diff --git a/camera/Camera.cpp b/camera/Camera.cpp
index c6c35ef..19849f8 100644
--- a/camera/Camera.cpp
+++ b/camera/Camera.cpp
@@ -347,6 +347,13 @@
     return c->setPreviewCallbackTarget(callbackProducer);
 }
 
+int32_t Camera::setAudioRestriction(int32_t mode)
+{
+    sp <::android::hardware::ICamera> c = mCamera;
+    if (c == 0) return NO_INIT;
+    return c->setAudioRestriction(mode);
+}
+
 // callback from camera service
 void Camera::notifyCallback(int32_t msgType, int32_t ext1, int32_t ext2)
 {
diff --git a/camera/ICamera.cpp b/camera/ICamera.cpp
index f0945c7..060e8e0 100644
--- a/camera/ICamera.cpp
+++ b/camera/ICamera.cpp
@@ -56,6 +56,7 @@
     SET_VIDEO_BUFFER_TARGET,
     RELEASE_RECORDING_FRAME_HANDLE,
     RELEASE_RECORDING_FRAME_HANDLE_BATCH,
+    SET_AUDIO_RESTRICTION,
 };
 
 class BpCamera: public BpInterface<ICamera>
@@ -191,6 +192,14 @@
         }
     }
 
+    int32_t setAudioRestriction(int32_t mode) {
+        Parcel data, reply;
+        data.writeInterfaceToken(ICamera::getInterfaceDescriptor());
+        data.writeInt32(mode);
+        remote()->transact(SET_AUDIO_RESTRICTION, data, &reply);
+        return reply.readInt32();
+    }
+
     status_t setVideoBufferMode(int32_t videoBufferMode)
     {
         ALOGV("setVideoBufferMode: %d", videoBufferMode);
@@ -494,6 +503,12 @@
             reply->writeInt32(setVideoTarget(st));
             return NO_ERROR;
         } break;
+        case SET_AUDIO_RESTRICTION: {
+            CHECK_INTERFACE(ICamera, data, reply);
+            int32_t mode = data.readInt32();
+            reply->writeInt32(setAudioRestriction(mode));
+            return NO_ERROR;
+        } break;
         default:
             return BBinder::onTransact(code, data, reply, flags);
     }
diff --git a/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl b/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl
index 49dfde8..5987b42 100644
--- a/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl
+++ b/camera/aidl/android/hardware/camera2/ICameraDeviceUser.aidl
@@ -155,4 +155,20 @@
     void updateOutputConfiguration(int streamId, in OutputConfiguration outputConfiguration);
 
     void finalizeOutputConfigurations(int streamId, in OutputConfiguration outputConfiguration);
+
+
+    // Keep in sync with public API in
+    // frameworks/base/core/java/android/hardware/camera2/CameraDevice.java
+    const int AUDIO_RESTRICTION_NONE = 0;
+    const int AUDIO_RESTRICTION_VIBRATION = 1;
+    const int AUDIO_RESTRICTION_VIBRATION_SOUND = 3;
+
+    /**
+      * Set audio restriction mode for this camera device.
+      *
+      * @param mode the audio restriction mode ID as above
+      *
+      * @return the resulting system-wide audio restriction mode
+      */
+    int setCameraAudioRestriction(int mode);
 }
diff --git a/camera/include/camera/Camera.h b/camera/include/camera/Camera.h
index 430aa1c..9800bb7 100644
--- a/camera/include/camera/Camera.h
+++ b/camera/include/camera/Camera.h
@@ -167,6 +167,8 @@
 
             sp<ICameraRecordingProxy> getRecordingProxy();
 
+            int32_t      setAudioRestriction(int32_t mode);
+
     // ICameraClient interface
     virtual void        notifyCallback(int32_t msgType, int32_t ext, int32_t ext2);
     virtual void        dataCallback(int32_t msgType, const sp<IMemory>& dataPtr,
diff --git a/camera/include/camera/android/hardware/ICamera.h b/camera/include/camera/android/hardware/ICamera.h
index 80823d6..eba9efe 100644
--- a/camera/include/camera/android/hardware/ICamera.h
+++ b/camera/include/camera/android/hardware/ICamera.h
@@ -140,6 +140,9 @@
     // Set the video buffer producer for camera to use in VIDEO_BUFFER_MODE_BUFFER_QUEUE mode.
     virtual status_t        setVideoTarget(
             const sp<IGraphicBufferProducer>& bufferProducer) = 0;
+
+    // Set the audio restriction mode
+    virtual int32_t         setAudioRestriction(int32_t mode) = 0;
 };
 
 // ----------------------------------------------------------------------------
diff --git a/camera/ndk/impl/ACameraCaptureSession.cpp b/camera/ndk/impl/ACameraCaptureSession.cpp
index d6f1412..68db233 100644
--- a/camera/ndk/impl/ACameraCaptureSession.cpp
+++ b/camera/ndk/impl/ACameraCaptureSession.cpp
@@ -33,7 +33,9 @@
         dev->unlockDevice();
     }
     // Fire onClosed callback
-    (*mUserSessionCallback.onClosed)(mUserSessionCallback.context, this);
+    if (mUserSessionCallback.onClosed != nullptr) {
+        (*mUserSessionCallback.onClosed)(mUserSessionCallback.context, this);
+    }
     ALOGV("~ACameraCaptureSession: %p is deleted", this);
 }
 
diff --git a/camera/ndk/include/camera/NdkCameraMetadataTags.h b/camera/ndk/include/camera/NdkCameraMetadataTags.h
index 8dd6e00..68fe045 100644
--- a/camera/ndk/include/camera/NdkCameraMetadataTags.h
+++ b/camera/ndk/include/camera/NdkCameraMetadataTags.h
@@ -1126,10 +1126,17 @@
      * </ul>
      * <p>For devices at the LIMITED level or above:</p>
      * <ul>
-     * <li>For YUV_420_888 burst capture use case, this list will always include (<code>min</code>, <code>max</code>)
-     * and (<code>max</code>, <code>max</code>) where <code>min</code> &lt;= 15 and <code>max</code> = the maximum output frame rate of the
+     * <li>For devices that advertise NIR color filter arrangement in
+     * ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT, this list will always include
+     * (<code>max</code>, <code>max</code>) where <code>max</code> = the maximum output frame rate of the maximum YUV_420_888
+     * output size.</li>
+     * <li>For devices advertising any color filter arrangement other than NIR, or devices not
+     * advertising color filter arrangement, this list will always include (<code>min</code>, <code>max</code>) and
+     * (<code>max</code>, <code>max</code>) where <code>min</code> &lt;= 15 and <code>max</code> = the maximum output frame rate of the
      * maximum YUV_420_888 output size.</li>
      * </ul>
+     *
+     * @see ACAMERA_SENSOR_INFO_COLOR_FILTER_ARRANGEMENT
      */
     ACAMERA_CONTROL_AE_AVAILABLE_TARGET_FPS_RANGES =            // int32[2*n]
             ACAMERA_CONTROL_START + 20,
@@ -3536,11 +3543,19 @@
      * output capture result.</p>
      * <p>This control is only effective if ACAMERA_CONTROL_AE_MODE or ACAMERA_CONTROL_MODE is set to
      * OFF; otherwise the auto-exposure algorithm will override this value.</p>
+     * <p>Note that for devices supporting postRawSensitivityBoost, the total sensitivity applied
+     * to the final processed image is the combination of ACAMERA_SENSOR_SENSITIVITY and
+     * ACAMERA_CONTROL_POST_RAW_SENSITIVITY_BOOST. In case the application uses the sensor
+     * sensitivity from last capture result of an auto request for a manual request, in order
+     * to achieve the same brightness in the output image, the application should also
+     * set postRawSensitivityBoost.</p>
      *
      * @see ACAMERA_CONTROL_AE_MODE
      * @see ACAMERA_CONTROL_MODE
+     * @see ACAMERA_CONTROL_POST_RAW_SENSITIVITY_BOOST
      * @see ACAMERA_SENSOR_INFO_SENSITIVITY_RANGE
      * @see ACAMERA_SENSOR_MAX_ANALOG_SENSITIVITY
+     * @see ACAMERA_SENSOR_SENSITIVITY
      */
     ACAMERA_SENSOR_SENSITIVITY =                                // int32
             ACAMERA_SENSOR_START + 2,
@@ -7751,6 +7766,13 @@
      */
     ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_SECURE_IMAGE_DATA         = 13,
 
+    /**
+     * <p>The camera device is only accessible by Android's system components and privileged
+     * applications. Processes need to have the android.permission.SYSTEM_CAMERA in
+     * addition to android.permission.CAMERA in order to connect to this camera device.</p>
+     */
+    ACAMERA_REQUEST_AVAILABLE_CAPABILITIES_SYSTEM_CAMERA             = 14,
+
 } acamera_metadata_enum_android_request_available_capabilities_t;
 
 
diff --git a/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp b/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
index 7ab0124..938b5f5 100644
--- a/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
+++ b/camera/ndk/ndk_vendor/tests/AImageReaderVendorTest.cpp
@@ -253,21 +253,9 @@
         return true;
     }
 
-    static void onDeviceDisconnected(void* /*obj*/, ACameraDevice* /*device*/) {}
-
-    static void onDeviceError(void* /*obj*/, ACameraDevice* /*device*/, int /*errorCode*/) {}
-
-    static void onSessionClosed(void* /*obj*/, ACameraCaptureSession* /*session*/) {}
-
-    static void onSessionReady(void* /*obj*/, ACameraCaptureSession* /*session*/) {}
-
-    static void onSessionActive(void* /*obj*/, ACameraCaptureSession* /*session*/) {}
-
    private:
-    ACameraDevice_StateCallbacks mDeviceCb{this, onDeviceDisconnected,
-                                           onDeviceError};
-    ACameraCaptureSession_stateCallbacks mSessionCb{
-        this, onSessionClosed, onSessionReady, onSessionActive};
+    ACameraDevice_StateCallbacks mDeviceCb{this, nullptr, nullptr};
+    ACameraCaptureSession_stateCallbacks mSessionCb{ this, nullptr, nullptr, nullptr};
 
     native_handle_t* mImgReaderAnw = nullptr;  // not owned by us.
 
diff --git a/camera/tests/CameraBinderTests.cpp b/camera/tests/CameraBinderTests.cpp
index 8fe029a..f07a1e6 100644
--- a/camera/tests/CameraBinderTests.cpp
+++ b/camera/tests/CameraBinderTests.cpp
@@ -57,7 +57,7 @@
 #include <algorithm>
 
 using namespace android;
-using ::android::hardware::ICameraServiceDefault;
+using ::android::hardware::ICameraService;
 using ::android::hardware::camera2::ICameraDeviceUser;
 
 #define ASSERT_NOT_NULL(x) \
@@ -507,7 +507,7 @@
         bool queryStatus;
         res = device->isSessionConfigurationSupported(sessionConfiguration, &queryStatus);
         EXPECT_TRUE(res.isOk() ||
-                (res.serviceSpecificErrorCode() == ICameraServiceDefault::ERROR_INVALID_OPERATION))
+                (res.serviceSpecificErrorCode() == ICameraService::ERROR_INVALID_OPERATION))
                 << res;
         if (res.isOk()) {
             EXPECT_TRUE(queryStatus);
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index df28842..e39f885 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -37,6 +37,7 @@
 
 #include <binder/IPCThreadState.h>
 #include <utils/Errors.h>
+#include <utils/SystemClock.h>
 #include <utils/Timers.h>
 #include <utils/Trace.h>
 
@@ -95,6 +96,8 @@
 static const uint32_t kFallbackWidth = 1280;        // 720p
 static const uint32_t kFallbackHeight = 720;
 static const char* kMimeTypeAvc = "video/avc";
+static const char* kMimeTypeApplicationOctetstream = "application/octet-stream";
+static const char* kWinscopeMagicString = "#VV1NSC0PET1ME!#";
 
 // Command-line parameters.
 static bool gVerbose = false;           // chatty on stdout
@@ -350,6 +353,50 @@
 }
 
 /*
+ * Writes an unsigned integer byte-by-byte in little endian order regardless
+ * of the platform endianness.
+ */
+template <typename UINT>
+static void writeValueLE(UINT value, uint8_t* buffer) {
+    for (int i = 0; i < sizeof(UINT); ++i) {
+        buffer[i] = static_cast<uint8_t>(value);
+        value >>= 8;
+    }
+}
+
+/*
+ * Saves frames presentation time relative to the elapsed realtime clock in microseconds
+ * preceded by a Winscope magic string and frame count to a metadata track.
+ * This metadata is used by the Winscope tool to sync video with SurfaceFlinger
+ * and WindowManager traces.
+ *
+ * The metadata is written as a binary array as follows:
+ * - winscope magic string (kWinscopeMagicString constant), without trailing null char,
+ * - the number of recorded frames (as little endian uint32),
+ * - for every frame its presentation time relative to the elapsed realtime clock in microseconds
+ *   (as little endian uint64).
+ */
+static status_t writeWinscopeMetadata(const Vector<int64_t>& timestamps,
+        const ssize_t metaTrackIdx, const sp<MediaMuxer>& muxer) {
+    ALOGV("Writing metadata");
+    int64_t systemTimeToElapsedTimeOffsetMicros = (android::elapsedRealtimeNano()
+        - systemTime(SYSTEM_TIME_MONOTONIC)) / 1000;
+    sp<ABuffer> buffer = new ABuffer(timestamps.size() * sizeof(int64_t)
+        + sizeof(uint32_t) + strlen(kWinscopeMagicString));
+    uint8_t* pos = buffer->data();
+    strcpy(reinterpret_cast<char*>(pos), kWinscopeMagicString);
+    pos += strlen(kWinscopeMagicString);
+    writeValueLE<uint32_t>(timestamps.size(), pos);
+    pos += sizeof(uint32_t);
+    for (size_t idx = 0; idx < timestamps.size(); ++idx) {
+        writeValueLE<uint64_t>(static_cast<uint64_t>(timestamps[idx]
+            + systemTimeToElapsedTimeOffsetMicros), pos);
+        pos += sizeof(uint64_t);
+    }
+    return muxer->writeSampleData(buffer, metaTrackIdx, timestamps[0], 0);
+}
+
+/*
  * Runs the MediaCodec encoder, sending the output to the MediaMuxer.  The
  * input frames are coming from the virtual display as fast as SurfaceFlinger
  * wants to send them.
@@ -364,10 +411,12 @@
     static int kTimeout = 250000;   // be responsive on signal
     status_t err;
     ssize_t trackIdx = -1;
+    ssize_t metaTrackIdx = -1;
     uint32_t debugNumFrames = 0;
     int64_t startWhenNsec = systemTime(CLOCK_MONOTONIC);
     int64_t endWhenNsec = startWhenNsec + seconds_to_nanoseconds(gTimeLimitSec);
     DisplayInfo mainDpyInfo;
+    Vector<int64_t> timestamps;
     bool firstFrame = true;
 
     assert((rawFp == NULL && muxer != NULL) || (rawFp != NULL && muxer == NULL));
@@ -471,6 +520,9 @@
                             "Failed writing data to muxer (err=%d)\n", err);
                         return err;
                     }
+                    if (gOutputFormat == FORMAT_MP4) {
+                        timestamps.add(ptsUsec);
+                    }
                 }
                 debugNumFrames++;
             }
@@ -497,6 +549,11 @@
                 encoder->getOutputFormat(&newFormat);
                 if (muxer != NULL) {
                     trackIdx = muxer->addTrack(newFormat);
+                    if (gOutputFormat == FORMAT_MP4) {
+                        sp<AMessage> metaFormat = new AMessage;
+                        metaFormat->setString(KEY_MIME, kMimeTypeApplicationOctetstream);
+                        metaTrackIdx = muxer->addTrack(metaFormat);
+                    }
                     ALOGV("Starting muxer");
                     err = muxer->start();
                     if (err != NO_ERROR) {
@@ -533,6 +590,13 @@
                         systemTime(CLOCK_MONOTONIC) - startWhenNsec));
         fflush(stdout);
     }
+    if (metaTrackIdx >= 0 && !timestamps.isEmpty()) {
+        err = writeWinscopeMetadata(timestamps, metaTrackIdx, muxer);
+        if (err != NO_ERROR) {
+            fprintf(stderr, "Failed writing metadata to muxer (err=%d)\n", err);
+            return err;
+        }
+    }
     return NO_ERROR;
 }
 
diff --git a/drm/libmediadrm/CryptoHal.cpp b/drm/libmediadrm/CryptoHal.cpp
index d62ccd6..954608f 100644
--- a/drm/libmediadrm/CryptoHal.cpp
+++ b/drm/libmediadrm/CryptoHal.cpp
@@ -19,9 +19,9 @@
 #include <utils/Log.h>
 
 #include <android/hardware/drm/1.0/types.h>
-#include <android/hidl/manager/1.0/IServiceManager.h>
-
+#include <android/hidl/manager/1.2/IServiceManager.h>
 #include <binder/IMemory.h>
+#include <hidl/ServiceManagement.h>
 #include <hidlmemory/FrameworkUtils.h>
 #include <media/hardware/CryptoAPI.h>
 #include <media/stagefright/foundation/ADebug.h>
@@ -47,7 +47,6 @@
 using ::android::hardware::hidl_vec;
 using ::android::hardware::Return;
 using ::android::hardware::Void;
-using ::android::hidl::manager::V1_0::IServiceManager;
 using ::android::sp;
 
 typedef drm::V1_2::Status Status_V1_2;
@@ -129,9 +128,9 @@
 Vector<sp<ICryptoFactory>> CryptoHal::makeCryptoFactories() {
     Vector<sp<ICryptoFactory>> factories;
 
-    auto manager = ::IServiceManager::getService();
+    auto manager = hardware::defaultServiceManager1_2();
     if (manager != NULL) {
-        manager->listByInterface(drm::V1_0::ICryptoFactory::descriptor,
+        manager->listManifestByInterface(drm::V1_0::ICryptoFactory::descriptor,
                 [&factories](const hidl_vec<hidl_string> &registered) {
                     for (const auto &instance : registered) {
                         auto factory = drm::V1_0::ICryptoFactory::getService(instance);
@@ -142,7 +141,7 @@
                     }
                 }
             );
-        manager->listByInterface(drm::V1_1::ICryptoFactory::descriptor,
+        manager->listManifestByInterface(drm::V1_1::ICryptoFactory::descriptor,
                 [&factories](const hidl_vec<hidl_string> &registered) {
                     for (const auto &instance : registered) {
                         auto factory = drm::V1_1::ICryptoFactory::getService(instance);
diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index 919f4ee..7cfe900 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -26,7 +26,6 @@
 #include <android/hardware/drm/1.2/types.h>
 #include <android/hidl/manager/1.2/IServiceManager.h>
 #include <hidl/ServiceManagement.h>
-
 #include <media/EventMetric.h>
 #include <media/PluginMetricsReporting.h>
 #include <media/drm/DrmAPI.h>
@@ -57,7 +56,6 @@
 using ::android::hardware::hidl_vec;
 using ::android::hardware::Return;
 using ::android::hardware::Void;
-using ::android::hidl::manager::V1_0::IServiceManager;
 using ::android::os::PersistableBundle;
 using ::android::sp;
 
@@ -394,7 +392,7 @@
                     }
                 }
             );
-        manager->listByInterface(drm::V1_2::IDrmFactory::descriptor,
+        manager->listManifestByInterface(drm::V1_2::IDrmFactory::descriptor,
                 [&factories](const hidl_vec<hidl_string> &registered) {
                     for (const auto &instance : registered) {
                         auto factory = drm::V1_2::IDrmFactory::getService(instance);
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
index eaa3390..cb69f91 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
@@ -89,7 +89,7 @@
     // asset_id change. If it sends an EcmContainer with 2 Ecms with different
     // asset_ids (old and new) then it might be best to prefetch the Emm.
     if ((asset_.id() != 0) && (*asset_id != asset_.id())) {
-        ALOGW("Asset_id change from %llu to %" PRIu64, asset_.id(), *asset_id);
+        ALOGW("Asset_id change from %" PRIu64 " to %" PRIu64, asset_.id(), *asset_id);
         asset_.Clear();
     }
 
diff --git a/drm/mediacas/plugins/clearkey/ecm.cpp b/drm/mediacas/plugins/clearkey/ecm.cpp
index 9fde13a..b3b5218 100644
--- a/drm/mediacas/plugins/clearkey/ecm.cpp
+++ b/drm/mediacas/plugins/clearkey/ecm.cpp
@@ -17,6 +17,8 @@
 //#define LOG_NDEBUG 0
 #define LOG_TAG "ecm"
 
+#include <inttypes.h>
+
 #include "ecm.h"
 #include "ecm_generator.h"
 #include "protos/license_protos.pb.h"
@@ -76,7 +78,7 @@
         return status;
     }
     if (asset.id() != asset_from_emm.id()) {
-        ALOGE("Asset_id from Emm (%llu) does not match asset_id from Ecm (%llu).",
+        ALOGE("Asset_id from Emm (%" PRIu64 ") does not match asset_id from Ecm (%" PRIu64 ").",
                 asset_from_emm.id(), asset.id());
         return CLEARKEY_STATUS_INVALID_PARAMETER;
     }
diff --git a/include/media/AudioMixer.h b/include/media/AudioMixer.h
index de839c6..85ee950 120000
--- a/include/media/AudioMixer.h
+++ b/include/media/AudioMixer.h
@@ -1 +1 @@
-../../media/libaudioclient/include/media/AudioMixer.h
\ No newline at end of file
+../../media/libaudioprocessing/include/media/AudioMixer.h
\ No newline at end of file
diff --git a/include/media/BufferProviders.h b/include/media/BufferProviders.h
index 779bb15..778e1d8 120000
--- a/include/media/BufferProviders.h
+++ b/include/media/BufferProviders.h
@@ -1 +1 @@
-../../media/libmedia/include/media/BufferProviders.h
\ No newline at end of file
+../../media/libaudioprocessing/include/media/BufferProviders.h
\ No newline at end of file
diff --git a/include/private/media/VideoFrame.h b/include/private/media/VideoFrame.h
index 712f118..16e794a 100644
--- a/include/private/media/VideoFrame.h
+++ b/include/private/media/VideoFrame.h
@@ -41,7 +41,7 @@
             uint32_t angle, uint32_t bpp, bool hasData, size_t iccSize):
         mWidth(width), mHeight(height),
         mDisplayWidth(displayWidth), mDisplayHeight(displayHeight),
-        mTileWidth(tileWidth), mTileHeight(tileHeight),
+        mTileWidth(tileWidth), mTileHeight(tileHeight), mDurationUs(0),
         mRotationAngle(angle), mBytesPerPixel(bpp), mRowBytes(bpp * width),
         mSize(hasData ? (bpp * width * height) : 0),
         mIccSize(iccSize), mReserved(0) {
@@ -78,6 +78,7 @@
     uint32_t mDisplayHeight;   // Display height before rotation
     uint32_t mTileWidth;       // Tile width (0 if image doesn't have grid)
     uint32_t mTileHeight;      // Tile height (0 if image doesn't have grid)
+    int64_t  mDurationUs;      // Frame duration in microseconds
     int32_t  mRotationAngle;   // Rotation angle, clockwise, should be multiple of 90
     uint32_t mBytesPerPixel;   // Number of bytes per pixel
     uint32_t mRowBytes;        // Number of bytes per row before rotation
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index f5f021b..6697cb5 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -9,6 +9,7 @@
 	libaaudioservice \
 	libaudioflinger \
 	libaudiopolicyservice \
+	libaudioprocessing \
 	libbinder \
 	libcutils \
 	liblog \
diff --git a/media/bufferpool/1.0/AccessorImpl.cpp b/media/bufferpool/1.0/AccessorImpl.cpp
index a5366f6..09006ca 100644
--- a/media/bufferpool/1.0/AccessorImpl.cpp
+++ b/media/bufferpool/1.0/AccessorImpl.cpp
@@ -248,7 +248,7 @@
     ALOGD("Destruction - bufferpool %p "
           "cached: %zu/%zuM, %zu/%d%% in use; "
           "allocs: %zu, %d%% recycled; "
-          "transfers: %zu, %d%% unfetced",
+          "transfers: %zu, %d%% unfetched",
           this, mStats.mBuffersCached, mStats.mSizeCached >> 20,
           mStats.mBuffersInUse, percentage(mStats.mBuffersInUse, mStats.mBuffersCached),
           mStats.mTotalAllocations, percentage(mStats.mTotalRecycles, mStats.mTotalAllocations),
diff --git a/media/bufferpool/2.0/AccessorImpl.cpp b/media/bufferpool/2.0/AccessorImpl.cpp
index cacd465..929a20e 100644
--- a/media/bufferpool/2.0/AccessorImpl.cpp
+++ b/media/bufferpool/2.0/AccessorImpl.cpp
@@ -304,7 +304,7 @@
     ALOGD("Destruction - bufferpool2 %p "
           "cached: %zu/%zuM, %zu/%d%% in use; "
           "allocs: %zu, %d%% recycled; "
-          "transfers: %zu, %d%% unfetced",
+          "transfers: %zu, %d%% unfetched",
           this, mStats.mBuffersCached, mStats.mSizeCached >> 20,
           mStats.mBuffersInUse, percentage(mStats.mBuffersInUse, mStats.mBuffersCached),
           mStats.mTotalAllocations, percentage(mStats.mTotalRecycles, mStats.mTotalAllocations),
diff --git a/media/codec2/components/aac/C2SoftAacEnc.cpp b/media/codec2/components/aac/C2SoftAacEnc.cpp
index 1dc676b..a8f39d5 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.cpp
+++ b/media/codec2/components/aac/C2SoftAacEnc.cpp
@@ -159,7 +159,8 @@
       mInputSize(0),
       mNextFrameTimestampUs(0),
       mSignalledError(false),
-      mOutIndex(0u) {
+      mOutIndex(0u),
+      mRemainderLen(0u) {
 }
 
 C2SoftAacEnc::~C2SoftAacEnc() {
@@ -185,6 +186,7 @@
     mInputSize = 0u;
     mNextFrameTimestampUs = 0;
     mSignalledError = false;
+    mRemainderLen = 0;
     return C2_OK;
 }
 
@@ -369,18 +371,21 @@
         mInputTimeSet = true;
     }
 
-    size_t numFrames = (capacity + mInputSize + (eos ? mNumBytesPerInputFrame - 1 : 0))
-            / mNumBytesPerInputFrame;
+    size_t numFrames =
+        (mRemainderLen + capacity + mInputSize + (eos ? mNumBytesPerInputFrame - 1 : 0))
+        / mNumBytesPerInputFrame;
     ALOGV("capacity = %zu; mInputSize = %zu; numFrames = %zu "
-          "mNumBytesPerInputFrame = %u inputTS = %lld",
+          "mNumBytesPerInputFrame = %u inputTS = %lld remaining = %zu",
           capacity, mInputSize, numFrames,
-          mNumBytesPerInputFrame, work->input.ordinal.timestamp.peekll());
+          mNumBytesPerInputFrame, work->input.ordinal.timestamp.peekll(),
+          mRemainderLen);
 
     std::shared_ptr<C2LinearBlock> block;
     std::unique_ptr<C2WriteView> wView;
     uint8_t *outPtr = temp;
     size_t outAvailable = 0u;
     uint64_t inputIndex = work->input.ordinal.frameIndex.peeku();
+    size_t bytesPerSample = channelCount * sizeof(int16_t);
 
     AACENC_InArgs inargs;
     AACENC_OutArgs outargs;
@@ -449,7 +454,25 @@
     };
     std::list<OutputBuffer> outputBuffers;
 
-    while (encoderErr == AACENC_OK && inargs.numInSamples > 0) {
+    if (mRemainderLen > 0) {
+        size_t offset = 0;
+        for (; mRemainderLen < bytesPerSample && offset < capacity; ++offset) {
+            mRemainder[mRemainderLen++] = data[offset];
+        }
+        data += offset;
+        capacity -= offset;
+        if (mRemainderLen == bytesPerSample) {
+            inBuffer[0] = mRemainder;
+            inBufferSize[0] = bytesPerSample;
+            inargs.numInSamples = channelCount;
+            mRemainderLen = 0;
+            ALOGV("Processing remainder");
+        } else {
+            // We have exhausted the input already
+            inargs.numInSamples = 0;
+        }
+    }
+    while (encoderErr == AACENC_OK && inargs.numInSamples >= channelCount) {
         if (numFrames && !block) {
             C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
             // TODO: error handling, proper usage, etc.
@@ -486,7 +509,7 @@
                 mNextFrameTimestampUs = work->input.ordinal.timestamp
                         + (consumed * 1000000ll / channelCount / sampleRate);
                 std::shared_ptr<C2Buffer> buffer = createLinearBuffer(block, 0, outargs.numOutBytes);
-#if defined(LOG_NDEBUG) && !LOG_NDEBUG
+#if 0
                 hexdump(outPtr, std::min(outargs.numOutBytes, 256));
 #endif
                 outPtr = temp;
@@ -503,12 +526,17 @@
                 inBufferSize[0] -= outargs.numInSamples * sizeof(int16_t);
                 inargs.numInSamples -= outargs.numInSamples;
             }
+
+            if (inBuffer[0] == mRemainder) {
+                inBuffer[0] = const_cast<uint8_t *>(data);
+                inBufferSize[0] = capacity;
+                inargs.numInSamples = capacity / sizeof(int16_t);
+            }
         }
         ALOGV("encoderErr = %d mInputSize = %zu "
               "inargs.numInSamples = %d, mNextFrameTimestampUs = %lld",
               encoderErr, mInputSize, inargs.numInSamples, mNextFrameTimestampUs.peekll());
     }
-
     if (eos && inBufferSize[0] > 0) {
         if (numFrames && !block) {
             C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
@@ -539,6 +567,14 @@
                            &outBufDesc,
                            &inargs,
                            &outargs);
+        inBufferSize[0] = 0;
+    }
+
+    if (inBufferSize[0] > 0) {
+        for (size_t i = 0; i < inBufferSize[0]; ++i) {
+            mRemainder[i] = static_cast<uint8_t *>(inBuffer[0])[i];
+        }
+        mRemainderLen = inBufferSize[0];
     }
 
     while (outputBuffers.size() > 1) {
diff --git a/media/codec2/components/aac/C2SoftAacEnc.h b/media/codec2/components/aac/C2SoftAacEnc.h
index 2655039..6ecfbdd 100644
--- a/media/codec2/components/aac/C2SoftAacEnc.h
+++ b/media/codec2/components/aac/C2SoftAacEnc.h
@@ -61,6 +61,10 @@
     bool mSignalledError;
     std::atomic_uint64_t mOutIndex;
 
+    // We support max 6 channels
+    uint8_t mRemainder[6 * sizeof(int16_t)];
+    size_t mRemainderLen;
+
     status_t initEncoder();
 
     status_t setAudioParams();
diff --git a/media/codec2/components/flac/Android.bp b/media/codec2/components/flac/Android.bp
index e5eb51d..48cc51b 100644
--- a/media/codec2/components/flac/Android.bp
+++ b/media/codec2/components/flac/Android.bp
@@ -23,8 +23,11 @@
 
     srcs: ["C2SoftFlacEnc.cpp"],
 
-    static_libs: [
+    shared_libs: [
         "libaudioutils",
+    ],
+
+    static_libs: [
         "libFLAC",
     ],
 }
diff --git a/media/codec2/components/g711/C2SoftG711Dec.cpp b/media/codec2/components/g711/C2SoftG711Dec.cpp
index 43b843a..b6cc32e 100644
--- a/media/codec2/components/g711/C2SoftG711Dec.cpp
+++ b/media/codec2/components/g711/C2SoftG711Dec.cpp
@@ -73,7 +73,7 @@
 
         addParameter(
                 DefineParam(mChannelCount, C2_PARAMKEY_CHANNEL_COUNT)
-                .withDefault(new C2StreamChannelCountInfo::output(0u, 1))
+                .withDefault(new C2StreamChannelCountInfo::output(0u, 6))
                 .withFields({C2F(mChannelCount, value).equalTo(1)})
                 .withSetter(Setter<decltype(*mChannelCount)>::StrictValueWithNoDeps)
                 .build());
diff --git a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
index df7b403..a1f8ff3 100644
--- a/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
+++ b/media/codec2/components/mpeg2/C2SoftMpeg2Dec.cpp
@@ -47,6 +47,12 @@
         noInputLatency();
         noTimeStretch();
 
+        // TODO: Proper support for reorder depth.
+        addParameter(
+                DefineParam(mActualOutputDelay, C2_PARAMKEY_OUTPUT_DELAY)
+                .withConstValue(new C2PortActualDelayTuning::output(3u))
+                .build());
+
         // TODO: output latency and reordering
 
         addParameter(
diff --git a/media/codec2/components/xaac/C2SoftXaacDec.cpp b/media/codec2/components/xaac/C2SoftXaacDec.cpp
index a3ebadb..60ae93c 100644
--- a/media/codec2/components/xaac/C2SoftXaacDec.cpp
+++ b/media/codec2/components/xaac/C2SoftXaacDec.cpp
@@ -1309,69 +1309,84 @@
                                 &ui_exec_done);
     RETURN_IF_FATAL(err_code,  "IA_CMD_TYPE_DONE_QUERY");
 
-    if (ui_exec_done != 1) {
-        VOID* p_array;        // ITTIAM:buffer to handle gain payload
-        WORD32 buf_size = 0;  // ITTIAM:gain payload length
-        WORD32 bit_str_fmt = 1;
-        WORD32 gain_stream_flag = 1;
-
-        err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
-                                    IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN, &buf_size);
-        RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN");
-
-        err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
-                                    IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF, &p_array);
-        RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF");
-
-        if (buf_size > 0) {
-            /*Set bitstream_split_format */
-            err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
-                                      IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT, &bit_str_fmt);
-            RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
-
-            memcpy(mDrcInBuf, p_array, buf_size);
-            /* Set number of bytes to be processed */
-            err_code =
-                ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES_BS, 0, &buf_size);
-            RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
-
-            err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
-                                      IA_DRC_DEC_CONFIG_GAIN_STREAM_FLAG, &gain_stream_flag);
-            RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
-
-            /* Execute process */
-            err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_INIT,
-                                      IA_CMD_TYPE_INIT_CPY_BSF_BUFF, nullptr);
-            RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
-
-            mMpegDDRCPresent = 1;
-        }
-    }
-
-    /* How much buffer is used in input buffers */
+    int32_t num_preroll = 0;
     err_code = ixheaacd_dec_api(mXheaacCodecHandle,
-                                IA_API_CMD_GET_CURIDX_INPUT_BUF,
-                                0,
-                                bytesConsumed);
-    RETURN_IF_FATAL(err_code,  "IA_API_CMD_GET_CURIDX_INPUT_BUF");
+                                IA_API_CMD_GET_CONFIG_PARAM,
+                                IA_ENHAACPLUS_DEC_CONFIG_GET_NUM_PRE_ROLL_FRAMES,
+                                &num_preroll);
+    RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GET_NUM_PRE_ROLL_FRAMES");
 
-    /* Get the output bytes */
-    err_code = ixheaacd_dec_api(mXheaacCodecHandle,
-                                IA_API_CMD_GET_OUTPUT_BYTES,
-                                0,
-                                outBytes);
-    RETURN_IF_FATAL(err_code,  "IA_API_CMD_GET_OUTPUT_BYTES");
+    {
+      int32_t preroll_frame_offset = 0;
 
-    if (mMpegDDRCPresent == 1) {
-        memcpy(mDrcInBuf, mOutputBuffer, *outBytes);
-        err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES, 0, outBytes);
-        RETURN_IF_FATAL(err_code, "IA_API_CMD_SET_INPUT_BYTES");
+        do {
+            if (ui_exec_done != 1) {
+                VOID* p_array;        // ITTIAM:buffer to handle gain payload
+                WORD32 buf_size = 0;  // ITTIAM:gain payload length
+                WORD32 bit_str_fmt = 1;
+                WORD32 gain_stream_flag = 1;
 
-        err_code =
-            ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_EXECUTE, IA_CMD_TYPE_DO_EXECUTE, nullptr);
-        RETURN_IF_FATAL(err_code, "IA_CMD_TYPE_DO_EXECUTE");
+                err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
+                                            IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN, &buf_size);
+                RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN");
 
-        memcpy(mOutputBuffer, mDrcOutBuf, *outBytes);
+                err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
+                                            IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF, &p_array);
+                RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF");
+
+                if (buf_size > 0) {
+                    /*Set bitstream_split_format */
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
+                                            IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT, &bit_str_fmt);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+
+                    memcpy(mDrcInBuf, p_array, buf_size);
+                    /* Set number of bytes to be processed */
+                    err_code =
+                        ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES_BS, 0, &buf_size);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
+                                            IA_DRC_DEC_CONFIG_GAIN_STREAM_FLAG, &gain_stream_flag);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+
+                    /* Execute process */
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_INIT,
+                                            IA_CMD_TYPE_INIT_CPY_BSF_BUFF, nullptr);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+
+                    mMpegDDRCPresent = 1;
+                }
+            }
+
+            /* How much buffer is used in input buffers */
+            err_code = ixheaacd_dec_api(mXheaacCodecHandle,
+                                        IA_API_CMD_GET_CURIDX_INPUT_BUF,
+                                        0,
+                                        bytesConsumed);
+            RETURN_IF_FATAL(err_code,  "IA_API_CMD_GET_CURIDX_INPUT_BUF");
+
+            /* Get the output bytes */
+            err_code = ixheaacd_dec_api(mXheaacCodecHandle,
+                                        IA_API_CMD_GET_OUTPUT_BYTES,
+                                        0,
+                                        outBytes);
+            RETURN_IF_FATAL(err_code,  "IA_API_CMD_GET_OUTPUT_BYTES");
+
+            if (mMpegDDRCPresent == 1) {
+                memcpy(mDrcInBuf, mOutputBuffer + preroll_frame_offset, *outBytes);
+                preroll_frame_offset += *outBytes;
+                err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES, 0, outBytes);
+                RETURN_IF_FATAL(err_code, "IA_API_CMD_SET_INPUT_BYTES");
+
+                err_code =
+                    ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_EXECUTE, IA_CMD_TYPE_DO_EXECUTE, nullptr);
+                RETURN_IF_FATAL(err_code, "IA_CMD_TYPE_DO_EXECUTE");
+
+                memcpy(mOutputBuffer, mDrcOutBuf, *outBytes);
+            }
+            num_preroll--;
+        } while (num_preroll > 0);
     }
     return IA_NO_ERROR;
 }
diff --git a/media/codec2/hidl/1.0/utils/InputBufferManager.cpp b/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
index a023a05..8c0d0a4 100644
--- a/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
+++ b/media/codec2/hidl/1.0/utils/InputBufferManager.cpp
@@ -70,7 +70,7 @@
                  << ".";
     std::lock_guard<std::mutex> lock(mMutex);
 
-    std::set<TrackedBuffer> &bufferIds =
+    std::set<TrackedBuffer*> &bufferIds =
             mTrackedBuffersMap[listener][frameIndex];
 
     for (size_t i = 0; i < input.buffers.size(); ++i) {
@@ -79,13 +79,14 @@
                          << "Input buffer at index " << i << " is null.";
             continue;
         }
-        const TrackedBuffer &bufferId =
-                *bufferIds.emplace(listener, frameIndex, i, input.buffers[i]).
-                first;
+        TrackedBuffer *bufferId =
+            new TrackedBuffer(listener, frameIndex, i, input.buffers[i]);
+        mTrackedBufferCache.emplace(bufferId);
+        bufferIds.emplace(bufferId);
 
         c2_status_t status = input.buffers[i]->registerOnDestroyNotify(
                 onBufferDestroyed,
-                const_cast<void*>(reinterpret_cast<const void*>(&bufferId)));
+                reinterpret_cast<void*>(bufferId));
         if (status != C2_OK) {
             LOG(DEBUG) << "InputBufferManager::_registerFrameData -- "
                        << "registerOnDestroyNotify() failed "
@@ -119,31 +120,32 @@
 
     auto findListener = mTrackedBuffersMap.find(listener);
     if (findListener != mTrackedBuffersMap.end()) {
-        std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds
+        std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds
                 = findListener->second;
         auto findFrameIndex = frameIndex2BufferIds.find(frameIndex);
         if (findFrameIndex != frameIndex2BufferIds.end()) {
-            std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
-            for (const TrackedBuffer& bufferId : bufferIds) {
-                std::shared_ptr<C2Buffer> buffer = bufferId.buffer.lock();
+            std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
+            for (TrackedBuffer* bufferId : bufferIds) {
+                std::shared_ptr<C2Buffer> buffer = bufferId->buffer.lock();
                 if (buffer) {
                     c2_status_t status = buffer->unregisterOnDestroyNotify(
                             onBufferDestroyed,
-                            const_cast<void*>(
-                            reinterpret_cast<const void*>(&bufferId)));
+                            reinterpret_cast<void*>(bufferId));
                     if (status != C2_OK) {
                         LOG(DEBUG) << "InputBufferManager::_unregisterFrameData "
                                    << "-- unregisterOnDestroyNotify() failed "
                                    << "(listener @ 0x"
                                         << std::hex
-                                        << bufferId.listener.unsafe_get()
+                                        << bufferId->listener.unsafe_get()
                                    << ", frameIndex = "
-                                        << std::dec << bufferId.frameIndex
-                                   << ", bufferIndex = " << bufferId.bufferIndex
+                                        << std::dec << bufferId->frameIndex
+                                   << ", bufferIndex = " << bufferId->bufferIndex
                                    << ") => status = " << status
                                    << ".";
                     }
                 }
+                mTrackedBufferCache.erase(bufferId);
+                delete bufferId;
             }
 
             frameIndex2BufferIds.erase(findFrameIndex);
@@ -179,31 +181,32 @@
 
     auto findListener = mTrackedBuffersMap.find(listener);
     if (findListener != mTrackedBuffersMap.end()) {
-        std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds =
+        std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds =
                 findListener->second;
         for (auto findFrameIndex = frameIndex2BufferIds.begin();
                 findFrameIndex != frameIndex2BufferIds.end();
                 ++findFrameIndex) {
-            std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
-            for (const TrackedBuffer& bufferId : bufferIds) {
-                std::shared_ptr<C2Buffer> buffer = bufferId.buffer.lock();
+            std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
+            for (TrackedBuffer* bufferId : bufferIds) {
+                std::shared_ptr<C2Buffer> buffer = bufferId->buffer.lock();
                 if (buffer) {
                     c2_status_t status = buffer->unregisterOnDestroyNotify(
                             onBufferDestroyed,
-                            const_cast<void*>(
-                            reinterpret_cast<const void*>(&bufferId)));
+                            reinterpret_cast<void*>(bufferId));
                     if (status != C2_OK) {
                         LOG(DEBUG) << "InputBufferManager::_unregisterFrameData "
                                    << "-- unregisterOnDestroyNotify() failed "
                                    << "(listener @ 0x"
                                         << std::hex
-                                        << bufferId.listener.unsafe_get()
+                                        << bufferId->listener.unsafe_get()
                                    << ", frameIndex = "
-                                        << std::dec << bufferId.frameIndex
-                                   << ", bufferIndex = " << bufferId.bufferIndex
+                                        << std::dec << bufferId->frameIndex
+                                   << ", bufferIndex = " << bufferId->bufferIndex
                                    << ") => status = " << status
                                    << ".";
                     }
+                    mTrackedBufferCache.erase(bufferId);
+                    delete bufferId;
                 }
             }
         }
@@ -236,50 +239,59 @@
                      << std::dec << ".";
         return;
     }
-    TrackedBuffer id(*reinterpret_cast<TrackedBuffer*>(arg));
+
+    std::lock_guard<std::mutex> lock(mMutex);
+    TrackedBuffer *bufferId = reinterpret_cast<TrackedBuffer*>(arg);
+
+    if (mTrackedBufferCache.find(bufferId) == mTrackedBufferCache.end()) {
+        LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- called with "
+                     << "unregistered buffer: "
+                     << "buf @ 0x" << std::hex << buf
+                     << ", arg @ 0x" << std::hex << arg
+                     << std::dec << ".";
+        return;
+    }
+
     LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- called with "
                  << "buf @ 0x" << std::hex << buf
                  << ", arg @ 0x" << std::hex << arg
                  << std::dec << " -- "
-                 << "listener @ 0x" << std::hex << id.listener.unsafe_get()
-                 << ", frameIndex = " << std::dec << id.frameIndex
-                 << ", bufferIndex = " << id.bufferIndex
+                 << "listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+                 << ", frameIndex = " << std::dec << bufferId->frameIndex
+                 << ", bufferIndex = " << bufferId->bufferIndex
                  << ".";
-
-    std::lock_guard<std::mutex> lock(mMutex);
-
-    auto findListener = mTrackedBuffersMap.find(id.listener);
+    auto findListener = mTrackedBuffersMap.find(bufferId->listener);
     if (findListener == mTrackedBuffersMap.end()) {
-        LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
-                   << "received invalid listener: "
-                   << "listener @ 0x" << std::hex << id.listener.unsafe_get()
-                   << " (frameIndex = " << std::dec << id.frameIndex
-                   << ", bufferIndex = " << id.bufferIndex
-                   << ").";
+        LOG(VERBOSE) << "InputBufferManager::_onBufferDestroyed -- "
+                     << "received invalid listener: "
+                     << "listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+                     << " (frameIndex = " << std::dec << bufferId->frameIndex
+                     << ", bufferIndex = " << bufferId->bufferIndex
+                     << ").";
         return;
     }
 
-    std::map<uint64_t, std::set<TrackedBuffer>> &frameIndex2BufferIds
+    std::map<uint64_t, std::set<TrackedBuffer*>> &frameIndex2BufferIds
             = findListener->second;
-    auto findFrameIndex = frameIndex2BufferIds.find(id.frameIndex);
+    auto findFrameIndex = frameIndex2BufferIds.find(bufferId->frameIndex);
     if (findFrameIndex == frameIndex2BufferIds.end()) {
         LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
                    << "received invalid frame index: "
-                   << "frameIndex = " << id.frameIndex
-                   << " (listener @ 0x" << std::hex << id.listener.unsafe_get()
-                   << ", bufferIndex = " << std::dec << id.bufferIndex
+                   << "frameIndex = " << bufferId->frameIndex
+                   << " (listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
+                   << ", bufferIndex = " << std::dec << bufferId->bufferIndex
                    << ").";
         return;
     }
 
-    std::set<TrackedBuffer> &bufferIds = findFrameIndex->second;
-    auto findBufferId = bufferIds.find(id);
+    std::set<TrackedBuffer*> &bufferIds = findFrameIndex->second;
+    auto findBufferId = bufferIds.find(bufferId);
     if (findBufferId == bufferIds.end()) {
         LOG(DEBUG) << "InputBufferManager::_onBufferDestroyed -- "
                    << "received invalid buffer index: "
-                   << "bufferIndex = " << id.bufferIndex
-                   << " (frameIndex = " << id.frameIndex
-                   << ", listener @ 0x" << std::hex << id.listener.unsafe_get()
+                   << "bufferIndex = " << bufferId->bufferIndex
+                   << " (frameIndex = " << bufferId->frameIndex
+                   << ", listener @ 0x" << std::hex << bufferId->listener.unsafe_get()
                    << std::dec << ").";
         return;
     }
@@ -292,10 +304,13 @@
         }
     }
 
-    DeathNotifications &deathNotifications = mDeathNotifications[id.listener];
-    deathNotifications.indices[id.frameIndex].emplace_back(id.bufferIndex);
+    DeathNotifications &deathNotifications = mDeathNotifications[bufferId->listener];
+    deathNotifications.indices[bufferId->frameIndex].emplace_back(bufferId->bufferIndex);
     ++deathNotifications.count;
     mOnBufferDestroyed.notify_one();
+
+    mTrackedBufferCache.erase(bufferId);
+    delete bufferId;
 }
 
 // Notify the clients about buffer destructions.
diff --git a/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h b/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
index b6857d5..42fa557 100644
--- a/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
+++ b/media/codec2/hidl/1.0/utils/include/codec2/hidl/1.0/InputBufferManager.h
@@ -196,13 +196,9 @@
                 frameIndex(frameIndex),
                 bufferIndex(bufferIndex),
                 buffer(buffer) {}
-        TrackedBuffer(const TrackedBuffer&) = default;
-        bool operator<(const TrackedBuffer& other) const {
-            return bufferIndex < other.bufferIndex;
-        }
     };
 
-    // Map: listener -> frameIndex -> set<TrackedBuffer>.
+    // Map: listener -> frameIndex -> set<TrackedBuffer*>.
     // Essentially, this is used to store triples (listener, frameIndex,
     // bufferIndex) that's searchable by listener and (listener, frameIndex).
     // However, the value of the innermost map is TrackedBuffer, which also
@@ -210,7 +206,7 @@
     // because onBufferDestroyed() needs to know listener and frameIndex too.
     typedef std::map<wp<IComponentListener>,
                      std::map<uint64_t,
-                              std::set<TrackedBuffer>>> TrackedBuffersMap;
+                              std::set<TrackedBuffer*>>> TrackedBuffersMap;
 
     // Storage for pending (unsent) death notifications for one listener.
     // Each pair in member named "indices" are (frameIndex, bufferIndex) from
@@ -247,6 +243,16 @@
     // Mutex for the management of all input buffers.
     std::mutex mMutex;
 
+    // Cache for all TrackedBuffers.
+    //
+    // Whenever registerOnDestroyNotify() is called, an argument of type
+    // TrackedBuffer is created and stored into this cache.
+    // Whenever unregisterOnDestroyNotify() or onBufferDestroyed() is called,
+    // the TrackedBuffer is removed from this cache.
+    //
+    // mTrackedBuffersMap stores references to TrackedBuffers inside this cache.
+    std::set<TrackedBuffer*> mTrackedBufferCache;
+
     // Tracked input buffers.
     TrackedBuffersMap mTrackedBuffersMap;
 
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index 0cbf62b..1883225 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -29,6 +29,7 @@
 #include <android/hardware/cas/native/1.0/IDescrambler.h>
 #include <android-base/stringprintf.h>
 #include <binder/MemoryDealer.h>
+#include <cutils/properties.h>
 #include <gui/Surface.h>
 #include <media/openmax/OMX_Core.h>
 #include <media/stagefright/foundation/ABuffer.h>
@@ -1080,8 +1081,7 @@
                     outputGeneration);
         }
 
-        if (oStreamFormat.value == C2BufferData::LINEAR
-                && mComponentName.find("c2.qti.") == std::string::npos) {
+        if (oStreamFormat.value == C2BufferData::LINEAR) {
             // WORKAROUND: if we're using early CSD workaround we convert to
             //             array mode, to appease apps assuming the output
             //             buffers to be of the same size.
@@ -1133,8 +1133,9 @@
     }
 
     C2StreamBufferTypeSetting::output oStreamFormat(0u);
-    c2_status_t err = mComponent->query({ &oStreamFormat }, {}, C2_DONT_BLOCK, nullptr);
-    if (err != C2_OK) {
+    C2PrependHeaderModeSetting prepend(PREPEND_HEADER_TO_NONE);
+    c2_status_t err = mComponent->query({ &oStreamFormat, &prepend }, {}, C2_DONT_BLOCK, nullptr);
+    if (err != C2_OK && err != C2_BAD_INDEX) {
         return UNKNOWN_ERROR;
     }
     size_t numInputSlots = mInput.lock()->numSlots;
@@ -1174,7 +1175,7 @@
                             mName, buffer->capacity(), config->size());
                 }
             } else if (oStreamFormat.value == C2BufferData::LINEAR && i == 0
-                    && mComponentName.find("c2.qti.") == std::string::npos) {
+                        && (!prepend || prepend.value == PREPEND_HEADER_TO_NONE)) {
                 // WORKAROUND: Some apps expect CSD available without queueing
                 //             any input. Queue an empty buffer to get the CSD.
                 buffer->setRange(0, 0);
diff --git a/media/codec2/sfplugin/Codec2InfoBuilder.cpp b/media/codec2/sfplugin/Codec2InfoBuilder.cpp
index 6b75eba..745d701 100644
--- a/media/codec2/sfplugin/Codec2InfoBuilder.cpp
+++ b/media/codec2/sfplugin/Codec2InfoBuilder.cpp
@@ -117,8 +117,9 @@
         }
     }
 
-    // For VP9, the static info is always propagated by framework.
+    // For VP9/AV1, the static info is always propagated by framework.
     supportsHdr |= (mediaType == MIMETYPE_VIDEO_VP9);
+    supportsHdr |= (mediaType == MIMETYPE_VIDEO_AV1);
 
     for (C2Value::Primitive profile : profileQuery[0].values.values) {
         pl.profile = (C2Config::profile_t)profile.ref<uint32_t>();
diff --git a/media/codec2/sfplugin/utils/Codec2Mapper.cpp b/media/codec2/sfplugin/utils/Codec2Mapper.cpp
index 40160c7..ef6af48 100644
--- a/media/codec2/sfplugin/utils/Codec2Mapper.cpp
+++ b/media/codec2/sfplugin/utils/Codec2Mapper.cpp
@@ -382,10 +382,11 @@
     // TODO: will need to disambiguate between Main8 and Main10
     { C2Config::PROFILE_AV1_0, AV1ProfileMain8 },
     { C2Config::PROFILE_AV1_0, AV1ProfileMain10 },
+    { C2Config::PROFILE_AV1_0, AV1ProfileMain10HDR10 },
+    { C2Config::PROFILE_AV1_0, AV1ProfileMain10HDR10Plus },
 };
 
 ALookup<C2Config::profile_t, int32_t> sAv1HdrProfiles = {
-    { C2Config::PROFILE_AV1_0, AV1ProfileMain10 },
     { C2Config::PROFILE_AV1_0, AV1ProfileMain10HDR10 },
 };
 
@@ -629,7 +630,7 @@
 // static
 std::shared_ptr<C2Mapper::ProfileLevelMapper>
 C2Mapper::GetProfileLevelMapper(std::string mediaType) {
-    std::transform(mediaType.begin(), mediaType.begin(), mediaType.end(), ::tolower);
+    std::transform(mediaType.begin(), mediaType.end(), mediaType.begin(), ::tolower);
     if (mediaType == MIMETYPE_AUDIO_AAC) {
         return std::make_shared<AacProfileLevelMapper>();
     } else if (mediaType == MIMETYPE_VIDEO_AVC) {
@@ -657,11 +658,13 @@
 // static
 std::shared_ptr<C2Mapper::ProfileLevelMapper>
 C2Mapper::GetHdrProfileLevelMapper(std::string mediaType, bool isHdr10Plus) {
-    std::transform(mediaType.begin(), mediaType.begin(), mediaType.end(), ::tolower);
+    std::transform(mediaType.begin(), mediaType.end(), mediaType.begin(), ::tolower);
     if (mediaType == MIMETYPE_VIDEO_HEVC) {
         return std::make_shared<HevcProfileLevelMapper>(true, isHdr10Plus);
     } else if (mediaType == MIMETYPE_VIDEO_VP9) {
         return std::make_shared<Vp9ProfileLevelMapper>(true, isHdr10Plus);
+    } else if (mediaType == MIMETYPE_VIDEO_AV1) {
+        return std::make_shared<Av1ProfileLevelMapper>(true, isHdr10Plus);
     }
     return nullptr;
 }
diff --git a/media/extractors/amr/AMRExtractor.cpp b/media/extractors/amr/AMRExtractor.cpp
index ffeff42..26431a4 100644
--- a/media/extractors/amr/AMRExtractor.cpp
+++ b/media/extractors/amr/AMRExtractor.cpp
@@ -144,6 +144,7 @@
 
 AMRExtractor::AMRExtractor(DataSourceHelper *source)
     : mDataSource(source),
+      mMeta(NULL),
       mInitCheck(NO_INIT),
       mOffsetTableLength(0) {
     float confidence;
@@ -191,7 +192,9 @@
 
 AMRExtractor::~AMRExtractor() {
     delete mDataSource;
-    AMediaFormat_delete(mMeta);
+    if (mMeta) {
+        AMediaFormat_delete(mMeta);
+    }
 }
 
 media_status_t AMRExtractor::getMetaData(AMediaFormat *meta) {
diff --git a/media/extractors/mkv/Android.bp b/media/extractors/mkv/Android.bp
index 1744d3d..38821fd 100644
--- a/media/extractors/mkv/Android.bp
+++ b/media/extractors/mkv/Android.bp
@@ -12,10 +12,10 @@
     shared_libs: [
         "liblog",
         "libmediandk",
+        "libstagefright_flacdec",
     ],
 
     static_libs: [
-        "libstagefright_flacdec",
         "libstagefright_foundation",
         "libstagefright_metadatautils",
         "libwebm",
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index 9d5890c..b91d16f 100755
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -4993,8 +4993,11 @@
 }
 
 status_t MPEG4Source::parseSampleAuxiliaryInformationSizes(
-        off64_t offset, off64_t /* size */) {
+        off64_t offset, off64_t size) {
     ALOGV("parseSampleAuxiliaryInformationSizes");
+    if (size < 9) {
+        return -EINVAL;
+    }
     // 14496-12 8.7.12
     uint8_t version;
     if (mDataSource->readAt(
@@ -5007,25 +5010,32 @@
         return ERROR_UNSUPPORTED;
     }
     offset++;
+    size--;
 
     uint32_t flags;
     if (!mDataSource->getUInt24(offset, &flags)) {
         return ERROR_IO;
     }
     offset += 3;
+    size -= 3;
 
     if (flags & 1) {
+        if (size < 13) {
+            return -EINVAL;
+        }
         uint32_t tmp;
         if (!mDataSource->getUInt32(offset, &tmp)) {
             return ERROR_MALFORMED;
         }
         mCurrentAuxInfoType = tmp;
         offset += 4;
+        size -= 4;
         if (!mDataSource->getUInt32(offset, &tmp)) {
             return ERROR_MALFORMED;
         }
         mCurrentAuxInfoTypeParameter = tmp;
         offset += 4;
+        size -= 4;
     }
 
     uint8_t defsize;
@@ -5034,6 +5044,7 @@
     }
     mCurrentDefaultSampleInfoSize = defsize;
     offset++;
+    size--;
 
     uint32_t smplcnt;
     if (!mDataSource->getUInt32(offset, &smplcnt)) {
@@ -5041,11 +5052,16 @@
     }
     mCurrentSampleInfoCount = smplcnt;
     offset += 4;
-
+    size -= 4;
     if (mCurrentDefaultSampleInfoSize != 0) {
         ALOGV("@@@@ using default sample info size of %d", mCurrentDefaultSampleInfoSize);
         return OK;
     }
+    if(smplcnt > size) {
+        ALOGW("b/124525515 - smplcnt(%u) > size(%ld)", (unsigned int)smplcnt, (unsigned long)size);
+        android_errorWriteLog(0x534e4554, "124525515");
+        return -EINVAL;
+    }
     if (smplcnt > mCurrentSampleInfoAllocSize) {
         uint8_t * newPtr =  (uint8_t*) realloc(mCurrentSampleInfoSizes, smplcnt);
         if (newPtr == NULL) {
@@ -5061,26 +5077,32 @@
 }
 
 status_t MPEG4Source::parseSampleAuxiliaryInformationOffsets(
-        off64_t offset, off64_t /* size */) {
+        off64_t offset, off64_t size) {
     ALOGV("parseSampleAuxiliaryInformationOffsets");
+    if (size < 8) {
+        return -EINVAL;
+    }
     // 14496-12 8.7.13
     uint8_t version;
     if (mDataSource->readAt(offset, &version, sizeof(version)) != 1) {
         return ERROR_IO;
     }
     offset++;
+    size--;
 
     uint32_t flags;
     if (!mDataSource->getUInt24(offset, &flags)) {
         return ERROR_IO;
     }
     offset += 3;
+    size -= 3;
 
     uint32_t entrycount;
     if (!mDataSource->getUInt32(offset, &entrycount)) {
         return ERROR_IO;
     }
     offset += 4;
+    size -= 4;
     if (entrycount == 0) {
         return OK;
     }
@@ -5106,19 +5128,31 @@
 
     for (size_t i = 0; i < entrycount; i++) {
         if (version == 0) {
+            if (size < 4) {
+                ALOGW("b/124526959");
+                android_errorWriteLog(0x534e4554, "124526959");
+                return -EINVAL;
+            }
             uint32_t tmp;
             if (!mDataSource->getUInt32(offset, &tmp)) {
                 return ERROR_IO;
             }
             mCurrentSampleInfoOffsets[i] = tmp;
             offset += 4;
+            size -= 4;
         } else {
+            if (size < 8) {
+                ALOGW("b/124526959");
+                android_errorWriteLog(0x534e4554, "124526959");
+                return -EINVAL;
+            }
             uint64_t tmp;
             if (!mDataSource->getUInt64(offset, &tmp)) {
                 return ERROR_IO;
             }
             mCurrentSampleInfoOffsets[i] = tmp;
             offset += 8;
+            size -= 8;
         }
     }
 
@@ -5405,20 +5439,30 @@
 
     if (flags & kSampleSizePresent) {
         bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleSizePresent) {
-        sampleSize = mTrackFragmentHeaderInfo.mDefaultSampleSize;
     } else {
         sampleSize = mTrackFragmentHeaderInfo.mDefaultSampleSize;
+#ifdef VERY_VERY_VERBOSE_LOGGING
+        // We don't expect this, but also want to avoid spamming the log if
+        // we hit this case.
+        if (!(mTrackFragmentHeaderInfo.mFlags
+              & TrackFragmentHeaderInfo::kDefaultSampleSizePresent)) {
+            ALOGW("No sample size specified");
+        }
+#endif
     }
 
     if (flags & kSampleFlagsPresent) {
         bytesPerSample += 4;
-    } else if (mTrackFragmentHeaderInfo.mFlags
-            & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent) {
-        sampleFlags = mTrackFragmentHeaderInfo.mDefaultSampleFlags;
     } else {
         sampleFlags = mTrackFragmentHeaderInfo.mDefaultSampleFlags;
+#ifdef VERY_VERY_VERBOSE_LOGGING
+        // We don't expect this, but also want to avoid spamming the log if
+        // we hit this case.
+        if (!(mTrackFragmentHeaderInfo.mFlags
+              & TrackFragmentHeaderInfo::kDefaultSampleFlagsPresent)) {
+            ALOGW("No sample flags specified");
+        }
+#endif
     }
 
     if (flags & kSampleCompositionTimeOffsetPresent) {
@@ -5440,16 +5484,12 @@
 
         // apply some sanity (vs strict legality) checks
         //
-        // clamp the count of entries in the trun box, to avoid spending forever parsing
-        // this box. Clamping (vs error) lets us play *something*.
-        // 1 million is about 400 msecs on a Pixel3, should be no more than a couple seconds
-        // on the slowest devices.
-        static constexpr uint32_t kMaxTrunSampleCount = 1000000;
+        static constexpr uint32_t kMaxTrunSampleCount = 10000;
         if (sampleCount > kMaxTrunSampleCount) {
-            ALOGW("b/123389881 clamp sampleCount(%u) @ kMaxTrunSampleCount(%u)",
+            ALOGW("b/123389881 sampleCount(%u) > kMaxTrunSampleCount(%u)",
                   sampleCount, kMaxTrunSampleCount);
             android_errorWriteLog(0x534e4554, "124389881 count");
-
+            return -EINVAL;
         }
     }
 
@@ -5493,7 +5533,12 @@
         tmp.duration = sampleDuration;
         tmp.compositionOffset = sampleCtsOffset;
         memset(tmp.iv, 0, sizeof(tmp.iv));
-        mCurrentSamples.add(tmp);
+        if (mCurrentSamples.add(tmp) < 0) {
+            ALOGW("b/123389881 failed saving sample(n=%zu)", mCurrentSamples.size());
+            android_errorWriteLog(0x534e4554, "124389881 allocation");
+            mCurrentSamples.clear();
+            return NO_MEMORY;
+        }
 
         dataOffset += sampleSize;
     }
@@ -5775,11 +5820,11 @@
                       meta, AMEDIAFORMAT_KEY_TIME_US, ((long double)cts * 1000000) / mTimescale);
                 AMediaFormat_setInt32(meta, AMEDIAFORMAT_KEY_IS_SYNC_FRAME, 1);
 
-                int32_t byteOrder;
-                AMediaFormat_getInt32(mFormat,
+                int32_t byteOrder = 0;
+                bool isGetBigEndian = AMediaFormat_getInt32(mFormat,
                         AMEDIAFORMAT_KEY_PCM_BIG_ENDIAN, &byteOrder);
 
-                if (byteOrder == 1) {
+                if (isGetBigEndian && byteOrder == 1) {
                     // Big-endian -> little-endian
                     uint16_t *dstData = (uint16_t *)buf;
                     uint16_t *srcData = (uint16_t *)buf;
diff --git a/media/extractors/mp4/SampleTable.cpp b/media/extractors/mp4/SampleTable.cpp
index bf29bf1..59c8200 100644
--- a/media/extractors/mp4/SampleTable.cpp
+++ b/media/extractors/mp4/SampleTable.cpp
@@ -391,20 +391,11 @@
     }
 
     mTimeToSampleCount = U32_AT(&header[4]);
-    if (mTimeToSampleCount > UINT32_MAX / (2 * sizeof(uint32_t))) {
-        // Choose this bound because
-        // 1) 2 * sizeof(uint32_t) is the amount of memory needed for one
-        //    time-to-sample entry in the time-to-sample table.
-        // 2) mTimeToSampleCount is the number of entries of the time-to-sample
-        //    table.
-        // 3) We hope that the table size does not exceed UINT32_MAX.
+    if (mTimeToSampleCount > (data_size - 8) / (2 * sizeof(uint32_t))) {
         ALOGE("Time-to-sample table size too large.");
         return ERROR_OUT_OF_RANGE;
     }
 
-    // Note: At this point, we know that mTimeToSampleCount * 2 will not
-    // overflow because of the above condition.
-
     uint64_t allocSize = (uint64_t)mTimeToSampleCount * 2 * sizeof(uint32_t);
     mTotalSize += allocSize;
     if (mTotalSize > kMaxTotalSize) {
@@ -540,6 +531,12 @@
     }
 
     uint64_t allocSize = (uint64_t)numSyncSamples * sizeof(uint32_t);
+    if (allocSize > data_size - 8) {
+        ALOGW("b/124771364 - allocSize(%lu) > size(%lu)",
+                (unsigned long)allocSize, (unsigned long)(data_size - 8));
+        android_errorWriteLog(0x534e4554, "124771364");
+        return ERROR_MALFORMED;
+    }
     if (allocSize > kMaxTotalSize) {
         ALOGE("Sync sample table size too large.");
         return ERROR_OUT_OF_RANGE;
@@ -655,6 +652,7 @@
     }
 
     mSampleTimeEntries = new (std::nothrow) SampleTimeEntry[mNumSampleSizes];
+    memset(mSampleTimeEntries, 0, sizeof(SampleTimeEntry) * mNumSampleSizes);
     if (!mSampleTimeEntries) {
         ALOGE("Cannot allocate sample entry table with %llu entries.",
                 (unsigned long long)mNumSampleSizes);
diff --git a/media/extractors/ogg/OggExtractor.cpp b/media/extractors/ogg/OggExtractor.cpp
index 72b94bb..4012ece 100644
--- a/media/extractors/ogg/OggExtractor.cpp
+++ b/media/extractors/ogg/OggExtractor.cpp
@@ -1062,8 +1062,15 @@
     size_t size = buffer->range_length();
 
     if (size < kOpusHeaderSize
-            || memcmp(data, "OpusHead", 8)
-            || /* version = */ data[8] != 1) {
+            || memcmp(data, "OpusHead", 8)) {
+        return AMEDIA_ERROR_MALFORMED;
+    }
+    // allow both version 0 and 1. Per the opus specification:
+    // An earlier draft of the specification described a version 0, but the only difference
+    // between version 1 and version 0 is that version 0 did not specify the semantics for
+    // handling the version field
+    if ( /* version = */ data[8] > 1) {
+        ALOGW("no support for opus version %d", data[8]);
         return AMEDIA_ERROR_MALFORMED;
     }
 
@@ -1384,7 +1391,7 @@
         return NULL;
     }
 
-    *confidence = 0.2f;
+    *confidence = 0.5f;
 
     return CreateExtractor;
 }
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index fb276c2..52eadd4 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -36,7 +36,6 @@
 #include "binding/AAudioStreamConfiguration.h"
 #include "binding/IAAudioService.h"
 #include "binding/AAudioServiceMessage.h"
-#include "core/AudioGlobal.h"
 #include "core/AudioStreamBuilder.h"
 #include "fifo/FifoBuffer.h"
 #include "utility/AudioClock.h"
diff --git a/media/libaudioclient/AudioProductStrategy.cpp b/media/libaudioclient/AudioProductStrategy.cpp
index 0e1dfac..cff72fd 100644
--- a/media/libaudioclient/AudioProductStrategy.cpp
+++ b/media/libaudioclient/AudioProductStrategy.cpp
@@ -70,6 +70,7 @@
     return NO_ERROR;
 }
 
+// Keep in sync with android/media/audiopolicy/AudioProductStrategy#attributeMatches
 bool AudioProductStrategy::attributesMatches(const audio_attributes_t refAttributes,
                                         const audio_attributes_t clientAttritubes)
 {
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index dd95e34..750fc21 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -340,11 +340,11 @@
         return reply.readInt32();
     }
 
-    virtual void setRecordSilenced(uid_t uid, bool silenced)
+    virtual void setRecordSilenced(audio_port_handle_t portId, bool silenced)
     {
         Parcel data, reply;
         data.writeInterfaceToken(IAudioFlinger::getInterfaceDescriptor());
-        data.writeInt32(uid);
+        data.writeInt32(portId);
         data.writeInt32(silenced ? 1 : 0);
         remote()->transact(SET_RECORD_SILENCED, data, &reply);
     }
@@ -1156,11 +1156,9 @@
         } break;
         case SET_RECORD_SILENCED: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            uid_t uid = data.readInt32();
-            audio_source_t source;
-            data.read(&source, sizeof(audio_source_t));
+            audio_port_handle_t portId = data.readInt32();
             bool silenced = data.readInt32() == 1;
-            setRecordSilenced(uid, silenced);
+            setRecordSilenced(portId, silenced);
             return NO_ERROR;
         } break;
         case SET_PARAMETERS: {
@@ -1339,10 +1337,14 @@
         }
         case GET_EFFECT_DESCRIPTOR: {
             CHECK_INTERFACE(IAudioFlinger, data, reply);
-            effect_uuid_t uuid;
-            data.read(&uuid, sizeof(effect_uuid_t));
-            effect_uuid_t type;
-            data.read(&type, sizeof(effect_uuid_t));
+            effect_uuid_t uuid = {};
+            if (data.read(&uuid, sizeof(effect_uuid_t)) != NO_ERROR) {
+                android_errorWriteLog(0x534e4554, "139417189");
+            }
+            effect_uuid_t type = {};
+            if (data.read(&type, sizeof(effect_uuid_t)) != NO_ERROR) {
+                android_errorWriteLog(0x534e4554, "139417189");
+            }
             uint32_t preferredTypeFlag = data.readUint32();
             effect_descriptor_t desc = {};
             status_t status = getEffectDescriptor(&uuid, &type, preferredTypeFlag, &desc);
diff --git a/media/libaudioclient/include/media/AudioMixer.h b/media/libaudioclient/include/media/AudioMixer.h
deleted file mode 100644
index 783eef3..0000000
--- a/media/libaudioclient/include/media/AudioMixer.h
+++ /dev/null
@@ -1,519 +0,0 @@
-/*
-**
-** Copyright 2007, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-**     http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-#ifndef ANDROID_AUDIO_MIXER_H
-#define ANDROID_AUDIO_MIXER_H
-
-#include <map>
-#include <pthread.h>
-#include <sstream>
-#include <stdint.h>
-#include <sys/types.h>
-#include <unordered_map>
-#include <vector>
-
-#include <android/os/IExternalVibratorService.h>
-#include <media/AudioBufferProvider.h>
-#include <media/AudioResampler.h>
-#include <media/AudioResamplerPublic.h>
-#include <media/BufferProviders.h>
-#include <system/audio.h>
-#include <utils/Compat.h>
-#include <utils/threads.h>
-
-// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
-#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
-
-// This must match frameworks/av/services/audioflinger/Configuration.h
-#define FLOAT_AUX
-
-namespace android {
-
-namespace NBLog {
-class Writer;
-}   // namespace NBLog
-
-// ----------------------------------------------------------------------------
-
-class AudioMixer
-{
-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_VOLUMES = FCC_2; // stereo volume only
-    // maximum number of channels supported for the content
-    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
-
-    static const uint16_t UNITY_GAIN_INT = 0x1000;
-    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
-
-    enum { // names
-        // setParameter targets
-        TRACK           = 0x3000,
-        RESAMPLE        = 0x3001,
-        RAMP_VOLUME     = 0x3002, // ramp to new volume
-        VOLUME          = 0x3003, // don't ramp
-        TIMESTRETCH     = 0x3004,
-
-        // set Parameter names
-        // for target TRACK
-        CHANNEL_MASK    = 0x4000,
-        FORMAT          = 0x4001,
-        MAIN_BUFFER     = 0x4002,
-        AUX_BUFFER      = 0x4003,
-        DOWNMIX_TYPE    = 0X4004,
-        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
-        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
-        // for haptic
-        HAPTIC_ENABLED  = 0x4007, // Set haptic data from this track should be played or not.
-        HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
-        // for target RESAMPLE
-        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
-                                  // parameter 'value' is the new sample rate in Hz.
-                                  // Only creates a sample rate converter the first time that
-                                  // the track sample rate is different from the mix sample rate.
-                                  // If the new sample rate is the same as the mix sample rate,
-                                  // and a sample rate converter already exists,
-                                  // then the sample rate converter remains present but is a no-op.
-        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
-                                  // This clears out the resampler's input buffer.
-        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
-                                  // the track is restored to the mix sample rate.
-        // for target RAMP_VOLUME and VOLUME (8 channels max)
-        // FIXME use float for these 3 to improve the dynamic range
-        VOLUME0         = 0x4200,
-        VOLUME1         = 0x4201,
-        AUXLEVEL        = 0x4210,
-        // for target TIMESTRETCH
-        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
-                                  // parameter 'value' is a pointer to the new playback rate.
-    };
-
-    typedef enum { // Haptic intensity, should keep consistent with VibratorService
-        HAPTIC_SCALE_MUTE = os::IExternalVibratorService::SCALE_MUTE,
-        HAPTIC_SCALE_VERY_LOW = os::IExternalVibratorService::SCALE_VERY_LOW,
-        HAPTIC_SCALE_LOW = os::IExternalVibratorService::SCALE_LOW,
-        HAPTIC_SCALE_NONE = os::IExternalVibratorService::SCALE_NONE,
-        HAPTIC_SCALE_HIGH = os::IExternalVibratorService::SCALE_HIGH,
-        HAPTIC_SCALE_VERY_HIGH = os::IExternalVibratorService::SCALE_VERY_HIGH,
-    } haptic_intensity_t;
-    static constexpr float HAPTIC_SCALE_VERY_LOW_RATIO = 2.0f / 3.0f;
-    static constexpr float HAPTIC_SCALE_LOW_RATIO = 3.0f / 4.0f;
-    static const constexpr float HAPTIC_MAX_AMPLITUDE_FLOAT = 1.0f;
-
-    static inline bool isValidHapticIntensity(haptic_intensity_t hapticIntensity) {
-        switch (hapticIntensity) {
-        case HAPTIC_SCALE_MUTE:
-        case HAPTIC_SCALE_VERY_LOW:
-        case HAPTIC_SCALE_LOW:
-        case HAPTIC_SCALE_NONE:
-        case HAPTIC_SCALE_HIGH:
-        case HAPTIC_SCALE_VERY_HIGH:
-            return true;
-        default:
-            return false;
-        }
-    }
-
-    AudioMixer(size_t frameCount, uint32_t sampleRate)
-        : mSampleRate(sampleRate)
-        , mFrameCount(frameCount) {
-        pthread_once(&sOnceControl, &sInitRoutine);
-    }
-
-    // Create a new track in the mixer.
-    //
-    // \param name        a unique user-provided integer associated with the track.
-    //                    If name already exists, the function will abort.
-    // \param channelMask output channel mask.
-    // \param format      PCM format
-    // \param sessionId   Session id for the track. Tracks with the same
-    //                    session id will be submixed together.
-    //
-    // \return OK        on success.
-    //         BAD_VALUE if the format does not satisfy isValidFormat()
-    //                   or the channelMask does not satisfy isValidChannelMask().
-    status_t    create(
-            int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
-
-    bool        exists(int name) const {
-        return mTracks.count(name) > 0;
-    }
-
-    // Free an allocated track by name.
-    void        destroy(int name);
-
-    // Enable or disable an allocated track by name
-    void        enable(int name);
-    void        disable(int name);
-
-    void        setParameter(int name, int target, int param, void *value);
-
-    void        setBufferProvider(int name, AudioBufferProvider* bufferProvider);
-
-    void        process() {
-        for (const auto &pair : mTracks) {
-            // Clear contracted buffer before processing if contracted channels are saved
-            const std::shared_ptr<Track> &t = pair.second;
-            if (t->mKeepContractedChannels) {
-                t->clearContractedBuffer();
-            }
-        }
-        (this->*mHook)();
-        processHapticData();
-    }
-
-    size_t      getUnreleasedFrames(int name) const;
-
-    std::string trackNames() const {
-        std::stringstream ss;
-        for (const auto &pair : mTracks) {
-            ss << pair.first << " ";
-        }
-        return ss.str();
-    }
-
-    void        setNBLogWriter(NBLog::Writer *logWriter) {
-        mNBLogWriter = logWriter;
-    }
-
-    static inline bool isValidFormat(audio_format_t format) {
-        switch (format) {
-        case AUDIO_FORMAT_PCM_8_BIT:
-        case AUDIO_FORMAT_PCM_16_BIT:
-        case AUDIO_FORMAT_PCM_24_BIT_PACKED:
-        case AUDIO_FORMAT_PCM_32_BIT:
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return true;
-        default:
-            return false;
-        }
-    }
-
-    static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
-        return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
-    }
-
-private:
-
-    /* For multi-format functions (calls template functions
-     * in AudioMixerOps.h).  The template parameters are as follows:
-     *
-     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
-     *   USEFLOATVOL (set to true if float volume is used)
-     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
-     *   TO: int32_t (Q4.27) or float
-     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
-     *   TA: int32_t (Q4.27)
-     */
-
-    enum {
-        // FIXME this representation permits up to 8 channels
-        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
-    };
-
-    enum {
-        NEEDS_CHANNEL_1             = 0x00000000,   // mono
-        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
-
-        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
-
-        NEEDS_MUTE                  = 0x00000100,
-        NEEDS_RESAMPLE              = 0x00001000,
-        NEEDS_AUX                   = 0x00010000,
-    };
-
-    // hook types
-    enum {
-        PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
-    };
-
-    enum {
-        TRACKTYPE_NOP,
-        TRACKTYPE_RESAMPLE,
-        TRACKTYPE_NORESAMPLE,
-        TRACKTYPE_NORESAMPLEMONO,
-    };
-
-    // process hook functionality
-    using process_hook_t = void(AudioMixer::*)();
-
-    struct Track;
-    using hook_t = void(Track::*)(int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
-
-    struct Track {
-        Track()
-            : bufferProvider(nullptr)
-        {
-            // TODO: move additional initialization here.
-        }
-
-        ~Track()
-        {
-            // bufferProvider, mInputBufferProvider need not be deleted.
-            mResampler.reset(nullptr);
-            // Ensure the order of destruction of buffer providers as they
-            // release the upstream provider in the destructor.
-            mTimestretchBufferProvider.reset(nullptr);
-            mPostDownmixReformatBufferProvider.reset(nullptr);
-            mDownmixerBufferProvider.reset(nullptr);
-            mReformatBufferProvider.reset(nullptr);
-            mContractChannelsNonDestructiveBufferProvider.reset(nullptr);
-            mAdjustChannelsBufferProvider.reset(nullptr);
-        }
-
-        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
-        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
-        bool        doesResample() const { return mResampler.get() != nullptr; }
-        void        resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
-        void        adjustVolumeRamp(bool aux, bool useFloat = false);
-        size_t      getUnreleasedFrames() const { return mResampler.get() != nullptr ?
-                                                    mResampler->getUnreleasedFrames() : 0; };
-
-        status_t    prepareForDownmix();
-        void        unprepareForDownmix();
-        status_t    prepareForReformat();
-        void        unprepareForReformat();
-        status_t    prepareForAdjustChannels();
-        void        unprepareForAdjustChannels();
-        status_t    prepareForAdjustChannelsNonDestructive(size_t frames);
-        void        unprepareForAdjustChannelsNonDestructive();
-        void        clearContractedBuffer();
-        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
-        void        reconfigureBufferProviders();
-
-        static hook_t getTrackHook(int trackType, uint32_t channelCount,
-                audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
-        void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
-        template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
-            typename TO, typename TI, typename TA>
-        void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
-
-        uint32_t    needs;
-
-        // TODO: Eventually remove legacy integer volume settings
-        union {
-        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
-        int32_t     volumeRL;
-        };
-
-        int32_t     prevVolume[MAX_NUM_VOLUMES];
-        int32_t     volumeInc[MAX_NUM_VOLUMES];
-        int32_t     auxInc;
-        int32_t     prevAuxLevel;
-        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
-
-        uint16_t    frameCount;
-
-        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
-        uint8_t     unused_padding; // formerly format, was always 16
-        uint16_t    enabled;        // actually bool
-        audio_channel_mask_t channelMask;
-
-        // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
-        //  for how the Track buffer provider is wrapped by another one when dowmixing is required
-        AudioBufferProvider*                bufferProvider;
-
-        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
-
-        hook_t      hook;
-        const void  *mIn;             // current location in buffer
-
-        std::unique_ptr<AudioResampler> mResampler;
-        uint32_t            sampleRate;
-        int32_t*           mainBuffer;
-        int32_t*           auxBuffer;
-
-        /* Buffer providers are constructed to translate the track input data as needed.
-         *
-         * TODO: perhaps make a single PlaybackConverterProvider class to move
-         * all pre-mixer track buffer conversions outside the AudioMixer class.
-         *
-         * 1) mInputBufferProvider: The AudioTrack buffer provider.
-         * 2) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved
-         *    channel format to another. Expanded channels are filled with zeros and put at the end
-         *    of each audio frame. Contracted channels are copied to the end of the buffer.
-         * 3) mContractChannelsNonDestructiveBufferProvider: Non-destructively contract sample data.
-         *    This is currently using at audio-haptic coupled playback to separate audio and haptic
-         *    data. Contracted channels could be written to given buffer.
-         * 4) mReformatBufferProvider: If not NULL, performs the audio reformat to
-         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
-         *    requires reformat. For example, it may convert floating point input to
-         *    PCM_16_bit if that's required by the downmixer.
-         * 5) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
-         *    the number of channels required by the mixer sink.
-         * 6) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
-         *    the downmixer requirements to the mixer engine input requirements.
-         * 7) mTimestretchBufferProvider: Adds timestretching for playback rate
-         */
-        AudioBufferProvider*     mInputBufferProvider;    // externally provided buffer provider.
-        // TODO: combine mAdjustChannelsBufferProvider and
-        // mContractChannelsNonDestructiveBufferProvider
-        std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mContractChannelsNonDestructiveBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
-        std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
-
-        int32_t     sessionId;
-
-        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
-        audio_format_t mFormat;          // input track format
-        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
-                                         // each track must be converted to this format.
-        audio_format_t mDownmixRequiresFormat;  // required downmixer format
-                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
-                                                // AUDIO_FORMAT_INVALID if no required format
-
-        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
-        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
-        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
-
-        float          mAuxLevel;                     // floating point set aux level
-        float          mPrevAuxLevel;                 // floating point prev aux level
-        float          mAuxInc;                       // floating point aux increment
-
-        audio_channel_mask_t mMixerChannelMask;
-        uint32_t             mMixerChannelCount;
-
-        AudioPlaybackRate    mPlaybackRate;
-
-        // Haptic
-        bool                 mHapticPlaybackEnabled;
-        haptic_intensity_t   mHapticIntensity;
-        audio_channel_mask_t mHapticChannelMask;
-        uint32_t             mHapticChannelCount;
-        audio_channel_mask_t mMixerHapticChannelMask;
-        uint32_t             mMixerHapticChannelCount;
-        uint32_t             mAdjustInChannelCount;
-        uint32_t             mAdjustOutChannelCount;
-        uint32_t             mAdjustNonDestructiveInChannelCount;
-        uint32_t             mAdjustNonDestructiveOutChannelCount;
-        bool                 mKeepContractedChannels;
-
-        float getHapticScaleGamma() const {
-        // Need to keep consistent with the value in VibratorService.
-        switch (mHapticIntensity) {
-        case HAPTIC_SCALE_VERY_LOW:
-            return 2.0f;
-        case HAPTIC_SCALE_LOW:
-            return 1.5f;
-        case HAPTIC_SCALE_HIGH:
-            return 0.5f;
-        case HAPTIC_SCALE_VERY_HIGH:
-            return 0.25f;
-        default:
-            return 1.0f;
-        }
-        }
-
-        float getHapticMaxAmplitudeRatio() const {
-        // Need to keep consistent with the value in VibratorService.
-        switch (mHapticIntensity) {
-        case HAPTIC_SCALE_VERY_LOW:
-            return HAPTIC_SCALE_VERY_LOW_RATIO;
-        case HAPTIC_SCALE_LOW:
-            return HAPTIC_SCALE_LOW_RATIO;
-        case HAPTIC_SCALE_NONE:
-        case HAPTIC_SCALE_HIGH:
-        case HAPTIC_SCALE_VERY_HIGH:
-            return 1.0f;
-        default:
-            return 0.0f;
-        }
-        }
-
-    private:
-        // hooks
-        void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-        void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-        void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
-        void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-        void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-
-        // multi-format track hooks
-        template <int MIXTYPE, typename TO, typename TI, typename TA>
-        void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
-        template <int MIXTYPE, typename TO, typename TI, typename TA>
-        void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
-    };
-
-    // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
-    static constexpr int BLOCKSIZE = 16;
-
-    bool setChannelMasks(int name,
-            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
-
-    // Called when track info changes and a new process hook should be determined.
-    void invalidate() {
-        mHook = &AudioMixer::process__validate;
-    }
-
-    void process__validate();
-    void process__nop();
-    void process__genericNoResampling();
-    void process__genericResampling();
-    void process__oneTrack16BitsStereoNoResampling();
-
-    template <int MIXTYPE, typename TO, typename TI, typename TA>
-    void process__noResampleOneTrack();
-
-    void processHapticData();
-
-    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
-            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
-    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
-            void *in, audio_format_t mixerInFormat, size_t sampleCount);
-
-    static void sInitRoutine();
-
-    // initialization constants
-    const uint32_t mSampleRate;
-    const size_t mFrameCount;
-
-    NBLog::Writer *mNBLogWriter = nullptr;   // associated NBLog::Writer
-
-    process_hook_t mHook = &AudioMixer::process__nop;   // one of process__*, never nullptr
-
-    // the size of the type (int32_t) should be the largest of all types supported
-    // by the mixer.
-    std::unique_ptr<int32_t[]> mOutputTemp;
-    std::unique_ptr<int32_t[]> mResampleTemp;
-
-    // track names grouped by main buffer, in no particular order of main buffer.
-    // however names for a particular main buffer are in order (by construction).
-    std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
-
-    // track names that are enabled, in increasing order (by construction).
-    std::vector<int /* name */> mEnabled;
-
-    // track smart pointers, by name, in increasing order of name.
-    std::map<int /* name */, std::shared_ptr<Track>> mTracks;
-
-    static pthread_once_t sOnceControl; // initialized in constructor by first new
-};
-
-// ----------------------------------------------------------------------------
-} // namespace android
-
-#endif // ANDROID_AUDIO_MIXER_H
diff --git a/media/libaudioclient/include/media/AudioParameter.h b/media/libaudioclient/include/media/AudioParameter.h
index 24837e3..3c190f2 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libaudioclient/include/media/AudioParameter.h
@@ -67,9 +67,12 @@
     //  keyAudioLanguagePreferred: Preferred audio language
     static const char * const keyAudioLanguagePreferred;
 
-    //  keyStreamConnect / Disconnect: value is an int in audio_devices_t
-    static const char * const keyStreamConnect;
-    static const char * const keyStreamDisconnect;
+    //  keyDeviceConnect / Disconnect: value is an int in audio_devices_t
+    static const char * const keyDeviceConnect;
+    static const char * const keyDeviceDisconnect;
+    //  Need to be here because vendors still use them.
+    static const char * const keyStreamConnect;  // Deprecated: DO NOT USE.
+    static const char * const keyStreamDisconnect;  // Deprecated: DO NOT USE.
 
     // For querying stream capabilities. All the returned values are lists.
     //   keyStreamSupportedFormats: audio_format_t
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 8ec8931..db09ddf 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -384,7 +384,7 @@
     // mic mute/state
     virtual     status_t    setMicMute(bool state) = 0;
     virtual     bool        getMicMute() const = 0;
-    virtual     void        setRecordSilenced(uid_t uid, bool silenced) = 0;
+    virtual     void        setRecordSilenced(audio_port_handle_t portId, bool silenced) = 0;
 
     virtual     status_t    setParameters(audio_io_handle_t ioHandle,
                                     const String8& keyValuePairs) = 0;
diff --git a/media/libaudiofoundation/Android.bp b/media/libaudiofoundation/Android.bp
new file mode 100644
index 0000000..c91b79e
--- /dev/null
+++ b/media/libaudiofoundation/Android.bp
@@ -0,0 +1,37 @@
+cc_library_headers {
+    name: "libaudiofoundation_headers",
+    vendor_available: true,
+    export_include_dirs: ["include"],
+}
+
+cc_library_shared {
+    name: "libaudiofoundation",
+    vendor_available: true,
+
+    srcs: [
+        "AudioGain.cpp",
+        "AudioPortBase.cpp",
+        "AudioProfile.cpp",
+    ],
+
+    shared_libs: [
+        "libbase",
+        "libbinder",
+        "liblog",
+        "libmedia_helper",
+        "libutils",
+    ],
+
+    header_libs: [
+        "libaudio_system_headers",
+        "libaudioclient_headers",
+        "libaudiofoundation_headers",
+    ],
+
+    export_header_lib_headers: ["libaudiofoundation_headers"],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
+}
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
new file mode 100644
index 0000000..9d1d6db
--- /dev/null
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -0,0 +1,174 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "AudioGain"
+//#define LOG_NDEBUG 0
+
+//#define VERY_VERBOSE_LOGGING
+#ifdef VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include <android-base/stringprintf.h>
+#include <media/AudioGain.h>
+#include <utils/Log.h>
+
+#include <math.h>
+
+namespace android {
+
+AudioGain::AudioGain(int index, bool useInChannelMask)
+{
+    mIndex = index;
+    mUseInChannelMask = useInChannelMask;
+    memset(&mGain, 0, sizeof(struct audio_gain));
+}
+
+void AudioGain::getDefaultConfig(struct audio_gain_config *config)
+{
+    config->index = mIndex;
+    config->mode = mGain.mode;
+    config->channel_mask = mGain.channel_mask;
+    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        config->values[0] = mGain.default_value;
+    } else {
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            config->values[i] = mGain.default_value;
+        }
+    }
+    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        config->ramp_duration_ms = mGain.min_ramp_ms;
+    }
+}
+
+status_t AudioGain::checkConfig(const struct audio_gain_config *config)
+{
+    if ((config->mode & ~mGain.mode) != 0) {
+        return BAD_VALUE;
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
+        if ((config->values[0] < mGain.min_value) ||
+                    (config->values[0] > mGain.max_value)) {
+            return BAD_VALUE;
+        }
+    } else {
+        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
+            return BAD_VALUE;
+        }
+        uint32_t numValues;
+        if (mUseInChannelMask) {
+            numValues = audio_channel_count_from_in_mask(config->channel_mask);
+        } else {
+            numValues = audio_channel_count_from_out_mask(config->channel_mask);
+        }
+        for (size_t i = 0; i < numValues; i++) {
+            if ((config->values[i] < mGain.min_value) ||
+                    (config->values[i] > mGain.max_value)) {
+                return BAD_VALUE;
+            }
+        }
+    }
+    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
+        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
+                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
+            return BAD_VALUE;
+        }
+    }
+    return NO_ERROR;
+}
+
+void AudioGain::dump(std::string *dst, int spaces, int index) const
+{
+    dst->append(base::StringPrintf("%*sGain %d:\n", spaces, "", index+1));
+    dst->append(base::StringPrintf("%*s- mode: %08x\n", spaces, "", mGain.mode));
+    dst->append(base::StringPrintf("%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask));
+    dst->append(base::StringPrintf("%*s- min_value: %d mB\n", spaces, "", mGain.min_value));
+    dst->append(base::StringPrintf("%*s- max_value: %d mB\n", spaces, "", mGain.max_value));
+    dst->append(base::StringPrintf("%*s- default_value: %d mB\n", spaces, "", mGain.default_value));
+    dst->append(base::StringPrintf("%*s- step_value: %d mB\n", spaces, "", mGain.step_value));
+    dst->append(base::StringPrintf("%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms));
+    dst->append(base::StringPrintf("%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms));
+}
+
+status_t AudioGain::writeToParcel(android::Parcel *parcel) const
+{
+    status_t status = NO_ERROR;
+    if ((status = parcel->writeInt32(mIndex)) != NO_ERROR) return status;
+    if ((status = parcel->writeBool(mUseInChannelMask)) != NO_ERROR) return status;
+    if ((status = parcel->writeBool(mUseForVolume)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.mode)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.channel_mask)) != NO_ERROR) return status;
+    if ((status = parcel->writeInt32(mGain.min_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeInt32(mGain.max_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeInt32(mGain.default_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.step_value)) != NO_ERROR) return status;
+    if ((status = parcel->writeUint32(mGain.min_ramp_ms)) != NO_ERROR) return status;
+    status = parcel->writeUint32(mGain.max_ramp_ms);
+    return status;
+}
+
+status_t AudioGain::readFromParcel(const android::Parcel *parcel)
+{
+    status_t status = NO_ERROR;
+    if ((status = parcel->readInt32(&mIndex)) != NO_ERROR) return status;
+    if ((status = parcel->readBool(&mUseInChannelMask)) != NO_ERROR) return status;
+    if ((status = parcel->readBool(&mUseForVolume)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.mode)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.channel_mask)) != NO_ERROR) return status;
+    if ((status = parcel->readInt32(&mGain.min_value)) != NO_ERROR) return status;
+    if ((status = parcel->readInt32(&mGain.max_value)) != NO_ERROR) return status;
+    if ((status = parcel->readInt32(&mGain.default_value)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.step_value)) != NO_ERROR) return status;
+    if ((status = parcel->readUint32(&mGain.min_ramp_ms)) != NO_ERROR) return status;
+    status = parcel->readUint32(&mGain.max_ramp_ms);
+    return status;
+}
+
+status_t AudioGains::writeToParcel(android::Parcel *parcel) const {
+    status_t status = NO_ERROR;
+    if ((status = parcel->writeUint64(this->size())) != NO_ERROR) return status;
+    for (const auto &audioGain : *this) {
+        if ((status = parcel->writeParcelable(*audioGain)) != NO_ERROR) {
+            break;
+        }
+    }
+    return status;
+}
+
+status_t AudioGains::readFromParcel(const android::Parcel *parcel) {
+    status_t status = NO_ERROR;
+    uint64_t count;
+    if ((status = parcel->readUint64(&count)) != NO_ERROR) return status;
+    for (uint64_t i = 0; i < count; i++) {
+        sp<AudioGain> audioGain = new AudioGain(0, false);
+        if ((status = parcel->readParcelable(audioGain.get())) != NO_ERROR) {
+            this->clear();
+            break;
+        }
+        this->push_back(audioGain);
+    }
+    return status;
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/AudioPortBase.cpp b/media/libaudiofoundation/AudioPortBase.cpp
new file mode 100644
index 0000000..922a82c
--- /dev/null
+++ b/media/libaudiofoundation/AudioPortBase.cpp
@@ -0,0 +1,87 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <algorithm>
+
+#include <android-base/stringprintf.h>
+#include <media/AudioPortBase.h>
+#include <utils/Log.h>
+
+namespace android {
+
+void AudioPortFoundation::toAudioPort(struct audio_port *port) const {
+    // TODO: update this function once audio_port structure reflects the new profile definition.
+    // For compatibility reason: flatening the AudioProfile into audio_port structure.
+    FormatSet flatenedFormats;
+    SampleRateSet flatenedRates;
+    ChannelMaskSet flatenedChannels;
+    for (const auto& profile : *getAudioProfileVectorBase()) {
+        if (profile->isValid()) {
+            audio_format_t formatToExport = profile->getFormat();
+            const SampleRateSet &ratesToExport = profile->getSampleRates();
+            const ChannelMaskSet &channelsToExport = profile->getChannels();
+
+            flatenedFormats.insert(formatToExport);
+            flatenedRates.insert(ratesToExport.begin(), ratesToExport.end());
+            flatenedChannels.insert(channelsToExport.begin(), channelsToExport.end());
+
+            if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
+                    flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
+                    flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
+                ALOGE("%s: bailing out: cannot export profiles to port config", __func__);
+                return;
+            }
+        }
+    }
+    port->role = mRole;
+    port->type = mType;
+    strlcpy(port->name, mName.c_str(), AUDIO_PORT_MAX_NAME_LEN);
+    port->num_sample_rates = flatenedRates.size();
+    port->num_channel_masks = flatenedChannels.size();
+    port->num_formats = flatenedFormats.size();
+    std::copy(flatenedRates.begin(), flatenedRates.end(), port->sample_rates);
+    std::copy(flatenedChannels.begin(), flatenedChannels.end(), port->channel_masks);
+    std::copy(flatenedFormats.begin(), flatenedFormats.end(), port->formats);
+
+    ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
+
+    port->num_gains = std::min(mGains.size(), (size_t) AUDIO_PORT_MAX_GAINS);
+    for (size_t i = 0; i < port->num_gains; i++) {
+        port->gains[i] = mGains[i]->getGain();
+    }
+}
+
+void AudioPortFoundation::dump(std::string *dst, int spaces, bool verbose) const {
+    if (!mName.empty()) {
+        dst->append(base::StringPrintf("%*s- name: %s\n", spaces, "", mName.c_str()));
+    }
+    if (verbose) {
+        std::string profilesStr;
+        getAudioProfileVectorBase()->dump(&profilesStr, spaces);
+        dst->append(profilesStr);
+
+        if (mGains.size() != 0) {
+            dst->append(base::StringPrintf("%*s- gains:\n", spaces, ""));
+            for (size_t i = 0; i < mGains.size(); i++) {
+                std::string gainStr;
+                mGains[i]->dump(&gainStr, spaces + 2, i);
+                dst->append(gainStr);
+            }
+        }
+    }
+}
+
+}
\ No newline at end of file
diff --git a/media/libaudiofoundation/AudioProfile.cpp b/media/libaudiofoundation/AudioProfile.cpp
new file mode 100644
index 0000000..aaaa7d1
--- /dev/null
+++ b/media/libaudiofoundation/AudioProfile.cpp
@@ -0,0 +1,222 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <set>
+
+#define LOG_TAG "AudioProfile"
+//#define LOG_NDEBUG 0
+
+#include <android-base/stringprintf.h>
+#include <media/AudioContainers.h>
+#include <media/AudioProfile.h>
+#include <media/TypeConverter.h>
+#include <utils/Errors.h>
+
+namespace android {
+
+bool operator == (const AudioProfile &left, const AudioProfile &right)
+{
+    return (left.getFormat() == right.getFormat()) &&
+            (left.getChannels() == right.getChannels()) &&
+            (left.getSampleRates() == right.getSampleRates());
+}
+
+// static
+sp<AudioProfile> AudioProfile::createFullDynamic(audio_format_t dynamicFormat)
+{
+    AudioProfile* dynamicProfile = new AudioProfile(dynamicFormat,
+            ChannelMaskSet(), SampleRateSet());
+    dynamicProfile->setDynamicFormat(true);
+    dynamicProfile->setDynamicChannels(true);
+    dynamicProfile->setDynamicRate(true);
+    return dynamicProfile;
+}
+
+AudioProfile::AudioProfile(audio_format_t format,
+                           audio_channel_mask_t channelMasks,
+                           uint32_t samplingRate) :
+        mName(""),
+        mFormat(format)
+{
+    mChannelMasks.insert(channelMasks);
+    mSamplingRates.insert(samplingRate);
+}
+
+AudioProfile::AudioProfile(audio_format_t format,
+                           const ChannelMaskSet &channelMasks,
+                           const SampleRateSet &samplingRateCollection) :
+        mName(""),
+        mFormat(format),
+        mChannelMasks(channelMasks),
+        mSamplingRates(samplingRateCollection) {}
+
+void AudioProfile::setChannels(const ChannelMaskSet &channelMasks)
+{
+    if (mIsDynamicChannels) {
+        mChannelMasks = channelMasks;
+    }
+}
+
+void AudioProfile::setSampleRates(const SampleRateSet &sampleRates)
+{
+    if (mIsDynamicRate) {
+        mSamplingRates = sampleRates;
+    }
+}
+
+void AudioProfile::clear()
+{
+    if (mIsDynamicChannels) {
+        mChannelMasks.clear();
+    }
+    if (mIsDynamicRate) {
+        mSamplingRates.clear();
+    }
+}
+
+void AudioProfile::dump(std::string *dst, int spaces) const
+{
+    dst->append(base::StringPrintf("%s%s%s\n", mIsDynamicFormat ? "[dynamic format]" : "",
+             mIsDynamicChannels ? "[dynamic channels]" : "",
+             mIsDynamicRate ? "[dynamic rates]" : ""));
+    if (mName.length() != 0) {
+        dst->append(base::StringPrintf("%*s- name: %s\n", spaces, "", mName.c_str()));
+    }
+    std::string formatLiteral;
+    if (FormatConverter::toString(mFormat, formatLiteral)) {
+        dst->append(base::StringPrintf("%*s- format: %s\n", spaces, "", formatLiteral.c_str()));
+    }
+    if (!mSamplingRates.empty()) {
+        dst->append(base::StringPrintf("%*s- sampling rates:", spaces, ""));
+        for (auto it = mSamplingRates.begin(); it != mSamplingRates.end();) {
+            dst->append(base::StringPrintf("%d", *it));
+            dst->append(++it == mSamplingRates.end() ? "" : ", ");
+        }
+        dst->append("\n");
+    }
+
+    if (!mChannelMasks.empty()) {
+        dst->append(base::StringPrintf("%*s- channel masks:", spaces, ""));
+        for (auto it = mChannelMasks.begin(); it != mChannelMasks.end();) {
+            dst->append(base::StringPrintf("0x%04x", *it));
+            dst->append(++it == mChannelMasks.end() ? "" : ", ");
+        }
+        dst->append("\n");
+    }
+}
+
+ssize_t AudioProfileVectorBase::add(const sp<AudioProfile> &profile)
+{
+    ssize_t index = size();
+    push_back(profile);
+    return index;
+}
+
+void AudioProfileVectorBase::clearProfiles()
+{
+    for (auto it = begin(); it != end();) {
+        if ((*it)->isDynamicFormat() && (*it)->hasValidFormat()) {
+            it = erase(it);
+        } else {
+            (*it)->clear();
+            ++it;
+        }
+    }
+}
+
+sp<AudioProfile> AudioProfileVectorBase::getFirstValidProfile() const
+{
+    for (const auto &profile : *this) {
+        if (profile->isValid()) {
+            return profile;
+        }
+    }
+    return nullptr;
+}
+
+sp<AudioProfile> AudioProfileVectorBase::getFirstValidProfileFor(audio_format_t format) const
+{
+    for (const auto &profile : *this) {
+        if (profile->isValid() && profile->getFormat() == format) {
+            return profile;
+        }
+    }
+    return nullptr;
+}
+
+FormatVector AudioProfileVectorBase::getSupportedFormats() const
+{
+    FormatVector supportedFormats;
+    for (const auto &profile : *this) {
+        if (profile->hasValidFormat()) {
+            supportedFormats.push_back(profile->getFormat());
+        }
+    }
+    return supportedFormats;
+}
+
+bool AudioProfileVectorBase::hasDynamicChannelsFor(audio_format_t format) const
+{
+    for (const auto &profile : *this) {
+        if (profile->getFormat() == format && profile->isDynamicChannels()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioProfileVectorBase::hasDynamicFormat() const
+{
+    for (const auto &profile : *this) {
+        if (profile->isDynamicFormat()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioProfileVectorBase::hasDynamicProfile() const
+{
+    for (const auto &profile : *this) {
+        if (profile->isDynamic()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool AudioProfileVectorBase::hasDynamicRateFor(audio_format_t format) const
+{
+    for (const auto &profile : *this) {
+        if (profile->getFormat() == format && profile->isDynamicRate()) {
+            return true;
+        }
+    }
+    return false;
+}
+
+void AudioProfileVectorBase::dump(std::string *dst, int spaces) const
+{
+    dst->append(base::StringPrintf("%*s- Profiles:\n", spaces, ""));
+    for (size_t i = 0; i < size(); i++) {
+        dst->append(base::StringPrintf("%*sProfile %zu:", spaces + 4, "", i));
+        std::string profileStr;
+        at(i)->dump(&profileStr, spaces + 8);
+        dst->append(profileStr);
+    }
+}
+
+} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioContainers.h b/media/libaudiofoundation/include/media/AudioContainers.h
new file mode 100644
index 0000000..3313224
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioContainers.h
@@ -0,0 +1,52 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <set>
+#include <vector>
+
+#include <system/audio.h>
+
+namespace android {
+
+using ChannelMaskSet = std::set<audio_channel_mask_t>;
+using FormatSet = std::set<audio_format_t>;
+using SampleRateSet = std::set<uint32_t>;
+
+using FormatVector = std::vector<audio_format_t>;
+
+static inline ChannelMaskSet asInMask(const ChannelMaskSet& channelMasks) {
+    ChannelMaskSet inMaskSet;
+    for (const auto &channel : channelMasks) {
+        if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
+            inMaskSet.insert(audio_channel_mask_out_to_in(channel));
+        }
+    }
+    return inMaskSet;
+}
+
+static inline ChannelMaskSet asOutMask(const ChannelMaskSet& channelMasks) {
+    ChannelMaskSet outMaskSet;
+    for (const auto &channel : channelMasks) {
+        if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
+            outMaskSet.insert(audio_channel_mask_in_to_out(channel));
+        }
+    }
+    return outMaskSet;
+}
+
+} // namespace android
\ No newline at end of file
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h b/media/libaudiofoundation/include/media/AudioGain.h
similarity index 84%
rename from services/audiopolicy/common/managerdefinitions/include/AudioGain.h
rename to media/libaudiofoundation/include/media/AudioGain.h
index 4af93e1..6a7fb55 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioGain.h
+++ b/media/libaudiofoundation/include/media/AudioGain.h
@@ -16,15 +16,17 @@
 
 #pragma once
 
+#include <binder/Parcel.h>
+#include <binder/Parcelable.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
-#include <utils/String8.h>
 #include <system/audio.h>
+#include <string>
 #include <vector>
 
 namespace android {
 
-class AudioGain: public RefBase
+class AudioGain: public RefBase, public Parcelable
 {
 public:
     AudioGain(int index, bool useInChannelMask);
@@ -55,7 +57,7 @@
     int getMaxRampInMs() const { return mGain.max_ramp_ms; }
 
     // TODO: remove dump from here (split serialization)
-    void dump(String8 *dst, int spaces, int index) const;
+    void dump(std::string *dst, int spaces, int index) const;
 
     void getDefaultConfig(struct audio_gain_config *config);
     status_t checkConfig(const struct audio_gain_config *config);
@@ -65,6 +67,9 @@
 
     const struct audio_gain &getGain() const { return mGain; }
 
+    status_t writeToParcel(Parcel* parcel) const override;
+    status_t readFromParcel(const Parcel* parcel) override;
+
 private:
     int               mIndex;
     struct audio_gain mGain;
@@ -72,7 +77,7 @@
     bool              mUseForVolume = false;
 };
 
-class AudioGains : public std::vector<sp<AudioGain> >
+class AudioGains : public std::vector<sp<AudioGain> >, public Parcelable
 {
 public:
     bool canUseForVolume() const
@@ -90,6 +95,9 @@
         push_back(gain);
         return 0;
     }
+
+    status_t writeToParcel(Parcel* parcel) const override;
+    status_t readFromParcel(const Parcel* parcel) override;
 };
 
 } // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioPortBase.h b/media/libaudiofoundation/include/media/AudioPortBase.h
new file mode 100644
index 0000000..5812c2c
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioPortBase.h
@@ -0,0 +1,130 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <string>
+
+#include <media/AudioGain.h>
+#include <media/AudioProfile.h>
+#include <utils/Errors.h>
+#include <utils/RefBase.h>
+#include <system/audio.h>
+#include <cutils/config_utils.h>
+
+namespace android {
+
+class AudioPortFoundation : public virtual RefBase
+{
+public:
+    AudioPortFoundation(const std::string& name, audio_port_type_t type,  audio_port_role_t role) :
+            mName(name), mType(type), mRole(role) {}
+
+    virtual ~AudioPortFoundation() = default;
+
+    void setName(const std::string &name) { mName = name; }
+    const std::string &getName() const { return mName; }
+
+    audio_port_type_t getType() const { return mType; }
+    audio_port_role_t getRole() const { return mRole; }
+
+    virtual const std::string getTagName() const = 0;
+
+    void setGains(const AudioGains &gains) { mGains = gains; }
+    const AudioGains &getGains() const { return mGains; }
+
+    virtual void toAudioPort(struct audio_port *port) const;
+
+    virtual AudioProfileVectorBase* getAudioProfileVectorBase() const = 0;
+    virtual void addAudioProfile(const sp<AudioProfile> &profile) {
+        getAudioProfileVectorBase()->add(profile);
+    }
+    virtual void clearAudioProfiles() {
+        getAudioProfileVectorBase()->clearProfiles();
+    }
+
+    bool hasValidAudioProfile() const { return getAudioProfileVectorBase()->hasValidProfile(); }
+
+    status_t checkGain(const struct audio_gain_config *gainConfig, int index) const {
+        if (index < 0 || (size_t)index >= mGains.size()) {
+            return BAD_VALUE;
+        }
+        return mGains[index]->checkConfig(gainConfig);
+    }
+
+    bool useInputChannelMask() const
+    {
+        return ((mType == AUDIO_PORT_TYPE_DEVICE) && (mRole == AUDIO_PORT_ROLE_SOURCE)) ||
+                ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SINK));
+    }
+
+    void dump(std::string *dst, int spaces, bool verbose = true) const;
+
+    AudioGains mGains; // gain controllers
+protected:
+    std::string  mName;
+    audio_port_type_t mType;
+    audio_port_role_t mRole;
+};
+
+template <typename ProfileVector,
+          typename = typename std::enable_if<std::is_base_of<
+                  AudioProfileVectorBase, ProfileVector>::value>::type>
+class AudioPortBase : public AudioPortFoundation
+{
+public:
+    AudioPortBase(const std::string& name, audio_port_type_t type,  audio_port_role_t role) :
+            AudioPortFoundation(name, type, role) {}
+
+    virtual ~AudioPortBase() {}
+
+    AudioProfileVectorBase* getAudioProfileVectorBase() const override {
+        return static_cast<AudioProfileVectorBase*>(const_cast<ProfileVector*>(&mProfiles));
+    }
+
+    void addAudioProfile(const sp<AudioProfile> &profile) override { mProfiles.add(profile); }
+    void clearAudioProfiles() override { return mProfiles.clearProfiles(); }
+
+    void setAudioProfiles(const ProfileVector &profiles) { mProfiles = profiles; }
+    ProfileVector &getAudioProfiles() { return mProfiles; }
+
+protected:
+    ProfileVector mProfiles; // AudioProfiles supported by this port (format, Rates, Channels)
+};
+
+
+class AudioPortConfigBase : public virtual RefBase
+{
+public:
+    virtual ~AudioPortConfigBase() = default;
+
+    virtual status_t applyAudioPortConfig(const struct audio_port_config *config,
+                                          struct audio_port_config *backupConfig = NULL) = 0;
+    virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
+                                   const struct audio_port_config *srcConfig = NULL) const = 0;
+
+    unsigned int getSamplingRate() const { return mSamplingRate; }
+    audio_format_t getFormat() const { return mFormat; }
+    audio_channel_mask_t getChannelMask() const { return mChannelMask; }
+
+protected:
+    unsigned int mSamplingRate = 0u;
+    audio_format_t mFormat = AUDIO_FORMAT_INVALID;
+    audio_channel_mask_t mChannelMask = AUDIO_CHANNEL_NONE;
+    struct audio_gain_config mGain = { .index = -1 };
+};
+
+} // namespace android
diff --git a/media/libaudiofoundation/include/media/AudioProfile.h b/media/libaudiofoundation/include/media/AudioProfile.h
new file mode 100644
index 0000000..20f35eb
--- /dev/null
+++ b/media/libaudiofoundation/include/media/AudioProfile.h
@@ -0,0 +1,106 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <string>
+#include <vector>
+
+#include <media/AudioContainers.h>
+#include <system/audio.h>
+#include <utils/RefBase.h>
+
+namespace android {
+
+class AudioProfile final : public RefBase
+{
+public:
+    static sp<AudioProfile> createFullDynamic(audio_format_t dynamicFormat = AUDIO_FORMAT_DEFAULT);
+
+    AudioProfile(audio_format_t format, audio_channel_mask_t channelMasks, uint32_t samplingRate);
+    AudioProfile(audio_format_t format,
+                 const ChannelMaskSet &channelMasks,
+                 const SampleRateSet &samplingRateCollection);
+
+    audio_format_t getFormat() const { return mFormat; }
+    const ChannelMaskSet &getChannels() const { return mChannelMasks; }
+    const SampleRateSet &getSampleRates() const { return mSamplingRates; }
+    void setChannels(const ChannelMaskSet &channelMasks);
+    void setSampleRates(const SampleRateSet &sampleRates);
+
+    void clear();
+    bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
+    bool supportsChannels(audio_channel_mask_t channels) const
+    {
+        return mChannelMasks.count(channels) != 0;
+    }
+    bool supportsRate(uint32_t rate) const { return mSamplingRates.count(rate) != 0; }
+
+    bool hasValidFormat() const { return mFormat != AUDIO_FORMAT_DEFAULT; }
+    bool hasValidRates() const { return !mSamplingRates.empty(); }
+    bool hasValidChannels() const { return !mChannelMasks.empty(); }
+
+    void setDynamicChannels(bool dynamic) { mIsDynamicChannels = dynamic; }
+    bool isDynamicChannels() const { return mIsDynamicChannels; }
+
+    void setDynamicRate(bool dynamic) { mIsDynamicRate = dynamic; }
+    bool isDynamicRate() const { return mIsDynamicRate; }
+
+    void setDynamicFormat(bool dynamic) { mIsDynamicFormat = dynamic; }
+    bool isDynamicFormat() const { return mIsDynamicFormat; }
+
+    bool isDynamic() { return mIsDynamicFormat || mIsDynamicChannels || mIsDynamicRate; }
+
+    void dump(std::string *dst, int spaces) const;
+
+private:
+    std::string  mName;
+    audio_format_t mFormat; // The format for an audio profile should only be set when initialized.
+    ChannelMaskSet mChannelMasks;
+    SampleRateSet mSamplingRates;
+
+    bool mIsDynamicFormat = false;
+    bool mIsDynamicChannels = false;
+    bool mIsDynamicRate = false;
+};
+
+class AudioProfileVectorBase : public std::vector<sp<AudioProfile> >
+{
+public:
+    virtual ~AudioProfileVectorBase() = default;
+
+    virtual ssize_t add(const sp<AudioProfile> &profile);
+
+    // If the profile is dynamic format and has valid format, it will be removed when doing
+    // clearProfiles(). Otherwise, AudioProfile::clear() will be called.
+    virtual void clearProfiles();
+
+    sp<AudioProfile> getFirstValidProfile() const;
+    sp<AudioProfile> getFirstValidProfileFor(audio_format_t format) const;
+    bool hasValidProfile() const { return getFirstValidProfile() != 0; }
+
+    FormatVector getSupportedFormats() const;
+    bool hasDynamicChannelsFor(audio_format_t format) const;
+    bool hasDynamicFormat() const;
+    bool hasDynamicProfile() const;
+    bool hasDynamicRateFor(audio_format_t format) const;
+
+    virtual void dump(std::string *dst, int spaces) const;
+};
+
+bool operator == (const AudioProfile &left, const AudioProfile &right);
+
+} // namespace android
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 5e22322..5837fcf 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -13,12 +13,6 @@
     ],
 
     shared_libs: [
-        "android.hardware.audio.effect@2.0",
-        "android.hardware.audio.effect@4.0",
-        "android.hardware.audio.effect@5.0",
-        "android.hardware.audio@2.0",
-        "android.hardware.audio@4.0",
-        "android.hardware.audio@5.0",
         "libaudiohal@2.0",
         "libaudiohal@4.0",
         "libaudiohal@5.0",
@@ -26,7 +20,8 @@
     ],
 
     header_libs: [
-        "libaudiohal_headers"
+        "libaudiohal_headers",
+        "libbase_headers",
     ]
 }
 
diff --git a/media/libaudiohal/DevicesFactoryHalInterface.cpp b/media/libaudiohal/DevicesFactoryHalInterface.cpp
index f86009c..d5336fa 100644
--- a/media/libaudiohal/DevicesFactoryHalInterface.cpp
+++ b/media/libaudiohal/DevicesFactoryHalInterface.cpp
@@ -14,26 +14,16 @@
  * limitations under the License.
  */
 
-#include <android/hardware/audio/2.0/IDevicesFactory.h>
-#include <android/hardware/audio/4.0/IDevicesFactory.h>
-#include <android/hardware/audio/5.0/IDevicesFactory.h>
-
 #include <libaudiohal/FactoryHalHidl.h>
 
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+
 namespace android {
 
 // static
 sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
-    if (hardware::audio::V5_0::IDevicesFactory::getService() != nullptr) {
-        return V5_0::createDevicesFactoryHal();
-    }
-    if (hardware::audio::V4_0::IDevicesFactory::getService() != nullptr) {
-        return V4_0::createDevicesFactoryHal();
-    }
-    if (hardware::audio::V2_0::IDevicesFactory::getService() != nullptr) {
-        return V2_0::createDevicesFactoryHal();
-    }
-    return nullptr;
+    return createPreferedImpl<DevicesFactoryHalInterface>();
 }
 
 } // namespace android
+
diff --git a/media/libaudiohal/EffectsFactoryHalInterface.cpp b/media/libaudiohal/EffectsFactoryHalInterface.cpp
index bd3ef61..d15b14e 100644
--- a/media/libaudiohal/EffectsFactoryHalInterface.cpp
+++ b/media/libaudiohal/EffectsFactoryHalInterface.cpp
@@ -1,5 +1,5 @@
 /*
- * Copyright (C) 2016 The Android Open Source Project
+ * Copyright (C) 2017 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
@@ -14,26 +14,15 @@
  * limitations under the License.
  */
 
-#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/5.0/IEffectsFactory.h>
-
 #include <libaudiohal/FactoryHalHidl.h>
 
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+
 namespace android {
 
 // static
 sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
-    if (hardware::audio::effect::V5_0::IEffectsFactory::getService() != nullptr) {
-        return effect::V5_0::createEffectsFactoryHal();
-    }
-    if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
-        return effect::V4_0::createEffectsFactoryHal();
-    }
-    if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
-        return effect::V2_0::createEffectsFactoryHal();
-    }
-    return nullptr;
+    return createPreferedImpl<EffectsFactoryHalInterface>();
 }
 
 // static
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index b25f82e..b07f21d 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -322,6 +322,14 @@
         const struct audio_port_config *sinks,
         audio_patch_handle_t *patch) {
     if (mDevice == 0) return NO_INIT;
+    if (patch == nullptr) return BAD_VALUE;
+
+    if (*patch != AUDIO_PATCH_HANDLE_NONE) {
+        status_t status = releaseAudioPatch(*patch);
+        ALOGW_IF(status != NO_ERROR, "%s error %d releasing patch handle %d",
+            __func__, status, *patch);
+    }
+
     hidl_vec<AudioPortConfig> hidlSources, hidlSinks;
     HidlUtils::audioPortConfigsFromHal(num_sources, sources, &hidlSources);
     HidlUtils::audioPortConfigsFromHal(num_sinks, sinks, &hidlSinks);
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
index 5e01e42..1335a0c 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
@@ -35,13 +35,10 @@
 namespace android {
 namespace CPP_VERSION {
 
-DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
-    sp<IDevicesFactory> defaultFactory{IDevicesFactory::getService()};
-    if (!defaultFactory) {
-        ALOGE("Failed to obtain IDevicesFactory/default service, terminating process.");
-        exit(1);
-    }
-    mDeviceFactories.push_back(defaultFactory);
+DevicesFactoryHalHidl::DevicesFactoryHalHidl(sp<IDevicesFactory> devicesFactory) {
+    ALOG_ASSERT(devicesFactory != nullptr, "Provided IDevicesFactory service is NULL");
+
+    mDeviceFactories.push_back(devicesFactory);
     if (MAJOR_VERSION >= 4) {
         // The MSD factory is optional and only available starting at HAL 4.0
         sp<IDevicesFactory> msdFactory{IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD)};
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.h b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
index 27e0649..8775e7b 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
@@ -32,18 +32,14 @@
 class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
 {
   public:
+    DevicesFactoryHalHidl(sp<IDevicesFactory> devicesFactory);
+
     // Opens a device with the specified name. To close the device, it is
     // necessary to release references to the returned object.
     virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
-
   private:
-    friend class DevicesFactoryHalHybrid;
-
     std::vector<sp<IDevicesFactory>> mDeviceFactories;
 
-    // Can not be constructed directly by clients.
-    DevicesFactoryHalHidl();
-
     virtual ~DevicesFactoryHalHidl() = default;
 };
 
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
index f337a8b..0e1f1bb 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
@@ -17,16 +17,17 @@
 #define LOG_TAG "DevicesFactoryHalHybrid"
 //#define LOG_NDEBUG 0
 
+#include "DevicesFactoryHalHidl.h"
 #include "DevicesFactoryHalHybrid.h"
 #include "DevicesFactoryHalLocal.h"
-#include "DevicesFactoryHalHidl.h"
+#include <libaudiohal/FactoryHalHidl.h>
 
 namespace android {
 namespace CPP_VERSION {
 
-DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
+DevicesFactoryHalHybrid::DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory)
         : mLocalFactory(new DevicesFactoryHalLocal()),
-          mHidlFactory(new DevicesFactoryHalHidl()) {
+          mHidlFactory(new DevicesFactoryHalHidl(hidlFactory)) {
 }
 
 status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
@@ -36,6 +37,12 @@
     }
     return mLocalFactory->openDevice(name, device);
 }
-
 } // namespace CPP_VERSION
+
+template <>
+sp<DevicesFactoryHalInterface> createFactoryHal<AudioHALVersion::CPP_VERSION>() {
+    auto service = hardware::audio::CPP_VERSION::IDevicesFactory::getService();
+    return service ? new CPP_VERSION::DevicesFactoryHalHybrid(service) : nullptr;
+}
+
 } // namespace android
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
index 5ac0d0d..545bb70 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
@@ -17,17 +17,20 @@
 #ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
 #define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
 
+#include PATH(android/hardware/audio/FILE_VERSION/IDevicesFactory.h)
 #include <media/audiohal/DevicesFactoryHalInterface.h>
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
 
+using ::android::hardware::audio::CPP_VERSION::IDevicesFactory;
+
 namespace android {
 namespace CPP_VERSION {
 
 class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
 {
   public:
-    DevicesFactoryHalHybrid();
+    DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory);
 
     // Opens a device with the specified name. To close the device, it is
     // necessary to release references to the returned object.
@@ -38,10 +41,6 @@
     sp<DevicesFactoryHalInterface> mHidlFactory;
 };
 
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal() {
-    return new DevicesFactoryHalHybrid();
-}
-
 } // namespace CPP_VERSION
 } // namespace android
 
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 7fd6bde..ba7b195 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -19,11 +19,12 @@
 
 #include <cutils/native_handle.h>
 
-#include "EffectsFactoryHalHidl.h"
 #include "ConversionHelperHidl.h"
 #include "EffectBufferHalHidl.h"
 #include "EffectHalHidl.h"
+#include "EffectsFactoryHalHidl.h"
 #include "HidlUtils.h"
+#include <libaudiohal/FactoryHalHidl.h>
 
 using ::android::hardware::audio::common::CPP_VERSION::implementation::HidlUtils;
 using ::android::hardware::Return;
@@ -35,12 +36,10 @@
 using namespace ::android::hardware::audio::common::CPP_VERSION;
 using namespace ::android::hardware::audio::effect::CPP_VERSION;
 
-EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
-    mEffectsFactory = IEffectsFactory::getService();
-    if (mEffectsFactory == 0) {
-        ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
-        exit(1);
-    }
+EffectsFactoryHalHidl::EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory)
+        : ConversionHelperHidl("EffectsFactory") {
+    ALOG_ASSERT(effectsFactory != nullptr, "Provided IDevicesFactory service is NULL");
+    mEffectsFactory = effectsFactory;
 }
 
 status_t EffectsFactoryHalHidl::queryAllDescriptors() {
@@ -147,4 +146,11 @@
 
 } // namespace CPP_VERSION
 } // namespace effect
+
+template<>
+sp<EffectsFactoryHalInterface> createFactoryHal<AudioHALVersion::CPP_VERSION>() {
+    auto service = hardware::audio::effect::CPP_VERSION::IEffectsFactory::getService();
+    return service ? new effect::CPP_VERSION::EffectsFactoryHalHidl(service) : nullptr;
+}
+
 } // namespace android
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 01178ff..2828513 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -18,7 +18,6 @@
 #define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
 
 #include PATH(android/hardware/audio/effect/FILE_VERSION/IEffectsFactory.h)
-#include PATH(android/hardware/audio/effect/FILE_VERSION/types.h)
 #include <media/audiohal/EffectsFactoryHalInterface.h>
 
 #include "ConversionHelperHidl.h"
@@ -34,7 +33,7 @@
 class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
 {
   public:
-    EffectsFactoryHalHidl();
+    EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory);
 
     // Returns the number of different effects in all loaded libraries.
     virtual status_t queryNumberEffects(uint32_t *pNumEffects);
@@ -66,10 +65,6 @@
     status_t queryAllDescriptors();
 };
 
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal() {
-    return new EffectsFactoryHalHidl();
-}
-
 } // namespace CPP_VERSION
 } // namespace effect
 } // namespace android
diff --git a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
index c7319d0..829f99c 100644
--- a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
+++ b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
@@ -23,33 +23,42 @@
 #include <media/audiohal/EffectsFactoryHalInterface.h>
 #include <utils/StrongPointer.h>
 
+#include <array>
+#include <utility>
+
 namespace android {
 
-namespace effect {
-namespace V2_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V2_0
+/** Supported HAL versions, in order of preference.
+ * Implementation should use specialize the `create*FactoryHal` for their version.
+ * Client should use `createPreferedImpl<*FactoryHal>()` to instantiate
+ * the preferred available impl.
+ */
+enum class AudioHALVersion {
+    V5_0,
+    V4_0,
+    V2_0,
+    end, // used for iterating over supported versions
+};
 
-namespace V4_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V4_0
+/** Template function to fully specialized for each version and each Interface. */
+template <AudioHALVersion, class Interface>
+sp<Interface> createFactoryHal();
 
-namespace V5_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V5_0
-} // namespace effect
+/** @Return the preferred available implementation or nullptr if none are available. */
+template <class Interface, AudioHALVersion version = AudioHALVersion{}>
+static sp<Interface> createPreferedImpl() {
+    if constexpr (version == AudioHALVersion::end) {
+        return nullptr; // tried all version, all returned nullptr
+    } else {
+        if (auto created = createFactoryHal<version, Interface>(); created != nullptr) {
+           return created;
+        }
 
-namespace V2_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V2_0
+        using Raw = std::underlying_type_t<AudioHALVersion>; // cast as enum class do not support ++
+        return createPreferedImpl<Interface, AudioHALVersion(Raw(version) + 1)>();
+    }
+}
 
-namespace V4_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V4_0
-
-namespace V5_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V5_0
 
 } // namespace android
 
diff --git a/media/libaudioprocessing/Android.bp b/media/libaudioprocessing/Android.bp
index cb78063..e8aa700 100644
--- a/media/libaudioprocessing/Android.bp
+++ b/media/libaudioprocessing/Android.bp
@@ -3,20 +3,13 @@
 
     export_include_dirs: ["include"],
 
+    header_libs: ["libaudioclient_headers"],
+
     shared_libs: [
-        "libaudiohal",
         "libaudioutils",
         "libcutils",
         "liblog",
-        "libnbaio",
-        "libnblog",
-        "libsonic",
         "libutils",
-        "libvibrator",
-    ],
-
-    header_libs: [
-        "libbase_headers",
     ],
 
     cflags: [
@@ -33,18 +26,31 @@
     defaults: ["libaudioprocessing_defaults"],
 
     srcs: [
+        "AudioMixer.cpp",
         "BufferProviders.cpp",
         "RecordBufferConverter.cpp",
     ],
-    whole_static_libs: ["libaudioprocessing_arm"],
+
+    header_libs: [
+        "libbase_headers",
+    ],
+
+    shared_libs: [
+        "libaudiohal",
+        "libsonic",
+        "libvibrator",
+    ],
+
+    whole_static_libs: ["libaudioprocessing_base"],
 }
 
 cc_library_static {
-    name: "libaudioprocessing_arm",
+    name: "libaudioprocessing_base",
     defaults: ["libaudioprocessing_defaults"],
+    vendor_available: true,
 
     srcs: [
-        "AudioMixer.cpp",
+        "AudioMixerBase.cpp",
         "AudioResampler.cpp",
         "AudioResamplerCubic.cpp",
         "AudioResamplerSinc.cpp",
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index f7cc096..c0b11a4 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -18,6 +18,7 @@
 #define LOG_TAG "AudioMixer"
 //#define LOG_NDEBUG 0
 
+#include <sstream>
 #include <stdint.h>
 #include <string.h>
 #include <stdlib.h>
@@ -27,9 +28,6 @@
 #include <utils/Errors.h>
 #include <utils/Log.h>
 
-#include <cutils/compiler.h>
-#include <utils/Debug.h>
-
 #include <system/audio.h>
 
 #include <audio_utils/primitives.h>
@@ -58,138 +56,15 @@
 #define ALOGVV(a...) do { } while (0)
 #endif
 
-#ifndef ARRAY_SIZE
-#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
-#endif
-
-// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
-// original code will be used for stereo sinks, the new mixer for multichannel.
-static constexpr bool kUseNewMixer = true;
-
-// Set kUseFloat to true to allow floating input into the mixer engine.
-// If kUseNewMixer is false, this is ignored or may be overridden internally
-// because of downmix/upmix support.
-static constexpr bool kUseFloat = true;
-
-#ifdef FLOAT_AUX
-using TYPE_AUX = float;
-static_assert(kUseNewMixer && kUseFloat,
-        "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
-#else
-using TYPE_AUX = int32_t; // q4.27
-#endif
-
 // Set to default copy buffer size in frames for input processing.
-static const size_t kCopyBufferFrameCount = 256;
+static constexpr size_t kCopyBufferFrameCount = 256;
 
 namespace android {
 
 // ----------------------------------------------------------------------------
 
-static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
-    return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-}
-
-status_t AudioMixer::create(
-        int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
-{
-    LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
-
-    if (!isValidChannelMask(channelMask)) {
-        ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
-        return BAD_VALUE;
-    }
-    if (!isValidFormat(format)) {
-        ALOGE("%s invalid format: %#x", __func__, format);
-        return BAD_VALUE;
-    }
-
-    auto t = std::make_shared<Track>();
-    {
-        // TODO: move initialization to the Track constructor.
-        // assume default parameters for the track, except where noted below
-        t->needs = 0;
-
-        // Integer volume.
-        // Currently integer volume is kept for the legacy integer mixer.
-        // Will be removed when the legacy mixer path is removed.
-        t->volume[0] = 0;
-        t->volume[1] = 0;
-        t->prevVolume[0] = 0 << 16;
-        t->prevVolume[1] = 0 << 16;
-        t->volumeInc[0] = 0;
-        t->volumeInc[1] = 0;
-        t->auxLevel = 0;
-        t->auxInc = 0;
-        t->prevAuxLevel = 0;
-
-        // Floating point volume.
-        t->mVolume[0] = 0.f;
-        t->mVolume[1] = 0.f;
-        t->mPrevVolume[0] = 0.f;
-        t->mPrevVolume[1] = 0.f;
-        t->mVolumeInc[0] = 0.;
-        t->mVolumeInc[1] = 0.;
-        t->mAuxLevel = 0.;
-        t->mAuxInc = 0.;
-        t->mPrevAuxLevel = 0.;
-
-        // no initialization needed
-        // t->frameCount
-        t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
-        t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
-        channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
-        t->channelCount = audio_channel_count_from_out_mask(channelMask);
-        t->enabled = false;
-        ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
-                "Non-stereo channel mask: %d\n", channelMask);
-        t->channelMask = channelMask;
-        t->sessionId = sessionId;
-        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
-        t->bufferProvider = NULL;
-        t->buffer.raw = NULL;
-        // no initialization needed
-        // t->buffer.frameCount
-        t->hook = NULL;
-        t->mIn = NULL;
-        t->sampleRate = mSampleRate;
-        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
-        t->mainBuffer = NULL;
-        t->auxBuffer = NULL;
-        t->mInputBufferProvider = NULL;
-        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
-        t->mFormat = format;
-        t->mMixerInFormat = selectMixerInFormat(format);
-        t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
-        t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
-                AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
-        t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
-        t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
-        // haptic
-        t->mHapticPlaybackEnabled = false;
-        t->mHapticIntensity = HAPTIC_SCALE_NONE;
-        t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
-        t->mMixerHapticChannelCount = 0;
-        t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
-        t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
-        t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
-        t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
-        t->mKeepContractedChannels = false;
-        // Check the downmixing (or upmixing) requirements.
-        status_t status = t->prepareForDownmix();
-        if (status != OK) {
-            ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
-            return BAD_VALUE;
-        }
-        // prepareForDownmix() may change mDownmixRequiresFormat
-        ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
-        t->prepareForReformat();
-        t->prepareForAdjustChannelsNonDestructive(mFrameCount);
-        t->prepareForAdjustChannels();
-
-        mTracks[name] = t;
-        return OK;
-    }
+bool AudioMixer::isValidChannelMask(audio_channel_mask_t channelMask) const {
+    return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
 }
 
 // Called when channel masks have changed for a track name
@@ -198,7 +73,7 @@
 bool AudioMixer::setChannelMasks(int name,
         audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
     LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
+    const std::shared_ptr<Track> &track = getTrack(name);
 
     if (trackChannelMask == (track->channelMask | track->mHapticChannelMask)
             && mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
@@ -255,14 +130,8 @@
     track->prepareForAdjustChannelsNonDestructive(mFrameCount);
     track->prepareForAdjustChannels();
 
-    if (track->mResampler.get() != nullptr) {
-        // resampler channels may have changed.
-        const uint32_t resetToSampleRate = track->sampleRate;
-        track->mResampler.reset(nullptr);
-        track->sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
-        // recreate the resampler with updated format, channels, saved sampleRate.
-        track->setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
-    }
+    // Resampler channels may have changed.
+    track->recreateResampler(mSampleRate);
     return true;
 }
 
@@ -477,171 +346,10 @@
     }
 }
 
-void AudioMixer::destroy(int name)
-{
-    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    ALOGV("deleteTrackName(%d)", name);
-
-    if (mTracks[name]->enabled) {
-        invalidate();
-    }
-    mTracks.erase(name); // deallocate track
-}
-
-void AudioMixer::enable(int name)
-{
-    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
-
-    if (!track->enabled) {
-        track->enabled = true;
-        ALOGV("enable(%d)", name);
-        invalidate();
-    }
-}
-
-void AudioMixer::disable(int name)
-{
-    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
-
-    if (track->enabled) {
-        track->enabled = false;
-        ALOGV("disable(%d)", name);
-        invalidate();
-    }
-}
-
-/* Sets the volume ramp variables for the AudioMixer.
- *
- * The volume ramp variables are used to transition from the previous
- * volume to the set volume.  ramp controls the duration of the transition.
- * Its value is typically one state framecount period, but may also be 0,
- * meaning "immediate."
- *
- * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
- * even if there is a nonzero floating point increment (in that case, the volume
- * change is immediate).  This restriction should be changed when the legacy mixer
- * is removed (see #2).
- * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
- * when no longer needed.
- *
- * @param newVolume set volume target in floating point [0.0, 1.0].
- * @param ramp number of frames to increment over. if ramp is 0, the volume
- * should be set immediately.  Currently ramp should not exceed 65535 (frames).
- * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
- * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
- * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
- * @param pSetVolume pointer to the float target volume, set on return.
- * @param pPrevVolume pointer to the float previous volume, set on return.
- * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
- * @return true if the volume has changed, false if volume is same.
- */
-static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
-        int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
-        float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
-    // check floating point volume to see if it is identical to the previously
-    // set volume.
-    // We do not use a tolerance here (and reject changes too small)
-    // as it may be confusing to use a different value than the one set.
-    // If the resulting volume is too small to ramp, it is a direct set of the volume.
-    if (newVolume == *pSetVolume) {
-        return false;
-    }
-    if (newVolume < 0) {
-        newVolume = 0; // should not have negative volumes
-    } else {
-        switch (fpclassify(newVolume)) {
-        case FP_SUBNORMAL:
-        case FP_NAN:
-            newVolume = 0;
-            break;
-        case FP_ZERO:
-            break; // zero volume is fine
-        case FP_INFINITE:
-            // Infinite volume could be handled consistently since
-            // floating point math saturates at infinities,
-            // but we limit volume to unity gain float.
-            // ramp = 0; break;
-            //
-            newVolume = AudioMixer::UNITY_GAIN_FLOAT;
-            break;
-        case FP_NORMAL:
-        default:
-            // Floating point does not have problems with overflow wrap
-            // that integer has.  However, we limit the volume to
-            // unity gain here.
-            // TODO: Revisit the volume limitation and perhaps parameterize.
-            if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
-                newVolume = AudioMixer::UNITY_GAIN_FLOAT;
-            }
-            break;
-        }
-    }
-
-    // set floating point volume ramp
-    if (ramp != 0) {
-        // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
-        // is no computational mismatch; hence equality is checked here.
-        ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
-                " prev:%f  set_to:%f", *pPrevVolume, *pSetVolume);
-        const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
-        // could be inf, cannot be nan, subnormal
-        const float maxv = std::max(newVolume, *pPrevVolume);
-
-        if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
-                && maxv + inc != maxv) { // inc must make forward progress
-            *pVolumeInc = inc;
-            // ramp is set now.
-            // Note: if newVolume is 0, then near the end of the ramp,
-            // it may be possible that the ramped volume may be subnormal or
-            // temporarily negative by a small amount or subnormal due to floating
-            // point inaccuracies.
-        } else {
-            ramp = 0; // ramp not allowed
-        }
-    }
-
-    // compute and check integer volume, no need to check negative values
-    // The integer volume is limited to "unity_gain" to avoid wrapping and other
-    // audio artifacts, so it never reaches the range limit of U4.28.
-    // We safely use signed 16 and 32 bit integers here.
-    const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
-    const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
-            AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
-
-    // set integer volume ramp
-    if (ramp != 0) {
-        // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
-        // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
-        // is no computational mismatch; hence equality is checked here.
-        ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
-                " prev:%d  set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
-        const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
-
-        if (inc != 0) { // inc must make forward progress
-            *pIntVolumeInc = inc;
-        } else {
-            ramp = 0; // ramp not allowed
-        }
-    }
-
-    // if no ramp, or ramp not allowed, then clear float and integer increments
-    if (ramp == 0) {
-        *pVolumeInc = 0;
-        *pPrevVolume = newVolume;
-        *pIntVolumeInc = 0;
-        *pIntPrevVolume = intVolume << 16;
-    }
-    *pSetVolume = newVolume;
-    *pIntSetVolume = intVolume;
-    return true;
-}
-
 void AudioMixer::setParameter(int name, int target, int param, void *value)
 {
     LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
+    const std::shared_ptr<Track> &track = getTrack(name);
 
     int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
     int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
@@ -670,11 +378,7 @@
             }
             break;
         case AUX_BUFFER:
-            if (track->auxBuffer != valueBuf) {
-                track->auxBuffer = valueBuf;
-                ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
-                invalidate();
-            }
+            AudioMixerBase::setParameter(name, target, param, value);
             break;
         case FORMAT: {
             audio_format_t format = static_cast<audio_format_t>(valueInt);
@@ -730,127 +434,38 @@
         break;
 
     case RESAMPLE:
-        switch (param) {
-        case SAMPLE_RATE:
-            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
-            if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
-                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
-                        uint32_t(valueInt));
-                invalidate();
-            }
-            break;
-        case RESET:
-            track->resetResampler();
-            invalidate();
-            break;
-        case REMOVE:
-            track->mResampler.reset(nullptr);
-            track->sampleRate = mSampleRate;
-            invalidate();
-            break;
-        default:
-            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
-        }
-        break;
-
     case RAMP_VOLUME:
     case VOLUME:
+        AudioMixerBase::setParameter(name, target, param, value);
+        break;
+    case TIMESTRETCH:
         switch (param) {
-        case AUXLEVEL:
-            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
-                    target == RAMP_VOLUME ? mFrameCount : 0,
-                    &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
-                    &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
-                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
-                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
-                invalidate();
+        case PLAYBACK_RATE: {
+            const AudioPlaybackRate *playbackRate =
+                    reinterpret_cast<AudioPlaybackRate*>(value);
+            ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
+                    "bad parameters speed %f, pitch %f",
+                    playbackRate->mSpeed, playbackRate->mPitch);
+            if (track->setPlaybackRate(*playbackRate)) {
+                ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
+                        "%f %f %d %d",
+                        playbackRate->mSpeed,
+                        playbackRate->mPitch,
+                        playbackRate->mStretchMode,
+                        playbackRate->mFallbackMode);
+                // invalidate();  (should not require reconfigure)
             }
-            break;
+        } break;
         default:
-            if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
-                if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
-                        target == RAMP_VOLUME ? mFrameCount : 0,
-                        &track->volume[param - VOLUME0],
-                        &track->prevVolume[param - VOLUME0],
-                        &track->volumeInc[param - VOLUME0],
-                        &track->mVolume[param - VOLUME0],
-                        &track->mPrevVolume[param - VOLUME0],
-                        &track->mVolumeInc[param - VOLUME0])) {
-                    ALOGV("setParameter(%s, VOLUME%d: %04x)",
-                            target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
-                                    track->volume[param - VOLUME0]);
-                    invalidate();
-                }
-            } else {
-                LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
-            }
+            LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
         }
         break;
-        case TIMESTRETCH:
-            switch (param) {
-            case PLAYBACK_RATE: {
-                const AudioPlaybackRate *playbackRate =
-                        reinterpret_cast<AudioPlaybackRate*>(value);
-                ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
-                        "bad parameters speed %f, pitch %f",
-                        playbackRate->mSpeed, playbackRate->mPitch);
-                if (track->setPlaybackRate(*playbackRate)) {
-                    ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
-                            "%f %f %d %d",
-                            playbackRate->mSpeed,
-                            playbackRate->mPitch,
-                            playbackRate->mStretchMode,
-                            playbackRate->mFallbackMode);
-                    // invalidate();  (should not require reconfigure)
-                }
-            } break;
-            default:
-                LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
-            }
-            break;
 
     default:
         LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
     }
 }
 
-bool AudioMixer::Track::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
-{
-    if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
-        if (sampleRate != trackSampleRate) {
-            sampleRate = trackSampleRate;
-            if (mResampler.get() == nullptr) {
-                ALOGV("Creating resampler from track %d Hz to device %d Hz",
-                        trackSampleRate, devSampleRate);
-                AudioResampler::src_quality quality;
-                // force lowest quality level resampler if use case isn't music or video
-                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
-                // quality level based on the initial ratio, but that could change later.
-                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
-                if (isMusicRate(trackSampleRate)) {
-                    quality = AudioResampler::DEFAULT_QUALITY;
-                } else {
-                    quality = AudioResampler::DYN_LOW_QUALITY;
-                }
-
-                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
-                // but if none exists, it is the channel count (1 for mono).
-                const int resamplerChannelCount = mDownmixerBufferProvider.get() != nullptr
-                        ? mMixerChannelCount : channelCount;
-                ALOGVV("Creating resampler:"
-                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
-                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
-                mResampler.reset(AudioResampler::create(
-                        mMixerInFormat,
-                        resamplerChannelCount,
-                        devSampleRate, quality));
-            }
-            return true;
-        }
-    }
-    return false;
-}
-
 bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
 {
     if ((mTimestretchBufferProvider.get() == nullptr &&
@@ -863,8 +478,7 @@
     if (mTimestretchBufferProvider.get() == nullptr) {
         // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
         // but if none exists, it is the channel count (1 for mono).
-        const int timestretchChannelCount = mDownmixerBufferProvider.get() != nullptr
-                ? mMixerChannelCount : channelCount;
+        const int timestretchChannelCount = getOutputChannelCount();
         mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
                 mMixerInFormat, sampleRate, playbackRate));
         reconfigureBufferProviders();
@@ -875,84 +489,10 @@
     return true;
 }
 
-/* Checks to see if the volume ramp has completed and clears the increment
- * variables appropriately.
- *
- * FIXME: There is code to handle int/float ramp variable switchover should it not
- * complete within a mixer buffer processing call, but it is preferred to avoid switchover
- * due to precision issues.  The switchover code is included for legacy code purposes
- * and can be removed once the integer volume is removed.
- *
- * It is not sufficient to clear only the volumeInc integer variable because
- * if one channel requires ramping, all channels are ramped.
- *
- * There is a bit of duplicated code here, but it keeps backward compatibility.
- */
-inline void AudioMixer::Track::adjustVolumeRamp(bool aux, bool useFloat)
-{
-    if (useFloat) {
-        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
-            if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
-                     (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
-                volumeInc[i] = 0;
-                prevVolume[i] = volume[i] << 16;
-                mVolumeInc[i] = 0.;
-                mPrevVolume[i] = mVolume[i];
-            } else {
-                //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
-                prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
-            }
-        }
-    } else {
-        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
-            if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
-                    ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
-                volumeInc[i] = 0;
-                prevVolume[i] = volume[i] << 16;
-                mVolumeInc[i] = 0.;
-                mPrevVolume[i] = mVolume[i];
-            } else {
-                //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
-                mPrevVolume[i]  = float_from_u4_28(prevVolume[i]);
-            }
-        }
-    }
-
-    if (aux) {
-#ifdef FLOAT_AUX
-        if (useFloat) {
-            if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
-                    (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
-                auxInc = 0;
-                prevAuxLevel = auxLevel << 16;
-                mAuxInc = 0.f;
-                mPrevAuxLevel = mAuxLevel;
-            }
-        } else
-#endif
-        if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
-                (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
-            auxInc = 0;
-            prevAuxLevel = auxLevel << 16;
-            mAuxInc = 0.f;
-            mPrevAuxLevel = mAuxLevel;
-        }
-    }
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name) const
-{
-    const auto it = mTracks.find(name);
-    if (it != mTracks.end()) {
-        return it->second->getUnreleasedFrames();
-    }
-    return 0;
-}
-
 void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
 {
     LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
-    const std::shared_ptr<Track> &track = mTracks[name];
+    const std::shared_ptr<Track> &track = getTrack(name);
 
     if (track->mInputBufferProvider == bufferProvider) {
         return; // don't reset any buffer providers if identical.
@@ -976,679 +516,6 @@
     track->reconfigureBufferProviders();
 }
 
-void AudioMixer::process__validate()
-{
-    // TODO: fix all16BitsStereNoResample logic to
-    // either properly handle muted tracks (it should ignore them)
-    // or remove altogether as an obsolete optimization.
-    bool all16BitsStereoNoResample = true;
-    bool resampling = false;
-    bool volumeRamp = false;
-
-    mEnabled.clear();
-    mGroups.clear();
-    for (const auto &pair : mTracks) {
-        const int name = pair.first;
-        const std::shared_ptr<Track> &t = pair.second;
-        if (!t->enabled) continue;
-
-        mEnabled.emplace_back(name);  // we add to mEnabled in order of name.
-        mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
-
-        uint32_t n = 0;
-        // FIXME can overflow (mask is only 3 bits)
-        n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
-        if (t->doesResample()) {
-            n |= NEEDS_RESAMPLE;
-        }
-        if (t->auxLevel != 0 && t->auxBuffer != NULL) {
-            n |= NEEDS_AUX;
-        }
-
-        if (t->volumeInc[0]|t->volumeInc[1]) {
-            volumeRamp = true;
-        } else if (!t->doesResample() && t->volumeRL == 0) {
-            n |= NEEDS_MUTE;
-        }
-        t->needs = n;
-
-        if (n & NEEDS_MUTE) {
-            t->hook = &Track::track__nop;
-        } else {
-            if (n & NEEDS_AUX) {
-                all16BitsStereoNoResample = false;
-            }
-            if (n & NEEDS_RESAMPLE) {
-                all16BitsStereoNoResample = false;
-                resampling = true;
-                t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
-                        t->mMixerInFormat, t->mMixerFormat);
-                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                        "Track %d needs downmix + resample", name);
-            } else {
-                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
-                    t->hook = Track::getTrackHook(
-                            (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
-                                    && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
-                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
-                            t->mMixerChannelCount,
-                            t->mMixerInFormat, t->mMixerFormat);
-                    all16BitsStereoNoResample = false;
-                }
-                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
-                    t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
-                            t->mMixerInFormat, t->mMixerFormat);
-                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
-                            "Track %d needs downmix", name);
-                }
-            }
-        }
-    }
-
-    // select the processing hooks
-    mHook = &AudioMixer::process__nop;
-    if (mEnabled.size() > 0) {
-        if (resampling) {
-            if (mOutputTemp.get() == nullptr) {
-                mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
-            }
-            if (mResampleTemp.get() == nullptr) {
-                mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
-            }
-            mHook = &AudioMixer::process__genericResampling;
-        } else {
-            // we keep temp arrays around.
-            mHook = &AudioMixer::process__genericNoResampling;
-            if (all16BitsStereoNoResample && !volumeRamp) {
-                if (mEnabled.size() == 1) {
-                    const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
-                    if ((t->needs & NEEDS_MUTE) == 0) {
-                        // The check prevents a muted track from acquiring a process hook.
-                        //
-                        // This is dangerous if the track is MONO as that requires
-                        // special case handling due to implicit channel duplication.
-                        // Stereo or Multichannel should actually be fine here.
-                        mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
-                                t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
-                    }
-                }
-            }
-        }
-    }
-
-    ALOGV("mixer configuration change: %zu "
-        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
-        mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
-
-   process();
-
-    // Now that the volume ramp has been done, set optimal state and
-    // track hooks for subsequent mixer process
-    if (mEnabled.size() > 0) {
-        bool allMuted = true;
-
-        for (const int name : mEnabled) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            if (!t->doesResample() && t->volumeRL == 0) {
-                t->needs |= NEEDS_MUTE;
-                t->hook = &Track::track__nop;
-            } else {
-                allMuted = false;
-            }
-        }
-        if (allMuted) {
-            mHook = &AudioMixer::process__nop;
-        } else if (all16BitsStereoNoResample) {
-            if (mEnabled.size() == 1) {
-                //const int i = 31 - __builtin_clz(enabledTracks);
-                const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
-                // Muted single tracks handled by allMuted above.
-                mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
-                        t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
-            }
-        }
-    }
-}
-
-void AudioMixer::Track::track__genericResample(
-        int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
-{
-    ALOGVV("track__genericResample\n");
-    mResampler->setSampleRate(sampleRate);
-
-    // ramp gain - resample to temp buffer and scale/mix in 2nd step
-    if (aux != NULL) {
-        // always resample with unity gain when sending to auxiliary buffer to be able
-        // to apply send level after resampling
-        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
-        mResampler->resample(temp, outFrameCount, bufferProvider);
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
-            volumeRampStereo(out, outFrameCount, temp, aux);
-        } else {
-            volumeStereo(out, outFrameCount, temp, aux);
-        }
-    } else {
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
-            mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
-            mResampler->resample(temp, outFrameCount, bufferProvider);
-            volumeRampStereo(out, outFrameCount, temp, aux);
-        }
-
-        // constant gain
-        else {
-            mResampler->setVolume(mVolume[0], mVolume[1]);
-            mResampler->resample(out, outFrameCount, bufferProvider);
-        }
-    }
-}
-
-void AudioMixer::Track::track__nop(int32_t* out __unused,
-        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
-{
-}
-
-void AudioMixer::Track::volumeRampStereo(
-        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
-    int32_t vl = prevVolume[0];
-    int32_t vr = prevVolume[1];
-    const int32_t vlInc = volumeInc[0];
-    const int32_t vrInc = volumeInc[1];
-
-    //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-    //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
-    //       (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-    // ramp volume
-    if (CC_UNLIKELY(aux != NULL)) {
-        int32_t va = prevAuxLevel;
-        const int32_t vaInc = auxInc;
-        int32_t l;
-        int32_t r;
-
-        do {
-            l = (*temp++ >> 12);
-            r = (*temp++ >> 12);
-            *out++ += (vl >> 16) * l;
-            *out++ += (vr >> 16) * r;
-            *aux++ += (va >> 17) * (l + r);
-            vl += vlInc;
-            vr += vrInc;
-            va += vaInc;
-        } while (--frameCount);
-        prevAuxLevel = va;
-    } else {
-        do {
-            *out++ += (vl >> 16) * (*temp++ >> 12);
-            *out++ += (vr >> 16) * (*temp++ >> 12);
-            vl += vlInc;
-            vr += vrInc;
-        } while (--frameCount);
-    }
-    prevVolume[0] = vl;
-    prevVolume[1] = vr;
-    adjustVolumeRamp(aux != NULL);
-}
-
-void AudioMixer::Track::volumeStereo(
-        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
-    const int16_t vl = volume[0];
-    const int16_t vr = volume[1];
-
-    if (CC_UNLIKELY(aux != NULL)) {
-        const int16_t va = auxLevel;
-        do {
-            int16_t l = (int16_t)(*temp++ >> 12);
-            int16_t r = (int16_t)(*temp++ >> 12);
-            out[0] = mulAdd(l, vl, out[0]);
-            int16_t a = (int16_t)(((int32_t)l + r) >> 1);
-            out[1] = mulAdd(r, vr, out[1]);
-            out += 2;
-            aux[0] = mulAdd(a, va, aux[0]);
-            aux++;
-        } while (--frameCount);
-    } else {
-        do {
-            int16_t l = (int16_t)(*temp++ >> 12);
-            int16_t r = (int16_t)(*temp++ >> 12);
-            out[0] = mulAdd(l, vl, out[0]);
-            out[1] = mulAdd(r, vr, out[1]);
-            out += 2;
-        } while (--frameCount);
-    }
-}
-
-void AudioMixer::Track::track__16BitsStereo(
-        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
-    ALOGVV("track__16BitsStereo\n");
-    const int16_t *in = static_cast<const int16_t *>(mIn);
-
-    if (CC_UNLIKELY(aux != NULL)) {
-        int32_t l;
-        int32_t r;
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            int32_t va = prevAuxLevel;
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-            const int32_t vaInc = auxInc;
-            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                l = (int32_t)*in++;
-                r = (int32_t)*in++;
-                *out++ += (vl >> 16) * l;
-                *out++ += (vr >> 16) * r;
-                *aux++ += (va >> 17) * (l + r);
-                vl += vlInc;
-                vr += vrInc;
-                va += vaInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            prevAuxLevel = va;
-            adjustVolumeRamp(true);
-        }
-
-        // constant gain
-        else {
-            const uint32_t vrl = volumeRL;
-            const int16_t va = (int16_t)auxLevel;
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
-                in += 2;
-                out[0] = mulAddRL(1, rl, vrl, out[0]);
-                out[1] = mulAddRL(0, rl, vrl, out[1]);
-                out += 2;
-                aux[0] = mulAdd(a, va, aux[0]);
-                aux++;
-            } while (--frameCount);
-        }
-    } else {
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-
-            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                *out++ += (vl >> 16) * (int32_t) *in++;
-                *out++ += (vr >> 16) * (int32_t) *in++;
-                vl += vlInc;
-                vr += vrInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            adjustVolumeRamp(false);
-        }
-
-        // constant gain
-        else {
-            const uint32_t vrl = volumeRL;
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                in += 2;
-                out[0] = mulAddRL(1, rl, vrl, out[0]);
-                out[1] = mulAddRL(0, rl, vrl, out[1]);
-                out += 2;
-            } while (--frameCount);
-        }
-    }
-    mIn = in;
-}
-
-void AudioMixer::Track::track__16BitsMono(
-        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
-    ALOGVV("track__16BitsMono\n");
-    const int16_t *in = static_cast<int16_t const *>(mIn);
-
-    if (CC_UNLIKELY(aux != NULL)) {
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            int32_t va = prevAuxLevel;
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-            const int32_t vaInc = auxInc;
-
-            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                int32_t l = *in++;
-                *out++ += (vl >> 16) * l;
-                *out++ += (vr >> 16) * l;
-                *aux++ += (va >> 16) * l;
-                vl += vlInc;
-                vr += vrInc;
-                va += vaInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            prevAuxLevel = va;
-            adjustVolumeRamp(true);
-        }
-        // constant gain
-        else {
-            const int16_t vl = volume[0];
-            const int16_t vr = volume[1];
-            const int16_t va = (int16_t)auxLevel;
-            do {
-                int16_t l = *in++;
-                out[0] = mulAdd(l, vl, out[0]);
-                out[1] = mulAdd(l, vr, out[1]);
-                out += 2;
-                aux[0] = mulAdd(l, va, aux[0]);
-                aux++;
-            } while (--frameCount);
-        }
-    } else {
-        // ramp gain
-        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
-            int32_t vl = prevVolume[0];
-            int32_t vr = prevVolume[1];
-            const int32_t vlInc = volumeInc[0];
-            const int32_t vrInc = volumeInc[1];
-
-            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
-            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
-            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
-
-            do {
-                int32_t l = *in++;
-                *out++ += (vl >> 16) * l;
-                *out++ += (vr >> 16) * l;
-                vl += vlInc;
-                vr += vrInc;
-            } while (--frameCount);
-
-            prevVolume[0] = vl;
-            prevVolume[1] = vr;
-            adjustVolumeRamp(false);
-        }
-        // constant gain
-        else {
-            const int16_t vl = volume[0];
-            const int16_t vr = volume[1];
-            do {
-                int16_t l = *in++;
-                out[0] = mulAdd(l, vl, out[0]);
-                out[1] = mulAdd(l, vr, out[1]);
-                out += 2;
-            } while (--frameCount);
-        }
-    }
-    mIn = in;
-}
-
-// no-op case
-void AudioMixer::process__nop()
-{
-    ALOGVV("process__nop\n");
-
-    for (const auto &pair : mGroups) {
-        // process by group of tracks with same output buffer to
-        // avoid multiple memset() on same buffer
-        const auto &group = pair.second;
-
-        const std::shared_ptr<Track> &t = mTracks[group[0]];
-        memset(t->mainBuffer, 0,
-                mFrameCount * audio_bytes_per_frame(
-                        t->mMixerChannelCount + t->mMixerHapticChannelCount, t->mMixerFormat));
-
-        // now consume data
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            size_t outFrames = mFrameCount;
-            while (outFrames) {
-                t->buffer.frameCount = outFrames;
-                t->bufferProvider->getNextBuffer(&t->buffer);
-                if (t->buffer.raw == NULL) break;
-                outFrames -= t->buffer.frameCount;
-                t->bufferProvider->releaseBuffer(&t->buffer);
-            }
-        }
-    }
-}
-
-// generic code without resampling
-void AudioMixer::process__genericNoResampling()
-{
-    ALOGVV("process__genericNoResampling\n");
-    int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
-
-    for (const auto &pair : mGroups) {
-        // process by group of tracks with same output main buffer to
-        // avoid multiple memset() on same buffer
-        const auto &group = pair.second;
-
-        // acquire buffer
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            t->buffer.frameCount = mFrameCount;
-            t->bufferProvider->getNextBuffer(&t->buffer);
-            t->frameCount = t->buffer.frameCount;
-            t->mIn = t->buffer.raw;
-        }
-
-        int32_t *out = (int *)pair.first;
-        size_t numFrames = 0;
-        do {
-            const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
-            memset(outTemp, 0, sizeof(outTemp));
-            for (const int name : group) {
-                const std::shared_ptr<Track> &t = mTracks[name];
-                int32_t *aux = NULL;
-                if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
-                    aux = t->auxBuffer + numFrames;
-                }
-                for (int outFrames = frameCount; outFrames > 0; ) {
-                    // t->in == nullptr can happen if the track was flushed just after having
-                    // been enabled for mixing.
-                    if (t->mIn == nullptr) {
-                        break;
-                    }
-                    size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
-                    if (inFrames > 0) {
-                        (t.get()->*t->hook)(
-                                outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
-                                inFrames, mResampleTemp.get() /* naked ptr */, aux);
-                        t->frameCount -= inFrames;
-                        outFrames -= inFrames;
-                        if (CC_UNLIKELY(aux != NULL)) {
-                            aux += inFrames;
-                        }
-                    }
-                    if (t->frameCount == 0 && outFrames) {
-                        t->bufferProvider->releaseBuffer(&t->buffer);
-                        t->buffer.frameCount = (mFrameCount - numFrames) -
-                                (frameCount - outFrames);
-                        t->bufferProvider->getNextBuffer(&t->buffer);
-                        t->mIn = t->buffer.raw;
-                        if (t->mIn == nullptr) {
-                            break;
-                        }
-                        t->frameCount = t->buffer.frameCount;
-                    }
-                }
-            }
-
-            const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-            convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
-                    frameCount * t1->mMixerChannelCount);
-            // TODO: fix ugly casting due to choice of out pointer type
-            out = reinterpret_cast<int32_t*>((uint8_t*)out
-                    + frameCount * t1->mMixerChannelCount
-                    * audio_bytes_per_sample(t1->mMixerFormat));
-            numFrames += frameCount;
-        } while (numFrames < mFrameCount);
-
-        // release each track's buffer
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            t->bufferProvider->releaseBuffer(&t->buffer);
-        }
-    }
-}
-
-// generic code with resampling
-void AudioMixer::process__genericResampling()
-{
-    ALOGVV("process__genericResampling\n");
-    int32_t * const outTemp = mOutputTemp.get(); // naked ptr
-    size_t numFrames = mFrameCount;
-
-    for (const auto &pair : mGroups) {
-        const auto &group = pair.second;
-        const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-
-        // clear temp buffer
-        memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
-        for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
-            int32_t *aux = NULL;
-            if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
-                aux = t->auxBuffer;
-            }
-
-            // this is a little goofy, on the resampling case we don't
-            // acquire/release the buffers because it's done by
-            // the resampler.
-            if (t->needs & NEEDS_RESAMPLE) {
-                (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
-            } else {
-
-                size_t outFrames = 0;
-
-                while (outFrames < numFrames) {
-                    t->buffer.frameCount = numFrames - outFrames;
-                    t->bufferProvider->getNextBuffer(&t->buffer);
-                    t->mIn = t->buffer.raw;
-                    // t->mIn == nullptr can happen if the track was flushed just after having
-                    // been enabled for mixing.
-                    if (t->mIn == nullptr) break;
-
-                    (t.get()->*t->hook)(
-                            outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
-                            mResampleTemp.get() /* naked ptr */,
-                            aux != nullptr ? aux + outFrames : nullptr);
-                    outFrames += t->buffer.frameCount;
-
-                    t->bufferProvider->releaseBuffer(&t->buffer);
-                }
-            }
-        }
-        convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
-                outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
-    }
-}
-
-// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__oneTrack16BitsStereoNoResampling()
-{
-    ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
-    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
-            "%zu != 1 tracks enabled", mEnabled.size());
-    const int name = mEnabled[0];
-    const std::shared_ptr<Track> &t = mTracks[name];
-
-    AudioBufferProvider::Buffer& b(t->buffer);
-
-    int32_t* out = t->mainBuffer;
-    float *fout = reinterpret_cast<float*>(out);
-    size_t numFrames = mFrameCount;
-
-    const int16_t vl = t->volume[0];
-    const int16_t vr = t->volume[1];
-    const uint32_t vrl = t->volumeRL;
-    while (numFrames) {
-        b.frameCount = numFrames;
-        t->bufferProvider->getNextBuffer(&b);
-        const int16_t *in = b.i16;
-
-        // in == NULL can happen if the track was flushed just after having
-        // been enabled for mixing.
-        if (in == NULL || (((uintptr_t)in) & 3)) {
-            if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
-                 memset((char*)fout, 0, numFrames
-                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
-            } else {
-                 memset((char*)out, 0, numFrames
-                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
-            }
-            ALOGE_IF((((uintptr_t)in) & 3),
-                    "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
-                    " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
-                    in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
-            return;
-        }
-        size_t outFrames = b.frameCount;
-
-        switch (t->mMixerFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            do {
-                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                in += 2;
-                int32_t l = mulRL(1, rl, vrl);
-                int32_t r = mulRL(0, rl, vrl);
-                *fout++ = float_from_q4_27(l);
-                *fout++ = float_from_q4_27(r);
-                // Note: In case of later int16_t sink output,
-                // conversion and clamping is done by memcpy_to_i16_from_float().
-            } while (--outFrames);
-            break;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
-                // volume is boosted, so we might need to clamp even though
-                // we process only one track.
-                do {
-                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                    in += 2;
-                    int32_t l = mulRL(1, rl, vrl) >> 12;
-                    int32_t r = mulRL(0, rl, vrl) >> 12;
-                    // clamping...
-                    l = clamp16(l);
-                    r = clamp16(r);
-                    *out++ = (r<<16) | (l & 0xFFFF);
-                } while (--outFrames);
-            } else {
-                do {
-                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
-                    in += 2;
-                    int32_t l = mulRL(1, rl, vrl) >> 12;
-                    int32_t r = mulRL(0, rl, vrl) >> 12;
-                    *out++ = (r<<16) | (l & 0xFFFF);
-                } while (--outFrames);
-            }
-            break;
-        default:
-            LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
-        }
-        numFrames -= b.frameCount;
-        t->bufferProvider->releaseBuffer(&b);
-    }
-}
-
 /*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
 
 /*static*/ void AudioMixer::sInitRoutine()
@@ -1656,211 +523,71 @@
     DownmixerBufferProvider::init(); // for the downmixer
 }
 
-/* TODO: consider whether this level of optimization is necessary.
- * Perhaps just stick with a single for loop.
- */
-
-// 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))
-
-/* 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
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
-        typename TO, typename TI, typename TV, typename TA, typename TAV>
-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)
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack()
 {
-    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;
-    }
+    return std::make_shared<Track>();
 }
 
-/* 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
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
-        typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
-        const TI* in, TA* aux, const TV *vol, TAV vola)
+status_t AudioMixer::postCreateTrack(TrackBase *track)
 {
-    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;
+    Track* t = static_cast<Track*>(track);
+
+    audio_channel_mask_t channelMask = t->channelMask;
+    t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
+    t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
+    channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
+    t->channelCount = audio_channel_count_from_out_mask(channelMask);
+    ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+            "Non-stereo channel mask: %d\n", channelMask);
+    t->channelMask = channelMask;
+    t->mInputBufferProvider = NULL;
+    t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
+    t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+    // haptic
+    t->mHapticPlaybackEnabled = false;
+    t->mHapticIntensity = HAPTIC_SCALE_NONE;
+    t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
+    t->mMixerHapticChannelCount = 0;
+    t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
+    t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
+    t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
+    t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
+    t->mKeepContractedChannels = false;
+    // Check the downmixing (or upmixing) requirements.
+    status_t status = t->prepareForDownmix();
+    if (status != OK) {
+        ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+        return BAD_VALUE;
     }
+    // prepareForDownmix() may change mDownmixRequiresFormat
+    ALOGVV("mMixerFormat:%#x  mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+    t->prepareForReformat();
+    t->prepareForAdjustChannelsNonDestructive(mFrameCount);
+    t->prepareForAdjustChannels();
+    return OK;
 }
 
-/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
-    typename TO, typename TI, typename TA>
-void AudioMixer::Track::volumeMix(TO *out, size_t outFrames,
-        const TI *in, TA *aux, bool ramp)
+void AudioMixer::preProcess()
 {
-    if (USEFLOATVOL) {
-        if (ramp) {
-            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    mPrevVolume, mVolumeInc,
-#ifdef FLOAT_AUX
-                    &mPrevAuxLevel, mAuxInc
-#else
-                    &prevAuxLevel, auxInc
-#endif
-                );
-            if (ADJUSTVOL) {
-                adjustVolumeRamp(aux != NULL, true);
-            }
-        } else {
-            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    mVolume,
-#ifdef FLOAT_AUX
-                    mAuxLevel
-#else
-                    auxLevel
-#endif
-            );
-        }
-    } else {
-        if (ramp) {
-            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    prevVolume, volumeInc, &prevAuxLevel, auxInc);
-            if (ADJUSTVOL) {
-                adjustVolumeRamp(aux != NULL);
-            }
-        } else {
-            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
-                    volume, auxLevel);
+    for (const auto &pair : mTracks) {
+        // Clear contracted buffer before processing if contracted channels are saved
+        const std::shared_ptr<TrackBase> &tb = pair.second;
+        Track *t = static_cast<Track*>(tb.get());
+        if (t->mKeepContractedChannels) {
+            t->clearContractedBuffer();
         }
     }
 }
 
-/* This process hook is called when there is a single track without
- * aux buffer, volume ramp, or resampling.
- * TODO: Update the hook selection: this can properly handle aux and ramp.
- *
- * 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
- * TA: int32_t (Q4.27)
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process__noResampleOneTrack()
+void AudioMixer::postProcess()
 {
-    ALOGVV("process__noResampleOneTrack\n");
-    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
-            "%zu != 1 tracks enabled", mEnabled.size());
-    const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
-    const uint32_t channels = t->mMixerChannelCount;
-    TO* out = reinterpret_cast<TO*>(t->mainBuffer);
-    TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
-    const bool ramp = t->needsRamp();
-
-    for (size_t numFrames = mFrameCount; numFrames > 0; ) {
-        AudioBufferProvider::Buffer& b(t->buffer);
-        // get input buffer
-        b.frameCount = numFrames;
-        t->bufferProvider->getNextBuffer(&b);
-        const TI *in = reinterpret_cast<TI*>(b.raw);
-
-        // in == NULL can happen if the track was flushed just after having
-        // been enabled for mixing.
-        if (in == NULL || (((uintptr_t)in) & 3)) {
-            memset(out, 0, numFrames
-                    * channels * audio_bytes_per_sample(t->mMixerFormat));
-            ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
-                    "buffer %p track %p, channels %d, needs %#x",
-                    in, &t, t->channelCount, t->needs);
-            return;
-        }
-
-        const size_t outFrames = b.frameCount;
-        t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
-                out, outFrames, in, aux, ramp);
-
-        out += outFrames * channels;
-        if (aux != NULL) {
-            aux += outFrames;
-        }
-        numFrames -= b.frameCount;
-
-        // release buffer
-        t->bufferProvider->releaseBuffer(&b);
-    }
-    if (ramp) {
-        t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
-    }
-}
-
-void AudioMixer::processHapticData()
-{
+    // Process haptic data.
     // Need to keep consistent with VibrationEffect.scale(int, float, int)
     for (const auto &pair : mGroups) {
         // process by group of tracks with same output main buffer.
         const auto &group = pair.second;
         for (const int name : group) {
-            const std::shared_ptr<Track> &t = mTracks[name];
+            const std::shared_ptr<Track> &t = getTrack(name);
             if (t->mHapticPlaybackEnabled) {
                 size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount;
                 float gamma = t->getHapticScaleGamma();
@@ -1887,225 +614,5 @@
     }
 }
 
-/* This track hook is called to do resampling then mixing,
- * pulling from the track's upstream AudioBufferProvider.
- *
- * 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
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
-{
-    ALOGVV("track__Resample\n");
-    mResampler->setSampleRate(sampleRate);
-    const bool ramp = needsRamp();
-    if (ramp || aux != NULL) {
-        // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
-        // if aux != NULL: resample with unity gain to temp buffer then apply send level.
-
-        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
-        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
-        mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
-
-        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
-                out, outFrameCount, temp, aux, ramp);
-
-    } else { // constant volume gain
-        mResampler->setVolume(mVolume[0], mVolume[1]);
-        mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
-    }
-}
-
-/* This track hook is called to mix a track, when no resampling is required.
- * The input buffer should be present in in.
- *
- * 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
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux)
-{
-    ALOGVV("track__NoResample\n");
-    const TI *in = static_cast<const TI *>(mIn);
-
-    volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
-            out, frameCount, in, aux, needsRamp());
-
-    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
-    // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
-    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
-    mIn = in;
-}
-
-/* The Mixer engine generates either int32_t (Q4_27) or float data.
- * We use this function to convert the engine buffers
- * to the desired mixer output format, either int16_t (Q.15) or float.
- */
-/* static */
-void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
-        void *in, audio_format_t mixerInFormat, size_t sampleCount)
-{
-    switch (mixerInFormat) {
-    case AUDIO_FORMAT_PCM_FLOAT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
-            break;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
-            break;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    case AUDIO_FORMAT_PCM_16_BIT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
-            break;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
-            break;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    default:
-        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-        break;
-    }
-}
-
-/* Returns the proper track hook to use for mixing the track into the output buffer.
- */
-/* static */
-AudioMixer::hook_t AudioMixer::Track::getTrackHook(int trackType, uint32_t channelCount,
-        audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
-{
-    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
-        switch (trackType) {
-        case TRACKTYPE_NOP:
-            return &Track::track__nop;
-        case TRACKTYPE_RESAMPLE:
-            return &Track::track__genericResample;
-        case TRACKTYPE_NORESAMPLEMONO:
-            return &Track::track__16BitsMono;
-        case TRACKTYPE_NORESAMPLE:
-            return &Track::track__16BitsStereo;
-        default:
-            LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
-            break;
-        }
-    }
-    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
-    switch (trackType) {
-    case TRACKTYPE_NOP:
-        return &Track::track__nop;
-    case TRACKTYPE_RESAMPLE:
-        switch (mixerInFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t) &Track::track__Resample<
-                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t) &Track::track__Resample<
-                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-            break;
-        }
-        break;
-    case TRACKTYPE_NORESAMPLEMONO:
-        switch (mixerInFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                            MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                            MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-            break;
-        }
-        break;
-    case TRACKTYPE_NORESAMPLE:
-        switch (mixerInFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return (AudioMixer::hook_t) &Track::track__NoResample<
-                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-            break;
-        }
-        break;
-    default:
-        LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
-        break;
-    }
-    return NULL;
-}
-
-/* Returns the proper process hook for mixing tracks. Currently works only for
- * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
- *
- * TODO: Due to the special mixing considerations of duplicating to
- * a stereo output track, the input track cannot be MONO.  This should be
- * prevented by the caller.
- */
-/* static */
-AudioMixer::process_hook_t AudioMixer::getProcessHook(
-        int processType, uint32_t channelCount,
-        audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
-{
-    if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
-        LOG_ALWAYS_FATAL("bad processType: %d", processType);
-        return NULL;
-    }
-    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
-        return &AudioMixer::process__oneTrack16BitsStereoNoResampling;
-    }
-    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
-    switch (mixerInFormat) {
-    case AUDIO_FORMAT_PCM_FLOAT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    case AUDIO_FORMAT_PCM_16_BIT:
-        switch (mixerOutFormat) {
-        case AUDIO_FORMAT_PCM_FLOAT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        case AUDIO_FORMAT_PCM_16_BIT:
-            return &AudioMixer::process__noResampleOneTrack<
-                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
-        default:
-            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
-            break;
-        }
-        break;
-    default:
-        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
-        break;
-    }
-    return NULL;
-}
-
 // ----------------------------------------------------------------------------
 } // namespace android
diff --git a/media/libaudioprocessing/AudioMixerBase.cpp b/media/libaudioprocessing/AudioMixerBase.cpp
new file mode 100644
index 0000000..75c077d
--- /dev/null
+++ b/media/libaudioprocessing/AudioMixerBase.cpp
@@ -0,0 +1,1692 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#define LOG_TAG "AudioMixer"
+//#define LOG_NDEBUG 0
+
+#include <sstream>
+#include <string.h>
+
+#include <audio_utils/primitives.h>
+#include <cutils/compiler.h>
+#include <media/AudioMixerBase.h>
+#include <utils/Log.h>
+
+#include "AudioMixerOps.h"
+
+// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+// Look for MONO_HACK for any Mono hack involving legacy mono channel to
+// stereo channel conversion.
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf  // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
+static constexpr int BLOCKSIZE = 16;
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+bool AudioMixerBase::isValidFormat(audio_format_t format) const
+{
+    switch (format) {
+    case AUDIO_FORMAT_PCM_8_BIT:
+    case AUDIO_FORMAT_PCM_16_BIT:
+    case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+    case AUDIO_FORMAT_PCM_32_BIT:
+    case AUDIO_FORMAT_PCM_FLOAT:
+        return true;
+    default:
+        return false;
+    }
+}
+
+bool AudioMixerBase::isValidChannelMask(audio_channel_mask_t channelMask) const
+{
+    return audio_channel_count_from_out_mask(channelMask) <= MAX_NUM_CHANNELS;
+}
+
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixerBase::preCreateTrack()
+{
+    return std::make_shared<TrackBase>();
+}
+
+status_t AudioMixerBase::create(
+        int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
+{
+    LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
+
+    if (!isValidChannelMask(channelMask)) {
+        ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
+        return BAD_VALUE;
+    }
+    if (!isValidFormat(format)) {
+        ALOGE("%s invalid format: %#x", __func__, format);
+        return BAD_VALUE;
+    }
+
+    auto t = preCreateTrack();
+    {
+        // TODO: move initialization to the Track constructor.
+        // assume default parameters for the track, except where noted below
+        t->needs = 0;
+
+        // Integer volume.
+        // Currently integer volume is kept for the legacy integer mixer.
+        // Will be removed when the legacy mixer path is removed.
+        t->volume[0] = 0;
+        t->volume[1] = 0;
+        t->prevVolume[0] = 0 << 16;
+        t->prevVolume[1] = 0 << 16;
+        t->volumeInc[0] = 0;
+        t->volumeInc[1] = 0;
+        t->auxLevel = 0;
+        t->auxInc = 0;
+        t->prevAuxLevel = 0;
+
+        // Floating point volume.
+        t->mVolume[0] = 0.f;
+        t->mVolume[1] = 0.f;
+        t->mPrevVolume[0] = 0.f;
+        t->mPrevVolume[1] = 0.f;
+        t->mVolumeInc[0] = 0.;
+        t->mVolumeInc[1] = 0.;
+        t->mAuxLevel = 0.;
+        t->mAuxInc = 0.;
+        t->mPrevAuxLevel = 0.;
+
+        // no initialization needed
+        // t->frameCount
+        t->channelCount = audio_channel_count_from_out_mask(channelMask);
+        t->enabled = false;
+        ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+                "Non-stereo channel mask: %d\n", channelMask);
+        t->channelMask = channelMask;
+        t->sessionId = sessionId;
+        // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+        t->bufferProvider = NULL;
+        t->buffer.raw = NULL;
+        // no initialization needed
+        // t->buffer.frameCount
+        t->hook = NULL;
+        t->mIn = NULL;
+        t->sampleRate = mSampleRate;
+        // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+        t->mainBuffer = NULL;
+        t->auxBuffer = NULL;
+        t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+        t->mFormat = format;
+        t->mMixerInFormat = kUseFloat && kUseNewMixer ?
+                AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+        t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
+                AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
+        t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+        status_t status = postCreateTrack(t.get());
+        if (status != OK) return status;
+        mTracks[name] = t;
+        return OK;
+    }
+}
+
+// Called when channel masks have changed for a track name
+bool AudioMixerBase::setChannelMasks(int name,
+        audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    if (trackChannelMask == track->channelMask && mixerChannelMask == track->mMixerChannelMask) {
+        return false;  // no need to change
+    }
+    // always recompute for both channel masks even if only one has changed.
+    const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
+    const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
+
+    ALOG_ASSERT(trackChannelCount && mixerChannelCount);
+    track->channelMask = trackChannelMask;
+    track->channelCount = trackChannelCount;
+    track->mMixerChannelMask = mixerChannelMask;
+    track->mMixerChannelCount = mixerChannelCount;
+
+    // Resampler channels may have changed.
+    track->recreateResampler(mSampleRate);
+    return true;
+}
+
+void AudioMixerBase::destroy(int name)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    ALOGV("deleteTrackName(%d)", name);
+
+    if (mTracks[name]->enabled) {
+        invalidate();
+    }
+    mTracks.erase(name); // deallocate track
+}
+
+void AudioMixerBase::enable(int name)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    if (!track->enabled) {
+        track->enabled = true;
+        ALOGV("enable(%d)", name);
+        invalidate();
+    }
+}
+
+void AudioMixerBase::disable(int name)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    if (track->enabled) {
+        track->enabled = false;
+        ALOGV("disable(%d)", name);
+        invalidate();
+    }
+}
+
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition from the previous
+ * volume to the set volume.  ramp controls the duration of the transition.
+ * Its value is typically one state framecount period, but may also be 0,
+ * meaning "immediate."
+ *
+ * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
+ * even if there is a nonzero floating point increment (in that case, the volume
+ * change is immediate).  This restriction should be changed when the legacy mixer
+ * is removed (see #2).
+ * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
+ * when no longer needed.
+ *
+ * @param newVolume set volume target in floating point [0.0, 1.0].
+ * @param ramp number of frames to increment over. if ramp is 0, the volume
+ * should be set immediately.  Currently ramp should not exceed 65535 (frames).
+ * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
+ * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
+ * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
+ * @param pSetVolume pointer to the float target volume, set on return.
+ * @param pPrevVolume pointer to the float previous volume, set on return.
+ * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
+ * @return true if the volume has changed, false if volume is same.
+ */
+static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
+        int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
+        float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
+    // check floating point volume to see if it is identical to the previously
+    // set volume.
+    // We do not use a tolerance here (and reject changes too small)
+    // as it may be confusing to use a different value than the one set.
+    // If the resulting volume is too small to ramp, it is a direct set of the volume.
+    if (newVolume == *pSetVolume) {
+        return false;
+    }
+    if (newVolume < 0) {
+        newVolume = 0; // should not have negative volumes
+    } else {
+        switch (fpclassify(newVolume)) {
+        case FP_SUBNORMAL:
+        case FP_NAN:
+            newVolume = 0;
+            break;
+        case FP_ZERO:
+            break; // zero volume is fine
+        case FP_INFINITE:
+            // Infinite volume could be handled consistently since
+            // floating point math saturates at infinities,
+            // but we limit volume to unity gain float.
+            // ramp = 0; break;
+            //
+            newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+            break;
+        case FP_NORMAL:
+        default:
+            // Floating point does not have problems with overflow wrap
+            // that integer has.  However, we limit the volume to
+            // unity gain here.
+            // TODO: Revisit the volume limitation and perhaps parameterize.
+            if (newVolume > AudioMixerBase::UNITY_GAIN_FLOAT) {
+                newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+            }
+            break;
+        }
+    }
+
+    // set floating point volume ramp
+    if (ramp != 0) {
+        // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
+        // is no computational mismatch; hence equality is checked here.
+        ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
+                " prev:%f  set_to:%f", *pPrevVolume, *pSetVolume);
+        const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
+        // could be inf, cannot be nan, subnormal
+        const float maxv = std::max(newVolume, *pPrevVolume);
+
+        if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
+                && maxv + inc != maxv) { // inc must make forward progress
+            *pVolumeInc = inc;
+            // ramp is set now.
+            // Note: if newVolume is 0, then near the end of the ramp,
+            // it may be possible that the ramped volume may be subnormal or
+            // temporarily negative by a small amount or subnormal due to floating
+            // point inaccuracies.
+        } else {
+            ramp = 0; // ramp not allowed
+        }
+    }
+
+    // compute and check integer volume, no need to check negative values
+    // The integer volume is limited to "unity_gain" to avoid wrapping and other
+    // audio artifacts, so it never reaches the range limit of U4.28.
+    // We safely use signed 16 and 32 bit integers here.
+    const float scaledVolume = newVolume * AudioMixerBase::UNITY_GAIN_INT; // not neg, subnormal, nan
+    const int32_t intVolume = (scaledVolume >= (float)AudioMixerBase::UNITY_GAIN_INT) ?
+            AudioMixerBase::UNITY_GAIN_INT : (int32_t)scaledVolume;
+
+    // set integer volume ramp
+    if (ramp != 0) {
+        // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
+        // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
+        // is no computational mismatch; hence equality is checked here.
+        ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
+                " prev:%d  set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
+        const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
+
+        if (inc != 0) { // inc must make forward progress
+            *pIntVolumeInc = inc;
+        } else {
+            ramp = 0; // ramp not allowed
+        }
+    }
+
+    // if no ramp, or ramp not allowed, then clear float and integer increments
+    if (ramp == 0) {
+        *pVolumeInc = 0;
+        *pPrevVolume = newVolume;
+        *pIntVolumeInc = 0;
+        *pIntPrevVolume = intVolume << 16;
+    }
+    *pSetVolume = newVolume;
+    *pIntSetVolume = intVolume;
+    return true;
+}
+
+void AudioMixerBase::setParameter(int name, int target, int param, void *value)
+{
+    LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+    const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+    int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+    int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
+
+    switch (target) {
+
+    case TRACK:
+        switch (param) {
+        case CHANNEL_MASK: {
+            const audio_channel_mask_t trackChannelMask =
+                static_cast<audio_channel_mask_t>(valueInt);
+            if (setChannelMasks(name, trackChannelMask, track->mMixerChannelMask)) {
+                ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
+                invalidate();
+            }
+            } break;
+        case MAIN_BUFFER:
+            if (track->mainBuffer != valueBuf) {
+                track->mainBuffer = valueBuf;
+                ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+                invalidate();
+            }
+            break;
+        case AUX_BUFFER:
+            if (track->auxBuffer != valueBuf) {
+                track->auxBuffer = valueBuf;
+                ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+                invalidate();
+            }
+            break;
+        case FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track->mFormat != format) {
+                ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
+                track->mFormat = format;
+                ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
+                invalidate();
+            }
+            } break;
+        case MIXER_FORMAT: {
+            audio_format_t format = static_cast<audio_format_t>(valueInt);
+            if (track->mMixerFormat != format) {
+                track->mMixerFormat = format;
+                ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
+            }
+            } break;
+        case MIXER_CHANNEL_MASK: {
+            const audio_channel_mask_t mixerChannelMask =
+                    static_cast<audio_channel_mask_t>(valueInt);
+            if (setChannelMasks(name, track->channelMask, mixerChannelMask)) {
+                ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
+                invalidate();
+            }
+            } break;
+        default:
+            LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
+        }
+        break;
+
+    case RESAMPLE:
+        switch (param) {
+        case SAMPLE_RATE:
+            ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
+            if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
+                ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+                        uint32_t(valueInt));
+                invalidate();
+            }
+            break;
+        case RESET:
+            track->resetResampler();
+            invalidate();
+            break;
+        case REMOVE:
+            track->mResampler.reset(nullptr);
+            track->sampleRate = mSampleRate;
+            invalidate();
+            break;
+        default:
+            LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
+        }
+        break;
+
+    case RAMP_VOLUME:
+    case VOLUME:
+        switch (param) {
+        case AUXLEVEL:
+            if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                    target == RAMP_VOLUME ? mFrameCount : 0,
+                    &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
+                    &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
+                ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+                        target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
+                invalidate();
+            }
+            break;
+        default:
+            if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
+                if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+                        target == RAMP_VOLUME ? mFrameCount : 0,
+                        &track->volume[param - VOLUME0],
+                        &track->prevVolume[param - VOLUME0],
+                        &track->volumeInc[param - VOLUME0],
+                        &track->mVolume[param - VOLUME0],
+                        &track->mPrevVolume[param - VOLUME0],
+                        &track->mVolumeInc[param - VOLUME0])) {
+                    ALOGV("setParameter(%s, VOLUME%d: %04x)",
+                            target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+                                    track->volume[param - VOLUME0]);
+                    invalidate();
+                }
+            } else {
+                LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
+            }
+        }
+        break;
+
+    default:
+        LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
+    }
+}
+
+bool AudioMixerBase::TrackBase::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+{
+    if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
+        if (sampleRate != trackSampleRate) {
+            sampleRate = trackSampleRate;
+            if (mResampler.get() == nullptr) {
+                ALOGV("Creating resampler from track %d Hz to device %d Hz",
+                        trackSampleRate, devSampleRate);
+                AudioResampler::src_quality quality;
+                // force lowest quality level resampler if use case isn't music or video
+                // FIXME this is flawed for dynamic sample rates, as we choose the resampler
+                // quality level based on the initial ratio, but that could change later.
+                // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
+                if (isMusicRate(trackSampleRate)) {
+                    quality = AudioResampler::DEFAULT_QUALITY;
+                } else {
+                    quality = AudioResampler::DYN_LOW_QUALITY;
+                }
+
+                // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+                // but if none exists, it is the channel count (1 for mono).
+                const int resamplerChannelCount = getOutputChannelCount();
+                ALOGVV("Creating resampler:"
+                        " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
+                        mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
+                mResampler.reset(AudioResampler::create(
+                        mMixerInFormat,
+                        resamplerChannelCount,
+                        devSampleRate, quality));
+            }
+            return true;
+        }
+    }
+    return false;
+}
+
+/* Checks to see if the volume ramp has completed and clears the increment
+ * variables appropriately.
+ *
+ * FIXME: There is code to handle int/float ramp variable switchover should it not
+ * complete within a mixer buffer processing call, but it is preferred to avoid switchover
+ * due to precision issues.  The switchover code is included for legacy code purposes
+ * and can be removed once the integer volume is removed.
+ *
+ * It is not sufficient to clear only the volumeInc integer variable because
+ * if one channel requires ramping, all channels are ramped.
+ *
+ * There is a bit of duplicated code here, but it keeps backward compatibility.
+ */
+void AudioMixerBase::TrackBase::adjustVolumeRamp(bool aux, bool useFloat)
+{
+    if (useFloat) {
+        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+            if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
+                     (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
+                volumeInc[i] = 0;
+                prevVolume[i] = volume[i] << 16;
+                mVolumeInc[i] = 0.;
+                mPrevVolume[i] = mVolume[i];
+            } else {
+                //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
+                prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
+            }
+        }
+    } else {
+        for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+            if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
+                    ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
+                volumeInc[i] = 0;
+                prevVolume[i] = volume[i] << 16;
+                mVolumeInc[i] = 0.;
+                mPrevVolume[i] = mVolume[i];
+            } else {
+                //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
+                mPrevVolume[i]  = float_from_u4_28(prevVolume[i]);
+            }
+        }
+    }
+
+    if (aux) {
+#ifdef FLOAT_AUX
+        if (useFloat) {
+            if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
+                    (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
+                auxInc = 0;
+                prevAuxLevel = auxLevel << 16;
+                mAuxInc = 0.f;
+                mPrevAuxLevel = mAuxLevel;
+            }
+        } else
+#endif
+        if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
+                (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
+            auxInc = 0;
+            prevAuxLevel = auxLevel << 16;
+            mAuxInc = 0.f;
+            mPrevAuxLevel = mAuxLevel;
+        }
+    }
+}
+
+void AudioMixerBase::TrackBase::recreateResampler(uint32_t devSampleRate)
+{
+    if (mResampler.get() != nullptr) {
+        const uint32_t resetToSampleRate = sampleRate;
+        mResampler.reset(nullptr);
+        sampleRate = devSampleRate; // without resampler, track rate is device sample rate.
+        // recreate the resampler with updated format, channels, saved sampleRate.
+        setResampler(resetToSampleRate /*trackSampleRate*/, devSampleRate);
+    }
+}
+
+size_t AudioMixerBase::getUnreleasedFrames(int name) const
+{
+    const auto it = mTracks.find(name);
+    if (it != mTracks.end()) {
+        return it->second->getUnreleasedFrames();
+    }
+    return 0;
+}
+
+std::string AudioMixerBase::trackNames() const
+{
+    std::stringstream ss;
+    for (const auto &pair : mTracks) {
+        ss << pair.first << " ";
+    }
+    return ss.str();
+}
+
+void AudioMixerBase::process__validate()
+{
+    // TODO: fix all16BitsStereNoResample logic to
+    // either properly handle muted tracks (it should ignore them)
+    // or remove altogether as an obsolete optimization.
+    bool all16BitsStereoNoResample = true;
+    bool resampling = false;
+    bool volumeRamp = false;
+
+    mEnabled.clear();
+    mGroups.clear();
+    for (const auto &pair : mTracks) {
+        const int name = pair.first;
+        const std::shared_ptr<TrackBase> &t = pair.second;
+        if (!t->enabled) continue;
+
+        mEnabled.emplace_back(name);  // we add to mEnabled in order of name.
+        mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
+
+        uint32_t n = 0;
+        // FIXME can overflow (mask is only 3 bits)
+        n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
+        if (t->doesResample()) {
+            n |= NEEDS_RESAMPLE;
+        }
+        if (t->auxLevel != 0 && t->auxBuffer != NULL) {
+            n |= NEEDS_AUX;
+        }
+
+        if (t->volumeInc[0]|t->volumeInc[1]) {
+            volumeRamp = true;
+        } else if (!t->doesResample() && t->volumeRL == 0) {
+            n |= NEEDS_MUTE;
+        }
+        t->needs = n;
+
+        if (n & NEEDS_MUTE) {
+            t->hook = &TrackBase::track__nop;
+        } else {
+            if (n & NEEDS_AUX) {
+                all16BitsStereoNoResample = false;
+            }
+            if (n & NEEDS_RESAMPLE) {
+                all16BitsStereoNoResample = false;
+                resampling = true;
+                t->hook = TrackBase::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
+                        t->mMixerInFormat, t->mMixerFormat);
+                ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+                        "Track %d needs downmix + resample", name);
+            } else {
+                if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
+                    t->hook = TrackBase::getTrackHook(
+                            (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO  // TODO: MONO_HACK
+                                    && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
+                                ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
+                            t->mMixerChannelCount,
+                            t->mMixerInFormat, t->mMixerFormat);
+                    all16BitsStereoNoResample = false;
+                }
+                if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
+                    t->hook = TrackBase::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
+                            t->mMixerInFormat, t->mMixerFormat);
+                    ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+                            "Track %d needs downmix", name);
+                }
+            }
+        }
+    }
+
+    // select the processing hooks
+    mHook = &AudioMixerBase::process__nop;
+    if (mEnabled.size() > 0) {
+        if (resampling) {
+            if (mOutputTemp.get() == nullptr) {
+                mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+            }
+            if (mResampleTemp.get() == nullptr) {
+                mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+            }
+            mHook = &AudioMixerBase::process__genericResampling;
+        } else {
+            // we keep temp arrays around.
+            mHook = &AudioMixerBase::process__genericNoResampling;
+            if (all16BitsStereoNoResample && !volumeRamp) {
+                if (mEnabled.size() == 1) {
+                    const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+                    if ((t->needs & NEEDS_MUTE) == 0) {
+                        // The check prevents a muted track from acquiring a process hook.
+                        //
+                        // This is dangerous if the track is MONO as that requires
+                        // special case handling due to implicit channel duplication.
+                        // Stereo or Multichannel should actually be fine here.
+                        mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                                t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+                    }
+                }
+            }
+        }
+    }
+
+    ALOGV("mixer configuration change: %zu "
+        "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
+        mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
+
+    process();
+
+    // Now that the volume ramp has been done, set optimal state and
+    // track hooks for subsequent mixer process
+    if (mEnabled.size() > 0) {
+        bool allMuted = true;
+
+        for (const int name : mEnabled) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            if (!t->doesResample() && t->volumeRL == 0) {
+                t->needs |= NEEDS_MUTE;
+                t->hook = &TrackBase::track__nop;
+            } else {
+                allMuted = false;
+            }
+        }
+        if (allMuted) {
+            mHook = &AudioMixerBase::process__nop;
+        } else if (all16BitsStereoNoResample) {
+            if (mEnabled.size() == 1) {
+                //const int i = 31 - __builtin_clz(enabledTracks);
+                const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+                // Muted single tracks handled by allMuted above.
+                mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+                        t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+            }
+        }
+    }
+}
+
+void AudioMixerBase::TrackBase::track__genericResample(
+        int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
+{
+    ALOGVV("track__genericResample\n");
+    mResampler->setSampleRate(sampleRate);
+
+    // ramp gain - resample to temp buffer and scale/mix in 2nd step
+    if (aux != NULL) {
+        // always resample with unity gain when sending to auxiliary buffer to be able
+        // to apply send level after resampling
+        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
+        mResampler->resample(temp, outFrameCount, bufferProvider);
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            volumeRampStereo(out, outFrameCount, temp, aux);
+        } else {
+            volumeStereo(out, outFrameCount, temp, aux);
+        }
+    } else {
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+            memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
+            mResampler->resample(temp, outFrameCount, bufferProvider);
+            volumeRampStereo(out, outFrameCount, temp, aux);
+        }
+
+        // constant gain
+        else {
+            mResampler->setVolume(mVolume[0], mVolume[1]);
+            mResampler->resample(out, outFrameCount, bufferProvider);
+        }
+    }
+}
+
+void AudioMixerBase::TrackBase::track__nop(int32_t* out __unused,
+        size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
+{
+}
+
+void AudioMixerBase::TrackBase::volumeRampStereo(
+        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+    int32_t vl = prevVolume[0];
+    int32_t vr = prevVolume[1];
+    const int32_t vlInc = volumeInc[0];
+    const int32_t vrInc = volumeInc[1];
+
+    //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+    //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
+    //       (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+    // ramp volume
+    if (CC_UNLIKELY(aux != NULL)) {
+        int32_t va = prevAuxLevel;
+        const int32_t vaInc = auxInc;
+        int32_t l;
+        int32_t r;
+
+        do {
+            l = (*temp++ >> 12);
+            r = (*temp++ >> 12);
+            *out++ += (vl >> 16) * l;
+            *out++ += (vr >> 16) * r;
+            *aux++ += (va >> 17) * (l + r);
+            vl += vlInc;
+            vr += vrInc;
+            va += vaInc;
+        } while (--frameCount);
+        prevAuxLevel = va;
+    } else {
+        do {
+            *out++ += (vl >> 16) * (*temp++ >> 12);
+            *out++ += (vr >> 16) * (*temp++ >> 12);
+            vl += vlInc;
+            vr += vrInc;
+        } while (--frameCount);
+    }
+    prevVolume[0] = vl;
+    prevVolume[1] = vr;
+    adjustVolumeRamp(aux != NULL);
+}
+
+void AudioMixerBase::TrackBase::volumeStereo(
+        int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+    const int16_t vl = volume[0];
+    const int16_t vr = volume[1];
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        const int16_t va = auxLevel;
+        do {
+            int16_t l = (int16_t)(*temp++ >> 12);
+            int16_t r = (int16_t)(*temp++ >> 12);
+            out[0] = mulAdd(l, vl, out[0]);
+            int16_t a = (int16_t)(((int32_t)l + r) >> 1);
+            out[1] = mulAdd(r, vr, out[1]);
+            out += 2;
+            aux[0] = mulAdd(a, va, aux[0]);
+            aux++;
+        } while (--frameCount);
+    } else {
+        do {
+            int16_t l = (int16_t)(*temp++ >> 12);
+            int16_t r = (int16_t)(*temp++ >> 12);
+            out[0] = mulAdd(l, vl, out[0]);
+            out[1] = mulAdd(r, vr, out[1]);
+            out += 2;
+        } while (--frameCount);
+    }
+}
+
+void AudioMixerBase::TrackBase::track__16BitsStereo(
+        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+    ALOGVV("track__16BitsStereo\n");
+    const int16_t *in = static_cast<const int16_t *>(mIn);
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        int32_t l;
+        int32_t r;
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            int32_t va = prevAuxLevel;
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+            const int32_t vaInc = auxInc;
+            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                l = (int32_t)*in++;
+                r = (int32_t)*in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * r;
+                *aux++ += (va >> 17) * (l + r);
+                vl += vlInc;
+                vr += vrInc;
+                va += vaInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            prevAuxLevel = va;
+            adjustVolumeRamp(true);
+        }
+
+        // constant gain
+        else {
+            const uint32_t vrl = volumeRL;
+            const int16_t va = (int16_t)auxLevel;
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
+                in += 2;
+                out[0] = mulAddRL(1, rl, vrl, out[0]);
+                out[1] = mulAddRL(0, rl, vrl, out[1]);
+                out += 2;
+                aux[0] = mulAdd(a, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+
+            // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //        t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //        (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                *out++ += (vl >> 16) * (int32_t) *in++;
+                *out++ += (vr >> 16) * (int32_t) *in++;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            adjustVolumeRamp(false);
+        }
+
+        // constant gain
+        else {
+            const uint32_t vrl = volumeRL;
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                in += 2;
+                out[0] = mulAddRL(1, rl, vrl, out[0]);
+                out[1] = mulAddRL(0, rl, vrl, out[1]);
+                out += 2;
+            } while (--frameCount);
+        }
+    }
+    mIn = in;
+}
+
+void AudioMixerBase::TrackBase::track__16BitsMono(
+        int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+    ALOGVV("track__16BitsMono\n");
+    const int16_t *in = static_cast<int16_t const *>(mIn);
+
+    if (CC_UNLIKELY(aux != NULL)) {
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            int32_t va = prevAuxLevel;
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+            const int32_t vaInc = auxInc;
+
+            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                int32_t l = *in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * l;
+                *aux++ += (va >> 16) * l;
+                vl += vlInc;
+                vr += vrInc;
+                va += vaInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            prevAuxLevel = va;
+            adjustVolumeRamp(true);
+        }
+        // constant gain
+        else {
+            const int16_t vl = volume[0];
+            const int16_t vr = volume[1];
+            const int16_t va = (int16_t)auxLevel;
+            do {
+                int16_t l = *in++;
+                out[0] = mulAdd(l, vl, out[0]);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+                aux[0] = mulAdd(l, va, aux[0]);
+                aux++;
+            } while (--frameCount);
+        }
+    } else {
+        // ramp gain
+        if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+            int32_t vl = prevVolume[0];
+            int32_t vr = prevVolume[1];
+            const int32_t vlInc = volumeInc[0];
+            const int32_t vrInc = volumeInc[1];
+
+            // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+            //         t, vlInc/65536.0f, vl/65536.0f, volume[0],
+            //         (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+            do {
+                int32_t l = *in++;
+                *out++ += (vl >> 16) * l;
+                *out++ += (vr >> 16) * l;
+                vl += vlInc;
+                vr += vrInc;
+            } while (--frameCount);
+
+            prevVolume[0] = vl;
+            prevVolume[1] = vr;
+            adjustVolumeRamp(false);
+        }
+        // constant gain
+        else {
+            const int16_t vl = volume[0];
+            const int16_t vr = volume[1];
+            do {
+                int16_t l = *in++;
+                out[0] = mulAdd(l, vl, out[0]);
+                out[1] = mulAdd(l, vr, out[1]);
+                out += 2;
+            } while (--frameCount);
+        }
+    }
+    mIn = in;
+}
+
+// no-op case
+void AudioMixerBase::process__nop()
+{
+    ALOGVV("process__nop\n");
+
+    for (const auto &pair : mGroups) {
+        // process by group of tracks with same output buffer to
+        // avoid multiple memset() on same buffer
+        const auto &group = pair.second;
+
+        const std::shared_ptr<TrackBase> &t = mTracks[group[0]];
+        memset(t->mainBuffer, 0,
+                mFrameCount * audio_bytes_per_frame(t->getMixerChannelCount(), t->mMixerFormat));
+
+        // now consume data
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            size_t outFrames = mFrameCount;
+            while (outFrames) {
+                t->buffer.frameCount = outFrames;
+                t->bufferProvider->getNextBuffer(&t->buffer);
+                if (t->buffer.raw == NULL) break;
+                outFrames -= t->buffer.frameCount;
+                t->bufferProvider->releaseBuffer(&t->buffer);
+            }
+        }
+    }
+}
+
+// generic code without resampling
+void AudioMixerBase::process__genericNoResampling()
+{
+    ALOGVV("process__genericNoResampling\n");
+    int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
+
+    for (const auto &pair : mGroups) {
+        // process by group of tracks with same output main buffer to
+        // avoid multiple memset() on same buffer
+        const auto &group = pair.second;
+
+        // acquire buffer
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            t->buffer.frameCount = mFrameCount;
+            t->bufferProvider->getNextBuffer(&t->buffer);
+            t->frameCount = t->buffer.frameCount;
+            t->mIn = t->buffer.raw;
+        }
+
+        int32_t *out = (int *)pair.first;
+        size_t numFrames = 0;
+        do {
+            const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
+            memset(outTemp, 0, sizeof(outTemp));
+            for (const int name : group) {
+                const std::shared_ptr<TrackBase> &t = mTracks[name];
+                int32_t *aux = NULL;
+                if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+                    aux = t->auxBuffer + numFrames;
+                }
+                for (int outFrames = frameCount; outFrames > 0; ) {
+                    // t->in == nullptr can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                    if (t->mIn == nullptr) {
+                        break;
+                    }
+                    size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
+                    if (inFrames > 0) {
+                        (t.get()->*t->hook)(
+                                outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
+                                inFrames, mResampleTemp.get() /* naked ptr */, aux);
+                        t->frameCount -= inFrames;
+                        outFrames -= inFrames;
+                        if (CC_UNLIKELY(aux != NULL)) {
+                            aux += inFrames;
+                        }
+                    }
+                    if (t->frameCount == 0 && outFrames) {
+                        t->bufferProvider->releaseBuffer(&t->buffer);
+                        t->buffer.frameCount = (mFrameCount - numFrames) -
+                                (frameCount - outFrames);
+                        t->bufferProvider->getNextBuffer(&t->buffer);
+                        t->mIn = t->buffer.raw;
+                        if (t->mIn == nullptr) {
+                            break;
+                        }
+                        t->frameCount = t->buffer.frameCount;
+                    }
+                }
+            }
+
+            const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+            convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
+                    frameCount * t1->mMixerChannelCount);
+            // TODO: fix ugly casting due to choice of out pointer type
+            out = reinterpret_cast<int32_t*>((uint8_t*)out
+                    + frameCount * t1->mMixerChannelCount
+                    * audio_bytes_per_sample(t1->mMixerFormat));
+            numFrames += frameCount;
+        } while (numFrames < mFrameCount);
+
+        // release each track's buffer
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            t->bufferProvider->releaseBuffer(&t->buffer);
+        }
+    }
+}
+
+// generic code with resampling
+void AudioMixerBase::process__genericResampling()
+{
+    ALOGVV("process__genericResampling\n");
+    int32_t * const outTemp = mOutputTemp.get(); // naked ptr
+    size_t numFrames = mFrameCount;
+
+    for (const auto &pair : mGroups) {
+        const auto &group = pair.second;
+        const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+
+        // clear temp buffer
+        memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
+        for (const int name : group) {
+            const std::shared_ptr<TrackBase> &t = mTracks[name];
+            int32_t *aux = NULL;
+            if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+                aux = t->auxBuffer;
+            }
+
+            // this is a little goofy, on the resampling case we don't
+            // acquire/release the buffers because it's done by
+            // the resampler.
+            if (t->needs & NEEDS_RESAMPLE) {
+                (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
+            } else {
+
+                size_t outFrames = 0;
+
+                while (outFrames < numFrames) {
+                    t->buffer.frameCount = numFrames - outFrames;
+                    t->bufferProvider->getNextBuffer(&t->buffer);
+                    t->mIn = t->buffer.raw;
+                    // t->mIn == nullptr can happen if the track was flushed just after having
+                    // been enabled for mixing.
+                    if (t->mIn == nullptr) break;
+
+                    (t.get()->*t->hook)(
+                            outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
+                            mResampleTemp.get() /* naked ptr */,
+                            aux != nullptr ? aux + outFrames : nullptr);
+                    outFrames += t->buffer.frameCount;
+
+                    t->bufferProvider->releaseBuffer(&t->buffer);
+                }
+            }
+        }
+        convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
+                outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
+    }
+}
+
+// one track, 16 bits stereo without resampling is the most common case
+void AudioMixerBase::process__oneTrack16BitsStereoNoResampling()
+{
+    ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
+    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
+            "%zu != 1 tracks enabled", mEnabled.size());
+    const int name = mEnabled[0];
+    const std::shared_ptr<TrackBase> &t = mTracks[name];
+
+    AudioBufferProvider::Buffer& b(t->buffer);
+
+    int32_t* out = t->mainBuffer;
+    float *fout = reinterpret_cast<float*>(out);
+    size_t numFrames = mFrameCount;
+
+    const int16_t vl = t->volume[0];
+    const int16_t vr = t->volume[1];
+    const uint32_t vrl = t->volumeRL;
+    while (numFrames) {
+        b.frameCount = numFrames;
+        t->bufferProvider->getNextBuffer(&b);
+        const int16_t *in = b.i16;
+
+        // in == NULL can happen if the track was flushed just after having
+        // been enabled for mixing.
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
+                 memset((char*)fout, 0, numFrames
+                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+            } else {
+                 memset((char*)out, 0, numFrames
+                         * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+            }
+            ALOGE_IF((((uintptr_t)in) & 3),
+                    "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
+                    " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
+                    in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
+            return;
+        }
+        size_t outFrames = b.frameCount;
+
+        switch (t->mMixerFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            do {
+                uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                in += 2;
+                int32_t l = mulRL(1, rl, vrl);
+                int32_t r = mulRL(0, rl, vrl);
+                *fout++ = float_from_q4_27(l);
+                *fout++ = float_from_q4_27(r);
+                // Note: In case of later int16_t sink output,
+                // conversion and clamping is done by memcpy_to_i16_from_float().
+            } while (--outFrames);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
+                // volume is boosted, so we might need to clamp even though
+                // we process only one track.
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    // clamping...
+                    l = clamp16(l);
+                    r = clamp16(r);
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            } else {
+                do {
+                    uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+                    in += 2;
+                    int32_t l = mulRL(1, rl, vrl) >> 12;
+                    int32_t r = mulRL(0, rl, vrl) >> 12;
+                    *out++ = (r<<16) | (l & 0xFFFF);
+                } while (--outFrames);
+            }
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
+        }
+        numFrames -= b.frameCount;
+        t->bufferProvider->releaseBuffer(&b);
+    }
+}
+
+/* TODO: consider whether this level of optimization is necessary.
+ * Perhaps just stick with a single for loop.
+ */
+
+// 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))
+
+/* 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
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+        typename TO, typename TI, typename TV, typename TA, typename TAV>
+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;
+    }
+}
+
+/* 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
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+        typename TO, typename TI, typename TV, typename TA, typename TAV>
+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;
+    }
+}
+
+/* MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+    typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::volumeMix(TO *out, size_t outFrames,
+        const TI *in, TA *aux, bool ramp)
+{
+    if (USEFLOATVOL) {
+        if (ramp) {
+            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    mPrevVolume, mVolumeInc,
+#ifdef FLOAT_AUX
+                    &mPrevAuxLevel, mAuxInc
+#else
+                    &prevAuxLevel, auxInc
+#endif
+                );
+            if (ADJUSTVOL) {
+                adjustVolumeRamp(aux != NULL, true);
+            }
+        } else {
+            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    mVolume,
+#ifdef FLOAT_AUX
+                    mAuxLevel
+#else
+                    auxLevel
+#endif
+            );
+        }
+    } else {
+        if (ramp) {
+            volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    prevVolume, volumeInc, &prevAuxLevel, auxInc);
+            if (ADJUSTVOL) {
+                adjustVolumeRamp(aux != NULL);
+            }
+        } else {
+            volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+                    volume, auxLevel);
+        }
+    }
+}
+
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ *
+ * 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
+ * TA: int32_t (Q4.27)
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::process__noResampleOneTrack()
+{
+    ALOGVV("process__noResampleOneTrack\n");
+    LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
+            "%zu != 1 tracks enabled", mEnabled.size());
+    const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+    const uint32_t channels = t->mMixerChannelCount;
+    TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+    TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+    const bool ramp = t->needsRamp();
+
+    for (size_t numFrames = mFrameCount; numFrames > 0; ) {
+        AudioBufferProvider::Buffer& b(t->buffer);
+        // get input buffer
+        b.frameCount = numFrames;
+        t->bufferProvider->getNextBuffer(&b);
+        const TI *in = reinterpret_cast<TI*>(b.raw);
+
+        // in == NULL can happen if the track was flushed just after having
+        // been enabled for mixing.
+        if (in == NULL || (((uintptr_t)in) & 3)) {
+            memset(out, 0, numFrames
+                    * channels * audio_bytes_per_sample(t->mMixerFormat));
+            ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
+                    "buffer %p track %p, channels %d, needs %#x",
+                    in, &t, t->channelCount, t->needs);
+            return;
+        }
+
+        const size_t outFrames = b.frameCount;
+        t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
+                out, outFrames, in, aux, ramp);
+
+        out += outFrames * channels;
+        if (aux != NULL) {
+            aux += outFrames;
+        }
+        numFrames -= b.frameCount;
+
+        // release buffer
+        t->bufferProvider->releaseBuffer(&b);
+    }
+    if (ramp) {
+        t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
+    }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ *
+ * 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
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+    ALOGVV("track__Resample\n");
+    mResampler->setSampleRate(sampleRate);
+    const bool ramp = needsRamp();
+    if (ramp || aux != NULL) {
+        // if ramp:        resample with unity gain to temp buffer and scale/mix in 2nd step.
+        // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+        mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+        memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
+        mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
+
+        volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+                out, outFrameCount, temp, aux, ramp);
+
+    } else { // constant volume gain
+        mResampler->setVolume(mVolume[0], mVolume[1]);
+        mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
+    }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in in.
+ *
+ * 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
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__NoResample(
+        TO* out, size_t frameCount, TO* temp __unused, TA* aux)
+{
+    ALOGVV("track__NoResample\n");
+    const TI *in = static_cast<const TI *>(mIn);
+
+    volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+            out, frameCount, in, aux, needsRamp());
+
+    // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+    // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+    in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
+    mIn = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+/* static */
+void AudioMixerBase::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+        void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+    switch (mixerInFormat) {
+    case AUDIO_FORMAT_PCM_FLOAT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    case AUDIO_FORMAT_PCM_16_BIT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
+            break;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
+            break;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+        break;
+    }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+/* static */
+AudioMixerBase::hook_t AudioMixerBase::TrackBase::getTrackHook(int trackType, uint32_t channelCount,
+        audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+        switch (trackType) {
+        case TRACKTYPE_NOP:
+            return &TrackBase::track__nop;
+        case TRACKTYPE_RESAMPLE:
+            return &TrackBase::track__genericResample;
+        case TRACKTYPE_NORESAMPLEMONO:
+            return &TrackBase::track__16BitsMono;
+        case TRACKTYPE_NORESAMPLE:
+            return &TrackBase::track__16BitsStereo;
+        default:
+            LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+            break;
+        }
+    }
+    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+    switch (trackType) {
+    case TRACKTYPE_NOP:
+        return &TrackBase::track__nop;
+    case TRACKTYPE_RESAMPLE:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    case TRACKTYPE_NORESAMPLEMONO:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                            MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                            MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    case TRACKTYPE_NORESAMPLE:
+        switch (mixerInFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                    MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+                    MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+        break;
+    }
+    return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ *
+ * TODO: Due to the special mixing considerations of duplicating to
+ * a stereo output track, the input track cannot be MONO.  This should be
+ * prevented by the caller.
+ */
+/* static */
+AudioMixerBase::process_hook_t AudioMixerBase::getProcessHook(
+        int processType, uint32_t channelCount,
+        audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+    if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+        LOG_ALWAYS_FATAL("bad processType: %d", processType);
+        return NULL;
+    }
+    if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+        return &AudioMixerBase::process__oneTrack16BitsStereoNoResampling;
+    }
+    LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+    switch (mixerInFormat) {
+    case AUDIO_FORMAT_PCM_FLOAT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    case AUDIO_FORMAT_PCM_16_BIT:
+        switch (mixerOutFormat) {
+        case AUDIO_FORMAT_PCM_FLOAT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        case AUDIO_FORMAT_PCM_16_BIT:
+            return &AudioMixerBase::process__noResampleOneTrack<
+                    MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+        default:
+            LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+            break;
+        }
+        break;
+    default:
+        LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+        break;
+    }
+    return NULL;
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android
diff --git a/media/libaudioprocessing/include/media/AudioMixer.h b/media/libaudioprocessing/include/media/AudioMixer.h
new file mode 100644
index 0000000..3f7cd48
--- /dev/null
+++ b/media/libaudioprocessing/include/media/AudioMixer.h
@@ -0,0 +1,238 @@
+/*
+**
+** Copyright 2007, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef ANDROID_AUDIO_MIXER_H
+#define ANDROID_AUDIO_MIXER_H
+
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <android/os/IExternalVibratorService.h>
+#include <media/AudioMixerBase.h>
+#include <media/BufferProviders.h>
+#include <utils/threads.h>
+
+// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
+#define MAX_GAIN_INT AudioMixerBase::UNITY_GAIN_INT
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioMixer extends AudioMixerBase by adding support for down- and up-mixing
+// and time stretch that are implemented via Effects HAL, and adding support
+// for haptic channels which depends on Vibrator service. This is the version
+// that is used by Audioflinger.
+
+class AudioMixer : public AudioMixerBase
+{
+public:
+    // maximum number of channels supported for the content
+    static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
+
+    enum { // extension of AudioMixerBase parameters
+        DOWNMIX_TYPE    = 0x4004,
+        // for haptic
+        HAPTIC_ENABLED  = 0x4007, // Set haptic data from this track should be played or not.
+        HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
+        // for target TIMESTRETCH
+        PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
+                                  // parameter 'value' is a pointer to the new playback rate.
+    };
+
+    typedef enum { // Haptic intensity, should keep consistent with VibratorService
+        HAPTIC_SCALE_MUTE = os::IExternalVibratorService::SCALE_MUTE,
+        HAPTIC_SCALE_VERY_LOW = os::IExternalVibratorService::SCALE_VERY_LOW,
+        HAPTIC_SCALE_LOW = os::IExternalVibratorService::SCALE_LOW,
+        HAPTIC_SCALE_NONE = os::IExternalVibratorService::SCALE_NONE,
+        HAPTIC_SCALE_HIGH = os::IExternalVibratorService::SCALE_HIGH,
+        HAPTIC_SCALE_VERY_HIGH = os::IExternalVibratorService::SCALE_VERY_HIGH,
+    } haptic_intensity_t;
+    static constexpr float HAPTIC_SCALE_VERY_LOW_RATIO = 2.0f / 3.0f;
+    static constexpr float HAPTIC_SCALE_LOW_RATIO = 3.0f / 4.0f;
+    static const constexpr float HAPTIC_MAX_AMPLITUDE_FLOAT = 1.0f;
+
+    static inline bool isValidHapticIntensity(haptic_intensity_t hapticIntensity) {
+        switch (hapticIntensity) {
+        case HAPTIC_SCALE_MUTE:
+        case HAPTIC_SCALE_VERY_LOW:
+        case HAPTIC_SCALE_LOW:
+        case HAPTIC_SCALE_NONE:
+        case HAPTIC_SCALE_HIGH:
+        case HAPTIC_SCALE_VERY_HIGH:
+            return true;
+        default:
+            return false;
+        }
+    }
+
+    AudioMixer(size_t frameCount, uint32_t sampleRate)
+            : AudioMixerBase(frameCount, sampleRate) {
+        pthread_once(&sOnceControl, &sInitRoutine);
+    }
+
+    bool isValidChannelMask(audio_channel_mask_t channelMask) const override;
+
+    void setParameter(int name, int target, int param, void *value) override;
+    void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
+
+private:
+
+    struct Track : public TrackBase {
+        Track() : TrackBase() {}
+
+        ~Track()
+        {
+            // mInputBufferProvider need not be deleted.
+            // Ensure the order of destruction of buffer providers as they
+            // release the upstream provider in the destructor.
+            mTimestretchBufferProvider.reset(nullptr);
+            mPostDownmixReformatBufferProvider.reset(nullptr);
+            mDownmixerBufferProvider.reset(nullptr);
+            mReformatBufferProvider.reset(nullptr);
+            mContractChannelsNonDestructiveBufferProvider.reset(nullptr);
+            mAdjustChannelsBufferProvider.reset(nullptr);
+        }
+
+        uint32_t getOutputChannelCount() override {
+            return mDownmixerBufferProvider.get() != nullptr ? mMixerChannelCount : channelCount;
+        }
+        uint32_t getMixerChannelCount() override {
+            return mMixerChannelCount + mMixerHapticChannelCount;
+        }
+
+        status_t    prepareForDownmix();
+        void        unprepareForDownmix();
+        status_t    prepareForReformat();
+        void        unprepareForReformat();
+        status_t    prepareForAdjustChannels();
+        void        unprepareForAdjustChannels();
+        status_t    prepareForAdjustChannelsNonDestructive(size_t frames);
+        void        unprepareForAdjustChannelsNonDestructive();
+        void        clearContractedBuffer();
+        bool        setPlaybackRate(const AudioPlaybackRate &playbackRate);
+        void        reconfigureBufferProviders();
+
+        /* Buffer providers are constructed to translate the track input data as needed.
+         * See DownmixerBufferProvider below for how the Track buffer provider
+         * is wrapped by another one when dowmixing is required.
+         *
+         * TODO: perhaps make a single PlaybackConverterProvider class to move
+         * all pre-mixer track buffer conversions outside the AudioMixer class.
+         *
+         * 1) mInputBufferProvider: The AudioTrack buffer provider.
+         * 2) mAdjustChannelsBufferProvider: Expands or contracts sample data from one interleaved
+         *    channel format to another. Expanded channels are filled with zeros and put at the end
+         *    of each audio frame. Contracted channels are copied to the end of the buffer.
+         * 3) mContractChannelsNonDestructiveBufferProvider: Non-destructively contract sample data.
+         *    This is currently using at audio-haptic coupled playback to separate audio and haptic
+         *    data. Contracted channels could be written to given buffer.
+         * 4) mReformatBufferProvider: If not NULL, performs the audio reformat to
+         *    match either mMixerInFormat or mDownmixRequiresFormat, if the downmixer
+         *    requires reformat. For example, it may convert floating point input to
+         *    PCM_16_bit if that's required by the downmixer.
+         * 5) mDownmixerBufferProvider: If not NULL, performs the channel remixing to match
+         *    the number of channels required by the mixer sink.
+         * 6) mPostDownmixReformatBufferProvider: If not NULL, performs reformatting from
+         *    the downmixer requirements to the mixer engine input requirements.
+         * 7) mTimestretchBufferProvider: Adds timestretching for playback rate
+         */
+        AudioBufferProvider* mInputBufferProvider;    // externally provided buffer provider.
+        // TODO: combine mAdjustChannelsBufferProvider and
+        // mContractChannelsNonDestructiveBufferProvider
+        std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mContractChannelsNonDestructiveBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mReformatBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mDownmixerBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
+        std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
+
+        audio_format_t mDownmixRequiresFormat;  // required downmixer format
+                                                // AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
+                                                // AUDIO_FORMAT_INVALID if no required format
+
+        AudioPlaybackRate    mPlaybackRate;
+
+        // Haptic
+        bool                 mHapticPlaybackEnabled;
+        haptic_intensity_t   mHapticIntensity;
+        audio_channel_mask_t mHapticChannelMask;
+        uint32_t             mHapticChannelCount;
+        audio_channel_mask_t mMixerHapticChannelMask;
+        uint32_t             mMixerHapticChannelCount;
+        uint32_t             mAdjustInChannelCount;
+        uint32_t             mAdjustOutChannelCount;
+        uint32_t             mAdjustNonDestructiveInChannelCount;
+        uint32_t             mAdjustNonDestructiveOutChannelCount;
+        bool                 mKeepContractedChannels;
+
+        float getHapticScaleGamma() const {
+        // Need to keep consistent with the value in VibratorService.
+        switch (mHapticIntensity) {
+        case HAPTIC_SCALE_VERY_LOW:
+            return 2.0f;
+        case HAPTIC_SCALE_LOW:
+            return 1.5f;
+        case HAPTIC_SCALE_HIGH:
+            return 0.5f;
+        case HAPTIC_SCALE_VERY_HIGH:
+            return 0.25f;
+        default:
+            return 1.0f;
+        }
+        }
+
+        float getHapticMaxAmplitudeRatio() const {
+        // Need to keep consistent with the value in VibratorService.
+        switch (mHapticIntensity) {
+        case HAPTIC_SCALE_VERY_LOW:
+            return HAPTIC_SCALE_VERY_LOW_RATIO;
+        case HAPTIC_SCALE_LOW:
+            return HAPTIC_SCALE_LOW_RATIO;
+        case HAPTIC_SCALE_NONE:
+        case HAPTIC_SCALE_HIGH:
+        case HAPTIC_SCALE_VERY_HIGH:
+            return 1.0f;
+        default:
+            return 0.0f;
+        }
+        }
+    };
+
+    inline std::shared_ptr<Track> getTrack(int name) {
+        return std::static_pointer_cast<Track>(mTracks[name]);
+    }
+
+    std::shared_ptr<TrackBase> preCreateTrack() override;
+    status_t postCreateTrack(TrackBase *track) override;
+
+    void preProcess() override;
+    void postProcess() override;
+
+    bool setChannelMasks(int name,
+            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) override;
+
+    static void sInitRoutine();
+
+    static pthread_once_t sOnceControl; // initialized in constructor by first new
+};
+
+// ----------------------------------------------------------------------------
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_H
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
new file mode 100644
index 0000000..805b6d0
--- /dev/null
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -0,0 +1,359 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+**     http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef ANDROID_AUDIO_MIXER_BASE_H
+#define ANDROID_AUDIO_MIXER_BASE_H
+
+#include <map>
+#include <memory>
+#include <string>
+#include <unordered_map>
+#include <vector>
+
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+
+// This must match frameworks/av/services/audioflinger/Configuration.h
+// when used with the Audio Framework.
+#define FLOAT_AUX
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioMixerBase is functional on its own if only mixing and resampling
+// is needed.
+
+class AudioMixerBase
+{
+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_VOLUMES = FCC_2; // stereo volume only
+
+    static const uint16_t UNITY_GAIN_INT = 0x1000;
+    static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+    enum { // names
+        // setParameter targets
+        TRACK           = 0x3000,
+        RESAMPLE        = 0x3001,
+        RAMP_VOLUME     = 0x3002, // ramp to new volume
+        VOLUME          = 0x3003, // don't ramp
+        TIMESTRETCH     = 0x3004,
+
+        // set Parameter names
+        // for target TRACK
+        CHANNEL_MASK    = 0x4000,
+        FORMAT          = 0x4001,
+        MAIN_BUFFER     = 0x4002,
+        AUX_BUFFER      = 0x4003,
+        // 0x4004 reserved
+        MIXER_FORMAT    = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+        // for target RESAMPLE
+        SAMPLE_RATE     = 0x4100, // Configure sample rate conversion on this track name;
+                                  // parameter 'value' is the new sample rate in Hz.
+                                  // Only creates a sample rate converter the first time that
+                                  // the track sample rate is different from the mix sample rate.
+                                  // If the new sample rate is the same as the mix sample rate,
+                                  // and a sample rate converter already exists,
+                                  // then the sample rate converter remains present but is a no-op.
+        RESET           = 0x4101, // Reset sample rate converter without changing sample rate.
+                                  // This clears out the resampler's input buffer.
+        REMOVE          = 0x4102, // Remove the sample rate converter on this track name;
+                                  // the track is restored to the mix sample rate.
+        // for target RAMP_VOLUME and VOLUME (8 channels max)
+        // FIXME use float for these 3 to improve the dynamic range
+        VOLUME0         = 0x4200,
+        VOLUME1         = 0x4201,
+        AUXLEVEL        = 0x4210,
+    };
+
+    AudioMixerBase(size_t frameCount, uint32_t sampleRate)
+        : mSampleRate(sampleRate)
+        , mFrameCount(frameCount) {
+    }
+
+    virtual ~AudioMixerBase() {}
+
+    virtual bool isValidFormat(audio_format_t format) const;
+    virtual bool isValidChannelMask(audio_channel_mask_t channelMask) const;
+
+    // Create a new track in the mixer.
+    //
+    // \param name        a unique user-provided integer associated with the track.
+    //                    If name already exists, the function will abort.
+    // \param channelMask output channel mask.
+    // \param format      PCM format
+    // \param sessionId   Session id for the track. Tracks with the same
+    //                    session id will be submixed together.
+    //
+    // \return OK        on success.
+    //         BAD_VALUE if the format does not satisfy isValidFormat()
+    //                   or the channelMask does not satisfy isValidChannelMask().
+    status_t    create(
+            int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
+
+    bool        exists(int name) const {
+        return mTracks.count(name) > 0;
+    }
+
+    // Free an allocated track by name.
+    void        destroy(int name);
+
+    // Enable or disable an allocated track by name
+    void        enable(int name);
+    void        disable(int name);
+
+    virtual void setParameter(int name, int target, int param, void *value);
+
+    void        process() {
+        preProcess();
+        (this->*mHook)();
+        postProcess();
+    }
+
+    size_t      getUnreleasedFrames(int name) const;
+
+    std::string trackNames() const;
+
+  protected:
+    // Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
+    // original code will be used for stereo sinks, the new mixer for everything else.
+    static constexpr bool kUseNewMixer = true;
+
+    // Set kUseFloat to true to allow floating input into the mixer engine.
+    // If kUseNewMixer is false, this is ignored or may be overridden internally
+    static constexpr bool kUseFloat = true;
+
+#ifdef FLOAT_AUX
+    using TYPE_AUX = float;
+    static_assert(kUseNewMixer && kUseFloat,
+            "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
+#else
+    using TYPE_AUX = int32_t; // q4.27
+#endif
+
+    /* For multi-format functions (calls template functions
+     * in AudioMixerOps.h).  The template parameters are as follows:
+     *
+     *   MIXTYPE     (see AudioMixerOps.h MIXTYPE_* enumeration)
+     *   USEFLOATVOL (set to true if float volume is used)
+     *   ADJUSTVOL   (set to true if volume ramp parameters needs adjustment afterwards)
+     *   TO: int32_t (Q4.27) or float
+     *   TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+     *   TA: int32_t (Q4.27)
+     */
+
+    enum {
+        // FIXME this representation permits up to 8 channels
+        NEEDS_CHANNEL_COUNT__MASK   = 0x00000007,
+    };
+
+    enum {
+        NEEDS_CHANNEL_1             = 0x00000000,   // mono
+        NEEDS_CHANNEL_2             = 0x00000001,   // stereo
+
+        // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+        NEEDS_MUTE                  = 0x00000100,
+        NEEDS_RESAMPLE              = 0x00001000,
+        NEEDS_AUX                   = 0x00010000,
+    };
+
+    // hook types
+    enum {
+        PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
+    };
+
+    enum {
+        TRACKTYPE_NOP,
+        TRACKTYPE_RESAMPLE,
+        TRACKTYPE_NORESAMPLE,
+        TRACKTYPE_NORESAMPLEMONO,
+    };
+
+    // process hook functionality
+    using process_hook_t = void(AudioMixerBase::*)();
+
+    struct TrackBase;
+    using hook_t = void(TrackBase::*)(
+            int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+
+    struct TrackBase {
+        TrackBase()
+            : bufferProvider(nullptr)
+        {
+            // TODO: move additional initialization here.
+        }
+        virtual ~TrackBase() {}
+
+        virtual uint32_t getOutputChannelCount() { return channelCount; }
+        virtual uint32_t getMixerChannelCount() { return mMixerChannelCount; }
+
+        bool        needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+        bool        setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+        bool        doesResample() const { return mResampler.get() != nullptr; }
+        void        recreateResampler(uint32_t devSampleRate);
+        void        resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
+        void        adjustVolumeRamp(bool aux, bool useFloat = false);
+        size_t      getUnreleasedFrames() const { return mResampler.get() != nullptr ?
+                                                    mResampler->getUnreleasedFrames() : 0; };
+
+        static hook_t getTrackHook(int trackType, uint32_t channelCount,
+                audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+        void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+        template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+            typename TO, typename TI, typename TA>
+        void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
+
+        uint32_t    needs;
+
+        // TODO: Eventually remove legacy integer volume settings
+        union {
+        int16_t     volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+        int32_t     volumeRL;
+        };
+
+        int32_t     prevVolume[MAX_NUM_VOLUMES];
+        int32_t     volumeInc[MAX_NUM_VOLUMES];
+        int32_t     auxInc;
+        int32_t     prevAuxLevel;
+        int16_t     auxLevel;       // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+
+        uint16_t    frameCount;
+
+        uint8_t     channelCount;   // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+        uint8_t     unused_padding; // formerly format, was always 16
+        uint16_t    enabled;        // actually bool
+        audio_channel_mask_t channelMask;
+
+        // actual buffer provider used by the track hooks
+        AudioBufferProvider*                bufferProvider;
+
+        mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+        hook_t      hook;
+        const void  *mIn;             // current location in buffer
+
+        std::unique_ptr<AudioResampler> mResampler;
+        uint32_t    sampleRate;
+        int32_t*    mainBuffer;
+        int32_t*    auxBuffer;
+
+        int32_t     sessionId;
+
+        audio_format_t mMixerFormat;     // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+        audio_format_t mFormat;          // input track format
+        audio_format_t mMixerInFormat;   // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+                                         // each track must be converted to this format.
+
+        float          mVolume[MAX_NUM_VOLUMES];     // floating point set volume
+        float          mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+        float          mVolumeInc[MAX_NUM_VOLUMES];  // floating point volume increment
+
+        float          mAuxLevel;                     // floating point set aux level
+        float          mPrevAuxLevel;                 // floating point prev aux level
+        float          mAuxInc;                       // floating point aux increment
+
+        audio_channel_mask_t mMixerChannelMask;
+        uint32_t             mMixerChannelCount;
+
+      protected:
+
+        // hooks
+        void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+        void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+        void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+        void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+        void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+
+        // multi-format track hooks
+        template <int MIXTYPE, typename TO, typename TI, typename TA>
+        void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+        template <int MIXTYPE, typename TO, typename TI, typename TA>
+        void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+    };
+
+    // preCreateTrack must create an instance of a proper TrackBase descendant.
+    // postCreateTrack is called after filling out fields of TrackBase. It can
+    // abort track creation by returning non-OK status. See the implementation
+    // of create() for details.
+    virtual std::shared_ptr<TrackBase> preCreateTrack();
+    virtual status_t postCreateTrack(TrackBase *track __unused) { return OK; }
+
+    // preProcess is called before the process hook, postProcess after,
+    // see the implementation of process() method.
+    virtual void preProcess() {}
+    virtual void postProcess() {}
+
+    virtual bool setChannelMasks(int name,
+            audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+    // Called when track info changes and a new process hook should be determined.
+    void invalidate() {
+        mHook = &AudioMixerBase::process__validate;
+    }
+
+    void process__validate();
+    void process__nop();
+    void process__genericNoResampling();
+    void process__genericResampling();
+    void process__oneTrack16BitsStereoNoResampling();
+
+    template <int MIXTYPE, typename TO, typename TI, typename TA>
+    void process__noResampleOneTrack();
+
+    static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+            audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+    static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+            void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+    // initialization constants
+    const uint32_t mSampleRate;
+    const size_t mFrameCount;
+
+    process_hook_t mHook = &AudioMixerBase::process__nop;   // one of process__*, never nullptr
+
+    // the size of the type (int32_t) should be the largest of all types supported
+    // by the mixer.
+    std::unique_ptr<int32_t[]> mOutputTemp;
+    std::unique_ptr<int32_t[]> mResampleTemp;
+
+    // track names grouped by main buffer, in no particular order of main buffer.
+    // however names for a particular main buffer are in order (by construction).
+    std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
+
+    // track names that are enabled, in increasing order (by construction).
+    std::vector<int /* name */> mEnabled;
+
+    // track smart pointers, by name, in increasing order of name.
+    std::map<int /* name */, std::shared_ptr<TrackBase>> mTracks;
+};
+
+}  // namespace android
+
+#endif  // ANDROID_AUDIO_MIXER_BASE_H
diff --git a/media/libmedia/include/media/BufferProviders.h b/media/libaudioprocessing/include/media/BufferProviders.h
similarity index 100%
rename from media/libmedia/include/media/BufferProviders.h
rename to media/libaudioprocessing/include/media/BufferProviders.h
diff --git a/media/libeffects/downmix/Android.bp b/media/libeffects/downmix/Android.bp
index 9c82b1d..2a2f36e 100644
--- a/media/libeffects/downmix/Android.bp
+++ b/media/libeffects/downmix/Android.bp
@@ -6,6 +6,7 @@
     srcs: ["EffectDownmix.c"],
 
     shared_libs: [
+        "libaudioutils",
         "libcutils",
         "liblog",
     ],
@@ -23,5 +24,4 @@
         "libaudioeffects",
         "libhardware_headers",
     ],
-    static_libs: ["libaudioutils" ],
 }
diff --git a/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h b/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h
index 7468a90..10eedd9 100644
--- a/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h
+++ b/media/libeffects/lvm/lib/Common/lib/VectorArithmetic.h
@@ -53,6 +53,7 @@
                                  LVM_INT16 NrFrames,
                                  LVM_INT32 NrChannels);
 void Copy_Float_Stereo_Mc(       const LVM_FLOAT *src,
+                                 LVM_FLOAT *StereoOut,
                                  LVM_FLOAT *dst,
                                  LVM_INT16 NrFrames,
                                  LVM_INT32 NrChannels);
diff --git a/media/libeffects/lvm/lib/Common/src/Copy_16.c b/media/libeffects/lvm/lib/Common/src/Copy_16.c
index 3858450..3eb3c14 100644
--- a/media/libeffects/lvm/lib/Common/src/Copy_16.c
+++ b/media/libeffects/lvm/lib/Common/src/Copy_16.c
@@ -117,30 +117,31 @@
     }
 }
 
-// Merge a multichannel source with stereo contained in dst, to dst.
+// Merge a multichannel source with stereo contained in StereoOut, to dst.
 void Copy_Float_Stereo_Mc(const LVM_FLOAT *src,
+                 LVM_FLOAT *StereoOut,
                  LVM_FLOAT *dst,
                  LVM_INT16 NrFrames, /* Number of frames*/
                  LVM_INT32 NrChannels)
 {
     LVM_INT16 ii, jj;
-    LVM_FLOAT *src_st = dst + 2 * (NrFrames - 1);
 
-    // repack dst which carries stereo information
+    // pack dst with stereo information of StereoOut
     // together with the upper channels of src.
+    StereoOut += 2 * (NrFrames - 1);
     dst += NrChannels * (NrFrames - 1);
     src += NrChannels * (NrFrames - 1);
     for (ii = NrFrames; ii != 0; ii--)
     {
-        dst[1] = src_st[1];
-        dst[0] = src_st[0]; // copy 1 before 0 is required for NrChannels == 3.
+        dst[1] = StereoOut[1];
+        dst[0] = StereoOut[0]; // copy 1 before 0 is required for NrChannels == 3.
         for (jj = 2; jj < NrChannels; jj++)
         {
             dst[jj] = src[jj];
         }
         dst    -= NrChannels;
         src    -= NrChannels;
-        src_st -= 2;
+        StereoOut -= 2;
     }
 }
 #endif
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h
index ab8ccd1..c8df8e4 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Private.h
@@ -60,7 +60,11 @@
 #define LVCS_COMPGAINFRAME          64          /* Compressor gain update interval */
 
 /* Memory */
+#ifdef SUPPORT_MC
+#define LVCS_SCRATCHBUFFERS              8      /* Number of buffers required for inplace processing */
+#else
 #define LVCS_SCRATCHBUFFERS              6      /* Number of buffers required for inplace processing */
+#endif
 #ifdef SUPPORT_MC
 /*
  * The Concert Surround module applies processing only on the first two
diff --git a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c
index ef1d9eb..56fb04f 100644
--- a/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c
+++ b/media/libeffects/lvm/lib/StereoWidening/src/LVCS_Process.c
@@ -106,7 +106,7 @@
      * The Concert Surround module carries out processing only on L, R.
      */
     pInput = pScratch + (2 * NrFrames);
-    pStIn  = pScratch + (LVCS_SCRATCHBUFFERS * NrFrames);
+    pStIn  = pScratch + ((LVCS_SCRATCHBUFFERS - 2) * NrFrames);
     /* The first two channel data is extracted from the input data and
      * copied into pInput buffer
      */
@@ -303,13 +303,45 @@
      */
     if (pInstance->Params.OperatingMode != LVCS_OFF)
     {
+#ifdef SUPPORT_MC
+        LVM_FLOAT *pStereoOut;
+        /*
+         * LVCS_Process_CS uses output buffer to store intermediate outputs of StereoEnhancer,
+         * Equalizer, ReverbGenerator and BypassMixer.
+         * So, to avoid i/o data overlapping, when i/o buffers are common, use scratch buffer
+         * to store intermediate outputs.
+         */
+        if (pOutData == pInData)
+        {
+          /*
+           * Scratch memory is used in 4 chunks of (2 * NrFrames) size.
+           * First chunk of memory is used by LVCS_StereoEnhancer and LVCS_ReverbGenerator,
+           * second and fourth are used as input buffers by pInput and pStIn in LVCS_Process_CS.
+           * Hence, pStereoOut is pointed to use unused third portion of scratch memory.
+           */
+            pStereoOut = (LVM_FLOAT *) \
+                          pInstance->MemoryTable. \
+                          Region[LVCS_MEMREGION_TEMPORARY_FAST].pBaseAddress +
+                          ((LVCS_SCRATCHBUFFERS - 4) * NrFrames);
+        }
+        else
+        {
+            pStereoOut = pOutData;
+        }
+
         /*
          * Call CS process function
          */
             err = LVCS_Process_CS(hInstance,
                                   pInData,
+                                  pStereoOut,
+                                  NrFrames);
+#else
+            err = LVCS_Process_CS(hInstance,
+                                  pInData,
                                   pOutData,
                                   NumSamples);
+#endif
 
 
         /*
@@ -329,10 +361,17 @@
 
             if(NumSamples < LVCS_COMPGAINFRAME)
             {
+#ifdef SUPPORT_MC
+                NonLinComp_Float(Gain,                    /* Compressor gain setting */
+                                 pStereoOut,
+                                 pStereoOut,
+                                 (LVM_INT32)(2 * NrFrames));
+#else
                 NonLinComp_Float(Gain,                    /* Compressor gain setting */
                                  pOutData,
                                  pOutData,
                                  (LVM_INT32)(2 * NumSamples));
+#endif
             }
             else
             {
@@ -361,7 +400,11 @@
 
                 FinalGain = Gain;
                 Gain = pInstance->CompressGain;
+#ifdef SUPPORT_MC
+                pOutPtr = pStereoOut;
+#else
                 pOutPtr = pOutData;
+#endif
 
                 while(SampleToProcess > 0)
                 {
@@ -428,6 +471,7 @@
         }
 #ifdef SUPPORT_MC
         Copy_Float_Stereo_Mc(pInData,
+                             pStereoOut,
                              pOutData,
                              NrFrames,
                              channels);
diff --git a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
index 10dda19..0a2850f 100644
--- a/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
+++ b/media/libeffects/lvm/wrapper/Bundle/EffectBundle.cpp
@@ -2710,7 +2710,7 @@
         name[*pValueSize - 1] = 0;
         *pValueSize = strlen(name) + 1;
         ALOGVV("%s EQ_PARAM_GET_PRESET_NAME preset %d, name %s len %d",
-                __func__, preset, gEqualizerPresets[preset].name, *pValueSize);
+               __func__, preset, name, *pValueSize);
 
     } break;
 
diff --git a/media/libheif/HeifDecoderImpl.cpp b/media/libheif/HeifDecoderImpl.cpp
index a977300..bad4210 100644
--- a/media/libheif/HeifDecoderImpl.cpp
+++ b/media/libheif/HeifDecoderImpl.cpp
@@ -31,6 +31,7 @@
 #include <private/media/VideoFrame.h>
 #include <utils/Log.h>
 #include <utils/RefBase.h>
+#include <vector>
 
 HeifDecoder* createHeifDecoder() {
     return new android::HeifDecoderImpl();
@@ -38,6 +39,22 @@
 
 namespace android {
 
+void initFrameInfo(HeifFrameInfo *info, const VideoFrame *videoFrame) {
+    info->mWidth = videoFrame->mWidth;
+    info->mHeight = videoFrame->mHeight;
+    info->mRotationAngle = videoFrame->mRotationAngle;
+    info->mBytesPerPixel = videoFrame->mBytesPerPixel;
+    info->mDurationUs = videoFrame->mDurationUs;
+    if (videoFrame->mIccSize > 0) {
+        info->mIccData.assign(
+                videoFrame->getFlattenedIccData(),
+                videoFrame->getFlattenedIccData() + videoFrame->mIccSize);
+    } else {
+        // clear old Icc data if there is no Icc data.
+        info->mIccData.clear();
+    }
+}
+
 /*
  * HeifDataSource
  *
@@ -293,11 +310,11 @@
     // it's not, default to HAL_PIXEL_FORMAT_RGB_565.
     mOutputColor(HAL_PIXEL_FORMAT_RGB_565),
     mCurScanline(0),
-    mWidth(0),
-    mHeight(0),
+    mTotalScanline(0),
     mFrameDecoded(false),
     mHasImage(false),
     mHasVideo(false),
+    mSequenceLength(0),
     mAvailableLines(0),
     mNumSlices(1),
     mSliceHeight(0),
@@ -336,48 +353,94 @@
 
     mHasImage = hasImage && !strcasecmp(hasImage, "yes");
     mHasVideo = hasVideo && !strcasecmp(hasVideo, "yes");
-    sp<IMemory> sharedMem;
+
+    HeifFrameInfo* defaultInfo = nullptr;
     if (mHasImage) {
         // image index < 0 to retrieve primary image
-        sharedMem = mRetriever->getImageAtIndex(
+        sp<IMemory> sharedMem = mRetriever->getImageAtIndex(
                 -1, mOutputColor, true /*metaOnly*/);
-    } else if (mHasVideo) {
-        sharedMem = mRetriever->getFrameAtTime(0,
-                MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC,
-                mOutputColor, true /*metaOnly*/);
+
+        if (sharedMem == nullptr || sharedMem->pointer() == nullptr) {
+            ALOGE("init: videoFrame is a nullptr");
+            return false;
+        }
+
+        VideoFrame* videoFrame = static_cast<VideoFrame*>(sharedMem->pointer());
+
+        ALOGV("Image dimension %dx%d, display %dx%d, angle %d, iccSize %d",
+                videoFrame->mWidth,
+                videoFrame->mHeight,
+                videoFrame->mDisplayWidth,
+                videoFrame->mDisplayHeight,
+                videoFrame->mRotationAngle,
+                videoFrame->mIccSize);
+
+        initFrameInfo(&mImageInfo, videoFrame);
+
+        if (videoFrame->mTileHeight >= 512) {
+            // Try decoding in slices only if the image has tiles and is big enough.
+            mSliceHeight = videoFrame->mTileHeight;
+            ALOGV("mSliceHeight %u", mSliceHeight);
+        }
+
+        defaultInfo = &mImageInfo;
     }
 
-    if (sharedMem == nullptr || sharedMem->pointer() == nullptr) {
-        ALOGE("getFrameAtTime: videoFrame is a nullptr");
+    if (mHasVideo) {
+        sp<IMemory> sharedMem = mRetriever->getFrameAtTime(0,
+                MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC,
+                mOutputColor, true /*metaOnly*/);
+
+        if (sharedMem == nullptr || sharedMem->pointer() == nullptr) {
+            ALOGE("init: videoFrame is a nullptr");
+            return false;
+        }
+
+        VideoFrame* videoFrame = static_cast<VideoFrame*>(sharedMem->pointer());
+
+        ALOGV("Sequence dimension %dx%d, display %dx%d, angle %d, iccSize %d",
+                videoFrame->mWidth,
+                videoFrame->mHeight,
+                videoFrame->mDisplayWidth,
+                videoFrame->mDisplayHeight,
+                videoFrame->mRotationAngle,
+                videoFrame->mIccSize);
+
+        initFrameInfo(&mSequenceInfo, videoFrame);
+
+        mSequenceLength = atoi(mRetriever->extractMetadata(METADATA_KEY_VIDEO_FRAME_COUNT));
+
+        if (defaultInfo == nullptr) {
+            defaultInfo = &mSequenceInfo;
+        }
+    }
+
+    if (defaultInfo == nullptr) {
+        ALOGD("No valid image or sequence available");
         return false;
     }
 
-    VideoFrame* videoFrame = static_cast<VideoFrame*>(sharedMem->pointer());
-
-    ALOGV("Meta dimension %dx%d, display %dx%d, angle %d, iccSize %d",
-            videoFrame->mWidth,
-            videoFrame->mHeight,
-            videoFrame->mDisplayWidth,
-            videoFrame->mDisplayHeight,
-            videoFrame->mRotationAngle,
-            videoFrame->mIccSize);
-
     if (frameInfo != nullptr) {
-        frameInfo->set(
-                videoFrame->mWidth,
-                videoFrame->mHeight,
-                videoFrame->mRotationAngle,
-                videoFrame->mBytesPerPixel,
-                videoFrame->mIccSize,
-                videoFrame->getFlattenedIccData());
+        *frameInfo = *defaultInfo;
     }
-    mWidth = videoFrame->mWidth;
-    mHeight = videoFrame->mHeight;
-    if (mHasImage && videoFrame->mTileHeight >= 512 && mWidth >= 3000 && mHeight >= 2000 ) {
-        // Try decoding in slices only if the image has tiles and is big enough.
-        mSliceHeight = videoFrame->mTileHeight;
-        mNumSlices = (videoFrame->mHeight + mSliceHeight - 1) / mSliceHeight;
-        ALOGV("mSliceHeight %u, mNumSlices %zu", mSliceHeight, mNumSlices);
+
+    // default total scanline, this might change if decodeSequence() is used
+    mTotalScanline = defaultInfo->mHeight;
+
+    return true;
+}
+
+bool HeifDecoderImpl::getSequenceInfo(
+        HeifFrameInfo* frameInfo, size_t *frameCount) {
+    ALOGV("%s", __FUNCTION__);
+    if (!mHasVideo) {
+        return false;
+    }
+    if (frameInfo != nullptr) {
+        *frameInfo = mSequenceInfo;
+    }
+    if (frameCount != nullptr) {
+        *frameCount = mSequenceLength;
     }
     return true;
 }
@@ -416,11 +479,11 @@
         ALOGV("decodeAsync(): decoding slice %zu", i);
         size_t top = i * mSliceHeight;
         size_t bottom = (i + 1) * mSliceHeight;
-        if (bottom > mHeight) {
-            bottom = mHeight;
+        if (bottom > mImageInfo.mHeight) {
+            bottom = mImageInfo.mHeight;
         }
         sp<IMemory> frameMemory = mRetriever->getImageRectAtIndex(
-                -1, mOutputColor, 0, top, mWidth, bottom);
+                -1, mOutputColor, 0, top, mImageInfo.mWidth, bottom);
         {
             Mutex::Autolock autolock(mLock);
 
@@ -452,42 +515,44 @@
     // See if we want to decode in slices to allow client to start
     // scanline processing in parallel with decode. If this fails
     // we fallback to decoding the full frame.
-    if (mHasImage && mNumSlices > 1) {
-        // get first slice and metadata
-        sp<IMemory> frameMemory = mRetriever->getImageRectAtIndex(
-                -1, mOutputColor, 0, 0, mWidth, mSliceHeight);
-
-        if (frameMemory == nullptr || frameMemory->pointer() == nullptr) {
-            ALOGE("decode: metadata is a nullptr");
-            return false;
+    if (mHasImage) {
+        if (mSliceHeight >= 512 &&
+                mImageInfo.mWidth >= 3000 &&
+                mImageInfo.mHeight >= 2000 ) {
+            // Try decoding in slices only if the image has tiles and is big enough.
+            mNumSlices = (mImageInfo.mHeight + mSliceHeight - 1) / mSliceHeight;
+            ALOGV("mSliceHeight %u, mNumSlices %zu", mSliceHeight, mNumSlices);
         }
 
-        VideoFrame* videoFrame = static_cast<VideoFrame*>(frameMemory->pointer());
+        if (mNumSlices > 1) {
+            // get first slice and metadata
+            sp<IMemory> frameMemory = mRetriever->getImageRectAtIndex(
+                    -1, mOutputColor, 0, 0, mImageInfo.mWidth, mSliceHeight);
 
-        if (frameInfo != nullptr) {
-            frameInfo->set(
-                    videoFrame->mWidth,
-                    videoFrame->mHeight,
-                    videoFrame->mRotationAngle,
-                    videoFrame->mBytesPerPixel,
-                    videoFrame->mIccSize,
-                    videoFrame->getFlattenedIccData());
+            if (frameMemory == nullptr || frameMemory->pointer() == nullptr) {
+                ALOGE("decode: metadata is a nullptr");
+                return false;
+            }
+
+            VideoFrame* videoFrame = static_cast<VideoFrame*>(frameMemory->pointer());
+
+            if (frameInfo != nullptr) {
+                initFrameInfo(frameInfo, videoFrame);
+            }
+            mFrameMemory = frameMemory;
+            mAvailableLines = mSliceHeight;
+            mThread = new DecodeThread(this);
+            if (mThread->run("HeifDecode", ANDROID_PRIORITY_FOREGROUND) == OK) {
+                mFrameDecoded = true;
+                return true;
+            }
+            // Fallback to decode without slicing
+            mThread.clear();
+            mNumSlices = 1;
+            mSliceHeight = 0;
+            mAvailableLines = 0;
+            mFrameMemory.clear();
         }
-
-        mFrameMemory = frameMemory;
-        mAvailableLines = mSliceHeight;
-        mThread = new DecodeThread(this);
-        if (mThread->run("HeifDecode", ANDROID_PRIORITY_FOREGROUND) == OK) {
-            mFrameDecoded = true;
-            return true;
-        }
-
-        // Fallback to decode without slicing
-        mThread.clear();
-        mNumSlices = 1;
-        mSliceHeight = 0;
-        mAvailableLines = 0;
-        mFrameMemory.clear();
     }
 
     if (mHasImage) {
@@ -520,13 +585,8 @@
             videoFrame->mSize);
 
     if (frameInfo != nullptr) {
-        frameInfo->set(
-                videoFrame->mWidth,
-                videoFrame->mHeight,
-                videoFrame->mRotationAngle,
-                videoFrame->mBytesPerPixel,
-                videoFrame->mIccSize,
-                videoFrame->getFlattenedIccData());
+        initFrameInfo(frameInfo, videoFrame);
+
     }
     mFrameDecoded = true;
 
@@ -536,6 +596,50 @@
     return true;
 }
 
+bool HeifDecoderImpl::decodeSequence(int frameIndex, HeifFrameInfo* frameInfo) {
+    ALOGV("%s: frame index %d", __FUNCTION__, frameIndex);
+    if (!mHasVideo) {
+        return false;
+    }
+
+    if (frameIndex < 0 || frameIndex >= mSequenceLength) {
+        ALOGE("invalid frame index: %d, total frames %zu", frameIndex, mSequenceLength);
+        return false;
+    }
+
+    mCurScanline = 0;
+
+    // set total scanline to sequence height now
+    mTotalScanline = mSequenceInfo.mHeight;
+
+    mFrameMemory = mRetriever->getFrameAtIndex(frameIndex, mOutputColor);
+    if (mFrameMemory == nullptr || mFrameMemory->pointer() == nullptr) {
+        ALOGE("decode: videoFrame is a nullptr");
+        return false;
+    }
+
+    VideoFrame* videoFrame = static_cast<VideoFrame*>(mFrameMemory->pointer());
+    if (videoFrame->mSize == 0 ||
+            mFrameMemory->size() < videoFrame->getFlattenedSize()) {
+        ALOGE("decode: videoFrame size is invalid");
+        return false;
+    }
+
+    ALOGV("Decoded dimension %dx%d, display %dx%d, angle %d, rowbytes %d, size %d",
+            videoFrame->mWidth,
+            videoFrame->mHeight,
+            videoFrame->mDisplayWidth,
+            videoFrame->mDisplayHeight,
+            videoFrame->mRotationAngle,
+            videoFrame->mRowBytes,
+            videoFrame->mSize);
+
+    if (frameInfo != nullptr) {
+        initFrameInfo(frameInfo, videoFrame);
+    }
+    return true;
+}
+
 bool HeifDecoderImpl::getScanlineInner(uint8_t* dst) {
     if (mFrameMemory == nullptr || mFrameMemory->pointer() == nullptr) {
         return false;
@@ -547,7 +651,7 @@
 }
 
 bool HeifDecoderImpl::getScanline(uint8_t* dst) {
-    if (mCurScanline >= mHeight) {
+    if (mCurScanline >= mTotalScanline) {
         ALOGE("no more scanline available");
         return false;
     }
@@ -567,8 +671,8 @@
 size_t HeifDecoderImpl::skipScanlines(size_t count) {
     uint32_t oldScanline = mCurScanline;
     mCurScanline += count;
-    if (mCurScanline > mHeight) {
-        mCurScanline = mHeight;
+    if (mCurScanline > mTotalScanline) {
+        mCurScanline = mTotalScanline;
     }
     return (mCurScanline > oldScanline) ? (mCurScanline - oldScanline) : 0;
 }
diff --git a/media/libheif/HeifDecoderImpl.h b/media/libheif/HeifDecoderImpl.h
index 528ee3b..69c74a7 100644
--- a/media/libheif/HeifDecoderImpl.h
+++ b/media/libheif/HeifDecoderImpl.h
@@ -40,12 +40,16 @@
 
     bool init(HeifStream* stream, HeifFrameInfo* frameInfo) override;
 
+    bool getSequenceInfo(HeifFrameInfo* frameInfo, size_t *frameCount) override;
+
     bool getEncodedColor(HeifEncodedColor* outColor) const override;
 
     bool setOutputColor(HeifColorFormat heifColor) override;
 
     bool decode(HeifFrameInfo* frameInfo) override;
 
+    bool decodeSequence(int frameIndex, HeifFrameInfo* frameInfo) override;
+
     bool getScanline(uint8_t* dst) override;
 
     size_t skipScanlines(size_t count) override;
@@ -56,13 +60,15 @@
     sp<IDataSource> mDataSource;
     sp<MediaMetadataRetriever> mRetriever;
     sp<IMemory> mFrameMemory;
+    HeifFrameInfo mImageInfo;
+    HeifFrameInfo mSequenceInfo;
     android_pixel_format_t mOutputColor;
     size_t mCurScanline;
-    uint32_t mWidth;
-    uint32_t mHeight;
+    size_t mTotalScanline;
     bool mFrameDecoded;
     bool mHasImage;
     bool mHasVideo;
+    size_t mSequenceLength;
 
     // Slice decoding only
     Mutex mLock;
diff --git a/media/libheif/include/HeifDecoderAPI.h b/media/libheif/include/HeifDecoderAPI.h
index aa10f33..9073672 100644
--- a/media/libheif/include/HeifDecoderAPI.h
+++ b/media/libheif/include/HeifDecoderAPI.h
@@ -17,7 +17,7 @@
 #ifndef _HEIF_DECODER_API_
 #define _HEIF_DECODER_API_
 
-#include <memory>
+#include <vector>
 
 /*
  * The output color pixel format of heif decoder.
@@ -40,41 +40,13 @@
 /*
  * Represents a color converted (RGB-based) video frame
  */
-struct HeifFrameInfo
-{
-    HeifFrameInfo() :
-        mWidth(0), mHeight(0), mRotationAngle(0), mBytesPerPixel(0),
-        mIccSize(0), mIccData(nullptr) {}
-
-    // update the frame info, will make a copy of |iccData| internally
-    void set(uint32_t width, uint32_t height, int32_t rotation, uint32_t bpp,
-            uint32_t iccSize, uint8_t* iccData) {
-        mWidth = width;
-        mHeight = height;
-        mRotationAngle = rotation;
-        mBytesPerPixel = bpp;
-
-        if (mIccData != nullptr) {
-            mIccData.reset(nullptr);
-        }
-        mIccSize = iccSize;
-        if (iccSize > 0) {
-            mIccData.reset(new uint8_t[iccSize]);
-            if (mIccData.get() != nullptr) {
-                memcpy(mIccData.get(), iccData, iccSize);
-            } else {
-                mIccSize = 0;
-            }
-        }
-    }
-
-    // Intentional public access modifiers:
+struct HeifFrameInfo {
     uint32_t mWidth;
     uint32_t mHeight;
     int32_t  mRotationAngle;           // Rotation angle, clockwise, should be multiple of 90
     uint32_t mBytesPerPixel;           // Number of bytes for one pixel
-    uint32_t mIccSize;                 // Number of bytes in mIccData
-    std::unique_ptr<uint8_t[]> mIccData; // Actual ICC data, memory is owned by this structure
+    int64_t mDurationUs;               // Duration of the frame in us
+    std::vector<uint8_t> mIccData;     // ICC data array
 };
 
 /*
@@ -113,8 +85,8 @@
     virtual size_t getLength() const = 0;
 
 private:
-    HeifStream(const HeifFrameInfo&) = delete;
-    HeifStream& operator=(const HeifFrameInfo&) = delete;
+    HeifStream(const HeifStream&) = delete;
+    HeifStream& operator=(const HeifStream&) = delete;
 };
 
 /*
@@ -146,6 +118,14 @@
     virtual bool init(HeifStream* stream, HeifFrameInfo* frameInfo) = 0;
 
     /*
+     * Returns true if the stream contains an image sequence and false otherwise.
+     * |frameInfo| will be filled with information of pictures in the sequence
+     * and |frameCount| the length of the sequence upon success and unmodified
+     * upon failure.
+     */
+    virtual bool getSequenceInfo(HeifFrameInfo* frameInfo, size_t *frameCount) = 0;
+
+    /*
      * Decode the picture internally, returning whether it succeeded. |frameInfo|
      * will be filled with information of the primary picture upon success and
      * unmodified upon failure.
@@ -156,6 +136,20 @@
     virtual bool decode(HeifFrameInfo* frameInfo) = 0;
 
     /*
+     * Decode the picture from the image sequence at index |frameIndex|.
+     * |frameInfo| will be filled with information of the decoded picture upon
+     * success and unmodified upon failure.
+     *
+     * |frameIndex| is the 0-based index of the video frame to retrieve. The frame
+     * index must be that of a valid frame. The total number of frames available for
+     * retrieval was reported via getSequenceInfo().
+     *
+     * After this succeeded, getScanline can be called to read the scanlines
+     * that were decoded.
+     */
+    virtual bool decodeSequence(int frameIndex, HeifFrameInfo* frameInfo) = 0;
+
+    /*
      * Read the next scanline (in top-down order), returns true upon success
      * and false otherwise.
      */
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 1c95e27..9f34035 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/AudioParameter.cpp
@@ -40,6 +40,8 @@
         AUDIO_PARAMETER_KEY_AUDIO_LANGUAGE_PREFERRED;
 const char * const AudioParameter::keyMonoOutput = AUDIO_PARAMETER_MONO_OUTPUT;
 const char * const AudioParameter::keyStreamHwAvSync = AUDIO_PARAMETER_STREAM_HW_AV_SYNC;
+const char * const AudioParameter::keyDeviceConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
+const char * const AudioParameter::keyDeviceDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
 const char * const AudioParameter::keyStreamConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
 const char * const AudioParameter::keyStreamDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
 const char * const AudioParameter::keyStreamSupportedFormats = AUDIO_PARAMETER_STREAM_SUP_FORMATS;
diff --git a/media/libmedia/IMediaMetadataRetriever.cpp b/media/libmedia/IMediaMetadataRetriever.cpp
index f9fa86e..d95bc8e 100644
--- a/media/libmedia/IMediaMetadataRetriever.cpp
+++ b/media/libmedia/IMediaMetadataRetriever.cpp
@@ -109,7 +109,7 @@
             data.writeInt32(0);
         } else {
             // serialize the headers
-            data.writeInt64(headers->size());
+            data.writeInt32(headers->size());
             for (size_t i = 0; i < headers->size(); ++i) {
                 data.writeString8(headers->keyAt(i));
                 data.writeString8(headers->valueAt(i));
@@ -213,15 +213,14 @@
         return interface_cast<IMemory>(reply.readStrongBinder());
     }
 
-    status_t getFrameAtIndex(std::vector<sp<IMemory> > *frames,
-            int frameIndex, int numFrames, int colorFormat, bool metaOnly)
+    sp<IMemory> getFrameAtIndex(
+            int index, int colorFormat, bool metaOnly)
     {
-        ALOGV("getFrameAtIndex: frameIndex(%d), numFrames(%d), colorFormat(%d) metaOnly(%d)",
-                frameIndex, numFrames, colorFormat, metaOnly);
+        ALOGV("getFrameAtIndex: index(%d), colorFormat(%d) metaOnly(%d)",
+                index, colorFormat, metaOnly);
         Parcel data, reply;
         data.writeInterfaceToken(IMediaMetadataRetriever::getInterfaceDescriptor());
-        data.writeInt32(frameIndex);
-        data.writeInt32(numFrames);
+        data.writeInt32(index);
         data.writeInt32(colorFormat);
         data.writeInt32(metaOnly);
 #ifndef DISABLE_GROUP_SCHEDULE_HACK
@@ -230,16 +229,9 @@
         remote()->transact(GET_FRAME_AT_INDEX, data, &reply);
         status_t ret = reply.readInt32();
         if (ret != NO_ERROR) {
-            return ret;
+            return NULL;
         }
-        int retNumFrames = reply.readInt32();
-        if (retNumFrames < numFrames) {
-            numFrames = retNumFrames;
-        }
-        for (int i = 0; i < numFrames; i++) {
-            frames->push_back(interface_cast<IMemory>(reply.readStrongBinder()));
-        }
-        return OK;
+        return interface_cast<IMemory>(reply.readStrongBinder());
     }
 
     sp<IMemory> extractAlbumArt()
@@ -318,11 +310,22 @@
             }
 
             KeyedVector<String8, String8> headers;
-            size_t numHeaders = (size_t) data.readInt64();
+            size_t numHeaders = (size_t) data.readInt32();
             for (size_t i = 0; i < numHeaders; ++i) {
-                String8 key = data.readString8();
-                String8 value = data.readString8();
-                headers.add(key, value);
+                String8 key;
+                String8 value;
+                status_t status;
+                status = data.readString8(&key);
+                if (status != OK) {
+                    return status;
+                }
+                status = data.readString8(&value);
+                if (status != OK) {
+                    return status;
+                }
+                if (headers.add(key, value) < 0) {
+                    return UNKNOWN_ERROR;
+                }
             }
 
             reply->writeInt32(
@@ -431,24 +434,20 @@
 
         case GET_FRAME_AT_INDEX: {
             CHECK_INTERFACE(IMediaMetadataRetriever, data, reply);
-            int frameIndex = data.readInt32();
-            int numFrames = data.readInt32();
+            int index = data.readInt32();
             int colorFormat = data.readInt32();
             bool metaOnly = (data.readInt32() != 0);
-            ALOGV("getFrameAtIndex: frameIndex(%d), numFrames(%d), colorFormat(%d), metaOnly(%d)",
-                    frameIndex, numFrames, colorFormat, metaOnly);
+            ALOGV("getFrameAtIndex: index(%d), colorFormat(%d), metaOnly(%d)",
+                    index, colorFormat, metaOnly);
 #ifndef DISABLE_GROUP_SCHEDULE_HACK
             setSchedPolicy(data);
 #endif
-            std::vector<sp<IMemory> > frames;
-            status_t err = getFrameAtIndex(
-                    &frames, frameIndex, numFrames, colorFormat, metaOnly);
-            reply->writeInt32(err);
-            if (OK == err) {
-                reply->writeInt32(frames.size());
-                for (size_t i = 0; i < frames.size(); i++) {
-                    reply->writeStrongBinder(IInterface::asBinder(frames[i]));
-                }
+            sp<IMemory> frame = getFrameAtIndex(index, colorFormat, metaOnly);
+            if (frame != nullptr) {  // Don't send NULL across the binder interface
+                reply->writeInt32(NO_ERROR);
+                reply->writeStrongBinder(IInterface::asBinder(frame));
+            } else {
+                reply->writeInt32(UNKNOWN_ERROR);
             }
 #ifndef DISABLE_GROUP_SCHEDULE_HACK
             restoreSchedPolicy();
diff --git a/media/libmedia/include/media/IMediaMetadataRetriever.h b/media/libmedia/include/media/IMediaMetadataRetriever.h
index c6f422d..28d2192 100644
--- a/media/libmedia/include/media/IMediaMetadataRetriever.h
+++ b/media/libmedia/include/media/IMediaMetadataRetriever.h
@@ -48,9 +48,8 @@
             int index, int colorFormat, bool metaOnly, bool thumbnail) = 0;
     virtual sp<IMemory>     getImageRectAtIndex(
             int index, int colorFormat, int left, int top, int right, int bottom) = 0;
-    virtual status_t        getFrameAtIndex(
-            std::vector<sp<IMemory> > *frames,
-            int frameIndex, int numFrames, int colorFormat, bool metaOnly) = 0;
+    virtual sp<IMemory>     getFrameAtIndex(
+            int index, int colorFormat, bool metaOnly) = 0;
     virtual sp<IMemory>     extractAlbumArt() = 0;
     virtual const char*     extractMetadata(int keyCode) = 0;
 };
diff --git a/media/libmedia/include/media/MediaMetadataRetrieverInterface.h b/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
index 98d300f..37dc401 100644
--- a/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
+++ b/media/libmedia/include/media/MediaMetadataRetrieverInterface.h
@@ -49,9 +49,8 @@
             int index, int colorFormat, bool metaOnly, bool thumbnail) = 0;
     virtual sp<IMemory> getImageRectAtIndex(
             int index, int colorFormat, int left, int top, int right, int bottom) = 0;
-    virtual status_t getFrameAtIndex(
-            std::vector<sp<IMemory> >* frames,
-            int frameIndex, int numFrames, int colorFormat, bool metaOnly) = 0;
+    virtual sp<IMemory> getFrameAtIndex(
+            int frameIndex, int colorFormat, bool metaOnly) = 0;
     virtual MediaAlbumArt* extractAlbumArt() = 0;
     virtual const char* extractMetadata(int keyCode) = 0;
 };
diff --git a/media/libmedia/include/media/TypeConverter.h b/media/libmedia/include/media/TypeConverter.h
index 2f8c209..011498a 100644
--- a/media/libmedia/include/media/TypeConverter.h
+++ b/media/libmedia/include/media/TypeConverter.h
@@ -17,10 +17,11 @@
 #ifndef ANDROID_TYPE_CONVERTER_H_
 #define ANDROID_TYPE_CONVERTER_H_
 
+#include <set>
 #include <string>
 #include <string.h>
-
 #include <vector>
+
 #include <system/audio.h>
 #include <utils/Log.h>
 #include <utils/Vector.h>
@@ -42,16 +43,6 @@
     }
 };
 template <typename T>
-struct VectorTraits
-{
-    typedef T Type;
-    typedef Vector<Type> Collection;
-    static void add(Collection &collection, Type value)
-    {
-        collection.add(value);
-    }
-};
-template <typename T>
 struct SortedVectorTraits
 {
     typedef T Type;
@@ -61,18 +52,28 @@
         collection.add(value);
     }
 };
+template <typename T>
+struct SetTraits
+{
+    typedef T Type;
+    typedef std::set<Type> Collection;
+    static void add(Collection &collection, Type value)
+    {
+        collection.insert(value);
+    }
+};
 
-using SampleRateTraits = SortedVectorTraits<uint32_t>;
+using SampleRateTraits = SetTraits<uint32_t>;
 using DeviceTraits = DefaultTraits<audio_devices_t>;
 struct OutputDeviceTraits : public DeviceTraits {};
 struct InputDeviceTraits : public DeviceTraits {};
-using ChannelTraits = SortedVectorTraits<audio_channel_mask_t>;
+using ChannelTraits = SetTraits<audio_channel_mask_t>;
 struct OutputChannelTraits : public ChannelTraits {};
 struct InputChannelTraits : public ChannelTraits {};
 struct ChannelIndexTraits : public ChannelTraits {};
 using InputFlagTraits = DefaultTraits<audio_input_flags_t>;
 using OutputFlagTraits = DefaultTraits<audio_output_flags_t>;
-using FormatTraits = VectorTraits<audio_format_t>;
+using FormatTraits = DefaultTraits<audio_format_t>;
 using GainModeTraits = DefaultTraits<audio_gain_mode_t>;
 using StreamTraits = DefaultTraits<audio_stream_type_t>;
 using AudioModeTraits = DefaultTraits<audio_mode_t>;
@@ -259,6 +260,7 @@
                                     || std::is_same<T, audio_source_t>::value
                                     || std::is_same<T, audio_stream_type_t>::value
                                     || std::is_same<T, audio_usage_t>::value
+                                    || std::is_same<T, audio_format_t>::value
                                     , int> = 0>
 static inline std::string toString(const T& value)
 {
@@ -291,14 +293,6 @@
     return result;
 }
 
-// TODO: Remove when FormatTraits uses DefaultTraits.
-static inline std::string toString(const audio_format_t& format)
-{
-    std::string result;
-    return TypeConverter<VectorTraits<audio_format_t>>::toString(format, result)
-            ? result : std::to_string(static_cast<int>(format));
-}
-
 static inline std::string toString(const audio_attributes_t& attributes)
 {
     std::ostringstream result;
diff --git a/media/libmedia/include/media/mediametadataretriever.h b/media/libmedia/include/media/mediametadataretriever.h
index d29e97d..138a014 100644
--- a/media/libmedia/include/media/mediametadataretriever.h
+++ b/media/libmedia/include/media/mediametadataretriever.h
@@ -98,9 +98,8 @@
             int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false, bool thumbnail = false);
     sp<IMemory> getImageRectAtIndex(
             int index, int colorFormat, int left, int top, int right, int bottom);
-    status_t getFrameAtIndex(
-            std::vector<sp<IMemory> > *frames, int frameIndex, int numFrames = 1,
-            int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false);
+    sp<IMemory>  getFrameAtIndex(
+            int index, int colorFormat = HAL_PIXEL_FORMAT_RGB_565, bool metaOnly = false);
     sp<IMemory> extractAlbumArt();
     const char* extractMetadata(int keyCode);
 
diff --git a/media/libmedia/mediametadataretriever.cpp b/media/libmedia/mediametadataretriever.cpp
index e61b04d..2ae76b3 100644
--- a/media/libmedia/mediametadataretriever.cpp
+++ b/media/libmedia/mediametadataretriever.cpp
@@ -179,18 +179,16 @@
             index, colorFormat, left, top, right, bottom);
 }
 
-status_t MediaMetadataRetriever::getFrameAtIndex(
-        std::vector<sp<IMemory> > *frames,
-        int frameIndex, int numFrames, int colorFormat, bool metaOnly) {
-    ALOGV("getFrameAtIndex: frameIndex(%d), numFrames(%d), colorFormat(%d) metaOnly(%d)",
-            frameIndex, numFrames, colorFormat, metaOnly);
+sp<IMemory>  MediaMetadataRetriever::getFrameAtIndex(
+        int index, int colorFormat, bool metaOnly) {
+    ALOGV("getFrameAtIndex: index(%d), colorFormat(%d) metaOnly(%d)",
+            index, colorFormat, metaOnly);
     Mutex::Autolock _l(mLock);
     if (mRetriever == 0) {
         ALOGE("retriever is not initialized");
-        return INVALID_OPERATION;
+        return NULL;
     }
-    return mRetriever->getFrameAtIndex(
-            frames, frameIndex, numFrames, colorFormat, metaOnly);
+    return mRetriever->getFrameAtIndex(index, colorFormat, metaOnly);
 }
 
 const char* MediaMetadataRetriever::extractMetadata(int keyCode)
diff --git a/media/libmediametrics/Android.bp b/media/libmediametrics/Android.bp
index 15ea578..9d348ec 100644
--- a/media/libmediametrics/Android.bp
+++ b/media/libmediametrics/Android.bp
@@ -37,6 +37,15 @@
             "1" ,
         ]
     },
+
+    header_abi_checker: {
+        enabled: true,
+        symbol_file: "libmediametrics.map.txt",
+    },
+
+    visibility: [
+        "//frameworks/av:__subpackages__",
+        "//frameworks/base/core/jni",
+        "//frameworks/base/media/jni",
+    ],
 }
-
-
diff --git a/media/libmediametrics/MediaAnalyticsItem.cpp b/media/libmediametrics/MediaAnalyticsItem.cpp
index 02c23b1..b7856a6 100644
--- a/media/libmediametrics/MediaAnalyticsItem.cpp
+++ b/media/libmediametrics/MediaAnalyticsItem.cpp
@@ -64,6 +64,16 @@
     return item;
 }
 
+MediaAnalyticsItem* MediaAnalyticsItem::convert(mediametrics_handle_t handle) {
+    MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
+    return item;
+}
+
+mediametrics_handle_t MediaAnalyticsItem::convert(MediaAnalyticsItem *item ) {
+    mediametrics_handle_t handle = (mediametrics_handle_t) item;
+    return handle;
+}
+
 // access functions for the class
 MediaAnalyticsItem::MediaAnalyticsItem()
     : mPid(-1),
diff --git a/media/libmediametrics/MediaMetrics.cpp b/media/libmediametrics/MediaMetrics.cpp
index 6109190..360ae0c 100644
--- a/media/libmediametrics/MediaMetrics.cpp
+++ b/media/libmediametrics/MediaMetrics.cpp
@@ -169,6 +169,11 @@
     return item->selfrecord();
 }
 
+mediametrics_handle_t mediametrics_dup(mediametrics_handle_t handle) {
+    android::MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
+    if (item == NULL) return android::MediaAnalyticsItem::convert(item);
+    return android::MediaAnalyticsItem::convert(item->dup());
+}
 
 const char *mediametrics_readable(mediametrics_handle_t handle) {
     android::MediaAnalyticsItem *item = (android::MediaAnalyticsItem *) handle;
diff --git a/media/libmediametrics/include/MediaAnalyticsItem.h b/media/libmediametrics/include/MediaAnalyticsItem.h
index 4a36f6a..42a2f5b 100644
--- a/media/libmediametrics/include/MediaAnalyticsItem.h
+++ b/media/libmediametrics/include/MediaAnalyticsItem.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_MEDIA_MEDIAANALYTICSITEM_H
 #define ANDROID_MEDIA_MEDIAANALYTICSITEM_H
 
+#include "MediaMetrics.h"
+
 #include <string>
 #include <sys/types.h>
 
@@ -94,6 +96,9 @@
         static MediaAnalyticsItem* create(Key key);
         static MediaAnalyticsItem* create();
 
+        static MediaAnalyticsItem* convert(mediametrics_handle_t);
+        static mediametrics_handle_t convert(MediaAnalyticsItem *);
+
         // access functions for the class
         ~MediaAnalyticsItem();
 
diff --git a/media/libmediametrics/include/MediaMetrics.h b/media/libmediametrics/include/MediaMetrics.h
index a4e1ed2..29fb241 100644
--- a/media/libmediametrics/include/MediaMetrics.h
+++ b/media/libmediametrics/include/MediaMetrics.h
@@ -79,6 +79,7 @@
 // # of attributes set within this record.
 int32_t mediametrics_count(mediametrics_handle_t handle);
 
+mediametrics_handle_t mediametrics_dup(mediametrics_handle_t handle);
 bool mediametrics_selfRecord(mediametrics_handle_t handle);
 
 const char *mediametrics_readable(mediametrics_handle_t handle);
diff --git a/media/libmediaplayerservice/MediaPlayerService.cpp b/media/libmediaplayerservice/MediaPlayerService.cpp
index dfd3933..8ac169f 100644
--- a/media/libmediaplayerservice/MediaPlayerService.cpp
+++ b/media/libmediaplayerservice/MediaPlayerService.cpp
@@ -58,6 +58,7 @@
 #include <media/AudioTrack.h>
 #include <media/MemoryLeakTrackUtil.h>
 #include <media/stagefright/InterfaceUtils.h>
+#include <media/stagefright/MediaCodecConstants.h>
 #include <media/stagefright/MediaCodecList.h>
 #include <media/stagefright/MediaErrors.h>
 #include <media/stagefright/Utils.h>
@@ -264,6 +265,172 @@
     return ok;
 }
 
+static void dumpCodecDetails(int fd, const sp<IMediaCodecList> &codecList, bool queryDecoders) {
+    const size_t SIZE = 256;
+    char buffer[SIZE];
+    String8 result;
+
+    const char *codecType = queryDecoders? "Decoder" : "Encoder";
+    snprintf(buffer, SIZE - 1, "\n%s infos by media types:\n"
+             "=============================\n", codecType);
+    result.append(buffer);
+
+    size_t numCodecs = codecList->countCodecs();
+
+    // gather all media types supported by codec class, and link to codecs that support them
+    KeyedVector<AString, Vector<sp<MediaCodecInfo>>> allMediaTypes;
+    for (size_t codec_ix = 0; codec_ix < numCodecs; ++codec_ix) {
+        sp<MediaCodecInfo> info = codecList->getCodecInfo(codec_ix);
+        if (info->isEncoder() == !queryDecoders) {
+            Vector<AString> supportedMediaTypes;
+            info->getSupportedMediaTypes(&supportedMediaTypes);
+            if (!supportedMediaTypes.size()) {
+                snprintf(buffer, SIZE - 1, "warning: %s does not support any media types\n",
+                        info->getCodecName());
+                result.append(buffer);
+            } else {
+                for (const AString &mediaType : supportedMediaTypes) {
+                    if (allMediaTypes.indexOfKey(mediaType) < 0) {
+                        allMediaTypes.add(mediaType, Vector<sp<MediaCodecInfo>>());
+                    }
+                    allMediaTypes.editValueFor(mediaType).add(info);
+                }
+            }
+        }
+    }
+
+    KeyedVector<AString, bool> visitedCodecs;
+    for (size_t type_ix = 0; type_ix < allMediaTypes.size(); ++type_ix) {
+        const AString &mediaType = allMediaTypes.keyAt(type_ix);
+        snprintf(buffer, SIZE - 1, "\nMedia type '%s':\n", mediaType.c_str());
+        result.append(buffer);
+
+        for (const sp<MediaCodecInfo> &info : allMediaTypes.valueAt(type_ix)) {
+            sp<MediaCodecInfo::Capabilities> caps = info->getCapabilitiesFor(mediaType.c_str());
+            if (caps == NULL) {
+                snprintf(buffer, SIZE - 1, "warning: %s does not have capabilities for type %s\n",
+                        info->getCodecName(), mediaType.c_str());
+                result.append(buffer);
+                continue;
+            }
+            snprintf(buffer, SIZE - 1, "  %s \"%s\" supports\n",
+                       codecType, info->getCodecName());
+            result.append(buffer);
+
+            auto printList = [&](const char *type, const Vector<AString> &values){
+                snprintf(buffer, SIZE - 1, "    %s: [", type);
+                result.append(buffer);
+                for (size_t j = 0; j < values.size(); ++j) {
+                    snprintf(buffer, SIZE - 1, "\n      %s%s", values[j].c_str(),
+                            j == values.size() - 1 ? " " : ",");
+                    result.append(buffer);
+                }
+                result.append("]\n");
+            };
+
+            if (visitedCodecs.indexOfKey(info->getCodecName()) < 0) {
+                visitedCodecs.add(info->getCodecName(), true);
+                {
+                    Vector<AString> aliases;
+                    info->getAliases(&aliases);
+                    // quote alias
+                    for (AString &alias : aliases) {
+                        alias.insert("\"", 1, 0);
+                        alias.append('"');
+                    }
+                    printList("aliases", aliases);
+                }
+                {
+                    uint32_t attrs = info->getAttributes();
+                    Vector<AString> list;
+                    list.add(AStringPrintf("encoder: %d",
+                                           !!(attrs & MediaCodecInfo::kFlagIsEncoder)));
+                    list.add(AStringPrintf("vendor: %d",
+                                           !!(attrs & MediaCodecInfo::kFlagIsVendor)));
+                    list.add(AStringPrintf("software-only: %d",
+                                           !!(attrs & MediaCodecInfo::kFlagIsSoftwareOnly)));
+                    list.add(AStringPrintf("hw-accelerated: %d",
+                                           !!(attrs & MediaCodecInfo::kFlagIsHardwareAccelerated)));
+                    printList(AStringPrintf("attributes: %#x", attrs).c_str(), list);
+                }
+
+                snprintf(buffer, SIZE - 1, "    owner: \"%s\"\n", info->getOwnerName());
+                result.append(buffer);
+                snprintf(buffer, SIZE - 1, "    rank: %u\n", info->getRank());
+                result.append(buffer);
+            } else {
+                result.append("    aliases, attributes, owner, rank: see above\n");
+            }
+
+            {
+                Vector<AString> list;
+                Vector<MediaCodecInfo::ProfileLevel> profileLevels;
+                caps->getSupportedProfileLevels(&profileLevels);
+                for (const MediaCodecInfo::ProfileLevel &pl : profileLevels) {
+                    const char *niceProfile =
+                        mediaType.equalsIgnoreCase(MIMETYPE_AUDIO_AAC)
+                            ? asString_AACObject(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_MPEG2)
+                            ? asString_MPEG2Profile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_H263)
+                            ? asString_H263Profile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_MPEG4)
+                            ? asString_MPEG4Profile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_AVC)
+                            ? asString_AVCProfile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_VP8)
+                            ? asString_VP8Profile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_HEVC)
+                            ? asString_HEVCProfile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_VP9)
+                            ? asString_VP9Profile(pl.mProfile) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_AV1)
+                            ? asString_AV1Profile(pl.mProfile) : "??";
+                    const char *niceLevel =
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_MPEG2)
+                            ? asString_MPEG2Level(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_H263)
+                            ? asString_H263Level(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_MPEG4)
+                            ? asString_MPEG4Level(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_AVC)
+                            ? asString_AVCLevel(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_VP8)
+                            ? asString_VP8Level(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_HEVC)
+                            ? asString_HEVCTierLevel(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_VP9)
+                            ? asString_VP9Level(pl.mLevel) :
+                        mediaType.equalsIgnoreCase(MIMETYPE_VIDEO_AV1)
+                            ? asString_AV1Level(pl.mLevel) : "??";
+
+                    list.add(AStringPrintf("% 5u/% 5u (%s/%s)",
+                            pl.mProfile, pl.mLevel, niceProfile, niceLevel));
+                }
+                printList("profile/levels", list);
+            }
+
+            {
+                Vector<AString> list;
+                Vector<uint32_t> colors;
+                caps->getSupportedColorFormats(&colors);
+                for (uint32_t color : colors) {
+                    list.add(AStringPrintf("%#x (%s)", color,
+                            asString_ColorFormat((int32_t)color)));
+                }
+                printList("colors", list);
+            }
+
+            snprintf(buffer, SIZE - 1, "    details: %s\n",
+                     caps->getDetails()->debugString(6).c_str());
+            result.append(buffer);
+        }
+    }
+    result.append("\n");
+    ::write(fd, result.string(), result.size());
+}
+
+
 // TODO: Find real cause of Audio/Video delay in PV framework and remove this workaround
 /* static */ int MediaPlayerService::AudioOutput::mMinBufferCount = 4;
 /* static */ bool MediaPlayerService::AudioOutput::mIsOnEmulator = false;
@@ -423,7 +590,7 @@
     SortedVector< sp<MediaRecorderClient> > mediaRecorderClients;
 
     if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
-        snprintf(buffer, SIZE, "Permission Denial: "
+        snprintf(buffer, SIZE - 1, "Permission Denial: "
                 "can't dump MediaPlayerService from pid=%d, uid=%d\n",
                 IPCThreadState::self()->getCallingPid(),
                 IPCThreadState::self()->getCallingUid());
@@ -452,11 +619,11 @@
         }
 
         result.append(" Files opened and/or mapped:\n");
-        snprintf(buffer, SIZE, "/proc/%d/maps", getpid());
+        snprintf(buffer, SIZE - 1, "/proc/%d/maps", getpid());
         FILE *f = fopen(buffer, "r");
         if (f) {
             while (!feof(f)) {
-                fgets(buffer, SIZE, f);
+                fgets(buffer, SIZE - 1, f);
                 if (strstr(buffer, " /storage/") ||
                     strstr(buffer, " /system/sounds/") ||
                     strstr(buffer, " /data/") ||
@@ -472,13 +639,13 @@
             result.append("\n");
         }
 
-        snprintf(buffer, SIZE, "/proc/%d/fd", getpid());
+        snprintf(buffer, SIZE - 1, "/proc/%d/fd", getpid());
         DIR *d = opendir(buffer);
         if (d) {
             struct dirent *ent;
             while((ent = readdir(d)) != NULL) {
                 if (strcmp(ent->d_name,".") && strcmp(ent->d_name,"..")) {
-                    snprintf(buffer, SIZE, "/proc/%d/fd/%s", getpid(), ent->d_name);
+                    snprintf(buffer, SIZE - 1, "/proc/%d/fd/%s", getpid(), ent->d_name);
                     struct stat s;
                     if (lstat(buffer, &s) == 0) {
                         if ((s.st_mode & S_IFMT) == S_IFLNK) {
@@ -543,6 +710,11 @@
         }
     }
     write(fd, result.string(), result.size());
+
+    sp<IMediaCodecList> codecList = getCodecList();
+    dumpCodecDetails(fd, codecList, true /* decoders */);
+    dumpCodecDetails(fd, codecList, false /* !decoders */);
+
     return NO_ERROR;
 }
 
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.cpp b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
index 40b17bf..4a3c65e 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.cpp
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.cpp
@@ -242,31 +242,27 @@
     sp<IMemory> frame = mRetriever->getImageRectAtIndex(
             index, colorFormat, left, top, right, bottom);
     if (frame == NULL) {
-        ALOGE("failed to extract image");
-        return NULL;
+        ALOGE("failed to extract image at index %d", index);
     }
     return frame;
 }
 
-status_t MetadataRetrieverClient::getFrameAtIndex(
-            std::vector<sp<IMemory> > *frames,
-            int frameIndex, int numFrames, int colorFormat, bool metaOnly) {
-    ALOGV("getFrameAtIndex: frameIndex(%d), numFrames(%d), colorFormat(%d), metaOnly(%d)",
-            frameIndex, numFrames, colorFormat, metaOnly);
+sp<IMemory> MetadataRetrieverClient::getFrameAtIndex(
+            int index, int colorFormat, bool metaOnly) {
+    ALOGV("getFrameAtIndex: index(%d), colorFormat(%d), metaOnly(%d)",
+            index, colorFormat, metaOnly);
     Mutex::Autolock lock(mLock);
     Mutex::Autolock glock(sLock);
     if (mRetriever == NULL) {
         ALOGE("retriever is not initialized");
-        return INVALID_OPERATION;
+        return NULL;
     }
 
-    status_t err = mRetriever->getFrameAtIndex(
-            frames, frameIndex, numFrames, colorFormat, metaOnly);
-    if (err != OK) {
-        frames->clear();
-        return err;
+    sp<IMemory> frame = mRetriever->getFrameAtIndex(index, colorFormat, metaOnly);
+    if (frame == NULL) {
+        ALOGE("failed to extract frame at index %d", index);
     }
-    return OK;
+    return frame;
 }
 
 sp<IMemory> MetadataRetrieverClient::extractAlbumArt()
diff --git a/media/libmediaplayerservice/MetadataRetrieverClient.h b/media/libmediaplayerservice/MetadataRetrieverClient.h
index 272d093..8020441 100644
--- a/media/libmediaplayerservice/MetadataRetrieverClient.h
+++ b/media/libmediaplayerservice/MetadataRetrieverClient.h
@@ -56,9 +56,8 @@
             int index, int colorFormat, bool metaOnly, bool thumbnail);
     virtual sp<IMemory>             getImageRectAtIndex(
             int index, int colorFormat, int left, int top, int right, int bottom);
-    virtual status_t getFrameAtIndex(
-                std::vector<sp<IMemory> > *frames,
-                int frameIndex, int numFrames, int colorFormat, bool metaOnly);
+    virtual sp<IMemory>             getFrameAtIndex(
+            int index, int colorFormat, bool metaOnly);
     virtual sp<IMemory>             extractAlbumArt();
     virtual const char*             extractMetadata(int keyCode);
 
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
index 2f0da2d..ee463ce 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.cpp
@@ -106,16 +106,17 @@
     releaseAndResetMediaBuffers();
 }
 
-sp<AMessage> NuPlayer::Decoder::getStats() const {
+sp<AMessage> NuPlayer::Decoder::getStats() {
 
+    Mutex::Autolock autolock(mStatsLock);
     mStats->setInt64("frames-total", mNumFramesTotal);
     mStats->setInt64("frames-dropped-input", mNumInputFramesDropped);
     mStats->setInt64("frames-dropped-output", mNumOutputFramesDropped);
     mStats->setFloat("frame-rate-total", mFrameRateTotal);
 
-    // i'm mutexed right now.
     // make our own copy, so we aren't victim to any later changes.
     sp<AMessage> copiedStats = mStats->dup();
+
     return copiedStats;
 }
 
@@ -362,13 +363,17 @@
     CHECK_EQ((status_t)OK, mCodec->getOutputFormat(&mOutputFormat));
     CHECK_EQ((status_t)OK, mCodec->getInputFormat(&mInputFormat));
 
-    mStats->setString("mime", mime.c_str());
-    mStats->setString("component-name", mComponentName.c_str());
+    {
+        Mutex::Autolock autolock(mStatsLock);
+        mStats->setString("mime", mime.c_str());
+        mStats->setString("component-name", mComponentName.c_str());
+    }
 
     if (!mIsAudio) {
         int32_t width, height;
         if (mOutputFormat->findInt32("width", &width)
                 && mOutputFormat->findInt32("height", &height)) {
+            Mutex::Autolock autolock(mStatsLock);
             mStats->setInt32("width", width);
             mStats->setInt32("height", height);
         }
@@ -799,6 +804,7 @@
         int32_t width, height;
         if (format->findInt32("width", &width)
                 && format->findInt32("height", &height)) {
+            Mutex::Autolock autolock(mStatsLock);
             mStats->setInt32("width", width);
             mStats->setInt32("height", height);
         }
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
index 3da2f0b..4a52b0c 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoder.h
@@ -34,7 +34,7 @@
             const sp<Surface> &surface = NULL,
             const sp<CCDecoder> &ccDecoder = NULL);
 
-    virtual sp<AMessage> getStats() const;
+    virtual sp<AMessage> getStats();
 
     // sets the output surface of video decoders.
     virtual status_t setVideoSurface(const sp<Surface> &surface);
diff --git a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h
index d44c396..a3e0046 100644
--- a/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h
+++ b/media/libmediaplayerservice/nuplayer/NuPlayerDecoderBase.h
@@ -47,7 +47,7 @@
     void signalResume(bool notifyComplete);
     void initiateShutdown();
 
-    virtual sp<AMessage> getStats() const {
+    virtual sp<AMessage> getStats() {
         return mStats;
     }
 
@@ -88,6 +88,7 @@
     int32_t mBufferGeneration;
     bool mPaused;
     sp<AMessage> mStats;
+    Mutex mStatsLock;
 
 private:
     enum {
diff --git a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
index bf14ec2..83da092 100644
--- a/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
+++ b/media/libmediaplayerservice/nuplayer/RTSPSource.cpp
@@ -144,6 +144,10 @@
     if (mLooper == NULL) {
         return;
     }
+
+    // Close socket before posting message to RTSPSource message handler.
+    close(mHandler->getARTSPConnection()->getSocket());
+
     sp<AMessage> msg = new AMessage(kWhatDisconnect, this);
 
     sp<AMessage> dummy;
diff --git a/media/libstagefright/Android.bp b/media/libstagefright/Android.bp
index 9170805..7eab230 100644
--- a/media/libstagefright/Android.bp
+++ b/media/libstagefright/Android.bp
@@ -129,7 +129,6 @@
         "CameraSource.cpp",
         "CameraSourceTimeLapse.cpp",
         "DataConverter.cpp",
-        "DataSourceBase.cpp",
         "DataSourceFactory.cpp",
         "DataURISource.cpp",
         "ClearFileSource.cpp",
@@ -267,7 +266,6 @@
     srcs: [
         "ClearFileSource.cpp",
         "DataURISource.cpp",
-        "DataSourceBase.cpp",
         "HTTPBase.cpp",
         "HevcUtils.cpp",
         "MediaClock.cpp",
diff --git a/media/libstagefright/DataSourceBase.cpp b/media/libstagefright/DataSourceBase.cpp
deleted file mode 100644
index 8f47ee5..0000000
--- a/media/libstagefright/DataSourceBase.cpp
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- * Copyright (C) 2009 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-//#define LOG_NDEBUG 0
-#define LOG_TAG "DataSourceBase"
-
-#include <media/DataSourceBase.h>
-#include <media/stagefright/foundation/ByteUtils.h>
-#include <media/stagefright/MediaErrors.h>
-#include <utils/String8.h>
-
-namespace android {
-
-bool DataSourceBase::getUInt16(off64_t offset, uint16_t *x) {
-    *x = 0;
-
-    uint8_t byte[2];
-    if (readAt(offset, byte, 2) != 2) {
-        return false;
-    }
-
-    *x = (byte[0] << 8) | byte[1];
-
-    return true;
-}
-
-bool DataSourceBase::getUInt24(off64_t offset, uint32_t *x) {
-    *x = 0;
-
-    uint8_t byte[3];
-    if (readAt(offset, byte, 3) != 3) {
-        return false;
-    }
-
-    *x = (byte[0] << 16) | (byte[1] << 8) | byte[2];
-
-    return true;
-}
-
-bool DataSourceBase::getUInt32(off64_t offset, uint32_t *x) {
-    *x = 0;
-
-    uint32_t tmp;
-    if (readAt(offset, &tmp, 4) != 4) {
-        return false;
-    }
-
-    *x = ntohl(tmp);
-
-    return true;
-}
-
-bool DataSourceBase::getUInt64(off64_t offset, uint64_t *x) {
-    *x = 0;
-
-    uint64_t tmp;
-    if (readAt(offset, &tmp, 8) != 8) {
-        return false;
-    }
-
-    *x = ntoh64(tmp);
-
-    return true;
-}
-
-bool DataSourceBase::getUInt16Var(off64_t offset, uint16_t *x, size_t size) {
-    if (size == 2) {
-        return getUInt16(offset, x);
-    }
-    if (size == 1) {
-        uint8_t tmp;
-        if (readAt(offset, &tmp, 1) == 1) {
-            *x = tmp;
-            return true;
-        }
-    }
-    return false;
-}
-
-bool DataSourceBase::getUInt32Var(off64_t offset, uint32_t *x, size_t size) {
-    if (size == 4) {
-        return getUInt32(offset, x);
-    }
-    if (size == 2) {
-        uint16_t tmp;
-        if (getUInt16(offset, &tmp)) {
-            *x = tmp;
-            return true;
-        }
-    }
-    return false;
-}
-
-bool DataSourceBase::getUInt64Var(off64_t offset, uint64_t *x, size_t size) {
-    if (size == 8) {
-        return getUInt64(offset, x);
-    }
-    if (size == 4) {
-        uint32_t tmp;
-        if (getUInt32(offset, &tmp)) {
-            *x = tmp;
-            return true;
-        }
-    }
-    return false;
-}
-
-status_t DataSourceBase::getSize(off64_t *size) {
-    *size = 0;
-
-    return ERROR_UNSUPPORTED;
-}
-
-bool DataSourceBase::getUri(char *uriString __unused, size_t bufferSize __unused) {
-    return false;
-}
-
-}  // namespace android
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index 18a6bd8..f99dd1c 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -21,6 +21,7 @@
 #include <binder/MemoryBase.h>
 #include <binder/MemoryHeapBase.h>
 #include <gui/Surface.h>
+#include <gui/SurfaceComposerClient.h>
 #include <inttypes.h>
 #include <media/ICrypto.h>
 #include <media/IMediaSource.h>
@@ -28,6 +29,7 @@
 #include <media/stagefright/foundation/avc_utils.h>
 #include <media/stagefright/foundation/ADebug.h>
 #include <media/stagefright/foundation/AMessage.h>
+#include <media/stagefright/foundation/ColorUtils.h>
 #include <media/stagefright/ColorConverter.h>
 #include <media/stagefright/MediaBuffer.h>
 #include <media/stagefright/MediaCodec.h>
@@ -41,10 +43,11 @@
 
 static const int64_t kBufferTimeOutUs = 10000LL; // 10 msec
 static const size_t kRetryCount = 50; // must be >0
+static const int64_t kDefaultSampleDurationUs = 33333LL; // 33ms
 
 sp<IMemory> allocVideoFrame(const sp<MetaData>& trackMeta,
         int32_t width, int32_t height, int32_t tileWidth, int32_t tileHeight,
-        int32_t dstBpp, bool metaOnly = false) {
+        int32_t dstBpp, bool allocRotated, bool metaOnly) {
     int32_t rotationAngle;
     if (!trackMeta->findInt32(kKeyRotation, &rotationAngle)) {
         rotationAngle = 0;  // By default, no rotation
@@ -74,6 +77,14 @@
         displayHeight = height;
     }
 
+    if (allocRotated && (rotationAngle == 90 || rotationAngle == 270)) {
+        int32_t tmp;
+        tmp = width; width = height; height = tmp;
+        tmp = displayWidth; displayWidth = displayHeight; displayHeight = tmp;
+        tmp = tileWidth; tileWidth = tileHeight; tileHeight = tmp;
+        rotationAngle = 0;
+    }
+
     VideoFrame frame(width, height, displayWidth, displayHeight,
             tileWidth, tileHeight, rotationAngle, dstBpp, !metaOnly, iccSize);
 
@@ -94,6 +105,20 @@
     return frameMem;
 }
 
+sp<IMemory> allocVideoFrame(const sp<MetaData>& trackMeta,
+        int32_t width, int32_t height, int32_t tileWidth, int32_t tileHeight,
+        int32_t dstBpp, bool allocRotated = false) {
+    return allocVideoFrame(trackMeta, width, height, tileWidth, tileHeight, dstBpp,
+            allocRotated, false /*metaOnly*/);
+}
+
+sp<IMemory> allocMetaFrame(const sp<MetaData>& trackMeta,
+        int32_t width, int32_t height, int32_t tileWidth, int32_t tileHeight,
+        int32_t dstBpp) {
+    return allocVideoFrame(trackMeta, width, height, tileWidth, tileHeight, dstBpp,
+            false /*allocRotated*/, true /*metaOnly*/);
+}
+
 bool findThumbnailInfo(
         const sp<MetaData> &trackMeta, int32_t *width, int32_t *height,
         uint32_t *type = NULL, const void **data = NULL, size_t *size = NULL) {
@@ -117,23 +142,27 @@
 bool getDstColorFormat(
         android_pixel_format_t colorFormat,
         OMX_COLOR_FORMATTYPE *dstFormat,
+        ui::PixelFormat *captureFormat,
         int32_t *dstBpp) {
     switch (colorFormat) {
         case HAL_PIXEL_FORMAT_RGB_565:
         {
             *dstFormat = OMX_COLOR_Format16bitRGB565;
+            *captureFormat = ui::PixelFormat::RGB_565;
             *dstBpp = 2;
             return true;
         }
         case HAL_PIXEL_FORMAT_RGBA_8888:
         {
             *dstFormat = OMX_COLOR_Format32BitRGBA8888;
+            *captureFormat = ui::PixelFormat::RGBA_8888;
             *dstBpp = 4;
             return true;
         }
         case HAL_PIXEL_FORMAT_BGRA_8888:
         {
             *dstFormat = OMX_COLOR_Format32bitBGRA8888;
+            *captureFormat = ui::PixelFormat::BGRA_8888;
             *dstBpp = 4;
             return true;
         }
@@ -150,9 +179,10 @@
 sp<IMemory> FrameDecoder::getMetadataOnly(
         const sp<MetaData> &trackMeta, int colorFormat, bool thumbnail) {
     OMX_COLOR_FORMATTYPE dstFormat;
+    ui::PixelFormat captureFormat;
     int32_t dstBpp;
-    if (!getDstColorFormat(
-            (android_pixel_format_t)colorFormat, &dstFormat, &dstBpp)) {
+    if (!getDstColorFormat((android_pixel_format_t)colorFormat,
+            &dstFormat, &captureFormat, &dstBpp)) {
         return NULL;
     }
 
@@ -170,8 +200,19 @@
             tileWidth = tileHeight = 0;
         }
     }
-    return allocVideoFrame(trackMeta,
-            width, height, tileWidth, tileHeight, dstBpp, true /*metaOnly*/);
+
+    sp<IMemory> metaMem = allocMetaFrame(trackMeta, width, height, tileWidth, tileHeight, dstBpp);
+
+    // try to fill sequence meta's duration based on average frame rate,
+    // default to 33ms if frame rate is unavailable.
+    int32_t frameRate;
+    VideoFrame* meta = static_cast<VideoFrame*>(metaMem->pointer());
+    if (trackMeta->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) {
+        meta->mDurationUs = 1000000LL / frameRate;
+    } else {
+        meta->mDurationUs = kDefaultSampleDurationUs;
+    }
+    return metaMem;
 }
 
 FrameDecoder::FrameDecoder(
@@ -194,15 +235,30 @@
     }
 }
 
+bool isHDR(const sp<AMessage> &format) {
+    uint32_t standard, range, transfer;
+    if (!format->findInt32("color-standard", (int32_t*)&standard)) {
+        standard = 0;
+    }
+    if (!format->findInt32("color-range", (int32_t*)&range)) {
+        range = 0;
+    }
+    if (!format->findInt32("color-transfer", (int32_t*)&transfer)) {
+        transfer = 0;
+    }
+    return standard == ColorUtils::kColorStandardBT2020 &&
+            transfer == ColorUtils::kColorTransferST2084;
+}
+
 status_t FrameDecoder::init(
-        int64_t frameTimeUs, size_t numFrames, int option, int colorFormat) {
-    if (!getDstColorFormat(
-            (android_pixel_format_t)colorFormat, &mDstFormat, &mDstBpp)) {
+        int64_t frameTimeUs, int option, int colorFormat) {
+    if (!getDstColorFormat((android_pixel_format_t)colorFormat,
+            &mDstFormat, &mCaptureFormat, &mDstBpp)) {
         return ERROR_UNSUPPORTED;
     }
 
     sp<AMessage> videoFormat = onGetFormatAndSeekOptions(
-            frameTimeUs, numFrames, option, &mReadOptions);
+            frameTimeUs, option, &mReadOptions, &mSurface);
     if (videoFormat == NULL) {
         ALOGE("video format or seek mode not supported");
         return ERROR_UNSUPPORTED;
@@ -219,7 +275,7 @@
     }
 
     err = decoder->configure(
-            videoFormat, NULL /* surface */, NULL /* crypto */, 0 /* flags */);
+            videoFormat, mSurface, NULL /* crypto */, 0 /* flags */);
     if (err != OK) {
         ALOGW("configure returned error %d (%s)", err, asString(err));
         decoder->release();
@@ -253,19 +309,7 @@
         return NULL;
     }
 
-    return mFrames.size() > 0 ? mFrames[0] : NULL;
-}
-
-status_t FrameDecoder::extractFrames(std::vector<sp<IMemory> >* frames) {
-    status_t err = extractInternal();
-    if (err != OK) {
-        return err;
-    }
-
-    for (size_t i = 0; i < mFrames.size(); i++) {
-        frames->push_back(mFrames[i]);
-    }
-    return OK;
+    return mFrameMemory;
 }
 
 status_t FrameDecoder::extractInternal() {
@@ -379,8 +423,13 @@
                         ALOGE("failed to get output buffer %zu", index);
                         break;
                     }
-                    err = onOutputReceived(videoFrameBuffer, mOutputFormat, ptsUs, &done);
-                    mDecoder->releaseOutputBuffer(index);
+                    if (mSurface != nullptr) {
+                        mDecoder->renderOutputBufferAndRelease(index);
+                        err = onOutputReceived(videoFrameBuffer, mOutputFormat, ptsUs, &done);
+                    } else {
+                        err = onOutputReceived(videoFrameBuffer, mOutputFormat, ptsUs, &done);
+                        mDecoder->releaseOutputBuffer(index);
+                    }
                 } else {
                     ALOGW("Received error %d (%s) instead of output", err, asString(err));
                     done = true;
@@ -404,22 +453,23 @@
         const sp<MetaData> &trackMeta,
         const sp<IMediaSource> &source)
     : FrameDecoder(componentName, trackMeta, source),
+      mFrame(NULL),
       mIsAvcOrHevc(false),
       mSeekMode(MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC),
       mTargetTimeUs(-1LL),
-      mNumFrames(0),
-      mNumFramesDecoded(0) {
+      mDefaultSampleDurationUs(0) {
 }
 
 sp<AMessage> VideoFrameDecoder::onGetFormatAndSeekOptions(
-        int64_t frameTimeUs, size_t numFrames, int seekMode, MediaSource::ReadOptions *options) {
+        int64_t frameTimeUs, int seekMode,
+        MediaSource::ReadOptions *options,
+        sp<Surface> *window) {
     mSeekMode = static_cast<MediaSource::ReadOptions::SeekMode>(seekMode);
     if (mSeekMode < MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC ||
             mSeekMode > MediaSource::ReadOptions::SEEK_FRAME_INDEX) {
         ALOGE("Unknown seek mode: %d", mSeekMode);
         return NULL;
     }
-    mNumFrames = numFrames;
 
     const char *mime;
     if (!trackMeta()->findCString(kKeyMIMEType, &mime)) {
@@ -460,6 +510,23 @@
         videoFormat->setInt32("android._num-input-buffers", 1);
         videoFormat->setInt32("android._num-output-buffers", 1);
     }
+
+    if (isHDR(videoFormat)) {
+        *window = initSurfaceControl();
+        if (*window == NULL) {
+            ALOGE("Failed to init surface control for HDR, fallback to non-hdr");
+        } else {
+            videoFormat->setInt32("color-format", OMX_COLOR_FormatAndroidOpaque);
+        }
+    }
+
+    int32_t frameRate;
+    if (trackMeta()->findInt32(kKeyFrameRate, &frameRate) && frameRate > 0) {
+        mDefaultSampleDurationUs = 1000000LL / frameRate;
+    } else {
+        mDefaultSampleDurationUs = kDefaultSampleDurationUs;
+    }
+
     return videoFormat;
 }
 
@@ -480,6 +547,12 @@
         // option, in which case we need to actually decode to targetTimeUs.
         *flags |= MediaCodec::BUFFER_FLAG_EOS;
     }
+    int64_t durationUs;
+    if (sampleMeta.findInt64(kKeyDuration, &durationUs)) {
+        mSampleDurations.push_back(durationUs);
+    } else {
+        mSampleDurations.push_back(mDefaultSampleDurationUs);
+    }
     return OK;
 }
 
@@ -487,6 +560,11 @@
         const sp<MediaCodecBuffer> &videoFrameBuffer,
         const sp<AMessage> &outputFormat,
         int64_t timeUs, bool *done) {
+    int64_t durationUs = mDefaultSampleDurationUs;
+    if (!mSampleDurations.empty()) {
+        durationUs = *mSampleDurations.begin();
+        mSampleDurations.erase(mSampleDurations.begin());
+    }
     bool shouldOutput = (mTargetTimeUs < 0LL) || (timeUs >= mTargetTimeUs);
 
     // If this is not the target frame, skip color convert.
@@ -495,7 +573,7 @@
         return OK;
     }
 
-    *done = (++mNumFramesDecoded >= mNumFrames);
+    *done = true;
 
     if (outputFormat == NULL) {
         return ERROR_MALFORMED;
@@ -504,13 +582,22 @@
     int32_t width, height, stride, srcFormat;
     if (!outputFormat->findInt32("width", &width) ||
             !outputFormat->findInt32("height", &height) ||
-            !outputFormat->findInt32("stride", &stride) ||
             !outputFormat->findInt32("color-format", &srcFormat)) {
         ALOGE("format missing dimension or color: %s",
                 outputFormat->debugString().c_str());
         return ERROR_MALFORMED;
     }
 
+    if (!outputFormat->findInt32("stride", &stride)) {
+        if (mSurfaceControl == NULL) {
+            ALOGE("format must have stride for byte buffer mode: %s",
+                    outputFormat->debugString().c_str());
+            return ERROR_MALFORMED;
+        }
+        // for surface output, set stride to width, we don't actually need it.
+        stride = width;
+    }
+
     int32_t crop_left, crop_top, crop_right, crop_bottom;
     if (!outputFormat->findRect("crop", &crop_left, &crop_top, &crop_right, &crop_bottom)) {
         crop_left = crop_top = 0;
@@ -518,15 +605,25 @@
         crop_bottom = height - 1;
     }
 
-    sp<IMemory> frameMem = allocVideoFrame(
-            trackMeta(),
-            (crop_right - crop_left + 1),
-            (crop_bottom - crop_top + 1),
-            0,
-            0,
-            dstBpp());
-    addFrame(frameMem);
-    VideoFrame* frame = static_cast<VideoFrame*>(frameMem->pointer());
+    if (mFrame == NULL) {
+        sp<IMemory> frameMem = allocVideoFrame(
+                trackMeta(),
+                (crop_right - crop_left + 1),
+                (crop_bottom - crop_top + 1),
+                0,
+                0,
+                dstBpp(),
+                mSurfaceControl != nullptr /*allocRotated*/);
+        mFrame = static_cast<VideoFrame*>(frameMem->pointer());
+
+        setFrame(frameMem);
+    }
+
+    mFrame->mDurationUs = durationUs;
+
+    if (mSurfaceControl != nullptr) {
+        return captureSurfaceControl();
+    }
 
     ColorConverter converter((OMX_COLOR_FORMATTYPE)srcFormat, dstFormat());
 
@@ -547,8 +644,8 @@
                 (const uint8_t *)videoFrameBuffer->data(),
                 width, height, stride,
                 crop_left, crop_top, crop_right, crop_bottom,
-                frame->getFlattenedData(),
-                frame->mWidth, frame->mHeight, frame->mRowBytes,
+                mFrame->getFlattenedData(),
+                mFrame->mWidth, mFrame->mHeight, mFrame->mRowBytes,
                 crop_left, crop_top, crop_right, crop_bottom);
         return OK;
     }
@@ -558,6 +655,101 @@
     return ERROR_UNSUPPORTED;
 }
 
+sp<Surface> VideoFrameDecoder::initSurfaceControl() {
+    sp<SurfaceComposerClient> client = new SurfaceComposerClient();
+    if (client->initCheck() != NO_ERROR) {
+        ALOGE("failed to get SurfaceComposerClient");
+        return NULL;
+    }
+
+    // create a container layer to hold the capture layer, so that we can
+    // use full frame drop. If without the container, the crop will be set
+    // to display size.
+    sp<SurfaceControl> parent = client->createSurface(
+            String8("parent"),
+            0 /* width */, 0 /* height */,
+            PIXEL_FORMAT_RGBA_8888,
+            ISurfaceComposerClient::eFXSurfaceContainer );
+
+    if (!parent) {
+        ALOGE("failed to get surface control parent");
+        return NULL;
+    }
+
+    // create the surface with unknown size 1x1 for now, real size will
+    // be set before the capture when we have output format info.
+    sp<SurfaceControl> surfaceControl = client->createSurface(
+            String8("thumbnail"),
+            1 /* width */, 1 /* height */,
+            PIXEL_FORMAT_RGBA_8888,
+            ISurfaceComposerClient::eFXSurfaceBufferQueue,
+            parent.get());
+
+    if (!surfaceControl) {
+        ALOGE("failed to get surface control");
+        return NULL;
+    }
+
+    SurfaceComposerClient::Transaction t;
+    t.hide(parent)
+            .show(surfaceControl)
+            .apply(true);
+
+    mSurfaceControl = surfaceControl;
+    mParent = parent;
+
+    return surfaceControl->getSurface();
+}
+
+status_t VideoFrameDecoder::captureSurfaceControl() {
+    // set the layer size to the output size before the capture
+    SurfaceComposerClient::Transaction()
+        .setSize(mSurfaceControl, mFrame->mWidth, mFrame->mHeight)
+        .apply(true);
+
+    sp<GraphicBuffer> outBuffer;
+    status_t err = ScreenshotClient::captureChildLayers(
+            mParent->getHandle(),
+            ui::Dataspace::V0_SRGB,
+            captureFormat(),
+            Rect(0, 0, mFrame->mWidth, mFrame->mHeight),
+            {},
+            1.0f /*frameScale*/,
+            &outBuffer);
+
+    if (err != OK) {
+        ALOGE("failed to captureLayers: err %d", err);
+        return err;
+    }
+
+    ALOGV("capture: %dx%d, format %d, stride %d",
+            outBuffer->getWidth(),
+            outBuffer->getHeight(),
+            outBuffer->getPixelFormat(),
+            outBuffer->getStride());
+
+    uint8_t *base;
+    int32_t outBytesPerPixel, outBytesPerStride;
+    err = outBuffer->lock(
+            GraphicBuffer::USAGE_SW_READ_OFTEN,
+            reinterpret_cast<void**>(&base),
+            &outBytesPerPixel,
+            &outBytesPerStride);
+    if (err != OK) {
+        ALOGE("failed to lock graphic buffer: err %d", err);
+        return err;
+    }
+
+    uint8_t *dst = mFrame->getFlattenedData();
+    for (size_t y = 0 ; y < fmin(mFrame->mHeight, outBuffer->getHeight()) ; y++) {
+        memcpy(dst, base, fmin(mFrame->mWidth, outBuffer->getWidth()) * mFrame->mBytesPerPixel);
+        dst += mFrame->mRowBytes;
+        base += outBuffer->getStride() * mFrame->mBytesPerPixel;
+    }
+    outBuffer->unlock();
+    return OK;
+}
+
 ////////////////////////////////////////////////////////////////////////
 
 ImageDecoder::ImageDecoder(
@@ -577,8 +769,8 @@
 }
 
 sp<AMessage> ImageDecoder::onGetFormatAndSeekOptions(
-        int64_t frameTimeUs, size_t /*numFrames*/,
-        int /*seekMode*/, MediaSource::ReadOptions *options) {
+        int64_t frameTimeUs, int /*seekMode*/,
+        MediaSource::ReadOptions *options, sp<Surface> * /*window*/) {
     sp<MetaData> overrideMeta;
     if (frameTimeUs < 0) {
         uint32_t type;
@@ -705,7 +897,7 @@
                 trackMeta(), mWidth, mHeight, mTileWidth, mTileHeight, dstBpp());
         mFrame = static_cast<VideoFrame*>(frameMem->pointer());
 
-        addFrame(frameMem);
+        setFrame(frameMem);
     }
 
     int32_t srcFormat;
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index eceb84e..b1404de 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -527,7 +527,7 @@
       mFlags(0),
       mStickyError(OK),
       mSoftRenderer(NULL),
-      mAnalyticsItem(NULL),
+      mMetricsHandle(0),
       mIsVideo(false),
       mVideoWidth(0),
       mVideoHeight(0),
@@ -548,19 +548,19 @@
     mResourceManagerClient = new ResourceManagerClient(this);
     mResourceManagerService = new ResourceManagerServiceProxy(pid, mUid);
 
-    initAnalyticsItem();
+    initMediametrics();
 }
 
 MediaCodec::~MediaCodec() {
     CHECK_EQ(mState, UNINITIALIZED);
     mResourceManagerService->removeClient(getId(mResourceManagerClient));
 
-    flushAnalyticsItem();
+    flushMediametrics();
 }
 
-void MediaCodec::initAnalyticsItem() {
-    if (mAnalyticsItem == NULL) {
-        mAnalyticsItem = MediaAnalyticsItem::create(kCodecKeyName);
+void MediaCodec::initMediametrics() {
+    if (mMetricsHandle == 0) {
+        mMetricsHandle = mediametrics_create(kCodecKeyName);
     }
 
     mLatencyHist.setup(kLatencyHistBuckets, kLatencyHistWidth, kLatencyHistFloor);
@@ -574,38 +574,39 @@
     }
 }
 
-void MediaCodec::updateAnalyticsItem() {
-    ALOGV("MediaCodec::updateAnalyticsItem");
-    if (mAnalyticsItem == NULL) {
+void MediaCodec::updateMediametrics() {
+    ALOGV("MediaCodec::updateMediametrics");
+    if (mMetricsHandle == 0) {
         return;
     }
 
+
     if (mLatencyHist.getCount() != 0 ) {
-        mAnalyticsItem->setInt64(kCodecLatencyMax, mLatencyHist.getMax());
-        mAnalyticsItem->setInt64(kCodecLatencyMin, mLatencyHist.getMin());
-        mAnalyticsItem->setInt64(kCodecLatencyAvg, mLatencyHist.getAvg());
-        mAnalyticsItem->setInt64(kCodecLatencyCount, mLatencyHist.getCount());
+        mediametrics_setInt64(mMetricsHandle, kCodecLatencyMax, mLatencyHist.getMax());
+        mediametrics_setInt64(mMetricsHandle, kCodecLatencyMin, mLatencyHist.getMin());
+        mediametrics_setInt64(mMetricsHandle, kCodecLatencyAvg, mLatencyHist.getAvg());
+        mediametrics_setInt64(mMetricsHandle, kCodecLatencyCount, mLatencyHist.getCount());
 
         if (kEmitHistogram) {
             // and the histogram itself
             std::string hist = mLatencyHist.emit();
-            mAnalyticsItem->setCString(kCodecLatencyHist, hist.c_str());
+            mediametrics_setCString(mMetricsHandle, kCodecLatencyHist, hist.c_str());
         }
     }
     if (mLatencyUnknown > 0) {
-        mAnalyticsItem->setInt64(kCodecLatencyUnknown, mLatencyUnknown);
+        mediametrics_setInt64(mMetricsHandle, kCodecLatencyUnknown, mLatencyUnknown);
     }
 
 #if 0
     // enable for short term, only while debugging
-    updateEphemeralAnalytics(mAnalyticsItem);
+    updateEphemeralMediametrics(mMetricsHandle);
 #endif
 }
 
-void MediaCodec::updateEphemeralAnalytics(MediaAnalyticsItem *item) {
-    ALOGD("MediaCodec::updateEphemeralAnalytics()");
+void MediaCodec::updateEphemeralMediametrics(mediametrics_handle_t item) {
+    ALOGD("MediaCodec::updateEphemeralMediametrics()");
 
-    if (item == NULL) {
+    if (item == 0) {
         return;
     }
 
@@ -628,28 +629,27 @@
 
     // spit the data (if any) into the supplied analytics record
     if (recentHist.getCount()!= 0 ) {
-        item->setInt64(kCodecRecentLatencyMax, recentHist.getMax());
-        item->setInt64(kCodecRecentLatencyMin, recentHist.getMin());
-        item->setInt64(kCodecRecentLatencyAvg, recentHist.getAvg());
-        item->setInt64(kCodecRecentLatencyCount, recentHist.getCount());
+        mediametrics_setInt64(item, kCodecRecentLatencyMax, recentHist.getMax());
+        mediametrics_setInt64(item, kCodecRecentLatencyMin, recentHist.getMin());
+        mediametrics_setInt64(item, kCodecRecentLatencyAvg, recentHist.getAvg());
+        mediametrics_setInt64(item, kCodecRecentLatencyCount, recentHist.getCount());
 
         if (kEmitHistogram) {
             // and the histogram itself
             std::string hist = recentHist.emit();
-            item->setCString(kCodecRecentLatencyHist, hist.c_str());
+            mediametrics_setCString(item, kCodecRecentLatencyHist, hist.c_str());
         }
     }
 }
 
-void MediaCodec::flushAnalyticsItem() {
-    updateAnalyticsItem();
-    if (mAnalyticsItem != NULL) {
-        // don't log empty records
-        if (mAnalyticsItem->count() > 0) {
-            mAnalyticsItem->selfrecord();
+void MediaCodec::flushMediametrics() {
+    updateMediametrics();
+    if (mMetricsHandle != 0) {
+        if (mediametrics_count(mMetricsHandle) > 0) {
+            mediametrics_selfRecord(mMetricsHandle);
         }
-        delete mAnalyticsItem;
-        mAnalyticsItem = NULL;
+        mediametrics_delete(mMetricsHandle);
+        mMetricsHandle = 0;
     }
 }
 
@@ -981,9 +981,10 @@
     // ".secure"
     msg->setString("name", name);
 
-    if (mAnalyticsItem != NULL) {
-        mAnalyticsItem->setCString(kCodecCodec, name.c_str());
-        mAnalyticsItem->setCString(kCodecMode, mIsVideo ? kCodecModeVideo : kCodecModeAudio);
+    if (mMetricsHandle != 0) {
+        mediametrics_setCString(mMetricsHandle, kCodecCodec, name.c_str());
+        mediametrics_setCString(mMetricsHandle, kCodecMode,
+                                mIsVideo ? kCodecModeVideo : kCodecModeAudio);
     }
 
     if (mIsVideo) {
@@ -1044,16 +1045,17 @@
         uint32_t flags) {
     sp<AMessage> msg = new AMessage(kWhatConfigure, this);
 
-    if (mAnalyticsItem != NULL) {
+    if (mMetricsHandle != 0) {
         int32_t profile = 0;
         if (format->findInt32("profile", &profile)) {
-            mAnalyticsItem->setInt32(kCodecProfile, profile);
+            mediametrics_setInt32(mMetricsHandle, kCodecProfile, profile);
         }
         int32_t level = 0;
         if (format->findInt32("level", &level)) {
-            mAnalyticsItem->setInt32(kCodecLevel, level);
+            mediametrics_setInt32(mMetricsHandle, kCodecLevel, level);
         }
-        mAnalyticsItem->setInt32(kCodecEncoder, (flags & CONFIGURE_FLAG_ENCODE) ? 1 : 0);
+        mediametrics_setInt32(mMetricsHandle, kCodecEncoder,
+                              (flags & CONFIGURE_FLAG_ENCODE) ? 1 : 0);
     }
 
     if (mIsVideo) {
@@ -1063,17 +1065,17 @@
             mRotationDegrees = 0;
         }
 
-        if (mAnalyticsItem != NULL) {
-            mAnalyticsItem->setInt32(kCodecWidth, mVideoWidth);
-            mAnalyticsItem->setInt32(kCodecHeight, mVideoHeight);
-            mAnalyticsItem->setInt32(kCodecRotation, mRotationDegrees);
+        if (mMetricsHandle != 0) {
+            mediametrics_setInt32(mMetricsHandle, kCodecWidth, mVideoWidth);
+            mediametrics_setInt32(mMetricsHandle, kCodecHeight, mVideoHeight);
+            mediametrics_setInt32(mMetricsHandle, kCodecRotation, mRotationDegrees);
             int32_t maxWidth = 0;
             if (format->findInt32("max-width", &maxWidth)) {
-                mAnalyticsItem->setInt32(kCodecMaxWidth, maxWidth);
+                mediametrics_setInt32(mMetricsHandle, kCodecMaxWidth, maxWidth);
             }
             int32_t maxHeight = 0;
             if (format->findInt32("max-height", &maxHeight)) {
-                mAnalyticsItem->setInt32(kCodecMaxHeight, maxHeight);
+                mediametrics_setInt32(mMetricsHandle, kCodecMaxHeight, maxHeight);
             }
         }
 
@@ -1095,8 +1097,8 @@
         } else {
             msg->setPointer("descrambler", descrambler.get());
         }
-        if (mAnalyticsItem != NULL) {
-            mAnalyticsItem->setInt32(kCodecCrypto, 1);
+        if (mMetricsHandle != 0) {
+            mediametrics_setInt32(mMetricsHandle, kCodecCrypto, 1);
         }
     } else if (mFlags & kFlagIsSecure) {
         ALOGW("Crypto or descrambler should be given for secure codec");
@@ -1561,22 +1563,22 @@
     return OK;
 }
 
-status_t MediaCodec::getMetrics(MediaAnalyticsItem * &reply) {
+status_t MediaCodec::getMetrics(mediametrics_handle_t &reply) {
 
-    reply = NULL;
+    reply = 0;
 
     // shouldn't happen, but be safe
-    if (mAnalyticsItem == NULL) {
+    if (mMetricsHandle == 0) {
         return UNKNOWN_ERROR;
     }
 
     // update any in-flight data that's not carried within the record
-    updateAnalyticsItem();
+    updateMediametrics();
 
     // send it back to the caller.
-    reply = mAnalyticsItem->dup();
+    reply = mediametrics_dup(mMetricsHandle);
 
-    updateEphemeralAnalytics(reply);
+    updateEphemeralMediametrics(reply);
 
     return OK;
 }
@@ -1890,10 +1892,11 @@
                         case CONFIGURING:
                         {
                             if (actionCode == ACTION_CODE_FATAL) {
-                                mAnalyticsItem->setInt32(kCodecError, err);
-                                mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
-                                flushAnalyticsItem();
-                                initAnalyticsItem();
+                                mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+                                mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+                                                        stateString(mState).c_str());
+                                flushMediametrics();
+                                initMediametrics();
                             }
                             setState(actionCode == ACTION_CODE_FATAL ?
                                     UNINITIALIZED : INITIALIZED);
@@ -1903,10 +1906,11 @@
                         case STARTING:
                         {
                             if (actionCode == ACTION_CODE_FATAL) {
-                                mAnalyticsItem->setInt32(kCodecError, err);
-                                mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
-                                flushAnalyticsItem();
-                                initAnalyticsItem();
+                                mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+                                mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+                                                        stateString(mState).c_str());
+                                flushMediametrics();
+                                initMediametrics();
                             }
                             setState(actionCode == ACTION_CODE_FATAL ?
                                     UNINITIALIZED : CONFIGURED);
@@ -1944,10 +1948,11 @@
                         case FLUSHING:
                         {
                             if (actionCode == ACTION_CODE_FATAL) {
-                                mAnalyticsItem->setInt32(kCodecError, err);
-                                mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
-                                flushAnalyticsItem();
-                                initAnalyticsItem();
+                                mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+                                mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+                                                        stateString(mState).c_str());
+                                flushMediametrics();
+                                initMediametrics();
 
                                 setState(UNINITIALIZED);
                             } else {
@@ -1977,10 +1982,11 @@
                                 setState(INITIALIZED);
                                 break;
                             default:
-                                mAnalyticsItem->setInt32(kCodecError, err);
-                                mAnalyticsItem->setCString(kCodecErrorState, stateString(mState).c_str());
-                                flushAnalyticsItem();
-                                initAnalyticsItem();
+                                mediametrics_setInt32(mMetricsHandle, kCodecError, err);
+                                mediametrics_setCString(mMetricsHandle, kCodecErrorState,
+                                                        stateString(mState).c_str());
+                                flushMediametrics();
+                                initMediametrics();
                                 setState(UNINITIALIZED);
                                 break;
                             }
@@ -2037,7 +2043,8 @@
                     CHECK(msg->findString("componentName", &mComponentName));
 
                     if (mComponentName.c_str()) {
-                        mAnalyticsItem->setCString(kCodecCodec, mComponentName.c_str());
+                        mediametrics_setCString(mMetricsHandle, kCodecCodec,
+                                                mComponentName.c_str());
                     }
 
                     const char *owner = mCodecInfo->getOwnerName();
@@ -2053,11 +2060,11 @@
                     if (mComponentName.endsWith(".secure")) {
                         mFlags |= kFlagIsSecure;
                         resourceType = MediaResource::kSecureCodec;
-                        mAnalyticsItem->setInt32(kCodecSecure, 1);
+                        mediametrics_setInt32(mMetricsHandle, kCodecSecure, 1);
                     } else {
                         mFlags &= ~kFlagIsSecure;
                         resourceType = MediaResource::kNonSecureCodec;
-                        mAnalyticsItem->setInt32(kCodecSecure, 0);
+                        mediametrics_setInt32(mMetricsHandle, kCodecSecure, 0);
                     }
 
                     if (mIsVideo) {
@@ -2105,14 +2112,15 @@
                     (new AMessage)->postReply(mReplyID);
 
                     // augment our media metrics info, now that we know more things
-                    if (mAnalyticsItem != NULL) {
+                    if (mMetricsHandle != 0) {
                         sp<AMessage> format;
                         if (mConfigureMsg != NULL &&
                             mConfigureMsg->findMessage("format", &format)) {
                                 // format includes: mime
                                 AString mime;
                                 if (format->findString("mime", &mime)) {
-                                    mAnalyticsItem->setCString(kCodecMime, mime.c_str());
+                                    mediametrics_setCString(mMetricsHandle, kCodecMime,
+                                                            mime.c_str());
                                 }
                             }
                     }
diff --git a/media/libstagefright/SimpleDecodingSource.cpp b/media/libstagefright/SimpleDecodingSource.cpp
index babdc7a..8b6262f 100644
--- a/media/libstagefright/SimpleDecodingSource.cpp
+++ b/media/libstagefright/SimpleDecodingSource.cpp
@@ -36,7 +36,7 @@
 using namespace android;
 
 const int64_t kTimeoutWaitForOutputUs = 500000; // 0.5 seconds
-const int64_t kTimeoutWaitForInputUs = 5000; // 5 milliseconds
+const int64_t kTimeoutWaitForInputUs = 0; // don't wait
 const int kTimeoutMaxRetries = 20;
 
 //static
diff --git a/media/libstagefright/StagefrightMetadataRetriever.cpp b/media/libstagefright/StagefrightMetadataRetriever.cpp
index fa3d372..6f536a9 100644
--- a/media/libstagefright/StagefrightMetadataRetriever.cpp
+++ b/media/libstagefright/StagefrightMetadataRetriever.cpp
@@ -44,7 +44,7 @@
 StagefrightMetadataRetriever::StagefrightMetadataRetriever()
     : mParsedMetaData(false),
       mAlbumArt(NULL),
-      mLastImageIndex(-1) {
+      mLastDecodedIndex(-1) {
     ALOGV("StagefrightMetadataRetriever()");
 }
 
@@ -143,8 +143,8 @@
 
     FrameRect rect = {left, top, right, bottom};
 
-    if (mImageDecoder != NULL && index == mLastImageIndex) {
-        return mImageDecoder->extractFrame(&rect);
+    if (mDecoder != NULL && index == mLastDecodedIndex) {
+        return mDecoder->extractFrame(&rect);
     }
 
     return getImageInternal(
@@ -153,6 +153,8 @@
 
 sp<IMemory> StagefrightMetadataRetriever::getImageInternal(
         int index, int colorFormat, bool metaOnly, bool thumbnail, FrameRect* rect) {
+    mDecoder.clear();
+    mLastDecodedIndex = -1;
 
     if (mExtractor.get() == NULL) {
         ALOGE("no extractor.");
@@ -227,14 +229,14 @@
         const AString &componentName = matchingCodecs[i];
         sp<ImageDecoder> decoder = new ImageDecoder(componentName, trackMeta, source);
         int64_t frameTimeUs = thumbnail ? -1 : 0;
-        if (decoder->init(frameTimeUs, 1 /*numFrames*/, 0 /*option*/, colorFormat) == OK) {
+        if (decoder->init(frameTimeUs, 0 /*option*/, colorFormat) == OK) {
             sp<IMemory> frame = decoder->extractFrame(rect);
 
             if (frame != NULL) {
                 if (rect != NULL) {
                     // keep the decoder if slice decoding
-                    mImageDecoder = decoder;
-                    mLastImageIndex = index;
+                    mDecoder = decoder;
+                    mLastDecodedIndex = index;
                 }
                 return frame;
             }
@@ -242,6 +244,7 @@
         ALOGV("%s failed to extract thumbnail, trying next decoder.", componentName.c_str());
     }
 
+    ALOGE("all codecs failed to extract frame.");
     return NULL;
 }
 
@@ -250,36 +253,40 @@
     ALOGV("getFrameAtTime: %" PRId64 " us option: %d colorFormat: %d, metaOnly: %d",
             timeUs, option, colorFormat, metaOnly);
 
-    sp<IMemory> frame;
-    status_t err = getFrameInternal(
-            timeUs, 1, option, colorFormat, metaOnly, &frame, NULL /*outFrames*/);
-    return (err == OK) ? frame : NULL;
+    return getFrameInternal(timeUs, option, colorFormat, metaOnly);
 }
 
-status_t StagefrightMetadataRetriever::getFrameAtIndex(
-        std::vector<sp<IMemory> >* frames,
-        int frameIndex, int numFrames, int colorFormat, bool metaOnly) {
-    ALOGV("getFrameAtIndex: frameIndex %d, numFrames %d, colorFormat: %d, metaOnly: %d",
-            frameIndex, numFrames, colorFormat, metaOnly);
+sp<IMemory> StagefrightMetadataRetriever::getFrameAtIndex(
+        int frameIndex, int colorFormat, bool metaOnly) {
+    ALOGV("getFrameAtIndex: frameIndex %d, colorFormat: %d, metaOnly: %d",
+            frameIndex, colorFormat, metaOnly);
+    if (mDecoder != NULL && frameIndex == mLastDecodedIndex + 1) {
+        sp<IMemory> frame = mDecoder->extractFrame();
+        if (frame != nullptr) {
+            mLastDecodedIndex = frameIndex;
+        }
+        return frame;
+    }
 
-    return getFrameInternal(
-            frameIndex, numFrames, MediaSource::ReadOptions::SEEK_FRAME_INDEX,
-            colorFormat, metaOnly, NULL /*outFrame*/, frames);
+    return getFrameInternal(frameIndex,
+            MediaSource::ReadOptions::SEEK_FRAME_INDEX, colorFormat, metaOnly);
 }
 
-status_t StagefrightMetadataRetriever::getFrameInternal(
-        int64_t timeUs, int numFrames, int option, int colorFormat, bool metaOnly,
-        sp<IMemory>* outFrame, std::vector<sp<IMemory> >* outFrames) {
+sp<IMemory> StagefrightMetadataRetriever::getFrameInternal(
+        int64_t timeUs, int option, int colorFormat, bool metaOnly) {
+    mDecoder.clear();
+    mLastDecodedIndex = -1;
+
     if (mExtractor.get() == NULL) {
         ALOGE("no extractor.");
-        return NO_INIT;
+        return NULL;
     }
 
     sp<MetaData> fileMeta = mExtractor->getMetaData();
 
     if (fileMeta == NULL) {
         ALOGE("extractor doesn't publish metadata, failed to initialize?");
-        return NO_INIT;
+        return NULL;
     }
 
     size_t n = mExtractor->countTracks();
@@ -300,30 +307,24 @@
 
     if (i == n) {
         ALOGE("no video track found.");
-        return INVALID_OPERATION;
+        return NULL;
     }
 
     sp<MetaData> trackMeta = mExtractor->getTrackMetaData(
             i, MediaExtractor::kIncludeExtensiveMetaData);
     if (!trackMeta) {
-        return UNKNOWN_ERROR;
+        return NULL;
     }
 
     if (metaOnly) {
-        if (outFrame != NULL) {
-            *outFrame = FrameDecoder::getMetadataOnly(trackMeta, colorFormat);
-            if (*outFrame != NULL) {
-                return OK;
-            }
-        }
-        return UNKNOWN_ERROR;
+        return FrameDecoder::getMetadataOnly(trackMeta, colorFormat);
     }
 
     sp<IMediaSource> source = mExtractor->getTrack(i);
 
     if (source.get() == NULL) {
         ALOGV("unable to instantiate video track.");
-        return UNKNOWN_ERROR;
+        return NULL;
     }
 
     const void *data;
@@ -350,24 +351,22 @@
     for (size_t i = 0; i < matchingCodecs.size(); ++i) {
         const AString &componentName = matchingCodecs[i];
         sp<VideoFrameDecoder> decoder = new VideoFrameDecoder(componentName, trackMeta, source);
-        if (decoder->init(timeUs, numFrames, option, colorFormat) == OK) {
-            if (outFrame != NULL) {
-                *outFrame = decoder->extractFrame();
-                if (*outFrame != NULL) {
-                    return OK;
+        if (decoder->init(timeUs, option, colorFormat) == OK) {
+            sp<IMemory> frame = decoder->extractFrame();
+            if (frame != nullptr) {
+                // keep the decoder if seeking by frame index
+                if (option == MediaSource::ReadOptions::SEEK_FRAME_INDEX) {
+                    mDecoder = decoder;
+                    mLastDecodedIndex = timeUs;
                 }
-            } else if (outFrames != NULL) {
-                status_t err = decoder->extractFrames(outFrames);
-                if (err == OK) {
-                    return OK;
-                }
+                return frame;
             }
         }
         ALOGV("%s failed to extract frame, trying next decoder.", componentName.c_str());
     }
 
     ALOGE("all codecs failed to extract frame.");
-    return UNKNOWN_ERROR;
+    return NULL;
 }
 
 MediaAlbumArt *StagefrightMetadataRetriever::extractAlbumArt() {
diff --git a/media/libstagefright/codecs/flac/enc/Android.bp b/media/libstagefright/codecs/flac/enc/Android.bp
index d7d871a..f35bce1 100644
--- a/media/libstagefright/codecs/flac/enc/Android.bp
+++ b/media/libstagefright/codecs/flac/enc/Android.bp
@@ -15,8 +15,10 @@
     },
 
     header_libs: ["libbase_headers"],
-    static_libs: [
+    shared_libs: [
         "libaudioutils",
+    ],
+    static_libs: [
         "libFLAC",
     ],
 }
diff --git a/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp b/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp
index da86758..87e8fd4 100644
--- a/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp
+++ b/media/libstagefright/codecs/xaacdec/SoftXAAC.cpp
@@ -1426,75 +1426,90 @@
     RETURN_IF_FATAL(err_code, "IA_CMD_TYPE_DO_EXECUTE");
 
     UWORD32 ui_exec_done;
+    WORD32 i_num_preroll = 0;
     /* Checking for end of processing */
     err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_EXECUTE, IA_CMD_TYPE_DONE_QUERY,
                                 &ui_exec_done);
     RETURN_IF_FATAL(err_code, "IA_CMD_TYPE_DONE_QUERY");
 
-#ifdef ENABLE_MPEG_D_DRC
+    err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
+                              IA_ENHAACPLUS_DEC_CONFIG_GET_NUM_PRE_ROLL_FRAMES,
+                              &i_num_preroll);
+
+    RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GET_NUM_PRE_ROLL_FRAMES");
     {
-        if (ui_exec_done != 1) {
-            VOID* p_array;        // ITTIAM:buffer to handle gain payload
-            WORD32 buf_size = 0;  // ITTIAM:gain payload length
-            WORD32 bit_str_fmt = 1;
-            WORD32 gain_stream_flag = 1;
+        int32_t pi_preroll_frame_offset = 0;
+        do {
+#ifdef ENABLE_MPEG_D_DRC
+            if (ui_exec_done != 1) {
+                VOID* p_array;        // ITTIAM:buffer to handle gain payload
+                WORD32 buf_size = 0;  // ITTIAM:gain payload length
+                WORD32 bit_str_fmt = 1;
+                WORD32 gain_stream_flag = 1;
 
-            err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
-                                        IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN, &buf_size);
-            RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN");
+                err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
+                                            IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN, &buf_size);
+                RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_LEN");
 
-            err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
-                                        IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF, &p_array);
-            RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF");
+                err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CONFIG_PARAM,
+                                            IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF, &p_array);
+                RETURN_IF_FATAL(err_code, "IA_ENHAACPLUS_DEC_CONFIG_GAIN_PAYLOAD_BUF");
 
-            if (buf_size > 0) {
-                /*Set bitstream_split_format */
-                err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
-                                          IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT, &bit_str_fmt);
-                RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+                if (buf_size > 0) {
+                    /*Set bitstream_split_format */
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
+                                              IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT, &bit_str_fmt);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
 
-                memcpy(mDrcInBuf, p_array, buf_size);
-                /* Set number of bytes to be processed */
-                err_code =
-                    ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES_BS, 0, &buf_size);
-                RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+                    memcpy(mDrcInBuf, p_array, buf_size);
+                    /* Set number of bytes to be processed */
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES_BS,
+                                              0, &buf_size);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
 
-                err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
-                                          IA_DRC_DEC_CONFIG_GAIN_STREAM_FLAG, &gain_stream_flag);
-                RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_CONFIG_PARAM,
+                                              IA_DRC_DEC_CONFIG_GAIN_STREAM_FLAG,
+                                              &gain_stream_flag);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
 
-                /* Execute process */
-                err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_INIT,
-                                          IA_CMD_TYPE_INIT_CPY_BSF_BUFF, NULL);
-                RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
+                    /* Execute process */
+                    err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_INIT,
+                                              IA_CMD_TYPE_INIT_CPY_BSF_BUFF, NULL);
+                    RETURN_IF_FATAL(err_code, "IA_DRC_DEC_CONFIG_PARAM_BITS_FORMAT");
 
-                mMpegDDRCPresent = 1;
+                    mMpegDDRCPresent = 1;
+                }
             }
-        }
-    }
 #endif
-    /* How much buffer is used in input buffers */
-    err_code =
-        ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CURIDX_INPUT_BUF, 0, bytesConsumed);
-    RETURN_IF_FATAL(err_code, "IA_API_CMD_GET_CURIDX_INPUT_BUF");
+            /* How much buffer is used in input buffers */
+            err_code =
+                ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_CURIDX_INPUT_BUF,
+                                 0, bytesConsumed);
+            RETURN_IF_FATAL(err_code, "IA_API_CMD_GET_CURIDX_INPUT_BUF");
 
-    /* Get the output bytes */
-    err_code = ixheaacd_dec_api(mXheaacCodecHandle, IA_API_CMD_GET_OUTPUT_BYTES, 0, outBytes);
-    RETURN_IF_FATAL(err_code, "IA_API_CMD_GET_OUTPUT_BYTES");
+            /* Get the output bytes */
+            err_code = ixheaacd_dec_api(mXheaacCodecHandle,
+                                        IA_API_CMD_GET_OUTPUT_BYTES, 0, outBytes);
+            RETURN_IF_FATAL(err_code, "IA_API_CMD_GET_OUTPUT_BYTES");
 #ifdef ENABLE_MPEG_D_DRC
 
-    if (mMpegDDRCPresent == 1) {
-        memcpy(mDrcInBuf, mOutputBuffer, *outBytes);
-        err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES, 0, outBytes);
-        RETURN_IF_FATAL(err_code, "IA_API_CMD_SET_INPUT_BYTES");
+            if (mMpegDDRCPresent == 1) {
+                memcpy(mDrcInBuf, mOutputBuffer + pi_preroll_frame_offset, *outBytes);
+                pi_preroll_frame_offset += *outBytes;
+                err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_SET_INPUT_BYTES,
+                                          0, outBytes);
+                RETURN_IF_FATAL(err_code, "IA_API_CMD_SET_INPUT_BYTES");
 
-        err_code =
-            ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_EXECUTE, IA_CMD_TYPE_DO_EXECUTE, NULL);
-        RETURN_IF_FATAL(err_code, "IA_CMD_TYPE_DO_EXECUTE");
+                err_code = ia_drc_dec_api(mMpegDDrcHandle, IA_API_CMD_EXECUTE,
+                                          IA_CMD_TYPE_DO_EXECUTE, NULL);
+                RETURN_IF_FATAL(err_code, "IA_CMD_TYPE_DO_EXECUTE");
 
-        memcpy(mOutputBuffer, mDrcOutBuf, *outBytes);
-    }
+                memcpy(mOutputBuffer, mDrcOutBuf, *outBytes);
+            }
 #endif
+            i_num_preroll--;
+        } while (i_num_preroll > 0);
+    }
     return IA_NO_ERROR;
 }
 
diff --git a/media/libstagefright/colorconversion/ColorConverter.cpp b/media/libstagefright/colorconversion/ColorConverter.cpp
index d685321..c7dc415 100644
--- a/media/libstagefright/colorconversion/ColorConverter.cpp
+++ b/media/libstagefright/colorconversion/ColorConverter.cpp
@@ -324,8 +324,8 @@
 }
 
 #define DECLARE_YUV2RGBFUNC(func, rgb) int (*func)(     \
-        const uint8*, int, const uint8*, int,           \
-        const uint8*, int, uint8*, int, int, int)       \
+        const uint8_t*, int, const uint8_t*, int,           \
+        const uint8_t*, int, uint8_t*, int, int, int)       \
         = mSrcColorSpace.isBt709() ? libyuv::H420To##rgb \
         : mSrcColorSpace.isJpeg() ? libyuv::J420To##rgb  \
         : libyuv::I420To##rgb
@@ -350,7 +350,7 @@
     {
         DECLARE_YUV2RGBFUNC(func, RGB565);
         (*func)(src_y, src.mStride, src_u, src.mStride / 2, src_v, src.mStride / 2,
-                (uint8 *)dst_ptr, dst.mStride, src.cropWidth(), src.cropHeight());
+                (uint8_t *)dst_ptr, dst.mStride, src.cropWidth(), src.cropHeight());
         break;
     }
 
@@ -358,7 +358,7 @@
     {
         DECLARE_YUV2RGBFUNC(func, ABGR);
         (*func)(src_y, src.mStride, src_u, src.mStride / 2, src_v, src.mStride / 2,
-                (uint8 *)dst_ptr, dst.mStride, src.cropWidth(), src.cropHeight());
+                (uint8_t *)dst_ptr, dst.mStride, src.cropWidth(), src.cropHeight());
         break;
     }
 
@@ -366,7 +366,7 @@
     {
         DECLARE_YUV2RGBFUNC(func, ARGB);
         (*func)(src_y, src.mStride, src_u, src.mStride / 2, src_v, src.mStride / 2,
-                (uint8 *)dst_ptr, dst.mStride, src.cropWidth(), src.cropHeight());
+                (uint8_t *)dst_ptr, dst.mStride, src.cropWidth(), src.cropHeight());
         break;
     }
 
@@ -391,17 +391,17 @@
 
     switch (mDstFormat) {
     case OMX_COLOR_Format16bitRGB565:
-        libyuv::NV12ToRGB565(src_y, src.mStride, src_u, src.mStride, (uint8 *)dst_ptr,
+        libyuv::NV12ToRGB565(src_y, src.mStride, src_u, src.mStride, (uint8_t *)dst_ptr,
                 dst.mStride, src.cropWidth(), src.cropHeight());
         break;
 
     case OMX_COLOR_Format32bitBGRA8888:
-        libyuv::NV12ToARGB(src_y, src.mStride, src_u, src.mStride, (uint8 *)dst_ptr,
+        libyuv::NV12ToARGB(src_y, src.mStride, src_u, src.mStride, (uint8_t *)dst_ptr,
                 dst.mStride, src.cropWidth(), src.cropHeight());
         break;
 
     case OMX_COLOR_Format32BitRGBA8888:
-        libyuv::NV12ToABGR(src_y, src.mStride, src_u, src.mStride, (uint8 *)dst_ptr,
+        libyuv::NV12ToABGR(src_y, src.mStride, src_u, src.mStride, (uint8_t *)dst_ptr,
                 dst.mStride, src.cropWidth(), src.cropHeight());
         break;
 
diff --git a/media/libstagefright/colorconversion/SoftwareRenderer.cpp b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
index cf91405..4711315 100644
--- a/media/libstagefright/colorconversion/SoftwareRenderer.cpp
+++ b/media/libstagefright/colorconversion/SoftwareRenderer.cpp
@@ -31,9 +31,14 @@
 
 namespace android {
 
-static int ALIGN(int x, int y) {
-    // y must be a power of 2.
-    return (x + y - 1) & ~(y - 1);
+inline void initDstYUV(
+        const android_ycbcr &ycbcr, int32_t cropTop, int32_t cropLeft,
+        uint8_t **dst_y, uint8_t **dst_u, uint8_t **dst_v) {
+    *dst_y = (uint8_t *)ycbcr.y + cropTop * ycbcr.ystride + cropLeft;
+
+    int32_t c_offset = (cropTop / 2) * ycbcr.cstride + cropLeft / 2;
+    *dst_v = (uint8_t *)ycbcr.cr + c_offset;
+    *dst_u = (uint8_t *)ycbcr.cb + c_offset;
 }
 
 SoftwareRenderer::SoftwareRenderer(
@@ -300,20 +305,14 @@
         const uint8_t *src_u = (const uint8_t *)data + mStride * mHeight + mCropTop * mStride / 4;
         const uint8_t *src_v = (const uint8_t *)src_u + mStride * mHeight / 4;
 
-        uint8_t *dst_y = (uint8_t *)ycbcr.y;
-        uint8_t *dst_v = (uint8_t *)ycbcr.cr;
-        uint8_t *dst_u = (uint8_t *)ycbcr.cb;
-        size_t dst_c_stride = ALIGN(buf->stride / 2, 16);
-
-        dst_y += mCropTop * buf->stride + mCropLeft;
-        dst_v += (mCropTop/2) * dst_c_stride + mCropLeft/2;
-        dst_u += (mCropTop/2) * dst_c_stride + mCropLeft/2;
+        uint8_t *dst_y, *dst_u, *dst_v;
+        initDstYUV(ycbcr, mCropTop, mCropLeft, &dst_y, &dst_u, &dst_v);
 
         for (int y = 0; y < mCropHeight; ++y) {
             memcpy(dst_y, src_y, mCropWidth);
 
             src_y += mStride;
-            dst_y += buf->stride;
+            dst_y += ycbcr.ystride;
         }
 
         for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
@@ -322,22 +321,16 @@
 
             src_u += mStride / 2;
             src_v += mStride / 2;
-            dst_u += dst_c_stride;
-            dst_v += dst_c_stride;
+            dst_u += ycbcr.cstride;
+            dst_v += ycbcr.cstride;
         }
     } else if (mColorFormat == OMX_COLOR_FormatYUV420Planar16) {
         const uint8_t *src_y = (const uint8_t *)data + mCropTop * mStride + mCropLeft * 2;
         const uint8_t *src_u = (const uint8_t *)data + mStride * mHeight + mCropTop * mStride / 4;
         const uint8_t *src_v = (const uint8_t *)src_u + mStride * mHeight / 4;
 
-        uint8_t *dst_y = (uint8_t *)ycbcr.y;
-        uint8_t *dst_v = (uint8_t *)ycbcr.cr;
-        uint8_t *dst_u = (uint8_t *)ycbcr.cb;
-        size_t dst_c_stride = ALIGN(buf->stride / 2, 16);
-
-        dst_y += mCropTop * buf->stride + mCropLeft;
-        dst_v += (mCropTop / 2) * dst_c_stride + mCropLeft / 2;
-        dst_u += (mCropTop / 2) * dst_c_stride + mCropLeft / 2;
+        uint8_t *dst_y, *dst_u, *dst_v;
+        initDstYUV(ycbcr, mCropTop, mCropLeft, &dst_y, &dst_u, &dst_v);
 
         for (int y = 0; y < mCropHeight; ++y) {
             for (int x = 0; x < mCropWidth; ++x) {
@@ -345,7 +338,7 @@
             }
 
             src_y += mStride;
-            dst_y += buf->stride;
+            dst_y += ycbcr.ystride;
         }
 
         for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
@@ -356,8 +349,8 @@
 
             src_u += mStride / 2;
             src_v += mStride / 2;
-            dst_u += dst_c_stride;
-            dst_v += dst_c_stride;
+            dst_u += ycbcr.cstride;
+            dst_v += ycbcr.cstride;
         }
     } else if (mColorFormat == OMX_TI_COLOR_FormatYUV420PackedSemiPlanar
             || mColorFormat == OMX_COLOR_FormatYUV420SemiPlanar) {
@@ -368,20 +361,14 @@
         src_y += mCropLeft + mCropTop * mWidth;
         src_uv += (mCropLeft + mCropTop * mWidth) / 2;
 
-        uint8_t *dst_y = (uint8_t *)ycbcr.y;
-        uint8_t *dst_v = (uint8_t *)ycbcr.cr;
-        uint8_t *dst_u = (uint8_t *)ycbcr.cb;
-        size_t dst_c_stride = ALIGN(buf->stride / 2, 16);
-
-        dst_y += mCropTop * buf->stride + mCropLeft;
-        dst_v += (mCropTop/2) * dst_c_stride + mCropLeft/2;
-        dst_u += (mCropTop/2) * dst_c_stride + mCropLeft/2;
+        uint8_t *dst_y, *dst_u, *dst_v;
+        initDstYUV(ycbcr, mCropTop, mCropLeft, &dst_y, &dst_u, &dst_v);
 
         for (int y = 0; y < mCropHeight; ++y) {
             memcpy(dst_y, src_y, mCropWidth);
 
             src_y += mWidth;
-            dst_y += buf->stride;
+            dst_y += ycbcr.ystride;
         }
 
         for (int y = 0; y < (mCropHeight + 1) / 2; ++y) {
@@ -392,8 +379,8 @@
             }
 
             src_uv += mWidth;
-            dst_u += dst_c_stride;
-            dst_v += dst_c_stride;
+            dst_u += ycbcr.cstride;
+            dst_v += ycbcr.cstride;
         }
     } else if (mColorFormat == OMX_COLOR_Format24bitRGB888) {
         uint8_t* srcPtr = (uint8_t*)data + mWidth * mCropTop * 3 + mCropLeft * 3;
diff --git a/media/libstagefright/exports.lds b/media/libstagefright/exports.lds
index aabc233..f5ddf1e 100644
--- a/media/libstagefright/exports.lds
+++ b/media/libstagefright/exports.lds
@@ -395,7 +395,6 @@
         ScaleFilterCols_NEON*;
         ScaleFilterReduce;
         ScaleFilterRows_NEON*;
-        ScaleOffset;
         ScalePlane;
         ScalePlane_16;
         ScalePlaneBilinearDown;
@@ -505,4 +504,8 @@
         YUY2ToYRow_Any_NEON*;
         YUY2ToYRow_C;
         YUY2ToYRow_NEON*;
+        ogg_packet_*;
+        ogg_page_*;
+        ogg_stream_*;
+        ogg_sync_*;
 };
diff --git a/media/libstagefright/flac/dec/Android.bp b/media/libstagefright/flac/dec/Android.bp
index b494e16..7ebe71f 100644
--- a/media/libstagefright/flac/dec/Android.bp
+++ b/media/libstagefright/flac/dec/Android.bp
@@ -1,4 +1,4 @@
-cc_library {
+cc_library_shared {
     name: "libstagefright_flacdec",
     vendor_available: true,
 
@@ -18,29 +18,20 @@
         cfi: true,
     },
 
-    static: {
-        whole_static_libs: [
-            "libFLAC",
-            "libaudioutils",
-        ],
-    },
-
-    shared: {
-        static_libs: [
-            "libFLAC",
-            "libaudioutils",
-        ],
-        export_static_lib_headers: [
-            "libFLAC",
-        ],
-    },
-
     shared_libs: [
+        "libaudioutils",
         "liblog",
     ],
 
+    static_libs: [
+        "libFLAC",
+    ],
+
+    export_static_lib_headers: [
+        "libFLAC",
+    ],
+
     header_libs: [
         "libmedia_headers",
-        "libFLAC-headers",
     ],
 }
diff --git a/media/libstagefright/foundation/avc_utils.cpp b/media/libstagefright/foundation/avc_utils.cpp
index e8a6083..f53d2c9 100644
--- a/media/libstagefright/foundation/avc_utils.cpp
+++ b/media/libstagefright/foundation/avc_utils.cpp
@@ -166,10 +166,21 @@
     unsigned pic_height_in_map_units_minus1 = parseUE(&br);
     unsigned frame_mbs_only_flag = br.getBits(1);
 
-    *width = pic_width_in_mbs_minus1 * 16 + 16;
+    //    *width = pic_width_in_mbs_minus1 * 16 + 16;
+    if (__builtin_mul_overflow(pic_width_in_mbs_minus1, 16, &pic_width_in_mbs_minus1) ||
+        __builtin_add_overflow(pic_width_in_mbs_minus1, 16, width)) {
+        *width = 0;
+    }
 
-    *height = (2 - frame_mbs_only_flag)
-        * (pic_height_in_map_units_minus1 * 16 + 16);
+    //    *height = (2 - frame_mbs_only_flag) * (pic_height_in_map_units_minus1 * 16 + 16);
+    if (__builtin_mul_overflow(
+                pic_height_in_map_units_minus1, 16, &pic_height_in_map_units_minus1) ||
+        __builtin_add_overflow(
+                pic_height_in_map_units_minus1, 16, &pic_height_in_map_units_minus1) ||
+        __builtin_mul_overflow(
+                pic_height_in_map_units_minus1, (2 - frame_mbs_only_flag), height)) {
+        *height = 0;
+    }
 
     if (!frame_mbs_only_flag) {
         br.getBits(1);  // mb_adaptive_frame_field_flag
@@ -202,17 +213,19 @@
 
 
         // *width -= (frame_crop_left_offset + frame_crop_right_offset) * cropUnitX;
-        if(__builtin_add_overflow(frame_crop_left_offset, frame_crop_right_offset, &frame_crop_left_offset) ||
-            __builtin_mul_overflow(frame_crop_left_offset, cropUnitX, &frame_crop_left_offset) ||
-            __builtin_sub_overflow(*width, frame_crop_left_offset, width) ||
+        if(__builtin_add_overflow(
+                   frame_crop_left_offset, frame_crop_right_offset, &frame_crop_left_offset) ||
+           __builtin_mul_overflow(frame_crop_left_offset, cropUnitX, &frame_crop_left_offset) ||
+           __builtin_sub_overflow(*width, frame_crop_left_offset, width) ||
             *width < 0) {
             *width = 0;
         }
 
         //*height -= (frame_crop_top_offset + frame_crop_bottom_offset) * cropUnitY;
-        if(__builtin_add_overflow(frame_crop_top_offset, frame_crop_bottom_offset, &frame_crop_top_offset) ||
-            __builtin_mul_overflow(frame_crop_top_offset, cropUnitY, &frame_crop_top_offset) ||
-            __builtin_sub_overflow(*height, frame_crop_top_offset, height) ||
+        if(__builtin_add_overflow(
+                   frame_crop_top_offset, frame_crop_bottom_offset, &frame_crop_top_offset) ||
+           __builtin_mul_overflow(frame_crop_top_offset, cropUnitY, &frame_crop_top_offset) ||
+           __builtin_sub_overflow(*height, frame_crop_top_offset, height) ||
             *height < 0) {
             *height = 0;
         }
diff --git a/media/libstagefright/httplive/PlaylistFetcher.cpp b/media/libstagefright/httplive/PlaylistFetcher.cpp
index 635ecfe..0950db0 100644
--- a/media/libstagefright/httplive/PlaylistFetcher.cpp
+++ b/media/libstagefright/httplive/PlaylistFetcher.cpp
@@ -2160,7 +2160,9 @@
             return ERROR_MALFORMED;
         }
 
-        CHECK_LE(offset + aac_frame_length, buffer->size());
+        if (aac_frame_length > buffer->size() - offset) {
+            return ERROR_MALFORMED;
+        }
 
         int64_t unitTimeUs = timeUs + numSamples * 1000000LL / sampleRate;
         offset += aac_frame_length;
diff --git a/media/libstagefright/include/FrameDecoder.h b/media/libstagefright/include/FrameDecoder.h
index dc58c15..8e42fcf 100644
--- a/media/libstagefright/include/FrameDecoder.h
+++ b/media/libstagefright/include/FrameDecoder.h
@@ -24,15 +24,17 @@
 #include <media/stagefright/foundation/ABase.h>
 #include <media/MediaSource.h>
 #include <media/openmax/OMX_Video.h>
-#include <system/graphics-base.h>
+#include <ui/GraphicTypes.h>
 
 namespace android {
 
 struct AMessage;
-class MediaCodecBuffer;
-class IMediaSource;
-class VideoFrame;
 struct MediaCodec;
+class IMediaSource;
+class MediaCodecBuffer;
+class Surface;
+class SurfaceControl;
+class VideoFrame;
 
 struct FrameRect {
     int32_t left, top, right, bottom;
@@ -44,13 +46,10 @@
             const sp<MetaData> &trackMeta,
             const sp<IMediaSource> &source);
 
-    status_t init(
-            int64_t frameTimeUs, size_t numFrames, int option, int colorFormat);
+    status_t init(int64_t frameTimeUs, int option, int colorFormat);
 
     sp<IMemory> extractFrame(FrameRect *rect = NULL);
 
-    status_t extractFrames(std::vector<sp<IMemory> >* frames);
-
     static sp<IMemory> getMetadataOnly(
             const sp<MetaData> &trackMeta, int colorFormat, bool thumbnail = false);
 
@@ -59,9 +58,9 @@
 
     virtual sp<AMessage> onGetFormatAndSeekOptions(
             int64_t frameTimeUs,
-            size_t numFrames,
             int seekMode,
-            MediaSource::ReadOptions *options) = 0;
+            MediaSource::ReadOptions *options,
+            sp<Surface> *window) = 0;
 
     virtual status_t onExtractRect(FrameRect *rect) = 0;
 
@@ -79,24 +78,24 @@
 
     sp<MetaData> trackMeta()     const      { return mTrackMeta; }
     OMX_COLOR_FORMATTYPE dstFormat() const  { return mDstFormat; }
+    ui::PixelFormat captureFormat() const   { return mCaptureFormat; }
     int32_t dstBpp()             const      { return mDstBpp; }
-
-    void addFrame(const sp<IMemory> &frame) {
-        mFrames.push_back(frame);
-    }
+    void setFrame(const sp<IMemory> &frameMem) { mFrameMemory = frameMem; }
 
 private:
     AString mComponentName;
     sp<MetaData> mTrackMeta;
     sp<IMediaSource> mSource;
     OMX_COLOR_FORMATTYPE mDstFormat;
+    ui::PixelFormat mCaptureFormat;
     int32_t mDstBpp;
-    std::vector<sp<IMemory> > mFrames;
+    sp<IMemory> mFrameMemory;
     MediaSource::ReadOptions mReadOptions;
     sp<MediaCodec> mDecoder;
     sp<AMessage> mOutputFormat;
     bool mHaveMoreInputs;
     bool mFirstSample;
+    sp<Surface> mSurface;
 
     status_t extractInternal();
 
@@ -112,9 +111,9 @@
 protected:
     virtual sp<AMessage> onGetFormatAndSeekOptions(
             int64_t frameTimeUs,
-            size_t numFrames,
             int seekMode,
-            MediaSource::ReadOptions *options) override;
+            MediaSource::ReadOptions *options,
+            sp<Surface> *window) override;
 
     virtual status_t onExtractRect(FrameRect *rect) override {
         // Rect extraction for sequences is not supported for now.
@@ -134,11 +133,17 @@
             bool *done) override;
 
 private:
+    sp<SurfaceControl> mSurfaceControl;
+    sp<SurfaceControl> mParent;
+    VideoFrame *mFrame;
     bool mIsAvcOrHevc;
     MediaSource::ReadOptions::SeekMode mSeekMode;
     int64_t mTargetTimeUs;
-    size_t mNumFrames;
-    size_t mNumFramesDecoded;
+    List<int64_t> mSampleDurations;
+    int64_t mDefaultSampleDurationUs;
+
+    sp<Surface> initSurfaceControl();
+    status_t captureSurfaceControl();
 };
 
 struct ImageDecoder : public FrameDecoder {
@@ -150,9 +155,9 @@
 protected:
     virtual sp<AMessage> onGetFormatAndSeekOptions(
             int64_t frameTimeUs,
-            size_t numFrames,
             int seekMode,
-            MediaSource::ReadOptions *options) override;
+            MediaSource::ReadOptions *options,
+            sp<Surface> *window) override;
 
     virtual status_t onExtractRect(FrameRect *rect) override;
 
diff --git a/media/libstagefright/include/StagefrightMetadataRetriever.h b/media/libstagefright/include/StagefrightMetadataRetriever.h
index c50677a..ee51290 100644
--- a/media/libstagefright/include/StagefrightMetadataRetriever.h
+++ b/media/libstagefright/include/StagefrightMetadataRetriever.h
@@ -26,7 +26,7 @@
 namespace android {
 
 class DataSource;
-struct ImageDecoder;
+struct FrameDecoder;
 struct FrameRect;
 
 struct StagefrightMetadataRetriever : public MediaMetadataRetrieverBase {
@@ -47,9 +47,8 @@
             int index, int colorFormat, bool metaOnly, bool thumbnail);
     virtual sp<IMemory> getImageRectAtIndex(
             int index, int colorFormat, int left, int top, int right, int bottom);
-    virtual status_t getFrameAtIndex(
-            std::vector<sp<IMemory> >* frames,
-            int frameIndex, int numFrames, int colorFormat, bool metaOnly);
+    virtual sp<IMemory> getFrameAtIndex(
+            int index, int colorFormat, bool metaOnly);
 
     virtual MediaAlbumArt *extractAlbumArt();
     virtual const char *extractMetadata(int keyCode);
@@ -62,17 +61,17 @@
     KeyedVector<int, String8> mMetaData;
     MediaAlbumArt *mAlbumArt;
 
-    sp<ImageDecoder> mImageDecoder;
-    int mLastImageIndex;
+    sp<FrameDecoder> mDecoder;
+    int mLastDecodedIndex;
     void parseMetaData();
     void parseColorAspects(const sp<MetaData>& meta);
     // Delete album art and clear metadata.
     void clearMetadata();
 
-    status_t getFrameInternal(
-            int64_t timeUs, int numFrames, int option, int colorFormat, bool metaOnly,
-            sp<IMemory>* outFrame, std::vector<sp<IMemory> >* outFrames);
-    virtual sp<IMemory> getImageInternal(
+    sp<IMemory> getFrameInternal(
+            int64_t timeUs, int option, int colorFormat, bool metaOnly);
+
+    sp<IMemory> getImageInternal(
             int index, int colorFormat, bool metaOnly, bool thumbnail, FrameRect* rect);
 
     StagefrightMetadataRetriever(const StagefrightMetadataRetriever &);
diff --git a/media/libstagefright/include/media/stagefright/DataSourceBase.h b/media/libstagefright/include/media/stagefright/DataSourceBase.h
index af5b83d..c607c91 100644
--- a/media/libstagefright/include/media/stagefright/DataSourceBase.h
+++ b/media/libstagefright/include/media/stagefright/DataSourceBase.h
@@ -18,6 +18,8 @@
 
 #define DATA_SOURCE_BASE_H_
 
+#include <media/stagefright/foundation/ByteUtils.h>
+#include <media/stagefright/MediaErrors.h>
 #include <sys/types.h>
 #include <utils/Errors.h>
 
@@ -45,20 +47,106 @@
     virtual ssize_t readAt(off64_t offset, void *data, size_t size) = 0;
 
     // Convenience methods:
-    bool getUInt16(off64_t offset, uint16_t *x);
-    bool getUInt24(off64_t offset, uint32_t *x); // 3 byte int, returned as a 32-bit int
-    bool getUInt32(off64_t offset, uint32_t *x);
-    bool getUInt64(off64_t offset, uint64_t *x);
+    bool getUInt16(off64_t offset, uint16_t *x) {
+        *x = 0;
+
+        uint8_t byte[2];
+        if (readAt(offset, byte, 2) != 2) {
+            return false;
+        }
+
+        *x = (byte[0] << 8) | byte[1];
+
+        return true;
+    }
+    // 3 byte int, returned as a 32-bit int
+    bool getUInt24(off64_t offset, uint32_t *x) {
+        *x = 0;
+
+        uint8_t byte[3];
+        if (readAt(offset, byte, 3) != 3) {
+            return false;
+        }
+
+        *x = (byte[0] << 16) | (byte[1] << 8) | byte[2];
+
+        return true;
+    }
+    bool getUInt32(off64_t offset, uint32_t *x) {
+        *x = 0;
+
+        uint32_t tmp;
+        if (readAt(offset, &tmp, 4) != 4) {
+            return false;
+        }
+
+        *x = ntohl(tmp);
+
+        return true;
+    }
+    bool getUInt64(off64_t offset, uint64_t *x) {
+        *x = 0;
+
+        uint64_t tmp;
+        if (readAt(offset, &tmp, 8) != 8) {
+            return false;
+        }
+
+        *x = ntoh64(tmp);
+
+        return true;
+    }
 
     // read either int<N> or int<2N> into a uint<2N>_t, size is the int size in bytes.
-    bool getUInt16Var(off64_t offset, uint16_t *x, size_t size);
-    bool getUInt32Var(off64_t offset, uint32_t *x, size_t size);
-    bool getUInt64Var(off64_t offset, uint64_t *x, size_t size);
+    bool getUInt16Var(off64_t offset, uint16_t *x, size_t size) {
+        if (size == 2) {
+            return getUInt16(offset, x);
+        }
+        if (size == 1) {
+            uint8_t tmp;
+            if (readAt(offset, &tmp, 1) == 1) {
+                *x = tmp;
+                return true;
+            }
+        }
+        return false;
+    }
+    bool getUInt32Var(off64_t offset, uint32_t *x, size_t size) {
+        if (size == 4) {
+            return getUInt32(offset, x);
+        }
+        if (size == 2) {
+            uint16_t tmp;
+            if (getUInt16(offset, &tmp)) {
+                *x = tmp;
+                return true;
+            }
+        }
+        return false;
+    }
+    bool getUInt64Var(off64_t offset, uint64_t *x, size_t size) {
+        if (size == 8) {
+            return getUInt64(offset, x);
+        }
+        if (size == 4) {
+            uint32_t tmp;
+            if (getUInt32(offset, &tmp)) {
+                *x = tmp;
+                return true;
+            }
+        }
+        return false;
+    }
 
     // May return ERROR_UNSUPPORTED.
-    virtual status_t getSize(off64_t *size);
+    virtual status_t getSize(off64_t *size) {
+        *size = 0;
+        return ERROR_UNSUPPORTED;
+    }
 
-    virtual bool getUri(char *uriString, size_t bufferSize);
+    virtual bool getUri(char * /*uriString*/, size_t /*bufferSize*/) {
+        return false;
+    }
 
     virtual uint32_t flags() {
         return 0;
diff --git a/media/libstagefright/include/media/stagefright/MediaCodec.h b/media/libstagefright/include/media/stagefright/MediaCodec.h
index cd30347..01d0325 100644
--- a/media/libstagefright/include/media/stagefright/MediaCodec.h
+++ b/media/libstagefright/include/media/stagefright/MediaCodec.h
@@ -25,7 +25,7 @@
 #include <media/hardware/CryptoAPI.h>
 #include <media/MediaCodecInfo.h>
 #include <media/MediaResource.h>
-#include <media/MediaAnalyticsItem.h>
+#include <media/MediaMetrics.h>
 #include <media/stagefright/foundation/AHandler.h>
 #include <media/stagefright/FrameRenderTracker.h>
 #include <utils/Vector.h>
@@ -189,7 +189,7 @@
 
     status_t getCodecInfo(sp<MediaCodecInfo> *codecInfo) const;
 
-    status_t getMetrics(MediaAnalyticsItem * &reply);
+    status_t getMetrics(mediametrics_handle_t &reply);
 
     status_t setParameters(const sp<AMessage> &params);
 
@@ -328,11 +328,11 @@
     sp<Surface> mSurface;
     SoftwareRenderer *mSoftRenderer;
 
-    MediaAnalyticsItem *mAnalyticsItem;
-    void initAnalyticsItem();
-    void updateAnalyticsItem();
-    void flushAnalyticsItem();
-    void updateEphemeralAnalytics(MediaAnalyticsItem *item);
+    mediametrics_handle_t mMetricsHandle;
+    void initMediametrics();
+    void updateMediametrics();
+    void flushMediametrics();
+    void updateEphemeralMediametrics(mediametrics_handle_t item);
 
     sp<AMessage> mOutputFormat;
     sp<AMessage> mInputFormat;
diff --git a/media/libstagefright/rtsp/ARTSPConnection.h b/media/libstagefright/rtsp/ARTSPConnection.h
index 8df2676..56b604d 100644
--- a/media/libstagefright/rtsp/ARTSPConnection.h
+++ b/media/libstagefright/rtsp/ARTSPConnection.h
@@ -46,6 +46,8 @@
             const char *url, AString *host, unsigned *port, AString *path,
             AString *user, AString *pass);
 
+    int getSocket() { return mSocket; }
+
 protected:
     virtual ~ARTSPConnection();
     virtual void onMessageReceived(const sp<AMessage> &msg);
diff --git a/media/libstagefright/rtsp/ASessionDescription.cpp b/media/libstagefright/rtsp/ASessionDescription.cpp
index 9263565..2b42040 100644
--- a/media/libstagefright/rtsp/ASessionDescription.cpp
+++ b/media/libstagefright/rtsp/ASessionDescription.cpp
@@ -141,6 +141,12 @@
                 AString key, value;
 
                 ssize_t equalPos = line.find("=");
+                /* The condition 'if (line.size() < 2 || line.c_str()[1] != '=')' a few lines above
+                 * ensures '=' is at position 1.  However for robustness we do the following check.
+                 */
+                if (equalPos < 0) {
+                    return false;
+                }
 
                 key = AString(line, 0, equalPos + 1);
                 value = AString(line, equalPos + 1, line.size() - equalPos - 1);
diff --git a/media/libstagefright/rtsp/MyHandler.h b/media/libstagefright/rtsp/MyHandler.h
index 48bc8ce..85ffba2 100644
--- a/media/libstagefright/rtsp/MyHandler.h
+++ b/media/libstagefright/rtsp/MyHandler.h
@@ -257,6 +257,10 @@
         msg->post();
     }
 
+    sp<ARTSPConnection> getARTSPConnection() {
+      return mConn;
+    }
+
     static void addRR(const sp<ABuffer> &buf) {
         uint8_t *ptr = buf->data() + buf->size();
         ptr[0] = 0x80 | 0;
diff --git a/media/libstagefright/timedtext/TextDescriptions2.cpp b/media/libstagefright/timedtext/TextDescriptions2.cpp
index f48eacc..fd42d3a 100644
--- a/media/libstagefright/timedtext/TextDescriptions2.cpp
+++ b/media/libstagefright/timedtext/TextDescriptions2.cpp
@@ -145,7 +145,7 @@
         tmpData += 8;
         size_t remaining = size - 8;
 
-        if (size < chunkSize) {
+        if (chunkSize <= 8 || size < chunkSize) {
             return OK;
         }
         switch(chunkType) {
diff --git a/media/mtp/MtpServer.cpp b/media/mtp/MtpServer.cpp
index ca8cb78..6adf563 100644
--- a/media/mtp/MtpServer.cpp
+++ b/media/mtp/MtpServer.cpp
@@ -44,6 +44,7 @@
 #include "MtpStringBuffer.h"
 
 namespace android {
+static const int SN_EVENT_LOG_ID = 0x534e4554;
 
 static const MtpOperationCode kSupportedOperationCodes[] = {
     MTP_OPERATION_GET_DEVICE_INFO,
@@ -961,9 +962,20 @@
     if (!parseDateTime(modified, modifiedTime))
         modifiedTime = 0;
 
+    if ((strcmp(name, ".") == 0) || (strcmp(name, "..") == 0) ||
+        (strcmp(name, "/") == 0) || (strcmp(basename(name), name) != 0)) {
+        char errMsg[80];
+
+        snprintf(errMsg, sizeof(errMsg), "Invalid name: %s", (const char *) name);
+        ALOGE("%s (b/130656917)", errMsg);
+        android_errorWriteWithInfoLog(SN_EVENT_LOG_ID, "130656917", -1, errMsg,
+                                      strlen(errMsg));
+
+        return MTP_RESPONSE_INVALID_PARAMETER;
+    }
     if (path[path.size() - 1] != '/')
         path.append("/");
-    path.append(name);
+    path.append(basename(name));
 
     // check space first
     if (mSendObjectFileSize > storage->getFreeSpace())
diff --git a/media/mtp/MtpServer.h b/media/mtp/MtpServer.h
index 1f8799f..8cc9a9a 100644
--- a/media/mtp/MtpServer.h
+++ b/media/mtp/MtpServer.h
@@ -34,8 +34,11 @@
 
 class IMtpDatabase;
 class MtpStorage;
+class MtpMockServer;
 
 class MtpServer {
+    // libFuzzer testing
+    friend class MtpMockServer;
 
 private:
     IMtpDatabase*       mDatabase;
diff --git a/media/mtp/MtpStringBuffer.cpp b/media/mtp/MtpStringBuffer.cpp
index cd379bf..d8d425b 100644
--- a/media/mtp/MtpStringBuffer.cpp
+++ b/media/mtp/MtpStringBuffer.cpp
@@ -26,14 +26,31 @@
 
 namespace {
 
-std::wstring_convert<std::codecvt_utf8_utf16<char16_t>,char16_t> gConvert;
+const char * utf16_cerror = "__CONVERSION_ERROR__";
+const char16_t * utf8_cerror = u"__CONVERSION_ERROR__";
+
+std::wstring_convert<std::codecvt_utf8_utf16<char16_t>,char16_t> gConvert(utf16_cerror, utf8_cerror);
 
 static std::string utf16ToUtf8(std::u16string input_str) {
-    return gConvert.to_bytes(input_str);
+    std::string conversion = gConvert.to_bytes(input_str);
+
+    if (conversion == utf16_cerror) {
+        ALOGE("Unable to convert UTF-16 string to UTF-8");
+        return "";
+    } else {
+        return conversion;
+    }
 }
 
 static std::u16string utf8ToUtf16(std::string input_str) {
-    return gConvert.from_bytes(input_str);
+    std::u16string conversion = gConvert.from_bytes(input_str);
+
+    if (conversion == utf8_cerror) {
+        ALOGE("Unable to convert UTF-8 string to UTF-16");
+        return u"";
+    } else {
+        return conversion;
+    }
 }
 
 } // namespace
diff --git a/media/mtp/MtpUtils.cpp b/media/mtp/MtpUtils.cpp
index 8564576..84a20d3 100644
--- a/media/mtp/MtpUtils.cpp
+++ b/media/mtp/MtpUtils.cpp
@@ -150,6 +150,7 @@
             ret += copyFile(oldFile.c_str(), newFile.c_str());
         }
     }
+    closedir(dir);
     return ret;
 }
 
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 0b745ac..aef0ade 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -1132,16 +1132,16 @@
     return mute;
 }
 
-void AudioFlinger::setRecordSilenced(uid_t uid, bool silenced)
+void AudioFlinger::setRecordSilenced(audio_port_handle_t portId, bool silenced)
 {
-    ALOGV("AudioFlinger::setRecordSilenced(uid:%d, silenced:%d)", uid, silenced);
+    ALOGV("AudioFlinger::setRecordSilenced(portId:%d, silenced:%d)", portId, silenced);
 
     AutoMutex lock(mLock);
     for (size_t i = 0; i < mRecordThreads.size(); i++) {
-        mRecordThreads[i]->setRecordSilenced(uid, silenced);
+        mRecordThreads[i]->setRecordSilenced(portId, silenced);
     }
     for (size_t i = 0; i < mMmapThreads.size(); i++) {
-        mMmapThreads[i]->setRecordSilenced(uid, silenced);
+        mMmapThreads[i]->setRecordSilenced(portId, silenced);
     }
 }
 
@@ -1357,8 +1357,8 @@
         String8(AudioParameter::keyFrameCount),
         String8(AudioParameter::keyInputSource),
         String8(AudioParameter::keyMonoOutput),
-        String8(AudioParameter::keyStreamConnect),
-        String8(AudioParameter::keyStreamDisconnect),
+        String8(AudioParameter::keyDeviceConnect),
+        String8(AudioParameter::keyDeviceDisconnect),
         String8(AudioParameter::keyStreamSupportedFormats),
         String8(AudioParameter::keyStreamSupportedChannels),
         String8(AudioParameter::keyStreamSupportedSamplingRates),
@@ -1933,7 +1933,8 @@
                                                   &output.notificationFrameCount,
                                                   callingPid, clientUid, &output.flags,
                                                   input.clientInfo.clientTid,
-                                                  &lStatus, portId);
+                                                  &lStatus, portId,
+                                                  input.opPackageName);
         LOG_ALWAYS_FATAL_IF((lStatus == NO_ERROR) && (recordTrack == 0));
 
         // lStatus == BAD_TYPE means FAST flag was rejected: request a new input from
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index 72e669a..5e4509f 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -162,7 +162,7 @@
     virtual     status_t    setMicMute(bool state);
     virtual     bool        getMicMute() const;
 
-    virtual     void        setRecordSilenced(uid_t uid, bool silenced);
+    virtual     void        setRecordSilenced(audio_port_handle_t portId, bool silenced);
 
     virtual     status_t    setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs);
     virtual     String8     getParameters(audio_io_handle_t ioHandle, const String8& keys) const;
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index c5b9953..3eacc8c 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -105,13 +105,8 @@
     return mSQ.poll();
 }
 
-void FastMixer::setNBLogWriter(NBLog::Writer *logWriter)
+void FastMixer::setNBLogWriter(NBLog::Writer *logWriter __unused)
 {
-    // FIXME If mMixer is set or changed prior to this, we don't inform correctly.
-    //       Should cache logWriter and re-apply it at the assignment to mMixer.
-    if (mMixer != NULL) {
-        mMixer->setNBLogWriter(logWriter);
-    }
 }
 
 void FastMixer::onIdle()
diff --git a/services/audioflinger/FastThread.cpp b/services/audioflinger/FastThread.cpp
index 04b32c2..8b7a124 100644
--- a/services/audioflinger/FastThread.cpp
+++ b/services/audioflinger/FastThread.cpp
@@ -124,7 +124,7 @@
             mDumpState = next->mDumpState != NULL ? next->mDumpState : mDummyDumpState;
             tlNBLogWriter = next->mNBLogWriter != NULL ?
                     next->mNBLogWriter : mDummyNBLogWriter.get();
-            setNBLogWriter(tlNBLogWriter); // FastMixer informs its AudioMixer, FastCapture ignores
+            setNBLogWriter(tlNBLogWriter); // This is used for debugging only
 
             // We want to always have a valid reference to the previous (non-idle) state.
             // However, the state queue only guarantees access to current and previous states.
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index 08660dd..c8397cd 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -19,6 +19,39 @@
     #error This header file should only be included from AudioFlinger.h
 #endif
 
+// Checks and monitors OP_RECORD_AUDIO
+class OpRecordAudioMonitor : public RefBase {
+public:
+    ~OpRecordAudioMonitor() override;
+    bool hasOpRecordAudio() const;
+
+    static sp<OpRecordAudioMonitor> createIfNeeded(uid_t uid, const String16& opPackageName);
+
+private:
+    OpRecordAudioMonitor(uid_t uid, const String16& opPackageName);
+    void onFirstRef() override;
+
+    AppOpsManager mAppOpsManager;
+
+    class RecordAudioOpCallback : public BnAppOpsCallback {
+    public:
+        explicit RecordAudioOpCallback(const wp<OpRecordAudioMonitor>& monitor);
+        void opChanged(int32_t op, const String16& packageName) override;
+
+    private:
+        const wp<OpRecordAudioMonitor> mMonitor;
+    };
+
+    sp<RecordAudioOpCallback> mOpCallback;
+    // called by RecordAudioOpCallback when OP_RECORD_AUDIO is updated in AppOp callback
+    // and in onFirstRef()
+    void checkRecordAudio();
+
+    std::atomic_bool mHasOpRecordAudio;
+    const uid_t mUid;
+    const String16 mPackage;
+};
+
 // record track
 class RecordTrack : public TrackBase {
 public:
@@ -36,6 +69,7 @@
                                 uid_t uid,
                                 audio_input_flags_t flags,
                                 track_type type,
+                                const String16& opPackageName,
                                 audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
     virtual             ~RecordTrack();
     virtual status_t    initCheck() const;
@@ -68,7 +102,7 @@
                                 { return (mFlags & AUDIO_INPUT_FLAG_DIRECT) != 0; }
 
             void        setSilenced(bool silenced) { if (!isPatchTrack()) mSilenced = silenced; }
-            bool        isSilenced() const { return mSilenced; }
+            bool        isSilenced() const;
 
             status_t    getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
 
@@ -111,6 +145,11 @@
             audio_input_flags_t                mFlags;
 
             bool                               mSilenced;
+
+            // used to enforce OP_RECORD_AUDIO
+            uid_t                              mUid;
+            String16                           mOpPackageName;
+            sp<OpRecordAudioMonitor>           mOpRecordAudioMonitor;
 };
 
 // playback track, used by PatchPanel
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 3f4a24e..6ca50a7 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -2609,7 +2609,7 @@
     LOG_ALWAYS_FATAL_IF(result != OK,
             "Error when retrieving output stream buffer size: %d", result);
     mFrameCount = mBufferSize / mFrameSize;
-    if (mFrameCount & 15) {
+    if ((mType == MIXER || mType == DUPLICATING) && (mFrameCount & 15)) {
         ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
                 mFrameCount);
     }
@@ -2953,9 +2953,11 @@
             ALOG_ASSERT(mCallbackThread != 0);
             mCallbackThread->setWriteBlocked(mWriteAckSequence);
         }
+        ATRACE_BEGIN("write");
         // FIXME We should have an implementation of timestamps for direct output threads.
         // They are used e.g for multichannel PCM playback over HDMI.
         bytesWritten = mOutput->write((char *)mSinkBuffer + offset, mBytesRemaining);
+        ATRACE_END();
 
         if (mUseAsyncWrite &&
                 ((bytesWritten < 0) || (bytesWritten == (ssize_t)mBytesRemaining))) {
@@ -5301,11 +5303,11 @@
         return false;
     }
     // Check validity as we don't call AudioMixer::create() here.
-    if (!AudioMixer::isValidFormat(format)) {
+    if (!mAudioMixer->isValidFormat(format)) {
         ALOGW("%s: invalid format: %#x", __func__, format);
         return false;
     }
-    if (!AudioMixer::isValidChannelMask(channelMask)) {
+    if (!mAudioMixer->isValidChannelMask(channelMask)) {
         ALOGW("%s: invalid channelMask: %#x", __func__, channelMask);
         return false;
     }
@@ -5658,10 +5660,17 @@
             minFrames = 1;
         }
 
-        if ((track->framesReady() >= minFrames) && track->isReady() && !track->isPaused() &&
+        const size_t framesReady = track->framesReady();
+        const int trackId = track->id();
+        if (ATRACE_ENABLED()) {
+            std::string traceName("nRdy");
+            traceName += std::to_string(trackId);
+            ATRACE_INT(traceName.c_str(), framesReady);
+        }
+        if ((framesReady >= minFrames) && track->isReady() && !track->isPaused() &&
                 !track->isStopping_2() && !track->isStopped())
         {
-            ALOGVV("track(%d) s=%08x [OK]", track->id(), cblk->mServer);
+            ALOGVV("track(%d) s=%08x [OK]", trackId, cblk->mServer);
 
             if (track->mFillingUpStatus == Track::FS_FILLED) {
                 track->mFillingUpStatus = Track::FS_ACTIVE;
@@ -5738,7 +5747,7 @@
                 // fill a buffer, then remove it from active list.
                 // Only consider last track started for mixer state control
                 if (--(track->mRetryCount) <= 0) {
-                    ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", track->id());
+                    ALOGV("BUFFER TIMEOUT: remove track(%d) from active list", trackId);
                     tracksToRemove->add(track);
                     // indicate to client process that the track was disabled because of underrun;
                     // it will then automatically call start() when data is available
@@ -5746,7 +5755,7 @@
                 } else if (last) {
                     ALOGW("pause because of UNDERRUN, framesReady = %zu,"
                             "minFrames = %u, mFormat = %#x",
-                            track->framesReady(), minFrames, mFormat);
+                            framesReady, minFrames, mFormat);
                     mixerStatus = MIXER_TRACKS_ENABLED;
                     if (mHwSupportsPause && !mHwPaused && !mStandby) {
                         doHwPause = true;
@@ -7293,7 +7302,7 @@
                         // Sanitize before releasing if the track has no access to the source data
                         // An idle UID receives silence from non virtual devices until active
                         if (activeTrack->isSilenced()) {
-                            memset(activeTrack->mSink.raw, 0, framesOut * mFrameSize);
+                            memset(activeTrack->mSink.raw, 0, framesOut * activeTrack->frameSize());
                         }
                         activeTrack->releaseBuffer(&activeTrack->mSink);
                     }
@@ -7454,7 +7463,8 @@
         audio_input_flags_t *flags,
         pid_t tid,
         status_t *status,
-        audio_port_handle_t portId)
+        audio_port_handle_t portId,
+        const String16& opPackageName)
 {
     size_t frameCount = *pFrameCount;
     size_t notificationFrameCount = *pNotificationFrameCount;
@@ -7588,7 +7598,7 @@
         track = new RecordTrack(this, client, attr, sampleRate,
                       format, channelMask, frameCount,
                       nullptr /* buffer */, (size_t)0 /* bufferSize */, sessionId, creatorPid, uid,
-                      *flags, TrackBase::TYPE_DEFAULT, portId);
+                      *flags, TrackBase::TYPE_DEFAULT, opPackageName, portId);
 
         lStatus = track->initCheck();
         if (lStatus != NO_ERROR) {
@@ -7924,12 +7934,12 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::RecordThread::setRecordSilenced(uid_t uid, bool silenced)
+void AudioFlinger::RecordThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
 {
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mTracks.size() ; i++) {
         sp<RecordTrack> track = mTracks[i];
-        if (track != 0 && track->uid() == uid) {
+        if (track != 0 && track->portId() == portId) {
             track->setSilenced(silenced);
         }
     }
@@ -9467,11 +9477,11 @@
     mInput->stream->updateSinkMetadata(metadata);
 }
 
-void AudioFlinger::MmapCaptureThread::setRecordSilenced(uid_t uid, bool silenced)
+void AudioFlinger::MmapCaptureThread::setRecordSilenced(audio_port_handle_t portId, bool silenced)
 {
     Mutex::Autolock _l(mLock);
     for (size_t i = 0; i < mActiveTracks.size() ; i++) {
-        if (mActiveTracks[i]->uid() == uid) {
+        if (mActiveTracks[i]->portId() == portId) {
             mActiveTracks[i]->setSilenced_l(silenced);
             broadcast_l();
         }
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index fc8aa13..6a9c0e7 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -1553,7 +1553,8 @@
                     audio_input_flags_t *flags,
                     pid_t tid,
                     status_t *status /*non-NULL*/,
-                    audio_port_handle_t portId);
+                    audio_port_handle_t portId,
+                    const String16& opPackageName);
 
             status_t    start(RecordTrack* recordTrack,
                               AudioSystem::sync_event_t event,
@@ -1615,7 +1616,7 @@
             void        checkBtNrec();
 
             // Sets the UID records silence
-            void        setRecordSilenced(uid_t uid, bool silenced);
+            void        setRecordSilenced(audio_port_handle_t portId, bool silenced);
 
             status_t    getActiveMicrophones(std::vector<media::MicrophoneInfo>* activeMicrophones);
 
@@ -1784,7 +1785,8 @@
     virtual     void        invalidateTracks(audio_stream_type_t streamType __unused) {}
 
                 // Sets the UID records silence
-    virtual     void        setRecordSilenced(uid_t uid __unused, bool silenced __unused) {}
+    virtual     void        setRecordSilenced(audio_port_handle_t portId __unused,
+                                              bool silenced __unused) {}
 
  protected:
                 void        dumpInternals_l(int fd, const Vector<String16>& args) override;
@@ -1871,7 +1873,8 @@
 
                 void           updateMetadata_l() override;
                 void           processVolume_l() override;
-                void           setRecordSilenced(uid_t uid, bool silenced) override;
+                void           setRecordSilenced(audio_port_handle_t portId,
+                                                 bool silenced) override;
 
     virtual     void           toAudioPortConfig(struct audio_port_config *config);
 
diff --git a/services/audioflinger/TrackBase.h b/services/audioflinger/TrackBase.h
index 8f720b5..7cf34c1 100644
--- a/services/audioflinger/TrackBase.h
+++ b/services/audioflinger/TrackBase.h
@@ -205,6 +205,16 @@
 protected:
     DISALLOW_COPY_AND_ASSIGN(TrackBase);
 
+    void releaseCblk() {
+        if (mCblk != nullptr) {
+            mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
+            if (mClient == 0) {
+                free(mCblk);
+            }
+            mCblk = nullptr;
+        }
+    }
+
     // AudioBufferProvider interface
     virtual status_t getNextBuffer(AudioBufferProvider::Buffer* buffer) = 0;
     virtual void releaseBuffer(AudioBufferProvider::Buffer* buffer);
@@ -215,6 +225,8 @@
 
     uint32_t channelCount() const { return mChannelCount; }
 
+    size_t frameSize() const { return mFrameSize; }
+
     audio_channel_mask_t channelMask() const { return mChannelMask; }
 
     virtual uint32_t sampleRate() const { return mSampleRate; }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 78db80c..fa35e5d 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -18,12 +18,14 @@
 
 #define LOG_TAG "AudioFlinger"
 //#define LOG_NDEBUG 0
+#define ATRACE_TAG ATRACE_TAG_AUDIO
 
 #include "Configuration.h"
 #include <linux/futex.h>
 #include <math.h>
 #include <sys/syscall.h>
 #include <utils/Log.h>
+#include <utils/Trace.h>
 
 #include <private/media/AudioTrackShared.h>
 
@@ -237,12 +239,7 @@
 {
     // delete the proxy before deleting the shared memory it refers to, to avoid dangling reference
     mServerProxy.clear();
-    if (mCblk != NULL) {
-        mCblk->~audio_track_cblk_t();   // destroy our shared-structure.
-        if (mClient == 0) {
-            free(mCblk);
-        }
-    }
+    releaseCblk();
     mCblkMemory.clear();    // free the shared memory before releasing the heap it belongs to
     if (mClient != 0) {
         // Client destructor must run with AudioFlinger client mutex locked
@@ -442,7 +439,7 @@
     return mHasOpPlayAudio.load();
 }
 
-// Note this method is never called (and never to be) for audio server / root track
+// Note this method is never called (and never to be) for audio server / patch record track
 // - not called from constructor due to check on UID,
 // - not called from PlayAudioOpCallback because the callback is not installed in this case
 void AudioFlinger::PlaybackThread::OpPlayAudioMonitor::checkPlayAudioForUsage()
@@ -549,6 +546,12 @@
         return;
     }
 
+    if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
+        ALOGE("%s(%d): no more tracks available", __func__, mId);
+        releaseCblk(); // this makes the track invalid.
+        return;
+    }
+
     if (sharedBuffer == 0) {
         mAudioTrackServerProxy = new AudioTrackServerProxy(mCblk, mBuffer, frameCount,
                 mFrameSize, !isExternalTrack(), sampleRate);
@@ -558,10 +561,6 @@
     }
     mServerProxy = mAudioTrackServerProxy;
 
-    if (!thread->isTrackAllowed_l(channelMask, format, sessionId, uid)) {
-        ALOGE("%s(%d): no more tracks available", __func__, mId);
-        return;
-    }
     // only allocate a fast track index if we were able to allocate a normal track name
     if (flags & AUDIO_OUTPUT_FLAG_FAST) {
         // FIXME: Not calling framesReadyIsCalledByMultipleThreads() exposes a potential
@@ -1828,9 +1827,19 @@
     ALOG_ASSERT(mPeerProxy != 0, "%s(%d): called without peer proxy", __func__, mId);
     Proxy::Buffer buf;
     buf.mFrameCount = buffer->frameCount;
+    if (ATRACE_ENABLED()) {
+        std::string traceName("PTnReq");
+        traceName += std::to_string(id());
+        ATRACE_INT(traceName.c_str(), buf.mFrameCount);
+    }
     status_t status = mPeerProxy->obtainBuffer(&buf, &mPeerTimeout);
     ALOGV_IF(status != NO_ERROR, "%s(%d): getNextBuffer status %d", __func__, mId, status);
     buffer->frameCount = buf.mFrameCount;
+    if (ATRACE_ENABLED()) {
+        std::string traceName("PTnObt");
+        traceName += std::to_string(id());
+        ATRACE_INT(traceName.c_str(), buf.mFrameCount);
+    }
     if (buf.mFrameCount == 0) {
         return WOULD_BLOCK;
     }
@@ -1883,6 +1892,105 @@
 // ----------------------------------------------------------------------------
 //      Record
 // ----------------------------------------------------------------------------
+
+
+// ----------------------------------------------------------------------------
+//      AppOp for audio recording
+// -------------------------------
+
+#undef LOG_TAG
+#define LOG_TAG "AF::OpRecordAudioMonitor"
+
+// static
+sp<AudioFlinger::RecordThread::OpRecordAudioMonitor>
+AudioFlinger::RecordThread::OpRecordAudioMonitor::createIfNeeded(
+            uid_t uid, const String16& opPackageName)
+{
+    if (isServiceUid(uid)) {
+        ALOGV("not silencing record for service uid:%d pack:%s",
+                uid, String8(opPackageName).string());
+        return nullptr;
+    }
+
+    if (opPackageName.size() == 0) {
+        Vector<String16> packages;
+        // no package name, happens with SL ES clients
+        // query package manager to find one
+        PermissionController permissionController;
+        permissionController.getPackagesForUid(uid, packages);
+        if (packages.isEmpty()) {
+            return nullptr;
+        } else {
+            ALOGV("using pack:%s for uid:%d", String8(packages[0]).string(), uid);
+            return new OpRecordAudioMonitor(uid, packages[0]);
+        }
+    }
+
+    return new OpRecordAudioMonitor(uid, opPackageName);
+}
+
+AudioFlinger::RecordThread::OpRecordAudioMonitor::OpRecordAudioMonitor(
+        uid_t uid, const String16& opPackageName)
+        : mHasOpRecordAudio(true), mUid(uid), mPackage(opPackageName)
+{
+}
+
+AudioFlinger::RecordThread::OpRecordAudioMonitor::~OpRecordAudioMonitor()
+{
+    if (mOpCallback != 0) {
+        mAppOpsManager.stopWatchingMode(mOpCallback);
+    }
+    mOpCallback.clear();
+}
+
+void AudioFlinger::RecordThread::OpRecordAudioMonitor::onFirstRef()
+{
+    checkRecordAudio();
+    mOpCallback = new RecordAudioOpCallback(this);
+    ALOGV("start watching OP_RECORD_AUDIO for pack:%s", String8(mPackage).string());
+    mAppOpsManager.startWatchingMode(AppOpsManager::OP_RECORD_AUDIO, mPackage, mOpCallback);
+}
+
+bool AudioFlinger::RecordThread::OpRecordAudioMonitor::hasOpRecordAudio() const {
+    return mHasOpRecordAudio.load();
+}
+
+// Called by RecordAudioOpCallback when OP_RECORD_AUDIO is updated in AppOp callback
+// and in onFirstRef()
+// Note this method is never called (and never to be) for audio server / root track
+// due to the UID in createIfNeeded(). As a result for those record track, it's:
+// - not called from constructor,
+// - not called from RecordAudioOpCallback because the callback is not installed in this case
+void AudioFlinger::RecordThread::OpRecordAudioMonitor::checkRecordAudio()
+{
+    const int32_t mode = mAppOpsManager.checkOp(AppOpsManager::OP_RECORD_AUDIO,
+            mUid, mPackage);
+    const bool hasIt =  (mode == AppOpsManager::MODE_ALLOWED);
+    // verbose logging only log when appOp changed
+    ALOGI_IF(hasIt != mHasOpRecordAudio.load(),
+            "OP_RECORD_AUDIO missing, %ssilencing record uid%d pack:%s",
+            hasIt ? "un" : "", mUid, String8(mPackage).string());
+    mHasOpRecordAudio.store(hasIt);
+}
+
+AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::RecordAudioOpCallback(
+        const wp<OpRecordAudioMonitor>& monitor) : mMonitor(monitor)
+{ }
+
+void AudioFlinger::RecordThread::OpRecordAudioMonitor::RecordAudioOpCallback::opChanged(int32_t op,
+            const String16& packageName) {
+    UNUSED(packageName);
+    if (op != AppOpsManager::OP_RECORD_AUDIO) {
+        return;
+    }
+    sp<OpRecordAudioMonitor> monitor = mMonitor.promote();
+    if (monitor != NULL) {
+        monitor->checkRecordAudio();
+    }
+}
+
+
+
 #undef LOG_TAG
 #define LOG_TAG "AF::RecordHandle"
 
@@ -1954,6 +2062,7 @@
             uid_t uid,
             audio_input_flags_t flags,
             track_type type,
+            const String16& opPackageName,
             audio_port_handle_t portId)
     :   TrackBase(thread, client, attr, sampleRate, format,
                   channelMask, frameCount, buffer, bufferSize, sessionId,
@@ -1967,7 +2076,8 @@
         mResamplerBufferProvider(NULL), // initialize in case of early constructor exit
         mRecordBufferConverter(NULL),
         mFlags(flags),
-        mSilenced(false)
+        mSilenced(false),
+        mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(uid, opPackageName))
 {
     if (mCblk == NULL) {
         return;
@@ -2218,6 +2328,14 @@
     mServerLatencyMs.store(latencyMs);
 }
 
+bool AudioFlinger::RecordThread::RecordTrack::isSilenced() const {
+    if (mSilenced) {
+        return true;
+    }
+    // The monitor is only created for record tracks that can be silenced.
+    return mOpRecordAudioMonitor ? !mOpRecordAudioMonitor->hasOpRecordAudio() : false;
+}
+
 status_t AudioFlinger::RecordThread::RecordTrack::getActiveMicrophones(
         std::vector<media::MicrophoneInfo>* activeMicrophones)
 {
@@ -2268,7 +2386,7 @@
                 audio_attributes_t{} /* currently unused for patch track */,
                 sampleRate, format, channelMask, frameCount,
                 buffer, bufferSize, AUDIO_SESSION_NONE, getpid(), AID_AUDIOSERVER,
-                flags, TYPE_PATCH),
+                flags, TYPE_PATCH, String16()),
         PatchTrackBase(new ClientProxy(mCblk, mBuffer, frameCount, mFrameSize, false, true),
                        *recordThread, timeout)
 {
@@ -2294,6 +2412,11 @@
     ALOGV_IF(status != NO_ERROR,
              "%s(%d): mPeerProxy->obtainBuffer status %d", __func__, mId, status);
     buffer->frameCount = buf.mFrameCount;
+    if (ATRACE_ENABLED()) {
+        std::string traceName("PRnObt");
+        traceName += std::to_string(id());
+        ATRACE_INT(traceName.c_str(), buf.mFrameCount);
+    }
     if (buf.mFrameCount == 0) {
         return WOULD_BLOCK;
     }
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 30f29d6..35126ad 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -258,7 +258,7 @@
     virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
                 std::vector<audio_format_t> *formats) = 0;
 
-    virtual void     setAppState(uid_t uid, app_state_t state) = 0;
+    virtual void     setAppState(audio_port_handle_t portId, app_state_t state) = 0;
 
     virtual status_t listAudioProductStrategies(AudioProductStrategyVector &strategies) = 0;
 
diff --git a/services/audiopolicy/audio_policy.conf b/services/audiopolicy/audio_policy.conf
deleted file mode 100644
index 9b83fef..0000000
--- a/services/audiopolicy/audio_policy.conf
+++ /dev/null
@@ -1,145 +0,0 @@
-#
-# Template audio policy configuration file
-#
-
-# Global configuration section:
-# - before audio HAL version 3.0:
-#   lists input and output devices always present on the device
-#   as well as the output device selected by default.
-#   Devices are designated by a string that corresponds to the enum in audio.h
-#
-#  global_configuration {
-#    attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
-#    default_output_device AUDIO_DEVICE_OUT_SPEAKER
-#    attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC|AUDIO_DEVICE_IN_REMOTE_SUBMIX
-#  }
-#
-# - after and including audio HAL 3.0 the global_configuration section is included in each
-#   hardware module section.
-#   it also includes the audio HAL version of this hw module:
-#  global_configuration {
-#    ...
-#     audio_hal_version <major.minor>  # audio HAL version in e.g. 3.0
-#  }
-#   other attributes (attached devices, default device) have to be included in the
-#   global_configuration section of each hardware module
-
-
-# audio hardware module section: contains descriptors for all audio hw modules present on the
-# device. Each hw module node is named after the corresponding hw module library base name.
-# For instance, "primary" corresponds to audio.primary.<device>.so.
-# The "primary" module is mandatory and must include at least one output with
-# AUDIO_OUTPUT_FLAG_PRIMARY flag.
-# Each module descriptor contains one or more output profile descriptors and zero or more
-# input profile descriptors. Each profile lists all the parameters supported by a given output
-# or input stream category.
-# The "channel_masks", "formats", "devices" and "flags" are specified using strings corresponding
-# to enums in audio.h and audio_policy.h. They are concatenated by use of "|" without space or "\n".
-#
-# For audio HAL version posterior to 3.0 the following sections or sub sections can be present in
-# a hw module section:
-# - A "global_configuration" section: see above
-# - Optionally a "devices" section:
-#   This section contains descriptors for audio devices with attributes like an address or a
-#   gain controller. The syntax for the devices section and device descriptor is as follows:
-#    devices {
-#      <device name> {              # <device name>: any string without space
-#        type <device type>         # <device type> e.g. AUDIO_DEVICE_OUT_SPEAKER
-#        address <address>          # optional: device address, char string less than 64 in length
-#      }
-#    }
-# - one or more "gains" sections can be present in a device descriptor section.
-#   If present, they describe the capabilities of gain controllers attached to this input or
-#   output device. e.g. :
-#   <device name> {                  # <device name>: any string without space
-#     type <device type>             # <device type> e.g. AUDIO_DEVICE_OUT_SPEAKER
-#     address <address>              # optional: device address, char string less than 64 in length
-#     gains {
-#       <gain name> {
-#         mode <gain modes supported>              # e.g. AUDIO_GAIN_MODE_CHANNELS
-#         channel_mask <controlled channels>       # needed if mode AUDIO_GAIN_MODE_CHANNELS
-#         min_value_mB <min value in millibel>
-#         max_value_mB <max value in millibel>
-#         default_value_mB <default value in millibel>
-#         step_value_mB <step value in millibel>
-#         min_ramp_ms <min duration in ms>         # needed if mode AUDIO_GAIN_MODE_RAMP
-#         max_ramp_ms <max duration ms>            # needed if mode AUDIO_GAIN_MODE_RAMP
-#       }
-#     }
-#   }
-# - when a device descriptor is present, output and input profiles can refer to this device by
-# its name in their "devices" section instead of specifying a device type. e.g. :
-#   outputs {
-#     primary {
-#       sampling_rates 44100
-#       channel_masks AUDIO_CHANNEL_OUT_STEREO
-#       formats AUDIO_FORMAT_PCM_16_BIT
-#       devices <device name>
-#       flags AUDIO_OUTPUT_FLAG_PRIMARY
-#     }
-#   }
-# sample audio_policy.conf file below
-
-audio_hw_modules {
-  primary {
-    global_configuration {
-      attached_output_devices AUDIO_DEVICE_OUT_SPEAKER
-      default_output_device AUDIO_DEVICE_OUT_SPEAKER
-      attached_input_devices AUDIO_DEVICE_IN_BUILTIN_MIC
-      audio_hal_version 3.0
-    }
-    devices {
-      speaker {
-        type AUDIO_DEVICE_OUT_SPEAKER
-        gains {
-          gain_1 {
-            mode AUDIO_GAIN_MODE_JOINT
-            min_value_mB -8400
-            max_value_mB 4000
-            default_value_mB 0
-            step_value_mB 100
-          }
-        }
-      }
-    }
-    outputs {
-      primary {
-        sampling_rates 48000
-        channel_masks AUDIO_CHANNEL_OUT_STEREO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices speaker
-        flags AUDIO_OUTPUT_FLAG_PRIMARY
-      }
-    }
-    inputs {
-      primary {
-        sampling_rates 8000|16000
-        channel_masks AUDIO_CHANNEL_IN_MONO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_IN_BUILTIN_MIC
-      }
-    }
-  }
-  r_submix {
-    global_configuration {
-      attached_input_devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
-      audio_hal_version 2.0
-    }
-    outputs {
-      submix {
-        sampling_rates 48000
-        channel_masks AUDIO_CHANNEL_OUT_STEREO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_OUT_REMOTE_SUBMIX
-      }
-    }
-    inputs {
-      submix {
-        sampling_rates 48000
-        channel_masks AUDIO_CHANNEL_IN_STEREO
-        formats AUDIO_FORMAT_PCM_16_BIT
-        devices AUDIO_DEVICE_IN_REMOTE_SUBMIX
-      }
-    }
-  }
-}
diff --git a/services/audiopolicy/common/managerdefinitions/Android.bp b/services/audiopolicy/common/managerdefinitions/Android.bp
index f02f3cf..71d5789 100644
--- a/services/audiopolicy/common/managerdefinitions/Android.bp
+++ b/services/audiopolicy/common/managerdefinitions/Android.bp
@@ -3,13 +3,12 @@
 
     srcs: [
         "src/AudioCollections.cpp",
-        "src/AudioGain.cpp",
         "src/AudioInputDescriptor.cpp",
         "src/AudioOutputDescriptor.cpp",
         "src/AudioPatch.cpp",
         "src/AudioPolicyMix.cpp",
         "src/AudioPort.cpp",
-        "src/AudioProfile.cpp",
+        "src/AudioProfileVector.cpp",
         "src/AudioRoute.cpp",
         "src/ClientDescriptor.cpp",
         "src/DeviceDescriptor.cpp",
@@ -21,6 +20,7 @@
         "src/TypeConverter.cpp",
     ],
     shared_libs: [
+        "libaudiofoundation",
         "libcutils",
         "libhidlbase",
         "liblog",
@@ -28,7 +28,10 @@
         "libutils",
         "libxml2",
     ],
-    export_shared_lib_headers: ["libmedia"],
+    export_shared_lib_headers: [
+        "libaudiofoundation",
+        "libmedia",
+    ],
     static_libs: [
         "libaudioutils",
     ],
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h b/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h
index a948ea9..646ef31 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioCollections.h
@@ -31,7 +31,7 @@
 class AudioPortVector : public Vector<sp<AudioPort> >
 {
 public:
-    sp<AudioPort> findByTagName(const String8 &tagName) const;
+    sp<AudioPort> findByTagName(const std::string &tagName) const;
 };
 
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
index 37f9d14..816498c 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioInputDescriptor.h
@@ -97,7 +97,7 @@
     RecordClientVector clientsList(bool activeOnly = false,
         audio_source_t source = AUDIO_SOURCE_DEFAULT, bool preferredDeviceOnly = false) const;
 
-    void setAppState(uid_t uid, app_state_t state);
+    void setAppState(audio_port_handle_t portId, app_state_t state);
 
     // implementation of ClientMapHandler<RecordClientDescriptor>
     void addClient(const sp<RecordClientDescriptor> &client) override;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index 2264d8f..c17f308 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -19,7 +19,6 @@
 #include <unordered_map>
 #include <unordered_set>
 
-#include <AudioGain.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
 #include <DeviceDescriptor.h>
@@ -30,6 +29,7 @@
 #include <AudioPolicyMix.h>
 #include <EffectDescriptor.h>
 #include <SoundTriggerSession.h>
+#include <media/AudioProfile.h>
 
 namespace android {
 
@@ -40,7 +40,8 @@
                       DeviceVector &availableOutputDevices,
                       DeviceVector &availableInputDevices,
                       sp<DeviceDescriptor> &defaultOutputDevice)
-        : mHwModules(hwModules),
+        : mEngineLibraryNameSuffix(kDefaultEngineLibraryNameSuffix),
+          mHwModules(hwModules),
           mAvailableOutputDevices(availableOutputDevices),
           mAvailableInputDevices(availableInputDevices),
           mDefaultOutputDevice(defaultOutputDevice),
@@ -55,6 +56,14 @@
         mSource = file;
     }
 
+    const std::string& getEngineLibraryNameSuffix() const {
+        return mEngineLibraryNameSuffix;
+    }
+
+    void setEngineLibraryNameSuffix(const std::string& suffix) {
+        mEngineLibraryNameSuffix = suffix;
+    }
+
     void setHwModules(const HwModuleCollection &hwModules)
     {
         mHwModules = hwModules;
@@ -108,10 +117,11 @@
     void setDefault(void)
     {
         mSource = "AudioPolicyConfig::setDefault";
+        mEngineLibraryNameSuffix = kDefaultEngineLibraryNameSuffix;
         mDefaultOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_SPEAKER);
-        mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic());
+        mDefaultOutputDevice->addAudioProfile(AudioProfile::createFullDynamic(gDynamicFormat));
         sp<DeviceDescriptor> defaultInputDevice = new DeviceDescriptor(AUDIO_DEVICE_IN_BUILTIN_MIC);
-        defaultInputDevice->addAudioProfile(AudioProfile::createFullDynamic());
+        defaultInputDevice->addAudioProfile(AudioProfile::createFullDynamic(gDynamicFormat));
         sp<AudioProfile> micProfile = new AudioProfile(
                 AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_MONO, 8000);
         defaultInputDevice->addAudioProfile(micProfile);
@@ -123,14 +133,14 @@
         mDefaultOutputDevice->attach(module);
         defaultInputDevice->attach(module);
 
-        sp<OutputProfile> outProfile = new OutputProfile(String8("primary"));
+        sp<OutputProfile> outProfile = new OutputProfile("primary");
         outProfile->addAudioProfile(
                 new AudioProfile(AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 44100));
         outProfile->addSupportedDevice(mDefaultOutputDevice);
         outProfile->setFlags(AUDIO_OUTPUT_FLAG_PRIMARY);
         module->addOutputProfile(outProfile);
 
-        sp<InputProfile> inProfile = new InputProfile(String8("primary"));
+        sp<InputProfile> inProfile = new InputProfile("primary");
         inProfile->addAudioProfile(micProfile);
         inProfile->addSupportedDevice(defaultInputDevice);
         module->addInputProfile(inProfile);
@@ -167,7 +177,10 @@
     }
 
 private:
+    static const constexpr char* const kDefaultEngineLibraryNameSuffix = "default";
+
     std::string mSource;
+    std::string mEngineLibraryNameSuffix;
     HwModuleCollection &mHwModules; /**< Collection of Module, with Profiles, i.e. Mix Ports. */
     DeviceVector &mAvailableOutputDevices;
     DeviceVector &mAvailableInputDevices;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
index d906f11..c26bffc 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPort.h
@@ -17,9 +17,10 @@
 #pragma once
 
 #include "AudioCollections.h"
-#include "AudioProfile.h"
-#include "AudioGain.h"
+#include "AudioProfileVector.h"
 #include "HandleGenerator.h"
+#include <media/AudioGain.h>
+#include <media/AudioPortBase.h>
 #include <utils/String8.h>
 #include <utils/Vector.h>
 #include <utils/RefBase.h>
@@ -32,25 +33,15 @@
 class HwModule;
 class AudioRoute;
 
-class AudioPort : public virtual RefBase, private HandleGenerator<audio_port_handle_t>
+class AudioPort : public virtual RefBase, public AudioPortBase<AudioProfileVector>,
+                  private HandleGenerator<audio_port_handle_t>
 {
 public:
-    AudioPort(const String8& name, audio_port_type_t type,  audio_port_role_t role) :
-        mName(name), mType(type), mRole(role), mFlags(AUDIO_OUTPUT_FLAG_NONE) {}
+    AudioPort(const std::string& name, audio_port_type_t type,  audio_port_role_t role) :
+            AudioPortBase(name, type, role), mFlags(AUDIO_OUTPUT_FLAG_NONE) {}
 
     virtual ~AudioPort() {}
 
-    void setName(const String8 &name) { mName = name; }
-    const String8 &getName() const { return mName; }
-
-    audio_port_type_t getType() const { return mType; }
-    audio_port_role_t getRole() const { return mRole; }
-
-    virtual const String8 getTagName() const = 0;
-
-    void setGains(const AudioGains &gains) { mGains = gains; }
-    const AudioGains &getGains() const { return mGains; }
-
     virtual void setFlags(uint32_t flags)
     {
         //force direct flag if offload flag is set: offloading implies a direct output stream
@@ -70,18 +61,9 @@
     // Audio port IDs are in a different namespace than AudioFlinger unique IDs
     static audio_port_handle_t getNextUniqueId();
 
-    virtual void toAudioPort(struct audio_port *port) const;
-
     virtual void importAudioPort(const sp<AudioPort>& port, bool force = false);
 
-    void addAudioProfile(const sp<AudioProfile> &profile) { mProfiles.add(profile); }
-
-    void setAudioProfiles(const AudioProfileVector &profiles) { mProfiles = profiles; }
-    AudioProfileVector &getAudioProfiles() { return mProfiles; }
-
-    bool hasValidAudioProfile() const { return mProfiles.hasValidProfile(); }
-
-    bool hasDynamicAudioProfile() const { return mProfiles.hasDynamicProfile(); }
+    bool hasDynamicAudioProfile() const { return getAudioProfileVectorBase()->hasDynamicProfile(); }
 
     // searches for an exact match
     virtual status_t checkExactAudioProfile(const struct audio_port_config *config) const;
@@ -95,10 +77,6 @@
         return mProfiles.checkCompatibleProfile(samplingRate, channelMask, format, mType, mRole);
     }
 
-    void clearAudioProfiles() { return mProfiles.clearProfiles(); }
-
-    status_t checkGain(const struct audio_gain_config *gainConfig, int index) const;
-
     void pickAudioProfile(uint32_t &samplingRate,
                           audio_channel_mask_t &channelMask,
                           audio_format_t &format) const;
@@ -110,8 +88,8 @@
     // Used to select an audio HAL output stream with a sample format providing the
     // less degradation for a given AudioTrack sample format.
     static bool isBetterFormatMatch(audio_format_t newFormat,
-                                        audio_format_t currentFormat,
-                                        audio_format_t targetFormat);
+                                    audio_format_t currentFormat,
+                                    audio_format_t targetFormat);
     static uint32_t formatDistance(audio_format_t format1,
                                    audio_format_t format2);
     static const uint32_t kFormatDistanceMax = 4;
@@ -121,12 +99,6 @@
     const char *getModuleName() const;
     sp<HwModule> getModule() const { return mModule; }
 
-    bool useInputChannelMask() const
-    {
-        return ((mType == AUDIO_PORT_TYPE_DEVICE) && (mRole == AUDIO_PORT_ROLE_SOURCE)) ||
-                ((mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SINK));
-    }
-
     inline bool isDirectOutput() const
     {
         return (mType == AUDIO_PORT_TYPE_MIX) && (mRole == AUDIO_PORT_ROLE_SOURCE) &&
@@ -136,43 +108,36 @@
     void addRoute(const sp<AudioRoute> &route) { mRoutes.add(route); }
     const AudioRouteVector &getRoutes() const { return mRoutes; }
 
-    void dump(String8 *dst, int spaces, bool verbose = true) const;
-
     void log(const char* indent) const;
 
-    AudioGains mGains; // gain controllers
-
 private:
-    void pickChannelMask(audio_channel_mask_t &channelMask, const ChannelsVector &channelMasks) const;
-    void pickSamplingRate(uint32_t &rate,const SampleRateVector &samplingRates) const;
+    void pickChannelMask(audio_channel_mask_t &channelMask,
+                         const ChannelMaskSet &channelMasks) const;
+    void pickSamplingRate(uint32_t &rate, const SampleRateSet &samplingRates) const;
 
-    sp<HwModule> mModule;                 // audio HW module exposing this I/O stream
-    String8  mName;
-    audio_port_type_t mType;
-    audio_port_role_t mRole;
     uint32_t mFlags; // attribute flags mask (e.g primary output, direct output...).
-    AudioProfileVector mProfiles; // AudioProfiles supported by this port (format, Rates, Channels)
+    sp<HwModule> mModule;     // audio HW module exposing this I/O stream
     AudioRouteVector mRoutes; // Routes involving this port
 };
 
-class AudioPortConfig : public virtual RefBase
+class AudioPortConfig : public AudioPortConfigBase
 {
 public:
     status_t applyAudioPortConfig(const struct audio_port_config *config,
-                                  struct audio_port_config *backupConfig = NULL);
+                                  struct audio_port_config *backupConfig = NULL) override;
+
     virtual void toAudioPortConfig(struct audio_port_config *dstConfig,
-                                   const struct audio_port_config *srcConfig = NULL) const = 0;
+            const struct audio_port_config *srcConfig = NULL) const override;
+
     virtual sp<AudioPort> getAudioPort() const = 0;
+
     virtual bool hasSameHwModuleAs(const sp<AudioPortConfig>& other) const {
         return (other != 0) && (other->getAudioPort() != 0) && (getAudioPort() != 0) &&
                 (other->getAudioPort()->getModuleHandle() == getAudioPort()->getModuleHandle());
     }
+
     bool hasGainController(bool canUseForVolume = false) const;
 
-    unsigned int mSamplingRate = 0u;
-    audio_format_t mFormat = AUDIO_FORMAT_INVALID;
-    audio_channel_mask_t mChannelMask = AUDIO_CHANNEL_NONE;
-    struct audio_gain_config mGain = { .index = -1 };
     union audio_io_flags mFlags = { AUDIO_INPUT_FLAG_NONE };
 };
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
deleted file mode 100644
index b588d57..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/AudioProfile.h
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-#include <vector>
-
-#include <system/audio.h>
-#include <utils/RefBase.h>
-#include <utils/SortedVector.h>
-#include <utils/String8.h>
-
-#include "policy.h"
-
-namespace android {
-
-typedef SortedVector<uint32_t> SampleRateVector;
-typedef Vector<audio_format_t> FormatVector;
-
-template <typename T>
-bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
-{
-    if (left.size() != right.size()) {
-        return false;
-    }
-    for (size_t index = 0; index < right.size(); index++) {
-        if (left[index] != right[index]) {
-            return false;
-        }
-    }
-    return true;
-}
-
-template <typename T>
-bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
-{
-    return !(left == right);
-}
-
-class ChannelsVector : public SortedVector<audio_channel_mask_t>
-{
-public:
-    ChannelsVector() = default;
-    ChannelsVector(const ChannelsVector&) = default;
-    ChannelsVector(const SortedVector<audio_channel_mask_t>& sv) :
-            SortedVector<audio_channel_mask_t>(sv) {}
-    ChannelsVector& operator=(const ChannelsVector&) = default;
-
-    // Applies audio_channel_mask_out_to_in to all elements and returns the result.
-    ChannelsVector asInMask() const;
-    // Applies audio_channel_mask_in_to_out to all elements and returns the result.
-    ChannelsVector asOutMask() const;
-};
-
-class AudioProfile : public virtual RefBase
-{
-public:
-    static sp<AudioProfile> createFullDynamic();
-
-    AudioProfile(audio_format_t format, audio_channel_mask_t channelMasks, uint32_t samplingRate);
-    AudioProfile(audio_format_t format,
-                 const ChannelsVector &channelMasks,
-                 const SampleRateVector &samplingRateCollection);
-
-    audio_format_t getFormat() const { return mFormat; }
-    const ChannelsVector &getChannels() const { return mChannelMasks; }
-    const SampleRateVector &getSampleRates() const { return mSamplingRates; }
-    void setChannels(const ChannelsVector &channelMasks);
-    void setSampleRates(const SampleRateVector &sampleRates);
-
-    void clear();
-    bool isValid() const { return hasValidFormat() && hasValidRates() && hasValidChannels(); }
-    bool supportsChannels(audio_channel_mask_t channels) const
-    {
-        return mChannelMasks.indexOf(channels) >= 0;
-    }
-    bool supportsRate(uint32_t rate) const { return mSamplingRates.indexOf(rate) >= 0; }
-
-    status_t checkExact(uint32_t rate, audio_channel_mask_t channels, audio_format_t format) const;
-    status_t checkCompatibleChannelMask(audio_channel_mask_t channelMask,
-                                        audio_channel_mask_t &updatedChannelMask,
-                                        audio_port_type_t portType,
-                                        audio_port_role_t portRole) const;
-    status_t checkCompatibleSamplingRate(uint32_t samplingRate,
-                                         uint32_t &updatedSamplingRate) const;
-
-    bool hasValidFormat() const { return mFormat != AUDIO_FORMAT_DEFAULT; }
-    bool hasValidRates() const { return !mSamplingRates.isEmpty(); }
-    bool hasValidChannels() const { return !mChannelMasks.isEmpty(); }
-
-    void setDynamicChannels(bool dynamic) { mIsDynamicChannels = dynamic; }
-    bool isDynamicChannels() const { return mIsDynamicChannels; }
-
-    void setDynamicRate(bool dynamic) { mIsDynamicRate = dynamic; }
-    bool isDynamicRate() const { return mIsDynamicRate; }
-
-    void setDynamicFormat(bool dynamic) { mIsDynamicFormat = dynamic; }
-    bool isDynamicFormat() const { return mIsDynamicFormat; }
-
-    bool isDynamic() { return mIsDynamicFormat || mIsDynamicChannels || mIsDynamicRate; }
-
-    void dump(String8 *dst, int spaces) const;
-
-private:
-    String8  mName;
-    audio_format_t mFormat;
-    ChannelsVector mChannelMasks;
-    SampleRateVector mSamplingRates;
-
-    bool mIsDynamicFormat = false;
-    bool mIsDynamicChannels = false;
-    bool mIsDynamicRate = false;
-};
-
-
-class AudioProfileVector : public Vector<sp<AudioProfile> >
-{
-public:
-    ssize_t add(const sp<AudioProfile> &profile);
-    // This API is intended to be used by the policy manager once retrieving capabilities
-    // for a profile with dynamic format, rate and channels attributes
-    ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd);
-
-    status_t checkExactProfile(uint32_t samplingRate, audio_channel_mask_t channelMask,
-                               audio_format_t format) const;
-    status_t checkCompatibleProfile(uint32_t &samplingRate, audio_channel_mask_t &channelMask,
-                                    audio_format_t &format,
-                                    audio_port_type_t portType,
-                                    audio_port_role_t portRole) const;
-    void clearProfiles();
-    // Assuming that this profile vector contains input profiles,
-    // find the best matching config from 'outputProfiles', according to
-    // the given preferences for audio formats and channel masks.
-    // Note: std::vectors are used because specialized containers for formats
-    //       and channels can be sorted and use their own ordering.
-    status_t findBestMatchingOutputConfig(const AudioProfileVector& outputProfiles,
-            const std::vector<audio_format_t>& preferredFormats, // order: most pref -> least pref
-            const std::vector<audio_channel_mask_t>& preferredOutputChannels,
-            bool preferHigherSamplingRates,
-            audio_config_base *bestOutputConfig) const;
-
-    sp<AudioProfile> getFirstValidProfile() const;
-    sp<AudioProfile> getFirstValidProfileFor(audio_format_t format) const;
-    bool hasValidProfile() const { return getFirstValidProfile() != 0; }
-
-    FormatVector getSupportedFormats() const;
-    bool hasDynamicChannelsFor(audio_format_t format) const;
-    bool hasDynamicFormat() const { return getProfileFor(gDynamicFormat) != 0; }
-    bool hasDynamicProfile() const;
-    bool hasDynamicRateFor(audio_format_t format) const;
-
-    // One audio profile will be added for each format supported by Audio HAL
-    void setFormats(const FormatVector &formats);
-
-    void dump(String8 *dst, int spaces) const;
-
-private:
-    sp<AudioProfile> getProfileFor(audio_format_t format) const;
-    void setSampleRatesFor(const SampleRateVector &sampleRates, audio_format_t format);
-    void setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format);
-
-    static int compareFormats(const sp<AudioProfile> *profile1, const sp<AudioProfile> *profile2);
-};
-
-bool operator == (const AudioProfile &left, const AudioProfile &right);
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioProfileVector.h b/services/audiopolicy/common/managerdefinitions/include/AudioProfileVector.h
new file mode 100644
index 0000000..2e7328d
--- /dev/null
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioProfileVector.h
@@ -0,0 +1,68 @@
+/*
+ * Copyright (C) 2015 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <media/AudioProfile.h>
+#include <system/audio.h>
+
+namespace android {
+
+class AudioProfileVector : public AudioProfileVectorBase {
+public:
+    virtual ~AudioProfileVector() = default;
+
+    ssize_t add(const sp<AudioProfile> &profile) override;
+
+    // This API is intended to be used by the policy manager once retrieving capabilities
+    // for a profile with dynamic format, rate and channels attributes
+    ssize_t addProfileFromHal(const sp<AudioProfile> &profileToAdd);
+    void appendProfiles(const AudioProfileVectorBase& audioProfiles) {
+        insert(end(), audioProfiles.begin(), audioProfiles.end());
+    }
+
+    status_t checkExactProfile(const uint32_t samplingRate,
+                               audio_channel_mask_t channelMask,
+                               audio_format_t format) const;
+
+    status_t checkCompatibleProfile(uint32_t &samplingRate,
+                                    audio_channel_mask_t &channelMask,
+                                    audio_format_t &format,
+                                    audio_port_type_t portType,
+                                    audio_port_role_t portRole) const;
+
+    // Assuming that this profile vector contains input profiles,
+    // find the best matching config from 'outputProfiles', according to
+    // the given preferences for audio formats and channel masks.
+    // Note: std::vectors are used because specialized containers for formats
+    //       and channels can be sorted and use their own ordering.
+    status_t findBestMatchingOutputConfig(
+            const AudioProfileVector &outputProfiles,
+            const std::vector<audio_format_t> &preferredFormats, // order: most pref -> least pref
+            const std::vector<audio_channel_mask_t> &preferredOutputChannels,
+            bool preferHigherSamplingRates,
+            audio_config_base *bestOutputConfig) const;
+
+    // One audio profile will be added for each format supported by Audio HAL
+    void setFormats(const FormatVector &formats);
+
+private:
+    sp<AudioProfile> getProfileFor(audio_format_t format) const;
+    void setSampleRatesFor(const SampleRateSet &sampleRates, audio_format_t format);
+    void setChannelsFor(const ChannelMaskSet &channelMasks, audio_format_t format);
+};
+
+} // namespace android
\ No newline at end of file
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 33e506f..c2f1d93 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -30,13 +30,13 @@
 {
 public:
      // Note that empty name refers by convention to a generic device.
-    explicit DeviceDescriptor(audio_devices_t type, const String8 &tagName = String8(""));
+    explicit DeviceDescriptor(audio_devices_t type, const std::string &tagName = "");
     DeviceDescriptor(audio_devices_t type, const FormatVector &encodedFormats,
-            const String8 &tagName = String8(""));
+            const std::string &tagName = "");
 
     virtual ~DeviceDescriptor() {}
 
-    virtual const String8 getTagName() const { return mTagName; }
+    virtual const std::string getTagName() const { return mTagName; }
 
     audio_devices_t type() const { return mDeviceType; }
     String8 address() const { return mAddress; }
@@ -75,7 +75,7 @@
 
 private:
     String8 mAddress{""};
-    String8 mTagName; // Unique human readable identifier for a device port found in conf file.
+    std::string mTagName; // Unique human readable identifier for a device port found in conf file.
     audio_devices_t     mDeviceType;
     FormatVector        mEncodedFormats;
     audio_port_handle_t mId = AUDIO_PORT_HANDLE_NONE;
@@ -112,10 +112,17 @@
      * equal to AUDIO_PORT_HANDLE_NONE, it also returns a nullptr.
      */
     sp<DeviceDescriptor> getDeviceFromId(audio_port_handle_t id) const;
-    sp<DeviceDescriptor> getDeviceFromTagName(const String8 &tagName) const;
+    sp<DeviceDescriptor> getDeviceFromTagName(const std::string &tagName) const;
     DeviceVector getDevicesFromHwModule(audio_module_handle_t moduleHandle) const;
     audio_devices_t getDeviceTypesFromHwModule(audio_module_handle_t moduleHandle) const;
 
+    DeviceVector getFirstDevicesFromTypes(std::vector<audio_devices_t> orderedTypes) const;
+    sp<DeviceDescriptor> getFirstExistingDevice(std::vector<audio_devices_t> orderedTypes) const;
+
+    // If there are devices with the given type and the devices to add is not empty,
+    // remove all the devices with the given type and add all the devices to add.
+    void replaceDevicesByType(audio_devices_t typeToRemove, const DeviceVector &devicesToAdd);
+
     bool contains(const sp<DeviceDescriptor>& item) const { return indexOf(item) >= 0; }
 
     /**
diff --git a/services/audiopolicy/common/managerdefinitions/include/HwModule.h b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
index eb34da4..65c886a 100644
--- a/services/audiopolicy/common/managerdefinitions/include/HwModule.h
+++ b/services/audiopolicy/common/managerdefinitions/include/HwModule.h
@@ -82,17 +82,17 @@
     status_t addInputProfile(const sp<IOProfile> &profile);
     status_t addProfile(const sp<IOProfile> &profile);
 
-    status_t addOutputProfile(const String8& name, const audio_config_t *config,
+    status_t addOutputProfile(const std::string& name, const audio_config_t *config,
             audio_devices_t device, const String8& address);
-    status_t removeOutputProfile(const String8& name);
-    status_t addInputProfile(const String8& name, const audio_config_t *config,
+    status_t removeOutputProfile(const std::string& name);
+    status_t addInputProfile(const std::string& name, const audio_config_t *config,
             audio_devices_t device, const String8& address);
-    status_t removeInputProfile(const String8& name);
+    status_t removeInputProfile(const std::string& name);
 
     audio_module_handle_t getHandle() const { return mHandle; }
     void setHandle(audio_module_handle_t handle);
 
-    sp<AudioPort> findPortByTagName(const String8 &tagName) const
+    sp<AudioPort> findPortByTagName(const std::string &tagName) const
     {
         return mPorts.findByTagName(tagName);
     }
diff --git a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
index e0b56d4..419dd35 100644
--- a/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
+++ b/services/audiopolicy/common/managerdefinitions/include/IOProfile.h
@@ -18,6 +18,7 @@
 
 #include "AudioPort.h"
 #include "DeviceDescriptor.h"
+#include "policy.h"
 #include <utils/String8.h>
 #include <system/audio.h>
 
@@ -33,7 +34,7 @@
 class IOProfile : public AudioPort
 {
 public:
-    IOProfile(const String8 &name, audio_port_role_t role)
+    IOProfile(const std::string &name, audio_port_role_t role)
         : AudioPort(name, AUDIO_PORT_TYPE_MIX, role),
           maxOpenCount(1),
           curOpenCount(0),
@@ -41,7 +42,7 @@
           curActiveCount(0) {}
 
     // For a Profile aka MixPort, tag name and name are equivalent.
-    virtual const String8 getTagName() const { return getName(); }
+    virtual const std::string getTagName() const { return getName(); }
 
     // FIXME: this is needed because shared MMAP stream clients use the same audio session.
     // Once capture clients are tracked individually and not per session this can be removed
@@ -183,13 +184,13 @@
 class InputProfile : public IOProfile
 {
 public:
-    explicit InputProfile(const String8 &name) : IOProfile(name, AUDIO_PORT_ROLE_SINK) {}
+    explicit InputProfile(const std::string &name) : IOProfile(name, AUDIO_PORT_ROLE_SINK) {}
 };
 
 class OutputProfile : public IOProfile
 {
 public:
-    explicit OutputProfile(const String8 &name) : IOProfile(name, AUDIO_PORT_ROLE_SOURCE) {}
+    explicit OutputProfile(const std::string &name) : IOProfile(name, AUDIO_PORT_ROLE_SOURCE) {}
 };
 
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h b/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
deleted file mode 100644
index 0a27947..0000000
--- a/services/audiopolicy/common/managerdefinitions/include/audio_policy_conf.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (C) 2012 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-
-/////////////////////////////////////////////////
-//      Definitions for audio policy configuration file (audio_policy.conf)
-/////////////////////////////////////////////////
-
-#define AUDIO_HARDWARE_MODULE_ID_MAX_LEN 32
-
-#define AUDIO_POLICY_CONFIG_FILE "/system/etc/audio_policy.conf"
-#define AUDIO_POLICY_VENDOR_CONFIG_FILE "/vendor/etc/audio_policy.conf"
-
-// global configuration
-#define GLOBAL_CONFIG_TAG "global_configuration"
-
-#define ATTACHED_OUTPUT_DEVICES_TAG "attached_output_devices"
-#define DEFAULT_OUTPUT_DEVICE_TAG "default_output_device"
-#define ATTACHED_INPUT_DEVICES_TAG "attached_input_devices"
-#define SPEAKER_DRC_ENABLED_TAG "speaker_drc_enabled"
-#define AUDIO_HAL_VERSION_TAG "audio_hal_version"
-
-// hw modules descriptions
-#define AUDIO_HW_MODULE_TAG "audio_hw_modules"
-
-#define OUTPUTS_TAG "outputs"
-#define INPUTS_TAG "inputs"
-
-#define SAMPLING_RATES_TAG "sampling_rates"
-#define FORMATS_TAG "formats"
-#define CHANNELS_TAG "channel_masks"
-#define DEVICES_TAG "devices"
-#define FLAGS_TAG "flags"
-
-#define APM_DEVICES_TAG "devices"
-#define APM_DEVICE_TYPE "type"
-#define APM_DEVICE_ADDRESS "address"
-
-#define MIXERS_TAG "mixers"
-#define MIXER_TYPE "type"
-#define MIXER_TYPE_MUX "mux"
-#define MIXER_TYPE_MIX "mix"
-
-#define GAINS_TAG "gains"
-#define GAIN_MODE "mode"
-#define GAIN_CHANNELS "channel_mask"
-#define GAIN_MIN_VALUE "min_value_mB"
-#define GAIN_MAX_VALUE "max_value_mB"
-#define GAIN_DEFAULT_VALUE "default_value_mB"
-#define GAIN_STEP_VALUE "step_value_mB"
-#define GAIN_MIN_RAMP_MS "min_ramp_ms"
-#define GAIN_MAX_RAMP_MS "max_ramp_ms"
-
-#define DYNAMIC_VALUE_TAG "dynamic" // special value for "channel_masks", "sampling_rates" and
-                                    // "formats" in outputs descriptors indicating that supported
-                                    // values should be queried after opening the output.
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
index c90a582..b391a09 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioCollections.cpp
@@ -21,11 +21,10 @@
 #include "AudioPort.h"
 #include "AudioRoute.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 
 namespace android {
 
-sp<AudioPort> AudioPortVector::findByTagName(const String8 &tagName) const
+sp<AudioPort> AudioPortVector::findByTagName(const std::string &tagName) const
 {
     for (const auto& port : *this) {
         if (port->getTagName() == tagName) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
deleted file mode 100644
index 2725870..0000000
--- a/services/audiopolicy/common/managerdefinitions/src/AudioGain.cpp
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "APM::AudioGain"
-//#define LOG_NDEBUG 0
-
-//#define VERY_VERBOSE_LOGGING
-#ifdef VERY_VERBOSE_LOGGING
-#define ALOGVV ALOGV
-#else
-#define ALOGVV(a...) do { } while(0)
-#endif
-
-#include "AudioGain.h"
-#include <utils/Log.h>
-#include <utils/String8.h>
-
-#include <math.h>
-
-namespace android {
-
-AudioGain::AudioGain(int index, bool useInChannelMask)
-{
-    mIndex = index;
-    mUseInChannelMask = useInChannelMask;
-    memset(&mGain, 0, sizeof(struct audio_gain));
-}
-
-void AudioGain::getDefaultConfig(struct audio_gain_config *config)
-{
-    config->index = mIndex;
-    config->mode = mGain.mode;
-    config->channel_mask = mGain.channel_mask;
-    if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
-        config->values[0] = mGain.default_value;
-    } else {
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
-        }
-        for (size_t i = 0; i < numValues; i++) {
-            config->values[i] = mGain.default_value;
-        }
-    }
-    if ((mGain.mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
-        config->ramp_duration_ms = mGain.min_ramp_ms;
-    }
-}
-
-status_t AudioGain::checkConfig(const struct audio_gain_config *config)
-{
-    if ((config->mode & ~mGain.mode) != 0) {
-        return BAD_VALUE;
-    }
-    if ((config->mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
-        if ((config->values[0] < mGain.min_value) ||
-                    (config->values[0] > mGain.max_value)) {
-            return BAD_VALUE;
-        }
-    } else {
-        if ((config->channel_mask & ~mGain.channel_mask) != 0) {
-            return BAD_VALUE;
-        }
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(config->channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(config->channel_mask);
-        }
-        for (size_t i = 0; i < numValues; i++) {
-            if ((config->values[i] < mGain.min_value) ||
-                    (config->values[i] > mGain.max_value)) {
-                return BAD_VALUE;
-            }
-        }
-    }
-    if ((config->mode & AUDIO_GAIN_MODE_RAMP) == AUDIO_GAIN_MODE_RAMP) {
-        if ((config->ramp_duration_ms < mGain.min_ramp_ms) ||
-                    (config->ramp_duration_ms > mGain.max_ramp_ms)) {
-            return BAD_VALUE;
-        }
-    }
-    return NO_ERROR;
-}
-
-void AudioGain::dump(String8 *dst, int spaces, int index) const
-{
-    dst->appendFormat("%*sGain %d:\n", spaces, "", index+1);
-    dst->appendFormat("%*s- mode: %08x\n", spaces, "", mGain.mode);
-    dst->appendFormat("%*s- channel_mask: %08x\n", spaces, "", mGain.channel_mask);
-    dst->appendFormat("%*s- min_value: %d mB\n", spaces, "", mGain.min_value);
-    dst->appendFormat("%*s- max_value: %d mB\n", spaces, "", mGain.max_value);
-    dst->appendFormat("%*s- default_value: %d mB\n", spaces, "", mGain.default_value);
-    dst->appendFormat("%*s- step_value: %d mB\n", spaces, "", mGain.step_value);
-    dst->appendFormat("%*s- min_ramp_ms: %d ms\n", spaces, "", mGain.min_ramp_ms);
-    dst->appendFormat("%*s- max_ramp_ms: %d ms\n", spaces, "", mGain.max_ramp_ms);
-}
-
-} // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
index a096e8f..5813937 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioInputDescriptor.cpp
@@ -22,7 +22,6 @@
 #include <policy.h>
 #include <AudioPolicyInterface.h>
 #include "AudioInputDescriptor.h"
-#include "AudioGain.h"
 #include "AudioPolicyMix.h"
 #include "HwModule.h"
 
@@ -35,8 +34,8 @@
 {
     if (profile != NULL) {
         profile->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
-        if (profile->mGains.size() > 0) {
-            profile->mGains[0]->getDefaultConfig(&mGain);
+        if (profile->getGains().size() > 0) {
+            profile->getGains()[0]->getDefaultConfig(&mGain);
         }
     }
 }
@@ -213,7 +212,7 @@
     mDevice = device;
 
     ALOGV("opening input for device %s profile %p name %s",
-          mDevice->toString().c_str(), mProfile.get(), mProfile->getName().string());
+          mDevice->toString().c_str(), mProfile.get(), mProfile->getName().c_str());
 
     audio_devices_t deviceType = mDevice->type();
 
@@ -451,13 +450,13 @@
     return enabledEffects;
 }
 
-void AudioInputDescriptor::setAppState(uid_t uid, app_state_t state)
+void AudioInputDescriptor::setAppState(audio_port_handle_t portId, app_state_t state)
 {
     RecordClientVector clients = clientsList(false /*activeOnly*/);
     RecordClientVector updatedClients;
 
     for (const auto& client : clients) {
-        if (uid == client->uid()) {
+        if (portId == client->portId()) {
             bool wasSilenced = client->isSilenced();
             client->setAppState(state);
             if (client->active() && wasSilenced != client->isSilenced()) {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 8a60cf2..6f0c3f5 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -21,10 +21,10 @@
 #include "AudioOutputDescriptor.h"
 #include "AudioPolicyMix.h"
 #include "IOProfile.h"
-#include "AudioGain.h"
 #include "Volume.h"
 #include "HwModule.h"
 #include "TypeConverter.h"
+#include <media/AudioGain.h>
 #include <media/AudioParameter.h>
 #include <media/AudioPolicy.h>
 
@@ -40,8 +40,8 @@
 {
     if (mPort.get() != nullptr) {
         mPort->pickAudioProfile(mSamplingRate, mChannelMask, mFormat);
-        if (mPort->mGains.size() > 0) {
-            mPort->mGains[0]->getDefaultConfig(&mGain);
+        if (mPort->getGains().size() > 0) {
+            mPort->getGains()[0]->getDefaultConfig(&mGain);
         }
     }
 }
@@ -483,7 +483,7 @@
     mFlags = (audio_output_flags_t)(mFlags | flags);
 
     ALOGV("opening output for device %s profile %p name %s",
-          mDevices.toString().c_str(), mProfile.get(), mProfile->getName().string());
+          mDevices.toString().c_str(), mProfile.get(), mProfile->getName().c_str());
 
     status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
                                                    output,
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
index 3a4db90..bf0cc94 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPatch.cpp
@@ -18,7 +18,6 @@
 //#define LOG_NDEBUG 0
 
 #include "AudioPatch.h"
-#include "AudioGain.h"
 #include "TypeConverter.h"
 
 #include <log/log.h>
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
index c42923a..0221348 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPolicyMix.cpp
@@ -22,7 +22,6 @@
 #include "HwModule.h"
 #include "AudioPort.h"
 #include "IOProfile.h"
-#include "AudioGain.h"
 #include <AudioOutputDescriptor.h>
 
 namespace android {
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
index c11490a..decfad1 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioPort.cpp
@@ -19,7 +19,6 @@
 #include "TypeConverter.h"
 #include "AudioPort.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 #include <policy.h>
 
 #ifndef ARRAY_SIZE
@@ -31,7 +30,7 @@
 // --- AudioPort class implementation
 void AudioPort::attach(const sp<HwModule>& module)
 {
-    ALOGV("%s: attaching module %s to port %s", __FUNCTION__, getModuleName(), mName.string());
+    ALOGV("%s: attaching module %s to port %s", __FUNCTION__, getModuleName(), mName.c_str());
     mModule = module;
 }
 
@@ -61,67 +60,6 @@
     return mModule != 0 ? mModule->getName() : "invalid module";
 }
 
-void AudioPort::toAudioPort(struct audio_port *port) const
-{
-    // TODO: update this function once audio_port structure reflects the new profile definition.
-    // For compatibility reason: flatening the AudioProfile into audio_port structure.
-    SortedVector<audio_format_t> flatenedFormats;
-    SampleRateVector flatenedRates;
-    ChannelsVector flatenedChannels;
-    for (const auto& profile : mProfiles) {
-        if (profile->isValid()) {
-            audio_format_t formatToExport = profile->getFormat();
-            const SampleRateVector &ratesToExport = profile->getSampleRates();
-            const ChannelsVector &channelsToExport = profile->getChannels();
-
-            if (flatenedFormats.indexOf(formatToExport) < 0) {
-                flatenedFormats.add(formatToExport);
-            }
-            for (size_t rateIndex = 0; rateIndex < ratesToExport.size(); rateIndex++) {
-                uint32_t rate = ratesToExport[rateIndex];
-                if (flatenedRates.indexOf(rate) < 0) {
-                    flatenedRates.add(rate);
-                }
-            }
-            for (size_t chanIndex = 0; chanIndex < channelsToExport.size(); chanIndex++) {
-                audio_channel_mask_t channels = channelsToExport[chanIndex];
-                if (flatenedChannels.indexOf(channels) < 0) {
-                    flatenedChannels.add(channels);
-                }
-            }
-            if (flatenedRates.size() > AUDIO_PORT_MAX_SAMPLING_RATES ||
-                    flatenedChannels.size() > AUDIO_PORT_MAX_CHANNEL_MASKS ||
-                    flatenedFormats.size() > AUDIO_PORT_MAX_FORMATS) {
-                ALOGE("%s: bailing out: cannot export profiles to port config", __FUNCTION__);
-                return;
-            }
-        }
-    }
-    port->role = mRole;
-    port->type = mType;
-    strlcpy(port->name, mName, AUDIO_PORT_MAX_NAME_LEN);
-    port->num_sample_rates = flatenedRates.size();
-    port->num_channel_masks = flatenedChannels.size();
-    port->num_formats = flatenedFormats.size();
-    for (size_t i = 0; i < flatenedRates.size(); i++) {
-        port->sample_rates[i] = flatenedRates[i];
-    }
-    for (size_t i = 0; i < flatenedChannels.size(); i++) {
-        port->channel_masks[i] = flatenedChannels[i];
-    }
-    for (size_t i = 0; i < flatenedFormats.size(); i++) {
-        port->formats[i] = flatenedFormats[i];
-    }
-
-    ALOGV("AudioPort::toAudioPort() num gains %zu", mGains.size());
-
-    uint32_t i;
-    for (i = 0; i < mGains.size() && i < AUDIO_PORT_MAX_GAINS; i++) {
-        port->gains[i] = mGains[i]->getGain();
-    }
-    port->num_gains = i;
-}
-
 void AudioPort::importAudioPort(const sp<AudioPort>& port, bool force __unused)
 {
     for (const auto& profileToImport : port->mProfiles) {
@@ -163,7 +101,7 @@
     return status;
 }
 
-void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateVector &samplingRates) const
+void AudioPort::pickSamplingRate(uint32_t &pickedRate,const SampleRateSet &samplingRates) const
 {
     pickedRate = 0;
     // For direct outputs, pick minimum sampling rate: this helps ensuring that the
@@ -171,9 +109,9 @@
     // sink
     if (isDirectOutput()) {
         uint32_t samplingRate = UINT_MAX;
-        for (size_t i = 0; i < samplingRates.size(); i ++) {
-            if ((samplingRates[i] < samplingRate) && (samplingRates[i] > 0)) {
-                samplingRate = samplingRates[i];
+        for (const auto rate : samplingRates) {
+            if ((rate < samplingRate) && (rate > 0)) {
+                samplingRate = rate;
             }
         }
         pickedRate = (samplingRate == UINT_MAX) ? 0 : samplingRate;
@@ -189,16 +127,16 @@
         // TODO: should mSamplingRates[] be ordered in terms of our preference
         // and we return the first (and hence most preferred) match?  This is of concern if
         // we want to choose 96kHz over 192kHz for USB driver stability or resource constraints.
-        for (size_t i = 0; i < samplingRates.size(); i ++) {
-            if ((samplingRates[i] > pickedRate) && (samplingRates[i] <= maxRate)) {
-                pickedRate = samplingRates[i];
+        for (const auto rate : samplingRates) {
+            if ((rate > pickedRate) && (rate <= maxRate)) {
+                pickedRate = rate;
             }
         }
     }
 }
 
 void AudioPort::pickChannelMask(audio_channel_mask_t &pickedChannelMask,
-                                const ChannelsVector &channelMasks) const
+                                const ChannelMaskSet &channelMasks) const
 {
     pickedChannelMask = AUDIO_CHANNEL_NONE;
     // For direct outputs, pick minimum channel count: this helps ensuring that the
@@ -206,15 +144,15 @@
     // sink
     if (isDirectOutput()) {
         uint32_t channelCount = UINT_MAX;
-        for (size_t i = 0; i < channelMasks.size(); i ++) {
+        for (const auto channelMask : channelMasks) {
             uint32_t cnlCount;
             if (useInputChannelMask()) {
-                cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_in_mask(channelMask);
             } else {
-                cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_out_mask(channelMask);
             }
             if ((cnlCount < channelCount) && (cnlCount > 0)) {
-                pickedChannelMask = channelMasks[i];
+                pickedChannelMask = channelMask;
                 channelCount = cnlCount;
             }
         }
@@ -227,15 +165,15 @@
         if (mType != AUDIO_PORT_TYPE_MIX) {
             maxCount = UINT_MAX;
         }
-        for (size_t i = 0; i < channelMasks.size(); i ++) {
+        for (const auto channelMask : channelMasks) {
             uint32_t cnlCount;
             if (useInputChannelMask()) {
-                cnlCount = audio_channel_count_from_in_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_in_mask(channelMask);
             } else {
-                cnlCount = audio_channel_count_from_out_mask(channelMasks[i]);
+                cnlCount = audio_channel_count_from_out_mask(channelMask);
             }
             if ((cnlCount > channelCount) && (cnlCount <= maxCount)) {
-                pickedChannelMask = channelMasks[i];
+                pickedChannelMask = channelMask;
                 channelCount = cnlCount;
             }
         }
@@ -343,38 +281,13 @@
             }
         }
     }
-    ALOGV("%s Port[nm:%s] profile rate=%d, format=%d, channels=%d", __FUNCTION__, mName.string(),
+    ALOGV("%s Port[nm:%s] profile rate=%d, format=%d, channels=%d", __FUNCTION__, mName.c_str(),
           samplingRate, channelMask, format);
 }
 
-status_t AudioPort::checkGain(const struct audio_gain_config *gainConfig, int index) const
-{
-    if (index < 0 || (size_t)index >= mGains.size()) {
-        return BAD_VALUE;
-    }
-    return mGains[index]->checkConfig(gainConfig);
-}
-
-void AudioPort::dump(String8 *dst, int spaces, bool verbose) const
-{
-    if (!mName.isEmpty()) {
-        dst->appendFormat("%*s- name: %s\n", spaces, "", mName.string());
-    }
-    if (verbose) {
-        mProfiles.dump(dst, spaces);
-
-        if (mGains.size() != 0) {
-            dst->appendFormat("%*s- gains:\n", spaces, "");
-            for (size_t i = 0; i < mGains.size(); i++) {
-                mGains[i]->dump(dst, spaces + 2, i);
-            }
-        }
-    }
-}
-
 void AudioPort::log(const char* indent) const
 {
-    ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.string(), mType, mRole);
+    ALOGI("%s Port[nm:%s, type:%d, role:%d]", indent, mName.c_str(), mType, mRole);
 }
 
 // --- AudioPortConfig class implementation
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioProfileVector.cpp
similarity index 60%
rename from services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
rename to services/audiopolicy/common/managerdefinitions/src/AudioProfileVector.cpp
index 69d6b0c..c17df37 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioProfileVector.cpp
@@ -21,122 +21,36 @@
 #define LOG_TAG "APM::AudioProfile"
 //#define LOG_NDEBUG 0
 
+#include <media/AudioContainers.h>
 #include <media/AudioResamplerPublic.h>
 #include <utils/Errors.h>
 
-#include "AudioGain.h"
 #include "AudioPort.h"
-#include "AudioProfile.h"
+#include "AudioProfileVector.h"
 #include "HwModule.h"
-#include "TypeConverter.h"
+#include "policy.h"
 
 namespace android {
 
-ChannelsVector ChannelsVector::asInMask() const
+status_t checkExact(const sp<AudioProfile> &audioProfile, uint32_t samplingRate,
+        audio_channel_mask_t channelMask, audio_format_t format)
 {
-    ChannelsVector inMaskVector;
-    for (const auto& channel : *this) {
-        if (audio_channel_mask_out_to_in(channel) != AUDIO_CHANNEL_INVALID) {
-            inMaskVector.add(audio_channel_mask_out_to_in(channel));
-        }
-    }
-    return inMaskVector;
-}
-
-ChannelsVector ChannelsVector::asOutMask() const
-{
-    ChannelsVector outMaskVector;
-    for (const auto& channel : *this) {
-        if (audio_channel_mask_in_to_out(channel) != AUDIO_CHANNEL_INVALID) {
-            outMaskVector.add(audio_channel_mask_in_to_out(channel));
-        }
-    }
-    return outMaskVector;
-}
-
-bool operator == (const AudioProfile &left, const AudioProfile &compareTo)
-{
-    return (left.getFormat() == compareTo.getFormat()) &&
-            (left.getChannels() == compareTo.getChannels()) &&
-            (left.getSampleRates() == compareTo.getSampleRates());
-}
-
-static AudioProfile* createFullDynamicImpl()
-{
-    AudioProfile* dynamicProfile = new AudioProfile(gDynamicFormat,
-            ChannelsVector(), SampleRateVector());
-    dynamicProfile->setDynamicFormat(true);
-    dynamicProfile->setDynamicChannels(true);
-    dynamicProfile->setDynamicRate(true);
-    return dynamicProfile;
-}
-
-// static
-sp<AudioProfile> AudioProfile::createFullDynamic()
-{
-    static sp<AudioProfile> dynamicProfile = createFullDynamicImpl();
-    return dynamicProfile;
-}
-
-AudioProfile::AudioProfile(audio_format_t format,
-                           audio_channel_mask_t channelMasks,
-                           uint32_t samplingRate) :
-        mName(String8("")),
-        mFormat(format)
-{
-    mChannelMasks.add(channelMasks);
-    mSamplingRates.add(samplingRate);
-}
-
-AudioProfile::AudioProfile(audio_format_t format,
-                           const ChannelsVector &channelMasks,
-                           const SampleRateVector &samplingRateCollection) :
-        mName(String8("")),
-        mFormat(format),
-        mChannelMasks(channelMasks),
-        mSamplingRates(samplingRateCollection) {}
-
-void AudioProfile::setChannels(const ChannelsVector &channelMasks)
-{
-    if (mIsDynamicChannels) {
-        mChannelMasks = channelMasks;
-    }
-}
-
-void AudioProfile::setSampleRates(const SampleRateVector &sampleRates)
-{
-    if (mIsDynamicRate) {
-        mSamplingRates = sampleRates;
-    }
-}
-
-void AudioProfile::clear()
-{
-    if (mIsDynamicChannels) {
-        mChannelMasks.clear();
-    }
-    if (mIsDynamicRate) {
-        mSamplingRates.clear();
-    }
-}
-
-status_t AudioProfile::checkExact(uint32_t samplingRate, audio_channel_mask_t channelMask,
-                                  audio_format_t format) const
-{
-    if (audio_formats_match(format, mFormat) &&
-            supportsChannels(channelMask) &&
-            supportsRate(samplingRate)) {
+    if (audio_formats_match(format, audioProfile->getFormat()) &&
+            audioProfile->supportsChannels(channelMask) &&
+            audioProfile->supportsRate(samplingRate)) {
         return NO_ERROR;
     }
     return BAD_VALUE;
 }
 
-status_t AudioProfile::checkCompatibleSamplingRate(uint32_t samplingRate,
-                                                   uint32_t &updatedSamplingRate) const
+status_t checkCompatibleSamplingRate(const sp<AudioProfile> &audioProfile,
+                                     uint32_t samplingRate,
+                                     uint32_t &updatedSamplingRate)
 {
     ALOG_ASSERT(samplingRate > 0);
 
-    if (mSamplingRates.isEmpty()) {
+    const SampleRateSet sampleRates = audioProfile->getSampleRates();
+    if (sampleRates.empty()) {
         updatedSamplingRate = samplingRate;
         return NO_ERROR;
     }
@@ -144,19 +58,18 @@
     // Search for the closest supported sampling rate that is above (preferred)
     // or below (acceptable) the desired sampling rate, within a permitted ratio.
     // The sampling rates are sorted in ascending order.
-    size_t orderOfDesiredRate = mSamplingRates.orderOf(samplingRate);
+    auto desiredRate = sampleRates.lower_bound(samplingRate);
 
     // Prefer to down-sample from a higher sampling rate, as we get the desired frequency spectrum.
-    if (orderOfDesiredRate < mSamplingRates.size()) {
-        uint32_t candidate = mSamplingRates[orderOfDesiredRate];
-        if (candidate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
-            updatedSamplingRate = candidate;
+    if (desiredRate != sampleRates.end()) {
+        if (*desiredRate / AUDIO_RESAMPLER_DOWN_RATIO_MAX <= samplingRate) {
+            updatedSamplingRate = *desiredRate;
             return NO_ERROR;
         }
     }
     // But if we have to up-sample from a lower sampling rate, that's OK.
-    if (orderOfDesiredRate != 0) {
-        uint32_t candidate = mSamplingRates[orderOfDesiredRate - 1];
+    if (desiredRate != sampleRates.begin()) {
+        uint32_t candidate = *(--desiredRate);
         if (candidate * AUDIO_RESAMPLER_UP_RATIO_MAX >= samplingRate) {
             updatedSamplingRate = candidate;
             return NO_ERROR;
@@ -166,12 +79,14 @@
     return BAD_VALUE;
 }
 
-status_t AudioProfile::checkCompatibleChannelMask(audio_channel_mask_t channelMask,
-                                                  audio_channel_mask_t &updatedChannelMask,
-                                                  audio_port_type_t portType,
-                                                  audio_port_role_t portRole) const
+status_t checkCompatibleChannelMask(const sp<AudioProfile> &audioProfile,
+                                    audio_channel_mask_t channelMask,
+                                    audio_channel_mask_t &updatedChannelMask,
+                                    audio_port_type_t portType,
+                                    audio_port_role_t portRole)
 {
-    if (mChannelMasks.isEmpty()) {
+    const ChannelMaskSet channelMasks = audioProfile->getChannels();
+    if (channelMasks.empty()) {
         updatedChannelMask = channelMask;
         return NO_ERROR;
     }
@@ -180,8 +95,7 @@
             == AUDIO_CHANNEL_REPRESENTATION_INDEX;
     const uint32_t channelCount = audio_channel_count_from_in_mask(channelMask);
     int bestMatch = 0;
-    for (size_t i = 0; i < mChannelMasks.size(); i ++) {
-        audio_channel_mask_t supported = mChannelMasks[i];
+    for (const auto &supported : channelMasks) {
         if (supported == channelMask) {
             // Exact matches always taken.
             updatedChannelMask = channelMask;
@@ -259,46 +173,16 @@
     return bestMatch > 0 ? NO_ERROR : BAD_VALUE;
 }
 
-void AudioProfile::dump(String8 *dst, int spaces) const
-{
-    dst->appendFormat("%s%s%s\n", mIsDynamicFormat ? "[dynamic format]" : "",
-             mIsDynamicChannels ? "[dynamic channels]" : "",
-             mIsDynamicRate ? "[dynamic rates]" : "");
-    if (mName.length() != 0) {
-        dst->appendFormat("%*s- name: %s\n", spaces, "", mName.string());
-    }
-    std::string formatLiteral;
-    if (FormatConverter::toString(mFormat, formatLiteral)) {
-        dst->appendFormat("%*s- format: %s\n", spaces, "", formatLiteral.c_str());
-    }
-    if (!mSamplingRates.isEmpty()) {
-        dst->appendFormat("%*s- sampling rates:", spaces, "");
-        for (size_t i = 0; i < mSamplingRates.size(); i++) {
-            dst->appendFormat("%d", mSamplingRates[i]);
-            dst->append(i == (mSamplingRates.size() - 1) ? "" : ", ");
-        }
-        dst->append("\n");
-    }
-
-    if (!mChannelMasks.isEmpty()) {
-        dst->appendFormat("%*s- channel masks:", spaces, "");
-        for (size_t i = 0; i < mChannelMasks.size(); i++) {
-            dst->appendFormat("0x%04x", mChannelMasks[i]);
-            dst->append(i == (mChannelMasks.size() - 1) ? "" : ", ");
-        }
-        dst->append("\n");
-    }
-}
-
 ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
 {
-    ssize_t index = Vector::add(profile);
+    ssize_t index = size();
+    push_back(profile);
     // we sort from worst to best, so that AUDIO_FORMAT_DEFAULT is always the first entry.
-    // TODO: compareFormats could be a lambda to convert between pointer-to-format to format:
-    // [](const audio_format_t *format1, const audio_format_t *format2) {
-    //     return compareFormats(*format1, *format2);
-    // }
-    sort(compareFormats);
+    std::sort(begin(), end(),
+            [](const sp<AudioProfile> & a, const sp<AudioProfile> & b)
+            {
+                return AudioPort::compareFormats(a->getFormat(), b->getFormat()) < 0;
+            });
     return index;
 }
 
@@ -310,7 +194,7 @@
     }
     if (!profileToAdd->hasValidChannels() && !profileToAdd->hasValidRates()) {
         FormatVector formats;
-        formats.add(profileToAdd->getFormat());
+        formats.push_back(profileToAdd->getFormat());
         setFormats(FormatVector(formats));
         return 0;
     }
@@ -324,7 +208,7 @@
     }
     // Go through the list of profile to avoid duplicates
     for (size_t profileIndex = 0; profileIndex < size(); profileIndex++) {
-        const sp<AudioProfile> &profile = itemAt(profileIndex);
+        const sp<AudioProfile> &profile = at(profileIndex);
         if (profile->isValid() && profile == profileToAdd) {
             // Nothing to do
             return profileIndex;
@@ -334,16 +218,16 @@
     return add(profileToAdd);
 }
 
-status_t AudioProfileVector::checkExactProfile(uint32_t samplingRate,
+status_t AudioProfileVector::checkExactProfile(const uint32_t samplingRate,
                                                audio_channel_mask_t channelMask,
                                                audio_format_t format) const
 {
-    if (isEmpty()) {
+    if (empty()) {
         return NO_ERROR;
     }
 
     for (const auto& profile : *this) {
-        if (profile->checkExact(samplingRate, channelMask, format) == NO_ERROR) {
+        if (checkExact(profile, samplingRate, channelMask, format) == NO_ERROR) {
             return NO_ERROR;
         }
     }
@@ -356,7 +240,7 @@
                                                     audio_port_type_t portType,
                                                     audio_port_role_t portRole) const
 {
-    if (isEmpty()) {
+    if (empty()) {
         return NO_ERROR;
     }
 
@@ -366,7 +250,7 @@
 
     // iterate from best format to worst format (reverse order)
     for (ssize_t i = size() - 1; i >= 0 ; --i) {
-        const sp<AudioProfile> profile = itemAt(i);
+        const sp<AudioProfile> profile = at(i);
         audio_format_t formatToCompare = profile->getFormat();
         if (formatToCompare == format ||
                 (checkInexact
@@ -376,9 +260,9 @@
             // rate and channels as well
             audio_channel_mask_t updatedChannels;
             uint32_t updatedRate;
-            if (profile->checkCompatibleChannelMask(channelMask, updatedChannels,
-                                                    portType, portRole) == NO_ERROR &&
-                    profile->checkCompatibleSamplingRate(samplingRate, updatedRate) == NO_ERROR) {
+            if (checkCompatibleChannelMask(profile, channelMask, updatedChannels,
+                                           portType, portRole) == NO_ERROR &&
+                    checkCompatibleSamplingRate(profile, samplingRate, updatedRate) == NO_ERROR) {
                 // for inexact checks we take the first linear pcm format due to sorting.
                 format = formatToCompare;
                 channelMask = updatedChannels;
@@ -390,18 +274,6 @@
     return BAD_VALUE;
 }
 
-void AudioProfileVector::clearProfiles()
-{
-    for (size_t i = size(); i != 0; ) {
-        sp<AudioProfile> profile = itemAt(--i);
-        if (profile->isDynamicFormat() && profile->hasValidFormat()) {
-            removeAt(i);
-            continue;
-        }
-        profile->clear();
-    }
-}
-
 // Returns an intersection between two possibly unsorted vectors and the contents of 'order'.
 // The result is ordered according to 'order'.
 template<typename T, typename Order>
@@ -449,16 +321,16 @@
         if (inputProfile == nullptr || outputProfile == nullptr) {
             continue;
         }
-        auto channels = intersectFilterAndOrder(inputProfile->getChannels().asOutMask(),
+        auto channels = intersectFilterAndOrder(asOutMask(inputProfile->getChannels()),
                 outputProfile->getChannels(), preferredOutputChannels);
         if (channels.empty()) {
             continue;
         }
         auto sampleRates = preferHigherSamplingRates ?
                 intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
-                        std::greater<typename SampleRateVector::value_type>()) :
+                        std::greater<typename SampleRateSet::value_type>()) :
                 intersectAndOrder(inputProfile->getSampleRates(), outputProfile->getSampleRates(),
-                        std::less<typename SampleRateVector::value_type>());
+                        std::less<typename SampleRateSet::value_type>());
         if (sampleRates.empty()) {
             continue;
         }
@@ -472,69 +344,6 @@
     return BAD_VALUE;
 }
 
-sp<AudioProfile> AudioProfileVector::getFirstValidProfile() const
-{
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->isValid()) {
-            return itemAt(i);
-        }
-    }
-    return 0;
-}
-
-sp<AudioProfile> AudioProfileVector::getFirstValidProfileFor(audio_format_t format) const
-{
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->isValid() && itemAt(i)->getFormat() == format) {
-            return itemAt(i);
-        }
-    }
-    return 0;
-}
-
-FormatVector AudioProfileVector::getSupportedFormats() const
-{
-    FormatVector supportedFormats;
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->hasValidFormat()) {
-            supportedFormats.add(itemAt(i)->getFormat());
-        }
-    }
-    return supportedFormats;
-}
-
-bool AudioProfileVector::hasDynamicChannelsFor(audio_format_t format) const
-{
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
-        if (profile->getFormat() == format && profile->isDynamicChannels()) {
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioProfileVector::hasDynamicProfile() const
-{
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->isDynamic()) {
-            return true;
-        }
-    }
-    return false;
-}
-
-bool AudioProfileVector::hasDynamicRateFor(audio_format_t format) const
-{
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
-        if (profile->getFormat() == format && profile->isDynamicRate()) {
-            return true;
-        }
-    }
-    return false;
-}
-
 void AudioProfileVector::setFormats(const FormatVector &formats)
 {
     // Only allow to change the format of dynamic profile
@@ -542,8 +351,8 @@
     if (dynamicFormatProfile == 0) {
         return;
     }
-    for (size_t i = 0; i < formats.size(); i++) {
-        sp<AudioProfile> profile = new AudioProfile(formats[i],
+    for (const auto &format : formats) {
+        sp<AudioProfile> profile = new AudioProfile(format,
                 dynamicFormatProfile->getChannels(),
                 dynamicFormatProfile->getSampleRates());
         profile->setDynamicFormat(true);
@@ -553,30 +362,20 @@
     }
 }
 
-void AudioProfileVector::dump(String8 *dst, int spaces) const
-{
-    dst->appendFormat("%*s- Profiles:\n", spaces, "");
-    for (size_t i = 0; i < size(); i++) {
-        dst->appendFormat("%*sProfile %zu:", spaces + 4, "", i);
-        itemAt(i)->dump(dst, spaces + 8);
-    }
-}
-
 sp<AudioProfile> AudioProfileVector::getProfileFor(audio_format_t format) const
 {
-    for (size_t i = 0; i < size(); i++) {
-        if (itemAt(i)->getFormat() == format) {
-            return itemAt(i);
+    for (const auto &profile : *this) {
+        if (profile->getFormat() == format) {
+            return profile;
         }
     }
-    return 0;
+    return nullptr;
 }
 
 void AudioProfileVector::setSampleRatesFor(
-        const SampleRateVector &sampleRates, audio_format_t format)
+        const SampleRateSet &sampleRates, audio_format_t format)
 {
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
+    for (const auto &profile : *this) {
         if (profile->getFormat() == format && profile->isDynamicRate()) {
             if (profile->hasValidRates()) {
                 // Need to create a new profile with same format
@@ -592,10 +391,9 @@
     }
 }
 
-void AudioProfileVector::setChannelsFor(const ChannelsVector &channelMasks, audio_format_t format)
+void AudioProfileVector::setChannelsFor(const ChannelMaskSet &channelMasks, audio_format_t format)
 {
-    for (size_t i = 0; i < size(); i++) {
-        sp<AudioProfile> profile = itemAt(i);
+    for (const auto &profile : *this) {
         if (profile->getFormat() == format && profile->isDynamicChannels()) {
             if (profile->hasValidChannels()) {
                 // Need to create a new profile with same format
@@ -611,11 +409,4 @@
     }
 }
 
-// static
-int AudioProfileVector::compareFormats(const sp<AudioProfile> *profile1,
-                                       const sp<AudioProfile> *profile2)
-{
-    return AudioPort::compareFormats((*profile1)->getFormat(), (*profile2)->getFormat());
-}
-
 } // namespace android
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
index 79f0919..0f35ff8 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioRoute.cpp
@@ -19,7 +19,6 @@
 
 #include "AudioRoute.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 
 namespace android
 {
@@ -27,11 +26,11 @@
 void AudioRoute::dump(String8 *dst, int spaces) const
 {
     dst->appendFormat("%*s- Type: %s\n", spaces, "", mType == AUDIO_ROUTE_MUX ? "Mux" : "Mix");
-    dst->appendFormat("%*s- Sink: %s\n", spaces, "", mSink->getTagName().string());
+    dst->appendFormat("%*s- Sink: %s\n", spaces, "", mSink->getTagName().c_str());
     if (mSources.size() != 0) {
         dst->appendFormat("%*s- Sources: \n", spaces, "");
         for (size_t i = 0; i < mSources.size(); i++) {
-            dst->appendFormat("%*s%s \n", spaces + 4, "", mSources[i]->getTagName().string());
+            dst->appendFormat("%*s%s \n", spaces + 4, "", mSources[i]->getTagName().c_str());
         }
     }
     dst->append("\n");
@@ -42,10 +41,10 @@
     if (mSink == 0 || dstPort == 0 || dstPort != mSink) {
         return false;
     }
-    ALOGV("%s: sinks %s matching", __FUNCTION__, mSink->getTagName().string());
+    ALOGV("%s: sinks %s matching", __FUNCTION__, mSink->getTagName().c_str());
     for (const auto &sourcePort : mSources) {
         if (sourcePort == srcPort) {
-            ALOGV("%s: sources %s matching", __FUNCTION__, sourcePort->getTagName().string());
+            ALOGV("%s: sources %s matching", __FUNCTION__, sourcePort->getTagName().c_str());
             return true;
         }
     }
diff --git a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
index ad07ab1..1dc7020 100644
--- a/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/ClientDescriptor.cpp
@@ -21,7 +21,6 @@
 #include <utils/Log.h>
 #include <utils/String8.h>
 #include <TypeConverter.h>
-#include "AudioGain.h"
 #include "AudioOutputDescriptor.h"
 #include "AudioPatch.h"
 #include "ClientDescriptor.h"
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index ecd5b34..018636d 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -22,19 +22,18 @@
 #include <set>
 #include "DeviceDescriptor.h"
 #include "TypeConverter.h"
-#include "AudioGain.h"
 #include "HwModule.h"
 
 namespace android {
 
-DeviceDescriptor::DeviceDescriptor(audio_devices_t type, const String8 &tagName) :
+DeviceDescriptor::DeviceDescriptor(audio_devices_t type, const std::string &tagName) :
         DeviceDescriptor(type, FormatVector{}, tagName)
 {
 }
 
 DeviceDescriptor::DeviceDescriptor(audio_devices_t type, const FormatVector &encodedFormats,
-        const String8 &tagName) :
-    AudioPort(String8(""), AUDIO_PORT_TYPE_DEVICE,
+        const std::string &tagName) :
+    AudioPort("", AUDIO_PORT_TYPE_DEVICE,
               audio_is_output_device(type) ? AUDIO_PORT_ROLE_SINK :
                                              AUDIO_PORT_ROLE_SOURCE),
     mTagName(tagName), mDeviceType(type), mEncodedFormats(encodedFormats)
@@ -48,9 +47,9 @@
      * For now, the workaround to remove AC3 and IEC61937 support on HDMI is to declare
      * something like 'encodedFormats="AUDIO_FORMAT_PCM_16_BIT"' on the HDMI devicePort.
      */
-    if (type == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.isEmpty()) {
-        mEncodedFormats.add(AUDIO_FORMAT_AC3);
-        mEncodedFormats.add(AUDIO_FORMAT_IEC61937);
+    if (type == AUDIO_DEVICE_OUT_HDMI && mEncodedFormats.empty()) {
+        mEncodedFormats.push_back(AUDIO_FORMAT_AC3);
+        mEncodedFormats.push_back(AUDIO_FORMAT_IEC61937);
     }
 }
 
@@ -97,7 +96,7 @@
     if (!device_has_encoding_capability(type())) {
         return true;
     }
-    if (mEncodedFormats.isEmpty()) {
+    if (mEncodedFormats.empty()) {
         return true;
     }
 
@@ -106,7 +105,7 @@
 
 bool DeviceDescriptor::supportsFormat(audio_format_t format)
 {
-    if (mEncodedFormats.isEmpty()) {
+    if (mEncodedFormats.empty()) {
         return true;
     }
 
@@ -256,7 +255,6 @@
         audio_devices_t curType = itemAt(i)->type() & ~AUDIO_DEVICE_BIT_IN;
         if ((isOutput == curIsOutput) && ((type & curType) != 0)) {
             devices.add(itemAt(i));
-            type &= ~curType;
             ALOGV("DeviceVector::%s() for type %08x found %p",
                     __func__, itemAt(i)->type(), itemAt(i).get());
         }
@@ -264,7 +262,7 @@
     return devices;
 }
 
-sp<DeviceDescriptor> DeviceVector::getDeviceFromTagName(const String8 &tagName) const
+sp<DeviceDescriptor> DeviceVector::getDeviceFromTagName(const std::string &tagName) const
 {
     for (const auto& device : *this) {
         if (device->getTagName() == tagName) {
@@ -274,6 +272,38 @@
     return nullptr;
 }
 
+DeviceVector DeviceVector::getFirstDevicesFromTypes(
+        std::vector<audio_devices_t> orderedTypes) const
+{
+    DeviceVector devices;
+    for (auto deviceType : orderedTypes) {
+        if (!(devices = getDevicesFromTypeMask(deviceType)).isEmpty()) {
+            break;
+        }
+    }
+    return devices;
+}
+
+sp<DeviceDescriptor> DeviceVector::getFirstExistingDevice(
+        std::vector<audio_devices_t> orderedTypes) const {
+    sp<DeviceDescriptor> device;
+    for (auto deviceType : orderedTypes) {
+        if ((device = getDevice(deviceType, String8(""), AUDIO_FORMAT_DEFAULT)) != nullptr) {
+            break;
+        }
+    }
+    return device;
+}
+
+void DeviceVector::replaceDevicesByType(
+        audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
+    DeviceVector devicesToRemove = getDevicesFromTypeMask(typeToRemove);
+    if (!devicesToRemove.isEmpty() && !devicesToAdd.isEmpty()) {
+        remove(devicesToRemove);
+        add(devicesToAdd);
+    }
+}
+
 void DeviceVector::dump(String8 *dst, const String8 &tag, int spaces, bool verbose) const
 {
     if (isEmpty()) {
@@ -344,8 +374,8 @@
     if (mId != 0) {
         dst->appendFormat("%*s- id: %2d\n", spaces, "", mId);
     }
-    if (!mTagName.isEmpty()) {
-        dst->appendFormat("%*s- tag name: %s\n", spaces, "", mTagName.string());
+    if (!mTagName.empty()) {
+        dst->appendFormat("%*s- tag name: %s\n", spaces, "", mTagName.c_str());
     }
 
     dst->appendFormat("%*s- type: %-48s\n", spaces, "", ::android::toString(mDeviceType).c_str());
@@ -353,7 +383,9 @@
     if (mAddress.size() != 0) {
         dst->appendFormat("%*s- address: %-32s\n", spaces, "", mAddress.string());
     }
-    AudioPort::dump(dst, spaces, verbose);
+    std::string portStr;
+    AudioPort::dump(&portStr, spaces, verbose);
+    dst->append(portStr.c_str());
 }
 
 std::string DeviceDescriptor::toString() const
diff --git a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
index 1f9b725..c232775 100644
--- a/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/HwModule.cpp
@@ -19,7 +19,6 @@
 
 #include "HwModule.h"
 #include "IOProfile.h"
-#include "AudioGain.h"
 #include <policy.h>
 #include <system/audio.h>
 
@@ -42,7 +41,7 @@
     }
 }
 
-status_t HwModule::addOutputProfile(const String8& name, const audio_config_t *config,
+status_t HwModule::addOutputProfile(const std::string& name, const audio_config_t *config,
                                     audio_devices_t device, const String8& address)
 {
     sp<IOProfile> profile = new OutputProfile(name);
@@ -96,7 +95,7 @@
     }
 }
 
-status_t HwModule::removeOutputProfile(const String8& name)
+status_t HwModule::removeOutputProfile(const std::string& name)
 {
     for (size_t i = 0; i < mOutputProfiles.size(); i++) {
         if (mOutputProfiles[i]->getName() == name) {
@@ -111,7 +110,7 @@
     return NO_ERROR;
 }
 
-status_t HwModule::addInputProfile(const String8& name, const audio_config_t *config,
+status_t HwModule::addInputProfile(const std::string& name, const audio_config_t *config,
                                    audio_devices_t device, const String8& address)
 {
     sp<IOProfile> profile = new InputProfile(name);
@@ -126,12 +125,12 @@
     profile->addSupportedDevice(devDesc);
 
     ALOGV("addInputProfile() name %s rate %d mask 0x%08x",
-          name.string(), config->sample_rate, config->channel_mask);
+          name.c_str(), config->sample_rate, config->channel_mask);
 
     return addInputProfile(profile);
 }
 
-status_t HwModule::removeInputProfile(const String8& name)
+status_t HwModule::removeInputProfile(const std::string& name)
 {
     for (size_t i = 0; i < mInputProfiles.size(); i++) {
         if (mInputProfiles[i]->getName() == name) {
@@ -194,13 +193,13 @@
             }
             DeviceVector sourceDevicesForRoute = getRouteSourceDevices(route);
             if (sourceDevicesForRoute.isEmpty()) {
-                ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().string());
+                ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().c_str());
                 continue;
             }
             sourceDevices.add(sourceDevicesForRoute);
         }
         if (sourceDevices.isEmpty()) {
-            ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().string());
+            ALOGE("%s: invalid source devices for %s", __FUNCTION__, stream->getName().c_str());
             continue;
         }
         stream->setSupportedDevices(sourceDevices);
@@ -215,7 +214,7 @@
             }
             sp<DeviceDescriptor> sinkDevice = getRouteSinkDevice(route);
             if (sinkDevice == 0) {
-                ALOGE("%s: invalid sink device for %s", __FUNCTION__, stream->getName().string());
+                ALOGE("%s: invalid sink device for %s", __FUNCTION__, stream->getName().c_str());
                 continue;
             }
             sinkDevices.add(sinkDevice);
@@ -336,7 +335,7 @@
             if (allowToCreate) {
                 moduleDevice->attach(hwModule);
                 moduleDevice->setAddress(devAddress);
-                moduleDevice->setName(String8(name));
+                moduleDevice->setName(name);
             }
             return moduleDevice;
         }
@@ -360,8 +359,8 @@
               address);
         return nullptr;
     }
-    sp<DeviceDescriptor> device = new DeviceDescriptor(type, String8(name));
-    device->setName(String8(name));
+    sp<DeviceDescriptor> device = new DeviceDescriptor(type, name);
+    device->setName(name);
     device->setAddress(String8(address));
     device->setEncodedFormat(encodedFormat);
 
diff --git a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
index fe2eaee..d4a7f31 100644
--- a/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/IOProfile.cpp
@@ -20,7 +20,6 @@
 #include <system/audio-base.h>
 #include "IOProfile.h"
 #include "HwModule.h"
-#include "AudioGain.h"
 #include "TypeConverter.h"
 
 namespace android {
@@ -105,7 +104,9 @@
 
 void IOProfile::dump(String8 *dst) const
 {
-    AudioPort::dump(dst, 4);
+    std::string portStr;
+    AudioPort::dump(&portStr, 4);
+    dst->append(portStr.c_str());
 
     dst->appendFormat("    - flags: 0x%04x", getFlags());
     std::string flagsLiteral;
diff --git a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
index 5f820c2..707169b 100644
--- a/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/Serializer.cpp
@@ -406,8 +406,8 @@
             samplingRatesFromString(samplingRates, ","));
 
     profile->setDynamicFormat(profile->getFormat() == gDynamicFormat);
-    profile->setDynamicChannels(profile->getChannels().isEmpty());
-    profile->setDynamicRate(profile->getSampleRates().isEmpty());
+    profile->setDynamicChannels(profile->getChannels().empty());
+    profile->setDynamicRate(profile->getSampleRates().empty());
 
     return profile;
 }
@@ -430,15 +430,15 @@
     audio_port_role_t portRole = (role == Attributes::roleSource) ?
             AUDIO_PORT_ROLE_SOURCE : AUDIO_PORT_ROLE_SINK;
 
-    Element mixPort = new IOProfile(String8(name.c_str()), portRole);
+    Element mixPort = new IOProfile(name, portRole);
 
     AudioProfileTraits::Collection profiles;
     status_t status = deserializeCollection<AudioProfileTraits>(child, &profiles, NULL);
     if (status != NO_ERROR) {
         return Status::fromStatusT(status);
     }
-    if (profiles.isEmpty()) {
-        profiles.add(AudioProfile::createFullDynamic());
+    if (profiles.empty()) {
+        profiles.add(AudioProfile::createFullDynamic(gDynamicFormat));
     }
     mixPort->setAudioProfiles(profiles);
 
@@ -508,7 +508,7 @@
     if (!encodedFormatsLiteral.empty()) {
         encodedFormats = formatsFromString(encodedFormatsLiteral, " ");
     }
-    Element deviceDesc = new DeviceDescriptor(type, encodedFormats, String8(name.c_str()));
+    Element deviceDesc = new DeviceDescriptor(type, encodedFormats, name);
 
     std::string address = getXmlAttribute(cur, Attributes::address);
     if (!address.empty()) {
@@ -521,8 +521,8 @@
     if (status != NO_ERROR) {
         return Status::fromStatusT(status);
     }
-    if (profiles.isEmpty()) {
-        profiles.add(AudioProfile::createFullDynamic());
+    if (profiles.empty()) {
+        profiles.add(AudioProfile::createFullDynamic(gDynamicFormat));
     }
     deviceDesc->setAudioProfiles(profiles);
 
@@ -532,7 +532,7 @@
         return Status::fromStatusT(status);
     }
     ALOGV("%s: adding device tag %s type %08x address %s", __func__,
-          deviceDesc->getName().string(), type, deviceDesc->address().string());
+          deviceDesc->getName().c_str(), type, deviceDesc->address().string());
     return deviceDesc;
 }
 
@@ -555,7 +555,7 @@
         return Status::fromStatusT(BAD_VALUE);
     }
     // Convert Sink name to port pointer
-    sp<AudioPort> sink = ctx->findPortByTagName(String8(sinkAttr.c_str()));
+    sp<AudioPort> sink = ctx->findPortByTagName(sinkAttr);
     if (sink == NULL) {
         ALOGE("%s: no sink found with name=%s", __func__, sinkAttr.c_str());
         return Status::fromStatusT(BAD_VALUE);
@@ -574,7 +574,7 @@
     char *devTag = strtok(sourcesLiteral.get(), ",");
     while (devTag != NULL) {
         if (strlen(devTag) != 0) {
-            sp<AudioPort> source = ctx->findPortByTagName(String8(devTag));
+            sp<AudioPort> source = ctx->findPortByTagName(devTag);
             if (source == NULL) {
                 ALOGE("%s: no source found with name=%s", __func__, devTag);
                 return Status::fromStatusT(BAD_VALUE);
@@ -648,7 +648,7 @@
                         ALOGV("%s: %s %s=%s", __func__, tag, childAttachedDeviceTag,
                                 reinterpret_cast<const char*>(attachedDevice.get()));
                         sp<DeviceDescriptor> device = module->getDeclaredDevices().
-                                getDeviceFromTagName(String8(reinterpret_cast<const char*>(
+                                getDeviceFromTagName(std::string(reinterpret_cast<const char*>(
                                                         attachedDevice.get())));
                         ctx->addAvailableDevice(device);
                     }
@@ -663,7 +663,7 @@
                 ALOGV("%s: %s %s=%s", __func__, tag, childDefaultOutputDeviceTag,
                         reinterpret_cast<const char*>(defaultOutputDevice.get()));
                 sp<DeviceDescriptor> device = module->getDeclaredDevices().getDeviceFromTagName(
-                        String8(reinterpret_cast<const char*>(defaultOutputDevice.get())));
+                        std::string(reinterpret_cast<const char*>(defaultOutputDevice.get())));
                 if (device != 0 && ctx->getDefaultOutputDevice() == 0) {
                     ctx->setDefaultOutputDevice(device);
                     ALOGV("%s: default is %08x",
diff --git a/services/audiopolicy/config/audio_policy_volumes.xml b/services/audiopolicy/config/audio_policy_volumes.xml
index ec64a7c..27bd3ff 100644
--- a/services/audiopolicy/config/audio_policy_volumes.xml
+++ b/services/audiopolicy/config/audio_policy_volumes.xml
@@ -44,7 +44,7 @@
     <volume stream="AUDIO_STREAM_VOICE_CALL" deviceCategory="DEVICE_CATEGORY_EXT_MEDIA"
                                              ref="DEFAULT_MEDIA_VOLUME_CURVE"/>
     <volume stream="AUDIO_STREAM_VOICE_CALL" deviceCategory="DEVICE_CATEGORY_HEARING_AID"
-                                             ref="DEFAULT_HEARING_AID_VOLUME_CURVE"/>
+                                             ref="DEFAULT_NON_MUTABLE_HEARING_AID_VOLUME_CURVE"/>
     <volume stream="AUDIO_STREAM_SYSTEM" deviceCategory="DEVICE_CATEGORY_HEADSET">
         <point>1,-3000</point>
         <point>33,-2600</point>
diff --git a/services/audiopolicy/engine/common/include/EngineBase.h b/services/audiopolicy/engine/common/include/EngineBase.h
index cedc78f..fca9a60 100644
--- a/services/audiopolicy/engine/common/include/EngineBase.h
+++ b/services/audiopolicy/engine/common/include/EngineBase.h
@@ -17,18 +17,18 @@
 #pragma once
 
 #include <EngineConfig.h>
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
 #include <ProductStrategy.h>
 #include <VolumeGroup.h>
 
 namespace android {
 namespace audio_policy {
 
-class EngineBase : public AudioPolicyManagerInterface
+class EngineBase : public EngineInterface
 {
 public:
     ///
-    /// from AudioPolicyManagerInterface
+    /// from EngineInterface
     ///
     android::status_t initCheck() override;
 
diff --git a/services/audiopolicy/engine/common/include/ProductStrategy.h b/services/audiopolicy/engine/common/include/ProductStrategy.h
index 1a2a198..c538f52 100644
--- a/services/audiopolicy/engine/common/include/ProductStrategy.h
+++ b/services/audiopolicy/engine/common/include/ProductStrategy.h
@@ -19,7 +19,6 @@
 #include "VolumeGroup.h"
 
 #include <system/audio.h>
-#include <AudioPolicyManagerInterface.h>
 #include <utils/RefBase.h>
 #include <HandleGenerator.h>
 #include <string>
@@ -27,6 +26,7 @@
 #include <map>
 #include <utils/Errors.h>
 #include <utils/String8.h>
+#include <media/AudioAttributes.h>
 
 namespace android {
 
diff --git a/services/audiopolicy/engine/common/include/VolumeCurve.h b/services/audiopolicy/engine/common/include/VolumeCurve.h
index 54314e3..d3d0904 100644
--- a/services/audiopolicy/engine/common/include/VolumeCurve.h
+++ b/services/audiopolicy/engine/common/include/VolumeCurve.h
@@ -18,7 +18,6 @@
 
 #include "IVolumeCurves.h"
 #include <policy.h>
-#include <AudioPolicyManagerInterface.h>
 #include <utils/RefBase.h>
 #include <HandleGenerator.h>
 #include <utils/String8.h>
diff --git a/services/audiopolicy/engine/common/include/VolumeGroup.h b/services/audiopolicy/engine/common/include/VolumeGroup.h
index c34b406..5378f64 100644
--- a/services/audiopolicy/engine/common/include/VolumeGroup.h
+++ b/services/audiopolicy/engine/common/include/VolumeGroup.h
@@ -16,7 +16,6 @@
 
 #pragma once
 
-#include <AudioPolicyManagerInterface.h>
 #include <VolumeCurve.h>
 #include <system/audio.h>
 #include <utils/RefBase.h>
diff --git a/services/audiopolicy/engine/common/src/ProductStrategy.cpp b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
index f74f190..ac3e462 100644
--- a/services/audiopolicy/engine/common/src/ProductStrategy.cpp
+++ b/services/audiopolicy/engine/common/src/ProductStrategy.cpp
@@ -19,6 +19,7 @@
 
 #include "ProductStrategy.h"
 
+#include <media/AudioProductStrategy.h>
 #include <media/TypeConverter.h>
 #include <utils/String8.h>
 #include <cstdint>
diff --git a/services/audiopolicy/engine/config/src/EngineConfig.cpp b/services/audiopolicy/engine/config/src/EngineConfig.cpp
index 1ad7739..d47fbd2 100644
--- a/services/audiopolicy/engine/config/src/EngineConfig.cpp
+++ b/services/audiopolicy/engine/config/src/EngineConfig.cpp
@@ -32,9 +32,9 @@
 #include <istream>
 
 #include <cstdint>
+#include <stdarg.h>
 #include <string>
 
-
 namespace android {
 
 using utilities::convertTo;
@@ -603,7 +603,39 @@
     return NO_ERROR;
 }
 
+namespace {
+
+class XmlErrorHandler {
+public:
+    XmlErrorHandler() {
+        xmlSetGenericErrorFunc(this, &xmlErrorHandler);
+    }
+    XmlErrorHandler(const XmlErrorHandler&) = delete;
+    XmlErrorHandler(XmlErrorHandler&&) = delete;
+    XmlErrorHandler& operator=(const XmlErrorHandler&) = delete;
+    XmlErrorHandler& operator=(XmlErrorHandler&&) = delete;
+    ~XmlErrorHandler() {
+        xmlSetGenericErrorFunc(NULL, NULL);
+        if (!mErrorMessage.empty()) {
+            ALOG(LOG_ERROR, "libxml2", "%s", mErrorMessage.c_str());
+        }
+    }
+    static void xmlErrorHandler(void* ctx, const char* msg, ...) {
+        char buffer[256];
+        va_list args;
+        va_start(args, msg);
+        vsnprintf(buffer, sizeof(buffer), msg, args);
+        va_end(args);
+        static_cast<XmlErrorHandler*>(ctx)->mErrorMessage += buffer;
+    }
+private:
+    std::string mErrorMessage;
+};
+
+}  // namespace
+
 ParsingResult parse(const char* path) {
+    XmlErrorHandler errorHandler;
     xmlDocPtr doc;
     doc = xmlParseFile(path);
     if (doc == NULL) {
@@ -641,6 +673,7 @@
 }
 
 android::status_t parseLegacyVolumeFile(const char* path, VolumeGroups &volumeGroups) {
+    XmlErrorHandler errorHandler;
     xmlDocPtr doc;
     doc = xmlParseFile(path);
     if (doc == NULL) {
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
index ebd82a7..ae3fc79 100644
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
+++ b/services/audiopolicy/engine/interface/AudioPolicyManagerObserver.h
@@ -16,7 +16,6 @@
 
 #pragma once
 
-#include <AudioGain.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
 #include <IOProfile.h>
diff --git a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h b/services/audiopolicy/engine/interface/EngineInterface.h
similarity index 97%
rename from services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
rename to services/audiopolicy/engine/interface/EngineInterface.h
index b7fd031..0c58a7c 100644
--- a/services/audiopolicy/engine/interface/AudioPolicyManagerInterface.h
+++ b/services/audiopolicy/engine/interface/EngineInterface.h
@@ -38,7 +38,7 @@
 /**
  * This interface is dedicated to the policy manager that a Policy Engine shall implement.
  */
-class AudioPolicyManagerInterface
+class EngineInterface
 {
 public:
     /**
@@ -295,7 +295,13 @@
     virtual void dump(String8 *dst) const = 0;
 
 protected:
-    virtual ~AudioPolicyManagerInterface() {}
+    virtual ~EngineInterface() {}
 };
 
+__attribute__((visibility("default")))
+extern "C" EngineInterface* createEngineInstance();
+
+__attribute__((visibility("default")))
+extern "C" void destroyEngineInstance(EngineInterface *engine);
+
 } // namespace android
diff --git a/services/audiopolicy/engineconfigurable/Android.bp b/services/audiopolicy/engineconfigurable/Android.bp
index c27dc88..8f522f0 100644
--- a/services/audiopolicy/engineconfigurable/Android.bp
+++ b/services/audiopolicy/engineconfigurable/Android.bp
@@ -33,6 +33,7 @@
 
     ],
     shared_libs: [
+        "libaudiofoundation",
         "liblog",
         "libcutils",
         "libutils",
diff --git a/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h b/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h
index efc69da..f52de21 100644
--- a/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h
+++ b/services/audiopolicy/engineconfigurable/include/AudioPolicyEngineInstance.h
@@ -16,7 +16,7 @@
 
 #pragma once
 
-class AudioPolicyManagerInterface;
+class EngineInterface;
 class AudioPolicyPluginInterface;
 
 namespace android {
@@ -69,7 +69,7 @@
  * Compile time error will claim if invalid interface is requested.
  */
 template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const;
+EngineInterface *EngineInstance::queryInterface() const;
 
 template <>
 AudioPolicyPluginInterface *EngineInstance::queryInterface() const;
diff --git a/services/audiopolicy/engineconfigurable/src/Engine.cpp b/services/audiopolicy/engineconfigurable/src/Engine.cpp
index cb45fcf..c37efca 100644
--- a/services/audiopolicy/engineconfigurable/src/Engine.cpp
+++ b/services/audiopolicy/engineconfigurable/src/Engine.cpp
@@ -361,7 +361,7 @@
 }
 
 template <>
-AudioPolicyManagerInterface *Engine::queryInterface()
+EngineInterface *Engine::queryInterface()
 {
     return this;
 }
diff --git a/services/audiopolicy/engineconfigurable/src/Engine.h b/services/audiopolicy/engineconfigurable/src/Engine.h
index 4662e7e..3b371d8 100644
--- a/services/audiopolicy/engineconfigurable/src/Engine.h
+++ b/services/audiopolicy/engineconfigurable/src/Engine.h
@@ -17,7 +17,7 @@
 #pragma once
 
 #include "EngineBase.h"
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
 #include <AudioPolicyPluginInterface.h>
 #include "Collection.h"
 
diff --git a/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp b/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp
index 2442590..b127796 100644
--- a/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp
+++ b/services/audiopolicy/engineconfigurable/src/EngineInstance.cpp
@@ -14,7 +14,7 @@
  * limitations under the License.
  */
 
-#include <AudioPolicyManagerInterface.h>
+#include <EngineInterface.h>
 #include <AudioPolicyPluginInterface.h>
 #include "AudioPolicyEngineInstance.h"
 #include "Engine.h"
@@ -45,9 +45,9 @@
 }
 
 template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const
+EngineInterface *EngineInstance::queryInterface() const
 {
-    return getEngine()->queryInterface<AudioPolicyManagerInterface>();
+    return getEngine()->queryInterface<EngineInterface>();
 }
 
 template <>
@@ -57,5 +57,16 @@
 }
 
 } // namespace audio_policy
+
+extern "C" EngineInterface* createEngineInstance()
+{
+    return audio_policy::EngineInstance::getInstance()->queryInterface<EngineInterface>();
+}
+
+extern "C" void destroyEngineInstance(EngineInterface*)
+{
+    // The engine is a singleton.
+}
+
 } // namespace android
 
diff --git a/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h b/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
index 5bfad29..72c8de1 100644
--- a/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
+++ b/services/audiopolicy/engineconfigurable/wrapper/include/ParameterManagerWrapper.h
@@ -16,7 +16,6 @@
 
 #pragma once
 
-#include <AudioGain.h>
 #include <AudioPort.h>
 #include <HwModule.h>
 #include <DeviceDescriptor.h>
diff --git a/services/audiopolicy/enginedefault/Android.bp b/services/audiopolicy/enginedefault/Android.bp
index 7b42c6a..aaf4158 100644
--- a/services/audiopolicy/enginedefault/Android.bp
+++ b/services/audiopolicy/enginedefault/Android.bp
@@ -1,16 +1,15 @@
 cc_library_shared {
     name: "libaudiopolicyenginedefault",
-    export_include_dirs: ["include"],
     srcs: [
         "src/Engine.cpp",
         "src/EngineInstance.cpp",
     ],
     cflags: [
+        "-fvisibility=hidden",
         "-Wall",
         "-Werror",
         "-Wextra",
     ],
-    local_include_dirs: ["include"],
     header_libs: [
         "libbase_headers",
         "libaudiopolicycommon",
@@ -22,6 +21,7 @@
         "libaudiopolicyengine_config",
     ],
     shared_libs: [
+        "libaudiofoundation",
         "liblog",
         "libcutils",
         "libutils",
diff --git a/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h b/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h
deleted file mode 100644
index 1e329f0..0000000
--- a/services/audiopolicy/enginedefault/include/AudioPolicyEngineInstance.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (C) 2015 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- *      http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#pragma once
-
-class AudioPolicyManagerInterface;
-
-namespace android
-{
-namespace audio_policy
-{
-
-class Engine;
-
-class EngineInstance
-{
-protected:
-    EngineInstance();
-
-public:
-    virtual ~EngineInstance();
-
-    /**
-     * Get Audio Policy Engine instance.
-     *
-     * @return pointer to Route Manager Instance object.
-     */
-    static EngineInstance *getInstance();
-
-    /**
-     * Interface query.
-     * The first client of an interface of the policy engine will start the singleton.
-     *
-     * @tparam RequestedInterface: interface that the client is wishing to retrieve.
-     *
-     * @return interface handle.
-     */
-    template <class RequestedInterface>
-    RequestedInterface *queryInterface() const;
-
-protected:
-    /**
-     * Get Audio Policy Engine instance.
-     *
-     * @return Audio Policy Engine singleton.
-     */
-    Engine *getEngine() const;
-
-private:
-    /* Copy facilities are put private to disable copy. */
-    EngineInstance(const EngineInstance &object);
-    EngineInstance &operator=(const EngineInstance &object);
-};
-
-/**
- * Limit template instantation to supported type interfaces.
- * Compile time error will claim if invalid interface is requested.
- */
-template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const;
-
-} // namespace audio_policy
-} // namespace android
diff --git a/services/audiopolicy/enginedefault/src/Engine.cpp b/services/audiopolicy/enginedefault/src/Engine.cpp
index 04170ac..b895c2f 100644
--- a/services/audiopolicy/enginedefault/src/Engine.cpp
+++ b/services/audiopolicy/enginedefault/src/Engine.cpp
@@ -136,27 +136,23 @@
     return EngineBase::setForceUse(usage, config);
 }
 
-audio_devices_t Engine::getDeviceForStrategyInt(legacy_strategy strategy,
-                                                DeviceVector availableOutputDevices,
-                                                DeviceVector availableInputDevices,
-                                                const SwAudioOutputCollection &outputs,
-                                                uint32_t outputDeviceTypesToIgnore) const
+DeviceVector Engine::getDevicesForStrategyInt(legacy_strategy strategy,
+                                              DeviceVector availableOutputDevices,
+                                              DeviceVector availableInputDevices,
+                                              const SwAudioOutputCollection &outputs) const
 {
-    uint32_t device = AUDIO_DEVICE_NONE;
-    uint32_t availableOutputDevicesType =
-            availableOutputDevices.types() & ~outputDeviceTypesToIgnore;
+    DeviceVector devices;
 
     switch (strategy) {
 
     case STRATEGY_TRANSMITTED_THROUGH_SPEAKER:
-        device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+        devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         break;
 
     case STRATEGY_SONIFICATION_RESPECTFUL:
         if (isInCall() || outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_VOICE_CALL))) {
-            device = getDeviceForStrategyInt(
-                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs);
         } else {
             bool media_active_locally =
                     outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_MUSIC),
@@ -165,17 +161,18 @@
                         toVolumeSource(AUDIO_STREAM_ACCESSIBILITY),
                         SONIFICATION_RESPECTFUL_AFTER_MUSIC_DELAY);
             // routing is same as media without the "remote" device
-            device = getDeviceForStrategyInt(STRATEGY_MEDIA,
+            availableOutputDevices.remove(availableOutputDevices.getDevicesFromTypeMask(
+                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX));
+            devices = getDevicesForStrategyInt(STRATEGY_MEDIA,
                     availableOutputDevices,
-                    availableInputDevices, outputs,
-                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX | outputDeviceTypesToIgnore);
+                    availableInputDevices, outputs);
             // if no media is playing on the device, check for mandatory use of "safe" speaker
             // when media would have played on speaker, and the safe speaker path is available
-            if (!media_active_locally
-                    && (device & AUDIO_DEVICE_OUT_SPEAKER)
-                    && (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-                device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
-                device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+            if (!media_active_locally) {
+                devices.replaceDevicesByType(
+                        AUDIO_DEVICE_OUT_SPEAKER,
+                        availableOutputDevices.getDevicesFromTypeMask(
+                                AUDIO_DEVICE_OUT_SPEAKER_SAFE));
             }
         }
         break;
@@ -183,9 +180,8 @@
     case STRATEGY_DTMF:
         if (!isInCall()) {
             // when off call, DTMF strategy follows the same rules as MEDIA strategy
-            device = getDeviceForStrategyInt(
-                    STRATEGY_MEDIA, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_MEDIA, availableOutputDevices, availableInputDevices, outputs);
             break;
         }
         // when in call, DTMF and PHONE strategies follow the same rules
@@ -197,24 +193,27 @@
         //   - cannot route from voice call RX OR
         //   - audio HAL version is < 3.0 and TX device is on the primary HW module
         if (getPhoneState() == AUDIO_MODE_IN_CALL) {
-            audio_devices_t txDevice = getDeviceForInputSource(AUDIO_SOURCE_VOICE_COMMUNICATION);
+            audio_devices_t txDevice = getDeviceForInputSource(
+                    AUDIO_SOURCE_VOICE_COMMUNICATION)->type();
             sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
-            audio_devices_t availPrimaryInputDevices =
-                 availableInputDevices.getDeviceTypesFromHwModule(primaryOutput->getModuleHandle());
+            DeviceVector availPrimaryInputDevices =
+                    availableInputDevices.getDevicesFromHwModule(primaryOutput->getModuleHandle());
 
             // TODO: getPrimaryOutput return only devices from first module in
             // audio_policy_configuration.xml, hearing aid is not there, but it's
             // a primary device
             // FIXME: this is not the right way of solving this problem
-            audio_devices_t availPrimaryOutputDevices =
-                (primaryOutput->supportedDevices().types() | AUDIO_DEVICE_OUT_HEARING_AID) &
-                availableOutputDevices.types();
+            DeviceVector availPrimaryOutputDevices = availableOutputDevices.getDevicesFromTypeMask(
+                    primaryOutput->supportedDevices().types());
+            availPrimaryOutputDevices.add(
+                    availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_HEARING_AID));
 
-            if (((availableInputDevices.types() &
-                    AUDIO_DEVICE_IN_TELEPHONY_RX & ~AUDIO_DEVICE_BIT_IN) == 0) ||
-                    (((txDevice & availPrimaryInputDevices & ~AUDIO_DEVICE_BIT_IN) != 0) &&
-                         (primaryOutput->getAudioPort()->getModuleVersionMajor() < 3))) {
-                availableOutputDevicesType = availPrimaryOutputDevices;
+            if ((availableInputDevices.getDevice(AUDIO_DEVICE_IN_TELEPHONY_RX,
+                    String8(""), AUDIO_FORMAT_DEFAULT) == nullptr) ||
+                    ((availPrimaryInputDevices.getDevice(
+                            txDevice, String8(""), AUDIO_FORMAT_DEFAULT) != nullptr) &&
+                            (primaryOutput->getAudioPort()->getModuleVersionMajor() < 3))) {
+                availableOutputDevices = availPrimaryOutputDevices;
             }
         }
         // for phone strategy, we first consider the forced use and then the available devices by
@@ -222,49 +221,40 @@
         switch (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)) {
         case AUDIO_POLICY_FORCE_BT_SCO:
             if (!isInCall() || strategy != STRATEGY_DTMF) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
-                if (device) break;
+                devices = availableOutputDevices.getDevicesFromTypeMask(
+                        AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT);
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
-            if (device) break;
+            devices = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET, AUDIO_DEVICE_OUT_BLUETOOTH_SCO});
+            if (!devices.isEmpty()) break;
             // if SCO device is requested but no SCO device is available, fall back to default case
             FALLTHROUGH_INTENDED;
 
         default:    // FORCE_NONE
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_HEARING_AID;
-            if (device) break;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_HEARING_AID);
+            if (!devices.isEmpty()) break;
             // when not in a phone call, phone strategy should route STREAM_VOICE_CALL to A2DP
             if (!isInCall() &&
                     (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
                      outputs.isA2dpSupported()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-                if (device) break;
+                devices = availableOutputDevices.getFirstDevicesFromTypes({
+                        AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+                        AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES});
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_HEADSET;
-            if (device) break;
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
-            if (device) break;
+            devices = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_WIRED_HEADPHONE, AUDIO_DEVICE_OUT_WIRED_HEADSET,
+                    AUDIO_DEVICE_OUT_LINE, AUDIO_DEVICE_OUT_USB_HEADSET,
+                    AUDIO_DEVICE_OUT_USB_DEVICE});
+            if (!devices.isEmpty()) break;
             if (!isInCall()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-                if (device) break;
+                devices = availableOutputDevices.getFirstDevicesFromTypes({
+                        AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,
+                        AUDIO_DEVICE_OUT_AUX_DIGITAL, AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET});
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_EARPIECE;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_EARPIECE);
             break;
 
         case AUDIO_POLICY_FORCE_SPEAKER:
@@ -273,22 +263,18 @@
             if (!isInCall() &&
                     (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
                      outputs.isA2dpSupported()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-                if (device) break;
+                devices = availableOutputDevices.getDevicesFromTypeMask(
+                        AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER);
+                if (!devices.isEmpty()) break;
             }
             if (!isInCall()) {
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
-                if (device) break;
-                device = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
-                if (device) break;
+                devices = availableOutputDevices.getFirstDevicesFromTypes({
+                        AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_DEVICE,
+                        AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, AUDIO_DEVICE_OUT_AUX_DIGITAL,
+                        AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET});
+                if (!devices.isEmpty()) break;
             }
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
             break;
         }
     break;
@@ -298,9 +284,8 @@
         // If incall, just select the STRATEGY_PHONE device
         if (isInCall() ||
                 outputs.isActiveLocally(toVolumeSource(AUDIO_STREAM_VOICE_CALL))) {
-            device = getDeviceForStrategyInt(
-                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
             break;
         }
         FALLTHROUGH_INTENDED;
@@ -313,41 +298,37 @@
 
         if ((strategy == STRATEGY_SONIFICATION) ||
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)) {
-            device = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+            devices = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         }
 
         // if SCO headset is connected and we are told to use it, play ringtone over
         // speaker and BT SCO
-        if ((availableOutputDevicesType & AUDIO_DEVICE_OUT_ALL_SCO) != 0) {
-            uint32_t device2 = AUDIO_DEVICE_NONE;
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT;
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET;
-            }
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_SCO;
-            }
+        if (!availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_ALL_SCO).isEmpty()) {
+            DeviceVector devices2;
+            devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET,
+                    AUDIO_DEVICE_OUT_BLUETOOTH_SCO});
             // Use ONLY Bluetooth SCO output when ringing in vibration mode
             if (!((getForceUse(AUDIO_POLICY_FORCE_FOR_SYSTEM) == AUDIO_POLICY_FORCE_SYSTEM_ENFORCED)
                     && (strategy == STRATEGY_ENFORCED_AUDIBLE))) {
                 if (getForceUse(AUDIO_POLICY_FORCE_FOR_VIBRATE_RINGING)
                         == AUDIO_POLICY_FORCE_BT_SCO) {
-                    if (device2 != AUDIO_DEVICE_NONE) {
-                        device = device2;
+                    if (!devices2.isEmpty()) {
+                        devices = devices2;
                         break;
                     }
                 }
             }
             // Use both Bluetooth SCO and phone default output when ringing in normal mode
             if (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION) == AUDIO_POLICY_FORCE_BT_SCO) {
-                if ((strategy == STRATEGY_SONIFICATION) &&
-                        (device & AUDIO_DEVICE_OUT_SPEAKER) &&
-                        (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-                    device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
-                    device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+                if (strategy == STRATEGY_SONIFICATION) {
+                    devices.replaceDevicesByType(
+                            AUDIO_DEVICE_OUT_SPEAKER,
+                            availableOutputDevices.getDevicesFromTypeMask(
+                                    AUDIO_DEVICE_OUT_SPEAKER_SAFE));
                 }
-                if (device2 != AUDIO_DEVICE_NONE) {
-                    device |= device2;
+                if (!devices2.isEmpty()) {
+                    devices.add(devices2);
                     break;
                 }
             }
@@ -361,25 +342,20 @@
             // compressed format as they would likely not be mixed and dropped.
             for (size_t i = 0; i < outputs.size(); i++) {
                 sp<AudioOutputDescriptor> desc = outputs.valueAt(i);
-                audio_devices_t devices = desc->devices().types() &
-                    (AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_HDMI_ARC);
-                if (desc->isActive() && !audio_is_linear_pcm(desc->mFormat) &&
-                        devices != AUDIO_DEVICE_NONE) {
-                    availableOutputDevicesType = availableOutputDevices.types() & ~devices;
+                if (desc->isActive() && !audio_is_linear_pcm(desc->getFormat())) {
+                    availableOutputDevices.remove(desc->devices().getDevicesFromTypeMask(
+                            AUDIO_DEVICE_OUT_HDMI | AUDIO_DEVICE_OUT_SPDIF
+                            | AUDIO_DEVICE_OUT_HDMI_ARC));
                 }
             }
-            availableOutputDevices =
-                    availableOutputDevices.getDevicesFromTypeMask(availableOutputDevicesType);
             if (outputs.isActive(toVolumeSource(AUDIO_STREAM_RING)) ||
                     outputs.isActive(toVolumeSource(AUDIO_STREAM_ALARM))) {
-                return getDeviceForStrategyInt(
-                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+                return getDevicesForStrategyInt(
+                    STRATEGY_SONIFICATION, availableOutputDevices, availableInputDevices, outputs);
             }
             if (isInCall()) {
-                return getDeviceForStrategyInt(
-                        STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
-                        outputDeviceTypesToIgnore);
+                return getDevicesForStrategyInt(
+                        STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
             }
         }
         // For other cases, STRATEGY_ACCESSIBILITY behaves like STRATEGY_MEDIA
@@ -388,128 +364,116 @@
     // FIXME: STRATEGY_REROUTING follow STRATEGY_MEDIA for now
     case STRATEGY_REROUTING:
     case STRATEGY_MEDIA: {
-        uint32_t device2 = AUDIO_DEVICE_NONE;
+        DeviceVector devices2;
         if (strategy != STRATEGY_SONIFICATION) {
             // no sonification on remote submix (e.g. WFD)
-            if (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
-                                                 String8("0"), AUDIO_FORMAT_DEFAULT) != 0) {
-                device2 = availableOutputDevices.types() & AUDIO_DEVICE_OUT_REMOTE_SUBMIX;
+            sp<DeviceDescriptor> remoteSubmix;
+            if ((remoteSubmix = availableOutputDevices.getDevice(
+                    AUDIO_DEVICE_OUT_REMOTE_SUBMIX, String8("0"),
+                    AUDIO_FORMAT_DEFAULT)) != nullptr) {
+                devices2.add(remoteSubmix);
             }
         }
         if (isInCall() && (strategy == STRATEGY_MEDIA)) {
-            device = getDeviceForStrategyInt(
-                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs,
-                    outputDeviceTypesToIgnore);
+            devices = getDevicesForStrategyInt(
+                    STRATEGY_PHONE, availableOutputDevices, availableInputDevices, outputs);
             break;
         }
         // FIXME: Find a better solution to prevent routing to BT hearing aid(b/122931261).
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP)) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HEARING_AID;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_HEARING_AID);
         }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) != AUDIO_POLICY_FORCE_NO_BT_A2DP) &&
                  outputs.isA2dpSupported()) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP;
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES;
-            }
-            if (device2 == AUDIO_DEVICE_NONE) {
-                device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER;
-            }
+            devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES,
+                    AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER});
         }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
             (getForceUse(AUDIO_POLICY_FORCE_FOR_MEDIA) == AUDIO_POLICY_FORCE_SPEAKER)) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
+        if (devices2.isEmpty()) {
+            devices2 = availableOutputDevices.getFirstDevicesFromTypes({
+                    AUDIO_DEVICE_OUT_WIRED_HEADPHONE, AUDIO_DEVICE_OUT_LINE,
+                    AUDIO_DEVICE_OUT_WIRED_HEADSET, AUDIO_DEVICE_OUT_USB_HEADSET,
+                    AUDIO_DEVICE_OUT_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_DEVICE,
+                    AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET});
         }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_LINE;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_WIRED_HEADSET;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_HEADSET;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_ACCESSORY;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_USB_DEVICE;
-        }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET;
-        }
-        if ((device2 == AUDIO_DEVICE_NONE) && (strategy != STRATEGY_SONIFICATION)) {
+        if ((devices2.isEmpty()) && (strategy != STRATEGY_SONIFICATION)) {
             // no sonification on aux digital (e.g. HDMI)
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_DIGITAL;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_AUX_DIGITAL);
         }
-        if ((device2 == AUDIO_DEVICE_NONE) &&
+        if ((devices2.isEmpty()) &&
                 (getForceUse(AUDIO_POLICY_FORCE_FOR_DOCK) == AUDIO_POLICY_FORCE_ANALOG_DOCK)) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET;
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(
+                    AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET);
         }
-        if (device2 == AUDIO_DEVICE_NONE) {
-            device2 = availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER;
+        if (devices2.isEmpty()) {
+            devices2 = availableOutputDevices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER);
         }
-        int device3 = AUDIO_DEVICE_NONE;
+        DeviceVector devices3;
         if (strategy == STRATEGY_MEDIA) {
             // ARC, SPDIF and AUX_LINE can co-exist with others.
-            device3 = availableOutputDevicesType & AUDIO_DEVICE_OUT_HDMI_ARC;
-            device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPDIF);
-            device3 |= (availableOutputDevicesType & AUDIO_DEVICE_OUT_AUX_LINE);
+            devices3 = availableOutputDevices.getDevicesFromTypeMask(
+                    AUDIO_DEVICE_OUT_HDMI_ARC | AUDIO_DEVICE_OUT_SPDIF | AUDIO_DEVICE_OUT_AUX_LINE);
         }
 
-        device2 |= device3;
+        devices2.add(devices3);
         // device is DEVICE_OUT_SPEAKER if we come from case STRATEGY_SONIFICATION or
         // STRATEGY_ENFORCED_AUDIBLE, AUDIO_DEVICE_NONE otherwise
-        device |= device2;
+        devices.add(devices2);
 
         // If hdmi system audio mode is on, remove speaker out of output list.
         if ((strategy == STRATEGY_MEDIA) &&
             (getForceUse(AUDIO_POLICY_FORCE_FOR_HDMI_SYSTEM_AUDIO) ==
                 AUDIO_POLICY_FORCE_HDMI_SYSTEM_AUDIO_ENFORCED)) {
-            device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+            devices.remove(devices.getDevicesFromTypeMask(AUDIO_DEVICE_OUT_SPEAKER));
         }
 
         // for STRATEGY_SONIFICATION:
         // if SPEAKER was selected, and SPEAKER_SAFE is available, use SPEAKER_SAFE instead
-        if ((strategy == STRATEGY_SONIFICATION) &&
-                (device & AUDIO_DEVICE_OUT_SPEAKER) &&
-                (availableOutputDevicesType & AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
-            device |= AUDIO_DEVICE_OUT_SPEAKER_SAFE;
-            device &= ~AUDIO_DEVICE_OUT_SPEAKER;
+        if (strategy == STRATEGY_SONIFICATION) {
+            devices.replaceDevicesByType(
+                    AUDIO_DEVICE_OUT_SPEAKER,
+                    availableOutputDevices.getDevicesFromTypeMask(
+                            AUDIO_DEVICE_OUT_SPEAKER_SAFE));
         }
         } break;
 
     default:
-        ALOGW("getDeviceForStrategy() unknown strategy: %d", strategy);
+        ALOGW("getDevicesForStrategy() unknown strategy: %d", strategy);
         break;
     }
 
-    if (device == AUDIO_DEVICE_NONE) {
-        ALOGV("getDeviceForStrategy() no device found for strategy %d", strategy);
-        device = getApmObserver()->getDefaultOutputDevice()->type();
-        ALOGE_IF(device == AUDIO_DEVICE_NONE,
-                 "getDeviceForStrategy() no default device defined");
+    if (devices.isEmpty()) {
+        ALOGV("getDevicesForStrategy() no device found for strategy %d", strategy);
+        sp<DeviceDescriptor> defaultOutputDevice = getApmObserver()->getDefaultOutputDevice();
+        if (defaultOutputDevice != nullptr) {
+            devices.add(defaultOutputDevice);
+        }
+        ALOGE_IF(devices.isEmpty(),
+                 "getDevicesForStrategy() no default device defined");
     }
-    ALOGVV("getDeviceForStrategy() strategy %d, device %x", strategy, device);
-    return device;
+
+    ALOGVV("getDevices"
+           "ForStrategy() strategy %d, device %x", strategy, devices.types());
+    return devices;
 }
 
 
-audio_devices_t Engine::getDeviceForInputSource(audio_source_t inputSource) const
+sp<DeviceDescriptor> Engine::getDeviceForInputSource(audio_source_t inputSource) const
 {
     const DeviceVector availableOutputDevices = getApmObserver()->getAvailableOutputDevices();
     const DeviceVector availableInputDevices = getApmObserver()->getAvailableInputDevices();
     const SwAudioOutputCollection &outputs = getApmObserver()->getOutputs();
-    audio_devices_t availableDeviceTypes = availableInputDevices.types() & ~AUDIO_DEVICE_BIT_IN;
+    DeviceVector availableDevices = availableInputDevices;
     sp<AudioOutputDescriptor> primaryOutput = outputs.getPrimaryOutput();
-    audio_devices_t availablePrimaryDeviceTypes = availableInputDevices.getDeviceTypesFromHwModule(
-        primaryOutput->getModuleHandle()) & ~AUDIO_DEVICE_BIT_IN;
-    uint32_t device = AUDIO_DEVICE_NONE;
+    DeviceVector availablePrimaryDevices = availableInputDevices.getDevicesFromHwModule(
+            primaryOutput->getModuleHandle());
+    sp<DeviceDescriptor> device;
 
     // when a call is active, force device selection to match source VOICE_COMMUNICATION
     // for most other input sources to avoid rerouting call TX audio
@@ -532,57 +496,47 @@
     switch (inputSource) {
     case AUDIO_SOURCE_DEFAULT:
     case AUDIO_SOURCE_MIC:
-    if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_A2DP) {
-        device = AUDIO_DEVICE_IN_BLUETOOTH_A2DP;
-    } else if ((getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) &&
-        (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET)) {
-        device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-        device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-        device = AUDIO_DEVICE_IN_USB_HEADSET;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-        device = AUDIO_DEVICE_IN_USB_DEVICE;
-    } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-        device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-    }
-    break;
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_BLUETOOTH_A2DP, String8(""), AUDIO_FORMAT_DEFAULT);
+        if (device != nullptr) break;
+        if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) {
+            device = availableDevices.getDevice(
+                    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+            if (device != nullptr) break;
+        }
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
+        break;
 
     case AUDIO_SOURCE_VOICE_COMMUNICATION:
         // Allow only use of devices on primary input if in call and HAL does not support routing
         // to voice call path.
         if ((getPhoneState() == AUDIO_MODE_IN_CALL) &&
-                (availableOutputDevices.types() & AUDIO_DEVICE_OUT_TELEPHONY_TX) == 0) {
-            availableDeviceTypes = availablePrimaryDeviceTypes;
+                (availableOutputDevices.getDevice(AUDIO_DEVICE_OUT_TELEPHONY_TX,
+                        String8(""), AUDIO_FORMAT_DEFAULT)) == nullptr) {
+            availableDevices = availablePrimaryDevices;
         }
 
         switch (getForceUse(AUDIO_POLICY_FORCE_FOR_COMMUNICATION)) {
         case AUDIO_POLICY_FORCE_BT_SCO:
             // if SCO device is requested but no SCO device is available, fall back to default case
-            if (availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-                device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
+            device = availableDevices.getDevice(
+                    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+            if (device != nullptr) {
                 break;
             }
             FALLTHROUGH_INTENDED;
 
         default:    // FORCE_NONE
-            if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-                device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-                device = AUDIO_DEVICE_IN_USB_HEADSET;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-                device = AUDIO_DEVICE_IN_USB_DEVICE;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-                device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-            }
+            device = availableDevices.getFirstExistingDevice({
+                    AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                    AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
             break;
 
         case AUDIO_POLICY_FORCE_SPEAKER:
-            if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
-                device = AUDIO_DEVICE_IN_BACK_MIC;
-            } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-                device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-            }
+            device = availableDevices.getFirstExistingDevice({
+                    AUDIO_DEVICE_IN_BACK_MIC, AUDIO_DEVICE_IN_BUILTIN_MIC});
             break;
         }
         break;
@@ -591,77 +545,60 @@
     case AUDIO_SOURCE_UNPROCESSED:
     case AUDIO_SOURCE_HOTWORD:
         if (inputSource == AUDIO_SOURCE_HOTWORD) {
-            availableDeviceTypes = availablePrimaryDeviceTypes;
+            availableDevices = availablePrimaryDevices;
         }
-        if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO &&
-                availableDeviceTypes & AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET) {
-            device = AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-            device = AUDIO_DEVICE_IN_USB_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
+        if (getForceUse(AUDIO_POLICY_FORCE_FOR_RECORD) == AUDIO_POLICY_FORCE_BT_SCO) {
+            device = availableDevices.getDevice(
+                    AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, String8(""), AUDIO_FORMAT_DEFAULT);
+            if (device != nullptr) break;
         }
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
         break;
     case AUDIO_SOURCE_CAMCORDER:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_BACK_MIC) {
-            device = AUDIO_DEVICE_IN_BACK_MIC;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-            // This is specifically for a device without built-in mic
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        }
+        // For a device without built-in mic, adding usb device
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_BACK_MIC, AUDIO_DEVICE_IN_BUILTIN_MIC,
+                AUDIO_DEVICE_IN_USB_DEVICE});
         break;
     case AUDIO_SOURCE_VOICE_DOWNLINK:
     case AUDIO_SOURCE_VOICE_CALL:
     case AUDIO_SOURCE_VOICE_UPLINK:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_VOICE_CALL) {
-            device = AUDIO_DEVICE_IN_VOICE_CALL;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_VOICE_CALL, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     case AUDIO_SOURCE_VOICE_PERFORMANCE:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_WIRED_HEADSET) {
-            device = AUDIO_DEVICE_IN_WIRED_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_HEADSET) {
-            device = AUDIO_DEVICE_IN_USB_HEADSET;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_USB_DEVICE) {
-            device = AUDIO_DEVICE_IN_USB_DEVICE;
-        } else if (availableDeviceTypes & AUDIO_DEVICE_IN_BUILTIN_MIC) {
-            device = AUDIO_DEVICE_IN_BUILTIN_MIC;
-        }
+        device = availableDevices.getFirstExistingDevice({
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_IN_USB_HEADSET,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_IN_BUILTIN_MIC});
         break;
     case AUDIO_SOURCE_REMOTE_SUBMIX:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_REMOTE_SUBMIX) {
-            device = AUDIO_DEVICE_IN_REMOTE_SUBMIX;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_REMOTE_SUBMIX, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     case AUDIO_SOURCE_FM_TUNER:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_FM_TUNER) {
-            device = AUDIO_DEVICE_IN_FM_TUNER;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_FM_TUNER, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     case AUDIO_SOURCE_ECHO_REFERENCE:
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_ECHO_REFERENCE) {
-            device = AUDIO_DEVICE_IN_ECHO_REFERENCE;
-        }
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_ECHO_REFERENCE, String8(""), AUDIO_FORMAT_DEFAULT);
         break;
     default:
         ALOGW("getDeviceForInputSource() invalid input source %d", inputSource);
         break;
     }
-    if (device == AUDIO_DEVICE_NONE) {
+    if (device == nullptr) {
         ALOGV("getDeviceForInputSource() no device found for source %d", inputSource);
-        if (availableDeviceTypes & AUDIO_DEVICE_IN_STUB) {
-            device = AUDIO_DEVICE_IN_STUB;
-        }
-        ALOGE_IF(device == AUDIO_DEVICE_NONE,
+        device = availableDevices.getDevice(
+                AUDIO_DEVICE_IN_STUB, String8(""), AUDIO_FORMAT_DEFAULT);
+        ALOGE_IF(device == nullptr,
                  "getDeviceForInputSource() no default device defined");
     }
-    ALOGV("getDeviceForInputSource()input source %d, device %08x", inputSource, device);
+    ALOGV_IF(device != nullptr,
+             "getDeviceForInputSource()input source %d, device %08x",
+             inputSource, device->type());
     return device;
 }
 
@@ -684,11 +621,9 @@
 
     auto legacyStrategy = mLegacyStrategyMap.find(strategy) != end(mLegacyStrategyMap) ?
                 mLegacyStrategyMap.at(strategy) : STRATEGY_NONE;
-    audio_devices_t devices = getDeviceForStrategyInt(legacyStrategy,
-                                                      availableOutputDevices,
-                                                      availableInputDevices, outputs,
-                                                      (uint32_t)AUDIO_DEVICE_NONE);
-    return availableOutputDevices.getDevicesFromTypeMask(devices);
+    return getDevicesForStrategyInt(legacyStrategy,
+                                    availableOutputDevices,
+                                    availableInputDevices, outputs);
 }
 
 DeviceVector Engine::getOutputDevicesForAttributes(const audio_attributes_t &attributes,
@@ -747,27 +682,25 @@
     if (device != nullptr) {
         return device;
     }
-    audio_devices_t deviceType = getDeviceForInputSource(attr.source);
 
-    if (audio_is_remote_submix_device(deviceType)) {
-        address = "0";
-        std::size_t pos;
-        std::string tags { attr.tags };
-        if ((pos = tags.find("addr=")) != std::string::npos) {
-            address = tags.substr(pos + std::strlen("addr="));
-        }
+    device = getDeviceForInputSource(attr.source);
+    if (device == nullptr || !audio_is_remote_submix_device(device->type())) {
+        // Return immediately if the device is null or it is not a remote submix device.
+        return device;
     }
-    return availableInputDevices.getDevice(deviceType,
+
+    // For remote submix device, try to find the device by address.
+    address = "0";
+    std::size_t pos;
+    std::string tags { attr.tags };
+    if ((pos = tags.find("addr=")) != std::string::npos) {
+        address = tags.substr(pos + std::strlen("addr="));
+    }
+    return availableInputDevices.getDevice(device->type(),
                                            String8(address.c_str()),
                                            AUDIO_FORMAT_DEFAULT);
 }
 
-template <>
-AudioPolicyManagerInterface *Engine::queryInterface()
-{
-    return this;
-}
-
 } // namespace audio_policy
 } // namespace android
 
diff --git a/services/audiopolicy/enginedefault/src/Engine.h b/services/audiopolicy/enginedefault/src/Engine.h
index d5dfacc..4360c6f 100644
--- a/services/audiopolicy/enginedefault/src/Engine.h
+++ b/services/audiopolicy/enginedefault/src/Engine.h
@@ -17,8 +17,7 @@
 #pragma once
 
 #include "EngineBase.h"
-#include "AudioPolicyManagerInterface.h"
-#include <AudioGain.h>
+#include "EngineInterface.h"
 #include <policy.h>
 
 namespace android
@@ -48,12 +47,9 @@
     Engine();
     virtual ~Engine() = default;
 
-    template <class RequestedInterface>
-    RequestedInterface *queryInterface();
-
 private:
     ///
-    /// from EngineBase, so from AudioPolicyManagerInterface
+    /// from EngineBase, so from EngineInterface
     ///
     status_t setForceUse(audio_policy_force_use_t usage,
                          audio_policy_forced_cfg_t config) override;
@@ -77,15 +73,14 @@
 
     status_t setDefaultDevice(audio_devices_t device);
 
-    audio_devices_t getDeviceForStrategyInt(legacy_strategy strategy,
-                                            DeviceVector availableOutputDevices,
-                                            DeviceVector availableInputDevices,
-                                            const SwAudioOutputCollection &outputs,
-                                            uint32_t outputDeviceTypesToIgnore) const;
+    DeviceVector getDevicesForStrategyInt(legacy_strategy strategy,
+                                          DeviceVector availableOutputDevices,
+                                          DeviceVector availableInputDevices,
+                                          const SwAudioOutputCollection &outputs) const;
 
     DeviceVector getDevicesForProductStrategy(product_strategy_t strategy) const;
 
-    audio_devices_t getDeviceForInputSource(audio_source_t inputSource) const;
+    sp<DeviceDescriptor> getDeviceForInputSource(audio_source_t inputSource) const;
 
     DeviceStrategyMap mDevicesForStrategies;
 
diff --git a/services/audiopolicy/enginedefault/src/EngineInstance.cpp b/services/audiopolicy/enginedefault/src/EngineInstance.cpp
index 17e9832..eeb3758 100644
--- a/services/audiopolicy/enginedefault/src/EngineInstance.cpp
+++ b/services/audiopolicy/enginedefault/src/EngineInstance.cpp
@@ -14,41 +14,21 @@
  * limitations under the License.
  */
 
-#include <AudioPolicyManagerInterface.h>
-#include "AudioPolicyEngineInstance.h"
+#include <EngineInterface.h>
 #include "Engine.h"
 
-namespace android
-{
-namespace audio_policy
-{
+namespace android {
+namespace audio_policy {
 
-EngineInstance::EngineInstance()
+extern "C" EngineInterface* createEngineInstance()
 {
+    return new (std::nothrow) Engine();
 }
 
-EngineInstance *EngineInstance::getInstance()
+extern "C" void destroyEngineInstance(EngineInterface *engine)
 {
-    static EngineInstance instance;
-    return &instance;
-}
-
-EngineInstance::~EngineInstance()
-{
-}
-
-Engine *EngineInstance::getEngine() const
-{
-    static Engine engine;
-    return &engine;
-}
-
-template <>
-AudioPolicyManagerInterface *EngineInstance::queryInterface() const
-{
-    return getEngine()->queryInterface<AudioPolicyManagerInterface>();
+    delete static_cast<Engine*>(engine);
 }
 
 } // namespace audio_policy
 } // namespace android
-
diff --git a/services/audiopolicy/manager/AudioPolicyFactory.cpp b/services/audiopolicy/manager/AudioPolicyFactory.cpp
index 7aff6a9..476a1ec 100644
--- a/services/audiopolicy/manager/AudioPolicyFactory.cpp
+++ b/services/audiopolicy/manager/AudioPolicyFactory.cpp
@@ -21,7 +21,13 @@
 extern "C" AudioPolicyInterface* createAudioPolicyManager(
         AudioPolicyClientInterface *clientInterface)
 {
-    return new AudioPolicyManager(clientInterface);
+    AudioPolicyManager *apm = new AudioPolicyManager(clientInterface);
+    status_t status = apm->initialize();
+    if (status != NO_ERROR) {
+        delete apm;
+        apm = nullptr;
+    }
+    return apm;
 }
 
 extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface)
diff --git a/services/audiopolicy/managerdefault/Android.bp b/services/audiopolicy/managerdefault/Android.bp
new file mode 100644
index 0000000..1fa0d19
--- /dev/null
+++ b/services/audiopolicy/managerdefault/Android.bp
@@ -0,0 +1,44 @@
+cc_library_shared {
+    name: "libaudiopolicymanagerdefault",
+
+    srcs: [
+        "AudioPolicyManager.cpp",
+        "EngineLibrary.cpp",
+    ],
+
+    export_include_dirs: ["."],
+
+    shared_libs: [
+        "libaudiofoundation",
+        "libcutils",
+        "libdl",
+        "libutils",
+        "liblog",
+        "libaudiopolicy",
+        "libsoundtrigger",
+        "libmedia_helper",
+        "libmediametrics",
+        "libbinder",
+        "libhidlbase",
+        "libxml2",
+        // The default audio policy engine is always present in the system image.
+        // libaudiopolicyengineconfigurable can be built in addition by specifying
+        // a dependency on it in the device makefile. There will be no build time
+        // conflict with libaudiopolicyenginedefault.
+        "libaudiopolicyenginedefault",
+    ],
+
+    header_libs: [
+        "libaudiopolicycommon",
+        "libaudiopolicyengine_interface_headers",
+        "libaudiopolicymanager_interface_headers",
+    ],
+
+    static_libs: ["libaudiopolicycomponents"],
+
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+
+}
diff --git a/services/audiopolicy/managerdefault/Android.mk b/services/audiopolicy/managerdefault/Android.mk
deleted file mode 100644
index 684fc9f..0000000
--- a/services/audiopolicy/managerdefault/Android.mk
+++ /dev/null
@@ -1,56 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= AudioPolicyManager.cpp
-
-LOCAL_EXPORT_C_INCLUDE_DIRS := $(LOCAL_PATH)
-
-LOCAL_SHARED_LIBRARIES := \
-    libcutils \
-    libutils \
-    liblog \
-    libaudiopolicy \
-    libsoundtrigger
-
-ifeq ($(USE_CONFIGURABLE_AUDIO_POLICY), 1)
-
-ifneq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-$(error Configurable policy does not support legacy conf file)
-endif #ifneq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-
-LOCAL_SHARED_LIBRARIES += libaudiopolicyengineconfigurable
-
-else
-
-LOCAL_SHARED_LIBRARIES += libaudiopolicyenginedefault
-
-endif # ifeq ($(USE_CONFIGURABLE_AUDIO_POLICY), 1)
-
-LOCAL_C_INCLUDES += \
-    $(call include-path-for, audio-utils)
-
-LOCAL_HEADER_LIBRARIES := \
-    libaudiopolicycommon \
-    libaudiopolicyengine_interface_headers \
-    libaudiopolicymanager_interface_headers
-
-LOCAL_STATIC_LIBRARIES := \
-    libaudiopolicycomponents
-
-LOCAL_SHARED_LIBRARIES += libmedia_helper
-LOCAL_SHARED_LIBRARIES += libmediametrics
-
-LOCAL_SHARED_LIBRARIES += libbinder libhidlbase libxml2
-
-ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-LOCAL_CFLAGS += -DUSE_XML_AUDIO_POLICY_CONF
-endif #ifeq ($(USE_XML_AUDIO_POLICY_CONF), 1)
-
-LOCAL_CFLAGS += -Wall -Werror
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_MODULE:= libaudiopolicymanagerdefault
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index c048de3..1f8ceec 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -42,15 +42,12 @@
 #include <set>
 #include <unordered_set>
 #include <vector>
-#include <AudioPolicyManagerInterface.h>
-#include <AudioPolicyEngineInstance.h>
 #include <cutils/properties.h>
 #include <utils/Log.h>
 #include <media/AudioParameter.h>
 #include <private/android_filesystem_config.h>
 #include <soundtrigger/SoundTrigger.h>
 #include <system/audio.h>
-#include <audio_policy_conf.h>
 #include "AudioPolicyManager.h"
 #include <Serializer.h>
 #include "TypeConverter.h"
@@ -76,6 +73,26 @@
         AUDIO_CHANNEL_OUT_2POINT1POINT2, AUDIO_CHANNEL_OUT_2POINT0POINT2,
         AUDIO_CHANNEL_OUT_5POINT1, AUDIO_CHANNEL_OUT_STEREO }};
 
+template <typename T>
+bool operator== (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+    if (left.size() != right.size()) {
+        return false;
+    }
+    for (size_t index = 0; index < right.size(); index++) {
+        if (left[index] != right[index]) {
+            return false;
+        }
+    }
+    return true;
+}
+
+template <typename T>
+bool operator!= (const SortedVector<T> &left, const SortedVector<T> &right)
+{
+    return !(left == right);
+}
+
 // ----------------------------------------------------------------------------
 // AudioPolicyInterface implementation
 // ----------------------------------------------------------------------------
@@ -97,7 +114,7 @@
 {
     AudioParameter param(device->address());
     const String8 key(state == AUDIO_POLICY_DEVICE_STATE_AVAILABLE ?
-                AudioParameter::keyStreamConnect : AudioParameter::keyStreamDisconnect);
+                AudioParameter::keyDeviceConnect : AudioParameter::keyDeviceDisconnect);
     param.addInt(key, device->type());
     mpClientInterface->setParameters(AUDIO_IO_HANDLE_NONE, param.toString());
 }
@@ -475,6 +492,10 @@
     std::unordered_set<audio_format_t> formatSet;
     sp<HwModule> primaryModule =
             mHwModules.getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY);
+    if (primaryModule == nullptr) {
+        ALOGE("%s() unable to get primary module", __func__);
+        return NO_INIT;
+    }
     DeviceVector declaredDevices = primaryModule->getDeclaredDevices().getDevicesFromTypeMask(
             AUDIO_DEVICE_OUT_ALL_A2DP);
     for (const auto& device : declaredDevices) {
@@ -839,7 +860,7 @@
         // if explicitly requested
         static const uint32_t kRelevantFlags =
                 (AUDIO_OUTPUT_FLAG_HW_AV_SYNC | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
-                 AUDIO_OUTPUT_FLAG_VOIP_RX);
+                 AUDIO_OUTPUT_FLAG_VOIP_RX | AUDIO_OUTPUT_FLAG_MMAP_NOIRQ);
         flags =
             (audio_output_flags_t)((flags & kRelevantFlags) | AUDIO_OUTPUT_FLAG_DIRECT);
     }
@@ -1184,9 +1205,9 @@
             if (!desc->isDuplicated() && (profile == desc->mProfile)) {
                 // reuse direct output if currently open by the same client
                 // and configured with same parameters
-                if ((config->sample_rate == desc->mSamplingRate) &&
-                    (config->format == desc->mFormat) &&
-                    (channelMask == desc->mChannelMask) &&
+                if ((config->sample_rate == desc->getSamplingRate()) &&
+                    (config->format == desc->getFormat()) &&
+                    (channelMask == desc->getChannelMask()) &&
                     (session == desc->mDirectClientSession)) {
                     desc->mDirectOpenCount++;
                     ALOGI("%s reusing direct output %d for session %d", __func__, 
@@ -1226,13 +1247,13 @@
 
         // only accept an output with the requested parameters
         if (status != NO_ERROR ||
-            (config->sample_rate != 0 && config->sample_rate != outputDesc->mSamplingRate) ||
-            (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->mFormat) ||
-            (channelMask != 0 && channelMask != outputDesc->mChannelMask)) {
+            (config->sample_rate != 0 && config->sample_rate != outputDesc->getSamplingRate()) ||
+            (config->format != AUDIO_FORMAT_DEFAULT && config->format != outputDesc->getFormat()) ||
+            (channelMask != 0 && channelMask != outputDesc->getChannelMask())) {
             ALOGV("%s failed opening direct output: output %d sample rate %d %d," 
                     "format %d %d, channel mask %04x %04x", __func__, output, config->sample_rate,
-                    outputDesc->mSamplingRate, config->format, outputDesc->mFormat,
-                    channelMask, outputDesc->mChannelMask);
+                    outputDesc->getSamplingRate(), config->format, outputDesc->getFormat(),
+                    channelMask, outputDesc->getChannelMask());
             if (output != AUDIO_IO_HANDLE_NONE) {
                 outputDesc->close();
             }
@@ -1338,13 +1359,13 @@
     // Each IOProfile represents a MixPort from audio_policy_configuration.xml
     for (const auto &inProfile : inputProfiles) {
         if (hwAvSync == ((inProfile->getFlags() & AUDIO_INPUT_FLAG_HW_AV_SYNC) != 0)) {
-            msdProfiles.appendVector(inProfile->getAudioProfiles());
+            msdProfiles.appendProfiles(inProfile->getAudioProfiles());
         }
     }
     AudioProfileVector deviceProfiles;
     for (const auto &outProfile : outputProfiles) {
         if (hwAvSync == ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0)) {
-            deviceProfiles.appendVector(outProfile->getAudioProfiles());
+            deviceProfiles.appendProfiles(outProfile->getAudioProfiles());
         }
     }
     struct audio_config_base bestSinkConfig;
@@ -1503,13 +1524,13 @@
         // If haptic channel is specified, use the haptic output if present.
         // When using haptic output, same audio format and sample rate are required.
         const uint32_t outputHapticChannelCount = audio_channel_count_from_out_mask(
-            outputDesc->mChannelMask & AUDIO_CHANNEL_HAPTIC_ALL);
+            outputDesc->getChannelMask() & AUDIO_CHANNEL_HAPTIC_ALL);
         if ((hapticChannelCount == 0) != (outputHapticChannelCount == 0)) {
             continue;
         }
         if (outputHapticChannelCount >= hapticChannelCount
-            && format == outputDesc->mFormat
-            && samplingRate == outputDesc->mSamplingRate) {
+            && format == outputDesc->getFormat()
+            && samplingRate == outputDesc->getSamplingRate()) {
                 currentMatchCriteria[0] = outputHapticChannelCount;
         }
 
@@ -1517,12 +1538,13 @@
         currentMatchCriteria[1] = popcount(outputDesc->mFlags & functionalFlags);
 
         // channel mask and channel count match
-        uint32_t outputChannelCount = audio_channel_count_from_out_mask(outputDesc->mChannelMask);
+        uint32_t outputChannelCount = audio_channel_count_from_out_mask(
+                outputDesc->getChannelMask());
         if (channelMask != AUDIO_CHANNEL_NONE && channelCount > 2 &&
             channelCount <= outputChannelCount) {
             if ((audio_channel_mask_get_representation(channelMask) ==
-                    audio_channel_mask_get_representation(outputDesc->mChannelMask)) &&
-                    ((channelMask & outputDesc->mChannelMask) == channelMask)) {
+                    audio_channel_mask_get_representation(outputDesc->getChannelMask())) &&
+                    ((channelMask & outputDesc->getChannelMask()) == channelMask)) {
                 currentMatchCriteria[2] = outputChannelCount;
             }
             currentMatchCriteria[3] = outputChannelCount;
@@ -1530,8 +1552,8 @@
 
         // sampling rate match
         if (samplingRate > SAMPLE_RATE_HZ_DEFAULT &&
-                samplingRate <= outputDesc->mSamplingRate) {
-            currentMatchCriteria[4] = outputDesc->mSamplingRate;
+                samplingRate <= outputDesc->getSamplingRate()) {
+            currentMatchCriteria[4] = outputDesc->getSamplingRate();
         }
 
         // performance flags match
@@ -1541,7 +1563,7 @@
         if (format != AUDIO_FORMAT_INVALID) {
             currentMatchCriteria[6] =
                 AudioPort::kFormatDistanceMax -
-                AudioPort::formatDistance(format, outputDesc->mFormat);
+                AudioPort::formatDistance(format, outputDesc->getFormat());
         }
 
         // primary output match
@@ -2239,16 +2261,22 @@
         return status;
     }
 
-  // increment activity count before calling getNewInputDevice() below as only active sessions
+    // increment activity count before calling getNewInputDevice() below as only active sessions
     // are considered for device selection
     inputDesc->setClientActive(client, true);
 
     // indicate active capture to sound trigger service if starting capture from a mic on
     // primary HW module
     sp<DeviceDescriptor> device = getNewInputDevice(inputDesc);
-    setInputDevice(input, device, true /* force */);
+    if (device != nullptr) {
+        status = setInputDevice(input, device, true /* force */);
+    } else {
+        ALOGW("%s no new input device can be found for descriptor %d",
+                __FUNCTION__, inputDesc->getId());
+        status = BAD_VALUE;
+    }
 
-    if (inputDesc->activeCount()  == 1) {
+    if (status == NO_ERROR && inputDesc->activeCount() == 1) {
         sp<AudioPolicyMix> policyMix = inputDesc->mPolicyMix.promote();
         // if input maps to a dynamic policy with an activity listener, notify of state change
         if ((policyMix != NULL)
@@ -2279,11 +2307,16 @@
                         address, "remote-submix", AUDIO_FORMAT_DEFAULT);
             }
         }
+    } else if (status != NO_ERROR) {
+        // Restore client activity state.
+        inputDesc->setClientActive(client, false);
+        inputDesc->stop();
     }
 
-    ALOGV("%s input %d source = %d exit", __FUNCTION__, input, client->source());
+    ALOGV("%s input %d source = %d status = %d exit",
+            __FUNCTION__, input, client->source(), status);
 
-    return NO_ERROR;
+    return status;
 }
 
 status_t AudioPolicyManager::stopInput(audio_port_handle_t portId)
@@ -2889,9 +2922,9 @@
             // stereo and let audio flinger do the channel conversion if needed.
             outputConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
             inputConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
-            rSubmixModule->addOutputProfile(address, &outputConfig,
+            rSubmixModule->addOutputProfile(address.c_str(), &outputConfig,
                     AUDIO_DEVICE_OUT_REMOTE_SUBMIX, address);
-            rSubmixModule->addInputProfile(address, &inputConfig,
+            rSubmixModule->addInputProfile(address.c_str(), &inputConfig,
                     AUDIO_DEVICE_IN_REMOTE_SUBMIX, address);
 
             if ((res = setDeviceConnectionStateInt(deviceTypeToMakeAvailable,
@@ -2987,8 +3020,8 @@
                     }
                 }
             }
-            rSubmixModule->removeOutputProfile(address);
-            rSubmixModule->removeInputProfile(address);
+            rSubmixModule->removeOutputProfile(address.c_str());
+            rSubmixModule->removeInputProfile(address.c_str());
 
         } else if ((mix.mRouteFlags & MIX_ROUTE_FLAG_RENDER) == MIX_ROUTE_FLAG_RENDER) {
             if (mPolicyMixes.unregisterMix(mix) != NO_ERROR) {
@@ -3200,7 +3233,7 @@
     ALOGV("%s() profile %sfound with name: %s, "
         "sample rate: %u, format: 0x%x, channel_mask: 0x%x, output flags: 0x%x",
         __FUNCTION__, profile != 0 ? "" : "NOT ",
-        (profile != 0 ? profile->getTagName().string() : "null"),
+        (profile != 0 ? profile->getTagName().c_str() : "null"),
         config.sample_rate, config.format, config.channel_mask, output_flags);
     return (profile != 0);
 }
@@ -3864,7 +3897,7 @@
     if (srcDevice->hasSameHwModuleAs(sinkDevice) &&
             srcDevice->getModuleVersionMajor() >= 3 &&
             sinkDevice->getModule()->supportsPatch(srcDevice, sinkDevice) &&
-            srcDevice->getAudioPort()->mGains.size() > 0) {
+            srcDevice->getAudioPort()->getGains().size() > 0) {
         ALOGV("%s Device to Device route supported by >=3.0 HAL", __FUNCTION__);
         // TODO: may explicitly specify whether we should use HW or SW patch
         //   create patch between src device and output device
@@ -4104,7 +4137,7 @@
     for (size_t i = 0; i < hdmiOutputDevices.size(); i++) {
         // Simulate reconnection to update enabled surround sound formats.
         String8 address = hdmiOutputDevices[i]->address();
-        String8 name = hdmiOutputDevices[i]->getName();
+        std::string name = hdmiOutputDevices[i]->getName();
         status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_OUT_HDMI,
                                                       AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
                                                       address.c_str(),
@@ -4126,7 +4159,7 @@
     for (size_t i = 0; i < hdmiInputDevices.size(); i++) {
         // Simulate reconnection to update enabled surround sound formats.
         String8 address = hdmiInputDevices[i]->address();
-        String8 name = hdmiInputDevices[i]->getName();
+        std::string name = hdmiInputDevices[i]->getName();
         status_t status = setDeviceConnectionStateInt(AUDIO_DEVICE_IN_HDMI,
                                                       AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
                                                       address.c_str(),
@@ -4151,11 +4184,11 @@
     return profileUpdated ? NO_ERROR : INVALID_OPERATION;
 }
 
-void AudioPolicyManager::setAppState(uid_t uid, app_state_t state)
+void AudioPolicyManager::setAppState(audio_port_handle_t portId, app_state_t state)
 {
-    ALOGV("%s(uid:%d, state:%d)", __func__, uid, state);
+    ALOGV("%s(portId:%d, state:%d)", __func__, portId, state);
     for (size_t i = 0; i < mInputs.size(); i++) {
-        mInputs.valueAt(i)->setAppState(uid, state);
+        mInputs.valueAt(i)->setAppState(portId, state);
     }
 }
 
@@ -4294,17 +4327,8 @@
         : AudioPolicyManager(clientInterface, false /*forTesting*/)
 {
     loadConfig();
-    initialize();
 }
 
-//  This check is to catch any legacy platform updating to Q without having
-//  switched to XML since its deprecation on O.
-// TODO: after Q release, remove this check and flag as XML is now the only
-//        option and all legacy platform should have transitioned to XML.
-#ifndef USE_XML_AUDIO_POLICY_CONF
-#error Audio policy no longer supports legacy .conf configuration format
-#endif
-
 void AudioPolicyManager::loadConfig() {
     if (deserializeAudioPolicyXmlConfig(getConfig()) != NO_ERROR) {
         ALOGE("could not load audio policy configuration file, setting defaults");
@@ -4313,17 +4337,18 @@
 }
 
 status_t AudioPolicyManager::initialize() {
-    // Once policy config has been parsed, retrieve an instance of the engine and initialize it.
-    audio_policy::EngineInstance *engineInstance = audio_policy::EngineInstance::getInstance();
-    if (!engineInstance) {
-        ALOGE("%s:  Could not get an instance of policy engine", __FUNCTION__);
-        return NO_INIT;
-    }
-    // Retrieve the Policy Manager Interface
-    mEngine = engineInstance->queryInterface<AudioPolicyManagerInterface>();
-    if (mEngine == NULL) {
-        ALOGE("%s: Failed to get Policy Engine Interface", __FUNCTION__);
-        return NO_INIT;
+    {
+        auto engLib = EngineLibrary::load(
+                        "libaudiopolicyengine" + getConfig().getEngineLibraryNameSuffix() + ".so");
+        if (!engLib) {
+            ALOGE("%s: Failed to load the engine library", __FUNCTION__);
+            return NO_INIT;
+        }
+        mEngine = engLib->createEngine();
+        if (mEngine == nullptr) {
+            ALOGE("%s: Failed to instantiate the APM engine", __FUNCTION__);
+            return NO_INIT;
+        }
     }
     mEngine->setObserver(this);
     status_t status = mEngine->initCheck();
@@ -4617,7 +4642,7 @@
             }
 
             ALOGV("opening output for device %08x with params %s profile %p name %s",
-                  deviceType, address.string(), profile.get(), profile->getName().string());
+                  deviceType, address.string(), profile.get(), profile->getName().c_str());
             desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
             status_t status = desc->open(nullptr, DeviceVector(device),
@@ -5039,10 +5064,12 @@
     // also take into account external policy-related changes: add all outputs which are
     // associated with policies in the "before" and "after" output vectors
     ALOGVV("%s(): policy related outputs", __func__);
+    bool hasDynamicPolicy = false;
     for (size_t i = 0 ; i < mPreviousOutputs.size() ; i++) {
         const sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueAt(i);
         if (desc != 0 && desc->mPolicyMix != NULL) {
             srcOutputs.add(desc->mIoHandle);
+            hasDynamicPolicy = true;
             ALOGVV(" previous outputs: adding %d", desc->mIoHandle);
         }
     }
@@ -5050,6 +5077,7 @@
         const sp<SwAudioOutputDescriptor> desc = mOutputs.valueAt(i);
         if (desc != 0 && desc->mPolicyMix != NULL) {
             dstOutputs.add(desc->mIoHandle);
+            hasDynamicPolicy = true;
             ALOGVV(" new outputs: adding %d", desc->mIoHandle);
         }
     }
@@ -5058,12 +5086,45 @@
         // get maximum latency of all source outputs to determine the minimum mute time guaranteeing
         // audio from invalidated tracks will be rendered when unmuting
         uint32_t maxLatency = 0;
+        bool invalidate = hasDynamicPolicy;
         for (audio_io_handle_t srcOut : srcOutputs) {
             sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueFor(srcOut);
-            if (desc != 0 && maxLatency < desc->latency()) {
+            if (desc == nullptr) continue;
+
+            if (desc->isStrategyActive(psId) && maxLatency < desc->latency()) {
                 maxLatency = desc->latency();
             }
+
+            if (invalidate) continue;
+
+            for (auto client : desc->clientsList(false /*activeOnly*/)) {
+                if (!desc->mProfile->isDirectOutput()) {
+                    // a client on a non direct outputs has necessarily a linear PCM format
+                    // so we can call selectOutput() safely
+                    const audio_io_handle_t newOutput = selectOutput(dstOutputs,
+                                                                     client->flags(),
+                                                                     client->config().format,
+                                                                     client->config().channel_mask,
+                                                                     client->config().sample_rate);
+                    if (newOutput != srcOut) {
+                        invalidate = true;
+                        break;
+                    }
+                } else {
+                    sp<IOProfile> profile = getProfileForOutput(newDevices,
+                                   client->config().sample_rate,
+                                   client->config().format,
+                                   client->config().channel_mask,
+                                   client->flags(),
+                                   true /* directOnly */);
+                    if (profile != desc->mProfile) {
+                        invalidate = true;
+                        break;
+                    }
+                }
+            }
         }
+
         ALOGV_IF(!(srcOutputs.isEmpty() || dstOutputs.isEmpty()),
               "%s: strategy %d, moving from output %s to output %s", __func__, psId,
               std::to_string(srcOutputs[0]).c_str(),
@@ -5071,7 +5132,9 @@
         // mute strategy while moving tracks from one output to another
         for (audio_io_handle_t srcOut : srcOutputs) {
             sp<SwAudioOutputDescriptor> desc = mPreviousOutputs.valueFor(srcOut);
-            if (desc != 0 && desc->isStrategyActive(psId)) {
+            if (desc == nullptr) continue;
+
+            if (desc->isStrategyActive(psId)) {
                 setStrategyMute(psId, true, desc);
                 setStrategyMute(psId, false, desc, maxLatency * LATENCY_MUTE_FACTOR,
                                 newDevices.types());
@@ -5087,8 +5150,10 @@
             selectOutputForMusicEffects();
         }
         // Move tracks associated to this stream (and linked) from previous output to new output
-        for (auto stream :  mEngine->getStreamTypesForProductStrategy(psId)) {
-            mpClientInterface->invalidateStream(stream);
+        if (invalidate) {
+            for (auto stream :  mEngine->getStreamTypesForProductStrategy(psId)) {
+                mpClientInterface->invalidateStream(stream);
+            }
         }
     }
 }
@@ -6080,24 +6145,24 @@
         formatSet.insert(enforcedSurround.begin(), enforcedSurround.end());
     }
     for (const auto& format : formatSet) {
-        formatsPtr->push(format);
+        formatsPtr->push_back(format);
     }
 }
 
-void AudioPolicyManager::modifySurroundChannelMasks(ChannelsVector *channelMasksPtr) {
-    ChannelsVector &channelMasks = *channelMasksPtr;
+void AudioPolicyManager::modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr) {
+    ChannelMaskSet &channelMasks = *channelMasksPtr;
     audio_policy_forced_cfg_t forceUse = mEngine->getForceUse(
             AUDIO_POLICY_FORCE_FOR_ENCODED_SURROUND);
 
     // If NEVER, then remove support for channelMasks > stereo.
     if (forceUse == AUDIO_POLICY_FORCE_ENCODED_SURROUND_NEVER) {
-        for (size_t maskIndex = 0; maskIndex < channelMasks.size(); ) {
-            audio_channel_mask_t channelMask = channelMasks[maskIndex];
+        for (auto it = channelMasks.begin(); it != channelMasks.end();) {
+            audio_channel_mask_t channelMask = *it;
             if (channelMask & ~AUDIO_CHANNEL_OUT_STEREO) {
                 ALOGI("%s: force NEVER, so remove channelMask 0x%08x", __FUNCTION__, channelMask);
-                channelMasks.removeAt(maskIndex);
+                it = channelMasks.erase(it);
             } else {
-                maskIndex++;
+                ++it;
             }
         }
     // If ALWAYS or MANUAL, then make sure we at least support 5.1
@@ -6113,7 +6178,7 @@
         }
         // If not then add 5.1 support.
         if (!supports5dot1) {
-            channelMasks.add(AUDIO_CHANNEL_OUT_5POINT1);
+            channelMasks.insert(AUDIO_CHANNEL_OUT_5POINT1);
             ALOGI("%s: force MANUAL or ALWAYS, so adding channelMask for 5.1 surround", __func__);
         }
     }
@@ -6146,8 +6211,8 @@
     }
 
     for (audio_format_t format : profiles.getSupportedFormats()) {
-        ChannelsVector channelMasks;
-        SampleRateVector samplingRates;
+        ChannelMaskSet channelMasks;
+        SampleRateSet samplingRates;
         AudioParameter requestedParameters;
         requestedParameters.addInt(String8(AudioParameter::keyFormat), format);
 
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 612bd8f..9d97195 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -34,13 +34,11 @@
 #include <media/PatchBuilder.h>
 #include "AudioPolicyInterface.h"
 
-#include <AudioPolicyManagerInterface.h>
 #include <AudioPolicyManagerObserver.h>
-#include <AudioGain.h>
 #include <AudioPolicyConfig.h>
 #include <AudioPort.h>
 #include <AudioPatch.h>
-#include <AudioProfile.h>
+#include <AudioProfileVector.h>
 #include <DeviceDescriptor.h>
 #include <IOProfile.h>
 #include <HwModule.h>
@@ -49,6 +47,7 @@
 #include <AudioPolicyMix.h>
 #include <EffectDescriptor.h>
 #include <SoundTriggerSession.h>
+#include "EngineLibrary.h"
 #include "TypeConverter.h"
 
 namespace android {
@@ -279,7 +278,7 @@
         virtual status_t getHwOffloadEncodingFormatsSupportedForA2DP(
                     std::vector<audio_format_t> *formats);
 
-        virtual void setAppState(uid_t uid, app_state_t state);
+        virtual void setAppState(audio_port_handle_t portId, app_state_t state);
 
         virtual bool isHapticPlaybackSupported();
 
@@ -307,6 +306,8 @@
             return volumeGroup != VOLUME_GROUP_NONE ? NO_ERROR : BAD_VALUE;
         }
 
+        status_t initialize();
+
 protected:
         // A constructor that allows more fine-grained control over initialization process,
         // used in automatic tests.
@@ -321,7 +322,6 @@
         //   - initialize.
         AudioPolicyConfig& getConfig() { return mConfig; }
         void loadConfig();
-        status_t initialize();
 
         // From AudioPolicyManagerObserver
         virtual const AudioPatchCollection &getAudioPatches() const
@@ -752,7 +752,7 @@
         uint32_t nextAudioPortGeneration();
 
         // Audio Policy Engine Interface.
-        AudioPolicyManagerInterface *mEngine;
+        EngineInstance mEngine;
 
         // Surround formats that are enabled manually. Taken into account when
         // "encoded surround" is forced into "manual" mode.
@@ -762,7 +762,7 @@
 private:
         // Add or remove AC3 DTS encodings based on user preferences.
         void modifySurroundFormats(const sp<DeviceDescriptor>& devDesc, FormatVector *formatsPtr);
-        void modifySurroundChannelMasks(ChannelsVector *channelMasksPtr);
+        void modifySurroundChannelMasks(ChannelMaskSet *channelMasksPtr);
 
         // Support for Multi-Stream Decoder (MSD) module
         sp<DeviceDescriptor> getMsdAudioInDevice() const;
diff --git a/services/audiopolicy/managerdefault/EngineLibrary.cpp b/services/audiopolicy/managerdefault/EngineLibrary.cpp
new file mode 100644
index 0000000..ef699aa
--- /dev/null
+++ b/services/audiopolicy/managerdefault/EngineLibrary.cpp
@@ -0,0 +1,78 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "APM_EngineLoader"
+
+#include <dlfcn.h>
+#include <utils/Log.h>
+
+#include "EngineLibrary.h"
+
+namespace android {
+
+// static
+std::shared_ptr<EngineLibrary> EngineLibrary::load(std::string libraryPath)
+{
+    std::shared_ptr<EngineLibrary> engLib(new EngineLibrary());
+    return engLib->init(std::move(libraryPath)) ? engLib : nullptr;
+}
+
+EngineLibrary::~EngineLibrary()
+{
+    close();
+}
+
+bool EngineLibrary::init(std::string libraryPath)
+{
+    mLibraryHandle = dlopen(libraryPath.c_str(), 0);
+    if (mLibraryHandle == nullptr) {
+        ALOGE("Could not dlopen %s: %s", libraryPath.c_str(), dlerror());
+        return false;
+    }
+    mCreateEngineInstance = (EngineInterface* (*)())dlsym(mLibraryHandle, "createEngineInstance");
+    mDestroyEngineInstance = (void (*)(EngineInterface*))dlsym(
+            mLibraryHandle, "destroyEngineInstance");
+    if (mCreateEngineInstance == nullptr || mDestroyEngineInstance == nullptr) {
+        ALOGE("Could not find engine interface functions in %s", libraryPath.c_str());
+        close();
+        return false;
+    }
+    ALOGD("Loaded engine from %s", libraryPath.c_str());
+    return true;
+}
+
+EngineInstance EngineLibrary::createEngine()
+{
+    if (mCreateEngineInstance == nullptr || mDestroyEngineInstance == nullptr) {
+        return EngineInstance();
+    }
+    return EngineInstance(mCreateEngineInstance(),
+            [lib = shared_from_this(), destroy = mDestroyEngineInstance] (EngineInterface* e) {
+                destroy(e);
+            });
+}
+
+void EngineLibrary::close()
+{
+    if (mLibraryHandle != nullptr) {
+        dlclose(mLibraryHandle);
+    }
+    mLibraryHandle = nullptr;
+    mCreateEngineInstance = nullptr;
+    mDestroyEngineInstance = nullptr;
+}
+
+}  // namespace android
diff --git a/services/audiopolicy/managerdefault/EngineLibrary.h b/services/audiopolicy/managerdefault/EngineLibrary.h
new file mode 100644
index 0000000..f143916
--- /dev/null
+++ b/services/audiopolicy/managerdefault/EngineLibrary.h
@@ -0,0 +1,51 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#pragma once
+
+#include <functional>
+#include <memory>
+#include <string>
+
+#include <EngineInterface.h>
+
+namespace android {
+
+using EngineInstance = std::unique_ptr<EngineInterface, std::function<void (EngineInterface*)>>;
+
+class EngineLibrary : public std::enable_shared_from_this<EngineLibrary> {
+public:
+    static std::shared_ptr<EngineLibrary> load(std::string libraryPath);
+    ~EngineLibrary();
+
+    EngineLibrary(const EngineLibrary&) = delete;
+    EngineLibrary(EngineLibrary&&) = delete;
+    EngineLibrary& operator=(const EngineLibrary&) = delete;
+    EngineLibrary& operator=(EngineLibrary&&) = delete;
+
+    EngineInstance createEngine();
+
+private:
+    EngineLibrary() = default;
+    bool init(std::string libraryPath);
+    void close();
+
+    void *mLibraryHandle = nullptr;
+    EngineInterface* (*mCreateEngineInstance)() = nullptr;
+    void (*mDestroyEngineInstance)(EngineInterface*) = nullptr;
+};
+
+}  // namespace android
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index fa8da89..389f861 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -222,7 +222,7 @@
 
     if (result == NO_ERROR) {
         sp <AudioPlaybackClient> client =
-            new AudioPlaybackClient(*attr, *output, uid, pid, session, *selectedDeviceId, *stream);
+            new AudioPlaybackClient(*attr, *output, uid, pid, session, *portId, *selectedDeviceId, *stream);
         mAudioPlaybackClients.add(*portId, client);
     }
     return result;
@@ -451,7 +451,7 @@
             return status;
         }
 
-        sp<AudioRecordClient> client = new AudioRecordClient(*attr, *input, uid, pid, session,
+        sp<AudioRecordClient> client = new AudioRecordClient(*attr, *input, uid, pid, session, *portId,
                                                              *selectedDeviceId, opPackageName,
                                                              canCaptureOutput, canCaptureHotword);
         mAudioRecordClients.add(*portId, client);
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 90939ce..7dfc205 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -406,8 +406,7 @@
 {
 //    Go over all active clients and allow capture (does not force silence) in the
 //    following cases:
-//    Another client in the same UID has already been allowed to capture
-//    OR The client is the assistant
+//    The client is the assistant
 //        AND an accessibility service is on TOP or a RTT call is active
 //                AND the source is VOICE_RECOGNITION or HOTWORD
 //            OR uses VOICE_RECOGNITION AND is on TOP
@@ -498,21 +497,12 @@
         topActive = latestActive;
     }
 
-    std::vector<uid_t> enabledUids;
-
     for (size_t i =0; i < mAudioRecordClients.size(); i++) {
         sp<AudioRecordClient> current = mAudioRecordClients[i];
         if (!current->active) {
             continue;
         }
 
-        // keep capture allowed if another client with the same UID has already
-        // been allowed to capture
-        if (std::find(enabledUids.begin(), enabledUids.end(), current->uid)
-                != enabledUids.end()) {
-            continue;
-        }
-
         audio_source_t source = current->attributes.source;
         bool isTopOrLatestActive = topActive == nullptr ? false : current->uid == topActive->uid;
         bool isLatestSensitive = latestSensitiveActive == nullptr ?
@@ -552,29 +542,24 @@
             }
         } else if (mUidPolicy->isA11yUid(current->uid)) {
             // For accessibility service allow capture if:
-            //     Is on TOP
-            //          AND the source is VOICE_RECOGNITION or HOTWORD
-            //     Or
-            //          The assistant is not on TOP
-            //          AND there is no active privacy sensitive capture or call
+            //     The assistant is not on TOP
+            //         AND there is no active privacy sensitive capture or call
             //             OR client has CAPTURE_AUDIO_OUTPUT privileged permission
+            //     OR
+            //         Is on TOP AND the source is VOICE_RECOGNITION or HOTWORD
+            if (!isAssistantOnTop
+                    && (!(isSensitiveActive || isInCall) || current->canCaptureOutput)) {
+                allowCapture = true;
+            }
             if (isA11yOnTop) {
                 if (source == AUDIO_SOURCE_VOICE_RECOGNITION || source == AUDIO_SOURCE_HOTWORD) {
                     allowCapture = true;
                 }
-            } else {
-                if (!isAssistantOnTop
-                        && (!(isSensitiveActive || isInCall) || current->canCaptureOutput)) {
-                    allowCapture = true;
-                }
             }
         }
-        setAppState_l(current->uid,
+        setAppState_l(current->portId,
                       allowCapture ? apmStatFromAmState(mUidPolicy->getUidState(current->uid)) :
                                 APP_STATE_IDLE);
-        if (allowCapture) {
-            enabledUids.push_back(current->uid);
-        }
     }
 }
 
@@ -582,7 +567,7 @@
     for (size_t i = 0; i < mAudioRecordClients.size(); i++) {
         sp<AudioRecordClient> current = mAudioRecordClients[i];
         if (!isVirtualSource(current->attributes.source)) {
-            setAppState_l(current->uid, APP_STATE_IDLE);
+            setAppState_l(current->portId, APP_STATE_IDLE);
         }
     }
 }
@@ -628,17 +613,17 @@
     return false;
 }
 
-void AudioPolicyService::setAppState_l(uid_t uid, app_state_t state)
+void AudioPolicyService::setAppState_l(audio_port_handle_t portId, app_state_t state)
 {
     AutoCallerClear acc;
 
     if (mAudioPolicyManager) {
-        mAudioPolicyManager->setAppState(uid, state);
+        mAudioPolicyManager->setAppState(portId, state);
     }
     sp<IAudioFlinger> af = AudioSystem::get_audio_flinger();
     if (af) {
         bool silenced = state == APP_STATE_IDLE;
-        af->setRecordSilenced(uid, silenced);
+        af->setRecordSilenced(portId, silenced);
     }
 }
 
@@ -1040,8 +1025,7 @@
 
 bool AudioPolicyService::UidPolicy::isA11yOnTop() {
     for (const auto &uid : mCachedUids) {
-        std::vector<uid_t>::iterator it = find(mA11yUids.begin(), mA11yUids.end(), uid.first);
-        if (it == mA11yUids.end()) {
+        if (!isA11yUid(uid.first)) {
             continue;
         }
         if (uid.second.second >= ActivityManager::PROCESS_STATE_TOP
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 74aea0d..939df2c 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -311,7 +311,7 @@
     virtual status_t shellCommand(int in, int out, int err, Vector<String16>& args);
 
     // Sets whether the given UID records only silence
-    virtual void setAppState_l(uid_t uid, app_state_t state);
+    virtual void setAppState_l(audio_port_handle_t portId, app_state_t state);
 
     // Overrides the UID state as if it is idle
     status_t handleSetUidState(Vector<String16>& args, int err);
@@ -759,9 +759,10 @@
     public:
                 AudioClient(const audio_attributes_t attributes,
                             const audio_io_handle_t io, uid_t uid, pid_t pid,
-                            const audio_session_t session, const audio_port_handle_t deviceId) :
+                            const audio_session_t session,  audio_port_handle_t portId,
+                            const audio_port_handle_t deviceId) :
                                 attributes(attributes), io(io), uid(uid), pid(pid),
-                                session(session), deviceId(deviceId), active(false) {}
+                                session(session), portId(portId), deviceId(deviceId), active(false) {}
                 ~AudioClient() override = default;
 
 
@@ -770,6 +771,7 @@
         const uid_t uid;                     // client UID
         const pid_t pid;                     // client PID
         const audio_session_t session;       // audio session ID
+        const audio_port_handle_t portId;
         const audio_port_handle_t deviceId;  // selected input device port ID
               bool active;                   // Playback/Capture is active or inactive
     };
@@ -781,10 +783,10 @@
     public:
                 AudioRecordClient(const audio_attributes_t attributes,
                           const audio_io_handle_t io, uid_t uid, pid_t pid,
-                          const audio_session_t session, const audio_port_handle_t deviceId,
-                          const String16& opPackageName,
+                          const audio_session_t session, audio_port_handle_t portId,
+                          const audio_port_handle_t deviceId, const String16& opPackageName,
                           bool canCaptureOutput, bool canCaptureHotword) :
-                    AudioClient(attributes, io, uid, pid, session, deviceId),
+                    AudioClient(attributes, io, uid, pid, session, portId, deviceId),
                     opPackageName(opPackageName), startTimeNs(0),
                     canCaptureOutput(canCaptureOutput), canCaptureHotword(canCaptureHotword) {}
                 ~AudioRecordClient() override = default;
@@ -802,9 +804,9 @@
     public:
                 AudioPlaybackClient(const audio_attributes_t attributes,
                       const audio_io_handle_t io, uid_t uid, pid_t pid,
-                            const audio_session_t session, audio_port_handle_t deviceId,
-                            audio_stream_type_t stream) :
-                    AudioClient(attributes, io, uid, pid, session, deviceId), stream(stream) {}
+                            const audio_session_t session, audio_port_handle_t portId,
+                            audio_port_handle_t deviceId, audio_stream_type_t stream) :
+                    AudioClient(attributes, io, uid, pid, session, portId, deviceId), stream(stream) {}
                 ~AudioPlaybackClient() override = default;
 
         const audio_stream_type_t stream;
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
new file mode 100644
index 0000000..df23410
--- /dev/null
+++ b/services/audiopolicy/tests/Android.bp
@@ -0,0 +1,66 @@
+cc_test {
+    name: "audiopolicy_tests",
+
+    include_dirs: [
+        "frameworks/av/services/audiopolicy",
+    ],
+
+    shared_libs: [
+        "libaudioclient",
+        "libaudiofoundation",
+        "libaudiopolicy",
+        "libaudiopolicymanagerdefault",
+        "libbase",
+        "libhidlbase",
+        "liblog",
+        "libmedia_helper",
+        "libutils",
+        "libxml2",
+    ],
+
+    static_libs: ["libaudiopolicycomponents"],
+
+    header_libs: [
+        "libaudiopolicycommon",
+        "libaudiopolicyengine_interface_headers",
+        "libaudiopolicymanager_interface_headers",
+    ],
+
+    srcs: ["audiopolicymanager_tests.cpp"],
+
+    data: [":audiopolicytest_configuration_files",],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
+
+    test_suites: ["device-tests"],
+
+}
+
+// system/audio.h utilities test
+
+cc_test {
+    name: "systemaudio_tests",
+
+    shared_libs: [
+        "libaudiofoundation",
+        "libbase",
+        "liblog",
+        "libmedia_helper",
+        "libutils",
+    ],
+
+    header_libs: ["libmedia_headers"],
+
+    srcs: ["systemaudio_tests.cpp"],
+
+    cflags: [
+        "-Werror",
+        "-Wall",
+    ],
+
+    test_suites: ["device-tests"],
+
+}
diff --git a/services/audiopolicy/tests/Android.mk b/services/audiopolicy/tests/Android.mk
deleted file mode 100644
index ab9f78b..0000000
--- a/services/audiopolicy/tests/Android.mk
+++ /dev/null
@@ -1,65 +0,0 @@
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_C_INCLUDES := \
-  frameworks/av/services/audiopolicy \
-  $(call include-path-for, audio-utils) \
-
-LOCAL_SHARED_LIBRARIES := \
-  libaudiopolicymanagerdefault \
-  libbase \
-  liblog \
-  libmedia_helper \
-  libutils \
-
-LOCAL_STATIC_LIBRARIES := \
-  libaudiopolicycomponents \
-
-LOCAL_HEADER_LIBRARIES := \
-    libaudiopolicycommon \
-    libaudiopolicyengine_interface_headers \
-    libaudiopolicymanager_interface_headers
-
-LOCAL_SRC_FILES := \
-  audiopolicymanager_tests.cpp \
-
-LOCAL_MODULE := audiopolicy_tests
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_CFLAGS := -Werror -Wall
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_COMPATIBILITY_SUITE := device-tests
-
-include $(BUILD_NATIVE_TEST)
-
-# system/audio.h utilities test
-
-include $(CLEAR_VARS)
-
-LOCAL_SHARED_LIBRARIES := \
-  libbase \
-  liblog \
-  libmedia_helper \
-  libutils
-
-LOCAL_HEADER_LIBRARIES := \
-  libmedia_headers
-
-LOCAL_SRC_FILES := \
-  systemaudio_tests.cpp \
-
-LOCAL_MODULE := systemaudio_tests
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_CFLAGS := -Werror -Wall
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-LOCAL_COMPATIBILITY_SUITE := device-tests
-
-include $(BUILD_NATIVE_TEST)
diff --git a/services/audiopolicy/tests/audiopolicymanager_tests.cpp b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
index de5670c..60b5009 100644
--- a/services/audiopolicy/tests/audiopolicymanager_tests.cpp
+++ b/services/audiopolicy/tests/audiopolicymanager_tests.cpp
@@ -14,23 +14,39 @@
  * limitations under the License.
  */
 
+#include <map>
 #include <memory>
-#include <set>
+#include <string>
 #include <sys/wait.h>
 #include <unistd.h>
 
 #include <gtest/gtest.h>
 
 #define LOG_TAG "APM_Test"
-#include <log/log.h>
+#include <Serializer.h>
+#include <android-base/file.h>
+#include <media/AudioPolicy.h>
 #include <media/PatchBuilder.h>
+#include <media/RecordingActivityTracker.h>
+#include <utils/Log.h>
+#include <utils/Vector.h>
 
+#include "AudioPolicyInterface.h"
 #include "AudioPolicyTestClient.h"
 #include "AudioPolicyTestManager.h"
 
 using namespace android;
 
-TEST(AudioPolicyManagerTestInit, Failure) {
+TEST(AudioPolicyManagerTestInit, EngineFailure) {
+    AudioPolicyTestClient client;
+    AudioPolicyTestManager manager(&client);
+    manager.getConfig().setDefault();
+    manager.getConfig().setEngineLibraryNameSuffix("non-existent");
+    ASSERT_EQ(NO_INIT, manager.initialize());
+    ASSERT_EQ(NO_INIT, manager.initCheck());
+}
+
+TEST(AudioPolicyManagerTestInit, ClientFailure) {
     AudioPolicyTestClient client;
     AudioPolicyTestManager manager(&client);
     manager.getConfig().setDefault();
@@ -64,6 +80,12 @@
         return NO_ERROR;
     }
 
+    audio_io_handle_t openDuplicateOutput(audio_io_handle_t /*output1*/,
+                                 audio_io_handle_t /*output2*/) override {
+        audio_io_handle_t id = mNextIoHandle++;
+        return id;
+    }
+
     status_t openInput(audio_module_handle_t module,
                        audio_io_handle_t* input,
                        audio_config_t* /*config*/,
@@ -80,11 +102,11 @@
         return NO_ERROR;
     }
 
-    status_t createAudioPatch(const struct audio_patch* /*patch*/,
+    status_t createAudioPatch(const struct audio_patch* patch,
                               audio_patch_handle_t* handle,
                               int /*delayMs*/) override {
         *handle = mNextPatchHandle++;
-        mActivePatches.insert(*handle);
+        mActivePatches.insert(std::make_pair(*handle, *patch));
         return NO_ERROR;
     }
 
@@ -105,11 +127,19 @@
     // Helper methods for tests
     size_t getActivePatchesCount() const { return mActivePatches.size(); }
 
+    const struct audio_patch* getLastAddedPatch() const {
+        if (mActivePatches.empty()) {
+            return nullptr;
+        }
+        auto it = --mActivePatches.end();
+        return &it->second;
+    };
+
   private:
     audio_module_handle_t mNextModuleHandle = AUDIO_MODULE_HANDLE_NONE + 1;
     audio_io_handle_t mNextIoHandle = AUDIO_IO_HANDLE_NONE + 1;
     audio_patch_handle_t mNextPatchHandle = AUDIO_PATCH_HANDLE_NONE + 1;
-    std::set<audio_patch_handle_t> mActivePatches;
+    std::map<audio_patch_handle_t, struct audio_patch> mActivePatches;
 };
 
 class PatchCountCheck {
@@ -134,18 +164,34 @@
   protected:
     void SetUp() override;
     void TearDown() override;
-    virtual void SetUpConfig(AudioPolicyConfig *config) { (void)config; }
+    virtual void SetUpManagerConfig();
 
     void dumpToLog();
+    // When explicitly routing is needed, selectedDeviceId need to be set as the wanted port
+    // id. Otherwise, selectedDeviceId need to be initialized as AUDIO_PORT_HANDLE_NONE.
     void getOutputForAttr(
             audio_port_handle_t *selectedDeviceId,
             audio_format_t format,
             int channelMask,
             int sampleRate,
             audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+            audio_port_handle_t *portId = nullptr,
+            audio_attributes_t attr = {});
+    void getInputForAttr(
+            const audio_attributes_t &attr,
+            audio_unique_id_t riid,
+            audio_port_handle_t *selectedDeviceId,
+            audio_format_t format,
+            int channelMask,
+            int sampleRate,
+            audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
             audio_port_handle_t *portId = nullptr);
     PatchCountCheck snapshotPatchCount() { return PatchCountCheck(mClient.get()); }
 
+    void findDevicePort(audio_port_role_t role, audio_devices_t deviceType,
+            const std::string &address, audio_port &foundPort);
+    static audio_port_handle_t getDeviceIdFromPatch(const struct audio_patch* patch);
+
     std::unique_ptr<AudioPolicyManagerTestClient> mClient;
     std::unique_ptr<AudioPolicyTestManager> mManager;
 };
@@ -153,8 +199,7 @@
 void AudioPolicyManagerTest::SetUp() {
     mClient.reset(new AudioPolicyManagerTestClient);
     mManager.reset(new AudioPolicyTestManager(mClient.get()));
-    mManager->getConfig().setDefault();
-    SetUpConfig(&mManager->getConfig());  // Subclasses may want to customize the config.
+    SetUpManagerConfig();  // Subclasses may want to customize the config.
     ASSERT_EQ(NO_ERROR, mManager->initialize());
     ASSERT_EQ(NO_ERROR, mManager->initCheck());
 }
@@ -164,6 +209,10 @@
     mClient.reset();
 }
 
+void AudioPolicyManagerTest::SetUpManagerConfig() {
+    mManager->getConfig().setDefault();
+}
+
 void AudioPolicyManagerTest::dumpToLog() {
     int pipefd[2];
     ASSERT_NE(-1, pipe(pipefd));
@@ -200,15 +249,14 @@
         int channelMask,
         int sampleRate,
         audio_output_flags_t flags,
-        audio_port_handle_t *portId) {
-    audio_attributes_t attr = {};
+        audio_port_handle_t *portId,
+        audio_attributes_t attr) {
     audio_io_handle_t output = AUDIO_PORT_HANDLE_NONE;
     audio_stream_type_t stream = AUDIO_STREAM_DEFAULT;
     audio_config_t config = AUDIO_CONFIG_INITIALIZER;
     config.sample_rate = sampleRate;
     config.channel_mask = channelMask;
     config.format = format;
-    *selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
     audio_port_handle_t localPortId;
     if (!portId) portId = &localPortId;
     *portId = AUDIO_PORT_HANDLE_NONE;
@@ -218,6 +266,71 @@
     ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
 }
 
+void AudioPolicyManagerTest::getInputForAttr(
+        const audio_attributes_t &attr,
+        audio_unique_id_t riid,
+        audio_port_handle_t *selectedDeviceId,
+        audio_format_t format,
+        int channelMask,
+        int sampleRate,
+        audio_input_flags_t flags,
+        audio_port_handle_t *portId) {
+    audio_io_handle_t input = AUDIO_PORT_HANDLE_NONE;
+    audio_config_base_t config = AUDIO_CONFIG_BASE_INITIALIZER;
+    config.sample_rate = sampleRate;
+    config.channel_mask = channelMask;
+    config.format = format;
+    audio_port_handle_t localPortId;
+    if (!portId) portId = &localPortId;
+    *portId = AUDIO_PORT_HANDLE_NONE;
+    AudioPolicyInterface::input_type_t inputType;
+    ASSERT_EQ(OK, mManager->getInputForAttr(
+            &attr, &input, riid, AUDIO_SESSION_NONE, 0 /*uid*/, &config, flags,
+            selectedDeviceId, &inputType, portId));
+    ASSERT_NE(AUDIO_PORT_HANDLE_NONE, *portId);
+}
+
+void AudioPolicyManagerTest::findDevicePort(audio_port_role_t role,
+        audio_devices_t deviceType, const std::string &address, audio_port &foundPort) {
+    uint32_t numPorts = 0;
+    uint32_t generation1;
+    status_t ret;
+
+    ret = mManager->listAudioPorts(role, AUDIO_PORT_TYPE_DEVICE, &numPorts, nullptr, &generation1);
+    ASSERT_EQ(NO_ERROR, ret);
+
+    uint32_t generation2;
+    struct audio_port ports[numPorts];
+    ret = mManager->listAudioPorts(role, AUDIO_PORT_TYPE_DEVICE, &numPorts, ports, &generation2);
+    ASSERT_EQ(NO_ERROR, ret);
+    ASSERT_EQ(generation1, generation2);
+
+    for (const auto &port : ports) {
+        if (port.role == role && port.ext.device.type == deviceType &&
+                (strncmp(port.ext.device.address, address.c_str(),
+                         AUDIO_DEVICE_MAX_ADDRESS_LEN) == 0)) {
+            foundPort = port;
+            return;
+        }
+    }
+    GTEST_FAIL();
+}
+
+audio_port_handle_t AudioPolicyManagerTest::getDeviceIdFromPatch(
+        const struct audio_patch* patch) {
+    // The logic here is the same as the one in AudioIoDescriptor.
+    // Note this function is aim to get routed device id for test.
+    // In that case, device to device patch is not expected here.
+    if (patch->num_sources != 0 && patch->num_sinks != 0) {
+        if (patch->sources[0].type == AUDIO_PORT_TYPE_MIX) {
+            return patch->sinks[0].id;
+        } else {
+            return patch->sources[0].id;
+        }
+    }
+    return AUDIO_PORT_HANDLE_NONE;
+}
+
 
 TEST_F(AudioPolicyManagerTest, InitSuccess) {
     // SetUp must finish with no assertions.
@@ -277,15 +390,17 @@
 
 class AudioPolicyManagerTestMsd : public AudioPolicyManagerTest {
   protected:
-    void SetUpConfig(AudioPolicyConfig *config) override;
+    void SetUpManagerConfig() override;
     void TearDown() override;
 
     sp<DeviceDescriptor> mMsdOutputDevice;
     sp<DeviceDescriptor> mMsdInputDevice;
 };
 
-void AudioPolicyManagerTestMsd::SetUpConfig(AudioPolicyConfig *config) {
+void AudioPolicyManagerTestMsd::SetUpManagerConfig() {
     // TODO: Consider using Serializer to load part of the config from a string.
+    AudioPolicyManagerTest::SetUpManagerConfig();
+    AudioPolicyConfig& config = mManager->getConfig();
     mMsdOutputDevice = new DeviceDescriptor(AUDIO_DEVICE_OUT_BUS);
     sp<AudioProfile> pcmOutputProfile = new AudioProfile(
             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
@@ -298,22 +413,21 @@
     sp<AudioProfile> pcmInputProfile = new AudioProfile(
             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO, 44100);
     mMsdInputDevice->addAudioProfile(pcmInputProfile);
-    config->addAvailableDevice(mMsdOutputDevice);
-    config->addAvailableDevice(mMsdInputDevice);
+    config.addAvailableDevice(mMsdOutputDevice);
+    config.addAvailableDevice(mMsdInputDevice);
 
     sp<HwModule> msdModule = new HwModule(AUDIO_HARDWARE_MODULE_ID_MSD, 2 /*halVersionMajor*/);
-    HwModuleCollection modules = config->getHwModules();
+    HwModuleCollection modules = config.getHwModules();
     modules.add(msdModule);
-    config->setHwModules(modules);
+    config.setHwModules(modules);
     mMsdOutputDevice->attach(msdModule);
     mMsdInputDevice->attach(msdModule);
 
-    sp<OutputProfile> msdOutputProfile = new OutputProfile(String8("msd input"));
+    sp<OutputProfile> msdOutputProfile = new OutputProfile("msd input");
     msdOutputProfile->addAudioProfile(pcmOutputProfile);
     msdOutputProfile->addSupportedDevice(mMsdOutputDevice);
     msdModule->addOutputProfile(msdOutputProfile);
-    sp<OutputProfile> msdCompressedOutputProfile =
-            new OutputProfile(String8("msd compressed input"));
+    sp<OutputProfile> msdCompressedOutputProfile = new OutputProfile("msd compressed input");
     msdCompressedOutputProfile->addAudioProfile(ac3OutputProfile);
     msdCompressedOutputProfile->setFlags(
             AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD |
@@ -321,7 +435,7 @@
     msdCompressedOutputProfile->addSupportedDevice(mMsdOutputDevice);
     msdModule->addOutputProfile(msdCompressedOutputProfile);
 
-    sp<InputProfile> msdInputProfile = new InputProfile(String8("msd output"));
+    sp<InputProfile> msdInputProfile = new InputProfile("msd output");
     msdInputProfile->addAudioProfile(pcmInputProfile);
     msdInputProfile->addSupportedDevice(mMsdInputDevice);
     msdModule->addInputProfile(msdInputProfile);
@@ -330,12 +444,12 @@
     // of streams that are not supported by MSD.
     sp<AudioProfile> dtsOutputProfile = new AudioProfile(
             AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000);
-    config->getDefaultOutputDevice()->addAudioProfile(dtsOutputProfile);
-    sp<OutputProfile> primaryEncodedOutputProfile = new OutputProfile(String8("encoded"));
+    config.getDefaultOutputDevice()->addAudioProfile(dtsOutputProfile);
+    sp<OutputProfile> primaryEncodedOutputProfile = new OutputProfile("encoded");
     primaryEncodedOutputProfile->addAudioProfile(dtsOutputProfile);
     primaryEncodedOutputProfile->setFlags(AUDIO_OUTPUT_FLAG_DIRECT);
-    primaryEncodedOutputProfile->addSupportedDevice(config->getDefaultOutputDevice());
-    config->getHwModules().getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)->
+    primaryEncodedOutputProfile->addSupportedDevice(config.getDefaultOutputDevice());
+    config.getHwModules().getModuleFromName(AUDIO_HARDWARE_MODULE_ID_PRIMARY)->
             addOutputProfile(primaryEncodedOutputProfile);
 }
 
@@ -363,7 +477,7 @@
 
 TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedRoutesToMsd) {
     const PatchCountCheck patchCount = snapshotPatchCount();
-    audio_port_handle_t selectedDeviceId;
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
     getOutputForAttr(&selectedDeviceId,
             AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
     ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
@@ -372,7 +486,7 @@
 
 TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrPcmRoutesToMsd) {
     const PatchCountCheck patchCount = snapshotPatchCount();
-    audio_port_handle_t selectedDeviceId;
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
     getOutputForAttr(&selectedDeviceId,
             AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO, 48000);
     ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
@@ -381,7 +495,7 @@
 
 TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrEncodedPlusPcmRoutesToMsd) {
     const PatchCountCheck patchCount = snapshotPatchCount();
-    audio_port_handle_t selectedDeviceId;
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
     getOutputForAttr(&selectedDeviceId,
             AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
     ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
@@ -394,7 +508,7 @@
 
 TEST_F(AudioPolicyManagerTestMsd, GetOutputForAttrUnsupportedFormatBypassesMsd) {
     const PatchCountCheck patchCount = snapshotPatchCount();
-    audio_port_handle_t selectedDeviceId;
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
     getOutputForAttr(&selectedDeviceId,
             AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
     ASSERT_NE(selectedDeviceId, mMsdOutputDevice->getId());
@@ -405,7 +519,8 @@
     // Switch between formats that are supported and not supported by MSD.
     {
         const PatchCountCheck patchCount = snapshotPatchCount();
-        audio_port_handle_t selectedDeviceId, portId;
+        audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+        audio_port_handle_t portId;
         getOutputForAttr(&selectedDeviceId,
                 AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
                 &portId);
@@ -416,7 +531,8 @@
     }
     {
         const PatchCountCheck patchCount = snapshotPatchCount();
-        audio_port_handle_t selectedDeviceId, portId;
+        audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+        audio_port_handle_t portId;
         getOutputForAttr(&selectedDeviceId,
                 AUDIO_FORMAT_DTS, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT,
                 &portId);
@@ -427,10 +543,599 @@
     }
     {
         const PatchCountCheck patchCount = snapshotPatchCount();
-        audio_port_handle_t selectedDeviceId;
+        audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
         getOutputForAttr(&selectedDeviceId,
                 AUDIO_FORMAT_AC3, AUDIO_CHANNEL_OUT_5POINT1, 48000, AUDIO_OUTPUT_FLAG_DIRECT);
         ASSERT_EQ(selectedDeviceId, mMsdOutputDevice->getId());
         ASSERT_EQ(0, patchCount.deltaFromSnapshot());
     }
 }
+
+class AudioPolicyManagerTestWithConfigurationFile : public AudioPolicyManagerTest {
+protected:
+    void SetUpManagerConfig() override;
+    virtual std::string getConfigFile() { return sDefaultConfig; }
+
+    static const std::string sExecutableDir;
+    static const std::string sDefaultConfig;
+};
+
+const std::string AudioPolicyManagerTestWithConfigurationFile::sExecutableDir =
+        base::GetExecutableDirectory() + "/";
+
+const std::string AudioPolicyManagerTestWithConfigurationFile::sDefaultConfig =
+        sExecutableDir + "test_audio_policy_configuration.xml";
+
+void AudioPolicyManagerTestWithConfigurationFile::SetUpManagerConfig() {
+    status_t status = deserializeAudioPolicyFile(getConfigFile().c_str(), &mManager->getConfig());
+    ASSERT_EQ(NO_ERROR, status);
+}
+
+TEST_F(AudioPolicyManagerTestWithConfigurationFile, InitSuccess) {
+    // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestWithConfigurationFile, Dump) {
+    dumpToLog();
+}
+
+using PolicyMixTuple = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
+
+class AudioPolicyManagerTestDynamicPolicy : public AudioPolicyManagerTestWithConfigurationFile {
+protected:
+    void TearDown() override;
+
+    status_t addPolicyMix(int mixType, int mixFlag, audio_devices_t deviceType,
+            std::string mixAddress, const audio_config_t& audioConfig,
+            const std::vector<PolicyMixTuple>& rules);
+    void clearPolicyMix();
+
+    Vector<AudioMix> mAudioMixes;
+    const std::string mMixAddress = "remote_submix_media";
+};
+
+void AudioPolicyManagerTestDynamicPolicy::TearDown() {
+    mManager->unregisterPolicyMixes(mAudioMixes);
+    AudioPolicyManagerTestWithConfigurationFile::TearDown();
+}
+
+status_t AudioPolicyManagerTestDynamicPolicy::addPolicyMix(int mixType, int mixFlag,
+        audio_devices_t deviceType, std::string mixAddress, const audio_config_t& audioConfig,
+        const std::vector<PolicyMixTuple>& rules) {
+    Vector<AudioMixMatchCriterion> myMixMatchCriteria;
+
+    for(const auto &rule: rules) {
+        myMixMatchCriteria.add(AudioMixMatchCriterion(
+                std::get<0>(rule), std::get<1>(rule), std::get<2>(rule)));
+    }
+
+    AudioMix myAudioMix(myMixMatchCriteria, mixType, audioConfig, mixFlag,
+            String8(mixAddress.c_str()), 0);
+    myAudioMix.mDeviceType = deviceType;
+    // Clear mAudioMix before add new one to make sure we don't add already exist mixes.
+    mAudioMixes.clear();
+    mAudioMixes.add(myAudioMix);
+
+    // As the policy mixes registration may fail at some case,
+    // caller need to check the returned status.
+    status_t ret = mManager->registerPolicyMixes(mAudioMixes);
+    return ret;
+}
+
+void AudioPolicyManagerTestDynamicPolicy::clearPolicyMix() {
+    if (mManager != nullptr) {
+        mManager->unregisterPolicyMixes(mAudioMixes);
+    }
+    mAudioMixes.clear();
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, InitSuccess) {
+    // SetUp must finish with no assertions
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, Dump) {
+    dumpToLog();
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, RegisterPolicyMixes) {
+    status_t ret;
+    audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+
+    // Only capture of playback is allowed in LOOP_BACK &RENDER mode
+    ret = addPolicyMix(MIX_TYPE_RECORDERS, MIX_ROUTE_FLAG_LOOP_BACK_AND_RENDER,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+    ASSERT_EQ(INVALID_OPERATION, ret);
+
+    // Fail due to the device is already connected.
+    clearPolicyMix();
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+    ASSERT_EQ(INVALID_OPERATION, ret);
+
+    // The first time to register policy mixes with valid parameter should succeed.
+    clearPolicyMix();
+    audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    audioConfig.sample_rate = 48000;
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig,
+            std::vector<PolicyMixTuple>());
+    ASSERT_EQ(NO_ERROR, ret);
+    // Registering the same policy mixes should fail.
+    ret = mManager->registerPolicyMixes(mAudioMixes);
+    ASSERT_EQ(INVALID_OPERATION, ret);
+
+    // Registration should fail due to device not found.
+    // Note that earpiece is not present in the test configuration file.
+    // This will need to be updated if earpiece is added in the test configuration file.
+    clearPolicyMix();
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+            AUDIO_DEVICE_OUT_EARPIECE, "", audioConfig, std::vector<PolicyMixTuple>());
+    ASSERT_EQ(INVALID_OPERATION, ret);
+
+    // Registration should fail due to output not found.
+    clearPolicyMix();
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+    ASSERT_EQ(INVALID_OPERATION, ret);
+
+    // The first time to register valid policy mixes should succeed.
+    clearPolicyMix();
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_RENDER,
+            AUDIO_DEVICE_OUT_SPEAKER, "", audioConfig, std::vector<PolicyMixTuple>());
+    ASSERT_EQ(NO_ERROR, ret);
+    // Registering the same policy mixes should fail.
+    ret = mManager->registerPolicyMixes(mAudioMixes);
+    ASSERT_EQ(INVALID_OPERATION, ret);
+}
+
+TEST_F(AudioPolicyManagerTestDynamicPolicy, UnregisterPolicyMixes) {
+    status_t ret;
+    audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+
+    audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    audioConfig.sample_rate = 48000;
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig,
+            std::vector<PolicyMixTuple>());
+    ASSERT_EQ(NO_ERROR, ret);
+
+    // After successfully registering policy mixes, it should be able to unregister.
+    ret = mManager->unregisterPolicyMixes(mAudioMixes);
+    ASSERT_EQ(NO_ERROR, ret);
+
+    // After unregistering policy mixes successfully, it should fail unregistering
+    // the same policy mixes as they are not registered.
+    ret = mManager->unregisterPolicyMixes(mAudioMixes);
+    ASSERT_EQ(INVALID_OPERATION, ret);
+}
+
+class AudioPolicyManagerTestDPNoRemoteSubmixModule : public AudioPolicyManagerTestDynamicPolicy {
+protected:
+    std::string getConfigFile() override { return sPrimaryOnlyConfig; }
+
+    static const std::string sPrimaryOnlyConfig;
+};
+
+const std::string AudioPolicyManagerTestDPNoRemoteSubmixModule::sPrimaryOnlyConfig =
+        sExecutableDir + "test_audio_policy_primary_only_configuration.xml";
+
+TEST_F(AudioPolicyManagerTestDPNoRemoteSubmixModule, InitSuccess) {
+    // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestDPNoRemoteSubmixModule, Dump) {
+    dumpToLog();
+}
+
+TEST_F(AudioPolicyManagerTestDPNoRemoteSubmixModule, RegistrationFailure) {
+    // Registration/Unregistration should fail due to module for remote submix not found.
+    status_t ret;
+    audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+    audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    audioConfig.sample_rate = 48000;
+    ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, "", audioConfig, std::vector<PolicyMixTuple>());
+    ASSERT_EQ(INVALID_OPERATION, ret);
+
+    ret = mManager->unregisterPolicyMixes(mAudioMixes);
+    ASSERT_EQ(INVALID_OPERATION, ret);
+}
+
+class AudioPolicyManagerTestDPPlaybackReRouting : public AudioPolicyManagerTestDynamicPolicy,
+        public testing::WithParamInterface<audio_attributes_t> {
+protected:
+    void SetUp() override;
+    void TearDown() override;
+
+    std::unique_ptr<RecordingActivityTracker> mTracker;
+
+    std::vector<PolicyMixTuple> mUsageRules = {
+            {AUDIO_USAGE_MEDIA, AUDIO_SOURCE_DEFAULT, RULE_MATCH_ATTRIBUTE_USAGE},
+            {AUDIO_USAGE_ALARM, AUDIO_SOURCE_DEFAULT, RULE_MATCH_ATTRIBUTE_USAGE}
+    };
+
+    struct audio_port mInjectionPort;
+    audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;
+};
+
+void AudioPolicyManagerTestDPPlaybackReRouting::SetUp() {
+    AudioPolicyManagerTestDynamicPolicy::SetUp();
+
+    mTracker.reset(new RecordingActivityTracker());
+
+    audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+    audioConfig.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+    audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    audioConfig.sample_rate = 48000;
+    status_t ret = addPolicyMix(MIX_TYPE_PLAYERS, MIX_ROUTE_FLAG_LOOP_BACK,
+            AUDIO_DEVICE_OUT_REMOTE_SUBMIX, mMixAddress, audioConfig, mUsageRules);
+    ASSERT_EQ(NO_ERROR, ret);
+
+    struct audio_port extractionPort;
+    findDevicePort(AUDIO_PORT_ROLE_SOURCE, AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+            mMixAddress, extractionPort);
+
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+    audio_source_t source = AUDIO_SOURCE_REMOTE_SUBMIX;
+    audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN, source, 0, ""};
+    std::string tags = "addr=" + mMixAddress;
+    strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+    getInputForAttr(attr, mTracker->getRiid(), &selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT,
+            AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &mPortId);
+    ASSERT_EQ(NO_ERROR, mManager->startInput(mPortId));
+    ASSERT_EQ(extractionPort.id, selectedDeviceId);
+
+    findDevicePort(AUDIO_PORT_ROLE_SINK, AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+            mMixAddress, mInjectionPort);
+}
+
+void AudioPolicyManagerTestDPPlaybackReRouting::TearDown() {
+    mManager->stopInput(mPortId);
+    AudioPolicyManagerTestDynamicPolicy::TearDown();
+}
+
+TEST_F(AudioPolicyManagerTestDPPlaybackReRouting, InitSuccess) {
+    // SetUp must finish with no assertions
+}
+
+TEST_F(AudioPolicyManagerTestDPPlaybackReRouting, Dump) {
+    dumpToLog();
+}
+
+TEST_P(AudioPolicyManagerTestDPPlaybackReRouting, PlaybackReRouting) {
+    const audio_attributes_t attr = GetParam();
+    const audio_usage_t usage = attr.usage;
+
+    audio_port_handle_t playbackRoutedPortId = AUDIO_PORT_HANDLE_NONE;
+    audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+    getOutputForAttr(&playbackRoutedPortId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, &portId, attr);
+    if (std::find_if(begin(mUsageRules), end(mUsageRules), [&usage](const auto &usageRule) {
+            return (std::get<0>(usageRule) == usage) &&
+            (std::get<2>(usageRule) == RULE_MATCH_ATTRIBUTE_USAGE);}) != end(mUsageRules) ||
+            (strncmp(attr.tags, "addr=", strlen("addr=")) == 0 &&
+                    strncmp(attr.tags + strlen("addr="), mMixAddress.c_str(),
+                    AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - strlen("addr=") - 1) == 0)) {
+        EXPECT_EQ(mInjectionPort.id, playbackRoutedPortId);
+    } else {
+        EXPECT_NE(mInjectionPort.id, playbackRoutedPortId);
+    }
+}
+
+INSTANTIATE_TEST_CASE_P(
+        PlaybackReroutingUsageMatch,
+        AudioPolicyManagerTestDPPlaybackReRouting,
+        testing::Values(
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_MEDIA,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ALARM,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""}
+                )
+        );
+
+INSTANTIATE_TEST_CASE_P(
+        PlaybackReroutingAddressPriorityMatch,
+        AudioPolicyManagerTestDPPlaybackReRouting,
+        testing::Values(
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_MEDIA,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VOICE_COMMUNICATION,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ALARM,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION_EVENT,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_ASSISTANCE_SONIFICATION,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_GAME,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VIRTUAL_SOURCE,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ASSISTANT,
+                                     AUDIO_SOURCE_DEFAULT, 0, "addr=remote_submix_media"}
+                )
+        );
+
+INSTANTIATE_TEST_CASE_P(
+        PlaybackReroutingUnHandledUsages,
+        AudioPolicyManagerTestDPPlaybackReRouting,
+        testing::Values(
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_VOICE_COMMUNICATION,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_TELEPHONY_RINGTONE,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_COMMUNICATION_REQUEST,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_COMMUNICATION_INSTANT,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_NOTIFICATION_COMMUNICATION_DELAYED,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_NOTIFICATION_EVENT,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_ASSISTANCE_ACCESSIBILITY,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC,
+                                     AUDIO_USAGE_ASSISTANCE_SONIFICATION,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_GAME,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_MUSIC, AUDIO_USAGE_ASSISTANT,
+                                     AUDIO_SOURCE_DEFAULT, 0, ""}
+                )
+        );
+
+class AudioPolicyManagerTestDPMixRecordInjection : public AudioPolicyManagerTestDynamicPolicy,
+        public testing::WithParamInterface<audio_attributes_t> {
+protected:
+    void SetUp() override;
+    void TearDown() override;
+
+    std::unique_ptr<RecordingActivityTracker> mTracker;
+
+    std::vector<PolicyMixTuple> mSourceRules = {
+        {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_CAMCORDER, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET},
+        {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_MIC, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET},
+        {AUDIO_USAGE_UNKNOWN, AUDIO_SOURCE_VOICE_COMMUNICATION, RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET}
+    };
+
+    struct audio_port mExtractionPort;
+    audio_port_handle_t mPortId = AUDIO_PORT_HANDLE_NONE;
+};
+
+void AudioPolicyManagerTestDPMixRecordInjection::SetUp() {
+    AudioPolicyManagerTestDynamicPolicy::SetUp();
+
+    mTracker.reset(new RecordingActivityTracker());
+
+    audio_config_t audioConfig = AUDIO_CONFIG_INITIALIZER;
+    audioConfig.channel_mask = AUDIO_CHANNEL_IN_STEREO;
+    audioConfig.format = AUDIO_FORMAT_PCM_16_BIT;
+    audioConfig.sample_rate = 48000;
+    status_t ret = addPolicyMix(MIX_TYPE_RECORDERS, MIX_ROUTE_FLAG_LOOP_BACK,
+            AUDIO_DEVICE_IN_REMOTE_SUBMIX, mMixAddress, audioConfig, mSourceRules);
+    ASSERT_EQ(NO_ERROR, ret);
+
+    struct audio_port injectionPort;
+    findDevicePort(AUDIO_PORT_ROLE_SINK, AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+            mMixAddress, injectionPort);
+
+    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE;
+    audio_usage_t usage = AUDIO_USAGE_VIRTUAL_SOURCE;
+    audio_attributes_t attr = {AUDIO_CONTENT_TYPE_UNKNOWN, usage, AUDIO_SOURCE_DEFAULT, 0, ""};
+    std::string tags = std::string("addr=") + mMixAddress;
+    strncpy(attr.tags, tags.c_str(), AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1);
+    getOutputForAttr(&selectedDeviceId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, &mPortId, attr);
+    ASSERT_EQ(NO_ERROR, mManager->startOutput(mPortId));
+    ASSERT_EQ(injectionPort.id, getDeviceIdFromPatch(mClient->getLastAddedPatch()));
+
+    findDevicePort(AUDIO_PORT_ROLE_SOURCE, AUDIO_DEVICE_IN_REMOTE_SUBMIX,
+            mMixAddress, mExtractionPort);
+}
+
+void AudioPolicyManagerTestDPMixRecordInjection::TearDown() {
+    mManager->stopOutput(mPortId);
+    AudioPolicyManagerTestDynamicPolicy::TearDown();
+}
+
+TEST_F(AudioPolicyManagerTestDPMixRecordInjection, InitSuccess) {
+    // SetUp mush finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestDPMixRecordInjection, Dump) {
+    dumpToLog();
+}
+
+TEST_P(AudioPolicyManagerTestDPMixRecordInjection, RecordingInjection) {
+    const audio_attributes_t attr = GetParam();
+    const audio_source_t source = attr.source;
+
+    audio_port_handle_t captureRoutedPortId = AUDIO_PORT_HANDLE_NONE;
+    audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+    getInputForAttr(attr, mTracker->getRiid(), &captureRoutedPortId, AUDIO_FORMAT_PCM_16_BIT,
+            AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &portId);
+    if (std::find_if(begin(mSourceRules), end(mSourceRules), [&source](const auto &sourceRule) {
+            return (std::get<1>(sourceRule) == source) &&
+            (std::get<2>(sourceRule) == RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET);})
+            != end(mSourceRules)) {
+        EXPECT_EQ(mExtractionPort.id, captureRoutedPortId);
+    } else {
+        EXPECT_NE(mExtractionPort.id, captureRoutedPortId);
+    }
+}
+
+// No address priority rule for remote recording, address is a "don't care"
+INSTANTIATE_TEST_CASE_P(
+        RecordInjectionSourceMatch,
+        AudioPolicyManagerTestDPMixRecordInjection,
+        testing::Values(
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_CAMCORDER, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_CAMCORDER, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_MIC, 0, "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_MIC, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_VOICE_COMMUNICATION, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_VOICE_COMMUNICATION, 0,
+                                     "addr=remote_submix_media"}
+                )
+        );
+
+// No address priority rule for remote recording
+INSTANTIATE_TEST_CASE_P(
+        RecordInjectionSourceNotMatch,
+        AudioPolicyManagerTestDPMixRecordInjection,
+        testing::Values(
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_VOICE_RECOGNITION, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_HOTWORD, 0, ""},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_VOICE_RECOGNITION, 0,
+                                     "addr=remote_submix_media"},
+                (audio_attributes_t){AUDIO_CONTENT_TYPE_UNKNOWN, AUDIO_USAGE_UNKNOWN,
+                                     AUDIO_SOURCE_HOTWORD, 0, "addr=remote_submix_media"}
+                )
+        );
+
+using DeviceConnectionTestParams =
+        std::tuple<audio_devices_t /*type*/, std::string /*name*/, std::string /*address*/>;
+
+class AudioPolicyManagerTestDeviceConnection : public AudioPolicyManagerTestWithConfigurationFile,
+        public testing::WithParamInterface<DeviceConnectionTestParams> {
+};
+
+TEST_F(AudioPolicyManagerTestDeviceConnection, InitSuccess) {
+    // SetUp must finish with no assertions.
+}
+
+TEST_F(AudioPolicyManagerTestDeviceConnection, Dump) {
+    dumpToLog();
+}
+
+TEST_P(AudioPolicyManagerTestDeviceConnection, SetDeviceConnectionState) {
+    const audio_devices_t type = std::get<0>(GetParam());
+    const std::string name = std::get<1>(GetParam());
+    const std::string address = std::get<2>(GetParam());
+
+    if (type == AUDIO_DEVICE_OUT_HDMI) {
+        // Set device connection state failed due to no device descriptor found
+        // For HDMI case, it is easier to simulate device descriptor not found error
+        // by using a undeclared encoded format.
+        ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+                type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+                address.c_str(), name.c_str(), AUDIO_FORMAT_MAT_2_1));
+    }
+    // Connect with valid parameters should succeed
+    ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+            address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+    // Try to connect with the same device again should fail
+    ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+            address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+    // Disconnect the connected device should succeed
+    ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+            address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+    // Disconnect device that is not connected should fail
+    ASSERT_EQ(INVALID_OPERATION, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+            address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+    // Try to set device connection state  with a invalid connection state should fail
+    ASSERT_EQ(BAD_VALUE, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_CNT,
+            "", "", AUDIO_FORMAT_DEFAULT));
+}
+
+TEST_P(AudioPolicyManagerTestDeviceConnection, ExplicitlyRoutingAfterConnection) {
+    const audio_devices_t type = std::get<0>(GetParam());
+    const std::string name = std::get<1>(GetParam());
+    const std::string address = std::get<2>(GetParam());
+
+    // Connect device to do explicitly routing test
+    ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_AVAILABLE,
+            address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+
+    audio_port devicePort;
+    const audio_port_role_t role = audio_is_output_device(type)
+            ? AUDIO_PORT_ROLE_SINK : AUDIO_PORT_ROLE_SOURCE;
+    findDevicePort(role, type, address, devicePort);
+
+    audio_port_handle_t routedPortId = devicePort.id;
+    audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE;
+    // Try start input or output according to the device type
+    if (audio_is_output_devices(type)) {
+        getOutputForAttr(&routedPortId, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+                48000 /*sampleRate*/, AUDIO_OUTPUT_FLAG_NONE, &portId);
+    } else if (audio_is_input_device(type)) {
+        RecordingActivityTracker tracker;
+        getInputForAttr({}, tracker.getRiid(), &routedPortId, AUDIO_FORMAT_PCM_16_BIT,
+                AUDIO_CHANNEL_IN_STEREO, 48000 /*sampleRate*/, AUDIO_INPUT_FLAG_NONE, &portId);
+    }
+    ASSERT_EQ(devicePort.id, routedPortId);
+
+    ASSERT_EQ(NO_ERROR, mManager->setDeviceConnectionState(
+            type, AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE,
+            address.c_str(), name.c_str(), AUDIO_FORMAT_DEFAULT));
+}
+
+INSTANTIATE_TEST_CASE_P(
+        DeviceConnectionState,
+        AudioPolicyManagerTestDeviceConnection,
+        testing::Values(
+                DeviceConnectionTestParams({AUDIO_DEVICE_IN_HDMI, "test_in_hdmi",
+                                            "audio_policy_test_in_hdmi"}),
+                DeviceConnectionTestParams({AUDIO_DEVICE_OUT_HDMI, "test_out_hdmi",
+                                            "audio_policy_test_out_hdmi"}),
+                DeviceConnectionTestParams({AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, "bt_hfp_in",
+                                            "hfp_client_in"}),
+                DeviceConnectionTestParams({AUDIO_DEVICE_OUT_BLUETOOTH_SCO, "bt_hfp_out",
+                                            "hfp_client_out"})
+                )
+        );
diff --git a/services/audiopolicy/tests/resources/Android.bp b/services/audiopolicy/tests/resources/Android.bp
new file mode 100644
index 0000000..41f5ee1
--- /dev/null
+++ b/services/audiopolicy/tests/resources/Android.bp
@@ -0,0 +1,7 @@
+filegroup {
+    name: "audiopolicytest_configuration_files",
+    srcs: [
+        "test_audio_policy_configuration.xml",
+        "test_audio_policy_primary_only_configuration.xml",
+    ],
+}
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
new file mode 100644
index 0000000..87f0ab9
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_audio_policy_configuration.xml
@@ -0,0 +1,111 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 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.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <globalConfiguration speaker_drc_enabled="true"/>
+
+    <modules>
+        <!-- Primary module -->
+        <module name="primary" halVersion="2.0">
+            <attachedDevices>
+                <item>Speaker</item>
+                <item>Built-In Mic</item>
+            </attachedDevices>
+            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+                <mixPort name="mixport_bt_hfp_output" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="mixport_bt_hfp_input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="8000,11025,16000,44100,48000"
+                             channelMasks="AUDIO_CHANNEL_IN_STEREO,AUDIO_CHANNEL_IN_MONO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+                </devicePort>
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                </devicePort>
+                <devicePort tagName="Hdmi" type="AUDIO_DEVICE_OUT_HDMI" role="sink">
+                </devicePort>
+                <devicePort tagName="Hdmi-In Mic" type="AUDIO_DEVICE_IN_HDMI" role="source">
+                </devicePort>
+                <devicePort tagName="BT SCO" type="AUDIO_DEVICE_OUT_BLUETOOTH_SCO"
+                            role="sink" address="hfp_client_out">
+                </devicePort>
+                <devicePort tagName="BT SCO Headset Mic" type="AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET"
+                            role="source" address="hfp_client_in">
+                </devicePort>
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="Speaker"
+                       sources="primary output"/>
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic,Hdmi-In Mic"/>
+                <route type="mix" sink="Hdmi"
+                       sources="primary output"/>
+                <route type="mix" sink="BT SCO"
+                       sources="mixport_bt_hfp_output"/>
+                <route type="mix" sink="mixport_bt_hfp_input"
+                       sources="BT SCO Headset Mic"/>
+            </routes>
+        </module>
+
+        <!-- Remote Submix module -->
+        <module name="r_submix" halVersion="2.0">
+            <attachedDevices>
+                <item>Remote Submix In</item>
+            </attachedDevices>
+            <mixPorts>
+                <mixPort name="r_submix output" role="source">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="r_submix input" role="sink">
+                   <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                            samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+           </mixPorts>
+           <devicePorts>
+               <devicePort tagName="Remote Submix Out" type="AUDIO_DEVICE_OUT_REMOTE_SUBMIX"  role="sink">
+                   <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                            samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+               </devicePort>
+               <devicePort tagName="Remote Submix In" type="AUDIO_DEVICE_IN_REMOTE_SUBMIX"  role="source">
+                   <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                            samplingRates="48000" channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </devicePort>
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="Remote Submix Out"
+                       sources="r_submix output"/>
+                <route type="mix" sink="r_submix input"
+                       sources="Remote Submix In"/>
+            </routes>
+        </module>
+    </modules>
+</audioPolicyConfiguration>
diff --git a/services/audiopolicy/tests/resources/test_audio_policy_primary_only_configuration.xml b/services/audiopolicy/tests/resources/test_audio_policy_primary_only_configuration.xml
new file mode 100644
index 0000000..edc0adb
--- /dev/null
+++ b/services/audiopolicy/tests/resources/test_audio_policy_primary_only_configuration.xml
@@ -0,0 +1,53 @@
+<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
+<!-- Copyright (C) 2019 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.
+-->
+
+<audioPolicyConfiguration version="1.0" xmlns:xi="http://www.w3.org/2001/XInclude">
+    <globalConfiguration speaker_drc_enabled="true"/>
+
+    <modules>
+        <!-- Primary module -->
+        <module name="primary" halVersion="2.0">
+            <attachedDevices>
+                <item>Speaker</item>
+                <item>Built-In Mic</item>
+            </attachedDevices>
+            <defaultOutputDevice>Speaker</defaultOutputDevice>
+            <mixPorts>
+                <mixPort name="primary output" role="source" flags="AUDIO_OUTPUT_FLAG_PRIMARY">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000" channelMasks="AUDIO_CHANNEL_OUT_STEREO"/>
+                </mixPort>
+                <mixPort name="primary input" role="sink">
+                    <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                             samplingRates="48000"
+                             channelMasks="AUDIO_CHANNEL_IN_STEREO"/>
+                </mixPort>
+            </mixPorts>
+            <devicePorts>
+                <devicePort tagName="Speaker" type="AUDIO_DEVICE_OUT_SPEAKER" role="sink">
+                </devicePort>
+                <devicePort tagName="Built-In Mic" type="AUDIO_DEVICE_IN_BUILTIN_MIC" role="source">
+                </devicePort>
+            </devicePorts>
+            <routes>
+                <route type="mix" sink="Speaker"
+                       sources="primary output"/>
+                <route type="mix" sink="primary input"
+                       sources="Built-In Mic"/>
+            </routes>
+        </module>
+    </modules>
+</audioPolicyConfiguration>
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index a4868bf..9ba6553 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -117,7 +117,12 @@
 
 // ----------------------------------------------------------------------------
 
+static const String16 sDumpPermission("android.permission.DUMP");
 static const String16 sManageCameraPermission("android.permission.MANAGE_CAMERA");
+static const String16 sCameraPermission("android.permission.CAMERA");
+static const String16 sSystemCameraPermission("android.permission.SYSTEM_CAMERA");
+static const String16
+        sCameraSendSystemEventsPermission("android.permission.CAMERA_SEND_SYSTEM_EVENTS");
 
 // Matches with PERCEPTIBLE_APP_ADJ in ProcessList.java
 static constexpr int32_t kVendorClientScore = 200;
@@ -130,7 +135,8 @@
 CameraService::CameraService() :
         mEventLog(DEFAULT_EVENT_LOG_LENGTH),
         mNumberOfCameras(0),
-        mSoundRef(0), mInitialized(false) {
+        mSoundRef(0), mInitialized(false),
+        mAudioRestriction(hardware::camera2::ICameraDeviceUser::AUDIO_RESTRICTION_NONE) {
     ALOGI("CameraService started (pid=%d)", getpid());
     mServiceLockWrapper = std::make_shared<WaitableMutexWrapper>(&mServiceLock);
 }
@@ -159,6 +165,7 @@
     mUidPolicy->registerSelf();
     mSensorPrivacyPolicy = new SensorPrivacyPolicy(this);
     mSensorPrivacyPolicy->registerSelf();
+    mAppOps.setCameraAudioRestriction(mAudioRestriction);
     sp<HidlCameraService> hcs = HidlCameraService::getInstance(this);
     if (hcs->registerAsService() != android::OK) {
         ALOGE("%s: Failed to register default android.frameworks.cameraservice.service@1.0",
@@ -239,7 +246,7 @@
     Mutex::Autolock lock(mStatusListenerLock);
 
     for (auto& i : mListenerList) {
-        i.second->getListener()->onTorchStatusChanged(mapToInterface(status), String16{cameraId});
+        i->getListener()->onTorchStatusChanged(mapToInterface(status), String16{cameraId});
     }
 }
 
@@ -514,6 +521,11 @@
                 "Camera subsystem is not available");;
     }
 
+    if (shouldRejectSystemCameraConnection(String8(cameraId))) {
+        return STATUS_ERROR_FMT(ERROR_INVALID_OPERATION, "Unable to retrieve camera"
+                "characteristics for system only device %s: ", String8(cameraId).string());
+    }
+
     Status ret{};
 
     status_t res = mCameraProviderManager->getCameraCharacteristics(
@@ -527,9 +539,12 @@
     int callingPid = CameraThreadState::getCallingPid();
     int callingUid = CameraThreadState::getCallingUid();
     std::vector<int32_t> tagsRemoved;
-    // If it's not calling from cameraserver, check the permission.
+    // If it's not calling from cameraserver, check the permission only if
+    // android.permission.CAMERA is required. If android.permission.SYSTEM_CAMERA was needed,
+    // it would've already been checked in shouldRejectSystemCameraConnection.
     if ((callingPid != getpid()) &&
-            !checkPermission(String16("android.permission.CAMERA"), callingPid, callingUid)) {
+            (getSystemCameraKind(String8(cameraId)) != SystemCameraKind::SYSTEM_ONLY_CAMERA) &&
+            !checkPermission(sCameraPermission, callingPid, callingUid)) {
         res = cameraInfo->removePermissionEntries(
                 mCameraProviderManager->getProviderTagIdLocked(String8(cameraId).string()),
                 &tagsRemoved);
@@ -969,9 +984,18 @@
                 clientName8.string(), clientUid, clientPid);
     }
 
-    // If it's not calling from cameraserver, check the permission.
+    if (shouldRejectSystemCameraConnection(cameraId)) {
+        ALOGW("Attempting to connect to system-only camera id %s, connection rejected",
+                cameraId.c_str());
+        return STATUS_ERROR_FMT(ERROR_DISCONNECTED, "No camera device with ID \"%s\" is"
+                                "available", cameraId.string());
+    }
+    // If it's not calling from cameraserver, check the permission if the
+    // device isn't a system only camera (shouldRejectSystemCameraConnection already checks for
+    // android.permission.SYSTEM_CAMERA for system only camera devices).
     if (callingPid != getpid() &&
-            !checkPermission(String16("android.permission.CAMERA"), clientPid, clientUid)) {
+                (getSystemCameraKind(cameraId) != SystemCameraKind::SYSTEM_ONLY_CAMERA) &&
+                !checkPermission(sCameraPermission, clientPid, clientUid)) {
         ALOGE("Permission Denial: can't use the camera pid=%d, uid=%d", clientPid, clientUid);
         return STATUS_ERROR_FMT(ERROR_PERMISSION_DENIED,
                 "Caller \"%s\" (PID %d, UID %d) cannot open camera \"%s\" without camera permission",
@@ -1324,18 +1348,66 @@
     return ret;
 }
 
-bool CameraService::shouldRejectHiddenCameraConnection(const String8 & cameraId) {
-    // If the thread serving this call is not a hwbinder thread and the caller
-    // isn't the cameraserver itself, and the camera id being requested is to be
-    // publically hidden, we should reject the connection.
-    if (!hardware::IPCThreadState::self()->isServingCall() &&
-            CameraThreadState::getCallingPid() != getpid() &&
-            mCameraProviderManager->isPublicallyHiddenSecureCamera(cameraId.c_str())) {
+static bool hasPermissionsForSystemCamera(int callingPid, int callingUid) {
+    return checkPermission(sSystemCameraPermission, callingPid, callingUid) &&
+            checkPermission(sCameraPermission, callingPid, callingUid);
+}
+
+bool CameraService::shouldSkipStatusUpdates(const String8& cameraId, bool isVendorListener,
+        int clientPid, int clientUid) const {
+    SystemCameraKind systemCameraKind = getSystemCameraKind(cameraId);
+    // If the client is not a vendor client, don't add listener if
+    //   a) the camera is a publicly hidden secure camera OR
+    //   b) the camera is a system only camera and the client doesn't
+    //      have android.permission.SYSTEM_CAMERA permissions.
+    if (!isVendorListener && (systemCameraKind == SystemCameraKind::HIDDEN_SECURE_CAMERA ||
+            (systemCameraKind == SystemCameraKind::SYSTEM_ONLY_CAMERA &&
+            !hasPermissionsForSystemCamera(clientPid, clientUid)))) {
         return true;
     }
     return false;
 }
 
+bool CameraService::shouldRejectSystemCameraConnection(const String8& cameraId) const {
+    // Rules for rejection:
+    // 1) If cameraserver tries to access this camera device, accept the
+    //    connection.
+    // 2) The camera device is a publicly hidden secure camera device AND some
+    //    component is trying to access it on a non-hwbinder thread (generally a non HAL client),
+    //    reject it.
+    // 3) if the camera device is advertised by the camera HAL as SYSTEM_ONLY
+    //    and the serving thread is a non hwbinder thread, the client must have
+    //    android.permission.SYSTEM_CAMERA permissions to connect.
+
+    int cPid = CameraThreadState::getCallingPid();
+    int cUid = CameraThreadState::getCallingUid();
+    SystemCameraKind systemCameraKind = getSystemCameraKind(cameraId);
+
+    // (1) Cameraserver trying to connect, accept.
+    if (CameraThreadState::getCallingPid() == getpid()) {
+        return false;
+    }
+    // (2)
+    if (!hardware::IPCThreadState::self()->isServingCall() &&
+            systemCameraKind == SystemCameraKind::HIDDEN_SECURE_CAMERA) {
+        ALOGW("Rejecting access to secure hidden camera %s", cameraId.c_str());
+        return true;
+    }
+    // (3) Here we only check for permissions if it is a system only camera device. This is since
+    //     getCameraCharacteristics() allows for calls to succeed (albeit after hiding some
+    //     characteristics) even if clients don't have android.permission.CAMERA. We do not want the
+    //     same behavior for system camera devices.
+    if (!hardware::IPCThreadState::self()->isServingCall() &&
+            systemCameraKind == SystemCameraKind::SYSTEM_ONLY_CAMERA &&
+            !hasPermissionsForSystemCamera(cPid, cUid)) {
+        ALOGW("Rejecting access to system only camera %s, inadequete permissions",
+                cameraId.c_str());
+        return true;
+    }
+
+    return false;
+}
+
 Status CameraService::connectDevice(
         const sp<hardware::camera2::ICameraDeviceCallbacks>& cameraCb,
         const String16& cameraId,
@@ -1385,14 +1457,6 @@
             (halVersion == -1) ? "default" : std::to_string(halVersion).c_str(),
             static_cast<int>(effectiveApiLevel));
 
-    if (shouldRejectHiddenCameraConnection(cameraId)) {
-        ALOGW("Attempting to connect to system-only camera id %s, connection rejected",
-              cameraId.c_str());
-        return STATUS_ERROR_FMT(ERROR_DISCONNECTED,
-                                "No camera device with ID \"%s\" currently available",
-                                cameraId.string());
-
-    }
     sp<CLIENT> client = nullptr;
     {
         // Acquire mServiceLock and prevent other clients from connecting
@@ -1668,8 +1732,7 @@
     if (pid != selfPid) {
         // Ensure we're being called by system_server, or similar process with
         // permissions to notify the camera service about system events
-        if (!checkCallingPermission(
-                String16("android.permission.CAMERA_SEND_SYSTEM_EVENTS"))) {
+        if (!checkCallingPermission(sCameraSendSystemEventsPermission)) {
             const int uid = CameraThreadState::getCallingUid();
             ALOGE("Permission Denial: cannot send updates to camera service about system"
                     " events from pid=%d, uid=%d", pid, uid);
@@ -1704,7 +1767,7 @@
     Mutex::Autolock lock(mStatusListenerLock);
 
     for (const auto& it : mListenerList) {
-        auto ret = it.second->getListener()->onCameraAccessPrioritiesChanged();
+        auto ret = it->getListener()->onCameraAccessPrioritiesChanged();
         if (!ret.isOk()) {
             ALOGE("%s: Failed to trigger permission callback: %d", __FUNCTION__,
                     ret.exceptionCode());
@@ -1720,8 +1783,7 @@
     if (pid != selfPid) {
         // Ensure we're being called by system_server, or similar process with
         // permissions to notify the camera service about system events
-        if (!checkCallingPermission(
-                String16("android.permission.CAMERA_SEND_SYSTEM_EVENTS"))) {
+        if (!checkCallingPermission(sCameraSendSystemEventsPermission)) {
             const int uid = CameraThreadState::getCallingUid();
             ALOGE("Permission Denial: cannot send updates to camera service about device"
                     " state changes from pid=%d, uid=%d", pid, uid);
@@ -1775,20 +1837,23 @@
         return STATUS_ERROR(ERROR_ILLEGAL_ARGUMENT, "Null listener given to addListener");
     }
 
+    auto clientUid = CameraThreadState::getCallingUid();
+    auto clientPid = CameraThreadState::getCallingPid();
+
     Mutex::Autolock lock(mServiceLock);
 
     {
         Mutex::Autolock lock(mStatusListenerLock);
         for (const auto &it : mListenerList) {
-            if (IInterface::asBinder(it.second->getListener()) == IInterface::asBinder(listener)) {
+            if (IInterface::asBinder(it->getListener()) == IInterface::asBinder(listener)) {
                 ALOGW("%s: Tried to add listener %p which was already subscribed",
                       __FUNCTION__, listener.get());
                 return STATUS_ERROR(ERROR_ALREADY_EXISTS, "Listener already registered");
             }
         }
 
-        auto clientUid = CameraThreadState::getCallingUid();
-        sp<ServiceListener> serviceListener = new ServiceListener(this, listener, clientUid);
+        sp<ServiceListener> serviceListener =
+                new ServiceListener(this, listener, clientUid, clientPid, isVendorListener);
         auto ret = serviceListener->initialize();
         if (ret != NO_ERROR) {
             String8 msg = String8::format("Failed to initialize service listener: %s (%d)",
@@ -1796,7 +1861,10 @@
             ALOGE("%s: %s", __FUNCTION__, msg.string());
             return STATUS_ERROR(ERROR_ILLEGAL_ARGUMENT, msg.string());
         }
-        mListenerList.emplace_back(isVendorListener, serviceListener);
+        // The listener still needs to be added to the list of listeners, regardless of what
+        // permissions the listener process has / whether it is a vendor listener. Since it might be
+        // eligible to listen to other camera ids.
+        mListenerList.emplace_back(serviceListener);
         mUidPolicy->registerMonitorUid(clientUid);
     }
 
@@ -1804,8 +1872,7 @@
     {
         Mutex::Autolock lock(mCameraStatesLock);
         for (auto& i : mCameraStates) {
-            if (!isVendorListener &&
-                mCameraProviderManager->isPublicallyHiddenSecureCamera(i.first.c_str())) {
+            if (shouldSkipStatusUpdates(i.first, isVendorListener, clientPid, clientUid)) {
                 ALOGV("Cannot add public listener for hidden system-only %s for pid %d",
                       i.first.c_str(), CameraThreadState::getCallingPid());
                 continue;
@@ -1844,9 +1911,9 @@
     {
         Mutex::Autolock lock(mStatusListenerLock);
         for (auto it = mListenerList.begin(); it != mListenerList.end(); it++) {
-            if (IInterface::asBinder(it->second->getListener()) == IInterface::asBinder(listener)) {
-                mUidPolicy->unregisterMonitorUid(it->second->getListenerUid());
-                IInterface::asBinder(listener)->unlinkToDeath(it->second);
+            if (IInterface::asBinder((*it)->getListener()) == IInterface::asBinder(listener)) {
+                mUidPolicy->unregisterMonitorUid((*it)->getListenerUid());
+                IInterface::asBinder(listener)->unlinkToDeath(*it);
                 mListenerList.erase(it);
                 return Status::ok();
             }
@@ -1962,6 +2029,7 @@
             mActiveClientManager.remove(i);
         }
     }
+    updateAudioRestrictionLocked();
 }
 
 bool CameraService::evictClientIdByRemote(const wp<IBinder>& remote) {
@@ -2335,6 +2403,7 @@
         mClientPackageName(clientPackageName), mClientPid(clientPid), mClientUid(clientUid),
         mServicePid(servicePid),
         mDisconnected(false),
+        mAudioRestriction(hardware::camera2::ICameraDeviceUser::AUDIO_RESTRICTION_NONE),
         mRemoteBinder(remoteCallback)
 {
     if (sCameraService == nullptr) {
@@ -2438,6 +2507,30 @@
     return level == API_2;
 }
 
+int32_t CameraService::BasicClient::setAudioRestriction(int32_t mode) {
+    {
+        Mutex::Autolock l(mAudioRestrictionLock);
+        mAudioRestriction = mode;
+    }
+    return sCameraService->updateAudioRestriction();
+}
+
+int32_t CameraService::BasicClient::getAudioRestriction() const {
+    Mutex::Autolock l(mAudioRestrictionLock);
+    return mAudioRestriction;
+}
+
+bool CameraService::BasicClient::isValidAudioRestriction(int32_t mode) {
+    switch (mode) {
+        case hardware::camera2::ICameraDeviceUser::AUDIO_RESTRICTION_NONE:
+        case hardware::camera2::ICameraDeviceUser::AUDIO_RESTRICTION_VIBRATION:
+        case hardware::camera2::ICameraDeviceUser::AUDIO_RESTRICTION_VIBRATION_SOUND:
+            return true;
+        default:
+            return false;
+    }
+}
+
 status_t CameraService::BasicClient::startCameraOps() {
     ATRACE_CALL();
 
@@ -3029,7 +3122,7 @@
 status_t CameraService::dump(int fd, const Vector<String16>& args) {
     ATRACE_CALL();
 
-    if (checkCallingPermission(String16("android.permission.DUMP")) == false) {
+    if (checkCallingPermission(sDumpPermission) == false) {
         dprintf(fd, "Permission Denial: can't dump CameraService from pid=%d, uid=%d\n",
                 CameraThreadState::getCallingPid(),
                 CameraThreadState::getCallingUid());
@@ -3261,13 +3354,13 @@
             Mutex::Autolock lock(mStatusListenerLock);
 
             for (auto& listener : mListenerList) {
-                if (!listener.first &&
-                    mCameraProviderManager->isPublicallyHiddenSecureCamera(cameraId.c_str())) {
+                if (shouldSkipStatusUpdates(cameraId, listener->isVendorListener(),
+                        listener->getListenerPid(), listener->getListenerUid())) {
                     ALOGV("Skipping camera discovery callback for system-only camera %s",
-                          cameraId.c_str());
+                            cameraId.c_str());
                     continue;
                 }
-                listener.second->getListener()->onStatusChanged(mapToInterface(status),
+                listener->getListener()->onStatusChanged(mapToInterface(status),
                         String16(cameraId));
             }
         });
@@ -3467,4 +3560,25 @@
         "  help print this message\n");
 }
 
+int32_t CameraService::updateAudioRestriction() {
+    Mutex::Autolock lock(mServiceLock);
+    return updateAudioRestrictionLocked();
+}
+
+int32_t CameraService::updateAudioRestrictionLocked() {
+    int32_t mode = 0;
+    // iterate through all active client
+    for (const auto& i : mActiveClientManager.getAll()) {
+        const auto clientSp = i->getValue();
+        mode |= clientSp->getAudioRestriction();
+    }
+
+    bool modeChanged = (mAudioRestriction != mode);
+    mAudioRestriction = mode;
+    if (modeChanged) {
+        mAppOps.setCameraAudioRestriction(mode);
+    }
+    return mode;
+}
+
 }; // namespace android
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index cf93a41..df8c17c 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -258,6 +258,14 @@
 
         // Block the client form using the camera
         virtual void block();
+
+        // set audio restriction from client
+        // Will call into camera service and hold mServiceLock
+        virtual int32_t setAudioRestriction(int32_t mode);
+
+        virtual int32_t getAudioRestriction() const;
+
+        static bool isValidAudioRestriction(int32_t mode);
     protected:
         BasicClient(const sp<CameraService>& cameraService,
                 const sp<IBinder>& remoteCallback,
@@ -286,6 +294,9 @@
         const pid_t                     mServicePid;
         bool                            mDisconnected;
 
+        mutable Mutex                   mAudioRestrictionLock;
+        int32_t                         mAudioRestriction;
+
         // - The app-side Binder interface to receive callbacks from us
         sp<IBinder>                     mRemoteBinder;   // immutable after constructor
 
@@ -439,6 +450,9 @@
 
     }; // class CameraClientManager
 
+    int32_t updateAudioRestriction();
+    int32_t updateAudioRestrictionLocked();
+
 private:
 
     typedef hardware::camera::common::V1_0::CameraDeviceStatus CameraDeviceStatus;
@@ -633,9 +647,20 @@
         sp<BasicClient>* client,
         std::shared_ptr<resource_policy::ClientDescriptor<String8, sp<BasicClient>>>* partial);
 
-    // Should an operation attempt on a cameraId be rejected, if the camera id is
-    // advertised as a publically hidden secure camera, by the camera HAL ?
-    bool shouldRejectHiddenCameraConnection(const String8 & cameraId);
+    // Should an operation attempt on a cameraId be rejected ? (this can happen
+    // under various conditions. For example if a camera device is advertised as
+    // system only or hidden secure camera, amongst possible others.
+    bool shouldRejectSystemCameraConnection(const String8 & cameraId) const;
+
+    // Should a device status update be skipped for a particular camera device ? (this can happen
+    // under various conditions. For example if a camera device is advertised as
+    // system only or hidden secure camera, amongst possible others.
+    bool shouldSkipStatusUpdates(const String8& cameraId, bool isVendorListener, int clientPid,
+            int clientUid) const;
+
+    inline SystemCameraKind getSystemCameraKind(const String8& cameraId) const {
+        return mCameraProviderManager->getSystemCameraKind(cameraId.c_str());
+    }
 
     // Single implementation shared between the various connect calls
     template<class CALLBACK, class CLIENT>
@@ -810,7 +835,9 @@
     class ServiceListener : public virtual IBinder::DeathRecipient {
         public:
             ServiceListener(sp<CameraService> parent, sp<hardware::ICameraServiceListener> listener,
-                    int uid) : mParent(parent), mListener(listener), mListenerUid(uid) {}
+                    int uid, int pid, bool isVendorClient)
+                    : mParent(parent), mListener(listener), mListenerUid(uid), mListenerPid(pid),
+                      mIsVendorListener(isVendorClient) { }
 
             status_t initialize() {
                 return IInterface::asBinder(mListener)->linkToDeath(this);
@@ -824,16 +851,20 @@
             }
 
             int getListenerUid() { return mListenerUid; }
+            int getListenerPid() { return mListenerPid; }
             sp<hardware::ICameraServiceListener> getListener() { return mListener; }
+            bool isVendorListener() { return mIsVendorListener; }
 
         private:
             wp<CameraService> mParent;
             sp<hardware::ICameraServiceListener> mListener;
-            int mListenerUid;
+            int mListenerUid = -1;
+            int mListenerPid = -1;
+            bool mIsVendorListener = false;
     };
 
     // Guarded by mStatusListenerMutex
-    std::vector<std::pair<bool, sp<ServiceListener>>> mListenerList;
+    std::vector<sp<ServiceListener>> mListenerList;
 
     Mutex       mStatusListenerLock;
 
@@ -949,6 +980,13 @@
 
     void broadcastTorchModeStatus(const String8& cameraId,
             hardware::camera::common::V1_0::TorchModeStatus status);
+
+    // TODO: right now each BasicClient holds one AppOpsManager instance.
+    // We can refactor the code so all of clients share this instance
+    AppOpsManager mAppOps;
+
+    // Aggreated audio restriction mode for all camera clients
+    int32_t mAudioRestriction;
 };
 
 } // namespace android
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index 162b50f..e996e83 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -959,6 +959,11 @@
         case Parameters::RECORD:
         case Parameters::PREVIEW:
             syncWithDevice();
+            // Due to flush a camera device sync is not a sufficient
+            // guarantee that the current client parameters are
+            // correctly applied. To resolve this wait for the current
+            // request id to return in the results.
+            waitUntilCurrentRequestIdLocked();
             res = stopStream();
             if (res != OK) {
                 ALOGE("%s: Camera %d: Can't stop streaming: %s (%d)",
@@ -2253,6 +2258,52 @@
     return OK;
 }
 
+int32_t Camera2Client::setAudioRestriction(int /*mode*/) {
+    // Empty implementation. setAudioRestriction is hidden interface and not
+    // supported by android.hardware.Camera API
+    return INVALID_OPERATION;
+}
+
+status_t Camera2Client::waitUntilCurrentRequestIdLocked() {
+    int32_t activeRequestId = mStreamingProcessor->getActiveRequestId();
+    if (activeRequestId != 0) {
+        auto res = waitUntilRequestIdApplied(activeRequestId,
+                mDevice->getExpectedInFlightDuration());
+        if (res == TIMED_OUT) {
+            ALOGE("%s: Camera %d: Timed out waiting for current request id to return in results!",
+                    __FUNCTION__, mCameraId);
+            return res;
+        } else if (res != OK) {
+            ALOGE("%s: Camera %d: Error while waiting for current request id to return in results!",
+                    __FUNCTION__, mCameraId);
+            return res;
+        }
+    }
+
+    return OK;
+}
+
+status_t Camera2Client::waitUntilRequestIdApplied(int32_t requestId, nsecs_t timeout) {
+    Mutex::Autolock l(mLatestRequestMutex);
+    while (mLatestRequestId != requestId) {
+        nsecs_t startTime = systemTime();
+
+        auto res = mLatestRequestSignal.waitRelative(mLatestRequestMutex, timeout);
+        if (res != OK) return res;
+
+        timeout -= (systemTime() - startTime);
+    }
+
+    return OK;
+}
+
+void Camera2Client::notifyRequestId(int32_t requestId) {
+    Mutex::Autolock al(mLatestRequestMutex);
+
+    mLatestRequestId = requestId;
+    mLatestRequestSignal.signal();
+}
+
 const char* Camera2Client::kAutofocusLabel = "autofocus";
 const char* Camera2Client::kTakepictureLabel = "take_picture";
 
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index a9ea271..e79a442 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -83,6 +83,7 @@
     virtual void            notifyError(int32_t errorCode,
                                         const CaptureResultExtras& resultExtras);
     virtual status_t        setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer);
+    virtual int32_t         setAudioRestriction(int mode);
 
     /**
      * Interface used by CameraService
@@ -122,6 +123,8 @@
 
     camera2::SharedParameters& getParameters();
 
+    void notifyRequestId(int32_t requestId);
+
     int getPreviewStreamId() const;
     int getCaptureStreamId() const;
     int getCallbackStreamId() const;
@@ -227,6 +230,12 @@
     status_t initializeImpl(TProviderPtr providerPtr, const String8& monitorTags);
 
     bool isZslEnabledInStillTemplate();
+
+    mutable Mutex mLatestRequestMutex;
+    Condition mLatestRequestSignal;
+    int32_t mLatestRequestId = -1;
+    status_t waitUntilRequestIdApplied(int32_t requestId, nsecs_t timeout);
+    status_t waitUntilCurrentRequestIdLocked();
 };
 
 }; // namespace android
diff --git a/services/camera/libcameraservice/api1/CameraClient.cpp b/services/camera/libcameraservice/api1/CameraClient.cpp
index d65ac7b..089f6cf 100644
--- a/services/camera/libcameraservice/api1/CameraClient.cpp
+++ b/services/camera/libcameraservice/api1/CameraClient.cpp
@@ -1171,4 +1171,18 @@
     return INVALID_OPERATION;
 }
 
+int32_t CameraClient::setAudioRestriction(int mode) {
+    if (!isValidAudioRestriction(mode)) {
+        ALOGE("%s: invalid audio restriction mode %d", __FUNCTION__, mode);
+        return BAD_VALUE;
+    }
+
+    Mutex::Autolock lock(mLock);
+    if (checkPidAndHardware() != NO_ERROR) {
+        return INVALID_OPERATION;
+    }
+    return BasicClient::setAudioRestriction(mode);
+}
+
+
 }; // namespace android
diff --git a/services/camera/libcameraservice/api1/CameraClient.h b/services/camera/libcameraservice/api1/CameraClient.h
index 9530b6c..fefa8c9 100644
--- a/services/camera/libcameraservice/api1/CameraClient.h
+++ b/services/camera/libcameraservice/api1/CameraClient.h
@@ -59,6 +59,7 @@
     virtual String8         getParameters() const;
     virtual status_t        sendCommand(int32_t cmd, int32_t arg1, int32_t arg2);
     virtual status_t        setVideoTarget(const sp<IGraphicBufferProducer>& bufferProducer);
+    virtual int32_t         setAudioRestriction(int mode);
 
     // Interface used by CameraService
     CameraClient(const sp<CameraService>& cameraService,
diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
index 683e84d..63e293a 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.cpp
@@ -86,6 +86,12 @@
         process3aState(frame, client);
     }
 
+    if (mCurrentRequestId != frame.mResultExtras.requestId) {
+        mCurrentRequestId = frame.mResultExtras.requestId;
+
+        client->notifyRequestId(mCurrentRequestId);
+    }
+
     return FrameProcessorBase::processSingleFrame(frame, device);
 }
 
diff --git a/services/camera/libcameraservice/api1/client2/FrameProcessor.h b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
index 8183c12..142b8cd 100644
--- a/services/camera/libcameraservice/api1/client2/FrameProcessor.h
+++ b/services/camera/libcameraservice/api1/client2/FrameProcessor.h
@@ -94,6 +94,7 @@
     };
 
     AlgState m3aState;
+    int32_t mCurrentRequestId = -1;
 
     // frame number -> pending 3A states that not all data are received yet.
     KeyedVector<int32_t, AlgState> mPending3AStates;
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index c7a4f2b..be188bc 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -1870,6 +1870,26 @@
     return res;
 }
 
+binder::Status CameraDeviceClient::setCameraAudioRestriction(int32_t mode,
+        /*out*/ int32_t* outMode) {
+    ATRACE_CALL();
+    binder::Status res;
+    if (!(res = checkPidStatus(__FUNCTION__)).isOk()) return res;
+
+    if (!isValidAudioRestriction(mode)) {
+        String8 msg = String8::format("Camera %s: invalid audio restriction mode %d",
+                mCameraIdStr.string(), mode);
+        ALOGW("%s: %s", __FUNCTION__, msg.string());
+        return STATUS_ERROR(CameraService::ERROR_ILLEGAL_ARGUMENT, msg.string());
+    }
+
+    Mutex::Autolock icl(mBinderSerializationLock);
+    if (outMode != nullptr) {
+        *outMode = BasicClient::setAudioRestriction(mode);
+    }
+    return binder::Status::ok();
+}
+
 status_t CameraDeviceClient::dump(int fd, const Vector<String16>& args) {
     return BasicClient::dump(fd, args);
 }
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index 1c5abb0..a9aa190 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -152,6 +152,10 @@
     virtual binder::Status finalizeOutputConfigurations(int32_t streamId,
             const hardware::camera2::params::OutputConfiguration &outputConfiguration) override;
 
+    virtual binder::Status setCameraAudioRestriction(int32_t mode,
+            /*out*/
+            int32_t* outMode = NULL) override;
+
     /**
      * Interface used by CameraService
      */
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index 98c1b5e..935bc37 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -383,6 +383,12 @@
      * drop buffers for stream of streamId.
      */
     virtual status_t dropStreamBuffers(bool /*dropping*/, int /*streamId*/) = 0;
+
+    /**
+     * Returns the maximum expected time it'll take for all currently in-flight
+     * requests to complete, based on their settings
+     */
+    virtual nsecs_t getExpectedInFlightDuration() = 0;
 };
 
 }; // namespace android
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.cpp b/services/camera/libcameraservice/common/CameraProviderManager.cpp
index c72029f..c21bd69 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.cpp
+++ b/services/camera/libcameraservice/common/CameraProviderManager.cpp
@@ -534,15 +534,23 @@
     }
 }
 
-bool CameraProviderManager::ProviderInfo::DeviceInfo3::isPublicallyHiddenSecureCamera() {
+SystemCameraKind CameraProviderManager::ProviderInfo::DeviceInfo3::getSystemCameraKind() {
     camera_metadata_entry_t entryCap;
     entryCap = mCameraCharacteristics.find(ANDROID_REQUEST_AVAILABLE_CAPABILITIES);
-    if (entryCap.count != 1) {
-        // Do NOT hide this camera device if the capabilities specify anything more
-        // than ANDROID_REQUEST_AVAILABLE_CAPABILITIES_SECURE_IMAGE_DATA.
-        return false;
+    if (entryCap.count == 1 &&
+            entryCap.data.u8[0] == ANDROID_REQUEST_AVAILABLE_CAPABILITIES_SECURE_IMAGE_DATA) {
+        return SystemCameraKind::HIDDEN_SECURE_CAMERA;
     }
-    return entryCap.data.u8[0] == ANDROID_REQUEST_AVAILABLE_CAPABILITIES_SECURE_IMAGE_DATA;
+
+    // Go through the capabilities and check if it has
+    // ANDROID_REQUEST_AVAILABLE_CAPABILITIES_SYSTEM_CAMERA
+    for (size_t i = 0; i < entryCap.count; ++i) {
+        uint8_t capability = entryCap.data.u8[i];
+        if (capability == ANDROID_REQUEST_AVAILABLE_CAPABILITIES_SYSTEM_CAMERA) {
+            return SystemCameraKind::SYSTEM_ONLY_CAMERA;
+        }
+    }
+    return SystemCameraKind::PUBLIC;
 }
 
 void CameraProviderManager::ProviderInfo::DeviceInfo3::getSupportedSizes(
@@ -1046,14 +1054,14 @@
     return deviceInfo->mIsLogicalCamera;
 }
 
-bool CameraProviderManager::isPublicallyHiddenSecureCamera(const std::string& id) {
+SystemCameraKind CameraProviderManager::getSystemCameraKind(const std::string& id) {
     std::lock_guard<std::mutex> lock(mInterfaceMutex);
 
     auto deviceInfo = findDeviceInfoLocked(id);
     if (deviceInfo == nullptr) {
-        return false;
+        return SystemCameraKind::PUBLIC;
     }
-    return deviceInfo->mIsPublicallyHiddenSecureCamera;
+    return deviceInfo->mSystemCameraKind;
 }
 
 bool CameraProviderManager::isHiddenPhysicalCamera(const std::string& cameraId) {
@@ -1937,7 +1945,7 @@
         return;
     }
 
-    mIsPublicallyHiddenSecureCamera = isPublicallyHiddenSecureCamera();
+    mSystemCameraKind = getSystemCameraKind();
 
     status_t res = fixupMonochromeTags();
     if (OK != res) {
diff --git a/services/camera/libcameraservice/common/CameraProviderManager.h b/services/camera/libcameraservice/common/CameraProviderManager.h
index 8cdfc24..801e978 100644
--- a/services/camera/libcameraservice/common/CameraProviderManager.h
+++ b/services/camera/libcameraservice/common/CameraProviderManager.h
@@ -54,6 +54,26 @@
             sp<VendorTagDescriptor>& descriptor);
 };
 
+enum SystemCameraKind {
+   /**
+    * These camera devices are visible to all apps and system components alike
+    */
+   PUBLIC = 0,
+
+   /**
+    * These camera devices are visible only to processes having the
+    * android.permission.SYSTEM_CAMERA permission. They are not exposed to 3P
+    * apps.
+    */
+   SYSTEM_ONLY_CAMERA,
+
+   /**
+    * These camera devices are visible only to HAL clients (that try to connect
+    * on a hwbinder thread).
+    */
+   HIDDEN_SECURE_CAMERA
+};
+
 /**
  * A manager for all camera providers available on an Android device.
  *
@@ -272,7 +292,7 @@
      */
     bool isLogicalCamera(const std::string& id, std::vector<std::string>* physicalCameraIds);
 
-    bool isPublicallyHiddenSecureCamera(const std::string& id);
+    SystemCameraKind getSystemCameraKind(const std::string& id);
     bool isHiddenPhysicalCamera(const std::string& cameraId);
 
     static const float kDepthARTolerance;
@@ -379,7 +399,7 @@
             std::vector<std::string> mPhysicalIds;
             hardware::CameraInfo mInfo;
             sp<IBase> mSavedInterface;
-            bool mIsPublicallyHiddenSecureCamera = false;
+            SystemCameraKind mSystemCameraKind = SystemCameraKind::PUBLIC;
 
             const hardware::camera::common::V1_0::CameraResourceCost mResourceCost;
 
@@ -497,7 +517,7 @@
             CameraMetadata mCameraCharacteristics;
             std::unordered_map<std::string, CameraMetadata> mPhysicalCameraCharacteristics;
             void queryPhysicalCameraIds();
-            bool isPublicallyHiddenSecureCamera();
+            SystemCameraKind getSystemCameraKind();
             status_t fixupMonochromeTags();
             status_t addDynamicDepthTags();
             static void getSupportedSizes(const CameraMetadata& ch, uint32_t tag,
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index 4227a3b..dd5a62b 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -3679,7 +3679,7 @@
         // Did we get the (final) result metadata for this capture?
         if (result->result != NULL && !isPartialResult) {
             if (request.physicalCameraIds.size() != result->num_physcam_metadata) {
-                SET_ERR("Requested physical Camera Ids %d not equal to number of metadata %d",
+                SET_ERR("Expected physical Camera metadata count %d not equal to actual count %d",
                         request.physicalCameraIds.size(), result->num_physcam_metadata);
                 return;
             }
@@ -3873,12 +3873,14 @@
                             errorCode) {
                         if (physicalCameraId.size() > 0) {
                             String8 cameraId(physicalCameraId);
-                            if (r.physicalCameraIds.find(cameraId) == r.physicalCameraIds.end()) {
+                            auto iter = r.physicalCameraIds.find(cameraId);
+                            if (iter == r.physicalCameraIds.end()) {
                                 ALOGE("%s: Reported result failure for physical camera device: %s "
                                         " which is not part of the respective request!",
                                         __FUNCTION__, cameraId.string());
                                 break;
                             }
+                            r.physicalCameraIds.erase(iter);
                             resultExtras.errorPhysicalCameraId = physicalCameraId;
                         } else {
                             logicalDeviceResultError = true;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index cae34ce..2573b48 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -194,6 +194,8 @@
      */
     status_t dropStreamBuffers(bool dropping, int streamId) override;
 
+    nsecs_t getExpectedInFlightDuration() override;
+
     /**
      * Helper functions to map between framework and HIDL values
      */
@@ -1111,12 +1113,6 @@
             const SurfaceMap& outputSurfaces);
 
     /**
-     * Returns the maximum expected time it'll take for all currently in-flight
-     * requests to complete, based on their settings
-     */
-    nsecs_t getExpectedInFlightDuration();
-
-    /**
      * Tracking for idle detection
      */
     sp<camera3::StatusTracker> mStatusTracker;
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
index acb8b3c..e1d35e8 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.cpp
@@ -54,9 +54,8 @@
         mState = STATE_ERROR;
     }
 
-    if (setId > CAMERA3_STREAM_SET_ID_INVALID) {
-        mBufferReleasedListener = new BufferReleasedListener(this);
-    }
+    bool needsReleaseNotify = setId > CAMERA3_STREAM_SET_ID_INVALID;
+    mBufferProducerListener = new BufferProducerListener(this, needsReleaseNotify);
 }
 
 Camera3OutputStream::Camera3OutputStream(int id,
@@ -87,9 +86,8 @@
         mState = STATE_ERROR;
     }
 
-    if (setId > CAMERA3_STREAM_SET_ID_INVALID) {
-        mBufferReleasedListener = new BufferReleasedListener(this);
-    }
+    bool needsReleaseNotify = setId > CAMERA3_STREAM_SET_ID_INVALID;
+    mBufferProducerListener = new BufferProducerListener(this, needsReleaseNotify);
 }
 
 Camera3OutputStream::Camera3OutputStream(int id,
@@ -124,10 +122,8 @@
     }
 
     mConsumerName = String8("Deferred");
-    if (setId > CAMERA3_STREAM_SET_ID_INVALID) {
-        mBufferReleasedListener = new BufferReleasedListener(this);
-    }
-
+    bool needsReleaseNotify = setId > CAMERA3_STREAM_SET_ID_INVALID;
+    mBufferProducerListener = new BufferProducerListener(this, needsReleaseNotify);
 }
 
 Camera3OutputStream::Camera3OutputStream(int id, camera3_stream_type_t type,
@@ -151,9 +147,8 @@
         mDropBuffers(false),
         mDequeueBufferLatency(kDequeueLatencyBinSize) {
 
-    if (setId > CAMERA3_STREAM_SET_ID_INVALID) {
-        mBufferReleasedListener = new BufferReleasedListener(this);
-    }
+    bool needsReleaseNotify = setId > CAMERA3_STREAM_SET_ID_INVALID;
+    mBufferProducerListener = new BufferProducerListener(this, needsReleaseNotify);
 
     // Subclasses expected to initialize mConsumer themselves
 }
@@ -261,7 +256,7 @@
         notifyBufferReleased(anwBuffer);
         if (mUseBufferManager) {
             // Return this buffer back to buffer manager.
-            mBufferReleasedListener->onBufferReleased();
+            mBufferProducerListener->onBufferReleased();
         }
     } else {
         if (mTraceFirstBuffer && (stream_type == CAMERA3_STREAM_OUTPUT)) {
@@ -387,8 +382,8 @@
     // Configure consumer-side ANativeWindow interface. The listener may be used
     // to notify buffer manager (if it is used) of the returned buffers.
     res = mConsumer->connect(NATIVE_WINDOW_API_CAMERA,
-            /*listener*/mBufferReleasedListener,
-            /*reportBufferRemoval*/true);
+            /*reportBufferRemoval*/true,
+            /*listener*/mBufferProducerListener);
     if (res != OK) {
         ALOGE("%s: Unable to connect to native window for stream %d",
                 __FUNCTION__, mId);
@@ -790,7 +785,7 @@
     return INVALID_OPERATION;
 }
 
-void Camera3OutputStream::BufferReleasedListener::onBufferReleased() {
+void Camera3OutputStream::BufferProducerListener::onBufferReleased() {
     sp<Camera3OutputStream> stream = mParent.promote();
     if (stream == nullptr) {
         ALOGV("%s: Parent camera3 output stream was destroyed", __FUNCTION__);
@@ -823,6 +818,25 @@
     }
 }
 
+void Camera3OutputStream::BufferProducerListener::onBuffersDiscarded(
+        const std::vector<sp<GraphicBuffer>>& buffers) {
+    sp<Camera3OutputStream> stream = mParent.promote();
+    if (stream == nullptr) {
+        ALOGV("%s: Parent camera3 output stream was destroyed", __FUNCTION__);
+        return;
+    }
+
+    if (buffers.size() > 0) {
+        Mutex::Autolock l(stream->mLock);
+        stream->onBuffersRemovedLocked(buffers);
+        if (stream->mUseBufferManager) {
+            stream->mBufferManager->onBuffersRemoved(stream->getId(),
+                    stream->getStreamSetId(), buffers.size());
+        }
+        ALOGV("Stream %d: %zu Buffers discarded.", stream->getId(), buffers.size());
+    }
+}
+
 void Camera3OutputStream::onBuffersRemovedLocked(
         const std::vector<sp<GraphicBuffer>>& removedBuffers) {
     sp<Camera3StreamBufferFreedListener> callback = mBufferFreedListener.promote();
diff --git a/services/camera/libcameraservice/device3/Camera3OutputStream.h b/services/camera/libcameraservice/device3/Camera3OutputStream.h
index 729c655..b4e49f9 100644
--- a/services/camera/libcameraservice/device3/Camera3OutputStream.h
+++ b/services/camera/libcameraservice/device3/Camera3OutputStream.h
@@ -146,18 +146,22 @@
      */
     virtual status_t setConsumers(const std::vector<sp<Surface>>& consumers);
 
-    class BufferReleasedListener : public BnProducerListener {
+    class BufferProducerListener : public SurfaceListener {
         public:
-          BufferReleasedListener(wp<Camera3OutputStream> parent) : mParent(parent) {}
+            BufferProducerListener(wp<Camera3OutputStream> parent, bool needsReleaseNotify)
+                    : mParent(parent), mNeedsReleaseNotify(needsReleaseNotify) {}
 
-          /**
-          * Implementation of IProducerListener, used to notify this stream that the consumer
-          * has returned a buffer and it is ready to return to Camera3BufferManager for reuse.
-          */
-          virtual void onBufferReleased();
+            /**
+            * Implementation of IProducerListener, used to notify this stream that the consumer
+            * has returned a buffer and it is ready to return to Camera3BufferManager for reuse.
+            */
+            virtual void onBufferReleased();
+            virtual bool needsReleaseNotify() { return mNeedsReleaseNotify; }
+            virtual void onBuffersDiscarded(const std::vector<sp<GraphicBuffer>>& buffers);
 
         private:
-          wp<Camera3OutputStream> mParent;
+            wp<Camera3OutputStream> mParent;
+            bool mNeedsReleaseNotify;
     };
 
     virtual status_t detachBuffer(sp<GraphicBuffer>* buffer, int* fenceFd);
@@ -262,10 +266,10 @@
     sp<Camera3BufferManager> mBufferManager;
 
     /**
-     * Buffer released listener, used to notify the buffer manager that a buffer is released
-     * from consumer side.
+     * Buffer producer listener, used to handle notification when a buffer is released
+     * from consumer side, or a set of buffers are discarded by the consumer.
      */
-    sp<BufferReleasedListener> mBufferReleasedListener;
+    sp<BufferProducerListener> mBufferProducerListener;
 
     /**
      * Flag indicating if the buffer manager is used to allocate the stream buffers
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.cpp b/services/camera/libcameraservice/device3/Camera3Stream.cpp
index fd9b4b0..f707ef8 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Stream.cpp
@@ -70,7 +70,7 @@
     mFormatOverridden(false),
     mOriginalFormat(format),
     mDataSpaceOverridden(false),
-    mOriginalDataSpace(HAL_DATASPACE_UNKNOWN),
+    mOriginalDataSpace(dataSpace),
     mPhysicalCameraId(physicalCameraId),
     mLastTimestamp(0) {
 
@@ -137,9 +137,6 @@
 
 void Camera3Stream::setDataSpaceOverride(bool dataSpaceOverridden) {
     mDataSpaceOverridden = dataSpaceOverridden;
-    if (dataSpaceOverridden && mOriginalDataSpace == HAL_DATASPACE_UNKNOWN) {
-        mOriginalDataSpace = camera3_stream::data_space;
-    }
 }
 
 bool Camera3Stream::isDataSpaceOverridden() const {
diff --git a/services/camera/libcameraservice/device3/Camera3Stream.h b/services/camera/libcameraservice/device3/Camera3Stream.h
index 67afd0f..805df82 100644
--- a/services/camera/libcameraservice/device3/Camera3Stream.h
+++ b/services/camera/libcameraservice/device3/Camera3Stream.h
@@ -588,7 +588,7 @@
 
     //Keep track of original dataSpace in case it gets overridden
     bool mDataSpaceOverridden;
-    android_dataspace mOriginalDataSpace;
+    const android_dataspace mOriginalDataSpace;
 
     String8 mPhysicalCameraId;
     nsecs_t mLastTimestamp;
diff --git a/services/camera/libcameraservice/device3/Camera3StreamSplitter.cpp b/services/camera/libcameraservice/device3/Camera3StreamSplitter.cpp
index 84c2ec7..80df7db 100644
--- a/services/camera/libcameraservice/device3/Camera3StreamSplitter.cpp
+++ b/services/camera/libcameraservice/device3/Camera3StreamSplitter.cpp
@@ -496,7 +496,7 @@
         mInputSlots[bufferItem.mSlot].mFrameNumber = bufferItem.mFrameNumber;
     } else {
         SP_LOGE("%s: Invalid input graphic buffer!", __FUNCTION__);
-        res = BAD_VALUE;
+        mOnFrameAvailableRes.store(BAD_VALUE);
         return;
     }
     bufferId = bufferItem.mGraphicBuffer->getId();
@@ -541,6 +541,11 @@
     mOnFrameAvailableRes.store(res);
 }
 
+void Camera3StreamSplitter::onFrameReplaced(const BufferItem& item) {
+    ATRACE_CALL();
+    onFrameAvailable(item);
+}
+
 void Camera3StreamSplitter::decrementBufRefCountLocked(uint64_t id, size_t surfaceId) {
     ATRACE_CALL();
 
diff --git a/services/camera/libcameraservice/device3/Camera3StreamSplitter.h b/services/camera/libcameraservice/device3/Camera3StreamSplitter.h
index 960f7aa..4eb455a 100644
--- a/services/camera/libcameraservice/device3/Camera3StreamSplitter.h
+++ b/services/camera/libcameraservice/device3/Camera3StreamSplitter.h
@@ -102,6 +102,13 @@
     void onFrameAvailable(const BufferItem& item) override;
 
     // From IConsumerListener
+    //
+    // Similar to onFrameAvailable, but buffer item is indeed replacing a buffer
+    // in the buffer queue. This can happen when buffer queue is in droppable
+    // mode.
+    void onFrameReplaced(const BufferItem& item) override;
+
+    // From IConsumerListener
     // We don't care about released buffers because we detach each buffer as
     // soon as we acquire it. See the comment for onBufferReleased below for
     // some clarifying notes about the name.
diff --git a/services/mediaanalytics/MediaAnalyticsService.cpp b/services/mediaanalytics/MediaAnalyticsService.cpp
index 0e7edfd..988c06b 100644
--- a/services/mediaanalytics/MediaAnalyticsService.cpp
+++ b/services/mediaanalytics/MediaAnalyticsService.cpp
@@ -169,6 +169,7 @@
     ALOGV("caller has uid=%d, embedded uid=%d", uid, uid_given);
 
     switch (uid)  {
+        case AID_DRM:
         case AID_MEDIA:
         case AID_MEDIA_CODEC:
         case AID_MEDIA_EX:
diff --git a/services/mediaanalytics/iface_statsd.cpp b/services/mediaanalytics/iface_statsd.cpp
index 6845f06..6fd4415 100644
--- a/services/mediaanalytics/iface_statsd.cpp
+++ b/services/mediaanalytics/iface_statsd.cpp
@@ -60,6 +60,7 @@
     { "audiotrack", statsd_audiotrack },
     { "codec", statsd_codec},
     { "drm.vendor.Google.WidevineCDM", statsd_widevineCDM },
+    { "drmmanager", statsd_drmmanager },
     { "extractor", statsd_extractor },
     { "mediadrm", statsd_mediadrm },
     { "nuplayer", statsd_nuplayer },
diff --git a/services/mediaanalytics/iface_statsd.h b/services/mediaanalytics/iface_statsd.h
index f85d303..014929b 100644
--- a/services/mediaanalytics/iface_statsd.h
+++ b/services/mediaanalytics/iface_statsd.h
@@ -30,5 +30,6 @@
 
 extern bool statsd_mediadrm(MediaAnalyticsItem *);
 extern bool statsd_widevineCDM(MediaAnalyticsItem *);
+extern bool statsd_drmmanager(MediaAnalyticsItem *);
 
 } // namespace android
diff --git a/services/mediaanalytics/statsd_drm.cpp b/services/mediaanalytics/statsd_drm.cpp
index 902483a..845383d 100644
--- a/services/mediaanalytics/statsd_drm.cpp
+++ b/services/mediaanalytics/statsd_drm.cpp
@@ -104,4 +104,38 @@
     return true;
 }
 
+// drmmanager
+bool statsd_drmmanager(MediaAnalyticsItem *item)
+{
+    if (item == NULL) return false;
+
+    nsecs_t timestamp = item->getTimestamp();
+    std::string pkgName = item->getPkgName();
+    int64_t pkgVersionCode = item->getPkgVersionCode();
+    int64_t mediaApexVersion = 0;
+
+    char *plugin_id = NULL;
+    (void) item->getCString("plugin_id", &plugin_id);
+    char *description = NULL;
+    (void) item->getCString("description", &description);
+    int32_t method_id = -1;
+    (void) item->getInt32("method_id", &method_id);
+    char *mime_types = NULL;
+    (void) item->getCString("mime_types", &mime_types);
+
+    if (enabled_statsd) {
+        android::util::stats_write(android::util::MEDIAMETRICS_DRMMANAGER_REPORTED,
+                                   timestamp, pkgName.c_str(), pkgVersionCode,
+                                   mediaApexVersion,
+                                   plugin_id, description,
+                                   method_id, mime_types);
+    } else {
+        ALOGV("NOT sending: drmmanager data");
+    }
+
+    free(plugin_id);
+    free(description);
+    free(mime_types);
+    return true;
+}
 } // namespace android
diff --git a/services/mediaextractor/Android.bp b/services/mediaextractor/Android.bp
index b812244..98cc69f 100644
--- a/services/mediaextractor/Android.bp
+++ b/services/mediaextractor/Android.bp
@@ -12,6 +12,7 @@
         "libstagefright",
         "libbinder",
         "libutils",
+        "liblog",
     ],
 }
 
diff --git a/services/oboeservice/AAudioService.cpp b/services/oboeservice/AAudioService.cpp
index e6a8375..af8c67b 100644
--- a/services/oboeservice/AAudioService.cpp
+++ b/services/oboeservice/AAudioService.cpp
@@ -78,6 +78,11 @@
     AAudioClientTracker::getInstance().registerClient(pid, client);
 }
 
+bool AAudioService::isCallerInService() {
+    return mAudioClient.clientPid == IPCThreadState::self()->getCallingPid() &&
+        mAudioClient.clientUid == IPCThreadState::self()->getCallingUid();
+}
+
 aaudio_handle_t AAudioService::openStream(const aaudio::AAudioStreamRequest &request,
                                           aaudio::AAudioStreamConfiguration &configurationOutput) {
     aaudio_result_t result = AAUDIO_OK;
@@ -105,8 +110,7 @@
     if (sharingMode == AAUDIO_SHARING_MODE_EXCLUSIVE) {
         // only trust audioserver for in service indication
         bool inService = false;
-        if (mAudioClient.clientPid == IPCThreadState::self()->getCallingPid() &&
-                mAudioClient.clientUid == IPCThreadState::self()->getCallingUid()) {
+        if (isCallerInService()) {
             inService = request.isInService();
         }
         serviceStream = new AAudioServiceStreamMMAP(*this, inService);
@@ -274,12 +278,14 @@
         result = AAUDIO_ERROR_INVALID_STATE;
     } else {
         const pid_t ownerPid = IPCThreadState::self()->getCallingPid(); // TODO review
+        int32_t priority = isCallerInService()
+            ? kRealTimeAudioPriorityService : kRealTimeAudioPriorityClient;
         serviceStream->setRegisteredThread(clientThreadId);
         int err = android::requestPriority(ownerPid, clientThreadId,
-                                           DEFAULT_AUDIO_PRIORITY, true /* isForApp */);
+                                           priority, true /* isForApp */);
         if (err != 0) {
             ALOGE("AAudioService::registerAudioThread(%d) failed, errno = %d, priority = %d",
-                  clientThreadId, errno, DEFAULT_AUDIO_PRIORITY);
+                  clientThreadId, errno, priority);
             result = AAUDIO_ERROR_INTERNAL;
         }
     }
diff --git a/services/oboeservice/AAudioService.h b/services/oboeservice/AAudioService.h
index d21b1cd..43a59c3 100644
--- a/services/oboeservice/AAudioService.h
+++ b/services/oboeservice/AAudioService.h
@@ -87,6 +87,10 @@
 
 private:
 
+    /** @return true if the client is the audioserver
+     */
+    bool isCallerInService();
+
     /**
      * Lookup stream and then validate access to the stream.
      * @param streamHandle
@@ -106,9 +110,10 @@
 
     aaudio::AAudioStreamTracker     mStreamTracker;
 
-    enum constants {
-        DEFAULT_AUDIO_PRIORITY = 2
-    };
+    // TODO  Extract the priority constants from services/audioflinger/Threads.cpp
+    // and share them with this code. Look for "kPriorityFastMixer".
+    static constexpr int32_t        kRealTimeAudioPriorityClient = 2;
+    static constexpr int32_t        kRealTimeAudioPriorityService = 3;
 };
 
 } /* namespace android */
diff --git a/services/oboeservice/Android.bp b/services/oboeservice/Android.bp
new file mode 100644
index 0000000..1b7a20c
--- /dev/null
+++ b/services/oboeservice/Android.bp
@@ -0,0 +1,57 @@
+// Copyright (C) 2019 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.
+
+cc_library_shared {
+
+    name: "libaaudioservice",
+
+    srcs: [
+        "AAudioClientTracker.cpp",
+        "AAudioEndpointManager.cpp",
+        "AAudioMixer.cpp",
+        "AAudioService.cpp",
+        "AAudioServiceEndpoint.cpp",
+        "AAudioServiceEndpointCapture.cpp",
+        "AAudioServiceEndpointMMAP.cpp",
+        "AAudioServiceEndpointPlay.cpp",
+        "AAudioServiceEndpointShared.cpp",
+        "AAudioServiceStreamBase.cpp",
+        "AAudioServiceStreamMMAP.cpp",
+        "AAudioServiceStreamShared.cpp",
+        "AAudioStreamTracker.cpp",
+        "AAudioThread.cpp",
+        "SharedMemoryProxy.cpp",
+        "SharedRingBuffer.cpp",
+        "TimestampScheduler.cpp",
+    ],
+
+    cflags: [
+        "-Wno-unused-parameter",
+        "-Wall",
+        "-Werror",
+    ],
+
+    shared_libs: [
+        "libaaudio_internal",
+        "libaudioclient",
+        "libaudioflinger",
+        "libbase",
+        "libbinder",
+        "libcutils",
+        "liblog",
+        "libmediautils",
+        "libutils",
+    ],
+
+}
diff --git a/services/oboeservice/Android.mk b/services/oboeservice/Android.mk
deleted file mode 100644
index 5e4cd39..0000000
--- a/services/oboeservice/Android.mk
+++ /dev/null
@@ -1,60 +0,0 @@
-LOCAL_PATH:= $(call my-dir)
-
-# AAudio Service
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := libaaudioservice
-LOCAL_MODULE_TAGS := optional
-
-LIBAAUDIO_DIR := ../../media/libaaudio
-LIBAAUDIO_SRC_DIR := $(LIBAAUDIO_DIR)/src
-
-LOCAL_C_INCLUDES := \
-    $(TOPDIR)frameworks/av/services/audioflinger \
-    $(call include-path-for, audio-utils) \
-    frameworks/native/include \
-    system/core/base/include \
-    $(TOP)/frameworks/av/media/libaaudio/include \
-    $(TOP)/frameworks/av/media/utils/include \
-    frameworks/native/include \
-    $(TOP)/external/tinyalsa/include \
-    $(TOP)/frameworks/av/media/libaaudio/src
-
-LOCAL_SRC_FILES += \
-    SharedMemoryProxy.cpp \
-    SharedRingBuffer.cpp \
-    AAudioClientTracker.cpp \
-    AAudioEndpointManager.cpp \
-    AAudioMixer.cpp \
-    AAudioService.cpp \
-    AAudioServiceEndpoint.cpp \
-    AAudioServiceEndpointCapture.cpp \
-    AAudioServiceEndpointMMAP.cpp \
-    AAudioServiceEndpointPlay.cpp \
-    AAudioServiceEndpointShared.cpp \
-    AAudioServiceStreamBase.cpp \
-    AAudioServiceStreamMMAP.cpp \
-    AAudioServiceStreamShared.cpp \
-    AAudioStreamTracker.cpp \
-    TimestampScheduler.cpp \
-    AAudioThread.cpp
-
-LOCAL_MULTILIB := $(AUDIOSERVER_MULTILIB)
-
-# LOCAL_CFLAGS += -fvisibility=hidden
-LOCAL_CFLAGS += -Wno-unused-parameter
-LOCAL_CFLAGS += -Wall -Werror
-
-LOCAL_SHARED_LIBRARIES :=  \
-    libaaudio_internal \
-    libaudioflinger \
-    libaudioclient \
-    libbinder \
-    libcutils \
-    libmediautils \
-    libutils \
-    liblog
-
-include $(BUILD_SHARED_LIBRARY)
-
-