diff --git a/camera/camera2/OutputConfiguration.cpp b/camera/camera2/OutputConfiguration.cpp
index d6642f3..2bccd87 100644
--- a/camera/camera2/OutputConfiguration.cpp
+++ b/camera/camera2/OutputConfiguration.cpp
@@ -161,7 +161,7 @@
     }
 
     std::vector<int32_t> sensorPixelModesUsed;
-    if ((err = parcel->readParcelableVector(&sensorPixelModesUsed)) != OK) {
+    if ((err = parcel->readInt32Vector(&sensorPixelModesUsed)) != OK) {
         ALOGE("%s: Failed to read sensor pixel mode(s) from parcel", __FUNCTION__);
         return err;
     }
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 03a8dc9..803c4a4 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -16,9 +16,9 @@
         libstagefright_foundation libjpeg libui libgui libcutils liblog \
         libhidlbase libdatasource libaudioclient \
         android.hardware.media.omx@1.0 \
-        media_permission-aidl-cpp
+        framework-permission-aidl-cpp
 
-LOCAL_STATIC_LIBRARIES := media_permission-aidl-cpp
+LOCAL_STATIC_LIBRARIES := framework-permission-aidl-cpp
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
@@ -52,7 +52,7 @@
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libmedia liblog libutils libbinder \
         libstagefright_foundation libdatasource libaudioclient \
-        media_permission-aidl-cpp
+        framework-permission-aidl-cpp
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/camera/include \
@@ -90,7 +90,7 @@
         frameworks/av/media/libstagefright \
         frameworks/native/include/media/openmax \
         frameworks/native/include/media/hardware \
-        media_permission-aidl-cpp
+        framework-permission-aidl-cpp
 
 LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
 
@@ -119,7 +119,7 @@
 LOCAL_SHARED_LIBRARIES := \
         libstagefright libmedia liblog libutils libbinder \
         libstagefright_foundation libaudioclient \
-        media_permission-aidl-cpp
+        framework-permission-aidl-cpp
 
 LOCAL_C_INCLUDES:= \
         frameworks/av/media/libstagefright \
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index c86a611..4b41ff8 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -24,7 +24,7 @@
 
 #include <utils/String16.h>
 
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <binder/ProcessState.h>
 #include <media/mediarecorder.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -39,7 +39,7 @@
 
 using namespace android;
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 static void usage(const char* name)
 {
@@ -113,10 +113,10 @@
         audio_attributes_t attr = AUDIO_ATTRIBUTES_INITIALIZER;
         attr.source = AUDIO_SOURCE_MIC;
 
-        // TODO b/182392769: use identity util
+        // TODO b/182392769: use attribution source util
         source = new AudioSource(
                 &attr,
-                Identity(),
+                AttributionSourceState(),
                 sampleRate,
                 channels);
     } else {
diff --git a/media/codec2/core/include/C2Config.h b/media/codec2/core/include/C2Config.h
index 9d9ed70..7caa457 100644
--- a/media/codec2/core/include/C2Config.h
+++ b/media/codec2/core/include/C2Config.h
@@ -262,6 +262,8 @@
     kParamIndexTunneledMode, // struct
     kParamIndexTunnelHandle, // int32[]
     kParamIndexTunnelSystemTime, // int64
+    kParamIndexTunnelHoldRender, // bool
+    kParamIndexTunnelStartRender, // bool
 
     // dmabuf allocator
     kParamIndexStoreDmaBufUsage,  // store, struct
@@ -2366,6 +2368,31 @@
         C2PortTunnelSystemTime;
 constexpr char C2_PARAMKEY_OUTPUT_RENDER_TIME[] = "output.render-time";
 
+
+/**
+ * Tunneled mode video peek signaling flag.
+ *
+ * When a video frame is pushed to the decoder with this parameter set to true,
+ * the decoder must decode the frame, signal partial completion, and hold on the
+ * frame until C2StreamTunnelStartRender is set to true (which resets this
+ * flag). Flush will also result in the frames being returned back to the
+ * client (but not rendered).
+ */
+typedef C2StreamParam<C2Info, C2EasyBoolValue, kParamIndexTunnelHoldRender>
+        C2StreamTunnelHoldRender;
+constexpr char C2_PARAMKEY_TUNNEL_HOLD_RENDER[] = "output.tunnel-hold-render";
+
+/**
+ * Tunneled mode video peek signaling flag.
+ *
+ * Upon receiving this flag, the decoder shall set C2StreamTunnelHoldRender to
+ * false, which shall cause any frames held for rendering to be immediately
+ * displayed, regardless of their timestamps.
+*/
+typedef C2StreamParam<C2Info, C2EasyBoolValue, kParamIndexTunnelStartRender>
+        C2StreamTunnelStartRender;
+constexpr char C2_PARAMKEY_TUNNEL_START_RENDER[] = "output.tunnel-start-render";
+
 C2ENUM(C2PlatformConfig::encoding_quality_level_t, uint32_t,
     NONE,
     S_HANDHELD,
diff --git a/media/codec2/sfplugin/CCodec.cpp b/media/codec2/sfplugin/CCodec.cpp
index ce15a30..16398a4 100644
--- a/media/codec2/sfplugin/CCodec.cpp
+++ b/media/codec2/sfplugin/CCodec.cpp
@@ -673,6 +673,10 @@
         mCodec->mCallback->onOutputBuffersChanged();
     }
 
+    void onFirstTunnelFrameReady() override {
+        mCodec->mCallback->onFirstTunnelFrameReady();
+    }
+
 private:
     CCodec *mCodec;
 };
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 3c3b41d..f88408e 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -209,6 +209,7 @@
     int32_t flags = 0;
     int32_t tmp = 0;
     bool eos = false;
+    bool tunnelFirstFrame = false;
     if (buffer->meta()->findInt32("eos", &tmp) && tmp) {
         eos = true;
         mInputMetEos = true;
@@ -217,6 +218,9 @@
     if (buffer->meta()->findInt32("csd", &tmp) && tmp) {
         flags |= C2FrameData::FLAG_CODEC_CONFIG;
     }
+    if (buffer->meta()->findInt32("tunnel-first-frame", &tmp) && tmp) {
+        tunnelFirstFrame = true;
+    }
     ALOGV("[%s] queueInputBuffer: buffer->size() = %zu", mName, buffer->size());
     std::list<std::unique_ptr<C2Work>> items;
     std::unique_ptr<C2Work> work(new C2Work);
@@ -288,6 +292,13 @@
         // TODO: fill info's
 
         work->input.configUpdate = std::move(mParamsToBeSet);
+        if (tunnelFirstFrame) {
+            C2StreamTunnelHoldRender::input tunnelHoldRender{
+                0u /* stream */,
+                C2_TRUE /* value */
+            };
+            work->input.configUpdate.push_back(C2Param::Copy(tunnelHoldRender));
+        }
         work->worklets.clear();
         work->worklets.emplace_back(new C2Worklet);
 
@@ -1724,6 +1735,15 @@
                 }
                 break;
             }
+            case C2StreamTunnelHoldRender::CORE_INDEX: {
+                C2StreamTunnelHoldRender::output firstTunnelFrameHoldRender;
+                if (!(worklet->output.flags & C2FrameData::FLAG_INCOMPLETE)) break;
+                if (!firstTunnelFrameHoldRender.updateFrom(*param)) break;
+                if (firstTunnelFrameHoldRender.value != C2_TRUE) break;
+                ALOGV("[%s] onWorkDone: first tunnel frame ready", mName);
+                mCCodecCallback->onFirstTunnelFrameReady();
+                break;
+            }
             default:
                 ALOGV("[%s] onWorkDone: unrecognized config update (%08X)",
                       mName, param->index());
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index 45da003..5a2aca2 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -45,6 +45,7 @@
     virtual void onError(status_t err, enum ActionCode actionCode) = 0;
     virtual void onOutputFramesRendered(int64_t mediaTimeUs, nsecs_t renderTimeNs) = 0;
     virtual void onOutputBuffersChanged() = 0;
+    virtual void onFirstTunnelFrameReady() = 0;
 };
 
 /**
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 27e87e6..2df0ba2 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -938,6 +938,14 @@
             return value == 0 ? C2_FALSE : C2_TRUE;
         }));
 
+    add(ConfigMapper("android._trigger-tunnel-peek", C2_PARAMKEY_TUNNEL_START_RENDER, "value")
+        .limitTo(D::PARAM & D::VIDEO & D::DECODER)
+        .withMapper([](C2Value v) -> C2Value {
+            int32_t value = 0;
+            (void)v.get(&value);
+            return value == 0 ? C2_FALSE : C2_TRUE;
+        }));
+
     /* still to do
     constexpr char KEY_PUSH_BLANK_BUFFERS_ON_STOP[] = "push-blank-buffers-on-shutdown";
 
diff --git a/media/codecs/m4v_h263/dec/test/Android.bp b/media/codecs/m4v_h263/dec/test/Android.bp
index 6eed66f..d8de569 100644
--- a/media/codecs/m4v_h263/dec/test/Android.bp
+++ b/media/codecs/m4v_h263/dec/test/Android.bp
@@ -47,6 +47,10 @@
         },
     },
 
+    // this unit test also runs within the mainline tests (MTS),
+    // so it must be compatible back to Android Q/10 (sdk 29)
+    min_sdk_version: "29",
+
     srcs: [
         "Mpeg4H263DecoderTest.cpp",
     ],
diff --git a/media/extractors/mkv/MatroskaExtractor.cpp b/media/extractors/mkv/MatroskaExtractor.cpp
index 4fd3a56..443e26c 100644
--- a/media/extractors/mkv/MatroskaExtractor.cpp
+++ b/media/extractors/mkv/MatroskaExtractor.cpp
@@ -1700,17 +1700,17 @@
         return ERROR_MALFORMED;
     }
 
-    size_t header_start = 0;
-    size_t header_lenth = 0;
+    long header_start = 0;
+    long header_length = 0;
     for (header_start = 0; header_start < frame.len - 4; header_start++) {
         if (ntohl(0x000001b3) == *(uint32_t*)((uint8_t*)tmpData.get() + header_start)) {
             break;
         }
     }
     bool isComplete_csd = false;
-    for (header_lenth = 0; header_lenth < frame.len - 4 - header_start; header_lenth++) {
+    for (header_length = 0; header_length < frame.len - 4 - header_start; header_length++) {
         if (ntohl(0x000001b8) == *(uint32_t*)((uint8_t*)tmpData.get()
-                                + header_start + header_lenth)) {
+                                + header_start + header_length)) {
             isComplete_csd = true;
             break;
         }
@@ -1720,7 +1720,7 @@
         return ERROR_MALFORMED;
     }
     addESDSFromCodecPrivate(trackInfo->mMeta, false,
-                              (uint8_t*)(tmpData.get()) + header_start, header_lenth);
+                            (uint8_t*)(tmpData.get()) + header_start, header_length);
 
     return OK;
 
diff --git a/media/extractors/mp3/MP3Extractor.cpp b/media/extractors/mp3/MP3Extractor.cpp
index 5bbabdf..248a39c 100644
--- a/media/extractors/mp3/MP3Extractor.cpp
+++ b/media/extractors/mp3/MP3Extractor.cpp
@@ -504,7 +504,14 @@
             }
 
             mCurrentTimeUs = seekTimeUs;
-            mCurrentPos = mFirstFramePos + seekTimeUs * bitrate / 8000000;
+            int64_t seekTimeUsTimesBitrate;
+            if (__builtin_mul_overflow(seekTimeUs, bitrate, &seekTimeUsTimesBitrate)) {
+              return AMEDIA_ERROR_UNSUPPORTED;
+            }
+            if (__builtin_add_overflow(
+                    mFirstFramePos, seekTimeUsTimesBitrate / 8000000, &mCurrentPos)) {
+                return AMEDIA_ERROR_UNSUPPORTED;
+            }
             seekCBR = true;
         } else {
             mCurrentTimeUs = actualSeekTimeUs;
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index fe2d98e..33a5c7f 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -52,6 +52,7 @@
         "libcutils",
         "libutils",
         "libbinder",
+        "libpermission",
     ],
 
     sanitize: {
@@ -86,7 +87,7 @@
     export_header_lib_headers: ["libaaudio_headers"],
 
     export_shared_lib_headers: [
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     shared_libs: [
@@ -99,15 +100,11 @@
         "libcutils",
         "libutils",
         "libbinder",
+        "framework-permission-aidl-cpp",
         "aaudio-aidl-cpp",
-        "media_permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
 
-    static_libs: [
-        "media_permission-aidl-cpp",
-    ],
-
     cflags: [
         "-Wno-unused-parameter",
         "-Wall",
@@ -177,7 +174,7 @@
     imports: [
         "audio_common-aidl",
         "shared-file-region-aidl",
-        "media_permission-aidl",
+        "framework-permission-aidl"
     ],
     backend:
     {
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.cpp b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
index 5e0a4bb..8d90034 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
@@ -31,7 +31,7 @@
 
 AAudioStreamRequest::AAudioStreamRequest(const StreamRequest& parcelable) :
         mConfiguration(std::move(parcelable.params)),
-        mIdentity(parcelable.identity),
+        mAttributionSource(parcelable.attributionSource),
         mSharingModeMatchRequired(parcelable.sharingModeMatchRequired),
         mInService(parcelable.inService) {
 }
@@ -39,7 +39,7 @@
 StreamRequest AAudioStreamRequest::parcelable() const {
     StreamRequest result;
     result.params = std::move(mConfiguration).parcelable();
-    result.identity = mIdentity;
+    result.attributionSource = mAttributionSource;
     result.sharingModeMatchRequired = mSharingModeMatchRequired;
     result.inService = mInService;
     return result;
@@ -50,7 +50,7 @@
 }
 
 void AAudioStreamRequest::dump() const {
-    ALOGD("mIdentity  = %s", mIdentity.toString().c_str());
+    ALOGD("mAttributionSource  = %s", mAttributionSource.toString().c_str());
     ALOGD("mSharingModeMatchRequired = %d", mSharingModeMatchRequired);
     ALOGD("mInService = %d", mInService);
     mConfiguration.dump();
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.h b/media/libaaudio/src/binding/AAudioStreamRequest.h
index 02341c8..cc43a48 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.h
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.h
@@ -23,10 +23,12 @@
 #include <aaudio/StreamRequest.h>
 
 #include "binding/AAudioStreamConfiguration.h"
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 namespace aaudio {
 
+using android::content::AttributionSourceState;
+
 class AAudioStreamRequest {
 public:
     AAudioStreamRequest() = default;
@@ -34,12 +36,12 @@
     // Construct based on a parcelable representation.
     explicit AAudioStreamRequest(const StreamRequest& parcelable);
 
-    const android::media::permission::Identity &getIdentity() const {
-        return mIdentity;
+    const AttributionSourceState &getAttributionSource() const {
+        return mAttributionSource;
     }
 
-    void setIdentity(const android::media::permission::Identity &identity) {
-        mIdentity = identity;
+    void setAttributionSource(const AttributionSourceState &attributionSource) {
+        mAttributionSource = attributionSource;
     }
 
     bool isSharingModeMatchRequired() const {
@@ -75,7 +77,7 @@
 
 private:
     AAudioStreamConfiguration  mConfiguration;
-    android::media::permission::Identity mIdentity;
+    AttributionSourceState mAttributionSource;
     bool                       mSharingModeMatchRequired = false;
     bool                       mInService = false; // Stream opened by AAudioservice
 };
diff --git a/media/libaaudio/src/binding/aidl/aaudio/StreamRequest.aidl b/media/libaaudio/src/binding/aidl/aaudio/StreamRequest.aidl
index 12802e6..53787a0 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/StreamRequest.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/StreamRequest.aidl
@@ -17,11 +17,11 @@
 package aaudio;
 
 import aaudio.StreamParameters;
-import android.media.permission.Identity;
+import android.content.AttributionSourceState;
 
 parcelable StreamRequest {
-    StreamParameters    params;
-    Identity            identity;
-    boolean             sharingModeMatchRequired; // = false;
-    boolean             inService; // = false; // Stream opened by AAudioservice
+    StreamParameters       params;
+    AttributionSourceState attributionSource;
+    boolean                sharingModeMatchRequired; // = false;
+    boolean                inService; // = false; // Stream opened by AAudioservice
 }
\ No newline at end of file
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index 0a19d17..ebc9f2b 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -166,6 +166,10 @@
                                   ? &mDataWriteCounter
                                   : descriptor->writeCounterAddress;
 
+    // Clear buffer to avoid an initial glitch on some devices.
+    size_t bufferSizeBytes = descriptor->capacityInFrames * descriptor->bytesPerFrame;
+    memset(descriptor->dataAddress, 0, bufferSizeBytes);
+
     mDataQueue = std::make_unique<FifoBufferIndirect>(
             descriptor->bytesPerFrame,
             descriptor->capacityInFrames,
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index d8b27c3..cf2abe8 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -51,7 +51,7 @@
 
 using android::Mutex;
 using android::WrappingBuffer;
-using android::media::permission::Identity;
+using android::content::AttributionSourceState;
 
 using namespace aaudio;
 
@@ -108,15 +108,16 @@
     // Request FLOAT for the shared mixer or the device.
     request.getConfiguration().setFormat(AUDIO_FORMAT_PCM_FLOAT);
 
-    // TODO b/182392769: use identity util
-    Identity identity;
-    identity.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
-    identity.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
-    identity.packageName = builder.getOpPackageName();
-    identity.attributionTag = builder.getAttributionTag();
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.packageName = builder.getOpPackageName();
+    attributionSource.attributionTag = builder.getAttributionTag();
+    attributionSource.token = sp<android::BBinder>::make();
 
     // Build the request to send to the server.
-    request.setIdentity(identity);
+    request.setAttributionSource(attributionSource);
     request.setSharingModeMatchRequired(isSharingModeMatchRequired());
     request.setInService(isInService());
 
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index 0d60120..acfac24 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -25,8 +25,7 @@
 
 // TODO These defines should be moved to a central place in audio.
 #define SAMPLES_PER_FRAME_MIN        1
-// TODO Remove 8 channel limitation.
-#define SAMPLES_PER_FRAME_MAX        FCC_8
+#define SAMPLES_PER_FRAME_MAX        FCC_LIMIT
 #define SAMPLE_RATE_HZ_MIN           8000
 // HDMI supports up to 32 channels at 1536000 Hz.
 #define SAMPLE_RATE_HZ_MAX           1600000
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 2135c54..e015592 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -43,8 +43,7 @@
 // on the edge of being ridiculous.
 // TODO These defines should be moved to a central place in audio.
 #define SAMPLES_PER_FRAME_MIN        1
-// TODO Remove 8 channel limitation.
-#define SAMPLES_PER_FRAME_MAX        FCC_8
+#define SAMPLES_PER_FRAME_MAX        FCC_LIMIT
 #define SAMPLE_RATE_HZ_MIN           8000
 // HDMI supports up to 32 channels at 1536000 Hz.
 #define SAMPLE_RATE_HZ_MAX           1600000
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index eca5392..dc66742 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -32,7 +32,7 @@
 #include "utility/AudioClock.h"
 #include "utility/FixedBlockWriter.h"
 
-using android::media::permission::Identity;
+using android::content::AttributionSourceState;
 
 using namespace android;
 using namespace aaudio;
@@ -157,12 +157,13 @@
             .tags = ""
     };
 
-    // TODO b/182392769: use identity util
-    Identity identity;
-    identity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
-    identity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
-    identity.packageName = builder.getOpPackageName();
-    identity.attributionTag = builder.getAttributionTag();
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.packageName = builder.getOpPackageName();
+    attributionSource.attributionTag = builder.getAttributionTag();
+    attributionSource.token = sp<BBinder>::make();
 
     // ----------- open the AudioRecord ---------------------
     // Might retry, but never more than once.
@@ -170,7 +171,7 @@
         const audio_format_t requestedInternalFormat = getDeviceFormat();
 
         mAudioRecord = new AudioRecord(
-                identity
+                attributionSource
         );
         mAudioRecord->set(
                 AUDIO_SOURCE_DEFAULT, // ignored because we pass attributes below
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index 7d0a197..692651d 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -25,7 +25,7 @@
 #include "AAudioLegacy.h"
 #include "legacy/AudioStreamLegacy.h"
 #include "utility/FixedBlockWriter.h"
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 namespace aaudio {
 
@@ -87,7 +87,7 @@
     FixedBlockWriter                 mFixedBlockWriter;
 
     // TODO add 64-bit position reporting to AudioRecord and use it.
-    android::media::permission::Identity mIdentity;
+    android::content::AttributionSourceState mAttributionSource;
 
     // Only one type of conversion buffer is used.
     std::unique_ptr<float[]>         mFormatConversionBufferFloat;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 04a9dec..1d412c0 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -33,7 +33,7 @@
 using namespace android;
 using namespace aaudio;
 
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 // Arbitrary and somewhat generous number of bursts.
 #define DEFAULT_BURSTS_PER_BUFFER_CAPACITY     8
@@ -151,7 +151,7 @@
     };
 
     mAudioTrack = new AudioTrack();
-    // TODO b/182392769: use identity util
+    // TODO b/182392769: use attribution source util
     mAudioTrack->set(
             AUDIO_STREAM_DEFAULT,  // ignored because we pass attributes below
             getSampleRate(),
@@ -167,7 +167,7 @@
             sessionId,
             streamTransferType,
             NULL,    // DEFAULT audio_offload_info_t
-            Identity(), // DEFAULT uid and pid
+            AttributionSourceState(), // DEFAULT uid and pid
             &attributes,
             // WARNING - If doNotReconnect set true then audio stops after plugging and unplugging
             // headphones a few times.
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index c77aeeb..321e7f9 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -1182,7 +1182,7 @@
         const media::AudioClient& aidl) {
     AudioClient legacy;
     legacy.clientTid = VALUE_OR_RETURN(aidl2legacy_int32_t_pid_t(aidl.clientTid));
-    legacy.identity = aidl.identity;
+    legacy.attributionSource = aidl.attributionSource;
     return legacy;
 }
 
@@ -1190,7 +1190,7 @@
         const AudioClient& legacy) {
     media::AudioClient aidl;
     aidl.clientTid = VALUE_OR_RETURN(legacy2aidl_pid_t_int32_t(legacy.clientTid));
-    aidl.identity = legacy.identity;
+    aidl.attributionSource = legacy.attributionSource;
     return aidl;
 }
 
@@ -2323,4 +2323,28 @@
     return unexpected(BAD_VALUE);
 }
 
+ConversionResult<TrackSecondaryOutputInfoPair>
+aidl2legacy_TrackSecondaryOutputInfo_TrackSecondaryOutputInfoPair(
+        const media::TrackSecondaryOutputInfo& aidl) {
+    TrackSecondaryOutputInfoPair trackSecondaryOutputInfoPair;
+    trackSecondaryOutputInfoPair.first =
+            VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
+    trackSecondaryOutputInfoPair.second =
+            VALUE_OR_RETURN(convertContainer<std::vector<audio_port_handle_t>>(
+                    aidl.secondaryOutputIds, aidl2legacy_int32_t_audio_io_handle_t));
+    return trackSecondaryOutputInfoPair;
+}
+
+ConversionResult<media::TrackSecondaryOutputInfo>
+legacy2aidl_TrackSecondaryOutputInfoPair_TrackSecondaryOutputInfo(
+        const TrackSecondaryOutputInfoPair& legacy) {
+    media::TrackSecondaryOutputInfo trackSecondaryOutputInfo;
+    trackSecondaryOutputInfo.portId =
+            VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy.first));
+    trackSecondaryOutputInfo.secondaryOutputIds =
+            VALUE_OR_RETURN(convertContainer<std::vector<int32_t>>(
+                    legacy.second, legacy2aidl_audio_io_handle_t_int32_t));
+    return trackSecondaryOutputInfo;
+}
+
 }  // namespace android
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 19d68a0..9c307ff 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -132,12 +132,12 @@
         "libshmemcompat",
         "libutils",
         "libvibrator",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
     export_shared_lib_headers: [
         "audioflinger-aidl-cpp",
         "audiopolicy-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libbinder",
     ],
 
@@ -164,7 +164,6 @@
         // for memory heap analysis
         "libc_malloc_debug_backtrace",
         "shared-file-region-aidl-cpp",
-        "media_permission-aidl-cpp",
     ],
     cflags: [
         "-Wall",
@@ -232,7 +231,7 @@
         "libshmemcompat",
         "libutils",
         "shared-file-region-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
     export_shared_lib_headers: [
         "audioclient-types-aidl-cpp",
@@ -351,10 +350,11 @@
         "aidl/android/media/AudioVibratorInfo.aidl",
         "aidl/android/media/EffectDescriptor.aidl",
         "aidl/android/media/ExtraAudioDescriptor.aidl",
+        "aidl/android/media/TrackSecondaryOutputInfo.aidl",
     ],
     imports: [
         "audio_common-aidl",
-        "media_permission-aidl",
+        "framework-permission-aidl",
     ],
     backend: {
         cpp: {
@@ -436,7 +436,7 @@
         "av-types-aidl",
         "effect-aidl",
         "shared-file-region-aidl",
-        "media_permission-aidl",
+        "framework-permission-aidl",
     ],
     double_loadable: true,
     backend: {
@@ -461,7 +461,6 @@
         "aidl/android/media/GetOutputForAttrResponse.aidl",
         "aidl/android/media/Int.aidl",
         "aidl/android/media/RecordClientInfo.aidl",
-
         "aidl/android/media/IAudioPolicyService.aidl",
         "aidl/android/media/IAudioPolicyServiceClient.aidl",
     ],
@@ -470,8 +469,9 @@
         "audioclient-types-aidl",
         "audiopolicy-types-aidl",
         "capture_state_listener-aidl",
-        "media_permission-aidl",
+        "framework-permission-aidl",
     ],
+
     double_loadable: true,
     backend: {
         cpp: {
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index d5047b1..6ad5483 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -42,7 +42,6 @@
 using aidl_utils::statusTFromBinderStatus;
 using binder::Status;
 using media::IAudioPolicyService;
-using media::permission::Identity;
 
 namespace {
 
@@ -58,8 +57,8 @@
 
 // ---------------------------------------------------------------------------
 
-AudioEffect::AudioEffect(const Identity& identity)
-    : mClientIdentity(identity)
+AudioEffect::AudioEffect(const android::content::AttributionSourceState& attributionSource)
+    : mClientAttributionSource(attributionSource)
 {
 }
 
@@ -108,12 +107,12 @@
     mDescriptor.type = *(type != NULL ? type : EFFECT_UUID_NULL);
     mDescriptor.uuid = *(uuid != NULL ? uuid : EFFECT_UUID_NULL);
 
-    // TODO b/182392769: use identity util
+    // TODO b/182392769: use attribution source util
     mIEffectClient = new EffectClient(this);
     pid_t pid = IPCThreadState::self()->getCallingPid();
-    mClientIdentity.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(pid));
+    mClientAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(pid));
     pid_t uid = IPCThreadState::self()->getCallingUid();
-    mClientIdentity.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(uid));
+    mClientAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(uid));
 
     media::CreateEffectRequest request;
     request.desc = VALUE_OR_RETURN_STATUS(
@@ -123,7 +122,7 @@
     request.output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(io));
     request.sessionId = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(mSessionId));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(device));
-    request.identity = mClientIdentity;
+    request.attributionSource = mClientAttributionSource;
     request.probe = probe;
 
     media::CreateEffectResponse response;
@@ -178,7 +177,7 @@
 
     IInterface::asBinder(iEffect)->linkToDeath(mIEffectClient);
     ALOGV("set() %p OK effect: %s id: %d status %d enabled %d pid %d", this, mDescriptor.name, mId,
-            mStatus, mEnabled, mClientIdentity.pid);
+            mStatus, mEnabled, mClientAttributionSource.pid);
 
     if (!audio_is_global_session(mSessionId)) {
         AudioSystem::acquireAudioSessionId(mSessionId, pid, uid);
@@ -223,7 +222,7 @@
     if (!mProbe && (mStatus == NO_ERROR || mStatus == ALREADY_EXISTS)) {
         if (!audio_is_global_session(mSessionId)) {
             AudioSystem::releaseAudioSessionId(mSessionId,
-                VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientIdentity.pid)));
+                VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid)));
         }
         if (mIEffect != NULL) {
             mIEffect->disconnect();
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 1a4bde9..a1d3bdb 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -39,7 +39,7 @@
 
 namespace android {
 
-using android::media::permission::Identity;
+using android::content::AttributionSourceState;
 using aidl_utils::statusTFromBinderStatus;
 
 // ---------------------------------------------------------------------------
@@ -126,11 +126,11 @@
     return NO_ERROR;
 }
 
-AudioRecord::AudioRecord(const Identity &client)
-    : mActive(false), mStatus(NO_INIT), mClientIdentity(client), mSessionId(AUDIO_SESSION_ALLOCATE),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL), mPreviousSchedulingGroup(SP_DEFAULT),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE), mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mSelectedMicDirection(MIC_DIRECTION_UNSPECIFIED),
+AudioRecord::AudioRecord(const AttributionSourceState &client)
+    : mActive(false), mStatus(NO_INIT), mClientAttributionSource(client),
+      mSessionId(AUDIO_SESSION_ALLOCATE), mPreviousPriority(ANDROID_PRIORITY_NORMAL),
+      mPreviousSchedulingGroup(SP_DEFAULT), mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
+      mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE), mSelectedMicDirection(MIC_DIRECTION_UNSPECIFIED),
       mSelectedMicFieldDimension(MIC_FIELD_DIMENSION_DEFAULT)
 {
 }
@@ -140,7 +140,7 @@
         uint32_t sampleRate,
         audio_format_t format,
         audio_channel_mask_t channelMask,
-        const Identity& client,
+        const AttributionSourceState& client,
         size_t frameCount,
         callback_t cbf,
         void* user,
@@ -154,14 +154,14 @@
         float microphoneFieldDimension)
     : mActive(false),
       mStatus(NO_INIT),
-      mClientIdentity(client),
+      mClientAttributionSource(client),
       mSessionId(AUDIO_SESSION_ALLOCATE),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
       mPreviousSchedulingGroup(SP_DEFAULT),
       mProxy(NULL)
 {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientIdentity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientIdentity.pid));
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientAttributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
     (void)set(inputSource, sampleRate, format, channelMask, frameCount, cbf, user,
             notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
             uid, pid, pAttributes, selectedDeviceId, selectedMicDirection,
@@ -191,7 +191,7 @@
         IPCThreadState::self()->flushCommands();
         ALOGV("%s(%d): releasing session id %d",
                 __func__, mPortId, mSessionId);
-        pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientIdentity.pid));
+        pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
         AudioSystem::releaseAudioSessionId(mSessionId, pid);
     }
 }
@@ -243,11 +243,11 @@
 
     // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
     ALOGV("%s(): inputSource %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
-          "notificationFrames %u, sessionId %d, transferType %d, flags %#x, identity %s"
+          "notificationFrames %u, sessionId %d, transferType %d, flags %#x, attributionSource %s"
           "uid %d, pid %d",
           __func__,
           inputSource, sampleRate, format, channelMask, frameCount, notificationFrames,
-          sessionId, transferType, flags, mClientIdentity.toString().c_str(), uid, pid);
+          sessionId, transferType, flags, mClientAttributionSource.toString().c_str(), uid, pid);
 
     // TODO b/182392553: refactor or remove
     pid_t callingPid = IPCThreadState::self()->getCallingPid();
@@ -256,13 +256,13 @@
     if (pid == -1 || (callingPid != myPid)) {
         adjPid = callingPid;
     }
-    mClientIdentity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(adjPid));
+    mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(adjPid));
 
     uid_t adjUid = uid;
     if (uid == -1 || (callingPid != myPid)) {
         adjUid = IPCThreadState::self()->getCallingUid();
     }
-    mClientIdentity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(adjUid));
+    mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(adjUid));
 
     mTracker.reset(new RecordingActivityTracker());
 
@@ -801,7 +801,7 @@
     input.config.sample_rate = mSampleRate;
     input.config.channel_mask = mChannelMask;
     input.config.format = mFormat;
-    input.clientInfo.identity = mClientIdentity;
+    input.clientInfo.attributionSource = mClientAttributionSource;
     input.clientInfo.clientTid = -1;
     if (mFlags & AUDIO_INPUT_FLAG_FAST) {
         if (mAudioRecordThread != 0) {
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 0bc592d..88e752b 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -52,7 +52,7 @@
 using aidl_utils::statusTFromBinderStatus;
 using binder::Status;
 using media::IAudioPolicyService;
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 // client singleton for AudioFlinger binder interface
 Mutex AudioSystem::gLock;
@@ -71,6 +71,25 @@
 Mutex gSoundTriggerCaptureStateListenerLock;
 sp<CaptureStateListenerImpl> gSoundTriggerCaptureStateListener = nullptr;
 
+// Binder for the AudioFlinger service that's passed to this client process from the system server.
+// This allows specific isolated processes to access the audio system. Currently used only for the
+// HotwordDetectionService.
+sp<IBinder> gAudioFlingerBinder = nullptr;
+
+void AudioSystem::setAudioFlingerBinder(const sp<IBinder>& audioFlinger) {
+    if (audioFlinger->getInterfaceDescriptor() != media::IAudioFlingerService::descriptor) {
+        ALOGE("setAudioFlingerBinder: received a binder of type %s",
+              String8(audioFlinger->getInterfaceDescriptor()).string());
+        return;
+    }
+    Mutex::Autolock _l(gLock);
+    if (gAudioFlinger != nullptr) {
+        ALOGW("setAudioFlingerBinder: ignoring; AudioFlinger connection already established.");
+        return;
+    }
+    gAudioFlingerBinder = audioFlinger;
+}
+
 // establish binder interface to AudioFlinger service
 const sp<IAudioFlinger> AudioSystem::get_audio_flinger() {
     sp<IAudioFlinger> af;
@@ -79,15 +98,19 @@
     {
         Mutex::Autolock _l(gLock);
         if (gAudioFlinger == 0) {
-            sp<IServiceManager> sm = defaultServiceManager();
             sp<IBinder> binder;
-            do {
-                binder = sm->getService(String16(IAudioFlinger::DEFAULT_SERVICE_NAME));
-                if (binder != 0)
-                    break;
-                ALOGW("AudioFlinger not published, waiting...");
-                usleep(500000); // 0.5 s
-            } while (true);
+            if (gAudioFlingerBinder != nullptr) {
+                binder = gAudioFlingerBinder;
+            } else {
+                sp<IServiceManager> sm = defaultServiceManager();
+                do {
+                    binder = sm->getService(String16(IAudioFlinger::DEFAULT_SERVICE_NAME));
+                    if (binder != 0)
+                        break;
+                    ALOGW("AudioFlinger not published, waiting...");
+                    usleep(500000); // 0.5 s
+                } while (true);
+            }
             if (gAudioFlingerClient == NULL) {
                 gAudioFlingerClient = new AudioFlingerClient();
             } else {
@@ -941,7 +964,7 @@
                                        audio_io_handle_t* output,
                                        audio_session_t session,
                                        audio_stream_type_t* stream,
-                                       const Identity& identity,
+                                       const AttributionSourceState& attributionSource,
                                        const audio_config_t* config,
                                        audio_output_flags_t flags,
                                        audio_port_handle_t* selectedDeviceId,
@@ -984,7 +1007,7 @@
     media::GetOutputForAttrResponse responseAidl;
 
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
-            aps->getOutputForAttr(attrAidl, sessionAidl, identity, configAidl, flagsAidl,
+            aps->getOutputForAttr(attrAidl, sessionAidl, attributionSource, configAidl, flagsAidl,
                                   selectedDeviceIdAidl, &responseAidl)));
 
     *output = VALUE_OR_RETURN_STATUS(
@@ -1038,7 +1061,7 @@
                                       audio_io_handle_t* input,
                                       audio_unique_id_t riid,
                                       audio_session_t session,
-                                      const Identity &identity,
+                                      const AttributionSourceState &attributionSource,
                                       const audio_config_base_t* config,
                                       audio_input_flags_t flags,
                                       audio_port_handle_t* selectedDeviceId,
@@ -1077,7 +1100,7 @@
     media::GetInputForAttrResponse response;
 
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
-            aps->getInputForAttr(attrAidl, inputAidl, riidAidl, sessionAidl, identity,
+            aps->getInputForAttr(attrAidl, inputAidl, riidAidl, sessionAidl, attributionSource,
                 configAidl, flagsAidl, selectedDeviceIdAidl, &response)));
 
     *input = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.input));
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 1bc3baa..6765bdb 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -48,7 +48,7 @@
 // ---------------------------------------------------------------------------
 
 using media::VolumeShaper;
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 // TODO: Move to a separate .h
 
@@ -225,11 +225,11 @@
     return NO_ERROR;
 }
 
-AudioTrack::AudioTrack() : AudioTrack(Identity())
+AudioTrack::AudioTrack() : AudioTrack(AttributionSourceState())
 {
 }
 
-AudioTrack::AudioTrack(const Identity& identity)
+AudioTrack::AudioTrack(const AttributionSourceState& attributionSource)
     : mStatus(NO_INIT),
       mState(STATE_STOPPED),
       mPreviousPriority(ANDROID_PRIORITY_NORMAL),
@@ -237,7 +237,7 @@
       mPausedPosition(0),
       mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
       mRoutedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mClientIdentity(identity),
+      mClientAttributionSource(attributionSource),
       mAudioTrackCallback(new AudioTrackCallback())
 {
     mAttributes.content_type = AUDIO_CONTENT_TYPE_UNKNOWN;
@@ -259,7 +259,7 @@
         audio_session_t sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
-        const Identity& identity,
+        const AttributionSourceState& attributionSource,
         const audio_attributes_t* pAttributes,
         bool doNotReconnect,
         float maxRequiredSpeed,
@@ -275,8 +275,8 @@
 
     (void)set(streamType, sampleRate, format, channelMask,
             frameCount, flags, cbf, user, notificationFrames,
-            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType,
-            offloadInfo, identity, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
+            0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
+            attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
 }
 
 AudioTrack::AudioTrack(
@@ -292,7 +292,7 @@
         audio_session_t sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
-        const Identity& identity,
+        const AttributionSourceState& attributionSource,
         const audio_attributes_t* pAttributes,
         bool doNotReconnect,
         float maxRequiredSpeed)
@@ -309,7 +309,7 @@
     (void)set(streamType, sampleRate, format, channelMask,
             0 /*frameCount*/, flags, cbf, user, notificationFrames,
             sharedBuffer, false /*threadCanCallJava*/, sessionId, transferType, offloadInfo,
-            identity, pAttributes, doNotReconnect, maxRequiredSpeed);
+            attributionSource, pAttributes, doNotReconnect, maxRequiredSpeed);
 }
 
 AudioTrack::~AudioTrack()
@@ -335,7 +335,7 @@
         mCblkMemory.clear();
         mSharedBuffer.clear();
         IPCThreadState::self()->flushCommands();
-        pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientIdentity.pid));
+        pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
         ALOGV("%s(%d), releasing session id %d from %d on behalf of %d",
                 __func__, mPortId,
                 mSessionId, IPCThreadState::self()->getCallingPid(), clientPid);
@@ -381,7 +381,7 @@
         audio_session_t sessionId,
         transfer_type transferType,
         const audio_offload_info_t *offloadInfo,
-        const Identity& identity,
+        const AttributionSourceState& attributionSource,
         const audio_attributes_t* pAttributes,
         bool doNotReconnect,
         float maxRequiredSpeed,
@@ -391,15 +391,15 @@
     uint32_t channelCount;
     pid_t callingPid;
     pid_t myPid;
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
 
     // Note mPortId is not valid until the track is created, so omit mPortId in ALOG for set.
     ALOGV("%s(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
           "flags #%x, notificationFrames %d, sessionId %d, transferType %d, uid %d, pid %d",
           __func__,
           streamType, sampleRate, format, channelMask, frameCount, flags, notificationFrames,
-          sessionId, transferType, identity.uid, identity.pid);
+          sessionId, transferType, attributionSource.uid, attributionSource.pid);
 
     mThreadCanCallJava = threadCanCallJava;
     mSelectedDeviceId = selectedDeviceId;
@@ -596,18 +596,15 @@
     }
     mNotificationFramesAct = 0;
     // TODO b/182392553: refactor or remove
+    mClientAttributionSource = AttributionSourceState(attributionSource);
     callingPid = IPCThreadState::self()->getCallingPid();
     myPid = getpid();
     if (uid == -1 || (callingPid != myPid)) {
-        mClientIdentity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
+        mClientAttributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
             IPCThreadState::self()->getCallingUid()));
-    } else {
-        mClientIdentity.uid = identity.uid;
     }
     if (pid == (pid_t)-1 || (callingPid != myPid)) {
-        mClientIdentity.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
-    } else {
-        mClientIdentity.pid = identity.pid;
+        mClientAttributionSource.pid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(callingPid));
     }
     mAuxEffectId = 0;
     mOrigFlags = mFlags = flags;
@@ -692,13 +689,14 @@
         float maxRequiredSpeed,
         audio_port_handle_t selectedDeviceId)
 {
-    Identity identity;
-    identity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
-    identity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
+    AttributionSourceState attributionSource;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
+    attributionSource.token = sp<BBinder>::make();
     return set(streamType, sampleRate, format,
             static_cast<audio_channel_mask_t>(channelMask),
             frameCount, flags, cbf, user, notificationFrames, sharedBuffer,
-            threadCanCallJava, sessionId, transferType, offloadInfo, identity,
+            threadCanCallJava, sessionId, transferType, offloadInfo, attributionSource,
             pAttributes, doNotReconnect, maxRequiredSpeed, selectedDeviceId);
 }
 
@@ -1700,7 +1698,7 @@
     input.config.channel_mask = mChannelMask;
     input.config.format = mFormat;
     input.config.offload_info = mOffloadInfoCopy;
-    input.clientInfo.identity = mClientIdentity;
+    input.clientInfo.attributionSource = mClientAttributionSource;
     input.clientInfo.clientTid = -1;
     if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
         // It is currently meaningless to request SCHED_FIFO for a Java thread.  Even if the
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index 389b73f..0564cdf 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -743,6 +743,16 @@
     return statusTFromBinderStatus(mDelegate->setVibratorInfos(vibratorInfos));
 }
 
+status_t AudioFlingerClientAdapter::updateSecondaryOutputs(
+        const TrackSecondaryOutputsMap& trackSecondaryOutputs) {
+    std::vector<media::TrackSecondaryOutputInfo> trackSecondaryOutputInfos =
+            VALUE_OR_RETURN_STATUS(
+                    convertContainer<std::vector<media::TrackSecondaryOutputInfo>>(
+                            trackSecondaryOutputs,
+                            legacy2aidl_TrackSecondaryOutputInfoPair_TrackSecondaryOutputInfo));
+    return statusTFromBinderStatus(mDelegate->updateSecondaryOutputs(trackSecondaryOutputInfos));
+}
+
 
 ////////////////////////////////////////////////////////////////////////////////////////////////////
 // AudioFlingerServerAdapter
@@ -1199,4 +1209,13 @@
     return Status::fromStatusT(mDelegate->setVibratorInfos(vibratorInfos));
 }
 
+Status AudioFlingerServerAdapter::updateSecondaryOutputs(
+        const std::vector<media::TrackSecondaryOutputInfo>& trackSecondaryOutputInfos) {
+    TrackSecondaryOutputsMap trackSecondaryOutputs =
+            VALUE_OR_RETURN_BINDER(convertContainer<TrackSecondaryOutputsMap>(
+                    trackSecondaryOutputInfos,
+                    aidl2legacy_TrackSecondaryOutputInfo_TrackSecondaryOutputInfoPair));
+    return Status::fromStatusT(mDelegate->updateSecondaryOutputs(trackSecondaryOutputs));
+}
+
 } // namespace android
diff --git a/media/libaudioclient/ToneGenerator.cpp b/media/libaudioclient/ToneGenerator.cpp
index 451c4b1..e5e8496 100644
--- a/media/libaudioclient/ToneGenerator.cpp
+++ b/media/libaudioclient/ToneGenerator.cpp
@@ -27,7 +27,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 // Descriptors for all available tones (See ToneGenerator::ToneDescriptor class declaration for details)
 const ToneGenerator::ToneDescriptor ToneGenerator::sToneDescriptors[] = {
@@ -1260,10 +1260,11 @@
 ////////////////////////////////////////////////////////////////////////////////
 bool ToneGenerator::initAudioTrack() {
     // Open audio track in mono, PCM 16bit, default sampling rate.
-    // TODO b/182392769: use identity util
-    Identity identity = Identity();
-    identity.packageName = mOpPackageName;
-    mpAudioTrack = new AudioTrack(identity);
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource = AttributionSourceState();
+    attributionSource.packageName = mOpPackageName;
+    attributionSource.token = sp<BBinder>::make();
+    mpAudioTrack = new AudioTrack(attributionSource);
     ALOGV("AudioTrack(%p) created", mpAudioTrack.get());
 
     audio_attributes_t attr;
@@ -1289,7 +1290,7 @@
             AUDIO_SESSION_ALLOCATE,
             AudioTrack::TRANSFER_CALLBACK,
             nullptr,
-            identity,
+            attributionSource,
             &attr);
     // Set caller name so it can be logged in destructor.
     // MediaMetricsConstants.h: AMEDIAMETRICS_PROP_CALLERNAME_VALUE_TONEGENERATOR
diff --git a/media/libaudioclient/aidl/android/media/AudioClient.aidl b/media/libaudioclient/aidl/android/media/AudioClient.aidl
index aa4d8f5..e98fed3 100644
--- a/media/libaudioclient/aidl/android/media/AudioClient.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioClient.aidl
@@ -16,7 +16,7 @@
 
 package android.media;
 
-import android.media.permission.Identity;
+import android.content.AttributionSourceState;
 
 /**
  * {@hide}
@@ -24,5 +24,5 @@
 parcelable AudioClient {
     /** Interpreted as pid_t. */
     int clientTid;
-    Identity identity;
+    AttributionSourceState attributionSource;
 }
diff --git a/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl b/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
index 5737fcd..2d274f4 100644
--- a/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateEffectRequest.aidl
@@ -19,7 +19,7 @@
 import android.media.AudioDevice;
 import android.media.EffectDescriptor;
 import android.media.IEffectClient;
-import android.media.permission.Identity;
+import android.content.AttributionSourceState;
 
 /**
  * Input arguments of the createEffect() method.
@@ -35,6 +35,6 @@
     /** Interpreted as audio_session_t. */
     int sessionId;
     AudioDevice device;
-    Identity identity;
+    AttributionSourceState attributionSource;
     boolean probe;
 }
diff --git a/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl b/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl
index 5b26d22..7e3c240 100644
--- a/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/CreateRecordRequest.aidl
@@ -19,7 +19,6 @@
 import android.media.AudioAttributesInternal;
 import android.media.AudioClient;
 import android.media.AudioConfigBase;
-import android.media.permission.Identity;
 
 /**
  * CreateRecordRequest contains all input arguments sent by AudioRecord to AudioFlinger
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index abbced5..d2cae6d 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -40,6 +40,7 @@
 import android.media.IAudioTrack;
 import android.media.MicrophoneInfoData;
 import android.media.RenderPosition;
+import android.media.TrackSecondaryOutputInfo;
 import android.media.audio.common.AudioFormat;
 
 /**
@@ -207,4 +208,9 @@
     // Set vibrators' information.
     // The value will be used to initialize HapticGenerator.
     void setVibratorInfos(in AudioVibratorInfo[] vibratorInfos);
+
+    // Update secondary outputs.
+    // This usually happens when there is a dynamic policy registered.
+    void updateSecondaryOutputs(
+            in TrackSecondaryOutputInfo[] trackSecondaryOutputInfos);
 }
diff --git a/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl b/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
index f8924f3..65bcd82 100644
--- a/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
@@ -16,6 +16,8 @@
 
 package android.media;
 
+import android.content.AttributionSourceState;
+
 import android.media.audio.common.AudioFormat;
 
 import android.media.AudioAttributesEx;
@@ -48,7 +50,6 @@
 import android.media.IAudioPolicyServiceClient;
 import android.media.ICaptureStateListener;
 import android.media.Int;
-import android.media.permission.Identity;
 import android.media.SoundTriggerSession;
 
 /**
@@ -81,7 +82,7 @@
 
     GetOutputForAttrResponse getOutputForAttr(in AudioAttributesInternal attr,
                                               int /* audio_session_t */ session,
-                                              in Identity identity,
+                                              in AttributionSourceState attributionSource,
                                               in AudioConfig config,
                                               int /* Bitmask, indexed by AudioOutputFlags */ flags,
                                               int /* audio_port_handle_t */ selectedDeviceId);
@@ -96,7 +97,7 @@
                                             int /* audio_io_handle_t */ input,
                                             int /* audio_unique_id_t */ riid,
                                             int /* audio_session_t */ session,
-                                            in Identity identity,
+                                            in AttributionSourceState attributionSource,
                                             in AudioConfigBase config,
                                             int /* Bitmask, indexed by AudioInputFlags */ flags,
                                             int /* audio_port_handle_t */ selectedDeviceId);
diff --git a/media/libaudioclient/aidl/android/media/TrackSecondaryOutputInfo.aidl b/media/libaudioclient/aidl/android/media/TrackSecondaryOutputInfo.aidl
new file mode 100644
index 0000000..113328e
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/TrackSecondaryOutputInfo.aidl
@@ -0,0 +1,27 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * This is a class that contains port handle for a track and handles for all secondary
+ * outputs of the track.
+ * @hide
+ */
+parcelable TrackSecondaryOutputInfo {
+    int portId; // audio_port_handle_t
+    int[] secondaryOutputIds; // audio_io_handle_t[]
+}
\ No newline at end of file
diff --git a/media/libaudioclient/fuzzer/Android.bp b/media/libaudioclient/fuzzer/Android.bp
index 21e25b9..b290aa8 100644
--- a/media/libaudioclient/fuzzer/Android.bp
+++ b/media/libaudioclient/fuzzer/Android.bp
@@ -65,7 +65,7 @@
         "libutils",
         "libxml2",
         "mediametricsservice-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
     header_libs: [
         "libaudiofoundation_headers",
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index 1b75917..d03c6fa 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -23,7 +23,7 @@
  */
 
 #include <android_audio_policy_configuration_V7_0-enums.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <binder/IServiceManager.h>
 #include <binder/MemoryDealer.h>
 #include <media/AidlConversion.h>
@@ -48,7 +48,7 @@
 using namespace ::android::audio::policy::configuration::V7_0;
 }
 
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 constexpr audio_unique_id_use_t kUniqueIds[] = {
     AUDIO_UNIQUE_ID_USE_UNSPECIFIED, AUDIO_UNIQUE_ID_USE_SESSION, AUDIO_UNIQUE_ID_USE_MODULE,
@@ -225,15 +225,16 @@
     attributes.usage = usage;
     sp<AudioTrack> track = new AudioTrack();
 
-    // TODO b/182392769: use identity util
-    Identity i;
-    i.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
-    i.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
     track->set(AUDIO_STREAM_DEFAULT, sampleRate, format, channelMask, frameCount, flags, nullptr,
                nullptr, notificationFrames, sharedBuffer, false, sessionId,
                ((fast && sharedBuffer == 0) || offload) ? AudioTrack::TRANSFER_CALLBACK
                                                         : AudioTrack::TRANSFER_DEFAULT,
-               offload ? &offloadInfo : nullptr, i, &attributes, false, 1.0f,
+               offload ? &offloadInfo : nullptr, attributionSource, &attributes, false, 1.0f,
                AUDIO_PORT_HANDLE_NONE);
 
     status_t status = track->initCheck();
@@ -308,10 +309,11 @@
 
     attributes.source = inputSource;
 
-    // TODO b/182392769: use identity util
-    Identity i;
-    i.packageName = std::string(mFdp.ConsumeRandomLengthString().c_str());
-    sp<AudioRecord> record = new AudioRecord(i);
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = std::string(mFdp.ConsumeRandomLengthString().c_str());
+    attributionSource.token = sp<BBinder>::make();
+    sp<AudioRecord> record = new AudioRecord(attributionSource);
     record->set(AUDIO_SOURCE_DEFAULT, sampleRate, format, channelMask, frameCount, nullptr, nullptr,
                 notificationFrames, false, sessionId,
                 fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT, flags,
@@ -418,9 +420,9 @@
     request.output = io;
     request.sessionId = sessionId;
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(device));
-    // TODO b/182392769: use identity util
-    request.identity.packageName = opPackageName;
-    request.identity.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(getpid()));
+    // TODO b/182392769: use attribution source util
+    request.attributionSource.packageName = opPackageName;
+    request.attributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(getpid()));
     request.probe = false;
 
     media::CreateEffectResponse response{};
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index 1dd9d60..4ec69c7 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -50,6 +50,7 @@
 #include <android/media/AudioUniqueIdUse.h>
 #include <android/media/EffectDescriptor.h>
 #include <android/media/ExtraAudioDescriptor.h>
+#include <android/media/TrackSecondaryOutputInfo.h>
 
 #include <android/media/SharedFileRegion.h>
 #include <binder/IMemory.h>
@@ -407,6 +408,13 @@
 legacy2aidl_audio_encapsulation_type_t_AudioEncapsulationType(
         const audio_encapsulation_type_t & legacy);
 
+using TrackSecondaryOutputInfoPair = std::pair<audio_port_handle_t, std::vector<audio_io_handle_t>>;
+ConversionResult<TrackSecondaryOutputInfoPair>
+aidl2legacy_TrackSecondaryOutputInfo_TrackSecondaryOutputInfoPair(
+        const media::TrackSecondaryOutputInfo& aidl);
+ConversionResult<media::TrackSecondaryOutputInfo>
+legacy2aidl_TrackSecondaryOutputInfoPair_TrackSecondaryOutputInfo(
+        const TrackSecondaryOutputInfoPair& legacy);
 
 
 }  // namespace android
diff --git a/media/libaudioclient/include/media/AudioClient.h b/media/libaudioclient/include/media/AudioClient.h
index 295fd4f..3be8ce2 100644
--- a/media/libaudioclient/include/media/AudioClient.h
+++ b/media/libaudioclient/include/media/AudioClient.h
@@ -19,7 +19,7 @@
 #define ANDROID_AUDIO_CLIENT_H
 
 #include <sys/types.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 namespace android {
 
@@ -29,7 +29,7 @@
         clientTid(-1) {}
 
     pid_t clientTid;
-    android::media::permission::Identity identity;
+    android::content::AttributionSourceState attributionSource;
 };
 
 }; // namespace android
diff --git a/media/libaudioclient/include/media/AudioCommonTypes.h b/media/libaudioclient/include/media/AudioCommonTypes.h
index 8e446ea..5dfe5fc 100644
--- a/media/libaudioclient/include/media/AudioCommonTypes.h
+++ b/media/libaudioclient/include/media/AudioCommonTypes.h
@@ -29,6 +29,8 @@
 using AttributesVector = std::vector<audio_attributes_t>;
 using StreamTypeVector = std::vector<audio_stream_type_t>;
 
+using TrackSecondaryOutputsMap = std::map<audio_port_handle_t, std::vector<audio_io_handle_t>>;
+
 constexpr bool operator==(const audio_attributes_t &lhs, const audio_attributes_t &rhs)
 {
     return lhs.usage == rhs.usage && lhs.content_type == rhs.content_type &&
diff --git a/media/libaudioclient/include/media/AudioEffect.h b/media/libaudioclient/include/media/AudioEffect.h
index 974ce62..3c19ec1 100644
--- a/media/libaudioclient/include/media/AudioEffect.h
+++ b/media/libaudioclient/include/media/AudioEffect.h
@@ -23,7 +23,7 @@
 #include <media/IAudioFlinger.h>
 #include <media/AudioSystem.h>
 #include <system/audio_effect.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <utils/RefBase.h>
 #include <utils/Errors.h>
@@ -337,9 +337,9 @@
      *
      * Parameters:
      *
-     * client:      Identity for app-op checks
+     * client:      Attribution source for app-op checks
      */
-    explicit AudioEffect(const media::permission::Identity& client);
+    explicit AudioEffect(const android::content::AttributionSourceState& client);
 
     /* Terminates the AudioEffect and unregisters it from AudioFlinger.
      * The effect engine is also destroyed if this AudioEffect was the last controlling
@@ -531,7 +531,7 @@
      static const uint32_t kMaxPreProcessing = 10;
 
 protected:
-     media::permission::Identity mClientIdentity; // Identity used for app op checks.
+     android::content::AttributionSourceState mClientAttributionSource; // source for app op checks.
      bool                    mEnabled = false;   // enable state
      audio_session_t         mSessionId = AUDIO_SESSION_OUTPUT_MIX; // audio session ID
      int32_t                 mPriority = 0;      // priority for effect control
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 9965e25..326919a 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -32,7 +32,7 @@
 #include <utils/threads.h>
 
 #include "android/media/IAudioRecord.h"
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 namespace android {
 
@@ -149,9 +149,9 @@
      *
      * Parameters:
      *
-     * clientIdentity:          The identity of the owner of the record
+     * client:          The attribution source of the owner of the record
      */
-                        AudioRecord(const media::permission::Identity& clientIdentity);
+                        AudioRecord(const android::content::AttributionSourceState& client);
 
     /* Creates an AudioRecord object and registers it with AudioFlinger.
      * Once created, the track needs to be started before it can be used.
@@ -164,7 +164,7 @@
      * format:             Audio format (e.g AUDIO_FORMAT_PCM_16_BIT for signed
      *                     16 bits per sample).
      * channelMask:        Channel mask, such that audio_is_input_channel(channelMask) is true.
-     * client:             The identity of the owner of the record
+     * client:             The attribution source of the owner of the record
      * frameCount:         Minimum size of track PCM buffer in frames. This defines the
      *                     application's contribution to the
      *                     latency of the track.  The actual size selected by the AudioRecord could
@@ -187,7 +187,7 @@
                                     uint32_t sampleRate,
                                     audio_format_t format,
                                     audio_channel_mask_t channelMask,
-                                    const media::permission::Identity& clientIdentity,
+                                    const android::content::AttributionSourceState& client,
                                     size_t frameCount = 0,
                                     callback_t cbf = NULL,
                                     void* user = NULL,
@@ -696,7 +696,7 @@
 
     status_t                mStatus;
 
-    media::permission::Identity mClientIdentity;    // The identity of the owner of this record
+    android::content::AttributionSourceState mClientAttributionSource; // Owner's attribution source
 
     size_t                  mFrameCount;            // corresponds to current IAudioRecord, value is
                                                     // reported back by AudioFlinger to the client
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index 4c99dbd..a9109c8 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -22,7 +22,7 @@
 #include <android/media/AudioVibratorInfo.h>
 #include <android/media/BnAudioFlingerClient.h>
 #include <android/media/BnAudioPolicyServiceClient.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <media/AidlConversionUtil.h>
 #include <media/AudioDeviceTypeAddr.h>
 #include <media/AudioPolicy.h>
@@ -38,6 +38,8 @@
 #include <utils/Mutex.h>
 #include <vector>
 
+using android::content::AttributionSourceState;
+
 namespace android {
 
 struct record_client_info {
@@ -146,6 +148,11 @@
     static void setRecordConfigCallback(record_config_callback);
     static void setRoutingCallback(routing_callback cb);
 
+    // Sets the binder to use for accessing the AudioFlinger service. This enables the system server
+    // to grant specific isolated processes access to the audio system. Currently used only for the
+    // HotwordDetectionService.
+    static void setAudioFlingerBinder(const sp<IBinder>& audioFlinger);
+
     // helper function to obtain AudioFlinger service handle
     static const sp<IAudioFlinger> get_audio_flinger();
 
@@ -264,7 +271,7 @@
                                      audio_io_handle_t *output,
                                      audio_session_t session,
                                      audio_stream_type_t *stream,
-                                     const media::permission::Identity& identity,
+                                     const AttributionSourceState& attributionSource,
                                      const audio_config_t *config,
                                      audio_output_flags_t flags,
                                      audio_port_handle_t *selectedDeviceId,
@@ -280,7 +287,7 @@
                                     audio_io_handle_t *input,
                                     audio_unique_id_t riid,
                                     audio_session_t session,
-                                    const media::permission::Identity& identity,
+                                     const AttributionSourceState& attributionSource,
                                     const audio_config_base_t *config,
                                     audio_input_flags_t flags,
                                     audio_port_handle_t *selectedDeviceId,
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index c293343..f61eef2 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -26,7 +26,7 @@
 #include <media/Modulo.h>
 #include <media/VolumeShaper.h>
 #include <utils/threads.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <string>
 
@@ -36,6 +36,8 @@
 
 namespace android {
 
+using content::AttributionSourceState;
+
 // ----------------------------------------------------------------------------
 
 struct audio_track_cblk_t;
@@ -182,7 +184,7 @@
      */
                         AudioTrack();
 
-                        AudioTrack(const media::permission::Identity& identity);
+                        AudioTrack(const AttributionSourceState& attributionSourceState);
 
     /* Creates an AudioTrack object and registers it with AudioFlinger.
      * Once created, the track needs to be started before it can be used.
@@ -230,7 +232,8 @@
      * transferType:       How data is transferred to AudioTrack.
      * offloadInfo:        If not NULL, provides offload parameters for
      *                     AudioSystem::getOutputForAttr().
-     * identity:           The identity of the app which initiallly requested this AudioTrack.
+     * attributionSource:  The attribution source of the app which initially requested this
+     *                     AudioTrack.
      *                     Includes the UID and PID for power management tracking, or -1 for
      *                     current user/process ID, plus the package name.
      * pAttributes:        If not NULL, supersedes streamType for use case selection.
@@ -259,8 +262,8 @@
                                     audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     const audio_offload_info_t *offloadInfo = NULL,
-                                    const media::permission::Identity& identity =
-                                        media::permission::Identity(),
+                                    const AttributionSourceState& attributionSource =
+                                        AttributionSourceState(),
                                     const audio_attributes_t* pAttributes = NULL,
                                     bool doNotReconnect = false,
                                     float maxRequiredSpeed = 1.0f,
@@ -290,8 +293,8 @@
                                     audio_session_t sessionId   = AUDIO_SESSION_ALLOCATE,
                                     transfer_type transferType = TRANSFER_DEFAULT,
                                     const audio_offload_info_t *offloadInfo = NULL,
-                                    const media::permission::Identity& identity =
-                                        media::permission::Identity(),
+                                    const AttributionSourceState& attributionSource =
+                                        AttributionSourceState(),
                                     const audio_attributes_t* pAttributes = NULL,
                                     bool doNotReconnect = false,
                                     float maxRequiredSpeed = 1.0f);
@@ -338,8 +341,8 @@
                             audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
                             transfer_type transferType = TRANSFER_DEFAULT,
                             const audio_offload_info_t *offloadInfo = NULL,
-                            const media::permission::Identity& identity =
-                                media::permission::Identity(),
+                            const AttributionSourceState& attributionSource =
+                                AttributionSourceState(),
                             const audio_attributes_t* pAttributes = NULL,
                             bool doNotReconnect = false,
                             float maxRequiredSpeed = 1.0f,
@@ -1347,7 +1350,7 @@
 
     sp<DeathNotifier>       mDeathNotifier;
     uint32_t                mSequence;              // incremented for each new IAudioTrack attempt
-    media::permission::Identity mClientIdentity;
+    AttributionSourceState mClientAttributionSource;
 
     wp<AudioSystem::AudioDeviceCallback> mDeviceCallback;
 
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 3a04569..327b37e 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -26,19 +26,21 @@
 #include <binder/IInterface.h>
 #include <media/AidlConversion.h>
 #include <media/AudioClient.h>
+#include <media/AudioCommonTypes.h>
 #include <media/DeviceDescriptorBase.h>
 #include <system/audio.h>
 #include <system/audio_effect.h>
 #include <system/audio_policy.h>
 #include <utils/String8.h>
 #include <media/MicrophoneInfo.h>
+#include <map>
 #include <string>
 #include <vector>
 
 #include <android/media/AudioVibratorInfo.h>
 #include <android/media/BnAudioFlingerService.h>
 #include <android/media/BpAudioFlingerService.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include "android/media/CreateEffectRequest.h"
 #include "android/media/CreateEffectResponse.h"
 #include "android/media/CreateRecordRequest.h"
@@ -55,6 +57,7 @@
 #include "android/media/OpenInputResponse.h"
 #include "android/media/OpenOutputRequest.h"
 #include "android/media/OpenOutputResponse.h"
+#include "android/media/TrackSecondaryOutputInfo.h"
 
 namespace android {
 
@@ -128,7 +131,6 @@
         audio_attributes_t attr;
         audio_config_base_t config;
         AudioClient clientInfo;
-        media::permission::Identity identity;
         audio_unique_id_t riid;
         int32_t maxSharedAudioHistoryMs;
 
@@ -338,6 +340,9 @@
     // The values will be used to initialize HapticGenerator.
     virtual status_t setVibratorInfos(
             const std::vector<media::AudioVibratorInfo>& vibratorInfos) = 0;
+
+    virtual status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs) = 0;
 };
 
 /**
@@ -430,6 +435,8 @@
     status_t getMicrophones(std::vector<media::MicrophoneInfo>* microphones) override;
     status_t setAudioHalPids(const std::vector<pid_t>& pids) override;
     status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos) override;
+    status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs) override;
 
 private:
     const sp<media::IAudioFlingerService> mDelegate;
@@ -513,6 +520,7 @@
             SET_EFFECT_SUSPENDED = media::BnAudioFlingerService::TRANSACTION_setEffectSuspended,
             SET_AUDIO_HAL_PIDS = media::BnAudioFlingerService::TRANSACTION_setAudioHalPids,
             SET_VIBRATOR_INFOS = media::BnAudioFlingerService::TRANSACTION_setVibratorInfos,
+            UPDATE_SECONDARY_OUTPUTS = media::BnAudioFlingerService::TRANSACTION_updateSecondaryOutputs,
         };
 
         /**
@@ -619,6 +627,8 @@
     Status getMicrophones(std::vector<media::MicrophoneInfoData>* _aidl_return) override;
     Status setAudioHalPids(const std::vector<int32_t>& pids) override;
     Status setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos) override;
+    Status updateSecondaryOutputs(
+            const std::vector<media::TrackSecondaryOutputInfo>& trackSecondaryOutputInfos) override;
 
 private:
     const sp<AudioFlingerServerAdapter::Delegate> mDelegate;
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 3c1da4d..def7ca6 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -37,6 +37,7 @@
         "libmediametrics_headers",
     ],
     shared_libs: [
+        "framework-permission-aidl-cpp",
         "libaudioclient",
         "libbinder",
         "libcutils",
@@ -59,6 +60,7 @@
         "libbinder",
         "libcutils",
         "libutils",
+        "framework-permission-aidl-cpp",
     ],
     data: ["record_test_input_*.txt"],
 }
diff --git a/media/libaudioclient/tests/test_create_audiorecord.cpp b/media/libaudioclient/tests/test_create_audiorecord.cpp
index 57676c1..1cbcb71 100644
--- a/media/libaudioclient/tests/test_create_audiorecord.cpp
+++ b/media/libaudioclient/tests/test_create_audiorecord.cpp
@@ -19,7 +19,7 @@
 #include <string.h>
 #include <unistd.h>
 
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <binder/MemoryBase.h>
 #include <binder/MemoryDealer.h>
 #include <binder/MemoryHeapBase.h>
@@ -33,7 +33,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 int testRecord(FILE *inputFile, int outputFileFd)
 {
@@ -41,16 +41,17 @@
     uint32_t testCount = 0;
     Vector<String16> args;
     int ret = 0;
-    // TODO b/182392769: use identity util
-    Identity identity;
-    identity.packageName = PACKAGE_NAME;
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = std::string(PACKAGE_NAME);
+    attributionSource.token = sp<BBinder>::make();
 
     if (inputFile == nullptr) {
         sp<AudioRecord> record = new AudioRecord(AUDIO_SOURCE_DEFAULT,
                                               0 /* sampleRate */,
                                               AUDIO_FORMAT_DEFAULT,
                                               AUDIO_CHANNEL_IN_MONO,
-                                              identity);
+                                              attributionSource);
         if (record == 0 || record->initCheck() != NO_ERROR) {
             write(outputFileFd, "Error creating AudioRecord\n",
                   sizeof("Error creating AudioRecord\n"));
@@ -96,7 +97,7 @@
         memset(&attributes, 0, sizeof(attributes));
         attributes.source = inputSource;
 
-        sp<AudioRecord> record = new AudioRecord(identity);
+        sp<AudioRecord> record = new AudioRecord(attributionSource);
 
         record->set(AUDIO_SOURCE_DEFAULT,
                    sampleRate,
diff --git a/media/libaudioprocessing/AudioMixerBase.cpp b/media/libaudioprocessing/AudioMixerBase.cpp
index a54e22f..f30eb54 100644
--- a/media/libaudioprocessing/AudioMixerBase.cpp
+++ b/media/libaudioprocessing/AudioMixerBase.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "AudioMixer"
 //#define LOG_NDEBUG 0
 
+#include <array>
 #include <sstream>
 #include <string.h>
 
@@ -1295,8 +1296,29 @@
 
 // Needs to derive a compile time constant (constexpr).  Could be targeted to go
 // to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
-#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
-        (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
+
+constexpr int MIXTYPE_MONOVOL(int mixtype, int channels) {
+    if (channels <= FCC_2) {
+        return mixtype;
+    } else if (mixtype == MIXTYPE_MULTI) {
+        return MIXTYPE_MULTI_MONOVOL;
+    } else if (mixtype == MIXTYPE_MULTI_SAVEONLY) {
+        return MIXTYPE_MULTI_SAVEONLY_MONOVOL;
+    } else {
+        return mixtype;
+    }
+}
+
+// Helper to make a functional array from volumeRampMulti.
+template <int MIXTYPE, typename TO, typename TI, typename TV, typename TA, typename TAV,
+          std::size_t ... Is>
+static constexpr auto makeVRMArray(std::index_sequence<Is...>)
+{
+    using F = void(*)(TO*, size_t, const TI*, TA*, TV*, const TV*, TAV*, TAV);
+    return std::array<F, sizeof...(Is)>{
+            { &volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE, Is + 1), Is + 1, TO, TI, TV, TA, TAV> ...}
+        };
+}
 
 /* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
  * TO: int32_t (Q4.27) or float
@@ -1308,40 +1330,26 @@
 static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
         const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
 {
-    switch (channels) {
-    case 1:
-        volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 2:
-        volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 3:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 4:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 5:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 6:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 7:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
-    case 8:
-        volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
-                frameCount, in, aux, vol, volinc, vola, volainc);
-        break;
+    static constexpr auto volumeRampMultiArray =
+            makeVRMArray<MIXTYPE, TO, TI, TV, TA, TAV>(std::make_index_sequence<FCC_LIMIT>());
+    if (channels > 0 && channels <= volumeRampMultiArray.size()) {
+        volumeRampMultiArray[channels - 1](out, frameCount, in, aux, vol, volinc, vola, volainc);
+    } else {
+        ALOGE("%s: invalid channel count:%d", __func__, channels);
     }
 }
 
+// Helper to make a functional array from volumeMulti.
+template <int MIXTYPE, typename TO, typename TI, typename TV, typename TA, typename TAV,
+          std::size_t ... Is>
+static constexpr auto makeVMArray(std::index_sequence<Is...>)
+{
+    using F = void(*)(TO*, size_t, const TI*, TA*, const TV*, TAV);
+    return std::array<F, sizeof...(Is)>{
+            { &volumeMulti<MIXTYPE_MONOVOL(MIXTYPE, Is + 1), Is + 1, TO, TI, TV, TA, TAV> ... }
+        };
+}
+
 /* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
  * TO: int32_t (Q4.27) or float
  * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
@@ -1352,31 +1360,12 @@
 static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
         const TI* in, TA* aux, const TV *vol, TAV vola)
 {
-    switch (channels) {
-    case 1:
-        volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 2:
-        volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 3:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 4:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 5:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 6:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 7:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
-        break;
-    case 8:
-        volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
-        break;
+    static constexpr auto volumeMultiArray =
+            makeVMArray<MIXTYPE, TO, TI, TV, TA, TAV>(std::make_index_sequence<FCC_LIMIT>());
+    if (channels > 0 && channels <= volumeMultiArray.size()) {
+        volumeMultiArray[channels - 1](out, frameCount, in, aux, vol, vola);
+    } else {
+        ALOGE("%s: invalid channel count:%d", __func__, channels);
     }
 }
 
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index 8d374c9..cd47dc6 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_MIXER_OPS_H
 #define ANDROID_AUDIO_MIXER_OPS_H
 
+#include <system/audio.h>
+
 namespace android {
 
 // Hack to make static_assert work in a constexpr
@@ -231,7 +233,7 @@
         typename TO, typename TI, typename TV,
         typename F>
 void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
-    static_assert(NCHAN > 0 && NCHAN <= 8);
+    static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
     static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
             || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
             || MIXTYPE == MIXTYPE_STEREOEXPAND
@@ -291,6 +293,16 @@
     // NCHAN == 8
     proc(*out++, f(inp(), vol[0])); // side left
     proc(*out++, f(inp(), vol[1])); // side right
+    if constexpr (NCHAN > FCC_8) {
+        // Mutes to zero extended surround channels.
+        // 7.1.4 has the correct behavior.
+        // 22.2 has the behavior that FLC and FRC will be mixed instead
+        // of SL and SR and LFE will be center, not left.
+        for (int i = 8; i < NCHAN; ++i) {
+            // TODO: Consider using android::audio_utils::channels::kSideFromChannelIdx
+            proc(*out++, f(inp(), 0.f));
+        }
+    }
 }
 
 /*
diff --git a/media/libaudioprocessing/AudioResampler.cpp b/media/libaudioprocessing/AudioResampler.cpp
index c761b38..51673d7 100644
--- a/media/libaudioprocessing/AudioResampler.cpp
+++ b/media/libaudioprocessing/AudioResampler.cpp
@@ -268,7 +268,7 @@
         mPhaseFraction(0),
         mQuality(quality) {
 
-    const int maxChannels = quality < DYN_LOW_QUALITY ? 2 : 8;
+    const int maxChannels = quality < DYN_LOW_QUALITY ? FCC_2 : FCC_LIMIT;
     if (inChannelCount < 1
             || inChannelCount > maxChannels) {
         LOG_ALWAYS_FATAL("Unsupported sample format %d quality %d channels",
diff --git a/media/libaudioprocessing/AudioResamplerDyn.cpp b/media/libaudioprocessing/AudioResamplerDyn.cpp
index 1aacfd1..2292b19 100644
--- a/media/libaudioprocessing/AudioResamplerDyn.cpp
+++ b/media/libaudioprocessing/AudioResamplerDyn.cpp
@@ -545,64 +545,76 @@
     // Note: A stride of 2 is achieved with non-SIMD processing.
     int stride = ((c.mHalfNumCoefs & 7) == 0) ? 16 : 2;
     LOG_ALWAYS_FATAL_IF(stride < 16, "Resampler stride must be 16 or more");
-    LOG_ALWAYS_FATAL_IF(mChannelCount < 1 || mChannelCount > 8,
-            "Resampler channels(%d) must be between 1 to 8", mChannelCount);
+    LOG_ALWAYS_FATAL_IF(mChannelCount < 1 || mChannelCount > FCC_LIMIT,
+            "Resampler channels(%d) must be between 1 to %d", mChannelCount, FCC_LIMIT);
     // stride 16 (falls back to stride 2 for machines that do not support NEON)
+
+
+// For now use a #define as a compiler generated function table requires renaming.
+#pragma push_macro("AUDIORESAMPLERDYN_CASE")
+#undef AUDIORESAMPLERDYN_CASE
+#define AUDIORESAMPLERDYN_CASE(CHANNEL, LOCKED) \
+    case CHANNEL: if constexpr (CHANNEL <= FCC_LIMIT) {\
+        mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<CHANNEL, LOCKED, 16>; \
+    } break
+
     if (locked) {
         switch (mChannelCount) {
-        case 1:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, true, 16>;
-            break;
-        case 2:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, true, 16>;
-            break;
-        case 3:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, true, 16>;
-            break;
-        case 4:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, true, 16>;
-            break;
-        case 5:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, true, 16>;
-            break;
-        case 6:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, true, 16>;
-            break;
-        case 7:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, true, 16>;
-            break;
-        case 8:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, true, 16>;
-            break;
+        AUDIORESAMPLERDYN_CASE(1, true);
+        AUDIORESAMPLERDYN_CASE(2, true);
+        AUDIORESAMPLERDYN_CASE(3, true);
+        AUDIORESAMPLERDYN_CASE(4, true);
+        AUDIORESAMPLERDYN_CASE(5, true);
+        AUDIORESAMPLERDYN_CASE(6, true);
+        AUDIORESAMPLERDYN_CASE(7, true);
+        AUDIORESAMPLERDYN_CASE(8, true);
+        AUDIORESAMPLERDYN_CASE(9, true);
+        AUDIORESAMPLERDYN_CASE(10, true);
+        AUDIORESAMPLERDYN_CASE(11, true);
+        AUDIORESAMPLERDYN_CASE(12, true);
+        AUDIORESAMPLERDYN_CASE(13, true);
+        AUDIORESAMPLERDYN_CASE(14, true);
+        AUDIORESAMPLERDYN_CASE(15, true);
+        AUDIORESAMPLERDYN_CASE(16, true);
+        AUDIORESAMPLERDYN_CASE(17, true);
+        AUDIORESAMPLERDYN_CASE(18, true);
+        AUDIORESAMPLERDYN_CASE(19, true);
+        AUDIORESAMPLERDYN_CASE(20, true);
+        AUDIORESAMPLERDYN_CASE(21, true);
+        AUDIORESAMPLERDYN_CASE(22, true);
+        AUDIORESAMPLERDYN_CASE(23, true);
+        AUDIORESAMPLERDYN_CASE(24, true);
         }
     } else {
         switch (mChannelCount) {
-        case 1:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<1, false, 16>;
-            break;
-        case 2:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<2, false, 16>;
-            break;
-        case 3:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<3, false, 16>;
-            break;
-        case 4:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<4, false, 16>;
-            break;
-        case 5:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<5, false, 16>;
-            break;
-        case 6:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<6, false, 16>;
-            break;
-        case 7:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<7, false, 16>;
-            break;
-        case 8:
-            mResampleFunc = &AudioResamplerDyn<TC, TI, TO>::resample<8, false, 16>;
-            break;
+        AUDIORESAMPLERDYN_CASE(1, false);
+        AUDIORESAMPLERDYN_CASE(2, false);
+        AUDIORESAMPLERDYN_CASE(3, false);
+        AUDIORESAMPLERDYN_CASE(4, false);
+        AUDIORESAMPLERDYN_CASE(5, false);
+        AUDIORESAMPLERDYN_CASE(6, false);
+        AUDIORESAMPLERDYN_CASE(7, false);
+        AUDIORESAMPLERDYN_CASE(8, false);
+        AUDIORESAMPLERDYN_CASE(9, false);
+        AUDIORESAMPLERDYN_CASE(10, false);
+        AUDIORESAMPLERDYN_CASE(11, false);
+        AUDIORESAMPLERDYN_CASE(12, false);
+        AUDIORESAMPLERDYN_CASE(13, false);
+        AUDIORESAMPLERDYN_CASE(14, false);
+        AUDIORESAMPLERDYN_CASE(15, false);
+        AUDIORESAMPLERDYN_CASE(16, false);
+        AUDIORESAMPLERDYN_CASE(17, false);
+        AUDIORESAMPLERDYN_CASE(18, false);
+        AUDIORESAMPLERDYN_CASE(19, false);
+        AUDIORESAMPLERDYN_CASE(20, false);
+        AUDIORESAMPLERDYN_CASE(21, false);
+        AUDIORESAMPLERDYN_CASE(22, false);
+        AUDIORESAMPLERDYN_CASE(23, false);
+        AUDIORESAMPLERDYN_CASE(24, false);
         }
     }
+#pragma pop_macro("AUDIORESAMPLERDYN_CASE")
+
 #ifdef DEBUG_RESAMPLER
     printf("channels:%d  %s  stride:%d  %s  coef:%d  shift:%d\n",
             mChannelCount, locked ? "locked" : "interpolated",
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
index cf84b83..3419816 100644
--- a/media/libaudioprocessing/include/media/AudioMixerBase.h
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -45,8 +45,7 @@
 {
 public:
     // Do not change these unless underlying code changes.
-    // This mixer has a hard-coded upper limit of 8 channels for output.
-    static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
+    static constexpr uint32_t MAX_NUM_CHANNELS = FCC_LIMIT;
     static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
 
     static const uint16_t UNITY_GAIN_INT = 0x1000;
diff --git a/media/libeffects/visualizer/EffectVisualizer.cpp b/media/libeffects/visualizer/EffectVisualizer.cpp
index f838892..1551e33 100644
--- a/media/libeffects/visualizer/EffectVisualizer.cpp
+++ b/media/libeffects/visualizer/EffectVisualizer.cpp
@@ -157,7 +157,7 @@
     if (pConfig->inputCfg.format != pConfig->outputCfg.format) return -EINVAL;
     const uint32_t channelCount = audio_channel_count_from_out_mask(pConfig->inputCfg.channels);
 #ifdef SUPPORT_MC
-    if (channelCount < 1 || channelCount > FCC_8) return -EINVAL;
+    if (channelCount < 1 || channelCount > FCC_LIMIT) return -EINVAL;
 #else
     if (channelCount != FCC_2) return -EINVAL;
 #endif
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index e471c7b..e98d7d8 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -365,7 +365,7 @@
         "libaudioclient",
         "libmedia_codeclist",
         "libmedia_omx",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     export_shared_lib_headers: [
@@ -374,17 +374,17 @@
         "libandroidicu",
         //"libsonivox",
         "libmedia_omx",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     static_libs: [
         "resourcemanager_aidl_interface-ndk_platform",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     export_static_lib_headers: [
         "resourcemanager_aidl_interface-ndk_platform",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     export_include_dirs: [
diff --git a/media/libmedia/IMediaPlayerService.cpp b/media/libmedia/IMediaPlayerService.cpp
index 0f189ee..07c0ac5 100644
--- a/media/libmedia/IMediaPlayerService.cpp
+++ b/media/libmedia/IMediaPlayerService.cpp
@@ -35,7 +35,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 enum {
     CREATE = IBinder::FIRST_CALL_TRANSACTION,
@@ -65,22 +65,22 @@
 
     virtual sp<IMediaPlayer> create(
             const sp<IMediaPlayerClient>& client, audio_session_t audioSessionId,
-            const Identity& identity) {
+            const AttributionSourceState& attributionSource) {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
         data.writeStrongBinder(IInterface::asBinder(client));
         data.writeInt32(audioSessionId);
-        data.writeParcelable(identity);
+        data.writeParcelable(attributionSource);
 
         remote()->transact(CREATE, data, &reply);
         return interface_cast<IMediaPlayer>(reply.readStrongBinder());
     }
 
-    virtual sp<IMediaRecorder> createMediaRecorder(const Identity& identity)
+    virtual sp<IMediaRecorder> createMediaRecorder(const AttributionSourceState& attributionSource)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IMediaPlayerService::getInterfaceDescriptor());
-        data.writeParcelable(identity);
+        data.writeParcelable(attributionSource);
         remote()->transact(CREATE_MEDIA_RECORDER, data, &reply);
         return interface_cast<IMediaRecorder>(reply.readStrongBinder());
     }
@@ -131,23 +131,23 @@
             sp<IMediaPlayerClient> client =
                 interface_cast<IMediaPlayerClient>(data.readStrongBinder());
             audio_session_t audioSessionId = (audio_session_t) data.readInt32();
-            Identity identity;
-            status_t status = data.readParcelable(&identity);
+            AttributionSourceState attributionSource;
+            status_t status = data.readParcelable(&attributionSource);
             if (status != NO_ERROR) {
                 return status;
             }
-            sp<IMediaPlayer> player = create(client, audioSessionId, identity);
+            sp<IMediaPlayer> player = create(client, audioSessionId, attributionSource);
             reply->writeStrongBinder(IInterface::asBinder(player));
             return NO_ERROR;
         } break;
         case CREATE_MEDIA_RECORDER: {
             CHECK_INTERFACE(IMediaPlayerService, data, reply);
-            Identity identity;
-            status_t status = data.readParcelable(&identity);
+            AttributionSourceState attributionSource;
+            status_t status = data.readParcelable(&attributionSource);
             if (status != NO_ERROR) {
                 return status;
             }
-            sp<IMediaRecorder> recorder = createMediaRecorder(identity);
+            sp<IMediaRecorder> recorder = createMediaRecorder(attributionSource);
             reply->writeStrongBinder(IInterface::asBinder(recorder));
             return NO_ERROR;
         } break;
diff --git a/media/libmedia/include/media/IMediaPlayerService.h b/media/libmedia/include/media/IMediaPlayerService.h
index 243e9c7..6070673 100644
--- a/media/libmedia/include/media/IMediaPlayerService.h
+++ b/media/libmedia/include/media/IMediaPlayerService.h
@@ -27,7 +27,7 @@
 
 #include <media/IMediaPlayerClient.h>
 #include <media/IMediaMetadataRetriever.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <string>
 
@@ -48,12 +48,12 @@
     DECLARE_META_INTERFACE(MediaPlayerService);
 
     virtual sp<IMediaRecorder> createMediaRecorder(
-        const android::media::permission::Identity &identity) = 0;
+        const android::content::AttributionSourceState &attributionSource) = 0;
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever() = 0;
     virtual sp<IMediaPlayer> create(const sp<IMediaPlayerClient>& client,
             audio_session_t audioSessionId = AUDIO_SESSION_ALLOCATE,
-            const android::media::permission::Identity& identity =
-                android::media::permission::Identity()) = 0;
+            const android::content::AttributionSourceState &attributionSource =
+                android::content::AttributionSourceState()) = 0;
     virtual sp<IMediaCodecList> getCodecList() const = 0;
 
     // Connects to a remote display.
diff --git a/media/libmedia/include/media/MediaRecorderBase.h b/media/libmedia/include/media/MediaRecorderBase.h
index b5325ce..2b7818d 100644
--- a/media/libmedia/include/media/MediaRecorderBase.h
+++ b/media/libmedia/include/media/MediaRecorderBase.h
@@ -21,7 +21,7 @@
 #include <media/AudioSystem.h>
 #include <media/MicrophoneInfo.h>
 #include <media/mediarecorder.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <system/audio.h>
 
@@ -34,8 +34,8 @@
 struct PersistentSurface;
 
 struct MediaRecorderBase {
-    explicit MediaRecorderBase(const media::permission::Identity &client)
-        : mClient(client) {}
+    explicit MediaRecorderBase(const android::content::AttributionSourceState &attributionSource)
+        : mAttributionSource(attributionSource) {}
     virtual ~MediaRecorderBase() {}
 
     virtual status_t init() = 0;
@@ -84,7 +84,7 @@
 
 protected:
 
-    media::permission::Identity mClient;
+    android::content::AttributionSourceState mAttributionSource;
 
 private:
     MediaRecorderBase(const MediaRecorderBase &);
diff --git a/media/libmedia/include/media/mediaplayer.h b/media/libmedia/include/media/mediaplayer.h
index fbba398..de4c7db 100644
--- a/media/libmedia/include/media/mediaplayer.h
+++ b/media/libmedia/include/media/mediaplayer.h
@@ -29,7 +29,7 @@
 #include <media/IMediaPlayer.h>
 #include <media/IMediaDeathNotifier.h>
 #include <media/IStreamSource.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <utils/KeyedVector.h>
 #include <utils/String8.h>
@@ -212,8 +212,8 @@
                     public virtual IMediaDeathNotifier
 {
 public:
-    explicit MediaPlayer(const android::media::permission::Identity& mIdentity =
-        android::media::permission::Identity());
+    explicit MediaPlayer(const android::content::AttributionSourceState& mAttributionSource =
+        android::content::AttributionSourceState());
     ~MediaPlayer();
             void            died();
             void            disconnect();
@@ -317,7 +317,7 @@
     float                       mSendLevel;
     struct sockaddr_in          mRetransmitEndpoint;
     bool                        mRetransmitEndpointValid;
-    const android::media::permission::Identity mIdentity;
+    const android::content::AttributionSourceState mAttributionSource;
 };
 
 }; // namespace android
diff --git a/media/libmedia/include/media/mediarecorder.h b/media/libmedia/include/media/mediarecorder.h
index 96a3293..d54ff32 100644
--- a/media/libmedia/include/media/mediarecorder.h
+++ b/media/libmedia/include/media/mediarecorder.h
@@ -25,7 +25,7 @@
 #include <media/IMediaRecorderClient.h>
 #include <media/IMediaDeathNotifier.h>
 #include <media/MicrophoneInfo.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 namespace android {
 
@@ -227,7 +227,7 @@
                       public virtual IMediaDeathNotifier
 {
 public:
-    explicit MediaRecorder(const media::permission::Identity& identity);
+    explicit MediaRecorder(const android::content::AttributionSourceState& attributionSource);
     ~MediaRecorder();
 
     void        died();
diff --git a/media/libmedia/mediaplayer.cpp b/media/libmedia/mediaplayer.cpp
index 7504787..1c9b9e4 100644
--- a/media/libmedia/mediaplayer.cpp
+++ b/media/libmedia/mediaplayer.cpp
@@ -40,9 +40,10 @@
 namespace android {
 
 using media::VolumeShaper;
-using media::permission::Identity;
+using content::AttributionSourceState;
 
-MediaPlayer::MediaPlayer(const Identity& identity) : mIdentity(identity)
+MediaPlayer::MediaPlayer(const AttributionSourceState& attributionSource)
+        : mAttributionSource(attributionSource)
 {
     ALOGV("constructor");
     mListener = NULL;
@@ -153,7 +154,7 @@
     if (url != NULL) {
         const sp<IMediaPlayerService> service(getMediaPlayerService());
         if (service != 0) {
-            sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mIdentity));
+            sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mAttributionSource));
             if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
                 (NO_ERROR != player->setDataSource(httpService, url, headers))) {
                 player.clear();
@@ -170,7 +171,7 @@
     status_t err = UNKNOWN_ERROR;
     const sp<IMediaPlayerService> service(getMediaPlayerService());
     if (service != 0) {
-        sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mIdentity));
+        sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mAttributionSource));
         if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
             (NO_ERROR != player->setDataSource(fd, offset, length))) {
             player.clear();
@@ -186,7 +187,7 @@
     status_t err = UNKNOWN_ERROR;
     const sp<IMediaPlayerService> service(getMediaPlayerService());
     if (service != 0) {
-        sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mIdentity));
+        sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mAttributionSource));
         if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
             (NO_ERROR != player->setDataSource(source))) {
             player.clear();
@@ -202,7 +203,7 @@
     status_t err = UNKNOWN_ERROR;
     const sp<IMediaPlayerService> service(getMediaPlayerService());
     if (service != 0) {
-        sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mIdentity));
+        sp<IMediaPlayer> player(service->create(this, mAudioSessionId, mAttributionSource));
         if ((NO_ERROR != doSetRetransmitEndpoint(player)) ||
             (NO_ERROR != player->setDataSource(rtpParams))) {
             player.clear();
diff --git a/media/libmedia/mediarecorder.cpp b/media/libmedia/mediarecorder.cpp
index da2b190..cf12c36 100644
--- a/media/libmedia/mediarecorder.cpp
+++ b/media/libmedia/mediarecorder.cpp
@@ -33,7 +33,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 status_t MediaRecorder::setCamera(const sp<hardware::ICamera>& camera,
         const sp<ICameraRecordingProxy>& proxy)
@@ -760,13 +760,14 @@
     return INVALID_OPERATION;
 }
 
-MediaRecorder::MediaRecorder(const Identity &identity) : mSurfaceMediaSource(NULL)
+MediaRecorder::MediaRecorder(const AttributionSourceState &attributionSource)
+        : mSurfaceMediaSource(NULL)
 {
     ALOGV("constructor");
 
     const sp<IMediaPlayerService> service(getMediaPlayerService());
     if (service != NULL) {
-        mMediaRecorder = service->createMediaRecorder(identity);
+        mMediaRecorder = service->createMediaRecorder(attributionSource);
     }
     if (mMediaRecorder != NULL) {
         mCurrentState = MEDIA_RECORDER_IDLE;
diff --git a/media/libmediametrics/MediaMetricsItem.cpp b/media/libmediametrics/MediaMetricsItem.cpp
index a8350ea..d597a4d 100644
--- a/media/libmediametrics/MediaMetricsItem.cpp
+++ b/media/libmediametrics/MediaMetricsItem.cpp
@@ -308,6 +308,17 @@
     switch (uid) {
     case AID_RADIO:     // telephony subsystem, RIL
         return false;
+    default:
+        // Some isolated processes can access the audio system; see
+        // AudioSystem::setAudioFlingerBinder (currently only the HotwordDetectionService). Instead
+        // of also allowing access to the MediaMetrics service, it's simpler to just disable it for
+        // now.
+        // TODO(b/190151205): Either allow the HotwordDetectionService to access MediaMetrics or
+        // make this disabling specific to that process.
+        if (uid >= AID_ISOLATED_START && uid <= AID_ISOLATED_END) {
+            return false;
+        }
+        break;
     }
 
     int enabled = property_get_int32(Item::EnabledProperty, -1);
diff --git a/media/libmediaplayerservice/Android.bp b/media/libmediaplayerservice/Android.bp
index 287317d..f55678d 100644
--- a/media/libmediaplayerservice/Android.bp
+++ b/media/libmediaplayerservice/Android.bp
@@ -35,7 +35,7 @@
         "android.hardware.media.c2@1.0",
         "android.hardware.media.omx@1.0",
         "av-types-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
         "libbase",
         "libactivitymanager_aidl",
@@ -76,12 +76,12 @@
         "libstagefright_nuplayer",
         "libstagefright_rtsp",
         "libstagefright_timedtext",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     export_shared_lib_headers: [
         "libmedia",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     include_dirs: [
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index dc4aea5..d278a01 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -95,7 +95,7 @@
 using android::NOT_ENOUGH_DATA;
 using android::Parcel;
 using android::media::VolumeShaper;
-using android::media::permission::Identity;
+using android::content::AttributionSourceState;
 
 // Max number of entries in the filter.
 const int kMaxFilterSize = 64;  // I pulled that out of thin air.
@@ -455,21 +455,22 @@
     ALOGV("MediaPlayerService destroyed");
 }
 
-sp<IMediaRecorder> MediaPlayerService::createMediaRecorder(const Identity& identity)
+sp<IMediaRecorder> MediaPlayerService::createMediaRecorder(
+        const AttributionSourceState& attributionSource)
 {
-    // TODO b/182392769: use identity util
-    Identity verifiedIdentity = identity;
-    verifiedIdentity.uid = VALUE_OR_FATAL(
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState verifiedAttributionSource = attributionSource;
+    verifiedAttributionSource.uid = VALUE_OR_FATAL(
       legacy2aidl_uid_t_int32_t(IPCThreadState::self()->getCallingUid()));
-    verifiedIdentity.pid = VALUE_OR_FATAL(
+    verifiedAttributionSource.pid = VALUE_OR_FATAL(
         legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
     sp<MediaRecorderClient> recorder =
-        new MediaRecorderClient(this, verifiedIdentity);
+        new MediaRecorderClient(this, verifiedAttributionSource);
     wp<MediaRecorderClient> w = recorder;
     Mutex::Autolock lock(mLock);
     mMediaRecorderClients.add(w);
     ALOGV("Create new media recorder client from pid %s",
-        verifiedIdentity.toString().c_str());
+        verifiedAttributionSource.toString().c_str());
     return recorder;
 }
 
@@ -489,21 +490,21 @@
 }
 
 sp<IMediaPlayer> MediaPlayerService::create(const sp<IMediaPlayerClient>& client,
-        audio_session_t audioSessionId, const Identity& identity)
+        audio_session_t audioSessionId, const AttributionSourceState& attributionSource)
 {
     int32_t connId = android_atomic_inc(&mNextConnId);
-    // TODO b/182392769: use identity util
-    Identity verifiedIdentity = identity;
-    verifiedIdentity.pid = VALUE_OR_FATAL(
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState verifiedAttributionSource = attributionSource;
+    verifiedAttributionSource.pid = VALUE_OR_FATAL(
         legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
-    verifiedIdentity.uid = VALUE_OR_FATAL(
+    verifiedAttributionSource.uid = VALUE_OR_FATAL(
         legacy2aidl_uid_t_int32_t(IPCThreadState::self()->getCallingUid()));
 
     sp<Client> c = new Client(
-            this, verifiedIdentity, connId, client, audioSessionId);
+            this, verifiedAttributionSource, connId, client, audioSessionId);
 
     ALOGV("Create new client(%d) from %s, ", connId,
-        verifiedIdentity.toString().c_str());
+        verifiedAttributionSource.toString().c_str());
 
     wp<Client> w = c;
     {
@@ -556,8 +557,8 @@
     char buffer[SIZE];
     String8 result;
     result.append(" Client\n");
-    snprintf(buffer, 255, "  Identity(%s), connId(%d), status(%d), looping(%s)\n",
-        mIdentity.toString().c_str(), mConnId, mStatus, mLoop?"true": "false");
+    snprintf(buffer, 255, "  AttributionSource(%s), connId(%d), status(%d), looping(%s)\n",
+        mAttributionSource.toString().c_str(), mConnId, mStatus, mLoop?"true": "false");
     result.append(buffer);
 
     sp<MediaPlayerBase> p;
@@ -621,7 +622,8 @@
             for (int i = 0, n = mMediaRecorderClients.size(); i < n; ++i) {
                 sp<MediaRecorderClient> c = mMediaRecorderClients[i].promote();
                 if (c != 0) {
-                    snprintf(buffer, 255, " MediaRecorderClient pid(%d)\n", c->mIdentity.pid);
+                    snprintf(buffer, 255, " MediaRecorderClient pid(%d)\n",
+                            c->mAttributionSource.pid);
                     result.append(buffer);
                     write(fd, result.string(), result.size());
                     result = "\n";
@@ -744,10 +746,10 @@
 }
 
 MediaPlayerService::Client::Client(
-        const sp<MediaPlayerService>& service, const Identity& identity,
+        const sp<MediaPlayerService>& service, const AttributionSourceState& attributionSource,
         int32_t connId, const sp<IMediaPlayerClient>& client,
         audio_session_t audioSessionId)
-        : mIdentity(identity)
+        : mAttributionSource(attributionSource)
 {
     ALOGV("Client(%d) constructor", connId);
     mConnId = connId;
@@ -768,7 +770,8 @@
 
 MediaPlayerService::Client::~Client()
 {
-    ALOGV("Client(%d) destructor identity = %s", mConnId, mIdentity.toString().c_str());
+    ALOGV("Client(%d) destructor AttributionSource = %s", mConnId,
+            mAttributionSource.toString().c_str());
     mAudioOutput.clear();
     wp<Client> client(this);
     disconnect();
@@ -781,7 +784,8 @@
 
 void MediaPlayerService::Client::disconnect()
 {
-    ALOGV("disconnect(%d) from identity %s", mConnId, mIdentity.toString().c_str());
+    ALOGV("disconnect(%d) from AttributionSource %s", mConnId,
+            mAttributionSource.toString().c_str());
     // grab local reference and clear main reference to prevent future
     // access to object
     sp<MediaPlayerBase> p;
@@ -822,11 +826,11 @@
     }
     if (p == NULL) {
         p = MediaPlayerFactory::createPlayer(playerType, mListener,
-            VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mIdentity.pid)));
+            VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mAttributionSource.pid)));
     }
 
     if (p != NULL) {
-        p->setUID(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mIdentity.uid)));
+        p->setUID(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mAttributionSource.uid)));
     }
 
     return p;
@@ -934,7 +938,7 @@
     mAudioDeviceUpdatedListener = new AudioDeviceUpdatedNotifier(p);
 
     if (!p->hardwareOutput()) {
-        mAudioOutput = new AudioOutput(mAudioSessionId, mIdentity,
+        mAudioOutput = new AudioOutput(mAudioSessionId, mAttributionSource,
                 mAudioAttributes, mAudioDeviceUpdatedListener);
         static_cast<MediaPlayerInterface*>(p.get())->setAudioSink(mAudioOutput);
     }
@@ -1784,8 +1788,9 @@
 
 #undef LOG_TAG
 #define LOG_TAG "AudioSink"
-MediaPlayerService::AudioOutput::AudioOutput(audio_session_t sessionId, const Identity& identity,
-        const audio_attributes_t* attr, const sp<AudioSystem::AudioDeviceCallback>& deviceCallback)
+MediaPlayerService::AudioOutput::AudioOutput(audio_session_t sessionId,
+        const AttributionSourceState& attributionSource, const audio_attributes_t* attr,
+        const sp<AudioSystem::AudioDeviceCallback>& deviceCallback)
     : mCallback(NULL),
       mCallbackCookie(NULL),
       mCallbackData(NULL),
@@ -1797,7 +1802,7 @@
       mMsecsPerFrame(0),
       mFrameSize(0),
       mSessionId(sessionId),
-      mIdentity(identity),
+      mAttributionSource(attributionSource),
       mSendLevel(0.0),
       mAuxEffectId(0),
       mFlags(AUDIO_OUTPUT_FLAG_NONE),
@@ -2193,7 +2198,7 @@
                     mSessionId,
                     AudioTrack::TRANSFER_CALLBACK,
                     offloadInfo,
-                    mIdentity,
+                    mAttributionSource,
                     mAttributes,
                     doNotReconnect,
                     1.0f,  // default value for maxRequiredSpeed
@@ -2220,7 +2225,7 @@
                     mSessionId,
                     AudioTrack::TRANSFER_DEFAULT,
                     NULL, // offload info
-                    mIdentity,
+                    mAttributionSource,
                     mAttributes,
                     doNotReconnect,
                     targetSpeed,
diff --git a/media/libmediaplayerservice/MediaPlayerService.h b/media/libmediaplayerservice/MediaPlayerService.h
index 35a65d3..98091be 100644
--- a/media/libmediaplayerservice/MediaPlayerService.h
+++ b/media/libmediaplayerservice/MediaPlayerService.h
@@ -33,12 +33,14 @@
 #include <media/MediaPlayerInterface.h>
 #include <media/Metadata.h>
 #include <media/stagefright/foundation/ABase.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <system/audio.h>
 
 namespace android {
 
+using content::AttributionSourceState;
+
 class AudioTrack;
 struct AVSyncSettings;
 class DeathNotifier;
@@ -80,7 +82,7 @@
      public:
                                 AudioOutput(
                                         audio_session_t sessionId,
-                                        const media::permission::Identity& identity,
+                                        const AttributionSourceState& attributionSource,
                                         const audio_attributes_t * attr,
                                         const sp<AudioSystem::AudioDeviceCallback>& deviceCallback);
         virtual                 ~AudioOutput();
@@ -169,7 +171,7 @@
         float                   mMsecsPerFrame;
         size_t                  mFrameSize;
         audio_session_t         mSessionId;
-        media::permission::Identity mIdentity;
+        AttributionSourceState  mAttributionSource;
         float                   mSendLevel;
         int                     mAuxEffectId;
         audio_output_flags_t    mFlags;
@@ -231,13 +233,13 @@
     static  void                instantiate();
 
     // IMediaPlayerService interface
-    virtual sp<IMediaRecorder>  createMediaRecorder(const media::permission::Identity &identity);
+    virtual sp<IMediaRecorder> createMediaRecorder(const AttributionSourceState &attributionSource);
     void    removeMediaRecorderClient(const wp<MediaRecorderClient>& client);
     virtual sp<IMediaMetadataRetriever> createMetadataRetriever();
 
     virtual sp<IMediaPlayer>    create(const sp<IMediaPlayerClient>& client,
                                        audio_session_t audioSessionId,
-                                       const media::permission::Identity& identity);
+                                       const AttributionSourceState& attributionSource);
 
     virtual sp<IMediaCodecList> getCodecList() const;
 
@@ -380,7 +382,7 @@
                 void            notify(int msg, int ext1, int ext2, const Parcel *obj);
 
                 pid_t           pid() const {
-                    return VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mIdentity.pid));
+                    return VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mAttributionSource.pid));
                 }
         virtual status_t        dump(int fd, const Vector<String16>& args);
 
@@ -411,7 +413,7 @@
 
         friend class MediaPlayerService;
                                 Client( const sp<MediaPlayerService>& service,
-                                        const media::permission::Identity& identity,
+                                        const AttributionSourceState& attributionSource,
                                         int32_t connId,
                                         const sp<IMediaPlayerClient>& client,
                                         audio_session_t audioSessionId);
@@ -458,7 +460,7 @@
                     sp<MediaPlayerService>        mService;
                     sp<IMediaPlayerClient>        mClient;
                     sp<AudioOutput>               mAudioOutput;
-                    const media::permission::Identity mIdentity;
+                    const AttributionSourceState  mAttributionSource;
                     status_t                      mStatus;
                     bool                          mLoop;
                     int32_t                       mConnId;
diff --git a/media/libmediaplayerservice/MediaRecorderClient.cpp b/media/libmediaplayerservice/MediaRecorderClient.cpp
index daa923e..a914006 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.cpp
+++ b/media/libmediaplayerservice/MediaRecorderClient.cpp
@@ -126,8 +126,9 @@
     }
 
     if ((as == AUDIO_SOURCE_FM_TUNER
-            && !(captureAudioOutputAllowed(mIdentity) || captureTunerAudioInputAllowed(mIdentity)))
-            || !recordingAllowed(mIdentity, (audio_source_t)as)) {
+            && !(captureAudioOutputAllowed(mAttributionSource)
+                    || captureTunerAudioInputAllowed(mAttributionSource)))
+            || !recordingAllowed(mAttributionSource, (audio_source_t)as)) {
         return PERMISSION_DENIED;
     }
     Mutex::Autolock lock(mLock);
@@ -377,12 +378,12 @@
 }
 
 MediaRecorderClient::MediaRecorderClient(const sp<MediaPlayerService>& service,
-        const Identity& identity)
+        const AttributionSourceState& attributionSource)
 {
     ALOGV("Client constructor");
-    // identity already validated in createMediaRecorder
-    mIdentity = identity;
-    mRecorder = new StagefrightRecorder(identity);
+    // attribution source already validated in createMediaRecorder
+    mAttributionSource = attributionSource;
+    mRecorder = new StagefrightRecorder(attributionSource);
     mMediaPlayerService = service;
 }
 
diff --git a/media/libmediaplayerservice/MediaRecorderClient.h b/media/libmediaplayerservice/MediaRecorderClient.h
index 24c6ee1..dcb9f82 100644
--- a/media/libmediaplayerservice/MediaRecorderClient.h
+++ b/media/libmediaplayerservice/MediaRecorderClient.h
@@ -22,7 +22,7 @@
 
 #include <media/AudioSystem.h>
 #include <media/IMediaRecorder.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <vector>
 
@@ -94,13 +94,13 @@
 
                            MediaRecorderClient(
                                    const sp<MediaPlayerService>& service,
-                                   const media::permission::Identity& identity);
+                                   const content::AttributionSourceState& attributionSource);
     virtual                ~MediaRecorderClient();
 
     std::vector<DeathNotifier> mDeathNotifiers;
     sp<AudioDeviceUpdatedNotifier> mAudioDeviceUpdatedNotifier;
 
-    media::permission::Identity mIdentity;
+    content::AttributionSourceState mAttributionSource;
     mutable Mutex          mLock;
     MediaRecorderBase      *mRecorder;
     sp<MediaPlayerService> mMediaPlayerService;
diff --git a/media/libmediaplayerservice/StagefrightRecorder.cpp b/media/libmediaplayerservice/StagefrightRecorder.cpp
index ce642f3..bffd7b3 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.cpp
+++ b/media/libmediaplayerservice/StagefrightRecorder.cpp
@@ -116,8 +116,8 @@
 }
 
 
-StagefrightRecorder::StagefrightRecorder(const Identity& clientIdentity)
-    : MediaRecorderBase(clientIdentity),
+StagefrightRecorder::StagefrightRecorder(const AttributionSourceState& client)
+    : MediaRecorderBase(client),
       mWriter(NULL),
       mOutputFd(-1),
       mAudioSource((audio_source_t)AUDIO_SOURCE_CNT), // initialize with invalid value
@@ -159,7 +159,7 @@
 
     // we run as part of the media player service; what we really want to
     // know is the app which requested the recording.
-    mMetricsItem->setUid(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClient.uid)));
+    mMetricsItem->setUid(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mAttributionSource.uid)));
 
     mMetricsItem->setCString(kRecorderLogSessionId, mLogSessionId.c_str());
 
@@ -1144,7 +1144,8 @@
 
 status_t StagefrightRecorder::setClientName(const String16& clientName) {
 
-    mClient.packageName = VALUE_OR_RETURN_STATUS(legacy2aidl_String16_string(clientName));
+    mAttributionSource.packageName = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_String16_string(clientName));
 
     return OK;
 }
@@ -1355,7 +1356,7 @@
     sp<AudioSource> audioSource =
         new AudioSource(
                 &attr,
-                mClient,
+                mAttributionSource,
                 sourceSampleRate,
                 mAudioChannels,
                 mSampleRate,
@@ -1880,10 +1881,10 @@
     Size videoSize;
     videoSize.width = mVideoWidth;
     videoSize.height = mVideoHeight;
-    uid_t uid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(mClient.uid));
-    pid_t pid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(mClient.pid));
+    uid_t uid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(mAttributionSource.uid));
+    pid_t pid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(mAttributionSource.pid));
     String16 clientName = VALUE_OR_RETURN_STATUS(
-        aidl2legacy_string_view_String16(mClient.packageName.value_or("")));
+        aidl2legacy_string_view_String16(mAttributionSource.packageName.value_or("")));
     if (mCaptureFpsEnable) {
         if (!(mCaptureFps > 0.)) {
             ALOGE("Invalid mCaptureFps value: %lf", mCaptureFps);
diff --git a/media/libmediaplayerservice/StagefrightRecorder.h b/media/libmediaplayerservice/StagefrightRecorder.h
index 59a080e..d6de47f 100644
--- a/media/libmediaplayerservice/StagefrightRecorder.h
+++ b/media/libmediaplayerservice/StagefrightRecorder.h
@@ -26,12 +26,12 @@
 #include <system/audio.h>
 
 #include <media/hardware/MetadataBufferType.h>
-#include <android/media/permission/Identity.h>
-
-using namespace android::media::permission;
+#include <android/content/AttributionSourceState.h>
 
 namespace android {
 
+using content::AttributionSourceState;
+
 class Camera;
 class ICameraRecordingProxy;
 class CameraSource;
@@ -45,7 +45,7 @@
 struct ALooper;
 
 struct StagefrightRecorder : public MediaRecorderBase {
-    explicit StagefrightRecorder(const Identity& clientIdentity);
+    explicit StagefrightRecorder(const AttributionSourceState& attributionSource);
     virtual ~StagefrightRecorder();
     virtual status_t init();
     virtual status_t setLogSessionId(const String8 &id);
diff --git a/media/libmediaplayerservice/tests/stagefrightRecorder/Android.bp b/media/libmediaplayerservice/tests/stagefrightRecorder/Android.bp
index 5b16911..92236ea 100644
--- a/media/libmediaplayerservice/tests/stagefrightRecorder/Android.bp
+++ b/media/libmediaplayerservice/tests/stagefrightRecorder/Android.bp
@@ -73,7 +73,7 @@
         "libstagefright",
         "libstagefright_foundation",
         "libutils",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
 
diff --git a/media/libmediaplayerservice/tests/stagefrightRecorder/StagefrightRecorderTest.cpp b/media/libmediaplayerservice/tests/stagefrightRecorder/StagefrightRecorderTest.cpp
index 6dea53d..162c187 100644
--- a/media/libmediaplayerservice/tests/stagefrightRecorder/StagefrightRecorderTest.cpp
+++ b/media/libmediaplayerservice/tests/stagefrightRecorder/StagefrightRecorderTest.cpp
@@ -59,10 +59,11 @@
     }
 
     void SetUp() override {
-        // TODO b/182392769: use identity util
-        Identity identity;
-        identity.packageName = std::string(LOG_TAG);
-        mStfRecorder = new StagefrightRecorder(identity);
+        // TODO b/182392769: use attribution source util
+        AttributionSourceState attributionSource;
+        attributionSource.packageName = std::string(LOG_TAG);
+        attributionSource.token = sp<BBinder>::make();
+        mStfRecorder = new StagefrightRecorder(attributionSource);
         ASSERT_NE(mStfRecorder, nullptr) << "Failed to create the instance of recorder";
 
         mOutputAudioFp = fopen(OUTPUT_FILE_NAME_AUDIO, "wb");
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index d6e36b9..a052a70 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -338,7 +338,7 @@
         "android.hardware.cas.native@1.0",
         "android.hardware.drm@1.0",
         "android.hardware.media.omx@1.0",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
 
@@ -352,7 +352,7 @@
         "libogg",
         "libwebm",
         "libstagefright_id3",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libmediandk_format",
         "libmedia_ndkformatpriv",
     ],
@@ -370,7 +370,7 @@
         "libhidlmemory",
         "libmedia",
         "android.hidl.allocator@1.0",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     export_include_dirs: [
diff --git a/media/libstagefright/AudioSource.cpp b/media/libstagefright/AudioSource.cpp
index 89fe56f..b6acdc8 100644
--- a/media/libstagefright/AudioSource.cpp
+++ b/media/libstagefright/AudioSource.cpp
@@ -34,7 +34,7 @@
 
 namespace android {
 
-using android::media::permission::Identity;
+using content::AttributionSourceState;
 
 static void AudioRecordCallbackFunction(int event, void *user, void *info) {
     AudioSource *source = (AudioSource *) user;
@@ -54,13 +54,13 @@
 }
 
 AudioSource::AudioSource(
-    const audio_attributes_t *attr, const Identity& identity,
+    const audio_attributes_t *attr, const AttributionSourceState& attributionSource,
     uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate,
     audio_port_handle_t selectedDeviceId,
     audio_microphone_direction_t selectedMicDirection,
     float selectedMicFieldDimension)
 {
-  set(attr, identity, sampleRate, channelCount, outSampleRate, selectedDeviceId,
+  set(attr, attributionSource, sampleRate, channelCount, outSampleRate, selectedDeviceId,
       selectedMicDirection, selectedMicFieldDimension);
 }
 
@@ -71,17 +71,18 @@
         audio_microphone_direction_t selectedMicDirection,
         float selectedMicFieldDimension)
 {
-  // TODO b/182392769: use identity util
-  Identity identity;
-  identity.packageName = VALUE_OR_FATAL(legacy2aidl_String16_string(opPackageName));
-  identity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
-  identity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
-  set(attr, identity, sampleRate, channelCount, outSampleRate, selectedDeviceId,
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = VALUE_OR_FATAL(legacy2aidl_String16_string(opPackageName));
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(uid));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(pid));
+    attributionSource.token = sp<BBinder>::make();
+    set(attr, attributionSource, sampleRate, channelCount, outSampleRate, selectedDeviceId,
       selectedMicDirection, selectedMicFieldDimension);
 }
 
 void AudioSource::set(
-   const audio_attributes_t *attr, const Identity& identity,
+   const audio_attributes_t *attr, const AttributionSourceState& attributionSource,
         uint32_t sampleRate, uint32_t channelCount, uint32_t outSampleRate,
         audio_port_handle_t selectedDeviceId,
         audio_microphone_direction_t selectedMicDirection,
@@ -126,7 +127,7 @@
     mRecord = new AudioRecord(
         AUDIO_SOURCE_DEFAULT, sampleRate, AUDIO_FORMAT_PCM_16_BIT,
         audio_channel_in_mask_from_count(channelCount),
-        identity,
+        attributionSource,
         (size_t) (bufCount * frameCount),
         AudioRecordCallbackFunction,
         this,
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index 8fa7463..1986272 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -976,6 +976,10 @@
             return "BufferDecoded";
         case TunnelPeekState::kBufferRendered:
             return "BufferRendered";
+        case TunnelPeekState::kDisabledQueued:
+            return "DisabledQueued";
+        case TunnelPeekState::kEnabledQueued:
+            return "EnabledQueued";
         default:
             return default_string;
     }
@@ -986,25 +990,39 @@
     if (!msg->findInt32("tunnel-peek", &tunnelPeek)){
         return;
     }
+
+    TunnelPeekState previousState = mTunnelPeekState;
     if(tunnelPeek == 0){
-        if (mTunnelPeekState == TunnelPeekState::kEnabledNoBuffer) {
-            mTunnelPeekState = TunnelPeekState::kDisabledNoBuffer;
-            ALOGV("TunnelPeekState: %s -> %s",
-                  asString(TunnelPeekState::kEnabledNoBuffer),
-                  asString(TunnelPeekState::kDisabledNoBuffer));
-            return;
+        switch (mTunnelPeekState) {
+            case TunnelPeekState::kEnabledNoBuffer:
+                mTunnelPeekState = TunnelPeekState::kDisabledNoBuffer;
+                break;
+            case TunnelPeekState::kEnabledQueued:
+                mTunnelPeekState = TunnelPeekState::kDisabledQueued;
+                break;
+            default:
+                ALOGV("Ignoring tunnel-peek=%d for %s", tunnelPeek, asString(mTunnelPeekState));
+                return;
         }
     } else {
-        if (mTunnelPeekState == TunnelPeekState::kDisabledNoBuffer) {
-            mTunnelPeekState = TunnelPeekState::kEnabledNoBuffer;
-            ALOGV("TunnelPeekState: %s -> %s",
-                  asString(TunnelPeekState::kDisabledNoBuffer),
-                  asString(TunnelPeekState::kEnabledNoBuffer));
-            return;
+        switch (mTunnelPeekState) {
+            case TunnelPeekState::kDisabledNoBuffer:
+                mTunnelPeekState = TunnelPeekState::kEnabledNoBuffer;
+                break;
+            case TunnelPeekState::kDisabledQueued:
+                mTunnelPeekState = TunnelPeekState::kEnabledQueued;
+                break;
+            case TunnelPeekState::kBufferDecoded:
+                msg->setInt32("android._trigger-tunnel-peek", 1);
+                mTunnelPeekState = TunnelPeekState::kBufferRendered;
+                break;
+            default:
+                ALOGV("Ignoring tunnel-peek=%d for %s", tunnelPeek, asString(mTunnelPeekState));
+                return;
         }
     }
 
-    ALOGV("Ignoring tunnel-peek=%d for %s", tunnelPeek, asString(mTunnelPeekState));
+    ALOGV("TunnelPeekState: %s -> %s", asString(previousState), asString(mTunnelPeekState));
 }
 
 void MediaCodec::updatePlaybackDuration(const sp<AMessage> &msg) {
@@ -3294,25 +3312,32 @@
                     if (mState != STARTED) {
                         break;
                     }
+                    TunnelPeekState previousState = mTunnelPeekState;
                     switch(mTunnelPeekState) {
                         case TunnelPeekState::kDisabledNoBuffer:
+                        case TunnelPeekState::kDisabledQueued:
                             mTunnelPeekState = TunnelPeekState::kBufferDecoded;
+                            ALOGV("First tunnel frame ready");
                             ALOGV("TunnelPeekState: %s -> %s",
-                                  asString(TunnelPeekState::kDisabledNoBuffer),
-                                  asString(TunnelPeekState::kBufferDecoded));
+                                  asString(previousState),
+                                  asString(mTunnelPeekState));
                             break;
                         case TunnelPeekState::kEnabledNoBuffer:
-                            mTunnelPeekState = TunnelPeekState::kBufferDecoded;
-                            ALOGV("TunnelPeekState: %s -> %s",
-                                  asString(TunnelPeekState::kEnabledNoBuffer),
-                                  asString(TunnelPeekState::kBufferDecoded));
+                        case TunnelPeekState::kEnabledQueued:
                             {
                                 sp<AMessage> parameters = new AMessage();
                                 parameters->setInt32("android._trigger-tunnel-peek", 1);
                                 mCodec->signalSetParameters(parameters);
                             }
+                            mTunnelPeekState = TunnelPeekState::kBufferRendered;
+                            ALOGV("First tunnel frame ready");
+                            ALOGV("TunnelPeekState: %s -> %s",
+                                  asString(previousState),
+                                  asString(mTunnelPeekState));
                             break;
                         default:
+                            ALOGV("Ignoring first tunnel frame ready, TunnelPeekState: %s",
+                                  asString(mTunnelPeekState));
                             break;
                     }
 
@@ -4777,6 +4802,28 @@
         buffer->meta()->setInt32("csd", true);
     }
 
+    if (mTunneled) {
+        TunnelPeekState previousState = mTunnelPeekState;
+        switch(mTunnelPeekState){
+            case TunnelPeekState::kEnabledNoBuffer:
+                buffer->meta()->setInt32("tunnel-first-frame", 1);
+                mTunnelPeekState = TunnelPeekState::kEnabledQueued;
+                ALOGV("TunnelPeekState: %s -> %s",
+                        asString(previousState),
+                        asString(mTunnelPeekState));
+                break;
+            case TunnelPeekState::kDisabledNoBuffer:
+                buffer->meta()->setInt32("tunnel-first-frame", 1);
+                mTunnelPeekState = TunnelPeekState::kDisabledQueued;
+                ALOGV("TunnelPeekState: %s -> %s",
+                        asString(previousState),
+                        asString(mTunnelPeekState));
+                break;
+            default:
+                break;
+        }
+    }
+
     status_t err = OK;
     if (hasCryptoOrDescrambler() && !c2Buffer && !memory) {
         AString *errorDetailMsg;
diff --git a/media/libstagefright/include/media/stagefright/AudioSource.h b/media/libstagefright/include/media/stagefright/AudioSource.h
index d1dcdb5..43d50f1 100644
--- a/media/libstagefright/include/media/stagefright/AudioSource.h
+++ b/media/libstagefright/include/media/stagefright/AudioSource.h
@@ -31,6 +31,8 @@
 
 namespace android {
 
+using content::AttributionSourceState;
+
 class AudioRecord;
 
 struct AudioSource : public MediaSource, public MediaBufferObserver {
@@ -38,7 +40,7 @@
     // _not_ a bitmask of audio_channels_t constants.
     AudioSource(
         const audio_attributes_t *attr,
-        const media::permission::Identity& identity,
+        const AttributionSourceState& attributionSource,
         uint32_t sampleRate,
         uint32_t channels,
         uint32_t outSampleRate = 0,
@@ -145,7 +147,7 @@
 
     void set(
         const audio_attributes_t *attr,
-        const media::permission::Identity& identity,
+        const AttributionSourceState& attributionSource,
         uint32_t sampleRate,
         uint32_t channels,
         uint32_t outSampleRate = 0,
diff --git a/media/libstagefright/include/media/stagefright/MediaCodec.h b/media/libstagefright/include/media/stagefright/MediaCodec.h
index 0e6f0b3..d372140 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodec.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodec.h
@@ -377,15 +377,23 @@
     // This type is used to track the tunnel mode video peek state machine:
     //
     // DisabledNoBuffer -> EnabledNoBuffer  when tunnel-peek = true
+    // DisabledQueued   -> EnabledQueued    when tunnel-peek = true
+    // DisabledNoBuffer -> DisabledQueued   when first frame queued
     // EnabledNoBuffer  -> DisabledNoBuffer when tunnel-peek = false
+    // EnabledQueued    -> DisabledQueued   when tunnel-peek = false
+    // EnabledNoBuffer  -> EnabledQueued    when first frame queued
     // DisabledNoBuffer -> BufferDecoded    when kWhatFirstTunnelFrameReady
+    // DisabledQueued   -> BufferDecoded    when kWhatFirstTunnelFrameReady
     // EnabledNoBuffer  -> BufferDecoded    when kWhatFirstTunnelFrameReady
+    // EnabledQueued    -> BufferDecoded    when kWhatFirstTunnelFrameReady
     // BufferDecoded    -> BufferRendered   when kWhatFrameRendered
     // <all states>     -> EnabledNoBuffer  when flush
     // <all states>     -> EnabledNoBuffer  when stop then configure then start
     enum struct TunnelPeekState {
         kDisabledNoBuffer,
         kEnabledNoBuffer,
+        kDisabledQueued,
+        kEnabledQueued,
         kBufferDecoded,
         kBufferRendered,
     };
diff --git a/media/libstagefright/tests/fuzzers/WriterFuzzer.cpp b/media/libstagefright/tests/fuzzers/WriterFuzzer.cpp
index 969c6e1..97d1160 100644
--- a/media/libstagefright/tests/fuzzers/WriterFuzzer.cpp
+++ b/media/libstagefright/tests/fuzzers/WriterFuzzer.cpp
@@ -17,7 +17,7 @@
 //          dylan.katz@leviathansecurity.com
 
 #include <android-base/file.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <ctype.h>
 #include <media/mediarecorder.h>
 #include <media/stagefright/MPEG4Writer.h>
@@ -40,7 +40,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 std::string getFourCC(FuzzedDataProvider *fdp) {
     std::string fourCC = fdp->ConsumeRandomLengthString(4);
@@ -166,11 +166,12 @@
     StandardWriters writerType = dataProvider.ConsumeEnum<StandardWriters>();
     sp<MediaWriter> writer = createWriter(tf.fd, writerType, fileMeta);
 
-    Identity i;
-    i.packageName = dataProvider.ConsumeRandomLengthString(kMaxPackageNameLen);
-    i.uid = dataProvider.ConsumeIntegral<int32_t>();
-    i.pid = dataProvider.ConsumeIntegral<int32_t>();
-    sp<MediaRecorder> mr = new MediaRecorder(i);
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = dataProvider.ConsumeRandomLengthString(kMaxPackageNameLen);
+    attributionSource.uid = dataProvider.ConsumeIntegral<int32_t>();
+    attributionSource.pid = dataProvider.ConsumeIntegral<int32_t>();
+    attributionSource.token = sp<BBinder>::make();
+    sp<MediaRecorder> mr = new MediaRecorder(attributionSource);
     writer->setListener(mr);
 
     uint8_t baseOpLen = operations.size();
diff --git a/media/libstagefright/webm/Android.bp b/media/libstagefright/webm/Android.bp
index 3ceacfe..32a22ba 100644
--- a/media/libstagefright/webm/Android.bp
+++ b/media/libstagefright/webm/Android.bp
@@ -40,7 +40,7 @@
         "libstagefright_foundation",
         "libutils",
         "liblog",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     header_libs: [
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index 9e48c1f..bfe73d5 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -46,12 +46,11 @@
         "libbinder",
         "libcutils",
         "liblog",
-        "libpermission",
         "libutils",
         "libhidlbase",
+        "libpermission",
         "android.hardware.graphics.bufferqueue@1.0",
         "android.hidl.token@1.0-utils",
-        "media_permission-aidl-cpp",
     ],
     export_static_lib_headers: [
         "libbatterystats_aidl",
@@ -71,7 +70,7 @@
     ],
 
     export_shared_lib_headers: [
-        "media_permission-aidl-cpp"
+        "libpermission",
     ],
 
     include_dirs: [
diff --git a/media/utils/ServiceUtilities.cpp b/media/utils/ServiceUtilities.cpp
index 214a174..9c7b863 100644
--- a/media/utils/ServiceUtilities.cpp
+++ b/media/utils/ServiceUtilities.cpp
@@ -25,7 +25,7 @@
 #include <system/audio-hal-enums.h>
 #include <media/AidlConversion.h>
 #include <media/AidlConversionUtil.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <iterator>
 #include <algorithm>
@@ -40,7 +40,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 static const String16 sAndroidPermissionRecordAudio("android.permission.RECORD_AUDIO");
 static const String16 sModifyPhoneState("android.permission.MODIFY_PHONE_STATE");
@@ -81,92 +81,101 @@
   }
 }
 
-static bool checkRecordingInternal(const Identity& identity, const String16& msg,
-        bool start, audio_source_t source) {
+std::optional<AttributionSourceState> resolveAttributionSource(
+        const AttributionSourceState& callerAttributionSource) {
+    AttributionSourceState nextAttributionSource = callerAttributionSource;
+
+    if (!nextAttributionSource.packageName.has_value()) {
+        nextAttributionSource = AttributionSourceState(nextAttributionSource);
+        PermissionController permissionController;
+        const uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(nextAttributionSource.uid));
+        nextAttributionSource.packageName = VALUE_OR_FATAL(legacy2aidl_String16_string(
+                resolveCallingPackage(permissionController, VALUE_OR_FATAL(
+                aidl2legacy_string_view_String16(nextAttributionSource.packageName.value_or(""))),
+                uid)));
+        if (!nextAttributionSource.packageName.has_value()) {
+            return std::nullopt;
+        }
+    }
+
+    AttributionSourceState myAttributionSource;
+    myAttributionSource.uid = VALUE_OR_FATAL(android::legacy2aidl_uid_t_int32_t(getuid()));
+    myAttributionSource.pid = VALUE_OR_FATAL(android::legacy2aidl_pid_t_int32_t(getpid()));
+    myAttributionSource.token = sp<BBinder>::make();
+    myAttributionSource.next.push_back(nextAttributionSource);
+
+    return std::optional<AttributionSourceState>{myAttributionSource};
+}
+
+static bool checkRecordingInternal(const AttributionSourceState& attributionSource,
+        const String16& msg, bool start, audio_source_t source) {
     // Okay to not track in app ops as audio server or media server is us and if
     // device is rooted security model is considered compromised.
     // system_server loses its RECORD_AUDIO permission when a secondary
     // user is active, but it is a core system service so let it through.
     // TODO(b/141210120): UserManager.DISALLOW_RECORD_AUDIO should not affect system user 0
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
     if (isAudioServerOrMediaServerOrSystemServerOrRootUid(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.
+    // may open a record track on behalf of a client. Note that pid may be a tid.
     // IMPORTANT: DON'T USE PermissionCache - RUNTIME PERMISSIONS CHANGE.
-    PermissionController permissionController;
-    const bool ok = permissionController.checkPermission(sAndroidPermissionRecordAudio,
-            identity.pid, identity.uid);
-    if (!ok) {
-        ALOGE("Request requires %s", String8(sAndroidPermissionRecordAudio).c_str());
+    const std::optional<AttributionSourceState> resolvedAttributionSource =
+            resolveAttributionSource(attributionSource);
+    if (!resolvedAttributionSource.has_value()) {
         return false;
     }
 
-    String16 resolvedOpPackageName = resolveCallingPackage(
-            permissionController, VALUE_OR_FATAL(aidl2legacy_string_view_String16(
-                identity.packageName.value_or(""))), uid);
-    if (resolvedOpPackageName.size() == 0) {
-        return false;
-    }
+    const int32_t attributedOpCode = getOpForSource(source);
 
-
-    AppOpsManager appOps;
-    const int32_t op = getOpForSource(source);
+    permission::PermissionChecker permissionChecker;
+    bool permitted = false;
     if (start) {
-        if (int32_t mode = appOps.startOpNoThrow(op, identity.uid,
-            resolvedOpPackageName, /*startIfModeDefault*/ false,
-            VALUE_OR_FATAL(aidl2legacy_optional_string_view_optional_String16(
-                identity.attributionTag)), msg) == AppOpsManager::MODE_ERRORED) {
-            ALOGE("Request start for \"%s\" (uid %d) denied by app op: %d, mode: %d",
-                String8(resolvedOpPackageName).c_str(), identity.uid, op, mode);
-            return false;
-        }
+        permitted = (permissionChecker.checkPermissionForStartDataDeliveryFromDatasource(
+                sAndroidPermissionRecordAudio, resolvedAttributionSource.value(), msg,
+                attributedOpCode) != permission::PermissionChecker::PERMISSION_HARD_DENIED);
     } else {
-        if (int32_t mode = appOps.checkOp(op, uid,
-            resolvedOpPackageName) == AppOpsManager::MODE_ERRORED) {
-            ALOGE("Request check for \"%s\" (uid %d) denied by app op: %d, mode: %d",
-                String8(resolvedOpPackageName).c_str(), identity.uid, op, mode);
-            return false;
-        }
+        permitted = (permissionChecker.checkPermissionForPreflightFromDatasource(
+                sAndroidPermissionRecordAudio, resolvedAttributionSource.value(), msg,
+                attributedOpCode) != permission::PermissionChecker::PERMISSION_HARD_DENIED);
     }
 
-    return true;
+    return permitted;
 }
 
-bool recordingAllowed(const Identity& identity, audio_source_t source) {
-    return checkRecordingInternal(identity, String16(), /*start*/ false, source);
+bool recordingAllowed(const AttributionSourceState& attributionSource, audio_source_t source) {
+    return checkRecordingInternal(attributionSource, String16(), /*start*/ false, source);
 }
 
-bool startRecording(const Identity& identity, const String16& msg, audio_source_t source) {
-     return checkRecordingInternal(identity, msg, /*start*/ true, source);
+bool startRecording(const AttributionSourceState& attributionSource, const String16& msg,
+        audio_source_t source) {
+    return checkRecordingInternal(attributionSource, msg, /*start*/ true, source);
 }
 
-void finishRecording(const Identity& identity, audio_source_t source) {
+void finishRecording(const AttributionSourceState& attributionSource, audio_source_t source) {
     // Okay to not track in app ops as audio server is us and if
     // device is rooted security model is considered compromised.
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    if (isAudioServerOrRootUid(uid)) return;
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    if (isAudioServerOrMediaServerOrSystemServerOrRootUid(uid)) return;
 
-    PermissionController permissionController;
-    String16 resolvedOpPackageName = resolveCallingPackage(
-            permissionController,
-            VALUE_OR_FATAL(aidl2legacy_string_view_String16(identity.packageName.value_or(""))),
-            VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid)));
-    if (resolvedOpPackageName.size() == 0) {
+    // 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.
+    // IMPORTANT: DON'T USE PermissionCache - RUNTIME PERMISSIONS CHANGE.
+    const std::optional<AttributionSourceState> resolvedAttributionSource =
+            resolveAttributionSource(attributionSource);
+    if (!resolvedAttributionSource.has_value()) {
         return;
     }
 
-    AppOpsManager appOps;
-
-    const int32_t op = getOpForSource(source);
-    appOps.finishOp(op, identity.uid, resolvedOpPackageName,
-        VALUE_OR_FATAL(aidl2legacy_optional_string_view_optional_String16(
-            identity.attributionTag)));
+    const int32_t attributedOpCode = getOpForSource(source);
+    permission::PermissionChecker permissionChecker;
+    permissionChecker.finishDataDeliveryFromDatasource(attributedOpCode,
+            resolvedAttributionSource.value());
 }
 
-bool captureAudioOutputAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool captureAudioOutputAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     if (isAudioServerOrRootUid(uid)) return true;
     static const String16 sCaptureAudioOutput("android.permission.CAPTURE_AUDIO_OUTPUT");
     bool ok = PermissionCache::checkPermission(sCaptureAudioOutput, pid, uid);
@@ -174,9 +183,9 @@
     return ok;
 }
 
-bool captureMediaOutputAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool captureMediaOutputAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     if (isAudioServerOrRootUid(uid)) return true;
     static const String16 sCaptureMediaOutput("android.permission.CAPTURE_MEDIA_OUTPUT");
     bool ok = PermissionCache::checkPermission(sCaptureMediaOutput, pid, uid);
@@ -184,9 +193,9 @@
     return ok;
 }
 
-bool captureTunerAudioInputAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool captureTunerAudioInputAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     if (isAudioServerOrRootUid(uid)) return true;
     static const String16 sCaptureTunerAudioInput("android.permission.CAPTURE_TUNER_AUDIO_INPUT");
     bool ok = PermissionCache::checkPermission(sCaptureTunerAudioInput, pid, uid);
@@ -194,9 +203,9 @@
     return ok;
 }
 
-bool captureVoiceCommunicationOutputAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    uid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool captureVoiceCommunicationOutputAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    uid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     if (isAudioServerOrRootUid(uid)) return true;
     static const String16 sCaptureVoiceCommOutput(
         "android.permission.CAPTURE_VOICE_COMMUNICATION_OUTPUT");
@@ -205,15 +214,18 @@
     return ok;
 }
 
-bool captureHotwordAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    uid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool captureHotwordAllowed(const AttributionSourceState& attributionSource) {
     // CAPTURE_AUDIO_HOTWORD permission implies RECORD_AUDIO permission
-    bool ok = recordingAllowed(identity);
+    bool ok = recordingAllowed(attributionSource);
 
     if (ok) {
         static const String16 sCaptureHotwordAllowed("android.permission.CAPTURE_AUDIO_HOTWORD");
-        ok = PermissionCache::checkPermission(sCaptureHotwordAllowed, pid, uid);
+        // Use PermissionChecker, which includes some logic for allowing the isolated
+        // HotwordDetectionService to hold certain permissions.
+        permission::PermissionChecker permissionChecker;
+        ok = (permissionChecker.checkPermissionForPreflight(
+                sCaptureHotwordAllowed, attributionSource, String16(),
+                AppOpsManager::OP_NONE) != permission::PermissionChecker::PERMISSION_HARD_DENIED);
     }
     if (!ok) ALOGV("android.permission.CAPTURE_AUDIO_HOTWORD");
     return ok;
@@ -230,12 +242,12 @@
 }
 
 bool modifyAudioRoutingAllowed() {
-    return modifyAudioRoutingAllowed(getCallingIdentity());
+    return modifyAudioRoutingAllowed(getCallingAttributionSource());
 }
 
-bool modifyAudioRoutingAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool modifyAudioRoutingAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     if (isAudioServerUid(IPCThreadState::self()->getCallingUid())) return true;
     // IMPORTANT: Use PermissionCache - not a runtime permission and may not change.
     bool ok = PermissionCache::checkPermission(sModifyAudioRouting, pid, uid);
@@ -245,12 +257,12 @@
 }
 
 bool modifyDefaultAudioEffectsAllowed() {
-    return modifyDefaultAudioEffectsAllowed(getCallingIdentity());
+    return modifyDefaultAudioEffectsAllowed(getCallingAttributionSource());
 }
 
-bool modifyDefaultAudioEffectsAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool modifyDefaultAudioEffectsAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     if (isAudioServerUid(IPCThreadState::self()->getCallingUid())) return true;
 
     static const String16 sModifyDefaultAudioEffectsAllowed(
@@ -271,18 +283,18 @@
     return ok;
 }
 
-bool modifyPhoneStateAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool modifyPhoneStateAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     bool ok = PermissionCache::checkPermission(sModifyPhoneState, pid, uid);
     ALOGE_IF(!ok, "Request requires %s", String8(sModifyPhoneState).c_str());
     return ok;
 }
 
 // privileged behavior needed by Dialer, Settings, SetupWizard and CellBroadcastReceiver
-bool bypassInterruptionPolicyAllowed(const Identity& identity) {
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(identity.pid));
+bool bypassInterruptionPolicyAllowed(const AttributionSourceState& attributionSource) {
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
+    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(attributionSource.pid));
     static const String16 sWriteSecureSettings("android.permission.WRITE_SECURE_SETTINGS");
     bool ok = PermissionCache::checkPermission(sModifyPhoneState, pid, uid)
         || PermissionCache::checkPermission(sWriteSecureSettings, pid, uid)
@@ -292,11 +304,14 @@
     return ok;
 }
 
-Identity getCallingIdentity() {
-  Identity identity = Identity();
-  identity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
-  identity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(IPCThreadState::self()->getCallingUid()));
-  return identity;
+AttributionSourceState getCallingAttributionSource() {
+    AttributionSourceState attributionSource = AttributionSourceState();
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(
+            IPCThreadState::self()->getCallingPid()));
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
+            IPCThreadState::self()->getCallingUid()));
+    attributionSource.token = sp<BBinder>::make();
+  return attributionSource;
 }
 
 void purgePermissionCache() {
diff --git a/media/utils/fuzzers/Android.bp b/media/utils/fuzzers/Android.bp
index b245834..c1698dc 100644
--- a/media/utils/fuzzers/Android.bp
+++ b/media/utils/fuzzers/Android.bp
@@ -16,7 +16,8 @@
         "liblog",
         "libmediautils",
         "libutils",
-        "media_permission-aidl-cpp",
+        "libbinder",
+        "framework-permission-aidl-cpp",
     ],
 
     cflags: [
diff --git a/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp b/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp
index 2f9e780..6e52512 100644
--- a/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp
+++ b/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp
@@ -17,16 +17,16 @@
 #include <fcntl.h>
 
 #include <functional>
-#include  <type_traits>
+#include <type_traits>
 
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include "fuzzer/FuzzedDataProvider.h"
 #include "mediautils/ServiceUtilities.h"
 
 static constexpr int kMaxOperations = 50;
 static constexpr int kMaxStringLen = 256;
 
-using android::media::permission::Identity;
+using android::content::AttributionSourceState;
 
 const std::vector<std::function<void(FuzzedDataProvider*, android::MediaPackageManager)>>
     operations = {
@@ -54,10 +54,11 @@
     std::string packageNameStr = data_provider.ConsumeRandomLengthString(kMaxStringLen);
     std::string msgStr = data_provider.ConsumeRandomLengthString(kMaxStringLen);
     android::String16 msgStr16(packageNameStr.c_str());
-    Identity identity;
-    identity.packageName = packageNameStr;
-    identity.uid = uid;
-    identity.pid = pid;
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageNameStr;
+    attributionSource.uid = uid;
+    attributionSource.pid = pid;
+    attributionSource.token = android::sp<android::BBinder>::make();
 
     // There is not state here, and order is not significant,
     // so we can simply call all of the target functions
@@ -65,14 +66,14 @@
     android::isAudioServerUid(uid);
     android::isAudioServerOrSystemServerUid(uid);
     android::isAudioServerOrMediaServerUid(uid);
-    android::recordingAllowed(identity);
-    android::startRecording(identity, msgStr16, source);
-    android::finishRecording(identity, source);
-    android::captureAudioOutputAllowed(identity);
-    android::captureMediaOutputAllowed(identity);
-    android::captureHotwordAllowed(identity);
-    android::modifyPhoneStateAllowed(identity);
-    android::bypassInterruptionPolicyAllowed(identity);
+    android::recordingAllowed(attributionSource);
+    android::startRecording(attributionSource, msgStr16, source);
+    android::finishRecording(attributionSource, source);
+    android::captureAudioOutputAllowed(attributionSource);
+    android::captureMediaOutputAllowed(attributionSource);
+    android::captureHotwordAllowed(attributionSource);
+    android::modifyPhoneStateAllowed(attributionSource);
+    android::bypassInterruptionPolicyAllowed(attributionSource);
     android::settingsAllowed();
     android::modifyAudioRoutingAllowed();
     android::modifyDefaultAudioEffectsAllowed();
diff --git a/media/utils/include/mediautils/ServiceUtilities.h b/media/utils/include/mediautils/ServiceUtilities.h
index e7132b8..734313c 100644
--- a/media/utils/include/mediautils/ServiceUtilities.h
+++ b/media/utils/include/mediautils/ServiceUtilities.h
@@ -25,7 +25,9 @@
 #include <cutils/multiuser.h>
 #include <private/android_filesystem_config.h>
 #include <system/audio-hal-enums.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
+#include <binder/PermissionController.h>
+#include <android/permission/PermissionChecker.h>
 
 #include <map>
 #include <optional>
@@ -35,6 +37,8 @@
 
 namespace android {
 
+using content::AttributionSourceState;
+
 // Audio permission utilities
 
 // Used for calls that should originate from system services.
@@ -80,28 +84,30 @@
     }
 }
 
-bool recordingAllowed(const media::permission::Identity& identity,
+bool recordingAllowed(const AttributionSourceState& attributionSource,
         audio_source_t source = AUDIO_SOURCE_DEFAULT);
-bool startRecording(const media::permission::Identity& identity,
+bool startRecording(const AttributionSourceState& attributionSource,
     const String16& msg, audio_source_t source);
-void finishRecording(const media::permission::Identity& identity, audio_source_t source);
-bool captureAudioOutputAllowed(const media::permission::Identity& identity);
-bool captureMediaOutputAllowed(const media::permission::Identity& identity);
-bool captureTunerAudioInputAllowed(const media::permission::Identity& identity);
-bool captureVoiceCommunicationOutputAllowed(const media::permission::Identity& identity);
-bool captureHotwordAllowed(const media::permission::Identity& identity);
+void finishRecording(const AttributionSourceState& attributionSource, audio_source_t source);
+std::optional<AttributionSourceState> resolveAttributionSource(
+    const AttributionSourceState& callerAttributionSource);
+bool captureAudioOutputAllowed(const AttributionSourceState& attributionSource);
+bool captureMediaOutputAllowed(const AttributionSourceState& attributionSource);
+bool captureTunerAudioInputAllowed(const AttributionSourceState& attributionSource);
+bool captureVoiceCommunicationOutputAllowed(const AttributionSourceState& attributionSource);
+bool captureHotwordAllowed(const AttributionSourceState& attributionSource);
 bool settingsAllowed();
 bool modifyAudioRoutingAllowed();
-bool modifyAudioRoutingAllowed(const media::permission::Identity& identity);
+bool modifyAudioRoutingAllowed(const AttributionSourceState& attributionSource);
 bool modifyDefaultAudioEffectsAllowed();
-bool modifyDefaultAudioEffectsAllowed(const media::permission::Identity& identity);
+bool modifyDefaultAudioEffectsAllowed(const AttributionSourceState& attributionSource);
 bool dumpAllowed();
-bool modifyPhoneStateAllowed(const media::permission::Identity& identity);
-bool bypassInterruptionPolicyAllowed(const media::permission::Identity& identity);
+bool modifyPhoneStateAllowed(const AttributionSourceState& attributionSource);
+bool bypassInterruptionPolicyAllowed(const AttributionSourceState& attributionSource);
 void purgePermissionCache();
 int32_t getOpForSource(audio_source_t source);
 
-media::permission::Identity getCallingIdentity();
+AttributionSourceState getCallingAttributionSource();
 
 status_t checkIMemory(const sp<IMemory>& iMemory);
 
diff --git a/services/audioflinger/Android.bp b/services/audioflinger/Android.bp
index a7d47fb..b91f302 100644
--- a/services/audioflinger/Android.bp
+++ b/services/audioflinger/Android.bp
@@ -81,12 +81,12 @@
         "libmedia_helper",
         "libshmemcompat",
         "libvibrator",
-        "media_permission-aidl-cpp",
     ],
 
     static_libs: [
         "libcpustats",
         "libsndfile",
+        "libpermission",
     ],
 
     header_libs: [
@@ -97,7 +97,6 @@
 
     export_shared_lib_headers: [
         "libpermission",
-        "media_permission-aidl-cpp",
     ],
 
     cflags: [
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 3562b00..54a6425 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -103,7 +103,7 @@
 namespace android {
 
 using media::IEffectClient;
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 static const char kDeadlockedString[] = "AudioFlinger may be deadlocked\n";
 static const char kHardwareLockedString[] = "Hardware lock is taken\n";
@@ -163,31 +163,32 @@
     }
 };
 
-// TODO b/182392769: use identity util
+// TODO b/182392769: use attribution source util
 /* static */
-media::permission::Identity AudioFlinger::checkIdentityPackage(
-        const media::permission::Identity& identity) {
+AttributionSourceState AudioFlinger::checkAttributionSourcePackage(
+        const AttributionSourceState& attributionSource) {
     Vector<String16> packages;
-    PermissionController{}.getPackagesForUid(identity.uid, packages);
+    PermissionController{}.getPackagesForUid(attributionSource.uid, packages);
 
-    Identity checkedIdentity = identity;
-    if (!identity.packageName.has_value() || identity.packageName.value().size() == 0) {
+    AttributionSourceState checkedAttributionSource = attributionSource;
+    if (!attributionSource.packageName.has_value()
+            || attributionSource.packageName.value().size() == 0) {
         if (!packages.isEmpty()) {
-            checkedIdentity.packageName =
+            checkedAttributionSource.packageName =
                 std::move(legacy2aidl_String16_string(packages[0]).value());
         }
     } else {
         String16 opPackageLegacy = VALUE_OR_FATAL(
-            aidl2legacy_string_view_String16(identity.packageName.value_or("")));
+            aidl2legacy_string_view_String16(attributionSource.packageName.value_or("")));
         if (std::find_if(packages.begin(), packages.end(),
                 [&opPackageLegacy](const auto& package) {
                 return opPackageLegacy == package; }) == packages.end()) {
             ALOGW("The package name(%s) provided does not correspond to the uid %d",
-                    identity.packageName.value_or("").c_str(), identity.uid);
-            checkedIdentity.packageName = std::optional<std::string>();
+                    attributionSource.packageName.value_or("").c_str(), attributionSource.uid);
+            checkedAttributionSource.packageName = std::optional<std::string>();
         }
     }
-    return checkedIdentity;
+    return checkedAttributionSource;
 }
 
 // ----------------------------------------------------------------------------
@@ -236,11 +237,12 @@
     timespec ts{};
     clock_gettime(CLOCK_MONOTONIC, &ts);
     // zero ID has a special meaning, so start allocation at least at AUDIO_UNIQUE_ID_USE_MAX
-    uint32_t sessionBase = (uint32_t)std::max((long)1, ts.tv_sec);
+    uint32_t movingBase = (uint32_t)std::max((long)1, ts.tv_sec);
     // unsigned instead of audio_unique_id_use_t, because ++ operator is unavailable for enum
     for (unsigned use = AUDIO_UNIQUE_ID_USE_UNSPECIFIED; use < AUDIO_UNIQUE_ID_USE_MAX; use++) {
         mNextUniqueIds[use] =
-                ((use == AUDIO_UNIQUE_ID_USE_SESSION) ? sessionBase : 1) * AUDIO_UNIQUE_ID_USE_MAX;
+                ((use == AUDIO_UNIQUE_ID_USE_SESSION || use == AUDIO_UNIQUE_ID_USE_CLIENT) ?
+                        movingBase : 1) * AUDIO_UNIQUE_ID_USE_MAX;
     }
 
 #if 1
@@ -312,6 +314,27 @@
     return NO_ERROR;
 }
 
+status_t AudioFlinger::updateSecondaryOutputs(
+        const TrackSecondaryOutputsMap& trackSecondaryOutputs) {
+    Mutex::Autolock _l(mLock);
+    for (const auto& [trackId, secondaryOutputs] : trackSecondaryOutputs) {
+        size_t i = 0;
+        for (; i < mPlaybackThreads.size(); ++i) {
+            PlaybackThread *thread = mPlaybackThreads.valueAt(i).get();
+            Mutex::Autolock _tl(thread->mLock);
+            sp<PlaybackThread::Track> track = thread->getTrackById_l(trackId);
+            if (track != nullptr) {
+                ALOGD("%s trackId: %u", __func__, trackId);
+                updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
+                break;
+            }
+        }
+        ALOGW_IF(i >= mPlaybackThreads.size(),
+                 "%s cannot find track with id %u", __func__, trackId);
+    }
+    return NO_ERROR;
+}
+
 // getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
 const media::AudioVibratorInfo* AudioFlinger::getDefaultVibratorInfo_l() {
     if (mAudioVibratorInfos.empty()) {
@@ -409,7 +432,7 @@
 
         ret = AudioSystem::getOutputForAttr(&localAttr, &io,
                                             actualSessionId,
-                                            &streamType, client.identity,
+                                            &streamType, client.attributionSource,
                                             &fullConfig,
                                             (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_MMAP_NOIRQ |
                                                     AUDIO_OUTPUT_FLAG_DIRECT),
@@ -420,7 +443,7 @@
         ret = AudioSystem::getInputForAttr(&localAttr, &io,
                                               RECORD_RIID_INVALID,
                                               actualSessionId,
-                                              client.identity,
+                                              client.attributionSource,
                                               config,
                                               AUDIO_INPUT_FLAG_MMAP_NOIRQ, deviceId, &portId);
     }
@@ -826,21 +849,21 @@
 
     // TODO b/182392553: refactor or make clearer
     pid_t clientPid =
-        VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(input.clientInfo.identity.pid));
+        VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(input.clientInfo.attributionSource.pid));
     bool updatePid = (clientPid == (pid_t)-1);
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
     uid_t clientUid =
-        VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(input.clientInfo.identity.uid));
+        VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(input.clientInfo.attributionSource.uid));
     audio_io_handle_t effectThreadId = AUDIO_IO_HANDLE_NONE;
     std::vector<int> effectIds;
     audio_attributes_t localAttr = input.attr;
 
-    Identity adjIdentity = input.clientInfo.identity;
+    AttributionSourceState adjAttributionSource = input.clientInfo.attributionSource;
     if (!isAudioServerOrMediaServerUid(callingUid)) {
         ALOGW_IF(clientUid != callingUid,
                 "%s uid %d tried to pass itself off as %d",
                 __FUNCTION__, callingUid, clientUid);
-        adjIdentity.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
+        adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
         clientUid = callingUid;
         updatePid = true;
     }
@@ -850,7 +873,7 @@
                  "%s uid %d pid %d tried to pass itself off as pid %d",
                  __func__, callingUid, callingPid, clientPid);
         clientPid = callingPid;
-        adjIdentity.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
+        adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
     }
 
     audio_session_t sessionId = input.sessionId;
@@ -865,7 +888,7 @@
     output.outputId = AUDIO_IO_HANDLE_NONE;
     output.selectedDeviceId = input.selectedDeviceId;
     lStatus = AudioSystem::getOutputForAttr(&localAttr, &output.outputId, sessionId, &streamType,
-                                            adjIdentity, &input.config, input.flags,
+                                            adjAttributionSource, &input.config, input.flags,
                                             &output.selectedDeviceId, &portId, &secondaryOutputs);
 
     if (lStatus != NO_ERROR || output.outputId == AUDIO_IO_HANDLE_NONE) {
@@ -930,7 +953,7 @@
                                       &output.frameCount, &output.notificationFrameCount,
                                       input.notificationsPerBuffer, input.speed,
                                       input.sharedBuffer, sessionId, &output.flags,
-                                      callingPid, adjIdentity, input.clientInfo.clientTid,
+                                      callingPid, adjAttributionSource, input.clientInfo.clientTid,
                                       &lStatus, portId, input.audioTrackCallback);
         LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (track == 0));
         // we don't abort yet if lStatus != NO_ERROR; there is still work to be done regardless
@@ -944,88 +967,7 @@
             // Connect secondary outputs. Failure on a secondary output must not imped the primary
             // Any secondary output setup failure will lead to a desync between the AP and AF until
             // the track is destroyed.
-            TeePatches teePatches;
-            for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
-                PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
-                if (secondaryThread == NULL) {
-                    ALOGE("no playback thread found for secondary output %d", output.outputId);
-                    continue;
-                }
-
-                size_t sourceFrameCount = thread->frameCount() * output.sampleRate
-                                          / thread->sampleRate();
-                size_t sinkFrameCount = secondaryThread->frameCount() * output.sampleRate
-                                          / secondaryThread->sampleRate();
-                // If the secondary output has just been opened, the first secondaryThread write
-                // will not block as it will fill the empty startup buffer of the HAL,
-                // so a second sink buffer needs to be ready for the immediate next blocking write.
-                // Additionally, have a margin of one main thread buffer as the scheduling jitter
-                // can reorder the writes (eg if thread A&B have the same write intervale,
-                // the scheduler could schedule AB...BA)
-                size_t frameCountToBeReady = 2 * sinkFrameCount + sourceFrameCount;
-                // Total secondary output buffer must be at least as the read frames plus
-                // the margin of a few buffers on both sides in case the
-                // threads scheduling has some jitter.
-                // That value should not impact latency as the secondary track is started before
-                // its buffer is full, see frameCountToBeReady.
-                size_t frameCount = frameCountToBeReady + 2 * (sourceFrameCount + sinkFrameCount);
-                // The frameCount should also not be smaller than the secondary thread min frame
-                // count
-                size_t minFrameCount = AudioSystem::calculateMinFrameCount(
-                            [&] { Mutex::Autolock _l(secondaryThread->mLock);
-                                  return secondaryThread->latency_l(); }(),
-                            secondaryThread->mNormalFrameCount,
-                            secondaryThread->mSampleRate,
-                            output.sampleRate,
-                            input.speed);
-                frameCount = std::max(frameCount, minFrameCount);
-
-                using namespace std::chrono_literals;
-                auto inChannelMask = audio_channel_mask_out_to_in(input.config.channel_mask);
-                sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
-                                                               output.sampleRate,
-                                                               inChannelMask,
-                                                               input.config.format,
-                                                               frameCount,
-                                                               NULL /* buffer */,
-                                                               (size_t)0 /* bufferSize */,
-                                                               AUDIO_INPUT_FLAG_DIRECT,
-                                                               0ns /* timeout */);
-                status_t status = patchRecord->initCheck();
-                if (status != NO_ERROR) {
-                    ALOGE("Secondary output patchRecord init failed: %d", status);
-                    continue;
-                }
-
-                // TODO: We could check compatibility of the secondaryThread with the PatchTrack
-                // for fast usage: thread has fast mixer, sample rate matches, etc.;
-                // for now, we exclude fast tracks by removing the Fast flag.
-                const audio_output_flags_t outputFlags =
-                        (audio_output_flags_t)(output.flags & ~AUDIO_OUTPUT_FLAG_FAST);
-                sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
-                                                               streamType,
-                                                               output.sampleRate,
-                                                               input.config.channel_mask,
-                                                               input.config.format,
-                                                               frameCount,
-                                                               patchRecord->buffer(),
-                                                               patchRecord->bufferSize(),
-                                                               outputFlags,
-                                                               0ns /* timeout */,
-                                                               frameCountToBeReady);
-                status = patchTrack->initCheck();
-                if (status != NO_ERROR) {
-                    ALOGE("Secondary output patchTrack init failed: %d", status);
-                    continue;
-                }
-                teePatches.push_back({patchRecord, patchTrack});
-                secondaryThread->addPatchTrack(patchTrack);
-                // In case the downstream patchTrack on the secondaryThread temporarily outlives
-                // our created track, ensure the corresponding patchRecord is still alive.
-                patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
-                patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
-            }
-            track->setTeePatches(std::move(teePatches));
+            updateSecondaryOutputsForTrack_l(track.get(), thread, secondaryOutputs);
         }
 
         // move effect chain to this output thread if an effect on same session was waiting
@@ -2090,24 +2032,26 @@
     output.inputId = AUDIO_IO_HANDLE_NONE;
 
     // TODO b/182392553: refactor or clean up
-    Identity adjIdentity = input.clientInfo.identity;
-    bool updatePid = (adjIdentity.pid == -1);
+    AttributionSourceState adjAttributionSource = input.clientInfo.attributionSource;
+    bool updatePid = (adjAttributionSource.pid == -1);
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    const uid_t currentUid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(adjIdentity.uid));
+    const uid_t currentUid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(
+           adjAttributionSource.uid));
     if (!isAudioServerOrMediaServerUid(callingUid)) {
         ALOGW_IF(currentUid != callingUid,
                 "%s uid %d tried to pass itself off as %d",
                 __FUNCTION__, callingUid, currentUid);
-        adjIdentity.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
+        adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
         updatePid = true;
     }
     const pid_t callingPid = IPCThreadState::self()->getCallingPid();
-    const pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(adjIdentity.pid));
+    const pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(
+            adjAttributionSource.pid));
     if (updatePid) {
         ALOGW_IF(currentPid != (pid_t)-1 && currentPid != callingPid,
                  "%s uid %d pid %d tried to pass itself off as pid %d",
                  __func__, callingUid, callingPid, currentPid);
-        adjIdentity.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
+        adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
     }
 
     // we don't yet support anything other than linear PCM
@@ -2135,7 +2079,7 @@
     output.selectedDeviceId = input.selectedDeviceId;
     output.flags = input.flags;
 
-    client = registerPid(VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(adjIdentity.pid)));
+    client = registerPid(VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(adjAttributionSource.pid)));
 
     // Not a conventional loop, but a retry loop for at most two iterations total.
     // Try first maybe with FAST flag then try again without FAST flag if that fails.
@@ -2155,7 +2099,7 @@
                                       input.riid,
                                       sessionId,
                                     // FIXME compare to AudioTrack
-                                      adjIdentity,
+                                      adjAttributionSource,
                                       &input.config,
                                       output.flags, &output.selectedDeviceId, &portId);
     if (lStatus != NO_ERROR) {
@@ -2182,7 +2126,7 @@
                                                   input.config.format, input.config.channel_mask,
                                                   &output.frameCount, sessionId,
                                                   &output.notificationFrameCount,
-                                                  callingPid, adjIdentity, &output.flags,
+                                                  callingPid, adjAttributionSource, &output.flags,
                                                   input.clientInfo.clientTid,
                                                   &lStatus, portId, input.maxSharedAudioHistoryMs);
         LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
@@ -2952,8 +2896,8 @@
         audio_is_linear_pcm(config->format) &&
         audio_is_linear_pcm(halconfig.format) &&
         (halconfig.sample_rate <= AUDIO_RESAMPLER_DOWN_RATIO_MAX * config->sample_rate) &&
-        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_8) &&
-        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_8)) {
+        (audio_channel_count_from_in_mask(halconfig.channel_mask) <= FCC_LIMIT) &&
+        (audio_channel_count_from_in_mask(config->channel_mask) <= FCC_LIMIT)) {
         // FIXME describe the change proposed by HAL (save old values so we can log them here)
         ALOGV("openInput_l() reopening with proposed sampling rate and channel mask");
         inStream.clear();
@@ -3441,6 +3385,94 @@
     return nullptr;
 }
 
+void AudioFlinger::updateSecondaryOutputsForTrack_l(
+        PlaybackThread::Track* track,
+        PlaybackThread* thread,
+        const std::vector<audio_io_handle_t> &secondaryOutputs) const {
+    TeePatches teePatches;
+    for (audio_io_handle_t secondaryOutput : secondaryOutputs) {
+        PlaybackThread *secondaryThread = checkPlaybackThread_l(secondaryOutput);
+        if (secondaryThread == nullptr) {
+            ALOGE("no playback thread found for secondary output %d", thread->id());
+            continue;
+        }
+
+        size_t sourceFrameCount = thread->frameCount() * track->sampleRate()
+                                  / thread->sampleRate();
+        size_t sinkFrameCount = secondaryThread->frameCount() * track->sampleRate()
+                                  / secondaryThread->sampleRate();
+        // If the secondary output has just been opened, the first secondaryThread write
+        // will not block as it will fill the empty startup buffer of the HAL,
+        // so a second sink buffer needs to be ready for the immediate next blocking write.
+        // Additionally, have a margin of one main thread buffer as the scheduling jitter
+        // can reorder the writes (eg if thread A&B have the same write intervale,
+        // the scheduler could schedule AB...BA)
+        size_t frameCountToBeReady = 2 * sinkFrameCount + sourceFrameCount;
+        // Total secondary output buffer must be at least as the read frames plus
+        // the margin of a few buffers on both sides in case the
+        // threads scheduling has some jitter.
+        // That value should not impact latency as the secondary track is started before
+        // its buffer is full, see frameCountToBeReady.
+        size_t frameCount = frameCountToBeReady + 2 * (sourceFrameCount + sinkFrameCount);
+        // The frameCount should also not be smaller than the secondary thread min frame
+        // count
+        size_t minFrameCount = AudioSystem::calculateMinFrameCount(
+                    [&] { Mutex::Autolock _l(secondaryThread->mLock);
+                          return secondaryThread->latency_l(); }(),
+                    secondaryThread->mNormalFrameCount,
+                    secondaryThread->mSampleRate,
+                    track->sampleRate(),
+                    track->getSpeed());
+        frameCount = std::max(frameCount, minFrameCount);
+
+        using namespace std::chrono_literals;
+        auto inChannelMask = audio_channel_mask_out_to_in(track->channelMask());
+        sp patchRecord = new RecordThread::PatchRecord(nullptr /* thread */,
+                                                       track->sampleRate(),
+                                                       inChannelMask,
+                                                       track->format(),
+                                                       frameCount,
+                                                       nullptr /* buffer */,
+                                                       (size_t)0 /* bufferSize */,
+                                                       AUDIO_INPUT_FLAG_DIRECT,
+                                                       0ns /* timeout */);
+        status_t status = patchRecord->initCheck();
+        if (status != NO_ERROR) {
+            ALOGE("Secondary output patchRecord init failed: %d", status);
+            continue;
+        }
+
+        // TODO: We could check compatibility of the secondaryThread with the PatchTrack
+        // for fast usage: thread has fast mixer, sample rate matches, etc.;
+        // for now, we exclude fast tracks by removing the Fast flag.
+        const audio_output_flags_t outputFlags =
+                (audio_output_flags_t)(track->getOutputFlags() & ~AUDIO_OUTPUT_FLAG_FAST);
+        sp patchTrack = new PlaybackThread::PatchTrack(secondaryThread,
+                                                       track->streamType(),
+                                                       track->sampleRate(),
+                                                       track->channelMask(),
+                                                       track->format(),
+                                                       frameCount,
+                                                       patchRecord->buffer(),
+                                                       patchRecord->bufferSize(),
+                                                       outputFlags,
+                                                       0ns /* timeout */,
+                                                       frameCountToBeReady);
+        status = patchTrack->initCheck();
+        if (status != NO_ERROR) {
+            ALOGE("Secondary output patchTrack init failed: %d", status);
+            continue;
+        }
+        teePatches.push_back({patchRecord, patchTrack});
+        secondaryThread->addPatchTrack(patchTrack);
+        // In case the downstream patchTrack on the secondaryThread temporarily outlives
+        // our created track, ensure the corresponding patchRecord is still alive.
+        patchTrack->setPeerProxy(patchRecord, true /* holdReference */);
+        patchRecord->setPeerProxy(patchTrack, false /* holdReference */);
+    }
+    track->setTeePatches(std::move(teePatches));
+}
+
 sp<AudioFlinger::SyncEvent> AudioFlinger::createSyncEvent(AudioSystem::sync_event_t type,
                                     audio_session_t triggerSession,
                                     audio_session_t listenerSession,
@@ -3577,7 +3609,7 @@
     const int32_t priority = request.priority;
     const AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
             aidl2legacy_AudioDeviceTypeAddress(request.device));
-    Identity adjIdentity = request.identity;
+    AttributionSourceState adjAttributionSource = request.attributionSource;
     const audio_session_t sessionId = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_session_t(request.sessionId));
     audio_io_handle_t io = VALUE_OR_RETURN_STATUS(
@@ -3595,19 +3627,20 @@
 
     // TODO b/182392553: refactor or make clearer
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    adjIdentity.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
-    pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(adjIdentity.pid));
+    adjAttributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
+    pid_t currentPid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_pid_t(adjAttributionSource.pid));
     if (currentPid == -1 || !isAudioServerOrMediaServerUid(callingUid)) {
         const pid_t callingPid = IPCThreadState::self()->getCallingPid();
         ALOGW_IF(currentPid != -1 && currentPid != callingPid,
                  "%s uid %d pid %d tried to pass itself off as pid %d",
                  __func__, callingUid, callingPid, currentPid);
-        adjIdentity.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
+        adjAttributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_pid_t_int32_t(callingPid));
         currentPid = callingPid;
     }
 
     ALOGV("createEffect pid %d, effectClient %p, priority %d, sessionId %d, io %d, factory %p",
-          adjIdentity.pid, effectClient.get(), priority, sessionId, io, mEffectsFactoryHal.get());
+          adjAttributionSource.pid, effectClient.get(), priority, sessionId, io,
+          mEffectsFactoryHal.get());
 
     if (mEffectsFactoryHal == 0) {
         ALOGE("%s: no effects factory hal", __func__);
@@ -3635,7 +3668,7 @@
             goto Exit;
         }
     } else if (sessionId == AUDIO_SESSION_DEVICE) {
-        if (!modifyDefaultAudioEffectsAllowed(adjIdentity)) {
+        if (!modifyDefaultAudioEffectsAllowed(adjAttributionSource)) {
             ALOGE("%s: device effect permission denied for uid %d", __func__, callingUid);
             lStatus = PERMISSION_DENIED;
             goto Exit;
@@ -3680,7 +3713,7 @@
         // check recording permission for visualizer
         if ((memcmp(&descOut.type, SL_IID_VISUALIZATION, sizeof(effect_uuid_t)) == 0) &&
             // TODO: Do we need to start/stop op - i.e. is there recording being performed?
-            !recordingAllowed(adjIdentity)) {
+            !recordingAllowed(adjAttributionSource)) {
             lStatus = PERMISSION_DENIED;
             goto Exit;
         }
@@ -3963,7 +3996,7 @@
         // if the move request is not received from audio policy manager, the effect must be
         // re-registered with the new strategy and output
         if (dstChain == 0) {
-            dstChain = effect->callback()->chain().promote();
+            dstChain = effect->getCallback()->chain().promote();
             if (dstChain == 0) {
                 ALOGW("moveEffectChain_l() cannot get chain from effect %p", effect.get());
                 status = NO_INIT;
@@ -4013,7 +4046,7 @@
             goto Exit;
         }
 
-        dstChain = effect->callback()->chain().promote();
+        dstChain = effect->getCallback()->chain().promote();
         if (dstChain == 0) {
             thread->addEffect_l(effect);
             status = INVALID_OPERATION;
@@ -4170,7 +4203,8 @@
         case TransactionCode::SET_LOW_RAM_DEVICE:
         case TransactionCode::SYSTEM_READY:
         case TransactionCode::SET_AUDIO_HAL_PIDS:
-        case TransactionCode::SET_VIBRATOR_INFOS: {
+        case TransactionCode::SET_VIBRATOR_INFOS:
+        case TransactionCode::UPDATE_SECONDARY_OUTPUTS: {
             if (!isServiceUid(IPCThreadState::self()->getCallingUid())) {
                 ALOGW("%s: transaction %d received from PID %d unauthorized UID %d",
                       __func__, code, IPCThreadState::self()->getCallingPid(),
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 4b03d10..fff61f8 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -37,6 +37,8 @@
 #include <android/media/IAudioFlingerClient.h>
 #include <android/media/IAudioTrackCallback.h>
 #include <android/os/BnExternalVibrationController.h>
+#include <android/content/AttributionSourceState.h>
+
 
 #include <android-base/macros.h>
 #include <cutils/atomic.h>
@@ -124,13 +126,15 @@
 
 #define INCLUDING_FROM_AUDIOFLINGER_H
 
+using android::content::AttributionSourceState;
+
 class AudioFlinger : public AudioFlingerServerAdapter::Delegate
 {
 public:
     static void instantiate() ANDROID_API;
 
-    static media::permission::Identity checkIdentityPackage(
-                                            const media::permission::Identity& identity);
+    static AttributionSourceState checkAttributionSourcePackage(
+        const AttributionSourceState& attributionSource);
 
     status_t dump(int fd, const Vector<String16>& args) override;
 
@@ -272,6 +276,9 @@
 
     virtual status_t setVibratorInfos(const std::vector<media::AudioVibratorInfo>& vibratorInfos);
 
+    virtual status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs);
+
     status_t onTransactWrapper(TransactionCode code, const Parcel& data, uint32_t flags,
         const std::function<status_t()>& delegate) override;
 
@@ -775,6 +782,11 @@
 
               ThreadBase *hapticPlaybackThread_l() const;
 
+              void updateSecondaryOutputsForTrack_l(
+                      PlaybackThread::Track* track,
+                      PlaybackThread* thread,
+                      const std::vector<audio_io_handle_t>& secondaryOutputs) const;
+
 
                 void        removeClient_l(pid_t pid);
                 void        removeNotificationClient(pid_t pid);
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index d75b13b..d3492d9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -152,12 +152,12 @@
     if (fromHandle) {
         if (enabled) {
             if (status != NO_ERROR) {
-                mCallback->checkSuspendOnEffectEnabled(this, false, false /*threadLocked*/);
+                getCallback()->checkSuspendOnEffectEnabled(this, false, false /*threadLocked*/);
             } else {
-                mCallback->onEffectEnable(this);
+                getCallback()->onEffectEnable(this);
             }
         } else {
-            mCallback->onEffectDisable(this);
+            getCallback()->onEffectDisable(this);
         }
     }
     return status;
@@ -247,8 +247,9 @@
             doRegister = true;
             mPolicyRegistered = mHandles.size() > 0;
             if (mPolicyRegistered) {
-                io = mCallback->io();
-                strategy = mCallback->strategy();
+                const auto callback = getCallback();
+                io = callback->io();
+                strategy = callback->strategy();
             }
         }
         // enable effect when registered according to enable state requested by controlling handle
@@ -349,8 +350,9 @@
 // unsafe method called when the effect parent thread has been destroyed
 ssize_t AudioFlinger::EffectBase::disconnectHandle(EffectHandle *handle, bool unpinIfLast)
 {
+    const auto callback = getCallback();
     ALOGV("disconnect() %p handle %p", this, handle);
-    if (mCallback->disconnectEffectHandle(handle, unpinIfLast)) {
+    if (callback->disconnectEffectHandle(handle, unpinIfLast)) {
         return mHandles.size();
     }
 
@@ -358,7 +360,7 @@
     ssize_t numHandles = removeHandle_l(handle);
     if ((numHandles == 0) && (!mPinned || unpinIfLast)) {
         mLock.unlock();
-        mCallback->updateOrphanEffectChains(this);
+        callback->updateOrphanEffectChains(this);
         mLock.lock();
     }
     return numHandles;
@@ -377,7 +379,7 @@
 }
 
 void AudioFlinger::EffectBase::checkSuspendOnEffectEnabled(bool enabled, bool threadLocked) {
-    mCallback->checkSuspendOnEffectEnabled(this, enabled, threadLocked);
+    getCallback()->checkSuspendOnEffectEnabled(this, enabled, threadLocked);
 }
 
 static String8 effectFlagsToString(uint32_t flags) {
@@ -835,7 +837,7 @@
                 mConfig.inputCfg.buffer.raw != mConfig.outputCfg.buffer.raw) {
         // If an insert effect is idle and input buffer is different from output buffer,
         // accumulate input onto output
-        if (mCallback->activeTrackCnt() != 0) {
+        if (getCallback()->activeTrackCnt() != 0) {
             // similar handling with data_bypass above.
             if (mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {
                 accumulateInputToOutput();
@@ -860,6 +862,7 @@
     status_t status;
     uint32_t size;
     audio_channel_mask_t channelMask;
+    sp<EffectCallbackInterface> callback;
 
     if (mEffectInterface == 0) {
         status = NO_INIT;
@@ -870,7 +873,8 @@
     // TODO: handle configuration of input (record) SW effects above the HAL,
     // similar to output EFFECT_FLAG_TYPE_INSERT/REPLACE,
     // in which case input channel masks should be used here.
-    channelMask = mCallback->channelMask();
+    callback = getCallback();
+    channelMask = callback->channelMask();
     mConfig.inputCfg.channels = channelMask;
     mConfig.outputCfg.channels = channelMask;
 
@@ -899,7 +903,7 @@
 #endif
     }
     if (isHapticGenerator()) {
-        audio_channel_mask_t hapticChannelMask = mCallback->hapticChannelMask();
+        audio_channel_mask_t hapticChannelMask = callback->hapticChannelMask();
         mConfig.inputCfg.channels |= hapticChannelMask;
         mConfig.outputCfg.channels |= hapticChannelMask;
     }
@@ -912,11 +916,11 @@
     mConfig.outputCfg.format = EFFECT_BUFFER_FORMAT;
 
     // Don't use sample rate for thread if effect isn't offloadable.
-    if (mCallback->isOffloadOrDirect() && !isOffloaded()) {
+    if (callback->isOffloadOrDirect() && !isOffloaded()) {
         mConfig.inputCfg.samplingRate = DEFAULT_OUTPUT_SAMPLE_RATE;
         ALOGV("Overriding effect input as 48kHz");
     } else {
-        mConfig.inputCfg.samplingRate = mCallback->sampleRate();
+        mConfig.inputCfg.samplingRate = callback->sampleRate();
     }
     mConfig.outputCfg.samplingRate = mConfig.inputCfg.samplingRate;
     mConfig.inputCfg.bufferProvider.cookie = NULL;
@@ -942,11 +946,11 @@
     }
     mConfig.inputCfg.mask = EFFECT_CONFIG_ALL;
     mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
-    mConfig.inputCfg.buffer.frameCount = mCallback->frameCount();
+    mConfig.inputCfg.buffer.frameCount = callback->frameCount();
     mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
 
     ALOGV("configure() %p chain %p buffer %p framecount %zu",
-          this, mCallback->chain().promote().get(),
+          this, callback->chain().promote().get(),
           mConfig.inputCfg.buffer.raw, mConfig.inputCfg.buffer.frameCount);
 
     status_t cmdStatus;
@@ -962,7 +966,7 @@
 
 #ifdef MULTICHANNEL_EFFECT_CHAIN
     if (status != NO_ERROR &&
-            mCallback->isOutput() &&
+            callback->isOutput() &&
             (mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO
                     || mConfig.outputCfg.channels != AUDIO_CHANNEL_OUT_STEREO)) {
         // Older effects may require exact STEREO position mask.
@@ -1029,7 +1033,7 @@
             size = sizeof(int);
             *(int32_t *)p->data = VISUALIZER_PARAM_LATENCY;
 
-            uint32_t latency = mCallback->latency();
+            uint32_t latency = callback->latency();
 
             *((int32_t *)p->data + 1)= latency;
             mEffectInterface->command(EFFECT_CMD_SET_PARAM,
@@ -1076,7 +1080,7 @@
 {
     if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
          (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
-        (void)mCallback->addEffectToHal(mEffectInterface);
+        (void)getCallback()->addEffectToHal(mEffectInterface);
     }
 }
 
@@ -1089,7 +1093,7 @@
         status = start_l();
     }
     if (status == NO_ERROR) {
-        mCallback->resetVolume();
+        getCallback()->resetVolume();
     }
     return status;
 }
@@ -1139,7 +1143,7 @@
         // We have the EffectChain and EffectModule lock, permit a reentrant call to setVolume:
         // resetVolume_l --> setVolume_l --> EffectModule::setVolume
         mSetVolumeReentrantTid = gettid();
-        mCallback->resetVolume();
+        getCallback()->resetVolume();
         mSetVolumeReentrantTid = INVALID_PID;
     }
 
@@ -1172,7 +1176,7 @@
 {
     if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC ||
              (mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_POST_PROC) {
-        mCallback->removeEffectFromHal(mEffectInterface);
+        getCallback()->removeEffectFromHal(mEffectInterface);
     }
     return NO_ERROR;
 }
@@ -1288,7 +1292,7 @@
 
 bool AudioFlinger::EffectModule::isOffloadedOrDirect() const
 {
-    return mCallback->isOffloadOrDirect();
+    return getCallback()->isOffloadOrDirect();
 }
 
 bool AudioFlinger::EffectModule::isVolumeControlEnabled() const
@@ -1332,7 +1336,7 @@
                 || size > mInConversionBuffer->getSize())) {
             mInConversionBuffer.clear();
             ALOGV("%s: allocating mInConversionBuffer %zu", __func__, size);
-            (void)mCallback->allocateHalBuffer(size, &mInConversionBuffer);
+            (void)getCallback()->allocateHalBuffer(size, &mInConversionBuffer);
         }
         if (mInConversionBuffer != nullptr) {
             mInConversionBuffer->setFrameCount(inFrameCount);
@@ -1376,7 +1380,7 @@
                 || size > mOutConversionBuffer->getSize())) {
             mOutConversionBuffer.clear();
             ALOGV("%s: allocating mOutConversionBuffer %zu", __func__, size);
-            (void)mCallback->allocateHalBuffer(size, &mOutConversionBuffer);
+            (void)getCallback()->allocateHalBuffer(size, &mOutConversionBuffer);
         }
         if (mOutConversionBuffer != nullptr) {
             mOutConversionBuffer->setFrameCount(outFrameCount);
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 9da95bc..661881e 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -138,8 +138,9 @@
                              int32_t __unused,
                              std::vector<uint8_t>* __unused) { return NO_ERROR; };
 
+    // mCallback is atomic so this can be lock-free.
     void setCallback(const sp<EffectCallbackInterface>& callback) { mCallback = callback; }
-    sp<EffectCallbackInterface>&     callback() { return mCallback; }
+    sp<EffectCallbackInterface> getCallback() const { return mCallback.load(); }
 
     status_t addHandle(EffectHandle *handle);
     ssize_t disconnectHandle(EffectHandle *handle, bool unpinIfLast);
@@ -170,7 +171,7 @@
     DISALLOW_COPY_AND_ASSIGN(EffectBase);
 
 mutable Mutex                 mLock;      // mutex for process, commands and handles list protection
-    sp<EffectCallbackInterface> mCallback; // parent effect chain
+    mediautils::atomic_sp<EffectCallbackInterface> mCallback; // parent effect chain
     const int                 mId;        // this instance unique ID
     const audio_session_t     mSessionId; // audio session ID
     const effect_descriptor_t mDescriptor;// effect descriptor received from effect engine
diff --git a/services/audioflinger/FastCapture.cpp b/services/audioflinger/FastCapture.cpp
index d6d6e25..2963202 100644
--- a/services/audioflinger/FastCapture.cpp
+++ b/services/audioflinger/FastCapture.cpp
@@ -107,7 +107,7 @@
             mSampleRate = Format_sampleRate(mFormat);
 #if !LOG_NDEBUG
             unsigned channelCount = Format_channelCount(mFormat);
-            ALOG_ASSERT(channelCount >= 1 && channelCount <= FCC_8);
+            ALOG_ASSERT(channelCount >= 1 && channelCount <= FCC_LIMIT);
 #endif
         }
         dumpState->mSampleRate = mSampleRate;
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 13e2ced..88d4eaf 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -48,6 +48,15 @@
 
 /*static*/ const FastMixerState FastMixer::sInitial;
 
+static audio_channel_mask_t getChannelMaskFromCount(size_t count) {
+    const audio_channel_mask_t mask = audio_channel_out_mask_from_count(count);
+    if (mask == AUDIO_CHANNEL_INVALID) {
+        // some counts have no positional masks. TODO: Update this to return index count?
+        return audio_channel_mask_for_index_assignment_from_count(count);
+    }
+    return mask;
+}
+
 FastMixer::FastMixer(audio_io_handle_t parentIoHandle)
     : FastThread("cycle_ms", "load_us"),
     // mFastTrackNames
@@ -79,7 +88,7 @@
     mDummyDumpState = &mDummyFastMixerDumpState;
     // TODO: Add channel mask to NBAIO_Format.
     // We assume that the channel mask must be a valid positional channel mask.
-    mSinkChannelMask = audio_channel_out_mask_from_count(mSinkChannelCount);
+    mSinkChannelMask = getChannelMaskFromCount(mSinkChannelCount);
 
     unsigned i;
     for (i = 0; i < FastMixerState::sMaxFastTracks; ++i) {
@@ -238,7 +247,7 @@
             LOG_ALWAYS_FATAL_IF(mSinkChannelCount > AudioMixer::MAX_NUM_CHANNELS);
 
             if (mSinkChannelMask == AUDIO_CHANNEL_NONE) {
-                mSinkChannelMask = audio_channel_out_mask_from_count(mSinkChannelCount);
+                mSinkChannelMask = getChannelMaskFromCount(mSinkChannelCount);
             }
             mAudioChannelCount = mSinkChannelCount - audio_channel_count_from_out_mask(
                     mSinkChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
diff --git a/services/audioflinger/MmapTracks.h b/services/audioflinger/MmapTracks.h
index ba868d7..eb640bb 100644
--- a/services/audioflinger/MmapTracks.h
+++ b/services/audioflinger/MmapTracks.h
@@ -29,7 +29,7 @@
                             audio_channel_mask_t channelMask,
                             audio_session_t sessionId,
                             bool isOut,
-                            const media::permission::Identity& identity,
+                            const android::content::AttributionSourceState& attributionSource,
                             pid_t creatorPid,
                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
     virtual             ~MmapTrack();
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 0af4c7b..51a41af 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -15,8 +15,6 @@
 ** limitations under the License.
 */
 
-#include <android/media/permission/Identity.h>
-
 #ifndef INCLUDING_FROM_AUDIOFLINGER_H
     #error This header file should only be included from AudioFlinger.h
 #endif
@@ -28,12 +26,12 @@
     bool hasOpPlayAudio() const;
 
     static sp<OpPlayAudioMonitor> createIfNeeded(
-            const android::media::permission::Identity& identity,
+            const AttributionSourceState& attributionSource,
             const audio_attributes_t& attr, int id,
             audio_stream_type_t streamType);
 
 private:
-    OpPlayAudioMonitor(const android::media::permission::Identity& identity,
+    OpPlayAudioMonitor(const AttributionSourceState& attributionSource,
         audio_usage_t usage, int id);
     void onFirstRef() override;
     static void getPackagesForUid(uid_t uid, Vector<String16>& packages);
@@ -54,7 +52,7 @@
     void checkPlayAudioForUsage();
 
     std::atomic_bool mHasOpPlayAudio;
-    const android::media::permission::Identity mIdentity;
+    const AttributionSourceState mAttributionSource;
     const int32_t mUsage; // on purpose not audio_usage_t because always checked in appOps as int32_t
     const int mId; // for logging purposes only
 };
@@ -75,13 +73,14 @@
                                 const sp<IMemory>& sharedBuffer,
                                 audio_session_t sessionId,
                                 pid_t creatorPid,
-                                const media::permission::Identity& identity,
+                                const AttributionSourceState& attributionSource,
                                 audio_output_flags_t flags,
                                 track_type type,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
                                 /** default behaviour is to start when there are as many frames
                                   * ready as possible (aka. Buffer is full). */
-                                size_t frameCountToBeReady = SIZE_MAX);
+                                size_t frameCountToBeReady = SIZE_MAX,
+                                float speed = 1.0f);
     virtual             ~Track();
     virtual status_t    initCheck() const;
 
@@ -183,6 +182,9 @@
                    mAudioTrackServerProxy->getUnderrunFrames());
        }
     }
+
+    audio_output_flags_t getOutputFlags() const { return mFlags; }
+    float getSpeed() const { return mSpeed; }
 protected:
     // for numerous
     friend class PlaybackThread;
@@ -311,6 +313,7 @@
     bool                mPauseHwPending = false; // direct/offload track request for thread pause
     audio_output_flags_t mFlags;
     TeePatches  mTeePatches;
+    const float         mSpeed;
 };  // end of Track
 
 
@@ -329,7 +332,7 @@
                                 audio_format_t format,
                                 audio_channel_mask_t channelMask,
                                 size_t frameCount,
-                                const android::media::permission::Identity& identity);
+                                const AttributionSourceState& attributionSource);
     virtual             ~OutputTrack();
 
     virtual status_t    start(AudioSystem::sync_event_t event =
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index a1c2de7..88aa7cb 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -15,6 +15,8 @@
 ** limitations under the License.
 */
 
+#include <android/content/AttributionSourceState.h>
+
 #ifndef INCLUDING_FROM_AUDIOFLINGER_H
     #error This header file should only be included from AudioFlinger.h
 #endif
@@ -26,11 +28,12 @@
     bool hasOp() const;
     int32_t getOp() const { return mAppOp; }
 
-    static sp<OpRecordAudioMonitor> createIfNeeded
-        (const media::permission::Identity& identity, const audio_attributes_t& attr);
+    static sp<OpRecordAudioMonitor> createIfNeeded(const AttributionSourceState& attributionSource,
+            const audio_attributes_t& attr);
 
 private:
-    OpRecordAudioMonitor(const media::permission::Identity& identity, int32_t appOp);
+    OpRecordAudioMonitor(const AttributionSourceState& attributionSource, int32_t appOp);
+
     void onFirstRef() override;
 
     AppOpsManager mAppOpsManager;
@@ -50,7 +53,7 @@
     void checkOp();
 
     std::atomic_bool mHasOp;
-    const media::permission::Identity mIdentity;
+    const AttributionSourceState mAttributionSource;
     const int32_t mAppOp;
 };
 
@@ -68,7 +71,7 @@
                                 size_t bufferSize,
                                 audio_session_t sessionId,
                                 pid_t creatorPid,
-                                const media::permission::Identity& identity,
+                                const AttributionSourceState& attributionSource,
                                 audio_input_flags_t flags,
                                 track_type type,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index d878611..f62082e 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -118,7 +118,7 @@
 namespace android {
 
 using media::IEffectClient;
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 // retry counts for buffer fill timeout
 // 50 * ~20msecs = 1 second
@@ -1638,7 +1638,7 @@
         detachAuxEffect_l(effect->id());
     }
 
-    sp<EffectChain> chain = effect->callback()->chain().promote();
+    sp<EffectChain> chain = effect->getCallback()->chain().promote();
     if (chain != 0) {
         // remove effect chain if removing last effect
         if (chain->removeEffect_l(effect, release) == 0) {
@@ -2129,7 +2129,7 @@
         audio_session_t sessionId,
         audio_output_flags_t *flags,
         pid_t creatorPid,
-        const Identity& identity,
+        const AttributionSourceState& attributionSource,
         pid_t tid,
         status_t *status,
         audio_port_handle_t portId,
@@ -2424,8 +2424,8 @@
         track = new Track(this, client, streamType, attr, sampleRate, format,
                           channelMask, frameCount,
                           nullptr /* buffer */, (size_t)0 /* bufferSize */, sharedBuffer,
-                          sessionId, creatorPid, identity, trackFlags, TrackBase::TYPE_DEFAULT,
-                          portId, SIZE_MAX /*frameCountToBeReady*/);
+                          sessionId, creatorPid, attributionSource, trackFlags,
+                          TrackBase::TYPE_DEFAULT, portId, SIZE_MAX /*frameCountToBeReady*/, speed);
 
         lStatus = track != 0 ? track->initCheck() : (status_t) NO_MEMORY;
         if (lStatus != NO_ERROR) {
@@ -3321,6 +3321,17 @@
     invalidateTracks_l(streamType);
 }
 
+// getTrackById_l must be called with holding thread lock
+AudioFlinger::PlaybackThread::Track* AudioFlinger::PlaybackThread::getTrackById_l(
+        audio_port_handle_t trackPortId) {
+    for (size_t i = 0; i < mTracks.size(); i++) {
+        if (mTracks[i]->portId() == trackPortId) {
+            return mTracks[i].get();
+        }
+    }
+    return nullptr;
+}
+
 status_t AudioFlinger::PlaybackThread::addEffectChain_l(const sp<EffectChain>& chain)
 {
     audio_session_t session = chain->sessionId();
@@ -6875,19 +6886,20 @@
     // from different OutputTracks and their associated MixerThreads (e.g. one may
     // nearly empty and the other may be dropping data).
 
-    // TODO b/182392769: use identity util, move to server edge
-    Identity identity = Identity();
-    identity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
+    // TODO b/182392769: use attribution source util, move to server edge
+    AttributionSourceState attributionSource = AttributionSourceState();
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(
         IPCThreadState::self()->getCallingUid()));
-    identity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(
       IPCThreadState::self()->getCallingPid()));
+    attributionSource.token = sp<BBinder>::make();
     sp<OutputTrack> outputTrack = new OutputTrack(thread,
                                             this,
                                             mSampleRate,
                                             mFormat,
                                             mChannelMask,
                                             frameCount,
-                                            identity);
+                                            attributionSource);
     status_t status = outputTrack != 0 ? outputTrack->initCheck() : (status_t) NO_MEMORY;
     if (status != NO_ERROR) {
         ALOGE("addOutputTrack() initCheck failed %d", status);
@@ -7811,7 +7823,7 @@
         audio_session_t sessionId,
         size_t *pNotificationFrameCount,
         pid_t creatorPid,
-        const Identity& identity,
+        const AttributionSourceState& attributionSource,
         audio_input_flags_t *flags,
         pid_t tid,
         status_t *status,
@@ -7825,7 +7837,8 @@
     audio_input_flags_t inputFlags = mInput->flags;
     audio_input_flags_t requestedFlags = *flags;
     uint32_t sampleRate;
-    Identity checkedIdentity = AudioFlinger::checkIdentityPackage(identity);
+    AttributionSourceState checkedAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+            attributionSource);
 
     lStatus = initCheck();
     if (lStatus != NO_ERROR) {
@@ -7840,7 +7853,7 @@
     }
 
     if (maxSharedAudioHistoryMs != 0) {
-        if (!captureHotwordAllowed(checkedIdentity)) {
+        if (!captureHotwordAllowed(checkedAttributionSource)) {
             lStatus = PERMISSION_DENIED;
             goto Exit;
         }
@@ -7962,16 +7975,17 @@
         Mutex::Autolock _l(mLock);
         int32_t startFrames = -1;
         if (!mSharedAudioPackageName.empty()
-                && mSharedAudioPackageName == checkedIdentity.packageName
+                && mSharedAudioPackageName == checkedAttributionSource.packageName
                 && mSharedAudioSessionId == sessionId
-                && captureHotwordAllowed(checkedIdentity)) {
+                && captureHotwordAllowed(checkedAttributionSource)) {
             startFrames = mSharedAudioStartFrames;
         }
 
         track = new RecordTrack(this, client, attr, sampleRate,
                       format, channelMask, frameCount,
                       nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid,
-                      checkedIdentity, *flags, TrackBase::TYPE_DEFAULT, portId, startFrames);
+                      checkedAttributionSource, *flags, TrackBase::TYPE_DEFAULT, portId,
+                      startFrames);
 
         lStatus = track->initCheck();
         if (lStatus != NO_ERROR) {
@@ -8559,7 +8573,7 @@
     if (param.getInt(String8(AudioParameter::keyChannels), value) == NO_ERROR) {
         audio_channel_mask_t mask = (audio_channel_mask_t) value;
         if (!audio_is_input_channel(mask) ||
-                audio_channel_count_from_in_mask(mask) > FCC_8) {
+                audio_channel_count_from_in_mask(mask) > FCC_LIMIT) {
             status = BAD_VALUE;
         } else {
             channelMask = mask;
@@ -8596,7 +8610,7 @@
                 if (mInput->stream->getAudioProperties(&config) == OK &&
                         audio_is_linear_pcm(config.format) && audio_is_linear_pcm(reqFormat) &&
                         config.sample_rate <= (AUDIO_RESAMPLER_DOWN_RATIO_MAX * samplingRate) &&
-                        audio_channel_count_from_in_mask(config.channel_mask) <= FCC_8) {
+                        audio_channel_count_from_in_mask(config.channel_mask) <= FCC_LIMIT) {
                     status = NO_ERROR;
                 }
             }
@@ -8658,10 +8672,10 @@
     mFormat = mHALFormat;
     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);
+        LOG_ALWAYS_FATAL_IF(mChannelCount > FCC_LIMIT, "HAL channel count %d > %d",
+                mChannelCount, FCC_LIMIT);
     } else {
-        // Can have more that FCC_8 channels in encoded streams.
+        // Can have more that FCC_LIMIT channels in encoded streams.
         ALOGI("HAL format %#x is not linear pcm", mFormat);
     }
     result = mInput->stream->getFrameSize(&mFrameSize);
@@ -9163,7 +9177,7 @@
                                          audio_port_handle_t *handle)
 {
     ALOGV("%s clientUid %d mStandby %d mPortId %d *handle %d", __FUNCTION__,
-          client.identity.uid, mStandby, mPortId, *handle);
+          client.attributionSource.uid, mStandby, mPortId, *handle);
     if (mHalStream == 0) {
         return NO_INIT;
     }
@@ -9195,7 +9209,7 @@
         ret = AudioSystem::getOutputForAttr(&mAttr, &io,
                                             mSessionId,
                                             &stream,
-                                            client.identity,
+                                            client.attributionSource,
                                             &config,
                                             flags,
                                             &deviceId,
@@ -9212,7 +9226,7 @@
         ret = AudioSystem::getInputForAttr(&mAttr, &io,
                                               RECORD_RIID_INVALID,
                                               mSessionId,
-                                              client.identity,
+                                              client.attributionSource,
                                               &config,
                                               AUDIO_INPUT_FLAG_MMAP_NOIRQ,
                                               &deviceId,
@@ -9252,7 +9266,8 @@
 
     // Given that MmapThread::mAttr is mutable, should a MmapTrack have attributes ?
     sp<MmapTrack> track = new MmapTrack(this, attr == nullptr ? mAttr : *attr, mSampleRate, mFormat,
-                                        mChannelMask, mSessionId, isOutput(), client.identity,
+                                        mChannelMask, mSessionId, isOutput(),
+                                        client.attributionSource,
                                         IPCThreadState::self()->getCallingPid(), portId);
 
     if (isOutput()) {
@@ -9260,7 +9275,7 @@
         mHalVolFloat = -1.0f;
     } else if (!track->isSilenced_l()) {
         for (const sp<MmapTrack> &t : mActiveTracks) {
-            if (t->isSilenced_l() && t->uid() != client.identity.uid)
+            if (t->isSilenced_l() && t->uid() != client.attributionSource.uid)
                 t->invalidate();
         }
     }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 17acb16..eee1f2b 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -918,7 +918,7 @@
                                 audio_session_t sessionId,
                                 audio_output_flags_t *flags,
                                 pid_t creatorPid,
-                                const media::permission::Identity& identity,
+                                const AttributionSourceState& attributionSource,
                                 pid_t tid,
                                 status_t *status /*non-NULL*/,
                                 audio_port_handle_t portId,
@@ -1013,6 +1013,8 @@
                     mDownStreamPatch = *patch;
                 }
 
+                PlaybackThread::Track* getTrackById_l(audio_port_handle_t trackId);
+
 protected:
     // updated by readOutputParameters_l()
     size_t                          mNormalFrameCount;  // normal mixer and effects
@@ -1696,7 +1698,7 @@
                     audio_session_t sessionId,
                     size_t *pNotificationFrameCount,
                     pid_t creatorPid,
-                    const media::permission::Identity& identity,
+                    const AttributionSourceState& attributionSource,
                     audio_input_flags_t *flags,
                     pid_t tid,
                     status_t *status /*non-NULL*/,
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 8be7c86..57ff0d7 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -65,7 +65,7 @@
 
 using ::android::aidl_utils::binderStatusFromStatusT;
 using binder::Status;
-using media::permission::Identity;
+using content::AttributionSourceState;
 using media::VolumeShaper;
 // ----------------------------------------------------------------------------
 //      TrackBase
@@ -238,12 +238,13 @@
     }
 }
 
-// TODO b/182392769: use identity util
-static Identity audioServerIdentity(pid_t pid) {
-   Identity i{};
-   i.uid = AID_AUDIOSERVER;
-   i.pid = pid;
-   return i;
+// TODO b/182392769: use attribution source util
+static AttributionSourceState audioServerAttributionSource(pid_t pid) {
+   AttributionSourceState attributionSource{};
+   attributionSource.uid = AID_AUDIOSERVER;
+   attributionSource.pid = pid;
+   attributionSource.token = sp<BBinder>::make();
+   return attributionSource;
 }
 
 status_t AudioFlinger::ThreadBase::TrackBase::initCheck() const
@@ -498,11 +499,11 @@
 // static
 sp<AudioFlinger::PlaybackThread::OpPlayAudioMonitor>
 AudioFlinger::PlaybackThread::OpPlayAudioMonitor::createIfNeeded(
-            const Identity& identity, const audio_attributes_t& attr, int id,
+            const AttributionSourceState& attributionSource, const audio_attributes_t& attr, int id,
             audio_stream_type_t streamType)
 {
     Vector <String16> packages;
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
     getPackagesForUid(uid, packages);
     if (isServiceUid(uid)) {
         if (packages.isEmpty()) {
@@ -525,13 +526,15 @@
         return nullptr;
     }
 
-    Identity checkedIdentity = AudioFlinger::checkIdentityPackage(identity);
-    return new OpPlayAudioMonitor(checkedIdentity, attr.usage, id);
+    AttributionSourceState checkedAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+            attributionSource);
+    return new OpPlayAudioMonitor(checkedAttributionSource, attr.usage, id);
 }
 
 AudioFlinger::PlaybackThread::OpPlayAudioMonitor::OpPlayAudioMonitor(
-        const Identity& identity, audio_usage_t usage, int id)
-        : mHasOpPlayAudio(true), mIdentity(identity), mUsage((int32_t) usage), mId(id)
+        const AttributionSourceState& attributionSource, audio_usage_t usage, int id)
+        : mHasOpPlayAudio(true), mAttributionSource(attributionSource), mUsage((int32_t) usage),
+        mId(id)
 {
 }
 
@@ -546,10 +549,11 @@
 void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::onFirstRef()
 {
     checkPlayAudioForUsage();
-    if (mIdentity.packageName.has_value()) {
+    if (mAttributionSource.packageName.has_value()) {
         mOpCallback = new PlayAudioOpCallback(this);
         mAppOpsManager.startWatchingMode(AppOpsManager::OP_PLAY_AUDIO,
-            VALUE_OR_FATAL(aidl2legacy_string_view_String16(mIdentity.packageName.value_or("")))
+            VALUE_OR_FATAL(aidl2legacy_string_view_String16(
+            mAttributionSource.packageName.value_or("")))
             , mOpCallback);
     }
 }
@@ -563,12 +567,12 @@
 // - not called from PlayAudioOpCallback because the callback is not installed in this case
 void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage()
 {
-    if (!mIdentity.packageName.has_value()) {
+    if (!mAttributionSource.packageName.has_value()) {
         mHasOpPlayAudio.store(false);
     } else {
-        uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mIdentity.uid));
+        uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mAttributionSource.uid));
         String16 packageName = VALUE_OR_FATAL(
-            aidl2legacy_string_view_String16(mIdentity.packageName.value_or("")));
+            aidl2legacy_string_view_String16(mAttributionSource.packageName.value_or("")));
         bool hasIt = mAppOpsManager.checkAudioOpNoThrow(AppOpsManager::OP_PLAY_AUDIO,
                     mUsage, uid, packageName) == AppOpsManager::MODE_ALLOWED;
         ALOGD("OpPlayAudio: track:%d usage:%d %smuted", mId, mUsage, hasIt ? "not " : "");
@@ -620,11 +624,12 @@
             const sp<IMemory>& sharedBuffer,
             audio_session_t sessionId,
             pid_t creatorPid,
-            const Identity& identity,
+            const AttributionSourceState& attributionSource,
             audio_output_flags_t flags,
             track_type type,
             audio_port_handle_t portId,
-            size_t frameCountToBeReady)
+            size_t frameCountToBeReady,
+            float speed)
     :   TrackBase(thread, client, attr, sampleRate, format, channelMask, frameCount,
                   // TODO: Using unsecurePointer() has some associated security pitfalls
                   //       (see declaration for details).
@@ -633,7 +638,7 @@
                   (sharedBuffer != 0) ? sharedBuffer->unsecurePointer() : buffer,
                   (sharedBuffer != 0) ? sharedBuffer->size() : bufferSize,
                   sessionId, creatorPid,
-                  VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid)), true /*isOut*/,
+                  VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid)), true /*isOut*/,
                   (type == TYPE_PATCH) ? ( buffer == NULL ? ALLOC_LOCAL : ALLOC_NONE) : ALLOC_CBLK,
                   type,
                   portId,
@@ -648,7 +653,7 @@
     mPresentationCompleteFrames(0),
     mFrameMap(16 /* sink-frame-to-track-frame map memory */),
     mVolumeHandler(new media::VolumeHandler(sampleRate)),
-    mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(identity, attr, id(),
+    mOpPlayAudioMonitor(OpPlayAudioMonitor::createIfNeeded(attributionSource, attr, id(),
         streamType)),
     // mSinkTimestamp
     mFastIndex(-1),
@@ -658,7 +663,8 @@
     mFinalVolume(0.f),
     mResumeToStopping(false),
     mFlushHwPending(false),
-    mFlags(flags)
+    mFlags(flags),
+    mSpeed(speed)
 {
     // client == 0 implies sharedBuffer == 0
     ALOG_ASSERT(!(client == 0 && sharedBuffer != 0));
@@ -670,7 +676,7 @@
         return;
     }
 
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid));
+    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid));
     if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
         ALOGE("%s(%d): no more tracks available", __func__, mId);
         releaseCblk(); // this makes the track invalid.
@@ -716,8 +722,8 @@
         // HapticGenerator effect, which will generate haptic data, on the track. In that case,
         // external vibration is always created for all tracks attached to haptic playback thread.
         mAudioVibrationController = new AudioVibrationController(this);
-        std::string packageName = identity.packageName.has_value() ?
-            identity.packageName.value() : "";
+        std::string packageName = attributionSource.packageName.has_value() ?
+            attributionSource.packageName.value() : "";
         mExternalVibration = new os::ExternalVibration(
                 mUid, packageName, mAttr, mAudioVibrationController);
     }
@@ -1404,6 +1410,10 @@
 void AudioFlinger::PlaybackThread::Track::setTeePatches(TeePatches teePatches) {
     forEachTeePatchTrack([](auto patchTrack) { patchTrack->destroy(); });
     mTeePatches = std::move(teePatches);
+    if (mState == TrackBase::ACTIVE || mState == TrackBase::RESUMING ||
+            mState == TrackBase::STOPPING_1) {
+        forEachTeePatchTrack([](auto patchTrack) { patchTrack->start(); });
+    }
 }
 
 status_t AudioFlinger::PlaybackThread::Track::getTimestamp(AudioTimestamp& timestamp)
@@ -1834,12 +1844,12 @@
             audio_format_t format,
             audio_channel_mask_t channelMask,
             size_t frameCount,
-            const Identity& identity)
+            const AttributionSourceState& attributionSource)
     :   Track(playbackThread, NULL, AUDIO_STREAM_PATCH,
               audio_attributes_t{} /* currently unused for output track */,
               sampleRate, format, channelMask, frameCount,
               nullptr /* buffer */, (size_t)0 /* bufferSize */, nullptr /* sharedBuffer */,
-              AUDIO_SESSION_NONE, getpid(), identity, AUDIO_OUTPUT_FLAG_NONE,
+              AUDIO_SESSION_NONE, getpid(), attributionSource, AUDIO_OUTPUT_FLAG_NONE,
               TYPE_OUTPUT),
     mActive(false), mSourceThread(sourceThread)
 {
@@ -2069,7 +2079,7 @@
               audio_attributes_t{} /* currently unused for patch track */,
               sampleRate, format, channelMask, frameCount,
               buffer, bufferSize, nullptr /* sharedBuffer */,
-              AUDIO_SESSION_NONE, getpid(), audioServerIdentity(getpid()), flags,
+              AUDIO_SESSION_NONE, getpid(), audioServerAttributionSource(getpid()), flags,
               TYPE_PATCH, AUDIO_PORT_HANDLE_NONE, frameCountToBeReady),
         PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, true, true),
                        *playbackThread, timeout)
@@ -2207,32 +2217,33 @@
 // static
 sp<AudioFlinger::RecordThread::OpRecordAudioMonitor>
 AudioFlinger::RecordThread::OpRecordAudioMonitor::createIfNeeded(
-            const Identity& identity, const audio_attributes_t& attr)
+            const AttributionSourceState& attributionSource, const audio_attributes_t& attr)
 {
-    if (isServiceUid(identity.uid)) {
+    if (isServiceUid(attributionSource.uid)) {
         ALOGV("not silencing record for service %s",
-                identity.toString().c_str());
+                attributionSource.toString().c_str());
         return nullptr;
     }
 
     // Capturing from FM TUNER output is not controlled by an app op
     // because it does not affect users privacy as does capturing from an actual microphone.
     if (attr.source == AUDIO_SOURCE_FM_TUNER) {
-        ALOGV("not muting FM TUNER capture for uid %d", identity.uid);
+        ALOGV("not muting FM TUNER capture for uid %d", attributionSource.uid);
         return nullptr;
     }
 
-    Identity checkedIdentity = AudioFlinger::checkIdentityPackage(identity);
-    if (!checkedIdentity.packageName.has_value()
-            || checkedIdentity.packageName.value().size() == 0) {
+    AttributionSourceState checkedAttributionSource = AudioFlinger::checkAttributionSourcePackage(
+            attributionSource);
+    if (!checkedAttributionSource.packageName.has_value()
+            || checkedAttributionSource.packageName.value().size() == 0) {
         return nullptr;
     }
-    return new OpRecordAudioMonitor(checkedIdentity, getOpForSource(attr.source));
+    return new OpRecordAudioMonitor(checkedAttributionSource, getOpForSource(attr.source));
 }
 
 AudioFlinger::RecordThread::OpRecordAudioMonitor::OpRecordAudioMonitor(
-        const Identity& identity, int32_t appOp)
-        : mHasOp(true), mIdentity(identity), mAppOp(appOp)
+        const AttributionSourceState& attributionSource, int32_t appOp)
+        : mHasOp(true), mAttributionSource(attributionSource), mAppOp(appOp)
 {
 }
 
@@ -2248,9 +2259,11 @@
 {
     checkOp();
     mOpCallback = new RecordAudioOpCallback(this);
-    ALOGV("start watching op %d for %s", mAppOp, mIdentity.toString().c_str());
-    mAppOpsManager.startWatchingMode(mAppOp,
-        VALUE_OR_FATAL(aidl2legacy_string_view_String16(mIdentity.packageName.value_or(""))),
+    ALOGV("start watching op %d for %s", mAppOp, mAttributionSource.toString().c_str());
+    // TODO: We need to always watch AppOpsManager::OP_RECORD_AUDIO too
+    // since it controls the mic permission for legacy apps.
+    mAppOpsManager.startWatchingMode(mAppOp, VALUE_OR_FATAL(aidl2legacy_string_view_String16(
+        mAttributionSource.packageName.value_or(""))),
         mOpCallback);
 }
 
@@ -2266,17 +2279,17 @@
 // - not called from RecordAudioOpCallback because the callback is not installed in this case
 void AudioFlinger::RecordThread::OpRecordAudioMonitor::checkOp()
 {
-
+    // TODO: We need to always check AppOpsManager::OP_RECORD_AUDIO too
+    // since it controls the mic permission for legacy apps.
     const int32_t mode = mAppOpsManager.checkOp(mAppOp,
-            mIdentity.uid, VALUE_OR_FATAL(aidl2legacy_string_view_String16(
-                mIdentity.packageName.value_or(""))));
-    const bool hasIt =  (mode == AppOpsManager::MODE_ALLOWED);
+            mAttributionSource.uid, VALUE_OR_FATAL(aidl2legacy_string_view_String16(
+                mAttributionSource.packageName.value_or(""))));
+    const bool hasIt = (mode == AppOpsManager::MODE_ALLOWED);
     // verbose logging only log when appOp changed
     ALOGI_IF(hasIt != mHasOp.load(),
             "App op %d missing, %ssilencing record %s",
-            mAppOp, hasIt ? "un" : "", mIdentity.toString().c_str());
+            mAppOp, hasIt ? "un" : "", mAttributionSource.toString().c_str());
     mHasOp.store(hasIt);
-
 }
 
 AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::RecordAudioOpCallback(
@@ -2376,7 +2389,7 @@
             size_t bufferSize,
             audio_session_t sessionId,
             pid_t creatorPid,
-            const Identity& identity,
+            const AttributionSourceState& attributionSource,
             audio_input_flags_t flags,
             track_type type,
             audio_port_handle_t portId,
@@ -2384,7 +2397,7 @@
     :   TrackBase(thread, client, attr, sampleRate, format,
                   channelMask, frameCount, buffer, bufferSize, sessionId,
                   creatorPid,
-                  VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid)),
+                  VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid)),
                   false /*isOut*/,
                   (type == TYPE_DEFAULT) ?
                           ((flags & AUDIO_INPUT_FLAG_FAST) ? ALLOC_PIPE : ALLOC_CBLK) :
@@ -2397,7 +2410,7 @@
         mRecordBufferConverter(NULL),
         mFlags(flags),
         mSilenced(false),
-        mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(identity, attr)),
+        mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(attributionSource, attr)),
         mStartFrames(startFrames)
 {
     if (mCblk == NULL) {
@@ -2706,10 +2719,11 @@
         return PERMISSION_DENIED;
     }
 
-    Identity identity{};
-    identity.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
-    identity.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingPid));
-    if (!captureHotwordAllowed(identity)) {
+    AttributionSourceState attributionSource{};
+    attributionSource.uid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
+    attributionSource.pid = VALUE_OR_RETURN_STATUS(legacy2aidl_uid_t_int32_t(callingPid));
+    attributionSource.token = sp<BBinder>::make();
+    if (!captureHotwordAllowed(attributionSource)) {
         return PERMISSION_DENIED;
     }
 
@@ -2745,7 +2759,7 @@
                 audio_attributes_t{} /* currently unused for patch track */,
                 sampleRate, format, channelMask, frameCount,
                 buffer, bufferSize, AUDIO_SESSION_NONE, getpid(),
-                audioServerIdentity(getpid()), flags, TYPE_PATCH),
+                audioServerAttributionSource(getpid()), flags, TYPE_PATCH),
         PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true),
                        *recordThread, timeout)
 {
@@ -3022,19 +3036,19 @@
         audio_channel_mask_t channelMask,
         audio_session_t sessionId,
         bool isOut,
-        const Identity& identity,
+        const AttributionSourceState& attributionSource,
         pid_t creatorPid,
         audio_port_handle_t portId)
     :   TrackBase(thread, NULL, attr, sampleRate, format,
                   channelMask, (size_t)0 /* frameCount */,
                   nullptr /* buffer */, (size_t)0 /* bufferSize */,
                   sessionId, creatorPid,
-                  VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.uid)),
+                  VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.uid)),
                   isOut,
                   ALLOC_NONE,
                   TYPE_DEFAULT, portId,
                   std::string(AMEDIAMETRICS_KEY_PREFIX_AUDIO_MMAP) + std::to_string(portId)),
-        mPid(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(identity.pid))),
+        mPid(VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(attributionSource.pid))),
             mSilenced(false), mSilencedNotified(false)
 {
     // Once this item is logged by the server, the client can add properties.
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 5f052a5..2e49e71 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -17,15 +17,18 @@
 #ifndef ANDROID_AUDIOPOLICY_INTERFACE_H
 #define ANDROID_AUDIOPOLICY_INTERFACE_H
 
+#include <media/AudioCommonTypes.h>
 #include <media/AudioDeviceTypeAddr.h>
 #include <media/AudioSystem.h>
 #include <media/AudioPolicy.h>
 #include <media/DeviceDescriptorBase.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <utils/String8.h>
 
 namespace android {
 
+using content::AttributionSourceState;
+
 // ----------------------------------------------------------------------------
 
 // The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
@@ -123,7 +126,7 @@
                                         audio_io_handle_t *output,
                                         audio_session_t session,
                                         audio_stream_type_t *stream,
-                                        const media::permission::Identity& identity,
+                                        const AttributionSourceState& attributionSouce,
                                         const audio_config_t *config,
                                         audio_output_flags_t *flags,
                                         audio_port_handle_t *selectedDeviceId,
@@ -142,7 +145,7 @@
                                      audio_io_handle_t *input,
                                      audio_unique_id_t riid,
                                      audio_session_t session,
-                                     const media::permission::Identity& identity,
+                                     const AttributionSourceState& attributionSouce,
                                      const audio_config_base_t *config,
                                      audio_input_flags_t flags,
                                      audio_port_handle_t *selectedDeviceId,
@@ -453,6 +456,9 @@
     virtual void setSoundTriggerCaptureState(bool active) = 0;
 
     virtual status_t getAudioPort(struct audio_port_v7 *port) = 0;
+
+    virtual status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs) = 0;
 };
 
     // These are the signatures of createAudioPolicyManager/destroyAudioPolicyManager
diff --git a/services/audiopolicy/common/include/policy.h b/services/audiopolicy/common/include/policy.h
index 552919d..577f641 100644
--- a/services/audiopolicy/common/include/policy.h
+++ b/services/audiopolicy/common/include/policy.h
@@ -42,7 +42,7 @@
 
 // For mixed output and inputs, the policy will use max mixer channel count.
 // Do not limit channel count otherwise
-#define MAX_MIXER_CHANNEL_COUNT FCC_8
+#define MAX_MIXER_CHANNEL_COUNT FCC_LIMIT
 
 /**
  * Alias to AUDIO_DEVICE_OUT_DEFAULT defined for clarification when this value is used by volume
diff --git a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
index 59876c6..74b3405 100644
--- a/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/ClientDescriptor.h
@@ -109,6 +109,9 @@
     const std::vector<wp<SwAudioOutputDescriptor>>& getSecondaryOutputs() const {
         return mSecondaryOutputs;
     };
+    void setSecondaryOutputs(std::vector<wp<SwAudioOutputDescriptor>>&& secondaryOutputs) {
+        mSecondaryOutputs = std::move(secondaryOutputs);
+    }
     VolumeSource volumeSource() const { return mVolumeSource; }
     const sp<AudioPolicyMix> getPrimaryMix() const {
         return mPrimaryMix.promote();
@@ -143,7 +146,7 @@
     const product_strategy_t mStrategy;
     const VolumeSource mVolumeSource;
     const audio_output_flags_t mFlags;
-    const std::vector<wp<SwAudioOutputDescriptor>> mSecondaryOutputs;
+    std::vector<wp<SwAudioOutputDescriptor>> mSecondaryOutputs;
     const wp<AudioPolicyMix> mPrimaryMix;
     /**
      * required for duplicating thread, prevent from removing active client from an output
diff --git a/services/audiopolicy/fuzzer/Android.bp b/services/audiopolicy/fuzzer/Android.bp
index 38bdedc..faf15d6 100644
--- a/services/audiopolicy/fuzzer/Android.bp
+++ b/services/audiopolicy/fuzzer/Android.bp
@@ -50,7 +50,7 @@
         "libbinder",
         "libaudiopolicy",
         "libaudiopolicymanagerdefault",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
     static_libs: [
         "android.hardware.audio.common@7.0-enums",
diff --git a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
index 1177b95..7000cd9 100644
--- a/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
+++ b/services/audiopolicy/fuzzer/audiopolicy_fuzzer.cpp
@@ -26,7 +26,7 @@
 
 #include <Serializer.h>
 #include <android-base/file.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <libxml/parser.h>
 #include <libxml/xinclude.h>
 #include <media/AudioPolicy.h>
@@ -47,7 +47,7 @@
 using namespace ::android::audio::policy::configuration::V7_0;
 }
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 static const std::vector<audio_format_t> kAudioFormats = [] {
     std::vector<audio_format_t> result;
@@ -249,11 +249,12 @@
     *portId = AUDIO_PORT_HANDLE_NONE;
     AudioPolicyInterface::output_type_t outputType;
 
-    // TODO b/182392769: use identity util
-    Identity i;
-    i.uid = 0;
-    if (mManager->getOutputForAttr(&attr, output, AUDIO_SESSION_NONE, &stream, i, &config,
-                                   &flags, selectedDeviceId, portId, {}, &outputType) != OK) {
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource;
+    attributionSource.uid = 0;
+    attributionSource.token = sp<BBinder>::make();
+    if (mManager->getOutputForAttr(&attr, output, AUDIO_SESSION_NONE, &stream, attributionSource,
+            &config, &flags, selectedDeviceId, portId, {}, &outputType) != OK) {
         return false;
     }
     if (*output == AUDIO_IO_HANDLE_NONE || *portId == AUDIO_PORT_HANDLE_NONE) {
@@ -276,10 +277,11 @@
     *portId = AUDIO_PORT_HANDLE_NONE;
     AudioPolicyInterface::input_type_t inputType;
 
-    Identity i;
-    i.uid = 0;
-    if (mManager->getInputForAttr(&attr, &input, riid, AUDIO_SESSION_NONE, i, &config,
-                                  flags, selectedDeviceId, &inputType, portId) != OK) {
+    AttributionSourceState attributionSource;
+    attributionSource.uid = 0;
+    attributionSource.token = sp<BBinder>::make();
+    if (mManager->getInputForAttr(&attr, &input, riid, AUDIO_SESSION_NONE, attributionSource,
+            &config, flags, selectedDeviceId, &inputType, portId) != OK) {
         return false;
     }
     if (*portId == AUDIO_PORT_HANDLE_NONE || input == AUDIO_IO_HANDLE_NONE) {
diff --git a/services/audiopolicy/managerdefault/Android.bp b/services/audiopolicy/managerdefault/Android.bp
index b111db4..0165dc8 100644
--- a/services/audiopolicy/managerdefault/Android.bp
+++ b/services/audiopolicy/managerdefault/Android.bp
@@ -34,7 +34,7 @@
         // a dependency on it in the device makefile. There will be no build time
         // conflict with libaudiopolicyenginedefault.
         "libaudiopolicyenginedefault",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 7185435..2e866ff 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -31,6 +31,7 @@
 
 #include <algorithm>
 #include <inttypes.h>
+#include <map>
 #include <math.h>
 #include <set>
 #include <unordered_set>
@@ -52,7 +53,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 //FIXME: workaround for truncated touch sounds
 // to be removed when the problem is handled by system UI
@@ -1132,7 +1133,7 @@
                                               audio_io_handle_t *output,
                                               audio_session_t session,
                                               audio_stream_type_t *stream,
-                                              const Identity& identity,
+                                              const AttributionSourceState& attributionSource,
                                               const audio_config_t *config,
                                               audio_output_flags_t *flags,
                                               audio_port_handle_t *selectedDeviceId,
@@ -1145,7 +1146,7 @@
         return INVALID_OPERATION;
     }
     const uid_t uid = VALUE_OR_RETURN_STATUS(
-        aidl2legacy_int32_t_uid_t(identity.uid));
+        aidl2legacy_int32_t_uid_t(attributionSource.uid));
     const audio_port_handle_t requestedPortId = *selectedDeviceId;
     audio_attributes_t resultAttr;
     bool isRequestedDeviceForExclusiveUse = false;
@@ -2114,7 +2115,7 @@
                                              audio_io_handle_t *input,
                                              audio_unique_id_t riid,
                                              audio_session_t session,
-                                             const Identity& identity,
+                                             const AttributionSourceState& attributionSource,
                                              const audio_config_base_t *config,
                                              audio_input_flags_t flags,
                                              audio_port_handle_t *selectedDeviceId,
@@ -2133,7 +2134,7 @@
     sp<AudioInputDescriptor> inputDesc;
     sp<RecordClientDescriptor> clientDesc;
     audio_port_handle_t requestedDeviceId = *selectedDeviceId;
-    uid_t uid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(identity.uid));
+    uid_t uid = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_uid_t(attributionSource.uid));
     bool isSoundTrigger;
 
     // The supplied portId must be AUDIO_PORT_HANDLE_NONE
@@ -5694,6 +5695,7 @@
 
 void AudioPolicyManager::checkSecondaryOutputs() {
     std::set<audio_stream_type_t> streamsToInvalidate;
+    TrackSecondaryOutputsMap trackSecondaryOutputs;
     for (size_t i = 0; i < mOutputs.size(); i++) {
         const sp<SwAudioOutputDescriptor>& outputDescriptor = mOutputs[i];
         for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
@@ -5710,16 +5712,28 @@
                 }
             }
 
-            if (status != OK ||
-                !std::equal(client->getSecondaryOutputs().begin(),
-                            client->getSecondaryOutputs().end(),
-                            secondaryDescs.begin(), secondaryDescs.end())) {
+            if (status != OK) {
                 streamsToInvalidate.insert(client->stream());
+            } else if (!std::equal(
+                    client->getSecondaryOutputs().begin(),
+                    client->getSecondaryOutputs().end(),
+                    secondaryDescs.begin(), secondaryDescs.end())) {
+                std::vector<wp<SwAudioOutputDescriptor>> weakSecondaryDescs;
+                std::vector<audio_io_handle_t> secondaryOutputIds;
+                for (const auto& secondaryDesc : secondaryDescs) {
+                    secondaryOutputIds.push_back(secondaryDesc->mIoHandle);
+                    weakSecondaryDescs.push_back(secondaryDesc);
+                }
+                trackSecondaryOutputs.emplace(client->portId(), secondaryOutputIds);
+                client->setSecondaryOutputs(std::move(weakSecondaryDescs));
             }
         }
     }
+    if (!trackSecondaryOutputs.empty()) {
+        mpClientInterface->updateSecondaryOutputs(trackSecondaryOutputs);
+    }
     for (audio_stream_type_t stream : streamsToInvalidate) {
-        ALOGD("%s Invalidate stream %d due to secondary output change", __func__, stream);
+        ALOGD("%s Invalidate stream %d due to fail getting output for attr", __func__, stream);
         mpClientInterface->invalidateStream(stream);
     }
 }
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index f5dd20c..98f96d1 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -52,6 +52,8 @@
 
 namespace android {
 
+using content::AttributionSourceState;
+
 // ----------------------------------------------------------------------------
 
 // Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
@@ -116,7 +118,7 @@
                                   audio_io_handle_t *output,
                                   audio_session_t session,
                                   audio_stream_type_t *stream,
-                                  const media::permission::Identity& identity,
+                                  const AttributionSourceState& attributionSource,
                                   const audio_config_t *config,
                                   audio_output_flags_t *flags,
                                   audio_port_handle_t *selectedDeviceId,
@@ -130,7 +132,7 @@
                                          audio_io_handle_t *input,
                                          audio_unique_id_t riid,
                                          audio_session_t session,
-                                         const media::permission::Identity& identity,
+                                         const AttributionSourceState& attributionSource,
                                          const audio_config_base_t *config,
                                          audio_input_flags_t flags,
                                          audio_port_handle_t *selectedDeviceId,
diff --git a/services/audiopolicy/service/Android.bp b/services/audiopolicy/service/Android.bp
index 14be671..0273d29 100644
--- a/services/audiopolicy/service/Android.bp
+++ b/services/audiopolicy/service/Android.bp
@@ -45,12 +45,12 @@
         "audiopolicy-aidl-cpp",
         "audiopolicy-types-aidl-cpp",
         "capture_state_listener-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     static_libs: [
         "libaudiopolicycomponents",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     header_libs: [
@@ -70,6 +70,6 @@
     export_shared_lib_headers: [
         "libactivitymanager_aidl",
         "libsensorprivacy",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 }
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index 77b5200..cd53073 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -291,4 +291,14 @@
     return af->getAudioPort(port);
 }
 
+status_t AudioPolicyService::AudioPolicyClient::updateSecondaryOutputs(
+        const TrackSecondaryOutputsMap& trackSecondaryOutputs) {
+    sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
+    if (af == nullptr) {
+        ALOGW("%s: could not get AudioFlinger", __func__);
+        return PERMISSION_DENIED;
+    }
+    return af->updateSecondaryOutputs(trackSecondaryOutputs);
+}
+
 } // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index 8426a77..3f01de9 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -35,7 +35,7 @@
 
 namespace android {
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 // ----------------------------------------------------------------------------
 // AudioPolicyEffects Implementation
@@ -123,9 +123,10 @@
         Vector <EffectDesc *> effects = mInputSources.valueAt(index)->mEffects;
         for (size_t i = 0; i < effects.size(); i++) {
             EffectDesc *effect = effects[i];
-            Identity identity;
-            identity.packageName = "android";
-            sp<AudioEffect> fx = new AudioEffect(identity);
+            AttributionSourceState attributionSource;
+            attributionSource.packageName = "android";
+            attributionSource.token = sp<BBinder>::make();
+            sp<AudioEffect> fx = new AudioEffect(attributionSource);
             fx->set(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
             status_t status = fx->initCheck();
             if (status != NO_ERROR && status != ALREADY_EXISTS) {
@@ -274,9 +275,10 @@
         Vector <EffectDesc *> effects = mOutputStreams.valueAt(index)->mEffects;
         for (size_t i = 0; i < effects.size(); i++) {
             EffectDesc *effect = effects[i];
-            Identity identity;
-            identity.packageName = "android";
-            sp<AudioEffect> fx = new AudioEffect(identity);
+            AttributionSourceState attributionSource;
+            attributionSource.packageName = "android";
+            attributionSource.token = sp<BBinder>::make();
+            sp<AudioEffect> fx = new AudioEffect(attributionSource);
             fx->set(NULL, &effect->mUuid, 0, 0, 0, audioSession, output);
             status_t status = fx->initCheck();
             if (status != NO_ERROR && status != ALREADY_EXISTS) {
@@ -976,9 +978,10 @@
     for (const auto& deviceEffectsIter : mDeviceEffects) {
         const auto& deviceEffects =  deviceEffectsIter.second;
         for (const auto& effectDesc : deviceEffects->mEffectDescriptors->mEffects) {
-            Identity identity;
-            identity.packageName = "android";
-            sp<AudioEffect> fx = new AudioEffect(identity);
+            AttributionSourceState attributionSource;
+            attributionSource.packageName = "android";
+            attributionSource.token = sp<BBinder>::make();
+            sp<AudioEffect> fx = new AudioEffect(attributionSource);
             fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0, nullptr,
                     nullptr, AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
                     AudioDeviceTypeAddr{deviceEffects->getDeviceType(),
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index b9c715e..3298f6b 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -25,7 +25,7 @@
 #include <media/MediaMetricsItem.h>
 #include <media/PolicyAidlConversion.h>
 #include <utils/Log.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #define VALUE_OR_RETURN_BINDER_STATUS(x) \
     ({ auto _tmp = (x); \
@@ -43,7 +43,7 @@
 namespace android {
 using binder::Status;
 using aidl_utils::binderStatusFromStatusT;
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 const std::vector<audio_usage_t>& SYSTEM_USAGES = {
     AUDIO_USAGE_CALL_ASSISTANT,
@@ -64,15 +64,16 @@
 }
 
 status_t AudioPolicyService::validateUsage(audio_usage_t usage) {
-     return validateUsage(usage, getCallingIdentity());
+     return validateUsage(usage, getCallingAttributionSource());
 }
 
-status_t AudioPolicyService::validateUsage(audio_usage_t usage, const Identity& identity) {
+status_t AudioPolicyService::validateUsage(audio_usage_t usage,
+        const AttributionSourceState& attributionSource) {
     if (isSystemUsage(usage)) {
         if (isSupportedSystemUsage(usage)) {
-            if (!modifyAudioRoutingAllowed(identity)) {
+            if (!modifyAudioRoutingAllowed(attributionSource)) {
                 ALOGE(("permission denied: modify audio routing not allowed "
-                       "for identity %s"), identity.toString().c_str());
+                       "for attributionSource %s"), attributionSource.toString().c_str());
                 return PERMISSION_DENIED;
             }
         } else {
@@ -279,7 +280,7 @@
 
 Status AudioPolicyService::getOutputForAttr(const media::AudioAttributesInternal& attrAidl,
                                             int32_t sessionAidl,
-                                            const Identity& identity,
+                                            const AttributionSourceState& attributionSource,
                                             const media::AudioConfig& configAidl,
                                             int32_t flagsAidl,
                                             int32_t selectedDeviceIdAidl,
@@ -307,28 +308,28 @@
 
     RETURN_IF_BINDER_ERROR(
             binderStatusFromStatusT(AudioValidator::validateAudioAttributes(attr, "68953950")));
-    RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(validateUsage(attr.usage, identity)));
+    RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(validateUsage(attr.usage, attributionSource)));
 
     ALOGV("%s()", __func__);
     Mutex::Autolock _l(mLock);
 
     // TODO b/182392553: refactor or remove
-    Identity adjIdentity = identity;
+    AttributionSourceState adjAttributionSource = attributionSource;
     const uid_t callingUid = IPCThreadState::self()->getCallingUid();
-    if (!isAudioServerOrMediaServerUid(callingUid) || identity.uid == -1) {
+    if (!isAudioServerOrMediaServerUid(callingUid) || attributionSource.uid == -1) {
         int32_t callingUidAidl = VALUE_OR_RETURN_BINDER_STATUS(
             legacy2aidl_uid_t_int32_t(callingUid));
-        ALOGW_IF(identity.uid != -1 && identity.uid != callingUidAidl,
+        ALOGW_IF(attributionSource.uid != -1 && attributionSource.uid != callingUidAidl,
                 "%s uid %d tried to pass itself off as %d", __func__,
-                callingUidAidl, identity.uid);
-        adjIdentity.uid = callingUidAidl;
+                callingUidAidl, attributionSource.uid);
+        adjAttributionSource.uid = callingUidAidl;
     }
     if (!mPackageManager.allowPlaybackCapture(VALUE_OR_RETURN_BINDER_STATUS(
-        aidl2legacy_int32_t_uid_t(adjIdentity.uid)))) {
+        aidl2legacy_int32_t_uid_t(adjAttributionSource.uid)))) {
         attr.flags = static_cast<audio_flags_mask_t>(attr.flags | AUDIO_FLAG_NO_MEDIA_PROJECTION);
     }
     if (((attr.flags & (AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE)) != 0)
-            && !bypassInterruptionPolicyAllowed(identity)) {
+            && !bypassInterruptionPolicyAllowed(adjAttributionSource)) {
         attr.flags = static_cast<audio_flags_mask_t>(
                 attr.flags & ~(AUDIO_FLAG_BYPASS_INTERRUPTION_POLICY|AUDIO_FLAG_BYPASS_MUTE));
     }
@@ -336,7 +337,7 @@
     AudioPolicyInterface::output_type_t outputType;
     status_t result = mAudioPolicyManager->getOutputForAttr(&attr, &output, session,
                                                             &stream,
-                                                            adjIdentity,
+                                                            adjAttributionSource,
                                                             &config,
                                                             &flags, &selectedDeviceId, &portId,
                                                             &secondaryOutputs,
@@ -349,16 +350,16 @@
         case AudioPolicyInterface::API_OUTPUT_LEGACY:
             break;
         case AudioPolicyInterface::API_OUTPUT_TELEPHONY_TX:
-            if (!modifyPhoneStateAllowed(adjIdentity)) {
+            if (!modifyPhoneStateAllowed(adjAttributionSource)) {
                 ALOGE("%s() permission denied: modify phone state not allowed for uid %d",
-                    __func__, adjIdentity.uid);
+                    __func__, adjAttributionSource.uid);
                 result = PERMISSION_DENIED;
             }
             break;
         case AudioPolicyInterface::API_OUT_MIX_PLAYBACK:
-            if (!modifyAudioRoutingAllowed(adjIdentity)) {
+            if (!modifyAudioRoutingAllowed(adjAttributionSource)) {
                 ALOGE("%s() permission denied: modify audio routing not allowed for uid %d",
-                    __func__, adjIdentity.uid);
+                    __func__, adjAttributionSource.uid);
                 result = PERMISSION_DENIED;
             }
             break;
@@ -371,7 +372,7 @@
 
     if (result == NO_ERROR) {
         sp<AudioPlaybackClient> client =
-                new AudioPlaybackClient(attr, output, adjIdentity, session,
+                new AudioPlaybackClient(attr, output, adjAttributionSource, session,
                     portId, selectedDeviceId, stream);
         mAudioPlaybackClients.add(portId, client);
 
@@ -508,7 +509,7 @@
                                            int32_t inputAidl,
                                            int32_t riidAidl,
                                            int32_t sessionAidl,
-                                           const Identity& identity,
+                                           const AttributionSourceState& attributionSource,
                                            const media::AudioConfigBase& configAidl,
                                            int32_t flagsAidl,
                                            int32_t selectedDeviceIdAidl,
@@ -551,42 +552,46 @@
         return binderStatusFromStatusT(BAD_VALUE);
     }
 
-    // Make sure identity represents the current caller
-    Identity adjIdentity = identity;
+    // Make sure attribution source represents the current caller
+    AttributionSourceState adjAttributionSource = attributionSource;
     // TODO b/182392553: refactor or remove
-    bool updatePid = (identity.pid == -1);
+    bool updatePid = (attributionSource.pid == -1);
     const uid_t callingUid =IPCThreadState::self()->getCallingUid();
-    const uid_t currentUid = VALUE_OR_RETURN_BINDER_STATUS(aidl2legacy_int32_t_uid_t(identity.uid));
+    const uid_t currentUid = VALUE_OR_RETURN_BINDER_STATUS(aidl2legacy_int32_t_uid_t(
+            attributionSource.uid));
     if (!isAudioServerOrMediaServerUid(callingUid)) {
         ALOGW_IF(currentUid != (uid_t)-1 && currentUid != callingUid,
                 "%s uid %d tried to pass itself off as %d", __FUNCTION__, callingUid,
                 currentUid);
-        adjIdentity.uid = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_uid_t_int32_t(callingUid));
+        adjAttributionSource.uid = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_uid_t_int32_t(
+                callingUid));
         updatePid = true;
     }
 
     if (updatePid) {
         const int32_t callingPid = VALUE_OR_RETURN_BINDER_STATUS(legacy2aidl_pid_t_int32_t(
             IPCThreadState::self()->getCallingPid()));
-        ALOGW_IF(identity.pid != -1 && identity.pid != callingPid,
+        ALOGW_IF(attributionSource.pid != -1 && attributionSource.pid != callingPid,
                  "%s uid %d pid %d tried to pass itself off as pid %d",
-                 __func__, adjIdentity.uid, callingPid, identity.pid);
-        adjIdentity.pid = callingPid;
+                 __func__, adjAttributionSource.uid, callingPid, attributionSource.pid);
+        adjAttributionSource.pid = callingPid;
     }
 
-    RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(validateUsage(attr.usage, adjIdentity)));
+    RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(validateUsage(attr.usage,
+            adjAttributionSource)));
 
     // check calling permissions.
     // Capturing from FM_TUNER source is controlled by captureTunerAudioInputAllowed() and
     // captureAudioOutputAllowed() (deprecated) as this does not affect users privacy
     // as does capturing from an actual microphone.
-    if (!(recordingAllowed(adjIdentity, attr.source) || attr.source == AUDIO_SOURCE_FM_TUNER)) {
+    if (!(recordingAllowed(adjAttributionSource, attr.source)
+            || attr.source == AUDIO_SOURCE_FM_TUNER)) {
         ALOGE("%s permission denied: recording not allowed for %s",
-                __func__, adjIdentity.toString().c_str());
+                __func__, adjAttributionSource.toString().c_str());
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
 
-    bool canCaptureOutput = captureAudioOutputAllowed(adjIdentity);
+    bool canCaptureOutput = captureAudioOutputAllowed(adjAttributionSource);
     if ((inputSource == AUDIO_SOURCE_VOICE_UPLINK ||
         inputSource == AUDIO_SOURCE_VOICE_DOWNLINK ||
         inputSource == AUDIO_SOURCE_VOICE_CALL ||
@@ -596,12 +601,12 @@
     }
 
     if (inputSource == AUDIO_SOURCE_FM_TUNER
-        && !captureTunerAudioInputAllowed(adjIdentity)
+        && !captureTunerAudioInputAllowed(adjAttributionSource)
         && !canCaptureOutput) {
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
 
-    bool canCaptureHotword = captureHotwordAllowed(adjIdentity);
+    bool canCaptureHotword = captureHotwordAllowed(adjAttributionSource);
     if ((inputSource == AUDIO_SOURCE_HOTWORD) && !canCaptureHotword) {
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
@@ -609,7 +614,7 @@
     if (((flags & AUDIO_INPUT_FLAG_HW_HOTWORD) != 0)
             && !canCaptureHotword) {
         ALOGE("%s: permission denied: hotword mode not allowed"
-              " for uid %d pid %d", __func__, adjIdentity.uid, adjIdentity.pid);
+              " for uid %d pid %d", __func__, adjAttributionSource.uid, adjAttributionSource.pid);
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
 
@@ -623,7 +628,7 @@
             AutoCallerClear acc;
             // the audio_in_acoustics_t parameter is ignored by get_input()
             status = mAudioPolicyManager->getInputForAttr(&attr, &input, riid, session,
-                                                          adjIdentity, &config,
+                                                          adjAttributionSource, &config,
                                                           flags, &selectedDeviceId,
                                                           &inputType, &portId);
 
@@ -647,7 +652,7 @@
                 }
                 break;
             case AudioPolicyInterface::API_INPUT_MIX_EXT_POLICY_REROUTE:
-                if (!modifyAudioRoutingAllowed(adjIdentity)) {
+                if (!modifyAudioRoutingAllowed(adjAttributionSource)) {
                     ALOGE("getInputForAttr() permission denied: modify audio routing not allowed");
                     status = PERMISSION_DENIED;
                 }
@@ -668,7 +673,7 @@
         }
 
         sp<AudioRecordClient> client = new AudioRecordClient(attr, input, session, portId,
-                                                             selectedDeviceId, adjIdentity,
+                                                             selectedDeviceId, adjAttributionSource,
                                                              canCaptureOutput, canCaptureHotword);
         mAudioRecordClients.add(portId, client);
     }
@@ -723,11 +728,11 @@
     msg << "Audio recording on session " << client->session;
 
     // check calling permissions
-    if (!(startRecording(client->identity, String16(msg.str().c_str()),
+    if (!(startRecording(client->attributionSource, String16(msg.str().c_str()),
         client->attributes.source)
             || client->attributes.source == AUDIO_SOURCE_FM_TUNER)) {
-        ALOGE("%s permission denied: recording not allowed for identity %s",
-                __func__, client->identity.toString().c_str());
+        ALOGE("%s permission denied: recording not allowed for attribution source %s",
+                __func__, client->attributionSource.toString().c_str());
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
 
@@ -771,13 +776,13 @@
             item->setCString(kAudioPolicyRqstSrc,
                              toString(client->attributes.source).c_str());
             item->setInt32(kAudioPolicyRqstSession, client->session);
-            if (client->identity.packageName.has_value() &&
-                client->identity.packageName.value().size() != 0) {
+            if (client->attributionSource.packageName.has_value() &&
+                client->attributionSource.packageName.value().size() != 0) {
                 item->setCString(kAudioPolicyRqstPkg,
-                    client->identity.packageName.value().c_str());
+                    client->attributionSource.packageName.value().c_str());
             } else {
                 item->setCString(kAudioPolicyRqstPkg,
-                    std::to_string(client->identity.uid).c_str());
+                    std::to_string(client->attributionSource.uid).c_str());
             }
             item->setCString(
                     kAudioPolicyRqstDevice, getDeviceTypeStrForPortId(client->deviceId).c_str());
@@ -793,13 +798,13 @@
                     item->setCString(kAudioPolicyActiveSrc,
                                      toString(other->attributes.source).c_str());
                     item->setInt32(kAudioPolicyActiveSession, other->session);
-                    if (other->identity.packageName.has_value() &&
-                        other->identity.packageName.value().size() != 0) {
+                    if (other->attributionSource.packageName.has_value() &&
+                        other->attributionSource.packageName.value().size() != 0) {
                         item->setCString(kAudioPolicyActivePkg,
-                            other->identity.packageName.value().c_str());
+                            other->attributionSource.packageName.value().c_str());
                     } else {
                         item->setCString(kAudioPolicyRqstPkg, std::to_string(
-                            other->identity.uid).c_str());
+                            other->attributionSource.uid).c_str());
                     }
                     item->setCString(kAudioPolicyActiveDevice,
                                      getDeviceTypeStrForPortId(other->deviceId).c_str());
@@ -815,7 +820,7 @@
         client->active = false;
         client->startTimeNs = 0;
         updateUidStates_l();
-        finishRecording(client->identity, client->attributes.source);
+        finishRecording(client->attributionSource, client->attributes.source);
     }
 
     return binderStatusFromStatusT(status);
@@ -844,7 +849,7 @@
     updateUidStates_l();
 
     // finish the recording app op
-    finishRecording(client->identity, client->attributes.source);
+    finishRecording(client->attributionSource, client->attributes.source);
     AutoCallerClear acc;
     return binderStatusFromStatusT(mAudioPolicyManager->stopInput(portId));
 }
@@ -1641,15 +1646,15 @@
     bool needCaptureMediaOutput = std::any_of(mixes.begin(), mixes.end(), [](auto& mix) {
             return mix.mAllowPrivilegedMediaPlaybackCapture; });
 
-    const Identity identity = getCallingIdentity();
+    const AttributionSourceState attributionSource = getCallingAttributionSource();
 
 
-    if (needCaptureMediaOutput && !captureMediaOutputAllowed(identity)) {
+    if (needCaptureMediaOutput && !captureMediaOutputAllowed(attributionSource)) {
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
 
     if (needCaptureVoiceCommunicationOutput &&
-        !captureVoiceCommunicationOutputAllowed(identity)) {
+        !captureVoiceCommunicationOutputAllowed(attributionSource)) {
         return binderStatusFromStatusT(PERMISSION_DENIED);
     }
 
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index fb38e3d..201273e 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -594,7 +594,8 @@
 
     for (size_t i =0; i < mAudioRecordClients.size(); i++) {
         sp<AudioRecordClient> current = mAudioRecordClients[i];
-        uid_t currentUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(current->identity.uid));
+        uid_t currentUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(
+                current->attributionSource.uid));
         if (!current->active) {
             continue;
         }
@@ -641,7 +642,7 @@
                             || (isInCommunication && currentUid == mPhoneStateOwnerUid)) {
                         if (!isInCommunication || latestSensitiveActiveOrComm == nullptr
                                 || VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(
-                                    latestSensitiveActiveOrComm->identity.uid))
+                                    latestSensitiveActiveOrComm->attributionSource.uid))
                                         != mPhoneStateOwnerUid) {
                             latestSensitiveActiveOrComm = current;
                             latestSensitiveStartNs = current->startTimeNs;
@@ -676,7 +677,7 @@
         // if audio mode is IN_COMMUNICATION, favor audio mode owner over an app with
         // foreground UI in case both are capturing with privacy sensitive flag.
         uid_t latestActiveUid = VALUE_OR_FATAL(
-            aidl2legacy_int32_t_uid_t(latestSensitiveActiveOrComm->identity.uid));
+            aidl2legacy_int32_t_uid_t(latestSensitiveActiveOrComm->attributionSource.uid));
         if (isInCommunication && latestActiveUid == mPhoneStateOwnerUid) {
             topSensitiveActive = latestSensitiveActiveOrComm;
             topSensitiveStartNs = latestSensitiveStartNs;
@@ -696,20 +697,20 @@
     for (size_t i =0; i < mAudioRecordClients.size(); i++) {
         sp<AudioRecordClient> current = mAudioRecordClients[i];
         uid_t currentUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(
-            current->identity.uid));
+            current->attributionSource.uid));
         if (!current->active) {
             continue;
         }
 
         audio_source_t source = current->attributes.source;
         bool isTopOrLatestActive = topActive == nullptr ? false :
-            current->identity.uid == topActive->identity.uid;
+            current->attributionSource.uid == topActive->attributionSource.uid;
         bool isTopOrLatestSensitive = topSensitiveActive == nullptr ? false :
-            current->identity.uid == topSensitiveActive->identity.uid;
+            current->attributionSource.uid == topSensitiveActive->attributionSource.uid;
 
         auto canCaptureIfInCallOrCommunication = [&](const auto &recordClient) REQUIRES(mLock) {
             uid_t recordUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(
-                recordClient->identity.uid));
+                recordClient->attributionSource.uid));
             bool canCaptureCall = recordClient->canCaptureOutput;
             bool canCaptureCommunication = recordClient->canCaptureOutput
                 || !isPhoneStateOwnerActive
@@ -785,7 +786,7 @@
                 allowCapture = true;
             }
         }
-        setAppState_l(current->portId,
+        setAppState_l(current,
                       allowCapture ? apmStatFromAmState(mUidPolicy->getUidState(currentUid)) :
                                 APP_STATE_IDLE);
     }
@@ -795,7 +796,7 @@
     for (size_t i = 0; i < mAudioRecordClients.size(); i++) {
         sp<AudioRecordClient> current = mAudioRecordClients[i];
         if (!isVirtualSource(current->attributes.source)) {
-            setAppState_l(current->portId, APP_STATE_IDLE);
+            setAppState_l(current, APP_STATE_IDLE);
         }
     }
 }
@@ -829,17 +830,32 @@
     return false;
 }
 
-void AudioPolicyService::setAppState_l(audio_port_handle_t portId, app_state_t state)
+void AudioPolicyService::setAppState_l(sp<AudioRecordClient> client, app_state_t state)
 {
     AutoCallerClear acc;
 
     if (mAudioPolicyManager) {
-        mAudioPolicyManager->setAppState(portId, state);
+        mAudioPolicyManager->setAppState(client->portId, state);
     }
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af) {
         bool silenced = state == APP_STATE_IDLE;
-        af->setRecordSilenced(portId, silenced);
+        if (client->silenced != silenced) {
+            if (client->active) {
+                if (silenced) {
+                    finishRecording(client->attributionSource, client->attributes.source);
+                } else {
+                    std::stringstream msg;
+                    msg << "Audio recording un-silenced on session " << client->session;
+                    if (!startRecording(client->attributionSource, String16(msg.str().c_str()),
+                            client->attributes.source)) {
+                        silenced = true;
+                    }
+                }
+            }
+            af->setRecordSilenced(client->portId, silenced);
+            client->silenced = silenced;
+        }
     }
 }
 
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 00d9670..ac9c20f 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -1,3 +1,4 @@
+
 /*
  * Copyright (C) 2009 The Android Open Source Project
  *
@@ -38,12 +39,14 @@
 #include "CaptureStateNotifier.h"
 #include <AudioPolicyInterface.h>
 #include <android/hardware/BnSensorPrivacyListener.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #include <unordered_map>
 
 namespace android {
 
+using content::AttributionSourceState;
+
 // ----------------------------------------------------------------------------
 
 class AudioPolicyService :
@@ -81,7 +84,7 @@
                                media::AudioPolicyForcedConfig* _aidl_return) override;
     binder::Status getOutput(media::AudioStreamType stream, int32_t* _aidl_return) override;
     binder::Status getOutputForAttr(const media::AudioAttributesInternal& attr, int32_t session,
-                                    const media::permission::Identity &identity,
+                                    const AttributionSourceState &attributionSource,
                                     const media::AudioConfig& config,
                                     int32_t flags, int32_t selectedDeviceId,
                                     media::GetOutputForAttrResponse* _aidl_return) override;
@@ -90,7 +93,7 @@
     binder::Status releaseOutput(int32_t portId) override;
     binder::Status getInputForAttr(const media::AudioAttributesInternal& attr, int32_t input,
                                    int32_t riid, int32_t session,
-                                   const media::permission::Identity &identity,
+                                   const AttributionSourceState &attributionSource,
                                    const media::AudioConfigBase& config, int32_t flags,
                                    int32_t selectedDeviceId,
                                    media::GetInputForAttrResponse* _aidl_return) override;
@@ -321,8 +324,10 @@
     // Handles binder shell commands
     virtual status_t shellCommand(int in, int out, int err, Vector<String16>& args);
 
+    class AudioRecordClient;
+
     // Sets whether the given UID records only silence
-    virtual void setAppState_l(audio_port_handle_t portId, app_state_t state) REQUIRES(mLock);
+    virtual void setAppState_l(sp<AudioRecordClient> client, app_state_t state) REQUIRES(mLock);
 
     // Overrides the UID state as if it is idle
     status_t handleSetUidState(Vector<String16>& args, int err);
@@ -344,7 +349,7 @@
 
     bool isSupportedSystemUsage(audio_usage_t usage);
     status_t validateUsage(audio_usage_t usage);
-    status_t validateUsage(audio_usage_t usage, const media::permission::Identity& identity);
+    status_t validateUsage(audio_usage_t usage, const AttributionSourceState& attributionSource);
 
     void updateUidStates();
     void updateUidStates_l() REQUIRES(mLock);
@@ -735,6 +740,9 @@
 
         status_t getAudioPort(struct audio_port_v7 *port) override;
 
+        status_t updateSecondaryOutputs(
+                const TrackSecondaryOutputsMap& trackSecondaryOutputs) override;
+
      private:
         AudioPolicyService *mAudioPolicyService;
     };
@@ -788,17 +796,18 @@
     public:
                 AudioClient(const audio_attributes_t attributes,
                             const audio_io_handle_t io,
-                            const media::permission::Identity& identity,
+                            const AttributionSourceState& attributionSource,
                             const audio_session_t session,  audio_port_handle_t portId,
                             const audio_port_handle_t deviceId) :
-                                attributes(attributes), io(io), identity(identity),
-                                session(session), portId(portId), deviceId(deviceId), active(false) {}
+                                attributes(attributes), io(io), attributionSource(
+                                attributionSource), session(session), portId(portId),
+                                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 media::permission::Identity& identity; //client identity
+        const AttributionSourceState& attributionSource; //client attributionsource
         const audio_session_t session;       // audio session ID
         const audio_port_handle_t portId;
         const audio_port_handle_t deviceId;  // selected input device port ID
@@ -814,17 +823,19 @@
                           const audio_io_handle_t io,
                           const audio_session_t session, audio_port_handle_t portId,
                           const audio_port_handle_t deviceId,
-                          const media::permission::Identity& identity,
+                          const AttributionSourceState& attributionSource,
                           bool canCaptureOutput, bool canCaptureHotword) :
-                    AudioClient(attributes, io, identity,
-                        session, portId, deviceId), identity(identity), startTimeNs(0),
-                    canCaptureOutput(canCaptureOutput), canCaptureHotword(canCaptureHotword) {}
+                    AudioClient(attributes, io, attributionSource,
+                        session, portId, deviceId), attributionSource(attributionSource),
+                        startTimeNs(0), canCaptureOutput(canCaptureOutput),
+                        canCaptureHotword(canCaptureHotword), silenced(false) {}
                 ~AudioRecordClient() override = default;
 
-        const media::permission::Identity identity;        // identity of client
+        const AttributionSourceState attributionSource; // attribution source of client
         nsecs_t startTimeNs;
         const bool canCaptureOutput;
         const bool canCaptureHotword;
+        bool silenced;
     };
 
     // --- AudioPlaybackClient ---
@@ -833,10 +844,10 @@
     class AudioPlaybackClient : public AudioClient {
     public:
                 AudioPlaybackClient(const audio_attributes_t attributes,
-                      const audio_io_handle_t io, media::permission::Identity identity,
+                      const audio_io_handle_t io, AttributionSourceState attributionSource,
                             const audio_session_t session, audio_port_handle_t portId,
                             audio_port_handle_t deviceId, audio_stream_type_t stream) :
-                    AudioClient(attributes, io, identity, session, portId,
+                    AudioClient(attributes, io, attributionSource, session, portId,
                         deviceId), stream(stream) {}
                 ~AudioPlaybackClient() override = default;
 
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
index f480210..b296fb0 100644
--- a/services/audiopolicy/tests/Android.bp
+++ b/services/audiopolicy/tests/Android.bp
@@ -25,7 +25,8 @@
         "libmedia_helper",
         "libutils",
         "libxml2",
-        "media_permission-aidl-cpp",
+        "libpermission",
+        "libbinder",
     ],
 
     static_libs: [
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index e2d7d17..f7b0565 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -134,6 +134,11 @@
     size_t getRoutingUpdatedCounter() const {
         return mRoutingUpdatedUpdateCount; }
 
+    status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs __unused) override {
+        return NO_ERROR;
+    }
+
 private:
     audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
     audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index d289e15..1384864 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -91,6 +91,10 @@
     status_t getAudioPort(struct audio_port_v7 *port __unused) override {
         return INVALID_OPERATION;
     };
+    status_t updateSecondaryOutputs(
+            const TrackSecondaryOutputsMap& trackSecondaryOutputs __unused) override {
+        return NO_INIT;
+    }
 };
 
 } // namespace android
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index 8f12ecf..a16ab7d 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -25,7 +25,7 @@
 #define LOG_TAG "APM_Test"
 #include <Serializer.h>
 #include <android-base/file.h>
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <media/AudioPolicy.h>
 #include <media/PatchBuilder.h>
 #include <media/RecordingActivityTracker.h>
@@ -40,7 +40,7 @@
 
 using namespace android;
 using testing::UnorderedElementsAre;
-using media::permission::Identity;
+using android::content::AttributionSourceState;
 
 TEST(AudioPolicyManagerTestInit, EngineFailure) {
     AudioPolicyTestClient client;
@@ -216,11 +216,12 @@
     if (!portId) portId = &localPortId;
     *portId = AUDIO_PORT_HANDLE_NONE;
     AudioPolicyInterface::output_type_t outputType;
-    // TODO b/182392769: use identity util
-    Identity i = Identity();
-    i.uid = 0;
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource = AttributionSourceState();
+    attributionSource.uid = 0;
+    attributionSource.token = sp<BBinder>::make();
     ASSERT_EQ(OK, mManager->getOutputForAttr(
-                    &attr, output, AUDIO_SESSION_NONE, &stream, i, &config, &flags,
+                    &attr, output, AUDIO_SESSION_NONE, &stream, attributionSource, &config, &flags,
                     selectedDeviceId, portId, {}, &outputType));
     ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
     ASSERT_NE(AUDIO_IO_HANDLE_NONE, *output);
@@ -244,11 +245,12 @@
     if (!portId) portId = &localPortId;
     *portId = AUDIO_PORT_HANDLE_NONE;
     AudioPolicyInterface::input_type_t inputType;
-    // TODO b/182392769: use identity util
-    Identity i = Identity();
-    i.uid = 0;
+    // TODO b/182392769: use attribution source util
+    AttributionSourceState attributionSource = AttributionSourceState();
+    attributionSource.uid = 0;
+    attributionSource.token = sp<BBinder>::make();
     ASSERT_EQ(OK, mManager->getInputForAttr(
-            &attr, &input, riid, AUDIO_SESSION_NONE, i, &config, flags,
+            &attr, &input, riid, AUDIO_SESSION_NONE, attributionSource, &config, flags,
             selectedDeviceId, &inputType, portId));
     ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
 }
diff --git a/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp b/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp
index 6dcf440..5afdfb9 100644
--- a/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp
+++ b/services/camera/libcameraservice/utils/SessionConfigurationUtils.cpp
@@ -669,15 +669,22 @@
         const std::vector<int32_t> &sensorPixelModesUsed, int format, int width, int height,
         const CameraMetadata &staticInfo, bool flexibleConsumer,
         std::unordered_set<int32_t> *overriddenSensorPixelModesUsed) {
+
+    const std::unordered_set<int32_t> &sensorPixelModesUsedSet =
+            convertToSet(sensorPixelModesUsed);
     if (!isUltraHighResolutionSensor(staticInfo)) {
+        if (sensorPixelModesUsedSet.find(ANDROID_SENSOR_PIXEL_MODE_MAXIMUM_RESOLUTION) !=
+                sensorPixelModesUsedSet.end()) {
+            // invalid value for non ultra high res sensors
+            return BAD_VALUE;
+        }
         overriddenSensorPixelModesUsed->clear();
         overriddenSensorPixelModesUsed->insert(ANDROID_SENSOR_PIXEL_MODE_DEFAULT);
         return OK;
     }
 
     StreamConfigurationPair streamConfigurationPair = getStreamConfigurationPair(staticInfo);
-    const std::unordered_set<int32_t> &sensorPixelModesUsedSet =
-            convertToSet(sensorPixelModesUsed);
+
     bool isInDefaultStreamConfigurationMap =
             inStreamConfigurationMap(format, width, height,
                     streamConfigurationPair.mDefaultStreamConfigurationMap);
diff --git a/services/mediametrics/statsd_recorder.cpp b/services/mediametrics/statsd_recorder.cpp
index 6edad7c..1b312b5 100644
--- a/services/mediametrics/statsd_recorder.cpp
+++ b/services/mediametrics/statsd_recorder.cpp
@@ -57,8 +57,8 @@
 
     // string kRecorderLogSessionId = "android.media.mediarecorder.log-session-id";
     std::string log_session_id;
-    if (item->getString("android.media.mediarecorder.log_session_id", &log_session_id)) {
-        metrics_proto.set_log_session_id(std::move(log_session_id));
+    if (item->getString("android.media.mediarecorder.log-session-id", &log_session_id)) {
+        metrics_proto.set_log_session_id(log_session_id);
     }
     // string kRecorderAudioMime = "android.media.mediarecorder.audio.mime";
     std::string audio_mime;
@@ -214,8 +214,7 @@
             << " video_bitrate:" << video_bitrate
 
             << " iframe_interval:" << iframe_interval
-            // TODO Recorder - add log_session_id
-            // << " log_session_id:" << log_session_id
+            << " log_session_id:" << log_session_id
             << " }";
     statsdLog->log(android::util::MEDIAMETRICS_RECORDER_REPORTED, log.str());
     return true;
diff --git a/services/oboeservice/AAudioEndpointManager.cpp b/services/oboeservice/AAudioEndpointManager.cpp
index 3224cfc..20e4cc5 100644
--- a/services/oboeservice/AAudioEndpointManager.cpp
+++ b/services/oboeservice/AAudioEndpointManager.cpp
@@ -184,7 +184,7 @@
             // An app can avoid having this happen by closing their streams when
             // the app is paused.
             pid_t pid = VALUE_OR_FATAL(
-                aidl2legacy_int32_t_pid_t(request.getIdentity().pid));
+                aidl2legacy_int32_t_pid_t(request.getAttributionSource().pid));
             AAudioClientTracker::getInstance().setExclusiveEnabled(pid, false);
             endpointToSteal = endpoint; // return it to caller
         }
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index 0b69bf6..40a664e 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -22,7 +22,7 @@
 #include <iostream>
 #include <sstream>
 
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 #include <aaudio/AAudio.h>
 #include <media/AidlConversion.h>
 #include <mediautils/ServiceUtilities.h>
@@ -47,18 +47,18 @@
        std::move(_tmp.value()); })
 
 using android::AAudioService;
-using android::media::permission::Identity;
+using android::content::AttributionSourceState;
 using binder::Status;
 
 android::AAudioService::AAudioService()
     : BnAAudioService(),
       mAdapter(this) {
     // TODO consider using geteuid()
-    // TODO b/182392769: use identity util
-    mAudioClient.identity.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
-    mAudioClient.identity.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
-    mAudioClient.identity.packageName = std::nullopt;
-    mAudioClient.identity.attributionTag = std::nullopt;
+    // TODO b/182392769: use attribution source util
+    mAudioClient.attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    mAudioClient.attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    mAudioClient.attributionSource.packageName = std::nullopt;
+    mAudioClient.attributionSource.attributionTag = std::nullopt;
     AAudioClientTracker::getInstance().setAAudioService(this);
 }
 
@@ -115,13 +115,14 @@
     aaudio_sharing_mode_t sharingMode = configurationInput.getSharingMode();
 
     // Enforce limit on client processes.
-    Identity callingIdentity = request.getIdentity();
+    AttributionSourceState attributionSource = request.getAttributionSource();
     pid_t pid = IPCThreadState::self()->getCallingPid();
-    callingIdentity.pid = VALUE_OR_RETURN_ILLEGAL_ARG_STATUS(
+    attributionSource.pid = VALUE_OR_RETURN_ILLEGAL_ARG_STATUS(
         legacy2aidl_pid_t_int32_t(pid));
-    callingIdentity.uid = VALUE_OR_RETURN_ILLEGAL_ARG_STATUS(
+    attributionSource.uid = VALUE_OR_RETURN_ILLEGAL_ARG_STATUS(
         legacy2aidl_uid_t_int32_t(IPCThreadState::self()->getCallingUid()));
-    if (callingIdentity.pid != mAudioClient.identity.pid) {
+    attributionSource.token = sp<BBinder>::make();
+    if (attributionSource.pid != mAudioClient.attributionSource.pid) {
         int32_t count = AAudioClientTracker::getInstance().getStreamCount(pid);
         if (count >= MAX_STREAMS_PER_PROCESS) {
             ALOGE("openStream(): exceeded max streams per process %d >= %d",
@@ -280,8 +281,8 @@
 }
 
 bool AAudioService::isCallerInService() {
-    pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mAudioClient.identity.pid));
-    uid_t clientUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mAudioClient.identity.uid));
+    pid_t clientPid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mAudioClient.attributionSource.pid));
+    uid_t clientUid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mAudioClient.attributionSource.uid));
     return clientPid == IPCThreadState::self()->getCallingPid() &&
         clientUid == IPCThreadState::self()->getCallingUid();
 }
@@ -307,7 +308,7 @@
         const uid_t callingUserId = IPCThreadState::self()->getCallingUid();
         const uid_t ownerUserId = serviceStream->getOwnerUserId();
         const uid_t clientUid = VALUE_OR_FATAL(
-            aidl2legacy_int32_t_uid_t(mAudioClient.identity.uid));
+            aidl2legacy_int32_t_uid_t(mAudioClient.attributionSource.uid));
         bool callerOwnsIt = callingUserId == ownerUserId;
         bool serverCalling = callingUserId == clientUid;
         bool serverOwnsIt = ownerUserId == clientUid;
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index b4efd1a..117218a 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -71,11 +71,11 @@
 aaudio_result_t AAudioServiceEndpointMMAP::open(const aaudio::AAudioStreamRequest &request) {
     aaudio_result_t result = AAUDIO_OK;
     copyFrom(request.getConstantConfiguration());
-    mMmapClient.identity = request.getIdentity();
-    // TODO b/182392769: use identity util
-    mMmapClient.identity.uid = VALUE_OR_FATAL(
+    mMmapClient.attributionSource = request.getAttributionSource();
+    // TODO b/182392769: use attribution source util
+    mMmapClient.attributionSource.uid = VALUE_OR_FATAL(
         legacy2aidl_uid_t_int32_t(IPCThreadState::self()->getCallingUid()));
-    mMmapClient.identity.pid = VALUE_OR_FATAL(
+    mMmapClient.attributionSource.pid = VALUE_OR_FATAL(
         legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
 
     audio_format_t audioFormat = getFormat();
@@ -165,8 +165,8 @@
                                                           this, // callback
                                                           mMmapStream,
                                                           &mPortHandle);
-    ALOGD("%s() mMapClient.identity = %s => portHandle = %d\n",
-          __func__, mMmapClient.identity.toString().c_str(), mPortHandle);
+    ALOGD("%s() mMapClient.attributionSource = %s => portHandle = %d\n",
+          __func__, mMmapClient.attributionSource.toString().c_str(), mPortHandle);
     if (status != OK) {
         // This can happen if the resource is busy or the config does
         // not match the hardware.
@@ -216,7 +216,7 @@
         // Exclusive mode can only be used by the service because the FD cannot be shared.
         int32_t audioServiceUid =
             VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
-        if ((mMmapClient.identity.uid != audioServiceUid) &&
+        if ((mMmapClient.attributionSource.uid != audioServiceUid) &&
             getSharingMode() == AAUDIO_SHARING_MODE_EXCLUSIVE) {
             ALOGW("%s() - exclusive FD cannot be used by client", __func__);
             result = AAUDIO_ERROR_UNAVAILABLE;
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index dbacd75..34ddd4d 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -39,7 +39,7 @@
 using namespace android;  // TODO just import names needed
 using namespace aaudio;   // TODO just import names needed
 
-using media::permission::Identity;
+using content::AttributionSourceState;
 
 /**
  * Base class for streams in the service.
@@ -50,7 +50,7 @@
         : mTimestampThread("AATime")
         , mAtomicStreamTimestamp()
         , mAudioService(audioService) {
-    mMmapClient.identity = Identity();
+    mMmapClient.attributionSource = AttributionSourceState();
 }
 
 AAudioServiceStreamBase::~AAudioServiceStreamBase() {
@@ -81,7 +81,7 @@
 
     result << "    0x" << std::setfill('0') << std::setw(8) << std::hex << mHandle
            << std::dec << std::setfill(' ') ;
-    result << std::setw(6) << mMmapClient.identity.uid;
+    result << std::setw(6) << mMmapClient.attributionSource.uid;
     result << std::setw(7) << mClientHandle;
     result << std::setw(4) << (isRunning() ? "yes" : " no");
     result << std::setw(6) << getState();
@@ -127,11 +127,11 @@
     AAudioEndpointManager &mEndpointManager = AAudioEndpointManager::getInstance();
     aaudio_result_t result = AAUDIO_OK;
 
-    mMmapClient.identity = request.getIdentity();
-    // TODO b/182392769: use identity util
-    mMmapClient.identity.uid = VALUE_OR_FATAL(
+    mMmapClient.attributionSource = request.getAttributionSource();
+    // TODO b/182392769: use attribution source util
+    mMmapClient.attributionSource.uid = VALUE_OR_FATAL(
         legacy2aidl_uid_t_int32_t(IPCThreadState::self()->getCallingUid()));
-    mMmapClient.identity.pid = VALUE_OR_FATAL(
+    mMmapClient.attributionSource.pid = VALUE_OR_FATAL(
         legacy2aidl_pid_t_int32_t(IPCThreadState::self()->getCallingPid()));
 
     // Limit scope of lock to avoid recursive lock in close().
diff --git a/services/oboeservice/AAudioServiceStreamBase.h b/services/oboeservice/AAudioServiceStreamBase.h
index c42df0f..976996d 100644
--- a/services/oboeservice/AAudioServiceStreamBase.h
+++ b/services/oboeservice/AAudioServiceStreamBase.h
@@ -160,11 +160,13 @@
     }
 
     uid_t getOwnerUserId() const {
-        return VALUE_OR_FATAL(android::aidl2legacy_int32_t_uid_t(mMmapClient.identity.uid));
+        return VALUE_OR_FATAL(android::aidl2legacy_int32_t_uid_t(
+                mMmapClient.attributionSource.uid));
     }
 
     pid_t getOwnerProcessId() const {
-        return VALUE_OR_FATAL(android::aidl2legacy_int32_t_pid_t(mMmapClient.identity.pid));
+        return VALUE_OR_FATAL(android::aidl2legacy_int32_t_pid_t(
+                mMmapClient.attributionSource.pid));
     }
 
     aaudio_handle_t getHandle() const {
diff --git a/services/oboeservice/Android.bp b/services/oboeservice/Android.bp
index a419dd5..4c58040 100644
--- a/services/oboeservice/Android.bp
+++ b/services/oboeservice/Android.bp
@@ -66,13 +66,13 @@
         "liblog",
         "libutils",
         "aaudio-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
 
     export_shared_lib_headers: [
         "libaaudio_internal",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
     ],
 
     header_libs: [
diff --git a/services/oboeservice/fuzzer/Android.bp b/services/oboeservice/fuzzer/Android.bp
index f4e8a81..605ac01 100644
--- a/services/oboeservice/fuzzer/Android.bp
+++ b/services/oboeservice/fuzzer/Android.bp
@@ -46,7 +46,7 @@
         "liblog",
         "libutils",
         "aaudio-aidl-cpp",
-        "media_permission-aidl-cpp",
+        "framework-permission-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
     static_libs: [
diff --git a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
index 8e508d3..4bc661c 100644
--- a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
+++ b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
@@ -23,7 +23,7 @@
 #include <AAudioService.h>
 #include <aaudio/AAudio.h>
 #include "aaudio/BnAAudioClient.h"
-#include <android/media/permission/Identity.h>
+#include <android/content/AttributionSourceState.h>
 
 #define UNUSED_PARAM __attribute__((unused))
 
@@ -295,11 +295,12 @@
             ? fdp.ConsumeIntegral<int32_t>()
             : kAAudioFormats[fdp.ConsumeIntegralInRange<int32_t>(0, kNumAAudioFormats - 1)]));
 
-    // TODO b/182392769: use identity util
-    media::permission::Identity identity;
-    identity.uid = getuid();
-    identity.pid = getpid();
-    request.setIdentity(identity);
+    // TODO b/182392769: use attribution source util
+    android::content::AttributionSourceState attributionSource;
+    attributionSource.uid = getuid();
+    attributionSource.pid = getpid();
+    attributionSource.token = sp<BBinder>::make();
+    request.setAttributionSource(attributionSource);
     request.setInService(fdp.ConsumeBool());
 
     request.getConfiguration().setDeviceId(fdp.ConsumeIntegral<int32_t>());
diff --git a/services/tuner/TunerService.cpp b/services/tuner/TunerService.cpp
index 77e1c40..5b4129a 100644
--- a/services/tuner/TunerService.cpp
+++ b/services/tuner/TunerService.cpp
@@ -445,90 +445,118 @@
     TunerFrontendCapabilities caps;
     switch (halInfo.type) {
         case FrontendType::ANALOG: {
-            TunerFrontendAnalogCapabilities analogCaps{
-                .typeCap = (int)halInfo.frontendCaps.analogCaps().typeCap,
-                .sifStandardCap = (int)halInfo.frontendCaps.analogCaps().sifStandardCap,
-            };
-            caps.set<TunerFrontendCapabilities::analogCaps>(analogCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::analogCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendAnalogCapabilities analogCaps{
+                    .typeCap = (int)halInfo.frontendCaps.analogCaps().typeCap,
+                    .sifStandardCap = (int)halInfo.frontendCaps.analogCaps().sifStandardCap,
+                };
+                caps.set<TunerFrontendCapabilities::analogCaps>(analogCaps);
+            }
             break;
         }
         case FrontendType::ATSC: {
-            TunerFrontendAtscCapabilities atscCaps{
-                .modulationCap = (int)halInfo.frontendCaps.atscCaps().modulationCap,
-            };
-            caps.set<TunerFrontendCapabilities::atscCaps>(atscCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::atscCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendAtscCapabilities atscCaps{
+                    .modulationCap = (int)halInfo.frontendCaps.atscCaps().modulationCap,
+                };
+                caps.set<TunerFrontendCapabilities::atscCaps>(atscCaps);
+            }
             break;
         }
         case FrontendType::ATSC3: {
-            TunerFrontendAtsc3Capabilities atsc3Caps{
-                .bandwidthCap = (int)halInfo.frontendCaps.atsc3Caps().bandwidthCap,
-                .modulationCap = (int)halInfo.frontendCaps.atsc3Caps().modulationCap,
-                .timeInterleaveModeCap =
-                        (int)halInfo.frontendCaps.atsc3Caps().timeInterleaveModeCap,
-                .codeRateCap = (int)halInfo.frontendCaps.atsc3Caps().codeRateCap,
-                .demodOutputFormatCap = (int)halInfo.frontendCaps.atsc3Caps().demodOutputFormatCap,
-                .fecCap = (int)halInfo.frontendCaps.atsc3Caps().fecCap,
-            };
-            caps.set<TunerFrontendCapabilities::atsc3Caps>(atsc3Caps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::atsc3Caps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendAtsc3Capabilities atsc3Caps{
+                    .bandwidthCap = (int)halInfo.frontendCaps.atsc3Caps().bandwidthCap,
+                    .modulationCap = (int)halInfo.frontendCaps.atsc3Caps().modulationCap,
+                    .timeInterleaveModeCap =
+                            (int)halInfo.frontendCaps.atsc3Caps().timeInterleaveModeCap,
+                    .codeRateCap = (int)halInfo.frontendCaps.atsc3Caps().codeRateCap,
+                    .demodOutputFormatCap
+                        = (int)halInfo.frontendCaps.atsc3Caps().demodOutputFormatCap,
+                    .fecCap = (int)halInfo.frontendCaps.atsc3Caps().fecCap,
+                };
+                caps.set<TunerFrontendCapabilities::atsc3Caps>(atsc3Caps);
+            }
             break;
         }
         case FrontendType::DVBC: {
-            TunerFrontendCableCapabilities cableCaps{
-                .modulationCap = (int)halInfo.frontendCaps.dvbcCaps().modulationCap,
-                .codeRateCap = (int64_t)halInfo.frontendCaps.dvbcCaps().fecCap,
-                .annexCap = (int)halInfo.frontendCaps.dvbcCaps().annexCap,
-            };
-            caps.set<TunerFrontendCapabilities::cableCaps>(cableCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::dvbcCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendCableCapabilities cableCaps{
+                    .modulationCap = (int)halInfo.frontendCaps.dvbcCaps().modulationCap,
+                    .codeRateCap = (int64_t)halInfo.frontendCaps.dvbcCaps().fecCap,
+                    .annexCap = (int)halInfo.frontendCaps.dvbcCaps().annexCap,
+                };
+                caps.set<TunerFrontendCapabilities::cableCaps>(cableCaps);
+            }
             break;
         }
         case FrontendType::DVBS: {
-            TunerFrontendDvbsCapabilities dvbsCaps{
-                .modulationCap = (int)halInfo.frontendCaps.dvbsCaps().modulationCap,
-                .codeRateCap = (long)halInfo.frontendCaps.dvbsCaps().innerfecCap,
-                .standard = (int)halInfo.frontendCaps.dvbsCaps().standard,
-            };
-            caps.set<TunerFrontendCapabilities::dvbsCaps>(dvbsCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::dvbsCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendDvbsCapabilities dvbsCaps{
+                    .modulationCap = (int)halInfo.frontendCaps.dvbsCaps().modulationCap,
+                    .codeRateCap = (long)halInfo.frontendCaps.dvbsCaps().innerfecCap,
+                    .standard = (int)halInfo.frontendCaps.dvbsCaps().standard,
+                };
+                caps.set<TunerFrontendCapabilities::dvbsCaps>(dvbsCaps);
+            }
             break;
         }
         case FrontendType::DVBT: {
-            TunerFrontendDvbtCapabilities dvbtCaps{
-                .transmissionModeCap = (int)halInfo.frontendCaps.dvbtCaps().transmissionModeCap,
-                .bandwidthCap = (int)halInfo.frontendCaps.dvbtCaps().bandwidthCap,
-                .constellationCap = (int)halInfo.frontendCaps.dvbtCaps().constellationCap,
-                .codeRateCap = (int)halInfo.frontendCaps.dvbtCaps().coderateCap,
-                .hierarchyCap = (int)halInfo.frontendCaps.dvbtCaps().hierarchyCap,
-                .guardIntervalCap = (int)halInfo.frontendCaps.dvbtCaps().guardIntervalCap,
-                .isT2Supported = (bool)halInfo.frontendCaps.dvbtCaps().isT2Supported,
-                .isMisoSupported = (bool)halInfo.frontendCaps.dvbtCaps().isMisoSupported,
-            };
-            caps.set<TunerFrontendCapabilities::dvbtCaps>(dvbtCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::dvbtCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendDvbtCapabilities dvbtCaps{
+                    .transmissionModeCap = (int)halInfo.frontendCaps.dvbtCaps().transmissionModeCap,
+                    .bandwidthCap = (int)halInfo.frontendCaps.dvbtCaps().bandwidthCap,
+                    .constellationCap = (int)halInfo.frontendCaps.dvbtCaps().constellationCap,
+                    .codeRateCap = (int)halInfo.frontendCaps.dvbtCaps().coderateCap,
+                    .hierarchyCap = (int)halInfo.frontendCaps.dvbtCaps().hierarchyCap,
+                    .guardIntervalCap = (int)halInfo.frontendCaps.dvbtCaps().guardIntervalCap,
+                    .isT2Supported = (bool)halInfo.frontendCaps.dvbtCaps().isT2Supported,
+                    .isMisoSupported = (bool)halInfo.frontendCaps.dvbtCaps().isMisoSupported,
+                };
+                caps.set<TunerFrontendCapabilities::dvbtCaps>(dvbtCaps);
+            }
             break;
         }
         case FrontendType::ISDBS: {
-            TunerFrontendIsdbsCapabilities isdbsCaps{
-                .modulationCap = (int)halInfo.frontendCaps.isdbsCaps().modulationCap,
-                .codeRateCap = (int)halInfo.frontendCaps.isdbsCaps().coderateCap,
-            };
-            caps.set<TunerFrontendCapabilities::isdbsCaps>(isdbsCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::isdbsCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendIsdbsCapabilities isdbsCaps{
+                    .modulationCap = (int)halInfo.frontendCaps.isdbsCaps().modulationCap,
+                    .codeRateCap = (int)halInfo.frontendCaps.isdbsCaps().coderateCap,
+                };
+                caps.set<TunerFrontendCapabilities::isdbsCaps>(isdbsCaps);
+            }
             break;
         }
         case FrontendType::ISDBS3: {
-            TunerFrontendIsdbs3Capabilities isdbs3Caps{
-                .modulationCap = (int)halInfo.frontendCaps.isdbs3Caps().modulationCap,
-                .codeRateCap = (int)halInfo.frontendCaps.isdbs3Caps().coderateCap,
-            };
-            caps.set<TunerFrontendCapabilities::isdbs3Caps>(isdbs3Caps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::isdbs3Caps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendIsdbs3Capabilities isdbs3Caps{
+                    .modulationCap = (int)halInfo.frontendCaps.isdbs3Caps().modulationCap,
+                    .codeRateCap = (int)halInfo.frontendCaps.isdbs3Caps().coderateCap,
+                };
+                caps.set<TunerFrontendCapabilities::isdbs3Caps>(isdbs3Caps);
+            }
             break;
         }
         case FrontendType::ISDBT: {
-            TunerFrontendIsdbtCapabilities isdbtCaps{
-                .modeCap = (int)halInfo.frontendCaps.isdbtCaps().modeCap,
-                .bandwidthCap = (int)halInfo.frontendCaps.isdbtCaps().bandwidthCap,
-                .modulationCap = (int)halInfo.frontendCaps.isdbtCaps().modulationCap,
-                .codeRateCap = (int)halInfo.frontendCaps.isdbtCaps().coderateCap,
-                .guardIntervalCap = (int)halInfo.frontendCaps.isdbtCaps().guardIntervalCap,
-            };
-            caps.set<TunerFrontendCapabilities::isdbtCaps>(isdbtCaps);
+            if (FrontendInfo::FrontendCapabilities::hidl_discriminator::isdbtCaps
+                    == halInfo.frontendCaps.getDiscriminator()) {
+                TunerFrontendIsdbtCapabilities isdbtCaps{
+                    .modeCap = (int)halInfo.frontendCaps.isdbtCaps().modeCap,
+                    .bandwidthCap = (int)halInfo.frontendCaps.isdbtCaps().bandwidthCap,
+                    .modulationCap = (int)halInfo.frontendCaps.isdbtCaps().modulationCap,
+                    .codeRateCap = (int)halInfo.frontendCaps.isdbtCaps().coderateCap,
+                    .guardIntervalCap = (int)halInfo.frontendCaps.isdbtCaps().guardIntervalCap,
+                };
+                caps.set<TunerFrontendCapabilities::isdbtCaps>(isdbtCaps);
+            }
             break;
         }
         default: