicRate) {
+        mSamplingRates = sampleRates;
+    }
+}
+
+void AudioProfile::clear()
+{
+    if (mIsDynamicChannels) {
+        mChannelMasks.clear();
+    }
+    if (mIsDynamicRate) {
+        mSamplingRates.clear();
+    }
+}
+
 status_t AudioProfile::checkExact(uint32_t samplingRate, audio_channel_mask_t channelMask,
                                   audio_format_t format) const
 {
@@ -39,27 +131,6 @@
     return BAD_VALUE;
 }
 
-template <typename T>
-bool operator == (const SortedVector<T> &left, const SortedVector<T> &right)
-{
-    if (left.size() != right.size()) {
-        return false;
-    }
-    for(size_t index = 0; index < right.size(); index++) {
-        if (left[index] != right[index]) {
-            return false;
-        }
-    }
-    return true;
-}
-
-bool operator == (const AudioProfile &left, const AudioProfile &compareTo)
-{
-    return (left.getFormat() == compareTo.getFormat()) &&
-            (left.getChannels() == compareTo.getChannels()) &&
-            (left.getSampleRates() == compareTo.getSampleRates());
-}
-
 status_t AudioProfile::checkCompatibleSamplingRate(uint32_t samplingRate,
                                                    uint32_t &updatedSamplingRate) const
 {
@@ -225,6 +296,50 @@
     write(fd, result.string(), result.size());
 }
 
+ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
+{
+    ssize_t index = Vector::add(profile);
+    // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
+    // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
+    // [](const audio_format_t *format1, const audio_format_t *format2) {
+    //     return compareFormats(*format1, *format2);
+    // }
+    sort(compareFormats);
+    return index;
+}
+
+ssize_t AudioProfileVector::addProfileFromHal(const sp<AudioProfile> &profileToAdd)
+{
+    // Check valid profile to add:
+    if (!profileToAdd->hasValidFormat()) {
+        return -1;
+    }
+    if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
+        FormatVector formats;
+        formats.add(profileToAdd->getFormat());
+        setFormats(FormatVector(formats));
+        return 0;
+    }
+    if (!profileToAdd->hasValidChannels() && profileToAdd->hasValidRates()) {
+        setSampleRatesFor(profileToAdd->getSampleRates(), profileToAdd->getFormat());
+        return 0;
+    }
+    if (profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
+        setChannelsFor(profileToAdd->getChannels(), profileToAdd->getFormat());
+        return 0;
+    }
+    // Go through the list of profile to avoid duplicates
+    for (size_t profileIndex = 0; profileIndex < size(); profileIndex++) {
+        const sp<AudioProfile> &profile = itemAt(profileIndex);
+        if (profile->isValid() && profile == profileToAdd) {
+            // Nothing to do
+            return profileIndex;
+        }
+    }
+    profileToAdd->setDynamicFormat(true); // set the format as dynamic to allow removal
+    return add(profileToAdd);
+}
+
 status_t AudioProfileVector::checkExactProfile(uint32_t samplingRate,
                                                audio_channel_mask_t channelMask,
                                                audio_format_t format) const
@@ -281,6 +396,233 @@
     return BAD_VALUE;
 }
 
+void AudioProfileVector::clearProfiles()
+{
+    for (size_t i = size(); i != 0; ) {
+        sp<AudioProfile> profile = itemAt(--i);
+        if (profile->isDynamicFormat() && profile->hasValidFormat()) {
+            removeAt(i);
+            continue;
+        }
+        profile->clear();
+    }
+}
+
+// Returns an intersection between two possibly unsorted vectors and the contents of 'order'.
+// The result is ordered according to 'order'.
+template<typename T, typename Order>
+std::vector<typename T::value_type> intersectFilterAndOrder(
+        const T& input1, const T& input2, const Order& order)
+{
+    std::set<typename T::value_type> set1{input1.begin(), input1.end()};
+    std::set<typename T::value_type> set2{input2.begin(), input2.end()};
+    std::set<typename T::value_type> common;
+    std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
+            std::inserter(common, common.begin()));
+    std::vector<typename T::value_type> result;
+    for (const auto& e : order) {
+        if (common.find(e) != common.end()) result.push_back(e);
+    }
+    return result;
+}
+
+// Intersect two possibly unsorted vectors, return common elements according to 'comp' ordering.
+// 'comp' is a comparator function.
+template<typename T, typename Compare>
+std::vector<typename T::value_type> intersectAndOrder(
+        const T& input1, const T& input2, Compare comp)
+{
+    std::set<typename T::value_type, Compare> set1{input1.begin(), input1.end(), comp};
+    std::set<typename T::value_type, Compare> set2{input2.begin(), input2.end(), comp};
+    std::vector<typename T::value_type> result;
+    std::set_intersection(set1.begin(), set1.end(), set2.begin(), set2.end(),
+            std::back_inserter(result), comp);
+    return result;
+}
+
+status_t AudioProfileVector::findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
+            const std::vector<audio_format_t>& preferredFormats,
+            const std::vector<audio_channel_mask_t>& preferredOutputChannels,
+            bool preferHigherSamplingRates,
+            audio_config_base *bestOutputConfig) const
+{
+    auto formats = intersectFilterAndOrder(getSupportedFormats(),
+            outputProfiles.getSupportedFormats(), preferredFormats);
+    // Pick the best compatible profile.
+    for (const auto& f : formats) {
+        sp<AudioProfile> inputProfile = getFirstValidProfileFor(f);
+        sp<AudioProfile> outputProfile = outputProfiles.getFirstValidProfileFor(f);
+        if (inputProfile == nullptr || outputProfile == nullptr) {
+            continue;
+        }
+        auto channels = intersectFilterAndOrder(inputProfile->getChannels().asOutMask(),
+                outputProfile->getChannels(), preferredOutputChannels);
+        if (channels.empty()) {
+            continue;
+        }
+        auto sampleRates = preferHigherSamplingRates ?
+                intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
+                        std::greater<typename SampleRateVector::value_type>()) :
+                intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
+                        std::less<typename SampleRateVector::value_type>());
+        if (sampleRates.empty()) {
+            continue;
+        }
+        ALOGD("%s() found channel mask %#x and sample rate %d for format %#x.",
+                __func__, *channels.begin(), *sampleRates.begin(), f);
+        bestOutputConfig->format = f;
+        bestOutputConfig->sample_rate = *sampleRates.begin();
+        bestOutputConfig->channel_mask = *channels.begin();
+        return NO_ERROR;
+    }
+    return BAD_VALUE;
+}
+
+sp<AudioProfile> AudioProfileVector::getFirstValidProfile() const
+{
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->isValid()) {
+            return itemAt(i);
+        }
+    }
+    return 0;
+}
+
+sp<AudioProfile> AudioProfileVector::getFirstValidProfileFor(audio_format_t format) const
+{
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->isValid() && itemAt(i)->getFormat() == format) {
+            return itemAt(i);
+        }
+    }
+    return 0;
+}
+
+FormatVector AudioProfileVector::getSupportedFormats() const
+{
+    FormatVector supportedFormats;
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->hasValidFormat()) {
+            supportedFormats.add(itemAt(i)->getFormat());
+        }
+    }
+    return supportedFormats;
+}
+
+bool AudioProfileVector::hasDynamicChannelsFor(audio_format_t format) const
+{
+    for (size_t i = 0; i < size(); i++) {
+        sp<AudioProfile> profile = itemAt(i);
+        if (profile->getFormat() == format && profile->isDynamicChannels()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioProfileVector::hasDynamicProfile() const
+{
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->isDynamic()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioProfileVector::hasDynamicRateFor(audio_format_t format) const
+{
+    for (size_t i = 0; i < size(); i++) {
+        sp<AudioProfile> profile = itemAt(i);
+        if (profile->getFormat() == format && profile->isDynamicRate()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+void AudioProfileVector::setFormats(const FormatVector &formats)
+{
+    // Only allow to change the format of dynamic profile
+    sp<AudioProfile> dynamicFormatProfile = getProfileFor(gDynamicFormat);
+    if (dynamicFormatProfile == 0) {
+        return;
+    }
+    for (size_t i = 0; i < formats.size(); i++) {
+        sp<AudioProfile> profile = new AudioProfile(formats[i],
+                dynamicFormatProfile->getChannels(),
+                dynamicFormatProfile->getSampleRates());
+        profile->setDynamicFormat(true);
+        profile->setDynamicChannels(dynamicFormatProfile->isDynamicChannels());
+        profile->setDynamicRate(dynamicFormatProfile->isDynamicRate());
+        add(profile);
+    }
+}
+
+void AudioProfileVector::dump(int fd, int spaces) const
+{
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+
+    snprintf(buffer, SIZE, "%*s- Profiles:\n", spaces, "");
+    write(fd, buffer, strlen(buffer));
+    for (size_t i = 0; i < size(); i++) {
+        snprintf(buffer, SIZE, "%*sProfile %zu:", spaces + 4, "", i);
+        write(fd, buffer, strlen(buffer));
+        itemAt(i)->dump(fd, spaces + 8);
+    }
+}
+
+sp<AudioProfile> AudioProfileVector::getProfileFor(audio_format_t format) const
+{
+    for (size_t i = 0; i < size(); i++) {
+        if (itemAt(i)->getFormat() == format) {
+            return itemAt(i);
+        }
+    }
+    return 0;
+}
+
+void AudioProfileVector::setSampleRatesFor(
+        const SampleRateVector &sampleRates, audio_format_t format)
+{
+    for (size_t i = 0; i < size(); i++) {
+        sp<AudioProfile> profile = itemAt(i);
+        if (profile->getFormat() == format && profile->isDynamicRate()) {
+            if (profile->hasValidRates()) {
+                // Need to create a new profile with same format
+                sp<AudioProfile> profileToAdd = new AudioProfile(format, profile->getChannels(),
+                        sampleRates);
+                profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
+                add(profileToAdd);
+            } else {
+                profile->setSampleRates(sampleRates);
+            }
+            return;
+        }
+    }
+}
+
+void AudioProfileVector::setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format)
+{
+    for (size_t i = 0; i < size(); i++) {
+        sp<AudioProfile> profile = itemAt(i);
+        if (profile->getFormat() == format && profile->isDynamicChannels()) {
+            if (profile->hasValidChannels()) {
+                // Need to create a new profile with same format
+                sp<AudioProfile> profileToAdd = new AudioProfile(format, channelMasks,
+                        profile->getSampleRates());
+                profileToAdd->setDynamicFormat(true); // need to set to allow cleaning
+                add(profileToAdd);
+            } else {
+                profile->setChannels(channelMasks);
+            }
+            return;
+        }
+    }
+}
+
+// static
 int AudioProfileVector::compareFormats(const sp<AudioProfile> *profile1,
                                        const sp<AudioProfile> *profile2)
 {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
index 7cda46b..5ea4c92 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioSession.cpp
@@ -35,14 +35,11 @@
                            audio_channel_mask_t channelMask,
                            audio_input_flags_t flags,
                            uid_t uid,
-                           bool isSoundTrigger,
-                           AudioMix* policyMix,
-                           AudioPolicyClientInterface *clientInterface) :
+                           bool isSoundTrigger) :
     mRecordClientInfo({ .uid = uid, .session = session, .source = inputSource}),
     mConfig({ .format = format, .sample_rate = sampleRate, .channel_mask = channelMask}),
     mFlags(flags), mIsSoundTrigger(isSoundTrigger),
-    mOpenCount(1), mActiveCount(0), mPolicyMix(policyMix), mClientInterface(clientInterface),
-    mInfoProvider(NULL)
+    mOpenCount(1), mActiveCount(0)
 {
 }
 
@@ -60,7 +57,6 @@
 
 uint32_t AudioSession::changeActiveCount(int delta)
 {
-    const uint32_t oldActiveCount = mActiveCount;
     if ((delta + (int)mActiveCount) < 0) {
         ALOGW("%s invalid delta %d, active count %d",
               __FUNCTION__, delta, mActiveCount);
@@ -68,34 +64,6 @@
     }
     mActiveCount += delta;
     ALOGV("%s active count %d", __FUNCTION__, mActiveCount);
-    int event = RECORD_CONFIG_EVENT_NONE;
-
-    if ((oldActiveCount == 0) && (mActiveCount > 0)) {
-        event = RECORD_CONFIG_EVENT_START;
-    } else if ((oldActiveCount > 0) && (mActiveCount == 0)) {
-        event = RECORD_CONFIG_EVENT_STOP;
-    }
-
-    if (event != RECORD_CONFIG_EVENT_NONE) {
-        // Dynamic policy callback:
-        // if input maps to a dynamic policy with an activity listener, notify of state change
-        if ((mPolicyMix != NULL) && ((mPolicyMix->mCbFlags & AudioMix::kCbFlagNotifyActivity) != 0))
-        {
-            mClientInterface->onDynamicPolicyMixStateUpdate(mPolicyMix->mDeviceAddress,
-                    (event == RECORD_CONFIG_EVENT_START) ? MIX_STATE_MIXING : MIX_STATE_IDLE);
-        }
-
-        // Recording configuration callback:
-        const AudioSessionInfoProvider* 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() :
-                AUDIO_PATCH_HANDLE_NONE;
-        if (patchHandle != AUDIO_PATCH_HANDLE_NONE) {
-            mClientInterface->onRecordingConfigurationUpdate(event, &mRecordClientInfo,
-                    &mConfig, &deviceConfig, patchHandle);
-        }
-    }
 
     return mActiveCount;
 }
@@ -114,27 +82,6 @@
     return false;
 }
 
-void AudioSession::setInfoProvider(AudioSessionInfoProvider *provider)
-{
-    mInfoProvider = provider;
-}
-
-void AudioSession::onSessionInfoUpdate() const
-{
-    if (mActiveCount > 0) {
-        // resend the callback after requerying the informations from the info provider
-        const AudioSessionInfoProvider* 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() :
-                AUDIO_PATCH_HANDLE_NONE;
-        if (patchHandle != AUDIO_PATCH_HANDLE_NONE) {
-            mClientInterface->onRecordingConfigurationUpdate(RECORD_CONFIG_EVENT_START,
-                    &mRecordClientInfo, &mConfig, &deviceConfig, patchHandle);
-        }
-    }
-}
-
 status_t AudioSession::dump(int fd, int spaces, int index) const
 {
     const size_t SIZE = 256;
@@ -169,8 +116,7 @@
 }
 
 status_t AudioSessionCollection::addSession(audio_session_t session,
-                                         const sp<AudioSession>& audioSession,
-                                         AudioSessionInfoProvider *provider)
+                                         const sp<AudioSession>& audioSession)
 {
     ssize_t index = indexOfKey(session);
 
@@ -178,7 +124,6 @@
         ALOGW("addSession() session %d already in", session);
         return ALREADY_EXISTS;
     }
-    audioSession->setInfoProvider(provider);
     add(session, audioSession);
     ALOGV("addSession() session %d  client %d source %d",
             session, audioSession->uid(), audioSession->inputSource());
@@ -194,7 +139,6 @@
         return ALREADY_EXISTS;
     }
     ALOGV("removeSession() session %d", session);
-    valueAt(index)->setInfoProvider(NULL);
     removeItemsAt(index);
     return NO_ERROR;
 }
@@ -271,13 +215,6 @@
     return source;
 }
 
-void AudioSessionCollection::onSessionInfoUpdate() const
-{
-    for (size_t i = 0; i < size(); i++) {
-        valueAt(i)->onSessionInfoUpdate();
-    }
-}
-
 status_t AudioSessionCollection::dump(int fd, int spaces) const
 {
     const size_t SIZE = 256;
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioSourceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioSourceDescriptor.cpp
deleted file mode 100644
index ba33e57..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioSourceDescriptor.cpp
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "APM::AudioSourceDescriptor"
-//#define LOG_NDEBUG 0
-
-#include <utils/Log.h>
-#include <utils/String8.h>
-#include <media/AudioPolicyHelper.h>
-#include <HwModule.h>
-#include <AudioGain.h>
-#include <AudioSourceDescriptor.h>
-#include <DeviceDescriptor.h>
-#include <IOProfile.h>
-#include <AudioOutputDescriptor.h>
-
-namespace android {
-
-status_t AudioSourceDescriptor::dump(int fd)
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-    String8 result;
-
-    snprintf(buffer, SIZE, "mStream: %d\n", audio_attributes_to_stream_type(&mAttributes));
-    result.append(buffer);
-    snprintf(buffer, SIZE, "mDevice:\n");
-    result.append(buffer);
-    write(fd, result.string(), result.size());
-    mDevice->dump(fd, 2 , 0);
-    return NO_ERROR;
-}
-
-
-status_t AudioSourceCollection::dump(int fd) const
-{
-    const size_t SIZE = 256;
-    char buffer[SIZE];
-
-    snprintf(buffer, SIZE, "\nAudio sources dump:\n");
-    write(fd, buffer, strlen(buffer));
-    for (size_t i = 0; i < size(); i++) {
-        snprintf(buffer, SIZE, "- Source %d dump:\n", keyAt(i));
-        write(fd, buffer, strlen(buffer));
-        valueAt(i)->dump(fd);
-    }
-
-    return NO_ERROR;
-}
-
-}; //namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
new file mode 100644
index 0000000..5aca3cc
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -0,0 +1,123 @@
+/*
+ * 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 "APM_ClientDescriptor"
+//#define LOG_NDEBUG 0
+
+#include <utils/Log.h>
+#include <utils/String8.h>
+#include "AudioGain.h"
+#include "AudioOutputDescriptor.h"
+#include "AudioPatch.h"
+#include "ClientDescriptor.h"
+#include "DeviceDescriptor.h"
+#include "HwModule.h"
+#include "IOProfile.h"
+
+namespace android {
+
+status_t ClientDescriptor::dump(int fd, int spaces, int index)
+{
+    String8 out;
+
+    // FIXME: use until other descriptor classes have a dump to String8 method
+    mDumpFd = fd;
+
+    status_t status = dump(out, spaces, index);
+    if (status == NO_ERROR) {
+        write(fd, out.string(), out.size());
+    }
+
+    return status;
+}
+
+status_t ClientDescriptor::dump(String8& out, int spaces, int index)
+{
+    out.appendFormat("%*sClient %d:\n", spaces, "", index+1);
+    out.appendFormat("%*s- Port ID: %d Session Id: %d UID: %d\n", spaces, "",
+             mPortId, mSessionId, mUid);
+    out.appendFormat("%*s- Format: %08x Sampling rate: %d Channels: %08x\n", spaces, "",
+             mConfig.format, mConfig.sample_rate, mConfig.channel_mask);
+    out.appendFormat("%*s- Preferred Device Id: %08x\n", spaces, "", mPreferredDeviceId);
+    out.appendFormat("%*s- State: %s\n", spaces, "", mActive ? "Active" : "Inactive");
+    return NO_ERROR;
+}
+
+status_t TrackClientDescriptor::dump(String8& out, int spaces, int index)
+{
+    ClientDescriptor::dump(out, spaces, index);
+
+    out.appendFormat("%*s- Stream: %d flags: %08x\n", spaces, "", mStream, mFlags);
+
+    return NO_ERROR;
+}
+
+status_t RecordClientDescriptor::dump(String8& out, int spaces, int index)
+{
+    ClientDescriptor::dump(out, spaces, index);
+
+    out.appendFormat("%*s- Source: %d flags: %08x\n", spaces, "", mSource, mFlags);
+
+    return NO_ERROR;
+}
+
+SourceClientDescriptor::SourceClientDescriptor(audio_port_handle_t portId, uid_t uid,
+         audio_attributes_t attributes, const sp<AudioPatch>& patchDesc,
+         const sp<DeviceDescriptor>& srcDevice, audio_stream_type_t stream) :
+    TrackClientDescriptor::TrackClientDescriptor(portId, uid, AUDIO_SESSION_NONE, attributes,
+        AUDIO_CONFIG_BASE_INITIALIZER, AUDIO_PORT_HANDLE_NONE, stream, AUDIO_OUTPUT_FLAG_NONE),
+        mPatchDesc(patchDesc), mSrcDevice(srcDevice)
+{
+}
+
+void SourceClientDescriptor::setSwOutput(const sp<SwAudioOutputDescriptor>& swOutput)
+{
+    mSwOutput = swOutput;
+}
+
+void SourceClientDescriptor::setHwOutput(const sp<HwAudioOutputDescriptor>& hwOutput)
+{
+    mHwOutput = hwOutput;
+}
+
+status_t SourceClientDescriptor::dump(String8& out, int spaces, int index)
+{
+    TrackClientDescriptor::dump(out, spaces, index);
+
+    if (mDumpFd >= 0) {
+        out.appendFormat("%*s- Device:\n", spaces, "");
+        write(mDumpFd, out.string(), out.size());
+
+        mSrcDevice->dump(mDumpFd, 2, 0);
+        mDumpFd = -1;
+    }
+
+    return NO_ERROR;
+}
+
+status_t SourceClientCollection::dump(int fd) const
+{
+    String8 out;
+    out.append("\nAudio sources:\n");
+    write(fd, out.string(), out.size());
+    for (size_t i = 0; i < size(); i++) {
+        valueAt(i)->dump(fd, 2, i);
+    }
+
+    return NO_ERROR;
+}
+
+}; //namespace android
diff --git a/services/audiopolicy/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..1638645 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -121,17 +121,28 @@
     return ret;
 }
 
-audio_devices_t DeviceVector::getDevicesFromHwModule(audio_module_handle_t moduleHandle) const
+DeviceVector DeviceVector::getDevicesFromHwModule(audio_module_handle_t moduleHandle) const
 {
-    audio_devices_t devices = AUDIO_DEVICE_NONE;
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->getModuleHandle() == moduleHandle) {
-            devices |= itemAt(i)->type();
+    DeviceVector devices;
+    for (const auto& device : *this) {
+        if (device->getModuleHandle() == moduleHandle) {
+            devices.add(device);
         }
     }
     return devices;
 }
 
+audio_devices_t DeviceVector::getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const
+{
+    audio_devices_t deviceTypes = AUDIO_DEVICE_NONE;
+    for (const auto& device : *this) {
+        if (device->getModuleHandle() == moduleHandle) {
+            deviceTypes |= device->type();
+        }
+    }
+    return deviceTypes;
+}
+
 sp<DeviceDescriptor> DeviceVector::getDevice(audio_devices_t type, const String8& address) const
 {
     sp<DeviceDescriptor> device;
@@ -145,8 +156,8 @@
             }
         }
     }
-    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 +171,7 @@
     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 +182,8 @@
         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 +252,7 @@
 
 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 +304,7 @@
 {
     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 2206526..440a4e7 100644
--- a/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/SessionRoute.cpp
@@ -127,7 +127,7 @@
         if (streamType == route->mStreamType && route->isActiveOrChanged()
                 && route->mDeviceDescriptor != 0) {
             audio_devices_t device = route->mDeviceDescriptor->type();
-            if (!availableDevices.getDevicesFromType(device).isEmpty()) {
+            if (!availableDevices.getDevicesFromTypeMask(device).isEmpty()) {
                 return device;
             }
         }
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 267996c..30f275f 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -313,7 +313,7 @@
             audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
             sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
             audio_devices_t availPrimaryInputDevices =
-                 availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle());
+                 availableInputDevices.getDeviceTypesFromHwModule(primaryOutput->getModuleHandle());
 
             // TODO: getPrimaryOutput return only devices from first module in
             // audio_policy_configuration.xml, hearing aid is not there, but it's
@@ -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(
@@ -669,9 +668,8 @@
         if ((getPhoneState() == AUDIO_MODE_IN_CALL) &&
                 (availableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
             sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
-            availableDeviceTypes =
-                    availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle())
-                    & ~AUDIO_DEVICE_BIT_IN;
+            availableDeviceTypes = availableInputDevices.getDeviceTypesFromHwModule(
+                    primaryOutput->getModuleHandle()) & ~AUDIO_DEVICE_BIT_IN;
         }
 
         switch (mForceUse[AUDIO_POLICY_FORCE_FOR_COMMUNICATION]) {
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 0d6cfda..96079cc 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -31,6 +31,7 @@
 
 #include <inttypes.h>
 #include <math.h>
+#include <vector>
 
 #include <AudioPolicyManagerInterface.h>
 #include <AudioPolicyEngineInstance.h>
@@ -38,6 +39,7 @@
 #include <utils/Log.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicyHelper.h>
+#include <private/android_filesystem_config.h>
 #include <soundtrigger/SoundTrigger.h>
 #include <system/audio.h>
 #include <audio_policy_conf.h>
@@ -80,6 +82,16 @@
     AUDIO_FORMAT_AAC_XHE,
 };
 
+// Compressed formats for MSD module, ordered from most preferred to least preferred.
+static const std::vector<audio_format_t> compressedFormatsOrder = {{
+        AUDIO_FORMAT_MAT_2_1, AUDIO_FORMAT_MAT_2_0, AUDIO_FORMAT_E_AC3,
+        AUDIO_FORMAT_AC3, AUDIO_FORMAT_PCM_16_BIT }};
+// Channel masks for MSD module, 3D > 2D > 1D ordering (most preferred to least preferred).
+static const std::vector<audio_channel_mask_t> surroundChannelMasksOrder = {{
+        AUDIO_CHANNEL_OUT_3POINT1POINT2, AUDIO_CHANNEL_OUT_3POINT0POINT2,
+        AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2,
+        AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }};
+
 // ----------------------------------------------------------------------------
 // AudioPolicyInterface implementation
 // ----------------------------------------------------------------------------
@@ -200,31 +212,30 @@
             return BAD_VALUE;
         }
 
-        // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
-        // output is suspended before any tracks are moved to it
-        checkA2dpSuspend();
-        checkOutputForAllStrategies();
-        // outputs must be closed after checkOutputForAllStrategies() is executed
-        if (!outputs.isEmpty()) {
-            for (audio_io_handle_t output : outputs) {
-                sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
-                // close unused outputs after device disconnection or direct outputs that have been
-                // opened by checkOutputsForDevice() to query dynamic parameters
-                if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
-                        (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
-                         (desc->mDirectOpenCount == 0))) {
-                    closeOutput(output);
+        checkForDeviceAndOutputChanges([&]() {
+            // outputs must be closed after checkOutputForAllStrategies() is executed
+            if (!outputs.isEmpty()) {
+                for (audio_io_handle_t output : outputs) {
+                    sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
+                    // close unused outputs after device disconnection or direct outputs that have been
+                    // opened by checkOutputsForDevice() to query dynamic parameters
+                    if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
+                            (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+                             (desc->mDirectOpenCount == 0))) {
+                        closeOutput(output);
+                    }
                 }
+                // check A2DP again after closing A2DP output to reset mA2dpSuspended if needed
+                return true;
             }
-            // check again after closing A2DP output to reset mA2dpSuspended if needed
-            checkA2dpSuspend();
-        }
+            return false;
+        });
 
-        updateDevicesAndOutputs();
         if (mEngine->getPhoneState() == AUDIO_MODE_IN_CALL && hasPrimaryOutput()) {
             audio_devices_t newDevice = getNewOutputDevice(mPrimaryOutput, false /*fromCache*/);
             updateCallRouting(newDevice);
         }
+        const audio_devices_t msdOutDevice = getMsdAudioOutDeviceTypes();
         for (size_t i = 0; i < mOutputs.size(); i++) {
             sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
             if ((mEngine->getPhoneState() != AUDIO_MODE_IN_CALL) || (desc != mPrimaryOutput)) {
@@ -232,7 +243,8 @@
                 // do not force device change on duplicated output because if device is 0, it will
                 // also force a device 0 for the two outputs it is duplicated to which may override
                 // a valid device selection on those outputs.
-                bool force = !desc->isDuplicated()
+                bool force = (msdOutDevice == AUDIO_DEVICE_NONE || msdOutDevice != desc->device())
+                        && !desc->isDuplicated()
                         && (!device_distinguishes_on_address(device)
                                 // always force when disconnecting (a non-duplicated device)
                                 || (state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE));
@@ -476,20 +488,15 @@
 
 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 +507,7 @@
         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) {
@@ -512,38 +517,31 @@
         // symmetric to the one in startInput()
         for (const auto& activeDesc : mInputs.getActiveInputs()) {
             if (activeDesc->hasSameHwModuleAs(txSourceDeviceDesc)) {
-                AudioSessionCollection activeSessions =
-                        activeDesc->getAudioSessions(true /*activeOnly*/);
-                for (size_t j = 0; j < activeSessions.size(); j++) {
-                    audio_session_t activeSession = activeSessions.keyAt(j);
-                    stopInput(activeDesc->mIoHandle, activeSession);
-                    releaseInput(activeDesc->mIoHandle, activeSession);
-                }
+                closeSessions(activeDesc, true  /*activeOnly*/);
             }
         }
     }
 
     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) const {
+    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 +555,8 @@
         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);
     }
@@ -577,9 +569,7 @@
                   || (is_state_in_call(state) && (state != oldState)));
 
     // check for device and output changes triggered by new phone state
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-    updateDevicesAndOutputs();
+    checkForDeviceAndOutputChanges();
 
     int delayMs = 0;
     if (isStateInCall(state)) {
@@ -643,14 +633,8 @@
         }
     }
 
-    // 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);
     }
@@ -685,9 +669,7 @@
             (usage == AUDIO_POLICY_FORCE_FOR_SYSTEM);
 
     // check for device and output changes triggered by new force usage
-    checkA2dpSuspend();
-    checkOutputForAllStrategies();
-    updateDevicesAndOutputs();
+    checkForDeviceAndOutputChanges();
 
     //FIXME: workaround for truncated touch sounds
     // to be removed when the problem is handled by system UI
@@ -803,6 +785,12 @@
                                               audio_port_handle_t *portId)
 {
     audio_attributes_t attributes;
+    DeviceVector outputDevices;
+    routing_strategy strategy;
+    audio_devices_t device;
+    audio_port_handle_t requestedDeviceId = *selectedDeviceId;
+    audio_devices_t msdDevice = getMsdAudioOutDeviceTypes();
+
     if (attr != NULL) {
         if (!isValidAttributes(attr)) {
             ALOGE("getOutputForAttr() invalid attributes: usage=%d content=%d flags=0x%x tags=[%s]",
@@ -819,43 +807,57 @@
         stream_type_to_audio_attributes(*stream, &attributes);
     }
 
-    // TODO: check for existing client for this port ID
-    if (*portId == AUDIO_PORT_HANDLE_NONE) {
-        *portId = AudioPort::getNextUniqueId();
-    }
-
-    sp<SwAudioOutputDescriptor> desc;
-    if (mPolicyMixes.getOutputForAttr(attributes, uid, desc) == NO_ERROR) {
-        ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr");
-        if (!audio_has_proportional_frames(config->format)) {
-            return BAD_VALUE;
-        }
-        *stream = streamTypefromAttributesInt(&attributes);
-        *output = desc->mIoHandle;
-        ALOGV("getOutputForAttr() returns output %d", *output);
-        return NO_ERROR;
-    }
-    if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
-        ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
-        return BAD_VALUE;
-    }
-
     ALOGV("getOutputForAttr() usage=%d, content=%d, tag=%s flags=%08x"
             " session %d selectedDeviceId %d",
             attributes.usage, attributes.content_type, attributes.tags, attributes.flags,
             session, *selectedDeviceId);
 
-    *stream = streamTypefromAttributesInt(&attributes);
+    // TODO: check for existing client for this port ID
+    if (*portId == AUDIO_PORT_HANDLE_NONE) {
+        *portId = AudioPort::getNextUniqueId();
+    }
 
-    // Explicit routing?
+    // First check for explicit routing (eg. setPreferredDevice)
     sp<DeviceDescriptor> deviceDesc;
     if (*selectedDeviceId != AUDIO_PORT_HANDLE_NONE) {
         deviceDesc = mAvailableOutputDevices.getDeviceFromId(*selectedDeviceId);
+    } else {
+        // If no explict route, is there a matching dynamic policy that applies?
+        sp<SwAudioOutputDescriptor> desc;
+        if (mPolicyMixes.getOutputForAttr(attributes, uid, desc) == NO_ERROR) {
+            ALOG_ASSERT(desc != 0, "Invalid desc returned by getOutputForAttr");
+            if (!audio_has_proportional_frames(config->format)) {
+                return BAD_VALUE;
+            }
+            *stream = streamTypefromAttributesInt(&attributes);
+            *output = desc->mIoHandle;
+            ALOGV("getOutputForAttr() returns output %d", *output);
+            goto exit;
+        }
+
+        // Virtual sources must always be dynamicaly or explicitly routed
+        if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+            ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+            return BAD_VALUE;
+        }
     }
+
+    // Virtual sources must always be dynamicaly or explicitly routed
+    if (attributes.usage == AUDIO_USAGE_VIRTUAL_SOURCE) {
+        ALOGW("getOutputForAttr() no policy mix found for usage AUDIO_USAGE_VIRTUAL_SOURCE");
+        return BAD_VALUE;
+    }
+
+    *stream = streamTypefromAttributesInt(&attributes);
+
+    // TODO:  Should this happen only if an explicit route is active?
+    // the previous code structure meant that this would always happen which
+    // would appear to result in adding a null deviceDesc when not using an
+    // explicit route.  Is that the intended and necessary behavior?
     mOutputRoutes.addRoute(session, *stream, SessionRoute::SOURCE_TYPE_NA, deviceDesc, uid);
 
-    routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
-    audio_devices_t device = getDeviceForStrategy(strategy, false /*fromCache*/);
+    strategy = (routing_strategy) getStrategyForAttr(&attributes);
+    device = getDeviceForStrategy(strategy, false /*fromCache*/);
 
     if ((attributes.flags & AUDIO_FLAG_HW_AV_SYNC) != 0) {
         *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
@@ -880,17 +882,41 @@
           "flags %#x",
           device, config->sample_rate, config->format, config->channel_mask, *flags);
 
-    *output = getOutputForDevice(device, session, *stream, config, flags);
+    *output = AUDIO_IO_HANDLE_NONE;
+    if (msdDevice != AUDIO_DEVICE_NONE) {
+        *output = getOutputForDevice(msdDevice, session, *stream, config, flags);
+        if (*output != AUDIO_IO_HANDLE_NONE && setMsdPatch(device) == NO_ERROR) {
+            ALOGV("%s() Using MSD device 0x%x instead of device 0x%x",
+                    __func__, msdDevice, device);
+            device = msdDevice;
+        } else {
+            *output = AUDIO_IO_HANDLE_NONE;
+        }
+    }
+    if (*output == AUDIO_IO_HANDLE_NONE) {
+        *output = getOutputForDevice(device, session, *stream, config, flags);
+    }
     if (*output == AUDIO_IO_HANDLE_NONE) {
         mOutputRoutes.removeRoute(session);
         return INVALID_OPERATION;
     }
 
-    DeviceVector outputDevices = mAvailableOutputDevices.getDevicesFromType(device);
+    outputDevices = mAvailableOutputDevices.getDevicesFromTypeMask(device);
     *selectedDeviceId = outputDevices.size() > 0 ? outputDevices.itemAt(0)->getId()
             : AUDIO_PORT_HANDLE_NONE;
 
-    ALOGV("  getOutputForAttr() returns output %d selectedDeviceId %d", *output, *selectedDeviceId);
+exit:
+    audio_config_base_t clientConfig = {.sample_rate = config->sample_rate,
+        .format = config->format,
+        .channel_mask = config->channel_mask };
+    sp<TrackClientDescriptor> clientDesc =
+        new TrackClientDescriptor(*portId, uid, session,
+                                  attributes, clientConfig, requestedDeviceId, *stream, *flags);
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueFor(*output);
+    outputDesc->clients().emplace(*portId, clientDesc);
+
+    ALOGV("  getOutputForAttr() returns output %d selectedDeviceId %d for port ID %d",
+          *output, *selectedDeviceId, *portId);
 
     return NO_ERROR;
 }
@@ -986,7 +1012,7 @@
         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("");
 
@@ -1052,6 +1078,164 @@
     return output;
 }
 
+sp<DeviceDescriptor> AudioPolicyManager::getMsdAudioInDevice() const {
+    sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
+    if (msdModule != 0) {
+        DeviceVector msdInputDevices = mAvailableInputDevices.getDevicesFromHwModule(
+                msdModule->getHandle());
+        if (!msdInputDevices.isEmpty()) return msdInputDevices.itemAt(0);
+    }
+    return 0;
+}
+
+audio_devices_t AudioPolicyManager::getMsdAudioOutDeviceTypes() const {
+    sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
+    if (msdModule != 0) {
+        return mAvailableOutputDevices.getDeviceTypesFromHwModule(msdModule->getHandle());
+    }
+    return AUDIO_DEVICE_NONE;
+}
+
+const AudioPatchCollection AudioPolicyManager::getMsdPatches() const {
+    AudioPatchCollection msdPatches;
+    audio_module_handle_t msdModuleHandle = mHwModules.getModuleFromName(
+            AUDIO_HARDWARE_MODULE_ID_MSD)->getHandle();
+    if (msdModuleHandle == AUDIO_MODULE_HANDLE_NONE) return msdPatches;
+    for (size_t i = 0; i < mAudioPatches.size(); ++i) {
+        sp<AudioPatch> patch = mAudioPatches.valueAt(i);
+        for (size_t j = 0; j < patch->mPatch.num_sources; ++j) {
+            const struct audio_port_config *source = &patch->mPatch.sources[j];
+            if (source->type == AUDIO_PORT_TYPE_DEVICE &&
+                    source->ext.device.hw_module == msdModuleHandle) {
+                msdPatches.addAudioPatch(patch->mHandle, patch);
+            }
+        }
+    }
+    return msdPatches;
+}
+
+status_t AudioPolicyManager::getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+        bool hwAvSync, audio_port_config *sourceConfig, audio_port_config *sinkConfig) const
+{
+    sp<HwModule> msdModule = mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD);
+    if (msdModule == nullptr) {
+        ALOGE("%s() unable to get MSD module", __func__);
+        return NO_INIT;
+    }
+    sp<HwModule> deviceModule = mHwModules.getModuleForDevice(outputDevice);
+    if (deviceModule == nullptr) {
+        ALOGE("%s() unable to get module for %#x", __func__, outputDevice);
+        return NO_INIT;
+    }
+    const InputProfileCollection &inputProfiles = msdModule->getInputProfiles();
+    if (inputProfiles.isEmpty()) {
+        ALOGE("%s() no input profiles for MSD module", __func__);
+        return NO_INIT;
+    }
+    const OutputProfileCollection &outputProfiles = deviceModule->getOutputProfiles();
+    if (outputProfiles.isEmpty()) {
+        ALOGE("%s() no output profiles for device %#x", __func__, outputDevice);
+        return NO_INIT;
+    }
+    AudioProfileVector msdProfiles;
+    // Each IOProfile represents a MixPort from audio_policy_configuration.xml
+    for (const auto &inProfile : inputProfiles) {
+        if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0)) {
+            msdProfiles.appendVector(inProfile->getAudioProfiles());
+        }
+    }
+    AudioProfileVector deviceProfiles;
+    for (const auto &outProfile : outputProfiles) {
+        if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0)) {
+            deviceProfiles.appendVector(outProfile->getAudioProfiles());
+        }
+    }
+    struct audio_config_base bestSinkConfig;
+    status_t result = msdProfiles.findBestMatchingOutputConfig(deviceProfiles,
+            compressedFormatsOrder, surroundChannelMasksOrder, true /*preferHigherSamplingRates*/,
+            &bestSinkConfig);
+    if (result != NO_ERROR) {
+        ALOGD("%s() no matching profiles found for device: %#x, hwAvSync: %d",
+                __func__, outputDevice, hwAvSync);
+        return result;
+    }
+    sinkConfig->sample_rate = bestSinkConfig.sample_rate;
+    sinkConfig->channel_mask = bestSinkConfig.channel_mask;
+    sinkConfig->format = bestSinkConfig.format;
+    // For encoded streams force direct flag to prevent downstream mixing.
+    sinkConfig->flags.output = static_cast<audio_output_flags_t>(
+            sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_DIRECT);
+    sourceConfig->sample_rate = bestSinkConfig.sample_rate;
+    // Specify exact channel mask to prevent guessing by bit count in PatchPanel.
+    sourceConfig->channel_mask = audio_channel_mask_out_to_in(bestSinkConfig.channel_mask);
+    sourceConfig->format = bestSinkConfig.format;
+    // Copy input stream directly without any processing (e.g. resampling).
+    sourceConfig->flags.input = static_cast<audio_input_flags_t>(
+            sourceConfig->flags.input | AUDIO_INPUT_FLAG_DIRECT);
+    if (hwAvSync) {
+        sinkConfig->flags.output = static_cast<audio_output_flags_t>(
+                sinkConfig->flags.output | AUDIO_OUTPUT_FLAG_HW_AV_SYNC);
+        sourceConfig->flags.input = static_cast<audio_input_flags_t>(
+                sourceConfig->flags.input | AUDIO_INPUT_FLAG_HW_AV_SYNC);
+    }
+    const unsigned int config_mask = AUDIO_PORT_CONFIG_SAMPLE_RATE |
+            AUDIO_PORT_CONFIG_CHANNEL_MASK | AUDIO_PORT_CONFIG_FORMAT | AUDIO_PORT_CONFIG_FLAGS;
+    sinkConfig->config_mask |= config_mask;
+    sourceConfig->config_mask |= config_mask;
+    return NO_ERROR;
+}
+
+PatchBuilder AudioPolicyManager::buildMsdPatch(audio_devices_t outputDevice) const
+{
+    PatchBuilder patchBuilder;
+    patchBuilder.addSource(getMsdAudioInDevice()).
+            addSink(findDevice(mAvailableOutputDevices, outputDevice));
+    audio_port_config sourceConfig = patchBuilder.patch()->sources[0];
+    audio_port_config sinkConfig = patchBuilder.patch()->sinks[0];
+    // TODO: Figure out whether MSD module has HW_AV_SYNC flag set in the AP config file.
+    // For now, we just forcefully try with HwAvSync first.
+    status_t res = getBestMsdAudioProfileFor(outputDevice, true /*hwAvSync*/,
+            &sourceConfig, &sinkConfig) == NO_ERROR ? NO_ERROR :
+            getBestMsdAudioProfileFor(
+                    outputDevice, false /*hwAvSync*/, &sourceConfig, &sinkConfig);
+    if (res == NO_ERROR) {
+        // Found a matching profile for encoded audio. Re-create PatchBuilder with this config.
+        return (PatchBuilder()).addSource(sourceConfig).addSink(sinkConfig);
+    }
+    ALOGV("%s() no matching profile found. Fall through to default PCM patch"
+            " supporting PCM format conversion.", __func__);
+    return patchBuilder;
+}
+
+status_t AudioPolicyManager::setMsdPatch(audio_devices_t outputDevice) {
+    ALOGV("%s() for outputDevice %#x", __func__, outputDevice);
+    if (outputDevice == AUDIO_DEVICE_NONE) {
+        // Use media strategy for unspecified output device. This should only
+        // occur on checkForDeviceAndOutputChanges(). Device connection events may
+        // therefore invalidate explicit routing requests.
+        outputDevice = getDeviceForStrategy(STRATEGY_MEDIA, false /*fromCache*/);
+    }
+    PatchBuilder patchBuilder = buildMsdPatch(outputDevice);
+    const struct audio_patch* patch = patchBuilder.patch();
+    const AudioPatchCollection msdPatches = getMsdPatches();
+    if (!msdPatches.isEmpty()) {
+        LOG_ALWAYS_FATAL_IF(msdPatches.size() > 1,
+                "The current MSD prototype only supports one output patch");
+        sp<AudioPatch> currentPatch = msdPatches.valueAt(0);
+        if (audio_patches_are_equal(&currentPatch->mPatch, patch)) {
+            return NO_ERROR;
+        }
+        releaseAudioPatch(currentPatch->mHandle, mUidCached);
+    }
+    status_t status = installPatch(__func__, -1 /*index*/, nullptr /*patchHandle*/,
+            patch, 0 /*delayMs*/, mUidCached, nullptr /*patchDescPtr*/);
+    ALOGE_IF(status != NO_ERROR, "%s() error %d creating MSD audio patch", __func__, status);
+    ALOGI_IF(status == NO_ERROR, "%s() Patch created from MSD_IN to "
+           "device:%#x (format:%#x channels:%#x samplerate:%d)", __func__, outputDevice,
+           patch->sources[0].format, patch->sources[0].channel_mask, patch->sources[0].sample_rate);
+    return status;
+}
+
 audio_io_handle_t AudioPolicyManager::selectOutput(const SortedVector<audio_io_handle_t>& outputs,
                                                        audio_output_flags_t flags,
                                                        audio_format_t format)
@@ -1132,19 +1316,21 @@
     return outputs[0];
 }
 
-status_t AudioPolicyManager::startOutput(audio_io_handle_t output,
-                                             audio_stream_type_t stream,
-                                             audio_session_t session)
+status_t AudioPolicyManager::startOutput(audio_port_handle_t portId)
 {
-    ALOGV("startOutput() output %d, stream %d, session %d",
-          output, stream, session);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        ALOGW("startOutput() unknown output %d", output);
+    ALOGV("%s portId %d", __FUNCTION__, portId);
+
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputForClient(portId);
+    if (outputDesc == 0) {
+        ALOGW("startOutput() no output for client %d", portId);
         return BAD_VALUE;
     }
+    sp<TrackClientDescriptor> client = outputDesc->clients()[portId];
+    audio_stream_type_t stream = client->stream();
+    audio_session_t session = client->session();
 
-    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+    ALOGV("startOutput() output %d, stream %d, session %d",
+          outputDesc->mIoHandle, stream, session);
 
     status_t status = outputDesc->start();
     if (status != NO_ERROR) {
@@ -1168,7 +1354,7 @@
     } else if (mOutputRoutes.getAndClearRouteChanged(session)) {
         newDevice = getNewOutputDevice(outputDesc, false /*fromCache*/);
         if (newDevice != outputDesc->device()) {
-            checkStrategyRoute(getStrategy(stream), output);
+            checkStrategyRoute(getStrategy(stream), outputDesc->mIoHandle);
         }
     } else {
         newDevice = AUDIO_DEVICE_NONE;
@@ -1295,11 +1481,6 @@
         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()),
@@ -1341,19 +1522,20 @@
     return NO_ERROR;
 }
 
-
-status_t AudioPolicyManager::stopOutput(audio_io_handle_t output,
-                                            audio_stream_type_t stream,
-                                            audio_session_t session)
+status_t AudioPolicyManager::stopOutput(audio_port_handle_t portId)
 {
-    ALOGV("stopOutput() output %d, stream %d, session %d", output, stream, session);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        ALOGW("stopOutput() unknown output %d", output);
+    ALOGV("%s portId %d", __FUNCTION__, portId);
+
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputForClient(portId);
+    if (outputDesc == 0) {
+        ALOGW("stopOutput() no output for client %d", portId);
         return BAD_VALUE;
     }
+    sp<TrackClientDescriptor> client = outputDesc->clients()[portId];
+    audio_stream_type_t stream = client->stream();
+    audio_session_t session = client->session();
 
-    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.valueAt(index);
+    ALOGV("stopOutput() output %d, stream %d, session %d", outputDesc->mIoHandle, stream, session);
 
     if (outputDesc->mRefCount[stream] == 1) {
         // Automatically disable the remote submix input when output is stopped on a
@@ -1394,11 +1576,6 @@
     // 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);
@@ -1455,32 +1632,35 @@
     }
 }
 
-void AudioPolicyManager::releaseOutput(audio_io_handle_t output,
-                                       audio_stream_type_t stream __unused,
-                                       audio_session_t session __unused)
+void AudioPolicyManager::releaseOutput(audio_port_handle_t portId)
 {
-    ALOGV("releaseOutput() %d", output);
-    ssize_t index = mOutputs.indexOfKey(output);
-    if (index < 0) {
-        ALOGW("releaseOutput() releasing unknown output %d", output);
+    ALOGV("%s portId %d", __FUNCTION__, portId);
+
+    sp<SwAudioOutputDescriptor> outputDesc = mOutputs.getOutputForClient(portId);
+    if (outputDesc == 0) {
+        ALOGW("releaseOutput() no output for client %d", portId);
         return;
     }
+    sp<TrackClientDescriptor> client = outputDesc->clients()[portId];
+    audio_session_t session = client->session();
+
+    ALOGV("releaseOutput() %d", outputDesc->mIoHandle);
 
     // Routing
     mOutputRoutes.removeRoute(session);
 
-    sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(index);
-    if (desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
-        if (desc->mDirectOpenCount <= 0) {
+    if (outputDesc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) {
+        if (outputDesc->mDirectOpenCount <= 0) {
             ALOGW("releaseOutput() invalid open count %d for output %d",
-                                                              desc->mDirectOpenCount, output);
+                  outputDesc->mDirectOpenCount, outputDesc->mIoHandle);
             return;
         }
-        if (--desc->mDirectOpenCount == 0) {
-            closeOutput(output);
+        if (--outputDesc->mDirectOpenCount == 0) {
+            closeOutput(outputDesc->mIoHandle);
             mpClientInterface->onAudioPortListUpdate();
         }
     }
+    outputDesc->clients().erase(portId);
 }
 
 
@@ -1505,6 +1685,9 @@
     audio_source_t inputSource = attr->source;
     AudioMix *policyMix = NULL;
     DeviceVector inputDevices;
+    sp<AudioInputDescriptor> inputDesc;
+    sp<RecordClientDescriptor> clientDesc;
+    audio_port_handle_t requestedDeviceId = *selectedDeviceId;
 
     if (inputSource == AUDIO_SOURCE_DEFAULT) {
         inputSource = AUDIO_SOURCE_MIC;
@@ -1555,12 +1738,12 @@
         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);
 
-        return NO_ERROR;
+        goto exit;
     }
 
     *input = AUDIO_IO_HANDLE_NONE;
@@ -1622,12 +1805,18 @@
         goto error;
     }
 
-    inputDevices = mAvailableInputDevices.getDevicesFromType(device);
+    inputDevices = mAvailableInputDevices.getDevicesFromTypeMask(device);
     *selectedDeviceId = inputDevices.size() > 0 ? inputDevices.itemAt(0)->getId()
             : AUDIO_PORT_HANDLE_NONE;
 
-    ALOGV("getInputForAttr() returns input %d type %d selectedDeviceId %d",
-            *input, *inputType, *selectedDeviceId);
+exit:
+    clientDesc = new RecordClientDescriptor(*portId, uid, session,
+                                  *attr, *config, requestedDeviceId, inputSource, flags);
+    inputDesc = mInputs.valueFor(*input);
+    inputDesc->clients().emplace(*portId, clientDesc);
+
+    ALOGV("getInputForAttr() returns input %d type %d selectedDeviceId %d for port ID %d",
+            *input, *inputType, *selectedDeviceId, *portId);
 
     return NO_ERROR;
 
@@ -1709,74 +1898,7 @@
                                                      config->channel_mask,
                                                      flags,
                                                      uid,
-                                                     isSoundTrigger,
-                                                     policyMix, mpClientInterface);
-
-// FIXME: disable concurrent capture until UI is ready
-#if 0
-    // reuse an open input if possible
-    sp<AudioInputDescriptor> reusedInputDesc;
-    for (size_t i = 0; i < mInputs.size(); i++) {
-        sp<AudioInputDescriptor> desc = mInputs.valueAt(i);
-        // reuse input if:
-        // - it shares the same profile
-        //      AND
-        // - it is not a reroute submix input
-        //      AND
-        // - it is: not used for sound trigger
-        //                OR
-        //          used for sound trigger and all clients use the same session ID
-        //
-        if ((profile == desc->mProfile) &&
-            (isSoundTrigger == desc->isSoundTrigger()) &&
-            !is_virtual_input_device(device)) {
-
-            sp<AudioSession> as = desc->getAudioSession(session);
-            if (as != 0) {
-                // do not allow unmatching properties on same session
-                if (as->matches(audioSession)) {
-                    as->changeOpenCount(1);
-                } else {
-                    ALOGW("getInputForDevice() record with different attributes"
-                          " exists for session %d", session);
-                    continue;
-                }
-            } else if (isSoundTrigger) {
-                continue;
-            }
-
-            // Reuse the already opened input stream on this profile if:
-            // - the new capture source is background OR
-            // - the path requested configurations match OR
-            // - the new source priority is less than the highest source priority on this input
-            // If the input stream cannot be reused, close it before opening a new stream
-            // on the same profile for the new client so that the requested path configuration
-            // can be selected.
-            if (!isConcurrentSource(inputSource) &&
-                    ((desc->mSamplingRate != samplingRate ||
-                    desc->mChannelMask != config->channel_mask ||
-                    !audio_formats_match(desc->mFormat, config->format)) &&
-                    (source_priority(desc->getHighestPrioritySource(false /*activeOnly*/)) <
-                     source_priority(inputSource)))) {
-                reusedInputDesc = desc;
-                continue;
-            } else {
-                desc->addAudioSession(session, audioSession);
-                ALOGV("%s: reusing input %d", __FUNCTION__, mInputs.keyAt(i));
-                return mInputs.keyAt(i);
-            }
-        }
-    }
-
-    if (reusedInputDesc != 0) {
-        AudioSessionCollection sessions = reusedInputDesc->getAudioSessions(false /*activeOnly*/);
-        for (size_t j = 0; j < sessions.size(); j++) {
-            audio_session_t currentSession = sessions.keyAt(j);
-            stopInput(reusedInputDesc->mIoHandle, currentSession);
-            releaseInput(reusedInputDesc->mIoHandle, currentSession);
-        }
-    }
-#endif
+                                                     isSoundTrigger);
 
     if (!profile->canOpenNewIo()) {
         return AUDIO_IO_HANDLE_NONE;
@@ -1790,7 +1912,7 @@
     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("");
     }
@@ -1897,44 +2019,32 @@
 }
 
 
-status_t AudioPolicyManager::startInput(audio_io_handle_t input,
-                                        audio_session_t session,
+status_t AudioPolicyManager::startInput(audio_port_handle_t portId,
                                         bool silenced,
                                         concurrency_type__mask_t *concurrency)
 {
+    *concurrency = API_INPUT_CONCURRENCY_NONE;
+
+    ALOGV("%s portId %d", __FUNCTION__, portId);
+
+    sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
+    if (inputDesc == 0) {
+        ALOGW("startInput() no input for client %d", portId);
+        return BAD_VALUE;
+    }
+    sp<RecordClientDescriptor> client = inputDesc->clients()[portId];
+    audio_session_t session = client->session();
+    audio_io_handle_t input = inputDesc->mIoHandle;
 
     ALOGV("AudioPolicyManager::startInput(input:%d, session:%d, silenced:%d, concurrency:%d)",
             input, session, silenced, *concurrency);
 
-    *concurrency = API_INPUT_CONCURRENCY_NONE;
-
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        ALOGW("startInput() unknown input %d", input);
-        return BAD_VALUE;
-    }
-    sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
-
     sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
     if (audioSession == 0) {
         ALOGW("startInput() unknown session %d on input %d", session, input);
         return BAD_VALUE;
     }
 
-// FIXME: disable concurrent capture until UI is ready
-#if 0
-    if (!isConcurentCaptureAllowed(inputDesc, audioSession)) {
-        ALOGW("startInput(%d) failed: other input already started", input);
-        return INVALID_OPERATION;
-    }
-
-    if (isInCall()) {
-        *concurrency |= API_INPUT_CONCURRENCY_CALL;
-    }
-    if (mInputs.activeInputsCountOnDevices() != 0) {
-        *concurrency |= API_INPUT_CONCURRENCY_CAPTURE;
-    }
-#else
     if (!is_virtual_input_device(inputDesc->mDevice)) {
         if (mCallTxPatch != 0 &&
             inputDesc->getModuleHandle() == mCallTxPatch->mPatch.sources[0].ext.device.hw_module) {
@@ -1959,11 +2069,8 @@
                         true /*activeOnly*/);
                 sp<AudioSession> activeSession = activeSessions.valueAt(0);
                 if (activeSession->isSilenced()) {
-                    audio_io_handle_t activeInput = activeDesc->mIoHandle;
-                    audio_session_t activeSessionId = activeSession->session();
-                    stopInput(activeInput, activeSessionId);
-                    releaseInput(activeInput, activeSessionId);
-                    ALOGV("startInput(%d) stopping silenced input %d", input, activeInput);
+                    closeSession(activeDesc, activeSession);
+                    ALOGV("startInput() session %d stopping silenced session %d", session, activeSession->session());
                     activeInputs = mInputs.getActiveInputs();
                 }
             }
@@ -2017,27 +2124,24 @@
             if (activeSource == AUDIO_SOURCE_HOTWORD) {
                 AudioSessionCollection activeSessions =
                         activeDesc->getAudioSessions(true /*activeOnly*/);
-                audio_session_t activeSession = activeSessions.keyAt(0);
-                audio_io_handle_t activeHandle = activeDesc->mIoHandle;
+                sp<AudioSession> activeSession = activeSessions[0];
                 SortedVector<audio_session_t> sessions = activeDesc->getPreemptedSessions();
                 *concurrency |= API_INPUT_CONCURRENCY_PREEMPT;
-                sessions.add(activeSession);
+                sessions.add(activeSession->session());
                 inputDesc->setPreemptedSessions(sessions);
-                stopInput(activeHandle, activeSession);
-                releaseInput(activeHandle, activeSession);
+                closeSession(inputDesc, activeSession);
                 ALOGV("startInput(%d) for HOTWORD preempting HOTWORD input %d",
                       input, activeDesc->mIoHandle);
             }
         }
     }
-#endif
 
     // Make sure we start with the correct silence state
     audioSession->setSilenced(silenced);
 
     // increment activity count before calling getNewInputDevice() below as only active sessions
     // are considered for device selection
-    audioSession->changeActiveCount(1);
+    inputDesc->changeRefCount(session, 1);
 
     // Routing?
     mInputRoutes.incRouteActivity(session);
@@ -2051,7 +2155,7 @@
         status_t status = inputDesc->start();
         if (status != NO_ERROR) {
             mInputRoutes.decRouteActivity(session);
-            audioSession->changeActiveCount(-1);
+            inputDesc->changeRefCount(session, -1);
             return status;
         }
 
@@ -2093,29 +2197,29 @@
     return NO_ERROR;
 }
 
-status_t AudioPolicyManager::stopInput(audio_io_handle_t input,
-                                       audio_session_t session)
+status_t AudioPolicyManager::stopInput(audio_port_handle_t portId)
 {
-    ALOGV("stopInput() input %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        ALOGW("stopInput() unknown input %d", input);
+    ALOGV("%s portId %d", __FUNCTION__, portId);
+
+    sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
+    if (inputDesc == 0) {
+        ALOGW("stopInput() no input for client %d", portId);
         return BAD_VALUE;
     }
-    sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
+    sp<RecordClientDescriptor> client = inputDesc->clients()[portId];
+    audio_session_t session = client->session();
+    audio_io_handle_t input = inputDesc->mIoHandle;
+
+    ALOGV("stopInput() input %d", input);
 
     sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
-    if (index < 0) {
-        ALOGW("stopInput() unknown session %d on input %d", session, input);
-        return BAD_VALUE;
-    }
 
     if (audioSession->activeCount() == 0) {
         ALOGW("stopInput() input %d already stopped", input);
         return INVALID_OPERATION;
     }
 
-    audioSession->changeActiveCount(-1);
+    inputDesc->changeRefCount(session, -1);
 
     // Routing?
     mInputRoutes.decRouteActivity(session);
@@ -2164,22 +2268,24 @@
     return NO_ERROR;
 }
 
-void AudioPolicyManager::releaseInput(audio_io_handle_t input,
-                                      audio_session_t session)
+void AudioPolicyManager::releaseInput(audio_port_handle_t portId)
 {
-    ALOGV("releaseInput() %d", input);
-    ssize_t index = mInputs.indexOfKey(input);
-    if (index < 0) {
-        ALOGW("releaseInput() releasing unknown input %d", input);
+    ALOGV("%s portId %d", __FUNCTION__, portId);
+
+    sp<AudioInputDescriptor> inputDesc = mInputs.getInputForClient(portId);
+    if (inputDesc == 0) {
+        ALOGW("releaseInput() no input for client %d", portId);
         return;
     }
+    sp<RecordClientDescriptor> client = inputDesc->clients()[portId];
+    audio_session_t session = client->session();
+    audio_io_handle_t input = inputDesc->mIoHandle;
+
+    ALOGV("releaseInput() %d", input);
 
     // Routing
     mInputRoutes.removeRoute(session);
 
-    sp<AudioInputDescriptor> inputDesc = mInputs.valueAt(index);
-    ALOG_ASSERT(inputDesc != 0);
-
     sp<AudioSession> audioSession = inputDesc->getAudioSession(session);
     if (audioSession == 0) {
         ALOGW("releaseInput() unknown session %d on input %d", session, input);
@@ -2202,10 +2308,31 @@
     }
 
     closeInput(input);
+    inputDesc->clients().erase(portId);
     mpClientInterface->onAudioPortListUpdate();
     ALOGV("releaseInput() exit");
 }
 
+void AudioPolicyManager::closeSessions(const sp<AudioInputDescriptor>& input, bool activeOnly)
+{
+    AudioSessionCollection sessions = input->getAudioSessions(activeOnly /*activeOnly*/);
+    for (size_t i = 0; i < sessions.size(); i++) {
+        closeSession(input, sessions[i]);
+    }
+}
+
+void AudioPolicyManager::closeSession(const sp<AudioInputDescriptor>& input,
+                                      const sp<AudioSession>& session)
+{
+    RecordClientVector clients = input->getClientsForSession(session->session());
+
+    for (const auto& client : clients) {
+        stopInput(client->portId());
+        releaseInput(client->portId());
+    }
+}
+
+
 void AudioPolicyManager::closeAllInputs() {
     bool patchRemoved = false;
 
@@ -2289,11 +2416,10 @@
         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))) {
                 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);
@@ -2311,13 +2437,15 @@
                 applyVolume = !mVolumeCurves->hasVolumeIndexForDevice(
                         stream, curStreamDevice);
             }
-
+            // rescale index before applying to curStream as ranges may be different for
+            // stream and curStream
+            int idx = rescaleVolumeIndex(index, stream, (audio_stream_type_t)curStream);
             if (applyVolume) {
                 //FIXME: workaround for truncated touch sounds
                 // delayed volume change for system stream to be removed when the problem is
                 // handled by system UI
                 status_t volStatus =
-                        checkAndSetVolume((audio_stream_type_t)curStream, index, desc, curDevice,
+                        checkAndSetVolume((audio_stream_type_t)curStream, idx, desc, curDevice,
                             (stream == AUDIO_STREAM_SYSTEM) ? TOUCH_SOUND_FIXED_DELAY_MS : 0);
                 if (volStatus != NO_ERROR) {
                     status = volStatus;
@@ -2637,42 +2765,24 @@
 
 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"));
@@ -2684,6 +2794,7 @@
     mEffects.dump(fd);
     mAudioPatches.dump(fd);
     mPolicyMixes.dump(fd);
+    mAudioSources.dump(fd);
 
     return NO_ERROR;
 }
@@ -2868,8 +2979,7 @@
     }
     ALOGV("createAudioPatch() num sources %d num sinks %d", patch->num_sources, patch->num_sinks);
 
-    if (patch->num_sources == 0 || patch->num_sources > AUDIO_PATCH_PORTS_MAX ||
-            patch->num_sinks == 0 || patch->num_sinks > AUDIO_PATCH_PORTS_MAX) {
+    if (!audio_patch_is_valid(patch)) {
         return BAD_VALUE;
     }
     // only one source per audio patch supported for now
@@ -3088,28 +3198,8 @@
             }
             // 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;
@@ -3242,10 +3332,10 @@
         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);
     }
@@ -3347,8 +3437,8 @@
 void AudioPolicyManager::clearAudioSources(uid_t uid)
 {
     for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--)  {
-        sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
-        if (sourceDesc->mUid == uid) {
+        sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        if (sourceDesc->uid() == uid) {
             stopAudioSource(mAudioSources.keyAt(i));
         }
     }
@@ -3366,20 +3456,23 @@
 }
 
 status_t AudioPolicyManager::startAudioSource(const struct audio_port_config *source,
-                                  const audio_attributes_t *attributes,
-                                  audio_patch_handle_t *handle,
-                                  uid_t uid)
+                                              const audio_attributes_t *attributes,
+                                              audio_port_handle_t *portId,
+                                              uid_t uid)
 {
-    ALOGV("%s source %p attributes %p handle %p", __FUNCTION__, source, attributes, handle);
-    if (source == NULL || attributes == NULL || handle == NULL) {
+    ALOGV("%s", __FUNCTION__);
+    *portId = AUDIO_PORT_HANDLE_NONE;
+
+    if (source == NULL || attributes == NULL || portId == NULL) {
+        ALOGW("%s invalid argument: source %p attributes %p handle %p",
+              __FUNCTION__, source, attributes, portId);
         return BAD_VALUE;
     }
 
-    *handle = AUDIO_PATCH_HANDLE_NONE;
-
     if (source->role != AUDIO_PORT_ROLE_SOURCE ||
             source->type != AUDIO_PORT_TYPE_DEVICE) {
-        ALOGV("%s INVALID_OPERATION source->role %d source->type %d", __FUNCTION__, source->role, source->type);
+        ALOGW("%s INVALID_OPERATION source->role %d source->type %d",
+              __FUNCTION__, source->role, source->type);
         return INVALID_OPERATION;
     }
 
@@ -3387,41 +3480,43 @@
             mAvailableInputDevices.getDevice(source->ext.device.type,
                                               String8(source->ext.device.address));
     if (srcDeviceDesc == 0) {
-        ALOGV("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
+        ALOGW("%s source->ext.device.type %08x not found", __FUNCTION__, source->ext.device.type);
         return BAD_VALUE;
     }
-    sp<AudioSourceDescriptor> sourceDesc =
-            new AudioSourceDescriptor(srcDeviceDesc, attributes, uid);
 
-    struct audio_patch dummyPatch;
+    *portId = AudioPort::getNextUniqueId();
+
+    struct audio_patch dummyPatch = {};
     sp<AudioPatch> patchDesc = new AudioPatch(&dummyPatch, uid);
-    sourceDesc->mPatchDesc = patchDesc;
+
+    sp<SourceClientDescriptor> sourceDesc =
+        new SourceClientDescriptor(*portId, uid, *attributes, patchDesc, srcDeviceDesc,
+                                   streamTypefromAttributesInt(attributes));
 
     status_t status = connectAudioSource(sourceDesc);
     if (status == NO_ERROR) {
-        mAudioSources.add(sourceDesc->getHandle(), sourceDesc);
-        *handle = sourceDesc->getHandle();
+        mAudioSources.add(*portId, sourceDesc);
     }
     return status;
 }
 
-status_t AudioPolicyManager::connectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc)
+status_t AudioPolicyManager::connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
 {
-    ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle());
+    ALOGV("%s handle %d", __FUNCTION__, sourceDesc->portId());
 
     // make sure we only have one patch per source.
     disconnectAudioSource(sourceDesc);
 
-    routing_strategy strategy = (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes);
-    audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes);
-    sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->mDevice;
+    audio_attributes_t attributes = sourceDesc->attributes();
+    routing_strategy strategy = (routing_strategy) getStrategyForAttr(&attributes);
+    audio_stream_type_t stream = sourceDesc->stream();
+    sp<DeviceDescriptor> srcDeviceDesc = sourceDesc->srcDevice();
 
     audio_devices_t sinkDevice = getDeviceForStrategy(strategy, true);
     sp<DeviceDescriptor> sinkDeviceDesc =
             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() &&
@@ -3453,16 +3548,14 @@
         // 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->patchDesc()->mPatch = *patchBuilder.patch();
         if (status != NO_ERROR) {
             ALOGW("%s patch panel could not connect device patch, error %d",
                   __FUNCTION__, status);
@@ -3472,32 +3565,32 @@
         status = startSource(outputDesc, stream, sinkDevice, NULL, &delayMs);
 
         if (status != NO_ERROR) {
-            mpClientInterface->releaseAudioPatch(sourceDesc->mPatchDesc->mAfPatchHandle, 0);
+            mpClientInterface->releaseAudioPatch(sourceDesc->patchDesc()->mAfPatchHandle, 0);
             return status;
         }
-        sourceDesc->mSwOutput = outputDesc;
+        sourceDesc->setSwOutput(outputDesc);
         if (delayMs != 0) {
             usleep(delayMs * 1000);
         }
     }
 
-    sourceDesc->mPatchDesc->mAfPatchHandle = afPatchHandle;
-    addAudioPatch(sourceDesc->mPatchDesc->mHandle, sourceDesc->mPatchDesc);
+    sourceDesc->patchDesc()->mAfPatchHandle = afPatchHandle;
+    addAudioPatch(sourceDesc->patchDesc()->mHandle, sourceDesc->patchDesc());
 
     return NO_ERROR;
 }
 
-status_t AudioPolicyManager::stopAudioSource(audio_patch_handle_t handle __unused)
+status_t AudioPolicyManager::stopAudioSource(audio_port_handle_t portId)
 {
-    sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueFor(handle);
-    ALOGV("%s handle %d", __FUNCTION__, handle);
+    sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueFor(portId);
+    ALOGV("%s port ID %d", __FUNCTION__, portId);
     if (sourceDesc == 0) {
-        ALOGW("%s unknown source for handle %d", __FUNCTION__, handle);
+        ALOGW("%s unknown source for port ID %d", __FUNCTION__, portId);
         return BAD_VALUE;
     }
     status_t status = disconnectAudioSource(sourceDesc);
 
-    mAudioSources.removeItem(handle);
+    mAudioSources.removeItem(portId);
     return status;
 }
 
@@ -3625,7 +3718,7 @@
             }
         }
         // 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;
@@ -3751,7 +3844,7 @@
 
     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.
@@ -3770,7 +3863,7 @@
                                              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.
@@ -3831,20 +3924,20 @@
     }
 }
 
-status_t AudioPolicyManager::disconnectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc)
+status_t AudioPolicyManager::disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc)
 {
-    ALOGV("%s handle %d", __FUNCTION__, sourceDesc->getHandle());
+    ALOGV("%s port Id %d", __FUNCTION__, sourceDesc->portId());
 
-    sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->mPatchDesc->mHandle);
+    sp<AudioPatch> patchDesc = mAudioPatches.valueFor(sourceDesc->patchDesc()->mHandle);
     if (patchDesc == 0) {
         ALOGW("%s source has no patch with handle %d", __FUNCTION__,
-              sourceDesc->mPatchDesc->mHandle);
+              sourceDesc->patchDesc()->mHandle);
         return BAD_VALUE;
     }
-    removeAudioPatch(sourceDesc->mPatchDesc->mHandle);
+    removeAudioPatch(sourceDesc->patchDesc()->mHandle);
 
-    audio_stream_type_t stream = streamTypefromAttributesInt(&sourceDesc->mAttributes);
-    sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->mSwOutput.promote();
+    audio_stream_type_t stream = sourceDesc->stream();
+    sp<SwAudioOutputDescriptor> swOutputDesc = sourceDesc->swOutput().promote();
     if (swOutputDesc != 0) {
         status_t status = stopSource(swOutputDesc, stream, false);
         if (status == NO_ERROR) {
@@ -3852,7 +3945,7 @@
         }
         mpClientInterface->releaseAudioPatch(patchDesc->mAfPatchHandle, 0);
     } else {
-        sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->mHwOutput.promote();
+        sp<HwAudioOutputDescriptor> hwOutputDesc = sourceDesc->hwOutput().promote();
         if (hwOutputDesc != 0) {
           //   release patch between src device and output device
           //   close Hwoutput and remove from mHwOutputs
@@ -3863,15 +3956,16 @@
     return NO_ERROR;
 }
 
-sp<AudioSourceDescriptor> AudioPolicyManager::getSourceForStrategyOnOutput(
+sp<SourceClientDescriptor> AudioPolicyManager::getSourceForStrategyOnOutput(
         audio_io_handle_t output, routing_strategy strategy)
 {
-    sp<AudioSourceDescriptor> source;
+    sp<SourceClientDescriptor> source;
     for (size_t i = 0; i < mAudioSources.size(); i++)  {
-        sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        audio_attributes_t attributes = sourceDesc->attributes();
         routing_strategy sourceStrategy =
-                (routing_strategy) getStrategyForAttr(&sourceDesc->mAttributes);
-        sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->mSwOutput.promote();
+                (routing_strategy) getStrategyForAttr(&attributes);
+        sp<SwAudioOutputDescriptor> outputDesc = sourceDesc->swOutput().promote();
         if (sourceStrategy == strategy && outputDesc != 0 && outputDesc->mIoHandle == output) {
             source = sourceDesc;
             break;
@@ -3914,6 +4008,7 @@
                      "%s/%s", kConfigLocationList[i], fileName);
             ret = serializer.deserialize(audioPolicyXmlConfigFile, config);
             if (ret == NO_ERROR) {
+                config.setSource(audioPolicyXmlConfigFile);
                 return ret;
             }
         }
@@ -3925,7 +4020,7 @@
 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),
@@ -4036,7 +4131,8 @@
             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;
@@ -4090,7 +4186,7 @@
             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("");
@@ -4665,6 +4761,7 @@
 
     nextAudioPortGeneration();
 
+    audio_devices_t device = inputDesc->mDevice;
     ssize_t index = mAudioPatches.indexOfKey(inputDesc->getPatchHandle());
     if (index >= 0) {
         sp<AudioPatch> patchDesc = mAudioPatches.valueAt(index);
@@ -4675,6 +4772,12 @@
 
     inputDesc->close();
     mInputs.removeItem(input);
+
+    audio_devices_t primaryInputDevices = availablePrimaryInputDevices();
+    if (((device & primaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
+            mInputs.activeInputsCountOnDevices(primaryInputDevices) == 0) {
+        SoundTrigger::setCaptureState(false);
+    }
 }
 
 SortedVector<audio_io_handle_t> AudioPolicyManager::getOutputsForDevice(
@@ -4696,18 +4799,17 @@
     return outputs;
 }
 
-bool AudioPolicyManager::vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
-                                      SortedVector<audio_io_handle_t>& outputs2)
+void AudioPolicyManager::checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked)
 {
-    if (outputs1.size() != outputs2.size()) {
-        return false;
+    // checkA2dpSuspend must run before checkOutputForAllStrategies so that A2DP
+    // output is suspended before any tracks are moved to it
+    checkA2dpSuspend();
+    checkOutputForAllStrategies();
+    if (onOutputsChecked != nullptr && onOutputsChecked()) checkA2dpSuspend();
+    updateDevicesAndOutputs();
+    if (mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_MSD) != 0) {
+        setMsdPatch();
     }
-    for (size_t i = 0; i < outputs1.size(); i++) {
-        if (outputs1[i] != outputs2[i]) {
-            return false;
-        }
-    }
-    return true;
 }
 
 void AudioPolicyManager::checkOutputForStrategy(routing_strategy strategy)
@@ -4735,7 +4837,7 @@
         }
     }
 
-    if (!vectorsEqual(srcOutputs,dstOutputs)) {
+    if (srcOutputs != dstOutputs) {
         // get maximum latency of all source outputs to determine the minimum mute time guaranteeing
         // audio from invalidated tracks will be rendered when unmuting
         uint32_t maxLatency = 0;
@@ -4754,7 +4856,7 @@
                 setStrategyMute(strategy, true, desc);
                 setStrategyMute(strategy, false, desc, maxLatency * LATENCY_MUTE_FACTOR, newDevice);
             }
-            sp<AudioSourceDescriptor> source =
+            sp<SourceClientDescriptor> source =
                     getSourceForStrategyOnOutput(srcOut, strategy);
             if (source != 0){
                 connectAudioSource(source);
@@ -5259,54 +5361,20 @@
     } 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
@@ -5364,53 +5432,21 @@
     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;
@@ -5544,8 +5580,8 @@
         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:
@@ -5557,9 +5593,9 @@
         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",
@@ -5630,6 +5666,21 @@
     return volumeDB;
 }
 
+int AudioPolicyManager::rescaleVolumeIndex(int srcIndex,
+                                           audio_stream_type_t srcStream,
+                                           audio_stream_type_t dstStream)
+{
+    if (srcStream == dstStream) {
+        return srcIndex;
+    }
+    float minSrc = (float)mVolumeCurves->getVolumeIndexMin(srcStream);
+    float maxSrc = (float)mVolumeCurves->getVolumeIndexMax(srcStream);
+    float minDst = (float)mVolumeCurves->getVolumeIndexMin(dstStream);
+    float maxDst = (float)mVolumeCurves->getVolumeIndexMax(dstStream);
+
+    return (int)(minDst + ((srcIndex - minSrc) * (maxDst - minDst)) / (maxSrc - minSrc));
+}
+
 status_t AudioPolicyManager::checkAndSetVolume(audio_stream_type_t stream,
                                                    int index,
                                                    const sp<AudioOutputDescriptor>& outputDesc,
@@ -5756,55 +5807,6 @@
     }
 }
 
-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
@@ -5818,39 +5820,7 @@
         return AUDIO_STREAM_TTS;
     }
 
-    // usage to stream type mapping
-    switch (attr->usage) {
-    case AUDIO_USAGE_MEDIA:
-    case AUDIO_USAGE_GAME:
-    case AUDIO_USAGE_ASSISTANT:
-    case AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE:
-        return AUDIO_STREAM_MUSIC;
-    case AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY:
-        return AUDIO_STREAM_ACCESSIBILITY;
-    case AUDIO_USAGE_ASSISTANCE_SONIFICATION:
-        return AUDIO_STREAM_SYSTEM;
-    case AUDIO_USAGE_VOICE_COMMUNICATION:
-        return AUDIO_STREAM_VOICE_CALL;
-
-    case AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING:
-        return AUDIO_STREAM_DTMF;
-
-    case AUDIO_USAGE_ALARM:
-        return AUDIO_STREAM_ALARM;
-    case AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE:
-        return AUDIO_STREAM_RING;
-
-    case AUDIO_USAGE_NOTIFICATION:
-    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST:
-    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT:
-    case AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED:
-    case AUDIO_USAGE_NOTIFICATION_EVENT:
-        return AUDIO_STREAM_NOTIFICATION;
-
-    case AUDIO_USAGE_UNKNOWN:
-    default:
-        return AUDIO_STREAM_MUSIC;
-    }
+    return audio_usage_to_stream_type(attr->usage);
 }
 
 bool AudioPolicyManager::isValidAttributes(const audio_attributes_t *paa)
@@ -5935,10 +5905,10 @@
 void AudioPolicyManager::cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc)
 {
     for (ssize_t i = (ssize_t)mAudioSources.size() - 1; i >= 0; i--)  {
-        sp<AudioSourceDescriptor> sourceDesc = mAudioSources.valueAt(i);
-        if (sourceDesc->mDevice->equals(deviceDesc)) {
-            ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->getHandle());
-            stopAudioSource(sourceDesc->getHandle());
+        sp<SourceClientDescriptor> sourceDesc = mAudioSources.valueAt(i);
+        if (sourceDesc->srcDevice()->equals(deviceDesc)) {
+            ALOGV("%s releasing audio source %d", __FUNCTION__, sourceDesc->portId());
+            stopAudioSource(sourceDesc->portId());
         }
     }
 
@@ -6180,4 +6150,58 @@
     }
 }
 
+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..9436767 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -17,6 +17,7 @@
 #pragma once
 
 #include <atomic>
+#include <functional>
 #include <memory>
 #include <unordered_set>
 
@@ -30,6 +31,7 @@
 #include <utils/SortedVector.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicy.h>
+#include <media/PatchBuilder.h>
 #include "AudioPolicyInterface.h"
 
 #include <AudioPolicyManagerInterface.h>
@@ -38,6 +40,7 @@
 #include <AudioPolicyConfig.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
+#include <AudioProfile.h>
 #include <DeviceDescriptor.h>
 #include <IOProfile.h>
 #include <HwModule.h>
@@ -118,15 +121,9 @@
                                           audio_output_flags_t *flags,
                                           audio_port_handle_t *selectedDeviceId,
                                           audio_port_handle_t *portId);
-        virtual status_t startOutput(audio_io_handle_t output,
-                                     audio_stream_type_t stream,
-                                     audio_session_t session);
-        virtual status_t stopOutput(audio_io_handle_t output,
-                                    audio_stream_type_t stream,
-                                    audio_session_t session);
-        virtual void releaseOutput(audio_io_handle_t output,
-                                   audio_stream_type_t stream,
-                                   audio_session_t session);
+        virtual status_t startOutput(audio_port_handle_t portId);
+        virtual status_t stopOutput(audio_port_handle_t portId);
+        virtual void releaseOutput(audio_port_handle_t portId);
         virtual status_t getInputForAttr(const audio_attributes_t *attr,
                                          audio_io_handle_t *input,
                                          audio_session_t session,
@@ -138,16 +135,13 @@
                                          audio_port_handle_t *portId);
 
         // indicates to the audio policy manager that the input starts being used.
-        virtual status_t startInput(audio_io_handle_t input,
-                                    audio_session_t session,
+        virtual status_t startInput(audio_port_handle_t portId,
                                     bool silenced,
                                     concurrency_type__mask_t *concurrency);
 
         // indicates to the audio policy manager that the input stops being used.
-        virtual status_t stopInput(audio_io_handle_t input,
-                                   audio_session_t session);
-        virtual void releaseInput(audio_io_handle_t input,
-                                  audio_session_t session);
+        virtual status_t stopInput(audio_port_handle_t portId);
+        virtual void releaseInput(audio_port_handle_t portId);
         virtual void closeAllInputs();
         virtual void initStreamVolume(audio_stream_type_t stream,
                                                     int indexMin,
@@ -229,9 +223,9 @@
 
         virtual status_t startAudioSource(const struct audio_port_config *source,
                                           const audio_attributes_t *attributes,
-                                          audio_patch_handle_t *handle,
+                                          audio_port_handle_t *portId,
                                           uid_t uid);
-        virtual status_t stopAudioSource(audio_patch_handle_t handle);
+        virtual status_t stopAudioSource(audio_port_handle_t portId);
 
         virtual status_t setMasterMono(bool mono);
         virtual status_t getMasterMono(bool *mono);
@@ -353,6 +347,10 @@
                                     int index,
                                     audio_devices_t device);
 
+        // rescale volume index from srcStream within range of dstStream
+        int rescaleVolumeIndex(int srcIndex,
+                               audio_stream_type_t srcStream,
+                               audio_stream_type_t dstStream);
         // check that volume change is permitted, compute and send new volume to audio hardware
         virtual status_t checkAndSetVolume(audio_stream_type_t stream, int index,
                                            const sp<AudioOutputDescriptor>& outputDesc,
@@ -377,10 +375,6 @@
                            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
@@ -410,6 +404,12 @@
         // close an input.
         void closeInput(audio_io_handle_t input);
 
+        // runs all the checks required for accomodating changes in devices and outputs
+        // if 'onOutputsChecked' callback is provided, it is executed after the outputs
+        // check via 'checkOutputForAllStrategies'. If the callback returns 'true',
+        // A2DP suspend status is rechecked.
+        void checkForDeviceAndOutputChanges(std::function<bool()> onOutputsChecked = nullptr);
+
         // checks and if necessary changes outputs used for all strategies.
         // must be called every time a condition that affects the output choice for a given strategy
         // changes: connected device, phone state, force use...
@@ -451,8 +451,6 @@
 
         SortedVector<audio_io_handle_t> getOutputsForDevice(audio_devices_t device,
                                                             const SwAudioOutputCollection& openOutputs);
-        bool vectorsEqual(SortedVector<audio_io_handle_t>& outputs1,
-                                           SortedVector<audio_io_handle_t>& outputs2);
 
         // mute/unmute strategies using an incompatible device combination
         // if muting, wait for the audio in pcm buffer to be drained before proceeding
@@ -501,13 +499,14 @@
             if (!hasPrimaryOutput()) {
                 return AUDIO_DEVICE_NONE;
             }
-            return mAvailableInputDevices.getDevicesFromHwModule(mPrimaryOutput->getModuleHandle());
+            return mAvailableInputDevices.getDeviceTypesFromHwModule(
+                    mPrimaryOutput->getModuleHandle());
         }
 
         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) const;
 
         // if argument "device" is different from AUDIO_DEVICE_NONE,  startSource() will force
         // the re-evaluation of the output device.
@@ -526,10 +525,10 @@
 
         status_t hasPrimaryOutput() const { return mPrimaryOutput != 0; }
 
-        status_t connectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc);
-        status_t disconnectAudioSource(const sp<AudioSourceDescriptor>& sourceDesc);
+        status_t connectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
+        status_t disconnectAudioSource(const sp<SourceClientDescriptor>& sourceDesc);
 
-        sp<AudioSourceDescriptor> getSourceForStrategyOnOutput(audio_io_handle_t output,
+        sp<SourceClientDescriptor> getSourceForStrategyOnOutput(audio_io_handle_t output,
                                                                routing_strategy strategy);
 
         void cleanUpForDevice(const sp<DeviceDescriptor>& deviceDesc);
@@ -544,7 +543,11 @@
         static bool streamsMatchForvolume(audio_stream_type_t stream1,
                                           audio_stream_type_t stream2);
 
-        uid_t mUidCached;
+        void closeSessions(const sp<AudioInputDescriptor>& input, bool activeOnly);
+        void closeSession(const sp<AudioInputDescriptor>& input,
+                          const sp<AudioSession>& session);
+
+        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
@@ -584,7 +587,7 @@
         sp<AudioPatch> mCallRxPatch;
 
         HwAudioOutputCollection mHwOutputs;
-        AudioSourceCollection mAudioSources;
+        SourceClientCollection mAudioSources;
 
         // for supporting "beacon" streams, i.e. streams that only play on speaker, and never
         // when something other than STREAM_TTS (a.k.a. "Transmitted Through Speaker") is playing
@@ -617,6 +620,17 @@
 
         status_t getSupportedFormats(audio_io_handle_t ioHandle, FormatVector& formats);
 
+        // Support for Multi-Stream Decoder (MSD) module
+        sp<DeviceDescriptor> getMsdAudioInDevice() const;
+        audio_devices_t getMsdAudioOutDeviceTypes() const;
+        const AudioPatchCollection getMsdPatches() const;
+        status_t getBestMsdAudioProfileFor(audio_devices_t outputDevice,
+                                           bool hwAvSync,
+                                           audio_port_config *sourceConfig,
+                                           audio_port_config *sinkConfig) const;
+        PatchBuilder buildMsdPatch(audio_devices_t outputDevice) const;
+        status_t setMsdPatch(audio_devices_t outputDevice = AUDIO_DEVICE_NONE);
+
         // If any, resolve any "dynamic" fields of an Audio Profiles collection
         void updateAudioProfiles(audio_devices_t device, audio_io_handle_t ioHandle,
                 AudioProfileVector &profiles);
@@ -684,6 +698,18 @@
             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..2858aad 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -23,7 +23,9 @@
 #include <memory>
 #include <cutils/misc.h>
 #include <media/AudioEffect.h>
+#include <media/AudioPolicyHelper.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 +33,6 @@
 #include <cutils/config_utils.h>
 #include <binder/IPCThreadState.h>
 #include "AudioPolicyEffects.h"
-#include "ServiceUtilities.h"
 
 namespace android {
 
@@ -317,6 +318,102 @@
     return status;
 }
 
+status_t AudioPolicyEffects::addStreamDefaultEffect(const effect_uuid_t *type,
+                                                    const String16& opPackageName,
+                                                    const effect_uuid_t *uuid,
+                                                    int32_t priority,
+                                                    audio_usage_t usage,
+                                                    audio_unique_id_t* id)
+{
+    if (uuid == NULL || type == NULL) {
+        ALOGE("addStreamDefaultEffect(): Null uuid or type uuid pointer");
+        return BAD_VALUE;
+    }
+
+    audio_stream_type_t stream = audio_usage_to_stream_type(usage);
+
+    if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_PUBLIC_CNT) {
+        ALOGE("addStreamDefaultEffect(): Unsupported stream type %d", stream);
+        return BAD_VALUE;
+    }
+
+    // Check that |uuid| or |type| corresponds to an effect on the system.
+    effect_descriptor_t descriptor = {};
+    status_t res = AudioEffect::getEffectDescriptor(
+            uuid, type, EFFECT_FLAG_TYPE_INSERT, &descriptor);
+    if (res != OK) {
+        ALOGE("addStreamDefaultEffect(): Failed to find effect descriptor matching uuid/type.");
+        return res;
+    }
+
+    // Only insert effects can be added dynamically as stream defaults.
+    if ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) != EFFECT_FLAG_TYPE_INSERT) {
+        ALOGE("addStreamDefaultEffect(): Desired effect cannot be attached "
+              "as a stream default effect.");
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock _l(mLock);
+
+    // Find the EffectDescVector for the given stream type, or create a new one if necessary.
+    ssize_t index = mOutputStreams.indexOfKey(stream);
+    EffectDescVector *desc = NULL;
+    if (index < 0) {
+        // No effects for this stream type yet.
+        desc = new EffectDescVector();
+        mOutputStreams.add(stream, desc);
+    } else {
+        desc = mOutputStreams.valueAt(index);
+    }
+
+    // Create a new effect and add it to the vector.
+    res = AudioEffect::newEffectUniqueId(id);
+    if (res != OK) {
+        ALOGE("addStreamDefaultEffect(): failed to get new unique id.");
+        return res;
+    }
+    EffectDesc *effect = new EffectDesc(
+            descriptor.name, *type, opPackageName, *uuid, priority, *id);
+    desc->mEffects.add(effect);
+    // TODO(b/71813697): Support setting params as well.
+
+    // TODO(b/71814300): Retroactively attach to any existing streams of the given type.
+    // This requires tracking the stream type of each session id in addition to what is
+    // already being tracked.
+
+    return NO_ERROR;
+}
+
+status_t AudioPolicyEffects::removeStreamDefaultEffect(audio_unique_id_t id)
+{
+    if (id == AUDIO_UNIQUE_ID_ALLOCATE) {
+        // ALLOCATE is not a unique identifier, but rather a reserved value indicating
+        // a real id has not been assigned. For default effects, this value is only used
+        // by system-owned defaults from the loaded config, which cannot be removed.
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock _l(mLock);
+
+    // Check each stream type.
+    size_t numStreams = mOutputStreams.size();
+    for (size_t i = 0; i < numStreams; ++i) {
+        // Check each effect for each stream.
+        EffectDescVector* descVector = mOutputStreams[i];
+        for (auto desc = descVector->mEffects.begin(); desc != descVector->mEffects.end(); ++desc) {
+            if ((*desc)->mId == id) {
+                // Found it!
+                // TODO(b/71814300): Remove from any streams the effect was attached to.
+                descVector->mEffects.erase(desc);
+                // Handles are unique; there can only be one match, so return early.
+                return NO_ERROR;
+            }
+        }
+    }
+
+    // Effect wasn't found, so it's been trivially removed successfully.
+    return NO_ERROR;
+}
 
 void AudioPolicyEffects::EffectVector::setProcessorEnabled(bool enabled)
 {
diff --git a/services/audiopolicy/service/AudioPolicyEffects.h b/services/audiopolicy/service/AudioPolicyEffects.h
index 623180e..69367b1 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.h
+++ b/services/audiopolicy/service/AudioPolicyEffects.h
@@ -64,7 +64,6 @@
     status_t releaseInputEffects(audio_io_handle_t input,
                                  audio_session_t audioSession);
 
-
     // Return a list of effect descriptors for default output effects
     // associated with audioSession
     status_t queryDefaultOutputSessionEffects(audio_session_t audioSession,
@@ -82,18 +81,49 @@
                              audio_stream_type_t stream,
                              audio_session_t audioSession);
 
+    // Add the effect to the list of default effects for streams of type |stream|.
+    status_t addStreamDefaultEffect(const effect_uuid_t *type,
+                                    const String16& opPackageName,
+                                    const effect_uuid_t *uuid,
+                                    int32_t priority,
+                                    audio_usage_t usage,
+                                    audio_unique_id_t* id);
+
+    // Remove the default stream effect from wherever it's attached.
+    status_t removeStreamDefaultEffect(audio_unique_id_t id);
+
 private:
 
     // class to store the description of an effects and its parameters
     // as defined in audio_effects.conf
     class EffectDesc {
     public:
-        EffectDesc(const char *name, const effect_uuid_t& uuid) :
+        EffectDesc(const char *name,
+                   const effect_uuid_t& typeUuid,
+                   const String16& opPackageName,
+                   const effect_uuid_t& uuid,
+                   uint32_t priority,
+                   audio_unique_id_t id) :
                         mName(strdup(name)),
-                        mUuid(uuid) { }
+                        mTypeUuid(typeUuid),
+                        mOpPackageName(opPackageName),
+                        mUuid(uuid),
+                        mPriority(priority),
+                        mId(id) { }
+        EffectDesc(const char *name, const effect_uuid_t& uuid) :
+                        EffectDesc(name,
+                                   *EFFECT_UUID_NULL,
+                                   String16(""),
+                                   uuid,
+                                   0,
+                                   AUDIO_UNIQUE_ID_ALLOCATE) { }
         EffectDesc(const EffectDesc& orig) :
                         mName(strdup(orig.mName)),
-                        mUuid(orig.mUuid) {
+                        mTypeUuid(orig.mTypeUuid),
+                        mOpPackageName(orig.mOpPackageName),
+                        mUuid(orig.mUuid),
+                        mPriority(orig.mPriority),
+                        mId(orig.mId) {
                             // deep copy mParams
                             for (size_t k = 0; k < orig.mParams.size(); k++) {
                                 effect_param_t *origParam = orig.mParams[k];
@@ -116,7 +146,11 @@
             }
         }
         char *mName;
+        effect_uuid_t mTypeUuid;
+        String16 mOpPackageName;
         effect_uuid_t mUuid;
+        int32_t mPriority;
+        audio_unique_id_t mId;
         Vector <effect_param_t *> mParams;
     };
 
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 7337f04..3439c9b 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -17,12 +17,12 @@
 #define LOG_TAG "AudioPolicyIntefaceImpl"
 //#define LOG_NDEBUG 0
 
-#include <utils/Log.h>
-#include <media/MediaAnalyticsItem.h>
-
 #include "AudioPolicyService.h"
-#include "ServiceUtilities.h"
 #include "TypeConverter.h"
+#include <media/AudioPolicyHelper.h>
+#include <media/MediaAnalyticsItem.h>
+#include <mediautils/ServiceUtilities.h>
+#include <utils/Log.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;
@@ -200,7 +200,7 @@
         !modifyPhoneStateAllowed(pid, uid)) {
         // If the app tries to play music through the telephony device and doesn't have permission
         // the fallback to the default output device.
-        mAudioPolicyManager->releaseOutput(*output, *stream, session);
+        mAudioPolicyManager->releaseOutput(*portId);
         flags = originalFlags;
         *selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
         *portId = AUDIO_PORT_HANDLE_NONE;
@@ -208,93 +208,116 @@
                                                  config,
                                                  &flags, selectedDeviceId, portId);
     }
+
+    if (result == NO_ERROR) {
+        sp <AudioPlaybackClient> client =
+            new AudioPlaybackClient(*attr, *output, uid, pid, session, *selectedDeviceId, *stream);
+        mAudioPlaybackClients.add(*portId, client);
+    }
     return result;
 }
 
-status_t AudioPolicyService::startOutput(audio_io_handle_t output,
-                                         audio_stream_type_t stream,
-                                         audio_session_t session)
+status_t AudioPolicyService::startOutput(audio_port_handle_t portId)
 {
-    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
-        return BAD_VALUE;
-    }
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
     ALOGV("startOutput()");
+    sp<AudioPlaybackClient> client;
     sp<AudioPolicyEffects>audioPolicyEffects;
     {
         Mutex::Autolock _l(mLock);
+        const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+        if (index < 0) {
+            ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+            return INVALID_OPERATION;
+        }
+        client = mAudioPlaybackClients.valueAt(index);
         audioPolicyEffects = mAudioPolicyEffects;
     }
     if (audioPolicyEffects != 0) {
         // create audio processors according to stream
-        status_t status = audioPolicyEffects->addOutputSessionEffects(output, stream, session);
+        status_t status = audioPolicyEffects->addOutputSessionEffects(
+            client->io, client->stream, client->session);
         if (status != NO_ERROR && status != ALREADY_EXISTS) {
-            ALOGW("Failed to add effects on session %d", session);
+            ALOGW("Failed to add effects on session %d", client->session);
         }
     }
     Mutex::Autolock _l(mLock);
     AutoCallerClear acc;
-    return mAudioPolicyManager->startOutput(output, stream, session);
+    status_t status = mAudioPolicyManager->startOutput(portId);
+    if (status == NO_ERROR) {
+        client->active = true;
+    }
+    return status;
 }
 
-status_t AudioPolicyService::stopOutput(audio_io_handle_t output,
-                                        audio_stream_type_t stream,
-                                        audio_session_t session)
+status_t AudioPolicyService::stopOutput(audio_port_handle_t portId)
 {
-    if (uint32_t(stream) >= AUDIO_STREAM_CNT) {
-        return BAD_VALUE;
-    }
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
     ALOGV("stopOutput()");
-    mOutputCommandThread->stopOutputCommand(output, stream, session);
+    mOutputCommandThread->stopOutputCommand(portId);
     return NO_ERROR;
 }
 
-status_t  AudioPolicyService::doStopOutput(audio_io_handle_t output,
-                                      audio_stream_type_t stream,
-                                      audio_session_t session)
+status_t  AudioPolicyService::doStopOutput(audio_port_handle_t portId)
 {
-    ALOGV("doStopOutput from tid %d", gettid());
+    ALOGV("doStopOutput");
+    sp<AudioPlaybackClient> client;
     sp<AudioPolicyEffects>audioPolicyEffects;
     {
         Mutex::Autolock _l(mLock);
+
+        const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+        if (index < 0) {
+            ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+            return INVALID_OPERATION;
+        }
+        client = mAudioPlaybackClients.valueAt(index);
         audioPolicyEffects = mAudioPolicyEffects;
     }
     if (audioPolicyEffects != 0) {
         // release audio processors from the stream
-        status_t status = audioPolicyEffects->releaseOutputSessionEffects(output, stream, session);
+        status_t status = audioPolicyEffects->releaseOutputSessionEffects(
+            client->io, client->stream, client->session);
         if (status != NO_ERROR && status != ALREADY_EXISTS) {
-            ALOGW("Failed to release effects on session %d", session);
+            ALOGW("Failed to release effects on session %d", client->session);
         }
     }
     Mutex::Autolock _l(mLock);
     AutoCallerClear acc;
-    return mAudioPolicyManager->stopOutput(output, stream, session);
+    status_t status = mAudioPolicyManager->stopOutput(portId);
+    if (status == NO_ERROR) {
+        client->active = false;
+    }
+    return status;
 }
 
-void AudioPolicyService::releaseOutput(audio_io_handle_t output,
-                                       audio_stream_type_t stream,
-                                       audio_session_t session)
+void AudioPolicyService::releaseOutput(audio_port_handle_t portId)
 {
     if (mAudioPolicyManager == NULL) {
         return;
     }
     ALOGV("releaseOutput()");
-    mOutputCommandThread->releaseOutputCommand(output, stream, session);
+    mOutputCommandThread->releaseOutputCommand(portId);
 }
 
-void AudioPolicyService::doReleaseOutput(audio_io_handle_t output,
-                                         audio_stream_type_t stream,
-                                         audio_session_t session)
+void AudioPolicyService::doReleaseOutput(audio_port_handle_t portId)
 {
     ALOGV("doReleaseOutput from tid %d", gettid());
     Mutex::Autolock _l(mLock);
+    const ssize_t index = mAudioPlaybackClients.indexOfKey(portId);
+    if (index < 0) {
+        ALOGE("%s AudioTrack client not found for portId %d", __FUNCTION__, portId);
+        return;
+    }
+    sp<AudioPlaybackClient> client = mAudioPlaybackClients.valueAt(index);
+    mAudioRecordClients.removeItem(portId);
+
     // called from internal thread: no need to clear caller identity
-    mAudioPolicyManager->releaseOutput(output, stream, session);
+    mAudioPolicyManager->releaseOutput(portId);
 }
 
 status_t AudioPolicyService::getInputForAttr(const audio_attributes_t *attr,
@@ -320,7 +343,7 @@
 
     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;
@@ -398,17 +421,13 @@
         if (status != NO_ERROR) {
             if (status == PERMISSION_DENIED) {
                 AutoCallerClear acc;
-                mAudioPolicyManager->releaseInput(*input, session);
+                mAudioPolicyManager->releaseInput(*portId);
             }
             return status;
         }
 
-        sp<AudioRecordClient> client =
-                new AudioRecordClient(*attr, *input, uid, pid, opPackageName, session);
-        client->active = false;
-        client->isConcurrent = false;
-        client->isVirtualDevice = false; //TODO : update from APM->getInputForAttr()
-        client->deviceId = *selectedDeviceId;
+        sp<AudioRecordClient> client = new AudioRecordClient(*attr, *input, uid, pid, session,
+                                                             *selectedDeviceId, opPackageName);
         mAudioRecordClients.add(*portId, client);
     }
 
@@ -496,8 +515,7 @@
     status_t status;
     {
         AutoCallerClear acc;
-        status = mAudioPolicyManager->startInput(
-                    client->input, client->session, *silenced, &concurrency);
+        status = mAudioPolicyManager->startInput(portId, *silenced, &concurrency);
 
     }
 
@@ -528,10 +546,13 @@
 
             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 +571,14 @@
                         // 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());
                     }
@@ -602,7 +628,7 @@
     // finish the recording app op
     finishRecording(client->opPackageName, client->uid);
     AutoCallerClear acc;
-    return mAudioPolicyManager->stopInput(client->input, client->session);
+    return mAudioPolicyManager->stopInput(portId);
 }
 
 void AudioPolicyService::releaseInput(audio_port_handle_t portId)
@@ -627,15 +653,15 @@
     }
     if (audioPolicyEffects != 0) {
         // release audio processors from the input
-        status_t status = audioPolicyEffects->releaseInputEffects(client->input, client->session);
+        status_t status = audioPolicyEffects->releaseInputEffects(client->io, client->session);
         if(status != NO_ERROR) {
-            ALOGW("Failed to release effects on input %d", client->input);
+            ALOGW("Failed to release effects on input %d", client->io);
         }
     }
     {
         Mutex::Autolock _l(mLock);
         AutoCallerClear acc;
-        mAudioPolicyManager->releaseInput(client->input, client->session);
+        mAudioPolicyManager->releaseInput(portId);
     }
 }
 
@@ -824,6 +850,50 @@
             (audio_session_t)audioSession, descriptors, count);
 }
 
+status_t AudioPolicyService::addStreamDefaultEffect(const effect_uuid_t *type,
+                                                    const String16& opPackageName,
+                                                    const effect_uuid_t *uuid,
+                                                    int32_t priority,
+                                                    audio_usage_t usage,
+                                                    audio_unique_id_t* id)
+{
+    if (mAudioPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!modifyDefaultAudioEffectsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    sp<AudioPolicyEffects>audioPolicyEffects;
+    {
+        Mutex::Autolock _l(mLock);
+        audioPolicyEffects = mAudioPolicyEffects;
+    }
+    if (audioPolicyEffects == 0) {
+        return NO_INIT;
+    }
+    return audioPolicyEffects->addStreamDefaultEffect(
+            type, opPackageName, uuid, priority, usage, id);
+}
+
+status_t AudioPolicyService::removeStreamDefaultEffect(audio_unique_id_t id)
+{
+    if (mAudioPolicyManager == NULL) {
+        return NO_INIT;
+    }
+    if (!modifyDefaultAudioEffectsAllowed()) {
+        return PERMISSION_DENIED;
+    }
+    sp<AudioPolicyEffects>audioPolicyEffects;
+    {
+        Mutex::Autolock _l(mLock);
+        audioPolicyEffects = mAudioPolicyEffects;
+    }
+    if (audioPolicyEffects == 0) {
+        return NO_INIT;
+    }
+    return audioPolicyEffects->removeStreamDefaultEffect(id);
+}
+
 bool AudioPolicyService::isOffloadSupported(const audio_offload_info_t& info)
 {
     if (mAudioPolicyManager == NULL) {
@@ -955,26 +1025,26 @@
 }
 
 status_t AudioPolicyService::startAudioSource(const struct audio_port_config *source,
-                                  const audio_attributes_t *attributes,
-                                  audio_patch_handle_t *handle)
+                                              const audio_attributes_t *attributes,
+                                              audio_port_handle_t *portId)
 {
     Mutex::Autolock _l(mLock);
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
     AutoCallerClear acc;
-    return mAudioPolicyManager->startAudioSource(source, attributes, handle,
+    return mAudioPolicyManager->startAudioSource(source, attributes, portId,
                                                  IPCThreadState::self()->getCallingUid());
 }
 
-status_t AudioPolicyService::stopAudioSource(audio_patch_handle_t handle)
+status_t AudioPolicyService::stopAudioSource(audio_port_handle_t portId)
 {
     Mutex::Autolock _l(mLock);
     if (mAudioPolicyManager == NULL) {
         return NO_INIT;
     }
     AutoCallerClear acc;
-    return mAudioPolicyManager->stopAudioSource(handle);
+    return mAudioPolicyManager->stopAudioSource(portId);
 }
 
 status_t AudioPolicyService::setMasterMono(bool mono)
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index f3cddc3..8bca221 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, \
@@ -716,26 +695,26 @@
                     }break;
                 case STOP_OUTPUT: {
                     StopOutputData *data = (StopOutputData *)command->mParam.get();
-                    ALOGV("AudioCommandThread() processing stop output %d",
-                            data->mIO);
+                    ALOGV("AudioCommandThread() processing stop output portId %d",
+                            data->mPortId);
                     svc = mService.promote();
                     if (svc == 0) {
                         break;
                     }
                     mLock.unlock();
-                    svc->doStopOutput(data->mIO, data->mStream, data->mSession);
+                    svc->doStopOutput(data->mPortId);
                     mLock.lock();
                     }break;
                 case RELEASE_OUTPUT: {
                     ReleaseOutputData *data = (ReleaseOutputData *)command->mParam.get();
-                    ALOGV("AudioCommandThread() processing release output %d",
-                            data->mIO);
+                    ALOGV("AudioCommandThread() processing release output portId %d",
+                            data->mPortId);
                     svc = mService.promote();
                     if (svc == 0) {
                         break;
                     }
                     mLock.unlock();
-                    svc->doReleaseOutput(data->mIO, data->mStream, data->mSession);
+                    svc->doReleaseOutput(data->mPortId);
                     mLock.lock();
                     }break;
                 case CREATE_AUDIO_PATCH: {
@@ -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,
@@ -967,33 +925,25 @@
     return sendCommand(command, delayMs);
 }
 
-void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_io_handle_t output,
-                                                               audio_stream_type_t stream,
-                                                               audio_session_t session)
+void AudioPolicyService::AudioCommandThread::stopOutputCommand(audio_port_handle_t portId)
 {
     sp<AudioCommand> command = new AudioCommand();
     command->mCommand = STOP_OUTPUT;
     sp<StopOutputData> data = new StopOutputData();
-    data->mIO = output;
-    data->mStream = stream;
-    data->mSession = session;
+    data->mPortId = portId;
     command->mParam = data;
-    ALOGV("AudioCommandThread() adding stop output %d", output);
+    ALOGV("AudioCommandThread() adding stop output portId %d", portId);
     sendCommand(command);
 }
 
-void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_io_handle_t output,
-                                                                  audio_stream_type_t stream,
-                                                                  audio_session_t session)
+void AudioPolicyService::AudioCommandThread::releaseOutputCommand(audio_port_handle_t portId)
 {
     sp<AudioCommand> command = new AudioCommand();
     command->mCommand = RELEASE_OUTPUT;
     sp<ReleaseOutputData> data = new ReleaseOutputData();
-    data->mIO = output;
-    data->mStream = stream;
-    data->mSession = session;
+    data->mPortId = portId;
     command->mParam = data;
-    ALOGV("AudioCommandThread() adding release output %d", output);
+    ALOGV("AudioCommandThread() adding release output portId %d", portId);
     sendCommand(command);
 }
 
@@ -1212,6 +1162,7 @@
                 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 +1207,6 @@
 
         } break;
 
-        case START_TONE:
-        case STOP_TONE:
         default:
             break;
         }
@@ -1330,27 +1279,6 @@
                                                    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 +1334,6 @@
 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..44c0347 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -38,8 +38,6 @@
 
 namespace android {
 
-using namespace std;
-
 // ----------------------------------------------------------------------------
 
 class AudioPolicyService :
@@ -83,15 +81,9 @@
                                       audio_output_flags_t flags,
                                       audio_port_handle_t *selectedDeviceId,
                                       audio_port_handle_t *portId);
-    virtual status_t startOutput(audio_io_handle_t output,
-                                 audio_stream_type_t stream,
-                                 audio_session_t session);
-    virtual status_t stopOutput(audio_io_handle_t output,
-                                audio_stream_type_t stream,
-                                audio_session_t session);
-    virtual void releaseOutput(audio_io_handle_t output,
-                               audio_stream_type_t stream,
-                               audio_session_t session);
+    virtual status_t startOutput(audio_port_handle_t portId);
+    virtual status_t stopOutput(audio_port_handle_t portId);
+    virtual void releaseOutput(audio_port_handle_t portId);
     virtual status_t getInputForAttr(const audio_attributes_t *attr,
                                      audio_io_handle_t *input,
                                      audio_session_t session,
@@ -134,6 +126,14 @@
     virtual status_t queryDefaultPreProcessing(audio_session_t audioSession,
                                               effect_descriptor_t *descriptors,
                                               uint32_t *count);
+    virtual status_t addStreamDefaultEffect(const effect_uuid_t *type,
+                                            const String16& opPackageName,
+                                            const effect_uuid_t *uuid,
+                                            int32_t priority,
+                                            audio_usage_t usage,
+                                            audio_unique_id_t* id);
+    virtual status_t removeStreamDefaultEffect(audio_unique_id_t id);
+
     virtual     status_t    onTransact(
                                 uint32_t code,
                                 const Parcel& data,
@@ -159,8 +159,6 @@
                                      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);
 
@@ -194,8 +192,8 @@
 
     virtual status_t startAudioSource(const struct audio_port_config *source,
                                       const audio_attributes_t *attributes,
-                                      audio_patch_handle_t *handle);
-    virtual status_t stopAudioSource(audio_patch_handle_t handle);
+                                      audio_port_handle_t *portId);
+    virtual status_t stopAudioSource(audio_port_handle_t portId);
 
     virtual status_t setMasterMono(bool mono);
     virtual status_t getMasterMono(bool *mono);
@@ -209,12 +207,8 @@
                                         bool reported);
     virtual status_t setSurroundFormatEnabled(audio_format_t audioFormat, bool enabled);
 
-            status_t doStopOutput(audio_io_handle_t output,
-                                  audio_stream_type_t stream,
-                                  audio_session_t session);
-            void doReleaseOutput(audio_io_handle_t output,
-                                 audio_stream_type_t stream,
-                                 audio_session_t session);
+            status_t doStopOutput(audio_port_handle_t portId);
+            void doReleaseOutput(audio_port_handle_t portId);
 
             status_t clientCreateAudioPatch(const struct audio_patch *patch,
                                       audio_patch_handle_t *handle,
@@ -224,7 +218,7 @@
             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 +287,6 @@
         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 +300,7 @@
         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 +309,6 @@
 
         // commands for tone AudioCommand
         enum {
-            START_TONE,
-            STOP_TONE,
             SET_VOLUME,
             SET_PARAMETERS,
             SET_VOICE_VOLUME,
@@ -345,20 +333,13 @@
         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,
                                             const char *keyValuePairs, int delayMs = 0);
                     status_t    voiceVolumeCommand(float volume, int delayMs = 0);
-                    void        stopOutputCommand(audio_io_handle_t output,
-                                                  audio_stream_type_t stream,
-                                                  audio_session_t session);
-                    void        releaseOutputCommand(audio_io_handle_t output,
-                                                     audio_stream_type_t stream,
-                                                     audio_session_t session);
+                    void        stopOutputCommand(audio_port_handle_t portId);
+                    void        releaseOutputCommand(audio_port_handle_t portId);
                     status_t    sendCommand(sp<AudioCommand>& command, int delayMs = 0);
                     void        insertCommand_l(sp<AudioCommand>& command, int delayMs = 0);
                     status_t    createAudioPatchCommand(const struct audio_patch *patch,
@@ -390,7 +371,7 @@
 
             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 +387,6 @@
             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;
@@ -432,16 +407,12 @@
 
         class StopOutputData : public AudioCommandData {
         public:
-            audio_io_handle_t mIO;
-            audio_stream_type_t mStream;
-            audio_session_t mSession;
+            audio_port_handle_t mPortId;
         };
 
         class ReleaseOutputData : public AudioCommandData {
         public:
-            audio_io_handle_t mIO;
-            audio_stream_type_t mStream;
-            audio_session_t mSession;
+            audio_port_handle_t mPortId;
         };
 
         class CreateAudioPatchData : public AudioCommandData {
@@ -478,7 +449,6 @@
         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 +523,6 @@
         // 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 +562,7 @@
     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 +575,10 @@
                                         audio_patch_handle_t patchHandle);
                             void      setAudioPortCallbacksEnabled(bool enabled);
 
+                            uid_t uid() {
+                                return mUid;
+                            }
+
                 // IBinder::DeathRecipient
                 virtual     void        binderDied(const wp<IBinder>& who);
 
@@ -619,34 +588,61 @@
 
         const wp<AudioPolicyService>        mService;
         const uid_t                         mUid;
+        const pid_t                         mPid;
         const sp<IAudioPolicyServiceClient> mAudioPolicyServiceClient;
               bool                          mAudioPortCallbacksEnabled;
     };
 
+    class AudioClient : public virtual RefBase {
+    public:
+                AudioClient(const audio_attributes_t attributes,
+                            const audio_io_handle_t io, uid_t uid, pid_t pid,
+                            const audio_session_t session, const audio_port_handle_t deviceId) :
+                                attributes(attributes), io(io), uid(uid), pid(pid),
+                                session(session), deviceId(deviceId), active(false) {}
+                ~AudioClient() override = default;
+
+
+        const audio_attributes_t attributes; // source, flags ...
+        const audio_io_handle_t io;          // audio HAL stream IO handle
+        const uid_t uid;                     // client UID
+        const pid_t pid;                     // client PID
+        const audio_session_t session;       // audio session ID
+        const audio_port_handle_t deviceId;  // selected input device port ID
+              bool active;                   // Playback/Capture is active or inactive
+    };
+
     // --- AudioRecordClient ---
     // Information about each registered AudioRecord client
     // (between calls to getInputForAttr() and releaseInput())
-    class AudioRecordClient : public RefBase {
+    class AudioRecordClient : public AudioClient {
     public:
                 AudioRecordClient(const audio_attributes_t attributes,
-                                  const audio_io_handle_t input, uid_t uid, pid_t pid,
-                                  const String16& opPackageName, const audio_session_t session) :
-                                      attributes(attributes),
-                                      input(input), uid(uid), pid(pid),
-                                      opPackageName(opPackageName), session(session),
-                                      active(false), isConcurrent(false), isVirtualDevice(false) {}
-        virtual ~AudioRecordClient() {}
+                          const audio_io_handle_t io, uid_t uid, pid_t pid,
+                          const audio_session_t session, const audio_port_handle_t deviceId,
+                          const String16& opPackageName) :
+                    AudioClient(attributes, io, uid, pid, session, deviceId),
+                    opPackageName(opPackageName), isConcurrent(false), isVirtualDevice(false) {}
+                ~AudioRecordClient() override = default;
 
-        const audio_attributes_t attributes; // source, flags ...
-        const audio_io_handle_t input;       // audio HAL input IO handle
-        const uid_t uid;                     // client UID
-        const pid_t pid;                     // client PID
         const String16 opPackageName;        // client package name
-        const audio_session_t session;       // audio session ID
-        bool active;                   // Capture is active or inactive
         bool isConcurrent;             // is allowed to concurrent capture
         bool isVirtualDevice;          // uses virtual device: updated by APM::getInputForAttr()
-        audio_port_handle_t deviceId;  // selected input device port ID
+    };
+
+    // --- AudioPlaybackClient ---
+    // Information about each registered AudioTrack client
+    // (between calls to getOutputForAttr() and releaseOutput())
+    class AudioPlaybackClient : public AudioClient {
+    public:
+                AudioPlaybackClient(const audio_attributes_t attributes,
+                      const audio_io_handle_t io, uid_t uid, pid_t pid,
+                            const audio_session_t session, audio_port_handle_t deviceId,
+                            audio_stream_type_t stream) :
+                    AudioClient(attributes, io, uid, pid, session, deviceId), stream(stream) {}
+                ~AudioPlaybackClient() override = default;
+
+        const audio_stream_type_t stream;
     };
 
     // A class automatically clearing and restoring binder caller identity inside
@@ -676,14 +672,13 @@
     // 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;
@@ -691,6 +686,7 @@
 
     sp<UidPolicy> mUidPolicy;
     DefaultKeyedVector< audio_port_handle_t, sp<AudioRecordClient> >   mAudioRecordClients;
+    DefaultKeyedVector< audio_port_handle_t, sp<AudioPlaybackClient> >   mAudioPlaybackClients;
 };
 
 } // namespace android
diff --git a/services/audiopolicy/tests/Android.mk b/services/audiopolicy/tests/Android.mk
index a43daea..b739b88 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 \
@@ -30,3 +29,26 @@
 LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
 
 include $(BUILD_NATIVE_TEST)
+
+# system/audio.h utilities test
+
+include $(CLEAR_VARS)
+
+LOCAL_SHARED_LIBRARIES := \
+  libbase \
+  liblog \
+  libmedia_helper \
+  libutils
+
+LOCAL_SRC_FILES := \
+  systemaudio_tests.cpp \
+
+LOCAL_MODULE := systemaudio_tests
+
+LOCAL_MODULE_TAGS := tests
+
+LOCAL_CFLAGS := -Werror -Wall
+
+LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
+
+include $(BUILD_NATIVE_TEST)
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/audiopolicy/tests/systemaudio_tests.cpp b/services/audiopolicy/tests/systemaudio_tests.cpp
new file mode 100644
index 0000000..abaae52
--- /dev/null
+++ b/services/audiopolicy/tests/systemaudio_tests.cpp
@@ -0,0 +1,117 @@
+/*
+ * 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.
+ */
+
+#include <gtest/gtest.h>
+
+#define LOG_TAG "SysAudio_Test"
+#include <log/log.h>
+#include <media/PatchBuilder.h>
+#include <system/audio.h>
+
+using namespace android;
+
+TEST(SystemAudioTest, PatchInvalid) {
+    audio_patch patch{};
+    ASSERT_FALSE(audio_patch_is_valid(&patch));
+    patch.num_sources = AUDIO_PATCH_PORTS_MAX + 1;
+    patch.num_sinks = 1;
+    ASSERT_FALSE(audio_patch_is_valid(&patch));
+    patch.num_sources = 1;
+    patch.num_sinks = AUDIO_PATCH_PORTS_MAX + 1;
+    ASSERT_FALSE(audio_patch_is_valid(&patch));
+    patch.num_sources = 0;
+    patch.num_sinks = 1;
+    ASSERT_FALSE(audio_patch_is_valid(&patch));
+}
+
+TEST(SystemAudioTest, PatchValid) {
+    const audio_port_config src = {
+        .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE };
+    // It's OK not to have sinks.
+    ASSERT_TRUE(audio_patch_is_valid((PatchBuilder{}).addSource(src).patch()));
+    const audio_port_config sink = {
+        .id = 2, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_DEVICE };
+    ASSERT_TRUE(audio_patch_is_valid((PatchBuilder{}).addSource(src).addSink(sink).patch()));
+    ASSERT_TRUE(audio_patch_is_valid(
+                    (PatchBuilder{}).addSource(src).addSource(src).addSink(sink).patch()));
+    ASSERT_TRUE(audio_patch_is_valid(
+                    (PatchBuilder{}).addSource(src).addSink(sink).addSink(sink).patch()));
+    ASSERT_TRUE(audio_patch_is_valid(
+                    (PatchBuilder{}).addSource(src).addSource(src).
+                    addSink(sink).addSink(sink).patch()));
+}
+
+TEST(SystemAudioTest, PatchHwAvSync) {
+    audio_port_config device_src_cfg = {
+        .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE };
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_src_cfg));
+    device_src_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_src_cfg));
+    device_src_cfg.flags.input = AUDIO_INPUT_FLAG_HW_AV_SYNC;
+    ASSERT_TRUE(audio_port_config_has_hw_av_sync(&device_src_cfg));
+
+    audio_port_config device_sink_cfg = {
+        .id = 1, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_DEVICE };
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_sink_cfg));
+    device_sink_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&device_sink_cfg));
+    device_sink_cfg.flags.output = AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
+    ASSERT_TRUE(audio_port_config_has_hw_av_sync(&device_sink_cfg));
+
+    audio_port_config mix_sink_cfg = {
+        .id = 1, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_MIX };
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_sink_cfg));
+    mix_sink_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_sink_cfg));
+    mix_sink_cfg.flags.input = AUDIO_INPUT_FLAG_HW_AV_SYNC;
+    ASSERT_TRUE(audio_port_config_has_hw_av_sync(&mix_sink_cfg));
+
+    audio_port_config mix_src_cfg = {
+        .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_MIX };
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_src_cfg));
+    mix_src_cfg.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+    ASSERT_FALSE(audio_port_config_has_hw_av_sync(&mix_src_cfg));
+    mix_src_cfg.flags.output = AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
+    ASSERT_TRUE(audio_port_config_has_hw_av_sync(&mix_src_cfg));
+}
+
+TEST(SystemAudioTest, PatchEqual) {
+    const audio_patch patch1{}, patch2{};
+    // Invalid patches are not equal.
+    ASSERT_FALSE(audio_patches_are_equal(&patch1, &patch2));
+    const audio_port_config src = {
+        .id = 1, .role = AUDIO_PORT_ROLE_SOURCE, .type = AUDIO_PORT_TYPE_DEVICE };
+    const audio_port_config sink = {
+        .id = 2, .role = AUDIO_PORT_ROLE_SINK, .type = AUDIO_PORT_TYPE_DEVICE };
+    ASSERT_FALSE(audio_patches_are_equal(
+                    (PatchBuilder{}).addSource(src).patch(),
+                    (PatchBuilder{}).addSource(src).addSink(sink).patch()));
+    ASSERT_TRUE(audio_patches_are_equal(
+                    (PatchBuilder{}).addSource(src).addSink(sink).patch(),
+                    (PatchBuilder{}).addSource(src).addSink(sink).patch()));
+    ASSERT_FALSE(audio_patches_are_equal(
+                    (PatchBuilder{}).addSource(src).addSink(sink).patch(),
+                    (PatchBuilder{}).addSource(src).addSource(src).addSink(sink).patch()));
+    audio_port_config sink_hw_av_sync = sink;
+    sink_hw_av_sync.config_mask |= AUDIO_PORT_CONFIG_FLAGS;
+    sink_hw_av_sync.flags.output = AUDIO_OUTPUT_FLAG_HW_AV_SYNC;
+    ASSERT_FALSE(audio_patches_are_equal(
+                    (PatchBuilder{}).addSource(src).addSink(sink).patch(),
+                    (PatchBuilder{}).addSource(src).addSink(sink_hw_av_sync).patch()));
+    ASSERT_TRUE(audio_patches_are_equal(
+                    (PatchBuilder{}).addSource(src).addSink(sink_hw_av_sync).patch(),
+                    (PatchBuilder{}).addSource(src).addSink(sink_hw_av_sync).patch()));
+}
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index c41de82..b85dd51 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -582,7 +582,7 @@
 Status CameraService::makeClient(const sp<CameraService>& cameraService,
         const sp<IInterface>& cameraCb, const String16& packageName, const String8& cameraId,
         int api1CameraId, int facing, int clientPid, uid_t clientUid, int servicePid,
-        bool legacyMode, int halVersion, int deviceVersion, apiLevel effectiveApiLevel,
+        int halVersion, int deviceVersion, apiLevel effectiveApiLevel,
         /*out*/sp<BasicClient>* client) {
 
     if (halVersion < 0 || halVersion == deviceVersion) {
@@ -594,7 +594,7 @@
                 sp<ICameraClient> tmp = static_cast<ICameraClient*>(cameraCb.get());
                 *client = new CameraClient(cameraService, tmp, packageName,
                         api1CameraId, facing, clientPid, clientUid,
-                        getpid(), legacyMode);
+                        getpid());
             } else { // Camera2 API route
                 ALOGW("Camera using old HAL version: %d", deviceVersion);
                 return STATUS_ERROR_FMT(ERROR_DEPRECATED_HAL,
@@ -612,7 +612,7 @@
                 *client = new Camera2Client(cameraService, tmp, packageName,
                         cameraId, api1CameraId,
                         facing, clientPid, clientUid,
-                        servicePid, legacyMode);
+                        servicePid);
             } else { // Camera2 API route
                 sp<hardware::camera2::ICameraDeviceCallbacks> tmp =
                         static_cast<hardware::camera2::ICameraDeviceCallbacks*>(cameraCb.get());
@@ -636,7 +636,7 @@
             sp<ICameraClient> tmp = static_cast<ICameraClient*>(cameraCb.get());
             *client = new CameraClient(cameraService, tmp, packageName,
                     api1CameraId, facing, clientPid, clientUid,
-                    servicePid, legacyMode);
+                    servicePid);
         } else {
             // Other combinations (e.g. HAL3.x open as HAL2.x) are not supported yet.
             ALOGE("Invalid camera HAL version %x: HAL %x device can only be"
@@ -735,8 +735,7 @@
             sp<ICameraClient>{nullptr}, id, cameraId,
             static_cast<int>(CAMERA_HAL_API_VERSION_UNSPECIFIED),
             internalPackageName, uid, USE_CALLING_PID,
-            API_1, /*legacyMode*/ false, /*shimUpdateOnly*/ true,
-            /*out*/ tmp)
+            API_1, /*shimUpdateOnly*/ true, /*out*/ tmp)
             ).isOk()) {
         ALOGE("%s: Error initializing shim metadata: %s", __FUNCTION__, ret.toString8().string());
     }
@@ -1200,8 +1199,7 @@
     sp<Client> client = nullptr;
     ret = connectHelper<ICameraClient,Client>(cameraClient, id, api1CameraId,
             CAMERA_HAL_API_VERSION_UNSPECIFIED, clientPackageName, clientUid, clientPid, API_1,
-            /*legacyMode*/ false, /*shimUpdateOnly*/ false,
-            /*out*/client);
+            /*shimUpdateOnly*/ false, /*out*/client);
 
     if(!ret.isOk()) {
         logRejected(id, getCallingPid(), String8(clientPackageName),
@@ -1227,8 +1225,7 @@
     Status ret = Status::ok();
     sp<Client> client = nullptr;
     ret = connectHelper<ICameraClient,Client>(cameraClient, id, api1CameraId, halVersion,
-            clientPackageName, clientUid, USE_CALLING_PID, API_1,
-            /*legacyMode*/ true, /*shimUpdateOnly*/ false,
+            clientPackageName, clientUid, USE_CALLING_PID, API_1, /*shimUpdateOnly*/ false,
             /*out*/client);
 
     if(!ret.isOk()) {
@@ -1256,9 +1253,7 @@
     ret = connectHelper<hardware::camera2::ICameraDeviceCallbacks,CameraDeviceClient>(cameraCb, id,
             /*api1CameraId*/-1,
             CAMERA_HAL_API_VERSION_UNSPECIFIED, clientPackageName,
-            clientUid, USE_CALLING_PID, API_2,
-            /*legacyMode*/ false, /*shimUpdateOnly*/ false,
-            /*out*/client);
+            clientUid, USE_CALLING_PID, API_2, /*shimUpdateOnly*/ false, /*out*/client);
 
     if(!ret.isOk()) {
         logRejected(id, getCallingPid(), String8(clientPackageName),
@@ -1273,7 +1268,7 @@
 template<class CALLBACK, class CLIENT>
 Status CameraService::connectHelper(const sp<CALLBACK>& cameraCb, const String8& cameraId,
         int api1CameraId, int halVersion, const String16& clientPackageName, int clientUid,
-        int clientPid, apiLevel effectiveApiLevel, bool legacyMode, bool shimUpdateOnly,
+        int clientPid, apiLevel effectiveApiLevel, bool shimUpdateOnly,
         /*out*/sp<CLIENT>& device) {
     binder::Status ret = binder::Status::ok();
 
@@ -1358,7 +1353,7 @@
         sp<BasicClient> tmp = nullptr;
         if(!(ret = makeClient(this, cameraCb, clientPackageName,
                 cameraId, api1CameraId, facing,
-                clientPid, clientUid, getpid(), legacyMode,
+                clientPid, clientUid, getpid(),
                 halVersion, deviceVersion, effectiveApiLevel,
                 /*out*/&tmp)).isOk()) {
             return ret;
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 8d4bcdb..e4a18d3 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -585,8 +585,7 @@
     template<class CALLBACK, class CLIENT>
     binder::Status connectHelper(const sp<CALLBACK>& cameraCb, const String8& cameraId,
             int api1CameraId, int halVersion, const String16& clientPackageName,
-            int clientUid, int clientPid,
-            apiLevel effectiveApiLevel, bool legacyMode, bool shimUpdateOnly,
+            int clientUid, int clientPid, apiLevel effectiveApiLevel, bool shimUpdateOnly,
             /*out*/sp<CLIENT>& device);
 
     // Lock guarding camera service state
@@ -844,7 +843,7 @@
     static binder::Status makeClient(const sp<CameraService>& cameraService,
             const sp<IInterface>& cameraCb, const String16& packageName, const String8& cameraId,
             int api1CameraId, int facing, int clientPid, uid_t clientUid, int servicePid,
-            bool legacyMode, int halVersion, int deviceVersion, apiLevel effectiveApiLevel,
+            int halVersion, int deviceVersion, apiLevel effectiveApiLevel,
             /*out*/sp<BasicClient>* client);
 
     status_t checkCameraAccess(const String16& opPackageName);
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index c8b3c2f..261cdbf 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -54,8 +54,7 @@
         int cameraFacing,
         int clientPid,
         uid_t clientUid,
-        int servicePid,
-        bool legacyMode):
+        int servicePid):
         Camera2ClientBase(cameraService, cameraClient, clientPackageName,
                 cameraDeviceId, api1CameraId, cameraFacing,
                 clientPid, clientUid, servicePid),
@@ -65,8 +64,6 @@
 
     SharedParameters::Lock l(mParameters);
     l.mParameters.state = Parameters::DISCONNECTED;
-
-    mLegacyMode = legacyMode;
 }
 
 status_t Camera2Client::initialize(sp<CameraProviderManager> manager, const String8& monitorTags) {
@@ -1443,7 +1440,7 @@
     return OK;
 }
 
-status_t Camera2Client::takePicture(int msgType) {
+status_t Camera2Client::takePicture(int /*msgType*/) {
     ATRACE_CALL();
     Mutex::Autolock icl(mBinderSerializationLock);
     status_t res;
@@ -1542,7 +1539,7 @@
     // Need HAL to have correct settings before (possibly) triggering precapture
     syncWithDevice();
 
-    res = mCaptureSequencer->startCapture(msgType);
+    res = mCaptureSequencer->startCapture();
     if (res != OK) {
         ALOGE("%s: Camera %d: Unable to start capture: %s (%d)",
                 __FUNCTION__, mCameraId, strerror(-res), res);
@@ -1662,27 +1659,6 @@
         return OK;
     }
 
-    // the camera2 api legacy mode can unconditionally disable the shutter sound
-    if (mLegacyMode) {
-        ALOGV("%s: Disable shutter sound in legacy mode", __FUNCTION__);
-        l.mParameters.playShutterSound = false;
-        return OK;
-    }
-
-    // Disabling shutter sound may not be allowed. In that case only
-    // allow the mediaserver process to disable the sound.
-    char value[PROPERTY_VALUE_MAX];
-    property_get("ro.camera.sound.forced", value, "0");
-    if (strncmp(value, "0", 2) != 0) {
-        // Disabling shutter sound is not allowed. Deny if the current
-        // process is not mediaserver.
-        if (getCallingPid() != getpid()) {
-            ALOGE("Failed to disable shutter sound. Permission denied (pid %d)",
-                    getCallingPid());
-            return PERMISSION_DENIED;
-        }
-    }
-
     l.mParameters.playShutterSound = false;
     return OK;
 }
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index 44929c3..a9ea271 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -96,8 +96,7 @@
             int cameraFacing,
             int clientPid,
             uid_t clientUid,
-            int servicePid,
-            bool legacyMode);
+            int servicePid);
 
     virtual ~Camera2Client();
 
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index f1203f9..ce44efe 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -40,7 +40,7 @@
         const String16& clientPackageName,
         int cameraId, int cameraFacing,
         int clientPid, int clientUid,
-        int servicePid, bool legacyMode):
+        int servicePid):
         Client(cameraService, cameraClient, clientPackageName,
                 String8::format("%d", cameraId), cameraId, cameraFacing, clientPid,
                 clientUid, servicePid)
@@ -57,7 +57,6 @@
     // Callback is disabled by default
     mPreviewCallbackFlag = CAMERA_FRAME_CALLBACK_FLAG_NOOP;
     mOrientation = getOrientation(0, mCameraFacing == CAMERA_FACING_FRONT);
-    mLegacyMode = legacyMode;
     mPlayShutterSound = true;
     LOG1("CameraClient::CameraClient X (pid %d, id %d)", callingPid, cameraId);
 }
@@ -715,26 +714,6 @@
         return OK;
     }
 
-    // the camera2 api legacy mode can unconditionally disable the shutter sound
-    if (mLegacyMode) {
-        ALOGV("%s: Disable shutter sound in legacy mode", __FUNCTION__);
-        mPlayShutterSound = false;
-        return OK;
-    }
-
-    // Disabling shutter sound may not be allowed. In that case only
-    // allow the mediaserver process to disable the sound.
-    char value[PROPERTY_VALUE_MAX];
-    property_get("ro.camera.sound.forced", value, "0");
-    if (strcmp(value, "0") != 0) {
-        // Disabling shutter sound is not allowed. Deny if the current
-        // process is not mediaserver.
-        if (getCallingPid() != getpid()) {
-            ALOGE("Failed to disable shutter sound. Permission denied (pid %d)", getCallingPid());
-            return PERMISSION_DENIED;
-        }
-    }
-
     mPlayShutterSound = false;
     return OK;
 }
diff --git a/services/camera/libcameraservice/api1/CameraClient.h b/services/camera/libcameraservice/api1/CameraClient.h
index 1910536..9530b6c 100644
--- a/services/camera/libcameraservice/api1/CameraClient.h
+++ b/services/camera/libcameraservice/api1/CameraClient.h
@@ -68,8 +68,7 @@
             int cameraFacing,
             int clientPid,
             int clientUid,
-            int servicePid,
-            bool legacyMode = false);
+            int servicePid);
     ~CameraClient();
 
     virtual status_t initialize(sp<CameraProviderManager> manager,
diff --git a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
index 1ee216f..f42cdd3 100644
--- a/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
+++ b/services/camera/libcameraservice/api1/client2/CaptureSequencer.cpp
@@ -50,8 +50,7 @@
         mStateTransitionCount(0),
         mTriggerId(0),
         mTimeoutCount(0),
-        mCaptureId(Camera2Client::kCaptureRequestIdStart),
-        mMsgType(0) {
+        mCaptureId(Camera2Client::kCaptureRequestIdStart) {
     ALOGV("%s", __FUNCTION__);
 }
 
@@ -64,7 +63,7 @@
     mZslProcessor = processor;
 }
 
-status_t CaptureSequencer::startCapture(int msgType) {
+status_t CaptureSequencer::startCapture() {
     ALOGV("%s", __FUNCTION__);
     ATRACE_CALL();
     Mutex::Autolock l(mInputMutex);
@@ -73,7 +72,6 @@
         return INVALID_OPERATION;
     }
     if (!mStartCapture) {
-        mMsgType = msgType;
         mStartCapture = true;
         mStartCaptureSignal.signal();
     }
@@ -386,7 +384,7 @@
 
     SharedParameters::Lock l(client->getParameters());
     /* warning: this also locks a SharedCameraCallbacks */
-    shutterNotifyLocked(l.mParameters, client, mMsgType);
+    shutterNotifyLocked(l.mParameters, client);
     mShutterNotified = true;
     mTimeoutCount = kMaxTimeoutsForCaptureEnd;
     return STANDARD_CAPTURE_WAIT;
@@ -610,7 +608,7 @@
         if (!mShutterNotified) {
             SharedParameters::Lock l(client->getParameters());
             /* warning: this also locks a SharedCameraCallbacks */
-            shutterNotifyLocked(l.mParameters, client, mMsgType);
+            shutterNotifyLocked(l.mParameters, client);
             mShutterNotified = true;
         }
     } else if (mTimeoutCount <= 0) {
@@ -715,12 +713,11 @@
 }
 
 /*static*/ void CaptureSequencer::shutterNotifyLocked(const Parameters &params,
-            const sp<Camera2Client>& client, int msgType) {
+            const sp<Camera2Client>& client) {
     ATRACE_CALL();
 
     if (params.state == Parameters::STILL_CAPTURE
-        && params.playShutterSound
-        && (msgType & CAMERA_MSG_SHUTTER)) {
+        && params.playShutterSound) {
         client->getCameraService()->playSound(CameraService::SOUND_SHUTTER);
     }
 
diff --git a/services/camera/libcameraservice/api1/client2/CaptureSequencer.h b/services/camera/libcameraservice/api1/client2/CaptureSequencer.h
index f2e3750..c23b12d 100644
--- a/services/camera/libcameraservice/api1/client2/CaptureSequencer.h
+++ b/services/camera/libcameraservice/api1/client2/CaptureSequencer.h
@@ -51,7 +51,7 @@
     void setZslProcessor(const wp<ZslProcessor>& processor);
 
     // Begin still image capture
-    status_t startCapture(int msgType);
+    status_t startCapture();
 
     // Wait until current image capture completes; returns immediately if no
     // capture is active. Returns TIMED_OUT if capture does not complete during
@@ -145,7 +145,6 @@
     bool mAeInPrecapture;
 
     int32_t mCaptureId;
-    int mMsgType;
 
     // Main internal methods
 
@@ -172,7 +171,7 @@
 
     // Emit Shutter/Raw callback to java, and maybe play a shutter sound
     static void shutterNotifyLocked(const Parameters &params,
-            const sp<Camera2Client>& client, int msgType);
+            const sp<Camera2Client>& client);
 };
 
 }; // namespace camera2
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/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 98d0534..84428c2 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -661,7 +661,8 @@
         }
 
         sp<Surface> surface;
-        res = createSurfaceFromGbp(streamInfo, isStreamInfoValid, surface, bufferProducer);
+        res = createSurfaceFromGbp(streamInfo, isStreamInfoValid, surface, bufferProducer,
+                physicalCameraId);
 
         if (!res.isOk())
             return res;
@@ -889,6 +890,8 @@
 
     const std::vector<sp<IGraphicBufferProducer> >& bufferProducers =
             outputConfiguration.getGraphicBufferProducers();
+    String8 physicalCameraId(outputConfiguration.getPhysicalCameraId());
+
     auto producerCount = bufferProducers.size();
     if (producerCount == 0) {
         ALOGE("%s: bufferProducers must not be empty", __FUNCTION__);
@@ -942,7 +945,7 @@
         OutputStreamInfo outInfo;
         sp<Surface> surface;
         res = createSurfaceFromGbp(outInfo, /*isStreamInfoValid*/ false, surface,
-                newOutputsMap.valueAt(i));
+                newOutputsMap.valueAt(i), physicalCameraId);
         if (!res.isOk())
             return res;
 
@@ -1021,7 +1024,8 @@
 
 binder::Status CameraDeviceClient::createSurfaceFromGbp(
         OutputStreamInfo& streamInfo, bool isStreamInfoValid,
-        sp<Surface>& surface, const sp<IGraphicBufferProducer>& gbp) {
+        sp<Surface>& surface, const sp<IGraphicBufferProducer>& gbp,
+        const String8& physicalId) {
 
     // bufferProducer must be non-null
     if (gbp == nullptr) {
@@ -1098,7 +1102,7 @@
     // Round dimensions to the nearest dimensions available for this format
     if (flexibleConsumer && isPublicFormat(format) &&
             !CameraDeviceClient::roundBufferDimensionNearest(width, height,
-            format, dataSpace, mDevice->info(), /*out*/&width, /*out*/&height)) {
+            format, dataSpace, mDevice->info(physicalId), /*out*/&width, /*out*/&height)) {
         String8 msg = String8::format("Camera %s: No supported stream configurations with "
                 "format %#x defined, failed to create output stream",
                 mCameraIdStr.string(), format);
@@ -1468,6 +1472,7 @@
 
     const std::vector<sp<IGraphicBufferProducer> >& bufferProducers =
             outputConfiguration.getGraphicBufferProducers();
+    String8 physicalId(outputConfiguration.getPhysicalCameraId());
 
     if (bufferProducers.size() == 0) {
         ALOGE("%s: bufferProducers must not be empty", __FUNCTION__);
@@ -1521,7 +1526,7 @@
 
         sp<Surface> surface;
         res = createSurfaceFromGbp(mStreamInfoMap[streamId], true /*isStreamInfoValid*/,
-                surface, bufferProducer);
+                surface, bufferProducer, physicalId);
 
         if (!res.isOk())
             return res;
@@ -1677,9 +1682,12 @@
 
     // WORKAROUND: HAL refuses to disconnect while there's streams in flight
     {
-        mDevice->clearStreamingRequest();
-
+        int64_t lastFrameNumber;
         status_t code;
+        if ((code = mDevice->flush(&lastFrameNumber)) != OK) {
+            ALOGE("%s: flush failed with code 0x%x", __FUNCTION__, code);
+        }
+
         if ((code = mDevice->waitUntilDrained()) != OK) {
             ALOGE("%s: waitUntilDrained failed with code 0x%x", __FUNCTION__,
                   code);
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index 5aaf5aa..c30561d 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -258,7 +258,8 @@
     // Create a Surface from an IGraphicBufferProducer. Returns error if
     // IGraphicBufferProducer's property doesn't match with streamInfo
     binder::Status createSurfaceFromGbp(OutputStreamInfo& streamInfo, bool isStreamInfoValid,
-            sp<Surface>& surface, const sp<IGraphicBufferProducer>& gbp);
+            sp<Surface>& surface, const sp<IGraphicBufferProducer>& gbp,
+            const String8& physicalCameraId);
 
 
     // Utility method to insert the surface into SurfaceMap
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index 0ba7403..98c1b5e 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -69,6 +69,10 @@
      * The device's static characteristics metadata buffer
      */
     virtual const CameraMetadata& info() const = 0;
+    /**
+     * The physical camera device's static characteristics metadata buffer
+     */
+    virtual const CameraMetadata& info(const String8& physicalId) const = 0;
 
     struct PhysicalCameraSettings {
         std::string cameraId;
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.cpp b/services/camera/libcameraservice/common/CameraProviderManager.cpp
index 0ce4318..3be6399 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.cpp
+++ b/services/camera/libcameraservice/common/CameraProviderManager.cpp
@@ -336,6 +336,7 @@
         const hardware::hidl_string& /*fqName*/,
         const hardware::hidl_string& name,
         bool /*preexisting*/) {
+    std::lock_guard<std::mutex> providerLock(mProviderLifecycleLock);
     {
         std::lock_guard<std::mutex> lock(mInterfaceMutex);
 
@@ -458,6 +459,7 @@
 }
 
 status_t CameraProviderManager::removeProvider(const std::string& provider) {
+    std::lock_guard<std::mutex> providerLock(mProviderLifecycleLock);
     std::unique_lock<std::mutex> lock(mInterfaceMutex);
     std::vector<String8> removedDeviceIds;
     status_t res = NAME_NOT_FOUND;
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.h b/services/camera/libcameraservice/common/CameraProviderManager.h
index b8b8b8c..c523c2d 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.h
+++ b/services/camera/libcameraservice/common/CameraProviderManager.h
@@ -246,6 +246,9 @@
     wp<StatusListener> mListener;
     ServiceInteractionProxy* mServiceProxy;
 
+    // mProviderLifecycleLock is locked during onRegistration and removeProvider
+    mutable std::mutex mProviderLifecycleLock;
+
     static HardwareServiceInteractionProxy sHardwareServiceInteractionProxy;
 
     struct ProviderInfo :
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 28ffc8b..491ed72 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -121,11 +121,25 @@
 
     res = manager->getCameraCharacteristics(mId.string(), &mDeviceInfo);
     if (res != OK) {
-        SET_ERR_L("Could not retrive camera characteristics: %s (%d)", strerror(-res), res);
+        SET_ERR_L("Could not retrieve camera characteristics: %s (%d)", strerror(-res), res);
         session->close();
         return res;
     }
 
+    std::vector<std::string> physicalCameraIds;
+    bool isLogical = CameraProviderManager::isLogicalCamera(mDeviceInfo, &physicalCameraIds);
+    if (isLogical) {
+        for (auto& physicalId : physicalCameraIds) {
+            res = manager->getCameraCharacteristics(physicalId, &mPhysicalDeviceInfoMap[physicalId]);
+            if (res != OK) {
+                SET_ERR_L("Could not retrieve camera %s characteristics: %s (%d)",
+                        physicalId.c_str(), strerror(-res), res);
+                session->close();
+                return res;
+            }
+        }
+    }
+
     std::shared_ptr<RequestMetadataQueue> queue;
     auto requestQueueRet = session->getCaptureRequestMetadataQueue(
         [&queue](const auto& descriptor) {
@@ -263,6 +277,7 @@
 status_t Camera3Device::disconnect() {
     ATRACE_CALL();
     Mutex::Autolock il(mInterfaceLock);
+    Mutex::Autolock stLock(mTrackerLock);
 
     ALOGI("%s: E", __FUNCTION__);
 
@@ -719,7 +734,7 @@
     return OK;
 }
 
-const CameraMetadata& Camera3Device::info() const {
+const CameraMetadata& Camera3Device::info(const String8& physicalId) const {
     ALOGVV("%s: E", __FUNCTION__);
     if (CC_UNLIKELY(mStatus == STATUS_UNINITIALIZED ||
                     mStatus == STATUS_ERROR)) {
@@ -727,7 +742,22 @@
                 mStatus == STATUS_ERROR ?
                 "when in error state" : "before init");
     }
-    return mDeviceInfo;
+    if (physicalId.isEmpty()) {
+        return mDeviceInfo;
+    } else {
+        std::string id(physicalId.c_str());
+        if (mPhysicalDeviceInfoMap.find(id) != mPhysicalDeviceInfoMap.end()) {
+            return mPhysicalDeviceInfoMap.at(id);
+        } else {
+            ALOGE("%s: Invalid physical camera id %s", __FUNCTION__, physicalId.c_str());
+            return mDeviceInfo;
+        }
+    }
+}
+
+const CameraMetadata& Camera3Device::info() const {
+    String8 emptyId;
+    return info(emptyId);
 }
 
 status_t Camera3Device::checkStatusOkToCaptureLocked() {
@@ -2718,8 +2748,9 @@
     if (res < 0) return res;
 
     if (mInFlightMap.size() == 1) {
-        // hold mLock to prevent race with disconnect
-        Mutex::Autolock l(mLock);
+        // Hold a separate dedicated tracker lock to prevent race with disconnect and also
+        // avoid a deadlock during reprocess requests.
+        Mutex::Autolock l(mTrackerLock);
         if (mStatusTracker != nullptr) {
             mStatusTracker->markComponentActive(mInFlightStatusId);
         }
@@ -2752,8 +2783,9 @@
 
     // Indicate idle inFlightMap to the status tracker
     if (mInFlightMap.size() == 0) {
-        // hold mLock to prevent race with disconnect
-        Mutex::Autolock l(mLock);
+        // Hold a separate dedicated tracker lock to prevent race with disconnect and also
+        // avoid a deadlock during reprocess requests.
+        Mutex::Autolock l(mTrackerLock);
         if (mStatusTracker != nullptr) {
             mStatusTracker->markComponentIdle(mInFlightStatusId, Fence::NO_FENCE);
         }
@@ -3908,18 +3940,17 @@
     }
 
     hardware::details::return_status err;
+    auto resultCallback =
+        [&status, &numRequestProcessed] (auto s, uint32_t n) {
+                status = s;
+                *numRequestProcessed = n;
+        };
     if (hidlSession_3_4 != nullptr) {
         err = hidlSession_3_4->processCaptureRequest_3_4(captureRequests_3_4, cachesToRemove,
-            [&status, &numRequestProcessed] (auto s, uint32_t n) {
-                status = s;
-                *numRequestProcessed = n;
-            });
+                                                         resultCallback);
     } else {
         err = mHidlSession->processCaptureRequest(captureRequests, cachesToRemove,
-            [&status, &numRequestProcessed] (auto s, uint32_t n) {
-                status = s;
-                *numRequestProcessed = n;
-            });
+                                                  resultCallback);
     }
     if (!err.isOk()) {
         ALOGE("%s: Transaction error: %s", __FUNCTION__, err.description().c_str());
@@ -4729,6 +4760,7 @@
 status_t Camera3Device::RequestThread::prepareHalRequests() {
     ATRACE_CALL();
 
+    bool batchedRequest = mNextRequests[0].captureRequest->mBatchSize > 1;
     for (size_t i = 0; i < mNextRequests.size(); i++) {
         auto& nextRequest = mNextRequests.editItemAt(i);
         sp<CaptureRequest> captureRequest = nextRequest.captureRequest;
@@ -4752,7 +4784,10 @@
         mPrevTriggers = triggerCount;
 
         // If the request is the same as last, or we had triggers last time
-        bool newRequest = mPrevRequest != captureRequest || triggersMixedIn;
+        bool newRequest = (mPrevRequest != captureRequest || triggersMixedIn) &&
+                // Request settings are all the same within one batch, so only treat the first
+                // request in a batch as new
+                !(batchedRequest && i > 0);
         if (newRequest) {
             /**
              * HAL workaround:
@@ -4901,7 +4936,7 @@
         // preview), and the current request is not the last one in the batch,
         // do not send callback to the app.
         bool hasCallback = true;
-        if (mNextRequests[0].captureRequest->mBatchSize > 1 && i != mNextRequests.size()-1) {
+        if (batchedRequest && i != mNextRequests.size()-1) {
             hasCallback = false;
         }
         res = parent->registerInFlight(halRequest->frame_number,
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index d8fe19f..96212ab 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -101,6 +101,7 @@
     status_t disconnect() override;
     status_t dump(int fd, const Vector<String16> &args) override;
     const CameraMetadata& info() const override;
+    const CameraMetadata& info(const String8& physicalId) const override;
 
     // Capture and setStreamingRequest will configure streams if currently in
     // idle state
@@ -379,6 +380,7 @@
     sp<HalInterface> mInterface;
 
     CameraMetadata             mDeviceInfo;
+    std::unordered_map<std::string, CameraMetadata> mPhysicalDeviceInfoMap;
 
     CameraMetadata             mRequestTemplateCache[CAMERA3_TEMPLATE_COUNT];
 
@@ -1208,6 +1210,9 @@
 
     static callbacks_notify_t sNotify;
 
+    // Synchronizes access to status tracker between inflight updates and disconnect.
+    // b/79972865
+    Mutex mTrackerLock;
 }; // class Camera3Device
 
 }; // namespace android
diff --git a/services/camera/libcameraservice/device3/DistortionMapper.cpp b/services/camera/libcameraservice/device3/DistortionMapper.cpp
index 4dafefd..ae7af8e 100644
--- a/services/camera/libcameraservice/device3/DistortionMapper.cpp
+++ b/services/camera/libcameraservice/device3/DistortionMapper.cpp
@@ -312,8 +312,8 @@
         int32_t coords[4] = {
             rects[i],
             rects[i + 1],
-            rects[i] + rects[i + 2],
-            rects[i + 1] + rects[i + 3]
+            rects[i] + rects[i + 2] - 1,
+            rects[i + 1] + rects[i + 3] - 1
         };
 
         mapRawToCorrected(coords, 2, clamp, simple);
@@ -321,8 +321,8 @@
         // Map back to (l, t, width, height)
         rects[i] = coords[0];
         rects[i + 1] = coords[1];
-        rects[i + 2] = coords[2] - coords[0];
-        rects[i + 3] = coords[3] - coords[1];
+        rects[i + 2] = coords[2] - coords[0] + 1;
+        rects[i + 3] = coords[3] - coords[1] + 1;
     }
 
     return OK;
@@ -400,8 +400,8 @@
         int32_t coords[4] = {
             rects[i],
             rects[i + 1],
-            rects[i] + rects[i + 2],
-            rects[i + 1] + rects[i + 3]
+            rects[i] + rects[i + 2] - 1,
+            rects[i + 1] + rects[i + 3] - 1
         };
 
         mapCorrectedToRaw(coords, 2, clamp, simple);
@@ -409,8 +409,8 @@
         // Map back to (l, t, width, height)
         rects[i] = coords[0];
         rects[i + 1] = coords[1];
-        rects[i + 2] = coords[2] - coords[0];
-        rects[i + 3] = coords[3] - coords[1];
+        rects[i + 2] = coords[2] - coords[0] + 1;
+        rects[i + 3] = coords[3] - coords[1] + 1;
     }
 
     return OK;
diff --git a/services/camera/libcameraservice/tests/DistortionMapperTest.cpp b/services/camera/libcameraservice/tests/DistortionMapperTest.cpp
index 2a689c6..54935c9 100644
--- a/services/camera/libcameraservice/tests/DistortionMapperTest.cpp
+++ b/services/camera/libcameraservice/tests/DistortionMapperTest.cpp
@@ -167,6 +167,30 @@
     }
 }
 
+TEST(DistortionMapperTest, ClampConsistency) {
+    status_t res;
+
+    std::array<int32_t, 4> activeArray = {0, 0, 4032, 3024};
+    DistortionMapper m;
+    setupTestMapper(&m, identityDistortion, testICal, /*activeArray*/ activeArray.data(),
+            /*preCorrectionActiveArray*/ activeArray.data());
+
+    auto rectsOrig = activeArray;
+    res = m.mapCorrectedRectToRaw(activeArray.data(), 1, /*clamp*/true, /*simple*/ true);
+    ASSERT_EQ(res, OK);
+
+    for (size_t i = 0; i < activeArray.size(); i++) {
+        EXPECT_EQ(activeArray[i], rectsOrig[i]);
+    }
+
+    res = m.mapRawRectToCorrected(activeArray.data(), 1, /*clamp*/true, /*simple*/ true);
+    ASSERT_EQ(res, OK);
+
+    for (size_t i = 0; i < activeArray.size(); i++) {
+        EXPECT_EQ(activeArray[i], rectsOrig[i]);
+    }
+}
+
 TEST(DistortionMapperTest, SimpleTransform) {
     status_t res;
 
diff --git a/services/camera/libcameraservice/utils/TagMonitor.cpp b/services/camera/libcameraservice/utils/TagMonitor.cpp
index c0a353f..f4c49ec 100644
--- a/services/camera/libcameraservice/utils/TagMonitor.cpp
+++ b/services/camera/libcameraservice/utils/TagMonitor.cpp
@@ -49,7 +49,8 @@
     std::lock_guard<std::mutex> lock(mMonitorMutex);
 
     // Expand shorthands
-    if (ssize_t idx = tagNames.find("3a") != -1) {
+    ssize_t idx = tagNames.find("3a");
+    if (idx != -1) {
         ssize_t end = tagNames.find(",", idx);
         char* start = tagNames.lockBuffer(tagNames.size());
         start[idx] = '\0';
diff --git a/services/mediacodec/seccomp_policy/mediacodec-arm.policy b/services/mediacodec/seccomp_policy/mediacodec-arm.policy
index 6ec8895..edf4dab 100644
--- a/services/mediacodec/seccomp_policy/mediacodec-arm.policy
+++ b/services/mediacodec/seccomp_policy/mediacodec-arm.policy
@@ -55,4 +55,8 @@
 getdents64: 1
 getrandom: 1
 
+# Used by UBSan diagnostic messages
+readlink: 1
+open: 1
+
 @include /system/etc/seccomp_policy/crash_dump.arm.policy
diff --git a/services/mediacodec/seccomp_policy/mediacodec-x86.policy b/services/mediacodec/seccomp_policy/mediacodec-x86.policy
index bbbe552..4031b11 100644
--- a/services/mediacodec/seccomp_policy/mediacodec-x86.policy
+++ b/services/mediacodec/seccomp_policy/mediacodec-x86.policy
@@ -24,6 +24,7 @@
 mmap2: 1
 fstat64: 1
 stat64: 1
+statfs64: 1
 madvise: 1
 fstatat64: 1
 futex: 1
@@ -55,4 +56,8 @@
 getpid: 1
 gettid: 1
 
+# Used by UBSan diagnostic messages
+readlink: 1
+open: 1
+
 @include /system/etc/seccomp_policy/crash_dump.x86.policy
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/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index f9e21fb..f30f9bb 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -189,6 +189,7 @@
         minSizeFrames = AAUDIO_BUFFER_CAPACITY_MIN;
     }
     status = mMmapStream->createMmapBuffer(minSizeFrames, &mMmapBufferinfo);
+    bool isBufferShareable = mMmapBufferinfo.flags & AUDIO_MMAP_APPLICATION_SHAREABLE;
     if (status != OK) {
         ALOGE("%s() - createMmapBuffer() failed with status %d %s",
               __func__, status, strerror(-status));
@@ -198,18 +199,13 @@
         ALOGD("%s() createMmapBuffer() returned = %d, buffer_size = %d, burst_size %d"
                       ", Sharable FD: %s",
               __func__, status,
-              abs(mMmapBufferinfo.buffer_size_frames),
+              mMmapBufferinfo.buffer_size_frames,
               mMmapBufferinfo.burst_size_frames,
-              mMmapBufferinfo.buffer_size_frames < 0 ? "Yes" : "No");
+              isBufferShareable ? "Yes" : "No");
     }
 
     setBufferCapacity(mMmapBufferinfo.buffer_size_frames);
-    // The audio HAL indicates if the shared memory fd can be shared outside of audioserver
-    // by returning a negative buffer size
-    if (getBufferCapacity() < 0) {
-        // Exclusive mode can be used by client or service.
-        setBufferCapacity(-getBufferCapacity());
-    } else {
+    if (!isBufferShareable) {
         // Exclusive mode can only be used by the service because the FD cannot be shared.
         uid_t audioServiceUid = getuid();
         if ((mMmapClient.clientUid != audioServiceUid) &&
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) {