diff --git a/PREUPLOAD.cfg b/PREUPLOAD.cfg
index 716b550..1f7083b 100644
--- a/PREUPLOAD.cfg
+++ b/PREUPLOAD.cfg
@@ -11,3 +11,4 @@
 clang_format = --commit ${PREUPLOAD_COMMIT} --style file --extensions c,h,cc,cpp
                media/libmediatranscoding/
                services/mediatranscoding/
+               media/libaudioclient/tests/
diff --git a/apex/manifest.json b/apex/manifest.json
index f908f97..932d28e 100644
--- a/apex/manifest.json
+++ b/apex/manifest.json
@@ -1,6 +1,6 @@
 {
   "name": "com.android.media",
-  "version": 330090000,
+  "version": 990090000,
   "requireNativeLibs": [
     "libandroid.so",
     "libbinder_ndk.so",
diff --git a/apex/manifest_codec.json b/apex/manifest_codec.json
index 01a85ae..f42f5a4 100644
--- a/apex/manifest_codec.json
+++ b/apex/manifest_codec.json
@@ -1,6 +1,6 @@
 {
   "name": "com.android.media.swcodec",
-  "version": 330090000,
+  "version": 990090000,
   "requireNativeLibs": [
     ":sphal"
   ]
diff --git a/camera/VendorTagDescriptor.cpp b/camera/VendorTagDescriptor.cpp
index b37803a..151b653 100644
--- a/camera/VendorTagDescriptor.cpp
+++ b/camera/VendorTagDescriptor.cpp
@@ -598,7 +598,6 @@
 status_t VendorTagDescriptor::setAsGlobalVendorTagDescriptor(const sp<VendorTagDescriptor>& desc) {
     status_t res = OK;
     Mutex::Autolock al(sLock);
-    sGlobalVendorTagDescriptor = desc;
 
     vendor_tag_ops_t* opsPtr = NULL;
     if (desc != NULL) {
@@ -613,6 +612,9 @@
         ALOGE("%s: Could not set vendor tag descriptor, received error %s (%d)."
                 , __FUNCTION__, strerror(-res), res);
     }
+
+    sGlobalVendorTagDescriptor = desc;
+
     return res;
 }
 
@@ -631,7 +633,6 @@
         const sp<VendorTagDescriptorCache>& cache) {
     status_t res = OK;
     Mutex::Autolock al(sLock);
-    sGlobalVendorTagDescriptorCache = cache;
 
     struct vendor_tag_cache_ops* opsPtr = NULL;
     if (cache != NULL) {
@@ -646,6 +647,9 @@
         ALOGE("%s: Could not set vendor tag cache, received error %s (%d)."
                 , __FUNCTION__, strerror(-res), res);
     }
+
+    sGlobalVendorTagDescriptorCache = cache;
+
     return res;
 }
 
diff --git a/drm/libmediadrm/DrmMetricsConsumer.cpp b/drm/libmediadrm/DrmMetricsConsumer.cpp
index c06f09b..fd095b7 100644
--- a/drm/libmediadrm/DrmMetricsConsumer.cpp
+++ b/drm/libmediadrm/DrmMetricsConsumer.cpp
@@ -42,7 +42,7 @@
         }
         return type_names[attribute];
     }
-    
+
     static const char *type_names[] = {"PROVISION_REQUIRED", "KEY_NEEDED",
                                        "KEY_EXPIRED", "VENDOR_DEFINED",
                                        "SESSION_RECLAIMED"};
diff --git a/drm/mediadrm/plugins/clearkey/aidl/Android.bp b/drm/mediadrm/plugins/clearkey/aidl/Android.bp
index 2d1f741..397e4c6 100644
--- a/drm/mediadrm/plugins/clearkey/aidl/Android.bp
+++ b/drm/mediadrm/plugins/clearkey/aidl/Android.bp
@@ -69,3 +69,78 @@
         "android.hardware.drm-service.clearkey",
     ],
 }
+
+cc_defaults {
+    name: "fuzz_aidl_clearkey_service_defaults",
+
+    srcs: [
+        "CreatePluginFactories.cpp",
+        "CryptoPlugin.cpp",
+        "DrmFactory.cpp",
+        "DrmPlugin.cpp",
+    ],
+
+    relative_install_path: "hw",
+
+    cflags: ["-Wall", "-Werror", "-Wthread-safety"],
+
+    include_dirs: ["frameworks/av/include"],
+
+    shared_libs: [
+        "libbase",
+        "libbinder_ndk",
+        "libcrypto",
+        "liblog",
+        "libprotobuf-cpp-lite",
+        "libutils",
+        "android.hardware.drm-V1-ndk",
+    ],
+
+    static_libs: [
+        "android.hardware.common-V2-ndk",
+        "libclearkeybase_fuzz",
+    ],
+
+    local_include_dirs: ["include"],
+
+    sanitize: {
+        integer_overflow: true,
+    },
+}
+
+cc_fuzz {
+    name: "android.hardware.drm-service.clearkey.aidl_fuzzer",
+    defaults: [
+        "fuzz_aidl_clearkey_service_defaults",
+    ],
+    static_libs: [
+        "libbase",
+        "libbinder_random_parcel",
+        "libcutils",
+        "liblog",
+        "libutils",
+    ],
+    target: {
+        android: {
+            shared_libs: [
+                "libbinder_ndk",
+                "libbinder",
+            ],
+        },
+        host: {
+            static_libs: [
+                "libbinder_ndk",
+                "libbinder",
+            ],
+        },
+        darwin: {
+            enabled: false,
+        },
+    },
+    srcs: ["fuzzer.cpp"],
+    fuzz_config: {
+        cc: [
+            "hamzeh@google.com",
+        ],
+    },
+}
\ No newline at end of file
diff --git a/drm/mediadrm/plugins/clearkey/aidl/fuzzer.cpp b/drm/mediadrm/plugins/clearkey/aidl/fuzzer.cpp
new file mode 100644
index 0000000..9ef331f
--- /dev/null
+++ b/drm/mediadrm/plugins/clearkey/aidl/fuzzer.cpp
@@ -0,0 +1,35 @@
+/*
+ * Copyright (C) 2022 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 <fuzzbinder/libbinder_ndk_driver.h>
+#include <fuzzer/FuzzedDataProvider.h>
+
+#include "CreatePluginFactories.h"
+
+#include <android/binder_manager.h>
+#include <android/binder_process.h>
+
+using ::aidl::android::hardware::drm::clearkey::createDrmFactory;
+using ::aidl::android::hardware::drm::clearkey::DrmFactory;
+
+using android::fuzzService;
+using ndk::SharedRefBase;
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t* data, size_t size) {
+    std::shared_ptr<DrmFactory> drmFactory = createDrmFactory();
+    fuzzService(drmFactory->asBinder().get(),  FuzzedDataProvider(data, size));
+
+    return 0;
+}
diff --git a/drm/mediadrm/plugins/clearkey/common/Android.bp b/drm/mediadrm/plugins/clearkey/common/Android.bp
index a6a5b28..6913df4 100644
--- a/drm/mediadrm/plugins/clearkey/common/Android.bp
+++ b/drm/mediadrm/plugins/clearkey/common/Android.bp
@@ -97,3 +97,54 @@
         integer_overflow: true,
     },
 }
+
+cc_library_static {
+    name: "libclearkeydevicefiles-protos.common_fuzz",
+
+    proto: {
+        export_proto_headers: true,
+        type: "lite",
+    },
+    srcs: ["protos/DeviceFiles.proto"],
+}
+
+cc_library_static {
+    name: "libclearkeybase_fuzz",
+
+    srcs: [
+        "AesCtrDecryptor.cpp",
+        "Base64.cpp",
+        "Buffer.cpp",
+        "ClearKeyUUID.cpp",
+        "DeviceFiles.cpp",
+        "InitDataParser.cpp",
+        "JsonWebKey.cpp",
+        "MemoryFileSystem.cpp",
+        "Session.cpp",
+        "SessionLibrary.cpp",
+        "Utils.cpp",
+    ],
+
+    cflags: ["-Wall", "-Werror"],
+
+    include_dirs: ["frameworks/av/include"],
+
+    shared_libs: [
+        "libutils",
+        "libcrypto",
+    ],
+
+    whole_static_libs: [
+        "libjsmn",
+        "libclearkeydevicefiles-protos.common_fuzz",
+    ],
+
+    export_include_dirs: [
+        "include",
+        "include/clearkeydrm",
+    ],
+
+    sanitize: {
+        integer_overflow: true,
+    },
+}
diff --git a/include/media/MmapStreamCallback.h b/include/media/MmapStreamCallback.h
index 31b8eb5..76ee6d7 100644
--- a/include/media/MmapStreamCallback.h
+++ b/include/media/MmapStreamCallback.h
@@ -37,12 +37,9 @@
 
     /**
      * The volume to be applied to the use case specified when opening the stream has changed
-     * \param[in] channels a channel mask containing all channels the volume should be applied to.
-     * \param[in] values the volume values to be applied to each channel. The size of the vector
-     *                   should correspond to the channel count retrieved with
-     *                   audio_channel_count_from_in_mask() or audio_channel_count_from_out_mask()
+     * \param[in] volume the new target volume
      */
-    virtual void onVolumeChanged(audio_channel_mask_t channels, Vector<float> values) = 0;
+    virtual void onVolumeChanged(float volume) = 0;
 
     /**
      * The device the stream is routed to/from has changed
diff --git a/include/private/media/VideoFrame.h b/include/private/media/VideoFrame.h
index d4025e5..97e0b1d 100644
--- a/include/private/media/VideoFrame.h
+++ b/include/private/media/VideoFrame.h
@@ -85,8 +85,6 @@
     uint32_t mSize;            // Number of bytes of frame data
     uint32_t mIccSize;         // Number of bytes of ICC data
     uint32_t mBitDepth;        // number of bits per R / G / B channel
-
-    // Adding new items must be 64-bit aligned.
 };
 
 }; // namespace android
diff --git a/media/TEST_MAPPING b/media/TEST_MAPPING
index 41fe080..3b7a314 100644
--- a/media/TEST_MAPPING
+++ b/media/TEST_MAPPING
@@ -45,6 +45,32 @@
     "platinum-postsubmit": [
         // runs regularly, independent of changes in this tree.
         // signals if changes elsewhere break media functionality
+        // @FlakyTest: in staged-postsubmit, but not postsubmit
+        {
+            "name": "CtsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.EncodeDecodeTest"
+                },
+                {
+                    "exclude-annotation": "androidx.test.filters.FlakyTest"
+                }
+            ]
+        },
+        {
+            "name": "CtsMediaCodecTestCases",
+            "options": [
+                {
+                    "include-filter": "android.media.codec.cts.DecodeEditEncodeTest"
+                },
+                {
+                    "exclude-annotation": "androidx.test.filters.FlakyTest"
+                }
+            ]
+        }
+    ],
+    "staged-platinum-postsubmit": [
+        // runs every four hours
         {
             "name": "CtsMediaCodecTestCases",
             "options": [
diff --git a/media/audioserver/audioserver.rc b/media/audioserver/audioserver.rc
index c4a6601..0bd0d88 100644
--- a/media/audioserver/audioserver.rc
+++ b/media/audioserver/audioserver.rc
@@ -7,11 +7,9 @@
     ioprio rt 4
     task_profiles ProcessCapacityHigh HighPerformance
     onrestart restart vendor.audio-hal
+    onrestart restart vendor.audio-hal-aidl
     onrestart restart vendor.audio-hal-4-0-msd
     onrestart restart audio_proxy_service
-    # Keep the original service names for backward compatibility
-    onrestart restart vendor.audio-hal-2-0
-    onrestart restart audio-hal-2-0
 
 on property:vts.native_server.on=1
     stop audioserver
@@ -20,42 +18,34 @@
 
 on property:init.svc.audioserver=stopped
     stop vendor.audio-hal
+    stop vendor.audio-hal-aidl
     stop vendor.audio-hal-4-0-msd
     stop audio_proxy_service
-    # Keep the original service names for backward compatibility
-    stop vendor.audio-hal-2-0
-    stop audio-hal-2-0
     # See b/155364397. Need to have HAL service running for VTS.
     # Can't use 'restart' because then HAL service would restart
     # audioserver bringing it back into running state.
     start vendor.audio-hal
+    start vendor.audio-hal-aidl
     start vendor.audio-hal-4-0-msd
     start audio_proxy_service
-    # Keep the original service names for backward compatibility
-    start vendor.audio-hal-2-0
-    start audio-hal-2-0
 
 on property:init.svc.audioserver=running
     start vendor.audio-hal
+    start vendor.audio-hal-aidl
     start vendor.audio-hal-4-0-msd
     start audio_proxy_service
-    # Keep the original service names for backward compatibility
-    start vendor.audio-hal-2-0
-    start audio-hal-2-0
 
 on property:sys.audio.restart.hal=1
     # See b/159966243. Avoid restart loop between audioserver and HAL.
     # Keep the original service names for backward compatibility
     stop vendor.audio-hal
+    stop vendor.audio-hal-aidl
     stop vendor.audio-hal-4-0-msd
     stop audio_proxy_service
-    stop vendor.audio-hal-2-0
-    stop audio-hal-2-0
     start vendor.audio-hal
+    start vendor.audio-hal-aidl
     start vendor.audio-hal-4-0-msd
     start audio_proxy_service
-    start vendor.audio-hal-2-0
-    start audio-hal-2-0
     # reset the property
     setprop sys.audio.restart.hal 0
 
diff --git a/media/codec2/components/vpx/C2SoftVpxDec.cpp b/media/codec2/components/vpx/C2SoftVpxDec.cpp
index c2ccfa0..5d7ef5c 100644
--- a/media/codec2/components/vpx/C2SoftVpxDec.cpp
+++ b/media/codec2/components/vpx/C2SoftVpxDec.cpp
@@ -68,8 +68,8 @@
                 DefineParam(mSize, C2_PARAMKEY_PICTURE_SIZE)
                 .withDefault(new C2StreamPictureSizeInfo::output(0u, 320, 240))
                 .withFields({
-                    C2F(mSize, width).inRange(2, 2048, 2),
-                    C2F(mSize, height).inRange(2, 2048, 2),
+                    C2F(mSize, width).inRange(2, 2048),
+                    C2F(mSize, height).inRange(2, 2048),
                 })
                 .withSetter(SizeSetter)
                 .build());
@@ -717,7 +717,12 @@
     }
 
     C2MemoryUsage usage = { C2MemoryUsage::CPU_READ, C2MemoryUsage::CPU_WRITE };
-    c2_status_t err = pool->fetchGraphicBlock(align(mWidth, 16), mHeight, format, usage, &block);
+    // We always create a graphic block that is width aligned to 16 and height
+    // aligned to 2. We set the correct "crop" value of the image in the call to
+    // createGraphicBuffer() by setting the correct image dimensions.
+    c2_status_t err = pool->fetchGraphicBlock(align(mWidth, 16),
+                                              align(mHeight, 2), format, usage,
+                                              &block);
     if (err != C2_OK) {
         ALOGE("fetchGraphicBlock for Output failed with status %d", err);
         work->result = err;
diff --git a/media/codec2/hidl/plugin/samples/SampleFilterPlugin.cpp b/media/codec2/hidl/plugin/samples/SampleFilterPlugin.cpp
index b942be7..da62111 100644
--- a/media/codec2/hidl/plugin/samples/SampleFilterPlugin.cpp
+++ b/media/codec2/hidl/plugin/samples/SampleFilterPlugin.cpp
@@ -807,7 +807,6 @@
     // affectedParams
     {
         C2StreamHdrStaticInfo::output::PARAM_TYPE,
-        C2StreamHdr10PlusInfo::output::PARAM_TYPE,
         C2StreamColorAspectsInfo::output::PARAM_TYPE,
     },
 };
diff --git a/media/codec2/sfplugin/CCodecConfig.cpp b/media/codec2/sfplugin/CCodecConfig.cpp
index 2643290..69efc5f 100644
--- a/media/codec2/sfplugin/CCodecConfig.cpp
+++ b/media/codec2/sfplugin/CCodecConfig.cpp
@@ -386,7 +386,7 @@
             // read back always as int
             float value;
             if (v.get(&value)) {
-                return (int32_t)value;
+                return (int32_t) (value + 0.5);
             }
             return C2Value();
         }));
@@ -932,7 +932,7 @@
         .limitTo(D::ENCODER & (D::CONFIG | D::PARAM)));
     add(ConfigMapper(KEY_FLAC_COMPRESSION_LEVEL, C2_PARAMKEY_COMPLEXITY, "value")
         .limitTo(D::AUDIO & D::ENCODER));
-    add(ConfigMapper("complexity", C2_PARAMKEY_COMPLEXITY, "value")
+    add(ConfigMapper(KEY_COMPLEXITY, C2_PARAMKEY_COMPLEXITY, "value")
         .limitTo(D::ENCODER & (D::CONFIG | D::PARAM)));
 
     add(ConfigMapper(KEY_GRID_COLUMNS, C2_PARAMKEY_TILE_LAYOUT, "columns")
diff --git a/media/extractors/mp4/MPEG4Extractor.cpp b/media/extractors/mp4/MPEG4Extractor.cpp
index 5ca874e..5a03992 100644
--- a/media/extractors/mp4/MPEG4Extractor.cpp
+++ b/media/extractors/mp4/MPEG4Extractor.cpp
@@ -4786,7 +4786,7 @@
         if (len2 == 0) {
             return ERROR_MALFORMED;
         }
-        if (offset >= csd_size || csd[offset] != 0x01) {
+        if (offset + len1 > csd_size || csd[offset] != 0x01) {
             return ERROR_MALFORMED;
         }
 
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 0c4a8f7..03930ac 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -1441,7 +1441,10 @@
 /**
  * Asynchronous request for the stream to flush.
  * Flushing will discard any pending data.
- * This call only works if the stream is pausing or paused. TODO review
+ * This call only works if the stream is OPEN, PAUSED, STOPPED, or FLUSHED.
+ * Calling this function when in other states,
+ * or calling from an AAudio callback function,
+ * will have no effect and an error will be returned.
  * Frame counters are not reset by a flush. They may be advanced.
  * After this call the state will be in {@link #AAUDIO_STREAM_STATE_FLUSHING} or
  * {@link #AAUDIO_STREAM_STATE_FLUSHED}.
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 9f0564f..34ecd25 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -605,13 +605,6 @@
     return AAUDIO_ERROR_INVALID_STATE;
 }
 
-aaudio_result_t AudioStreamInternal::updateStateMachine() {
-    if (isDataCallbackActive()) {
-        return AAUDIO_OK; // state is getting updated by the callback thread read/write call
-    }
-    return processCommands();
-}
-
 void AudioStreamInternal::logTimestamp(AAudioServiceMessage &command) {
     static int64_t oldPosition = 0;
     static int64_t oldTime = 0;
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index 2367572..4ea61d2 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -48,7 +48,7 @@
                                        int64_t *framePosition,
                                        int64_t *timeNanoseconds) override;
 
-    virtual aaudio_result_t updateStateMachine() override;
+    virtual aaudio_result_t processCommands() override;
 
     aaudio_result_t open(const AudioStreamBuilder &builder) override;
 
@@ -110,8 +110,6 @@
 
     aaudio_result_t drainTimestampsFromService();
 
-    aaudio_result_t processCommands();
-
     aaudio_result_t stopCallback_l();
 
     virtual void prepareBuffersForStart() {}
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 06f05b0..8a5186a 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -21,7 +21,9 @@
 #include <atomic>
 #include <stdint.h>
 
+#include <linux/futex.h>
 #include <media/MediaMetricsItem.h>
+#include <sys/syscall.h>
 
 #include <aaudio/AAudio.h>
 
@@ -362,34 +364,37 @@
 }
 
 void AudioStream::setState(aaudio_stream_state_t state) {
-    ALOGD("%s(s#%d) from %d to %d", __func__, getId(), mState, state);
-    if (state == mState) {
+    aaudio_stream_state_t oldState = mState.load();
+    ALOGD("%s(s#%d) from %d to %d", __func__, getId(), oldState, state);
+    if (state == oldState) {
         return; // no change
     }
     // Track transition to DISCONNECTED state.
     if (state == AAUDIO_STREAM_STATE_DISCONNECTED) {
         android::mediametrics::LogItem(mMetricsId)
                 .set(AMEDIAMETRICS_PROP_EVENT, AMEDIAMETRICS_PROP_EVENT_VALUE_DISCONNECT)
-                .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(getState()))
+                .set(AMEDIAMETRICS_PROP_STATE, AudioGlobal_convertStreamStateToText(oldState))
                 .record();
     }
     // CLOSED is a final state
-    if (mState == AAUDIO_STREAM_STATE_CLOSED) {
+    if (oldState == AAUDIO_STREAM_STATE_CLOSED) {
         ALOGW("%s(%d) tried to set to %d but already CLOSED", __func__, getId(), state);
 
     // Once CLOSING, we can only move to CLOSED state.
-    } else if (mState == AAUDIO_STREAM_STATE_CLOSING
+    } else if (oldState == AAUDIO_STREAM_STATE_CLOSING
                && state != AAUDIO_STREAM_STATE_CLOSED) {
         ALOGW("%s(%d) tried to set to %d but already CLOSING", __func__, getId(), state);
 
     // Once DISCONNECTED, we can only move to CLOSING or CLOSED state.
-    } else if (mState == AAUDIO_STREAM_STATE_DISCONNECTED
+    } else if (oldState == AAUDIO_STREAM_STATE_DISCONNECTED
                && !(state == AAUDIO_STREAM_STATE_CLOSING
                    || state == AAUDIO_STREAM_STATE_CLOSED)) {
         ALOGW("%s(%d) tried to set to %d but already DISCONNECTED", __func__, getId(), state);
 
     } else {
-        mState = state;
+        mState.store(state);
+        // Wake up a wakeForStateChange thread if it exists.
+        syscall(SYS_futex, &mState, FUTEX_WAKE_PRIVATE, INT_MAX, NULL, NULL, 0);
     }
 }
 
@@ -408,9 +413,15 @@
         if (durationNanos > timeoutNanoseconds) {
             durationNanos = timeoutNanoseconds;
         }
-        AudioClock::sleepForNanos(durationNanos);
-        timeoutNanoseconds -= durationNanos;
+        struct timespec time;
+        time.tv_sec = durationNanos / AAUDIO_NANOS_PER_SECOND;
+        // Add the fractional nanoseconds.
+        time.tv_nsec = durationNanos - (time.tv_sec * AAUDIO_NANOS_PER_SECOND);
 
+        // Sleep for durationNanos. If mState changes from the callback
+        // thread, this thread will wake up earlier.
+        syscall(SYS_futex, &mState, FUTEX_WAIT_PRIVATE, currentState, &time, NULL, 0);
+        timeoutNanoseconds -= durationNanos;
         aaudio_result_t result = updateStateMachine();
         if (result != AAUDIO_OK) {
             return result;
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 5fb4528..8dd5538 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -100,10 +100,17 @@
                                        int64_t *timeNanoseconds) = 0;
 
     /**
-     * Update state machine.()
-     * @return
+     * Update state machine.
+     * @return result of the operation.
      */
-    virtual aaudio_result_t updateStateMachine() = 0;
+    aaudio_result_t updateStateMachine() {
+        if (isDataCallbackActive()) {
+            return AAUDIO_OK; // state is getting updated by the callback thread read/write call
+        }
+        return processCommands();
+    };
+
+    virtual aaudio_result_t processCommands() = 0;
 
     // =========== End ABSTRACT methods ===========================
 
@@ -184,7 +191,7 @@
     // ============== Queries ===========================
 
     aaudio_stream_state_t getState() const {
-        return mState;
+        return mState.load();
     }
 
     virtual int32_t getBufferSize() const {
@@ -674,6 +681,8 @@
 
     const android::sp<MyPlayerBase>   mPlayerBase;
 
+    std::atomic<aaudio_stream_state_t>          mState{AAUDIO_STREAM_STATE_UNINITIALIZED};
+
     // These do not change after open().
     int32_t                     mSamplesPerFrame = AAUDIO_UNSPECIFIED;
     aaudio_channel_mask_t       mChannelMask = AAUDIO_UNSPECIFIED;
@@ -682,7 +691,6 @@
     aaudio_sharing_mode_t       mSharingMode = AAUDIO_SHARING_MODE_SHARED;
     bool                        mSharingModeMatchRequired = false; // must match sharing mode requested
     audio_format_t              mFormat = AUDIO_FORMAT_DEFAULT;
-    aaudio_stream_state_t       mState = AAUDIO_STREAM_STATE_UNINITIALIZED;
     aaudio_performance_mode_t   mPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE;
     int32_t                     mFramesPerBurst = 0;
     int32_t                     mBufferCapacity = 0;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index dd11169..f32ef65 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -127,7 +127,7 @@
             mCallbackEnabled.store(false);
         }
 
-        if (updateStateMachine() != AAUDIO_OK) {
+        if (processCommands() != AAUDIO_OK) {
             forceDisconnect();
             mCallbackEnabled.store(false);
         }
@@ -192,7 +192,7 @@
             mCallbackEnabled.store(false);
         }
 
-        if (updateStateMachine() != AAUDIO_OK) {
+        if (processCommands() != AAUDIO_OK) {
             forceDisconnect();
             mCallbackEnabled.store(false);
         }
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index 1e39e0f..9a136a7 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -364,8 +364,7 @@
     return checkForDisconnectRequest(false);
 }
 
-aaudio_result_t AudioStreamRecord::updateStateMachine()
-{
+aaudio_result_t AudioStreamRecord::processCommands() {
     aaudio_result_t result = AAUDIO_OK;
     aaudio_wrapping_frames_t position;
     status_t err;
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.h b/media/libaaudio/src/legacy/AudioStreamRecord.h
index 5ce73f9..252ff3c 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.h
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.h
@@ -58,7 +58,7 @@
 
     int64_t getFramesWritten() override;
 
-    aaudio_result_t updateStateMachine() override;
+    aaudio_result_t processCommands() override;
 
     aaudio_direction_t getDirection() const override {
         return AAUDIO_DIRECTION_INPUT;
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 6f1dc92..09caa5c 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -378,8 +378,7 @@
     return checkForDisconnectRequest(false);;
 }
 
-aaudio_result_t AudioStreamTrack::updateStateMachine()
-{
+aaudio_result_t AudioStreamTrack::processCommands() {
     status_t err;
     aaudio_wrapping_frames_t position;
     switch (getState()) {
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.h b/media/libaaudio/src/legacy/AudioStreamTrack.h
index 0f4d72b..1f877b5 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.h
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.h
@@ -79,7 +79,7 @@
         return AAUDIO_DIRECTION_OUTPUT;
     }
 
-    aaudio_result_t updateStateMachine() override;
+    aaudio_result_t processCommands() override;
 
     int64_t incrementClientFrameCounter(int32_t frames) override {
         return incrementFramesWritten(frames);
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index 11724e0..b0c9a0c 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -1932,7 +1932,7 @@
         case media::AudioPortType::SESSION:
             legacy.session = VALUE_OR_RETURN(
                     aidl2legacy_int32_t_audio_port_config_session_ext(
-                            VALUE_OR_RETURN(UNION_GET(aidl, session))));
+                            VALUE_OR_RETURN(UNION_GET(aidlSys, session))));
             return legacy;
 
     }
@@ -1966,9 +1966,9 @@
             return OK;
         }
         case AUDIO_PORT_TYPE_SESSION:
-            UNION_SET(*aidl, session, VALUE_OR_RETURN_STATUS(
+            UNION_SET(*aidl, unspecified, false);
+            UNION_SET(*aidlSys, session, VALUE_OR_RETURN_STATUS(
                             legacy2aidl_audio_port_config_session_ext_int32_t(legacy.session)));
-            UNION_SET(*aidlSys, unspecified, false);
             return OK;
     }
     LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
@@ -2816,7 +2816,7 @@
         case media::AudioPortType::SESSION:
             legacy.session = VALUE_OR_RETURN(
                     aidl2legacy_int32_t_audio_port_session_ext(
-                            VALUE_OR_RETURN(UNION_GET(aidl, session))));
+                            VALUE_OR_RETURN(UNION_GET(aidlSys, session))));
             return legacy;
 
     }
@@ -2852,9 +2852,9 @@
             return OK;
         }
         case AUDIO_PORT_TYPE_SESSION:
-            UNION_SET(*aidl, session, VALUE_OR_RETURN_STATUS(
+            UNION_SET(*aidl, unspecified, false);
+            UNION_SET(*aidlSys, session, VALUE_OR_RETURN_STATUS(
                             legacy2aidl_audio_port_session_ext_int32_t(legacy.session)));
-            UNION_SET(*aidlSys, unspecified, false);
             return OK;
     }
     LOG_ALWAYS_FATAL("Shouldn't get here"); // with -Werror,-Wswitch may compile-time fail
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 69a9c68..4c27d52 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -302,12 +302,26 @@
     double_loadable: true,
     vendor_available: true,
     srcs: [
+        "aidl/android/media/EffectConfig.aidl",
         "aidl/android/media/IEffect.aidl",
         "aidl/android/media/IEffectClient.aidl",
     ],
     imports: [
+        "android.media.audio.common.types-V1",
         "shared-file-region-aidl",
     ],
+    backend: {
+        cpp: {
+            min_sdk_version: "29",
+            apex_available: [
+                "//apex_available:platform",
+                "com.android.media",
+            ],
+        },
+        java: {
+            sdk_version: "module_current",
+        },
+    },
 }
 
 aidl_interface {
diff --git a/media/libaudioclient/AudioEffect.cpp b/media/libaudioclient/AudioEffect.cpp
index d447f0c..2870c4c 100644
--- a/media/libaudioclient/AudioEffect.cpp
+++ b/media/libaudioclient/AudioEffect.cpp
@@ -527,6 +527,36 @@
     return status;
 }
 
+status_t AudioEffect::getConfigs(
+        audio_config_base_t *inputCfg, audio_config_base_t *outputCfg)
+{
+    if (mProbe) {
+        return INVALID_OPERATION;
+    }
+    if (mStatus != NO_ERROR && mStatus != ALREADY_EXISTS) {
+        return mStatus;
+    }
+    if (inputCfg == NULL || outputCfg == NULL) {
+        return BAD_VALUE;
+    }
+    status_t status;
+    media::EffectConfig cfg;
+    Status bs = mIEffect->getConfig(&cfg, &status);
+    if (!bs.isOk()) {
+        status = statusTFromBinderStatus(bs);
+        ALOGW("%s received status %d from binder transaction", __func__, status);
+        return status;
+    }
+    if (status == NO_ERROR) {
+        *inputCfg = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfigBase_audio_config_base_t(
+                        cfg.inputCfg, cfg.isOnInputStream));
+        *outputCfg = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfigBase_audio_config_base_t(
+                        cfg.outputCfg, cfg.isOnInputStream));
+    } else {
+        ALOGW("%s received status %d from the effect", __func__, status);
+    }
+    return status;
+}
 
 // -------------------------------------------------------------------------
 
@@ -538,7 +568,6 @@
     if (cb != nullptr) {
         cb->onError(mStatus);
     }
-    mIEffect.clear();
 }
 
 // -------------------------------------------------------------------------
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index 15203d6..69d73ad 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -146,39 +146,6 @@
         audio_channel_mask_t channelMask,
         const AttributionSourceState& client,
         size_t frameCount,
-        legacy_callback_t callback,
-        void* user,
-        uint32_t notificationFrames,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        audio_input_flags_t flags,
-        const audio_attributes_t* pAttributes,
-        audio_port_handle_t selectedDeviceId,
-        audio_microphone_direction_t selectedMicDirection,
-        float microphoneFieldDimension)
-    : mActive(false),
-      mStatus(NO_INIT),
-      mClientAttributionSource(client),
-      mSessionId(AUDIO_SESSION_ALLOCATE),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT),
-      mProxy(NULL)
-{
-    uid_t uid = VALUE_OR_FATAL(aidl2legacy_int32_t_uid_t(mClientAttributionSource.uid));
-    pid_t pid = VALUE_OR_FATAL(aidl2legacy_int32_t_pid_t(mClientAttributionSource.pid));
-    (void)set(inputSource, sampleRate, format, channelMask, frameCount, callback, user,
-            notificationFrames, false /*threadCanCallJava*/, sessionId, transferType, flags,
-            uid, pid, pAttributes, selectedDeviceId, selectedMicDirection,
-            microphoneFieldDimension);
-}
-
-AudioRecord::AudioRecord(
-        audio_source_t inputSource,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        const AttributionSourceState& client,
-        size_t frameCount,
         const wp<IAudioRecordCallback>& callback,
         uint32_t notificationFrames,
         audio_session_t sessionId,
@@ -255,37 +222,6 @@
         mDeviceCallback.clear();
     }
 }
-namespace {
-class LegacyCallbackWrapper : public AudioRecord::IAudioRecordCallback {
-    const AudioRecord::legacy_callback_t mCallback;
-    void* const mData;
-
-  public:
-    LegacyCallbackWrapper(AudioRecord::legacy_callback_t callback, void* user)
-        : mCallback(callback), mData(user) {}
-
-    size_t onMoreData(const AudioRecord::Buffer& buffer) override {
-        AudioRecord::Buffer copy = buffer;
-        mCallback(AudioRecord::EVENT_MORE_DATA, mData, &copy);
-        return copy.size();
-    }
-
-    void onOverrun() override { mCallback(AudioRecord::EVENT_OVERRUN, mData, nullptr); }
-
-    void onMarker(uint32_t markerPosition) override {
-        mCallback(AudioRecord::EVENT_MARKER, mData, &markerPosition);
-    }
-
-    void onNewPos(uint32_t newPos) override {
-        mCallback(AudioRecord::EVENT_NEW_POS, mData, &newPos);
-    }
-
-    void onNewIAudioRecord() override {
-        mCallback(AudioRecord::EVENT_NEW_IAUDIORECORD, mData, nullptr);
-    }
-};
-}  // namespace
-
 status_t AudioRecord::set(
         audio_source_t inputSource,
         uint32_t sampleRate,
@@ -479,37 +415,6 @@
     return status;
 }
 
-status_t AudioRecord::set(
-        audio_source_t inputSource,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        size_t frameCount,
-        legacy_callback_t callback,
-        void* user,
-        uint32_t notificationFrames,
-        bool threadCanCallJava,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        audio_input_flags_t flags,
-        uid_t uid,
-        pid_t pid,
-        const audio_attributes_t* pAttributes,
-        audio_port_handle_t selectedDeviceId,
-        audio_microphone_direction_t selectedMicDirection,
-        float microphoneFieldDimension,
-        int32_t maxSharedAudioHistoryMs)
-{
-    if (callback != nullptr) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    return set(inputSource, sampleRate, format, channelMask, frameCount, mLegacyCallbackWrapper,
-        notificationFrames, threadCanCallJava, sessionId, transferType, flags, uid, pid,
-        pAttributes, selectedDeviceId, selectedMicDirection, microphoneFieldDimension,
-        maxSharedAudioHistoryMs);
-}
 // -------------------------------------------------------------------------
 
 status_t AudioRecord::start(AudioSystem::sync_event_t event, audio_session_t triggerSession)
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 6ab8339..ffbb32f 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -325,45 +325,6 @@
         }
     };
 }
-
-AudioTrack::AudioTrack(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        size_t frameCount,
-        audio_output_flags_t flags,
-        legacy_callback_t callback,
-        void* user,
-        int32_t notificationFrames,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        const audio_offload_info_t *offloadInfo,
-        const AttributionSourceState& attributionSource,
-        const audio_attributes_t* pAttributes,
-        bool doNotReconnect,
-        float maxRequiredSpeed,
-        audio_port_handle_t selectedDeviceId)
-    : mStatus(NO_INIT),
-      mState(STATE_STOPPED),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT),
-      mPausedPosition(0),
-      mAudioTrackCallback(new AudioTrackCallback())
-{
-    mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
-    if (callback != nullptr) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    mSetParams = std::unique_ptr<SetParams>{new SetParams{
-            streamType, sampleRate, format, channelMask, frameCount, flags, mLegacyCallbackWrapper,
-            notificationFrames, 0 /*sharedBuffer*/, false /*threadCanCallJava*/, sessionId,
-            transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
-            maxRequiredSpeed, selectedDeviceId}};
-}
-
 AudioTrack::AudioTrack(
         audio_stream_type_t streamType,
         uint32_t sampleRate,
@@ -397,44 +358,6 @@
                           doNotReconnect, maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
 }
 
-AudioTrack::AudioTrack(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        const sp<IMemory>& sharedBuffer,
-        audio_output_flags_t flags,
-        legacy_callback_t callback,
-        void* user,
-        int32_t notificationFrames,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        const audio_offload_info_t *offloadInfo,
-        const AttributionSourceState& attributionSource,
-        const audio_attributes_t* pAttributes,
-        bool doNotReconnect,
-        float maxRequiredSpeed)
-    : mStatus(NO_INIT),
-      mState(STATE_STOPPED),
-      mPreviousPriority(ANDROID_PRIORITY_NORMAL),
-      mPreviousSchedulingGroup(SP_DEFAULT),
-      mPausedPosition(0),
-      mSelectedDeviceId(AUDIO_PORT_HANDLE_NONE),
-      mAudioTrackCallback(new AudioTrackCallback())
-{
-    mAttributes = AUDIO_ATTRIBUTES_INITIALIZER;
-    if (callback) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    mSetParams = std::unique_ptr<SetParams>{new SetParams{
-            streamType, sampleRate, format, channelMask, 0 /*frameCount*/, flags,
-            mLegacyCallbackWrapper, notificationFrames, sharedBuffer, false /*threadCanCallJava*/,
-            sessionId, transferType, offloadInfo, attributionSource, pAttributes, doNotReconnect,
-            maxRequiredSpeed, AUDIO_PORT_HANDLE_NONE}};
-}
-
 void AudioTrack::onFirstRef() {
     if (mSetParams) {
         set(*mSetParams);
@@ -496,38 +419,6 @@
         mDeviceCallback.clear();
     }
 }
-
-status_t AudioTrack::set(
-        audio_stream_type_t streamType,
-        uint32_t sampleRate,
-        audio_format_t format,
-        audio_channel_mask_t channelMask,
-        size_t frameCount,
-        audio_output_flags_t flags,
-        legacy_callback_t callback,
-        void * user,
-        int32_t notificationFrames,
-        const sp<IMemory>& sharedBuffer,
-        bool threadCanCallJava,
-        audio_session_t sessionId,
-        transfer_type transferType,
-        const audio_offload_info_t *offloadInfo,
-        const AttributionSourceState& attributionSource,
-        const audio_attributes_t* pAttributes,
-        bool doNotReconnect,
-        float maxRequiredSpeed,
-        audio_port_handle_t selectedDeviceId)
-{
-    if (callback) {
-        mLegacyCallbackWrapper = sp<LegacyCallbackWrapper>::make(callback, user);
-    } else if (user) {
-        LOG_ALWAYS_FATAL("Callback data provided without callback pointer!");
-    }
-    return set(streamType, sampleRate,format, channelMask, frameCount, flags,
-               mLegacyCallbackWrapper, notificationFrames, sharedBuffer, threadCanCallJava,
-               sessionId, transferType, offloadInfo, attributionSource, pAttributes,
-               doNotReconnect, maxRequiredSpeed, selectedDeviceId);
-}
 status_t AudioTrack::set(
         audio_stream_type_t streamType,
         uint32_t sampleRate,
diff --git a/media/libaudioclient/TEST_MAPPING b/media/libaudioclient/TEST_MAPPING
index 3751f80..d36cf10 100644
--- a/media/libaudioclient/TEST_MAPPING
+++ b/media/libaudioclient/TEST_MAPPING
@@ -4,6 +4,9 @@
       "name": "audio_aidl_conversion_tests"
     },
     {
+      "name": "audio_aidl_status_tests"
+    },
+    {
       "name": "CtsNativeMediaAAudioTestCases",
       "options" : [
         {
@@ -11,5 +14,22 @@
         }
       ]
     }
+  ],
+  "postsubmit": [
+    {
+      "name": "audieorecord_tests"
+    },
+    {
+      "name": "audioeffect_tests"
+    },
+    {
+      "name": "audiorouting_tests"
+    },
+    {
+      "name": "audioclient_serialization_tests"
+    },
+    {
+      "name": "trackplayerbase_tests"
+    }
   ]
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl b/media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl
index 2cdf4f6..d9c6df4 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortExtSys.aidl
@@ -31,4 +31,6 @@
     AudioPortDeviceExtSys device;
     /** System-only parameters when the port is an audio mix. */
     AudioPortMixExtSys mix;
+    /** Framework audio session identifier. */
+    int session;
 }
diff --git a/media/libaudioclient/aidl/android/media/EffectConfig.aidl b/media/libaudioclient/aidl/android/media/EffectConfig.aidl
new file mode 100644
index 0000000..5f62b73
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/EffectConfig.aidl
@@ -0,0 +1,38 @@
+/*
+ * Copyright (C) 2022 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.audio.common.AudioConfigBase;
+
+/**
+ * Describes configuration of an audio effect. Input and output
+ * audio configurations are described separately because the effect
+ * can perform transformations on channel layouts, for example.
+ *
+ * {@hide}
+ */
+parcelable EffectConfig {
+    /** Configuration of the audio input of the effect. */
+    AudioConfigBase inputCfg;
+    /** Configuration of the audio output of the effect. */
+    AudioConfigBase outputCfg;
+    /**
+     * Specifies whether the effect is instantiated on an input stream,
+     * e.g. on the input from a microphone.
+     */
+    boolean isOnInputStream;
+}
diff --git a/media/libaudioclient/aidl/android/media/IEffect.aidl b/media/libaudioclient/aidl/android/media/IEffect.aidl
index 6ec0405..b7f70a6 100644
--- a/media/libaudioclient/aidl/android/media/IEffect.aidl
+++ b/media/libaudioclient/aidl/android/media/IEffect.aidl
@@ -16,6 +16,7 @@
 
 package android.media;
 
+import android.media.EffectConfig;
 import android.media.SharedFileRegion;
 
 /**
@@ -63,6 +64,14 @@
      */
     SharedFileRegion getCblk();
 
+    /**
+     * Provides audio configurations for effect's input and output,
+     * see EffectConfig parcelable for more details.
+     *
+     * @return a status_t code.
+     */
+    int getConfig(out EffectConfig config);
+
     // When adding a new method, please review and update
     // Effects.cpp AudioFlinger::EffectHandle::onTransact()
     // Effects.cpp IEFFECT_BINDER_METHOD_MACRO_LIST
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index 036e72e..5536bcb 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -84,13 +84,15 @@
 };
 
 template <typename T, typename X, typename FUNC>
-std::vector<T> getFlags(const xsdc_enum_range<X> &range, const FUNC &func,
-                        const std::string &findString = {}) {
+std::vector<T> getFlags(const xsdc_enum_range<X>& range, const FUNC& func,
+                        const std::string& findString = {},
+                        const std::set<X>& excludedValues = {}) {
     std::vector<T> vec;
     for (const auto &xsdEnumVal : range) {
         T enumVal;
         std::string enumString = toString(xsdEnumVal);
         if (enumString.find(findString) != std::string::npos &&
+            (excludedValues.find(xsdEnumVal) == excludedValues.end()) &&
             func(enumString.c_str(), &enumVal)) {
             vec.push_back(enumVal);
         }
@@ -102,13 +104,29 @@
     getFlags<audio_stream_type_t, xsd::AudioStreamType, decltype(audio_stream_type_from_string)>(
         xsdc_enum_range<xsd::AudioStreamType>{}, audio_stream_type_from_string);
 
+/**
+ * AudioFormat - AUDIO_FORMAT_HE_AAC_V1 and AUDIO_FORMAT_HE_AAC_V2
+ * are excluded from kFormats[] in order to avoid the abort triggered
+ * for these two types of AudioFormat in
+ * AidlConversion::legacy2aidl_audio_format_t_AudioFormatDescription()
+ */
 static const std::vector<audio_format_t> kFormats =
-    getFlags<audio_format_t, xsd::AudioFormat, decltype(audio_format_from_string)>(
-        xsdc_enum_range<xsd::AudioFormat>{}, audio_format_from_string);
+        getFlags<audio_format_t, xsd::AudioFormat, decltype(audio_format_from_string)>(
+                xsdc_enum_range<xsd::AudioFormat>{}, audio_format_from_string, {},
+                {xsd::AudioFormat::AUDIO_FORMAT_HE_AAC_V1,
+                 xsd::AudioFormat::AUDIO_FORMAT_HE_AAC_V2});
 
+/**
+ * AudioChannelMask - AUDIO_CHANNEL_IN_6
+ * is excluded from kChannelMasks[] in order to avoid the abort triggered
+ * for this type of AudioChannelMask in
+ * AidlConversion::legacy2aidl_audio_channel_mask_t_AudioChannelLayout()
+ */
 static const std::vector<audio_channel_mask_t> kChannelMasks =
-    getFlags<audio_channel_mask_t, xsd::AudioChannelMask, decltype(audio_channel_mask_from_string)>(
-        xsdc_enum_range<xsd::AudioChannelMask>{}, audio_channel_mask_from_string);
+        getFlags<audio_channel_mask_t, xsd::AudioChannelMask,
+                 decltype(audio_channel_mask_from_string)>(
+                xsdc_enum_range<xsd::AudioChannelMask>{}, audio_channel_mask_from_string, {},
+                {xsd::AudioChannelMask::AUDIO_CHANNEL_IN_6});
 
 static const std::vector<audio_usage_t> kUsages =
     getFlags<audio_usage_t, xsd::AudioUsage, decltype(audio_usage_from_string)>(
@@ -126,9 +144,17 @@
     getFlags<audio_gain_mode_t, xsd::AudioGainMode, decltype(audio_gain_mode_from_string)>(
         xsdc_enum_range<xsd::AudioGainMode>{}, audio_gain_mode_from_string);
 
+/**
+ * AudioDevice - AUDIO_DEVICE_IN_AMBIENT and AUDIO_DEVICE_IN_COMMUNICATION
+ * are excluded from kDevices[] in order to avoid the abort triggered
+ * for these two types of AudioDevice in
+ * AidlConversion::aidl2legacy_AudioDeviceDescription_audio_devices_t()
+ */
 static const std::vector<audio_devices_t> kDevices =
-    getFlags<audio_devices_t, xsd::AudioDevice, decltype(audio_device_from_string)>(
-        xsdc_enum_range<xsd::AudioDevice>{}, audio_device_from_string);
+        getFlags<audio_devices_t, xsd::AudioDevice, decltype(audio_device_from_string)>(
+                xsdc_enum_range<xsd::AudioDevice>{}, audio_device_from_string, {},
+                {xsd::AudioDevice::AUDIO_DEVICE_IN_AMBIENT,
+                 xsd::AudioDevice::AUDIO_DEVICE_IN_COMMUNICATION});
 
 static const std::vector<audio_input_flags_t> kInputFlags =
     getFlags<audio_input_flags_t, xsd::AudioInOutFlag, decltype(audio_input_flag_from_string)>(
diff --git a/media/libaudioclient/include/media/AudioEffect.h b/media/libaudioclient/include/media/AudioEffect.h
index ca35543..72c050b 100644
--- a/media/libaudioclient/include/media/AudioEffect.h
+++ b/media/libaudioclient/include/media/AudioEffect.h
@@ -578,6 +578,25 @@
                               uint32_t *replySize,
                               void *replyData);
 
+    /* Retrieves the configuration of the effect.
+     *
+     * Parameters:
+     *      inputCfg:  pointer to audio_config_base_t structure receiving input
+     *          configuration of the effect
+     *      outputCfg: pointer to audio_config_base_t structure receiving output
+     *          configuration of the effect
+     *
+     * Channel masks of the returned configs are "input" or "output" depending
+     * on the direction of the stream that the effect is attached to.
+     *
+     * Returned status (from utils/Errors.h) can be:
+     *  - NO_ERROR: successful operation.
+     *  - INVALID_OPERATION: the AudioEffect was not successfully initialized.
+     *  - BAD_VALUE: null config pointers
+     *  - DEAD_OBJECT: the effect engine has been deleted.
+     */
+     virtual status_t   getConfigs(audio_config_base_t *inputCfg,
+                                   audio_config_base_t *outputCfg);
 
      /*
       * Utility functions.
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index cb05dd9..5a1ff65 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -46,27 +46,6 @@
 {
 public:
 
-    /* Events used by AudioRecord callback function (legacy_callback_t).
-     * Keep in sync with frameworks/base/media/java/android/media/AudioRecord.java NATIVE_EVENT_*.
-     */
-    enum event_type {
-        EVENT_MORE_DATA = 0,        // Request to read available data from buffer.
-                                    // If this event is delivered but the callback handler
-                                    // does not want to read the available data, the handler must
-                                    // explicitly ignore the event by setting frameCount to zero.
-        EVENT_OVERRUN = 1,          // Buffer overrun occurred.
-        EVENT_MARKER = 2,           // Record head is at the specified marker position
-                                    // (See setMarkerPosition()).
-        EVENT_NEW_POS = 3,          // Record head is at a new position
-                                    // (See setPositionUpdatePeriod()).
-        EVENT_NEW_IAUDIORECORD = 4, // IAudioRecord was re-created, either due to re-routing and
-                                    // voluntary invalidation by mediaserver, or mediaserver crash.
-    };
-
-    /* Client should declare a Buffer and pass address to obtainBuffer()
-     * and releaseBuffer().  See also legacy_callback_t for EVENT_MORE_DATA.
-     */
-
     class Buffer
     {
       friend AudioRecord;
@@ -122,7 +101,6 @@
      *          - EVENT_NEW_IAUDIORECORD: unused.
      */
 
-    typedef void (*legacy_callback_t)(int event, void* user, void *info);
 
     class IAudioRecordCallback : public virtual RefBase {
         friend AudioRecord;
@@ -226,24 +204,6 @@
                                     float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT);
 
 
-                        AudioRecord(audio_source_t inputSource,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    const android::content::AttributionSourceState& client,
-                                    size_t frameCount,
-                                    legacy_callback_t callback,
-                                    void* user,
-                                    uint32_t notificationFrames = 0,
-                                    audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
-                                    transfer_type transferType = TRANSFER_DEFAULT,
-                                    audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
-                                    const audio_attributes_t* pAttributes = nullptr,
-                                    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
-                                    audio_microphone_direction_t
-                                        selectedMicDirection = MIC_DIRECTION_UNSPECIFIED,
-                                    float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT);
-
     /* Terminates the AudioRecord and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioRecord.
      */
@@ -286,27 +246,6 @@
                             float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT,
                             int32_t maxSharedAudioHistoryMs = 0);
 
-           status_t    set(audio_source_t inputSource,
-                            uint32_t sampleRate,
-                            audio_format_t format,
-                            audio_channel_mask_t channelMask,
-                            size_t frameCount,
-                            legacy_callback_t callback,
-                            void* user,
-                            uint32_t notificationFrames = 0,
-                            bool threadCanCallJava = false,
-                            audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
-                            transfer_type transferType = TRANSFER_DEFAULT,
-                            audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
-                            uid_t uid = AUDIO_UID_INVALID,
-                            pid_t pid = -1,
-                            const audio_attributes_t* pAttributes = nullptr,
-                            audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE,
-                            audio_microphone_direction_t
-                                selectedMicDirection = MIC_DIRECTION_UNSPECIFIED,
-                            float selectedMicFieldDimension = MIC_FIELD_DIMENSION_DEFAULT,
-                            int32_t maxSharedAudioHistoryMs = 0);
-
     /* Result of constructing the AudioRecord. This must be checked for successful initialization
      * before using any AudioRecord API (except for set()), because using
      * an uninitialized AudioRecord produces undefined results.
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index 9f540e6..b6ee483 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -148,7 +148,6 @@
      *          - EVENT_NEW_TIMESTAMP: pointer to const AudioTimestamp.
      */
 
-    typedef void (*legacy_callback_t)(int event, void* user, void* info);
     class IAudioTrackCallback : public virtual RefBase {
       friend AudioTrack;
       protected:
@@ -343,26 +342,6 @@
                                     float maxRequiredSpeed = 1.0f,
                                     audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
 
-
-                        AudioTrack( audio_stream_type_t streamType,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    size_t frameCount,
-                                    audio_output_flags_t flags,
-                                    legacy_callback_t cbf,
-                                    void* user = nullptr,
-                                    int32_t notificationFrames = 0,
-                                    audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
-                                    transfer_type transferType = TRANSFER_DEFAULT,
-                                    const audio_offload_info_t *offloadInfo = nullptr,
-                                    const AttributionSourceState& attributionSource =
-                                        AttributionSourceState(),
-                                    const audio_attributes_t* pAttributes = nullptr,
-                                    bool doNotReconnect = false,
-                                    float maxRequiredSpeed = 1.0f,
-                                    audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
     /* Creates an audio track and registers it with AudioFlinger.
      * With this constructor, the track is configured for static buffer mode.
      * Data to be rendered is passed in a shared memory buffer
@@ -391,25 +370,6 @@
                                     bool doNotReconnect = false,
                                     float maxRequiredSpeed = 1.0f);
 
-
-                        AudioTrack( audio_stream_type_t streamType,
-                                    uint32_t sampleRate,
-                                    audio_format_t format,
-                                    audio_channel_mask_t channelMask,
-                                    const sp<IMemory>& sharedBuffer,
-                                    audio_output_flags_t flags,
-                                    legacy_callback_t cbf,
-                                    void* user          = nullptr,
-                                    int32_t notificationFrames = 0,
-                                    audio_session_t sessionId   = AUDIO_SESSION_ALLOCATE,
-                                    transfer_type transferType = TRANSFER_DEFAULT,
-                                    const audio_offload_info_t *offloadInfo = nullptr,
-                                    const AttributionSourceState& attributionSource =
-                                        AttributionSourceState(),
-                                    const audio_attributes_t* pAttributes = nullptr,
-                                    bool doNotReconnect = false,
-                                    float maxRequiredSpeed = 1.0f);
-
     /* Terminates the AudioTrack and unregisters it from AudioFlinger.
      * Also destroys all resources associated with the AudioTrack.
      */
@@ -490,28 +450,8 @@
                         }
             void       onFirstRef() override;
         public:
-            status_t    set(audio_stream_type_t streamType,
-                            uint32_t sampleRate,
-                            audio_format_t format,
-                            audio_channel_mask_t channelMask,
-                            size_t frameCount,
-                            audio_output_flags_t flags,
-                            legacy_callback_t callback,
-                            void * user = nullptr,
-                            int32_t notificationFrames = 0,
-                            const sp<IMemory>& sharedBuffer = 0,
-                            bool threadCanCallJava = false,
-                            audio_session_t sessionId  = AUDIO_SESSION_ALLOCATE,
-                            transfer_type transferType = TRANSFER_DEFAULT,
-                            const audio_offload_info_t *offloadInfo = nullptr,
-                            const AttributionSourceState& attributionSource =
-                                AttributionSourceState(),
-                            const audio_attributes_t* pAttributes = nullptr,
-                            bool doNotReconnect = false,
-                            float maxRequiredSpeed = 1.0f,
-                            audio_port_handle_t selectedDeviceId = AUDIO_PORT_HANDLE_NONE);
-
-    // FIXME(b/169889714): Vendor code depends on the old method signature at link time
+            typedef void (*legacy_callback_t)(int event, void* user, void* info);
+            // FIXME(b/169889714): Vendor code depends on the old method signature at link time
             status_t    set(audio_stream_type_t streamType,
                             uint32_t sampleRate,
                             audio_format_t format,
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 891293e..6535b5b 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -93,3 +93,107 @@
     ],
     data: ["record_test_input_*.txt"],
 }
+
+cc_defaults {
+    name: "libaudioclient_gtests_defaults",
+    cflags: [
+        "-Wall",
+        "-Werror",
+    ],
+    shared_libs: [
+        "capture_state_listener-aidl-cpp",
+        "framework-permission-aidl-cpp",
+        "libbase",
+        "libbinder",
+        "libcgrouprc",
+        "libcutils",
+        "libdl",
+        "liblog",
+        "libmedia",
+        "libmediametrics",
+        "libmediautils",
+        "libmedia_helper",
+        "libnblog",
+        "libprocessgroup",
+        "libshmemcompat",
+        "libstagefright_foundation",
+        "libutils",
+        "libvibrator",
+        "mediametricsservice-aidl-cpp",
+        "packagemanager_aidl-cpp",
+        "shared-file-region-aidl-cpp",
+    ],
+    static_libs: [
+        "android.hardware.audio.common@7.0-enums",
+        "android.media.audio.common.types-V1-cpp",
+        "audioclient-types-aidl-cpp",
+        "audioflinger-aidl-cpp",
+        "audiopolicy-aidl-cpp",
+        "audiopolicy-types-aidl-cpp",
+        "av-types-aidl-cpp",
+        "effect-aidl-cpp",
+        "libaudioclient",
+        "libaudioclient_aidl_conversion",
+        "libaudiofoundation",
+        "libaudiomanager",
+        "libaudiopolicy",
+        "libaudioutils",
+    ],
+    data: ["bbb*.raw"],
+    test_config_template: "audio_test_template.xml",
+    test_suites: ["device-tests"],
+}
+
+cc_test {
+    name: "audiorecord_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiorecord_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "audiotrack_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiotrack_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "audioeffect_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audioeffect_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "audiorouting_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audiorouting_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+    shared_libs: [
+        "libxml2",
+    ],
+}
+
+cc_test {
+    name: "audioclient_serialization_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: [
+        "audioclient_serialization_tests.cpp",
+        "audio_test_utils.cpp",
+    ],
+}
+
+cc_test {
+    name: "trackplayerbase_tests",
+    defaults: ["libaudioclient_gtests_defaults"],
+    srcs: ["trackplayerbase_tests.cpp"],
+}
diff --git a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
index 997f62a..f3361c1 100644
--- a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
@@ -16,8 +16,8 @@
 
 #include <gtest/gtest.h>
 
-#include <media/AudioCommonTypes.h>
 #include <media/AidlConversion.h>
+#include <media/AudioCommonTypes.h>
 
 using namespace android;
 using namespace android::aidl_utils;
@@ -31,7 +31,8 @@
 
 namespace {
 
-template<typename T> size_t hash(const T& t) {
+template <typename T>
+size_t hash(const T& t) {
     return std::hash<T>{}(t);
 }
 
@@ -52,10 +53,8 @@
     return AudioChannelLayout::make<AudioChannelLayout::Tag::layoutMask>(
             // Use channels that exist both for input and output,
             // but doesn't form a known layout mask.
-            AudioChannelLayout::CHANNEL_FRONT_LEFT |
-            AudioChannelLayout::CHANNEL_FRONT_RIGHT |
-            AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT |
-            AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT);
+            AudioChannelLayout::CHANNEL_FRONT_LEFT | AudioChannelLayout::CHANNEL_FRONT_RIGHT |
+            AudioChannelLayout::CHANNEL_TOP_SIDE_LEFT | AudioChannelLayout::CHANNEL_TOP_SIDE_RIGHT);
 }
 
 AudioChannelLayout make_ACL_ChannelIndex2() {
@@ -74,7 +73,7 @@
 }
 
 AudioDeviceDescription make_AudioDeviceDescription(AudioDeviceType type,
-        const std::string& connection = "") {
+                                                   const std::string& connection = "") {
     AudioDeviceDescription result;
     result.type = type;
     result.connection = connection;
@@ -95,12 +94,12 @@
 
 AudioDeviceDescription make_ADD_WiredHeadset() {
     return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
-            AudioDeviceDescription::CONNECTION_ANALOG());
+                                       AudioDeviceDescription::CONNECTION_ANALOG());
 }
 
 AudioDeviceDescription make_ADD_BtScoHeadset() {
     return make_AudioDeviceDescription(AudioDeviceType::OUT_HEADSET,
-            AudioDeviceDescription::CONNECTION_BT_SCO());
+                                       AudioDeviceDescription::CONNECTION_BT_SCO());
 }
 
 AudioFormatDescription make_AudioFormatDescription(AudioFormatType type) {
@@ -121,8 +120,7 @@
     return result;
 }
 
-AudioFormatDescription make_AudioFormatDescription(PcmType transport,
-        const std::string& encoding) {
+AudioFormatDescription make_AudioFormatDescription(PcmType transport, const std::string& encoding) {
     auto result = make_AudioFormatDescription(encoding);
     result.pcm = transport;
     return result;
@@ -163,7 +161,8 @@
 // is identical to the same format description constructed by the framework.
 class HashIdentityTest : public ::testing::Test {
   public:
-    template<typename T> void verifyHashIdentity(const std::vector<std::function<T()>>& valueGens) {
+    template <typename T>
+    void verifyHashIdentity(const std::vector<std::function<T()>>& valueGens) {
         for (size_t i = 0; i < valueGens.size(); ++i) {
             for (size_t j = 0; j < valueGens.size(); ++j) {
                 if (i == j) {
@@ -177,27 +176,25 @@
 };
 
 TEST_F(HashIdentityTest, AudioChannelLayoutHashIdentity) {
-    verifyHashIdentity<AudioChannelLayout>({
-            make_ACL_None, make_ACL_Invalid, make_ACL_Stereo,
-            make_ACL_LayoutArbitrary, make_ACL_ChannelIndex2,
-            make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
+    verifyHashIdentity<AudioChannelLayout>({make_ACL_None, make_ACL_Invalid, make_ACL_Stereo,
+                                            make_ACL_LayoutArbitrary, make_ACL_ChannelIndex2,
+                                            make_ACL_ChannelIndexArbitrary, make_ACL_VoiceCall});
 }
 
 TEST_F(HashIdentityTest, AudioDeviceDescriptionHashIdentity) {
-    verifyHashIdentity<AudioDeviceDescription>({
-            make_ADD_None, make_ADD_DefaultIn, make_ADD_DefaultOut, make_ADD_WiredHeadset,
-            make_ADD_BtScoHeadset});
+    verifyHashIdentity<AudioDeviceDescription>({make_ADD_None, make_ADD_DefaultIn,
+                                                make_ADD_DefaultOut, make_ADD_WiredHeadset,
+                                                make_ADD_BtScoHeadset});
 }
 
 TEST_F(HashIdentityTest, AudioFormatDescriptionHashIdentity) {
-    verifyHashIdentity<AudioFormatDescription>({
-            make_AFD_Default, make_AFD_Invalid, make_AFD_Pcm16Bit, make_AFD_Bitstream,
-            make_AFD_Encap, make_AFD_Encap_with_Enc});
+    verifyHashIdentity<AudioFormatDescription>({make_AFD_Default, make_AFD_Invalid,
+                                                make_AFD_Pcm16Bit, make_AFD_Bitstream,
+                                                make_AFD_Encap, make_AFD_Encap_with_Enc});
 }
 
 using ChannelLayoutParam = std::tuple<AudioChannelLayout, bool /*isInput*/>;
-class AudioChannelLayoutRoundTripTest :
-        public testing::TestWithParam<ChannelLayoutParam> {};
+class AudioChannelLayoutRoundTripTest : public testing::TestWithParam<ChannelLayoutParam> {};
 TEST_P(AudioChannelLayoutRoundTripTest, Aidl2Legacy2Aidl) {
     const auto initial = std::get<0>(GetParam());
     const bool isInput = std::get<1>(GetParam());
@@ -207,21 +204,20 @@
     ASSERT_TRUE(convBack.ok());
     EXPECT_EQ(initial, convBack.value());
 }
-INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutRoundTrip,
-        AudioChannelLayoutRoundTripTest,
-        testing::Combine(
-                testing::Values(AudioChannelLayout{}, make_ACL_Invalid(), make_ACL_Stereo(),
-                        make_ACL_LayoutArbitrary(), make_ACL_ChannelIndex2(),
-                        make_ACL_ChannelIndexArbitrary()),
-                testing::Values(false, true)));
-INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip,
-        AudioChannelLayoutRoundTripTest,
-        // In legacy constants the voice call is only defined for input.
-        testing::Combine(testing::Values(make_ACL_VoiceCall()), testing::Values(true)));
+INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutRoundTrip, AudioChannelLayoutRoundTripTest,
+                         testing::Combine(testing::Values(AudioChannelLayout{}, make_ACL_Invalid(),
+                                                          make_ACL_Stereo(),
+                                                          make_ACL_LayoutArbitrary(),
+                                                          make_ACL_ChannelIndex2(),
+                                                          make_ACL_ChannelIndexArbitrary()),
+                                          testing::Values(false, true)));
+INSTANTIATE_TEST_SUITE_P(AudioChannelVoiceRoundTrip, AudioChannelLayoutRoundTripTest,
+                         // In legacy constants the voice call is only defined for input.
+                         testing::Combine(testing::Values(make_ACL_VoiceCall()),
+                                          testing::Values(true)));
 
 using ChannelLayoutEdgeCaseParam = std::tuple<int /*legacy*/, bool /*isInput*/, bool /*isValid*/>;
-class AudioChannelLayoutEdgeCaseTest :
-        public testing::TestWithParam<ChannelLayoutEdgeCaseParam> {};
+class AudioChannelLayoutEdgeCaseTest : public testing::TestWithParam<ChannelLayoutEdgeCaseParam> {};
 TEST_P(AudioChannelLayoutEdgeCaseTest, Legacy2Aidl) {
     const audio_channel_mask_t legacy = static_cast<audio_channel_mask_t>(std::get<0>(GetParam()));
     const bool isInput = std::get<1>(GetParam());
@@ -229,8 +225,8 @@
     auto conv = legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy, isInput);
     EXPECT_EQ(isValid, conv.ok());
 }
-INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutEdgeCase,
-        AudioChannelLayoutEdgeCaseTest,
+INSTANTIATE_TEST_SUITE_P(
+        AudioChannelLayoutEdgeCase, AudioChannelLayoutEdgeCaseTest,
         testing::Values(
                 // Valid legacy input masks.
                 std::make_tuple(AUDIO_CHANNEL_IN_VOICE_UPLINK_MONO, true, true),
@@ -240,25 +236,26 @@
                 std::make_tuple(
                         // This has the same numerical representation as Mask 'A' below
                         AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
-                        AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT, false, true),
+                                AUDIO_CHANNEL_OUT_TOP_FRONT_RIGHT,
+                        false, true),
                 std::make_tuple(
                         // This has the same numerical representation as Mask 'B' below
                         AUDIO_CHANNEL_OUT_FRONT_CENTER | AUDIO_CHANNEL_OUT_LOW_FREQUENCY |
-                        AUDIO_CHANNEL_OUT_TOP_BACK_LEFT, false, true),
+                                AUDIO_CHANNEL_OUT_TOP_BACK_LEFT,
+                        false, true),
                 // Invalid legacy input masks.
                 std::make_tuple(AUDIO_CHANNEL_IN_6, true, false),
-                std::make_tuple(
-                        AUDIO_CHANNEL_IN_6 | AUDIO_CHANNEL_IN_FRONT_PROCESSED, true, false),
-                std::make_tuple(
-                        AUDIO_CHANNEL_IN_PRESSURE | AUDIO_CHANNEL_IN_X_AXIS |
-                        AUDIO_CHANNEL_IN_Y_AXIS | AUDIO_CHANNEL_IN_Z_AXIS, true, false),
+                std::make_tuple(AUDIO_CHANNEL_IN_6 | AUDIO_CHANNEL_IN_FRONT_PROCESSED, true, false),
+                std::make_tuple(AUDIO_CHANNEL_IN_PRESSURE | AUDIO_CHANNEL_IN_X_AXIS |
+                                        AUDIO_CHANNEL_IN_Y_AXIS | AUDIO_CHANNEL_IN_Z_AXIS,
+                                true, false),
                 std::make_tuple(  // Mask 'A'
                         AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_VOICE_UPLINK, true, false),
                 std::make_tuple(  // Mask 'B'
                         AUDIO_CHANNEL_IN_STEREO | AUDIO_CHANNEL_IN_VOICE_DNLINK, true, false)));
 
-class AudioDeviceDescriptionRoundTripTest :
-        public testing::TestWithParam<AudioDeviceDescription> {};
+class AudioDeviceDescriptionRoundTripTest : public testing::TestWithParam<AudioDeviceDescription> {
+};
 TEST_P(AudioDeviceDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
     const auto initial = GetParam();
     auto conv = aidl2legacy_AudioDeviceDescription_audio_devices_t(initial);
@@ -267,13 +264,13 @@
     ASSERT_TRUE(convBack.ok());
     EXPECT_EQ(initial, convBack.value());
 }
-INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip,
-        AudioDeviceDescriptionRoundTripTest,
-        testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
-                make_ADD_DefaultOut(), make_ADD_WiredHeadset(), make_ADD_BtScoHeadset()));
+INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip, AudioDeviceDescriptionRoundTripTest,
+                         testing::Values(AudioDeviceDescription{}, make_ADD_DefaultIn(),
+                                         make_ADD_DefaultOut(), make_ADD_WiredHeadset(),
+                                         make_ADD_BtScoHeadset()));
 
-class AudioFormatDescriptionRoundTripTest :
-        public testing::TestWithParam<AudioFormatDescription> {};
+class AudioFormatDescriptionRoundTripTest : public testing::TestWithParam<AudioFormatDescription> {
+};
 TEST_P(AudioFormatDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
     const auto initial = GetParam();
     auto conv = aidl2legacy_AudioFormatDescription_audio_format_t(initial);
@@ -282,6 +279,6 @@
     ASSERT_TRUE(convBack.ok());
     EXPECT_EQ(initial, convBack.value());
 }
-INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip,
-        AudioFormatDescriptionRoundTripTest,
-        testing::Values(make_AFD_Invalid(), AudioFormatDescription{}, make_AFD_Pcm16Bit()));
+INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip, AudioFormatDescriptionRoundTripTest,
+                         testing::Values(make_AFD_Invalid(), AudioFormatDescription{},
+                                         make_AFD_Pcm16Bit()));
diff --git a/media/libaudioclient/tests/audio_aidl_status_tests.cpp b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
index 5517091..8a7e6c1 100644
--- a/media/libaudioclient/tests/audio_aidl_status_tests.cpp
+++ b/media/libaudioclient/tests/audio_aidl_status_tests.cpp
@@ -37,25 +37,10 @@
 
 // Special status values are preserved on round trip.
 TEST(audio_aidl_status_tests, statusRoundTripSpecialValues) {
-    for (status_t status : {
-            OK,
-            UNKNOWN_ERROR,
-            NO_MEMORY,
-            INVALID_OPERATION,
-            BAD_VALUE,
-            BAD_TYPE,
-            NAME_NOT_FOUND,
-            PERMISSION_DENIED,
-            NO_INIT,
-            ALREADY_EXISTS,
-            DEAD_OBJECT,
-            FAILED_TRANSACTION,
-            BAD_INDEX,
-            NOT_ENOUGH_DATA,
-            WOULD_BLOCK,
-            TIMED_OUT,
-            UNKNOWN_TRANSACTION,
-            FDS_NOT_ALLOWED}) {
+    for (status_t status :
+         {OK, UNKNOWN_ERROR, NO_MEMORY, INVALID_OPERATION, BAD_VALUE, BAD_TYPE, NAME_NOT_FOUND,
+          PERMISSION_DENIED, NO_INIT, ALREADY_EXISTS, DEAD_OBJECT, FAILED_TRANSACTION, BAD_INDEX,
+          NOT_ENOUGH_DATA, WOULD_BLOCK, TIMED_OUT, UNKNOWN_TRANSACTION, FDS_NOT_ALLOWED}) {
         ASSERT_EQ(status, statusTFromBinderStatus(binderStatusFromStatusT(status)));
     }
 }
@@ -63,47 +48,29 @@
 // Binder exceptions show as an error (not fixed at this time); these come fromExceptionCode().
 TEST(audio_aidl_status_tests, binderStatusExceptions) {
     for (int exceptionCode : {
-            //Status::EX_NONE,
-            Status::EX_SECURITY,
-            Status::EX_BAD_PARCELABLE,
-            Status::EX_ILLEGAL_ARGUMENT,
-            Status::EX_NULL_POINTER,
-            Status::EX_ILLEGAL_STATE,
-            Status::EX_NETWORK_MAIN_THREAD,
-            Status::EX_UNSUPPORTED_OPERATION,
-            //Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
-            Status::EX_PARCELABLE,
-            // This is special and Java specific; see Parcel.java.
-            Status::EX_HAS_REPLY_HEADER,
-            // This is special, and indicates to C++ binder proxies that the
-            // transaction has failed at a low level.
-            //Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
-            }) {
+                 // Status::EX_NONE,
+                 Status::EX_SECURITY, Status::EX_BAD_PARCELABLE, Status::EX_ILLEGAL_ARGUMENT,
+                 Status::EX_NULL_POINTER, Status::EX_ILLEGAL_STATE, Status::EX_NETWORK_MAIN_THREAD,
+                 Status::EX_UNSUPPORTED_OPERATION,
+                 // Status::EX_SERVICE_SPECIFIC, -- tested fromServiceSpecificError()
+                 Status::EX_PARCELABLE,
+                 // This is special and Java specific; see Parcel.java.
+                 Status::EX_HAS_REPLY_HEADER,
+                 // This is special, and indicates to C++ binder proxies that the
+                 // transaction has failed at a low level.
+                 // Status::EX_TRANSACTION_FAILED, -- tested fromStatusT().
+         }) {
         ASSERT_NE(OK, statusTFromBinderStatus(Status::fromExceptionCode(exceptionCode)));
     }
 }
 
 // Binder transaction errors show exactly in status_t; these come fromStatusT().
 TEST(audio_aidl_status_tests, binderStatusTransactionError) {
-    for (status_t status : {
-            OK, // Note: fromStatusT does check if this is 0, so this is no error.
-            UNKNOWN_ERROR,
-            NO_MEMORY,
-            INVALID_OPERATION,
-            BAD_VALUE,
-            BAD_TYPE,
-            NAME_NOT_FOUND,
-            PERMISSION_DENIED,
-            NO_INIT,
-            ALREADY_EXISTS,
-            DEAD_OBJECT,
-            FAILED_TRANSACTION,
-            BAD_INDEX,
-            NOT_ENOUGH_DATA,
-            WOULD_BLOCK,
-            TIMED_OUT,
-            UNKNOWN_TRANSACTION,
-            FDS_NOT_ALLOWED}) {
+    for (status_t status :
+         {OK,  // Note: fromStatusT does check if this is 0, so this is no error.
+          UNKNOWN_ERROR, NO_MEMORY, INVALID_OPERATION, BAD_VALUE, BAD_TYPE, NAME_NOT_FOUND,
+          PERMISSION_DENIED, NO_INIT, ALREADY_EXISTS, DEAD_OBJECT, FAILED_TRANSACTION, BAD_INDEX,
+          NOT_ENOUGH_DATA, WOULD_BLOCK, TIMED_OUT, UNKNOWN_TRANSACTION, FDS_NOT_ALLOWED}) {
         ASSERT_EQ(status, statusTFromBinderStatus(Status::fromStatusT(status)));
     }
 }
diff --git a/media/libaudioclient/tests/audio_test_template.xml b/media/libaudioclient/tests/audio_test_template.xml
new file mode 100644
index 0000000..ed0cb21
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_template.xml
@@ -0,0 +1,32 @@
+<?xml version="1.0" encoding="utf-8"?>
+<!-- Copyright (C) 2022 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.
+-->
+<configuration description="Unit test configuration for {MODULE}">
+    <target_preparer class="com.android.tradefed.targetprep.RootTargetPreparer" />
+
+    <target_preparer class="com.android.tradefed.targetprep.PushFilePreparer">
+        <option name="cleanup" value="true" />
+        <option name="push-file" key="{MODULE}" value="/data/local/tmp/{MODULE}" />
+
+        <!-- Files used for audio testing -->
+        <option name="push-file" key="bbb_1ch_8kHz_s16le.raw" value="/data/local/tmp/bbb_1ch_8kHz_s16le.raw" />
+        <option name="push-file" key="bbb_2ch_24kHz_s16le.raw" value="/data/local/tmp/bbb_2ch_24kHz_s16le.raw" />
+    </target_preparer>
+
+    <test class="com.android.tradefed.testtype.GTest" >
+        <option name="native-test-device-path" value="/data/local/tmp" />
+        <option name="module-name" value="{MODULE}" />
+    </test>
+</configuration>
diff --git a/media/libaudioclient/tests/audio_test_utils.cpp b/media/libaudioclient/tests/audio_test_utils.cpp
new file mode 100644
index 0000000..d7ce014
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.cpp
@@ -0,0 +1,787 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioTestUtils"
+
+#include <utils/Log.h>
+
+#include "audio_test_utils.h"
+
+// Generates a random string.
+void CreateRandomFile(int& fd) {
+    std::string filename = "/data/local/tmp/record-XXXXXX";
+    fd = mkstemp(filename.data());
+}
+
+void OnAudioDeviceUpdateNotifier::onAudioDeviceUpdate(audio_io_handle_t audioIo,
+                                                      audio_port_handle_t deviceId) {
+    std::unique_lock<std::mutex> lock{mMutex};
+    ALOGD("%s  audioIo=%d deviceId=%d", __func__, audioIo, deviceId);
+    mAudioIo = audioIo;
+    mDeviceId = deviceId;
+    mCondition.notify_all();
+}
+
+status_t OnAudioDeviceUpdateNotifier::waitForAudioDeviceCb() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    if (mAudioIo == AUDIO_IO_HANDLE_NONE) {
+        mCondition.wait_for(lock, std::chrono::milliseconds(500));
+        if (mAudioIo == AUDIO_IO_HANDLE_NONE) return TIMED_OUT;
+    }
+    return OK;
+}
+
+AudioPlayback::AudioPlayback(uint32_t sampleRate, audio_format_t format,
+                             audio_channel_mask_t channelMask, audio_output_flags_t flags,
+                             audio_session_t sessionId, AudioTrack::transfer_type transferType,
+                             audio_attributes_t* attributes)
+    : mSampleRate(sampleRate),
+      mFormat(format),
+      mChannelMask(channelMask),
+      mFlags(flags),
+      mSessionId(sessionId),
+      mTransferType(transferType),
+      mAttributes(attributes) {
+    mStopPlaying = false;
+    mBytesUsedSoFar = 0;
+    mState = PLAY_NO_INIT;
+    mMemCapacity = 0;
+    mMemoryDealer = nullptr;
+    mMemory = nullptr;
+}
+
+AudioPlayback::~AudioPlayback() {
+    stop();
+}
+
+status_t AudioPlayback::create() {
+    if (mState != PLAY_NO_INIT) return INVALID_OPERATION;
+    std::string packageName{"AudioPlayback"};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    if (mTransferType == AudioTrack::TRANSFER_OBTAIN) {
+        mTrack = new AudioTrack(attributionSource);
+        mTrack->set(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, 0 /* frameCount */,
+                    mFlags, nullptr /* callback */, 0 /* notificationFrames */,
+                    nullptr /* sharedBuffer */, false /*canCallJava */, mSessionId, mTransferType,
+                    nullptr /* offloadInfo */, attributionSource, mAttributes);
+    } else if (mTransferType == AudioTrack::TRANSFER_SHARED) {
+        mTrack = new AudioTrack(AUDIO_STREAM_MUSIC, mSampleRate, mFormat, mChannelMask, mMemory,
+                                mFlags, wp<AudioTrack::IAudioTrackCallback>::fromExisting(this), 0,
+                                mSessionId, mTransferType, nullptr, attributionSource, mAttributes);
+    } else {
+        ALOGE("Required Transfer type not existed");
+        return INVALID_OPERATION;
+    }
+    mTrack->setCallerName(packageName);
+    status_t status = mTrack->initCheck();
+    if (NO_ERROR == status) mState = PLAY_READY;
+    return status;
+}
+
+status_t AudioPlayback::loadResource(const char* name) {
+    status_t status = OK;
+    FILE* fp = fopen(name, "rbe");
+    struct stat buf {};
+    if (fp && !fstat(fileno(fp), &buf)) {
+        mMemCapacity = buf.st_size;
+        mMemoryDealer = new MemoryDealer(mMemCapacity, "AudioPlayback");
+        if (nullptr == mMemoryDealer.get()) {
+            ALOGE("couldn't get MemoryDealer!");
+            fclose(fp);
+            return NO_MEMORY;
+        }
+        mMemory = mMemoryDealer->allocate(mMemCapacity);
+        if (nullptr == mMemory.get()) {
+            ALOGE("couldn't get IMemory!");
+            fclose(fp);
+            return NO_MEMORY;
+        }
+        uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+        fread(ipBuffer, sizeof(uint8_t), mMemCapacity, fp);
+    } else {
+        ALOGE("unable to open input file %s", name);
+        status = NAME_NOT_FOUND;
+    }
+    if (fp) fclose(fp);
+    return status;
+}
+
+sp<AudioTrack> AudioPlayback::getAudioTrackHandle() {
+    return (PLAY_NO_INIT != mState) ? mTrack : nullptr;
+}
+
+status_t AudioPlayback::start() {
+    status_t status;
+    if (PLAY_READY != mState) {
+        return INVALID_OPERATION;
+    } else {
+        status = mTrack->start();
+        if (OK == status) {
+            mState = PLAY_STARTED;
+            LOG_FATAL_IF(false != mTrack->stopped());
+        }
+    }
+    return status;
+}
+
+void AudioPlayback::onBufferEnd() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    mStopPlaying = true;
+    mCondition.notify_all();
+}
+
+status_t AudioPlayback::fillBuffer() {
+    if (PLAY_STARTED != mState && PLAY_STOPPED != mState) return INVALID_OPERATION;
+    int retry = 25;
+    uint8_t* ipBuffer = static_cast<uint8_t*>(static_cast<void*>(mMemory->unsecurePointer()));
+    size_t nonContig = 0;
+    size_t bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+    while (bytesAvailable > 0) {
+        AudioTrack::Buffer trackBuffer;
+        trackBuffer.frameCount = mTrack->frameCount() * 2;
+        status_t status = mTrack->obtainBuffer(&trackBuffer, retry, &nonContig);
+        if (OK == status) {
+            size_t bytesToCopy = std::min(bytesAvailable, trackBuffer.size());
+            if (bytesToCopy > 0) {
+                memcpy(trackBuffer.data(), ipBuffer + mBytesUsedSoFar, bytesToCopy);
+            }
+            mTrack->releaseBuffer(&trackBuffer);
+            mBytesUsedSoFar += bytesToCopy;
+            bytesAvailable = mMemCapacity - mBytesUsedSoFar;
+            if (bytesAvailable == 0) {
+                stop();
+            }
+        } else if (WOULD_BLOCK == status) {
+            if (mStopPlaying)
+                return OK;
+            else
+                return TIMED_OUT;
+        }
+    }
+    return OK;
+}
+
+status_t AudioPlayback::waitForConsumption(bool testSeek) {
+    if (PLAY_STARTED != mState) return INVALID_OPERATION;
+    // in static buffer mode, lets not play clips with duration > 30 sec
+    int retry = 30;
+    // Total number of frames in the input file.
+    size_t totalFrameCount = mMemCapacity / mTrack->frameSize();
+    while (!mStopPlaying && retry > 0) {
+        // Get the total numbers of frames played.
+        uint32_t currPosition;
+        mTrack->getPosition(&currPosition);
+        if (testSeek && (currPosition > totalFrameCount * 0.6)) {
+            testSeek = false;
+            if (!mTrack->hasStarted()) return BAD_VALUE;
+            mTrack->pauseAndWait(std::chrono::seconds(2));
+            if (mTrack->hasStarted()) return BAD_VALUE;
+            mTrack->reload();
+            mTrack->getPosition(&currPosition);
+            if (currPosition != 0) return BAD_VALUE;
+            mTrack->start();
+            while (currPosition < totalFrameCount * 0.3) {
+                mTrack->getPosition(&currPosition);
+            }
+            mTrack->pauseAndWait(std::chrono::seconds(2));
+            uint32_t setPosition = totalFrameCount * 0.9;
+            mTrack->setPosition(setPosition);
+            uint32_t bufferPosition;
+            mTrack->getBufferPosition(&bufferPosition);
+            if (bufferPosition != setPosition) return BAD_VALUE;
+            mTrack->start();
+        }
+        std::this_thread::sleep_for(std::chrono::milliseconds(300));
+        retry--;
+    }
+    if (!mStopPlaying) return TIMED_OUT;
+    return OK;
+}
+
+status_t AudioPlayback::onProcess(bool testSeek) {
+    if (mTransferType == AudioTrack::TRANSFER_SHARED)
+        return waitForConsumption(testSeek);
+    else if (mTransferType == AudioTrack::TRANSFER_OBTAIN)
+        return fillBuffer();
+    else
+        return INVALID_OPERATION;
+}
+
+void AudioPlayback::stop() {
+    std::unique_lock<std::mutex> lock{mMutex};
+    mStopPlaying = true;
+    if (mState != PLAY_STOPPED) {
+        mTrack->stopAndJoinCallbacks();
+        LOG_FATAL_IF(true != mTrack->stopped());
+        mState = PLAY_STOPPED;
+    }
+}
+
+// hold pcm data sent by AudioRecord
+RawBuffer::RawBuffer(int64_t ptsPipeline, int64_t ptsManual, int32_t capacity)
+    : mData(capacity > 0 ? new uint8_t[capacity] : nullptr),
+      mPtsPipeline(ptsPipeline),
+      mPtsManual(ptsManual),
+      mCapacity(capacity) {}
+
+// Simple AudioCapture
+size_t AudioCapture::onMoreData(const AudioRecord::Buffer& buffer) {
+    if (mState != REC_STARTED) {
+        ALOGE("Unexpected Callback from audiorecord, not reading data");
+        return 0;
+    }
+
+    // no more frames to read
+    if (mNumFramesReceived > mNumFramesToRecord || mStopRecording) {
+        mStopRecording = true;
+        return 0;
+    }
+
+    int64_t timeUs = 0, position = 0, timeNs = 0;
+    ExtendedTimestamp ts;
+    ExtendedTimestamp::Location location;
+    const int32_t usPerSec = 1000000;
+
+    if (mRecord->getTimestamp(&ts) == OK &&
+        ts.getBestTimestamp(&position, &timeNs, ExtendedTimestamp::TIMEBASE_MONOTONIC, &location) ==
+                OK) {
+        // Use audio timestamp.
+        timeUs = timeNs / 1000 -
+                 (position - mNumFramesReceived + mNumFramesLost) * usPerSec / mSampleRate;
+    } else {
+        // This should not happen in normal case.
+        ALOGW("Failed to get audio timestamp, fallback to use systemclock");
+        timeUs = systemTime() / 1000LL;
+        // Estimate the real sampling time of the 1st sample in this buffer
+        // from AudioRecord's latency. (Apply this adjustment first so that
+        // the start time logic is not affected.)
+        timeUs -= mRecord->latency() * 1000LL;
+    }
+
+    ALOGV("dataCallbackTimestamp: %" PRId64 " us", timeUs);
+
+    const size_t frameSize = mRecord->frameSize();
+    uint64_t numLostBytes = (uint64_t)mRecord->getInputFramesLost() * frameSize;
+    if (numLostBytes > 0) {
+        ALOGW("Lost audio record data: %" PRIu64 " bytes", numLostBytes);
+    }
+    std::deque<RawBuffer> tmpQueue;
+    while (numLostBytes > 0) {
+        uint64_t bufferSize = numLostBytes;
+        if (numLostBytes > mMaxBytesPerCallback) {
+            numLostBytes -= mMaxBytesPerCallback;
+            bufferSize = mMaxBytesPerCallback;
+        } else {
+            numLostBytes = 0;
+        }
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer emptyBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+        memset(emptyBuffer.mData.get(), 0, bufferSize);
+        mNumFramesLost += bufferSize / frameSize;
+        mNumFramesReceived += bufferSize / frameSize;
+        tmpQueue.push_back(std::move(emptyBuffer));
+    }
+
+    if (buffer.size() == 0) {
+        ALOGW("Nothing is available from AudioRecord callback buffer");
+    } else {
+        const size_t bufferSize = buffer.size();
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer audioBuffer{timeUs, timestampUs, static_cast<int32_t>(bufferSize)};
+        memcpy(audioBuffer.mData.get(), buffer.data(), bufferSize);
+        mNumFramesReceived += bufferSize / frameSize;
+        tmpQueue.push_back(std::move(audioBuffer));
+    }
+
+    if (tmpQueue.size() > 0) {
+        std::unique_lock<std::mutex> lock{mMutex};
+        for (auto it = tmpQueue.begin(); it != tmpQueue.end(); it++)
+            mBuffersReceived.push_back(std::move(*it));
+        mCondition.notify_all();
+    }
+    return buffer.size();
+}
+
+void AudioCapture::onOverrun() {
+    ALOGV("received event overrun");
+    mBufferOverrun = true;
+}
+
+void AudioCapture::onMarker(uint32_t markerPosition) {
+    ALOGV("received Callback at position %d", markerPosition);
+    mReceivedCbMarkerAtPosition = markerPosition;
+}
+
+void AudioCapture::onNewPos(uint32_t markerPosition) {
+    ALOGV("received Callback at position %d", markerPosition);
+    mReceivedCbMarkerCount++;
+}
+
+void AudioCapture::onNewIAudioRecord() {
+    ALOGV("IAudioRecord is re-created");
+}
+
+AudioCapture::AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+                           audio_channel_mask_t channelMask, audio_input_flags_t flags,
+                           audio_session_t sessionId, AudioRecord::transfer_type transferType)
+    : mInputSource(inputSource),
+      mSampleRate(sampleRate),
+      mFormat(format),
+      mChannelMask(channelMask),
+      mFlags(flags),
+      mSessionId(sessionId),
+      mTransferType(transferType) {
+    mFrameCount = 0;
+    mNotificationFrames = 0;
+    mNumFramesToRecord = 0;
+    mNumFramesReceived = 0;
+    mNumFramesLost = 0;
+    mBufferOverrun = false;
+    mMarkerPosition = 0;
+    mMarkerPeriod = 0;
+    mReceivedCbMarkerAtPosition = -1;
+    mReceivedCbMarkerCount = 0;
+    mState = REC_NO_INIT;
+    mStopRecording = false;
+#if RECORD_TO_FILE
+    CreateRandomFile(mOutFileFd);
+#endif
+}
+
+AudioCapture::~AudioCapture() {
+    if (mOutFileFd > 0) close(mOutFileFd);
+    stop();
+}
+
+status_t AudioCapture::create() {
+    if (mState != REC_NO_INIT) return INVALID_OPERATION;
+    // get Min Frame Count
+    size_t minFrameCount;
+    status_t status =
+            AudioRecord::getMinFrameCount(&minFrameCount, mSampleRate, mFormat, mChannelMask);
+    if (NO_ERROR != status) return status;
+    // Limit notificationFrames basing on client bufferSize
+    const int samplesPerFrame = audio_channel_count_from_in_mask(mChannelMask);
+    const int bytesPerSample = audio_bytes_per_sample(mFormat);
+    mNotificationFrames = mMaxBytesPerCallback / (samplesPerFrame * bytesPerSample);
+    // select frameCount to be at least minFrameCount
+    mFrameCount = 2 * mNotificationFrames;
+    while (mFrameCount < minFrameCount) {
+        mFrameCount += mNotificationFrames;
+    }
+    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+        ALOGW("Overriding all previous computations");
+        mFrameCount = 0;
+        mNotificationFrames = 0;
+    }
+    mNumFramesToRecord = (mSampleRate * 0.25);  // record .25 sec
+    std::string packageName{"AudioCapture"};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    if (mTransferType == AudioRecord::TRANSFER_OBTAIN) {
+        if (mSampleRate == 48000) {  // test all available constructors
+            mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+                                      attributionSource, mFrameCount, nullptr /* callback */,
+                                      mNotificationFrames, mSessionId, mTransferType, mFlags);
+        } else {
+            mRecord = new AudioRecord(attributionSource);
+            status = mRecord->set(mInputSource, mSampleRate, mFormat, mChannelMask, mFrameCount,
+                                  nullptr /* callback */, 0 /* notificationFrames */,
+                                  false /* canCallJava */, mSessionId, mTransferType, mFlags,
+                                  attributionSource.uid, attributionSource.pid);
+        }
+        if (NO_ERROR != status) return status;
+    } else if (mTransferType == AudioRecord::TRANSFER_CALLBACK) {
+        mRecord = new AudioRecord(mInputSource, mSampleRate, mFormat, mChannelMask,
+                                  attributionSource, mFrameCount, this, mNotificationFrames,
+                                  mSessionId, mTransferType, mFlags);
+    } else {
+        ALOGE("Test application is not handling transfer type %s",
+              AudioRecord::convertTransferToText(mTransferType));
+        return NO_INIT;
+    }
+    mRecord->setCallerName(packageName);
+    status = mRecord->initCheck();
+    if (NO_ERROR == status) mState = REC_READY;
+    if (mFlags & AUDIO_INPUT_FLAG_FAST) {
+        mFrameCount = mRecord->frameCount();
+        mNotificationFrames = mRecord->getNotificationPeriodInFrames();
+        mMaxBytesPerCallback = mNotificationFrames * samplesPerFrame * bytesPerSample;
+    }
+    return status;
+}
+
+sp<AudioRecord> AudioCapture::getAudioRecordHandle() {
+    return (REC_NO_INIT == mState) ? nullptr : mRecord;
+}
+
+status_t AudioCapture::start(AudioSystem::sync_event_t event, audio_session_t triggerSession) {
+    status_t status;
+    if (REC_READY != mState) {
+        return INVALID_OPERATION;
+    } else {
+        status = mRecord->start(event, triggerSession);
+        if (OK == status) {
+            mState = REC_STARTED;
+            LOG_FATAL_IF(false != mRecord->stopped());
+        }
+    }
+    return status;
+}
+
+status_t AudioCapture::stop() {
+    status_t status = OK;
+    mStopRecording = true;
+    if (mState != REC_STOPPED) {
+        mRecord->stopAndJoinCallbacks();
+        mState = REC_STOPPED;
+        LOG_FATAL_IF(true != mRecord->stopped());
+    }
+    return status;
+}
+
+status_t AudioCapture::obtainBuffer(RawBuffer& buffer) {
+    if (REC_STARTED != mState && REC_STOPPED != mState) return INVALID_OPERATION;
+    int retry = 25;
+    AudioRecord::Buffer recordBuffer;
+    recordBuffer.frameCount = mNotificationFrames;
+    size_t nonContig = 0;
+    status_t status = mRecord->obtainBuffer(&recordBuffer, retry, &nonContig);
+    if (OK == status) {
+        const int64_t timestampUs =
+                ((1000000LL * mNumFramesReceived) + (mRecord->getSampleRate() >> 1)) /
+                mRecord->getSampleRate();
+        RawBuffer buff{-1, timestampUs, static_cast<int32_t>(recordBuffer.size())};
+        memcpy(buff.mData.get(), recordBuffer.data(), recordBuffer.size());
+        buffer = std::move(buff);
+        mNumFramesReceived += recordBuffer.size() / mRecord->frameSize();
+        mRecord->releaseBuffer(&recordBuffer);
+        if (mNumFramesReceived > mNumFramesToRecord) {
+            stop();
+        }
+    } else if (status == WOULD_BLOCK) {
+        if (mStopRecording)
+            return WOULD_BLOCK;
+        else
+            return TIMED_OUT;
+    }
+    return OK;
+}
+
+status_t AudioCapture::obtainBufferCb(RawBuffer& buffer) {
+    if (REC_STARTED != mState) return INVALID_OPERATION;
+    int retry = 10;
+    std::unique_lock<std::mutex> lock{mMutex};
+    while (mBuffersReceived.empty() && !mStopRecording && retry > 0) {
+        mCondition.wait_for(lock, std::chrono::milliseconds(100));
+        retry--;
+    }
+    if (!mBuffersReceived.empty()) {
+        auto it = mBuffersReceived.begin();
+        buffer = std::move(*it);
+        mBuffersReceived.erase(it);
+    } else {
+        if (retry == 0) return TIMED_OUT;
+        if (mStopRecording)
+            return WOULD_BLOCK;
+        else
+            return UNKNOWN_ERROR;
+    }
+    return OK;
+}
+
+status_t AudioCapture::audioProcess() {
+    RawBuffer buffer;
+    while (true) {
+        status_t status;
+        if (mTransferType == AudioRecord::TRANSFER_CALLBACK)
+            status = obtainBufferCb(buffer);
+        else
+            status = obtainBuffer(buffer);
+        switch (status) {
+            case OK:
+                if (mOutFileFd > 0) {
+                    const char* ptr =
+                            static_cast<const char*>(static_cast<void*>(buffer.mData.get()));
+                    write(mOutFileFd, ptr, buffer.mCapacity);
+                }
+                break;
+            case WOULD_BLOCK:
+                return OK;
+            case TIMED_OUT:          // "recorder application timed out from receiving buffers"
+            case NO_INIT:            // "recorder not initialized"
+            case INVALID_OPERATION:  // "recorder not started"
+            case UNKNOWN_ERROR:      // "Unknown error"
+            default:
+                return status;
+        }
+    }
+}
+
+status_t listAudioPorts(std::vector<audio_port_v7>& portsVec) {
+    int attempts = 5;
+    status_t status;
+    unsigned int generation1, generation;
+    unsigned int numPorts = 0;
+    do {
+        if (attempts-- < 0) {
+            status = TIMED_OUT;
+            break;
+        }
+        status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts,
+                                             nullptr, &generation1);
+        if (status != NO_ERROR) {
+            ALOGE("AudioSystem::listAudioPorts returned error %d", status);
+            break;
+        }
+        portsVec.resize(numPorts);
+        status = AudioSystem::listAudioPorts(AUDIO_PORT_ROLE_NONE, AUDIO_PORT_TYPE_NONE, &numPorts,
+                                             portsVec.data(), &generation);
+    } while (generation1 != generation && status == NO_ERROR);
+    if (status != NO_ERROR) {
+        numPorts = 0;
+        portsVec.clear();
+    }
+    return status;
+}
+
+status_t getPortById(const audio_port_handle_t portId, audio_port_v7& port) {
+    std::vector<struct audio_port_v7> ports;
+    status_t status = listAudioPorts(ports);
+    if (status != OK) return status;
+    for (auto i = 0; i < ports.size(); i++) {
+        if (ports[i].id == portId) {
+            port = ports[i];
+            return OK;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type,
+                             audio_devices_t deviceType, audio_port_v7& port) {
+    std::vector<struct audio_port_v7> ports;
+    status_t status = listAudioPorts(ports);
+    if (status != OK) return status;
+    for (auto i = 0; i < ports.size(); i++) {
+        if (ports[i].role == role && ports[i].type == type &&
+            ports[i].ext.device.type == deviceType) {
+            port = ports[i];
+            return OK;
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec) {
+    int attempts = 5;
+    status_t status;
+    unsigned int generation1, generation;
+    unsigned int numPatches = 0;
+    do {
+        if (attempts-- < 0) {
+            status = TIMED_OUT;
+            break;
+        }
+        status = AudioSystem::listAudioPatches(&numPatches, nullptr, &generation1);
+        if (status != NO_ERROR) {
+            ALOGE("AudioSystem::listAudioPatches returned error %d", status);
+            break;
+        }
+        patchesVec.resize(numPatches);
+        status = AudioSystem::listAudioPatches(&numPatches, patchesVec.data(), &generation);
+    } while (generation1 != generation && status == NO_ERROR);
+    if (status != NO_ERROR) {
+        numPatches = 0;
+        patchesVec.clear();
+    }
+    return status;
+}
+
+status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch) {
+    std::vector<struct audio_patch> patches;
+    status_t status = listAudioPatches(patches);
+    if (status != OK) return status;
+
+    for (auto i = 0; i < patches.size(); i++) {
+        for (auto j = 0; j < patches[i].num_sources; j++) {
+            if (patches[i].sources[j].type == AUDIO_PORT_TYPE_MIX &&
+                patches[i].sources[j].ext.mix.handle == audioIo) {
+                patch = patches[i];
+                return OK;
+            }
+        }
+    }
+    return BAD_VALUE;
+}
+
+status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch) {
+    std::vector<struct audio_patch> patches;
+    status_t status = listAudioPatches(patches);
+    if (status != OK) return status;
+
+    for (auto i = 0; i < patches.size(); i++) {
+        for (auto j = 0; j < patches[i].num_sinks; j++) {
+            if (patches[i].sinks[j].type == AUDIO_PORT_TYPE_MIX &&
+                patches[i].sinks[j].ext.mix.handle == audioIo) {
+                patch = patches[i];
+                return OK;
+            }
+        }
+    }
+    return BAD_VALUE;
+}
+
+bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch) {
+    for (auto j = 0; j < patch.num_sinks; j++) {
+        if (patch.sinks[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sinks[j].id == deviceId) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch) {
+    for (auto j = 0; j < patch.num_sources; j++) {
+        if (patch.sources[j].type == AUDIO_PORT_TYPE_DEVICE && patch.sources[j].id == deviceId) {
+            return true;
+        }
+    }
+    return false;
+}
+
+bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId) {
+    struct audio_patch patch;
+    if (getPatchForOutputMix(audioIo, patch) == OK) {
+        return patchContainsOutputDevice(deviceId, patch);
+    }
+    return false;
+}
+
+bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId) {
+    struct audio_patch patch;
+    if (getPatchForInputMix(audioIo, patch) == OK) {
+        return patchContainsInputDevice(deviceId, patch);
+    }
+    return false;
+}
+
+std::string dumpPortConfig(const audio_port_config& port) {
+    std::ostringstream result;
+    std::string deviceInfo;
+    if (port.type == AUDIO_PORT_TYPE_DEVICE) {
+        if (port.ext.device.type & AUDIO_DEVICE_BIT_IN) {
+            InputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        } else {
+            OutputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        }
+        deviceInfo += std::string(", address = ") + port.ext.device.address;
+    }
+    result << "audio_port_handle_t = " << port.id << ", "
+           << "Role = " << (port.role == AUDIO_PORT_ROLE_SOURCE ? "source" : "sink") << ", "
+           << "Type = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? "device" : "mix") << ", "
+           << "deviceInfo = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? deviceInfo : "") << ", "
+           << "config_mask = 0x" << std::hex << port.config_mask << std::dec << ", ";
+    if (port.config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
+        result << "sample rate = " << port.sample_rate << ", ";
+    }
+    if (port.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
+        result << "channel mask = " << port.channel_mask << ", ";
+    }
+    if (port.config_mask & AUDIO_PORT_CONFIG_FORMAT) {
+        result << "format = " << port.format << ", ";
+    }
+    result << "input flags = " << port.flags.input << ", ";
+    result << "output flags = " << port.flags.output << ", ";
+    result << "mix io handle = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? 0 : port.ext.mix.handle)
+           << "\n";
+    return result.str();
+}
+
+std::string dumpPatch(const audio_patch& patch) {
+    std::ostringstream result;
+    result << "----------------- Dumping Patch ------------ \n";
+    result << "Patch Handle: " << patch.id << ", sources: " << patch.num_sources
+           << ", sink: " << patch.num_sinks << "\n";
+    audio_port_v7 port;
+    for (uint32_t i = 0; i < patch.num_sources; i++) {
+        result << "----------------- Dumping Source Port Config @ index " << i
+               << " ------------ \n";
+        result << dumpPortConfig(patch.sources[i]);
+        result << "----------------- Dumping Source Port for id " << patch.sources[i].id
+               << " ------------ \n";
+        getPortById(patch.sources[i].id, port);
+        result << dumpPort(port);
+    }
+    for (uint32_t i = 0; i < patch.num_sinks; i++) {
+        result << "----------------- Dumping Sink Port Config @ index " << i << " ------------ \n";
+        result << dumpPortConfig(patch.sinks[i]);
+        result << "----------------- Dumping Sink Port for id " << patch.sinks[i].id
+               << " ------------ \n";
+        getPortById(patch.sinks[i].id, port);
+        result << dumpPort(port);
+    }
+    return result.str();
+}
+
+std::string dumpPort(const audio_port_v7& port) {
+    std::ostringstream result;
+    std::string deviceInfo;
+    if (port.type == AUDIO_PORT_TYPE_DEVICE) {
+        if (port.ext.device.type & AUDIO_DEVICE_BIT_IN) {
+            InputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        } else {
+            OutputDeviceConverter::maskToString(port.ext.device.type, deviceInfo);
+        }
+        deviceInfo += std::string(", address = ") + port.ext.device.address;
+    }
+    result << "audio_port_handle_t = " << port.id << ", "
+           << "Role = " << (port.role == AUDIO_PORT_ROLE_SOURCE ? "source" : "sink") << ", "
+           << "Type = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? "device" : "mix") << ", "
+           << "deviceInfo = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? deviceInfo : "") << ", "
+           << "Name = " << port.name << ", "
+           << "num profiles = " << port.num_audio_profiles << ", "
+           << "mix io handle = " << (port.type == AUDIO_PORT_TYPE_DEVICE ? 0 : port.ext.mix.handle)
+           << ", ";
+    for (int i = 0; i < port.num_audio_profiles; i++) {
+        result << "AudioProfile = " << i << " {";
+        result << "format = " << port.audio_profiles[i].format << ", ";
+        result << "samplerates = ";
+        for (int j = 0; j < port.audio_profiles[i].num_sample_rates; j++) {
+            result << port.audio_profiles[i].sample_rates[j] << ", ";
+        }
+        result << "channelmasks = ";
+        for (int j = 0; j < port.audio_profiles[i].num_channel_masks; j++) {
+            result << "0x" << std::hex << port.audio_profiles[i].channel_masks[j] << std::dec
+                   << ", ";
+        }
+        result << "} ";
+    }
+    result << dumpPortConfig(port.active_config);
+    return result.str();
+}
diff --git a/media/libaudioclient/tests/audio_test_utils.h b/media/libaudioclient/tests/audio_test_utils.h
new file mode 100644
index 0000000..fc269ed
--- /dev/null
+++ b/media/libaudioclient/tests/audio_test_utils.h
@@ -0,0 +1,186 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#ifndef AUDIO_TEST_UTILS_H_
+#define AUDIO_TEST_UTILS_H_
+
+#include <sys/stat.h>
+#include <unistd.h>
+#include <atomic>
+#include <chrono>
+#include <cinttypes>
+#include <deque>
+#include <memory>
+#include <mutex>
+#include <thread>
+
+#include <binder/MemoryDealer.h>
+#include <media/AidlConversion.h>
+#include <media/AudioRecord.h>
+#include <media/AudioTrack.h>
+
+#define RECORD_TO_FILE 0
+
+using namespace android;
+
+void CreateRandomFile(int& fd);
+status_t listAudioPorts(std::vector<audio_port_v7>& portsVec);
+status_t listAudioPatches(std::vector<struct audio_patch>& patchesVec);
+status_t getPortByAttributes(audio_port_role_t role, audio_port_type_t type,
+                             audio_devices_t deviceType, audio_port_v7& port);
+status_t getPatchForOutputMix(audio_io_handle_t audioIo, audio_patch& patch);
+status_t getPatchForInputMix(audio_io_handle_t audioIo, audio_patch& patch);
+bool patchContainsOutputDevice(audio_port_handle_t deviceId, audio_patch patch);
+bool patchContainsInputDevice(audio_port_handle_t deviceId, audio_patch patch);
+bool checkPatchPlayback(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+bool checkPatchCapture(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+std::string dumpPort(const audio_port_v7& port);
+std::string dumpPortConfig(const audio_port_config& port);
+std::string dumpPatch(const audio_patch& patch);
+
+class OnAudioDeviceUpdateNotifier : public AudioSystem::AudioDeviceCallback {
+  public:
+    audio_io_handle_t mAudioIo = AUDIO_IO_HANDLE_NONE;
+    audio_port_handle_t mDeviceId = AUDIO_PORT_HANDLE_NONE;
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+
+    void onAudioDeviceUpdate(audio_io_handle_t audioIo, audio_port_handle_t deviceId);
+    status_t waitForAudioDeviceCb();
+};
+
+// Simple AudioPlayback class.
+class AudioPlayback : public AudioTrack::IAudioTrackCallback {
+  friend sp<AudioPlayback>;
+    AudioPlayback(uint32_t sampleRate, audio_format_t format, audio_channel_mask_t channelMask,
+                  audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE,
+                  audio_session_t sessionId = AUDIO_SESSION_NONE,
+                  AudioTrack::transfer_type transferType = AudioTrack::TRANSFER_SHARED,
+                  audio_attributes_t* attributes = nullptr);
+  public:
+    status_t loadResource(const char* name);
+    status_t create();
+    sp<AudioTrack> getAudioTrackHandle();
+    status_t start();
+    status_t waitForConsumption(bool testSeek = false);
+    status_t fillBuffer();
+    status_t onProcess(bool testSeek = false);
+    virtual void onBufferEnd() override;
+    void stop();
+
+    bool mStopPlaying;
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+
+    enum State {
+        PLAY_NO_INIT,
+        PLAY_READY,
+        PLAY_STARTED,
+        PLAY_STOPPED,
+    };
+
+  private:
+    ~AudioPlayback();
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_output_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const AudioTrack::transfer_type mTransferType;
+    const audio_attributes_t* mAttributes;
+
+    size_t mBytesUsedSoFar;
+    State mState;
+    size_t mMemCapacity;
+    sp<MemoryDealer> mMemoryDealer;
+    sp<IMemory> mMemory;
+
+    sp<AudioTrack> mTrack;
+};
+
+// hold pcm data sent by AudioRecord
+class RawBuffer {
+  public:
+    RawBuffer(int64_t ptsPipeline = -1, int64_t ptsManual = -1, int32_t capacity = 0);
+
+    std::unique_ptr<uint8_t[]> mData;
+    int64_t mPtsPipeline;
+    int64_t mPtsManual;
+    int32_t mCapacity;
+};
+
+// Simple AudioCapture
+class AudioCapture : public AudioRecord::IAudioRecordCallback {
+  public:
+    AudioCapture(audio_source_t inputSource, uint32_t sampleRate, audio_format_t format,
+                 audio_channel_mask_t channelMask,
+                 audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE,
+                 audio_session_t sessionId = AUDIO_SESSION_ALLOCATE,
+                 AudioRecord::transfer_type transferType = AudioRecord::TRANSFER_CALLBACK);
+    ~AudioCapture();
+    size_t onMoreData(const AudioRecord::Buffer& buffer) override;
+    void onOverrun() override;
+    void onMarker(uint32_t markerPosition) override;
+    void onNewPos(uint32_t newPos) override;
+    void onNewIAudioRecord() override;
+    status_t create();
+    sp<AudioRecord> getAudioRecordHandle();
+    status_t start(AudioSystem::sync_event_t event = AudioSystem::SYNC_EVENT_NONE,
+                   audio_session_t triggerSession = AUDIO_SESSION_NONE);
+    status_t obtainBufferCb(RawBuffer& buffer);
+    status_t obtainBuffer(RawBuffer& buffer);
+    status_t audioProcess();
+    status_t stop();
+
+    uint32_t mFrameCount;
+    uint32_t mNotificationFrames;
+    int64_t mNumFramesToRecord;
+    int64_t mNumFramesReceived;
+    int64_t mNumFramesLost;
+    uint32_t mMarkerPosition;
+    uint32_t mMarkerPeriod;
+    uint32_t mReceivedCbMarkerAtPosition;
+    uint32_t mReceivedCbMarkerCount;
+    bool mBufferOverrun;
+
+    enum State {
+        REC_NO_INIT,
+        REC_READY,
+        REC_STARTED,
+        REC_STOPPED,
+    };
+
+  private:
+    const audio_source_t mInputSource;
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_input_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const AudioRecord::transfer_type mTransferType;
+
+    size_t mMaxBytesPerCallback = 2048;
+    sp<AudioRecord> mRecord;
+    State mState;
+    bool mStopRecording;
+    int mOutFileFd = -1;
+
+    std::mutex mMutex;
+    std::condition_variable mCondition;
+    std::deque<RawBuffer> mBuffersReceived;
+};
+
+#endif  // AUDIO_TEST_UTILS_H_
diff --git a/media/libaudioclient/tests/audioclient_serialization_tests.cpp b/media/libaudioclient/tests/audioclient_serialization_tests.cpp
new file mode 100644
index 0000000..93baefd6
--- /dev/null
+++ b/media/libaudioclient/tests/audioclient_serialization_tests.cpp
@@ -0,0 +1,310 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioClientSerializationUnitTests"
+
+#include <cstdint>
+#include <cstdlib>
+#include <ctime>
+
+#include <gtest/gtest.h>
+
+#include <android_audio_policy_configuration_V7_0-enums.h>
+#include <xsdc/XsdcSupport.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+namespace xsd {
+using namespace ::android::audio::policy::configuration::V7_0;
+}
+
+template <typename T, typename X, typename FUNC>
+std::vector<T> getFlags(const xsdc_enum_range<X>& range, const FUNC& func,
+                        const std::string& findString = {}) {
+    std::vector<T> vec;
+    for (const auto& xsdEnumVal : range) {
+        T enumVal;
+        std::string enumString = toString(xsdEnumVal);
+        if (enumString.find(findString) != std::string::npos &&
+            func(enumString.c_str(), &enumVal)) {
+            vec.push_back(enumVal);
+        }
+    }
+    return vec;
+}
+
+static const std::vector<audio_usage_t> kUsages =
+        getFlags<audio_usage_t, xsd::AudioUsage, decltype(audio_usage_from_string)>(
+                xsdc_enum_range<xsd::AudioUsage>{}, audio_usage_from_string);
+
+static const std::vector<audio_content_type_t> kContentType =
+        getFlags<audio_content_type_t, xsd::AudioContentType,
+                 decltype(audio_content_type_from_string)>(xsdc_enum_range<xsd::AudioContentType>{},
+                                                           audio_content_type_from_string);
+
+static const std::vector<audio_source_t> kInputSources =
+        getFlags<audio_source_t, xsd::AudioSource, decltype(audio_source_from_string)>(
+                xsdc_enum_range<xsd::AudioSource>{}, audio_source_from_string);
+
+static const std::vector<audio_stream_type_t> kStreamtypes =
+        getFlags<audio_stream_type_t, xsd::AudioStreamType,
+                 decltype(audio_stream_type_from_string)>(xsdc_enum_range<xsd::AudioStreamType>{},
+                                                          audio_stream_type_from_string);
+
+static const std::vector<uint32_t> kMixMatchRules = {
+        RULE_MATCH_ATTRIBUTE_USAGE,
+        RULE_EXCLUDE_ATTRIBUTE_USAGE,
+        RULE_MATCH_ATTRIBUTE_CAPTURE_PRESET,
+        RULE_EXCLUDE_ATTRIBUTE_CAPTURE_PRESET,
+        RULE_MATCH_UID,
+        RULE_EXCLUDE_UID,
+        RULE_MATCH_USERID,
+        RULE_EXCLUDE_USERID,
+};
+
+// Generates a random string.
+std::string CreateRandomString(size_t n) {
+    std::string data =
+            "abcdefghijklmnopqrstuvwxyz"
+            "ABCDEFGHIJKLMNOPQRSTUVWXYZ"
+            "0123456789";
+    srand(static_cast<unsigned int>(time(0)));
+    std::string s(n, ' ');
+    for (size_t i = 0; i < n; ++i) {
+        s[i] = data[rand() % data.size()];
+    }
+    return s;
+}
+
+class FillAudioAttributes {
+  public:
+    void fillAudioAttributes(audio_attributes_t& attr);
+
+    unsigned int mSeed;
+};
+
+void FillAudioAttributes::fillAudioAttributes(audio_attributes_t& attr) {
+    attr.content_type = kContentType[rand() % kContentType.size()];
+    attr.usage = kUsages[rand() % kUsages.size()];
+    attr.source = kInputSources[rand() % kInputSources.size()];
+    // attr.flags -> [0, (1 << (CAPTURE_PRIVATE + 1) - 1)]
+    attr.flags = static_cast<audio_flags_mask_t>(rand() & 0x3fff);
+    sprintf(attr.tags, "%s",
+            CreateRandomString((int)rand() % (AUDIO_ATTRIBUTES_TAGS_MAX_SIZE - 1)).c_str());
+}
+
+class SerializationTest : public FillAudioAttributes, public ::testing::Test {
+    void SetUp() override {
+        mSeed = static_cast<unsigned int>(time(0));
+        srand(mSeed);
+    }
+};
+
+// UNIT TESTS
+TEST_F(SerializationTest, AudioProductStrategyBinderization) {
+    for (int j = 0; j < 512; j++) {
+        const std::string name{"Test APSBinderization for seed::" + std::to_string(mSeed)};
+        std::vector<AudioAttributes> audioattributesvector;
+        for (auto i = 0; i < 16; i++) {
+            audio_attributes_t attributes;
+            fillAudioAttributes(attributes);
+            AudioAttributes audioattributes{static_cast<volume_group_t>(rand()),
+                                            kStreamtypes[rand() % kStreamtypes.size()], attributes};
+            audioattributesvector.push_back(audioattributes);
+        }
+        product_strategy_t psId = static_cast<product_strategy_t>(rand());
+        AudioProductStrategy aps{name, audioattributesvector, psId};
+
+        Parcel p;
+        EXPECT_EQ(NO_ERROR, aps.writeToParcel(&p)) << name;
+
+        AudioProductStrategy apsCopy;
+        p.setDataPosition(0);
+        EXPECT_EQ(NO_ERROR, apsCopy.readFromParcel(&p)) << name;
+        EXPECT_EQ(apsCopy.getName(), name) << name;
+        EXPECT_EQ(apsCopy.getId(), psId) << name;
+        auto avec = apsCopy.getAudioAttributes();
+        EXPECT_EQ(avec.size(), audioattributesvector.size()) << name;
+        for (int i = 0; i < audioattributesvector.size(); i++) {
+            EXPECT_EQ(avec[i].getGroupId(), audioattributesvector[i].getGroupId()) << name;
+            EXPECT_EQ(avec[i].getStreamType(), audioattributesvector[i].getStreamType()) << name;
+            EXPECT_TRUE(avec[i].getAttributes() == audioattributesvector[i].getAttributes())
+                    << name;
+        }
+    }
+}
+
+TEST_F(SerializationTest, AudioVolumeGroupBinderization) {
+    for (int j = 0; j < 512; j++) {
+        const std::string name{"Test AVGBinderization for seed::" + std::to_string(mSeed)};
+        volume_group_t groupId = static_cast<volume_group_t>(rand());
+        std::vector<audio_attributes_t> attributesvector;
+        for (auto i = 0; i < 16; i++) {
+            audio_attributes_t attributes;
+            fillAudioAttributes(attributes);
+            attributesvector.push_back(attributes);
+        }
+        std::vector<audio_stream_type_t> streamsvector;
+        for (auto i = 0; i < 8; i++) {
+            streamsvector.push_back(kStreamtypes[rand() % kStreamtypes.size()]);
+        }
+        AudioVolumeGroup avg{name, groupId, attributesvector, streamsvector};
+
+        Parcel p;
+        EXPECT_EQ(NO_ERROR, avg.writeToParcel(&p));
+
+        AudioVolumeGroup avgCopy;
+        p.setDataPosition(0);
+        EXPECT_EQ(NO_ERROR, avgCopy.readFromParcel(&p)) << name;
+        EXPECT_EQ(avgCopy.getName(), name) << name;
+        EXPECT_EQ(avgCopy.getId(), groupId) << name;
+        auto avec = avgCopy.getAudioAttributes();
+        EXPECT_EQ(avec.size(), attributesvector.size()) << name;
+        for (int i = 0; i < avec.size(); i++) {
+            EXPECT_TRUE(avec[i] == attributesvector[i]) << name;
+        }
+        StreamTypeVector svec = avgCopy.getStreamTypes();
+        EXPECT_EQ(svec.size(), streamsvector.size()) << name;
+        for (int i = 0; i < svec.size(); i++) {
+            EXPECT_EQ(svec[i], streamsvector[i]) << name;
+        }
+    }
+}
+
+TEST_F(SerializationTest, AudioMixBinderization) {
+    for (int j = 0; j < 512; j++) {
+        const std::string msg{"Test AMBinderization for seed::" + std::to_string(mSeed)};
+        Vector<AudioMixMatchCriterion> criteria;
+        for (int i = 0; i < 16; i++) {
+            AudioMixMatchCriterion ammc{kUsages[rand() % kUsages.size()],
+                                        kInputSources[rand() % kInputSources.size()],
+                                        kMixMatchRules[rand() % kMixMatchRules.size()]};
+            criteria.add(ammc);
+        }
+        audio_config_t config{};
+        config.sample_rate = 48000;
+        config.channel_mask = AUDIO_CHANNEL_IN_MONO;
+        config.format = AUDIO_FORMAT_PCM_16_BIT;
+        config.offload_info = AUDIO_INFO_INITIALIZER;
+        config.frame_count = 4800;
+        AudioMix am{criteria,
+                    static_cast<uint32_t>(rand()),
+                    config,
+                    static_cast<uint32_t>(rand()),
+                    String8(msg.c_str()),
+                    static_cast<uint32_t>(rand())};
+
+        Parcel p;
+        EXPECT_EQ(NO_ERROR, am.writeToParcel(&p)) << msg;
+
+        AudioMix amCopy;
+        p.setDataPosition(0);
+        EXPECT_EQ(NO_ERROR, amCopy.readFromParcel(&p)) << msg;
+        EXPECT_EQ(amCopy.mMixType, am.mMixType) << msg;
+        EXPECT_EQ(amCopy.mFormat.sample_rate, am.mFormat.sample_rate) << msg;
+        EXPECT_EQ(amCopy.mFormat.channel_mask, am.mFormat.channel_mask) << msg;
+        EXPECT_EQ(amCopy.mFormat.format, am.mFormat.format) << msg;
+        EXPECT_EQ(amCopy.mRouteFlags, am.mRouteFlags) << msg;
+        EXPECT_EQ(amCopy.mDeviceAddress, am.mDeviceAddress) << msg;
+        EXPECT_EQ(amCopy.mCbFlags, am.mCbFlags) << msg;
+        EXPECT_EQ(amCopy.mCriteria.size(), am.mCriteria.size()) << msg;
+        for (auto i = 0; i < amCopy.mCriteria.size(); i++) {
+            EXPECT_EQ(amCopy.mCriteria[i].mRule, am.mCriteria[i].mRule) << msg;
+            EXPECT_EQ(amCopy.mCriteria[i].mValue.mUserId, am.mCriteria[i].mValue.mUserId) << msg;
+        }
+    }
+}
+
+using MMCTestParams = std::tuple<audio_usage_t, audio_source_t, uint32_t>;
+
+class MMCParameterizedTest : public FillAudioAttributes,
+                             public ::testing::TestWithParam<MMCTestParams> {
+  public:
+    MMCParameterizedTest()
+        : mAudioUsage(std::get<0>(GetParam())),
+          mAudioSource(std::get<1>(GetParam())),
+          mAudioMixMatchRules(std::get<2>(GetParam())){};
+
+    const audio_usage_t mAudioUsage;
+    const audio_source_t mAudioSource;
+    const uint32_t mAudioMixMatchRules;
+
+    void SetUp() override {
+        mSeed = static_cast<unsigned int>(time(0));
+        srand(mSeed);
+    }
+};
+
+TEST_P(MMCParameterizedTest, AudioMixMatchCriterionBinderization) {
+    const std::string msg{"Test AMMCBinderization for seed::" + std::to_string(mSeed)};
+    AudioMixMatchCriterion ammc{mAudioUsage, mAudioSource, mAudioMixMatchRules};
+
+    Parcel p;
+    EXPECT_EQ(NO_ERROR, ammc.writeToParcel(&p)) << msg;
+
+    AudioMixMatchCriterion ammcCopy;
+    p.setDataPosition(0);
+    EXPECT_EQ(NO_ERROR, ammcCopy.readFromParcel(&p)) << msg;
+    EXPECT_EQ(ammcCopy.mRule, ammc.mRule) << msg;
+    EXPECT_EQ(ammcCopy.mValue.mUserId, ammc.mValue.mUserId) << msg;
+}
+
+// audioUsage, audioSource, audioMixMatchRules
+INSTANTIATE_TEST_SUITE_P(SerializationParameterizedTests, MMCParameterizedTest,
+                         ::testing::Combine(testing::ValuesIn(kUsages),
+                                            testing::ValuesIn(kInputSources),
+                                            testing::ValuesIn(kMixMatchRules)));
+
+using AudioAttributesTestParams = std::tuple<audio_stream_type_t>;
+
+class AudioAttributesParameterizedTest
+    : public FillAudioAttributes,
+      public ::testing::TestWithParam<AudioAttributesTestParams> {
+  public:
+    AudioAttributesParameterizedTest() : mAudioStream(std::get<0>(GetParam())){};
+
+    const audio_stream_type_t mAudioStream;
+
+    void SetUp() override {
+        mSeed = static_cast<unsigned int>(time(0));
+        srand(mSeed);
+    }
+};
+
+TEST_P(AudioAttributesParameterizedTest, AudioAttributesBinderization) {
+    const std::string msg{"Test AABinderization for seed::" + std::to_string(mSeed)};
+    volume_group_t groupId = static_cast<volume_group_t>(rand());
+    audio_stream_type_t stream = mAudioStream;
+    audio_attributes_t attributes;
+    fillAudioAttributes(attributes);
+    AudioAttributes audioattributes{groupId, stream, attributes};
+
+    Parcel p;
+    EXPECT_EQ(NO_ERROR, audioattributes.writeToParcel(&p)) << msg;
+
+    AudioAttributes audioattributesCopy;
+    p.setDataPosition(0);
+    EXPECT_EQ(NO_ERROR, audioattributesCopy.readFromParcel(&p)) << msg;
+    EXPECT_EQ(audioattributesCopy.getGroupId(), audioattributes.getGroupId()) << msg;
+    EXPECT_EQ(audioattributesCopy.getStreamType(), audioattributes.getStreamType()) << msg;
+    EXPECT_TRUE(audioattributesCopy.getAttributes() == attributes) << msg;
+}
+
+// audioStream
+INSTANTIATE_TEST_SUITE_P(SerializationParameterizedTests, AudioAttributesParameterizedTest,
+                         ::testing::Combine(testing::ValuesIn(kStreamtypes)));
diff --git a/media/libaudioclient/tests/audioeffect_tests.cpp b/media/libaudioclient/tests/audioeffect_tests.cpp
new file mode 100644
index 0000000..93fe306
--- /dev/null
+++ b/media/libaudioclient/tests/audioeffect_tests.cpp
@@ -0,0 +1,335 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioEffectUnitTests"
+
+#include <gtest/gtest.h>
+#include <media/AudioEffect.h>
+#include <system/audio_effects/effect_visualizer.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+static constexpr int kDefaultInputEffectPriority = -1;
+static constexpr int kDefaultOutputEffectPriority = 0;
+
+static const char* gPackageName = "AudioEffectTest";
+
+bool isEffectExistsOnAudioSession(const effect_uuid_t* type, int priority,
+                                  audio_session_t sessionId) {
+    std::string packageName{gPackageName};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    sp<AudioEffect> effect = new AudioEffect(attributionSource);
+    effect->set(type, nullptr /* uid */, priority, nullptr /* callback */, sessionId);
+    return effect->initCheck() == ALREADY_EXISTS;
+}
+
+bool isEffectDefaultOnRecord(const effect_uuid_t* type, const sp<AudioRecord>& audioRecord) {
+    effect_descriptor_t descriptors[AudioEffect::kMaxPreProcessing];
+    uint32_t numEffects = AudioEffect::kMaxPreProcessing;
+    status_t ret = AudioEffect::queryDefaultPreProcessing(audioRecord->getSessionId(), descriptors,
+                                                          &numEffects);
+    if (ret != OK) {
+        return false;
+    }
+    for (int i = 0; i < numEffects; i++) {
+        if (memcmp(&descriptors[i].type, type, sizeof(effect_uuid_t)) == 0) {
+            return true;
+        }
+    }
+    return false;
+}
+
+void listEffectsAvailable(std::vector<effect_descriptor_t>& descriptors) {
+    uint32_t numEffects = 0;
+    if (NO_ERROR == AudioEffect::queryNumberEffects(&numEffects)) {
+        for (auto i = 0; i < numEffects; i++) {
+            effect_descriptor_t des;
+            if (NO_ERROR == AudioEffect::queryEffect(i, &des)) descriptors.push_back(des);
+        }
+    }
+}
+
+bool isPreprocessing(effect_descriptor_t& descriptor) {
+    return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_PRE_PROC);
+}
+
+bool isInsert(effect_descriptor_t& descriptor) {
+    return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT);
+}
+
+bool isAux(effect_descriptor_t& descriptor) {
+    return ((descriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY);
+}
+
+bool isFastCompatible(effect_descriptor_t& descriptor) {
+    return !(((descriptor.flags & EFFECT_FLAG_HW_ACC_MASK) == 0) &&
+             ((descriptor.flags & EFFECT_FLAG_NO_PROCESS) == 0));
+}
+
+// UNIT TESTS
+TEST(AudioEffectTest, getEffectDescriptor) {
+    effect_uuid_t randomType = {
+            0x81781c08, 0x93dd, 0x11ec, 0xb909, {0x02, 0x42, 0xac, 0x12, 0x00, 0x02}};
+    effect_uuid_t randomUuid = {
+            0x653730e1, 0x1be1, 0x438e, 0xa35a, {0xfc, 0x9b, 0xa1, 0x2a, 0x5e, 0xc9}};
+    effect_uuid_t empty = EFFECT_UUID_INITIALIZER;
+
+    effect_descriptor_t descriptor;
+    EXPECT_EQ(NAME_NOT_FOUND, AudioEffect::getEffectDescriptor(&randomUuid, &randomType,
+                                                               EFFECT_FLAG_TYPE_MASK, &descriptor));
+
+    std::vector<effect_descriptor_t> descriptors;
+    listEffectsAvailable(descriptors);
+
+    for (auto i = 0; i < descriptors.size(); i++) {
+        EXPECT_EQ(NO_ERROR,
+                  AudioEffect::getEffectDescriptor(&descriptors[i].uuid, &descriptors[i].type,
+                                                   EFFECT_FLAG_TYPE_MASK, &descriptor));
+        EXPECT_EQ(0, memcmp(&descriptor, &descriptors[i], sizeof(effect_uuid_t)));
+    }
+    // negative tests
+    if (descriptors.size() > 0) {
+        EXPECT_EQ(BAD_VALUE,
+                  AudioEffect::getEffectDescriptor(&descriptors[0].uuid, &descriptors[0].type,
+                                                   EFFECT_FLAG_TYPE_MASK, nullptr));
+    }
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(nullptr, nullptr,
+                                                          EFFECT_FLAG_TYPE_PRE_PROC, &descriptor));
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(&empty, &randomType,
+                                                          EFFECT_FLAG_TYPE_MASK, nullptr));
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(nullptr, &randomType,
+                                                          EFFECT_FLAG_TYPE_POST_PROC, &descriptor));
+    EXPECT_EQ(BAD_VALUE, AudioEffect::getEffectDescriptor(&randomUuid, nullptr,
+                                                          EFFECT_FLAG_TYPE_INSERT, &descriptor));
+}
+
+TEST(AudioEffectTest, DISABLED_GetSetParameterForEffect) {
+    std::string packageName{gPackageName};
+    AttributionSourceState attributionSource;
+    attributionSource.packageName = packageName;
+    attributionSource.uid = VALUE_OR_FATAL(legacy2aidl_uid_t_int32_t(getuid()));
+    attributionSource.pid = VALUE_OR_FATAL(legacy2aidl_pid_t_int32_t(getpid()));
+    attributionSource.token = sp<BBinder>::make();
+    sp<AudioEffect> visualizer = new AudioEffect(attributionSource);
+    ASSERT_NE(visualizer, nullptr) << "effect not created";
+    visualizer->set(SL_IID_VISUALIZATION);
+    status_t status = visualizer->initCheck();
+    ASSERT_TRUE(status == NO_ERROR || status == ALREADY_EXISTS) << "Init check error";
+    ASSERT_EQ(NO_ERROR, visualizer->setEnabled(true)) << "visualizer not enabled";
+
+    uint32_t buf32[3][sizeof(effect_param_t) / sizeof(uint32_t) + 2];
+    effect_param_t* vis_none = (effect_param_t*)(buf32[0]);
+    effect_param_t* vis_rms = (effect_param_t*)(buf32[1]);
+    effect_param_t* vis_tmp = (effect_param_t*)(buf32[2]);
+
+    // Visualizer::setMeasurementMode()
+    vis_none->psize = sizeof(uint32_t);
+    vis_none->vsize = sizeof(uint32_t);
+    *(int32_t*)vis_none->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+    *((int32_t*)vis_none->data + 1) = MEASUREMENT_MODE_NONE;
+    EXPECT_EQ(NO_ERROR, visualizer->setParameter(vis_none))
+            << "setMeasurementMode doesn't report success";
+
+    // Visualizer::getMeasurementMode()
+    vis_tmp->psize = sizeof(uint32_t);
+    vis_tmp->vsize = sizeof(uint32_t);
+    *(int32_t*)vis_tmp->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+    *((int32_t*)vis_tmp->data + 1) = 23;
+    EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+            << "getMeasurementMode doesn't report success";
+    EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_none->data + 1))
+            << "target mode does not match set mode";
+
+    // Visualizer::setMeasurementModeDeferred()
+    vis_rms->psize = sizeof(uint32_t);
+    vis_rms->vsize = sizeof(uint32_t);
+    *(int32_t*)vis_rms->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
+    *((int32_t*)vis_rms->data + 1) = MEASUREMENT_MODE_PEAK_RMS;
+    EXPECT_EQ(NO_ERROR, visualizer->setParameterDeferred(vis_rms))
+            << "setMeasurementModeDeferred doesn't report success";
+
+    *((int32_t*)vis_tmp->data + 1) = 23;
+    EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+            << "getMeasurementMode doesn't report success";
+    EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_none->data + 1))
+            << "target mode does not match set mode";
+
+    // setParameterCommit
+    EXPECT_EQ(NO_ERROR, visualizer->setParameterCommit())
+            << "setMeasurementModeCommit does not report success";
+
+    // validate Params
+    *((int32_t*)vis_tmp->data + 1) = 23;
+    EXPECT_EQ(NO_ERROR, visualizer->getParameter(vis_tmp))
+            << "getMeasurementMode doesn't report success";
+    EXPECT_EQ(*((int32_t*)vis_tmp->data + 1), *((int32_t*)vis_rms->data + 1))
+            << "target mode does not match set mode";
+}
+
+TEST(AudioEffectTest, ManageSourceDefaultEffects) {
+    int32_t selectedEffect = -1;
+
+    const uint32_t sampleRate = 44100;
+    const audio_format_t format = AUDIO_FORMAT_PCM_16_BIT;
+    const audio_channel_mask_t channelMask = AUDIO_CHANNEL_IN_STEREO;
+    sp<AudioCapture> capture = nullptr;
+
+    std::vector<effect_descriptor_t> descriptors;
+    listEffectsAvailable(descriptors);
+    for (auto i = 0; i < descriptors.size(); i++) {
+        if (isPreprocessing(descriptors[i])) {
+            capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+            ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+            EXPECT_EQ(NO_ERROR, capture->create());
+            EXPECT_EQ(NO_ERROR, capture->start());
+            if (!isEffectDefaultOnRecord(&descriptors[i].type, capture->getAudioRecordHandle())) {
+                selectedEffect = i;
+                break;
+            }
+        }
+    }
+    if (selectedEffect == -1) GTEST_SKIP() << " expected at least one preprocessing effect";
+    effect_uuid_t selectedEffectType = descriptors[selectedEffect].type;
+
+    char type[512];
+    AudioEffect::guidToString(&selectedEffectType, type, sizeof(type));
+
+    capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+    ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+    EXPECT_EQ(NO_ERROR, capture->create());
+    EXPECT_EQ(NO_ERROR, capture->start());
+    EXPECT_FALSE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+            << "Effect should not have been default on record. " << type;
+    EXPECT_FALSE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+                                              capture->getAudioRecordHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(OK, capture->audioProcess());
+    EXPECT_EQ(OK, capture->stop());
+
+    String16 name{gPackageName};
+    audio_unique_id_t effectId;
+    status_t status = AudioEffect::addSourceDefaultEffect(
+            type, name, nullptr, kDefaultInputEffectPriority, AUDIO_SOURCE_MIC, &effectId);
+    EXPECT_EQ(NO_ERROR, status) << "Adding default effect failed: " << type;
+
+    capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+    ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+    EXPECT_EQ(NO_ERROR, capture->create());
+    EXPECT_EQ(NO_ERROR, capture->start());
+    EXPECT_TRUE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+            << "Effect should have been default on record. " << type;
+    EXPECT_TRUE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+                                             capture->getAudioRecordHandle()->getSessionId()))
+            << "Effect should have been added. " << type;
+    EXPECT_EQ(OK, capture->audioProcess());
+    EXPECT_EQ(OK, capture->stop());
+
+    status = AudioEffect::removeSourceDefaultEffect(effectId);
+    EXPECT_EQ(NO_ERROR, status);
+    capture = new AudioCapture(AUDIO_SOURCE_MIC, sampleRate, format, channelMask);
+    ASSERT_NE(capture, nullptr) << "Unable to create Record Application";
+    EXPECT_EQ(NO_ERROR, capture->create());
+    EXPECT_EQ(NO_ERROR, capture->start());
+    EXPECT_FALSE(isEffectDefaultOnRecord(&selectedEffectType, capture->getAudioRecordHandle()))
+            << "Effect should not have been default on record. " << type;
+    EXPECT_FALSE(isEffectExistsOnAudioSession(&selectedEffectType, kDefaultInputEffectPriority - 1,
+                                              capture->getAudioRecordHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(OK, capture->audioProcess());
+    EXPECT_EQ(OK, capture->stop());
+}
+
+TEST(AudioEffectTest, ManageStreamDefaultEffects) {
+    int32_t selectedEffect = -1;
+
+    std::vector<effect_descriptor_t> descriptors;
+    listEffectsAvailable(descriptors);
+    for (auto i = 0; i < descriptors.size(); i++) {
+        if (isAux(descriptors[i])) {
+            selectedEffect = i;
+            break;
+        }
+    }
+    if (selectedEffect == -1) GTEST_SKIP() << " expected at least one Aux effect";
+    effect_uuid_t* selectedEffectType = &descriptors[selectedEffect].type;
+
+    char type[512];
+    AudioEffect::guidToString(selectedEffectType, type, sizeof(type));
+    // create track
+    audio_attributes_t attributes;
+    attributes.usage = AUDIO_USAGE_MEDIA;
+    attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+    auto playback = sp<AudioPlayback>::make(
+            44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+    EXPECT_EQ(NO_ERROR, playback->create());
+    EXPECT_EQ(NO_ERROR, playback->start());
+    EXPECT_FALSE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+                                              playback->getAudioTrackHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+    playback->stop();
+    playback.clear();
+
+    String16 name{gPackageName};
+    audio_unique_id_t id;
+    status_t status = AudioEffect::addStreamDefaultEffect(
+            type, name, nullptr, kDefaultOutputEffectPriority, AUDIO_USAGE_MEDIA, &id);
+    EXPECT_EQ(NO_ERROR, status) << "Adding default effect failed: " << type;
+
+    playback = sp<AudioPlayback>::make(
+            44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+    EXPECT_EQ(NO_ERROR, playback->create());
+    float level = 0.2f, levelGot;
+    playback->getAudioTrackHandle()->setAuxEffectSendLevel(level);
+    EXPECT_EQ(NO_ERROR, playback->start());
+    EXPECT_TRUE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+                                             playback->getAudioTrackHandle()->getSessionId()))
+            << "Effect should have been added. " << type;
+    EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+    playback->getAudioTrackHandle()->getAuxEffectSendLevel(&levelGot);
+    EXPECT_EQ(level, levelGot);
+    playback->stop();
+    playback.clear();
+
+    status = AudioEffect::removeStreamDefaultEffect(id);
+    EXPECT_EQ(NO_ERROR, status);
+    playback = sp<AudioPlayback>::make(
+            44100 /*sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED, &attributes);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(NO_ERROR, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"));
+    EXPECT_EQ(NO_ERROR, playback->create());
+    EXPECT_EQ(NO_ERROR, playback->start());
+    EXPECT_FALSE(isEffectExistsOnAudioSession(selectedEffectType, kDefaultOutputEffectPriority - 1,
+                                              playback->getAudioTrackHandle()->getSessionId()))
+            << "Effect should not have been added. " << type;
+    EXPECT_EQ(NO_ERROR, playback->waitForConsumption());
+    playback->stop();
+    playback.clear();
+}
diff --git a/media/libaudioclient/tests/audiorecord_tests.cpp b/media/libaudioclient/tests/audiorecord_tests.cpp
new file mode 100644
index 0000000..754e6cc
--- /dev/null
+++ b/media/libaudioclient/tests/audiorecord_tests.cpp
@@ -0,0 +1,235 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+#define LOG_TAG "AudioRecordTest"
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+class AudioRecordTest : public ::testing::Test {
+  public:
+    virtual void SetUp() override {
+        mAC = new AudioCapture(AUDIO_SOURCE_DEFAULT, 44100, AUDIO_FORMAT_PCM_16_BIT,
+                               AUDIO_CHANNEL_IN_FRONT);
+        ASSERT_NE(nullptr, mAC);
+        ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAC) ASSERT_EQ(OK, mAC->stop());
+    }
+
+    sp<AudioCapture> mAC;
+};
+
+class AudioRecordCreateTest
+    : public ::testing::TestWithParam<
+              std::tuple<uint32_t, audio_format_t, audio_channel_mask_t, audio_input_flags_t,
+                         audio_session_t, audio_source_t>> {
+  public:
+    AudioRecordCreateTest()
+        : mSampleRate(std::get<0>(GetParam())),
+          mFormat(std::get<1>(GetParam())),
+          mChannelMask(std::get<2>(GetParam())),
+          mFlags(std::get<3>(GetParam())),
+          mSessionId(std::get<4>(GetParam())),
+          mInputSource(std::get<5>(GetParam())){};
+
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_input_flags_t mFlags;
+    const audio_session_t mSessionId;
+    const audio_source_t mInputSource;
+    const AudioRecord::transfer_type mTransferType = AudioRecord::TRANSFER_OBTAIN;
+
+    sp<AudioCapture> mAC;
+
+    virtual void SetUp() override {
+        mAC = new AudioCapture(mInputSource, mSampleRate, mFormat, mChannelMask, mFlags, mSessionId,
+                               mTransferType);
+        ASSERT_NE(nullptr, mAC);
+        ASSERT_EQ(OK, mAC->create()) << "record creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAC) ASSERT_EQ(OK, mAC->stop());
+    }
+};
+
+TEST_F(AudioRecordTest, TestSimpleRecord) {
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestAudioCbNotifier) {
+    EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->addAudioDeviceCallback(nullptr));
+    sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+    sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->addAudioDeviceCallback(cb));
+    EXPECT_EQ(OK, mAC->start()) << "record creation failed";
+    EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+    EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+    EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+    EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+    EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+    EXPECT_EQ(BAD_VALUE, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(nullptr));
+    EXPECT_EQ(INVALID_OPERATION, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->removeAudioDeviceCallback(cb));
+    mAC->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackPause) {
+    const auto playback = sp<AudioPlayback>::make(
+            8000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO);
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+    audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+    EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+            << "record creation failed";
+    EXPECT_EQ(OK, playback->start());
+    RawBuffer buffer;
+    status_t status = mAC->obtainBufferCb(buffer);
+    EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+    playback->getAudioTrackHandle()->pause();
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    playback->stop();
+}
+
+TEST_F(AudioRecordTest, TestEventRecordTrackStop) {
+    const auto playback = sp<AudioPlayback>::make(
+            8000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_MONO);
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_1ch_8kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, playback->create()) << "AudioTrack Creation failed";
+    audio_session_t audioTrackSession = playback->getAudioTrackHandle()->getSessionId();
+    EXPECT_EQ(OK, mAC->start(AudioSystem::SYNC_EVENT_PRESENTATION_COMPLETE, audioTrackSession))
+            << "record creation failed";
+    EXPECT_EQ(OK, playback->start());
+    RawBuffer buffer;
+    status_t status = mAC->obtainBufferCb(buffer);
+    EXPECT_EQ(status, TIMED_OUT) << "Not expecting any callbacks until track sends Sync event";
+    playback->stop();
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarker) {
+    mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition))
+            << "setMarkerPosition() failed";
+    uint32_t marker;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker))
+            << "getMarkerPosition() failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(marker, mAC->mMarkerPosition)
+            << "configured marker and received marker are different";
+    EXPECT_EQ(mAC->mReceivedCbMarkerAtPosition, mAC->mMarkerPosition)
+            << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetSetMarkerPeriodical) {
+    mAC->mMarkerPeriod = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPeriod))
+            << "setPositionUpdatePeriod() failed";
+    uint32_t marker;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker))
+            << "getPositionUpdatePeriod() failed";
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(marker, mAC->mMarkerPeriod) << "configured marker and received marker are different";
+    EXPECT_EQ(mAC->mReceivedCbMarkerCount, mAC->mNumFramesToRecord / mAC->mMarkerPeriod)
+            << "configured marker and received cb marker are different";
+}
+
+TEST_F(AudioRecordTest, TestGetPosition) {
+    uint32_t position;
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+    EXPECT_EQ(0, position);
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+    EXPECT_EQ(OK, mAC->stop());
+    EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPosition(&position)) << "getPosition() failed";
+}
+
+// TODO: Add checkPatchCapture(), verify the information of patch via dumpPort() and dumpPatch()
+TEST_P(AudioRecordCreateTest, TestCreateRecord) {
+    EXPECT_EQ(mFormat, mAC->getAudioRecordHandle()->format());
+    EXPECT_EQ(audio_channel_count_from_in_mask(mChannelMask),
+              mAC->getAudioRecordHandle()->channelCount());
+    if (mAC->mFrameCount != 0)
+        EXPECT_LE(mAC->mFrameCount, mAC->getAudioRecordHandle()->frameCount());
+    EXPECT_EQ(mInputSource, mAC->getAudioRecordHandle()->inputSource());
+    if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAC->getAudioRecordHandle()->getSampleRate());
+    if (mSessionId != AUDIO_SESSION_NONE)
+        EXPECT_EQ(mSessionId, mAC->getAudioRecordHandle()->getSessionId());
+    if (mTransferType != AudioRecord::TRANSFER_CALLBACK) {
+        uint32_t marker;
+        mAC->mMarkerPosition = (mAC->mNotificationFrames << 3) + (mAC->mNotificationFrames >> 1);
+        EXPECT_EQ(INVALID_OPERATION,
+                  mAC->getAudioRecordHandle()->setMarkerPosition(mAC->mMarkerPosition));
+        EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getMarkerPosition(&marker));
+        EXPECT_EQ(INVALID_OPERATION,
+                  mAC->getAudioRecordHandle()->setPositionUpdatePeriod(mAC->mMarkerPosition));
+        EXPECT_EQ(OK, mAC->getAudioRecordHandle()->getPositionUpdatePeriod(&marker));
+    }
+    EXPECT_EQ(OK, mAC->start()) << "start recording failed";
+    EXPECT_EQ(OK, mAC->audioProcess()) << "audioProcess failed";
+}
+
+// for port primary input
+INSTANTIATE_TEST_SUITE_P(AudioRecordPrimaryInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+                                                              24000, 32000, 44100, 48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+                                                              AUDIO_CHANNEL_IN_STEREO,
+                                                              AUDIO_CHANNEL_IN_FRONT_BACK),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// for port fast input
+INSTANTIATE_TEST_SUITE_P(AudioRecordFastInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(8000, 11025, 12000, 16000, 22050,
+                                                              24000, 32000, 44100, 48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_8_24_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO,
+                                                              AUDIO_CHANNEL_IN_STEREO,
+                                                              AUDIO_CHANNEL_IN_FRONT_BACK),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_FAST),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_DEFAULT)));
+
+// misc
+INSTANTIATE_TEST_SUITE_P(AudioRecordMiscInput, AudioRecordCreateTest,
+                         ::testing::Combine(::testing::Values(48000),
+                                            ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+                                            ::testing::Values(AUDIO_CHANNEL_IN_MONO),
+                                            ::testing::Values(AUDIO_INPUT_FLAG_NONE),
+                                            ::testing::Values(AUDIO_SESSION_NONE),
+                                            ::testing::Values(AUDIO_SOURCE_MIC,
+                                                              AUDIO_SOURCE_CAMCORDER,
+                                                              AUDIO_SOURCE_VOICE_RECOGNITION,
+                                                              AUDIO_SOURCE_VOICE_COMMUNICATION,
+                                                              AUDIO_SOURCE_UNPROCESSED)));
diff --git a/media/libaudioclient/tests/audiorouting_tests.cpp b/media/libaudioclient/tests/audiorouting_tests.cpp
new file mode 100644
index 0000000..32ba597
--- /dev/null
+++ b/media/libaudioclient/tests/audiorouting_tests.cpp
@@ -0,0 +1,253 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+
+#include <cutils/properties.h>
+#include <gtest/gtest.h>
+#include <libxml/parser.h>
+#include <libxml/xinclude.h>
+#include <string.h>
+#include <system/audio_config.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+template <class T>
+constexpr void (*xmlDeleter)(T* t);
+template <>
+constexpr auto xmlDeleter<xmlDoc> = xmlFreeDoc;
+template <>
+constexpr auto xmlDeleter<xmlChar> = [](xmlChar* s) { xmlFree(s); };
+
+/** @return a unique_ptr with the correct deleter for the libxml2 object. */
+template <class T>
+constexpr auto make_xmlUnique(T* t) {
+    // Wrap deleter in lambda to enable empty base optimization
+    auto deleter = [](T* t) { xmlDeleter<T>(t); };
+    return std::unique_ptr<T, decltype(deleter)>{t, deleter};
+}
+
+std::string getXmlAttribute(const xmlNode* cur, const char* attribute) {
+    auto charPtr = make_xmlUnique(xmlGetProp(cur, reinterpret_cast<const xmlChar*>(attribute)));
+    if (charPtr == NULL) {
+        return "";
+    }
+    std::string value(reinterpret_cast<const char*>(charPtr.get()));
+    return value;
+}
+
+struct MixPort {
+    std::string name;
+    std::string role;
+    std::string flags;
+};
+
+struct Route {
+    std::string name;
+    std::string sources;
+    std::string sink;
+};
+
+status_t parse_audio_policy_configuration_xml(std::vector<std::string>& attachedDevices,
+                                              std::vector<MixPort>& mixPorts,
+                                              std::vector<Route>& routes) {
+    std::string path = audio_find_readable_configuration_file("audio_policy_configuration.xml");
+    if (path.length() == 0) return UNKNOWN_ERROR;
+    auto doc = make_xmlUnique(xmlParseFile(path.c_str()));
+    if (doc == nullptr) return UNKNOWN_ERROR;
+    xmlNode* root = xmlDocGetRootElement(doc.get());
+    if (root == nullptr) return UNKNOWN_ERROR;
+    if (xmlXIncludeProcess(doc.get()) < 0) return UNKNOWN_ERROR;
+    mixPorts.clear();
+    if (!xmlStrcmp(root->name, reinterpret_cast<const xmlChar*>("audioPolicyConfiguration"))) {
+        std::string raw{getXmlAttribute(root, "version")};
+        for (auto* child = root->xmlChildrenNode; child != nullptr; child = child->next) {
+            if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>("modules"))) {
+                xmlNode* root = child;
+                for (auto* child = root->xmlChildrenNode; child != nullptr; child = child->next) {
+                    if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>("module"))) {
+                        xmlNode* root = child;
+                        for (auto* child = root->xmlChildrenNode; child != nullptr;
+                             child = child->next) {
+                            if (!xmlStrcmp(child->name,
+                                           reinterpret_cast<const xmlChar*>("mixPorts"))) {
+                                xmlNode* root = child;
+                                for (auto* child = root->xmlChildrenNode; child != nullptr;
+                                     child = child->next) {
+                                    if (!xmlStrcmp(child->name,
+                                                   reinterpret_cast<const xmlChar*>("mixPort"))) {
+                                        MixPort mixPort;
+                                        xmlNode* root = child;
+                                        mixPort.name = getXmlAttribute(root, "name");
+                                        mixPort.role = getXmlAttribute(root, "role");
+                                        mixPort.flags = getXmlAttribute(root, "flags");
+                                        if (mixPort.role == "source") mixPorts.push_back(mixPort);
+                                    }
+                                }
+                            } else if (!xmlStrcmp(child->name, reinterpret_cast<const xmlChar*>(
+                                                                       "attachedDevices"))) {
+                                xmlNode* root = child;
+                                for (auto* child = root->xmlChildrenNode; child != nullptr;
+                                     child = child->next) {
+                                    if (!xmlStrcmp(child->name,
+                                                   reinterpret_cast<const xmlChar*>("item"))) {
+                                        auto xmlValue = make_xmlUnique(xmlNodeListGetString(
+                                                child->doc, child->xmlChildrenNode, 1));
+                                        if (xmlValue == nullptr) {
+                                            raw = "";
+                                        } else {
+                                            raw = reinterpret_cast<const char*>(xmlValue.get());
+                                        }
+                                        std::string& value = raw;
+                                        attachedDevices.push_back(std::move(value));
+                                    }
+                                }
+                            } else if (!xmlStrcmp(child->name,
+                                                  reinterpret_cast<const xmlChar*>("routes"))) {
+                                xmlNode* root = child;
+                                for (auto* child = root->xmlChildrenNode; child != nullptr;
+                                     child = child->next) {
+                                    if (!xmlStrcmp(child->name,
+                                                   reinterpret_cast<const xmlChar*>("route"))) {
+                                        Route route;
+                                        xmlNode* root = child;
+                                        route.name = getXmlAttribute(root, "name");
+                                        route.sources = getXmlAttribute(root, "sources");
+                                        route.sink = getXmlAttribute(root, "sink");
+                                        routes.push_back(route);
+                                    }
+                                }
+                            }
+                        }
+                    }
+                }
+            }
+        }
+    }
+    return OK;
+}
+
+// UNIT TEST
+TEST(AudioTrackTest, TestPerformanceMode) {
+    std::vector<std::string> attachedDevices;
+    std::vector<MixPort> mixPorts;
+    std::vector<Route> routes;
+    EXPECT_EQ(OK, parse_audio_policy_configuration_xml(attachedDevices, mixPorts, routes));
+    std::string output_flags_string[] = {"AUDIO_OUTPUT_FLAG_FAST", "AUDIO_OUTPUT_FLAG_DEEP_BUFFER"};
+    audio_output_flags_t output_flags[] = {AUDIO_OUTPUT_FLAG_FAST, AUDIO_OUTPUT_FLAG_DEEP_BUFFER};
+    audio_flags_mask_t flags[] = {AUDIO_FLAG_LOW_LATENCY, AUDIO_FLAG_DEEP_BUFFER};
+    bool hasFlag = false;
+    for (int i = 0; i < sizeof(flags) / sizeof(flags[0]); i++) {
+        hasFlag = false;
+        for (int j = 0; j < mixPorts.size() && !hasFlag; j++) {
+            MixPort port = mixPorts[j];
+            if (port.role == "source" && port.flags.find(output_flags_string[i]) != -1) {
+                for (int k = 0; k < routes.size() && !hasFlag; k++) {
+                    if (routes[k].sources.find(port.name) != -1 &&
+                        std::find(attachedDevices.begin(), attachedDevices.end(), routes[k].sink) !=
+                                attachedDevices.end()) {
+                        hasFlag = true;
+                        std::cerr << "found port with flag " << output_flags_string[i] << "@ "
+                                  << " port :: name : " << port.name << " role : " << port.role
+                                  << " port :: flags : " << port.flags
+                                  << " connected via route name : " << routes[k].name
+                                  << " route sources : " << routes[k].sources
+                                  << " route sink : " << routes[k].sink << std::endl;
+                    }
+                }
+            }
+        }
+        if (!hasFlag) continue;
+        audio_attributes_t attributes = AUDIO_ATTRIBUTES_INITIALIZER;
+        attributes.usage = AUDIO_USAGE_MEDIA;
+        attributes.content_type = AUDIO_CONTENT_TYPE_MUSIC;
+        attributes.flags = flags[i];
+        sp<AudioPlayback> ap = sp<AudioPlayback>::make(
+                0 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+                AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE, AudioTrack::TRANSFER_OBTAIN,
+                &attributes);
+        ASSERT_NE(nullptr, ap);
+        ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+                << "Unable to open Resource";
+        EXPECT_EQ(OK, ap->create()) << "track creation failed";
+        sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+        EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cb));
+        EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+        EXPECT_EQ(OK, ap->onProcess());
+        EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+        EXPECT_TRUE(checkPatchPlayback(cb->mAudioIo, cb->mDeviceId));
+        EXPECT_NE(0, ap->getAudioTrackHandle()->getFlags() & output_flags[i]);
+        audio_patch patch;
+        EXPECT_EQ(OK, getPatchForOutputMix(cb->mAudioIo, patch));
+        for (auto j = 0; j < patch.num_sources; j++) {
+            if (patch.sources[j].type == AUDIO_PORT_TYPE_MIX &&
+                patch.sources[j].ext.mix.handle == cb->mAudioIo) {
+                if ((patch.sources[j].flags.output & output_flags[i]) == 0) {
+                    ADD_FAILURE() << "expected output flag " << output_flags[i] << " is absent";
+                    std::cerr << dumpPortConfig(patch.sources[j]);
+                }
+            }
+        }
+        ap->stop();
+        ap->getAudioTrackHandle()->removeAudioDeviceCallback(cb);
+    }
+}
+
+TEST(AudioTrackTest, TestRemoteSubmix) {
+    std::vector<std::string> attachedDevices;
+    std::vector<MixPort> mixPorts;
+    std::vector<Route> routes;
+    EXPECT_EQ(OK, parse_audio_policy_configuration_xml(attachedDevices, mixPorts, routes));
+    bool hasFlag = false;
+    for (int j = 0; j < attachedDevices.size() && !hasFlag; j++) {
+        if (attachedDevices[j].find("Remote Submix") != -1) hasFlag = true;
+    }
+    if (!hasFlag) GTEST_SKIP() << " Device does not have Remote Submix port.";
+    sp<AudioCapture> capture = new AudioCapture(AUDIO_SOURCE_REMOTE_SUBMIX, 48000,
+                                                AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_IN_STEREO);
+    ASSERT_NE(nullptr, capture);
+    ASSERT_EQ(OK, capture->create()) << "record creation failed";
+
+    sp<AudioPlayback> playback = sp<AudioPlayback>::make(
+            48000 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO,
+            AUDIO_OUTPUT_FLAG_NONE, AUDIO_SESSION_NONE);
+    ASSERT_NE(nullptr, playback);
+    ASSERT_EQ(OK, playback->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    ASSERT_EQ(OK, playback->create()) << "track creation failed";
+
+    audio_port_v7 port;
+    status_t status = getPortByAttributes(AUDIO_PORT_ROLE_SOURCE, AUDIO_PORT_TYPE_DEVICE,
+                                          AUDIO_DEVICE_IN_REMOTE_SUBMIX, port);
+    EXPECT_EQ(OK, status) << "Could not find port";
+
+    EXPECT_EQ(OK, capture->start()) << "start recording failed";
+    EXPECT_EQ(port.id, capture->getAudioRecordHandle()->getRoutedDeviceId())
+            << "Capture NOT routed on expected port";
+
+    status = getPortByAttributes(AUDIO_PORT_ROLE_SINK, AUDIO_PORT_TYPE_DEVICE,
+                                 AUDIO_DEVICE_OUT_REMOTE_SUBMIX, port);
+    EXPECT_EQ(OK, status) << "Could not find port";
+
+    EXPECT_EQ(OK, playback->start()) << "audio track start failed";
+    EXPECT_EQ(OK, playback->onProcess());
+    ASSERT_EQ(port.id, playback->getAudioTrackHandle()->getRoutedDeviceId())
+            << "Playback NOT routed on expected port";
+    capture->stop();
+    playback->stop();
+}
diff --git a/media/libaudioclient/tests/audiotrack_tests.cpp b/media/libaudioclient/tests/audiotrack_tests.cpp
new file mode 100644
index 0000000..a49af96
--- /dev/null
+++ b/media/libaudioclient/tests/audiotrack_tests.cpp
@@ -0,0 +1,132 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+//#define LOG_NDEBUG 0
+
+#include <gtest/gtest.h>
+
+#include "audio_test_utils.h"
+
+using namespace android;
+
+TEST(AudioTrackTest, TestPlayTrack) {
+    const auto ap = sp<AudioPlayback>::make(44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT,
+                                            AUDIO_CHANNEL_OUT_STEREO, AUDIO_OUTPUT_FLAG_NONE,
+                                            AUDIO_SESSION_NONE, AudioTrack::TRANSFER_OBTAIN);
+    ASSERT_NE(nullptr, ap);
+    ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+    EXPECT_EQ(OK, ap->onProcess());
+    ap->stop();
+}
+
+TEST(AudioTrackTest, TestSeek) {
+    const auto ap = sp<AudioPlayback>::make(
+            44100 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT, AUDIO_CHANNEL_OUT_STEREO);
+    ASSERT_NE(nullptr, ap);
+    ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+    EXPECT_EQ(OK, ap->onProcess(true));
+    ap->stop();
+}
+
+TEST(AudioTrackTest, TestAudioCbNotifier) {
+    const auto ap = sp<AudioPlayback>::make(0 /* sampleRate */, AUDIO_FORMAT_PCM_16_BIT,
+                                            AUDIO_CHANNEL_OUT_STEREO, AUDIO_OUTPUT_FLAG_FAST,
+                                            AUDIO_SESSION_NONE, AudioTrack::TRANSFER_SHARED);
+    ASSERT_NE(nullptr, ap);
+    ASSERT_EQ(OK, ap->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+            << "Unable to open Resource";
+    EXPECT_EQ(OK, ap->create()) << "track creation failed";
+    EXPECT_EQ(BAD_VALUE, ap->getAudioTrackHandle()->addAudioDeviceCallback(nullptr));
+    sp<OnAudioDeviceUpdateNotifier> cb = new OnAudioDeviceUpdateNotifier();
+    sp<OnAudioDeviceUpdateNotifier> cbOld = new OnAudioDeviceUpdateNotifier();
+    EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(INVALID_OPERATION, ap->getAudioTrackHandle()->addAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, ap->getAudioTrackHandle()->addAudioDeviceCallback(cb));
+    EXPECT_EQ(OK, ap->start()) << "audio track start failed";
+    EXPECT_EQ(OK, ap->onProcess());
+    EXPECT_EQ(OK, cb->waitForAudioDeviceCb());
+    EXPECT_EQ(AUDIO_IO_HANDLE_NONE, cbOld->mAudioIo);
+    EXPECT_EQ(AUDIO_PORT_HANDLE_NONE, cbOld->mDeviceId);
+    EXPECT_NE(AUDIO_IO_HANDLE_NONE, cb->mAudioIo);
+    EXPECT_NE(AUDIO_PORT_HANDLE_NONE, cb->mDeviceId);
+    EXPECT_TRUE(checkPatchPlayback(cb->mAudioIo, cb->mDeviceId));
+    EXPECT_EQ(BAD_VALUE, ap->getAudioTrackHandle()->removeAudioDeviceCallback(nullptr));
+    EXPECT_EQ(INVALID_OPERATION, ap->getAudioTrackHandle()->removeAudioDeviceCallback(cbOld));
+    EXPECT_EQ(OK, ap->getAudioTrackHandle()->removeAudioDeviceCallback(cb));
+    ap->stop();
+}
+
+class AudioTrackCreateTest
+    : public ::testing::TestWithParam<std::tuple<uint32_t, audio_format_t, audio_channel_mask_t,
+                                                 audio_output_flags_t, audio_session_t>> {
+  public:
+    AudioTrackCreateTest()
+        : mSampleRate(std::get<0>(GetParam())),
+          mFormat(std::get<1>(GetParam())),
+          mChannelMask(std::get<2>(GetParam())),
+          mFlags(std::get<3>(GetParam())),
+          mSessionId(std::get<4>(GetParam())){};
+
+    const uint32_t mSampleRate;
+    const audio_format_t mFormat;
+    const audio_channel_mask_t mChannelMask;
+    const audio_output_flags_t mFlags;
+    const audio_session_t mSessionId;
+
+    sp<AudioPlayback> mAP;
+
+    virtual void SetUp() override {
+        mAP = sp<AudioPlayback>::make(mSampleRate, mFormat, mChannelMask, mFlags,
+                                              mSessionId);
+        ASSERT_NE(nullptr, mAP);
+        ASSERT_EQ(OK, mAP->loadResource("/data/local/tmp/bbb_2ch_24kHz_s16le.raw"))
+                << "Unable to open Resource";
+        ASSERT_EQ(OK, mAP->create()) << "track creation failed";
+    }
+
+    virtual void TearDown() override {
+        if (mAP) mAP->stop();
+    }
+};
+
+TEST_P(AudioTrackCreateTest, TestCreateTrack) {
+    EXPECT_EQ(mFormat, mAP->getAudioTrackHandle()->format());
+    EXPECT_EQ(audio_channel_count_from_out_mask(mChannelMask),
+              mAP->getAudioTrackHandle()->channelCount());
+    if (mSampleRate != 0) EXPECT_EQ(mSampleRate, mAP->getAudioTrackHandle()->getSampleRate());
+    if (mSessionId != AUDIO_SESSION_NONE)
+        EXPECT_EQ(mSessionId, mAP->getAudioTrackHandle()->getSessionId());
+    EXPECT_EQ(mSampleRate, mAP->getAudioTrackHandle()->getOriginalSampleRate());
+    EXPECT_EQ(OK, mAP->start()) << "audio track start failed";
+    EXPECT_EQ(OK, mAP->onProcess());
+}
+
+// sampleRate, format, channelMask, flags, sessionId
+INSTANTIATE_TEST_SUITE_P(
+        AudioTrackParameterizedTest, AudioTrackCreateTest,
+        ::testing::Combine(::testing::Values(48000), ::testing::Values(AUDIO_FORMAT_PCM_16_BIT),
+                           ::testing::Values(AUDIO_CHANNEL_OUT_STEREO),
+                           ::testing::Values(AUDIO_OUTPUT_FLAG_NONE,
+                                             AUDIO_OUTPUT_FLAG_PRIMARY | AUDIO_OUTPUT_FLAG_FAST,
+                                             AUDIO_OUTPUT_FLAG_RAW | AUDIO_OUTPUT_FLAG_FAST,
+                                             AUDIO_OUTPUT_FLAG_DEEP_BUFFER),
+                           ::testing::Values(AUDIO_SESSION_NONE)));
diff --git a/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
new file mode 100644
index 0000000..2d1e4bf
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_1ch_8kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
new file mode 100644
index 0000000..c8ac5f7
--- /dev/null
+++ b/media/libaudioclient/tests/bbb_2ch_24kHz_s16le.raw
Binary files differ
diff --git a/media/libaudioclient/tests/test_create_audiorecord.cpp b/media/libaudioclient/tests/test_create_audiorecord.cpp
index 2e0883b..277110b 100644
--- a/media/libaudioclient/tests/test_create_audiorecord.cpp
+++ b/media/libaudioclient/tests/test_create_audiorecord.cpp
@@ -29,14 +29,13 @@
 
 #define NUM_ARGUMENTS 8
 #define VERSION_VALUE "1.0"
-#define PACKAGE_NAME  "AudioRecord test"
+#define PACKAGE_NAME "AudioRecord test"
 
 namespace android {
 
 using android::content::AttributionSourceState;
 
-int testRecord(FILE *inputFile, int outputFileFd)
-{
+int testRecord(FILE* inputFile, int outputFileFd) {
     char line[MAX_INPUT_FILE_LINE_LENGTH];
     uint32_t testCount = 0;
     Vector<String16> args;
@@ -47,11 +46,9 @@
     attributionSource.token = sp<BBinder>::make();
 
     if (inputFile == nullptr) {
-        sp<AudioRecord> record = new AudioRecord(AUDIO_SOURCE_DEFAULT,
-                                              0 /* sampleRate */,
-                                              AUDIO_FORMAT_DEFAULT,
-                                              AUDIO_CHANNEL_IN_MONO,
-                                              attributionSource);
+        sp<AudioRecord> record =
+                new AudioRecord(AUDIO_SOURCE_DEFAULT, 0 /* sampleRate */, AUDIO_FORMAT_DEFAULT,
+                                AUDIO_CHANNEL_IN_MONO, attributionSource);
         if (record == 0 || record->initCheck() != NO_ERROR) {
             write(outputFileFd, "Error creating AudioRecord\n",
                   sizeof("Error creating AudioRecord\n"));
@@ -80,11 +77,10 @@
         char statusStr[MAX_OUTPUT_FILE_LINE_LENGTH];
         bool fast = false;
 
-        if (sscanf(line, " %u %x %x %zu %d %x %u %u",
-                   &sampleRate, &format, &channelMask,
-                   &frameCount, &notificationFrames,
-                   &flags, &sessionId, &inputSource) != NUM_ARGUMENTS) {
-            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+        if (sscanf(line, " %u %x %x %zu %d %x %u %u", &sampleRate, &format, &channelMask,
+                   &frameCount, &notificationFrames, &flags, &sessionId,
+                   &inputSource) != NUM_ARGUMENTS) {
+            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount + 1);
             ret = 1;
             continue;
         }
@@ -100,21 +96,10 @@
         sp<AudioRecord> record = new AudioRecord(attributionSource);
         const auto emptyCallback = sp<AudioRecord::IAudioRecordCallback>::make();
 
-        record->set(AUDIO_SOURCE_DEFAULT,
-                   sampleRate,
-                   format,
-                   channelMask,
-                   frameCount,
-                   fast ? emptyCallback : nullptr,
-                   notificationFrames,
-                   false,
-                   sessionId,
-                   fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT,
-                   flags,
-                   getuid(),
-                   getpid(),
-                   &attributes,
-                   AUDIO_PORT_HANDLE_NONE);
+        record->set(AUDIO_SOURCE_DEFAULT, sampleRate, format, channelMask, frameCount,
+                    fast ? emptyCallback : nullptr, notificationFrames, false, sessionId,
+                    fast ? AudioRecord::TRANSFER_CALLBACK : AudioRecord::TRANSFER_DEFAULT, flags,
+                    getuid(), getpid(), &attributes, AUDIO_PORT_HANDLE_NONE);
         status = record->initCheck();
         sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
         write(outputFileFd, statusStr, strlen(statusStr));
@@ -126,11 +111,8 @@
     return ret;
 }
 
-}; // namespace android
+};  // namespace android
 
-
-int main(int argc, char **argv)
-{
+int main(int argc, char** argv) {
     return android::main(argc, argv, android::testRecord);
 }
-
diff --git a/media/libaudioclient/tests/test_create_audiotrack.cpp b/media/libaudioclient/tests/test_create_audiotrack.cpp
index e7231d3..4e09e21 100644
--- a/media/libaudioclient/tests/test_create_audiotrack.cpp
+++ b/media/libaudioclient/tests/test_create_audiotrack.cpp
@@ -32,18 +32,15 @@
 
 namespace android {
 
-int testTrack(FILE *inputFile, int outputFileFd)
-{
+int testTrack(FILE* inputFile, int outputFileFd) {
     char line[MAX_INPUT_FILE_LINE_LENGTH];
     uint32_t testCount = 0;
     Vector<String16> args;
     int ret = 0;
 
     if (inputFile == nullptr) {
-        sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT,
-                                              0 /* sampleRate */,
-                                              AUDIO_FORMAT_DEFAULT,
-                                              AUDIO_CHANNEL_OUT_STEREO);
+        sp<AudioTrack> track = new AudioTrack(AUDIO_STREAM_DEFAULT, 0 /* sampleRate */,
+                                              AUDIO_FORMAT_DEFAULT, AUDIO_CHANNEL_OUT_STEREO);
         if (track == 0 || track->initCheck() != NO_ERROR) {
             write(outputFileFd, "Error creating AudioTrack\n",
                   sizeof("Error creating AudioTrack\n"));
@@ -78,11 +75,10 @@
         bool offload = false;
         bool fast = false;
 
-        if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u",
-                   &sampleRate, &format, &channelMask,
-                   &frameCount, &notificationFrames, &useSharedBuffer,
-                   &flags, &sessionId, &usage, &contentType) != NUM_ARGUMENTS) {
-            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount+1);
+        if (sscanf(line, " %u %x %x %zu %d %u %x %u %u %u", &sampleRate, &format, &channelMask,
+                   &frameCount, &notificationFrames, &useSharedBuffer, &flags, &sessionId, &usage,
+                   &contentType) != NUM_ARGUMENTS) {
+            fprintf(stderr, "Malformed line for test #%u in input file\n", testCount + 1);
             ret = 1;
             continue;
         }
@@ -90,7 +86,7 @@
 
         if (useSharedBuffer != 0) {
             size_t heapSize = audio_channel_count_from_out_mask(channelMask) *
-                    audio_bytes_per_sample(format) * frameCount;
+                              audio_bytes_per_sample(format) * frameCount;
             heap = new MemoryDealer(heapSize, "AudioTrack Heap Base");
             sharedBuffer = heap->allocate(heapSize);
             frameCount = 0;
@@ -111,25 +107,13 @@
         attributes.usage = usage;
         sp<AudioTrack> track = new AudioTrack();
         const auto emptyCallback = sp<AudioTrack::IAudioTrackCallback>::make();
-        track->set(AUDIO_STREAM_DEFAULT,
-                   sampleRate,
-                   format,
-                   channelMask,
-                   frameCount,
-                   flags,
-                   (fast || offload) ? emptyCallback : nullptr,
-                   notificationFrames,
-                   sharedBuffer,
-                   false,
-                   sessionId,
-                   ((fast && sharedBuffer == 0) || offload) ?
-                           AudioTrack::TRANSFER_CALLBACK : AudioTrack::TRANSFER_DEFAULT,
-                   offload ? &offloadInfo : nullptr,
-                   AttributionSourceState(),
-                   &attributes,
-                   false,
-                   1.0f,
-                   AUDIO_PORT_HANDLE_NONE);
+        track->set(AUDIO_STREAM_DEFAULT, sampleRate, format, channelMask, frameCount, flags,
+                   (fast || offload) ? emptyCallback : nullptr, notificationFrames, sharedBuffer,
+                   false, sessionId,
+                   ((fast && sharedBuffer == 0) || offload) ? AudioTrack::TRANSFER_CALLBACK
+                                                            : AudioTrack::TRANSFER_DEFAULT,
+                   offload ? &offloadInfo : nullptr, AttributionSourceState(), &attributes, false,
+                   1.0f, AUDIO_PORT_HANDLE_NONE);
         status = track->initCheck();
         sprintf(statusStr, "\n#### Test %u status %d\n", testCount, status);
         write(outputFileFd, statusStr, strlen(statusStr));
@@ -141,11 +125,8 @@
     return ret;
 }
 
-}; // namespace android
+};  // namespace android
 
-
-int main(int argc, char **argv)
-{
+int main(int argc, char** argv) {
     return android::main(argc, argv, android::testTrack);
 }
-
diff --git a/media/libaudioclient/tests/test_create_utils.cpp b/media/libaudioclient/tests/test_create_utils.cpp
index caf5227..c2c2e8b 100644
--- a/media/libaudioclient/tests/test_create_utils.cpp
+++ b/media/libaudioclient/tests/test_create_utils.cpp
@@ -23,10 +23,10 @@
 
 namespace android {
 
-int readLine(FILE *inputFile, char *line, int size) {
+int readLine(FILE* inputFile, char* line, int size) {
     int ret = 0;
     while (true) {
-        char *str = fgets(line, size, inputFile);
+        char* str = fgets(line, size, inputFile);
         if (str == nullptr) {
             ret = -1;
             break;
@@ -42,8 +42,7 @@
     return ret;
 }
 
-bool checkVersion(FILE *inputFile, const char *version)
-{
+bool checkVersion(FILE* inputFile, const char* version) {
     char line[MAX_INPUT_FILE_LINE_LENGTH];
     char versionKey[MAX_INPUT_FILE_LINE_LENGTH];
     char versionValue[MAX_INPUT_FILE_LINE_LENGTH];
@@ -68,9 +67,8 @@
     return true;
 }
 
-int main(int argc, char **argv, test_func_t testFunc)
-{
-    FILE *inputFile = nullptr;
+int main(int argc, char** argv, test_func_t testFunc) {
+    FILE* inputFile = nullptr;
     int outputFileFd = STDOUT_FILENO;
     mode_t mode = S_IRUSR | S_IWUSR | S_IRGRP | S_IROTH;
     int ret = 0;
@@ -96,7 +94,7 @@
         if (strcmp(*argv, "-o") == 0) {
             argv++;
             if (*argv) {
-                outputFileFd = open(*argv, O_WRONLY|O_CREAT, mode);
+                outputFileFd = open(*argv, O_WRONLY | O_CREAT, mode);
                 if (outputFileFd < 0) {
                     ret = 1;
                 }
@@ -126,5 +124,4 @@
     return ret;
 }
 
-}; // namespace android
-
+};  // namespace android
diff --git a/media/libaudioclient/tests/test_create_utils.h b/media/libaudioclient/tests/test_create_utils.h
index 9a6f9fa..110baf7 100644
--- a/media/libaudioclient/tests/test_create_utils.h
+++ b/media/libaudioclient/tests/test_create_utils.h
@@ -27,13 +27,12 @@
 
 namespace android {
 
-int readLine(FILE *inputFile, char *line, int size);
+int readLine(FILE* inputFile, char* line, int size);
 
-bool checkVersion(FILE *inputFile, const char *version);
+bool checkVersion(FILE* inputFile, const char* version);
 
+typedef int (*test_func_t)(FILE* inputFile, int outputFileFd);
 
-typedef int (*test_func_t)(FILE *inputFile, int outputFileFd);
+int main(int argc, char** argv, test_func_t testFunc);
 
-int main(int argc, char **argv, test_func_t testFunc);
-
-}; // namespace android
+};  // namespace android
diff --git a/media/libaudioclient/tests/trackplayerbase_tests.cpp b/media/libaudioclient/tests/trackplayerbase_tests.cpp
new file mode 100644
index 0000000..c9b704d
--- /dev/null
+++ b/media/libaudioclient/tests/trackplayerbase_tests.cpp
@@ -0,0 +1,161 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "TrackPlayerBaseTest"
+
+#include <gtest/gtest.h>
+
+#include <media/TrackPlayerBase.h>
+
+using namespace android;
+using namespace android::media;
+
+class TrackPlayer : public TrackPlayerBase, public AudioTrack::IAudioTrackCallback {
+  public:
+    // methods protected in base class
+    using TrackPlayerBase::playerPause;
+    using TrackPlayerBase::playerSetVolume;
+    using TrackPlayerBase::playerStart;
+    using TrackPlayerBase::playerStop;
+};
+
+class TrackPlayerBaseTest
+    : public ::testing::TestWithParam<std::tuple<double, double, uint32_t, uint32_t>> {
+  public:
+    TrackPlayerBaseTest()
+        : mDuration(std::get<0>(GetParam())),
+          mPulseFreq(std::get<1>(GetParam())),
+          mChannelCount(std::get<2>(GetParam())),
+          mSampleRate(std::get<3>(GetParam())){};
+
+    virtual void SetUp() override {
+        mFrameCount = mDuration * mSampleRate;
+        audio_channel_mask_t channelMask = audio_channel_out_mask_from_count(mChannelCount);
+        sp<AudioTrack> track = new AudioTrack(mStreamType, mSampleRate, mFormat, channelMask,
+                                              mFrameCount, mFlags, nullptr /* callback */,
+                                              0 /* notificationFrames */, AUDIO_SESSION_NONE);
+        ASSERT_EQ(track->initCheck(), NO_ERROR);
+
+        mPlayer = new TrackPlayer();
+        mPlayer->init(track.get(), mPlayer, PLAYER_TYPE_AAUDIO, AUDIO_USAGE_MEDIA,
+                      AUDIO_SESSION_NONE);
+        sp<AudioTrack> playerTrack = mPlayer->mAudioTrack;
+        ASSERT_EQ(playerTrack->initCheck(), NO_ERROR);
+
+        mBufferSize = mFrameCount * playerTrack->frameSize();
+        mBuffer.resize(mBufferSize, 0);
+
+        // populate buffer
+        ASSERT_NE(mPulseFreq, 0);
+        int32_t nPulseSamples = mSampleRate / mPulseFreq;
+        int32_t pulseSize = nPulseSamples * playerTrack->frameSize();
+
+        int32_t marker = 0;
+        while (marker + pulseSize <= mBufferSize) {
+            memset(mBuffer.data() + marker, 127, pulseSize / 2);
+            marker += pulseSize;
+        }
+    }
+
+    void playBuffer() {
+        bool blocking = true;
+        ssize_t nbytes = mPlayer->mAudioTrack->write(mBuffer.data(), mBufferSize, blocking);
+        EXPECT_EQ(nbytes, mBufferSize) << "Did not write all data in blocking mode";
+    }
+
+    const double mDuration;  // seconds
+    sp<TrackPlayer> mPlayer;
+
+  private:
+    const double mPulseFreq;
+    const uint32_t mChannelCount;
+    const uint32_t mSampleRate;
+
+    const audio_format_t mFormat = AUDIO_FORMAT_PCM_16_BIT;
+    const audio_output_flags_t mFlags = AUDIO_OUTPUT_FLAG_NONE;
+    const audio_stream_type_t mStreamType = AUDIO_STREAM_MUSIC;
+
+    int32_t mBufferSize;
+    int32_t mFrameCount;
+    std::vector<uint8_t> mBuffer;
+};
+
+class PlaybackTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(PlaybackTestParam, PlaybackTest) {
+    // no-op implementation
+    EXPECT_TRUE(mPlayer->setStartDelayMs(0).isOk());
+
+    ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+    EXPECT_EQ(mPlayer->playerStop(), NO_ERROR);
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, PlaybackTestParam,
+                         ::testing::Values(std::make_tuple(2.5, 25.0, 2, 48000)));
+
+class ChangeVolumeTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(ChangeVolumeTestParam, ChangeVolumeTest) {
+    float volume = 1.0f;
+    (void)mPlayer->setPlayerVolume(volume / 2, volume);
+
+    ASSERT_TRUE(mPlayer->start().isOk());
+    ASSERT_EQ(mPlayer->playerSetVolume(), NO_ERROR);
+
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+    EXPECT_TRUE(mPlayer->stop().isOk());
+
+    std::vector<float> setVol = {0.95f, 0.05f, 0.5f, 0.25f, -1.0f, 1.0f, 1.0f};
+    std::vector<float> setPan = {0.0f, 0.0f, 1.0f, -1.0f, -1.0f, 0.5f, -0.5f};
+
+    ASSERT_TRUE(mPlayer->start().isOk());
+
+    for (int32_t i = 0; i < setVol.size(); i++) {
+        EXPECT_TRUE(mPlayer->setVolume(setVol[i]).isOk());
+        EXPECT_TRUE(mPlayer->setPan(setPan[i]).isOk());
+        ASSERT_NO_FATAL_FAILURE(playBuffer());
+    }
+    EXPECT_TRUE(mPlayer->stop().isOk());
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, ChangeVolumeTestParam,
+                         ::testing::Values(std::make_tuple(1.0, 100.0, 1, 24000)));
+
+class PauseTestParam : public TrackPlayerBaseTest {};
+
+TEST_P(PauseTestParam, PauseTest) {
+    ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+    ASSERT_EQ(mPlayer->playerPause(), NO_ERROR);
+    ASSERT_EQ(mPlayer->playerStart(), NO_ERROR);
+
+    ASSERT_NO_FATAL_FAILURE(playBuffer());
+
+    EXPECT_EQ(mPlayer->playerStop(), NO_ERROR);
+
+    for (int32_t i = 0; i < 5; i++) {
+        ASSERT_TRUE(mPlayer->start().isOk());
+        ASSERT_NO_FATAL_FAILURE(playBuffer());
+        ASSERT_TRUE(mPlayer->pause().isOk());
+    }
+    EXPECT_TRUE(mPlayer->stop().isOk());
+}
+
+INSTANTIATE_TEST_SUITE_P(TrackPlayerTest, PauseTestParam,
+                         ::testing::Values(std::make_tuple(1.0, 75.0, 2, 24000)));
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
index 5ffbffc..91efb96 100644
--- a/media/libaudiofoundation/DeviceDescriptorBase.cpp
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -126,7 +126,14 @@
                     "%*sEncapsulation modes: %u, metadata types: %u\n", spaces, "",
                     mEncapsulationModes, mEncapsulationMetadataTypes));
 
-    AudioPort::dump(dst, spaces, nullptr, verbose);
+    std::string portStr;
+    AudioPort::dump(&portStr, spaces, nullptr, verbose);
+    if (!portStr.empty()) {
+        if (!mName.empty()) {
+            dst->append(base::StringPrintf("%*s", spaces, ""));
+        }
+        dst->append(portStr);
+    }
 }
 
 std::string DeviceDescriptorBase::toString(bool includeSensitiveInfo) const
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
index 3137e13..d8cf20e 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
@@ -22,6 +22,7 @@
 
 #include <algorithm>
 #include <memory>
+#include <sstream>
 #include <string>
 #include <utility>
 
@@ -94,6 +95,33 @@
     return defaultValue;
 }
 
+std::string hapticParamToString(const struct HapticGeneratorParam& param) {
+    std::stringstream ss;
+    ss << "\t\tHapticGenerator Parameters:\n";
+    ss << "\t\t- resonant frequency: " << param.resonantFrequency << '\n';
+    ss << "\t\t- bpf Q: " << param.bpfQ << '\n';
+    ss << "\t\t- slow env normalization power: " << param.slowEnvNormalizationPower << '\n';
+    ss << "\t\t- bsf zero Q: " << param.bsfZeroQ << '\n';
+    ss << "\t\t- bsf pole Q: " << param.bsfPoleQ << '\n';
+    ss << "\t\t- distortion corner frequency: " << param.distortionCornerFrequency << '\n';
+    ss << "\t\t- distortion input gain: " << param.distortionInputGain << '\n';
+    ss << "\t\t- distortion cube threshold: " << param.distortionCubeThreshold << '\n';
+    ss << "\t\t- distortion output gain: " << param.distortionOutputGain << '\n';
+    return ss.str();
+}
+
+std::string hapticSettingToString(const struct HapticGeneratorParam& param) {
+    std::stringstream ss;
+    ss << "\t\tHaptic setting:\n";
+    ss << "\t\t- tracks intensity map:\n";
+    for (const auto&[id, intensity] : param.id2Intensity) {
+        ss << "\t\t\t- id=" << id << ", intensity=" << (int) intensity;
+    }
+    ss << "\t\t- max intensity: " << (int) param.maxHapticIntensity << '\n';
+    ss << "\t\t- max haptic amplitude: " << param.maxHapticAmplitude << '\n';
+    return ss.str();
+}
+
 int HapticGenerator_Init(struct HapticGeneratorContext *context) {
     context->itfe = &gHapticGeneratorInterface;
 
@@ -129,7 +157,7 @@
     context->param.distortionCubeThreshold = 0.1f;
     context->param.distortionOutputGain = getFloatProperty(
             "vendor.audio.hapticgenerator.distortion.output.gain", DEFAULT_DISTORTION_OUTPUT_GAIN);
-    ALOGD("Using distortion output gain as %f", context->param.distortionOutputGain);
+    ALOGD("%s\n%s", __func__, hapticParamToString(context->param).c_str());
 
     context->state = HAPTICGENERATOR_STATE_INITIALIZED;
     return 0;
@@ -289,6 +317,7 @@
         }
         int id = *(int *) value;
         os::HapticScale hapticIntensity = static_cast<os::HapticScale>(*((int *) value + 1));
+        ALOGD("Setting haptic intensity as %d", hapticIntensity);
         if (hapticIntensity == os::HapticScale::MUTE) {
             context->param.id2Intensity.erase(id);
         } else {
@@ -313,6 +342,10 @@
         context->param.bsfZeroQ = isnan(qFactor) ? DEFAULT_BSF_POLE_Q : qFactor;
         context->param.bsfPoleQ = context->param.bsfZeroQ / 2.0f;
         context->param.maxHapticAmplitude = maxAmplitude;
+        ALOGD("Updating vibrator info, resonantFrequency=%f, bsfZeroQ=%f, bsfPoleQ=%f, "
+              "maxHapticAmplitude=%f",
+              context->param.resonantFrequency, context->param.bsfZeroQ, context->param.bsfPoleQ,
+              context->param.maxHapticAmplitude);
 
         if (context->processorsRecord.bpf != nullptr) {
             context->processorsRecord.bpf->setCoefficients(
@@ -358,6 +391,11 @@
     return in;
 }
 
+void HapticGenerator_Dump(int32_t fd, const struct HapticGeneratorParam& param) {
+    dprintf(fd, "%s", hapticParamToString(param).c_str());
+    dprintf(fd, "%s", hapticSettingToString(param).c_str());
+}
+
 } // namespace (anonymous)
 
 //-----------------------------------------------------------------------------
@@ -562,6 +600,10 @@
         case EFFECT_CMD_SET_AUDIO_MODE:
             break;
 
+        case EFFECT_CMD_DUMP:
+            HapticGenerator_Dump(*(reinterpret_cast<int32_t*>(cmdData)), context->param);
+            break;
+
         default:
             ALOGW("HapticGenerator_Command invalid command %u", cmdCode);
             return -EINVAL;
diff --git a/media/libheif/HeifDecoderImpl.cpp b/media/libheif/HeifDecoderImpl.cpp
index 1b8656d..923f5c1 100644
--- a/media/libheif/HeifDecoderImpl.cpp
+++ b/media/libheif/HeifDecoderImpl.cpp
@@ -735,9 +735,9 @@
     HeifFrameInfo* info = &mImageInfo;
     if (info != nullptr) {
         return mImageInfo.mBitDepth;
+    } else {
+        return 0;
     }
-
-    return 0;
 }
 
 } // namespace android
diff --git a/media/libheif/include/HeifDecoderAPI.h b/media/libheif/include/HeifDecoderAPI.h
index dc12486..56f4765 100644
--- a/media/libheif/include/HeifDecoderAPI.h
+++ b/media/libheif/include/HeifDecoderAPI.h
@@ -47,7 +47,7 @@
     int32_t  mRotationAngle;           // Rotation angle, clockwise, should be multiple of 90
     uint32_t mBytesPerPixel;           // Number of bytes for one pixel
     int64_t  mDurationUs;              // Duration of the frame in us
-    uint32_t mBitDepth;                // Number of bits for each of the R/G/B channels
+    uint32_t mBitDepth;                // Number of bits of R/G/B channel
     std::vector<uint8_t> mIccData;     // ICC data array
 };
 
@@ -164,7 +164,7 @@
     virtual size_t skipScanlines(size_t count) = 0;
 
     /*
-     * Returns color depth in bits for each of the R/G/B channels.
+     * Returns color depth of R/G/B channel.
      */
     virtual uint32_t getColorDepth() = 0;
 
diff --git a/media/libmediaplayerservice/fuzzer/mediaplayer_fuzzer.cpp b/media/libmediaplayerservice/fuzzer/mediaplayer_fuzzer.cpp
index 7799f44..a189d04 100644
--- a/media/libmediaplayerservice/fuzzer/mediaplayer_fuzzer.cpp
+++ b/media/libmediaplayerservice/fuzzer/mediaplayer_fuzzer.cpp
@@ -26,6 +26,8 @@
 #include <media/IMediaRecorder.h>
 #include <media/IRemoteDisplay.h>
 #include <media/IRemoteDisplayClient.h>
+#include <media/MediaHTTPConnection.h>
+#include <media/MediaHTTPService.h>
 #include <media/stagefright/RemoteDataSource.h>
 #include <media/stagefright/foundation/base64.h>
 #include <thread>
@@ -102,6 +104,42 @@
     IBinder *onAsBinder() { return nullptr; };
 };
 
+struct TestMediaHTTPConnection : public MediaHTTPConnection {
+  public:
+    TestMediaHTTPConnection() {}
+    virtual ~TestMediaHTTPConnection() {}
+
+    virtual bool connect(const char* /*uri*/, const KeyedVector<String8, String8>* /*headers*/) {
+        return true;
+    }
+
+    virtual void disconnect() { return; }
+
+    virtual ssize_t readAt(off64_t /*offset*/, void* /*data*/, size_t size) { return size; }
+
+    virtual off64_t getSize() { return 0; }
+    virtual status_t getMIMEType(String8* /*mimeType*/) { return NO_ERROR; }
+    virtual status_t getUri(String8* /*uri*/) { return NO_ERROR; }
+
+  private:
+    DISALLOW_EVIL_CONSTRUCTORS(TestMediaHTTPConnection);
+};
+
+struct TestMediaHTTPService : public BnInterface<IMediaHTTPService> {
+  public:
+    TestMediaHTTPService() {}
+    ~TestMediaHTTPService(){};
+
+    virtual sp<MediaHTTPConnection> makeHTTPConnection() {
+        mMediaHTTPConnection = sp<TestMediaHTTPConnection>::make();
+        return mMediaHTTPConnection;
+    }
+
+  private:
+    sp<TestMediaHTTPConnection> mMediaHTTPConnection = nullptr;
+    DISALLOW_EVIL_CONSTRUCTORS(TestMediaHTTPService);
+};
+
 class BinderDeathNotifier : public IBinder::DeathRecipient {
    public:
     void binderDied(const wp<IBinder> &) { abort(); }
@@ -140,7 +178,9 @@
             AString out;
             encodeBase64(uriSuffix.data(), uriSuffix.size(), &out);
             uri += out.c_str();
-            status = mMediaPlayer->setDataSource(nullptr /*httpService*/, uri.c_str(), &headers);
+            sp<TestMediaHTTPService> testService = sp<TestMediaHTTPService>::make();
+            status =
+                    mMediaPlayer->setDataSource(testService /*httpService*/, uri.c_str(), &headers);
             break;
         }
         case fd: {
diff --git a/media/libnblog/Reader.cpp b/media/libnblog/Reader.cpp
index 67d028d..d6232d4 100644
--- a/media/libnblog/Reader.cpp
+++ b/media/libnblog/Reader.cpp
@@ -208,11 +208,14 @@
     }
     while (back + Entry::kPreviousLengthOffset >= front) {
         const uint8_t *prev = back - back[Entry::kPreviousLengthOffset] - Entry::kOverhead;
-        const Event type = (const Event)prev[offsetof(entry, type)];
         if (prev < front
-                || prev + prev[offsetof(entry, length)] + Entry::kOverhead != back
-                || type <= EVENT_RESERVED || type >= EVENT_UPPER_BOUND) {
-            // prev points to an out of limits or inconsistent entry
+                || prev + prev[offsetof(entry, length)] + Entry::kOverhead != back) {
+            // prev points to an out of limits entry
+            return nullptr;
+        }
+        const Event type = (const Event)prev[offsetof(entry, type)];
+        if (type <= EVENT_RESERVED || type >= EVENT_UPPER_BOUND) {
+            // prev points to an inconsistent entry
             return nullptr;
         }
         // if invalidTypes does not contain the type, then the type is valid.
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index ce4b4e6..6de112a 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -917,7 +917,7 @@
         return ERROR_MALFORMED;
     }
 
-    int32_t width, height, stride, srcFormat;
+    int32_t width, height, stride;
     if (outputFormat->findInt32("width", &width) == false) {
         ALOGE("MediaImageDecoder::onOutputReceived:width is missing in outputFormat");
         return ERROR_MALFORMED;
@@ -930,10 +930,9 @@
         ALOGE("MediaImageDecoder::onOutputReceived:stride is missing in outputFormat");
         return ERROR_MALFORMED;
     }
-    if (outputFormat->findInt32("color-format", &srcFormat) == false) {
-        ALOGE("MediaImageDecoder::onOutputReceived: color format is missing in outputFormat");
-        return ERROR_MALFORMED;
-    }
+
+    int32_t srcFormat;
+    CHECK(outputFormat->findInt32("color-format", &srcFormat));
 
     uint32_t bitDepth = 8;
     if (COLOR_FormatYUVP010 == srcFormat) {
diff --git a/media/libstagefright/MediaClock.cpp b/media/libstagefright/MediaClock.cpp
index 24608a7..ed0819d 100644
--- a/media/libstagefright/MediaClock.cpp
+++ b/media/libstagefright/MediaClock.cpp
@@ -110,8 +110,12 @@
     if (mAnchorTimeRealUs != -1) {
         int64_t oldNowMediaUs =
             mAnchorTimeMediaUs + (nowUs - mAnchorTimeRealUs) * (double)mPlaybackRate;
-        if (nowMediaUs < oldNowMediaUs + kAnchorFluctuationAllowedUs
-                && nowMediaUs > oldNowMediaUs - kAnchorFluctuationAllowedUs) {
+        // earlier, we ensured that the anchor times are non-negative and the
+        // math to calculate the now/oldNow times stays non-negative.
+        // by casting into uint64_t, we gain headroom to avoid any overflows at the upper end
+        // when adding the fluctuation allowance.
+        if ((uint64_t)nowMediaUs < (uint64_t)oldNowMediaUs + kAnchorFluctuationAllowedUs
+                && (uint64_t)nowMediaUs + kAnchorFluctuationAllowedUs > (uint64_t)oldNowMediaUs) {
             return;
         }
     }
diff --git a/media/libstagefright/bqhelper/Android.bp b/media/libstagefright/bqhelper/Android.bp
index 0e2b472..a4f0fb3 100644
--- a/media/libstagefright/bqhelper/Android.bp
+++ b/media/libstagefright/bqhelper/Android.bp
@@ -12,6 +12,7 @@
     double_loadable: true,
 
     srcs: [
+        ":libgui_frame_event_aidl",
         "FrameDropper.cpp",
         "GraphicBufferSource.cpp",
     ],
diff --git a/media/libstagefright/data/media_codecs_google_c2_video.xml b/media/libstagefright/data/media_codecs_google_c2_video.xml
index 3509ef8..03d8b78 100644
--- a/media/libstagefright/data/media_codecs_google_c2_video.xml
+++ b/media/libstagefright/data/media_codecs_google_c2_video.xml
@@ -60,7 +60,7 @@
         <MediaCodec name="c2.android.vp8.decoder" type="video/x-vnd.on2.vp8">
             <Alias name="OMX.google.vp8.decoder" />
             <Limit name="size" min="2x2" max="2048x2048" />
-            <Limit name="alignment" value="2x2" />
+            <Limit name="alignment" value="1x1" />
             <Limit name="block-size" value="16x16" />
             <Limit name="block-count" range="1-16384" />
             <Limit name="blocks-per-second" range="1-1000000" />
@@ -70,7 +70,7 @@
         <MediaCodec name="c2.android.vp9.decoder" type="video/x-vnd.on2.vp9">
             <Alias name="OMX.google.vp9.decoder" />
             <Limit name="size" min="2x2" max="2048x2048" />
-            <Limit name="alignment" value="2x2" />
+            <Limit name="alignment" value="1x1" />
             <Limit name="block-size" value="16x16" />
             <Limit name="block-count" range="1-16384" />
             <Limit name="blocks-per-second" range="1-500000" />
diff --git a/media/libstagefright/data/media_codecs_sw.xml b/media/libstagefright/data/media_codecs_sw.xml
index d7e2d18..fb4f9a0 100644
--- a/media/libstagefright/data/media_codecs_sw.xml
+++ b/media/libstagefright/data/media_codecs_sw.xml
@@ -150,7 +150,7 @@
         <MediaCodec name="c2.android.vp8.decoder" type="video/x-vnd.on2.vp8" variant="slow-cpu,!slow-cpu">
             <Alias name="OMX.google.vp8.decoder" />
             <Limit name="size" min="2x2" max="2048x2048" />
-            <Limit name="alignment" value="2x2" />
+            <Limit name="alignment" value="1x1" />
             <Limit name="block-size" value="16x16" />
             <Variant name="!slow-cpu">
                 <Limit name="block-count" range="1-16384" />
@@ -166,7 +166,7 @@
         </MediaCodec>
         <MediaCodec name="c2.android.vp9.decoder" type="video/x-vnd.on2.vp9" variant="slow-cpu,!slow-cpu">
             <Alias name="OMX.google.vp9.decoder" />
-            <Limit name="alignment" value="2x2" />
+            <Limit name="alignment" value="1x1" />
             <Limit name="block-size" value="16x16" />
             <Variant name="!slow-cpu">
                 <Limit name="size" min="2x2" max="2048x2048" />
diff --git a/media/ndk/NdkImage.cpp b/media/ndk/NdkImage.cpp
index 12a0d53..0bdb41b 100644
--- a/media/ndk/NdkImage.cpp
+++ b/media/ndk/NdkImage.cpp
@@ -24,6 +24,7 @@
 
 #include <android_media_Utils.h>
 #include <private/android/AHardwareBufferHelpers.h>
+#include <ui/PublicFormat.h>
 #include <utils/Log.h>
 
 using namespace android;
@@ -34,6 +35,8 @@
         int64_t timestamp, int32_t width, int32_t height, int32_t numPlanes) :
         mReader(reader), mFormat(format), mUsage(usage), mBuffer(buffer), mLockedBuffer(nullptr),
         mTimestamp(timestamp), mWidth(width), mHeight(height), mNumPlanes(numPlanes) {
+    PublicFormat publicFormat = static_cast<PublicFormat>(format);
+    mHalDataSpace = mapPublicFormatToHalDataspace(publicFormat);
     LOG_FATAL_IF(reader == nullptr, "AImageReader shouldn't be null while creating AImage");
 }
 
@@ -156,6 +159,20 @@
     return AMEDIA_OK;
 }
 
+media_status_t
+AImage::getDataSpace(android_dataspace* dataSpace) const {
+    if (dataSpace == nullptr) {
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+    *dataSpace = static_cast<android_dataspace>(-1);
+    if (isClosed()) {
+        ALOGE("%s: image %p has been closed!", __FUNCTION__, this);
+        return AMEDIA_ERROR_INVALID_OBJECT;
+    }
+    *dataSpace = mHalDataSpace;
+    return AMEDIA_OK;
+}
+
 media_status_t AImage::lockImage() {
     if (mBuffer == nullptr || mBuffer->mGraphicBuffer == nullptr) {
         LOG_ALWAYS_FATAL("%s: AImage %p has no buffer.", __FUNCTION__, this);
@@ -762,3 +779,15 @@
     }
     return image->getHardwareBuffer(buffer);
 }
+
+EXPORT
+media_status_t AImage_getDataSpace(
+    const AImage* image, /*out*/int32_t* dataSpace) {
+    ALOGV("%s", __FUNCTION__);
+
+    if (image == nullptr || dataSpace == nullptr) {
+        ALOGE("%s: bad argument. image %p dataSpace %p", __FUNCTION__, image, dataSpace);
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+    return image->getDataSpace((android_dataspace*)(dataSpace));
+}
\ No newline at end of file
diff --git a/media/ndk/NdkImagePriv.h b/media/ndk/NdkImagePriv.h
index 05115b9..dc10a6a 100644
--- a/media/ndk/NdkImagePriv.h
+++ b/media/ndk/NdkImagePriv.h
@@ -82,6 +82,7 @@
     media_status_t getPlaneRowStride(int planeIdx, /*out*/int32_t* rowStride) const;
     media_status_t getPlaneData(int planeIdx,/*out*/uint8_t** data, /*out*/int* dataLength) const;
     media_status_t getHardwareBuffer(/*out*/AHardwareBuffer** buffer) const;
+    media_status_t getDataSpace(/*out*/android_dataspace* dataSpace) const;
 
   private:
     // AImage should be deleted through free() API.
@@ -101,6 +102,7 @@
     const int32_t              mWidth;
     const int32_t              mHeight;
     const int32_t              mNumPlanes;
+    android_dataspace          mHalDataSpace = HAL_DATASPACE_UNKNOWN;
     bool                       mIsClosed = false;
     mutable Mutex              mLock;
 };
diff --git a/media/ndk/NdkImageReader.cpp b/media/ndk/NdkImageReader.cpp
index 1067e24..ac5cba8 100644
--- a/media/ndk/NdkImageReader.cpp
+++ b/media/ndk/NdkImageReader.cpp
@@ -46,6 +46,9 @@
 
 static constexpr int kWindowHalTokenSizeMax = 256;
 
+static media_status_t validateParameters(int32_t width, int32_t height, int32_t format,
+                                         uint64_t usage, int32_t maxImages,
+                                         /*out*/ AImageReader**& reader);
 static native_handle_t *convertHalTokenToNativeHandle(const HalToken &halToken);
 
 bool
@@ -263,13 +266,17 @@
                            int32_t height,
                            int32_t format,
                            uint64_t usage,
-                           int32_t maxImages)
+                           int32_t maxImages,
+                           uint32_t hardwareBufferFormat,
+                           android_dataspace dataSpace)
     : mWidth(width),
       mHeight(height),
       mFormat(format),
       mUsage(usage),
       mMaxImages(maxImages),
       mNumPlanes(getNumPlanesForFormat(format)),
+      mHalFormat(hardwareBufferFormat),
+      mHalDataSpace(dataSpace),
       mFrameListener(new FrameListener(this)),
       mBufferRemovedListener(new BufferRemovedListener(this)) {}
 
@@ -280,9 +287,6 @@
 
 media_status_t
 AImageReader::init() {
-    PublicFormat publicFormat = static_cast<PublicFormat>(mFormat);
-    mHalFormat = mapPublicFormatToHalFormat(publicFormat);
-    mHalDataSpace = mapPublicFormatToHalDataspace(publicFormat);
     mHalUsage = AHardwareBuffer_convertToGrallocUsageBits(mUsage);
 
     sp<IGraphicBufferProducer> gbProducer;
@@ -646,6 +650,41 @@
     }
 }
 
+static
+media_status_t validateParameters(int32_t width, int32_t height, int32_t format,
+                                  uint64_t usage, int32_t maxImages,
+                                  /*out*/ AImageReader**& reader) {
+    if (reader == nullptr) {
+        ALOGE("%s: reader argument is null", __FUNCTION__);
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+
+    if (width < 1 || height < 1) {
+        ALOGE("%s: image dimension must be positive: w:%d h:%d",
+                __FUNCTION__, width, height);
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+
+    if (maxImages < 1) {
+        ALOGE("%s: max outstanding image count must be at least 1 (%d)",
+                __FUNCTION__, maxImages);
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+
+    if (maxImages > BufferQueueDefs::NUM_BUFFER_SLOTS) {
+        ALOGE("%s: max outstanding image count (%d) cannot be larget than %d.",
+              __FUNCTION__, maxImages, BufferQueueDefs::NUM_BUFFER_SLOTS);
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+
+    if (!AImageReader::isSupportedFormatAndUsage(format, usage)) {
+        ALOGE("%s: format %d is not supported with usage 0x%" PRIx64 " by AImageReader",
+                __FUNCTION__, format, usage);
+        return AMEDIA_ERROR_INVALID_PARAMETER;
+    }
+    return AMEDIA_OK;
+}
+
 static native_handle_t *convertHalTokenToNativeHandle(
         const HalToken &halToken) {
     // We attempt to store halToken in the ints of the native_handle_t after its
@@ -696,42 +735,32 @@
 } //extern "C"
 
 EXPORT
+media_status_t AImageReader_newWithDataSpace(
+        int32_t width, int32_t height, uint64_t usage, int32_t maxImages,
+        uint32_t hardwareBufferFormat, int32_t dataSpace,
+        /*out*/ AImageReader** reader) {
+    ALOGV("%s", __FUNCTION__);
+
+    android_dataspace halDataSpace = static_cast<android_dataspace>(dataSpace);
+    int32_t format = static_cast<int32_t>(
+          mapHalFormatDataspaceToPublicFormat(hardwareBufferFormat, halDataSpace));
+    return AImageReader_newWithUsage(width, height, format, usage, maxImages, reader);
+}
+
+EXPORT
 media_status_t AImageReader_newWithUsage(
         int32_t width, int32_t height, int32_t format, uint64_t usage,
         int32_t maxImages, /*out*/ AImageReader** reader) {
     ALOGV("%s", __FUNCTION__);
 
-    if (width < 1 || height < 1) {
-        ALOGE("%s: image dimension must be positive: w:%d h:%d",
-                __FUNCTION__, width, height);
-        return AMEDIA_ERROR_INVALID_PARAMETER;
-    }
+    validateParameters(width, height, format, usage, maxImages, reader);
 
-    if (maxImages < 1) {
-        ALOGE("%s: max outstanding image count must be at least 1 (%d)",
-                __FUNCTION__, maxImages);
-        return AMEDIA_ERROR_INVALID_PARAMETER;
-    }
-
-    if (maxImages > BufferQueueDefs::NUM_BUFFER_SLOTS) {
-        ALOGE("%s: max outstanding image count (%d) cannot be larget than %d.",
-              __FUNCTION__, maxImages, BufferQueueDefs::NUM_BUFFER_SLOTS);
-        return AMEDIA_ERROR_INVALID_PARAMETER;
-    }
-
-    if (!AImageReader::isSupportedFormatAndUsage(format, usage)) {
-        ALOGE("%s: format %d is not supported with usage 0x%" PRIx64 " by AImageReader",
-                __FUNCTION__, format, usage);
-        return AMEDIA_ERROR_INVALID_PARAMETER;
-    }
-
-    if (reader == nullptr) {
-        ALOGE("%s: reader argument is null", __FUNCTION__);
-        return AMEDIA_ERROR_INVALID_PARAMETER;
-    }
+    PublicFormat publicFormat = static_cast<PublicFormat>(format);
+    uint32_t halFormat = mapPublicFormatToHalFormat(publicFormat);
+    android_dataspace halDataSpace = mapPublicFormatToHalDataspace(publicFormat);
 
     AImageReader* tmpReader = new AImageReader(
-        width, height, format, usage, maxImages);
+        width, height, format, usage, maxImages, halFormat, halDataSpace);
     if (tmpReader == nullptr) {
         ALOGE("%s: AImageReader allocation failed", __FUNCTION__);
         return AMEDIA_ERROR_UNKNOWN;
diff --git a/media/ndk/NdkImageReaderPriv.h b/media/ndk/NdkImageReaderPriv.h
index 37c606e..0199616 100644
--- a/media/ndk/NdkImageReaderPriv.h
+++ b/media/ndk/NdkImageReaderPriv.h
@@ -56,10 +56,12 @@
                  int32_t height,
                  int32_t format,
                  uint64_t usage,
-                 int32_t maxImages);
+                 int32_t maxImages,
+                 uint32_t hardwareBufferFormat,
+                 android_dataspace dataSpace);
     ~AImageReader();
 
-    // Inintialize AImageReader, uninitialized or failed to initialize AImageReader
+    // Initialize AImageReader, uninitialized or failed to initialize AImageReader
     // should never be passed to application
     media_status_t init();
 
@@ -79,7 +81,6 @@
     void           close();
 
   private:
-
     friend struct AImage; // for grabing reader lock
 
     BufferItem* getBufferItemLocked();
@@ -118,13 +119,16 @@
 
     const int32_t mWidth;
     const int32_t mHeight;
-    const int32_t mFormat;
+    int32_t mFormat;
     const uint64_t mUsage;  // AHARDWAREBUFFER_USAGE_* flags.
     const int32_t mMaxImages;
 
     // TODO(jwcai) Seems completely unused in AImageReader class.
     const int32_t mNumPlanes;
 
+    uint32_t mHalFormat = AHARDWAREBUFFER_FORMAT_R8G8B8A8_UNORM;
+    android_dataspace mHalDataSpace = HAL_DATASPACE_UNKNOWN;
+
     struct FrameListener : public ConsumerBase::FrameAvailableListener {
       public:
         explicit FrameListener(AImageReader* parent) : mReader(parent) {}
@@ -155,8 +159,6 @@
     };
     sp<BufferRemovedListener> mBufferRemovedListener;
 
-    int mHalFormat;
-    android_dataspace mHalDataSpace;
     uint64_t mHalUsage;
 
     sp<IGraphicBufferProducer> mProducer;
diff --git a/media/ndk/include/media/NdkImage.h b/media/ndk/include/media/NdkImage.h
index 71bc6d9..814a327 100644
--- a/media/ndk/include/media/NdkImage.h
+++ b/media/ndk/include/media/NdkImage.h
@@ -583,7 +583,7 @@
  * Available since API level 24.
  *
  * @param image the {@link AImage} of interest.
- * @param width the width of the image will be filled here if the method call succeeeds.
+ * @param width the width of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -599,7 +599,7 @@
  * Available since API level 24.
  *
  * @param image the {@link AImage} of interest.
- * @param height the height of the image will be filled here if the method call succeeeds.
+ * @param height the height of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -617,7 +617,7 @@
  * Available since API level 24.
  *
  * @param image the {@link AImage} of interest.
- * @param format the format of the image will be filled here if the method call succeeeds.
+ * @param format the format of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -636,7 +636,7 @@
  * Available since API level 24.
  *
  * @param image the {@link AImage} of interest.
- * @param rect the cropped rectangle of the image will be filled here if the method call succeeeds.
+ * @param rect the cropped rectangle of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -662,7 +662,7 @@
  * Available since API level 24.
  *
  * @param image the {@link AImage} of interest.
- * @param timestampNs the timestamp of the image will be filled here if the method call succeeeds.
+ * @param timestampNs the timestamp of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -682,7 +682,7 @@
  *
  * @param image the {@link AImage} of interest.
  * @param numPlanes the number of planes of the image will be filled here if the method call
- *         succeeeds.
+ *         succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -706,7 +706,7 @@
  *
  * @param image the {@link AImage} of interest.
  * @param planeIdx the index of the plane. Must be less than the number of planes of input image.
- * @param pixelStride the pixel stride of the image will be filled here if the method call succeeeds.
+ * @param pixelStride the pixel stride of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -735,7 +735,7 @@
  *
  * @param image the {@link AImage} of interest.
  * @param planeIdx the index of the plane. Must be less than the number of planes of input image.
- * @param rowStride the row stride of the image will be filled here if the method call succeeeds.
+ * @param rowStride the row stride of the image will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -762,8 +762,8 @@
  *
  * @param image the {@link AImage} of interest.
  * @param planeIdx the index of the plane. Must be less than the number of planes of input image.
- * @param data the data pointer of the image will be filled here if the method call succeeeds.
- * @param dataLength the valid length of data will be filled here if the method call succeeeds.
+ * @param data the data pointer of the image will be filled here if the method call succeeds.
+ * @param dataLength the valid length of data will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -826,6 +826,25 @@
  */
 media_status_t AImage_getHardwareBuffer(const AImage* image, /*out*/AHardwareBuffer** buffer) __INTRODUCED_IN(26);
 
+/**
+ * Query the dataspace of the input {@link AImage}.
+ *
+ * Available since API level 33.
+ *
+ * @param image the {@link AImage} of interest.
+ * @param dataSpace the dataspace of the image will be filled here if the method call succeeds.
+ *                  This must be one of the ADATASPACE_* enum value defined in
+ *                  {@link ADataSpace}.
+ *
+ * @return <ul>
+ *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
+ *         <li>{@link AMEDIA_ERROR_INVALID_PARAMETER} if image or dataSpace is NULL.</li>
+ *         <li>{@link AMEDIA_ERROR_INVALID_OBJECT} if the {@link AImageReader} generated this
+ *                 image has been deleted.</li></ul>
+ */
+media_status_t AImage_getDataSpace(const AImage* image,
+                                   /*out*/int32_t* dataSpace) __INTRODUCED_IN(33);
+
 __END_DECLS
 
 #endif //_NDK_IMAGE_H
diff --git a/media/ndk/include/media/NdkImageReader.h b/media/ndk/include/media/NdkImageReader.h
index 4bd7f2a..992955b 100644
--- a/media/ndk/include/media/NdkImageReader.h
+++ b/media/ndk/include/media/NdkImageReader.h
@@ -79,7 +79,7 @@
  *            by the user, one of them has to be released before a new {@link AImage} will become
  *            available for access through {@link AImageReader_acquireLatestImage} or
  *            {@link AImageReader_acquireNextImage}. Must be greater than 0.
- * @param reader The created image reader will be filled here if the method call succeeeds.
+ * @param reader The created image reader will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -133,7 +133,7 @@
  * Available since API level 24.
  *
  * @param reader The image reader of interest.
- * @param width the default width of the reader will be filled here if the method call succeeeds.
+ * @param width the default width of the reader will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -151,7 +151,7 @@
  * Available since API level 24.
  *
  * @param reader The image reader of interest.
- * @param height the default height of the reader will be filled here if the method call succeeeds.
+ * @param height the default height of the reader will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -165,7 +165,7 @@
  * Available since API level 24.
  *
  * @param reader The image reader of interest.
- * @param format the fromat of the reader will be filled here if the method call succeeeds. The
+ * @param format the format of the reader will be filled here if the method call succeeds. The
  *                value will be one of the AIMAGE_FORMAT_* enum value defiend in {@link NdkImage.h}.
  *
  * @return <ul>
@@ -181,7 +181,7 @@
  *
  * @param reader The image reader of interest.
  * @param maxImages the maximum number of concurrently acquired images of the reader will be filled
- *                here if the method call succeeeds.
+ *                here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -212,7 +212,7 @@
  * Available since API level 24.
  *
  * @param reader The image reader of interest.
- * @param image the acquired {@link AImage} will be filled here if the method call succeeeds.
+ * @param image the acquired {@link AImage} will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -257,7 +257,7 @@
  * Available since API level 24.
  *
  * @param reader The image reader of interest.
- * @param image the acquired {@link AImage} will be filled here if the method call succeeeds.
+ * @param image the acquired {@link AImage} will be filled here if the method call succeeds.
  *
  * @return <ul>
  *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
@@ -387,6 +387,44 @@
         /*out*/ AImageReader** reader) __INTRODUCED_IN(26);
 
 /**
+ * AImageReader constructor similar to {@link AImageReader_newWithUsage} that takes
+ * two additional parameters to build the format of the Image. All other parameters
+ * and the return values are identical to those passed to {@link AImageReader_newWithUsage}.
+ *
+ * <p>Instead of passing {@code format} parameter, this constructor accepts
+ * the combination of {@code hardwareBufferFormat} and {@code dataSpace} for the
+ * format of the Image that the reader will produce.</p>
+ *
+ * Available since API level 33.
+ *
+ * @param width The default width in pixels of the Images that this reader will produce.
+ * @param height The default height in pixels of the Images that this reader will produce.
+ * @param usage specifies how the consumer will access the AImage.
+ *              See {@link AImageReader_newWithUsage} parameter description for more details.
+ * @param maxImages The maximum number of images the user will want to access simultaneously.
+ *                  See {@link AImageReader_newWithUsage} parameter description for more details.
+ * @param hardwareBufferFormat The hardware buffer format passed by the producer.
+ *                             This must be one of the AHARDWAREBUFFER_FORMAT_* enum values defined
+ *                             in {@link hardware_buffer.h}.
+ * @param dataSpace The dataspace of the Image passed by the producer.
+ *                  This must be one of the ADATASPACE_* enum values defined in
+ *                  {@link ADataSpace}.
+ * @param reader The created image reader will be filled here if the method call succeeds.
+ *
+ * @return <ul>
+ *         <li>{@link AMEDIA_OK} if the method call succeeds.</li>
+ *         <li>{@link AMEDIA_ERROR_INVALID_PARAMETER} if reader is NULL, or one or more of width,
+ *                 height, maxImages, hardwareBufferFormat or dataSpace arguments
+ *                 is not supported.</li>
+ *         <li>{@link AMEDIA_ERROR_UNKNOWN} if the method fails for some other reasons.</li></ul>
+ *
+ * @see AImageReader_newWithUsage
+ */
+media_status_t AImageReader_newWithDataSpace(int32_t width, int32_t height, uint64_t usage,
+        int32_t maxImages, uint32_t hardwareBufferFormat, int32_t dataSpace,
+        /*out*/ AImageReader** reader) __INTRODUCED_IN(33);
+
+/**
  * Acquire the next {@link AImage} from the image reader's queue asynchronously.
  *
  * <p>AImageReader acquire method similar to {@link AImageReader_acquireNextImage} that takes an
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index c8faced..bac4b22 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -13,11 +13,13 @@
     AImageReader_getWindow; # introduced=24
     AImageReader_new; # introduced=24
     AImageReader_newWithUsage; # introduced=26
+    AImageReader_newWithDataSpace; # introduced=Tiramisu
     AImageReader_setBufferRemovedListener; # introduced=26
     AImageReader_setImageListener; # introduced=24
     AImage_delete; # introduced=24
     AImage_deleteAsync; # introduced=26
     AImage_getCropRect; # introduced=24
+    AImage_getDataSpace; # introduced=Tiramisu
     AImage_getFormat; # introduced=24
     AImage_getHardwareBuffer; # introduced=26
     AImage_getHeight; # introduced=24
diff --git a/media/utils/include/mediautils/ExtendedAccumulator.h b/media/utils/include/mediautils/ExtendedAccumulator.h
new file mode 100644
index 0000000..7e3e170
--- /dev/null
+++ b/media/utils/include/mediautils/ExtendedAccumulator.h
@@ -0,0 +1,90 @@
+/*
+ * Copyright (C) 2022 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 <atomic>
+#include <cstdint>
+#include <tuple>
+#include <type_traits>
+
+#include <log/log.h>
+
+namespace android::mediautils {
+
+// The goal of this class is to detect and accumulate wraparound occurrences on a
+// lower sized integer.
+
+// This class assumes that the underlying unsigned type is either incremented or
+// decremented by at most the underlying signed type between any two subsequent
+// polls (or construction). This is well-defined as the modular nature of
+// unsigned arithmetic ensures that every new value maps 1-1 to an
+// increment/decrement over the same sized signed type. It also ensures that our
+// counter will be equivalent mod the size of the integer even if the underlying
+// type is modified outside of this range.
+//
+// For convenience, this class is thread compatible. Additionally, it is safe
+// as long as there is only one writer.
+template <typename Integral = uint32_t, typename AccumulatingType = uint64_t>
+class ExtendedAccumulator {
+    static_assert(sizeof(Integral) < sizeof(AccumulatingType),
+                  "Accumulating type should be larger than underlying type");
+    static_assert(std::is_integral_v<Integral> && std::is_unsigned_v<Integral>,
+                  "Wraparound behavior is only well-defiend for unsigned ints");
+    static_assert(std::is_integral_v<AccumulatingType>);
+
+  public:
+    enum class Wrap {
+        NORMAL = 0,
+        UNDERFLOW = 1,
+        OVERFLOW = 2,
+    };
+
+    using UnsignedInt = Integral;
+    using SignedInt = std::make_signed_t<UnsignedInt>;
+
+    explicit ExtendedAccumulator(AccumulatingType initial = 0) : mAccumulated(initial) {}
+
+    // Returns a pair of the calculated change on the accumulating value, and a
+    // Wrap type representing the type of wraparound (if any) which occurred.
+    std::pair<SignedInt, Wrap> poll(UnsignedInt value) {
+        auto acc = mAccumulated.load(std::memory_order_relaxed);
+        const auto bottom_bits = static_cast<UnsignedInt>(acc);
+        std::pair<SignedInt, Wrap> res = {0, Wrap::NORMAL};
+        const bool overflow = __builtin_sub_overflow(value, bottom_bits, &res.first);
+
+        if (overflow) {
+            res.second = (res.first > 0) ? Wrap::OVERFLOW : Wrap::UNDERFLOW;
+        }
+
+        const bool acc_overflow = __builtin_add_overflow(acc, res.first, &acc);
+        // If our *accumulating* type overflows or underflows (depending on its
+        // signedness), we should abort.
+        if (acc_overflow) LOG_ALWAYS_FATAL("Unexpected overflow/underflow in %s", __func__);
+
+        mAccumulated.store(acc, std::memory_order_relaxed);
+        return res;
+    }
+
+    AccumulatingType getValue() const { return mAccumulated.load(std::memory_order_relaxed); }
+
+  private:
+    // Invariant - the bottom underlying bits of accumulated are the same as the
+    // last value provided to poll.
+    std::atomic<AccumulatingType> mAccumulated;
+};
+
+}  // namespace android::mediautils
diff --git a/media/utils/tests/Android.bp b/media/utils/tests/Android.bp
index 1024018..82dd0dc 100644
--- a/media/utils/tests/Android.bp
+++ b/media/utils/tests/Android.bp
@@ -15,11 +15,11 @@
         "-Wextra",
     ],
 
-    sanitize:{
-       address: true,
-       cfi: true,
-       integer_overflow: true,
-       memtag_heap: true,
+    sanitize: {
+        address: true,
+        cfi: true,
+        integer_overflow: true,
+        memtag_heap: true,
     },
 
     shared_libs: [
@@ -28,7 +28,7 @@
 
     srcs: [
         "sharedtest.cpp",
-    ]
+    ],
 }
 
 cc_test {
@@ -40,11 +40,11 @@
         "-Wextra",
     ],
 
-    sanitize:{
-       address: true,
-       cfi: true,
-       integer_overflow: true,
-       memtag_heap: true,
+    sanitize: {
+        address: true,
+        cfi: true,
+        integer_overflow: true,
+        memtag_heap: true,
     },
 
     shared_libs: [
@@ -174,11 +174,11 @@
         "-Wextra",
     ],
 
-    sanitize:{
-       address: true,
-       cfi: true,
-       integer_overflow: true,
-       memtag_heap: true,
+    sanitize: {
+        address: true,
+        cfi: true,
+        integer_overflow: true,
+        memtag_heap: true,
     },
 
     shared_libs: [
@@ -191,3 +191,30 @@
         "timecheck_tests.cpp",
     ],
 }
+
+cc_test {
+    name: "extended_accumulator_tests",
+    cflags: [
+        "-Wall",
+        "-Werror",
+        "-Wextra",
+    ],
+
+    sanitize: {
+        address: true,
+        cfi: true,
+        integer_overflow: true,
+        memtag_heap: true,
+    },
+
+    shared_libs: [
+        "libbase",
+        "liblog",
+        "libmediautils",
+        "libutils",
+    ],
+
+    srcs: [
+        "extended_accumulator_tests.cpp",
+    ],
+}
diff --git a/media/utils/tests/extended_accumulator_tests.cpp b/media/utils/tests/extended_accumulator_tests.cpp
new file mode 100644
index 0000000..e243e7e
--- /dev/null
+++ b/media/utils/tests/extended_accumulator_tests.cpp
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2022 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 "extended_accumulator_tests"
+
+#include <mediautils/ExtendedAccumulator.h>
+
+#include <type_traits>
+#include <cstdint>
+#include <limits.h>
+
+#include <gtest/gtest.h>
+#include <log/log.h>
+
+using namespace android;
+using namespace android::mediautils;
+
+// Conditionally choose a base accumulating counter value in order to prevent
+// unsigned underflow on the accumulator from aborting the tests.
+template <typename TType, typename CType>
+static constexpr CType getBase() {
+  static_assert(sizeof(TType) < sizeof(CType));
+  if constexpr (std::is_unsigned_v<CType>) {
+      return std::numeric_limits<TType>::max() + 1;
+  } else {
+      return 0;
+  }
+}
+
+// Since the entire state of this utility is the previous value, and the
+// behavior is isomorphic mod the underlying type on the previous value, we can
+// test combinations of the previous value of the underlying type and a
+// hypothetical signed update to that type and ensure the accumulator moves
+// correctly and reports overflow correctly.
+template <typename TestUInt, typename CType>
+void testPair(TestUInt prevVal, std::make_signed_t<TestUInt> delta) {
+    using TestDetect = ExtendedAccumulator<TestUInt, CType>;
+    using TestInt = typename TestDetect::SignedInt;
+    static_assert(std::is_same_v<typename TestDetect::UnsignedInt, TestUInt>);
+    static_assert(std::is_same_v<TestInt, std::make_signed_t<TestUInt>>);
+    static_assert(sizeof(TestUInt) < sizeof(CType));
+
+    // To safely detect underflow/overflow for testing
+    // Should be 0 mod TestUInt, max + 1 is convenient
+    static constexpr CType base = getBase<TestUInt, CType>();
+    const CType prev = base + prevVal;
+    TestDetect test{prev};
+    EXPECT_EQ(test.getValue(), prev);
+    // Prevent unsigned wraparound abort
+    CType next;
+    const auto err =  __builtin_add_overflow(prev, delta, &next);
+    LOG_ALWAYS_FATAL_IF(err, "Unexpected wrap in tests");
+    const auto [result, status] = test.poll(static_cast<TestUInt>(next));
+    EXPECT_EQ(test.getValue(), next);
+    EXPECT_EQ(result, delta);
+
+    // Test overflow/underflow event reporting.
+    if (next < base) EXPECT_EQ(TestDetect::Wrap::UNDERFLOW, status);
+    else if (next > base + std::numeric_limits<TestUInt>::max())
+        EXPECT_EQ(TestDetect::Wrap::OVERFLOW, status);
+    else EXPECT_EQ(TestDetect::Wrap::NORMAL, status);
+}
+
+// Test this utility on every combination of prior and update value for the
+// type uint8_t, with an unsigned containing type.
+TEST(wraparound_tests, cover_u8_u64) {
+    using TType = uint8_t;
+    using CType = uint64_t;
+    static constexpr CType max = std::numeric_limits<TType>::max();
+    for (CType i = 0; i <= max; i++) {
+        for (CType j = 0; j <= max; j++) {
+            testPair<TType, CType>(i, static_cast<int64_t>(j));
+        }
+    }
+}
+
+// Test this utility on every combination of prior and update value for the
+// type uint8_t, with a signed containing type.
+TEST(wraparound_tests, cover_u8_s64) {
+    using TType = uint8_t;
+    using CType = int64_t;
+    static constexpr CType max = std::numeric_limits<TType>::max();
+    for (CType i = 0; i <= max; i++) {
+        for (CType j = 0; j <= max; j++) {
+            testPair<TType, CType>(i, static_cast<int64_t>(j));
+        }
+    }
+}
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 894b31c..5661206 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -654,6 +654,7 @@
 void AudioFlinger::onExternalVibrationStop(const sp<os::ExternalVibration>& externalVibration) {
     sp<os::IExternalVibratorService> evs = getExternalVibratorService();
     if (evs != 0) {
+        ALOGD("%s, stopping external vibration", __func__);
         evs->onExternalVibrationStop(*externalVibration);
     }
 }
@@ -765,6 +766,11 @@
                             (uint32_t)(mStandbyTimeInNsecs / 1000000));
     result.append(buffer);
     write(fd, result.string(), result.size());
+
+    dprintf(fd, "Vibrator infos(size=%zu):\n", mAudioVibratorInfos.size());
+    for (const auto& vibratorInfo : mAudioVibratorInfos) {
+        dprintf(fd, "  - %s\n", vibratorInfo.toString().c_str());
+    }
 }
 
 void AudioFlinger::dumpPermissionDenial(int fd, const Vector<String16>& args __unused)
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index e6d7cf7..72c378d 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -569,7 +569,8 @@
       mMaxDisableWaitCnt(1), // set by configure(), should be >= 1
       mDisableWaitCnt(0),    // set by process() and updateState()
       mOffloaded(false),
-      mAddedToHal(false)
+      mAddedToHal(false),
+      mIsOutput(false)
 #ifdef FLOAT_EFFECT_CHAIN
       , mSupportsFloat(false)
 #endif
@@ -962,6 +963,7 @@
     mConfig.outputCfg.mask = EFFECT_CONFIG_ALL;
     mConfig.inputCfg.buffer.frameCount = callback->frameCount();
     mConfig.outputCfg.buffer.frameCount = mConfig.inputCfg.buffer.frameCount;
+    mIsOutput = callback->isOutput();
 
     ALOGV("configure() %p chain %p buffer %p framecount %zu",
           this, callback->chain().promote().get(),
@@ -980,7 +982,7 @@
 
 #ifdef MULTICHANNEL_EFFECT_CHAIN
     if (status != NO_ERROR &&
-            callback->isOutput() &&
+            mIsOutput &&
             (mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO
                     || mConfig.outputCfg.channels != AUDIO_CHANNEL_OUT_STEREO)) {
         // Older effects may require exact STEREO position mask.
@@ -1643,6 +1645,22 @@
     return status;
 }
 
+status_t AudioFlinger::EffectModule::getConfigs(
+        audio_config_base_t* inputCfg, audio_config_base_t* outputCfg, bool* isOutput) const {
+    Mutex::Autolock _l(mLock);
+    if (mConfig.inputCfg.mask == 0 || mConfig.outputCfg.mask == 0) {
+        return NO_INIT;
+    }
+    inputCfg->sample_rate = mConfig.inputCfg.samplingRate;
+    inputCfg->channel_mask = static_cast<audio_channel_mask_t>(mConfig.inputCfg.channels);
+    inputCfg->format = static_cast<audio_format_t>(mConfig.inputCfg.format);
+    outputCfg->sample_rate = mConfig.outputCfg.samplingRate;
+    outputCfg->channel_mask = static_cast<audio_channel_mask_t>(mConfig.outputCfg.channels);
+    outputCfg->format = static_cast<audio_format_t>(mConfig.outputCfg.format);
+    *isOutput = mIsOutput;
+    return NO_ERROR;
+}
+
 static std::string dumpInOutBuffer(bool isInput, const sp<EffectBufferHalInterface> &buffer) {
     std::stringstream ss;
 
@@ -1760,6 +1778,7 @@
 BINDER_METHOD_ENTRY(command) \
 BINDER_METHOD_ENTRY(disconnect) \
 BINDER_METHOD_ENTRY(getCblk) \
+BINDER_METHOD_ENTRY(getConfig) \
 
 // singleton for Binder Method Statistics for IEffect
 mediautils::MethodStatistics<int>& getIEffectStatistics() {
@@ -1803,6 +1822,13 @@
   *_aidl_return = (code); \
   return Status::ok();
 
+#define VALUE_OR_RETURN_STATUS_AS_OUT(exp)              \
+    ({                                                  \
+        auto _tmp = (exp);                              \
+        if (!_tmp.ok()) { RETURN(_tmp.error()); }       \
+        std::move(_tmp.value());                        \
+    })
+
 Status AudioFlinger::EffectHandle::enable(int32_t* _aidl_return)
 {
     AutoMutex _l(mLock);
@@ -1914,6 +1940,32 @@
     return Status::ok();
 }
 
+Status AudioFlinger::EffectHandle::getConfig(
+        media::EffectConfig* _config, int32_t* _aidl_return) {
+    AutoMutex _l(mLock);
+    sp<EffectBase> effect = mEffect.promote();
+    if (effect == nullptr || mDisconnected) {
+        RETURN(DEAD_OBJECT);
+    }
+    sp<EffectModule> effectModule = effect->asEffectModule();
+    if (effectModule == nullptr) {
+        RETURN(INVALID_OPERATION);
+    }
+    audio_config_base_t inputCfg = AUDIO_CONFIG_BASE_INITIALIZER;
+    audio_config_base_t outputCfg = AUDIO_CONFIG_BASE_INITIALIZER;
+    bool isOutput;
+    status_t status = effectModule->getConfigs(&inputCfg, &outputCfg, &isOutput);
+    if (status == NO_ERROR) {
+        constexpr bool isInput = false; // effects always use 'OUT' channel masks.
+        _config->inputCfg = VALUE_OR_RETURN_STATUS_AS_OUT(
+                legacy2aidl_audio_config_base_t_AudioConfigBase(inputCfg, isInput));
+        _config->outputCfg = VALUE_OR_RETURN_STATUS_AS_OUT(
+                legacy2aidl_audio_config_base_t_AudioConfigBase(outputCfg, isInput));
+        _config->isOnInputStream = !isOutput;
+    }
+    RETURN(status);
+}
+
 Status AudioFlinger::EffectHandle::command(int32_t cmdCode,
                        const std::vector<uint8_t>& cmdData,
                        int32_t maxResponseSize,
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index 42614cc..a89a814 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -283,6 +283,10 @@
     status_t         setHapticIntensity(int id, int intensity);
     status_t         setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo);
 
+    status_t         getConfigs(audio_config_base_t* inputCfg,
+                                audio_config_base_t* outputCfg,
+                                bool* isOutput) const;
+
     void             dump(int fd, const Vector<String16>& args);
 
 private:
@@ -314,6 +318,7 @@
     uint32_t mDisableWaitCnt;       // current process() calls count during disable period.
     bool     mOffloaded;            // effect is currently offloaded to the audio DSP
     bool     mAddedToHal;           // effect has been added to the audio HAL
+    bool     mIsOutput;             // direction of the AF thread
 
 #ifdef FLOAT_EFFECT_CHAIN
     bool    mSupportsFloat;         // effect supports float processing
@@ -370,6 +375,8 @@
                                     int32_t* _aidl_return) override;
     android::binder::Status disconnect() override;
     android::binder::Status getCblk(media::SharedFileRegion* _aidl_return) override;
+    android::binder::Status getConfig(media::EffectConfig* _config,
+                                      int32_t* _aidl_return) override;
 
     sp<Client> client() const { return mClient; }
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 56ebb6e..2d00dbe 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -10248,20 +10248,10 @@
         } else {
             sp<MmapStreamCallback> callback = mCallback.promote();
             if (callback != 0) {
-                int channelCount;
-                if (isOutput()) {
-                    channelCount = audio_channel_count_from_out_mask(mChannelMask);
-                } else {
-                    channelCount = audio_channel_count_from_in_mask(mChannelMask);
-                }
-                Vector<float> values;
-                for (int i = 0; i < channelCount; i++) {
-                    values.add(volume);
-                }
                 mHalVolFloat = volume; // SW volume control worked, so update value.
                 mNoCallbackWarningCount = 0;
                 mLock.unlock();
-                callback->onVolumeChanged(mChannelMask, values);
+                callback->onVolumeChanged(volume);
                 mLock.lock();
             } else {
                 if (mNoCallbackWarningCount < kMaxNoCallbackWarnings) {
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 6135020..a9720da 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1908,6 +1908,7 @@
         PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
         if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
                 && playbackThread->mHapticChannelCount > 0) {
+            ALOGD("%s, haptic playback was muted for track %d", __func__, mTrack->id());
             mTrack->setHapticPlaybackEnabled(false);
             *ret = true;
         }
@@ -1925,6 +1926,7 @@
         PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
         if ((mTrack->channelMask() & AUDIO_CHANNEL_HAPTIC_ALL) != AUDIO_CHANNEL_NONE
                 && playbackThread->mHapticChannelCount > 0) {
+            ALOGD("%s, haptic playback was unmuted for track %d", __func__, mTrack->id());
             mTrack->setHapticPlaybackEnabled(true);
             *ret = true;
         }
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 744609f..84a015b 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -2595,31 +2595,19 @@
         flags = (audio_input_flags_t)(flags | AUDIO_INPUT_FLAG_ULTRASOUND);
     }
 
-    // find a compatible input profile (not necessarily identical in parameters)
-    sp<IOProfile> profile;
     // sampling rate and flags may be updated by getInputProfile
     uint32_t profileSamplingRate = (config->sample_rate == 0) ?
             SAMPLE_RATE_HZ_DEFAULT : config->sample_rate;
-    audio_format_t profileFormat;
+    audio_format_t profileFormat = config->format;
     audio_channel_mask_t profileChannelMask = config->channel_mask;
     audio_input_flags_t profileFlags = flags;
-    for (;;) {
-        profileFormat = config->format; // reset each time through loop, in case it is updated
-        profile = getInputProfile(device, profileSamplingRate, profileFormat, profileChannelMask,
-                                  profileFlags);
-        if (profile != 0) {
-            break; // success
-        } else if (profileFlags & AUDIO_INPUT_FLAG_RAW) {
-            profileFlags = (audio_input_flags_t) (profileFlags & ~AUDIO_INPUT_FLAG_RAW); // retry
-        } else if (profileFlags != AUDIO_INPUT_FLAG_NONE && audio_is_linear_pcm(config->format)) {
-            profileFlags = AUDIO_INPUT_FLAG_NONE; // retry
-        } else { // fail
-            ALOGW("%s could not find profile for device %s, sampling rate %u, format %#x, "
-                  "channel mask 0x%X, flags %#x", __func__, device->toString().c_str(),
-                  config->sample_rate, config->format, config->channel_mask, flags);
-            return input;
-        }
+    // find a compatible input profile (not necessarily identical in parameters)
+    sp<IOProfile> profile = getInputProfile(
+            device, profileSamplingRate, profileFormat, profileChannelMask, profileFlags);
+    if (profile == nullptr) {
+        return input;
     }
+
     // Pick input sampling rate if not specified by client
     uint32_t samplingRate = config->sample_rate;
     if (samplingRate == 0) {
@@ -7094,51 +7082,68 @@
 {
     // Choose an input profile based on the requested capture parameters: select the first available
     // profile supporting all requested parameters.
+    // The flags can be ignored if it doesn't contain a much match flag.
     //
     // TODO: perhaps isCompatibleProfile should return a "matching" score so we can return
     // the best matching profile, not the first one.
 
-    sp<IOProfile> firstInexact;
-    uint32_t updatedSamplingRate = 0;
-    audio_format_t updatedFormat = AUDIO_FORMAT_INVALID;
-    audio_channel_mask_t updatedChannelMask = AUDIO_CHANNEL_INVALID;
-    for (const auto& hwModule : mHwModules) {
-        for (const auto& profile : hwModule->getInputProfiles()) {
-            // profile->log();
-            //updatedFormat = format;
-            if (profile->isCompatibleProfile(DeviceVector(device), samplingRate,
-                                             &samplingRate  /*updatedSamplingRate*/,
-                                             format,
-                                             &format,       /*updatedFormat*/
-                                             channelMask,
-                                             &channelMask   /*updatedChannelMask*/,
-                                             // FIXME ugly cast
-                                             (audio_output_flags_t) flags,
-                                             true /*exactMatchRequiredForInputFlags*/)) {
-                return profile;
-            }
-            if (firstInexact == nullptr && profile->isCompatibleProfile(DeviceVector(device),
-                                             samplingRate,
-                                             &updatedSamplingRate,
-                                             format,
-                                             &updatedFormat,
-                                             channelMask,
-                                             &updatedChannelMask,
-                                             // FIXME ugly cast
-                                             (audio_output_flags_t) flags,
-                                             false /*exactMatchRequiredForInputFlags*/)) {
-                firstInexact = profile;
-            }
+    const audio_input_flags_t mustMatchFlag = AUDIO_INPUT_FLAG_MMAP_NOIRQ;
+    const audio_input_flags_t oriFlags = flags;
 
+    for (;;) {
+        sp<IOProfile> firstInexact = nullptr;
+        uint32_t updatedSamplingRate = 0;
+        audio_format_t updatedFormat = AUDIO_FORMAT_INVALID;
+        audio_channel_mask_t updatedChannelMask = AUDIO_CHANNEL_INVALID;
+        for (const auto& hwModule : mHwModules) {
+            for (const auto& profile : hwModule->getInputProfiles()) {
+                // profile->log();
+                //updatedFormat = format;
+                if (profile->isCompatibleProfile(DeviceVector(device), samplingRate,
+                                                 &samplingRate  /*updatedSamplingRate*/,
+                                                 format,
+                                                 &format,       /*updatedFormat*/
+                                                 channelMask,
+                                                 &channelMask   /*updatedChannelMask*/,
+                                                 // FIXME ugly cast
+                                                 (audio_output_flags_t) flags,
+                                                 true /*exactMatchRequiredForInputFlags*/)) {
+                    return profile;
+                }
+                if (firstInexact == nullptr && profile->isCompatibleProfile(DeviceVector(device),
+                                                 samplingRate,
+                                                 &updatedSamplingRate,
+                                                 format,
+                                                 &updatedFormat,
+                                                 channelMask,
+                                                 &updatedChannelMask,
+                                                 // FIXME ugly cast
+                                                 (audio_output_flags_t) flags,
+                                                 false /*exactMatchRequiredForInputFlags*/)) {
+                    firstInexact = profile;
+                }
+            }
+        }
+
+        if (firstInexact != nullptr) {
+            samplingRate = updatedSamplingRate;
+            format = updatedFormat;
+            channelMask = updatedChannelMask;
+            return firstInexact;
+        } else if (flags & AUDIO_INPUT_FLAG_RAW) {
+            flags = (audio_input_flags_t) (flags & ~AUDIO_INPUT_FLAG_RAW); // retry
+        } else if ((flags & mustMatchFlag) == AUDIO_INPUT_FLAG_NONE &&
+                flags != AUDIO_INPUT_FLAG_NONE && audio_is_linear_pcm(format)) {
+            flags = AUDIO_INPUT_FLAG_NONE;
+        } else { // fail
+            ALOGW("%s could not find profile for device %s, sampling rate %u, format %#x, "
+                  "channel mask 0x%X, flags %#x", __func__, device->toString().c_str(),
+                  samplingRate, format, channelMask, oriFlags);
+            break;
         }
     }
-    if (firstInexact != nullptr) {
-        samplingRate = updatedSamplingRate;
-        format = updatedFormat;
-        channelMask = updatedChannelMask;
-        return firstInexact;
-    }
-    return NULL;
+
+    return nullptr;
 }
 
 float AudioPolicyManager::computeVolume(IVolumeCurves &curves,
diff --git a/services/audiopolicy/service/AudioPolicyEffects.cpp b/services/audiopolicy/service/AudioPolicyEffects.cpp
index 70fdfcb..c7a60c2 100644
--- a/services/audiopolicy/service/AudioPolicyEffects.cpp
+++ b/services/audiopolicy/service/AudioPolicyEffects.cpp
@@ -127,7 +127,8 @@
             attributionSource.packageName = "android";
             attributionSource.token = sp<BBinder>::make();
             sp<AudioEffect> fx = new AudioEffect(attributionSource);
-            fx->set(NULL, &effect->mUuid, -1, 0, 0, audioSession, input);
+            fx->set(nullptr /*type */, &effect->mUuid, -1 /* priority */, nullptr /* callback */,
+                    audioSession, input);
             status_t status = fx->initCheck();
             if (status != NO_ERROR && status != ALREADY_EXISTS) {
                 ALOGW("addInputEffects(): failed to create Fx %s on source %d",
@@ -279,7 +280,8 @@
             attributionSource.packageName = "android";
             attributionSource.token = sp<BBinder>::make();
             sp<AudioEffect> fx = new AudioEffect(attributionSource);
-            fx->set(NULL, &effect->mUuid, 0, 0, 0, audioSession, output);
+            fx->set(nullptr /* type */, &effect->mUuid, 0 /* priority */, nullptr /* callback */,
+                    audioSession, output);
             status_t status = fx->initCheck();
             if (status != NO_ERROR && status != ALREADY_EXISTS) {
                 ALOGE("addOutputSessionEffects(): failed to create Fx  %s on session %d",
@@ -984,8 +986,8 @@
             attributionSource.packageName = "android";
             attributionSource.token = sp<BBinder>::make();
             sp<AudioEffect> fx = new AudioEffect(attributionSource);
-            fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0, nullptr,
-                    nullptr, AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
+            fx->set(EFFECT_UUID_NULL, &effectDesc->mUuid, 0 /* priority */, nullptr /* callback */,
+                    AUDIO_SESSION_DEVICE, AUDIO_IO_HANDLE_NONE,
                     AudioDeviceTypeAddr{deviceEffects->getDeviceType(),
                                         deviceEffects->getDeviceAddress()});
             status_t status = fx->initCheck();
diff --git a/services/audiopolicy/service/Spatializer.cpp b/services/audiopolicy/service/Spatializer.cpp
index c199a76..a98d474 100644
--- a/services/audiopolicy/service/Spatializer.cpp
+++ b/services/audiopolicy/service/Spatializer.cpp
@@ -718,9 +718,10 @@
         // create FX instance on output
         AttributionSourceState attributionSource = AttributionSourceState();
         mEngine = new AudioEffect(attributionSource);
-        mEngine->set(nullptr, &mEngineDescriptor.uuid, 0, Spatializer::engineCallback /* cbf */,
-                     this /* user */, AUDIO_SESSION_OUTPUT_STAGE, output, {} /* device */,
-                     false /* probe */, true /* notifyFramesProcessed */);
+        mEngine->set(nullptr /* type */, &mEngineDescriptor.uuid, 0 /* priority */,
+                     wp<AudioEffect::IAudioEffectCallback>::fromExisting(this),
+                     AUDIO_SESSION_OUTPUT_STAGE, output, {} /* device */, false /* probe */,
+                     true /* notifyFramesProcessed */);
         status_t status = mEngine->initCheck();
         ALOGV("%s mEngine create status %d", __func__, (int)status);
         if (status != NO_ERROR) {
@@ -837,27 +838,10 @@
     }
 }
 
-void Spatializer::engineCallback(int32_t event, void *user, void *info) {
-    if (user == nullptr) {
-        return;
-    }
-    Spatializer* const me = reinterpret_cast<Spatializer *>(user);
-    switch (event) {
-        case AudioEffect::EVENT_FRAMES_PROCESSED: {
-            int frames = info == nullptr ? 0 : *(int*)info;
-            ALOGV("%s frames processed %d for me %p", __func__, frames, me);
-            me->postFramesProcessedMsg(frames);
-        } break;
-        default:
-            ALOGV("%s event %d", __func__, event);
-            break;
-    }
-}
-
-void Spatializer::postFramesProcessedMsg(int frames) {
+void Spatializer::onFramesProcessed(int32_t framesProcessed) {
     sp<AMessage> msg =
             new AMessage(EngineCallbackHandler::kWhatOnFramesProcessed, mHandler);
-    msg->setInt32(EngineCallbackHandler::kNumFramesKey, frames);
+    msg->setInt32(EngineCallbackHandler::kNumFramesKey, framesProcessed);
     msg->post();
 }
 
diff --git a/services/audiopolicy/service/Spatializer.h b/services/audiopolicy/service/Spatializer.h
index 29f4b08..ad45fa9 100644
--- a/services/audiopolicy/service/Spatializer.h
+++ b/services/audiopolicy/service/Spatializer.h
@@ -84,6 +84,7 @@
  * spatializer mixer thread is destroyed.
  */
 class Spatializer : public media::BnSpatializer,
+                    public AudioEffect::IAudioEffectCallback,
                     public IBinder::DeathRecipient,
                     private SpatializerPoseController::Listener {
   public:
@@ -274,7 +275,7 @@
         return NO_ERROR;
     }
 
-    void postFramesProcessedMsg(int frames);
+    virtual void onFramesProcessed(int32_t framesProcessed) override;
 
     /**
      * Checks if head and screen sensors must be actively monitored based on
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index c4b5a6c..9aa6b82 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -269,7 +269,10 @@
                     cameraId.c_str());
             continue;
         }
-        i->getListener()->onTorchStatusChanged(mapToInterface(status), String16{cameraId});
+        auto ret = i->getListener()->onTorchStatusChanged(mapToInterface(status),
+                String16{cameraId});
+        i->handleBinderStatus(ret, "%s: Failed to trigger onTorchStatusChanged for %d:%d: %d",
+                __FUNCTION__, i->getListenerUid(), i->getListenerPid(), ret.exceptionCode());
     }
 }
 
@@ -538,8 +541,12 @@
                         id.c_str());
                 continue;
             }
-            listener->getListener()->onPhysicalCameraStatusChanged(mapToInterface(newStatus),
-                    id16, physicalId16);
+            auto ret = listener->getListener()->onPhysicalCameraStatusChanged(
+                    mapToInterface(newStatus), id16, physicalId16);
+            listener->handleBinderStatus(ret,
+                    "%s: Failed to trigger onPhysicalCameraStatusChanged for %d:%d: %d",
+                    __FUNCTION__, listener->getListenerUid(), listener->getListenerPid(),
+                    ret.exceptionCode());
         }
     }
 }
@@ -580,8 +587,11 @@
         int32_t newStrengthLevel) {
     Mutex::Autolock lock(mStatusListenerLock);
     for (auto& i : mListenerList) {
-        i->getListener()->onTorchStrengthLevelChanged(String16{cameraId},
+        auto ret = i->getListener()->onTorchStrengthLevelChanged(String16{cameraId},
                 newStrengthLevel);
+        i->handleBinderStatus(ret,
+                "%s: Failed to trigger onTorchStrengthLevelChanged for %d:%d: %d", __FUNCTION__,
+                i->getListenerUid(), i->getListenerPid(), ret.exceptionCode());
     }
 }
 
@@ -1890,6 +1900,9 @@
             }
         }
 
+        // Enable/disable camera service watchdog
+        client->setCameraServiceWatchdog(mCameraServiceWatchdogEnabled);
+
         // Set rotate-and-crop override behavior
         if (mOverrideRotateAndCropMode != ANDROID_SCALER_ROTATE_AND_CROP_AUTO) {
             client->setRotateAndCropOverride(mOverrideRotateAndCropMode);
@@ -2371,10 +2384,8 @@
 
     for (const auto& it : mListenerList) {
         auto ret = it->getListener()->onCameraAccessPrioritiesChanged();
-        if (!ret.isOk()) {
-            ALOGE("%s: Failed to trigger permission callback: %d", __FUNCTION__,
-                    ret.exceptionCode());
-        }
+        it->handleBinderStatus(ret, "%s: Failed to trigger permission callback for %d:%d: %d",
+                __FUNCTION__, it->getListenerUid(), it->getListenerPid(), ret.exceptionCode());
     }
 }
 
@@ -4633,8 +4644,12 @@
                             cameraId.c_str());
                     continue;
                 }
-                listener->getListener()->onStatusChanged(mapToInterface(status),
+                auto ret = listener->getListener()->onStatusChanged(mapToInterface(status),
                         String16(cameraId));
+                listener->handleBinderStatus(ret,
+                        "%s: Failed to trigger onStatusChanged callback for %d:%d: %d",
+                        __FUNCTION__, listener->getListenerUid(), listener->getListenerPid(),
+                        ret.exceptionCode());
             }
         });
 }
@@ -4667,10 +4682,10 @@
         } else {
             ret = it->getListener()->onCameraClosed(cameraId64);
         }
-        if (!ret.isOk()) {
-            ALOGE("%s: Failed to trigger onCameraOpened/onCameraClosed callback: %d", __FUNCTION__,
-                    ret.exceptionCode());
-        }
+
+        it->handleBinderStatus(ret,
+                "%s: Failed to trigger onCameraOpened/onCameraClosed callback for %d:%d: %d",
+                __FUNCTION__, it->getListenerUid(), it->getListenerPid(), ret.exceptionCode());
     }
 }
 
@@ -4771,8 +4786,12 @@
                         String8(physicalCameraId).c_str());
                 continue;
             }
-            listener->getListener()->onPhysicalCameraStatusChanged(status,
+            auto ret = listener->getListener()->onPhysicalCameraStatusChanged(status,
                     logicalCameraId, physicalCameraId);
+            listener->handleBinderStatus(ret,
+                    "%s: Failed to trigger onPhysicalCameraStatusChanged for %d:%d: %d",
+                    __FUNCTION__, listener->getListenerUid(), listener->getListenerPid(),
+                    ret.exceptionCode());
         }
     }
 }
@@ -4828,6 +4847,8 @@
         return handleSetCameraMute(args);
     } else if (args.size() >= 2 && args[0] == String16("watch")) {
         return handleWatchCommand(args, in, out);
+    } else if (args.size() >= 2 && args[0] == String16("set-watchdog")) {
+        return handleSetCameraServiceWatchdog(args);
     } else if (args.size() == 1 && args[0] == String16("help")) {
         printHelp(out);
         return OK;
@@ -4921,6 +4942,28 @@
     return OK;
 }
 
+status_t CameraService::handleSetCameraServiceWatchdog(const Vector<String16>& args) {
+    int enableWatchdog = atoi(String8(args[1]));
+
+    if (enableWatchdog < 0 || enableWatchdog > 1) return BAD_VALUE;
+
+    Mutex::Autolock lock(mServiceLock);
+
+    mCameraServiceWatchdogEnabled = enableWatchdog;
+
+    const auto clients = mActiveClientManager.getAll();
+    for (auto& current : clients) {
+        if (current != nullptr) {
+            const auto basicClient = current->getValue();
+            if (basicClient.get() != nullptr) {
+                basicClient->setCameraServiceWatchdog(enableWatchdog);
+            }
+        }
+    }
+
+    return OK;
+}
+
 status_t CameraService::handleGetRotateAndCrop(int out) {
     Mutex::Autolock lock(mServiceLock);
 
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 89a537d..30c00a5 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -338,6 +338,9 @@
         // Set/reset camera mute
         virtual status_t setCameraMute(bool enabled) = 0;
 
+        // Set Camera service watchdog
+        virtual status_t setCameraServiceWatchdog(bool enabled) = 0;
+
         // The injection camera session to replace the internal camera
         // session.
         virtual status_t injectCamera(const String8& injectedCamId,
@@ -1054,6 +1057,29 @@
                 return IInterface::asBinder(mListener)->linkToDeath(this);
             }
 
+            template<typename... args_t>
+            void handleBinderStatus(const binder::Status &ret, const char *logOnError,
+                    args_t... args) {
+                if (!ret.isOk() &&
+                        (ret.exceptionCode() != binder::Status::Exception::EX_TRANSACTION_FAILED
+                        || !mLastTransactFailed)) {
+                    ALOGE(logOnError, args...);
+                }
+
+                // If the transaction failed, the process may have died (or other things, see
+                // b/28321379). Mute consecutive errors from this listener to avoid log spam.
+                if (ret.exceptionCode() == binder::Status::Exception::EX_TRANSACTION_FAILED) {
+                    if (!mLastTransactFailed) {
+                        ALOGE("%s: Muting similar errors from listener %d:%d", __FUNCTION__,
+                                mListenerUid, mListenerPid);
+                    }
+                    mLastTransactFailed = true;
+                } else {
+                    // Reset mLastTransactFailed when binder becomes healthy again.
+                    mLastTransactFailed = false;
+                }
+            }
+
             virtual void binderDied(const wp<IBinder> &/*who*/) {
                 auto parent = mParent.promote();
                 if (parent.get() != nullptr) {
@@ -1074,6 +1100,9 @@
             int mListenerPid = -1;
             bool mIsVendorListener = false;
             bool mOpenCloseCallbackAllowed = false;
+
+            // Flag for preventing log spam when binder becomes unhealthy
+            bool mLastTransactFailed = false;
     };
 
     // Guarded by mStatusListenerMutex
@@ -1197,6 +1226,9 @@
     // Handle 'watch' command as passed through 'cmd'
     status_t handleWatchCommand(const Vector<String16> &args, int inFd, int outFd);
 
+    // Set the camera service watchdog
+    status_t handleSetCameraServiceWatchdog(const Vector<String16>& args);
+
     // Enable tag monitoring of the given tags in provided clients
     status_t startWatchingTags(const Vector<String16> &args, int outFd);
 
@@ -1282,6 +1314,9 @@
     // Current camera mute mode
     bool mOverrideCameraMuteMode = false;
 
+    // Camera Service watchdog flag
+    bool mCameraServiceWatchdogEnabled = true;
+
     /**
      * A listener class that implements the IBinder::DeathRecipient interface
      * for use to call back the error state injected by the external camera, and
diff --git a/services/camera/libcameraservice/CameraServiceWatchdog.cpp b/services/camera/libcameraservice/CameraServiceWatchdog.cpp
index fcd6ebe..950da6a 100644
--- a/services/camera/libcameraservice/CameraServiceWatchdog.cpp
+++ b/services/camera/libcameraservice/CameraServiceWatchdog.cpp
@@ -64,6 +64,17 @@
     }
 }
 
+void CameraServiceWatchdog::setEnabled(bool enable)
+{
+    AutoMutex _l(mEnabledLock);
+
+    if (enable) {
+        mEnabled = true;
+    } else {
+        mEnabled = false;
+    }
+}
+
 void CameraServiceWatchdog::stop(uint32_t tid)
 {
     AutoMutex _l(mWatchdogLock);
diff --git a/services/camera/libcameraservice/CameraServiceWatchdog.h b/services/camera/libcameraservice/CameraServiceWatchdog.h
index f4955e2..f7c90a9 100644
--- a/services/camera/libcameraservice/CameraServiceWatchdog.h
+++ b/services/camera/libcameraservice/CameraServiceWatchdog.h
@@ -21,10 +21,12 @@
  * expected duration has exceeded.
  * Notes on multi-threaded behaviors:
  *    - The threadloop is blocked/paused when there are no calls being
- *   monitored.
+ *   monitored (when the TID cycle to counter map is empty).
  *   - The start and stop functions handle simultaneous call monitoring
  *   and single call monitoring differently. See function documentation for
  *   more details.
+ * To disable/enable:
+ *   - adb shell cmd media.camera set-cameraservice-watchdog [0/1]
  */
 
 #include <chrono>
@@ -49,15 +51,19 @@
 
 public:
     explicit CameraServiceWatchdog() : mPause(true), mMaxCycles(kMaxCycles),
-            mCycleLengthMs(kCycleLengthMs) {};
+            mCycleLengthMs(kCycleLengthMs), mEnabled(true) {};
 
-    explicit CameraServiceWatchdog (size_t maxCycles, uint32_t cycleLengthMs) :
-            mPause(true), mMaxCycles(maxCycles), mCycleLengthMs(cycleLengthMs) {};
+    explicit CameraServiceWatchdog (size_t maxCycles, uint32_t cycleLengthMs, bool enabled) :
+            mPause(true), mMaxCycles(maxCycles), mCycleLengthMs(cycleLengthMs), mEnabled(enabled)
+                    {};
 
     virtual ~CameraServiceWatchdog() {};
 
     virtual void requestExit();
 
+    /** Enables/disables the watchdog */
+    void setEnabled(bool enable);
+
     /** Used to wrap monitored calls in start and stop functions using custom timer values */
     template<typename T>
     auto watchThread(T func, uint32_t tid, uint32_t cycles, uint32_t cycleLength) {
@@ -66,8 +72,13 @@
         if (cycles != mMaxCycles || cycleLength != mCycleLengthMs) {
             // Create another instance of the watchdog to prevent disruption
             // of timer for current monitored calls
+
+            // Lock for mEnabled
+            mEnabledLock.lock();
             sp<CameraServiceWatchdog> tempWatchdog =
-                    new CameraServiceWatchdog(cycles, cycleLength);
+                    new CameraServiceWatchdog(cycles, cycleLength, mEnabled);
+            mEnabledLock.unlock();
+
             tempWatchdog->run("CameraServiceWatchdog");
             res = tempWatchdog->watchThread(func, tid);
             tempWatchdog->requestExit();
@@ -84,11 +95,17 @@
     /** Used to wrap monitored calls in start and stop functions using class timer values */
     template<typename T>
     auto watchThread(T func, uint32_t tid) {
+        AutoMutex _l(mEnabledLock);
+
         auto res = NULL;
 
-        start(tid);
-        res = func();
-        stop(tid);
+        if (mEnabled) {
+            start(tid);
+            res = func();
+            stop(tid);
+        } else {
+            res = func();
+        }
 
         return res;
     }
@@ -109,11 +126,13 @@
 
     virtual bool    threadLoop();
 
-    Mutex           mWatchdogLock;        // Lock for condition variable
-    Condition       mWatchdogCondition;   // Condition variable for stop/start
-    bool            mPause;               // True if thread is currently paused
-    uint32_t        mMaxCycles;           // Max cycles
-    uint32_t        mCycleLengthMs;       // Length of time elapsed per cycle
+    Mutex           mWatchdogLock;      // Lock for condition variable
+    Mutex           mEnabledLock;       // Lock for enabled status
+    Condition       mWatchdogCondition; // Condition variable for stop/start
+    bool            mPause;             // True if tid map is empty
+    uint32_t        mMaxCycles;         // Max cycles
+    uint32_t        mCycleLengthMs;     // Length of time elapsed per cycle
+    bool            mEnabled;           // True if watchdog is enabled
 
     std::unordered_map<uint32_t, uint32_t> tidToCycleCounterMap; // Thread Id to cycle counter map
 };
diff --git a/services/camera/libcameraservice/api1/Camera2Client.cpp b/services/camera/libcameraservice/api1/Camera2Client.cpp
index a6a89af..74806a4 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.cpp
+++ b/services/camera/libcameraservice/api1/Camera2Client.cpp
@@ -2313,6 +2313,10 @@
     return INVALID_OPERATION;
 }
 
+status_t Camera2Client::setCameraServiceWatchdog(bool enabled) {
+    return mDevice->setCameraServiceWatchdog(enabled);
+}
+
 status_t Camera2Client::setRotateAndCropOverride(uint8_t rotateAndCrop) {
     if (rotateAndCrop > ANDROID_SCALER_ROTATE_AND_CROP_AUTO) return BAD_VALUE;
 
diff --git a/services/camera/libcameraservice/api1/Camera2Client.h b/services/camera/libcameraservice/api1/Camera2Client.h
index 58e0e19..da49f56 100644
--- a/services/camera/libcameraservice/api1/Camera2Client.h
+++ b/services/camera/libcameraservice/api1/Camera2Client.h
@@ -90,6 +90,8 @@
     virtual bool            supportsCameraMute();
     virtual status_t        setCameraMute(bool enabled);
 
+    virtual status_t        setCameraServiceWatchdog(bool enabled);
+
     /**
      * Interface used by CameraService
      */
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
index 5e91501..15df981 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.cpp
@@ -1730,6 +1730,10 @@
     return binder::Status::ok();
 }
 
+status_t CameraDeviceClient::setCameraServiceWatchdog(bool enabled) {
+    return mDevice->setCameraServiceWatchdog(enabled);
+}
+
 status_t CameraDeviceClient::setRotateAndCropOverride(uint8_t rotateAndCrop) {
     if (rotateAndCrop > ANDROID_SCALER_ROTATE_AND_CROP_AUTO) return BAD_VALUE;
 
diff --git a/services/camera/libcameraservice/api2/CameraDeviceClient.h b/services/camera/libcameraservice/api2/CameraDeviceClient.h
index c5aad6b..45915ba 100644
--- a/services/camera/libcameraservice/api2/CameraDeviceClient.h
+++ b/services/camera/libcameraservice/api2/CameraDeviceClient.h
@@ -206,6 +206,8 @@
     virtual status_t      stopWatchingTags(int out);
     virtual status_t      dumpWatchedEventsToVector(std::vector<std::string> &out);
 
+    virtual status_t      setCameraServiceWatchdog(bool enabled);
+
     /**
      * Device listener interface
      */
diff --git a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp
index 9303fd2..beb655b 100644
--- a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp
@@ -66,6 +66,10 @@
     return OK;
 }
 
+status_t CameraOfflineSessionClient::setCameraServiceWatchdog(bool) {
+    return OK;
+}
+
 status_t CameraOfflineSessionClient::setRotateAndCropOverride(uint8_t /*rotateAndCrop*/) {
     // Since we're not submitting more capture requests, changes to rotateAndCrop override
     // make no difference.
diff --git a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h
index f2c42d8..9ea1093 100644
--- a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h
+++ b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h
@@ -84,6 +84,8 @@
     bool supportsCameraMute() override;
     status_t setCameraMute(bool enabled) override;
 
+    status_t setCameraServiceWatchdog(bool enabled) override;
+
     // permissions management
     status_t startCameraOps() override;
     status_t finishCameraOps() override;
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index 7e2f93c..69514f3 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -449,6 +449,11 @@
     virtual status_t setCameraMute(bool enabled) = 0;
 
     /**
+     * Enable/disable camera service watchdog
+     */
+    virtual status_t setCameraServiceWatchdog(bool enabled) = 0;
+
+    /**
      * Get the status tracker of the camera device
      */
     virtual wp<camera3::StatusTracker> getStatusTracker() = 0;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index dec439f..7f83849 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -113,10 +113,6 @@
 
 status_t Camera3Device::initializeCommonLocked() {
 
-    /** Start watchdog thread */
-    mCameraServiceWatchdog = new CameraServiceWatchdog();
-    mCameraServiceWatchdog->run("CameraServiceWatchdog");
-
     /** Start up status tracker thread */
     mStatusTracker = new StatusTracker(this);
     status_t res = mStatusTracker->run(String8::format("C3Dev-%s-Status", mId.string()).string());
@@ -230,6 +226,10 @@
     // Hidl/AidlCamera3DeviceInjectionMethods
     mInjectionMethods = createCamera3DeviceInjectionMethods(this);
 
+    /** Start watchdog thread */
+    mCameraServiceWatchdog = new CameraServiceWatchdog();
+    mCameraServiceWatchdog->run("CameraServiceWatchdog");
+
     return OK;
 }
 
@@ -4091,6 +4091,17 @@
     }
 }
 
+status_t Camera3Device::setCameraServiceWatchdog(bool enabled) {
+    Mutex::Autolock il(mInterfaceLock);
+    Mutex::Autolock l(mLock);
+
+    if (mCameraServiceWatchdog != NULL) {
+        mCameraServiceWatchdog->setEnabled(enabled);
+    }
+
+    return OK;
+}
+
 void Camera3Device::RequestThread::cleanUpFailedRequests(bool sendRequestError) {
     if (mNextRequests.empty()) {
         return;
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index f927b4d..3c5cb78 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -281,6 +281,11 @@
      */
     status_t setCameraMute(bool enabled);
 
+    /**
+     * Enables/disables camera service watchdog
+     */
+    status_t setCameraServiceWatchdog(bool enabled);
+
     // Get the status trackeer for the camera device
     wp<camera3::StatusTracker> getStatusTracker() { return mStatusTracker; }
 
diff --git a/services/camera/libcameraservice/libcameraservice_fuzzer/Android.bp b/services/camera/libcameraservice/libcameraservice_fuzzer/Android.bp
index 3c98a5e..1df730d 100644
--- a/services/camera/libcameraservice/libcameraservice_fuzzer/Android.bp
+++ b/services/camera/libcameraservice/libcameraservice_fuzzer/Android.bp
@@ -65,6 +65,7 @@
     fuzz_config: {
         cc: [
             "android-media-fuzzing-reports@google.com",
+            "android-camera-fwk-eng@google.com",
         ],
         componentid: 155276,
         libfuzzer_options: [
diff --git a/services/camera/libcameraservice/libcameraservice_fuzzer/camera_service_fuzzer.cpp b/services/camera/libcameraservice/libcameraservice_fuzzer/camera_service_fuzzer.cpp
index 97d7bf4..7a61f38 100644
--- a/services/camera/libcameraservice/libcameraservice_fuzzer/camera_service_fuzzer.cpp
+++ b/services/camera/libcameraservice/libcameraservice_fuzzer/camera_service_fuzzer.cpp
@@ -111,12 +111,15 @@
     size_t mPreviewBufferCount = 0;
     bool mAutoFocusMessage = false;
     bool mSnapshotNotification = false;
+    bool mRecordingNotification = false;
     mutable Mutex mPreviewLock;
     mutable Condition mPreviewCondition;
     mutable Mutex mAutoFocusLock;
     mutable Condition mAutoFocusCondition;
     mutable Mutex mSnapshotLock;
     mutable Condition mSnapshotCondition;
+    mutable Mutex mRecordingLock;
+    mutable Condition mRecordingCondition;
 
     void getNumCameras();
     void getCameraInformation(int32_t cameraId);
@@ -125,6 +128,7 @@
     void invokeDump();
     void invokeShellCommand();
     void invokeNotifyCalls();
+    void invokeTorchAPIs(int32_t cameraId);
 
     // CameraClient interface
     void notifyCallback(int32_t msgType, int32_t, int32_t) override;
@@ -152,6 +156,8 @@
             Mutex::Autolock l(mPreviewLock);
             ++mPreviewBufferCount;
             mPreviewCondition.broadcast();
+            mRecordingNotification = true;
+            mRecordingCondition.broadcast();
             break;
         }
         case CAMERA_MSG_COMPRESSED_IMAGE: {
@@ -311,114 +317,138 @@
     mCameraService->notifySystemEvent(eventId, args);
 }
 
+void CameraFuzzer::invokeTorchAPIs(int32_t cameraId) {
+    String16 cameraIdStr = String16(String8::format("%d", cameraId));
+    sp<IBinder> binder = new BBinder;
+
+    mCameraService->setTorchMode(cameraIdStr, true, binder);
+    ALOGV("Turned torch on.");
+    int32_t torchStrength = rand() % 5 + 1;
+    ALOGV("Changing torch strength level to %d", torchStrength);
+    mCameraService->turnOnTorchWithStrengthLevel(cameraIdStr, torchStrength, binder);
+    mCameraService->setTorchMode(cameraIdStr, false, binder);
+    ALOGV("Turned torch off.");
+}
+
 void CameraFuzzer::invokeCameraAPIs() {
-    for (int32_t cameraId = 0; cameraId < mNumCameras; ++cameraId) {
-        getCameraInformation(cameraId);
+    /** In order to avoid the timeout issue caused due to multiple iteration of loops, the 'for'
+     * loops are removed and the 'cameraId', 'pictureSize' and 'videoSize' are derived using the
+     * FuzzedDataProvider from the available cameras and vectors of 'pictureSizes' and 'videoSizes'
+     */
+    int32_t cameraId = mFuzzedDataProvider->ConsumeIntegralInRange<int32_t>(0, mNumCameras - 1);
+    getCameraInformation(cameraId);
+    invokeTorchAPIs(cameraId);
 
-        ::android::binder::Status rc;
-        sp<ICamera> cameraDevice;
+    ::android::binder::Status rc;
+    sp<ICamera> cameraDevice;
 
-        rc = mCameraService->connect(this, cameraId, String16(),
-                android::CameraService::USE_CALLING_UID, android::CameraService::USE_CALLING_PID,
-                /*targetSdkVersion*/__ANDROID_API_FUTURE__, &cameraDevice);
-        if (!rc.isOk()) {
-            // camera not connected
-            return;
+    rc = mCameraService->connect(this, cameraId, String16(),
+                                 android::CameraService::USE_CALLING_UID,
+                                 android::CameraService::USE_CALLING_PID,
+                                 /*targetSdkVersion*/ __ANDROID_API_FUTURE__, &cameraDevice);
+    if (!rc.isOk()) {
+        // camera not connected
+        return;
+    }
+    if (cameraDevice) {
+        sp<Surface> previewSurface;
+        sp<SurfaceControl> surfaceControl;
+        CameraParameters params(cameraDevice->getParameters());
+        String8 focusModes(params.get(CameraParameters::KEY_SUPPORTED_FOCUS_MODES));
+        bool isAFSupported = false;
+        const char* focusMode = nullptr;
+
+        if (focusModes.contains(CameraParameters::FOCUS_MODE_AUTO)) {
+            isAFSupported = true;
+        } else if (focusModes.contains(CameraParameters::FOCUS_MODE_CONTINUOUS_PICTURE)) {
+            isAFSupported = true;
+            focusMode = CameraParameters::FOCUS_MODE_CONTINUOUS_PICTURE;
+        } else if (focusModes.contains(CameraParameters::FOCUS_MODE_CONTINUOUS_VIDEO)) {
+            isAFSupported = true;
+            focusMode = CameraParameters::FOCUS_MODE_CONTINUOUS_VIDEO;
+        } else if (focusModes.contains(CameraParameters::FOCUS_MODE_MACRO)) {
+            isAFSupported = true;
+            focusMode = CameraParameters::FOCUS_MODE_MACRO;
         }
-        if (cameraDevice) {
-            sp<Surface> previewSurface;
-            sp<SurfaceControl> surfaceControl;
-            CameraParameters params(cameraDevice->getParameters());
-            String8 focusModes(params.get(CameraParameters::KEY_SUPPORTED_FOCUS_MODES));
-            bool isAFSupported = false;
-            const char *focusMode = nullptr;
+        if (nullptr != focusMode) {
+            params.set(CameraParameters::KEY_FOCUS_MODE, focusMode);
+            cameraDevice->setParameters(params.flatten());
+        }
+        int previewWidth, previewHeight;
+        params.getPreviewSize(&previewWidth, &previewHeight);
 
-            if (focusModes.contains(CameraParameters::FOCUS_MODE_AUTO)) {
-                isAFSupported = true;
-            } else if (focusModes.contains(CameraParameters::FOCUS_MODE_CONTINUOUS_PICTURE)) {
-                isAFSupported = true;
-                focusMode = CameraParameters::FOCUS_MODE_CONTINUOUS_PICTURE;
-            } else if (focusModes.contains(CameraParameters::FOCUS_MODE_CONTINUOUS_VIDEO)) {
-                isAFSupported = true;
-                focusMode = CameraParameters::FOCUS_MODE_CONTINUOUS_VIDEO;
-            } else if (focusModes.contains(CameraParameters::FOCUS_MODE_MACRO)) {
-                isAFSupported = true;
-                focusMode = CameraParameters::FOCUS_MODE_MACRO;
-            }
-            if (nullptr != focusMode) {
-                params.set(CameraParameters::KEY_FOCUS_MODE, focusMode);
-                cameraDevice->setParameters(params.flatten());
-            }
-            int previewWidth, previewHeight;
-            params.getPreviewSize(&previewWidth, &previewHeight);
+        mComposerClient = new SurfaceComposerClient;
+        mComposerClient->initCheck();
 
-            mComposerClient = new SurfaceComposerClient;
-            mComposerClient->initCheck();
-
-            bool shouldPassInvalidLayerMetaData = mFuzzedDataProvider->ConsumeBool();
-            int layerMetaData;
-            if (shouldPassInvalidLayerMetaData) {
-                layerMetaData = mFuzzedDataProvider->ConsumeIntegral<int>();
-            } else {
-                layerMetaData = kLayerMetadata[mFuzzedDataProvider->ConsumeIntegralInRange<size_t>(
+        bool shouldPassInvalidLayerMetaData = mFuzzedDataProvider->ConsumeBool();
+        int layerMetaData;
+        if (shouldPassInvalidLayerMetaData) {
+            layerMetaData = mFuzzedDataProvider->ConsumeIntegral<int>();
+        } else {
+            layerMetaData = kLayerMetadata[mFuzzedDataProvider->ConsumeIntegralInRange<size_t>(
                     0, kNumLayerMetaData - 1)];
-            }
-            surfaceControl = mComposerClient->createSurface(
+        }
+        surfaceControl = mComposerClient->createSurface(
                 String8("Test Surface"), previewWidth, previewHeight,
                 CameraParameters::previewFormatToEnum(params.getPreviewFormat()), layerMetaData);
 
-            if (surfaceControl.get() != nullptr) {
-                SurfaceComposerClient::Transaction{}
+        if (surfaceControl.get() != nullptr) {
+            SurfaceComposerClient::Transaction{}
                     .setLayer(surfaceControl, 0x7fffffff)
                     .show(surfaceControl)
                     .apply();
 
-                previewSurface = surfaceControl->getSurface();
-                cameraDevice->setPreviewTarget(previewSurface->getIGraphicBufferProducer());
-            }
-            cameraDevice->setPreviewCallbackFlag(CAMERA_FRAME_CALLBACK_FLAG_CAMCORDER);
-
-            Vector<Size> pictureSizes;
-            params.getSupportedPictureSizes(pictureSizes);
-
-            for (size_t i = 0; i < pictureSizes.size(); ++i) {
-                params.setPictureSize(pictureSizes[i].width, pictureSizes[i].height);
-                cameraDevice->setParameters(params.flatten());
-                cameraDevice->startPreview();
-                waitForPreviewStart();
-                cameraDevice->autoFocus();
-                waitForEvent(mAutoFocusLock, mAutoFocusCondition, mAutoFocusMessage);
-                bool shouldPassInvalidCameraMsg = mFuzzedDataProvider->ConsumeBool();
-                int msgType;
-                if (shouldPassInvalidCameraMsg) {
-                    msgType = mFuzzedDataProvider->ConsumeIntegral<int>();
-                } else {
-                    msgType = kCameraMsg[mFuzzedDataProvider->ConsumeIntegralInRange<size_t>(
-                        0, kNumCameraMsg - 1)];
-                }
-                cameraDevice->takePicture(msgType);
-
-                waitForEvent(mSnapshotLock, mSnapshotCondition, mSnapshotNotification);
-            }
-
-            Vector<Size> videoSizes;
-            params.getSupportedVideoSizes(videoSizes);
-
-            for (size_t i = 0; i < videoSizes.size(); ++i) {
-                params.setVideoSize(videoSizes[i].width, videoSizes[i].height);
-
-                cameraDevice->setParameters(params.flatten());
-                cameraDevice->startPreview();
-                waitForPreviewStart();
-                cameraDevice->setVideoBufferMode(
-                    android::hardware::BnCamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE);
-                cameraDevice->setVideoTarget(previewSurface->getIGraphicBufferProducer());
-                cameraDevice->startRecording();
-                cameraDevice->stopRecording();
-            }
-            cameraDevice->stopPreview();
-            cameraDevice->disconnect();
+            previewSurface = surfaceControl->getSurface();
+            cameraDevice->setPreviewTarget(previewSurface->getIGraphicBufferProducer());
         }
+        cameraDevice->setPreviewCallbackFlag(CAMERA_FRAME_CALLBACK_FLAG_CAMCORDER);
+
+        Vector<Size> pictureSizes;
+        params.getSupportedPictureSizes(pictureSizes);
+
+        if (pictureSizes.size()) {
+            Size pictureSize = pictureSizes[mFuzzedDataProvider->ConsumeIntegralInRange<size_t>(
+                    0, pictureSizes.size() - 1)];
+            params.setPictureSize(pictureSize.width, pictureSize.height);
+            cameraDevice->setParameters(params.flatten());
+            cameraDevice->startPreview();
+            waitForPreviewStart();
+            cameraDevice->autoFocus();
+            waitForEvent(mAutoFocusLock, mAutoFocusCondition, mAutoFocusMessage);
+            bool shouldPassInvalidCameraMsg = mFuzzedDataProvider->ConsumeBool();
+            int msgType;
+            if (shouldPassInvalidCameraMsg) {
+                msgType = mFuzzedDataProvider->ConsumeIntegral<int>();
+            } else {
+                msgType = kCameraMsg[mFuzzedDataProvider->ConsumeIntegralInRange<size_t>(
+                        0, kNumCameraMsg - 1)];
+            }
+            cameraDevice->takePicture(msgType);
+
+            waitForEvent(mSnapshotLock, mSnapshotCondition, mSnapshotNotification);
+            cameraDevice->stopPreview();
+        }
+
+        Vector<Size> videoSizes;
+        params.getSupportedVideoSizes(videoSizes);
+
+        if (videoSizes.size()) {
+            Size videoSize = videoSizes[mFuzzedDataProvider->ConsumeIntegralInRange<size_t>(
+                    0, videoSizes.size() - 1)];
+            params.setVideoSize(videoSize.width, videoSize.height);
+
+            cameraDevice->setParameters(params.flatten());
+            cameraDevice->startPreview();
+            waitForPreviewStart();
+            cameraDevice->setVideoBufferMode(
+                    android::hardware::BnCamera::VIDEO_BUFFER_MODE_BUFFER_QUEUE);
+            cameraDevice->setVideoTarget(previewSurface->getIGraphicBufferProducer());
+            cameraDevice->stopPreview();
+            cameraDevice->startRecording();
+            waitForEvent(mRecordingLock, mRecordingCondition, mRecordingNotification);
+            cameraDevice->stopRecording();
+        }
+        cameraDevice->disconnect();
     }
 }
 
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index 3f18b95..bbfa0b7 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -228,9 +228,6 @@
     if (mMmapStream != nullptr) {
         // Needs to be explicitly cleared or CTS will fail but it is not clear why.
         mMmapStream.clear();
-        // Apparently the above close is asynchronous. An attempt to open a new device
-        // right after a close can fail. Also some callbacks may still be in flight!
-        // FIXME Make closing synchronous.
         AudioClock::sleepForNanos(100 * AAUDIO_NANOS_PER_MILLISECOND);
     }
 }
@@ -362,12 +359,8 @@
     asyncTask.detach();
 }
 
-void AAudioServiceEndpointMMAP::onVolumeChanged(audio_channel_mask_t channels,
-                                              android::Vector<float> values) {
-    // TODO Do we really need a different volume for each channel?
-    // We get called with an array filled with a single value!
-    float volume = values[0];
-    ALOGD("%s() volume[0] = %f", __func__, volume);
+void AAudioServiceEndpointMMAP::onVolumeChanged(float volume) {
+    ALOGD("%s() volume = %f", __func__, volume);
     std::lock_guard<std::mutex> lock(mLockStreams);
     for(const auto& stream : mRegisteredStreams) {
         stream->onVolumeChanged(volume);
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.h b/services/oboeservice/AAudioServiceEndpointMMAP.h
index 3e7f2c7..3658c58 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.h
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.h
@@ -77,8 +77,7 @@
     // -------------- Callback functions for MmapStreamCallback ---------------------
     void onTearDown(audio_port_handle_t portHandle) override;
 
-    void onVolumeChanged(audio_channel_mask_t channels,
-                         android::Vector<float> values) override;
+    void onVolumeChanged(float volume) override;
 
     void onRoutingChanged(audio_port_handle_t portHandle) override;
     // ------------------------------------------------------------------------------