diff --git a/apex/OWNERS b/apex/OWNERS
index 5587f5f..54802d4 100644
--- a/apex/OWNERS
+++ b/apex/OWNERS
@@ -1,6 +1,7 @@
-chz@google.com
-dwkang@google.com
+essick@google.com
 jiyong@google.com
 lajos@google.com
-marcone@google.com
-wjia@google.com
+nchalko@google.com
+
+include platform/packages/modules/common:/MODULES_OWNERS
+
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index e6e3473..cd8fe97 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -1067,7 +1067,7 @@
 
     std::optional<PhysicalDisplayId> displayId = SurfaceComposerClient::getInternalDisplayId();
     if (!displayId) {
-        fprintf(stderr, "Failed to get token for internal display\n");
+        fprintf(stderr, "Failed to get ID for internal display\n");
         return 1;
     }
 
@@ -1168,17 +1168,14 @@
             }
             break;
         case 'd':
-            gPhysicalDisplayId = PhysicalDisplayId(atoll(optarg));
-            if (gPhysicalDisplayId.value == 0) {
-                fprintf(stderr, "Please specify a valid physical display id\n");
-                return 2;
-            } else if (SurfaceComposerClient::
-                    getPhysicalDisplayToken(gPhysicalDisplayId) == nullptr) {
-                fprintf(stderr, "Invalid physical display id: %s\n",
-                        to_string(gPhysicalDisplayId).c_str());
-                return 2;
+            if (const auto id = android::DisplayId::fromValue<PhysicalDisplayId>(atoll(optarg));
+                id && SurfaceComposerClient::getPhysicalDisplayToken(*id)) {
+                gPhysicalDisplayId = *id;
+                break;
             }
-            break;
+
+            fprintf(stderr, "Invalid physical display ID\n");
+            return 2;
         default:
             if (ic != '?') {
                 fprintf(stderr, "getopt_long returned unexpected value 0x%x\n", ic);
diff --git a/drm/mediadrm/plugins/TEST_MAPPING b/drm/mediadrm/plugins/TEST_MAPPING
index 7bd1568..87becb6 100644
--- a/drm/mediadrm/plugins/TEST_MAPPING
+++ b/drm/mediadrm/plugins/TEST_MAPPING
@@ -11,6 +11,9 @@
         },
         {
           "include-filter": "android.media.cts.MediaDrmMetricsTest"
+        },
+        {
+          "include-filter": "android.media.cts.NativeMediaDrmClearkeyTest"
         }
       ]
     }
diff --git a/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
index 6ac3510..089eb1c 100644
--- a/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/default/DrmPlugin.cpp
@@ -207,6 +207,7 @@
     }
 
     infoMap.clear();
+    android::Mutex::Autolock lock(mPlayPolicyLock);
     for (size_t i = 0; i < mPlayPolicy.size(); ++i) {
         infoMap.add(mPlayPolicy.keyAt(i), mPlayPolicy.valueAt(i));
     }
diff --git a/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
index aa9b59d..95f15ca 100644
--- a/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/default/include/DrmPlugin.h
@@ -262,7 +262,7 @@
     void initProperties();
     void setPlayPolicy();
 
-    android::Mutex mPlayPolicyLock;
+    mutable android::Mutex mPlayPolicyLock;
     android::KeyedVector<String8, String8> mPlayPolicy;
     android::KeyedVector<String8, String8> mStringProperties;
     android::KeyedVector<String8, Vector<uint8_t>> mByteArrayProperties;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
index bc7c3f2..0cd9375 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/DrmPlugin.cpp
@@ -221,7 +221,6 @@
         if (requestString.find(kOfflineLicense) != std::string::npos) {
             std::string emptyResponse;
             std::string keySetIdString(keySetId.begin(), keySetId.end());
-            Mutex::Autolock lock(mFileHandleLock);
             if (!mFileHandle.StoreLicense(keySetIdString,
                     DeviceFiles::kLicenseStateReleasing,
                     emptyResponse)) {
@@ -337,7 +336,6 @@
         }
         *keySetId = kKeySetIdPrefix + ByteArrayToHexString(
                 reinterpret_cast<const uint8_t*>(randomData.data()), randomData.size());
-        Mutex::Autolock lock(mFileHandleLock);
         if (mFileHandle.LicenseExists(*keySetId)) {
             // collision, regenerate
             ALOGV("Retry generating KeySetId");
@@ -395,7 +393,6 @@
     if (status == Status::OK) {
         if (isOfflineLicense) {
             if (isRelease) {
-                Mutex::Autolock lock(mFileHandleLock);
                 mFileHandle.DeleteLicense(keySetId);
                 mSessionLibrary->destroySession(session);
             } else {
@@ -404,7 +401,6 @@
                     return Void();
                 }
 
-                Mutex::Autolock lock(mFileHandleLock);
                 bool ok = mFileHandle.StoreLicense(
                         keySetId,
                         DeviceFiles::kLicenseStateActive,
@@ -459,7 +455,6 @@
         DeviceFiles::LicenseState licenseState;
         std::string offlineLicense;
         Status status = Status::OK;
-        Mutex::Autolock lock(mFileHandleLock);
         if (!mFileHandle.RetrieveLicense(std::string(keySetId.begin(), keySetId.end()),
                 &licenseState, &offlineLicense)) {
             ALOGE("Failed to restore offline license");
@@ -582,7 +577,6 @@
 Return<void> DrmPlugin::queryKeyStatus(
         const hidl_vec<uint8_t>& sessionId,
         queryKeyStatus_cb _hidl_cb) {
-
     if (sessionId.size() == 0) {
         // Returns empty key status KeyValue pair
         _hidl_cb(Status::BAD_VALUE, hidl_vec<KeyValue>());
@@ -592,12 +586,14 @@
     std::vector<KeyValue> infoMapVec;
     infoMapVec.clear();
 
+    mPlayPolicyLock.lock();
     KeyValue keyValuePair;
     for (size_t i = 0; i < mPlayPolicy.size(); ++i) {
         keyValuePair.key = mPlayPolicy[i].key;
         keyValuePair.value = mPlayPolicy[i].value;
         infoMapVec.push_back(keyValuePair);
     }
+    mPlayPolicyLock.unlock();
     _hidl_cb(Status::OK, toHidlVec(infoMapVec));
     return Void();
 }
@@ -768,8 +764,6 @@
 }
 
 Return<void> DrmPlugin::getOfflineLicenseKeySetIds(getOfflineLicenseKeySetIds_cb _hidl_cb) {
-    Mutex::Autolock lock(mFileHandleLock);
-
     std::vector<std::string> licenseNames = mFileHandle.ListLicenses();
     std::vector<KeySetId> keySetIds;
     if (mMockError != Status_V1_2::OK) {
@@ -790,7 +784,6 @@
         return toStatus_1_0(mMockError);
     }
     std::string licenseName(keySetId.begin(), keySetId.end());
-    Mutex::Autolock lock(mFileHandleLock);
     if (mFileHandle.DeleteLicense(licenseName)) {
         return Status::OK;
     }
@@ -799,8 +792,6 @@
 
 Return<void> DrmPlugin::getOfflineLicenseState(const KeySetId& keySetId,
         getOfflineLicenseState_cb _hidl_cb) {
-    Mutex::Autolock lock(mFileHandleLock);
-
     std::string licenseName(keySetId.begin(), keySetId.end());
     DeviceFiles::LicenseState state;
     std::string license;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/MemoryFileSystem.cpp b/drm/mediadrm/plugins/clearkey/hidl/MemoryFileSystem.cpp
index e61db3f..56910be 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/MemoryFileSystem.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/MemoryFileSystem.cpp
@@ -24,13 +24,11 @@
 }
 
 bool MemoryFileSystem::FileExists(const std::string& fileName) const {
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     auto result = mMemoryFileSystem.find(fileName);
     return result != mMemoryFileSystem.end();
 }
 
 ssize_t MemoryFileSystem::GetFileSize(const std::string& fileName) const {
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     auto result = mMemoryFileSystem.find(fileName);
     if (result != mMemoryFileSystem.end()) {
         return static_cast<ssize_t>(result->second.getFileSize());
@@ -42,7 +40,6 @@
 
 std::vector<std::string> MemoryFileSystem::ListFiles() const {
     std::vector<std::string> list;
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     for (const auto& filename : mMemoryFileSystem) {
         list.push_back(filename.first);
     }
@@ -51,7 +48,6 @@
 
 size_t MemoryFileSystem::Read(const std::string& path, std::string* buffer) {
     std::string key = GetFileName(path);
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     auto result = mMemoryFileSystem.find(key);
     if (result != mMemoryFileSystem.end()) {
         std::string serializedHashFile = result->second.getContent();
@@ -65,7 +61,6 @@
 
 size_t MemoryFileSystem::Write(const std::string& path, const MemoryFile& memoryFile) {
     std::string key = GetFileName(path);
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     auto result = mMemoryFileSystem.find(key);
     if (result != mMemoryFileSystem.end()) {
         mMemoryFileSystem.erase(key);
@@ -75,7 +70,6 @@
 }
 
 bool MemoryFileSystem::RemoveFile(const std::string& fileName) {
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     auto result = mMemoryFileSystem.find(fileName);
     if (result != mMemoryFileSystem.end()) {
         mMemoryFileSystem.erase(result);
@@ -87,7 +81,6 @@
 }
 
 bool MemoryFileSystem::RemoveAllFiles() {
-    std::lock_guard<std::mutex> lock(mMemoryFileSystemLock);
     mMemoryFileSystem.clear();
     return mMemoryFileSystem.empty();
 }
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
index 5d6e3da..cb5c9fe 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/DrmPlugin.h
@@ -432,8 +432,7 @@
         mMockError = Status_V1_2::OK;
     }
 
-    DeviceFiles mFileHandle GUARDED_BY(mFileHandleLock);
-    Mutex mFileHandleLock;
+    DeviceFiles mFileHandle;
     Mutex mSecureStopLock;
 
     CLEARKEY_DISALLOW_COPY_AND_ASSIGN_AND_NEW(DrmPlugin);
diff --git a/drm/mediadrm/plugins/clearkey/hidl/include/MemoryFileSystem.h b/drm/mediadrm/plugins/clearkey/hidl/include/MemoryFileSystem.h
index a90d818..1d98860 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/include/MemoryFileSystem.h
+++ b/drm/mediadrm/plugins/clearkey/hidl/include/MemoryFileSystem.h
@@ -5,9 +5,7 @@
 #ifndef CLEARKEY_MEMORY_FILE_SYSTEM_H_
 #define CLEARKEY_MEMORY_FILE_SYSTEM_H_
 
-#include <android-base/thread_annotations.h>
 #include <map>
-#include <mutex>
 #include <string>
 
 #include "ClearKeyTypes.h"
@@ -51,12 +49,10 @@
     size_t Write(const std::string& pathName, const MemoryFile& memoryFile);
 
  private:
-    mutable std::mutex mMemoryFileSystemLock;
-
     // License file name is made up of a unique keySetId, therefore,
     // the filename can be used as the key to locate licenses in the
     // memory file system.
-    std::map<std::string, MemoryFile> mMemoryFileSystem GUARDED_BY(mMemoryFileSystemLock);
+    std::map<std::string, MemoryFile> mMemoryFileSystem;
 
     std::string GetFileName(const std::string& path);
 
diff --git a/media/codec2/components/hevc/C2SoftHevcEnc.cpp b/media/codec2/components/hevc/C2SoftHevcEnc.cpp
index 4bc1777..b7a5686 100644
--- a/media/codec2/components/hevc/C2SoftHevcEnc.cpp
+++ b/media/codec2/components/hevc/C2SoftHevcEnc.cpp
@@ -245,6 +245,19 @@
                 })
                 .withSetter(CodedColorAspectsSetter, mColorAspects)
                 .build());
+
+        addParameter(
+                DefineParam(mPictureQuantization, C2_PARAMKEY_PICTURE_QUANTIZATION)
+                .withDefault(C2StreamPictureQuantizationTuning::output::AllocShared(
+                        0 /* flexCount */, 0u /* stream */))
+                .withFields({C2F(mPictureQuantization, m.values[0].type_).oneOf(
+                                {C2Config::picture_type_t(I_FRAME),
+                                  C2Config::picture_type_t(P_FRAME),
+                                  C2Config::picture_type_t(B_FRAME)}),
+                             C2F(mPictureQuantization, m.values[0].min).any(),
+                             C2F(mPictureQuantization, m.values[0].max).any()})
+                .withSetter(PictureQuantizationSetter)
+                .build());
     }
 
     static C2R InputDelaySetter(
@@ -464,9 +477,69 @@
         me.set().matrix = coded.v.matrix;
         return C2R::Ok();
     }
+    static C2R PictureQuantizationSetter(bool mayBlock,
+                                         C2P<C2StreamPictureQuantizationTuning::output> &me) {
+        (void)mayBlock;
+
+        // these are the ones we're going to set, so want them to default
+        // to the DEFAULT values for the codec
+        int32_t iMin = HEVC_QP_MIN, pMin = HEVC_QP_MIN, bMin = HEVC_QP_MIN;
+        int32_t iMax = HEVC_QP_MAX, pMax = HEVC_QP_MAX, bMax = HEVC_QP_MAX;
+
+        for (size_t i = 0; i < me.v.flexCount(); ++i) {
+            const C2PictureQuantizationStruct &layer = me.v.m.values[i];
+
+            // layerMin is clamped to [HEVC_QP_MIN, layerMax] to avoid error
+            // cases where layer.min > layer.max
+            int32_t layerMax = std::clamp(layer.max, HEVC_QP_MIN, HEVC_QP_MAX);
+            int32_t layerMin = std::clamp(layer.min, HEVC_QP_MIN, layerMax);
+            if (layer.type_ == C2Config::picture_type_t(I_FRAME)) {
+                iMax = layerMax;
+                iMin = layerMin;
+                ALOGV("iMin %d iMax %d", iMin, iMax);
+            } else if (layer.type_ == C2Config::picture_type_t(P_FRAME)) {
+                pMax = layerMax;
+                pMin = layerMin;
+                ALOGV("pMin %d pMax %d", pMin, pMax);
+            } else if (layer.type_ == C2Config::picture_type_t(B_FRAME)) {
+                bMax = layerMax;
+                bMin = layerMin;
+                ALOGV("bMin %d bMax %d", bMin, bMax);
+            }
+        }
+
+        ALOGV("PictureQuantizationSetter(entry): i %d-%d p %d-%d b %d-%d",
+              iMin, iMax, pMin, pMax, bMin, bMax);
+
+        int32_t maxFrameQP = std::min(std::min(iMax, pMax), bMax);
+        int32_t minFrameQP = std::max(std::max(iMin, pMin), bMin);
+        if (minFrameQP > maxFrameQP) {
+            minFrameQP = maxFrameQP;
+        }
+
+        // put them back into the structure
+        for (size_t i = 0; i < me.v.flexCount(); ++i) {
+            const C2PictureQuantizationStruct &layer = me.v.m.values[i];
+
+            if (layer.type_ == C2Config::picture_type_t(I_FRAME) ||
+                layer.type_ == C2Config::picture_type_t(P_FRAME) ||
+                layer.type_ == C2Config::picture_type_t(B_FRAME)) {
+                me.set().m.values[i].max = maxFrameQP;
+                me.set().m.values[i].min = minFrameQP;
+            }
+        }
+
+        ALOGV("PictureQuantizationSetter(exit): i = p = b = %d-%d",
+              minFrameQP, maxFrameQP);
+
+        return C2R::Ok();
+    }
     std::shared_ptr<C2StreamColorAspectsInfo::output> getCodedColorAspects_l() {
         return mCodedColorAspects;
     }
+    std::shared_ptr<C2StreamPictureQuantizationTuning::output> getPictureQuantization_l() const {
+        return mPictureQuantization;
+    }
 
    private:
     std::shared_ptr<C2StreamUsageTuning::input> mUsage;
@@ -482,6 +555,7 @@
     std::shared_ptr<C2StreamGopTuning::output> mGop;
     std::shared_ptr<C2StreamColorAspectsInfo::input> mColorAspects;
     std::shared_ptr<C2StreamColorAspectsInfo::output> mCodedColorAspects;
+    std::shared_ptr<C2StreamPictureQuantizationTuning::output> mPictureQuantization;
 };
 
 static size_t GetCPUCoreCount() {
@@ -654,12 +728,41 @@
         mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 3;
     }
 
-    switch (mBitrateMode->value) {
-        case C2Config::BITRATE_IGNORE:
-            mEncParams.s_config_prms.i4_rate_control_mode = 3;
-            mEncParams.s_tgt_lyr_prms.as_tgt_params[0].ai4_frame_qp[0] =
-                getQpFromQuality(mQuality->value);
+    // we resolved out-of-bound and unspecified values in PictureQuantizationSetter()
+    // so we can start with defaults that are overridden as needed.
+    int32_t maxFrameQP = mEncParams.s_config_prms.i4_max_frame_qp;
+    int32_t minFrameQP = mEncParams.s_config_prms.i4_min_frame_qp;
+
+    for (size_t i = 0; i < mQpBounds->flexCount(); ++i) {
+        const C2PictureQuantizationStruct &layer = mQpBounds->m.values[i];
+
+        // no need to loop, hevc library takes same range for I/P/B picture type
+        if (layer.type_ == C2Config::picture_type_t(I_FRAME) ||
+            layer.type_ == C2Config::picture_type_t(P_FRAME) ||
+            layer.type_ == C2Config::picture_type_t(B_FRAME)) {
+
+            maxFrameQP = layer.max;
+            minFrameQP = layer.min;
             break;
+        }
+    }
+    mEncParams.s_config_prms.i4_max_frame_qp = maxFrameQP;
+    mEncParams.s_config_prms.i4_min_frame_qp = minFrameQP;
+
+    ALOGV("MaxFrameQp: %d MinFrameQp: %d", maxFrameQP, minFrameQP);
+
+    mEncParams.s_tgt_lyr_prms.as_tgt_params[0].ai4_frame_qp[0] =
+        std::clamp(kDefaultInitQP, minFrameQP, maxFrameQP);
+
+    switch (mBitrateMode->value) {
+        case C2Config::BITRATE_IGNORE: {
+            mEncParams.s_config_prms.i4_rate_control_mode = 3;
+            // ensure initial qp values are within our newly configured bounds
+            int32_t frameQp = getQpFromQuality(mQuality->value);
+            mEncParams.s_tgt_lyr_prms.as_tgt_params[0].ai4_frame_qp[0] =
+                std::clamp(frameQp, minFrameQP, maxFrameQP);
+            break;
+        }
         case C2Config::BITRATE_CONST:
             mEncParams.s_config_prms.i4_rate_control_mode = 5;
             break;
@@ -723,6 +826,7 @@
         mGop = mIntf->getGop_l();
         mRequestSync = mIntf->getRequestSync_l();
         mColorAspects = mIntf->getCodedColorAspects_l();
+        mQpBounds = mIntf->getPictureQuantization_l();;
     }
 
     c2_status_t status = initEncParams();
diff --git a/media/codec2/components/hevc/C2SoftHevcEnc.h b/media/codec2/components/hevc/C2SoftHevcEnc.h
index 9dbf682..4217a8b 100644
--- a/media/codec2/components/hevc/C2SoftHevcEnc.h
+++ b/media/codec2/components/hevc/C2SoftHevcEnc.h
@@ -42,6 +42,11 @@
 #define DEFAULT_B_FRAMES     0
 #define DEFAULT_RC_LOOKAHEAD 0
 
+#define HEVC_QP_MIN 1
+#define HEVC_QP_MAX 51
+
+constexpr int32_t kDefaultInitQP = 32;
+
 struct C2SoftHevcEnc : public SimpleC2Component {
     class IntfImpl;
 
@@ -90,6 +95,7 @@
     std::shared_ptr<C2StreamGopTuning::output> mGop;
     std::shared_ptr<C2StreamRequestSyncFrameTuning::output> mRequestSync;
     std::shared_ptr<C2StreamColorAspectsInfo::output> mColorAspects;
+    std::shared_ptr<C2StreamPictureQuantizationTuning::output> mQpBounds;
 #ifdef FILE_DUMP_ENABLE
     char mInFile[200];
     char mOutFile[200];
diff --git a/media/codec2/fuzzer/C2Fuzzer.cpp b/media/codec2/fuzzer/C2Fuzzer.cpp
index 51e1013..e35ee48 100644
--- a/media/codec2/fuzzer/C2Fuzzer.cpp
+++ b/media/codec2/fuzzer/C2Fuzzer.cpp
@@ -194,12 +194,12 @@
   }
 
   std::vector<C2Param*> configParams;
+  C2StreamPictureSizeInfo::input inputSize(0u, kWidthOfVideo, kHeightOfVideo);
+  C2StreamSampleRateInfo::output sampleRateInfo(0u, kSamplingRateOfAudio);
+  C2StreamChannelCountInfo::output channelCountInfo(0u, kChannelsOfAudio);
   if (domain.value == DOMAIN_VIDEO) {
-    C2StreamPictureSizeInfo::input inputSize(0u, kWidthOfVideo, kHeightOfVideo);
     configParams.push_back(&inputSize);
   } else if (domain.value == DOMAIN_AUDIO) {
-    C2StreamSampleRateInfo::output sampleRateInfo(0u, kSamplingRateOfAudio);
-    C2StreamChannelCountInfo::output channelCountInfo(0u, kChannelsOfAudio);
     configParams.push_back(&sampleRateInfo);
     configParams.push_back(&channelCountInfo);
   }
diff --git a/media/codecs/m4v_h263/enc/src/mp4enc_api.cpp b/media/codecs/m4v_h263/enc/src/mp4enc_api.cpp
index 234faef..b295258 100644
--- a/media/codecs/m4v_h263/enc/src/mp4enc_api.cpp
+++ b/media/codecs/m4v_h263/enc/src/mp4enc_api.cpp
@@ -501,13 +501,16 @@
     /* check frame rate */
     for (i = 0; i < encParams->nLayers; i++)
     {
+        if (encOption->encFrameRate[i] <= 0. || encOption->encFrameRate[i] > 120)
+        {
+            goto CLEAN_UP;
+        }
         encParams->LayerFrameRate[i] = encOption->encFrameRate[i];
     }
 
     if (encParams->nLayers > 1)
     {
-        if (encOption->encFrameRate[0] == encOption->encFrameRate[1] ||
-                encOption->encFrameRate[0] == 0. || encOption->encFrameRate[1] == 0.) /* 7/31/03 */
+        if (encOption->encFrameRate[0] == encOption->encFrameRate[1])
             goto CLEAN_UP;
     }
     /* set max frame rate */
@@ -610,18 +613,19 @@
     /* Find the maximum width*height for memory allocation of the VOPs */
     for (idx = 0; idx < nLayers; idx++)
     {
-        temp_w = video->encParams->LayerWidth[idx];
-        temp_h = video->encParams->LayerHeight[idx];
+        temp_w = ((video->encParams->LayerWidth[idx] + 15) >> 4) << 4;
+        temp_h = ((video->encParams->LayerHeight[idx] + 15) >> 4) << 4;
+
+        if (temp_w > 2048 || temp_h > 2048) {
+            goto CLEAN_UP;
+        }
 
         if ((temp_w*temp_h) > max)
         {
             max = temp_w * temp_h;
-            max_width = ((temp_w + 15) >> 4) << 4;
-            max_height = ((temp_h + 15) >> 4) << 4;
-            if (((uint64_t)max_width * max_height) > (uint64_t)INT32_MAX
-                    || temp_w > INT32_MAX - 15 || temp_h > INT32_MAX - 15) {
-                goto CLEAN_UP;
-            }
+            max_width = temp_w;
+            max_height = temp_h;
+
             nTotalMB = ((max_width * max_height) >> 8);
         }
 
diff --git a/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp b/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp
index 4338c43..c04f7f3 100644
--- a/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp
+++ b/media/codecs/mp3dec/src/pvmp3_stereo_proc.cpp
@@ -219,6 +219,9 @@
 ; FUNCTION CODE
 ----------------------------------------------------------------------------*/
 
+#if __has_attribute(no_sanitize)
+__attribute__((no_sanitize("integer")))
+#endif
 void pvmp3_st_intensity(int32 xr[SUBBANDS_NUMBER*FILTERBANK_BANDS],
                         int32 xl[SUBBANDS_NUMBER*FILTERBANK_BANDS],
                         int32 is_pos,
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index 4b08295..fc86001 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -565,6 +565,145 @@
 };
 typedef int32_t aaudio_session_id_t;
 
+/**
+ * Defines the audio channel mask.
+ * Channel masks are used to describe the samples and their
+ * arrangement in the audio frame. They are also used in the endpoint
+ * (e.g. a USB audio interface, a DAC connected to headphones) to
+ * specify allowable configurations of a particular device.
+ *
+ * Added in API level 32.
+ */
+enum {
+    /**
+     * Invalid channel mask
+     */
+    AAUDIO_CHANNEL_INVALID = -1,
+
+    /**
+     * Output audio channel mask
+     */
+    AAUDIO_CHANNEL_FRONT_LEFT = 1 << 0,
+    AAUDIO_CHANNEL_FRONT_RIGHT = 1 << 1,
+    AAUDIO_CHANNEL_FRONT_CENTER = 1 << 2,
+    AAUDIO_CHANNEL_LOW_FREQUENCY = 1 << 3,
+    AAUDIO_CHANNEL_BACK_LEFT = 1 << 4,
+    AAUDIO_CHANNEL_BACK_RIGHT = 1 << 5,
+    AAUDIO_CHANNEL_FRONT_LEFT_OF_CENTER = 1 << 6,
+    AAUDIO_CHANNEL_FRONT_RIGHT_OF_CENTER = 1 << 7,
+    AAUDIO_CHANNEL_BACK_CENTER = 1 << 8,
+    AAUDIO_CHANNEL_SIDE_LEFT = 1 << 9,
+    AAUDIO_CHANNEL_SIDE_RIGHT = 1 << 10,
+    AAUDIO_CHANNEL_TOP_CENTER = 1 << 11,
+    AAUDIO_CHANNEL_TOP_FRONT_LEFT = 1 << 12,
+    AAUDIO_CHANNEL_TOP_FRONT_CENTER = 1 << 13,
+    AAUDIO_CHANNEL_TOP_FRONT_RIGHT = 1 << 14,
+    AAUDIO_CHANNEL_TOP_BACK_LEFT = 1 << 15,
+    AAUDIO_CHANNEL_TOP_BACK_CENTER = 1 << 16,
+    AAUDIO_CHANNEL_TOP_BACK_RIGHT = 1 << 17,
+    AAUDIO_CHANNEL_TOP_SIDE_LEFT = 1 << 18,
+    AAUDIO_CHANNEL_TOP_SIDE_RIGHT = 1 << 19,
+    AAUDIO_CHANNEL_BOTTOM_FRONT_LEFT = 1 << 20,
+    AAUDIO_CHANNEL_BOTTOM_FRONT_CENTER = 1 << 21,
+    AAUDIO_CHANNEL_BOTTOM_FRONT_RIGHT = 1 << 22,
+    AAUDIO_CHANNEL_LOW_FREQUENCY_2 = 1 << 23,
+    AAUDIO_CHANNEL_FRONT_WIDE_LEFT = 1 << 24,
+    AAUDIO_CHANNEL_FRONT_WIDE_RIGHT = 1 << 25,
+
+    AAUDIO_CHANNEL_MONO = AAUDIO_CHANNEL_FRONT_LEFT,
+    AAUDIO_CHANNEL_STEREO = AAUDIO_CHANNEL_FRONT_LEFT |
+                            AAUDIO_CHANNEL_FRONT_RIGHT,
+    AAUDIO_CHANNEL_2POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_TRI = AAUDIO_CHANNEL_FRONT_LEFT |
+                         AAUDIO_CHANNEL_FRONT_RIGHT |
+                         AAUDIO_CHANNEL_FRONT_CENTER,
+    AAUDIO_CHANNEL_TRI_BACK = AAUDIO_CHANNEL_FRONT_LEFT |
+                              AAUDIO_CHANNEL_FRONT_RIGHT |
+                              AAUDIO_CHANNEL_BACK_CENTER,
+    AAUDIO_CHANNEL_3POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_FRONT_CENTER |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_2POINT0POINT2 = AAUDIO_CHANNEL_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_2POINT1POINT2 = AAUDIO_CHANNEL_2POINT0POINT2 |
+                                   AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_3POINT0POINT2 = AAUDIO_CHANNEL_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_FRONT_CENTER |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_3POINT1POINT2 = AAUDIO_CHANNEL_3POINT0POINT2 |
+                                   AAUDIO_CHANNEL_LOW_FREQUENCY,
+    AAUDIO_CHANNEL_QUAD = AAUDIO_CHANNEL_FRONT_LEFT |
+                          AAUDIO_CHANNEL_FRONT_RIGHT |
+                          AAUDIO_CHANNEL_BACK_LEFT |
+                          AAUDIO_CHANNEL_BACK_RIGHT,
+    AAUDIO_CHANNEL_QUAD_SIDE = AAUDIO_CHANNEL_FRONT_LEFT |
+                               AAUDIO_CHANNEL_FRONT_RIGHT |
+                               AAUDIO_CHANNEL_SIDE_LEFT |
+                               AAUDIO_CHANNEL_SIDE_RIGHT,
+    AAUDIO_CHANNEL_SURROUND = AAUDIO_CHANNEL_FRONT_LEFT |
+                              AAUDIO_CHANNEL_FRONT_RIGHT |
+                              AAUDIO_CHANNEL_FRONT_CENTER |
+                              AAUDIO_CHANNEL_BACK_CENTER,
+    AAUDIO_CHANNEL_PENTA = AAUDIO_CHANNEL_QUAD |
+                           AAUDIO_CHANNEL_FRONT_CENTER,
+    // aka 5POINT1_BACK
+    AAUDIO_CHANNEL_5POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_FRONT_CENTER |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY |
+                             AAUDIO_CHANNEL_BACK_LEFT |
+                             AAUDIO_CHANNEL_BACK_RIGHT,
+    AAUDIO_CHANNEL_5POINT1_SIDE = AAUDIO_CHANNEL_FRONT_LEFT |
+                                  AAUDIO_CHANNEL_FRONT_RIGHT |
+                                  AAUDIO_CHANNEL_FRONT_CENTER |
+                                  AAUDIO_CHANNEL_LOW_FREQUENCY |
+                                  AAUDIO_CHANNEL_SIDE_LEFT |
+                                  AAUDIO_CHANNEL_SIDE_RIGHT,
+    AAUDIO_CHANNEL_6POINT1 = AAUDIO_CHANNEL_FRONT_LEFT |
+                             AAUDIO_CHANNEL_FRONT_RIGHT |
+                             AAUDIO_CHANNEL_FRONT_CENTER |
+                             AAUDIO_CHANNEL_LOW_FREQUENCY |
+                             AAUDIO_CHANNEL_BACK_LEFT |
+                             AAUDIO_CHANNEL_BACK_RIGHT |
+                             AAUDIO_CHANNEL_BACK_CENTER,
+    AAUDIO_CHANNEL_7POINT1 = AAUDIO_CHANNEL_5POINT1 |
+                             AAUDIO_CHANNEL_SIDE_LEFT |
+                             AAUDIO_CHANNEL_SIDE_RIGHT,
+    AAUDIO_CHANNEL_5POINT1POINT2 = AAUDIO_CHANNEL_5POINT1 |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_5POINT1POINT4 = AAUDIO_CHANNEL_5POINT1 |
+                                   AAUDIO_CHANNEL_TOP_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_TOP_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_TOP_BACK_LEFT |
+                                   AAUDIO_CHANNEL_TOP_BACK_RIGHT,
+    AAUDIO_CHANNEL_7POINT1POINT2 = AAUDIO_CHANNEL_7POINT1 |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+    AAUDIO_CHANNEL_7POINT1POINT4 = AAUDIO_CHANNEL_7POINT1 |
+                                   AAUDIO_CHANNEL_TOP_FRONT_LEFT |
+                                   AAUDIO_CHANNEL_TOP_FRONT_RIGHT |
+                                   AAUDIO_CHANNEL_TOP_BACK_LEFT |
+                                   AAUDIO_CHANNEL_TOP_BACK_RIGHT,
+    AAUDIO_CHANNEL_9POINT1POINT4 = AAUDIO_CHANNEL_7POINT1POINT4 |
+                                   AAUDIO_CHANNEL_FRONT_WIDE_LEFT |
+                                   AAUDIO_CHANNEL_FRONT_WIDE_RIGHT,
+    AAUDIO_CHANNEL_9POINT1POINT6 = AAUDIO_CHANNEL_9POINT1POINT4 |
+                                   AAUDIO_CHANNEL_TOP_SIDE_LEFT |
+                                   AAUDIO_CHANNEL_TOP_SIDE_RIGHT,
+
+    AAUDIO_CHANNEL_FRONT_BACK = AAUDIO_CHANNEL_FRONT_CENTER |
+                                AAUDIO_CHANNEL_BACK_CENTER,
+};
+typedef uint32_t aaudio_channel_mask_t;
+
 typedef struct AAudioStreamStruct         AAudioStream;
 typedef struct AAudioStreamBuilderStruct  AAudioStreamBuilder;
 
@@ -643,8 +782,11 @@
  * This is usually {@code Context#getPackageName()}.
  *
  * The default, if you do not call this function, is a random package in the calling uid.
- * The vast majority of apps have only one package per calling UID. If the package
- * name does not match the calling UID, then requests will be rejected.
+ * The vast majority of apps have only one package per calling UID.
+ * If an invalid package name is set, input streams may not be given permission to
+ * record when started.
+ *
+ * The package name is usually the applicationId in your app's build.gradle file.
  *
  * Available since API level 31.
  *
@@ -699,6 +841,11 @@
  * If an exact value is specified then an opened stream will use that value.
  * If a stream cannot be opened with the specified value then the open will fail.
  *
+ * As the channel count provided here may be different from the corresponding channel count
+ * of channel mask used in {@link AAudioStreamBuilder_setChannelMask}, the last called function
+ * will be respected if both this function and {@link AAudioStreamBuilder_setChannelMask} are
+ * called.
+ *
  * Available since API level 26.
  *
  * @param builder reference provided by AAudio_createStreamBuilder()
@@ -714,6 +861,8 @@
  *
  * @param builder reference provided by AAudio_createStreamBuilder()
  * @param samplesPerFrame Number of samples in a frame.
+ *
+ * @deprecated use {@link AAudioStreamBuilder_setChannelCount}
  */
 AAUDIO_API void AAudioStreamBuilder_setSamplesPerFrame(AAudioStreamBuilder* builder,
                                                        int32_t samplesPerFrame) __INTRODUCED_IN(26);
@@ -1136,6 +1285,32 @@
 AAUDIO_API aaudio_result_t  AAudioStreamBuilder_delete(AAudioStreamBuilder* builder)
     __INTRODUCED_IN(26);
 
+/**
+ * Set audio channel mask for the stream.
+ *
+ * The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
+ * If both channel mask and count are not set, then stereo will then be chosen when the
+ * stream is opened.
+ * After opening a stream with an unspecified value, the application must query for the
+ * actual value, which may vary by device.
+ *
+ * If an exact value is specified then an opened stream will use that value.
+ * If a stream cannot be opened with the specified value then the open will fail.
+ *
+ * As the corresponding channel count of provided channel mask here may be different
+ * from the channel count used in {@link AAudioStreamBuilder_setChannelCount} or
+ * {@link AAudioStreamBuilder_setSamplesPerFrame}, the last called function will be
+ * respected if this function and {@link AAudioStreamBuilder_setChannelCount} or
+ * {@link AAudioStreamBuilder_setSamplesPerFrame} are called.
+ *
+ * Available since API level 32.
+ *
+ * @param builder reference provided by AAudio_createStreamBuilder()
+ * @param channelMask Audio channel mask desired.
+ */
+AAUDIO_API void AAudioStreamBuilder_setChannelMask(AAudioStreamBuilder* builder,
+        aaudio_channel_mask_t channelMask) __INTRODUCED_IN(32);
+
 // ============================================================
 // Stream Control
 // ============================================================
@@ -1652,6 +1827,18 @@
 AAUDIO_API bool AAudioStream_isPrivacySensitive(AAudioStream* stream)
         __INTRODUCED_IN(30);
 
+/**
+ * Return the channel mask for the stream. This will be the mask set using
+ * {@link #AAudioStreamBuilder_setChannelMask}, or {@link #AAUDIO_UNSPECIFIED} otherwise.
+ *
+ * Available since API level 32.
+ *
+ * @param stream reference provided by AAudioStreamBuilder_openStream()
+ * @return actual channel mask
+ */
+AAUDIO_API aaudio_channel_mask_t AAudioStream_getChannelMask(AAudioStream* stream)
+        __INTRODUCED_IN(32);
+
 #ifdef __cplusplus
 }
 #endif
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index 33a5c7f..da59da6 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -7,6 +7,65 @@
     default_applicable_licenses: ["frameworks_av_license"],
 }
 
+tidy_errors = [
+    // https://clang.llvm.org/extra/clang-tidy/checks/list.html
+    // For many categories, the checks are too many to specify individually.
+    // Feel free to disable as needed - as warnings are generally ignored,
+    // we treat warnings as errors.
+    "android-*",
+    "bugprone-*",
+    "cert-*",
+    "clang-analyzer-security*",
+    "google-*",
+    "misc-*",
+    //"modernize-*",  // explicitly list the modernize as they can be subjective.
+    "modernize-avoid-bind",
+    //"modernize-avoid-c-arrays", // std::array<> can be verbose
+    "modernize-concat-nested-namespaces",
+    //"modernize-deprecated-headers", // C headers still ok even if there is C++ equivalent.
+    "modernize-deprecated-ios-base-aliases",
+    "modernize-loop-convert",
+    "modernize-make-shared",
+    "modernize-make-unique",
+    "modernize-pass-by-value",
+    "modernize-raw-string-literal",
+    "modernize-redundant-void-arg",
+    "modernize-replace-auto-ptr",
+    "modernize-replace-random-shuffle",
+    "modernize-return-braced-init-list",
+    "modernize-shrink-to-fit",
+    "modernize-unary-static-assert",
+    // "modernize-use-auto", // found in AAudioAudio.cpp
+    "modernize-use-bool-literals",
+    "modernize-use-default-member-init",
+    "modernize-use-emplace",
+    "modernize-use-equals-default",
+    "modernize-use-equals-delete",
+    // "modernize-use-nodiscard", // found in aidl generated files
+    "modernize-use-noexcept",
+    "modernize-use-nullptr",
+    // "modernize-use-override", // found in aidl generated files
+    // "modernize-use-trailing-return-type", // not necessarily more readable
+    "modernize-use-transparent-functors",
+    "modernize-use-uncaught-exceptions",
+    // "modernize-use-using", // found typedef in several files
+    "performance-*",
+
+    // Remove some pedantic stylistic requirements.
+    "-android-cloexec-dup", // found in SharedMemoryParcelable.cpp
+    "-bugprone-macro-parentheses", // found in SharedMemoryParcelable.h
+    "-bugprone-narrowing-conversions", // found in several interface from size_t to int32_t
+
+    "-google-readability-casting", // C++ casts not always necessary and may be verbose
+    "-google-readability-todo", // do not require TODO(info)
+    "-google-build-using-namespace", // Reenable and fix later.
+    "-google-global-names-in-headers", // found in several files
+
+    "-misc-non-private-member-variables-in-classes", // found in aidl generated files
+
+    "-performance-no-int-to-ptr", // found in SharedMemoryParcelable.h
+]
+
 cc_library {
     name: "libaaudio",
 
@@ -52,7 +111,7 @@
         "libcutils",
         "libutils",
         "libbinder",
-        "libpermission",
+        "framework-permission-aidl-cpp",
     ],
 
     sanitize: {
@@ -64,6 +123,13 @@
         symbol_file: "libaaudio.map.txt",
         versions: ["28"],
     },
+
+    tidy: true,
+    tidy_checks: tidy_errors,
+    tidy_checks_as_errors: tidy_errors,
+    tidy_flags: [
+        "-format-style=file",
+    ]
 }
 
 cc_library {
@@ -102,6 +168,7 @@
         "libbinder",
         "framework-permission-aidl-cpp",
         "aaudio-aidl-cpp",
+        "audioclient-types-aidl-cpp",
         "libaudioclient_aidl_conversion",
     ],
 
@@ -156,6 +223,13 @@
         integer_overflow: true,
         misc_undefined: ["bounds"],
     },
+
+    tidy: true,
+    tidy_checks: tidy_errors,
+    tidy_checks_as_errors: tidy_errors,
+    tidy_flags: [
+        "-format-style=file",
+    ]
 }
 
 aidl_interface {
@@ -172,19 +246,14 @@
         "binding/aidl/aaudio/IAAudioService.aidl",
     ],
     imports: [
-        "audio_common-aidl",
+        "audioclient-types-aidl",
         "shared-file-region-aidl",
-        "framework-permission-aidl"
+        "framework-permission-aidl",
     ],
     backend:
     {
-        cpp: {
-            enabled: true,
-        },
         java: {
-            // TODO: need to have audio_common-aidl available in Java to enable
-            //       this.
-            enabled: false,
+            sdk_version: "module_current",
         },
     },
 }
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.cpp b/media/libaaudio/src/binding/AAudioBinderClient.cpp
index fa5a2da..135bac3 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.cpp
+++ b/media/libaaudio/src/binding/AAudioBinderClient.cpp
@@ -36,13 +36,10 @@
 using android::IServiceManager;
 using android::defaultServiceManager;
 using android::interface_cast;
-using android::IInterface;
 using android::Mutex;
 using android::ProcessState;
 using android::sp;
 using android::status_t;
-using android::wp;
-using android::binder::Status;
 
 using namespace aaudio;
 
@@ -93,7 +90,7 @@
                     ALOGE("%s() - linkToDeath() returned %d", __func__, status);
                 }
                 aaudioService = interface_cast<IAAudioService>(binder);
-                mAdapter.reset(new Adapter(aaudioService, mAAudioClient));
+                mAdapter = std::make_shared<Adapter>(aaudioService, mAAudioClient);
                 needToRegister = true;
                 // Make sure callbacks can be received by mAAudioClient
                 ProcessState::self()->startThreadPool();
diff --git a/media/libaaudio/src/binding/AAudioBinderClient.h b/media/libaaudio/src/binding/AAudioBinderClient.h
index 6a7b639..557ced5 100644
--- a/media/libaaudio/src/binding/AAudioBinderClient.h
+++ b/media/libaaudio/src/binding/AAudioBinderClient.h
@@ -108,7 +108,7 @@
         return AAUDIO_ERROR_UNAVAILABLE;
     }
 
-    void onStreamChange(aaudio_handle_t handle, int32_t opcode, int32_t value) {
+    void onStreamChange(aaudio_handle_t /*handle*/, int32_t /*opcode*/, int32_t /*value*/) {
         // TODO This is just a stub so we can have a client Binder to pass to the service.
         // TODO Implemented in a later CL.
         ALOGW("onStreamChange called!");
@@ -116,7 +116,7 @@
 
     class AAudioClient : public android::IBinder::DeathRecipient, public BnAAudioClient {
     public:
-        AAudioClient(android::wp<AAudioBinderClient> aaudioBinderClient)
+        explicit AAudioClient(const android::wp<AAudioBinderClient>& aaudioBinderClient)
                 : mBinderClient(aaudioBinderClient) {
         }
 
@@ -150,10 +150,10 @@
     class Adapter : public AAudioBinderAdapter {
     public:
         Adapter(const android::sp<IAAudioService>& delegate,
-                const android::sp<AAudioClient>& aaudioClient)
+                android::sp<AAudioClient> aaudioClient)
                 : AAudioBinderAdapter(delegate.get()),
                   mDelegate(delegate),
-                  mAAudioClient(aaudioClient) {}
+                  mAAudioClient(std::move(aaudioClient)) {}
 
         virtual ~Adapter() {
             if (mDelegate != nullptr) {
diff --git a/media/libaaudio/src/binding/AAudioServiceInterface.h b/media/libaaudio/src/binding/AAudioServiceInterface.h
index 5d11512..bf94774 100644
--- a/media/libaaudio/src/binding/AAudioServiceInterface.h
+++ b/media/libaaudio/src/binding/AAudioServiceInterface.h
@@ -37,7 +37,7 @@
 class AAudioServiceInterface {
 public:
 
-    AAudioServiceInterface() {};
+    AAudioServiceInterface() = default;
     virtual ~AAudioServiceInterface() = default;
 
     virtual void registerClient(const android::sp<IAAudioClient>& client) = 0;
diff --git a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
index 2d501ef..cf76225 100644
--- a/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamConfiguration.cpp
@@ -23,20 +23,23 @@
 #include <sys/mman.h>
 #include <aaudio/AAudio.h>
 
+#include <media/AidlConversion.h>
+
 #include "binding/AAudioStreamConfiguration.h"
 
 using namespace aaudio;
 
-using android::media::audio::common::AudioFormat;
+using android::media::AudioFormatDescription;
 
 AAudioStreamConfiguration::AAudioStreamConfiguration(const StreamParameters& parcelable) {
-    setSamplesPerFrame(parcelable.samplesPerFrame);
+    setChannelMask(parcelable.channelMask);
     setSampleRate(parcelable.sampleRate);
     setDeviceId(parcelable.deviceId);
     static_assert(sizeof(aaudio_sharing_mode_t) == sizeof(parcelable.sharingMode));
     setSharingMode(parcelable.sharingMode);
-    static_assert(sizeof(audio_format_t) == sizeof(parcelable.audioFormat));
-    setFormat(static_cast<audio_format_t>(parcelable.audioFormat));
+    auto convFormat = android::aidl2legacy_AudioFormatDescription_audio_format_t(
+            parcelable.audioFormat);
+    setFormat(convFormat.ok() ? convFormat.value() : AUDIO_FORMAT_INVALID);
     static_assert(sizeof(aaudio_direction_t) == sizeof(parcelable.direction));
     setDirection(parcelable.direction);
     static_assert(sizeof(audio_usage_t) == sizeof(parcelable.usage));
@@ -63,13 +66,18 @@
 
 StreamParameters AAudioStreamConfiguration::parcelable() const {
     StreamParameters result;
-    result.samplesPerFrame = getSamplesPerFrame();
+    result.channelMask = getChannelMask();
     result.sampleRate = getSampleRate();
     result.deviceId = getDeviceId();
     static_assert(sizeof(aaudio_sharing_mode_t) == sizeof(result.sharingMode));
     result.sharingMode = getSharingMode();
-    static_assert(sizeof(audio_format_t) == sizeof(result.audioFormat));
-    result.audioFormat = static_cast<AudioFormat>(getFormat());
+    auto convAudioFormat = android::legacy2aidl_audio_format_t_AudioFormatDescription(getFormat());
+    if (convAudioFormat.ok()) {
+        result.audioFormat = convAudioFormat.value();
+    } else {
+        result.audioFormat = AudioFormatDescription{};
+        result.audioFormat.type = android::media::AudioFormatType::SYS_RESERVED_INVALID;
+    }
     static_assert(sizeof(aaudio_direction_t) == sizeof(result.direction));
     result.direction = getDirection();
     static_assert(sizeof(audio_usage_t) == sizeof(result.usage));
diff --git a/media/libaaudio/src/binding/AAudioStreamRequest.cpp b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
index 8d90034..a4cc2bd 100644
--- a/media/libaaudio/src/binding/AAudioStreamRequest.cpp
+++ b/media/libaaudio/src/binding/AAudioStreamRequest.cpp
@@ -30,7 +30,7 @@
 using namespace aaudio;
 
 AAudioStreamRequest::AAudioStreamRequest(const StreamRequest& parcelable) :
-        mConfiguration(std::move(parcelable.params)),
+        mConfiguration(parcelable.params),
         mAttributionSource(parcelable.attributionSource),
         mSharingModeMatchRequired(parcelable.sharingModeMatchRequired),
         mInService(parcelable.inService) {
@@ -38,7 +38,7 @@
 
 StreamRequest AAudioStreamRequest::parcelable() const {
     StreamRequest result;
-    result.params = std::move(mConfiguration).parcelable();
+    result.params = mConfiguration.parcelable();
     result.attributionSource = mAttributionSource;
     result.sharingModeMatchRequired = mSharingModeMatchRequired;
     result.inService = mInService;
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
index aa4ac27..dea3e4a 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.cpp
@@ -29,17 +29,15 @@
 #include "binding/AudioEndpointParcelable.h"
 
 using android::base::unique_fd;
-using android::media::SharedFileRegion;
-using android::NO_ERROR;
 using android::status_t;
 
 using namespace aaudio;
 
 AudioEndpointParcelable::AudioEndpointParcelable(Endpoint&& parcelable)
-        : mUpMessageQueueParcelable(std::move(parcelable.upMessageQueueParcelable)),
-          mDownMessageQueueParcelable(std::move(parcelable.downMessageQueueParcelable)),
-          mUpDataQueueParcelable(std::move(parcelable.upDataQueueParcelable)),
-          mDownDataQueueParcelable(std::move(parcelable.downDataQueueParcelable)),
+        : mUpMessageQueueParcelable(parcelable.upMessageQueueParcelable),
+          mDownMessageQueueParcelable(parcelable.downMessageQueueParcelable),
+          mUpDataQueueParcelable(parcelable.upDataQueueParcelable),
+          mDownDataQueueParcelable(parcelable.downDataQueueParcelable),
           mNumSharedMemories(parcelable.sharedMemories.size()) {
     for (size_t i = 0; i < parcelable.sharedMemories.size() && i < MAX_SHARED_MEMORIES; ++i) {
         // Re-construct.
@@ -56,10 +54,10 @@
 
 Endpoint AudioEndpointParcelable::parcelable()&& {
     Endpoint result;
-    result.upMessageQueueParcelable = std::move(mUpMessageQueueParcelable).parcelable();
-    result.downMessageQueueParcelable = std::move(mDownMessageQueueParcelable).parcelable();
-    result.upDataQueueParcelable = std::move(mUpDataQueueParcelable).parcelable();
-    result.downDataQueueParcelable = std::move(mDownDataQueueParcelable).parcelable();
+    result.upMessageQueueParcelable = mUpMessageQueueParcelable.parcelable();
+    result.downMessageQueueParcelable = mDownMessageQueueParcelable.parcelable();
+    result.upDataQueueParcelable = mUpDataQueueParcelable.parcelable();
+    result.downDataQueueParcelable = mDownDataQueueParcelable.parcelable();
     result.sharedMemories.reserve(std::min(mNumSharedMemories, MAX_SHARED_MEMORIES));
     for (size_t i = 0; i < mNumSharedMemories && i < MAX_SHARED_MEMORIES; ++i) {
         result.sharedMemories.emplace_back(std::move(mSharedMemories[i]).parcelable());
diff --git a/media/libaaudio/src/binding/AudioEndpointParcelable.h b/media/libaaudio/src/binding/AudioEndpointParcelable.h
index 5237a1a..544aa92 100644
--- a/media/libaaudio/src/binding/AudioEndpointParcelable.h
+++ b/media/libaaudio/src/binding/AudioEndpointParcelable.h
@@ -43,7 +43,7 @@
     // Ctor/assignment from a parcelable representation.
     // Since the parcelable object owns unique FDs (for shared memory blocks), move semantics are
     // provided to avoid the need to dupe.
-    AudioEndpointParcelable(Endpoint&& parcelable);
+    explicit AudioEndpointParcelable(Endpoint&& parcelable);
     AudioEndpointParcelable& operator=(Endpoint&& parcelable);
 
     /**
diff --git a/media/libaaudio/src/binding/RingBufferParcelable.cpp b/media/libaaudio/src/binding/RingBufferParcelable.cpp
index a4b3cec..fa7ca72 100644
--- a/media/libaaudio/src/binding/RingBufferParcelable.cpp
+++ b/media/libaaudio/src/binding/RingBufferParcelable.cpp
@@ -30,9 +30,9 @@
 using namespace aaudio;
 
 RingBufferParcelable::RingBufferParcelable(const RingBuffer& parcelable)
-        : mReadCounterParcelable(std::move(parcelable.readCounterParcelable)),
-          mWriteCounterParcelable(std::move(parcelable.writeCounterParcelable)),
-          mDataParcelable(std::move(parcelable.dataParcelable)),
+        : mReadCounterParcelable(parcelable.readCounterParcelable),
+          mWriteCounterParcelable(parcelable.writeCounterParcelable),
+          mDataParcelable(parcelable.dataParcelable),
           mBytesPerFrame(parcelable.bytesPerFrame),
           mFramesPerBurst(parcelable.framesPerBurst),
           mCapacityInFrames(parcelable.capacityInFrames),
@@ -42,9 +42,9 @@
 
 RingBuffer RingBufferParcelable::parcelable() const {
     RingBuffer result;
-    result.readCounterParcelable = std::move(mReadCounterParcelable).parcelable();
-    result.writeCounterParcelable = std::move(mWriteCounterParcelable).parcelable();
-    result.dataParcelable = std::move(mDataParcelable).parcelable();
+    result.readCounterParcelable = mReadCounterParcelable.parcelable();
+    result.writeCounterParcelable = mWriteCounterParcelable.parcelable();
+    result.dataParcelable = mDataParcelable.parcelable();
     result.bytesPerFrame = mBytesPerFrame;
     result.framesPerBurst = mFramesPerBurst;
     result.capacityInFrames = mCapacityInFrames;
diff --git a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
index eef238f..3a49655 100644
--- a/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedMemoryParcelable.cpp
@@ -32,7 +32,6 @@
 #include "binding/SharedMemoryParcelable.h"
 
 using android::base::unique_fd;
-using android::NO_ERROR;
 using android::status_t;
 using android::media::SharedFileRegion;
 
@@ -78,7 +77,7 @@
 }
 
 aaudio_result_t SharedMemoryParcelable::resolveSharedMemory(const unique_fd& fd) {
-    mResolvedAddress = (uint8_t *) mmap(0, mSizeInBytes, PROT_READ | PROT_WRITE,
+    mResolvedAddress = (uint8_t *) mmap(nullptr, mSizeInBytes, PROT_READ | PROT_WRITE,
                                         MAP_SHARED, fd.get(), 0);
     if (mResolvedAddress == MMAP_UNRESOLVED_ADDRESS) {
         ALOGE("mmap() failed for fd = %d, nBytes = %" PRId64 ", errno = %s",
diff --git a/media/libaaudio/src/binding/SharedRegionParcelable.cpp b/media/libaaudio/src/binding/SharedRegionParcelable.cpp
index 56b99c0..6fa109b 100644
--- a/media/libaaudio/src/binding/SharedRegionParcelable.cpp
+++ b/media/libaaudio/src/binding/SharedRegionParcelable.cpp
@@ -29,10 +29,7 @@
 #include "binding/SharedMemoryParcelable.h"
 #include "binding/SharedRegionParcelable.h"
 
-using android::NO_ERROR;
 using android::status_t;
-using android::Parcel;
-using android::Parcelable;
 
 using namespace aaudio;
 
diff --git a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
index b7c4f70..3b274f2 100644
--- a/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
+++ b/media/libaaudio/src/binding/aidl/aaudio/StreamParameters.aidl
@@ -16,14 +16,14 @@
 
 package aaudio;
 
-import android.media.audio.common.AudioFormat;
+import android.media.AudioFormatDescription;
 
 parcelable StreamParameters {
-    int                                       samplesPerFrame;  //      = AAUDIO_UNSPECIFIED;
+    int                                       channelMask;  //          = AAUDIO_UNSPECIFIED;
     int                                       sampleRate;  //           = AAUDIO_UNSPECIFIED;
     int                                       deviceId;  //             = AAUDIO_UNSPECIFIED;
     int /* aaudio_sharing_mode_t */           sharingMode;  //          = AAUDIO_SHARING_MODE_SHARED;
-    AudioFormat                               audioFormat;  //          = AUDIO_FORMAT_DEFAULT;
+    AudioFormatDescription                    audioFormat;  //          = AUDIO_FORMAT_DEFAULT;
     int /* aaudio_direction_t */              direction;  //            = AAUDIO_DIRECTION_OUTPUT;
     int /* aaudio_usage_t */                  usage;  //                = AAUDIO_UNSPECIFIED;
     int /* aaudio_content_type_t */           contentType;  //          = AAUDIO_UNSPECIFIED;
diff --git a/media/libaaudio/src/client/AudioEndpoint.cpp b/media/libaaudio/src/client/AudioEndpoint.cpp
index ebc9f2b..24888de 100644
--- a/media/libaaudio/src/client/AudioEndpoint.cpp
+++ b/media/libaaudio/src/client/AudioEndpoint.cpp
@@ -31,13 +31,6 @@
 #define RIDICULOUSLY_LARGE_BUFFER_CAPACITY   (256 * 1024)
 #define RIDICULOUSLY_LARGE_FRAME_SIZE        4096
 
-AudioEndpoint::AudioEndpoint()
-    : mFreeRunning(false)
-    , mDataReadCounter(0)
-    , mDataWriteCounter(0)
-{
-}
-
 // TODO Consider moving to a method in RingBufferDescriptor
 static aaudio_result_t AudioEndpoint_validateQueueDescriptor(const char *type,
                                                   const RingBufferDescriptor *descriptor) {
diff --git a/media/libaaudio/src/client/AudioEndpoint.h b/media/libaaudio/src/client/AudioEndpoint.h
index 4c8d60f..b3dbc20 100644
--- a/media/libaaudio/src/client/AudioEndpoint.h
+++ b/media/libaaudio/src/client/AudioEndpoint.h
@@ -34,7 +34,7 @@
 class AudioEndpoint {
 
 public:
-    AudioEndpoint();
+    AudioEndpoint() = default;
 
     /**
      * Configure based on the EndPointDescriptor_t.
@@ -95,9 +95,9 @@
 private:
     std::unique_ptr<android::FifoBufferIndirect> mUpCommandQueue;
     std::unique_ptr<android::FifoBufferIndirect> mDataQueue;
-    bool                    mFreeRunning;
-    android::fifo_counter_t mDataReadCounter; // only used if free-running
-    android::fifo_counter_t mDataWriteCounter; // only used if free-running
+    bool                    mFreeRunning{false};
+    android::fifo_counter_t mDataReadCounter{0}; // only used if free-running
+    android::fifo_counter_t mDataWriteCounter{0}; // only used if free-running
 };
 
 } // namespace aaudio
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index cf2abe8..91f3004 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -49,8 +49,6 @@
 // This is needed to make sense of the logs more easily.
 #define LOG_TAG (mInService ? "AudioStreamInternal_Service" : "AudioStreamInternal_Client")
 
-using android::Mutex;
-using android::WrappingBuffer;
 using android::content::AttributionSourceState;
 
 using namespace aaudio;
@@ -123,9 +121,9 @@
 
     request.getConfiguration().setDeviceId(getDeviceId());
     request.getConfiguration().setSampleRate(getSampleRate());
-    request.getConfiguration().setSamplesPerFrame(getSamplesPerFrame());
     request.getConfiguration().setDirection(getDirection());
     request.getConfiguration().setSharingMode(getSharingMode());
+    request.getConfiguration().setChannelMask(getChannelMask());
 
     request.getConfiguration().setUsage(getUsage());
     request.getConfiguration().setContentType(getContentType());
@@ -138,7 +136,8 @@
 
     mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
     if (mServiceStreamHandle < 0
-            && request.getConfiguration().getSamplesPerFrame() == 1 // mono?
+            && (request.getConfiguration().getSamplesPerFrame() == 1
+                    || request.getConfiguration().getChannelMask() == AAUDIO_CHANNEL_MONO)
             && getDirection() == AAUDIO_DIRECTION_OUTPUT
             && !isInService()) {
         // if that failed then try switching from mono to stereo if OUTPUT.
@@ -146,7 +145,7 @@
         // that writes to a stereo MMAP stream.
         ALOGD("%s() - openStream() returned %d, try switching from MONO to STEREO",
               __func__, mServiceStreamHandle);
-        request.getConfiguration().setSamplesPerFrame(2); // stereo
+        request.getConfiguration().setChannelMask(AAUDIO_CHANNEL_STEREO);
         mServiceStreamHandle = mServiceInterface.openStream(request, configurationOutput);
     }
     if (mServiceStreamHandle < 0) {
@@ -171,9 +170,10 @@
         goto error;
     }
     // Save results of the open.
-    if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
-        setSamplesPerFrame(configurationOutput.getSamplesPerFrame());
+    if (getChannelMask() == AAUDIO_UNSPECIFIED) {
+        setChannelMask(configurationOutput.getChannelMask());
     }
+
     mDeviceChannelCount = configurationOutput.getSamplesPerFrame();
 
     setSampleRate(configurationOutput.getSampleRate());
@@ -329,10 +329,10 @@
 {
     AudioStreamInternal *stream = (AudioStreamInternal *)context;
     //LOGD("oboe_callback_thread, stream = %p", stream);
-    if (stream != NULL) {
+    if (stream != nullptr) {
         return stream->callbackLoop();
     } else {
-        return NULL;
+        return nullptr;
     }
 }
 
@@ -421,7 +421,7 @@
     if (isDataCallbackSet()
             && (isActive() || getState() == AAUDIO_STREAM_STATE_DISCONNECTED)) {
         mCallbackEnabled.store(false);
-        aaudio_result_t result = joinThread_l(NULL); // may temporarily unlock mStreamLock
+        aaudio_result_t result = joinThread_l(nullptr); // may temporarily unlock mStreamLock
         if (result == AAUDIO_ERROR_INVALID_HANDLE) {
             ALOGD("%s() INVALID_HANDLE, stream was probably stolen", __func__);
             result = AAUDIO_OK;
@@ -508,7 +508,7 @@
     return result;
 }
 
-aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t clockId,
+aaudio_result_t AudioStreamInternal::getTimestamp(clockid_t /*clockId*/,
                            int64_t *framePosition,
                            int64_t *timeNanoseconds) {
     // Generated in server and passed to client. Return latest.
diff --git a/media/libaaudio/src/client/AudioStreamInternal.h b/media/libaaudio/src/client/AudioStreamInternal.h
index fbe4c13..eab1382 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.h
+++ b/media/libaaudio/src/client/AudioStreamInternal.h
@@ -116,7 +116,7 @@
 
     virtual void prepareBuffersForStart() {}
 
-    virtual void advanceClientToMatchServerPosition(int32_t serverMargin = 0) = 0;
+    virtual void advanceClientToMatchServerPosition(int32_t serverMargin) = 0;
 
     virtual void onFlushFromServer() {}
 
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
index 2da5406..1efccb1 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.cpp
@@ -46,8 +46,6 @@
 
 }
 
-AudioStreamInternalCapture::~AudioStreamInternalCapture() {}
-
 void AudioStreamInternalCapture::advanceClientToMatchServerPosition(int32_t serverMargin) {
     int64_t readCounter = mAudioEndpoint->getDataReadCounter();
     int64_t writeCounter = mAudioEndpoint->getDataWriteCounter() + serverMargin;
@@ -109,7 +107,7 @@
     if (mNeedCatchUp.isRequested()) {
         // Catch an MMAP pointer that is already advancing.
         // This will avoid initial underruns caused by a slow cold start.
-        advanceClientToMatchServerPosition();
+        advanceClientToMatchServerPosition(0 /*serverMargin*/);
         mNeedCatchUp.acknowledge();
     }
 
@@ -228,7 +226,7 @@
 void *AudioStreamInternalCapture::callbackLoop() {
     aaudio_result_t result = AAUDIO_OK;
     aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
-    if (!isDataCallbackSet()) return NULL;
+    if (!isDataCallbackSet()) return nullptr;
 
     // result might be a frame count
     while (mCallbackEnabled.load() && isActive() && (result >= 0)) {
@@ -260,5 +258,5 @@
 
     ALOGD("callbackLoop() exiting, result = %d, isActive() = %d",
           result, (int) isActive());
-    return NULL;
+    return nullptr;
 }
diff --git a/media/libaaudio/src/client/AudioStreamInternalCapture.h b/media/libaaudio/src/client/AudioStreamInternalCapture.h
index 251a7f2..87017de 100644
--- a/media/libaaudio/src/client/AudioStreamInternalCapture.h
+++ b/media/libaaudio/src/client/AudioStreamInternalCapture.h
@@ -28,8 +28,9 @@
 
 class AudioStreamInternalCapture : public AudioStreamInternal {
 public:
-    AudioStreamInternalCapture(AAudioServiceInterface  &serviceInterface, bool inService = false);
-    virtual ~AudioStreamInternalCapture();
+    explicit AudioStreamInternalCapture(AAudioServiceInterface  &serviceInterface,
+                                        bool inService = false);
+    virtual ~AudioStreamInternalCapture() = default;
 
     aaudio_result_t read(void *buffer,
                          int32_t numFrames,
@@ -45,7 +46,7 @@
     }
 protected:
 
-    void advanceClientToMatchServerPosition(int32_t serverOffset = 0) override;
+    void advanceClientToMatchServerPosition(int32_t serverOffset) override;
 
 /**
  * Low level data processing that will not block. It will just read or write as much as it can.
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
index 71bde90..5921799 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.cpp
@@ -44,8 +44,6 @@
 
 }
 
-AudioStreamInternalPlay::~AudioStreamInternalPlay() {}
-
 constexpr int kRampMSec = 10; // time to apply a change in volume
 
 aaudio_result_t AudioStreamInternalPlay::open(const AudioStreamBuilder &builder) {
@@ -115,7 +113,7 @@
 }
 
 void AudioStreamInternalPlay::onFlushFromServer() {
-    advanceClientToMatchServerPosition();
+    advanceClientToMatchServerPosition(0 /*serverMargin*/);
 }
 
 // Write the data, block if needed and timeoutMillis > 0
@@ -281,7 +279,7 @@
     ALOGD("%s() entering >>>>>>>>>>>>>>>", __func__);
     aaudio_result_t result = AAUDIO_OK;
     aaudio_data_callback_result_t callbackResult = AAUDIO_CALLBACK_RESULT_CONTINUE;
-    if (!isDataCallbackSet()) return NULL;
+    if (!isDataCallbackSet()) return nullptr;
     int64_t timeoutNanos = calculateReasonableTimeout(mCallbackFrames);
 
     // result might be a frame count
@@ -309,7 +307,7 @@
 
     ALOGD("%s() exiting, result = %d, isActive() = %d <<<<<<<<<<<<<<",
           __func__, result, (int) isActive());
-    return NULL;
+    return nullptr;
 }
 
 //------------------------------------------------------------------------------
diff --git a/media/libaaudio/src/client/AudioStreamInternalPlay.h b/media/libaaudio/src/client/AudioStreamInternalPlay.h
index 03c957d..e761807 100644
--- a/media/libaaudio/src/client/AudioStreamInternalPlay.h
+++ b/media/libaaudio/src/client/AudioStreamInternalPlay.h
@@ -30,8 +30,9 @@
 
 class AudioStreamInternalPlay : public AudioStreamInternal {
 public:
-    AudioStreamInternalPlay(AAudioServiceInterface  &serviceInterface, bool inService = false);
-    virtual ~AudioStreamInternalPlay();
+    explicit AudioStreamInternalPlay(AAudioServiceInterface  &serviceInterface,
+                                     bool inService = false);
+    virtual ~AudioStreamInternalPlay() = default;
 
     aaudio_result_t open(const AudioStreamBuilder &builder) override;
 
@@ -66,7 +67,7 @@
 
     void prepareBuffersForStart() override;
 
-    void advanceClientToMatchServerPosition(int32_t serverMargin = 0) override;
+    void advanceClientToMatchServerPosition(int32_t serverMargin) override;
 
     void onFlushFromServer() override;
 
diff --git a/media/libaaudio/src/client/IsochronousClockModel.cpp b/media/libaaudio/src/client/IsochronousClockModel.cpp
index f0dcd44..6921271 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.cpp
+++ b/media/libaaudio/src/client/IsochronousClockModel.cpp
@@ -43,14 +43,7 @@
 // and dumped to the log when the stream is stopped.
 
 IsochronousClockModel::IsochronousClockModel()
-        : mMarkerFramePosition(0)
-        , mMarkerNanoTime(0)
-        , mSampleRate(48000)
-        , mFramesPerBurst(48)
-        , mBurstPeriodNanos(0) // this will be updated before use
-        , mMaxMeasuredLatenessNanos(0)
-        , mLatenessForDriftNanos(kInitialLatenessForDriftNanos)
-        , mState(STATE_STOPPED)
+        : mLatenessForDriftNanos(kInitialLatenessForDriftNanos)
 {
     if ((AAudioProperty_getLogMask() & AAUDIO_LOG_CLOCK_MODEL_HISTOGRAM) != 0) {
         mHistogramMicros = std::make_unique<Histogram>(kHistogramBinCount,
diff --git a/media/libaaudio/src/client/IsochronousClockModel.h b/media/libaaudio/src/client/IsochronousClockModel.h
index 6280013..3007237 100644
--- a/media/libaaudio/src/client/IsochronousClockModel.h
+++ b/media/libaaudio/src/client/IsochronousClockModel.h
@@ -149,16 +149,16 @@
     static constexpr int32_t   kHistogramBinWidthMicros = 50;
     static constexpr int32_t   kHistogramBinCount = 128;
 
-    int64_t             mMarkerFramePosition; // Estimated HW position.
-    int64_t             mMarkerNanoTime;      // Estimated HW time.
-    int32_t             mSampleRate;
-    int32_t             mFramesPerBurst;      // number of frames transferred at one time.
-    int32_t             mBurstPeriodNanos;    // Time between HW bursts.
+    int64_t             mMarkerFramePosition{0}; // Estimated HW position.
+    int64_t             mMarkerNanoTime{0};      // Estimated HW time.
+    int32_t             mSampleRate{48000};
+    int32_t             mFramesPerBurst{48};     // number of frames transferred at one time.
+    int32_t             mBurstPeriodNanos{0};    // Time between HW bursts.
     // Includes mBurstPeriodNanos because we sample randomly over time.
-    int32_t             mMaxMeasuredLatenessNanos;
+    int32_t             mMaxMeasuredLatenessNanos{0};
     // Threshold for lateness that triggers a drift later in time.
     int32_t             mLatenessForDriftNanos;
-    clock_model_state_t mState;               // State machine handles startup sequence.
+    clock_model_state_t mState{STATE_STOPPED};   // State machine handles startup sequence.
 
     int32_t             mTimestampCount = 0;  // For logging.
 
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index d103aca..02e7f5f 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -128,7 +128,8 @@
                                                        int32_t samplesPerFrame)
 {
     AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
-    streamBuilder->setSamplesPerFrame(samplesPerFrame);
+    const aaudio_channel_mask_t channelMask = AAudioConvert_channelCountToMask(samplesPerFrame);
+    streamBuilder->setChannelMask(channelMask);
 }
 
 AAUDIO_API void AAudioStreamBuilder_setDirection(AAudioStreamBuilder* builder,
@@ -223,6 +224,13 @@
     streamBuilder->setFramesPerDataCallback(frames);
 }
 
+AAUDIO_API void AAudioStreamBuilder_setChannelMask(AAudioStreamBuilder* builder,
+                                                   aaudio_channel_mask_t channelMask)
+{
+    AudioStreamBuilder *streamBuilder = convertAAudioBuilderToStreamBuilder(builder);
+    streamBuilder->setChannelMask(channelMask);
+}
+
 AAUDIO_API aaudio_result_t  AAudioStreamBuilder_openStream(AAudioStreamBuilder* builder,
                                                      AAudioStream** streamPtr)
 {
@@ -562,3 +570,11 @@
     AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
     return audioStream->isPrivacySensitive();
 }
+
+AAUDIO_API aaudio_channel_mask_t AAudioStream_getChannelMask(AAudioStream* stream)
+{
+    AudioStream *audioStream = convertAAudioStreamToAudioStream(stream);
+    const aaudio_channel_mask_t channelMask = audioStream->getChannelMask();
+    // Do not return channel index masks as they are not public.
+    return AAudio_isChannelIndexMask(channelMask) ? AAUDIO_UNSPECIFIED : channelMask;
+}
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.cpp b/media/libaaudio/src/core/AAudioStreamParameters.cpp
index acfac24..84133f4 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.cpp
+++ b/media/libaaudio/src/core/AAudioStreamParameters.cpp
@@ -30,9 +30,6 @@
 // HDMI supports up to 32 channels at 1536000 Hz.
 #define SAMPLE_RATE_HZ_MAX           1600000
 
-AAudioStreamParameters::AAudioStreamParameters() {}
-AAudioStreamParameters::~AAudioStreamParameters() {}
-
 void AAudioStreamParameters::copyFrom(const AAudioStreamParameters &other) {
     mSamplesPerFrame      = other.mSamplesPerFrame;
     mSampleRate           = other.mSampleRate;
@@ -49,6 +46,7 @@
     mIsPrivacySensitive   = other.mIsPrivacySensitive;
     mOpPackageName        = other.mOpPackageName;
     mAttributionTag       = other.mAttributionTag;
+    mChannelMask          = other.mChannelMask;
 }
 
 static aaudio_result_t isFormatValid(audio_format_t format) {
@@ -187,7 +185,94 @@
             // break;
     }
 
-    return AAUDIO_OK;
+    return validateChannelMask();
+}
+
+bool AAudioStreamParameters::validateChannelMask() const {
+    if (mChannelMask == AAUDIO_UNSPECIFIED) {
+        return AAUDIO_OK;
+    }
+
+    if (mChannelMask & AAUDIO_CHANNEL_BIT_INDEX) {
+        switch (mChannelMask) {
+            case AAUDIO_CHANNEL_INDEX_MASK_1:
+            case AAUDIO_CHANNEL_INDEX_MASK_2:
+            case AAUDIO_CHANNEL_INDEX_MASK_3:
+            case AAUDIO_CHANNEL_INDEX_MASK_4:
+            case AAUDIO_CHANNEL_INDEX_MASK_5:
+            case AAUDIO_CHANNEL_INDEX_MASK_6:
+            case AAUDIO_CHANNEL_INDEX_MASK_7:
+            case AAUDIO_CHANNEL_INDEX_MASK_8:
+            case AAUDIO_CHANNEL_INDEX_MASK_9:
+            case AAUDIO_CHANNEL_INDEX_MASK_10:
+            case AAUDIO_CHANNEL_INDEX_MASK_11:
+            case AAUDIO_CHANNEL_INDEX_MASK_12:
+            case AAUDIO_CHANNEL_INDEX_MASK_13:
+            case AAUDIO_CHANNEL_INDEX_MASK_14:
+            case AAUDIO_CHANNEL_INDEX_MASK_15:
+            case AAUDIO_CHANNEL_INDEX_MASK_16:
+            case AAUDIO_CHANNEL_INDEX_MASK_17:
+            case AAUDIO_CHANNEL_INDEX_MASK_18:
+            case AAUDIO_CHANNEL_INDEX_MASK_19:
+            case AAUDIO_CHANNEL_INDEX_MASK_20:
+            case AAUDIO_CHANNEL_INDEX_MASK_21:
+            case AAUDIO_CHANNEL_INDEX_MASK_22:
+            case AAUDIO_CHANNEL_INDEX_MASK_23:
+            case AAUDIO_CHANNEL_INDEX_MASK_24:
+                return AAUDIO_OK;
+            default:
+                ALOGD("Invalid channel index mask %#x", mChannelMask);
+                return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+        }
+    }
+
+    if (getDirection() == AAUDIO_DIRECTION_INPUT) {
+        switch (mChannelMask) {
+            case AAUDIO_CHANNEL_MONO:
+            case AAUDIO_CHANNEL_STEREO:
+            case AAUDIO_CHANNEL_FRONT_BACK:
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+            case AAUDIO_CHANNEL_5POINT1:
+                return AAUDIO_OK;
+            default:
+                ALOGD("Invalid channel mask %#x, IN", mChannelMask);
+                return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+        }
+    } else {
+        switch (mChannelMask) {
+            case AAUDIO_CHANNEL_MONO:
+            case AAUDIO_CHANNEL_STEREO:
+            case AAUDIO_CHANNEL_2POINT1:
+            case AAUDIO_CHANNEL_TRI:
+            case AAUDIO_CHANNEL_TRI_BACK:
+            case AAUDIO_CHANNEL_3POINT1:
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+            case AAUDIO_CHANNEL_QUAD:
+            case AAUDIO_CHANNEL_QUAD_SIDE:
+            case AAUDIO_CHANNEL_SURROUND:
+            case AAUDIO_CHANNEL_PENTA:
+            case AAUDIO_CHANNEL_5POINT1:
+            case AAUDIO_CHANNEL_5POINT1_SIDE:
+            case AAUDIO_CHANNEL_5POINT1POINT2:
+            case AAUDIO_CHANNEL_5POINT1POINT4:
+            case AAUDIO_CHANNEL_6POINT1:
+            case AAUDIO_CHANNEL_7POINT1:
+            case AAUDIO_CHANNEL_7POINT1POINT2:
+            case AAUDIO_CHANNEL_7POINT1POINT4:
+            case AAUDIO_CHANNEL_9POINT1POINT4:
+            case AAUDIO_CHANNEL_9POINT1POINT6:
+                return AAUDIO_OK;
+            default:
+                ALOGD("Invalid channel mask %#x. OUT", mChannelMask);
+                return AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+        }
+    }
 }
 
 void AAudioStreamParameters::dump() const {
@@ -195,6 +280,7 @@
     ALOGD("mSessionId            = %6d", mSessionId);
     ALOGD("mSampleRate           = %6d", mSampleRate);
     ALOGD("mSamplesPerFrame      = %6d", mSamplesPerFrame);
+    ALOGD("mChannelMask          = %#x", mChannelMask);
     ALOGD("mSharingMode          = %6d", (int)mSharingMode);
     ALOGD("mAudioFormat          = %6d", (int)mAudioFormat);
     ALOGD("mDirection            = %6d", mDirection);
diff --git a/media/libaaudio/src/core/AAudioStreamParameters.h b/media/libaaudio/src/core/AAudioStreamParameters.h
index 5737052..7a6ed3e 100644
--- a/media/libaaudio/src/core/AAudioStreamParameters.h
+++ b/media/libaaudio/src/core/AAudioStreamParameters.h
@@ -26,8 +26,8 @@
 
 class AAudioStreamParameters {
 public:
-    AAudioStreamParameters();
-    virtual ~AAudioStreamParameters();
+    AAudioStreamParameters() = default;
+    virtual ~AAudioStreamParameters() = default;
 
     int32_t getDeviceId() const {
         return mDeviceId;
@@ -49,13 +49,6 @@
         return mSamplesPerFrame;
     }
 
-    /**
-     * This is also known as channelCount.
-     */
-    void setSamplesPerFrame(int32_t samplesPerFrame) {
-        mSamplesPerFrame = samplesPerFrame;
-    }
-
     audio_format_t getFormat() const {
         return mAudioFormat;
     }
@@ -141,7 +134,7 @@
     }
 
     // TODO b/182392769: reexamine if Identity can be used
-    void setOpPackageName(const std::optional<std::string> opPackageName) {
+    void setOpPackageName(const std::optional<std::string>& opPackageName) {
         mOpPackageName = opPackageName;
     }
 
@@ -149,10 +142,19 @@
         return mAttributionTag;
     }
 
-    void setAttributionTag(const std::optional<std::string> attributionTag) {
+    void setAttributionTag(const std::optional<std::string>& attributionTag) {
         mAttributionTag = attributionTag;
     }
 
+    aaudio_channel_mask_t getChannelMask() const {
+        return mChannelMask;
+    }
+
+    void setChannelMask(aaudio_channel_mask_t channelMask) {
+        mChannelMask = channelMask;
+        mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
+    }
+
     /**
      * @return bytes per frame of getFormat()
      */
@@ -171,6 +173,8 @@
     void dump() const;
 
 private:
+    bool validateChannelMask() const;
+
     int32_t                         mSamplesPerFrame      = AAUDIO_UNSPECIFIED;
     int32_t                         mSampleRate           = AAUDIO_UNSPECIFIED;
     int32_t                         mDeviceId             = AAUDIO_UNSPECIFIED;
@@ -186,6 +190,7 @@
     bool                            mIsPrivacySensitive   = false;
     std::optional<std::string>      mOpPackageName        = {};
     std::optional<std::string>      mAttributionTag       = {};
+    aaudio_channel_mask_t           mChannelMask          = AAUDIO_UNSPECIFIED;
 };
 
 } /* namespace aaudio */
diff --git a/media/libaaudio/src/core/AudioGlobal.h b/media/libaaudio/src/core/AudioGlobal.h
index 1e88d15..6c22744 100644
--- a/media/libaaudio/src/core/AudioGlobal.h
+++ b/media/libaaudio/src/core/AudioGlobal.h
@@ -31,7 +31,8 @@
 const char* AudioGlobal_convertResultToText(aaudio_result_t returnCode);
 const char* AudioGlobal_convertSharingModeToText(aaudio_sharing_mode_t mode);
 const char* AudioGlobal_convertStreamStateToText(aaudio_stream_state_t state);
-}
+
+} // namespace aaudio
 
 #endif  // AAUDIO_AUDIOGLOBAL_H
 
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 09d9535..ffc3b9d 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -76,6 +76,7 @@
     // Copy parameters from the Builder because the Builder may be deleted after this call.
     // TODO AudioStream should be a subclass of AudioStreamParameters
     mSamplesPerFrame = builder.getSamplesPerFrame();
+    mChannelMask = builder.getChannelMask();
     mSampleRate = builder.getSampleRate();
     mDeviceId = builder.getDeviceId();
     mFormat = builder.getFormat();
diff --git a/media/libaaudio/src/core/AudioStream.h b/media/libaaudio/src/core/AudioStream.h
index 9835c8c..8bb9b19 100644
--- a/media/libaaudio/src/core/AudioStream.h
+++ b/media/libaaudio/src/core/AudioStream.h
@@ -270,7 +270,8 @@
     }
 
     /**
-     * This is only valid after setSamplesPerFrame() and setFormat() have been called.
+     * This is only valid after setChannelMask() and setFormat()
+     * have been called.
      */
     int32_t getBytesPerFrame() const {
         return mSamplesPerFrame * getBytesPerSample();
@@ -284,7 +285,7 @@
     }
 
     /**
-     * This is only valid after setSamplesPerFrame() and setDeviceFormat() have been called.
+     * This is only valid after setChannelMask() and setDeviceFormat() have been called.
      */
     int32_t getBytesPerDeviceFrame() const {
         return getSamplesPerFrame() * audio_bytes_per_sample(getDeviceFormat());
@@ -318,6 +319,15 @@
         return mFramesPerDataCallback;
     }
 
+    aaudio_channel_mask_t getChannelMask() const {
+        return mChannelMask;
+    }
+
+    void setChannelMask(aaudio_channel_mask_t channelMask) {
+        mChannelMask = channelMask;
+        mSamplesPerFrame = AAudioConvert_channelMaskToCount(channelMask);
+    }
+
     /**
      * @return true if data callback has been specified
      */
@@ -429,7 +439,7 @@
     // PlayerBase allows the system to control the stream volume.
     class MyPlayerBase : public android::PlayerBase {
     public:
-        MyPlayerBase() {};
+        MyPlayerBase() = default;
 
         virtual ~MyPlayerBase() = default;
 
@@ -495,11 +505,6 @@
     }
 
     // This should not be called after the open() call.
-    void setSamplesPerFrame(int32_t samplesPerFrame) {
-        mSamplesPerFrame = samplesPerFrame;
-    }
-
-    // This should not be called after the open() call.
     void setFramesPerBurst(int32_t framesPerBurst) {
         mFramesPerBurst = framesPerBurst;
     }
@@ -563,7 +568,7 @@
      * @param numFrames
      * @return original pointer or the conversion buffer
      */
-    virtual const void * maybeConvertDeviceData(const void *audioData, int32_t numFrames) {
+    virtual const void * maybeConvertDeviceData(const void *audioData, int32_t /*numFrames*/) {
         return audioData;
     }
 
@@ -633,6 +638,7 @@
 
     // These do not change after open().
     int32_t                     mSamplesPerFrame = AAUDIO_UNSPECIFIED;
+    aaudio_channel_mask_t       mChannelMask = AAUDIO_UNSPECIFIED;
     int32_t                     mSampleRate = AAUDIO_UNSPECIFIED;
     int32_t                     mDeviceId = AAUDIO_UNSPECIFIED;
     aaudio_sharing_mode_t       mSharingMode = AAUDIO_SHARING_MODE_SHARED;
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index e015592..a8fd0d9 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -53,16 +53,10 @@
 /*
  * AudioStreamBuilder
  */
-AudioStreamBuilder::AudioStreamBuilder() {
-}
-
-AudioStreamBuilder::~AudioStreamBuilder() {
-}
-
 static aaudio_result_t builder_createStream(aaudio_direction_t direction,
-                                         aaudio_sharing_mode_t sharingMode,
-                                         bool tryMMap,
-                                         android::sp<AudioStream> &stream) {
+                                            aaudio_sharing_mode_t /*sharingMode*/,
+                                            bool tryMMap,
+                                            android::sp<AudioStream> &stream) {
     aaudio_result_t result = AAUDIO_OK;
 
     switch (direction) {
@@ -268,8 +262,8 @@
 
 void AudioStreamBuilder::logParameters() const {
     // This is very helpful for debugging in the future. Please leave it in.
-    ALOGI("rate   = %6d, channels  = %d, format   = %d, sharing = %s, dir = %s",
-          getSampleRate(), getSamplesPerFrame(), getFormat(),
+    ALOGI("rate   = %6d, channels  = %d, channelMask = %#x, format   = %d, sharing = %s, dir = %s",
+          getSampleRate(), getSamplesPerFrame(), getChannelMask(), getFormat(),
           AAudio_convertSharingModeToShortText(getSharingMode()),
           AAudio_convertDirectionToText(getDirection()));
     ALOGI("device = %6d, sessionId = %d, perfMode = %d, callback: %s with frames = %d",
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.h b/media/libaaudio/src/core/AudioStreamBuilder.h
index 9f93341..f91c25a 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.h
+++ b/media/libaaudio/src/core/AudioStreamBuilder.h
@@ -31,9 +31,9 @@
  */
 class AudioStreamBuilder : public AAudioStreamParameters {
 public:
-    AudioStreamBuilder();
+    AudioStreamBuilder() = default;
 
-    ~AudioStreamBuilder();
+    ~AudioStreamBuilder() = default;
 
     bool isSharingModeMatchRequired() const {
         return mSharingModeMatchRequired;
diff --git a/media/libaaudio/src/fifo/FifoBuffer.h b/media/libaaudio/src/fifo/FifoBuffer.h
index 37548f0..7b0aca1 100644
--- a/media/libaaudio/src/fifo/FifoBuffer.h
+++ b/media/libaaudio/src/fifo/FifoBuffer.h
@@ -38,7 +38,7 @@
 
 class FifoBuffer {
 public:
-    FifoBuffer(int32_t bytesPerFrame);
+    explicit FifoBuffer(int32_t bytesPerFrame);
 
     virtual ~FifoBuffer() = default;
 
@@ -162,6 +162,6 @@
     uint8_t *mExternalStorage = nullptr;
 };
 
-}  // android
+}  // namespace android
 
 #endif //FIFO_FIFO_BUFFER_H
diff --git a/media/libaaudio/src/fifo/FifoController.h b/media/libaaudio/src/fifo/FifoController.h
index 057a94e..e15d444 100644
--- a/media/libaaudio/src/fifo/FifoController.h
+++ b/media/libaaudio/src/fifo/FifoController.h
@@ -36,7 +36,7 @@
     , mWriteCounter(0)
     {}
 
-    virtual ~FifoController() {}
+    virtual ~FifoController() = default;
 
     // TODO review use of memory barriers, probably incorrect
     virtual fifo_counter_t getReadCounter() override {
@@ -57,6 +57,6 @@
     std::atomic<fifo_counter_t> mWriteCounter;
 };
 
-}  // android
+}  // namespace android
 
 #endif //FIFO_FIFO_CONTROLLER_H
diff --git a/media/libaaudio/src/fifo/FifoControllerBase.cpp b/media/libaaudio/src/fifo/FifoControllerBase.cpp
index 1dece0e..ad6d041 100644
--- a/media/libaaudio/src/fifo/FifoControllerBase.cpp
+++ b/media/libaaudio/src/fifo/FifoControllerBase.cpp
@@ -29,9 +29,6 @@
 {
 }
 
-FifoControllerBase::~FifoControllerBase() {
-}
-
 fifo_frames_t FifoControllerBase::getFullFramesAvailable() {
     fifo_frames_t temp = 0;
     __builtin_sub_overflow(getWriteCounter(), getReadCounter(), &temp);
diff --git a/media/libaaudio/src/fifo/FifoControllerBase.h b/media/libaaudio/src/fifo/FifoControllerBase.h
index 1edb8a3..2a6173b 100644
--- a/media/libaaudio/src/fifo/FifoControllerBase.h
+++ b/media/libaaudio/src/fifo/FifoControllerBase.h
@@ -43,7 +43,7 @@
      */
     FifoControllerBase(fifo_frames_t capacity, fifo_frames_t threshold);
 
-    virtual ~FifoControllerBase();
+    virtual ~FifoControllerBase() = default;
 
     // Abstract methods to be implemented in subclasses.
     /**
@@ -123,6 +123,6 @@
     fifo_frames_t mThreshold;
 };
 
-}  // android
+}  // namespace android
 
 #endif // FIFO_FIFO_CONTROLLER_BASE_H
diff --git a/media/libaaudio/src/fifo/FifoControllerIndirect.h b/media/libaaudio/src/fifo/FifoControllerIndirect.h
index ec48e57..a59225a 100644
--- a/media/libaaudio/src/fifo/FifoControllerIndirect.h
+++ b/media/libaaudio/src/fifo/FifoControllerIndirect.h
@@ -44,7 +44,7 @@
         setReadCounter(0);
         setWriteCounter(0);
     }
-    virtual ~FifoControllerIndirect() {};
+    virtual ~FifoControllerIndirect() = default;
 
     // TODO review use of memory barriers, probably incorrect
     virtual fifo_counter_t getReadCounter() override {
@@ -68,6 +68,6 @@
     std::atomic<fifo_counter_t> * mWriteCounterAddress;
 };
 
-}  // android
+}  // namespace android
 
 #endif //FIFO_FIFO_CONTROLLER_INDIRECT_H
diff --git a/media/libaaudio/src/flowgraph/AudioProcessorBase.cpp b/media/libaaudio/src/flowgraph/AudioProcessorBase.cpp
index 5667fdb..d8ffd00 100644
--- a/media/libaaudio/src/flowgraph/AudioProcessorBase.cpp
+++ b/media/libaaudio/src/flowgraph/AudioProcessorBase.cpp
@@ -34,8 +34,7 @@
                                int32_t samplesPerFrame,
                                int32_t framesPerBlock)
         : AudioPort(parent, samplesPerFrame)
-        , mFramesPerBlock(framesPerBlock)
-        , mSampleBlock(NULL) {
+        , mFramesPerBlock(framesPerBlock) {
     int32_t numFloats = framesPerBlock * getSamplesPerFrame();
     mSampleBlock = new float[numFloats]{0.0f};
 }
@@ -61,13 +60,13 @@
 
 /***************************************************************************/
 int32_t AudioFloatInputPort::pullData(int64_t framePosition, int32_t numFrames) {
-    return (mConnected == NULL)
+    return (mConnected == nullptr)
             ? std::min(getFramesPerBlock(), numFrames)
             : mConnected->pullData(framePosition, numFrames);
 }
 
 float *AudioFloatInputPort::getBlock() {
-    if (mConnected == NULL) {
+    if (mConnected == nullptr) {
         return AudioFloatBlockPort::getBlock(); // loaded using setValue()
     } else {
         return mConnected->getBlock();
diff --git a/media/libaaudio/src/flowgraph/MonoToMultiConverter.cpp b/media/libaaudio/src/flowgraph/MonoToMultiConverter.cpp
index 78aad52..c6fcac6 100644
--- a/media/libaaudio/src/flowgraph/MonoToMultiConverter.cpp
+++ b/media/libaaudio/src/flowgraph/MonoToMultiConverter.cpp
@@ -26,8 +26,6 @@
         , output(*this, channelCount) {
 }
 
-MonoToMultiConverter::~MonoToMultiConverter() { }
-
 int32_t MonoToMultiConverter::onProcess(int64_t framePosition, int32_t numFrames) {
     int32_t framesToProcess = input.pullData(framePosition, numFrames);
 
diff --git a/media/libaaudio/src/flowgraph/MonoToMultiConverter.h b/media/libaaudio/src/flowgraph/MonoToMultiConverter.h
index 34d53c7..5058ae0 100644
--- a/media/libaaudio/src/flowgraph/MonoToMultiConverter.h
+++ b/media/libaaudio/src/flowgraph/MonoToMultiConverter.h
@@ -29,7 +29,7 @@
 public:
     explicit MonoToMultiConverter(int32_t channelCount);
 
-    virtual ~MonoToMultiConverter();
+    virtual ~MonoToMultiConverter() = default;
 
     int32_t onProcess(int64_t framePosition, int32_t numFrames) override;
 
diff --git a/media/libaaudio/src/flowgraph/RampLinear.cpp b/media/libaaudio/src/flowgraph/RampLinear.cpp
index a260828..0cc32e5 100644
--- a/media/libaaudio/src/flowgraph/RampLinear.cpp
+++ b/media/libaaudio/src/flowgraph/RampLinear.cpp
@@ -37,6 +37,10 @@
 
 void RampLinear::setTarget(float target) {
     mTarget.store(target);
+    // If the ramp has not been used then start immediately at this level.
+    if (mLastFramePosition < 0) {
+        forceCurrent(target);
+    }
 }
 
 float RampLinear::interpolateCurrent() {
diff --git a/media/libaaudio/src/flowgraph/SourceFloat.cpp b/media/libaaudio/src/flowgraph/SourceFloat.cpp
index 4bb674f..5b3a51e 100644
--- a/media/libaaudio/src/flowgraph/SourceFloat.cpp
+++ b/media/libaaudio/src/flowgraph/SourceFloat.cpp
@@ -25,7 +25,7 @@
         : AudioSource(channelCount) {
 }
 
-int32_t SourceFloat::onProcess(int64_t framePosition, int32_t numFrames) {
+int32_t SourceFloat::onProcess(int64_t /*framePosition*/, int32_t numFrames) {
 
     float *outputBuffer = output.getBlock();
     int32_t channelCount = output.getSamplesPerFrame();
diff --git a/media/libaaudio/src/flowgraph/SourceI16.cpp b/media/libaaudio/src/flowgraph/SourceI16.cpp
index c3fcec2..a645cc2 100644
--- a/media/libaaudio/src/flowgraph/SourceI16.cpp
+++ b/media/libaaudio/src/flowgraph/SourceI16.cpp
@@ -30,7 +30,7 @@
         : AudioSource(channelCount) {
 }
 
-int32_t SourceI16::onProcess(int64_t framePosition, int32_t numFrames) {
+int32_t SourceI16::onProcess(int64_t /*framePosition*/, int32_t numFrames) {
     float *floatData = output.getBlock();
     int32_t channelCount = output.getSamplesPerFrame();
 
diff --git a/media/libaaudio/src/flowgraph/SourceI24.cpp b/media/libaaudio/src/flowgraph/SourceI24.cpp
index 097954e..50fb98e 100644
--- a/media/libaaudio/src/flowgraph/SourceI24.cpp
+++ b/media/libaaudio/src/flowgraph/SourceI24.cpp
@@ -32,7 +32,7 @@
         : AudioSource(channelCount) {
 }
 
-int32_t SourceI24::onProcess(int64_t framePosition, int32_t numFrames) {
+int32_t SourceI24::onProcess(int64_t /*framePosition*/, int32_t numFrames) {
     float *floatData = output.getBlock();
     int32_t channelCount = output.getSamplesPerFrame();
 
diff --git a/media/libaaudio/src/flowgraph/SourceI32.cpp b/media/libaaudio/src/flowgraph/SourceI32.cpp
index e8177ad..95bfd8f 100644
--- a/media/libaaudio/src/flowgraph/SourceI32.cpp
+++ b/media/libaaudio/src/flowgraph/SourceI32.cpp
@@ -30,7 +30,7 @@
         : AudioSource(channelCount) {
 }
 
-int32_t SourceI32::onProcess(int64_t framePosition, int32_t numFrames) {
+int32_t SourceI32::onProcess(int64_t /*framePosition*/, int32_t numFrames) {
     float *floatData = output.getBlock();
     int32_t channelCount = output.getSamplesPerFrame();
 
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
index e96e134..0f24771 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.cpp
@@ -37,9 +37,6 @@
         : AudioStream() {
 }
 
-AudioStreamLegacy::~AudioStreamLegacy() {
-}
-
 // Called from AudioTrack.cpp or AudioRecord.cpp
 static void AudioStreamLegacy_callback(int event, void* userData, void *info) {
     AudioStreamLegacy *streamLegacy = (AudioStreamLegacy *) userData;
diff --git a/media/libaaudio/src/legacy/AudioStreamLegacy.h b/media/libaaudio/src/legacy/AudioStreamLegacy.h
index 88ef270..d9ba990 100644
--- a/media/libaaudio/src/legacy/AudioStreamLegacy.h
+++ b/media/libaaudio/src/legacy/AudioStreamLegacy.h
@@ -60,7 +60,7 @@
 public:
     AudioStreamLegacy();
 
-    virtual ~AudioStreamLegacy();
+    virtual ~AudioStreamLegacy() = default;
 
     aaudio_legacy_callback_t getLegacyCallback();
 
diff --git a/media/libaaudio/src/legacy/AudioStreamRecord.cpp b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
index dc66742..12771cc 100644
--- a/media/libaaudio/src/legacy/AudioStreamRecord.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamRecord.cpp
@@ -65,11 +65,8 @@
     const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
 
     // TODO Support UNSPECIFIED in AudioRecord. For now, use stereo if unspecified.
-    int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
-                              ? 2 : getSamplesPerFrame();
-    audio_channel_mask_t channelMask = samplesPerFrame <= 2 ?
-                               audio_channel_in_mask_from_count(samplesPerFrame) :
-                               audio_channel_mask_for_index_assignment_from_count(samplesPerFrame);
+    audio_channel_mask_t channelMask =
+            AAudio_getChannelMaskForOpen(getChannelMask(), getSamplesPerFrame(), true /*isInput*/);
 
     size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0
                         : builder.getBufferCapacity();
@@ -115,7 +112,7 @@
     constexpr int32_t kMostLikelySampleRateForFast = 48000;
     if (getFormat() == AUDIO_FORMAT_PCM_FLOAT
             && perfMode == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY
-            && (samplesPerFrame <= 2) // FAST only for mono and stereo
+            && (audio_channel_count_from_in_mask(channelMask) <= 2) // FAST only for mono and stereo
             && (getSampleRate() == kMostLikelySampleRateForFast
                 || getSampleRate() == AAUDIO_UNSPECIFIED)) {
         setDeviceFormat(AUDIO_FORMAT_PCM_16_BIT);
@@ -228,7 +225,9 @@
             .set(AMEDIAMETRICS_PROP_ENCODINGCLIENT, toString(requestedFormat).c_str()).record();
 
     // Get the actual values from the AudioRecord.
-    setSamplesPerFrame(mAudioRecord->channelCount());
+    setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+            mAudioRecord->channelMask(), true /*isInput*/,
+            AAudio_isChannelIndexMask(getChannelMask())));
     setSampleRate(mAudioRecord->getSampleRate());
     setBufferCapacity(getBufferCapacityFromDevice());
     setFramesPerBurst(getFramesPerBurstFromDevice());
@@ -505,7 +504,7 @@
     return (aaudio_result_t) framesRead;
 }
 
-aaudio_result_t AudioStreamRecord::setBufferSize(int32_t requestedFrames)
+aaudio_result_t AudioStreamRecord::setBufferSize(int32_t /*requestedFrames*/)
 {
     return getBufferSize();
 }
diff --git a/media/libaaudio/src/legacy/AudioStreamTrack.cpp b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
index 1d412c0..118c004 100644
--- a/media/libaaudio/src/legacy/AudioStreamTrack.cpp
+++ b/media/libaaudio/src/legacy/AudioStreamTrack.cpp
@@ -66,13 +66,8 @@
     const aaudio_session_id_t requestedSessionId = builder.getSessionId();
     const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId);
 
-    // Try to create an AudioTrack
-    // Use stereo if unspecified.
-    int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED)
-                              ? 2 : getSamplesPerFrame();
-    audio_channel_mask_t channelMask = samplesPerFrame <= 2 ?
-                            audio_channel_out_mask_from_count(samplesPerFrame) :
-                            audio_channel_mask_for_index_assignment_from_count(samplesPerFrame);
+    audio_channel_mask_t channelMask =
+            AAudio_getChannelMaskForOpen(getChannelMask(), getSamplesPerFrame(), false /*isInput*/);
 
     audio_output_flags_t flags;
     aaudio_performance_mode_t perfMode = getPerformanceMode();
@@ -162,11 +157,11 @@
             callback,
             callbackData,
             notificationFrames,
-            0,       // DEFAULT sharedBuffer*/,
+            nullptr,       // DEFAULT sharedBuffer*/,
             false,   // DEFAULT threadCanCallJava
             sessionId,
             streamTransferType,
-            NULL,    // DEFAULT audio_offload_info_t
+            nullptr,    // DEFAULT audio_offload_info_t
             AttributionSourceState(), // DEFAULT uid and pid
             &attributes,
             // WARNING - If doNotReconnect set true then audio stops after plugging and unplugging
@@ -199,7 +194,9 @@
     doSetVolume();
 
     // Get the actual values from the AudioTrack.
-    setSamplesPerFrame(mAudioTrack->channelCount());
+    setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+        mAudioTrack->channelMask(), false /*isInput*/,
+        AAudio_isChannelIndexMask(getChannelMask())));
     setFormat(mAudioTrack->format());
     setDeviceFormat(mAudioTrack->format());
     setSampleRate(mAudioTrack->getSampleRate());
diff --git a/media/libaaudio/src/libaaudio.map.txt b/media/libaaudio/src/libaaudio.map.txt
index 1dd44d1..8fa9e38 100644
--- a/media/libaaudio/src/libaaudio.map.txt
+++ b/media/libaaudio/src/libaaudio.map.txt
@@ -25,6 +25,7 @@
     AAudioStreamBuilder_setPrivacySensitive;   # introduced=30
     AAudioStreamBuilder_setPackageName;   # introduced=31
     AAudioStreamBuilder_setAttributionTag;   # introduced=31
+    AAudioStreamBuilder_setChannelMask;    # introduced=32
     AAudioStreamBuilder_openStream;
     AAudioStreamBuilder_delete;
     AAudioStream_close;
@@ -61,6 +62,7 @@
     AAudioStream_isMMapUsed;
     AAudioStream_isPrivacySensitive;   # introduced=30
     AAudioStream_release;        # introduced=30
+    AAudioStream_getChannelMask;  # introduced=32
   local:
     *;
 };
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index d795725..d829934 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -256,6 +256,248 @@
     return privacySensitive ? AUDIO_FLAG_CAPTURE_PRIVATE : AUDIO_FLAG_NONE;
 }
 
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelLayoutMask(
+        aaudio_channel_mask_t channelMask, bool isInput) {
+    if (isInput) {
+        switch (channelMask) {
+            case AAUDIO_CHANNEL_MONO:
+                return AUDIO_CHANNEL_IN_MONO;
+            case AAUDIO_CHANNEL_STEREO:
+                return AUDIO_CHANNEL_IN_STEREO;
+            case AAUDIO_CHANNEL_FRONT_BACK:
+                return AUDIO_CHANNEL_IN_FRONT_BACK;
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+                return AUDIO_CHANNEL_IN_2POINT0POINT2;
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+                return AUDIO_CHANNEL_IN_2POINT1POINT2;
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+                return AUDIO_CHANNEL_IN_3POINT0POINT2;
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+                return AUDIO_CHANNEL_IN_3POINT1POINT2;
+            case AAUDIO_CHANNEL_5POINT1:
+                return AUDIO_CHANNEL_IN_5POINT1;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AUDIO_CHANNEL_INVALID;
+        }
+    } else {
+        switch (channelMask) {
+            case AAUDIO_CHANNEL_MONO:
+                return AUDIO_CHANNEL_OUT_MONO;
+            case AAUDIO_CHANNEL_STEREO:
+                return AUDIO_CHANNEL_OUT_STEREO;
+            case AAUDIO_CHANNEL_2POINT1:
+                return AUDIO_CHANNEL_OUT_2POINT1;
+            case AAUDIO_CHANNEL_TRI:
+                return AUDIO_CHANNEL_OUT_TRI;
+            case AAUDIO_CHANNEL_TRI_BACK:
+                return AUDIO_CHANNEL_OUT_TRI_BACK;
+            case AAUDIO_CHANNEL_3POINT1:
+                return AUDIO_CHANNEL_OUT_3POINT1;
+            case AAUDIO_CHANNEL_2POINT0POINT2:
+                return AUDIO_CHANNEL_OUT_2POINT0POINT2;
+            case AAUDIO_CHANNEL_2POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_2POINT1POINT2;
+            case AAUDIO_CHANNEL_3POINT0POINT2:
+                return AUDIO_CHANNEL_OUT_3POINT0POINT2;
+            case AAUDIO_CHANNEL_3POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_3POINT1POINT2;
+            case AAUDIO_CHANNEL_QUAD:
+                return AUDIO_CHANNEL_OUT_QUAD;
+            case AAUDIO_CHANNEL_QUAD_SIDE:
+                return AUDIO_CHANNEL_OUT_QUAD_SIDE;
+            case AAUDIO_CHANNEL_SURROUND:
+                return AUDIO_CHANNEL_OUT_SURROUND;
+            case AAUDIO_CHANNEL_PENTA:
+                return AUDIO_CHANNEL_OUT_PENTA;
+            case AAUDIO_CHANNEL_5POINT1:
+                return AUDIO_CHANNEL_OUT_5POINT1;
+            case AAUDIO_CHANNEL_5POINT1_SIDE:
+                return AUDIO_CHANNEL_OUT_5POINT1_SIDE;
+            case AAUDIO_CHANNEL_5POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_5POINT1POINT2;
+            case AAUDIO_CHANNEL_5POINT1POINT4:
+                return AUDIO_CHANNEL_OUT_5POINT1POINT4;
+            case AAUDIO_CHANNEL_6POINT1:
+                return AUDIO_CHANNEL_OUT_6POINT1;
+            case AAUDIO_CHANNEL_7POINT1:
+                return AUDIO_CHANNEL_OUT_7POINT1;
+            case AAUDIO_CHANNEL_7POINT1POINT2:
+                return AUDIO_CHANNEL_OUT_7POINT1POINT2;
+            case AAUDIO_CHANNEL_7POINT1POINT4:
+                return AUDIO_CHANNEL_OUT_7POINT1POINT4;
+            // TODO: add 9point1point4 and 9point1point6 when they are added in audio-hal-enums.h
+            // case AAUDIO_CHANNEL_9POINT1POINT4:
+            //     return AUDIO_CHANNEL_OUT_9POINT1POINT4;
+            // case AAUDIO_CHANNEL_9POINT1POINT6:
+            //     return AUDIO_CHANNEL_OUT_9POINT1POINT6;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AUDIO_CHANNEL_INVALID;
+        }
+    }
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelLayoutMask(
+        audio_channel_mask_t channelMask, bool isInput) {
+    if (isInput) {
+        switch (channelMask) {
+            case AUDIO_CHANNEL_IN_MONO:
+                return AAUDIO_CHANNEL_MONO;
+            case AUDIO_CHANNEL_IN_STEREO:
+                return AAUDIO_CHANNEL_STEREO;
+            case AUDIO_CHANNEL_IN_FRONT_BACK:
+                return AAUDIO_CHANNEL_FRONT_BACK;
+            case AUDIO_CHANNEL_IN_2POINT0POINT2:
+                return AAUDIO_CHANNEL_2POINT0POINT2;
+            case AUDIO_CHANNEL_IN_2POINT1POINT2:
+                return AAUDIO_CHANNEL_2POINT1POINT2;
+            case AUDIO_CHANNEL_IN_3POINT0POINT2:
+                return AAUDIO_CHANNEL_3POINT0POINT2;
+            case AUDIO_CHANNEL_IN_3POINT1POINT2:
+                return AAUDIO_CHANNEL_3POINT1POINT2;
+            case AUDIO_CHANNEL_IN_5POINT1:
+                return AAUDIO_CHANNEL_5POINT1;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AAUDIO_CHANNEL_INVALID;
+        }
+    } else {
+        switch (channelMask) {
+            case AUDIO_CHANNEL_OUT_MONO:
+                return AAUDIO_CHANNEL_MONO;
+            case AUDIO_CHANNEL_OUT_STEREO:
+                return AAUDIO_CHANNEL_STEREO;
+            case AUDIO_CHANNEL_OUT_2POINT1:
+                return AAUDIO_CHANNEL_2POINT1;
+            case AUDIO_CHANNEL_OUT_TRI:
+                return AAUDIO_CHANNEL_TRI;
+            case AUDIO_CHANNEL_OUT_TRI_BACK:
+                return AAUDIO_CHANNEL_TRI_BACK;
+            case AUDIO_CHANNEL_OUT_3POINT1:
+                return AAUDIO_CHANNEL_3POINT1;
+            case AUDIO_CHANNEL_OUT_2POINT0POINT2:
+                return AAUDIO_CHANNEL_2POINT0POINT2;
+            case AUDIO_CHANNEL_OUT_2POINT1POINT2:
+                return AAUDIO_CHANNEL_2POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_3POINT0POINT2:
+                return AAUDIO_CHANNEL_3POINT0POINT2;
+            case AUDIO_CHANNEL_OUT_3POINT1POINT2:
+                return AAUDIO_CHANNEL_3POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_QUAD:
+                return AAUDIO_CHANNEL_QUAD;
+            case AUDIO_CHANNEL_OUT_QUAD_SIDE:
+                return AAUDIO_CHANNEL_QUAD_SIDE;
+            case AUDIO_CHANNEL_OUT_SURROUND:
+                return AAUDIO_CHANNEL_SURROUND;
+            case AUDIO_CHANNEL_OUT_PENTA:
+                return AAUDIO_CHANNEL_PENTA;
+            case AUDIO_CHANNEL_OUT_5POINT1:
+                return AAUDIO_CHANNEL_5POINT1;
+            case AUDIO_CHANNEL_OUT_5POINT1_SIDE:
+                return AAUDIO_CHANNEL_5POINT1_SIDE;
+            case AUDIO_CHANNEL_OUT_5POINT1POINT2:
+                return AAUDIO_CHANNEL_5POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_5POINT1POINT4:
+                return AAUDIO_CHANNEL_5POINT1POINT4;
+            case AUDIO_CHANNEL_OUT_6POINT1:
+                return AAUDIO_CHANNEL_6POINT1;
+            case AUDIO_CHANNEL_OUT_7POINT1:
+                return AAUDIO_CHANNEL_7POINT1;
+            case AUDIO_CHANNEL_OUT_7POINT1POINT2:
+                return AAUDIO_CHANNEL_7POINT1POINT2;
+            case AUDIO_CHANNEL_OUT_7POINT1POINT4:
+                return AAUDIO_CHANNEL_7POINT1POINT4;
+            // TODO: add 9point1point4 and 9point1point6 when they are added in audio-hal-enums.h
+            // case AUDIO_CHANNEL_OUT_9POINT1POINT4:
+            //     return AAUDIO_CHANNEL_9POINT1POINT4;
+            // case AUDIO_CHANNEL_OUT_9POINT1POINT6:
+            //     return AAUDIO_CHANNEL_9POINT1POINT6;
+            default:
+                ALOGE("%s() %#x unrecognized", __func__, channelMask);
+                return AAUDIO_CHANNEL_INVALID;
+        }
+    }
+}
+
+int32_t AAudioConvert_channelMaskToCount(aaudio_channel_mask_t channelMask) {
+    return __builtin_popcount(channelMask & ~AAUDIO_CHANNEL_BIT_INDEX);
+}
+
+aaudio_channel_mask_t AAudioConvert_channelCountToMask(int32_t channelCount) {
+    if (channelCount < 0 || channelCount > AUDIO_CHANNEL_COUNT_MAX) {
+        return AAUDIO_CHANNEL_INVALID;
+    }
+
+    if (channelCount == 0) {
+        return AAUDIO_UNSPECIFIED;
+    }
+
+    // Return index mask if the channel count is greater than 2.
+    return AAUDIO_CHANNEL_BIT_INDEX | ((1 << channelCount) - 1);
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelIndexMask(
+        audio_channel_mask_t channelMask) {
+    if (audio_channel_mask_get_representation(channelMask) != AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+        ALOGE("%s() %#x not an index mask", __func__, channelMask);
+        return AAUDIO_CHANNEL_INVALID;
+    }
+    return (channelMask & ~AUDIO_CHANNEL_INDEX_HDR) | AAUDIO_CHANNEL_BIT_INDEX;
+}
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelIndexMask(
+        aaudio_channel_mask_t channelMask) {
+    if (!AAudio_isChannelIndexMask(channelMask)) {
+        ALOGE("%s() %#x not an index mask", __func__, channelMask);
+        return AUDIO_CHANNEL_INVALID;
+    }
+    return audio_channel_mask_for_index_assignment_from_count(
+            AAudioConvert_channelMaskToCount(channelMask));
+}
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelMask(
+        audio_channel_mask_t channelMask, bool isInput, bool indexMaskRequired) {
+    if (audio_channel_mask_get_representation(channelMask) == AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+        return AAudioConvert_androidToAAudioChannelIndexMask(channelMask);
+    }
+    if (indexMaskRequired) {
+        // Require index mask, `channelMask` here is a position mask.
+        const int channelCount = isInput ? audio_channel_count_from_in_mask(channelMask)
+                                         : audio_channel_count_from_out_mask(channelMask);
+        return AAudioConvert_channelCountToMask(channelCount);
+    }
+    return AAudioConvert_androidToAAudioChannelLayoutMask(channelMask, isInput);
+}
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelMask(
+        aaudio_channel_mask_t channelMask, bool isInput) {
+    return AAudio_isChannelIndexMask(channelMask)
+            ? AAudioConvert_aaudioToAndroidChannelIndexMask(channelMask)
+            : AAudioConvert_aaudioToAndroidChannelLayoutMask(channelMask, isInput);
+}
+
+bool AAudio_isChannelIndexMask(aaudio_channel_mask_t channelMask) {
+    return (channelMask & AAUDIO_CHANNEL_BIT_INDEX) == AAUDIO_CHANNEL_BIT_INDEX;
+}
+
+audio_channel_mask_t AAudio_getChannelMaskForOpen(
+        aaudio_channel_mask_t channelMask, int32_t samplesPerFrame, bool isInput) {
+    if (channelMask != AAUDIO_UNSPECIFIED) {
+        if (AAudio_isChannelIndexMask(channelMask) && samplesPerFrame <= 2) {
+            // When it is index mask and the count is less than 3, use position mask
+            // instead of index mask for opening a stream. This may need to be revisited
+            // when making channel index mask public.
+            return isInput ? audio_channel_in_mask_from_count(samplesPerFrame)
+                           : audio_channel_out_mask_from_count(samplesPerFrame);
+        }
+        return AAudioConvert_aaudioToAndroidChannelMask(channelMask, isInput);
+    }
+
+    // Return stereo when unspecified.
+    return isInput ? AUDIO_CHANNEL_IN_STEREO : AUDIO_CHANNEL_OUT_STEREO;
+}
+
 int32_t AAudioConvert_framesToBytes(int32_t numFrames,
                                     int32_t bytesPerFrame,
                                     int32_t *sizeInBytes) {
diff --git a/media/libaaudio/src/utility/AAudioUtilities.h b/media/libaaudio/src/utility/AAudioUtilities.h
index 82eb77d..52e57da 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.h
+++ b/media/libaaudio/src/utility/AAudioUtilities.h
@@ -96,6 +96,33 @@
 audio_flags_mask_t AAudioConvert_privacySensitiveToAudioFlagsMask(
         bool privacySensitive);
 
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelLayoutMask(
+        aaudio_channel_mask_t channelMask, bool isInput);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelLayoutMask(
+        audio_channel_mask_t channelMask, bool isInput);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelIndexMask(
+        audio_channel_mask_t channelMask);
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelIndexMask(
+        aaudio_channel_mask_t channelMask);
+
+aaudio_channel_mask_t AAudioConvert_androidToAAudioChannelMask(
+        audio_channel_mask_t channelMask, bool isInput, bool indexMaskRequired);
+
+audio_channel_mask_t AAudioConvert_aaudioToAndroidChannelMask(
+        aaudio_channel_mask_t channelMask, bool isInput);
+
+bool AAudio_isChannelIndexMask(aaudio_channel_mask_t channelMask);
+
+int32_t AAudioConvert_channelMaskToCount(aaudio_channel_mask_t channelMask);
+
+aaudio_channel_mask_t AAudioConvert_channelCountToMask(int32_t channelCount);
+
+audio_channel_mask_t AAudio_getChannelMaskForOpen(
+        aaudio_channel_mask_t channelMask, int32_t samplesPerFrame, bool isInput);
+
 // Note that this code may be replaced by Settings or by some other system configuration tool.
 
 /**
@@ -198,7 +225,7 @@
  * @return true if f() eventually returns true.
  */
 static inline bool AAudio_tryUntilTrue(
-        std::function<bool()> f, int times, int sleepMs) {
+        const std::function<bool()>& f, int times, int sleepMs) {
     static const useconds_t US_PER_MS = 1000;
 
     sleepMs = std::max(sleepMs, 0);
@@ -270,9 +297,7 @@
 
 class Timestamp {
 public:
-    Timestamp()
-            : mPosition(0)
-            , mNanoseconds(0) {}
+    Timestamp() = default;
     Timestamp(int64_t position, int64_t nanoseconds)
             : mPosition(position)
             , mNanoseconds(nanoseconds) {}
@@ -283,8 +308,8 @@
 
 private:
     // These cannot be const because we need to implement the copy assignment operator.
-    int64_t mPosition;
-    int64_t mNanoseconds;
+    int64_t mPosition{0};
+    int64_t mNanoseconds{0};
 };
 
 
@@ -318,4 +343,36 @@
     std::atomic<int> mRequested{0};
     std::atomic<int> mAcknowledged{0};
 };
+
+enum {
+    /**
+     * Audio channel index mask, only used internally.
+     */
+    AAUDIO_CHANNEL_BIT_INDEX = 0x80000000,
+    AAUDIO_CHANNEL_INDEX_MASK_1 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 1) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_2 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 2) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_3 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 3) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_4 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 4) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_5 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 5) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_6 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 6) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_7 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 7) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_8 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 8) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_9 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 9) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_10 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 10) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_11 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 11) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_12 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 12) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_13 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 13) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_14 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 14) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_15 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 15) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_16 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 16) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_17 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 17) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_18 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 18) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_19 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 19) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_20 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 20) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_21 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 21) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_22 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 22) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_23 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 23) - 1,
+    AAUDIO_CHANNEL_INDEX_MASK_24 = AAUDIO_CHANNEL_BIT_INDEX | (1 << 24) - 1,
+};
+
 #endif //UTILITY_AAUDIO_UTILITIES_H
diff --git a/media/libaaudio/src/utility/FixedBlockAdapter.h b/media/libaaudio/src/utility/FixedBlockAdapter.h
index 4dc7e68..290e473 100644
--- a/media/libaaudio/src/utility/FixedBlockAdapter.h
+++ b/media/libaaudio/src/utility/FixedBlockAdapter.h
@@ -35,7 +35,7 @@
 class FixedBlockAdapter
 {
 public:
-    FixedBlockAdapter(FixedBlockProcessor &fixedBlockProcessor)
+    explicit FixedBlockAdapter(FixedBlockProcessor &fixedBlockProcessor)
     : mFixedBlockProcessor(fixedBlockProcessor) {}
 
     virtual ~FixedBlockAdapter() = default;
diff --git a/media/libaaudio/src/utility/FixedBlockReader.h b/media/libaaudio/src/utility/FixedBlockReader.h
index 128dd52..dc82416 100644
--- a/media/libaaudio/src/utility/FixedBlockReader.h
+++ b/media/libaaudio/src/utility/FixedBlockReader.h
@@ -30,7 +30,7 @@
 class FixedBlockReader : public FixedBlockAdapter
 {
 public:
-    FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor);
+    explicit FixedBlockReader(FixedBlockProcessor &fixedBlockProcessor);
 
     virtual ~FixedBlockReader() = default;
 
diff --git a/media/libaaudio/src/utility/FixedBlockWriter.h b/media/libaaudio/src/utility/FixedBlockWriter.h
index f1d917c..3e89b5d 100644
--- a/media/libaaudio/src/utility/FixedBlockWriter.h
+++ b/media/libaaudio/src/utility/FixedBlockWriter.h
@@ -28,7 +28,7 @@
 class FixedBlockWriter : public FixedBlockAdapter
 {
 public:
-    FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor);
+    explicit FixedBlockWriter(FixedBlockProcessor &fixedBlockProcessor);
 
     virtual ~FixedBlockWriter() = default;
 
diff --git a/media/libaaudio/src/utility/MonotonicCounter.h b/media/libaaudio/src/utility/MonotonicCounter.h
index 63add4e..313ccbd 100644
--- a/media/libaaudio/src/utility/MonotonicCounter.h
+++ b/media/libaaudio/src/utility/MonotonicCounter.h
@@ -30,8 +30,8 @@
 class MonotonicCounter {
 
 public:
-    MonotonicCounter() {};
-    virtual ~MonotonicCounter() {};
+    MonotonicCounter() = default;
+    virtual ~MonotonicCounter() = default;
 
     /**
      * @return current value of the counter
diff --git a/media/libaaudio/tests/test_flowgraph.cpp b/media/libaaudio/tests/test_flowgraph.cpp
index d563a7e..611cbf7 100644
--- a/media/libaaudio/tests/test_flowgraph.cpp
+++ b/media/libaaudio/tests/test_flowgraph.cpp
@@ -76,31 +76,40 @@
 }
 
 TEST(test_flowgraph, module_ramp_linear) {
+    constexpr int singleNumOutput = 1;
     constexpr int rampSize = 5;
     constexpr int numOutput = 100;
     constexpr float value = 1.0f;
-    constexpr float target = 100.0f;
+    constexpr float initialTarget = 10.0f;
+    constexpr float finalTarget = 100.0f;
+    constexpr float tolerance = 0.0001f; // arbitrary
     float output[numOutput] = {};
     RampLinear rampLinear{1};
     SinkFloat sinkFloat{1};
 
     rampLinear.input.setValue(value);
     rampLinear.setLengthInFrames(rampSize);
-    rampLinear.setTarget(target);
-    rampLinear.forceCurrent(0.0f);
-
     rampLinear.output.connect(&sinkFloat.input);
 
+    // Check that the values go to the initial target instantly.
+    rampLinear.setTarget(initialTarget);
+    int32_t singleNumRead = sinkFloat.read(output, singleNumOutput);
+    ASSERT_EQ(singleNumRead, singleNumOutput);
+    EXPECT_NEAR(value * initialTarget, output[0], tolerance);
+
+    // Now set target and check that the linear ramp works as expected.
+    rampLinear.setTarget(finalTarget);
     int32_t numRead = sinkFloat.read(output, numOutput);
+    const float incrementSize = (finalTarget - initialTarget) / rampSize;
     ASSERT_EQ(numOutput, numRead);
-    constexpr float tolerance = 0.0001f; // arbitrary
+
     int i = 0;
     for (; i < rampSize; i++) {
-        float expected = i * value * target / rampSize;
+        float expected = value * (initialTarget + i * incrementSize);
         EXPECT_NEAR(expected, output[i], tolerance);
     }
     for (; i < numOutput; i++) {
-        float expected = value * target;
+        float expected = value * finalTarget;
         EXPECT_NEAR(expected, output[i], tolerance);
     }
 }
diff --git a/media/libaudioclient/AidlConversion.cpp b/media/libaudioclient/AidlConversion.cpp
index 321e7f9..4c155fe 100644
--- a/media/libaudioclient/AidlConversion.cpp
+++ b/media/libaudioclient/AidlConversion.cpp
@@ -14,6 +14,11 @@
  * limitations under the License.
  */
 
+#include <algorithm>
+#include <unordered_map>
+#include <utility>
+#include <vector>
+
 #define LOG_TAG "AidlConversion"
 //#define LOG_NDEBUG 0
 #include <utils/Log.h>
@@ -21,6 +26,7 @@
 #include "media/AidlConversion.h"
 
 #include <media/ShmemCompat.h>
+#include <media/stagefright/foundation/MediaDefs.h>
 
 ////////////////////////////////////////////////////////////////////////////////////////////////////
 // Utilities
@@ -273,21 +279,7 @@
             enumToMask_index<int32_t, media::AudioPortConfigType>);
 }
 
-ConversionResult<audio_channel_mask_t> aidl2legacy_int32_t_audio_channel_mask_t(int32_t aidl) {
-    // TODO(ytai): should we convert bit-by-bit?
-    // One problem here is that the representation is both opaque and is different based on the
-    // context (input vs. output). Can determine based on type and role, as per useInChannelMask().
-    return convertReinterpret<audio_channel_mask_t>(aidl);
-}
-
-ConversionResult<int32_t> legacy2aidl_audio_channel_mask_t_int32_t(audio_channel_mask_t legacy) {
-    // TODO(ytai): should we convert bit-by-bit?
-    // One problem here is that the representation is both opaque and is different based on the
-    // context (input vs. output). Can determine based on type and role, as per useInChannelMask().
-    return convertReinterpret<int32_t>(legacy);
-}
-
-ConversionResult<audio_io_config_event> aidl2legacy_AudioIoConfigEvent_audio_io_config_event(
+ConversionResult<audio_io_config_event_t> aidl2legacy_AudioIoConfigEvent_audio_io_config_event_t(
         media::AudioIoConfigEvent aidl) {
     switch (aidl) {
         case media::AudioIoConfigEvent::OUTPUT_REGISTERED:
@@ -312,8 +304,8 @@
     return unexpected(BAD_VALUE);
 }
 
-ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_AudioIoConfigEvent(
-        audio_io_config_event legacy) {
+ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_t_AudioIoConfigEvent(
+        audio_io_config_event_t legacy) {
     switch (legacy) {
         case AUDIO_OUTPUT_REGISTERED:
             return media::AudioIoConfigEvent::OUTPUT_REGISTERED;
@@ -393,18 +385,836 @@
     return unexpected(BAD_VALUE);
 }
 
-ConversionResult<audio_format_t> aidl2legacy_AudioFormat_audio_format_t(
-        media::audio::common::AudioFormat aidl) {
-    // This relies on AudioFormat being kept in sync with audio_format_t.
-    static_assert(sizeof(media::audio::common::AudioFormat) == sizeof(audio_format_t));
-    return static_cast<audio_format_t>(aidl);
+namespace {
+
+namespace detail {
+using AudioChannelPair = std::pair<audio_channel_mask_t, media::AudioChannelLayout>;
+using AudioChannelPairs = std::vector<AudioChannelPair>;
+using AudioDevicePair = std::pair<audio_devices_t, media::AudioDeviceDescription>;
+using AudioDevicePairs = std::vector<AudioDevicePair>;
+using AudioFormatPair = std::pair<audio_format_t, media::AudioFormatDescription>;
+using AudioFormatPairs = std::vector<AudioFormatPair>;
 }
 
-ConversionResult<media::audio::common::AudioFormat> legacy2aidl_audio_format_t_AudioFormat(
+const detail::AudioChannelPairs& getIndexAudioChannelPairs() {
+    static const detail::AudioChannelPairs pairs = {
+#define DEFINE_INDEX_MASK(n)                                                                \
+            {                                                                               \
+                AUDIO_CHANNEL_INDEX_MASK_##n,                                               \
+                media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::indexMask>( \
+                        media::AudioChannelLayout::INDEX_MASK_##n)                          \
+            }
+
+            DEFINE_INDEX_MASK(1),
+            DEFINE_INDEX_MASK(2),
+            DEFINE_INDEX_MASK(3),
+            DEFINE_INDEX_MASK(4),
+            DEFINE_INDEX_MASK(5),
+            DEFINE_INDEX_MASK(6),
+            DEFINE_INDEX_MASK(7),
+            DEFINE_INDEX_MASK(8),
+            DEFINE_INDEX_MASK(9),
+            DEFINE_INDEX_MASK(10),
+            DEFINE_INDEX_MASK(11),
+            DEFINE_INDEX_MASK(12),
+            DEFINE_INDEX_MASK(13),
+            DEFINE_INDEX_MASK(14),
+            DEFINE_INDEX_MASK(15),
+            DEFINE_INDEX_MASK(16),
+            DEFINE_INDEX_MASK(17),
+            DEFINE_INDEX_MASK(18),
+            DEFINE_INDEX_MASK(19),
+            DEFINE_INDEX_MASK(20),
+            DEFINE_INDEX_MASK(21),
+            DEFINE_INDEX_MASK(22),
+            DEFINE_INDEX_MASK(23),
+            DEFINE_INDEX_MASK(24)
+#undef DEFINE_INDEX_MASK
+    };
+    return pairs;
+}
+
+const detail::AudioChannelPairs& getInAudioChannelPairs() {
+    static const detail::AudioChannelPairs pairs = {
+#define DEFINE_INPUT_LAYOUT(n)                                                               \
+            {                                                                                \
+                AUDIO_CHANNEL_IN_##n,                                                        \
+                media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::layoutMask>( \
+                        media::AudioChannelLayout::LAYOUT_##n)                               \
+            }
+
+        DEFINE_INPUT_LAYOUT(MONO),
+        DEFINE_INPUT_LAYOUT(STEREO),
+        DEFINE_INPUT_LAYOUT(FRONT_BACK),
+        // AUDIO_CHANNEL_IN_6 not supported
+        DEFINE_INPUT_LAYOUT(2POINT0POINT2),
+        DEFINE_INPUT_LAYOUT(2POINT1POINT2),
+        DEFINE_INPUT_LAYOUT(3POINT0POINT2),
+        DEFINE_INPUT_LAYOUT(3POINT1POINT2),
+        DEFINE_INPUT_LAYOUT(5POINT1)
+#undef DEFINE_INPUT_LAYOUT
+    };
+    return pairs;
+}
+
+const detail::AudioChannelPairs& getOutAudioChannelPairs() {
+    static const detail::AudioChannelPairs pairs = {
+#define DEFINE_OUTPUT_LAYOUT(n)                                                              \
+            {                                                                                \
+                AUDIO_CHANNEL_OUT_##n,                                                       \
+                media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::layoutMask>( \
+                        media::AudioChannelLayout::LAYOUT_##n)                               \
+            }
+
+        DEFINE_OUTPUT_LAYOUT(MONO),
+        DEFINE_OUTPUT_LAYOUT(STEREO),
+        DEFINE_OUTPUT_LAYOUT(2POINT1),
+        DEFINE_OUTPUT_LAYOUT(TRI),
+        DEFINE_OUTPUT_LAYOUT(TRI_BACK),
+        DEFINE_OUTPUT_LAYOUT(3POINT1),
+        DEFINE_OUTPUT_LAYOUT(2POINT0POINT2),
+        DEFINE_OUTPUT_LAYOUT(2POINT1POINT2),
+        DEFINE_OUTPUT_LAYOUT(3POINT0POINT2),
+        DEFINE_OUTPUT_LAYOUT(3POINT1POINT2),
+        DEFINE_OUTPUT_LAYOUT(QUAD),
+        DEFINE_OUTPUT_LAYOUT(QUAD_SIDE),
+        DEFINE_OUTPUT_LAYOUT(SURROUND),
+        DEFINE_OUTPUT_LAYOUT(PENTA),
+        DEFINE_OUTPUT_LAYOUT(5POINT1),
+        DEFINE_OUTPUT_LAYOUT(5POINT1_SIDE),
+        DEFINE_OUTPUT_LAYOUT(5POINT1POINT2),
+        DEFINE_OUTPUT_LAYOUT(5POINT1POINT4),
+        DEFINE_OUTPUT_LAYOUT(6POINT1),
+        DEFINE_OUTPUT_LAYOUT(7POINT1),
+        DEFINE_OUTPUT_LAYOUT(7POINT1POINT2),
+        DEFINE_OUTPUT_LAYOUT(7POINT1POINT4),
+        DEFINE_OUTPUT_LAYOUT(13POINT_360RA),
+        DEFINE_OUTPUT_LAYOUT(22POINT2),
+        DEFINE_OUTPUT_LAYOUT(MONO_HAPTIC_A),
+        DEFINE_OUTPUT_LAYOUT(STEREO_HAPTIC_A),
+        DEFINE_OUTPUT_LAYOUT(HAPTIC_AB),
+        DEFINE_OUTPUT_LAYOUT(MONO_HAPTIC_AB),
+        DEFINE_OUTPUT_LAYOUT(STEREO_HAPTIC_AB)
+#undef DEFINE_OUTPUT_LAYOUT
+    };
+    return pairs;
+}
+
+const detail::AudioChannelPairs& getVoiceAudioChannelPairs() {
+    static const detail::AudioChannelPairs pairs = {
+#define DEFINE_VOICE_LAYOUT(n)                                                               \
+            {                                                                                \
+                AUDIO_CHANNEL_IN_VOICE_##n,                                                  \
+                media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::voiceMask>(  \
+                        media::AudioChannelLayout::VOICE_##n)                                \
+            }
+        DEFINE_VOICE_LAYOUT(UPLINK_MONO),
+        DEFINE_VOICE_LAYOUT(DNLINK_MONO),
+        DEFINE_VOICE_LAYOUT(CALL_MONO)
+#undef DEFINE_VOICE_LAYOUT
+    };
+    return pairs;
+}
+
+media::AudioDeviceDescription make_AudioDeviceDescription(media::AudioDeviceType type,
+        const std::string& connection = "") {
+    media::AudioDeviceDescription result;
+    result.type = type;
+    result.connection = connection;
+    return result;
+}
+
+void append_AudioDeviceDescription(detail::AudioDevicePairs& pairs,
+        audio_devices_t inputType, audio_devices_t outputType,
+        media::AudioDeviceType inType, media::AudioDeviceType outType,
+        const std::string& connection = "") {
+    pairs.push_back(std::make_pair(inputType, make_AudioDeviceDescription(inType, connection)));
+    pairs.push_back(std::make_pair(outputType, make_AudioDeviceDescription(outType, connection)));
+}
+
+const detail::AudioDevicePairs& getAudioDevicePairs() {
+    static const detail::AudioDevicePairs pairs = []() {
+        detail::AudioDevicePairs pairs = {{
+            {
+                AUDIO_DEVICE_NONE, media::AudioDeviceDescription{}
+            },
+            {
+                AUDIO_DEVICE_OUT_EARPIECE, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_SPEAKER_EARPIECE)
+            },
+            {
+                AUDIO_DEVICE_OUT_SPEAKER, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_SPEAKER)
+            },
+            {
+                AUDIO_DEVICE_OUT_WIRED_HEADPHONE, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_HEADPHONE,
+                        media::AudioDeviceDescription::CONNECTION_ANALOG())
+            },
+            {
+                AUDIO_DEVICE_OUT_BLUETOOTH_SCO, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_DEVICE,
+                        media::AudioDeviceDescription::CONNECTION_BT_SCO())
+            },
+            {
+                AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_CARKIT,
+                        media::AudioDeviceDescription::CONNECTION_BT_SCO())
+            },
+            {
+                AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_HEADPHONES, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_HEADPHONE,
+                        media::AudioDeviceDescription::CONNECTION_BT_A2DP())
+            },
+            {
+                AUDIO_DEVICE_OUT_BLUETOOTH_A2DP_SPEAKER, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_SPEAKER,
+                        media::AudioDeviceDescription::CONNECTION_BT_A2DP())
+            },
+            {
+                AUDIO_DEVICE_OUT_TELEPHONY_TX, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_TELEPHONY_TX)
+            },
+            {
+                AUDIO_DEVICE_OUT_AUX_LINE, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_LINE_AUX)
+            },
+            {
+                AUDIO_DEVICE_OUT_SPEAKER_SAFE, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_SPEAKER_SAFE)
+            },
+            {
+                AUDIO_DEVICE_OUT_HEARING_AID, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_HEARING_AID,
+                        media::AudioDeviceDescription::CONNECTION_WIRELESS())
+            },
+            {
+                AUDIO_DEVICE_OUT_ECHO_CANCELLER, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_ECHO_CANCELLER)
+            },
+            {
+                AUDIO_DEVICE_OUT_BLE_SPEAKER, make_AudioDeviceDescription(
+                        media::AudioDeviceType::OUT_SPEAKER,
+                        media::AudioDeviceDescription::CONNECTION_BT_LE())
+            },
+            // AUDIO_DEVICE_IN_AMBIENT and IN_COMMUNICATION are removed since they were deprecated.
+            {
+                AUDIO_DEVICE_IN_BUILTIN_MIC, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_MICROPHONE)
+            },
+            {
+                AUDIO_DEVICE_IN_BACK_MIC, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_MICROPHONE_BACK)
+            },
+            {
+                AUDIO_DEVICE_IN_TELEPHONY_RX, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_TELEPHONY_RX)
+            },
+            {
+                AUDIO_DEVICE_IN_TV_TUNER, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_TV_TUNER)
+            },
+            {
+                AUDIO_DEVICE_IN_LOOPBACK, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_LOOPBACK)
+            },
+            {
+                AUDIO_DEVICE_IN_BLUETOOTH_BLE, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_DEVICE,
+                        media::AudioDeviceDescription::CONNECTION_BT_LE())
+            },
+            {
+                AUDIO_DEVICE_IN_ECHO_REFERENCE, make_AudioDeviceDescription(
+                        media::AudioDeviceType::IN_ECHO_REFERENCE)
+            }
+        }};
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_DEFAULT, AUDIO_DEVICE_OUT_DEFAULT,
+                media::AudioDeviceType::IN_DEFAULT, media::AudioDeviceType::OUT_DEFAULT);
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_WIRED_HEADSET, AUDIO_DEVICE_OUT_WIRED_HEADSET,
+                media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
+                media::AudioDeviceDescription::CONNECTION_ANALOG());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET, AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET,
+                media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
+                media::AudioDeviceDescription::CONNECTION_BT_SCO());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_HDMI, AUDIO_DEVICE_OUT_HDMI,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_HDMI());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_REMOTE_SUBMIX, AUDIO_DEVICE_OUT_REMOTE_SUBMIX,
+                media::AudioDeviceType::IN_SUBMIX, media::AudioDeviceType::OUT_SUBMIX);
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_ANLG_DOCK_HEADSET, AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET,
+                media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
+                media::AudioDeviceDescription::CONNECTION_ANALOG_DOCK());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_DGTL_DOCK_HEADSET, AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET,
+                media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
+                media::AudioDeviceDescription::CONNECTION_DIGITAL_DOCK());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_USB_ACCESSORY, AUDIO_DEVICE_OUT_USB_ACCESSORY,
+                media::AudioDeviceType::IN_ACCESSORY, media::AudioDeviceType::OUT_ACCESSORY,
+                media::AudioDeviceDescription::CONNECTION_USB());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_USB_DEVICE, AUDIO_DEVICE_OUT_USB_DEVICE,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_USB());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_FM_TUNER, AUDIO_DEVICE_OUT_FM,
+                media::AudioDeviceType::IN_FM_TUNER, media::AudioDeviceType::OUT_FM);
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_LINE, AUDIO_DEVICE_OUT_LINE,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_ANALOG());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_SPDIF, AUDIO_DEVICE_OUT_SPDIF,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_SPDIF());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_BLUETOOTH_A2DP, AUDIO_DEVICE_OUT_BLUETOOTH_A2DP,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_BT_A2DP());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_IP, AUDIO_DEVICE_OUT_IP,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_IP_V4());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_BUS, AUDIO_DEVICE_OUT_BUS,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_BUS());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_PROXY, AUDIO_DEVICE_OUT_PROXY,
+                media::AudioDeviceType::IN_AFE_PROXY, media::AudioDeviceType::OUT_AFE_PROXY);
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_USB_HEADSET, AUDIO_DEVICE_OUT_USB_HEADSET,
+                media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
+                media::AudioDeviceDescription::CONNECTION_USB());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_HDMI_ARC, AUDIO_DEVICE_OUT_HDMI_ARC,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_HDMI_ARC());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_HDMI_EARC, AUDIO_DEVICE_OUT_HDMI_EARC,
+                media::AudioDeviceType::IN_DEVICE, media::AudioDeviceType::OUT_DEVICE,
+                media::AudioDeviceDescription::CONNECTION_HDMI_EARC());
+        append_AudioDeviceDescription(pairs,
+                AUDIO_DEVICE_IN_BLE_HEADSET, AUDIO_DEVICE_OUT_BLE_HEADSET,
+                media::AudioDeviceType::IN_HEADSET, media::AudioDeviceType::OUT_HEADSET,
+                media::AudioDeviceDescription::CONNECTION_BT_LE());
+        return pairs;
+    }();
+    return pairs;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(media::AudioFormatType type) {
+    media::AudioFormatDescription result;
+    result.type = type;
+    return result;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(media::PcmType pcm) {
+    auto result = make_AudioFormatDescription(media::AudioFormatType::PCM);
+    result.pcm = pcm;
+    return result;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(const std::string& encoding) {
+    media::AudioFormatDescription result;
+    result.encoding = encoding;
+    return result;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(media::PcmType transport,
+        const std::string& encoding) {
+    auto result = make_AudioFormatDescription(encoding);
+    result.pcm = transport;
+    return result;
+}
+
+const detail::AudioFormatPairs& getAudioFormatPairs() {
+    static const detail::AudioFormatPairs pairs = {{
+        {
+            AUDIO_FORMAT_INVALID,
+            make_AudioFormatDescription(media::AudioFormatType::SYS_RESERVED_INVALID)
+        },
+        {
+            AUDIO_FORMAT_DEFAULT, media::AudioFormatDescription{}
+        },
+        {
+            AUDIO_FORMAT_PCM_16_BIT, make_AudioFormatDescription(media::PcmType::INT_16_BIT)
+        },
+        {
+            AUDIO_FORMAT_PCM_8_BIT, make_AudioFormatDescription(media::PcmType::UINT_8_BIT)
+        },
+        {
+            AUDIO_FORMAT_PCM_32_BIT, make_AudioFormatDescription(media::PcmType::INT_32_BIT)
+        },
+        {
+            AUDIO_FORMAT_PCM_8_24_BIT, make_AudioFormatDescription(media::PcmType::FIXED_Q_8_24)
+        },
+        {
+            AUDIO_FORMAT_PCM_FLOAT, make_AudioFormatDescription(media::PcmType::FLOAT_32_BIT)
+        },
+        {
+            AUDIO_FORMAT_PCM_24_BIT_PACKED, make_AudioFormatDescription(media::PcmType::INT_24_BIT)
+        },
+        {
+            // See the comment in MediaDefs.h.
+            AUDIO_FORMAT_MP3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEG)
+        },
+        {
+            AUDIO_FORMAT_AMR_NB, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AMR_NB)
+        },
+        {
+            AUDIO_FORMAT_AMR_WB, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AMR_WB)
+        },
+        {
+            // Note: in MediaDefs.cpp MEDIA_MIMETYPE_AUDIO_AAC = "audio/mp4a-latm".
+            AUDIO_FORMAT_AAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_FORMAT)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_MAIN, make_AudioFormatDescription("audio/aac.main")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_LC, make_AudioFormatDescription("audio/aac.lc")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_SSR, make_AudioFormatDescription("audio/aac.ssr")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_LTP, make_AudioFormatDescription("audio/aac.ltp")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_HE_V1, make_AudioFormatDescription("audio/aac.he.v1")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_SCALABLE, make_AudioFormatDescription("audio/aac.scalable")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ERLC, make_AudioFormatDescription("audio/aac.erlc")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_LD, make_AudioFormatDescription("audio/aac.ld")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_HE_V2, make_AudioFormatDescription("audio/aac.he.v2")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ELD, make_AudioFormatDescription("audio/aac.eld")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_XHE, make_AudioFormatDescription("audio/aac.xhe")
+        },
+        // AUDIO_FORMAT_HE_AAC_V1 and HE_AAC_V2 are removed since they were deprecated long time
+        // ago.
+        {
+            AUDIO_FORMAT_VORBIS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_VORBIS)
+        },
+        {
+            AUDIO_FORMAT_OPUS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_OPUS)
+        },
+        {
+            AUDIO_FORMAT_AC3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AC3)
+        },
+        {
+            AUDIO_FORMAT_E_AC3, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EAC3)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_E_AC3_JOC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EAC3_JOC)
+        },
+        {
+            AUDIO_FORMAT_DTS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DTS)
+        },
+        {
+            AUDIO_FORMAT_DTS_HD, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DTS_HD)
+        },
+        // In the future, we would like to represent encapsulated bitstreams as
+        // nested AudioFormatDescriptions. The legacy 'AUDIO_FORMAT_IEC61937' type doesn't
+        // specify the format of the encapsulated bitstream.
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_IEC61937,
+            make_AudioFormatDescription(media::PcmType::INT_16_BIT, "audio/x-iec61937")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_DOLBY_TRUEHD, make_AudioFormatDescription("audio/vnd.dolby.truehd")
+        },
+        {
+            AUDIO_FORMAT_EVRC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EVRC)
+        },
+        {
+            AUDIO_FORMAT_EVRCB, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EVRCB)
+        },
+        {
+            AUDIO_FORMAT_EVRCWB, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EVRCWB)
+        },
+        {
+            AUDIO_FORMAT_EVRCNW, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_EVRCNW)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADIF, make_AudioFormatDescription("audio/aac.adif")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_WMA, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_WMA)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_WMA_PRO, make_AudioFormatDescription("audio/x-ms-wma.pro")
+        },
+        {
+            AUDIO_FORMAT_AMR_WB_PLUS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AMR_WB_PLUS)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MP2, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_MPEG_LAYER_II)
+        },
+        {
+            AUDIO_FORMAT_QCELP, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_QCELP)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_DSD, make_AudioFormatDescription("audio/vnd.sony.dsd")
+        },
+        {
+            AUDIO_FORMAT_FLAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_FLAC)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_ALAC, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_ALAC)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_APE, make_AudioFormatDescription("audio/x-ape")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC_ADTS)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_MAIN, make_AudioFormatDescription("audio/aac-adts.main")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_LC, make_AudioFormatDescription("audio/aac-adts.lc")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_SSR, make_AudioFormatDescription("audio/aac-adts.ssr")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_LTP, make_AudioFormatDescription("audio/aac-adts.ltp")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_HE_V1, make_AudioFormatDescription("audio/aac-adts.he.v1")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_SCALABLE, make_AudioFormatDescription("audio/aac-adts.scalable")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_ERLC, make_AudioFormatDescription("audio/aac-adts.erlc")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_LD, make_AudioFormatDescription("audio/aac-adts.ld")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_HE_V2, make_AudioFormatDescription("audio/aac-adts.he.v2")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_ELD, make_AudioFormatDescription("audio/aac-adts.eld")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_ADTS_XHE, make_AudioFormatDescription("audio/aac-adts.xhe")
+        },
+        {
+            // Note: not in the IANA registry. "vnd.octel.sbc" is not BT SBC.
+            AUDIO_FORMAT_SBC, make_AudioFormatDescription("audio/x-sbc")
+        },
+        {
+            AUDIO_FORMAT_APTX, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_APTX)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_APTX_HD, make_AudioFormatDescription("audio/vnd.qcom.aptx.hd")
+        },
+        {
+            // Note: not in the IANA registry. Matches MediaDefs.cpp.
+            AUDIO_FORMAT_AC4, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AC4)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_LDAC, make_AudioFormatDescription("audio/vnd.sony.ldac")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MAT, make_AudioFormatDescription("audio/vnd.dolby.mat")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MAT_1_0, make_AudioFormatDescription("audio/vnd.dolby.mat.1.0")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MAT_2_0, make_AudioFormatDescription("audio/vnd.dolby.mat.2.0")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MAT_2_1, make_AudioFormatDescription("audio/vnd.dolby.mat.2.1")
+        },
+        {
+            AUDIO_FORMAT_AAC_LATM, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_AAC)
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_LATM_LC, make_AudioFormatDescription("audio/mp4a-latm.lc")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_LATM_HE_V1, make_AudioFormatDescription("audio/mp4a-latm.he.v1")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_AAC_LATM_HE_V2, make_AudioFormatDescription("audio/mp4a-latm.he.v2")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_CELT, make_AudioFormatDescription("audio/x-celt")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_APTX_ADAPTIVE, make_AudioFormatDescription("audio/vnd.qcom.aptx.adaptive")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_LHDC, make_AudioFormatDescription("audio/vnd.savitech.lhdc")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_LHDC_LL, make_AudioFormatDescription("audio/vnd.savitech.lhdc.ll")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_APTX_TWSP, make_AudioFormatDescription("audio/vnd.qcom.aptx.twsp")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_LC3, make_AudioFormatDescription("audio/x-lc3")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MPEGH, make_AudioFormatDescription("audio/x-mpegh")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MPEGH_BL_L3, make_AudioFormatDescription("audio/x-mpegh.bl.l3")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MPEGH_BL_L4, make_AudioFormatDescription("audio/x-mpegh.bl.l4")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MPEGH_LC_L3, make_AudioFormatDescription("audio/x-mpegh.lc.l3")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_MPEGH_LC_L4, make_AudioFormatDescription("audio/x-mpegh.lc.l4")
+        },
+        {
+            // Note: not in the IANA registry.
+            AUDIO_FORMAT_IEC60958,
+            make_AudioFormatDescription(media::PcmType::INT_24_BIT, "audio/x-iec60958")
+        },
+        {
+            AUDIO_FORMAT_DTS_UHD, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DTS_UHD)
+        },
+        {
+            AUDIO_FORMAT_DRA, make_AudioFormatDescription(MEDIA_MIMETYPE_AUDIO_DRA)
+        },
+    }};
+    return pairs;
+}
+
+template<typename S, typename T>
+std::unordered_map<S, T> make_DirectMap(const std::vector<std::pair<S, T>>& v) {
+    std::unordered_map<S, T> result(v.begin(), v.end());
+    LOG_ALWAYS_FATAL_IF(result.size() != v.size(), "Duplicate key elements detected");
+    return result;
+}
+
+template<typename S, typename T>
+std::unordered_map<S, T> make_DirectMap(
+        const std::vector<std::pair<S, T>>& v1, const std::vector<std::pair<S, T>>& v2) {
+    std::unordered_map<S, T> result(v1.begin(), v1.end());
+    LOG_ALWAYS_FATAL_IF(result.size() != v1.size(), "Duplicate key elements detected in v1");
+    result.insert(v2.begin(), v2.end());
+    LOG_ALWAYS_FATAL_IF(result.size() != v1.size() + v2.size(),
+            "Duplicate key elements detected in v1+v2");
+    return result;
+}
+
+template<typename S, typename T>
+std::unordered_map<T, S> make_ReverseMap(const std::vector<std::pair<S, T>>& v) {
+    std::unordered_map<T, S> result;
+    std::transform(v.begin(), v.end(), std::inserter(result, result.begin()),
+            [](const std::pair<S, T>& p) {
+                return std::make_pair(p.second, p.first);
+            });
+    LOG_ALWAYS_FATAL_IF(result.size() != v.size(), "Duplicate key elements detected");
+    return result;
+}
+
+}  // namespace
+
+ConversionResult<audio_channel_mask_t> aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+        const media::AudioChannelLayout& aidl, bool isInput) {
+    using ReverseMap = std::unordered_map<media::AudioChannelLayout, audio_channel_mask_t>;
+    using Tag = media::AudioChannelLayout::Tag;
+    static const ReverseMap mIdx = make_ReverseMap(getIndexAudioChannelPairs());
+    static const ReverseMap mIn = make_ReverseMap(getInAudioChannelPairs());
+    static const ReverseMap mOut = make_ReverseMap(getOutAudioChannelPairs());
+    static const ReverseMap mVoice = make_ReverseMap(getVoiceAudioChannelPairs());
+
+    auto convert = [](const media::AudioChannelLayout& aidl, const ReverseMap& m,
+            const char* func, const char* type) -> ConversionResult<audio_channel_mask_t> {
+        if (auto it = m.find(aidl); it != m.end()) {
+            return it->second;
+        } else {
+            ALOGE("%s: no legacy %s audio_channel_mask_t found for %s", func, type,
+                    aidl.toString().c_str());
+            return unexpected(BAD_VALUE);
+        }
+    };
+
+    switch (aidl.getTag()) {
+        case Tag::none:
+            return AUDIO_CHANNEL_NONE;
+        case Tag::invalid:
+            return AUDIO_CHANNEL_INVALID;
+        case Tag::indexMask:
+            return convert(aidl, mIdx, __func__, "index");
+        case Tag::layoutMask:
+            return convert(aidl, isInput ? mIn : mOut, __func__, isInput ? "input" : "output");
+        case Tag::voiceMask:
+            return convert(aidl, mVoice, __func__, "voice");
+    }
+    ALOGE("%s: unexpected tag value %d", __func__, aidl.getTag());
+    return unexpected(BAD_VALUE);
+}
+
+ConversionResult<media::AudioChannelLayout> legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
+        audio_channel_mask_t legacy, bool isInput) {
+    using DirectMap = std::unordered_map<audio_channel_mask_t, media::AudioChannelLayout>;
+    using Tag = media::AudioChannelLayout::Tag;
+    static const DirectMap mIdx = make_DirectMap(getIndexAudioChannelPairs());
+    static const DirectMap mInAndVoice = make_DirectMap(
+            getInAudioChannelPairs(), getVoiceAudioChannelPairs());
+    static const DirectMap mOut = make_DirectMap(getOutAudioChannelPairs());
+
+    auto convert = [](const audio_channel_mask_t legacy, const DirectMap& m,
+            const char* func, const char* type) -> ConversionResult<media::AudioChannelLayout> {
+        if (auto it = m.find(legacy); it != m.end()) {
+            return it->second;
+        } else {
+            ALOGE("%s: no AudioChannelLayout found for legacy %s audio_channel_mask_t value 0x%x",
+                    func, type, legacy);
+            return unexpected(BAD_VALUE);
+        }
+    };
+
+    if (legacy == AUDIO_CHANNEL_NONE) {
+        return media::AudioChannelLayout{};
+    } else if (legacy == AUDIO_CHANNEL_INVALID) {
+        return media::AudioChannelLayout::make<Tag::invalid>(0);
+    }
+
+    const audio_channel_representation_t repr = audio_channel_mask_get_representation(legacy);
+    if (repr == AUDIO_CHANNEL_REPRESENTATION_INDEX) {
+        return convert(legacy, mIdx, __func__, "index");
+    } else if (repr == AUDIO_CHANNEL_REPRESENTATION_POSITION) {
+        return convert(legacy, isInput ? mInAndVoice : mOut, __func__,
+                isInput ? "input / voice" : "output");
+    }
+
+    ALOGE("%s: unknown representation %d in audio_channel_mask_t value 0x%x",
+            __func__, repr, legacy);
+    return unexpected(BAD_VALUE);
+}
+
+ConversionResult<audio_devices_t> aidl2legacy_AudioDeviceDescription_audio_devices_t(
+        const media::AudioDeviceDescription& aidl) {
+    static const std::unordered_map<media::AudioDeviceDescription, audio_devices_t> m =
+            make_ReverseMap(getAudioDevicePairs());
+    if (auto it = m.find(aidl); it != m.end()) {
+        return it->second;
+    } else {
+        ALOGE("%s: no legacy audio_devices_t found for %s", __func__, aidl.toString().c_str());
+        return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioDeviceDescription> legacy2aidl_audio_devices_t_AudioDeviceDescription(
+        audio_devices_t legacy) {
+    static const std::unordered_map<audio_devices_t, media::AudioDeviceDescription> m =
+            make_DirectMap(getAudioDevicePairs());
+    if (auto it = m.find(legacy); it != m.end()) {
+        return it->second;
+    } else {
+        ALOGE("%s: no AudioDeviceDescription found for legacy audio_devices_t value 0x%x",
+                __func__, legacy);
+        return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<audio_format_t> aidl2legacy_AudioFormatDescription_audio_format_t(
+        const media::AudioFormatDescription& aidl) {
+    static const std::unordered_map<media::AudioFormatDescription, audio_format_t> m =
+            make_ReverseMap(getAudioFormatPairs());
+    if (auto it = m.find(aidl); it != m.end()) {
+        return it->second;
+    } else {
+        ALOGE("%s: no legacy audio_format_t found for %s", __func__, aidl.toString().c_str());
+        return unexpected(BAD_VALUE);
+    }
+}
+
+ConversionResult<media::AudioFormatDescription> legacy2aidl_audio_format_t_AudioFormatDescription(
         audio_format_t legacy) {
-    // This relies on AudioFormat being kept in sync with audio_format_t.
-    static_assert(sizeof(media::audio::common::AudioFormat) == sizeof(audio_format_t));
-    return static_cast<media::audio::common::AudioFormat>(legacy);
+    static const std::unordered_map<audio_format_t, media::AudioFormatDescription> m =
+            make_DirectMap(getAudioFormatPairs());
+    if (auto it = m.find(legacy); it != m.end()) {
+        return it->second;
+    } else {
+        ALOGE("%s: no AudioFormatDescription found for legacy audio_format_t value 0x%x",
+                __func__, legacy);
+        return unexpected(BAD_VALUE);
+    }
 }
 
 ConversionResult<audio_gain_mode_t> aidl2legacy_AudioGainMode_audio_gain_mode_t(media::AudioGainMode aidl) {
@@ -449,24 +1259,14 @@
             enumToMask_index<int32_t, media::AudioGainMode>);
 }
 
-ConversionResult<audio_devices_t> aidl2legacy_int32_t_audio_devices_t(int32_t aidl) {
-    // TODO(ytai): bitfield?
-    return convertReinterpret<audio_devices_t>(aidl);
-}
-
-ConversionResult<int32_t> legacy2aidl_audio_devices_t_int32_t(audio_devices_t legacy) {
-    // TODO(ytai): bitfield?
-    return convertReinterpret<int32_t>(legacy);
-}
-
 ConversionResult<audio_gain_config> aidl2legacy_AudioGainConfig_audio_gain_config(
         const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type) {
     audio_gain_config legacy;
     legacy.index = VALUE_OR_RETURN(convertIntegral<int>(aidl.index));
     legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t_mask(aidl.mode));
-    legacy.channel_mask =
-            VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
     const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
+    legacy.channel_mask = VALUE_OR_RETURN(
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(aidl.channelMask, isInput));
     const bool isJoint = bitmaskIsSet(aidl.mode, media::AudioGainMode::JOINT);
     size_t numValues = isJoint ? 1
                                : isInput ? audio_channel_count_from_in_mask(legacy.channel_mask)
@@ -486,9 +1286,9 @@
     media::AudioGainConfig aidl;
     aidl.index = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.index));
     aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t_mask(legacy.mode));
-    aidl.channelMask =
-            VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
     const bool isInput = VALUE_OR_RETURN(direction(role, type)) == Direction::INPUT;
+    aidl.channelMask = VALUE_OR_RETURN(
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
     const bool isJoint = (legacy.mode & AUDIO_GAIN_MODE_JOINT) != 0;
     size_t numValues = isJoint ? 1
                                : isInput ? audio_channel_count_from_in_mask(legacy.channel_mask)
@@ -722,7 +1522,7 @@
         const media::AudioPortConfigDeviceExt& aidl) {
     audio_port_config_device_ext legacy;
     legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
-    legacy.type = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.type));
+    legacy.type = VALUE_OR_RETURN(aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
     RETURN_IF_ERROR(aidl2legacy_string(aidl.address, legacy.address, AUDIO_DEVICE_MAX_ADDRESS_LEN));
     return legacy;
 }
@@ -732,7 +1532,7 @@
         const audio_port_config_device_ext& legacy) {
     media::AudioPortConfigDeviceExt aidl;
     aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
-    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.type));
+    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.type));
     aidl.address = VALUE_OR_RETURN(
             legacy2aidl_string(legacy.address, AUDIO_DEVICE_MAX_ADDRESS_LEN));
     return aidl;
@@ -1053,11 +1853,15 @@
         legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<unsigned int>(aidl.sampleRate));
     }
     if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::CHANNEL_MASK)) {
+        const bool isInput = VALUE_OR_RETURN(direction(aidl.role, aidl.type)) == Direction::INPUT;
         legacy.channel_mask =
-                VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
+                VALUE_OR_RETURN(
+                        aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+                                aidl.channelMask, isInput));
     }
     if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::FORMAT)) {
-        legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+        legacy.format = VALUE_OR_RETURN(
+                aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
     }
     if (bitmaskIsSet(aidl.configMask, media::AudioPortConfigType::GAIN)) {
         legacy.gain = VALUE_OR_RETURN(
@@ -1082,11 +1886,14 @@
         aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
     }
     if (legacy.config_mask & AUDIO_PORT_CONFIG_CHANNEL_MASK) {
-        aidl.channelMask =
-                VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
+        const bool isInput = VALUE_OR_RETURN(
+                direction(legacy.role, legacy.type)) == Direction::INPUT;
+        aidl.channelMask = VALUE_OR_RETURN(
+                legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
     }
     if (legacy.config_mask & AUDIO_PORT_CONFIG_FORMAT) {
-        aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+        aidl.format = VALUE_OR_RETURN(
+                legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
     }
     if (legacy.config_mask & AUDIO_PORT_CONFIG_GAIN) {
         aidl.gain = VALUE_OR_RETURN(legacy2aidl_audio_gain_config_AudioGainConfig(
@@ -1148,33 +1955,40 @@
 
 ConversionResult<sp<AudioIoDescriptor>> aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(
         const media::AudioIoDescriptor& aidl) {
-    sp<AudioIoDescriptor> legacy(new AudioIoDescriptor());
-    legacy->mIoHandle = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_io_handle_t(aidl.ioHandle));
-    legacy->mPatch = VALUE_OR_RETURN(aidl2legacy_AudioPatch_audio_patch(aidl.patch));
-    legacy->mSamplingRate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.samplingRate));
-    legacy->mFormat = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
-    legacy->mChannelMask =
-            VALUE_OR_RETURN(aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
-    legacy->mFrameCount = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
-    legacy->mFrameCountHAL = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCountHAL));
-    legacy->mLatency = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.latency));
-    legacy->mPortId = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
-    return legacy;
+    const audio_io_handle_t io_handle = VALUE_OR_RETURN(
+            aidl2legacy_int32_t_audio_io_handle_t(aidl.ioHandle));
+    const struct audio_patch patch = VALUE_OR_RETURN(
+            aidl2legacy_AudioPatch_audio_patch(aidl.patch));
+    const bool isInput = aidl.isInput;
+    const uint32_t sampling_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.samplingRate));
+    const audio_format_t format = VALUE_OR_RETURN(
+            aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
+    const audio_channel_mask_t channel_mask = VALUE_OR_RETURN(
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(aidl.channelMask, isInput));
+    const size_t frame_count = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCount));
+    const size_t frame_count_hal = VALUE_OR_RETURN(convertIntegral<size_t>(aidl.frameCountHAL));
+    const uint32_t latency = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.latency));
+    const audio_port_handle_t port_id = VALUE_OR_RETURN(
+            aidl2legacy_int32_t_audio_port_handle_t(aidl.portId));
+    return sp<AudioIoDescriptor>::make(io_handle, patch, isInput, sampling_rate, format,
+            channel_mask, frame_count, frame_count_hal, latency, port_id);
 }
 
 ConversionResult<media::AudioIoDescriptor> legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(
         const sp<AudioIoDescriptor>& legacy) {
     media::AudioIoDescriptor aidl;
-    aidl.ioHandle = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(legacy->mIoHandle));
-    aidl.patch = VALUE_OR_RETURN(legacy2aidl_audio_patch_AudioPatch(legacy->mPatch));
-    aidl.samplingRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy->mSamplingRate));
-    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy->mFormat));
-    aidl.channelMask = VALUE_OR_RETURN(
-            legacy2aidl_audio_channel_mask_t_int32_t(legacy->mChannelMask));
-    aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy->mFrameCount));
-    aidl.frameCountHAL = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy->mFrameCountHAL));
-    aidl.latency = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy->mLatency));
-    aidl.portId = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy->mPortId));
+    aidl.ioHandle = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(legacy->getIoHandle()));
+    aidl.patch = VALUE_OR_RETURN(legacy2aidl_audio_patch_AudioPatch(legacy->getPatch()));
+    aidl.isInput = legacy->getIsInput();
+    aidl.samplingRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy->getSamplingRate()));
+    aidl.format = VALUE_OR_RETURN(
+            legacy2aidl_audio_format_t_AudioFormatDescription(legacy->getFormat()));
+    aidl.channelMask = VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
+                    legacy->getChannelMask(), legacy->getIsInput()));
+    aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy->getFrameCount()));
+    aidl.frameCountHAL = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy->getFrameCountHAL()));
+    aidl.latency = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy->getLatency()));
+    aidl.portId = VALUE_OR_RETURN(legacy2aidl_audio_port_handle_t_int32_t(legacy->getPortId()));
     return aidl;
 }
 
@@ -1473,7 +2287,7 @@
     legacy.version = VALUE_OR_RETURN(convertIntegral<uint16_t>(aidl.version));
     legacy.size = sizeof(audio_offload_info_t);
     audio_config_base_t config = VALUE_OR_RETURN(
-            aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.config));
+            aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.config, false /*isInput*/));
     legacy.sample_rate = config.sample_rate;
     legacy.channel_mask = config.channel_mask;
     legacy.format = config.format;
@@ -1502,9 +2316,10 @@
     }
     aidl.version = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.version));
     aidl.config.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
-    aidl.config.channelMask = VALUE_OR_RETURN(
-            legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
-    aidl.config.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+    aidl.config.channelMask = VALUE_OR_RETURN(legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
+                    legacy.channel_mask, false /*isInput*/));
+    aidl.config.format = VALUE_OR_RETURN(
+            legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
     aidl.streamType = VALUE_OR_RETURN(
             legacy2aidl_audio_stream_type_t_AudioStreamType(legacy.stream_type));
     aidl.bitRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.bit_rate));
@@ -1531,12 +2346,12 @@
 }
 
 ConversionResult<audio_config_t>
-aidl2legacy_AudioConfig_audio_config_t(const media::AudioConfig& aidl) {
+aidl2legacy_AudioConfig_audio_config_t(const media::AudioConfig& aidl, bool isInput) {
     audio_config_t legacy;
     legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.sampleRate));
     legacy.channel_mask = VALUE_OR_RETURN(
-            aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
-    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(aidl.channelMask, isInput));
+    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
     legacy.offload_info = VALUE_OR_RETURN(
             aidl2legacy_AudioOffloadInfo_audio_offload_info_t(aidl.offloadInfo));
     legacy.frame_count = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.frameCount));
@@ -1544,12 +2359,12 @@
 }
 
 ConversionResult<media::AudioConfig>
-legacy2aidl_audio_config_t_AudioConfig(const audio_config_t& legacy) {
+legacy2aidl_audio_config_t_AudioConfig(const audio_config_t& legacy, bool isInput) {
     media::AudioConfig aidl;
     aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
     aidl.channelMask = VALUE_OR_RETURN(
-            legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
-    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
+    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
     aidl.offloadInfo = VALUE_OR_RETURN(
             legacy2aidl_audio_offload_info_t_AudioOffloadInfo(legacy.offload_info));
     aidl.frameCount = VALUE_OR_RETURN(convertIntegral<int64_t>(legacy.frame_count));
@@ -1557,22 +2372,22 @@
 }
 
 ConversionResult<audio_config_base_t>
-aidl2legacy_AudioConfigBase_audio_config_base_t(const media::AudioConfigBase& aidl) {
+aidl2legacy_AudioConfigBase_audio_config_base_t(const media::AudioConfigBase& aidl, bool isInput) {
     audio_config_base_t legacy;
     legacy.sample_rate = VALUE_OR_RETURN(convertIntegral<uint32_t>(aidl.sampleRate));
     legacy.channel_mask = VALUE_OR_RETURN(
-            aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
-    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(aidl.channelMask, isInput));
+    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
     return legacy;
 }
 
 ConversionResult<media::AudioConfigBase>
-legacy2aidl_audio_config_base_t_AudioConfigBase(const audio_config_base_t& legacy) {
+legacy2aidl_audio_config_base_t_AudioConfigBase(const audio_config_base_t& legacy, bool isInput) {
     media::AudioConfigBase aidl;
     aidl.sampleRate = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.sample_rate));
     aidl.channelMask = VALUE_OR_RETURN(
-            legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
-    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
+    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
     return aidl;
 }
 
@@ -1785,7 +2600,8 @@
 aidl2legacy_AudioPortDeviceExt_audio_port_device_ext(const media::AudioPortDeviceExt& aidl) {
     audio_port_device_ext legacy;
     legacy.hw_module = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_module_handle_t(aidl.hwModule));
-    legacy.type = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.device.type));
+    legacy.type = VALUE_OR_RETURN(
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.device.type));
     RETURN_IF_ERROR(
             aidl2legacy_string(aidl.device.address, legacy.address, sizeof(legacy.address)));
     legacy.encapsulation_modes = VALUE_OR_RETURN(
@@ -1799,7 +2615,8 @@
 legacy2aidl_audio_port_device_ext_AudioPortDeviceExt(const audio_port_device_ext& legacy) {
     media::AudioPortDeviceExt aidl;
     aidl.hwModule = VALUE_OR_RETURN(legacy2aidl_audio_module_handle_t_int32_t(legacy.hw_module));
-    aidl.device.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.type));
+    aidl.device.type = VALUE_OR_RETURN(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.type));
     aidl.device.address = VALUE_OR_RETURN(
             legacy2aidl_string(legacy.address, sizeof(legacy.address)));
     aidl.encapsulationModes = VALUE_OR_RETURN(
@@ -1899,9 +2716,9 @@
 }
 
 ConversionResult<audio_profile>
-aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl) {
+aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl, bool isInput) {
     audio_profile legacy;
-    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(aidl.format));
+    legacy.format = VALUE_OR_RETURN(aidl2legacy_AudioFormatDescription_audio_format_t(aidl.format));
 
     if (aidl.samplingRates.size() > std::size(legacy.sample_rates)) {
         return unexpected(BAD_VALUE);
@@ -1916,7 +2733,9 @@
     }
     RETURN_IF_ERROR(
             convertRange(aidl.channelMasks.begin(), aidl.channelMasks.end(), legacy.channel_masks,
-                         aidl2legacy_int32_t_audio_channel_mask_t));
+                    [isInput](const media::AudioChannelLayout& l) {
+                        return aidl2legacy_AudioChannelLayout_audio_channel_mask_t(l, isInput);
+                    }));
     legacy.num_channel_masks = aidl.channelMasks.size();
 
     legacy.encapsulation_type = VALUE_OR_RETURN(
@@ -1925,9 +2744,9 @@
 }
 
 ConversionResult<media::AudioProfile>
-legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy) {
+legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy, bool isInput) {
     media::AudioProfile aidl;
-    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(legacy.format));
+    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(legacy.format));
 
     if (legacy.num_sample_rates > std::size(legacy.sample_rates)) {
         return unexpected(BAD_VALUE);
@@ -1943,7 +2762,9 @@
     RETURN_IF_ERROR(
             convertRange(legacy.channel_masks, legacy.channel_masks + legacy.num_channel_masks,
                          std::back_inserter(aidl.channelMasks),
-                         legacy2aidl_audio_channel_mask_t_int32_t));
+                    [isInput](audio_channel_mask_t m) {
+                        return legacy2aidl_audio_channel_mask_t_AudioChannelLayout(m, isInput);
+                    }));
 
     aidl.encapsulationType = VALUE_OR_RETURN(
             legacy2aidl_audio_encapsulation_type_t_AudioEncapsulationType(
@@ -1955,8 +2776,8 @@
 aidl2legacy_AudioGain_audio_gain(const media::AudioGain& aidl) {
     audio_gain legacy;
     legacy.mode = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_gain_mode_t_mask(aidl.mode));
-    legacy.channel_mask = VALUE_OR_RETURN(
-            aidl2legacy_int32_t_audio_channel_mask_t(aidl.channelMask));
+    legacy.channel_mask = VALUE_OR_RETURN(aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+                    aidl.channelMask, aidl.isInput));
     legacy.min_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.minValue));
     legacy.max_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.maxValue));
     legacy.default_value = VALUE_OR_RETURN(convertIntegral<int>(aidl.defaultValue));
@@ -1967,11 +2788,12 @@
 }
 
 ConversionResult<media::AudioGain>
-legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy) {
+legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy, bool isInput) {
     media::AudioGain aidl;
     aidl.mode = VALUE_OR_RETURN(legacy2aidl_audio_gain_mode_t_int32_t_mask(legacy.mode));
+    aidl.isInput = isInput;
     aidl.channelMask = VALUE_OR_RETURN(
-            legacy2aidl_audio_channel_mask_t_int32_t(legacy.channel_mask));
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(legacy.channel_mask, isInput));
     aidl.minValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.min_value));
     aidl.maxValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.max_value));
     aidl.defaultValue = VALUE_OR_RETURN(convertIntegral<int32_t>(legacy.default_value));
@@ -1992,8 +2814,11 @@
     if (aidl.profiles.size() > std::size(legacy.audio_profiles)) {
         return unexpected(BAD_VALUE);
     }
+    const bool isInput = VALUE_OR_RETURN(direction(aidl.role, aidl.type)) == Direction::INPUT;
     RETURN_IF_ERROR(convertRange(aidl.profiles.begin(), aidl.profiles.end(), legacy.audio_profiles,
-                                 aidl2legacy_AudioProfile_audio_profile));
+                                 [isInput](const media::AudioProfile& p) {
+                                     return aidl2legacy_AudioProfile_audio_profile(p, isInput);
+                                 }));
     legacy.num_audio_profiles = aidl.profiles.size();
 
     if (aidl.extraAudioDescriptors.size() > std::size(legacy.extra_audio_descriptors)) {
@@ -2029,10 +2854,13 @@
     if (legacy.num_audio_profiles > std::size(legacy.audio_profiles)) {
         return unexpected(BAD_VALUE);
     }
+    const bool isInput = VALUE_OR_RETURN(direction(legacy.role, legacy.type)) == Direction::INPUT;
     RETURN_IF_ERROR(
             convertRange(legacy.audio_profiles, legacy.audio_profiles + legacy.num_audio_profiles,
                          std::back_inserter(aidl.profiles),
-                         legacy2aidl_audio_profile_AudioProfile));
+                         [isInput](const audio_profile& p) {
+                             return legacy2aidl_audio_profile_AudioProfile(p, isInput);
+                         }));
 
     if (legacy.num_extra_audio_descriptors > std::size(legacy.extra_audio_descriptors)) {
         return unexpected(BAD_VALUE);
@@ -2049,7 +2877,9 @@
     RETURN_IF_ERROR(
             convertRange(legacy.gains, legacy.gains + legacy.num_gains,
                          std::back_inserter(aidl.gains),
-                         legacy2aidl_audio_gain_AudioGain));
+                         [isInput](const audio_gain& g) {
+                             return legacy2aidl_audio_gain_AudioGain(g, isInput);
+                         }));
 
     aidl.activeConfig = VALUE_OR_RETURN(
             legacy2aidl_audio_port_config_AudioPortConfig(legacy.active_config));
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 9c307ff..63da4d1 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -229,6 +229,7 @@
         "libbinder",
         "liblog",
         "libshmemcompat",
+        "libstagefright_foundation",
         "libutils",
         "shared-file-region-aidl-cpp",
         "framework-permission-aidl-cpp",
@@ -302,15 +303,20 @@
     local_include_dir: "aidl",
     srcs: [
         "aidl/android/media/AudioAttributesInternal.aidl",
+        "aidl/android/media/AudioChannelLayout.aidl",
         "aidl/android/media/AudioClient.aidl",
         "aidl/android/media/AudioConfig.aidl",
         "aidl/android/media/AudioConfigBase.aidl",
         "aidl/android/media/AudioContentType.aidl",
         "aidl/android/media/AudioDevice.aidl",
+        "aidl/android/media/AudioDeviceDescription.aidl",
+        "aidl/android/media/AudioDeviceType.aidl",
         "aidl/android/media/AudioDualMonoMode.aidl",
         "aidl/android/media/AudioEncapsulationMode.aidl",
         "aidl/android/media/AudioEncapsulationMetadataType.aidl",
         "aidl/android/media/AudioEncapsulationType.aidl",
+        "aidl/android/media/AudioFormatDescription.aidl",
+        "aidl/android/media/AudioFormatType.aidl",
         "aidl/android/media/AudioFlag.aidl",
         "aidl/android/media/AudioGain.aidl",
         "aidl/android/media/AudioGainConfig.aidl",
@@ -350,10 +356,11 @@
         "aidl/android/media/AudioVibratorInfo.aidl",
         "aidl/android/media/EffectDescriptor.aidl",
         "aidl/android/media/ExtraAudioDescriptor.aidl",
+        "aidl/android/media/PcmType.aidl",
         "aidl/android/media/TrackSecondaryOutputInfo.aidl",
     ],
     imports: [
-        "audio_common-aidl",
+        "android.media.audio.common.types",
         "framework-permission-aidl",
     ],
     backend: {
@@ -364,6 +371,9 @@
                 "com.android.media",
             ],
         },
+        java: {
+            sdk_version: "module_current",
+        },
     },
 }
 aidl_interface {
@@ -391,7 +401,6 @@
         "aidl/android/media/SoundTriggerSession.aidl",
     ],
     imports: [
-        "audio_common-aidl",
         "audioclient-types-aidl",
     ],
     backend: {
@@ -402,6 +411,9 @@
                 "com.android.media",
             ],
         },
+        java: {
+            sdk_version: "module_current",
+        },
     },
 }
 
@@ -431,7 +443,6 @@
         "aidl/android/media/IAudioTrackCallback.aidl",
     ],
     imports: [
-        "audio_common-aidl",
         "audioclient-types-aidl",
         "av-types-aidl",
         "effect-aidl",
@@ -447,6 +458,9 @@
                 "com.android.media",
             ],
         },
+        java: {
+            sdk_version: "module_current",
+        },
     },
 }
 
@@ -465,7 +479,6 @@
         "aidl/android/media/IAudioPolicyServiceClient.aidl",
     ],
     imports: [
-        "audio_common-aidl",
         "audioclient-types-aidl",
         "audiopolicy-types-aidl",
         "capture_state_listener-aidl",
@@ -481,5 +494,8 @@
                 "com.android.media",
             ],
         },
+        java: {
+            sdk_version: "module_current",
+        },
     },
 }
diff --git a/media/libaudioclient/AudioRecord.cpp b/media/libaudioclient/AudioRecord.cpp
index a1d3bdb..a02a373 100644
--- a/media/libaudioclient/AudioRecord.cpp
+++ b/media/libaudioclient/AudioRecord.cpp
@@ -668,6 +668,8 @@
 // ---- Explicit Routing ---------------------------------------------------
 status_t AudioRecord::setInputDevice(audio_port_handle_t deviceId) {
     AutoMutex lock(mLock);
+    ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
+            __func__, mPortId, deviceId, mSelectedDeviceId);
     if (mSelectedDeviceId != deviceId) {
         mSelectedDeviceId = deviceId;
         if (mStatus == NO_ERROR) {
diff --git a/media/libaudioclient/AudioSystem.cpp b/media/libaudioclient/AudioSystem.cpp
index 88e752b..2d376b9 100644
--- a/media/libaudioclient/AudioSystem.cpp
+++ b/media/libaudioclient/AudioSystem.cpp
@@ -336,7 +336,7 @@
     if (desc == 0) {
         *samplingRate = af->sampleRate(ioHandle);
     } else {
-        *samplingRate = desc->mSamplingRate;
+        *samplingRate = desc->getSamplingRate();
     }
     if (*samplingRate == 0) {
         ALOGE("AudioSystem::getSamplingRate failed for ioHandle %d", ioHandle);
@@ -371,7 +371,7 @@
     if (desc == 0) {
         *frameCount = af->frameCount(ioHandle);
     } else {
-        *frameCount = desc->mFrameCount;
+        *frameCount = desc->getFrameCount();
     }
     if (*frameCount == 0) {
         ALOGE("AudioSystem::getFrameCount failed for ioHandle %d", ioHandle);
@@ -406,7 +406,7 @@
     if (outputDesc == 0) {
         *latency = af->latency(output);
     } else {
-        *latency = outputDesc->mLatency;
+        *latency = outputDesc->getLatency();
     }
 
     ALOGV("getLatency() output %d, latency %d", output, *latency);
@@ -480,6 +480,12 @@
     return af->systemReady();
 }
 
+status_t AudioSystem::audioPolicyReady() {
+    const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
+    if (af == 0) return NO_INIT;
+    return af->audioPolicyReady();
+}
+
 status_t AudioSystem::getFrameCountHAL(audio_io_handle_t ioHandle,
                                        size_t* frameCount) {
     const sp<IAudioFlinger>& af = AudioSystem::get_audio_flinger();
@@ -488,7 +494,7 @@
     if (desc == 0) {
         *frameCount = af->frameCountHAL(ioHandle);
     } else {
-        *frameCount = desc->mFrameCountHAL;
+        *frameCount = desc->getFrameCountHAL();
     }
     if (*frameCount == 0) {
         ALOGE("AudioSystem::getFrameCountHAL failed for ioHandle %d", ioHandle);
@@ -529,15 +535,15 @@
 Status AudioSystem::AudioFlingerClient::ioConfigChanged(
         media::AudioIoConfigEvent _event,
         const media::AudioIoDescriptor& _ioDesc) {
-    audio_io_config_event event = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioIoConfigEvent_audio_io_config_event(_event));
+    audio_io_config_event_t event = VALUE_OR_RETURN_BINDER_STATUS(
+            aidl2legacy_AudioIoConfigEvent_audio_io_config_event_t(_event));
     sp<AudioIoDescriptor> ioDesc(
             VALUE_OR_RETURN_BINDER_STATUS(
                     aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(_ioDesc)));
 
     ALOGV("ioConfigChanged() event %d", event);
 
-    if (ioDesc->mIoHandle == AUDIO_IO_HANDLE_NONE) return Status::ok();
+    if (ioDesc->getIoHandle() == AUDIO_IO_HANDLE_NONE) return Status::ok();
 
     audio_port_handle_t deviceId = AUDIO_PORT_HANDLE_NONE;
     std::vector<sp<AudioDeviceCallback>> callbacksToCall;
@@ -550,93 +556,88 @@
             case AUDIO_OUTPUT_REGISTERED:
             case AUDIO_INPUT_OPENED:
             case AUDIO_INPUT_REGISTERED: {
-                sp<AudioIoDescriptor> oldDesc = getIoDescriptor_l(ioDesc->mIoHandle);
+                sp<AudioIoDescriptor> oldDesc = getIoDescriptor_l(ioDesc->getIoHandle());
                 if (oldDesc == 0) {
-                    mIoDescriptors.add(ioDesc->mIoHandle, ioDesc);
+                    mIoDescriptors.add(ioDesc->getIoHandle(), ioDesc);
                 } else {
                     deviceId = oldDesc->getDeviceId();
-                    mIoDescriptors.replaceValueFor(ioDesc->mIoHandle, ioDesc);
+                    mIoDescriptors.replaceValueFor(ioDesc->getIoHandle(), ioDesc);
                 }
 
                 if (ioDesc->getDeviceId() != AUDIO_PORT_HANDLE_NONE) {
                     deviceId = ioDesc->getDeviceId();
                     if (event == AUDIO_OUTPUT_OPENED || event == AUDIO_INPUT_OPENED) {
-                        auto it = mAudioDeviceCallbacks.find(ioDesc->mIoHandle);
+                        auto it = mAudioDeviceCallbacks.find(ioDesc->getIoHandle());
                         if (it != mAudioDeviceCallbacks.end()) {
                             callbacks = it->second;
                         }
                     }
                 }
-                ALOGV("ioConfigChanged() new %s %s %d samplingRate %u, format %#x channel mask %#x "
-                      "frameCount %zu deviceId %d",
+                ALOGV("ioConfigChanged() new %s %s %s",
                       event == AUDIO_OUTPUT_OPENED || event == AUDIO_OUTPUT_REGISTERED ?
                       "output" : "input",
                       event == AUDIO_OUTPUT_OPENED || event == AUDIO_INPUT_OPENED ?
                       "opened" : "registered",
-                      ioDesc->mIoHandle, ioDesc->mSamplingRate, ioDesc->mFormat,
-                      ioDesc->mChannelMask,
-                      ioDesc->mFrameCount, ioDesc->getDeviceId());
+                      ioDesc->toDebugString().c_str());
             }
                 break;
             case AUDIO_OUTPUT_CLOSED:
             case AUDIO_INPUT_CLOSED: {
-                if (getIoDescriptor_l(ioDesc->mIoHandle) == 0) {
+                if (getIoDescriptor_l(ioDesc->getIoHandle()) == 0) {
                     ALOGW("ioConfigChanged() closing unknown %s %d",
-                          event == AUDIO_OUTPUT_CLOSED ? "output" : "input", ioDesc->mIoHandle);
+                          event == AUDIO_OUTPUT_CLOSED ? "output" : "input", ioDesc->getIoHandle());
                     break;
                 }
                 ALOGV("ioConfigChanged() %s %d closed",
-                      event == AUDIO_OUTPUT_CLOSED ? "output" : "input", ioDesc->mIoHandle);
+                      event == AUDIO_OUTPUT_CLOSED ? "output" : "input", ioDesc->getIoHandle());
 
-                mIoDescriptors.removeItem(ioDesc->mIoHandle);
-                mAudioDeviceCallbacks.erase(ioDesc->mIoHandle);
+                mIoDescriptors.removeItem(ioDesc->getIoHandle());
+                mAudioDeviceCallbacks.erase(ioDesc->getIoHandle());
             }
                 break;
 
             case AUDIO_OUTPUT_CONFIG_CHANGED:
             case AUDIO_INPUT_CONFIG_CHANGED: {
-                sp<AudioIoDescriptor> oldDesc = getIoDescriptor_l(ioDesc->mIoHandle);
+                sp<AudioIoDescriptor> oldDesc = getIoDescriptor_l(ioDesc->getIoHandle());
                 if (oldDesc == 0) {
                     ALOGW("ioConfigChanged() modifying unknown %s! %d",
                           event == AUDIO_OUTPUT_CONFIG_CHANGED ? "output" : "input",
-                          ioDesc->mIoHandle);
+                          ioDesc->getIoHandle());
                     break;
                 }
 
                 deviceId = oldDesc->getDeviceId();
-                mIoDescriptors.replaceValueFor(ioDesc->mIoHandle, ioDesc);
+                mIoDescriptors.replaceValueFor(ioDesc->getIoHandle(), ioDesc);
 
                 if (deviceId != ioDesc->getDeviceId()) {
                     deviceId = ioDesc->getDeviceId();
-                    auto it = mAudioDeviceCallbacks.find(ioDesc->mIoHandle);
+                    auto it = mAudioDeviceCallbacks.find(ioDesc->getIoHandle());
                     if (it != mAudioDeviceCallbacks.end()) {
                         callbacks = it->second;
                     }
                 }
-                ALOGV("ioConfigChanged() new config for %s %d samplingRate %u, format %#x "
-                      "channel mask %#x frameCount %zu frameCountHAL %zu deviceId %d",
+                ALOGV("ioConfigChanged() new config for %s %s",
                       event == AUDIO_OUTPUT_CONFIG_CHANGED ? "output" : "input",
-                      ioDesc->mIoHandle, ioDesc->mSamplingRate, ioDesc->mFormat,
-                      ioDesc->mChannelMask, ioDesc->mFrameCount, ioDesc->mFrameCountHAL,
-                      ioDesc->getDeviceId());
+                      ioDesc->toDebugString().c_str());
 
             }
                 break;
             case AUDIO_CLIENT_STARTED: {
-                sp<AudioIoDescriptor> oldDesc = getIoDescriptor_l(ioDesc->mIoHandle);
+                sp<AudioIoDescriptor> oldDesc = getIoDescriptor_l(ioDesc->getIoHandle());
                 if (oldDesc == 0) {
-                    ALOGW("ioConfigChanged() start client on unknown io! %d", ioDesc->mIoHandle);
+                    ALOGW("ioConfigChanged() start client on unknown io! %d",
+                            ioDesc->getIoHandle());
                     break;
                 }
                 ALOGV("ioConfigChanged() AUDIO_CLIENT_STARTED  io %d port %d num callbacks %zu",
-                      ioDesc->mIoHandle, ioDesc->mPortId, mAudioDeviceCallbacks.size());
-                oldDesc->mPatch = ioDesc->mPatch;
-                auto it = mAudioDeviceCallbacks.find(ioDesc->mIoHandle);
+                      ioDesc->getIoHandle(), ioDesc->getPortId(), mAudioDeviceCallbacks.size());
+                oldDesc->setPatch(ioDesc->getPatch());
+                auto it = mAudioDeviceCallbacks.find(ioDesc->getIoHandle());
                 if (it != mAudioDeviceCallbacks.end()) {
                     auto cbks = it->second;
-                    auto it2 = cbks.find(ioDesc->mPortId);
+                    auto it2 = cbks.find(ioDesc->getPortId());
                     if (it2 != cbks.end()) {
-                        callbacks.emplace(ioDesc->mPortId, it2->second);
+                        callbacks.emplace(ioDesc->getPortId(), it2->second);
                         deviceId = oldDesc->getDeviceId();
                     }
                 }
@@ -655,8 +656,8 @@
     // Callbacks must be called without mLock held. May lead to dead lock if calling for
     // example getRoutedDevice that updates the device and tries to acquire mLock.
     for (auto cb  : callbacksToCall) {
-        // If callbacksToCall is not empty, it implies ioDesc->mIoHandle and deviceId are valid
-        cb->onAudioDeviceUpdate(ioDesc->mIoHandle, deviceId);
+        // If callbacksToCall is not empty, it implies ioDesc->getIoHandle() and deviceId are valid
+        cb->onAudioDeviceUpdate(ioDesc->getIoHandle(), deviceId);
     }
 
     return Status::ok();
@@ -846,7 +847,8 @@
     }
 
     media::AudioDevice deviceAidl;
-    deviceAidl.type = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_devices_t_int32_t(device));
+    deviceAidl.type = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     deviceAidl.address = address;
 
     return statusTFromBinderStatus(
@@ -855,7 +857,8 @@
                     VALUE_OR_RETURN_STATUS(
                             legacy2aidl_audio_policy_dev_state_t_AudioPolicyDeviceState(state)),
                     name,
-                    VALUE_OR_RETURN_STATUS(legacy2aidl_audio_format_t_AudioFormat(encodedFormat))));
+                    VALUE_OR_RETURN_STATUS(
+                            legacy2aidl_audio_format_t_AudioFormatDescription(encodedFormat))));
 }
 
 audio_policy_dev_state_t AudioSystem::getDeviceConnectionState(audio_devices_t device,
@@ -865,7 +868,8 @@
 
     auto result = [&]() -> ConversionResult<audio_policy_dev_state_t> {
         media::AudioDevice deviceAidl;
-        deviceAidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(device));
+        deviceAidl.type = VALUE_OR_RETURN(
+                legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
         deviceAidl.address = device_address;
 
         media::AudioPolicyDeviceState result;
@@ -895,12 +899,13 @@
     }
 
     media::AudioDevice deviceAidl;
-    deviceAidl.type = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_devices_t_int32_t(device));
+    deviceAidl.type = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     deviceAidl.address = address;
 
     return statusTFromBinderStatus(
             aps->handleDeviceConfigChange(deviceAidl, name, VALUE_OR_RETURN_STATUS(
-                    legacy2aidl_audio_format_t_AudioFormat(encodedFormat))));
+                    legacy2aidl_audio_format_t_AudioFormatDescription(encodedFormat))));
 }
 
 status_t AudioSystem::setPhoneState(audio_mode_t state, uid_t uid) {
@@ -998,7 +1003,7 @@
             legacy2aidl_audio_attributes_t_AudioAttributesInternal(*attr));
     int32_t sessionAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(session));
     media::AudioConfig configAidl = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_config_t_AudioConfig(*config));
+            legacy2aidl_audio_config_t_AudioConfig(*config, false /*isInput*/));
     int32_t flagsAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
     int32_t selectedDeviceIdAidl = VALUE_OR_RETURN_STATUS(
@@ -1092,7 +1097,7 @@
     int32_t riidAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_unique_id_t_int32_t(riid));
     int32_t sessionAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_session_t_int32_t(session));
     media::AudioConfigBase configAidl = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_config_base_t_AudioConfigBase(*config));
+            legacy2aidl_audio_config_base_t_AudioConfigBase(*config, true /*isInput*/));
     int32_t flagsAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
     int32_t selectedDeviceIdAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_port_handle_t_int32_t(*selectedDeviceId));
@@ -1165,7 +1170,8 @@
     media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
     int32_t indexAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(index));
-    int32_t deviceAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_devices_t_int32_t(device));
+    media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     return statusTFromBinderStatus(
             aps->setStreamVolumeIndex(streamAidl, deviceAidl, indexAidl));
 }
@@ -1178,7 +1184,8 @@
 
     media::AudioStreamType streamAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
-    int32_t deviceAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_devices_t_int32_t(device));
+    media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     int32_t indexAidl;
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
             aps->getStreamVolumeIndex(streamAidl, deviceAidl, &indexAidl)));
@@ -1197,7 +1204,8 @@
     media::AudioAttributesInternal attrAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
     int32_t indexAidl = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(index));
-    int32_t deviceAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_devices_t_int32_t(device));
+    media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     return statusTFromBinderStatus(
             aps->setVolumeIndexForAttributes(attrAidl, deviceAidl, indexAidl));
 }
@@ -1210,7 +1218,8 @@
 
     media::AudioAttributesInternal attrAidl = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
-    int32_t deviceAidl = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_devices_t_int32_t(device));
+    media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     int32_t indexAidl;
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
             aps->getVolumeIndexForAttributes(attrAidl, deviceAidl, &indexAidl)));
@@ -1259,19 +1268,20 @@
     return result.value_or(PRODUCT_STRATEGY_NONE);
 }
 
-audio_devices_t AudioSystem::getDevicesForStream(audio_stream_type_t stream) {
+DeviceTypeSet AudioSystem::getDevicesForStream(audio_stream_type_t stream) {
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
-    if (aps == 0) return AUDIO_DEVICE_NONE;
+    if (aps == 0) return DeviceTypeSet{};
 
-    auto result = [&]() -> ConversionResult<audio_devices_t> {
+    auto result = [&]() -> ConversionResult<DeviceTypeSet> {
         media::AudioStreamType streamAidl = VALUE_OR_RETURN(
                 legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
-        int32_t resultAidl;
+        std::vector<media::AudioDeviceDescription> resultAidl;
         RETURN_IF_ERROR(statusTFromBinderStatus(
                 aps->getDevicesForStream(streamAidl, &resultAidl)));
-        return aidl2legacy_int32_t_audio_devices_t(resultAidl);
+        return convertContainer<DeviceTypeSet>(resultAidl,
+                aidl2legacy_AudioDeviceDescription_audio_devices_t);
     }();
-    return result.value_or(AUDIO_DEVICE_NONE);
+    return result.value_or(DeviceTypeSet{});
 }
 
 status_t AudioSystem::getDevicesForAttributes(const AudioAttributes& aa,
@@ -1690,7 +1700,8 @@
             statusTFromBinderStatus(aps->acquireSoundTriggerSession(&retAidl)));
     *session = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_session_t(retAidl.session));
     *ioHandle = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(retAidl.ioHandle));
-    *device = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_devices_t(retAidl.device));
+    *device = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(retAidl.device));
     return OK;
 }
 
@@ -1830,7 +1841,8 @@
         media::AudioStreamType streamAidl = VALUE_OR_RETURN(
                 legacy2aidl_audio_stream_type_t_AudioStreamType(stream));
         int32_t indexAidl = VALUE_OR_RETURN(convertIntegral<int32_t>(index));
-        int32_t deviceAidl = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(device));
+        media::AudioDeviceDescription deviceAidl = VALUE_OR_RETURN(
+                legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
         float retAidl;
         RETURN_IF_ERROR(statusTFromBinderStatus(
                 aps->getStreamVolumeDB(streamAidl, indexAidl, deviceAidl, &retAidl)));
@@ -1865,7 +1877,7 @@
     media::Int numSurroundFormatsAidl;
     numSurroundFormatsAidl.value =
             VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*numSurroundFormats));
-    std::vector<media::audio::common::AudioFormat> surroundFormatsAidl;
+    std::vector<media::AudioFormatDescription> surroundFormatsAidl;
     std::vector<bool> surroundFormatsEnabledAidl;
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
             aps->getSurroundFormats(&numSurroundFormatsAidl, &surroundFormatsAidl,
@@ -1875,7 +1887,7 @@
             convertIntegral<unsigned int>(numSurroundFormatsAidl.value));
     RETURN_STATUS_IF_ERROR(
             convertRange(surroundFormatsAidl.begin(), surroundFormatsAidl.end(), surroundFormats,
-                         aidl2legacy_AudioFormat_audio_format_t));
+                         aidl2legacy_AudioFormatDescription_audio_format_t));
     std::copy(surroundFormatsEnabledAidl.begin(), surroundFormatsEnabledAidl.end(),
             surroundFormatsEnabled);
     return OK;
@@ -1892,7 +1904,7 @@
     media::Int numSurroundFormatsAidl;
     numSurroundFormatsAidl.value =
             VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(*numSurroundFormats));
-    std::vector<media::audio::common::AudioFormat> surroundFormatsAidl;
+    std::vector<media::AudioFormatDescription> surroundFormatsAidl;
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
             aps->getReportedSurroundFormats(&numSurroundFormatsAidl, &surroundFormatsAidl)));
 
@@ -1900,7 +1912,7 @@
             convertIntegral<unsigned int>(numSurroundFormatsAidl.value));
     RETURN_STATUS_IF_ERROR(
             convertRange(surroundFormatsAidl.begin(), surroundFormatsAidl.end(), surroundFormats,
-                         aidl2legacy_AudioFormat_audio_format_t));
+                         aidl2legacy_AudioFormatDescription_audio_format_t));
     return OK;
 }
 
@@ -1908,8 +1920,8 @@
     const sp<IAudioPolicyService>& aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
 
-    media::audio::common::AudioFormat audioFormatAidl = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_format_t_AudioFormat(audioFormat));
+    media::AudioFormatDescription audioFormatAidl = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_format_t_AudioFormatDescription(audioFormat));
     return statusTFromBinderStatus(
             aps->setSurroundFormatEnabled(audioFormatAidl, enabled));
 }
@@ -1962,12 +1974,13 @@
             & aps = AudioSystem::get_audio_policy_service();
     if (aps == 0) return PERMISSION_DENIED;
 
-    std::vector<media::audio::common::AudioFormat> formatsAidl;
+    std::vector<media::AudioFormatDescription> formatsAidl;
     RETURN_STATUS_IF_ERROR(statusTFromBinderStatus(
             aps->getHwOffloadEncodingFormatsSupportedForA2DP(&formatsAidl)));
     *formats = VALUE_OR_RETURN_STATUS(
-            convertContainer<std::vector<audio_format_t>>(formatsAidl,
-                                                          aidl2legacy_AudioFormat_audio_format_t));
+            convertContainer<std::vector<audio_format_t>>(
+                    formatsAidl,
+                    aidl2legacy_AudioFormatDescription_audio_format_t));
     return OK;
 }
 
@@ -2418,13 +2431,13 @@
         record_client_info_t clientInfoLegacy = VALUE_OR_RETURN_BINDER_STATUS(
                 aidl2legacy_RecordClientInfo_record_client_info_t(clientInfo));
         audio_config_base_t clientConfigLegacy = VALUE_OR_RETURN_BINDER_STATUS(
-                aidl2legacy_AudioConfigBase_audio_config_base_t(clientConfig));
+                aidl2legacy_AudioConfigBase_audio_config_base_t(clientConfig, true /*isInput*/));
         std::vector<effect_descriptor_t> clientEffectsLegacy = VALUE_OR_RETURN_BINDER_STATUS(
                 convertContainer<std::vector<effect_descriptor_t>>(
                         clientEffects,
                         aidl2legacy_EffectDescriptor_effect_descriptor_t));
         audio_config_base_t deviceConfigLegacy = VALUE_OR_RETURN_BINDER_STATUS(
-                aidl2legacy_AudioConfigBase_audio_config_base_t(deviceConfig));
+                aidl2legacy_AudioConfigBase_audio_config_base_t(deviceConfig, true /*isInput*/));
         std::vector<effect_descriptor_t> effectsLegacy = VALUE_OR_RETURN_BINDER_STATUS(
                 convertContainer<std::vector<effect_descriptor_t>>(
                         effects,
diff --git a/media/libaudioclient/AudioTrack.cpp b/media/libaudioclient/AudioTrack.cpp
index 5f802de..5b7760b 100644
--- a/media/libaudioclient/AudioTrack.cpp
+++ b/media/libaudioclient/AudioTrack.cpp
@@ -171,7 +171,7 @@
 
     auto result = [&]() -> ConversionResult<bool> {
         media::AudioConfigBase configAidl = VALUE_OR_RETURN(
-                legacy2aidl_audio_config_base_t_AudioConfigBase(config));
+                legacy2aidl_audio_config_base_t_AudioConfigBase(config, false /*isInput*/));
         media::AudioAttributesInternal attributesAidl = VALUE_OR_RETURN(
                 legacy2aidl_audio_attributes_t_AudioAttributesInternal(attributes));
         bool retAidl;
@@ -1275,10 +1275,6 @@
     if (mOutput == AUDIO_IO_HANDLE_NONE || mProxy.get() == 0) {
         return NO_INIT;
     }
-    // Reject if timed track or compressed audio.
-    if (!audio_is_linear_pcm(mFormat)) {
-        return INVALID_OPERATION;
-    }
 
     ssize_t originalBufferSize = mProxy->getBufferSizeInFrames();
     ssize_t finalBufferSize  = mProxy->setBufferSizeInFrames((uint32_t) bufferSizeInFrames);
@@ -1555,6 +1551,8 @@
 
 status_t AudioTrack::setOutputDevice(audio_port_handle_t deviceId) {
     AutoMutex lock(mLock);
+    ALOGV("%s(%d): deviceId=%d mSelectedDeviceId=%d",
+            __func__, mPortId, deviceId, mSelectedDeviceId);
     if (mSelectedDeviceId != deviceId) {
         mSelectedDeviceId = deviceId;
         if (mStatus == NO_ERROR) {
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index cae81f0..5f12f71 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -55,7 +55,9 @@
 ConversionResult<media::CreateTrackRequest> IAudioFlinger::CreateTrackInput::toAidl() const {
     media::CreateTrackRequest aidl;
     aidl.attr = VALUE_OR_RETURN(legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
-    aidl.config = VALUE_OR_RETURN(legacy2aidl_audio_config_t_AudioConfig(config));
+    // Do not be mislead by 'Input'--this is an input to 'createTrack', which creates output tracks.
+    aidl.config = VALUE_OR_RETURN(legacy2aidl_audio_config_t_AudioConfig(
+                    config, false /*isInput*/));
     aidl.clientInfo = VALUE_OR_RETURN(legacy2aidl_AudioClient_AudioClient(clientInfo));
     aidl.sharedBuffer = VALUE_OR_RETURN(legacy2aidl_NullableIMemory_SharedFileRegion(sharedBuffer));
     aidl.notificationsPerBuffer = VALUE_OR_RETURN(convertIntegral<int32_t>(notificationsPerBuffer));
@@ -74,7 +76,9 @@
 IAudioFlinger::CreateTrackInput::fromAidl(const media::CreateTrackRequest& aidl) {
     IAudioFlinger::CreateTrackInput legacy;
     legacy.attr = VALUE_OR_RETURN(aidl2legacy_AudioAttributesInternal_audio_attributes_t(aidl.attr));
-    legacy.config = VALUE_OR_RETURN(aidl2legacy_AudioConfig_audio_config_t(aidl.config));
+    // Do not be mislead by 'Input'--this is an input to 'createTrack', which creates output tracks.
+    legacy.config = VALUE_OR_RETURN(
+            aidl2legacy_AudioConfig_audio_config_t(aidl.config, false /*isInput*/));
     legacy.clientInfo = VALUE_OR_RETURN(aidl2legacy_AudioClient_AudioClient(aidl.clientInfo));
     legacy.sharedBuffer = VALUE_OR_RETURN(aidl2legacy_NullableSharedFileRegion_IMemory(aidl.sharedBuffer));
     legacy.notificationsPerBuffer = VALUE_OR_RETURN(
@@ -139,7 +143,8 @@
 IAudioFlinger::CreateRecordInput::toAidl() const {
     media::CreateRecordRequest aidl;
     aidl.attr = VALUE_OR_RETURN(legacy2aidl_audio_attributes_t_AudioAttributesInternal(attr));
-    aidl.config = VALUE_OR_RETURN(legacy2aidl_audio_config_base_t_AudioConfigBase(config));
+    aidl.config = VALUE_OR_RETURN(
+            legacy2aidl_audio_config_base_t_AudioConfigBase(config, true /*isInput*/));
     aidl.clientInfo = VALUE_OR_RETURN(legacy2aidl_AudioClient_AudioClient(clientInfo));
     aidl.riid = VALUE_OR_RETURN(legacy2aidl_audio_unique_id_t_int32_t(riid));
     aidl.maxSharedAudioHistoryMs = VALUE_OR_RETURN(
@@ -159,7 +164,8 @@
     IAudioFlinger::CreateRecordInput legacy;
     legacy.attr = VALUE_OR_RETURN(
             aidl2legacy_AudioAttributesInternal_audio_attributes_t(aidl.attr));
-    legacy.config = VALUE_OR_RETURN(aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.config));
+    legacy.config = VALUE_OR_RETURN(
+            aidl2legacy_AudioConfigBase_audio_config_base_t(aidl.config, true /*isInput*/));
     legacy.clientInfo = VALUE_OR_RETURN(aidl2legacy_AudioClient_AudioClient(aidl.clientInfo));
     legacy.riid = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_unique_id_t(aidl.riid));
     legacy.maxSharedAudioHistoryMs = VALUE_OR_RETURN(
@@ -242,9 +248,9 @@
 audio_format_t AudioFlingerClientAdapter::format(audio_io_handle_t output) const {
     auto result = [&]() -> ConversionResult<audio_format_t> {
         int32_t outputAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(output));
-        media::audio::common::AudioFormat aidlRet;
+        media::AudioFormatDescription aidlRet;
         RETURN_IF_ERROR(statusTFromBinderStatus(mDelegate->format(outputAidl, &aidlRet)));
-        return aidl2legacy_AudioFormat_audio_format_t(aidlRet);
+        return aidl2legacy_AudioFormatDescription_audio_format_t(aidlRet);
     }();
     return result.value_or(AUDIO_FORMAT_INVALID);
 }
@@ -410,10 +416,10 @@
                                                      audio_channel_mask_t channelMask) const {
     auto result = [&]() -> ConversionResult<size_t> {
         int32_t sampleRateAidl = VALUE_OR_RETURN(convertIntegral<int32_t>(sampleRate));
-        media::audio::common::AudioFormat formatAidl = VALUE_OR_RETURN(
-                legacy2aidl_audio_format_t_AudioFormat(format));
-        int32_t channelMaskAidl = VALUE_OR_RETURN(
-                legacy2aidl_audio_channel_mask_t_int32_t(channelMask));
+        media::AudioFormatDescription formatAidl = VALUE_OR_RETURN(
+                legacy2aidl_audio_format_t_AudioFormatDescription(format));
+        media::AudioChannelLayout channelMaskAidl = VALUE_OR_RETURN(
+                legacy2aidl_audio_channel_mask_t_AudioChannelLayout(channelMask, true /*isInput*/));
         int64_t aidlRet;
         RETURN_IF_ERROR(statusTFromBinderStatus(
                 mDelegate->getInputBufferSize(sampleRateAidl, formatAidl, channelMaskAidl,
@@ -715,6 +721,10 @@
     return statusTFromBinderStatus(mDelegate->systemReady());
 }
 
+status_t AudioFlingerClientAdapter::audioPolicyReady() {
+    return statusTFromBinderStatus(mDelegate->audioPolicyReady());
+}
+
 size_t AudioFlingerClientAdapter::frameCountHAL(audio_io_handle_t ioHandle) const {
     auto result = [&]() -> ConversionResult<size_t> {
         int32_t ioHandleAidl = VALUE_OR_RETURN(legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
@@ -806,11 +816,11 @@
 }
 
 Status AudioFlingerServerAdapter::format(int32_t output,
-                                         media::audio::common::AudioFormat* _aidl_return) {
+                                         media::AudioFormatDescription* _aidl_return) {
     audio_io_handle_t outputLegacy = VALUE_OR_RETURN_BINDER(
             aidl2legacy_int32_t_audio_io_handle_t(output));
     *_aidl_return = VALUE_OR_RETURN_BINDER(
-            legacy2aidl_audio_format_t_AudioFormat(mDelegate->format(outputLegacy)));
+            legacy2aidl_audio_format_t_AudioFormatDescription(mDelegate->format(outputLegacy)));
     return Status::ok();
 }
 
@@ -934,13 +944,14 @@
 }
 
 Status AudioFlingerServerAdapter::getInputBufferSize(int32_t sampleRate,
-                                                     media::audio::common::AudioFormat format,
-                                                     int32_t channelMask, int64_t* _aidl_return) {
+                                                     const media::AudioFormatDescription& format,
+                                                     const media::AudioChannelLayout& channelMask,
+                                                     int64_t* _aidl_return) {
     uint32_t sampleRateLegacy = VALUE_OR_RETURN_BINDER(convertIntegral<uint32_t>(sampleRate));
     audio_format_t formatLegacy = VALUE_OR_RETURN_BINDER(
-            aidl2legacy_AudioFormat_audio_format_t(format));
+            aidl2legacy_AudioFormatDescription_audio_format_t(format));
     audio_channel_mask_t channelMaskLegacy = VALUE_OR_RETURN_BINDER(
-            aidl2legacy_int32_t_audio_channel_mask_t(channelMask));
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(channelMask, true /*isInput*/));
     size_t size = mDelegate->getInputBufferSize(sampleRateLegacy, formatLegacy, channelMaskLegacy);
     *_aidl_return = VALUE_OR_RETURN_BINDER(convertIntegral<int64_t>(size));
     return Status::ok();
@@ -1189,6 +1200,11 @@
     return Status::fromStatusT(mDelegate->systemReady());
 }
 
+Status AudioFlingerServerAdapter::audioPolicyReady() {
+    mDelegate->audioPolicyReady();
+    return Status::ok();
+}
+
 Status AudioFlingerServerAdapter::frameCountHAL(int32_t ioHandle, int64_t* _aidl_return) {
     audio_io_handle_t ioHandleLegacy = VALUE_OR_RETURN_BINDER(
             aidl2legacy_int32_t_audio_io_handle_t(ioHandle));
diff --git a/media/libaudioclient/PolicyAidlConversion.cpp b/media/libaudioclient/PolicyAidlConversion.cpp
index 25fdb49..676bb37 100644
--- a/media/libaudioclient/PolicyAidlConversion.cpp
+++ b/media/libaudioclient/PolicyAidlConversion.cpp
@@ -232,10 +232,14 @@
                                  std::back_inserter(legacy.mCriteria),
                                  aidl2legacy_AudioMixMatchCriterion));
     legacy.mMixType = VALUE_OR_RETURN(aidl2legacy_AudioMixType_uint32_t(aidl.mixType));
-    legacy.mFormat = VALUE_OR_RETURN(aidl2legacy_AudioConfig_audio_config_t(aidl.format));
+    // See 'convertAudioMixToNative' in 'android_media_AudioSystem.cpp' -- only
+    // an output mask is expected here.
+    legacy.mFormat = VALUE_OR_RETURN(aidl2legacy_AudioConfig_audio_config_t(
+                    aidl.format, false /*isInput*/));
     legacy.mRouteFlags = VALUE_OR_RETURN(
             aidl2legacy_AudioMixRouteFlag_uint32_t_mask(aidl.routeFlags));
-    legacy.mDeviceType = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.device.type));
+    legacy.mDeviceType = VALUE_OR_RETURN(
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.device.type));
     legacy.mDeviceAddress = VALUE_OR_RETURN(aidl2legacy_string_view_String8(aidl.device.address));
     legacy.mCbFlags = VALUE_OR_RETURN(aidl2legacy_AudioMixCallbackFlag_uint32_t_mask(aidl.cbFlags));
     legacy.mAllowPrivilegedMediaPlaybackCapture = aidl.allowPrivilegedMediaPlaybackCapture;
@@ -251,10 +255,14 @@
                     legacy.mCriteria,
                     legacy2aidl_AudioMixMatchCriterion));
     aidl.mixType = VALUE_OR_RETURN(legacy2aidl_uint32_t_AudioMixType(legacy.mMixType));
-    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_config_t_AudioConfig(legacy.mFormat));
+    // See 'convertAudioMixToNative' in 'android_media_AudioSystem.cpp' -- only
+    // an output mask is expected here.
+    aidl.format = VALUE_OR_RETURN(legacy2aidl_audio_config_t_AudioConfig(
+                    legacy.mFormat, false /*isInput*/));
     aidl.routeFlags = VALUE_OR_RETURN(
             legacy2aidl_uint32_t_AudioMixRouteFlag_mask(legacy.mRouteFlags));
-    aidl.device.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.mDeviceType));
+    aidl.device.type = VALUE_OR_RETURN(
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.mDeviceType));
     aidl.device.address = VALUE_OR_RETURN(legacy2aidl_String8_string(legacy.mDeviceAddress));
     aidl.cbFlags = VALUE_OR_RETURN(legacy2aidl_uint32_t_AudioMixCallbackFlag_mask(legacy.mCbFlags));
     aidl.allowPrivilegedMediaPlaybackCapture = legacy.mAllowPrivilegedMediaPlaybackCapture;
diff --git a/media/libaudioclient/TEST_MAPPING b/media/libaudioclient/TEST_MAPPING
new file mode 100644
index 0000000..d8c18c0
--- /dev/null
+++ b/media/libaudioclient/TEST_MAPPING
@@ -0,0 +1,7 @@
+{
+  "presubmit": [
+    {
+       "name": "audio_aidl_conversion_tests"
+    }
+  ]
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioChannelLayout.aidl b/media/libaudioclient/aidl/android/media/AudioChannelLayout.aidl
new file mode 100644
index 0000000..fdc8184
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioChannelLayout.aidl
@@ -0,0 +1,268 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * This structure describes a layout of a multi-channel stream.
+ * There are two possible ways for representing a layout:
+ *
+ * - indexed mask, which tells what channels of an audio frame are used, but
+ *   doesn't label them in any way, thus a correspondence between channels in
+ *   the same position of frames originating from different streams must be
+ *   established externally;
+ *
+ * - layout mask, which gives a label to each channel, thus allowing to
+ *   match channels between streams of different layouts.
+ *
+ * Both representations are agnostic of the direction of audio transfer. Also,
+ * by construction, the number of bits set to '1' in the mask indicates the
+ * number of channels in the audio frame. A channel mask per se only defines the
+ * presence or absence of a channel, not the order. Please see 'INTERLEAVE_*'
+ * constants for the platform convention of order.
+ *
+ * The structure also defines a "voice mask" which is a special case of
+ * layout mask, intended for processing voice audio from telecommunication
+ * use cases.
+ */
+union AudioChannelLayout {
+    /**
+     * This variant is used for representing the "null" ("none") value
+     * for the channel layout. The field value must always be '0'.
+     */
+    int none = 0;
+    /**
+     * This variant is used for indicating an "invalid" layout for use by the
+     * framework only. HAL implementations must not accept or emit
+     * AudioChannelLayout values for this variant. The field value must always
+     * be '0'.
+     */
+    int invalid = 0;
+    /**
+     * This variant is used for representing indexed masks. The value
+     * must be one of the 'INDEX_MASK_*' constants. The 'indexMask' field
+     * must have at least one bit set.
+     */
+    int indexMask;
+    /**
+     * This variant is used for representing layout masks.
+     * It is recommended to use one of 'LAYOUT_*' values. The 'layoutMask' field
+     * must have at least one bit set.
+     */
+    int layoutMask;
+    /**
+     * This variant is used for processing of voice audio input and output.
+     * It is recommended to use one of 'VOICE_*' values. The 'voiceMask' field
+     * must have at least one bit set.
+     */
+    int voiceMask;
+
+    /**
+     * 'INDEX_MASK_' constants define how many channels are used.
+     */
+    const int INDEX_MASK_1 = (1 << 1) - 1;
+    const int INDEX_MASK_2 = (1 << 2) - 1;
+    const int INDEX_MASK_3 = (1 << 3) - 1;
+    const int INDEX_MASK_4 = (1 << 4) - 1;
+    const int INDEX_MASK_5 = (1 << 5) - 1;
+    const int INDEX_MASK_6 = (1 << 6) - 1;
+    const int INDEX_MASK_7 = (1 << 7) - 1;
+    const int INDEX_MASK_8 = (1 << 8) - 1;
+    const int INDEX_MASK_9 = (1 << 9) - 1;
+    const int INDEX_MASK_10 = (1 << 10) - 1;
+    const int INDEX_MASK_11 = (1 << 11) - 1;
+    const int INDEX_MASK_12 = (1 << 12) - 1;
+    const int INDEX_MASK_13 = (1 << 13) - 1;
+    const int INDEX_MASK_14 = (1 << 14) - 1;
+    const int INDEX_MASK_15 = (1 << 15) - 1;
+    const int INDEX_MASK_16 = (1 << 16) - 1;
+    const int INDEX_MASK_17 = (1 << 17) - 1;
+    const int INDEX_MASK_18 = (1 << 18) - 1;
+    const int INDEX_MASK_19 = (1 << 19) - 1;
+    const int INDEX_MASK_20 = (1 << 20) - 1;
+    const int INDEX_MASK_21 = (1 << 21) - 1;
+    const int INDEX_MASK_22 = (1 << 22) - 1;
+    const int INDEX_MASK_23 = (1 << 23) - 1;
+    const int INDEX_MASK_24 = (1 << 24) - 1;
+
+    /**
+     * 'LAYOUT_*' constants define channel layouts recognized by
+     * the audio system. The order of the channels in the frame is assumed
+     * to be from the LSB to MSB for all the bits set to '1'.
+     */
+    const int LAYOUT_MONO = CHANNEL_FRONT_LEFT;
+    const int LAYOUT_STEREO =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT;
+    const int LAYOUT_2POINT1 =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_LOW_FREQUENCY;
+    const int LAYOUT_TRI =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_FRONT_CENTER;
+    const int LAYOUT_TRI_BACK =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_BACK_CENTER;
+    const int LAYOUT_3POINT1 =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT | CHANNEL_FRONT_CENTER |
+            CHANNEL_LOW_FREQUENCY;
+    const int LAYOUT_2POINT0POINT2 =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
+    const int LAYOUT_2POINT1POINT2 =
+            LAYOUT_2POINT0POINT2 | CHANNEL_LOW_FREQUENCY;
+    const int LAYOUT_3POINT0POINT2 =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_FRONT_CENTER |
+            CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
+    const int LAYOUT_3POINT1POINT2 =
+            LAYOUT_3POINT0POINT2 | CHANNEL_LOW_FREQUENCY;
+    const int LAYOUT_QUAD =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT;
+    const int LAYOUT_QUAD_SIDE =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT;
+    const int LAYOUT_SURROUND =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER;
+    const int LAYOUT_PENTA = LAYOUT_QUAD | CHANNEL_FRONT_CENTER;
+    const int LAYOUT_5POINT1 =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
+            CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT;
+    const int LAYOUT_5POINT1_SIDE =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
+            CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT;
+    const int LAYOUT_5POINT1POINT2 = LAYOUT_5POINT1 |
+            CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
+    const int LAYOUT_5POINT1POINT4 = LAYOUT_5POINT1 |
+            CHANNEL_TOP_FRONT_LEFT | CHANNEL_TOP_FRONT_RIGHT |
+            CHANNEL_TOP_BACK_LEFT | CHANNEL_TOP_BACK_RIGHT;
+    const int LAYOUT_6POINT1 =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_FRONT_CENTER | CHANNEL_LOW_FREQUENCY |
+            CHANNEL_BACK_LEFT | CHANNEL_BACK_RIGHT | CHANNEL_BACK_CENTER;
+    const int LAYOUT_7POINT1 = LAYOUT_5POINT1 |
+            CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT;
+    const int LAYOUT_7POINT1POINT2 = LAYOUT_7POINT1 |
+            CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
+    const int LAYOUT_7POINT1POINT4 = LAYOUT_7POINT1 |
+            CHANNEL_TOP_FRONT_LEFT | CHANNEL_TOP_FRONT_RIGHT |
+            CHANNEL_TOP_BACK_LEFT | CHANNEL_TOP_BACK_RIGHT;
+    const int LAYOUT_9POINT1POINT4 = LAYOUT_7POINT1POINT4 |
+            CHANNEL_FRONT_WIDE_LEFT | CHANNEL_FRONT_WIDE_RIGHT;
+    const int LAYOUT_9POINT1POINT6 = LAYOUT_9POINT1POINT4 |
+            CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT;
+    const int LAYOUT_13POINT_360RA =
+            CHANNEL_FRONT_LEFT | CHANNEL_FRONT_RIGHT |
+            CHANNEL_FRONT_CENTER |
+            CHANNEL_SIDE_LEFT | CHANNEL_SIDE_RIGHT |
+            CHANNEL_TOP_FRONT_LEFT | CHANNEL_TOP_FRONT_RIGHT |
+            CHANNEL_TOP_FRONT_CENTER |
+            CHANNEL_TOP_BACK_LEFT | CHANNEL_TOP_BACK_RIGHT |
+            CHANNEL_BOTTOM_FRONT_LEFT | CHANNEL_BOTTOM_FRONT_RIGHT |
+            CHANNEL_BOTTOM_FRONT_CENTER;
+    const int LAYOUT_22POINT2 = LAYOUT_7POINT1POINT4 |
+            CHANNEL_FRONT_LEFT_OF_CENTER | CHANNEL_FRONT_RIGHT_OF_CENTER |
+            CHANNEL_BACK_CENTER | CHANNEL_TOP_CENTER |
+            CHANNEL_TOP_FRONT_CENTER | CHANNEL_TOP_BACK_CENTER |
+            CHANNEL_TOP_SIDE_LEFT | CHANNEL_TOP_SIDE_RIGHT |
+            CHANNEL_BOTTOM_FRONT_LEFT | CHANNEL_BOTTOM_FRONT_RIGHT |
+            CHANNEL_BOTTOM_FRONT_CENTER |
+            CHANNEL_LOW_FREQUENCY_2;
+    const int LAYOUT_MONO_HAPTIC_A =
+            LAYOUT_MONO | CHANNEL_HAPTIC_A;
+    const int LAYOUT_STEREO_HAPTIC_A =
+            LAYOUT_STEREO | CHANNEL_HAPTIC_A;
+    const int LAYOUT_HAPTIC_AB =
+            CHANNEL_HAPTIC_A | CHANNEL_HAPTIC_B;
+    const int LAYOUT_MONO_HAPTIC_AB =
+            LAYOUT_MONO | LAYOUT_HAPTIC_AB;
+    const int LAYOUT_STEREO_HAPTIC_AB =
+            LAYOUT_STEREO | LAYOUT_HAPTIC_AB;
+    const int LAYOUT_FRONT_BACK =
+            CHANNEL_FRONT_CENTER | CHANNEL_BACK_CENTER;
+
+    /**
+     * Expresses the convention when stereo audio samples are stored interleaved
+     * in an array.  This should improve readability by allowing code to use
+     * symbolic indices instead of hard-coded [0] and [1].
+     *
+     * For multi-channel beyond stereo, the platform convention is that channels
+     * are interleaved in order from least significant channel mask bit to most
+     * significant channel mask bit, with unused bits skipped. Any exceptions
+     * to this convention will be noted at the appropriate API.
+     */
+    const int INTERLEAVE_LEFT = 0;
+    const int INTERLEAVE_RIGHT = 1;
+
+    /**
+     * 'CHANNEL_*' constants are used to build 'LAYOUT_*' masks. Each constant
+     * must have exactly one bit set. The values do not match
+     * 'android.media.AudioFormat.CHANNEL_OUT_*' constants from the SDK
+     * for better efficiency in masks processing.
+     */
+    const int CHANNEL_FRONT_LEFT = 1 << 0;
+    const int CHANNEL_FRONT_RIGHT = 1 << 1;
+    const int CHANNEL_FRONT_CENTER = 1 << 2;
+    const int CHANNEL_LOW_FREQUENCY = 1 << 3;
+    const int CHANNEL_BACK_LEFT = 1 << 4;
+    const int CHANNEL_BACK_RIGHT = 1 << 5;
+    const int CHANNEL_FRONT_LEFT_OF_CENTER = 1 << 6;
+    const int CHANNEL_FRONT_RIGHT_OF_CENTER = 1 << 7;
+    const int CHANNEL_BACK_CENTER = 1 << 8;
+    const int CHANNEL_SIDE_LEFT = 1 << 9;
+    const int CHANNEL_SIDE_RIGHT = 1 << 10;
+    const int CHANNEL_TOP_CENTER = 1 << 11;
+    const int CHANNEL_TOP_FRONT_LEFT = 1 << 12;
+    const int CHANNEL_TOP_FRONT_CENTER = 1 << 13;
+    const int CHANNEL_TOP_FRONT_RIGHT = 1 << 14;
+    const int CHANNEL_TOP_BACK_LEFT = 1 << 15;
+    const int CHANNEL_TOP_BACK_CENTER = 1 << 16;
+    const int CHANNEL_TOP_BACK_RIGHT = 1 << 17;
+    const int CHANNEL_TOP_SIDE_LEFT = 1 << 18;
+    const int CHANNEL_TOP_SIDE_RIGHT = 1 << 19;
+    const int CHANNEL_BOTTOM_FRONT_LEFT = 1 << 20;
+    const int CHANNEL_BOTTOM_FRONT_CENTER = 1 << 21;
+    const int CHANNEL_BOTTOM_FRONT_RIGHT = 1 << 22;
+    const int CHANNEL_LOW_FREQUENCY_2 = 1 << 23;
+    const int CHANNEL_FRONT_WIDE_LEFT = 1 << 24;
+    const int CHANNEL_FRONT_WIDE_RIGHT = 1 << 25;
+    /**
+     * Haptic channels are not part of multichannel standards, however they
+     * enhance user experience when playing so they are packed together with the
+     * channels of the program. To avoid collision with positional channels the
+     * values for haptic channels start at the MSB of an integer (after the sign
+     * bit) and move down to LSB.
+     */
+    const int CHANNEL_HAPTIC_B = 1 << 29;
+    const int CHANNEL_HAPTIC_A = 1 << 30;
+
+    /**
+     * 'VOICE_*' constants define layouts for voice audio. The order of the
+     * channels in the frame is assumed to be from the LSB to MSB for all the
+     * bits set to '1'.
+     */
+    const int VOICE_UPLINK_MONO = CHANNEL_VOICE_UPLINK;
+    const int VOICE_DNLINK_MONO = CHANNEL_VOICE_DNLINK;
+    const int VOICE_CALL_MONO = CHANNEL_VOICE_UPLINK | CHANNEL_VOICE_DNLINK;
+
+    /**
+     * 'CHANNEL_VOICE_*' constants are used to build 'VOICE_*' masks. Each
+     * constant must have exactly one bit set. Use the same values as
+     * 'android.media.AudioFormat.CHANNEL_IN_VOICE_*' constants from the SDK.
+     */
+    const int CHANNEL_VOICE_UPLINK = 0x4000;
+    const int CHANNEL_VOICE_DNLINK = 0x8000;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioConfig.aidl b/media/libaudioclient/aidl/android/media/AudioConfig.aidl
index 8dc97d3..6996d42 100644
--- a/media/libaudioclient/aidl/android/media/AudioConfig.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioConfig.aidl
@@ -16,20 +16,17 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
+import android.media.AudioFormatDescription;
 import android.media.AudioOffloadInfo;
-import android.media.audio.common.AudioFormat;
 
 /**
  * {@hide}
  */
 parcelable AudioConfig {
     int sampleRate;
-    /**
-     * Interpreted as audio_channel_mask_t.
-     * TODO(ytai): Create a designated type.
-     */
-    int channelMask;
-    AudioFormat format;
+    AudioChannelLayout channelMask;
+    AudioFormatDescription format;
     AudioOffloadInfo offloadInfo;
     long frameCount;
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl b/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl
index 8353c0d..e84161b 100644
--- a/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioConfigBase.aidl
@@ -16,14 +16,14 @@
 
 package android.media;
 
-import android.media.audio.common.AudioFormat;
+import android.media.AudioChannelLayout;
+import android.media.AudioFormatDescription;
 
 /**
  * {@hide}
  */
 parcelable AudioConfigBase {
     int sampleRate;
-    /** Interpreted as audio_channel_mask_t. */
-    int channelMask;
-    AudioFormat format;
+    AudioChannelLayout channelMask;
+    AudioFormatDescription format;
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioDevice.aidl b/media/libaudioclient/aidl/android/media/AudioDevice.aidl
index b200697..a815874 100644
--- a/media/libaudioclient/aidl/android/media/AudioDevice.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioDevice.aidl
@@ -16,11 +16,12 @@
 
 package android.media;
 
+import android.media.AudioDeviceDescription;
+
 /**
  * {@hide}
  */
 parcelable AudioDevice {
-    /** Interpreted as audio_devices_t. */
-    int type;
+    AudioDeviceDescription type;
     @utf8InCpp String address;
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioDeviceDescription.aidl b/media/libaudioclient/aidl/android/media/AudioDeviceDescription.aidl
new file mode 100644
index 0000000..f7548b9
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioDeviceDescription.aidl
@@ -0,0 +1,101 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.AudioDeviceType;
+
+parcelable AudioDeviceDescription {
+    /**
+     * Type and directionality of the device. For bidirectional audio devices
+     * two descriptions need to be created, having the same value for
+     * the 'connection' field.
+     *
+     * See 'AudioDeviceType' for the list of supported values.
+     */
+    AudioDeviceType type = AudioDeviceType.NONE;
+    /**
+     * Specifies the type of the connection of the device to the audio system.
+     * Usually it's some kind of a communication protocol, e.g. Bluetooth SCO or
+     * USB. There is a list of connection types recognized by the framework,
+     * defined using 'CONNECTION_' constants. Vendors can add their own
+     * connection types with "vx.<vendor>." prefix.
+     *
+     * When the 'connection' field is left empty and 'type != NONE | DEFAULT',
+     * it is assumed that the device is permanently attached to the audio
+     * system, e.g. a built-in speaker or microphone.
+     *
+     * The 'connection' field must be left empty if 'type' is 'NONE' or
+     * '{IN|OUT}_DEFAULT'.
+     */
+    @utf8InCpp String connection;
+    /**
+     * Analog connection, for example, via 3.5 mm analog jack.
+     */
+    const @utf8InCpp String CONNECTION_ANALOG = "analog";
+    /**
+     * Low-End (Analog) Desk Dock.
+     */
+    const @utf8InCpp String CONNECTION_ANALOG_DOCK = "analog-dock";
+    /**
+     * Bluetooth A2DP connection.
+     */
+    const @utf8InCpp String CONNECTION_BT_A2DP = "bt-a2dp";
+    /**
+     * Bluetooth Low Energy (LE) connection.
+     */
+    const @utf8InCpp String CONNECTION_BT_LE = "bt-le";
+    /**
+     * Bluetooth SCO connection.
+     */
+    const @utf8InCpp String CONNECTION_BT_SCO = "bt-sco";
+    /**
+     * Bus connection. Mostly used in automotive scenarios.
+     */
+    const @utf8InCpp String CONNECTION_BUS = "bus";
+    /**
+     * High-End (Digital) Desk Dock.
+     */
+    const @utf8InCpp String CONNECTION_DIGITAL_DOCK = "digital-dock";
+    /**
+     * HDMI connection.
+     */
+    const @utf8InCpp String CONNECTION_HDMI = "hdmi";
+    /**
+     * HDMI ARC connection.
+     */
+    const @utf8InCpp String CONNECTION_HDMI_ARC = "hdmi-arc";
+    /**
+     * HDMI eARC connection.
+     */
+    const @utf8InCpp String CONNECTION_HDMI_EARC = "hdmi-earc";
+    /**
+     * IP v4 connection.
+     */
+    const @utf8InCpp String CONNECTION_IP_V4 = "ip-v4";
+    /**
+     * SPDIF connection.
+     */
+    const @utf8InCpp String CONNECTION_SPDIF = "spdif";
+    /**
+     * A wireless connection when the actual protocol is unspecified.
+     */
+    const @utf8InCpp String CONNECTION_WIRELESS = "wireless";
+    /**
+     * USB connection.
+     */
+    const @utf8InCpp String CONNECTION_USB = "usb";
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioDeviceType.aidl b/media/libaudioclient/aidl/android/media/AudioDeviceType.aidl
new file mode 100644
index 0000000..4da9fd6
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioDeviceType.aidl
@@ -0,0 +1,158 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * The type of the audio device. Only used as part of 'AudioDeviceDescription'
+ * structure.
+ *
+ * Types are divided into "input" and "output" categories. Audio devices that
+ * have both audio input and output, for example, headsets, are represented by a
+ * pair of input and output device types.
+ *
+ * The 'AudioDeviceType' intentionally binds together directionality and 'kind'
+ * of the device to avoid making them fully orthogonal. This is because not all
+ * types of devices are bidirectional, for example, speakers can only be used
+ * for output and microphones can only be used for input (at least, in the
+ * context of the audio framework).
+ */
+@Backing(type="int")
+enum AudioDeviceType {
+    /**
+     * "None" type is a "null" value. All fields of 'AudioDeviceDescription'
+     * must have default / empty / null values.
+     */
+    NONE = 0,
+    /**
+     * The "default" device is used when the client does not have any
+     * preference for a particular device.
+     */
+    IN_DEFAULT = 1,
+    /**
+     * A device implementing Android Open Accessory protocol.
+     */
+    IN_ACCESSORY = 2,
+    /**
+     * Input from a DSP front-end proxy device.
+     */
+    IN_AFE_PROXY = 3,
+    /**
+     * Used when only the connection protocol is known, e.g. a "HDMI Device."
+     */
+    IN_DEVICE = 4,
+    /**
+     * A device providing reference input for echo canceller.
+     */
+    IN_ECHO_REFERENCE = 5,
+    /**
+     * FM Tuner input.
+     */
+    IN_FM_TUNER = 6,
+    /**
+     * A microphone of a headset.
+     */
+    IN_HEADSET = 7,
+    /**
+     * Loopback input.
+     */
+    IN_LOOPBACK = 8,
+    /**
+     * The main microphone (the frontal mic on mobile devices).
+     */
+    IN_MICROPHONE = 9,
+    /**
+     * The secondary microphone (the back mic on mobile devices).
+     */
+    IN_MICROPHONE_BACK = 10,
+    /**
+     * Input from a submix of other streams.
+     */
+    IN_SUBMIX = 11,
+    /**
+     * Audio received via the telephone line.
+     */
+    IN_TELEPHONY_RX = 12,
+    /**
+     * TV Tuner audio input.
+     */
+    IN_TV_TUNER = 13,
+    /**
+     * The "default" device is used when the client does not have any
+     * preference for a particular device.
+     */
+    OUT_DEFAULT = 129,
+    /**
+     * A device implementing Android Open Accessory protocol.
+     */
+    OUT_ACCESSORY = 130,
+    /**
+     * Output from a DSP front-end proxy device.
+     */
+    OUT_AFE_PROXY = 131,
+    /**
+     * Car audio system.
+     */
+    OUT_CARKIT = 132,
+    /**
+     * Used when only the connection protocol is known, e.g. a "HDMI Device."
+     */
+    OUT_DEVICE = 133,
+    /**
+     * The echo canceller device.
+     */
+    OUT_ECHO_CANCELLER = 134,
+    /**
+     * The FM Tuner device.
+     */
+    OUT_FM = 135,
+    /**
+     * Headphones.
+     */
+    OUT_HEADPHONE = 136,
+    /**
+     * Headphones of a headset.
+     */
+    OUT_HEADSET = 137,
+    /**
+     * Hearing aid.
+     */
+    OUT_HEARING_AID = 138,
+    /**
+     * Secondary line level output.
+     */
+    OUT_LINE_AUX = 139,
+    /**
+     * The main speaker.
+     */
+    OUT_SPEAKER = 140,
+    /**
+     * The speaker of a mobile device in the case when it is close to the ear.
+     */
+    OUT_SPEAKER_EARPIECE = 141,
+    /**
+     * The main speaker with overload / overheating protection.
+     */
+    OUT_SPEAKER_SAFE = 142,
+    /**
+     * Output into a submix.
+     */
+    OUT_SUBMIX = 143,
+    /**
+     * Output into a telephone line.
+     */
+    OUT_TELEPHONY_TX = 144,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioFormatDescription.aidl b/media/libaudioclient/aidl/android/media/AudioFormatDescription.aidl
new file mode 100644
index 0000000..a656348
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioFormatDescription.aidl
@@ -0,0 +1,84 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+import android.media.AudioFormatType;
+import android.media.PcmType;
+
+/**
+ * An extensible type for specifying audio formats. All formats are largely
+ * divided into two classes: PCM and non-PCM (bitstreams). Bitstreams can
+ * be encapsulated into PCM streams.
+ *
+ * The type defined in a way to make each format uniquely identifiable, so
+ * that if the framework and the HAL construct a value for the same type
+ * (e.g. PCM 16 bit), they will produce identical parcelables which will have
+ * identical hashes. This makes possible deduplicating type descriptions
+ * by the framework when they are received from different HAL modules without
+ * relying on having some centralized registry of enumeration values.
+ *
+ * {@hide}
+ */
+parcelable AudioFormatDescription {
+    /**
+     * The type of the audio format. See the 'AudioFormatType' for the
+     * list of supported values.
+     */
+    AudioFormatType type = AudioFormatType.DEFAULT;
+    /**
+     * The type of the PCM stream or the transport stream for PCM
+     * encapsulations.  See 'PcmType' for the list of supported values.
+     */
+    PcmType pcm = PcmType.DEFAULT;
+    /**
+     * Optional encoding specification. Must be left empty when:
+     *
+     *  - 'type == DEFAULT && pcm == DEFAULT' -- that means "default" type;
+     *  - 'type == PCM' -- that means a regular PCM stream (not an encapsulation
+     *    of an encoded bitstream).
+     *
+     * For PCM encapsulations of encoded bitstreams (e.g. an encapsulation
+     * according to IEC-61937 standard), the value of the 'pcm' field must
+     * be set accordingly, as an example, PCM_INT_16_BIT must be used for
+     * IEC-61937. Note that 'type == NON_PCM' in this case.
+     *
+     * Encoding names mostly follow IANA standards for media types (MIME), and
+     * frameworks/av/media/libstagefright/foundation/MediaDefs.cpp with the
+     * latter having priority.  Since there are still many audio types not found
+     * in any of these lists, the following rules are applied:
+     *
+     *   - If there is a direct MIME type for the encoding, the MIME type name
+     *     is used as is, e.g. "audio/eac3" for the EAC-3 format.
+     *   - If the encoding is a "subformat" of a MIME-registered format,
+     *     the latter is augmented with a suffix, e.g. "audio/eac3-joc" for the
+     *     JOC extension of EAC-3.
+     *   - If it's a proprietary format, a "vnd." prefix is added, similar to
+     *     IANA rules, e.g. "audio/vnd.dolby.truehd".
+     *   - Otherwise, "x-" prefix is added, e.g. "audio/x-iec61937".
+     *   - All MIME types not found in the IANA formats list have an associated
+     *     comment.
+     *
+     * For PCM encapsulations with a known bitstream format, the latter
+     * is added to the encapsulation encoding as a suffix, after a "+" char.
+     * For example, an IEC61937 encapsulation of AC3 has the following
+     * representation:
+     *   type = NON_PCM,
+     *   pcm = PcmType.INT_16_BIT,
+     *   encoding = "audio/x-iec61937+audio/ac3"
+     */
+    @utf8InCpp String encoding;
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioFormatType.aidl b/media/libaudioclient/aidl/android/media/AudioFormatType.aidl
new file mode 100644
index 0000000..31ed2be
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/AudioFormatType.aidl
@@ -0,0 +1,48 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * The type of the audio format. Only used as part of 'AudioFormatDescription'
+ * structure.
+ */
+@Backing(type="byte")
+enum AudioFormatType {
+    /**
+     * "Default" type is used when the client does not care about the actual
+     * format. All fields of 'AudioFormatDescription' must have default / empty
+     * / null values.
+     */
+    DEFAULT = 0,
+    /**
+     * When the 'encoding' field of 'AudioFormatDescription' is not empty, it
+     * specifies the codec used for bitstream (non-PCM) data. It is also used
+     * in the case when the bitstream data is encapsulated into a PCM stream,
+     * see the documentation for 'AudioFormatDescription'.
+     */
+    NON_PCM = DEFAULT,
+    /**
+     * PCM type. The 'pcm' field of 'AudioFormatDescription' is used to specify
+     * the actual sample size and representation.
+     */
+    PCM = 1,
+    /**
+     * Value reserved for system use only. HALs must never return this value to
+     * the system or accept it from the system.
+     */
+    SYS_RESERVED_INVALID = -1,
+}
diff --git a/media/libaudioclient/aidl/android/media/AudioGain.aidl b/media/libaudioclient/aidl/android/media/AudioGain.aidl
index 048b295..4cfa96e 100644
--- a/media/libaudioclient/aidl/android/media/AudioGain.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioGain.aidl
@@ -16,17 +16,18 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
+
 /**
  * {@hide}
  */
 parcelable AudioGain {
     int index;
-    boolean useInChannelMask;
+    boolean isInput;
     boolean useForVolume;
     /** Bitmask, indexed by AudioGainMode. */
     int mode;
-    /** Interpreted as audio_channel_mask_t. */
-    int channelMask;
+    AudioChannelLayout channelMask;
     int minValue;
     int maxValue;
     int defaultValue;
diff --git a/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl b/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
index b93c2dc..afa3aca 100644
--- a/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioGainConfig.aidl
@@ -16,6 +16,8 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
+
 /**
  * {@hide}
  */
@@ -28,9 +30,8 @@
 
     /**
      * Channels which gain value follows. N/A in joint mode.
-     * Interpreted as audio_channel_mask_t.
      */
-    int channelMask;
+    AudioChannelLayout channelMask;
 
     /**
      * Gain values in millibels.
diff --git a/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl b/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
index 876ef9b..efdf99b 100644
--- a/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioIoDescriptor.aidl
@@ -16,8 +16,9 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
+import android.media.AudioFormatDescription;
 import android.media.AudioPatch;
-import android.media.audio.common.AudioFormat;
 
 /**
  * {@hide}
@@ -26,10 +27,10 @@
     /** Interpreted as audio_io_handle_t. */
     int ioHandle;
     AudioPatch patch;
+    boolean isInput;
     int samplingRate;
-    AudioFormat format;
-    /** Interpreted as audio_channel_mask_t. */
-    int channelMask;
+    AudioFormatDescription format;
+    AudioChannelLayout channelMask;
     long frameCount;
     long frameCountHAL;
     /** Only valid for output. */
diff --git a/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl b/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl
index c86b3f0..f326305 100644
--- a/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioOffloadInfo.aidl
@@ -20,7 +20,6 @@
 import android.media.AudioEncapsulationMode;
 import android.media.AudioStreamType;
 import android.media.AudioUsage;
-import android.media.audio.common.AudioFormat;
 
 /**
  * {@hide}
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
index 2dd30a4..be32a69 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfig.aidl
@@ -16,13 +16,14 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
 import android.media.AudioGainConfig;
 import android.media.AudioIoFlags;
 import android.media.AudioPortConfigExt;
 import android.media.AudioPortConfigType;
 import android.media.AudioPortRole;
 import android.media.AudioPortType;
-import android.media.audio.common.AudioFormat;
+import android.media.AudioFormatDescription;
 
 /**
  * {@hide}
@@ -43,14 +44,12 @@
     int sampleRate;
     /**
      * Channel mask, if applicable.
-     * Interpreted as audio_channel_mask_t.
-     * TODO: bitmask?
      */
-    int channelMask;
+    AudioChannelLayout channelMask;
     /**
      * Format, if applicable.
      */
-    AudioFormat format;
+    AudioFormatDescription format;
     /** Gain to apply, if applicable. */
     AudioGainConfig gain;
     /** Framework only: HW_AV_SYNC, DIRECT, ... */
diff --git a/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl b/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
index a99aa9b..31e5330 100644
--- a/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioPortConfigDeviceExt.aidl
@@ -16,6 +16,8 @@
 
 package android.media;
 
+import android.media.AudioDeviceDescription;
+
 /**
  * {@hide}
  */
@@ -26,11 +28,9 @@
      */
     int hwModule;
     /**
-     * Device type (e.g AUDIO_DEVICE_OUT_SPEAKER).
-     * Interpreted as audio_devices_t.
-     * TODO: Convert to a standalone AIDL representation.
+     * Device type.
      */
-    int type;
+    AudioDeviceDescription type;
     /** Device address. "" if N/A. */
     @utf8InCpp String address;
 }
diff --git a/media/libaudioclient/aidl/android/media/AudioProfile.aidl b/media/libaudioclient/aidl/android/media/AudioProfile.aidl
index afb288f..9fb8d49 100644
--- a/media/libaudioclient/aidl/android/media/AudioProfile.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioProfile.aidl
@@ -16,8 +16,9 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
 import android.media.AudioEncapsulationType;
-import android.media.audio.common.AudioFormat;
+import android.media.AudioFormatDescription;
 
 /**
  * {@hide}
@@ -25,9 +26,8 @@
 parcelable AudioProfile {
     @utf8InCpp String name;
     /** The format for an audio profile should only be set when initialized. */
-    AudioFormat format;
-    /** Interpreted as audio_channel_mask_t. */
-    int[] channelMasks;
+    AudioFormatDescription format;
+    AudioChannelLayout[] channelMasks;
     int[] samplingRates;
     boolean isDynamicFormat;
     boolean isDynamicChannels;
diff --git a/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl b/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
index f88fc3c..8538d8a 100644
--- a/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
+++ b/media/libaudioclient/aidl/android/media/AudioVibratorInfo.aidl
@@ -24,4 +24,5 @@
     int id;
     float resonantFrequency;
     float qFactor;
+    float maxAmplitude;
 }
diff --git a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
index d2cae6d..16f70c1 100644
--- a/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioFlingerService.aidl
@@ -16,6 +16,7 @@
 
 package android.media;
 
+import android.media.AudioChannelLayout;
 import android.media.AudioMode;
 import android.media.AudioPatch;
 import android.media.AudioPort;
@@ -41,7 +42,7 @@
 import android.media.MicrophoneInfoData;
 import android.media.RenderPosition;
 import android.media.TrackSecondaryOutputInfo;
-import android.media.audio.common.AudioFormat;
+import android.media.AudioFormatDescription;
 
 /**
  * {@hide}
@@ -62,7 +63,7 @@
      */
     int sampleRate(int /* audio_io_handle_t */ ioHandle);
 
-    AudioFormat format(int /* audio_io_handle_t */ output);
+    AudioFormatDescription format(int /* audio_io_handle_t */ output);
 
     long frameCount(int /* audio_io_handle_t */ ioHandle);
 
@@ -115,8 +116,8 @@
     // Retrieve the audio recording buffer size in bytes.
     // FIXME This API assumes a route, and so should be deprecated.
     long getInputBufferSize(int sampleRate,
-                            AudioFormat format,
-                            int /* audio_channel_mask_t */ channelMask);
+                            in AudioFormatDescription format,
+                            in AudioChannelLayout channelMask);
 
     OpenOutputResponse openOutput(in OpenOutputRequest request);
     int /* audio_io_handle_t */ openDuplicateOutput(int /* audio_io_handle_t */ output1,
@@ -197,6 +198,9 @@
     /* Indicate JAVA services are ready (scheduling, power management ...) */
     oneway void systemReady();
 
+    /* Indicate audio policy service is ready */
+    oneway void audioPolicyReady();
+
     // Returns the number of frames per audio HAL buffer.
     long frameCountHAL(int /* audio_io_handle_t */ ioHandle);
 
diff --git a/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl b/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
index 65bcd82..d54bd5e 100644
--- a/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
+++ b/media/libaudioclient/aidl/android/media/IAudioPolicyService.aidl
@@ -16,15 +16,15 @@
 
 package android.media;
 
+import android.media.AudioFormatDescription;
 import android.content.AttributionSourceState;
 
-import android.media.audio.common.AudioFormat;
-
 import android.media.AudioAttributesEx;
 import android.media.AudioAttributesInternal;
 import android.media.AudioConfig;
 import android.media.AudioConfigBase;
 import android.media.AudioDevice;
+import android.media.AudioDeviceDescription;
 import android.media.AudioMix;
 import android.media.AudioMode;
 import android.media.AudioOffloadInfo;
@@ -63,13 +63,13 @@
     void setDeviceConnectionState(in AudioDevice device,
                                   in AudioPolicyDeviceState state,
                                   @utf8InCpp String deviceName,
-                                  in AudioFormat encodedFormat);
+                                  in AudioFormatDescription encodedFormat);
 
     AudioPolicyDeviceState getDeviceConnectionState(in AudioDevice device);
 
     void handleDeviceConfigChange(in AudioDevice device,
                                   @utf8InCpp String deviceName,
-                                  in AudioFormat encodedFormat);
+                                  in AudioFormatDescription encodedFormat);
 
     void setPhoneState(AudioMode state, int /* uid_t */ uid);
 
@@ -114,18 +114,18 @@
                           int indexMax);
 
     void setStreamVolumeIndex(AudioStreamType stream,
-                              int /* audio_devices_t */ device,
+                              in AudioDeviceDescription device,
                               int index);
 
     int getStreamVolumeIndex(AudioStreamType stream,
-                             int /* audio_devices_t */ device);
+                             in AudioDeviceDescription device);
 
     void setVolumeIndexForAttributes(in AudioAttributesInternal attr,
-                                     int /* audio_devices_t */ device,
+                                     in AudioDeviceDescription device,
                                      int index);
 
     int getVolumeIndexForAttributes(in AudioAttributesInternal attr,
-                                    int /* audio_devices_t */ device);
+                                    in AudioDeviceDescription device);
 
     int getMaxVolumeIndexForAttributes(in AudioAttributesInternal attr);
 
@@ -133,7 +133,7 @@
 
     int /* product_strategy_t */ getStrategyForStream(AudioStreamType stream);
 
-    int /* bitmask of audio_devices_t */ getDevicesForStream(AudioStreamType stream);
+    AudioDeviceDescription[] getDevicesForStream(AudioStreamType stream);
 
     AudioDevice[] getDevicesForAttributes(in AudioAttributesEx attr);
 
@@ -268,7 +268,7 @@
 
     boolean getMasterMono();
 
-    float getStreamVolumeDB(AudioStreamType stream, int index, int /* audio_devices_t */ device);
+    float getStreamVolumeDB(AudioStreamType stream, int index, in AudioDeviceDescription device);
 
     /**
      * Populates supported surround formats and their enabled state in formats and formatsEnabled.
@@ -279,7 +279,7 @@
      * number of elements without actually retrieving them.
      */
     void getSurroundFormats(inout Int count,
-                            out AudioFormat[] formats,
+                            out AudioFormatDescription[] formats,
                             out boolean[] formatsEnabled);
 
     /**
@@ -291,11 +291,11 @@
      * number of elements without actually retrieving them.
      */
     void getReportedSurroundFormats(inout Int count,
-                                    out AudioFormat[] formats);
+                                    out AudioFormatDescription[] formats);
 
-    AudioFormat[] getHwOffloadEncodingFormatsSupportedForA2DP();
+    AudioFormatDescription[] getHwOffloadEncodingFormatsSupportedForA2DP();
 
-    void setSurroundFormatEnabled(AudioFormat audioFormat, boolean enabled);
+    void setSurroundFormatEnabled(in AudioFormatDescription audioFormat, boolean enabled);
 
     void setAssistantUid(int /* uid_t */ uid);
 
diff --git a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
index 06b12e9..1541948 100644
--- a/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
+++ b/media/libaudioclient/aidl/android/media/OpenOutputRequest.aidl
@@ -17,6 +17,7 @@
 package android.media;
 
 import android.media.AudioConfig;
+import android.media.AudioConfigBase;
 import android.media.AudioPort;
 
 /**
@@ -25,7 +26,8 @@
 parcelable OpenOutputRequest {
     /** Interpreted as audio_module_handle_t. */
     int module;
-    AudioConfig config;
+    AudioConfig halConfig;
+    AudioConfigBase mixerConfig;
     /** Type must be DEVICE. */
     AudioPort device;
     /** Bitmask, indexed by AudioOutputFlag. */
diff --git a/media/libaudioclient/aidl/android/media/PcmType.aidl b/media/libaudioclient/aidl/android/media/PcmType.aidl
new file mode 100644
index 0000000..c9e327c
--- /dev/null
+++ b/media/libaudioclient/aidl/android/media/PcmType.aidl
@@ -0,0 +1,53 @@
+/*
+ * Copyright (C) 2021 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ *      http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+package android.media;
+
+/**
+ * The type of the encoding used for representing PCM samples. Only used as
+ * part of 'AudioFormatDescription' structure.
+ */
+@Backing(type="byte")
+enum PcmType {
+    /**
+     * "Default" value used when the type 'AudioFormatDescription' is "default".
+     */
+    DEFAULT = 0,
+    /**
+     * Unsigned 8-bit integer.
+     */
+    UINT_8_BIT = DEFAULT,
+    /**
+     * Signed 16-bit integer.
+     */
+    INT_16_BIT = 1,
+    /**
+     * Signed 32-bit integer.
+     */
+    INT_32_BIT = 2,
+    /**
+     * Q8.24 fixed point format.
+     */
+    FIXED_Q_8_24 = 3,
+    /**
+     * IEEE 754 32-bit floating point format.
+     */
+    FLOAT_32_BIT = 4,
+    /**
+     * Signed 24-bit integer.
+     */
+    INT_24_BIT = 5,
+}
diff --git a/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl b/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl
index a829e59..b9708b2 100644
--- a/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl
+++ b/media/libaudioclient/aidl/android/media/SoundTriggerSession.aidl
@@ -16,6 +16,8 @@
 
 package android.media;
 
+import android.media.AudioDeviceDescription;
+
 /**
  * {@hide}
  */
@@ -24,6 +26,6 @@
     int session;
     /** Interpreted as audio_io_handle_t. */
     int ioHandle;
-    /** Interpreted as audio_devices_t. */
-    int device;
+    /** Device type. */
+    AudioDeviceDescription device;
 }
diff --git a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
index d03c6fa..bdd72dd 100644
--- a/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
+++ b/media/libaudioclient/fuzzer/audioflinger_fuzzer.cpp
@@ -597,7 +597,8 @@
     media::OpenInputRequest request{};
     request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
     request.input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
-    request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+    request.config = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_t_AudioConfig(config, true /*isInput*/));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
     request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
     request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
@@ -648,11 +649,16 @@
     sp<DeviceDescriptorBase> device = new DeviceDescriptorBase(getValue(&mFdp, kDevices));
     audio_output_flags_t flags = getValue(&mFdp, kOutputFlags);
 
+    audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+
     media::OpenOutputRequest request{};
     media::OpenOutputResponse response{};
 
     request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
-    request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+    request.halConfig = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_t_AudioConfig(config, false /*isInput*/));
+    request.mixerConfig = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_base_t_AudioConfigBase(mixerConfig, false /*isInput*/));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
     request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
 
diff --git a/media/libaudioclient/include/media/AidlConversion.h b/media/libaudioclient/include/media/AidlConversion.h
index 4ec69c7..2cf127c 100644
--- a/media/libaudioclient/include/media/AidlConversion.h
+++ b/media/libaudioclient/include/media/AidlConversion.h
@@ -22,14 +22,17 @@
 #include <system/audio.h>
 
 #include <android/media/AudioAttributesInternal.h>
+#include <android/media/AudioChannelLayout.h>
 #include <android/media/AudioClient.h>
 #include <android/media/AudioConfig.h>
 #include <android/media/AudioConfigBase.h>
+#include <android/media/AudioDeviceDescription.h>
 #include <android/media/AudioDualMonoMode.h>
 #include <android/media/AudioEncapsulationMode.h>
 #include <android/media/AudioEncapsulationMetadataType.h>
 #include <android/media/AudioEncapsulationType.h>
 #include <android/media/AudioFlag.h>
+#include <android/media/AudioFormatDescription.h>
 #include <android/media/AudioGain.h>
 #include <android/media/AudioGainMode.h>
 #include <android/media/AudioInputFlags.h>
@@ -96,9 +99,6 @@
 ConversionResult<unsigned int> aidl2legacy_int32_t_config_mask(int32_t aidl);
 ConversionResult<int32_t> legacy2aidl_config_mask_int32_t(unsigned int legacy);
 
-ConversionResult<audio_channel_mask_t> aidl2legacy_int32_t_audio_channel_mask_t(int32_t aidl);
-ConversionResult<int32_t> legacy2aidl_audio_channel_mask_t_int32_t(audio_channel_mask_t legacy);
-
 ConversionResult<pid_t> aidl2legacy_int32_t_pid_t(int32_t aidl);
 ConversionResult<int32_t> legacy2aidl_pid_t_int32_t(pid_t legacy);
 
@@ -116,10 +116,10 @@
 ConversionResult<std::optional<std::string_view>>
 legacy2aidl_optional_String16_optional_string(std::optional<String16> legacy);
 
-ConversionResult<audio_io_config_event> aidl2legacy_AudioIoConfigEvent_audio_io_config_event(
+ConversionResult<audio_io_config_event_t> aidl2legacy_AudioIoConfigEvent_audio_io_config_event_t(
         media::AudioIoConfigEvent aidl);
-ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_AudioIoConfigEvent(
-        audio_io_config_event legacy);
+ConversionResult<media::AudioIoConfigEvent> legacy2aidl_audio_io_config_event_t_AudioIoConfigEvent(
+        audio_io_config_event_t legacy);
 
 ConversionResult<audio_port_role_t> aidl2legacy_AudioPortRole_audio_port_role_t(
         media::AudioPortRole aidl);
@@ -131,9 +131,19 @@
 ConversionResult<media::AudioPortType> legacy2aidl_audio_port_type_t_AudioPortType(
         audio_port_type_t legacy);
 
-ConversionResult<audio_format_t> aidl2legacy_AudioFormat_audio_format_t(
-        media::audio::common::AudioFormat aidl);
-ConversionResult<media::audio::common::AudioFormat> legacy2aidl_audio_format_t_AudioFormat(
+ConversionResult<audio_channel_mask_t> aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+        const media::AudioChannelLayout& aidl, bool isInput);
+ConversionResult<media::AudioChannelLayout> legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
+        audio_channel_mask_t legacy, bool isInput);
+
+ConversionResult<audio_devices_t> aidl2legacy_AudioDeviceDescription_audio_devices_t(
+        const media::AudioDeviceDescription& aidl);
+ConversionResult<media::AudioDeviceDescription> legacy2aidl_audio_devices_t_AudioDeviceDescription(
+        audio_devices_t legacy);
+
+ConversionResult<audio_format_t> aidl2legacy_AudioFormatDescription_audio_format_t(
+        const media::AudioFormatDescription& aidl);
+ConversionResult<media::AudioFormatDescription> legacy2aidl_audio_format_t_AudioFormatDescription(
         audio_format_t legacy);
 
 ConversionResult<audio_gain_mode_t>
@@ -144,9 +154,6 @@
 ConversionResult<audio_gain_mode_t> aidl2legacy_int32_t_audio_gain_mode_t_mask(int32_t aidl);
 ConversionResult<int32_t> legacy2aidl_audio_gain_mode_t_int32_t_mask(audio_gain_mode_t legacy);
 
-ConversionResult<audio_devices_t> aidl2legacy_int32_t_audio_devices_t(int32_t aidl);
-ConversionResult<int32_t> legacy2aidl_audio_devices_t_int32_t(audio_devices_t legacy);
-
 ConversionResult<audio_gain_config> aidl2legacy_AudioGainConfig_audio_gain_config(
         const media::AudioGainConfig& aidl, media::AudioPortRole role, media::AudioPortType type);
 ConversionResult<media::AudioGainConfig> legacy2aidl_audio_gain_config_AudioGainConfig(
@@ -221,7 +228,6 @@
 
 ConversionResult<sp<AudioIoDescriptor>> aidl2legacy_AudioIoDescriptor_AudioIoDescriptor(
         const media::AudioIoDescriptor& aidl);
-
 ConversionResult<media::AudioIoDescriptor> legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(
         const sp<AudioIoDescriptor>& legacy);
 
@@ -266,14 +272,14 @@
 legacy2aidl_audio_offload_info_t_AudioOffloadInfo(const audio_offload_info_t& legacy);
 
 ConversionResult<audio_config_t>
-aidl2legacy_AudioConfig_audio_config_t(const media::AudioConfig& aidl);
+aidl2legacy_AudioConfig_audio_config_t(const media::AudioConfig& aidl, bool isInput);
 ConversionResult<media::AudioConfig>
-legacy2aidl_audio_config_t_AudioConfig(const audio_config_t& legacy);
+legacy2aidl_audio_config_t_AudioConfig(const audio_config_t& legacy, bool isInput);
 
 ConversionResult<audio_config_base_t>
-aidl2legacy_AudioConfigBase_audio_config_base_t(const media::AudioConfigBase& aidl);
+aidl2legacy_AudioConfigBase_audio_config_base_t(const media::AudioConfigBase& aidl, bool isInput);
 ConversionResult<media::AudioConfigBase>
-legacy2aidl_audio_config_base_t_AudioConfigBase(const audio_config_base_t& legacy);
+legacy2aidl_audio_config_base_t_AudioConfigBase(const audio_config_base_t& legacy, bool isInput);
 
 ConversionResult<sp<IMemory>>
 aidl2legacy_SharedFileRegion_IMemory(const media::SharedFileRegion& aidl);
@@ -340,14 +346,15 @@
 legacy2aidl_audio_port_session_ext_AudioPortSessionExt(const audio_port_session_ext& legacy);
 
 ConversionResult<audio_profile>
-aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl);
+aidl2legacy_AudioProfile_audio_profile(const media::AudioProfile& aidl, bool isInput);
 ConversionResult<media::AudioProfile>
-legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy);
+legacy2aidl_audio_profile_AudioProfile(const audio_profile& legacy, bool isInput);
 
 ConversionResult<audio_gain>
 aidl2legacy_AudioGain_audio_gain(const media::AudioGain& aidl);
+// The AIDL structure provides a flag for direction indication while the legacy type doesn't.
 ConversionResult<media::AudioGain>
-legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy);
+legacy2aidl_audio_gain_AudioGain(const audio_gain& legacy, bool isInput);
 
 ConversionResult<audio_port_v7>
 aidl2legacy_AudioPort_audio_port_v7(const media::AudioPort& aidl);
diff --git a/media/libaudioclient/include/media/AudioCommonTypes.h b/media/libaudioclient/include/media/AudioCommonTypes.h
index 5dfe5fc..bd341e3 100644
--- a/media/libaudioclient/include/media/AudioCommonTypes.h
+++ b/media/libaudioclient/include/media/AudioCommonTypes.h
@@ -17,9 +17,72 @@
 
 #pragma once
 
+#include <functional>
+
+#include <android/media/AudioChannelLayout.h>
+#include <android/media/AudioDeviceDescription.h>
+#include <android/media/AudioFormatDescription.h>
+#include <binder/Parcelable.h>
 #include <system/audio.h>
 #include <system/audio_policy.h>
-#include <binder/Parcelable.h>
+
+namespace {
+// see boost::hash_combine
+#if defined(__clang__)
+__attribute__((no_sanitize("unsigned-integer-overflow")))
+#endif
+static size_t hash_combine(size_t seed, size_t v) {
+    return std::hash<size_t>{}(v) + 0x9e3779b9 + (seed << 6) + (seed >> 2);
+}
+}
+
+namespace std {
+
+// Note: when extending the types hashed below we need to account for the
+// possibility of processing types belonging to different versions of the type,
+// e.g. a HAL may be using a previous version of the AIDL interface.
+
+template<> struct hash<android::media::AudioChannelLayout>
+{
+    std::size_t operator()(const android::media::AudioChannelLayout& acl) const noexcept {
+        using Tag = android::media::AudioChannelLayout::Tag;
+        const size_t seed = std::hash<Tag>{}(acl.getTag());
+        switch (acl.getTag()) {
+            case Tag::none:
+                return hash_combine(seed, std::hash<int32_t>{}(acl.get<Tag::none>()));
+            case Tag::invalid:
+                return hash_combine(seed, std::hash<int32_t>{}(acl.get<Tag::invalid>()));
+            case Tag::indexMask:
+                return hash_combine(seed, std::hash<int32_t>{}(acl.get<Tag::indexMask>()));
+            case Tag::layoutMask:
+                return hash_combine(seed, std::hash<int32_t>{}(acl.get<Tag::layoutMask>()));
+            case Tag::voiceMask:
+                return hash_combine(seed, std::hash<int32_t>{}(acl.get<Tag::voiceMask>()));
+        }
+        return seed;
+    }
+};
+
+template<> struct hash<android::media::AudioDeviceDescription>
+{
+    std::size_t operator()(const android::media::AudioDeviceDescription& add) const noexcept {
+        return hash_combine(
+                std::hash<android::media::AudioDeviceType>{}(add.type),
+                std::hash<std::string>{}(add.connection));
+    }
+};
+
+template<> struct hash<android::media::AudioFormatDescription>
+{
+    std::size_t operator()(const android::media::AudioFormatDescription& afd) const noexcept {
+        return hash_combine(
+                std::hash<android::media::AudioFormatType>{}(afd.type),
+                hash_combine(
+                        std::hash<android::media::PcmType>{}(afd.pcm),
+                        std::hash<std::string>{}(afd.encoding)));
+    }
+};
+}  // namespace std
 
 namespace android {
 
@@ -41,8 +104,43 @@
     return !(lhs==rhs);
 }
 
+constexpr bool operator==(const audio_offload_info_t &lhs, const audio_offload_info_t &rhs)
+{
+    return lhs.version == rhs.version && lhs.size == rhs.size &&
+           lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+           lhs.format == rhs.format && lhs.stream_type == rhs.stream_type &&
+           lhs.bit_rate == rhs.bit_rate && lhs.duration_us == rhs.duration_us &&
+           lhs.has_video == rhs.has_video && lhs.is_streaming == rhs.is_streaming &&
+           lhs.bit_width == rhs.bit_width && lhs.offload_buffer_size == rhs.offload_buffer_size &&
+           lhs.usage == rhs.usage && lhs.encapsulation_mode == rhs.encapsulation_mode &&
+           lhs.content_id == rhs.content_id && lhs.sync_id == rhs.sync_id;
+}
+constexpr bool operator!=(const audio_offload_info_t &lhs, const audio_offload_info_t &rhs)
+{
+    return !(lhs==rhs);
+}
+
+constexpr bool operator==(const audio_config_t &lhs, const audio_config_t &rhs)
+{
+    return lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+           lhs.format == rhs.format && lhs.offload_info == rhs.offload_info;
+}
+constexpr bool operator!=(const audio_config_t &lhs, const audio_config_t &rhs)
+{
+    return !(lhs==rhs);
+}
+
+constexpr bool operator==(const audio_config_base_t &lhs, const audio_config_base_t &rhs)
+{
+    return lhs.sample_rate == rhs.sample_rate && lhs.channel_mask == rhs.channel_mask &&
+           lhs.format == rhs.format;
+}
+constexpr bool operator!=(const audio_config_base_t &lhs, const audio_config_base_t &rhs)
+{
+    return !(lhs==rhs);
+}
+
 enum volume_group_t : uint32_t;
 static const volume_group_t VOLUME_GROUP_NONE = static_cast<volume_group_t>(-1);
 
 } // namespace android
-
diff --git a/media/libaudioclient/include/media/AudioIoDescriptor.h b/media/libaudioclient/include/media/AudioIoDescriptor.h
index 981d33a..ef729ed 100644
--- a/media/libaudioclient/include/media/AudioIoDescriptor.h
+++ b/media/libaudioclient/include/media/AudioIoDescriptor.h
@@ -17,9 +17,15 @@
 #ifndef ANDROID_AUDIO_IO_DESCRIPTOR_H
 #define ANDROID_AUDIO_IO_DESCRIPTOR_H
 
+#include <sstream>
+#include <string>
+
+#include <system/audio.h>
+#include <utils/RefBase.h>
+
 namespace android {
 
-enum audio_io_config_event {
+enum audio_io_config_event_t {
     AUDIO_OUTPUT_REGISTERED,
     AUDIO_OUTPUT_OPENED,
     AUDIO_OUTPUT_CLOSED,
@@ -35,39 +41,68 @@
 // frequent calls through IAudioFlinger
 class AudioIoDescriptor : public RefBase {
 public:
-    AudioIoDescriptor() :
-        mIoHandle(AUDIO_IO_HANDLE_NONE),
-        mSamplingRate(0), mFormat(AUDIO_FORMAT_DEFAULT), mChannelMask(AUDIO_CHANNEL_NONE),
-        mFrameCount(0), mFrameCountHAL(0), mLatency(0), mPortId(AUDIO_PORT_HANDLE_NONE)
-    {
-        memset(&mPatch, 0, sizeof(struct audio_patch));
-    }
+    AudioIoDescriptor() = default;
+    // For AUDIO_{INPUT|OUTPUT}_CLOSED events.
+    AudioIoDescriptor(audio_io_handle_t ioHandle) : mIoHandle(ioHandle) {}
+    // For AUDIO_CLIENT_STARTED events.
+    AudioIoDescriptor(
+            audio_io_handle_t ioHandle, const audio_patch& patch, audio_port_handle_t portId) :
+            mIoHandle(ioHandle), mPatch(patch), mPortId(portId) {}
+    // For everything else.
+    AudioIoDescriptor(
+            audio_io_handle_t ioHandle, const audio_patch& patch, bool isInput,
+            uint32_t samplingRate, audio_format_t format, audio_channel_mask_t channelMask,
+            size_t frameCount, size_t frameCountHal, uint32_t latency = 0,
+            audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) :
+            mIoHandle(ioHandle), mPatch(patch), mIsInput(isInput),
+            mSamplingRate(samplingRate), mFormat(format), mChannelMask(channelMask),
+            mFrameCount(frameCount), mFrameCountHAL(frameCountHal), mLatency(latency),
+            mPortId(portId) {}
 
-    virtual ~AudioIoDescriptor() {}
-
-    audio_port_handle_t getDeviceId() {
+    audio_io_handle_t getIoHandle() const { return mIoHandle; }
+    const audio_patch& getPatch() const { return mPatch; }
+    bool getIsInput() const { return mIsInput; }
+    uint32_t getSamplingRate() const { return mSamplingRate; }
+    audio_format_t getFormat() const { return mFormat; }
+    audio_channel_mask_t getChannelMask() const { return mChannelMask; }
+    size_t getFrameCount() const { return mFrameCount; }
+    size_t getFrameCountHAL() const { return mFrameCountHAL; }
+    uint32_t getLatency() const { return mLatency; }
+    audio_port_handle_t getPortId() const { return mPortId; }
+    audio_port_handle_t getDeviceId() const {
         if (mPatch.num_sources != 0 && mPatch.num_sinks != 0) {
-            if (mPatch.sources[0].type == AUDIO_PORT_TYPE_MIX) {
-                // this is an output mix
-                // FIXME: the API only returns the first device in case of multiple device selection
-                return mPatch.sinks[0].id;
-            } else {
-                // this is an input mix
-                return mPatch.sources[0].id;
-            }
+            // FIXME: the API only returns the first device in case of multiple device selection
+            return mIsInput ? mPatch.sources[0].id : mPatch.sinks[0].id;
         }
         return AUDIO_PORT_HANDLE_NONE;
     }
+    void setPatch(const audio_patch& patch) { mPatch = patch; }
 
-    audio_io_handle_t       mIoHandle;
-    struct audio_patch      mPatch;
-    uint32_t                mSamplingRate;
-    audio_format_t          mFormat;
-    audio_channel_mask_t    mChannelMask;
-    size_t                  mFrameCount;
-    size_t                  mFrameCountHAL;
-    uint32_t                mLatency;   // only valid for output
-    audio_port_handle_t     mPortId;    // valid for event AUDIO_CLIENT_STARTED
+    std::string toDebugString() const {
+        std::ostringstream ss;
+        ss << mIoHandle << ", samplingRate " << mSamplingRate << ", "
+           << audio_format_to_string(mFormat) << ", "
+           << (audio_channel_mask_get_representation(mChannelMask) ==
+                   AUDIO_CHANNEL_REPRESENTATION_INDEX ?
+                   audio_channel_index_mask_to_string(mChannelMask) :
+                   (mIsInput ? audio_channel_in_mask_to_string(mChannelMask) :
+                           audio_channel_out_mask_to_string(mChannelMask)))
+           << ", frameCount " << mFrameCount << ", frameCountHAL " << mFrameCountHAL
+           << ", deviceId " << getDeviceId();
+        return ss.str();
+    }
+
+  private:
+    const audio_io_handle_t    mIoHandle = AUDIO_IO_HANDLE_NONE;
+          struct audio_patch   mPatch = {};
+    const bool                 mIsInput = false;
+    const uint32_t             mSamplingRate = 0;
+    const audio_format_t       mFormat = AUDIO_FORMAT_DEFAULT;
+    const audio_channel_mask_t mChannelMask = AUDIO_CHANNEL_NONE;
+    const size_t               mFrameCount = 0;
+    const size_t               mFrameCountHAL = 0;
+    const uint32_t             mLatency = 0;
+    const audio_port_handle_t  mPortId = AUDIO_PORT_HANDLE_NONE;
 };
 
 
diff --git a/media/libaudioclient/include/media/AudioRecord.h b/media/libaudioclient/include/media/AudioRecord.h
index 326919a..f17ee3a 100644
--- a/media/libaudioclient/include/media/AudioRecord.h
+++ b/media/libaudioclient/include/media/AudioRecord.h
@@ -264,6 +264,7 @@
             size_t      frameCount() const  { return mFrameCount; }
             size_t      frameSize() const   { return mFrameSize; }
             audio_source_t inputSource() const  { return mAttributes.source; }
+            audio_channel_mask_t channelMask() const { return mChannelMask; }
 
     /*
      * Return the period of the notification callback in frames.
diff --git a/media/libaudioclient/include/media/AudioSystem.h b/media/libaudioclient/include/media/AudioSystem.h
index a9109c8..c92b924 100644
--- a/media/libaudioclient/include/media/AudioSystem.h
+++ b/media/libaudioclient/include/media/AudioSystem.h
@@ -24,6 +24,7 @@
 #include <android/media/BnAudioPolicyServiceClient.h>
 #include <android/content/AttributionSourceState.h>
 #include <media/AidlConversionUtil.h>
+#include <media/AudioContainers.h>
 #include <media/AudioDeviceTypeAddr.h>
 #include <media/AudioPolicy.h>
 #include <media/AudioProductStrategy.h>
@@ -225,6 +226,9 @@
     // Indicate JAVA services are ready (scheduling, power management ...)
     static status_t systemReady();
 
+    // Indicate audio policy service is ready
+    static status_t audioPolicyReady();
+
     // Returns the number of frames per audio HAL buffer.
     // Corresponds to audio_stream->get_buffer_size()/audio_stream_in_frame_size() for input.
     // See also getFrameCount().
@@ -318,7 +322,7 @@
     static status_t getMinVolumeIndexForAttributes(const audio_attributes_t &attr, int &index);
 
     static product_strategy_t getStrategyForStream(audio_stream_type_t stream);
-    static audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+    static DeviceTypeSet getDevicesForStream(audio_stream_type_t stream);
     static status_t getDevicesForAttributes(const AudioAttributes &aa,
                                             AudioDeviceTypeAddrVector *devices);
 
diff --git a/media/libaudioclient/include/media/AudioTrack.h b/media/libaudioclient/include/media/AudioTrack.h
index cb00990..a2cd12f 100644
--- a/media/libaudioclient/include/media/AudioTrack.h
+++ b/media/libaudioclient/include/media/AudioTrack.h
@@ -401,6 +401,7 @@
 
             uint32_t    channelCount() const { return mChannelCount; }
             size_t      frameCount() const  { return mFrameCount; }
+            audio_channel_mask_t channelMask() const { return mChannelMask; }
 
     /*
      * Return the period of the notification callback in frames.
@@ -427,8 +428,7 @@
      * less than or equal to the getBufferCapacityInFrames().
      * It may also be adjusted slightly for internal reasons.
      *
-     * Return the final size or a negative error if the track is unitialized
-     * or does not support variable sizes.
+     * Return the final size or a negative value (NO_INIT) if the track is uninitialized.
      */
             ssize_t     setBufferSizeInFrames(size_t size);
 
diff --git a/media/libaudioclient/include/media/IAudioFlinger.h b/media/libaudioclient/include/media/IAudioFlinger.h
index 0e059f7..8632abb 100644
--- a/media/libaudioclient/include/media/IAudioFlinger.h
+++ b/media/libaudioclient/include/media/IAudioFlinger.h
@@ -329,6 +329,9 @@
     /* Indicate JAVA services are ready (scheduling, power management ...) */
     virtual status_t systemReady() = 0;
 
+    // Indicate audio policy service is ready
+    virtual status_t audioPolicyReady() = 0;
+
     // Returns the number of frames per audio HAL buffer.
     virtual size_t frameCountHAL(audio_io_handle_t ioHandle) const = 0;
 
@@ -432,6 +435,8 @@
     status_t setAudioPortConfig(const struct audio_port_config* config) override;
     audio_hw_sync_t getAudioHwSyncForSession(audio_session_t sessionId) override;
     status_t systemReady() override;
+    status_t audioPolicyReady() override;
+
     size_t frameCountHAL(audio_io_handle_t ioHandle) const override;
     status_t getMicrophones(std::vector<media::MicrophoneInfo>* microphones) override;
     status_t setAudioHalPids(const std::vector<pid_t>& pids) override;
@@ -514,6 +519,7 @@
             SET_AUDIO_PORT_CONFIG = media::BnAudioFlingerService::TRANSACTION_setAudioPortConfig,
             GET_AUDIO_HW_SYNC_FOR_SESSION = media::BnAudioFlingerService::TRANSACTION_getAudioHwSyncForSession,
             SYSTEM_READY = media::BnAudioFlingerService::TRANSACTION_systemReady,
+            AUDIO_POLICY_READY = media::BnAudioFlingerService::TRANSACTION_audioPolicyReady,
             FRAME_COUNT_HAL = media::BnAudioFlingerService::TRANSACTION_frameCountHAL,
             GET_MICROPHONES = media::BnAudioFlingerService::TRANSACTION_getMicrophones,
             SET_MASTER_BALANCE = media::BnAudioFlingerService::TRANSACTION_setMasterBalance,
@@ -563,7 +569,7 @@
     Status createRecord(const media::CreateRecordRequest& request,
                         media::CreateRecordResponse* _aidl_return) override;
     Status sampleRate(int32_t ioHandle, int32_t* _aidl_return) override;
-    Status format(int32_t output, media::audio::common::AudioFormat* _aidl_return) override;
+    Status format(int32_t output, media::AudioFormatDescription* _aidl_return) override;
     Status frameCount(int32_t ioHandle, int64_t* _aidl_return) override;
     Status latency(int32_t output, int32_t* _aidl_return) override;
     Status setMasterVolume(float value) override;
@@ -585,8 +591,9 @@
     Status
     getParameters(int32_t ioHandle, const std::string& keys, std::string* _aidl_return) override;
     Status registerClient(const sp<media::IAudioFlingerClient>& client) override;
-    Status getInputBufferSize(int32_t sampleRate, media::audio::common::AudioFormat format,
-                              int32_t channelMask, int64_t* _aidl_return) override;
+    Status getInputBufferSize(int32_t sampleRate, const media::AudioFormatDescription& format,
+                              const media::AudioChannelLayout& channelMask,
+                              int64_t* _aidl_return) override;
     Status openOutput(const media::OpenOutputRequest& request,
                       media::OpenOutputResponse* _aidl_return) override;
     Status openDuplicateOutput(int32_t output1, int32_t output2, int32_t* _aidl_return) override;
@@ -624,6 +631,7 @@
     Status setAudioPortConfig(const media::AudioPortConfig& config) override;
     Status getAudioHwSyncForSession(int32_t sessionId, int32_t* _aidl_return) override;
     Status systemReady() override;
+    Status audioPolicyReady() override;
     Status frameCountHAL(int32_t ioHandle, int64_t* _aidl_return) override;
     Status getMicrophones(std::vector<media::MicrophoneInfoData>* _aidl_return) override;
     Status setAudioHalPids(const std::vector<int32_t>& pids) override;
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index def7ca6..2279244 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -9,10 +9,34 @@
 
 cc_defaults {
     name: "libaudioclient_tests_defaults",
+    test_suites: ["device-tests"],
     cflags: [
         "-Wall",
         "-Werror",
     ],
+    sanitize: {
+        misc_undefined: [
+            "unsigned-integer-overflow",
+            "signed-integer-overflow",
+        ],
+    },
+}
+
+cc_test {
+    name: "audio_aidl_conversion_tests",
+    defaults: ["libaudioclient_tests_defaults"],
+    srcs: ["audio_aidl_legacy_conversion_tests.cpp"],
+    shared_libs: [
+        "libbinder",
+        "libcutils",
+        "liblog",
+        "libutils",
+    ],
+    static_libs: [
+        "audioclient-types-aidl-cpp",
+        "libaudioclient_aidl_conversion",
+        "libstagefright_foundation",
+    ],
 }
 
 cc_test {
@@ -30,8 +54,10 @@
 cc_test {
     name: "test_create_audiotrack",
     defaults: ["libaudioclient_tests_defaults"],
-    srcs: ["test_create_audiotrack.cpp",
-           "test_create_utils.cpp"],
+    srcs: [
+        "test_create_audiotrack.cpp",
+        "test_create_utils.cpp",
+    ],
     header_libs: [
         "libmedia_headers",
         "libmediametrics_headers",
@@ -49,8 +75,10 @@
 cc_test {
     name: "test_create_audiorecord",
     defaults: ["libaudioclient_tests_defaults"],
-    srcs: ["test_create_audiorecord.cpp",
-           "test_create_utils.cpp"],
+    srcs: [
+        "test_create_audiorecord.cpp",
+        "test_create_utils.cpp",
+    ],
     header_libs: [
         "libmedia_headers",
         "libmediametrics_headers",
diff --git a/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
new file mode 100644
index 0000000..7f8af53
--- /dev/null
+++ b/media/libaudioclient/tests/audio_aidl_legacy_conversion_tests.cpp
@@ -0,0 +1,225 @@
+/*
+ * 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.
+ */
+
+#include <gtest/gtest.h>
+
+#include <media/AudioCommonTypes.h>
+#include <media/AidlConversion.h>
+
+using namespace android;
+using namespace android::aidl_utils;
+
+namespace {
+
+template<typename T> size_t hash(const T& t) {
+    return std::hash<T>{}(t);
+}
+
+media::AudioChannelLayout make_ACL_None() {
+    return media::AudioChannelLayout{};
+}
+
+media::AudioChannelLayout make_ACL_Invalid() {
+    return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::invalid>(0);
+}
+
+media::AudioChannelLayout make_ACL_Stereo() {
+    return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::layoutMask>(
+            media::AudioChannelLayout::LAYOUT_STEREO);
+}
+
+media::AudioChannelLayout make_ACL_ChannelIndex2() {
+    return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::indexMask>(
+            media::AudioChannelLayout::INDEX_MASK_2);
+}
+
+media::AudioChannelLayout make_ACL_VoiceCall() {
+    return media::AudioChannelLayout::make<media::AudioChannelLayout::Tag::voiceMask>(
+            media::AudioChannelLayout::VOICE_CALL_MONO);
+}
+
+media::AudioDeviceDescription make_AudioDeviceDescription(media::AudioDeviceType type,
+        const std::string& connection = "") {
+    media::AudioDeviceDescription result;
+    result.type = type;
+    result.connection = connection;
+    return result;
+}
+
+media::AudioDeviceDescription make_ADD_None() {
+    return media::AudioDeviceDescription{};
+}
+
+media::AudioDeviceDescription make_ADD_DefaultIn() {
+    return make_AudioDeviceDescription(media::AudioDeviceType::IN_DEFAULT);
+}
+
+media::AudioDeviceDescription make_ADD_DefaultOut() {
+    return make_AudioDeviceDescription(media::AudioDeviceType::OUT_DEFAULT);
+}
+
+media::AudioDeviceDescription make_ADD_WiredHeadset() {
+    return make_AudioDeviceDescription(media::AudioDeviceType::OUT_HEADSET,
+            media::AudioDeviceDescription::CONNECTION_ANALOG());
+}
+
+media::AudioDeviceDescription make_ADD_BtScoHeadset() {
+    return make_AudioDeviceDescription(media::AudioDeviceType::OUT_HEADSET,
+            media::AudioDeviceDescription::CONNECTION_BT_SCO());
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(media::AudioFormatType type) {
+    media::AudioFormatDescription result;
+    result.type = type;
+    return result;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(media::PcmType pcm) {
+    auto result = make_AudioFormatDescription(media::AudioFormatType::PCM);
+    result.pcm = pcm;
+    return result;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(const std::string& encoding) {
+    media::AudioFormatDescription result;
+    result.encoding = encoding;
+    return result;
+}
+
+media::AudioFormatDescription make_AudioFormatDescription(media::PcmType transport,
+        const std::string& encoding) {
+    auto result = make_AudioFormatDescription(encoding);
+    result.pcm = transport;
+    return result;
+}
+
+media::AudioFormatDescription make_AFD_Default() {
+    return media::AudioFormatDescription{};
+}
+
+media::AudioFormatDescription make_AFD_Invalid() {
+    return make_AudioFormatDescription(media::AudioFormatType::SYS_RESERVED_INVALID);
+}
+
+media::AudioFormatDescription make_AFD_Pcm16Bit() {
+    return make_AudioFormatDescription(media::PcmType::INT_16_BIT);
+}
+
+media::AudioFormatDescription make_AFD_Bitstream() {
+    return make_AudioFormatDescription("example");
+}
+
+media::AudioFormatDescription make_AFD_Encap() {
+    return make_AudioFormatDescription(media::PcmType::INT_16_BIT, "example.encap");
+}
+
+media::AudioFormatDescription make_AFD_Encap_with_Enc() {
+    auto afd = make_AFD_Encap();
+    afd.encoding += "+example";
+    return afd;
+}
+
+}  // namespace
+
+// Verify that two independently constructed ADDs/AFDs have the same hash.
+// This ensures that regardless of whether the ADD/AFD instance originates
+// from, it can be correctly compared to other ADD/AFD instance. Thus,
+// for example, a 16-bit integer format description provided by HAL
+// 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) {
+        for (size_t i = 0; i < valueGens.size(); ++i) {
+            for (size_t j = 0; j < valueGens.size(); ++j) {
+                if (i == j) {
+                    EXPECT_EQ(hash(valueGens[i]()), hash(valueGens[i]())) << i;
+                } else {
+                    EXPECT_NE(hash(valueGens[i]()), hash(valueGens[j]())) << i << ", " << j;
+                }
+            }
+        }
+    }
+};
+
+TEST_F(HashIdentityTest, AudioChannelLayoutHashIdentity) {
+    verifyHashIdentity<media::AudioChannelLayout>({
+            make_ACL_None, make_ACL_Invalid, make_ACL_Stereo, make_ACL_ChannelIndex2,
+            make_ACL_VoiceCall});
+}
+
+TEST_F(HashIdentityTest, AudioDeviceDescriptionHashIdentity) {
+    verifyHashIdentity<media::AudioDeviceDescription>({
+            make_ADD_None, make_ADD_DefaultIn, make_ADD_DefaultOut, make_ADD_WiredHeadset,
+            make_ADD_BtScoHeadset});
+}
+
+TEST_F(HashIdentityTest, AudioFormatDescriptionHashIdentity) {
+    verifyHashIdentity<media::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<media::AudioChannelLayout, bool /*isInput*/>;
+class AudioChannelLayoutRoundTripTest :
+        public testing::TestWithParam<ChannelLayoutParam> {};
+TEST_P(AudioChannelLayoutRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = std::get<0>(GetParam());
+    const bool isInput = std::get<1>(GetParam());
+    auto conv = aidl2legacy_AudioChannelLayout_audio_channel_mask_t(initial, isInput);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_channel_mask_t_AudioChannelLayout(conv.value(), isInput);
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioChannelLayoutRoundTrip,
+        AudioChannelLayoutRoundTripTest,
+        testing::Combine(
+                testing::Values(media::AudioChannelLayout{}, make_ACL_Invalid(), make_ACL_Stereo(),
+                        make_ACL_ChannelIndex2()),
+                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)));
+
+class AudioDeviceDescriptionRoundTripTest :
+        public testing::TestWithParam<media::AudioDeviceDescription> {};
+TEST_P(AudioDeviceDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = GetParam();
+    auto conv = aidl2legacy_AudioDeviceDescription_audio_devices_t(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_devices_t_AudioDeviceDescription(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioDeviceDescriptionRoundTrip,
+        AudioDeviceDescriptionRoundTripTest,
+        testing::Values(media::AudioDeviceDescription{}, make_ADD_DefaultIn(),
+                make_ADD_DefaultOut(), make_ADD_WiredHeadset(), make_ADD_BtScoHeadset()));
+
+class AudioFormatDescriptionRoundTripTest :
+        public testing::TestWithParam<media::AudioFormatDescription> {};
+TEST_P(AudioFormatDescriptionRoundTripTest, Aidl2Legacy2Aidl) {
+    const auto initial = GetParam();
+    auto conv = aidl2legacy_AudioFormatDescription_audio_format_t(initial);
+    ASSERT_TRUE(conv.ok());
+    auto convBack = legacy2aidl_audio_format_t_AudioFormatDescription(conv.value());
+    ASSERT_TRUE(convBack.ok());
+    EXPECT_EQ(initial, convBack.value());
+}
+INSTANTIATE_TEST_SUITE_P(AudioFormatDescriptionRoundTrip,
+        AudioFormatDescriptionRoundTripTest,
+        testing::Values(make_AFD_Invalid(), media::AudioFormatDescription{}, make_AFD_Pcm16Bit()));
diff --git a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
index c5d7da8..7f54474 100644
--- a/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
+++ b/media/libaudiofoundation/AudioDeviceTypeAddr.cpp
@@ -158,14 +158,15 @@
 
 ConversionResult<AudioDeviceTypeAddr>
 aidl2legacy_AudioDeviceTypeAddress(const media::AudioDevice& aidl) {
-    audio_devices_t type = VALUE_OR_RETURN(aidl2legacy_int32_t_audio_devices_t(aidl.type));
+    audio_devices_t type = VALUE_OR_RETURN(
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(aidl.type));
     return AudioDeviceTypeAddr(type, aidl.address);
 }
 
 ConversionResult<media::AudioDevice>
 legacy2aidl_AudioDeviceTypeAddress(const AudioDeviceTypeAddr& legacy) {
     media::AudioDevice aidl;
-    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_int32_t(legacy.mType));
+    aidl.type = VALUE_OR_RETURN(legacy2aidl_audio_devices_t_AudioDeviceDescription(legacy.mType));
     aidl.address = legacy.getAddress();
     return aidl;
 }
diff --git a/media/libaudiofoundation/AudioGain.cpp b/media/libaudiofoundation/AudioGain.cpp
index 1dee938..ea0258a 100644
--- a/media/libaudiofoundation/AudioGain.cpp
+++ b/media/libaudiofoundation/AudioGain.cpp
@@ -34,10 +34,10 @@
 
 namespace android {
 
-AudioGain::AudioGain(int index, bool useInChannelMask)
+AudioGain::AudioGain(int index, bool isInput)
 {
     mIndex = index;
-    mUseInChannelMask = useInChannelMask;
+    mIsInput = isInput;
     memset(&mGain, 0, sizeof(struct audio_gain));
 }
 
@@ -49,12 +49,9 @@
     if ((mGain.mode & AUDIO_GAIN_MODE_JOINT) == AUDIO_GAIN_MODE_JOINT) {
         config->values[0] = mGain.default_value;
     } else {
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(mGain.channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(mGain.channel_mask);
-        }
+        const uint32_t numValues = mIsInput ?
+                audio_channel_count_from_in_mask(mGain.channel_mask) :
+                audio_channel_count_from_out_mask(mGain.channel_mask);
         for (size_t i = 0; i < numValues; i++) {
             config->values[i] = mGain.default_value;
         }
@@ -78,12 +75,9 @@
         if ((config->channel_mask & ~mGain.channel_mask) != 0) {
             return BAD_VALUE;
         }
-        uint32_t numValues;
-        if (mUseInChannelMask) {
-            numValues = audio_channel_count_from_in_mask(config->channel_mask);
-        } else {
-            numValues = audio_channel_count_from_out_mask(config->channel_mask);
-        }
+        const uint32_t numValues = mIsInput ?
+                audio_channel_count_from_in_mask(config->channel_mask) :
+                audio_channel_count_from_out_mask(config->channel_mask);
         for (size_t i = 0; i < numValues; i++) {
             if ((config->values[i] < mGain.min_value) ||
                     (config->values[i] > mGain.max_value)) {
@@ -116,7 +110,7 @@
 bool AudioGain::equals(const sp<AudioGain>& other) const
 {
     return other != nullptr &&
-           mUseInChannelMask == other->mUseInChannelMask &&
+           mIsInput == other->mIsInput &&
            mUseForVolume == other->mUseForVolume &&
            // Compare audio gain
            mGain.mode == other->mGain.mode &&
@@ -137,12 +131,13 @@
 
 status_t AudioGain::writeToParcelable(media::AudioGain* parcelable) const {
     parcelable->index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mIndex));
-    parcelable->useInChannelMask = mUseInChannelMask;
+    parcelable->isInput = mIsInput;
     parcelable->useForVolume = mUseForVolume;
     parcelable->mode = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_gain_mode_t_int32_t_mask(mGain.mode));
     parcelable->channelMask = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(
+                    mGain.channel_mask, mIsInput));
     parcelable->minValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.min_value));
     parcelable->maxValue = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.max_value));
     parcelable->defaultValue = VALUE_OR_RETURN_STATUS(
@@ -161,12 +156,13 @@
 
 status_t AudioGain::readFromParcelable(const media::AudioGain& parcelable) {
     mIndex = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.index));
-    mUseInChannelMask = parcelable.useInChannelMask;
+    mIsInput = parcelable.isInput;
     mUseForVolume = parcelable.useForVolume;
     mGain.mode = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_gain_mode_t_mask(parcelable.mode));
     mGain.channel_mask = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_int32_t_audio_channel_mask_t(parcelable.channelMask));
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+                    parcelable.channelMask, parcelable.isInput));
     mGain.min_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.minValue));
     mGain.max_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.maxValue));
     mGain.default_value = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.defaultValue));
diff --git a/media/libaudiofoundation/AudioPort.cpp b/media/libaudiofoundation/AudioPort.cpp
index fafabd9..c70a6c2 100644
--- a/media/libaudiofoundation/AudioPort.cpp
+++ b/media/libaudiofoundation/AudioPort.cpp
@@ -210,7 +210,8 @@
     parcelable->name = mName;
     parcelable->type = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_type_t_AudioPortType(mType));
     parcelable->role = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_role_t_AudioPortRole(mRole));
-    parcelable->profiles = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioProfileVector(mProfiles));
+    parcelable->profiles = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_AudioProfileVector(mProfiles, useInputChannelMask()));
     parcelable->extraAudioDescriptors = mExtraAudioDescriptors;
     parcelable->gains = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioGains(mGains));
     return OK;
@@ -226,7 +227,8 @@
     mName = parcelable.name;
     mType = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPortType_audio_port_type_t(parcelable.type));
     mRole = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioPortRole_audio_port_role_t(parcelable.role));
-    mProfiles = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioProfileVector(parcelable.profiles));
+    mProfiles = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioProfileVector(parcelable.profiles, useInputChannelMask()));
     mExtraAudioDescriptors = parcelable.extraAudioDescriptors;
     mGains = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioGains(parcelable.gains));
     return OK;
@@ -330,23 +332,19 @@
            mGain.ramp_duration_ms == other->mGain.ramp_duration_ms;
 }
 
-status_t AudioPortConfig::writeToParcel(Parcel *parcel) const {
-    media::AudioPortConfig parcelable;
-    return writeToParcelable(&parcelable)
-        ?: parcelable.writeToParcel(parcel);
-}
-
-status_t AudioPortConfig::writeToParcelable(media::AudioPortConfig* parcelable) const {
+status_t AudioPortConfig::writeToParcelable(
+        media::AudioPortConfig* parcelable, bool isInput) const {
     parcelable->sampleRate = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mSamplingRate));
-    parcelable->format = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_format_t_AudioFormat(mFormat));
+    parcelable->format = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_format_t_AudioFormatDescription(mFormat));
     parcelable->channelMask = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_channel_mask_t_int32_t(mChannelMask));
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(mChannelMask, isInput));
     parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
     parcelable->gain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(mGain.index));
     parcelable->gain.mode = VALUE_OR_RETURN_STATUS(
             legacy2aidl_audio_gain_mode_t_int32_t_mask(mGain.mode));
     parcelable->gain.channelMask = VALUE_OR_RETURN_STATUS(
-            legacy2aidl_audio_channel_mask_t_int32_t(mGain.channel_mask));
+            legacy2aidl_audio_channel_mask_t_AudioChannelLayout(mGain.channel_mask, isInput));
     parcelable->gain.rampDurationMs = VALUE_OR_RETURN_STATUS(
             convertIntegral<int32_t>(mGain.ramp_duration_ms));
     parcelable->gain.values = VALUE_OR_RETURN_STATUS(convertContainer<std::vector<int32_t>>(
@@ -354,23 +352,20 @@
     return OK;
 }
 
-status_t AudioPortConfig::readFromParcel(const Parcel *parcel) {
-    media::AudioPortConfig parcelable;
-    return parcelable.readFromParcel(parcel)
-        ?: readFromParcelable(parcelable);
-}
-
-status_t AudioPortConfig::readFromParcelable(const media::AudioPortConfig& parcelable) {
+status_t AudioPortConfig::readFromParcelable(
+        const media::AudioPortConfig& parcelable, bool isInput) {
     mSamplingRate = VALUE_OR_RETURN_STATUS(convertIntegral<unsigned int>(parcelable.sampleRate));
-    mFormat = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioFormat_audio_format_t(parcelable.format));
+    mFormat = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioFormatDescription_audio_format_t(parcelable.format));
     mChannelMask = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_int32_t_audio_channel_mask_t(parcelable.channelMask));
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(parcelable.channelMask, isInput));
     mId = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_port_handle_t(parcelable.id));
     mGain.index = VALUE_OR_RETURN_STATUS(convertIntegral<int>(parcelable.gain.index));
     mGain.mode = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_gain_mode_t_mask(parcelable.gain.mode));
     mGain.channel_mask = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_int32_t_audio_channel_mask_t(parcelable.gain.channelMask));
+            aidl2legacy_AudioChannelLayout_audio_channel_mask_t(
+                    parcelable.gain.channelMask, isInput));
     mGain.ramp_duration_ms = VALUE_OR_RETURN_STATUS(
             convertIntegral<unsigned int>(parcelable.gain.rampDurationMs));
     if (parcelable.gain.values.size() > std::size(mGain.values)) {
diff --git a/media/libaudiofoundation/AudioProfile.cpp b/media/libaudiofoundation/AudioProfile.cpp
index 8ac3f73..47b2d54 100644
--- a/media/libaudiofoundation/AudioProfile.cpp
+++ b/media/libaudiofoundation/AudioProfile.cpp
@@ -154,19 +154,17 @@
     return *this;
 }
 
-status_t AudioProfile::writeToParcel(Parcel *parcel) const {
-    media::AudioProfile parcelable = VALUE_OR_RETURN_STATUS(toParcelable());
-    return parcelable.writeToParcel(parcel);
- }
-
 ConversionResult<media::AudioProfile>
-AudioProfile::toParcelable() const {
+AudioProfile::toParcelable(bool isInput) const {
     media::AudioProfile parcelable;
     parcelable.name = mName;
-    parcelable.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormat(mFormat));
+    parcelable.format = VALUE_OR_RETURN(legacy2aidl_audio_format_t_AudioFormatDescription(mFormat));
     parcelable.channelMasks = VALUE_OR_RETURN(
-            convertContainer<std::vector<int32_t>>(mChannelMasks,
-                                                   legacy2aidl_audio_channel_mask_t_int32_t));
+            convertContainer<std::vector<media::AudioChannelLayout>>(
+                    mChannelMasks,
+                    [isInput](audio_channel_mask_t m) {
+                        return legacy2aidl_audio_channel_mask_t_AudioChannelLayout(m, isInput);
+                    }));
     parcelable.samplingRates = VALUE_OR_RETURN(
             convertContainer<std::vector<int32_t>>(mSamplingRates,
                                                    convertIntegral<int32_t, uint32_t>));
@@ -178,23 +176,17 @@
     return parcelable;
 }
 
-status_t AudioProfile::readFromParcel(const Parcel *parcel) {
-    media::AudioProfile parcelable;
-    if (status_t status = parcelable.readFromParcel(parcel); status != OK) {
-        return status;
-    }
-    *this = *VALUE_OR_RETURN_STATUS(fromParcelable(parcelable));
-    return OK;
-}
-
 ConversionResult<sp<AudioProfile>>
-AudioProfile::fromParcelable(const media::AudioProfile& parcelable) {
+AudioProfile::fromParcelable(const media::AudioProfile& parcelable, bool isInput) {
     sp<AudioProfile> legacy = new AudioProfile();
     legacy->mName = parcelable.name;
-    legacy->mFormat = VALUE_OR_RETURN(aidl2legacy_AudioFormat_audio_format_t(parcelable.format));
+    legacy->mFormat = VALUE_OR_RETURN(
+            aidl2legacy_AudioFormatDescription_audio_format_t(parcelable.format));
     legacy->mChannelMasks = VALUE_OR_RETURN(
             convertContainer<ChannelMaskSet>(parcelable.channelMasks,
-                                             aidl2legacy_int32_t_audio_channel_mask_t));
+                    [isInput](const media::AudioChannelLayout& l) {
+                        return aidl2legacy_AudioChannelLayout_audio_channel_mask_t(l, isInput);
+                    }));
     legacy->mSamplingRates = VALUE_OR_RETURN(
             convertContainer<SampleRateSet>(parcelable.samplingRates,
                                             convertIntegral<uint32_t, int32_t>));
@@ -208,13 +200,13 @@
 }
 
 ConversionResult<sp<AudioProfile>>
-aidl2legacy_AudioProfile(const media::AudioProfile& aidl) {
-    return AudioProfile::fromParcelable(aidl);
+aidl2legacy_AudioProfile(const media::AudioProfile& aidl, bool isInput) {
+    return AudioProfile::fromParcelable(aidl, isInput);
 }
 
 ConversionResult<media::AudioProfile>
-legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy) {
-    return legacy->toParcelable();
+legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy, bool isInput) {
+    return legacy->toParcelable(isInput);
 }
 
 ssize_t AudioProfileVector::add(const sp<AudioProfile> &profile)
@@ -328,33 +320,6 @@
     }
 }
 
-status_t AudioProfileVector::writeToParcel(Parcel *parcel) const
-{
-    status_t status = NO_ERROR;
-    if ((status = parcel->writeVectorSize(*this)) != NO_ERROR) return status;
-    for (const auto &audioProfile : *this) {
-        if ((status = parcel->writeParcelable(*audioProfile)) != NO_ERROR) {
-            break;
-        }
-    }
-    return status;
-}
-
-status_t AudioProfileVector::readFromParcel(const Parcel *parcel)
-{
-    status_t status = NO_ERROR;
-    this->clear();
-    if ((status = parcel->resizeOutVector(this)) != NO_ERROR) return status;
-    for (size_t i = 0; i < this->size(); ++i) {
-        this->at(i) = new AudioProfile(AUDIO_FORMAT_DEFAULT, AUDIO_CHANNEL_NONE, 0 /*sampleRate*/);
-        if ((status = parcel->readParcelable(this->at(i).get())) != NO_ERROR) {
-            this->clear();
-            break;
-        }
-    }
-    return status;
-}
-
 bool AudioProfileVector::equals(const AudioProfileVector& other) const
 {
     return std::equal(begin(), end(), other.begin(), other.end(),
@@ -364,13 +329,19 @@
 }
 
 ConversionResult<AudioProfileVector>
-aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl) {
-    return convertContainer<AudioProfileVector>(aidl, aidl2legacy_AudioProfile);
+aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl, bool isInput) {
+    return convertContainer<AudioProfileVector>(aidl,
+            [isInput](const media::AudioProfile& p) {
+                return aidl2legacy_AudioProfile(p, isInput);
+            });
 }
 
 ConversionResult<std::vector<media::AudioProfile>>
-legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy) {
-    return convertContainer<std::vector<media::AudioProfile>>(legacy, legacy2aidl_AudioProfile);
+legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy, bool isInput) {
+    return convertContainer<std::vector<media::AudioProfile>>(legacy,
+            [isInput](const sp<AudioProfile>& p) {
+                return legacy2aidl_AudioProfile(p, isInput);
+            });
 }
 
 AudioProfileVector intersectAudioProfiles(const AudioProfileVector& profiles1,
diff --git a/media/libaudiofoundation/DeviceDescriptorBase.cpp b/media/libaudiofoundation/DeviceDescriptorBase.cpp
index 5cfea81..3cce722 100644
--- a/media/libaudiofoundation/DeviceDescriptorBase.cpp
+++ b/media/libaudiofoundation/DeviceDescriptorBase.cpp
@@ -166,7 +166,7 @@
 
 status_t DeviceDescriptorBase::writeToParcelable(media::AudioPort* parcelable) const {
     AudioPort::writeToParcelable(parcelable);
-    AudioPortConfig::writeToParcelable(&parcelable->activeConfig);
+    AudioPortConfig::writeToParcelable(&parcelable->activeConfig, useInputChannelMask());
     parcelable->id = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_port_handle_t_int32_t(mId));
 
     media::AudioPortDeviceExt ext;
@@ -190,7 +190,7 @@
         return BAD_VALUE;
     }
     status_t status = AudioPort::readFromParcelable(parcelable)
-                      ?: AudioPortConfig::readFromParcelable(parcelable.activeConfig);
+            ?: AudioPortConfig::readFromParcelable(parcelable.activeConfig, useInputChannelMask());
     if (status != OK) {
         return status;
     }
diff --git a/media/libaudiofoundation/include/media/AudioGain.h b/media/libaudiofoundation/include/media/AudioGain.h
index a06b686..28769d2 100644
--- a/media/libaudiofoundation/include/media/AudioGain.h
+++ b/media/libaudiofoundation/include/media/AudioGain.h
@@ -31,7 +31,7 @@
 class AudioGain: public RefBase, public Parcelable
 {
 public:
-    AudioGain(int index, bool useInChannelMask);
+    AudioGain(int index, bool isInput);
     virtual ~AudioGain() {}
 
     void setMode(audio_gain_mode_t mode) { mGain.mode = mode; }
@@ -80,7 +80,7 @@
 private:
     int               mIndex;
     struct audio_gain mGain;
-    bool              mUseInChannelMask;
+    bool              mIsInput;
     bool              mUseForVolume = false;
 };
 
diff --git a/media/libaudiofoundation/include/media/AudioPort.h b/media/libaudiofoundation/include/media/AudioPort.h
index 1cee1c9..6e1d032 100644
--- a/media/libaudiofoundation/include/media/AudioPort.h
+++ b/media/libaudiofoundation/include/media/AudioPort.h
@@ -130,7 +130,7 @@
 };
 
 
-class AudioPortConfig : public virtual RefBase, public virtual Parcelable
+class AudioPortConfig : public virtual RefBase
 {
 public:
     virtual ~AudioPortConfig() = default;
@@ -152,10 +152,8 @@
 
     bool equals(const sp<AudioPortConfig>& other) const;
 
-    status_t writeToParcel(Parcel* parcel) const override;
-    status_t readFromParcel(const Parcel* parcel) override;
-    status_t writeToParcelable(media::AudioPortConfig* parcelable) const;
-    status_t readFromParcelable(const media::AudioPortConfig& parcelable);
+    status_t writeToParcelable(media::AudioPortConfig* parcelable, bool isInput) const;
+    status_t readFromParcelable(const media::AudioPortConfig& parcelable, bool isInput);
 
 protected:
     unsigned int mSamplingRate = 0u;
diff --git a/media/libaudiofoundation/include/media/AudioProfile.h b/media/libaudiofoundation/include/media/AudioProfile.h
index 6a36e78..e34a49f 100644
--- a/media/libaudiofoundation/include/media/AudioProfile.h
+++ b/media/libaudiofoundation/include/media/AudioProfile.h
@@ -29,7 +29,7 @@
 
 namespace android {
 
-class AudioProfile final : public RefBase, public Parcelable
+class AudioProfile final : public RefBase
 {
 public:
     static sp<AudioProfile> createFullDynamic(audio_format_t dynamicFormat = AUDIO_FORMAT_DEFAULT);
@@ -81,11 +81,9 @@
 
     bool equals(const sp<AudioProfile>& other) const;
 
-    status_t writeToParcel(Parcel* parcel) const override;
-    status_t readFromParcel(const Parcel* parcel) override;
-
-    ConversionResult<media::AudioProfile> toParcelable() const;
-    static ConversionResult<sp<AudioProfile>> fromParcelable(const media::AudioProfile& parcelable);
+    ConversionResult<media::AudioProfile> toParcelable(bool isInput) const;
+    static ConversionResult<sp<AudioProfile>> fromParcelable(
+            const media::AudioProfile& parcelable, bool isInput);
 
 private:
 
@@ -106,11 +104,11 @@
 
 // Conversion routines, according to AidlConversion.h conventions.
 ConversionResult<sp<AudioProfile>>
-aidl2legacy_AudioProfile(const media::AudioProfile& aidl);
+aidl2legacy_AudioProfile(const media::AudioProfile& aidl, bool isInput);
 ConversionResult<media::AudioProfile>
-legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy);
+legacy2aidl_AudioProfile(const sp<AudioProfile>& legacy, bool isInput);
 
-class AudioProfileVector : public std::vector<sp<AudioProfile>>, public Parcelable
+class AudioProfileVector : public std::vector<sp<AudioProfile>>
 {
 public:
     virtual ~AudioProfileVector() = default;
@@ -136,18 +134,15 @@
     virtual void dump(std::string *dst, int spaces) const;
 
     bool equals(const AudioProfileVector& other) const;
-
-    status_t writeToParcel(Parcel* parcel) const override;
-    status_t readFromParcel(const Parcel* parcel) override;
 };
 
 bool operator == (const AudioProfile &left, const AudioProfile &right);
 
 // Conversion routines, according to AidlConversion.h conventions.
 ConversionResult<AudioProfileVector>
-aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl);
+aidl2legacy_AudioProfileVector(const std::vector<media::AudioProfile>& aidl, bool isInput);
 ConversionResult<std::vector<media::AudioProfile>>
-legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy);
+legacy2aidl_AudioProfileVector(const AudioProfileVector& legacy, bool isInput);
 
 AudioProfileVector intersectAudioProfiles(const AudioProfileVector& profiles1,
                                           const AudioProfileVector& profiles2);
diff --git a/media/libaudiofoundation/tests/Android.bp b/media/libaudiofoundation/tests/Android.bp
index bb9a5f2..f3cd446 100644
--- a/media/libaudiofoundation/tests/Android.bp
+++ b/media/libaudiofoundation/tests/Android.bp
@@ -11,12 +11,19 @@
     name: "audiofoundation_parcelable_test",
 
     shared_libs: [
-        "libaudiofoundation",
+        "libbase",
         "libbinder",
         "liblog",
         "libutils",
     ],
 
+    static_libs: [
+        "audioclient-types-aidl-cpp",
+        "libaudioclient_aidl_conversion",
+        "libaudiofoundation",
+        "libstagefright_foundation",
+    ],
+
     header_libs: [
         "libaudio_system_headers",
     ],
diff --git a/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp b/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
index 068b5d8..1696980 100644
--- a/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
+++ b/media/libaudiofoundation/tests/audiofoundation_parcelable_test.cpp
@@ -53,7 +53,7 @@
 
 AudioGains getAudioGainsForTest() {
     AudioGains audioGains;
-    sp<AudioGain> audioGain = new AudioGain(0 /*index*/, false /*useInChannelMask*/);
+    sp<AudioGain> audioGain = new AudioGain(0 /*index*/, false /*isInput*/);
     audioGain->setMode(AUDIO_GAIN_MODE_JOINT);
     audioGain->setChannelMask(AUDIO_CHANNEL_OUT_STEREO);
     audioGain->setMinValueInMb(-3200);
@@ -86,17 +86,6 @@
     ASSERT_TRUE(audioGainsFromParcel.equals(audioGains));
 }
 
-TEST(AudioFoundationParcelableTest, ParcelingAudioProfileVector) {
-    Parcel data;
-    AudioProfileVector audioProfiles = getAudioProfileVectorForTest();
-
-    ASSERT_EQ(data.writeParcelable(audioProfiles), NO_ERROR);
-    data.setDataPosition(0);
-    AudioProfileVector audioProfilesFromParcel;
-    ASSERT_EQ(data.readParcelable(&audioProfilesFromParcel), NO_ERROR);
-    ASSERT_TRUE(audioProfilesFromParcel.equals(audioProfiles));
-}
-
 TEST(AudioFoundationParcelableTest, ParcelingAudioPort) {
     Parcel data;
     sp<AudioPort> audioPort = new AudioPort(
@@ -116,11 +105,15 @@
     Parcel data;
     sp<AudioPortConfig> audioPortConfig = new AudioPortConfigTestStub();
     audioPortConfig->applyAudioPortConfig(&TEST_AUDIO_PORT_CONFIG);
-
-    ASSERT_EQ(data.writeParcelable(*audioPortConfig), NO_ERROR);
+    media::AudioPortConfig parcelable{};
+    ASSERT_EQ(NO_ERROR, audioPortConfig->writeToParcelable(&parcelable, false /*isInput*/));
+    ASSERT_EQ(NO_ERROR, data.writeParcelable(parcelable));
     data.setDataPosition(0);
+    media::AudioPortConfig parcelableFromParcel{};
+    ASSERT_EQ(NO_ERROR, data.readParcelable(&parcelableFromParcel));
     sp<AudioPortConfig> audioPortConfigFromParcel = new AudioPortConfigTestStub();
-    ASSERT_EQ(data.readParcelable(audioPortConfigFromParcel.get()), NO_ERROR);
+    ASSERT_EQ(NO_ERROR, audioPortConfigFromParcel->readFromParcelable(
+                    parcelableFromParcel, false /*isInput*/));
     ASSERT_TRUE(audioPortConfigFromParcel->equals(audioPortConfig));
 }
 
diff --git a/media/libaudiohal/FactoryHalHidl.cpp b/media/libaudiohal/FactoryHalHidl.cpp
index e420d07..c19d2c2 100644
--- a/media/libaudiohal/FactoryHalHidl.cpp
+++ b/media/libaudiohal/FactoryHalHidl.cpp
@@ -94,7 +94,7 @@
 }  // namespace
 
 void* createPreferredImpl(const std::string& package, const std::string& interface) {
-    for (auto version = detail::sAudioHALVersions; version != nullptr; ++version) {
+    for (auto version = detail::sAudioHALVersions; *version != nullptr; ++version) {
         void* rawInterface = nullptr;
         if (hasHalService(package, *version, interface)
                 && createHalService(*version, interface, &rawInterface)) {
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.cpp b/media/libaudiohal/impl/ConversionHelperHidl.cpp
index 32eaa31..0503698 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.cpp
+++ b/media/libaudiohal/impl/ConversionHelperHidl.cpp
@@ -105,6 +105,15 @@
 }
 
 // static
+void ConversionHelperHidl::argsFromHal(
+        const Vector<String16>& args, hidl_vec<hidl_string> *hidlArgs) {
+    hidlArgs->resize(args.size());
+    for (size_t i = 0; i < args.size(); ++i) {
+        (*hidlArgs)[i] = String8(args[i]).c_str();
+    }
+}
+
+// static
 status_t ConversionHelperHidl::analyzeResult(const Result& result) {
     switch (result) {
         case Result::OK: return OK;
diff --git a/media/libaudiohal/impl/ConversionHelperHidl.h b/media/libaudiohal/impl/ConversionHelperHidl.h
index 59122c7..2909013 100644
--- a/media/libaudiohal/impl/ConversionHelperHidl.h
+++ b/media/libaudiohal/impl/ConversionHelperHidl.h
@@ -21,6 +21,8 @@
 #include <hidl/HidlSupport.h>
 #include <system/audio.h>
 #include <utils/String8.h>
+#include <utils/String16.h>
+#include <utils/Vector.h>
 
 using ::android::hardware::audio::CPP_VERSION::ParameterValue;
 using CoreResult = ::android::hardware::audio::CPP_VERSION::Result;
@@ -37,6 +39,7 @@
     static status_t keysFromHal(const String8& keys, hidl_vec<hidl_string> *hidlKeys);
     static status_t parametersFromHal(const String8& kvPairs, hidl_vec<ParameterValue> *hidlParams);
     static void parametersToHal(const hidl_vec<ParameterValue>& parameters, String8 *values);
+    static void argsFromHal(const Vector<String16>& args, hidl_vec<hidl_string> *hidlArgs);
 
     ConversionHelperHidl(const char* className);
 
diff --git a/media/libaudiohal/impl/DeviceHalHidl.cpp b/media/libaudiohal/impl/DeviceHalHidl.cpp
index ca4f663..f86df1e 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.cpp
+++ b/media/libaudiohal/impl/DeviceHalHidl.cpp
@@ -457,11 +457,13 @@
 }
 #endif
 
-status_t DeviceHalHidl::dump(int fd) {
+status_t DeviceHalHidl::dump(int fd, const Vector<String16>& args) {
     if (mDevice == 0) return NO_INIT;
     native_handle_t* hidlHandle = native_handle_create(1, 0);
     hidlHandle->data[0] = fd;
-    Return<void> ret = mDevice->debug(hidlHandle, {} /* options */);
+    hidl_vec<hidl_string> hidlArgs;
+    argsFromHal(args, &hidlArgs);
+    Return<void> ret = mDevice->debug(hidlHandle, hidlArgs);
     native_handle_delete(hidlHandle);
     return processReturn("dump", ret);
 }
diff --git a/media/libaudiohal/impl/DeviceHalHidl.h b/media/libaudiohal/impl/DeviceHalHidl.h
index 2c847cf..2694ab3 100644
--- a/media/libaudiohal/impl/DeviceHalHidl.h
+++ b/media/libaudiohal/impl/DeviceHalHidl.h
@@ -119,7 +119,7 @@
     status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
     status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
 
-    virtual status_t dump(int fd);
+    status_t dump(int fd, const Vector<String16>& args) override;
 
   private:
     friend class DevicesFactoryHalHidl;
diff --git a/media/libaudiohal/impl/DeviceHalLocal.cpp b/media/libaudiohal/impl/DeviceHalLocal.cpp
index af7dc1a..e0304af 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.cpp
+++ b/media/libaudiohal/impl/DeviceHalLocal.cpp
@@ -233,7 +233,7 @@
     return INVALID_OPERATION;
 }
 
-status_t DeviceHalLocal::dump(int fd) {
+status_t DeviceHalLocal::dump(int fd, const Vector<String16>& /* args */) {
     return mDev->dump(mDev, fd);
 }
 
diff --git a/media/libaudiohal/impl/DeviceHalLocal.h b/media/libaudiohal/impl/DeviceHalLocal.h
index 46b510b..2fde936 100644
--- a/media/libaudiohal/impl/DeviceHalLocal.h
+++ b/media/libaudiohal/impl/DeviceHalLocal.h
@@ -112,7 +112,7 @@
     status_t addDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
     status_t removeDeviceEffect(audio_port_handle_t device, sp<EffectHalInterface> effect) override;
 
-    virtual status_t dump(int fd);
+    status_t dump(int fd, const Vector<String16>& args) override;
 
     void closeOutputStream(struct audio_stream_out *stream_out);
     void closeInputStream(struct audio_stream_in *stream_in);
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 9c4363c..f75bbf3 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -73,7 +73,9 @@
         uint32_t index, effect_descriptor_t *pDescriptor) {
     // TODO: We need somehow to track the changes on the server side
     // or figure out how to convert everybody to query all the descriptors at once.
-    // TODO: check for nullptr
+    if (pDescriptor == nullptr) {
+        return BAD_VALUE;
+    }
     if (mLastDescriptors.size() == 0) {
         status_t queryResult = queryAllDescriptors();
         if (queryResult != OK) return queryResult;
@@ -85,7 +87,9 @@
 
 status_t EffectsFactoryHalHidl::getDescriptor(
         const effect_uuid_t *pEffectUuid, effect_descriptor_t *pDescriptor) {
-    // TODO: check for nullptr
+    if (pDescriptor == nullptr || pEffectUuid == nullptr) {
+        return BAD_VALUE;
+    }
     if (mEffectsFactory == 0) return NO_INIT;
     Uuid hidlUuid;
     UuidUtils::uuidFromHal(*pEffectUuid, &hidlUuid);
@@ -105,6 +109,33 @@
     return processReturn(__FUNCTION__, ret);
 }
 
+status_t EffectsFactoryHalHidl::getDescriptors(const effect_uuid_t *pEffectType,
+                                               std::vector<effect_descriptor_t> *descriptors) {
+    if (pEffectType == nullptr || descriptors == nullptr) {
+        return BAD_VALUE;
+    }
+
+    uint32_t numEffects = 0;
+    status_t status = queryNumberEffects(&numEffects);
+    if (status != NO_ERROR) {
+        ALOGW("%s error %d from FactoryHal queryNumberEffects", __func__, status);
+        return status;
+    }
+
+    for (uint32_t i = 0; i < numEffects; i++) {
+        effect_descriptor_t descriptor;
+        status = getDescriptor(i, &descriptor);
+        if (status != NO_ERROR) {
+            ALOGW("%s error %d from FactoryHal getDescriptor", __func__, status);
+            continue;
+        }
+        if (memcmp(&descriptor.type, pEffectType, sizeof(effect_uuid_t)) == 0) {
+            descriptors->push_back(descriptor);
+        }
+    }
+    return descriptors->empty() ? NAME_NOT_FOUND : NO_ERROR;
+}
+
 status_t EffectsFactoryHalHidl::createEffect(
         const effect_uuid_t *pEffectUuid, int32_t sessionId, int32_t ioId,
         int32_t deviceId __unused, sp<EffectHalInterface> *effect) {
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 5fa85e7..ff26d9f 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -45,6 +45,9 @@
     virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
             effect_descriptor_t *pDescriptor);
 
+    virtual status_t getDescriptors(const effect_uuid_t *pEffectType,
+                                    std::vector<effect_descriptor_t> *descriptors);
+
     // Creates an effect engine of the specified type.
     // To release the effect engine, it is necessary to release references
     // to the returned effect object.
diff --git a/media/libaudiohal/impl/StreamHalHidl.cpp b/media/libaudiohal/impl/StreamHalHidl.cpp
index 539a149..452f84d 100644
--- a/media/libaudiohal/impl/StreamHalHidl.cpp
+++ b/media/libaudiohal/impl/StreamHalHidl.cpp
@@ -152,11 +152,13 @@
     return processReturn("standby", mStream->standby());
 }
 
-status_t StreamHalHidl::dump(int fd) {
+status_t StreamHalHidl::dump(int fd, const Vector<String16>& args) {
     if (!mStream) return NO_INIT;
     native_handle_t* hidlHandle = native_handle_create(1, 0);
     hidlHandle->data[0] = fd;
-    Return<void> ret = mStream->debug(hidlHandle, {} /* options */);
+    hidl_vec<hidl_string> hidlArgs;
+    argsFromHal(args, &hidlArgs);
+    Return<void> ret = mStream->debug(hidlHandle, hidlArgs);
     native_handle_delete(hidlHandle);
     mStreamPowerLog.dump(fd);
     return processReturn("dump", ret);
diff --git a/media/libaudiohal/impl/StreamHalHidl.h b/media/libaudiohal/impl/StreamHalHidl.h
index 970903b..6f5dd04 100644
--- a/media/libaudiohal/impl/StreamHalHidl.h
+++ b/media/libaudiohal/impl/StreamHalHidl.h
@@ -71,7 +71,7 @@
     // Put the audio hardware input/output into standby mode.
     virtual status_t standby();
 
-    virtual status_t dump(int fd);
+    virtual status_t dump(int fd, const Vector<String16>& args) override;
 
     // Start a stream operating in mmap mode.
     virtual status_t start();
diff --git a/media/libaudiohal/impl/StreamHalLocal.cpp b/media/libaudiohal/impl/StreamHalLocal.cpp
index 34bd5df..11fac61 100644
--- a/media/libaudiohal/impl/StreamHalLocal.cpp
+++ b/media/libaudiohal/impl/StreamHalLocal.cpp
@@ -87,7 +87,8 @@
     return mStream->standby(mStream);
 }
 
-status_t StreamHalLocal::dump(int fd) {
+status_t StreamHalLocal::dump(int fd, const Vector<String16>& args) {
+    (void) args;
     status_t status = mStream->dump(mStream, fd);
     mStreamPowerLog.dump(fd);
     return status;
diff --git a/media/libaudiohal/impl/StreamHalLocal.h b/media/libaudiohal/impl/StreamHalLocal.h
index b260495..493c521 100644
--- a/media/libaudiohal/impl/StreamHalLocal.h
+++ b/media/libaudiohal/impl/StreamHalLocal.h
@@ -50,7 +50,7 @@
     // Put the audio hardware input/output into standby mode.
     virtual status_t standby();
 
-    virtual status_t dump(int fd);
+    virtual status_t dump(int fd, const Vector<String16>& args) override;
 
     // Start a stream operating in mmap mode.
     virtual status_t start() = 0;
diff --git a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
index 29ef011..69cbcec 100644
--- a/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/DeviceHalInterface.h
@@ -120,7 +120,7 @@
     virtual status_t removeDeviceEffect(
             audio_port_handle_t device, sp<EffectHalInterface> effect) = 0;
 
-    virtual status_t dump(int fd) = 0;
+    virtual status_t dump(int fd, const Vector<String16>& args) = 0;
 
   protected:
     // Subclasses can not be constructed directly by clients.
diff --git a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
index 9fb56ae..3e505bd 100644
--- a/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/EffectsFactoryHalInterface.h
@@ -37,6 +37,9 @@
     virtual status_t getDescriptor(const effect_uuid_t *pEffectUuid,
             effect_descriptor_t *pDescriptor) = 0;
 
+    virtual status_t getDescriptors(const effect_uuid_t *pEffectType,
+                                    std::vector<effect_descriptor_t> *descriptors) = 0;
+
     // Creates an effect engine of the specified type.
     // To release the effect engine, it is necessary to release references
     // to the returned effect object.
diff --git a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
index 2be12fb..2b5b2db 100644
--- a/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
+++ b/media/libaudiohal/include/media/audiohal/StreamHalInterface.h
@@ -25,6 +25,7 @@
 #include <utils/Errors.h>
 #include <utils/RefBase.h>
 #include <utils/String8.h>
+#include <utils/Vector.h>
 
 namespace android {
 
@@ -69,7 +70,7 @@
     // Put the audio hardware input/output into standby mode.
     virtual status_t standby() = 0;
 
-    virtual status_t dump(int fd) = 0;
+    virtual status_t dump(int fd, const Vector<String16>& args = {}) = 0;
 
     // Start a stream operating in mmap mode.
     virtual status_t start() = 0;
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index d85e2e9..e68c002 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -434,6 +434,12 @@
                 track->mHapticIntensity = hapticIntensity;
             }
             } break;
+        case HAPTIC_MAX_AMPLITUDE: {
+            const float hapticMaxAmplitude = *reinterpret_cast<float*>(value);
+            if (track->mHapticMaxAmplitude != hapticMaxAmplitude) {
+                track->mHapticMaxAmplitude = hapticMaxAmplitude;
+            }
+            } break;
         default:
             LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
         }
@@ -553,6 +559,7 @@
     // haptic
     t->mHapticPlaybackEnabled = false;
     t->mHapticIntensity = os::HapticScale::NONE;
+    t->mHapticMaxAmplitude = NAN;
     t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
     t->mMixerHapticChannelCount = 0;
     t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
@@ -602,7 +609,8 @@
                 switch (t->mMixerFormat) {
                 // Mixer format should be AUDIO_FORMAT_PCM_FLOAT.
                 case AUDIO_FORMAT_PCM_FLOAT: {
-                    os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity);
+                    os::scaleHapticData((float*) buffer, sampleCount, t->mHapticIntensity,
+                                        t->mHapticMaxAmplitude);
                 } break;
                 default:
                     LOG_ALWAYS_FATAL("bad mMixerFormat: %#x", t->mMixerFormat);
diff --git a/media/libaudioprocessing/AudioMixerOps.h b/media/libaudioprocessing/AudioMixerOps.h
index cd47dc6..2988c67 100644
--- a/media/libaudioprocessing/AudioMixerOps.h
+++ b/media/libaudioprocessing/AudioMixerOps.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_MIXER_OPS_H
 #define ANDROID_AUDIO_MIXER_OPS_H
 
+#include <audio_utils/channels.h>
+#include <audio_utils/primitives.h>
 #include <system/audio.h>
 
 namespace android {
@@ -229,15 +231,26 @@
  * complexity of working on interleaved streams is now getting
  * too high, and likely limits compiler optimization.
  */
-template <int MIXTYPE, int NCHAN,
+
+// compile-time function.
+constexpr inline bool usesCenterChannel(audio_channel_mask_t mask) {
+    using namespace audio_utils::channels;
+    for (size_t i = 0; i < std::size(kSideFromChannelIdx); ++i) {
+        if ((mask & (1 << i)) != 0 && kSideFromChannelIdx[i] == AUDIO_GEOMETRY_SIDE_CENTER) {
+            return true;
+        }
+    }
+    return false;
+}
+
+/*
+ * Applies stereo volume to the audio data based on proper left right channel affinity
+ * (templated channel MASK parameter).
+ */
+template <int MIXTYPE, audio_channel_mask_t MASK,
         typename TO, typename TI, typename TV,
         typename F>
-void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
-    static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
-    static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
-            || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
-            || MIXTYPE == MIXTYPE_STEREOEXPAND
-            || MIXTYPE == MIXTYPE_MONOEXPAND);
+void stereoVolumeHelperWithChannelMask(TO*& out, const TI*& in, const TV *vol, F f) {
     auto proc = [](auto& a, const auto& b) {
         if constexpr (MIXTYPE == MIXTYPE_MULTI_STEREOVOL
                 || MIXTYPE == MIXTYPE_STEREOEXPAND
@@ -250,59 +263,109 @@
     auto inp = [&in]() -> const TI& {
         if constexpr (MIXTYPE == MIXTYPE_STEREOEXPAND
                 || MIXTYPE == MIXTYPE_MONOEXPAND) {
-            return *in;
+            return *in; // note STEREOEXPAND assumes replicated L/R channels (see doc below).
         } else {
             return *in++;
         }
     };
 
-    // HALs should only expose the canonical channel masks.
-    proc(*out++, f(inp(), vol[0])); // front left
-    if constexpr (NCHAN == 1) return;
-    proc(*out++, f(inp(), vol[1])); // front right
-    if constexpr (NCHAN == 2)  return;
-    if constexpr (NCHAN == 4) {
-        proc(*out++, f(inp(), vol[0])); // back left
-        proc(*out++, f(inp(), vol[1])); // back right
-        return;
-    }
-
-    // TODO: Precompute center volume if not ramping.
     std::decay_t<TV> center;
-    if constexpr (std::is_floating_point_v<TV>) {
-        center = (vol[0] + vol[1]) * 0.5;       // do not use divide
-    } else {
-        center = (vol[0] >> 1) + (vol[1] >> 1); // rounds to 0.
-    }
-    proc(*out++, f(inp(), center)); // center (or 2.1 LFE)
-    if constexpr (NCHAN == 3) return;
-    if constexpr (NCHAN == 5) {
-        proc(*out++, f(inp(), vol[0]));  // back left
-        proc(*out++, f(inp(), vol[1]));  // back right
-        return;
-    }
-
-    proc(*out++, f(inp(), center)); // lfe
-    proc(*out++, f(inp(), vol[0])); // back left
-    proc(*out++, f(inp(), vol[1])); // back right
-    if constexpr (NCHAN == 6) return;
-    if constexpr (NCHAN == 7) {
-        proc(*out++, f(inp(), center)); // back center
-        return;
-    }
-    // NCHAN == 8
-    proc(*out++, f(inp(), vol[0])); // side left
-    proc(*out++, f(inp(), vol[1])); // side right
-    if constexpr (NCHAN > FCC_8) {
-        // Mutes to zero extended surround channels.
-        // 7.1.4 has the correct behavior.
-        // 22.2 has the behavior that FLC and FRC will be mixed instead
-        // of SL and SR and LFE will be center, not left.
-        for (int i = 8; i < NCHAN; ++i) {
-            // TODO: Consider using android::audio_utils::channels::kSideFromChannelIdx
-            proc(*out++, f(inp(), 0.f));
+    constexpr bool USES_CENTER_CHANNEL = usesCenterChannel(MASK);
+    if constexpr (USES_CENTER_CHANNEL) {
+        if constexpr (std::is_floating_point_v<TV>) {
+            center = (vol[0] + vol[1]) * 0.5;       // do not use divide
+        } else {
+            center = (vol[0] >> 1) + (vol[1] >> 1); // rounds to 0.
         }
     }
+
+    using namespace audio_utils::channels;
+
+    // if LFE and LFE2 are both present, they take left and right volume respectively.
+    constexpr unsigned LFE_LFE2 = \
+             AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2;
+    constexpr bool has_LFE_LFE2 = (MASK & LFE_LFE2) == LFE_LFE2;
+
+#pragma push_macro("DO_CHANNEL_POSITION")
+#undef DO_CHANNEL_POSITION
+#define DO_CHANNEL_POSITION(BIT_INDEX) \
+    if constexpr ((MASK & (1 << BIT_INDEX)) != 0) { \
+        constexpr auto side = kSideFromChannelIdx[BIT_INDEX]; \
+        if constexpr (side == AUDIO_GEOMETRY_SIDE_LEFT || \
+               has_LFE_LFE2 && (1 << BIT_INDEX) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) { \
+            proc(*out++, f(inp(), vol[0])); \
+        } else if constexpr (side == AUDIO_GEOMETRY_SIDE_RIGHT || \
+               has_LFE_LFE2 && (1 << BIT_INDEX) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) { \
+            proc(*out++, f(inp(), vol[1])); \
+        } else /* constexpr */ { \
+            proc(*out++, f(inp(), center)); \
+        } \
+    }
+
+    DO_CHANNEL_POSITION(0);
+    DO_CHANNEL_POSITION(1);
+    DO_CHANNEL_POSITION(2);
+    DO_CHANNEL_POSITION(3);
+    DO_CHANNEL_POSITION(4);
+    DO_CHANNEL_POSITION(5);
+    DO_CHANNEL_POSITION(6);
+    DO_CHANNEL_POSITION(7);
+
+    DO_CHANNEL_POSITION(8);
+    DO_CHANNEL_POSITION(9);
+    DO_CHANNEL_POSITION(10);
+    DO_CHANNEL_POSITION(11);
+    DO_CHANNEL_POSITION(12);
+    DO_CHANNEL_POSITION(13);
+    DO_CHANNEL_POSITION(14);
+    DO_CHANNEL_POSITION(15);
+
+    DO_CHANNEL_POSITION(16);
+    DO_CHANNEL_POSITION(17);
+    DO_CHANNEL_POSITION(18);
+    DO_CHANNEL_POSITION(19);
+    DO_CHANNEL_POSITION(20);
+    DO_CHANNEL_POSITION(21);
+    DO_CHANNEL_POSITION(22);
+    DO_CHANNEL_POSITION(23);
+    static_assert(FCC_LIMIT <= FCC_24); // Note: this may need to change.
+#pragma pop_macro("DO_CHANNEL_POSITION")
+}
+
+// These are the channel position masks we expect from the HAL.
+// See audio_channel_out_mask_from_count() but this is constexpr
+constexpr inline audio_channel_mask_t canonicalChannelMaskFromCount(size_t channelCount) {
+    constexpr audio_channel_mask_t canonical[] = {
+        [0] = AUDIO_CHANNEL_NONE,
+        [1] = AUDIO_CHANNEL_OUT_MONO,
+        [2] = AUDIO_CHANNEL_OUT_STEREO,
+        [3] = AUDIO_CHANNEL_OUT_2POINT1,
+        [4] = AUDIO_CHANNEL_OUT_QUAD,
+        [5] = AUDIO_CHANNEL_OUT_PENTA,
+        [6] = AUDIO_CHANNEL_OUT_5POINT1,
+        [7] = AUDIO_CHANNEL_OUT_6POINT1,
+        [8] = AUDIO_CHANNEL_OUT_7POINT1,
+        [12] = AUDIO_CHANNEL_OUT_7POINT1POINT4,
+        [24] = AUDIO_CHANNEL_OUT_22POINT2,
+    };
+    return channelCount < std::size(canonical) ? canonical[channelCount] : AUDIO_CHANNEL_NONE;
+}
+
+template <int MIXTYPE, int NCHAN,
+        typename TO, typename TI, typename TV,
+        typename F>
+void stereoVolumeHelper(TO*& out, const TI*& in, const TV *vol, F f) {
+    static_assert(NCHAN > 0 && NCHAN <= FCC_LIMIT);
+    static_assert(MIXTYPE == MIXTYPE_MULTI_STEREOVOL
+            || MIXTYPE == MIXTYPE_MULTI_SAVEONLY_STEREOVOL
+            || MIXTYPE == MIXTYPE_STEREOEXPAND
+            || MIXTYPE == MIXTYPE_MONOEXPAND);
+    constexpr audio_channel_mask_t MASK{canonicalChannelMaskFromCount(NCHAN)};
+    if constexpr (MASK == AUDIO_CHANNEL_NONE) {
+        ALOGE("%s: Invalid position count %d", __func__, NCHAN);
+        return; // not a valid system mask, ignore.
+    }
+    stereoVolumeHelperWithChannelMask<MIXTYPE, MASK, TO, TI, TV, F>(out, in, vol, f);
 }
 
 /*
diff --git a/media/libaudioprocessing/include/media/AudioMixer.h b/media/libaudioprocessing/include/media/AudioMixer.h
index 70eafe3..5a9fa07 100644
--- a/media/libaudioprocessing/include/media/AudioMixer.h
+++ b/media/libaudioprocessing/include/media/AudioMixer.h
@@ -50,6 +50,7 @@
         // for haptic
         HAPTIC_ENABLED  = 0x4007, // Set haptic data from this track should be played or not.
         HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
+        HAPTIC_MAX_AMPLITUDE = 0x4009, // Set the max amplitude allowed for haptic data.
         // for target TIMESTRETCH
         PLAYBACK_RATE   = 0x4300, // Configure timestretch on this track name;
                                   // parameter 'value' is a pointer to the new playback rate.
@@ -145,6 +146,7 @@
         // Haptic
         bool                 mHapticPlaybackEnabled;
         os::HapticScale      mHapticIntensity;
+        float                mHapticMaxAmplitude;
         audio_channel_mask_t mHapticChannelMask;
         uint32_t             mHapticChannelCount;
         audio_channel_mask_t mMixerHapticChannelMask;
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index 3856817..ad402db 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -76,6 +76,7 @@
 //
 cc_binary {
     name: "mixerops_objdump",
+    header_libs: ["libaudioutils_headers"],
     srcs: ["mixerops_objdump.cpp"],
 }
 
@@ -84,6 +85,16 @@
 //
 cc_benchmark {
     name: "mixerops_benchmark",
+    header_libs: ["libaudioutils_headers"],
     srcs: ["mixerops_benchmark.cpp"],
     static_libs: ["libgoogle-benchmark"],
 }
+
+//
+// mixerops unit test
+//
+cc_test {
+    name: "mixerops_tests",
+    defaults: ["libaudioprocessing_test_defaults"],
+    srcs: ["mixerops_tests.cpp"],
+}
diff --git a/media/libaudioprocessing/tests/mixerops_benchmark.cpp b/media/libaudioprocessing/tests/mixerops_benchmark.cpp
index 7a4c5c7..f866b1a 100644
--- a/media/libaudioprocessing/tests/mixerops_benchmark.cpp
+++ b/media/libaudioprocessing/tests/mixerops_benchmark.cpp
@@ -16,11 +16,9 @@
 
 #include <inttypes.h>
 #include <type_traits>
-#include "../../../../system/media/audio_utils/include/audio_utils/primitives.h"
 #define LOG_ALWAYS_FATAL(...)
 
 #include <../AudioMixerOps.h>
-
 #include <benchmark/benchmark.h>
 
 using namespace android;
diff --git a/media/libaudioprocessing/tests/mixerops_tests.cpp b/media/libaudioprocessing/tests/mixerops_tests.cpp
new file mode 100644
index 0000000..2500ba9
--- /dev/null
+++ b/media/libaudioprocessing/tests/mixerops_tests.cpp
@@ -0,0 +1,175 @@
+/*
+ * 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 "mixerop_tests"
+#include <log/log.h>
+
+#include <inttypes.h>
+#include <type_traits>
+
+#include <../AudioMixerOps.h>
+#include <gtest/gtest.h>
+
+using namespace android;
+
+// Note: gtest templated tests require typenames, not integers.
+template <int MIXTYPE, int NCHAN>
+class MixerOpsBasicTest {
+public:
+    static void testStereoVolume() {
+        using namespace android::audio_utils::channels;
+
+        constexpr size_t FRAME_COUNT = 1000;
+        constexpr size_t SAMPLE_COUNT = FRAME_COUNT * NCHAN;
+
+        const float in[SAMPLE_COUNT] = {[0 ... (SAMPLE_COUNT - 1)] = 1.f};
+
+        AUDIO_GEOMETRY_SIDE sides[NCHAN];
+        size_t i = 0;
+        unsigned channel = canonicalChannelMaskFromCount(NCHAN);
+        constexpr unsigned LFE_LFE2 =
+                AUDIO_CHANNEL_OUT_LOW_FREQUENCY | AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2;
+        bool has_LFE_LFE2 = (channel & LFE_LFE2) == LFE_LFE2;
+        while (channel != 0) {
+            const int index = __builtin_ctz(channel);
+            if (has_LFE_LFE2 && (1 << index) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY) {
+                sides[i++] = AUDIO_GEOMETRY_SIDE_LEFT; // special case
+            } else if (has_LFE_LFE2 && (1 << index) == AUDIO_CHANNEL_OUT_LOW_FREQUENCY_2) {
+                sides[i++] = AUDIO_GEOMETRY_SIDE_RIGHT; // special case
+            } else {
+                sides[i++] = sideFromChannelIdx(index);
+            }
+            channel &= ~(1 << index);
+        }
+
+        float vola[2] = {1.f, 0.f}; // left volume at max.
+        float out[SAMPLE_COUNT]{};
+        float aux[FRAME_COUNT]{};
+        float volaux = 0.5;
+        {
+            volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, vola, volaux);
+            const float *outp = out;
+            const float *auxp = aux;
+            const float left = vola[0];
+            const float center = (vola[0] + vola[1]) * 0.5;
+            const float right = vola[1];
+            for (size_t i = 0; i < FRAME_COUNT; ++i) {
+                for (size_t j = 0; j < NCHAN; ++j) {
+                    const float audio = *outp++;
+                    if (sides[j] == AUDIO_GEOMETRY_SIDE_LEFT) {
+                        EXPECT_EQ(left, audio);
+                    } else if (sides[j] == AUDIO_GEOMETRY_SIDE_CENTER) {
+                        EXPECT_EQ(center, audio);
+                    } else {
+                        EXPECT_EQ(right, audio);
+                    }
+                }
+                EXPECT_EQ(volaux, *auxp++);  // works if all channels contain 1.f
+            }
+        }
+        float volb[2] = {0.f, 0.5f}; // right volume at half max.
+        {
+            // this accumulates into out, aux.
+            // float out[SAMPLE_COUNT]{};
+            // float aux[FRAME_COUNT]{};
+            volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, volb, volaux);
+            const float *outp = out;
+            const float *auxp = aux;
+            const float left = vola[0] + volb[0];
+            const float center = (vola[0] + vola[1] + volb[0] + volb[1]) * 0.5;
+            const float right = vola[1] + volb[1];
+            for (size_t i = 0; i < FRAME_COUNT; ++i) {
+                for (size_t j = 0; j < NCHAN; ++j) {
+                    const float audio = *outp++;
+                    if (sides[j] == AUDIO_GEOMETRY_SIDE_LEFT) {
+                        EXPECT_EQ(left, audio);
+                    } else if (sides[j] == AUDIO_GEOMETRY_SIDE_CENTER) {
+                        EXPECT_EQ(center, audio);
+                    } else {
+                        EXPECT_EQ(right, audio);
+                    }
+                }
+                // aux is accumulated so 2x the amplitude
+                EXPECT_EQ(volaux * 2.f, *auxp++);  // works if all channels contain 1.f
+            }
+        }
+
+        { // test aux as derived from out.
+            // AUX channel is the weighted sum of all of the output channels prior to volume
+            // adjustment.  We must set L and R to the same volume to allow computation
+            // of AUX from the output values.
+            const float volmono = 0.25f;
+            const float vollr[2] = {volmono, volmono}; // all the same.
+            float out[SAMPLE_COUNT]{};
+            float aux[FRAME_COUNT]{};
+            volumeMulti<MIXTYPE, NCHAN>(out, FRAME_COUNT, in, aux, vollr, volaux);
+            const float *outp = out;
+            const float *auxp = aux;
+            for (size_t i = 0; i < FRAME_COUNT; ++i) {
+                float accum = 0.f;
+                for (size_t j = 0; j < NCHAN; ++j) {
+                    accum += *outp++;
+                }
+                EXPECT_EQ(accum / NCHAN * volaux / volmono, *auxp++);
+            }
+        }
+    }
+};
+
+TEST(mixerops, stereovolume_1) { // Note: mono not used for output sinks yet.
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 1>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_2) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 2>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_3) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 3>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_4) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 4>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_5) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 5>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_6) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 6>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_7) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 7>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_8) {
+    MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 8>::testStereoVolume();
+}
+TEST(mixerops, stereovolume_12) {
+    if constexpr (FCC_LIMIT >= 12) { // NOTE: FCC_LIMIT is an enum, so can't #if
+        MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 12>::testStereoVolume();
+    }
+}
+TEST(mixerops, stereovolume_24) {
+    if constexpr (FCC_LIMIT >= 24) {
+        MixerOpsBasicTest<MIXTYPE_MULTI_STEREOVOL, 24>::testStereoVolume();
+    }
+}
+TEST(mixerops, channel_equivalence) {
+    // we must match the constexpr function with the system determined channel mask from count.
+    for (size_t i = 0; i < FCC_LIMIT; ++i) {
+        const audio_channel_mask_t actual = canonicalChannelMaskFromCount(i);
+        const audio_channel_mask_t system = audio_channel_out_mask_from_count(i);
+        if (system == AUDIO_CHANNEL_INVALID) continue;
+        EXPECT_EQ(system, actual);
+    }
+}
diff --git a/media/libeffects/hapticgenerator/Android.bp b/media/libeffects/hapticgenerator/Android.bp
index a660957..03ce329 100644
--- a/media/libeffects/hapticgenerator/Android.bp
+++ b/media/libeffects/hapticgenerator/Android.bp
@@ -45,6 +45,7 @@
 
     shared_libs: [
         "libaudioutils",
+        "libbase",
         "libbinder",
         "liblog",
         "libutils",
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
index 65a20a7..3137e13 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.cpp
@@ -22,12 +22,15 @@
 
 #include <algorithm>
 #include <memory>
+#include <string>
 #include <utility>
 
 #include <errno.h>
 #include <inttypes.h>
 #include <math.h>
 
+#include <android-base/parsedouble.h>
+#include <android-base/properties.h>
 #include <audio_effects/effect_hapticgenerator.h>
 #include <audio_utils/format.h>
 #include <system/audio.h>
@@ -35,6 +38,7 @@
 static constexpr float DEFAULT_RESONANT_FREQUENCY = 150.0f;
 static constexpr float DEFAULT_BSF_ZERO_Q = 8.0f;
 static constexpr float DEFAULT_BSF_POLE_Q = 4.0f;
+static constexpr float DEFAULT_DISTORTION_OUTPUT_GAIN = 1.5f;
 
 // This is the only symbol that needs to be exported
 __attribute__ ((visibility ("default")))
@@ -81,6 +85,15 @@
 
 namespace {
 
+float getFloatProperty(const std::string& key, float defaultValue) {
+    float result;
+    std::string value = android::base::GetProperty(key, "");
+    if (!value.empty() && android::base::ParseFloat(value, &result)) {
+        return result;
+    }
+    return defaultValue;
+}
+
 int HapticGenerator_Init(struct HapticGeneratorContext *context) {
     context->itfe = &gHapticGeneratorInterface;
 
@@ -114,7 +127,9 @@
     context->param.distortionCornerFrequency = 300.0f;
     context->param.distortionInputGain = 0.3f;
     context->param.distortionCubeThreshold = 0.1f;
-    context->param.distortionOutputGain = 1.5f;
+    context->param.distortionOutputGain = getFloatProperty(
+            "vendor.audio.hapticgenerator.distortion.output.gain", DEFAULT_DISTORTION_OUTPUT_GAIN);
+    ALOGD("Using distortion output gain as %f", context->param.distortionOutputGain);
 
     context->state = HAPTICGENERATOR_STATE_INITIALIZED;
     return 0;
@@ -287,15 +302,17 @@
         break;
     }
     case HG_PARAM_VIBRATOR_INFO: {
-        if (value == nullptr || size != 2 * sizeof(float)) {
+        if (value == nullptr || size != 3 * sizeof(float)) {
             return -EINVAL;
         }
         const float resonantFrequency = *(float*) value;
         const float qFactor = *((float *) value + 1);
+        const float maxAmplitude = *((float *) value + 2);
         context->param.resonantFrequency =
                 isnan(resonantFrequency) ? DEFAULT_RESONANT_FREQUENCY : resonantFrequency;
         context->param.bsfZeroQ = isnan(qFactor) ? DEFAULT_BSF_POLE_Q : qFactor;
         context->param.bsfPoleQ = context->param.bsfZeroQ / 2.0f;
+        context->param.maxHapticAmplitude = maxAmplitude;
 
         if (context->processorsRecord.bpf != nullptr) {
             context->processorsRecord.bpf->setCoefficients(
@@ -448,7 +465,8 @@
     float* hapticOutBuffer = HapticGenerator_runProcessingChain(
             context->processingChain, context->inputBuffer.data(),
             context->outputBuffer.data(), inBuffer->frameCount);
-    os::scaleHapticData(hapticOutBuffer, hapticSampleCount, context->param.maxHapticIntensity);
+    os::scaleHapticData(hapticOutBuffer, hapticSampleCount, context->param.maxHapticIntensity,
+                        context->param.maxHapticAmplitude);
 
     // For haptic data, the haptic playback thread will copy the data from effect input buffer,
     // which contains haptic data at the end of the buffer, directly to sink buffer.
diff --git a/media/libeffects/hapticgenerator/EffectHapticGenerator.h b/media/libeffects/hapticgenerator/EffectHapticGenerator.h
index 96b744a..85e961f 100644
--- a/media/libeffects/hapticgenerator/EffectHapticGenerator.h
+++ b/media/libeffects/hapticgenerator/EffectHapticGenerator.h
@@ -51,6 +51,7 @@
     // A map from track id to haptic intensity.
     std::map<int, os::HapticScale> id2Intensity;
     os::HapticScale maxHapticIntensity; // max intensity will be used to scale haptic data.
+    float maxHapticAmplitude; // max amplitude will be used to limit haptic data absolute values.
 
     float resonantFrequency;
     float bpfQ;
diff --git a/media/libmedia/IMediaPlayer.cpp b/media/libmedia/IMediaPlayer.cpp
index c89c023..c9f361e 100644
--- a/media/libmedia/IMediaPlayer.cpp
+++ b/media/libmedia/IMediaPlayer.cpp
@@ -17,7 +17,6 @@
 
 #include <arpa/inet.h>
 #include <stdint.h>
-#include <sys/types.h>
 
 #include <android/IDataSource.h>
 #include <binder/IPCThreadState.h>
diff --git a/media/libmedia/tests/codeclist/Android.bp b/media/libmedia/tests/codeclist/Android.bp
index 7dd0caa..57af9a9 100644
--- a/media/libmedia/tests/codeclist/Android.bp
+++ b/media/libmedia/tests/codeclist/Android.bp
@@ -25,7 +25,7 @@
 
 cc_test {
     name: "CodecListTest",
-    test_suites: ["device-tests"],
+    test_suites: ["device-tests", "mts"],
     gtest: true,
 
     srcs: [
@@ -41,7 +41,7 @@
         "libstagefright_xmlparser",
         "libutils",
     ],
-
+    compile_multilib: "first",
     cflags: [
         "-Werror",
         "-Wall",
diff --git a/media/libstagefright/FrameDecoder.cpp b/media/libstagefright/FrameDecoder.cpp
index efd4070..94a0424 100644
--- a/media/libstagefright/FrameDecoder.cpp
+++ b/media/libstagefright/FrameDecoder.cpp
@@ -262,13 +262,10 @@
 }
 
 bool isHDR(const sp<AMessage> &format) {
-    uint32_t standard, range, transfer;
+    uint32_t standard, transfer;
     if (!format->findInt32("color-standard", (int32_t*)&standard)) {
         standard = 0;
     }
-    if (!format->findInt32("color-range", (int32_t*)&range)) {
-        range = 0;
-    }
     if (!format->findInt32("color-transfer", (int32_t*)&transfer)) {
         transfer = 0;
     }
diff --git a/media/libstagefright/MediaCodec.cpp b/media/libstagefright/MediaCodec.cpp
index c03236a..a366506 100644
--- a/media/libstagefright/MediaCodec.cpp
+++ b/media/libstagefright/MediaCodec.cpp
@@ -4775,8 +4775,8 @@
     }
     const CryptoPlugin::SubSample *subSamples;
     size_t numSubSamples;
-    const uint8_t *key;
-    const uint8_t *iv;
+    const uint8_t *key = NULL;
+    const uint8_t *iv = NULL;
     CryptoPlugin::Mode mode = CryptoPlugin::kMode_Unencrypted;
 
     // We allow the simpler queueInputBuffer API to be used even in
@@ -4791,8 +4791,6 @@
 
             subSamples = &ss;
             numSubSamples = 1;
-            key = NULL;
-            iv = NULL;
             pattern.mEncryptBlocks = 0;
             pattern.mSkipBlocks = 0;
         }
diff --git a/media/libstagefright/foundation/MediaDefs.cpp b/media/libstagefright/foundation/MediaDefs.cpp
index ada5d81..0ec5ad5 100644
--- a/media/libstagefright/foundation/MediaDefs.cpp
+++ b/media/libstagefright/foundation/MediaDefs.cpp
@@ -65,7 +65,17 @@
 const char *MEDIA_MIMETYPE_AUDIO_WMA = "audio/x-ms-wma";
 const char *MEDIA_MIMETYPE_AUDIO_MS_ADPCM = "audio/x-adpcm-ms";
 const char *MEDIA_MIMETYPE_AUDIO_DVI_IMA_ADPCM = "audio/x-adpcm-dvi-ima";
-
+const char *MEDIA_MIMETYPE_AUDIO_DTS = "audio/vnd.dts";
+const char *MEDIA_MIMETYPE_AUDIO_DTS_HD = "audio/vnd.dts.hd";
+const char *MEDIA_MIMETYPE_AUDIO_DTS_UHD = "audio/vnd.dts.uhd";
+const char *MEDIA_MIMETYPE_AUDIO_EVRC = "audio/evrc";
+const char *MEDIA_MIMETYPE_AUDIO_EVRCB = "audio/evrcb";
+const char *MEDIA_MIMETYPE_AUDIO_EVRCWB = "audio/evrcwb";
+const char *MEDIA_MIMETYPE_AUDIO_EVRCNW = "audio/evrcnw";
+const char *MEDIA_MIMETYPE_AUDIO_AMR_WB_PLUS = "audio/amr-wb+";
+const char *MEDIA_MIMETYPE_AUDIO_APTX = "audio/aptx";
+const char *MEDIA_MIMETYPE_AUDIO_DRA = "audio/vnd.dra";
+const char *MEDIA_MIMETYPE_AUDIO_AAC_FORMAT = "audio/aac";
 
 const char *MEDIA_MIMETYPE_CONTAINER_MPEG4 = "video/mp4";
 const char *MEDIA_MIMETYPE_CONTAINER_WAV = "audio/x-wav";
diff --git a/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h b/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
index f5cecef..afa0c6d 100644
--- a/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
+++ b/media/libstagefright/foundation/include/media/stagefright/foundation/MediaDefs.h
@@ -67,7 +67,17 @@
 extern const char *MEDIA_MIMETYPE_AUDIO_WMA;
 extern const char *MEDIA_MIMETYPE_AUDIO_MS_ADPCM;
 extern const char *MEDIA_MIMETYPE_AUDIO_DVI_IMA_ADPCM;
-
+extern const char *MEDIA_MIMETYPE_AUDIO_DTS;
+extern const char *MEDIA_MIMETYPE_AUDIO_DTS_HD;
+extern const char *MEDIA_MIMETYPE_AUDIO_DTS_UHD;
+extern const char *MEDIA_MIMETYPE_AUDIO_EVRC;
+extern const char *MEDIA_MIMETYPE_AUDIO_EVRCB;
+extern const char *MEDIA_MIMETYPE_AUDIO_EVRCWB;
+extern const char *MEDIA_MIMETYPE_AUDIO_EVRCNW;
+extern const char *MEDIA_MIMETYPE_AUDIO_AMR_WB_PLUS;
+extern const char *MEDIA_MIMETYPE_AUDIO_APTX;
+extern const char *MEDIA_MIMETYPE_AUDIO_DRA;
+extern const char *MEDIA_MIMETYPE_AUDIO_AAC_FORMAT;
 
 extern const char *MEDIA_MIMETYPE_CONTAINER_MPEG4;
 extern const char *MEDIA_MIMETYPE_CONTAINER_WAV;
diff --git a/media/libstagefright/tests/fuzzers/Android.bp b/media/libstagefright/tests/fuzzers/Android.bp
index 0097830..ea17a4d 100644
--- a/media/libstagefright/tests/fuzzers/Android.bp
+++ b/media/libstagefright/tests/fuzzers/Android.bp
@@ -86,9 +86,6 @@
     dictionary: "dictionaries/formats.dict",
     defaults: ["libstagefright_fuzzer_defaults"],
     static_libs: [
-        "libstagefright_webm",
         "libdatasource",
-        "libstagefright_esds",
-        "libogg",
     ],
 }
diff --git a/media/ndk/NdkMediaFormat.cpp b/media/ndk/NdkMediaFormat.cpp
index c1793ce..b035e5a 100644
--- a/media/ndk/NdkMediaFormat.cpp
+++ b/media/ndk/NdkMediaFormat.cpp
@@ -351,6 +351,11 @@
 EXPORT const char* AMEDIAFORMAT_KEY_MIME = "mime";
 EXPORT const char* AMEDIAFORMAT_KEY_MPEG_USER_DATA = "mpeg-user-data";
 EXPORT const char* AMEDIAFORMAT_KEY_MPEG2_STREAM_HEADER = "mpeg2-stream-header";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS = "mpegh-compatible-sets";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION =
+        "mpegh-profile-level-indication";
+EXPORT const char* AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT =
+        "mpegh-reference-channel-layout";
 EXPORT const char* AMEDIAFORMAT_KEY_OPERATING_RATE = "operating-rate";
 EXPORT const char* AMEDIAFORMAT_KEY_PCM_ENCODING = "pcm-encoding";
 EXPORT const char* AMEDIAFORMAT_KEY_PRIORITY = "priority";
diff --git a/media/ndk/include/media/NdkMediaFormat.h b/media/ndk/include/media/NdkMediaFormat.h
index fbd855d..2d2fcc0 100644
--- a/media/ndk/include/media/NdkMediaFormat.h
+++ b/media/ndk/include/media/NdkMediaFormat.h
@@ -320,6 +320,34 @@
 extern const char* AMEDIAFORMAT_VIDEO_QP_P_MAX __INTRODUCED_IN(31);
 extern const char* AMEDIAFORMAT_VIDEO_QP_P_MIN __INTRODUCED_IN(31);
 
+/**
+ * MPEG-H audio profile and level compatibility.
+ *
+ * See FDAmd_2 of ISO_IEC_23008-3;2019 MHAProfileAndLevelCompatibilitySetBox.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS __INTRODUCED_IN(32);
+
+/**
+ * MPEG-H audio profile level indication.
+ *
+ * See ISO_IEC_23008-3;2019 MHADecoderConfigurationRecord mpegh3daProfileLevelIndication.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION __INTRODUCED_IN(32);
+
+/**
+ * MPEG-H audio reference channel layout.
+ *
+ * See ISO_IEC_23008-3;2019 MHADecoderConfigurationRecord referenceChannelLayout
+ * and ISO_IEC_23001‐8 ChannelConfiguration value.
+ *
+ * Available since API level 32.
+ */
+extern const char* AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT __INTRODUCED_IN(32);
+
 __END_DECLS
 
 #endif // _NDK_MEDIA_FORMAT_H
diff --git a/media/ndk/libmediandk.map.txt b/media/ndk/libmediandk.map.txt
index 7e9e57e..6f275c7 100644
--- a/media/ndk/libmediandk.map.txt
+++ b/media/ndk/libmediandk.map.txt
@@ -126,6 +126,9 @@
     AMEDIAFORMAT_KEY_MIME; # var introduced=21
     AMEDIAFORMAT_KEY_MPEG_USER_DATA; # var introduced=28
     AMEDIAFORMAT_KEY_MPEG2_STREAM_HEADER; # var introduced=29
+    AMEDIAFORMAT_KEY_MPEGH_COMPATIBLE_SETS; # var introduced=32
+    AMEDIAFORMAT_KEY_MPEGH_PROFILE_LEVEL_INDICATION; # var introduced=32
+    AMEDIAFORMAT_KEY_MPEGH_REFERENCE_CHANNEL_LAYOUT; # var introduced=32
     AMEDIAFORMAT_KEY_OPERATING_RATE; # var introduced=28
     AMEDIAFORMAT_KEY_PCM_BIG_ENDIAN; # var introduced=29
     AMEDIAFORMAT_KEY_PCM_ENCODING; # var introduced=28
diff --git a/media/tests/benchmark/MediaBenchmarkTest/Android.bp b/media/tests/benchmark/MediaBenchmarkTest/Android.bp
index 2e06da5..4b44dcf 100644
--- a/media/tests/benchmark/MediaBenchmarkTest/Android.bp
+++ b/media/tests/benchmark/MediaBenchmarkTest/Android.bp
@@ -69,7 +69,6 @@
 java_defaults {
     name: "MediaBenchmark-defaults",
 
-    sdk_version: "system_current",
     min_sdk_version: "28",
-    target_sdk_version: "29",
+    target_sdk_version: "30",
 }
diff --git a/media/tests/benchmark/MediaBenchmarkTest/build.gradle b/media/tests/benchmark/MediaBenchmarkTest/build.gradle
index b2aee1a..b222d47 100644
--- a/media/tests/benchmark/MediaBenchmarkTest/build.gradle
+++ b/media/tests/benchmark/MediaBenchmarkTest/build.gradle
@@ -17,21 +17,21 @@
 buildscript {
     repositories {
         google()
-        jcenter()
+        mavenCentral()
     }
     dependencies {
-        classpath 'com.android.tools.build:gradle:3.5.0'
+        classpath 'com.android.tools.build:gradle:4.2.1'
     }
 }
 
 apply plugin: 'com.android.application'
 
 android {
-    compileSdkVersion 29
+    compileSdkVersion 30
     defaultConfig {
         applicationId "com.android.media.benchmark"
         minSdkVersion 28
-        targetSdkVersion 29
+        targetSdkVersion 30
         versionCode 1
         versionName "1.0"
         testInstrumentationRunner "androidx.test.runner.AndroidJUnitRunner"
@@ -57,20 +57,20 @@
     externalNativeBuild {
         cmake {
             path "src/main/cpp/CMakeLists.txt"
-            version "3.10.2"
+            version "3.18.1"
         }
     }
 }
 
 repositories {
     google()
-    jcenter()
+    mavenCentral()
 }
 
 dependencies {
     implementation fileTree(dir: 'libs', include: ['*.jar'])
-    implementation 'androidx.appcompat:appcompat:1.1.0'
-    testImplementation 'junit:junit:4.12'
-    androidTestImplementation 'androidx.test:runner:1.2.0'
-    androidTestImplementation 'androidx.test.ext:junit:1.1.1'
+    implementation 'androidx.appcompat:appcompat:1.3.0'
+    testImplementation 'junit:junit:4.13.2'
+    androidTestImplementation 'androidx.test:runner:1.3.0'
+    androidTestImplementation 'androidx.test.ext:junit:1.1.2'
 }
\ No newline at end of file
diff --git a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp
index af92424..0192d68 100644
--- a/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp
+++ b/media/tests/benchmark/MediaBenchmarkTest/src/main/cpp/Android.bp
@@ -9,7 +9,6 @@
 
 cc_test_library {
     name: "libmediabenchmark_jni",
-    sdk_version: "current",
 
     defaults: [
         "libmediabenchmark_common-defaults",
diff --git a/media/tests/benchmark/src/native/common/Android.bp b/media/tests/benchmark/src/native/common/Android.bp
index 6b54c6a..718d217 100644
--- a/media/tests/benchmark/src/native/common/Android.bp
+++ b/media/tests/benchmark/src/native/common/Android.bp
@@ -55,7 +55,6 @@
 
 cc_defaults {
     name: "libmediabenchmark-defaults",
-    sdk_version: "current",
     stl: "c++_shared",
 
     shared_libs: [
diff --git a/media/tests/benchmark/src/native/extractor/Extractor.cpp b/media/tests/benchmark/src/native/extractor/Extractor.cpp
index f0bb3b9..3bdfbad 100644
--- a/media/tests/benchmark/src/native/extractor/Extractor.cpp
+++ b/media/tests/benchmark/src/native/extractor/Extractor.cpp
@@ -124,9 +124,7 @@
 
     int64_t sTime = mStats->getCurTime();
     if (mExtractor) {
-        // TODO: (b/140128505) Multiple calls result in DoS.
-        // Uncomment call to AMediaExtractor_delete() once this is resolved
-        // AMediaExtractor_delete(mExtractor);
+        AMediaExtractor_delete(mExtractor);
         mExtractor = nullptr;
     }
     int64_t eTime = mStats->getCurTime();
diff --git a/media/tests/benchmark/tests/Android.bp b/media/tests/benchmark/tests/Android.bp
index 0fbd20d..9a8caa3 100644
--- a/media/tests/benchmark/tests/Android.bp
+++ b/media/tests/benchmark/tests/Android.bp
@@ -33,7 +33,12 @@
 
     srcs: ["ExtractorTest.cpp"],
 
-    static_libs: ["libmediabenchmark_extractor"]
+    static_libs: ["libmediabenchmark_extractor"],
+
+    shared_libs: [
+        "libbase",
+        "libbinder_ndk",
+    ],
 }
 
 cc_test {
@@ -50,6 +55,11 @@
         "libmediabenchmark_extractor",
         "libmediabenchmark_decoder",
     ],
+
+    shared_libs: [
+        "libbase",
+        "libbinder_ndk",
+    ],
 }
 
 cc_test {
diff --git a/media/tests/benchmark/tests/DecoderTest.cpp b/media/tests/benchmark/tests/DecoderTest.cpp
index 81ef02a..3666724 100644
--- a/media/tests/benchmark/tests/DecoderTest.cpp
+++ b/media/tests/benchmark/tests/DecoderTest.cpp
@@ -21,6 +21,8 @@
 #include <iostream>
 #include <limits>
 
+#include <android/binder_process.h>
+
 #include "BenchmarkTestEnvironment.h"
 #include "Decoder.h"
 
@@ -175,6 +177,7 @@
                                             "c2.android.hevc.decoder", true)));
 
 int main(int argc, char **argv) {
+    ABinderProcess_startThreadPool();
     gEnv = new BenchmarkTestEnvironment();
     ::testing::AddGlobalTestEnvironment(gEnv);
     ::testing::InitGoogleTest(&argc, argv);
diff --git a/media/tests/benchmark/tests/ExtractorTest.cpp b/media/tests/benchmark/tests/ExtractorTest.cpp
index d14d15b..27ee9ba 100644
--- a/media/tests/benchmark/tests/ExtractorTest.cpp
+++ b/media/tests/benchmark/tests/ExtractorTest.cpp
@@ -19,6 +19,8 @@
 
 #include <gtest/gtest.h>
 
+#include <android/binder_process.h>
+
 #include "BenchmarkTestEnvironment.h"
 #include "Extractor.h"
 
@@ -73,6 +75,7 @@
                                                      0)));
 
 int main(int argc, char **argv) {
+    ABinderProcess_startThreadPool();
     gEnv = new BenchmarkTestEnvironment();
     ::testing::AddGlobalTestEnvironment(gEnv);
     ::testing::InitGoogleTest(&argc, argv);
diff --git a/media/utils/Android.bp b/media/utils/Android.bp
index bfe73d5..73c4e3b 100644
--- a/media/utils/Android.bp
+++ b/media/utils/Android.bp
@@ -81,6 +81,36 @@
     export_include_dirs: ["include"],
 }
 
+cc_library {
+    name: "libmediautils_vendor",
+    vendor_available: true,  // required for platform/hardware/interfaces
+    srcs: [
+        "MemoryLeakTrackUtil.cpp",
+    ],
+
+    cflags: [
+        "-Wall",
+        "-Wextra",
+        "-Werror",
+    ],
+    shared_libs: [
+        "liblog",
+        "libutils",
+    ],
+
+    static_libs: [
+        "libc_malloc_debug_backtrace",
+    ],
+
+    header_libs: [
+        "bionic_libc_platform_headers",
+    ],
+
+    local_include_dirs: ["include"],
+    export_include_dirs: ["include"],
+}
+
+
 cc_library_headers {
     name: "libmediautils_headers",
     vendor_available: true,  // required for platform/hardware/interfaces
diff --git a/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp b/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp
index 6e52512..51e8d7a 100644
--- a/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp
+++ b/media/utils/fuzzers/ServiceUtilitiesFuzz.cpp
@@ -17,7 +17,7 @@
 #include <fcntl.h>
 
 #include <functional>
-#include <type_traits>
+#include  <type_traits>
 
 #include <android/content/AttributionSourceState.h>
 #include "fuzzer/FuzzedDataProvider.h"
diff --git a/services/audioflinger/AudioFlinger.cpp b/services/audioflinger/AudioFlinger.cpp
index 65a163f..a1fb304 100644
--- a/services/audioflinger/AudioFlinger.cpp
+++ b/services/audioflinger/AudioFlinger.cpp
@@ -336,11 +336,11 @@
 }
 
 // getDefaultVibratorInfo_l must be called with AudioFlinger lock held.
-const media::AudioVibratorInfo* AudioFlinger::getDefaultVibratorInfo_l() {
+std::optional<media::AudioVibratorInfo> AudioFlinger::getDefaultVibratorInfo_l() {
     if (mAudioVibratorInfos.empty()) {
-        return nullptr;
+        return {};
     }
-    return &mAudioVibratorInfos.front();
+    return mAudioVibratorInfos.front();
 }
 
 AudioFlinger::~AudioFlinger()
@@ -695,7 +695,7 @@
         // dump all hardware devs
         for (size_t i = 0; i < mAudioHwDevs.size(); i++) {
             sp<DeviceHalInterface> dev = mAudioHwDevs.valueAt(i)->hwDevice();
-            dev->dump(fd);
+            dev->dump(fd, args);
         }
 
         mPatchPanel.dump(fd);
@@ -1873,13 +1873,13 @@
     }
 }
 
-void AudioFlinger::ioConfigChanged(audio_io_config_event event,
+void AudioFlinger::ioConfigChanged(audio_io_config_event_t event,
                                    const sp<AudioIoDescriptor>& ioDesc,
                                    pid_t pid) {
+    media::AudioIoConfigEvent eventAidl = VALUE_OR_FATAL(
+            legacy2aidl_audio_io_config_event_t_AudioIoConfigEvent(event));
     media::AudioIoDescriptor descAidl = VALUE_OR_FATAL(
             legacy2aidl_AudioIoDescriptor_AudioIoDescriptor(ioDesc));
-    media::AudioIoConfigEvent eventAidl = VALUE_OR_FATAL(
-            legacy2aidl_audio_io_config_event_AudioIoConfigEvent(event));
 
     Mutex::Autolock _l(mClientLock);
     size_t size = mNotificationClients.size();
@@ -2455,6 +2455,10 @@
         ThreadBase *thread = (ThreadBase *)mRecordThreads.valueAt(i).get();
         thread->systemReady();
     }
+    for (size_t i = 0; i < mMmapThreads.size(); i++) {
+        ThreadBase *thread = (ThreadBase *)mMmapThreads.valueAt(i).get();
+        thread->systemReady();
+    }
     return NO_ERROR;
 }
 
@@ -2501,7 +2505,8 @@
 
 sp<AudioFlinger::ThreadBase> AudioFlinger::openOutput_l(audio_module_handle_t module,
                                                         audio_io_handle_t *output,
-                                                        audio_config_t *config,
+                                                        audio_config_t *halConfig,
+                                                        audio_config_base_t *mixerConfig __unused,
                                                         audio_devices_t deviceType,
                                                         const String8& address,
                                                         audio_output_flags_t flags)
@@ -2529,16 +2534,16 @@
         // Check only for Normal Mixing mode
         if (kEnableExtendedPrecision) {
             // Specify format (uncomment one below to choose)
-            //config->format = AUDIO_FORMAT_PCM_FLOAT;
-            //config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
-            //config->format = AUDIO_FORMAT_PCM_32_BIT;
-            //config->format = AUDIO_FORMAT_PCM_8_24_BIT;
-            // ALOGV("openOutput_l() upgrading format to %#08x", config->format);
+            //halConfig->format = AUDIO_FORMAT_PCM_FLOAT;
+            //halConfig->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
+            //halConfig->format = AUDIO_FORMAT_PCM_32_BIT;
+            //halConfig->format = AUDIO_FORMAT_PCM_8_24_BIT;
+            // ALOGV("openOutput_l() upgrading format to %#08x", halConfig->format);
         }
         if (kEnableExtendedChannels) {
             // Specify channel mask (uncomment one below to choose)
-            //config->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
-            //config->channel_mask = audio_channel_mask_from_representation_and_bits(
+            //halConfig->channel_mask = audio_channel_out_mask_from_count(4);  // for USB 4ch
+            //halConfig->channel_mask = audio_channel_mask_from_representation_and_bits(
             //        AUDIO_CHANNEL_REPRESENTATION_INDEX, (1 << 4) - 1);  // another 4ch example
         }
     }
@@ -2549,7 +2554,7 @@
             *output,
             deviceType,
             flags,
-            config,
+            halConfig,
             address.string());
 
     mHardwareStatus = AUDIO_HW_IDLE;
@@ -2564,13 +2569,20 @@
             return thread;
         } else {
             sp<PlaybackThread> thread;
-            if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+            //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+            if (flags == (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST
+                                                    | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+                thread = new VirtualizerStageThread(this, outputStream, *output,
+                                                    mSystemReady, mixerConfig);
+                ALOGD("openOutput_l() created virtualizer output: ID %d thread %p",
+                      *output, thread.get());
+            } else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
                 thread = new OffloadThread(this, outputStream, *output, mSystemReady);
                 ALOGV("openOutput_l() created offload output: ID %d thread %p",
                       *output, thread.get());
             } else if ((flags & AUDIO_OUTPUT_FLAG_DIRECT)
-                    || !isValidPcmSinkFormat(config->format)
-                    || !isValidPcmSinkChannelMask(config->channel_mask)) {
+                    || !isValidPcmSinkFormat(halConfig->format)
+                    || !isValidPcmSinkChannelMask(halConfig->channel_mask)) {
                 thread = new DirectOutputThread(this, outputStream, *output, mSystemReady);
                 ALOGV("openOutput_l() created direct output: ID %d thread %p",
                       *output, thread.get());
@@ -2597,8 +2609,10 @@
 {
     audio_module_handle_t module = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_module_handle_t(request.module));
-    audio_config_t config = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioConfig_audio_config_t(request.config));
+    audio_config_t halConfig = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioConfig_audio_config_t(request.halConfig, false /*isInput*/));
+    audio_config_base_t mixerConfig = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioConfigBase_audio_config_base_t(request.mixerConfig, false/*isInput*/));
     sp<DeviceDescriptorBase> device = VALUE_OR_RETURN_STATUS(
             aidl2legacy_DeviceDescriptorBase(request.device));
     audio_output_flags_t flags = VALUE_OR_RETURN_STATUS(
@@ -2611,9 +2625,9 @@
               "Channels %#x, flags %#x",
               this, module,
               device->toString().c_str(),
-              config.sample_rate,
-              config.format,
-              config.channel_mask,
+              halConfig.sample_rate,
+              halConfig.format,
+              halConfig.channel_mask,
               flags);
 
     audio_devices_t deviceType = device->type();
@@ -2625,7 +2639,8 @@
 
     Mutex::Autolock _l(mLock);
 
-    sp<ThreadBase> thread = openOutput_l(module, &output, &config, deviceType, address, flags);
+    sp<ThreadBase> thread = openOutput_l(module, &output, &halConfig,
+            &mixerConfig, deviceType, address, flags);
     if (thread != 0) {
         if ((flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0) {
             PlaybackThread *playbackThread = (PlaybackThread *)thread.get();
@@ -2650,7 +2665,8 @@
             mmapThread->ioConfigChanged(AUDIO_OUTPUT_OPENED);
         }
         response->output = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(output));
-        response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+        response->config = VALUE_OR_RETURN_STATUS(
+                legacy2aidl_audio_config_t_AudioConfig(halConfig, false /*isInput*/));
         response->latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<int32_t>(latencyMs));
         response->flags = VALUE_OR_RETURN_STATUS(
                 legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
@@ -2736,9 +2752,7 @@
             mMmapThreads.removeItem(output);
             ALOGD("closing mmapThread %p", mmapThread.get());
         }
-        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
-        ioDesc->mIoHandle = output;
-        ioConfigChanged(AUDIO_OUTPUT_CLOSED, ioDesc);
+        ioConfigChanged(AUDIO_OUTPUT_CLOSED, sp<AudioIoDescriptor>::make(output));
         mPatchPanel.notifyStreamClosed(output);
     }
     // The thread entity (active unit of execution) is no longer running here,
@@ -2811,16 +2825,16 @@
 {
     Mutex::Autolock _l(mLock);
 
-    if (request.device.type == AUDIO_DEVICE_NONE) {
+    AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
+            aidl2legacy_AudioDeviceTypeAddress(request.device));
+    if (device.mType == AUDIO_DEVICE_NONE) {
         return BAD_VALUE;
     }
 
     audio_io_handle_t input = VALUE_OR_RETURN_STATUS(
             aidl2legacy_int32_t_audio_io_handle_t(request.input));
     audio_config_t config = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioConfig_audio_config_t(request.config));
-    AudioDeviceTypeAddr device = VALUE_OR_RETURN_STATUS(
-            aidl2legacy_AudioDeviceTypeAddress(request.device));
+            aidl2legacy_AudioConfig_audio_config_t(request.config, true /*isInput*/));
 
     sp<ThreadBase> thread = openInput_l(
             VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_module_handle_t(request.module)),
@@ -2834,7 +2848,8 @@
             String8{});
 
     response->input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(input));
-    response->config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(config));
+    response->config = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_t_AudioConfig(config, true /*isInput*/));
     response->device = request.device;
 
     if (thread != 0) {
@@ -2996,9 +3011,7 @@
             dumpToThreadLog_l(mmapThread);
             mMmapThreads.removeItem(input);
         }
-        const sp<AudioIoDescriptor> ioDesc = new AudioIoDescriptor();
-        ioDesc->mIoHandle = input;
-        ioConfigChanged(AUDIO_INPUT_CLOSED, ioDesc);
+        ioConfigChanged(AUDIO_INPUT_CLOSED, sp<AudioIoDescriptor>::make(input));
     }
     // FIXME: calling thread->exit() without mLock held should not be needed anymore now that
     // we have a different lock for notification client
@@ -3797,7 +3810,8 @@
                 io = mPlaybackThreads.keyAt(0);
             }
             ALOGV("createEffect() got io %d for effect %s", io, descOut.name);
-        } else if (checkPlaybackThread_l(io) != nullptr) {
+        } else if (checkPlaybackThread_l(io) != nullptr
+                        && sessionId != AUDIO_SESSION_OUTPUT_STAGE) {
             // allow only one effect chain per sessionId on mPlaybackThreads.
             for (size_t i = 0; i < mPlaybackThreads.size(); i++) {
                 const audio_io_handle_t checkIo = mPlaybackThreads.keyAt(i);
@@ -4178,6 +4192,7 @@
         case TransactionCode::LIST_AUDIO_PATCHES:
         case TransactionCode::SET_AUDIO_PORT_CONFIG:
         case TransactionCode::SET_RECORD_SILENCED:
+        case TransactionCode::AUDIO_POLICY_READY:
             ALOGW("%s: transaction %d received from PID %d",
                   __func__, code, IPCThreadState::self()->getCallingPid());
             // return status only for non void methods
diff --git a/services/audioflinger/AudioFlinger.h b/services/audioflinger/AudioFlinger.h
index fff61f8..d6bf0ae 100644
--- a/services/audioflinger/AudioFlinger.h
+++ b/services/audioflinger/AudioFlinger.h
@@ -269,6 +269,9 @@
 
     /* Indicate JAVA services are ready (scheduling, power management ...) */
     virtual status_t systemReady();
+    virtual status_t audioPolicyReady() { mAudioPolicyReady.store(true); return NO_ERROR; }
+            bool isAudioPolicyReady() const { return mAudioPolicyReady.load(); }
+
 
     virtual status_t getMicrophones(std::vector<media::MicrophoneInfo> *microphones);
 
@@ -309,7 +312,7 @@
     void updateDownStreamPatches_l(const struct audio_patch *patch,
                                    const std::set<audio_io_handle_t> streams);
 
-    const media::AudioVibratorInfo* getDefaultVibratorInfo_l();
+    std::optional<media::AudioVibratorInfo> getDefaultVibratorInfo_l();
 
 private:
     // FIXME The 400 is temporarily too high until a leak of writers in media.log is fixed.
@@ -735,7 +738,8 @@
                                            const String8& outputDeviceAddress);
               sp<ThreadBase> openOutput_l(audio_module_handle_t module,
                                           audio_io_handle_t *output,
-                                          audio_config_t *config,
+                                          audio_config_t *halConfig,
+                                          audio_config_base_t *mixerConfig,
                                           audio_devices_t deviceType,
                                           const String8& address,
                                           audio_output_flags_t flags);
@@ -746,7 +750,7 @@
               // no range check, AudioFlinger::mLock held
               bool streamMute_l(audio_stream_type_t stream) const
                                 { return mStreamTypes[stream].mute; }
-              void ioConfigChanged(audio_io_config_event event,
+              void ioConfigChanged(audio_io_config_event_t event,
                                    const sp<AudioIoDescriptor>& ioDesc,
                                    pid_t pid = 0);
 
@@ -986,6 +990,7 @@
     DeviceEffectManager mDeviceEffectManager;
 
     bool       mSystemReady;
+    std::atomic_bool mAudioPolicyReady{};
 
     mediautils::UidInfo mUidInfo;
 
diff --git a/services/audioflinger/DeviceEffectManager.h b/services/audioflinger/DeviceEffectManager.h
index a05f5fe..c222de8 100644
--- a/services/audioflinger/DeviceEffectManager.h
+++ b/services/audioflinger/DeviceEffectManager.h
@@ -163,8 +163,13 @@
     bool isOffloadOrMmap() const override { return false; }
 
     uint32_t  sampleRate() const override { return 0; }
-    audio_channel_mask_t channelMask() const override { return AUDIO_CHANNEL_NONE; }
-    uint32_t channelCount() const override { return 0; }
+    audio_channel_mask_t inChannelMask(int id __unused) const override {
+        return AUDIO_CHANNEL_NONE;
+    }
+    uint32_t inChannelCount(int id __unused) const override { return 0; }
+    audio_channel_mask_t outChannelMask() const override { return AUDIO_CHANNEL_NONE; }
+    uint32_t outChannelCount() const override { return 0; }
+
     audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
     size_t    frameCount() const override  { return 0; }
     uint32_t  latency() const override  { return 0; }
@@ -190,6 +195,10 @@
 
     wp<EffectChain> chain() const override { return nullptr; }
 
+    bool isAudioPolicyReady() const override {
+        return mManager.audioFlinger().isAudioPolicyReady();
+    }
+
     int newEffectId() { return mManager.audioFlinger().nextUniqueId(AUDIO_UNIQUE_ID_USE_EFFECT); }
 
     status_t addEffectToHal(audio_port_handle_t deviceId,
diff --git a/services/audioflinger/Effects.cpp b/services/audioflinger/Effects.cpp
index b267d88..bd661f9 100644
--- a/services/audioflinger/Effects.cpp
+++ b/services/audioflinger/Effects.cpp
@@ -242,6 +242,12 @@
 
     {
         Mutex::Autolock _l(mLock);
+
+        if ((isInternal_l() && !mPolicyRegistered)
+                || !getCallback()->isAudioPolicyReady()) {
+            return NO_ERROR;
+        }
+
         // register effect when first handle is attached and unregister when last handle is removed
         if (mPolicyRegistered != mHandles.size() > 0) {
             doRegister = true;
@@ -875,9 +881,9 @@
     // similar to output EFFECT_FLAG_TYPE_INSERT/REPLACE,
     // in which case input channel masks should be used here.
     callback = getCallback();
-    channelMask = callback->channelMask();
+    channelMask = callback->inChannelMask(mId);
     mConfig.inputCfg.channels = channelMask;
-    mConfig.outputCfg.channels = channelMask;
+    mConfig.outputCfg.channels = callback->outChannelMask();
 
     if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_AUXILIARY) {
         if (mConfig.inputCfg.channels != AUDIO_CHANNEL_OUT_MONO) {
@@ -1600,7 +1606,7 @@
     return status;
 }
 
-status_t AudioFlinger::EffectModule::setVibratorInfo(const media::AudioVibratorInfo* vibratorInfo)
+status_t AudioFlinger::EffectModule::setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo)
 {
     if (mStatus != NO_ERROR) {
         return mStatus;
@@ -1610,15 +1616,17 @@
         return INVALID_OPERATION;
     }
 
+    const size_t paramCount = 3;
     std::vector<uint8_t> request(
-            sizeof(effect_param_t) + sizeof(int32_t) + 2 * sizeof(float));
+            sizeof(effect_param_t) + sizeof(int32_t) + paramCount * sizeof(float));
     effect_param_t *param = (effect_param_t*) request.data();
     param->psize = sizeof(int32_t);
-    param->vsize = 2 * sizeof(float);
+    param->vsize = paramCount * sizeof(float);
     *(int32_t*)param->data = HG_PARAM_VIBRATOR_INFO;
     float* vibratorInfoPtr = reinterpret_cast<float*>(param->data + sizeof(int32_t));
-    vibratorInfoPtr[0] = vibratorInfo->resonantFrequency;
-    vibratorInfoPtr[1] = vibratorInfo->qFactor;
+    vibratorInfoPtr[0] = vibratorInfo.resonantFrequency;
+    vibratorInfoPtr[1] = vibratorInfo.qFactor;
+    vibratorInfoPtr[2] = vibratorInfo.maxAmplitude;
     std::vector<uint8_t> response;
     status_t status = command(EFFECT_CMD_SET_PARAM, request, sizeof(int32_t), &response);
     if (status == NO_ERROR) {
@@ -2048,11 +2056,11 @@
       mNewLeftVolume(UINT_MAX), mNewRightVolume(UINT_MAX),
       mEffectCallback(new EffectCallback(wp<EffectChain>(this), thread))
 {
-    mStrategy = AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
     sp<ThreadBase> p = thread.promote();
     if (p == nullptr) {
         return;
     }
+    mStrategy = p->getStrategyForStream(AUDIO_STREAM_MUSIC);
     mMaxTailBuffers = ((kProcessTailDurationMs * p->sampleRate()) / 1000) /
                                     p->frameCount();
 }
@@ -2125,8 +2133,8 @@
     if (mInBuffer == NULL) {
         return;
     }
-    const size_t frameSize =
-            audio_bytes_per_sample(EFFECT_BUFFER_FORMAT) * mEffectCallback->channelCount();
+    const size_t frameSize = audio_bytes_per_sample(EFFECT_BUFFER_FORMAT)
+            * mEffectCallback->inChannelCount(mEffects[0]->id());
 
     memset(mInBuffer->audioBuffer()->raw, 0, mEffectCallback->frameCount() * frameSize);
     mInBuffer->commit();
@@ -2236,6 +2244,9 @@
                 numSamples * sizeof(int32_t), &halBuffer);
 #endif
         if (result != OK) return result;
+
+        effect->configure();
+
         effect->setInBuffer(halBuffer);
         // auxiliary effects output samples to chain input buffer for further processing
         // by insert effects
@@ -2303,6 +2314,10 @@
             }
         }
 
+        mEffects.insertAt(effect, idx_insert);
+
+        effect->configure();
+
         // always read samples from chain input buffer
         effect->setInBuffer(mInBuffer);
 
@@ -2310,14 +2325,13 @@
         // output buffer, otherwise to chain input buffer
         if (idx_insert == size) {
             if (idx_insert != 0) {
-                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
                 mEffects[idx_insert-1]->configure();
+                mEffects[idx_insert-1]->setOutBuffer(mInBuffer);
             }
             effect->setOutBuffer(mOutBuffer);
         } else {
             effect->setOutBuffer(mInBuffer);
         }
-        mEffects.insertAt(effect, idx_insert);
 
         ALOGV("addEffect_l() effect %p, added in chain %p at rank %zu", effect.get(), this,
                 idx_insert);
@@ -2350,14 +2364,21 @@
 
             if (type != EFFECT_FLAG_TYPE_AUXILIARY) {
                 if (i == size - 1 && i != 0) {
-                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
                     mEffects[i - 1]->configure();
+                    mEffects[i - 1]->setOutBuffer(mOutBuffer);
                 }
             }
             mEffects.removeAt(i);
+
+            // make sure the input buffer configuration for the new first effect in the chain
+            // is updated if needed (can switch from HAL channel mask to mixer channel mask)
+            if (i == 0 && size > 1) {
+                mEffects[0]->configure();
+                mEffects[0]->setInBuffer(mInBuffer);
+            }
+
             ALOGV("removeEffect_l() effect %p, removed from chain %p at rank %zu", effect.get(),
                     this, i);
-
             break;
         }
     }
@@ -2932,7 +2953,43 @@
     return t->sampleRate();
 }
 
-audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::inChannelMask(int id) const {
+    sp<ThreadBase> t = thread().promote();
+    if (t == nullptr) {
+        return AUDIO_CHANNEL_NONE;
+    }
+    sp<EffectChain> c = chain().promote();
+    if (c == nullptr) {
+        return AUDIO_CHANNEL_NONE;
+    }
+
+    if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+            || c->isFirstEffect(id)) {
+        return t->mixerChannelMask();
+    } else {
+        return t->channelMask();
+    }
+}
+
+uint32_t AudioFlinger::EffectChain::EffectCallback::inChannelCount(int id) const {
+    sp<ThreadBase> t = thread().promote();
+    if (t == nullptr) {
+        return 0;
+    }
+    sp<EffectChain> c = chain().promote();
+    if (c == nullptr) {
+        return 0;
+    }
+
+    if (c->sessionId() != AUDIO_SESSION_OUTPUT_STAGE
+            || c->isFirstEffect(id)) {
+        return audio_channel_count_from_out_mask(t->mixerChannelMask());
+    } else {
+        return t->channelCount();
+    }
+}
+
+audio_channel_mask_t AudioFlinger::EffectChain::EffectCallback::outChannelMask() const {
     sp<ThreadBase> t = thread().promote();
     if (t == nullptr) {
         return AUDIO_CHANNEL_NONE;
@@ -2940,7 +2997,7 @@
     return t->channelMask();
 }
 
-uint32_t AudioFlinger::EffectChain::EffectCallback::channelCount() const {
+uint32_t AudioFlinger::EffectChain::EffectCallback::outChannelCount() const {
     sp<ThreadBase> t = thread().promote();
     if (t == nullptr) {
         return 0;
@@ -3364,7 +3421,8 @@
     return proxy->sampleRate();
 }
 
-audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelMask() const {
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelMask(
+        int id __unused) const {
     sp<DeviceEffectProxy> proxy = mProxy.promote();
     if (proxy == nullptr) {
         return AUDIO_CHANNEL_OUT_STEREO;
@@ -3372,7 +3430,23 @@
     return proxy->channelMask();
 }
 
-uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::channelCount() const {
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::inChannelCount(int id __unused) const {
+    sp<DeviceEffectProxy> proxy = mProxy.promote();
+    if (proxy == nullptr) {
+        return 2;
+    }
+    return proxy->channelCount();
+}
+
+audio_channel_mask_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelMask() const {
+    sp<DeviceEffectProxy> proxy = mProxy.promote();
+    if (proxy == nullptr) {
+        return AUDIO_CHANNEL_OUT_STEREO;
+    }
+    return proxy->channelMask();
+}
+
+uint32_t AudioFlinger::DeviceEffectProxy::ProxyCallback::outChannelCount() const {
     sp<DeviceEffectProxy> proxy = mProxy.promote();
     if (proxy == nullptr) {
         return 2;
diff --git a/services/audioflinger/Effects.h b/services/audioflinger/Effects.h
index a727e04..1d0d00d 100644
--- a/services/audioflinger/Effects.h
+++ b/services/audioflinger/Effects.h
@@ -34,8 +34,10 @@
     virtual bool isOffloadOrDirect() const = 0;
     virtual bool isOffloadOrMmap() const = 0;
     virtual uint32_t sampleRate() const = 0;
-    virtual audio_channel_mask_t channelMask() const = 0;
-    virtual uint32_t channelCount() const = 0;
+    virtual audio_channel_mask_t inChannelMask(int id) const = 0;
+    virtual uint32_t inChannelCount(int id) const = 0;
+    virtual audio_channel_mask_t outChannelMask() const = 0;
+    virtual uint32_t outChannelCount() const = 0;
     virtual audio_channel_mask_t hapticChannelMask() const = 0;
     virtual size_t frameCount() const = 0;
 
@@ -64,6 +66,8 @@
     virtual void resetVolume() = 0;
 
     virtual wp<EffectChain> chain() const = 0;
+
+    virtual bool isAudioPolicyReady() const = 0;
 };
 
 // EffectBase(EffectModule) and EffectChain classes both have their own mutex to protect
@@ -164,6 +168,16 @@
 
     void             dump(int fd, const Vector<String16>& args);
 
+protected:
+    bool             isInternal_l() const {
+                         for (auto handle : mHandles) {
+                            if (handle->client() != nullptr) {
+                                return false;
+                            }
+                         }
+                         return true;
+                     }
+
 private:
     friend class AudioFlinger;      // for mHandles
     bool             mPinned = false;
@@ -259,7 +273,7 @@
     bool             isHapticGenerator() const;
 
     status_t         setHapticIntensity(int id, int intensity);
-    status_t         setVibratorInfo(const media::AudioVibratorInfo* vibratorInfo);
+    status_t         setVibratorInfo(const media::AudioVibratorInfo& vibratorInfo);
 
     void             dump(int fd, const Vector<String16>& args);
 
@@ -342,6 +356,8 @@
     android::binder::Status disconnect() override;
     android::binder::Status getCblk(media::SharedFileRegion* _aidl_return) override;
 
+    sp<Client> client() const { return mClient; }
+
 private:
     void disconnect(bool unpinIfLast);
 
@@ -511,6 +527,8 @@
     sp<EffectCallbackInterface> effectCallback() const { return mEffectCallback; }
     wp<ThreadBase> thread() const { return mEffectCallback->thread(); }
 
+    bool isFirstEffect(int id) const { return !mEffects.isEmpty() && id == mEffects[0]->id(); }
+
     void dump(int fd, const Vector<String16>& args);
 
 private:
@@ -544,8 +562,10 @@
         bool isOffloadOrMmap() const override;
 
         uint32_t sampleRate() const override;
-        audio_channel_mask_t channelMask() const override;
-        uint32_t channelCount() const override;
+        audio_channel_mask_t inChannelMask(int id) const override;
+        uint32_t inChannelCount(int id) const override;
+        audio_channel_mask_t outChannelMask() const override;
+        uint32_t outChannelCount() const override;
         audio_channel_mask_t hapticChannelMask() const override;
         size_t frameCount() const override;
         uint32_t latency() const override;
@@ -566,6 +586,10 @@
 
         wp<EffectChain> chain() const override { return mChain; }
 
+        bool isAudioPolicyReady() const override {
+            return mAudioFlinger.isAudioPolicyReady();
+        }
+
         wp<ThreadBase> thread() const { return mThread.load(); }
 
         void setThread(const wp<ThreadBase>& thread) {
@@ -694,8 +718,10 @@
         bool isOffloadOrMmap() const override { return false; }
 
         uint32_t sampleRate() const override;
-        audio_channel_mask_t channelMask() const override;
-        uint32_t channelCount() const override;
+        audio_channel_mask_t inChannelMask(int id) const override;
+        uint32_t inChannelCount(int id) const override;
+        audio_channel_mask_t outChannelMask() const override;
+        uint32_t outChannelCount() const override;
         audio_channel_mask_t hapticChannelMask() const override { return AUDIO_CHANNEL_NONE; }
         size_t frameCount() const override  { return 0; }
         uint32_t latency() const override  { return 0; }
@@ -716,6 +742,10 @@
 
         wp<EffectChain> chain() const override { return nullptr; }
 
+        bool isAudioPolicyReady() const override {
+            return mManagerCallback->isAudioPolicyReady();
+        }
+
         int newEffectId();
 
     private:
diff --git a/services/audioflinger/FastMixer.cpp b/services/audioflinger/FastMixer.cpp
index 88d4eaf..fc34d95 100644
--- a/services/audioflinger/FastMixer.cpp
+++ b/services/audioflinger/FastMixer.cpp
@@ -204,6 +204,8 @@
                 (void *)(uintptr_t)fastTrack->mHapticPlaybackEnabled);
         mMixer->setParameter(index, AudioMixer::TRACK, AudioMixer::HAPTIC_INTENSITY,
                 (void *)(uintptr_t)fastTrack->mHapticIntensity);
+        mMixer->setParameter(index, AudioMixer::TRACK, AudioMixer::HAPTIC_MAX_AMPLITUDE,
+                (void *)(&(fastTrack->mHapticMaxAmplitude)));
 
         mMixer->enable(index);
         break;
diff --git a/services/audioflinger/FastMixerState.h b/services/audioflinger/FastMixerState.h
index 857d3de..ce3cc14 100644
--- a/services/audioflinger/FastMixerState.h
+++ b/services/audioflinger/FastMixerState.h
@@ -17,6 +17,8 @@
 #ifndef ANDROID_AUDIO_FAST_MIXER_STATE_H
 #define ANDROID_AUDIO_FAST_MIXER_STATE_H
 
+#include <math.h>
+
 #include <audio_utils/minifloat.h>
 #include <system/audio.h>
 #include <media/AudioMixer.h>
@@ -51,6 +53,7 @@
     int                     mGeneration;     // increment when any field is assigned
     bool                    mHapticPlaybackEnabled = false; // haptic playback is enabled or not
     os::HapticScale         mHapticIntensity = os::HapticScale::MUTE; // intensity of haptic data
+    float                   mHapticMaxAmplitude = NAN; // max amplitude allowed for haptic data
 };
 
 // Represents a single state of the fast mixer
diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index a381c7d..93118b8 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -258,6 +258,7 @@
                             reinterpret_cast<PlaybackThread*>(thread.get()), false /*closeThread*/);
                 } else {
                     audio_config_t config = AUDIO_CONFIG_INITIALIZER;
+                    audio_config_base_t mixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
                     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
                     audio_output_flags_t flags = AUDIO_OUTPUT_FLAG_NONE;
                     if (patch->sinks[0].config_mask & AUDIO_PORT_CONFIG_SAMPLE_RATE) {
@@ -276,6 +277,7 @@
                                                             patch->sinks[0].ext.device.hw_module,
                                                             &output,
                                                             &config,
+                                                            &mixerConfig,
                                                             outputDevice,
                                                             outputDeviceAddress,
                                                             flags);
diff --git a/services/audioflinger/PlaybackTracks.h b/services/audioflinger/PlaybackTracks.h
index 0929055..e9e98ca 100644
--- a/services/audioflinger/PlaybackTracks.h
+++ b/services/audioflinger/PlaybackTracks.h
@@ -19,6 +19,8 @@
     #error This header file should only be included from AudioFlinger.h
 #endif
 
+#include <math.h>
+
 // Checks and monitors OP_PLAY_AUDIO
 class OpPlayAudioMonitor : public RefBase {
 public:
@@ -161,6 +163,8 @@
             }
             /** Return at what intensity to play haptics, used in mixer. */
             os::HapticScale getHapticIntensity() const { return mHapticIntensity; }
+            /** Return the maximum amplitude allowed for haptics data, used in mixer. */
+            float getHapticMaxAmplitude() const { return mHapticMaxAmplitude; }
             /** Set intensity of haptic playback, should be set after querying vibrator service. */
             void    setHapticIntensity(os::HapticScale hapticIntensity) {
                 if (os::isValidHapticScale(hapticIntensity)) {
@@ -168,6 +172,12 @@
                     setHapticPlaybackEnabled(mHapticIntensity != os::HapticScale::MUTE);
                 }
             }
+            /** Set maximum amplitude allowed for haptic data, should be set after querying
+             *  vibrator service.
+             */
+            void    setHapticMaxAmplitude(float maxAmplitude) {
+                mHapticMaxAmplitude = maxAmplitude;
+            }
             sp<os::ExternalVibration> getExternalVibration() const { return mExternalVibration; }
 
             void    setTeePatches(TeePatches teePatches);
@@ -282,6 +292,8 @@
     bool                mHapticPlaybackEnabled = false; // indicates haptic playback enabled or not
     // intensity to play haptic data
     os::HapticScale mHapticIntensity = os::HapticScale::MUTE;
+    // max amplitude allowed for haptic data
+    float mHapticMaxAmplitude = NAN;
     class AudioVibrationController : public os::BnExternalVibrationController {
     public:
         explicit AudioVibrationController(Track* track) : mTrack(track) {}
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index b9cdab8..9665424 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -50,8 +50,10 @@
 #include <audio_utils/format.h>
 #include <audio_utils/minifloat.h>
 #include <audio_utils/safe_math.h>
-#include <system/audio_effects/effect_ns.h>
 #include <system/audio_effects/effect_aec.h>
+#include <system/audio_effects/effect_downmix.h>
+#include <system/audio_effects/effect_ns.h>
+#include <system/audio_effects/effect_virtualizer_stage.h>
 #include <system/audio.h>
 
 // NBAIO implementations
@@ -507,6 +509,8 @@
         return "MMAP_PLAYBACK";
     case MMAP_CAPTURE:
         return "MMAP_CAPTURE";
+    case VIRTUALIZER_STAGE:
+        return "VIRTUALIZER_STAGE";
     default:
         return "unknown";
     }
@@ -622,7 +626,7 @@
     return status;
 }
 
-void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event event, pid_t pid,
+void AudioFlinger::ThreadBase::sendIoConfigEvent(audio_io_config_event_t event, pid_t pid,
                                                  audio_port_handle_t portId)
 {
     Mutex::Autolock _l(mLock);
@@ -630,7 +634,7 @@
 }
 
 // sendIoConfigEvent_l() must be called with ThreadBase::mLock held
-void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event event, pid_t pid,
+void AudioFlinger::ThreadBase::sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid,
                                                    audio_port_handle_t portId)
 {
     // The audio statistics history is exponentially weighted to forget events
@@ -722,6 +726,19 @@
     sendConfigEvent_l(configEvent);
 }
 
+void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent()
+{
+    Mutex::Autolock _l(mLock);
+    sendCheckOutputStageEffectsEvent_l();
+}
+
+void AudioFlinger::ThreadBase::sendCheckOutputStageEffectsEvent_l()
+{
+    sp<ConfigEvent> configEvent =
+            (ConfigEvent *)new CheckOutputStageEffectsEvent();
+    sendConfigEvent_l(configEvent);
+}
+
 // post condition: mConfigEvents.isEmpty()
 void AudioFlinger::ThreadBase::processConfigEvents_l()
 {
@@ -784,6 +801,11 @@
                     (ResizeBufferConfigEventData *)event->mData.get();
             resizeInputBuffer_l(data->mMaxSharedAudioHistoryMs);
         } break;
+
+        case CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS: {
+            setCheckOutputStageEffects();
+        } break;
+
         default:
             ALOG_ASSERT(false, "processConfigEvents_l() unknown event type %d", event->mType);
             break;
@@ -1008,6 +1030,8 @@
         return String16("MmapPlayback");
     case MMAP_CAPTURE:
         return String16("MmapCapture");
+    case VIRTUALIZER_STAGE:
+        return String16("AudioVirt");
     default:
         ALOG_ASSERT(false);
         return String16("AudioUnknown");
@@ -1401,6 +1425,13 @@
             return BAD_VALUE;
         }
         break;
+    case VIRTUALIZER_STAGE:
+        if (!audio_is_global_session(sessionId)) {
+            ALOGW("checkEffectCompatibility_l(): non global effect %s on VIRTUALIZER_STAGE"
+                    " thread %s", desc->name, mThreadName);
+            return BAD_VALUE;
+        }
+        break;
     default:
         LOG_ALWAYS_FATAL("checkEffectCompatibility_l(): wrong thread type %d", mType);
     }
@@ -1477,11 +1508,11 @@
         if (effect->isHapticGenerator()) {
             // TODO(b/184194057): Use the vibrator information from the vibrator that will be used
             // for the HapticGenerator.
-            const media::AudioVibratorInfo* defaultVibratorInfo =
-                    mAudioFlinger->getDefaultVibratorInfo_l();
-            if (defaultVibratorInfo != nullptr) {
+            const std::optional<media::AudioVibratorInfo> defaultVibratorInfo =
+                    std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+            if (defaultVibratorInfo) {
                 // Only set the vibrator info when it is a valid one.
-                effect->setVibratorInfo(defaultVibratorInfo);
+                effect->setVibratorInfo(*defaultVibratorInfo);
             }
         }
         // create effect handle and connect it to effect module
@@ -1489,6 +1520,7 @@
         lStatus = handle->initCheck();
         if (lStatus == OK) {
             lStatus = effect->addHandle(handle.get());
+            sendCheckOutputStageEffectsEvent_l();
         }
         if (enabled != NULL) {
             *enabled = (int)effect->isEnabled();
@@ -1531,6 +1563,7 @@
         if (remove) {
             removeEffect_l(effect, true);
         }
+        sendCheckOutputStageEffectsEvent_l();
     }
     if (remove) {
         mAudioFlinger->updateOrphanEffectChains(effect);
@@ -1888,6 +1921,14 @@
     item->selfrecord();
 }
 
+product_strategy_t AudioFlinger::ThreadBase::getStrategyForStream(audio_stream_type_t stream) const
+{
+    if (!mAudioFlinger->isAudioPolicyReady()) {
+        return PRODUCT_STRATEGY_NONE;
+    }
+    return AudioSystem::getStrategyForStream(stream);
+}
+
 // ----------------------------------------------------------------------------
 //      Playback
 // ----------------------------------------------------------------------------
@@ -1896,15 +1937,16 @@
                                              AudioStreamOut* output,
                                              audio_io_handle_t id,
                                              type_t type,
-                                             bool systemReady)
+                                             bool systemReady,
+                                             audio_config_base_t *mixerConfig)
     :   ThreadBase(audioFlinger, id, type, systemReady, true /* isOut */),
         mNormalFrameCount(0), mSinkBuffer(NULL),
-        mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
+        mMixerBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == VIRTUALIZER_STAGE),
         mMixerBuffer(NULL),
         mMixerBufferSize(0),
         mMixerBufferFormat(AUDIO_FORMAT_INVALID),
         mMixerBufferValid(false),
-        mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision),
+        mEffectBufferEnabled(AudioFlinger::kEnableExtendedPrecision || type == VIRTUALIZER_STAGE),
         mEffectBuffer(NULL),
         mEffectBufferSize(0),
         mEffectBufferFormat(AUDIO_FORMAT_INVALID),
@@ -1956,8 +1998,18 @@
                 mOutput->audioHwDev->moduleName(), AUDIO_HARDWARE_MODULE_ID_MSD) == 0;
     }
 
+    if (mixerConfig != nullptr && mixerConfig->channel_mask != AUDIO_CHANNEL_NONE) {
+        mMixerChannelMask = mixerConfig->channel_mask;
+    }
+
     readOutputParameters_l();
 
+    if (mType != VIRTUALIZER_STAGE
+            && mMixerChannelMask != mChannelMask) {
+        LOG_ALWAYS_FATAL("HAL channel mask %#x does not match mixer channel mask %#x",
+                mChannelMask, mMixerChannelMask);
+    }
+
     // TODO: We may also match on address as well as device type for
     // AUDIO_DEVICE_OUT_BUS, AUDIO_DEVICE_OUT_ALL_A2DP, AUDIO_DEVICE_OUT_REMOTE_SUBMIX
     if (type == MIXER || type == DIRECT || type == OFFLOAD) {
@@ -2080,10 +2132,12 @@
     write(fd, result.string(), result.size());
 }
 
-void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args __unused)
+void AudioFlinger::PlaybackThread::dumpInternals_l(int fd, const Vector<String16>& args)
 {
     dprintf(fd, "  Master volume: %f\n", mMasterVolume);
     dprintf(fd, "  Master mute: %s\n", mMasterMute ? "on" : "off");
+    dprintf(fd, "  Mixer channel Mask: %#x (%s)\n",
+            mMixerChannelMask, channelMaskToString(mMixerChannelMask, true /* output */).c_str());
     if (mHapticChannelMask != AUDIO_CHANNEL_NONE) {
         dprintf(fd, "  Haptic channel mask: %#x (%s)\n", mHapticChannelMask,
                 channelMaskToString(mHapticChannelMask, true /* output */).c_str());
@@ -2109,7 +2163,7 @@
     }
     if (output != nullptr) {
         dprintf(fd, "  Hal stream dump:\n");
-        (void)output->stream->dump(fd);
+        (void)output->stream->dump(fd, args);
     }
 }
 
@@ -2397,11 +2451,11 @@
         // all tracks in same audio session must share the same routing strategy otherwise
         // conflicts will happen when tracks are moved from one output to another by audio policy
         // manager
-        product_strategy_t strategy = AudioSystem::getStrategyForStream(streamType);
+        product_strategy_t strategy = getStrategyForStream(streamType);
         for (size_t i = 0; i < mTracks.size(); ++i) {
             sp<Track> t = mTracks[i];
             if (t != 0 && t->isExternalTrack()) {
-                product_strategy_t actual = AudioSystem::getStrategyForStream(t->streamType());
+                product_strategy_t actual = getStrategyForStream(t->streamType());
                 if (sessionId == t->sessionId() && strategy != actual) {
                     ALOGE("createTrack_l() mismatched strategy; expected %u but found %u",
                             strategy, actual);
@@ -2445,7 +2499,7 @@
         if (chain != 0) {
             ALOGV("createTrack_l() setting main buffer %p", chain->inBuffer());
             track->setMainBuffer(chain->inBuffer());
-            chain->setStrategy(AudioSystem::getStrategyForStream(track->streamType()));
+            chain->setStrategy(getStrategyForStream(track->streamType()));
             chain->incTrackCnt();
         }
 
@@ -2613,8 +2667,19 @@
             mLock.unlock();
             const int intensity = AudioFlinger::onExternalVibrationStart(
                     track->getExternalVibration());
+            std::optional<media::AudioVibratorInfo> vibratorInfo;
+            {
+                // TODO(b/184194780): Use the vibrator information from the vibrator that will be
+                // used to play this track.
+                Mutex::Autolock _l(mAudioFlinger->mLock);
+                vibratorInfo = std::move(mAudioFlinger->getDefaultVibratorInfo_l());
+            }
             mLock.lock();
             track->setHapticIntensity(static_cast<os::HapticScale>(intensity));
+            if (vibratorInfo) {
+                track->setHapticMaxAmplitude(vibratorInfo->maxAmplitude);
+            }
+
             // Haptic playback should be enabled by vibrator service.
             if (track->getHapticPlaybackEnabled()) {
                 // Disable haptic playback of all active track to ensure only
@@ -2707,36 +2772,26 @@
     return mOutput->stream->selectPresentation(presentationId, programId);
 }
 
-void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event event, pid_t pid,
+void AudioFlinger::PlaybackThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
                                                    audio_port_handle_t portId) {
-    sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
     ALOGV("PlaybackThread::ioConfigChanged, thread %p, event %d", this, event);
-
-    desc->mIoHandle = mId;
-    struct audio_patch patch = mPatch;
-    if (isMsdDevice()) {
-        patch = mDownStreamPatch;
-    }
-
+    sp<AudioIoDescriptor> desc;
+    const struct audio_patch patch = isMsdDevice() ? mDownStreamPatch : mPatch;
     switch (event) {
     case AUDIO_OUTPUT_OPENED:
     case AUDIO_OUTPUT_REGISTERED:
     case AUDIO_OUTPUT_CONFIG_CHANGED:
-        desc->mPatch = patch;
-        desc->mChannelMask = mChannelMask;
-        desc->mSamplingRate = mSampleRate;
-        desc->mFormat = mFormat;
-        desc->mFrameCount = mNormalFrameCount; // FIXME see
-                                             // AudioFlinger::frameCount(audio_io_handle_t)
-        desc->mFrameCountHAL = mFrameCount;
-        desc->mLatency = latency_l();
+        desc = sp<AudioIoDescriptor>::make(mId, patch, false /*isInput*/,
+                mSampleRate, mFormat, mChannelMask,
+                // FIXME AudioFlinger::frameCount(audio_io_handle_t) instead of mNormalFrameCount?
+                mNormalFrameCount, mFrameCount, latency_l());
         break;
     case AUDIO_CLIENT_STARTED:
-        desc->mPatch = patch;
-        desc->mPortId = portId;
+        desc = sp<AudioIoDescriptor>::make(mId, patch, portId);
         break;
     case AUDIO_OUTPUT_CLOSED:
     default:
+        desc = sp<AudioIoDescriptor>::make(mId);
         break;
     }
     mAudioFlinger->ioConfigChanged(event, desc, pid);
@@ -2814,14 +2869,20 @@
     if (!audio_is_output_channel(mChannelMask)) {
         LOG_ALWAYS_FATAL("HAL channel mask %#x not valid for output", mChannelMask);
     }
-    if ((mType == MIXER || mType == DUPLICATING)
-            && !isValidPcmSinkChannelMask(mChannelMask)) {
+    if (hasMixer() && !isValidPcmSinkChannelMask(mChannelMask)) {
         LOG_ALWAYS_FATAL("HAL channel mask %#x not supported for mixed output",
                 mChannelMask);
     }
+
+    if (mMixerChannelMask == AUDIO_CHANNEL_NONE) {
+        mMixerChannelMask = mChannelMask;
+    }
+
     mChannelCount = audio_channel_count_from_out_mask(mChannelMask);
     mBalance.setChannelMask(mChannelMask);
 
+    uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mMixerChannelMask);
+
     // Get actual HAL format.
     status_t result = mOutput->stream->getAudioProperties(nullptr, nullptr, &mHALFormat);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error when retrieving output stream format: %d", result);
@@ -2831,8 +2892,7 @@
     if (!audio_is_valid_format(mFormat)) {
         LOG_ALWAYS_FATAL("HAL format %#x not valid for output", mFormat);
     }
-    if ((mType == MIXER || mType == DUPLICATING)
-            && !isValidPcmSinkFormat(mFormat)) {
+    if (hasMixer() && !isValidPcmSinkFormat(mFormat)) {
         LOG_FATAL("HAL format %#x not supported for mixed output",
                 mFormat);
     }
@@ -2841,7 +2901,7 @@
     LOG_ALWAYS_FATAL_IF(result != OK,
             "Error when retrieving output stream buffer size: %d", result);
     mFrameCount = mBufferSize / mFrameSize;
-    if ((mType == MIXER || mType == DUPLICATING) && (mFrameCount & 15)) {
+    if (hasMixer() && (mFrameCount & 15)) {
         ALOGW("HAL output buffer size is %zu frames but AudioMixer requires multiples of 16 frames",
                 mFrameCount);
     }
@@ -2914,7 +2974,7 @@
     }
     mNormalFrameCount = multiplier * mFrameCount;
     // round up to nearest 16 frames to satisfy AudioMixer
-    if (mType == MIXER || mType == DUPLICATING) {
+    if (hasMixer()) {
         mNormalFrameCount = (mNormalFrameCount + 15) & ~15;
     }
     ALOGI("HAL output buffer size %zu frames, normal sink buffer size %zu frames", mFrameCount,
@@ -2941,7 +3001,7 @@
     mMixerBuffer = NULL;
     if (mMixerBufferEnabled) {
         mMixerBufferFormat = AUDIO_FORMAT_PCM_FLOAT; // no longer valid: AUDIO_FORMAT_PCM_16_BIT.
-        mMixerBufferSize = mNormalFrameCount * mChannelCount
+        mMixerBufferSize = mNormalFrameCount * mixerChannelCount
                 * audio_bytes_per_sample(mMixerBufferFormat);
         (void)posix_memalign(&mMixerBuffer, 32, mMixerBufferSize);
     }
@@ -2949,7 +3009,7 @@
     mEffectBuffer = NULL;
     if (mEffectBufferEnabled) {
         mEffectBufferFormat = EFFECT_BUFFER_FORMAT;
-        mEffectBufferSize = mNormalFrameCount * mChannelCount
+        mEffectBufferSize = mNormalFrameCount * mixerChannelCount
                 * audio_bytes_per_sample(mEffectBufferFormat);
         (void)posix_memalign(&mEffectBuffer, 32, mEffectBufferSize);
     }
@@ -2958,6 +3018,7 @@
     mChannelMask = static_cast<audio_channel_mask_t>(mChannelMask & ~mHapticChannelMask);
     mHapticChannelCount = audio_channel_count_from_out_mask(mHapticChannelMask);
     mChannelCount -= mHapticChannelCount;
+    mMixerChannelMask = static_cast<audio_channel_mask_t>(mMixerChannelMask & ~mHapticChannelMask);
 
     // force reconfiguration of effect chains and engines to take new buffer size and audio
     // parameters into account
@@ -3051,15 +3112,15 @@
     // session AUDIO_SESSION_OUTPUT_MIX is placed in same strategy as MUSIC stream so that
     // it is moved to correct output by audio policy manager when A2DP is connected or disconnected
     if (sessionId == AUDIO_SESSION_OUTPUT_MIX) {
-        return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+        return getStrategyForStream(AUDIO_STREAM_MUSIC);
     }
     for (size_t i = 0; i < mTracks.size(); i++) {
         sp<Track> track = mTracks[i];
         if (sessionId == track->sessionId() && !track->isInvalid()) {
-            return AudioSystem::getStrategyForStream(track->streamType());
+            return getStrategyForStream(track->streamType());
         }
     }
-    return AudioSystem::getStrategyForStream(AUDIO_STREAM_MUSIC);
+    return getStrategyForStream(AUDIO_STREAM_MUSIC);
 }
 
 
@@ -3348,7 +3409,8 @@
         // Only one effect chain can be present in direct output thread and it uses
         // the sink buffer as input
         if (mType != DIRECT) {
-            size_t numSamples = mNormalFrameCount * (mChannelCount + mHapticChannelCount);
+            size_t numSamples = mNormalFrameCount
+                    * (audio_channel_count_from_out_mask(mMixerChannelMask) + mHapticChannelCount);
             status_t result = mAudioFlinger->mEffectsFactoryHal->allocateBuffer(
                     numSamples * sizeof(effect_buffer_t),
                     &halInBuffer);
@@ -3531,6 +3593,8 @@
 
     audio_patch_handle_t lastDownstreamPatchHandle = AUDIO_PATCH_HANDLE_NONE;
 
+    sendCheckOutputStageEffectsEvent();
+
     // loopCount is used for statistics and diagnostics.
     for (int64_t loopCount = 0; !exitPending(); ++loopCount)
     {
@@ -3587,11 +3651,18 @@
             }
         }
 
+        if (mCheckOutputStageEffects.exchange(false)) {
+            checkOutputStageEffects();
+        }
+
         { // scope for mLock
 
             Mutex::Autolock _l(mLock);
 
             processConfigEvents_l();
+            if (mCheckOutputStageEffects.load()) {
+                continue;
+            }
 
             // See comment at declaration of logString for why this is done under mLock
             if (logString != NULL) {
@@ -3757,6 +3828,8 @@
             if (mMixerBufferValid) {
                 void *buffer = mEffectBufferValid ? mEffectBuffer : mSinkBuffer;
                 audio_format_t format = mEffectBufferValid ? mEffectBufferFormat : mFormat;
+                uint32_t channelCount = mEffectBufferValid ?
+                            audio_channel_count_from_out_mask(mMixerChannelMask) : mChannelCount;
 
                 // mono blend occurs for mixer threads only (not direct or offloaded)
                 // and is handled here if we're going directly to the sink.
@@ -3774,7 +3847,7 @@
                 }
 
                 memcpy_by_audio_format(buffer, format, mMixerBuffer, mMixerBufferFormat,
-                        mNormalFrameCount * (mChannelCount + mHapticChannelCount));
+                        mNormalFrameCount * (channelCount + mHapticChannelCount));
 
                 // If we're going directly to the sink and there are haptic channels,
                 // we should adjust channels as the sample data is partially interleaved
@@ -4448,8 +4521,8 @@
 // ----------------------------------------------------------------------------
 
 AudioFlinger::MixerThread::MixerThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-        audio_io_handle_t id, bool systemReady, type_t type)
-    :   PlaybackThread(audioFlinger, output, id, type, systemReady),
+        audio_io_handle_t id, bool systemReady, type_t type, audio_config_base_t *mixerConfig)
+    :   PlaybackThread(audioFlinger, output, id, type, systemReady, mixerConfig),
         // mAudioMixer below
         // mFastMixer below
         mFastMixerFutex(0),
@@ -4566,6 +4639,7 @@
         fastTrack->mFormat = mFormat; // mPipeSink format for audio to FastMixer
         fastTrack->mHapticPlaybackEnabled = mHapticChannelMask != AUDIO_CHANNEL_NONE;
         fastTrack->mHapticIntensity = os::HapticScale::NONE;
+        fastTrack->mHapticMaxAmplitude = NAN;
         fastTrack->mGeneration++;
         state->mFastTracksGen++;
         state->mTrackMask = 1;
@@ -5103,6 +5177,7 @@
                     fastTrack->mFormat = track->mFormat;
                     fastTrack->mHapticPlaybackEnabled = track->getHapticPlaybackEnabled();
                     fastTrack->mHapticIntensity = track->getHapticIntensity();
+                    fastTrack->mHapticMaxAmplitude = track->getHapticMaxAmplitude();
                     fastTrack->mGeneration++;
                     state->mTrackMask |= 1 << j;
                     didModify = true;
@@ -5356,7 +5431,7 @@
                 trackId,
                 AudioMixer::TRACK,
                 AudioMixer::MIXER_CHANNEL_MASK,
-                (void *)(uintptr_t)(mChannelMask | mHapticChannelMask));
+                (void *)(uintptr_t)(mMixerChannelMask | mHapticChannelMask));
             // limit track sample rate to 2 x output sample rate, which changes at re-configuration
             uint32_t maxSampleRate = mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX;
             uint32_t reqSampleRate = proxy->getSampleRate();
@@ -5425,6 +5500,10 @@
                 trackId,
                 AudioMixer::TRACK,
                 AudioMixer::HAPTIC_INTENSITY, (void *)(uintptr_t)track->getHapticIntensity());
+            mAudioMixer->setParameter(
+                trackId,
+                AudioMixer::TRACK,
+                AudioMixer::HAPTIC_MAX_AMPLITUDE, (void *)(&(track->mHapticMaxAmplitude)));
 
             // reset retry count
             track->mRetryCount = kMaxTrackRetries;
@@ -5575,7 +5654,8 @@
     // remove all the tracks that need to be...
     removeTracks_l(*tracksToRemove);
 
-    if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0) {
+    if (getEffectChain_l(AUDIO_SESSION_OUTPUT_MIX) != 0 ||
+            getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE) != 0) {
         mEffectBufferValid = true;
     }
 
@@ -6971,6 +7051,69 @@
     MixerThread::cacheParameters_l();
 }
 
+// ----------------------------------------------------------------------------
+
+AudioFlinger::VirtualizerStageThread::VirtualizerStageThread(const sp<AudioFlinger>& audioFlinger,
+                                                             AudioStreamOut* output,
+                                                             audio_io_handle_t id,
+                                                             bool systemReady,
+                                                             audio_config_base_t *mixerConfig)
+    : MixerThread(audioFlinger, output, id, systemReady, VIRTUALIZER_STAGE, mixerConfig)
+{
+}
+
+void AudioFlinger::VirtualizerStageThread::checkOutputStageEffects()
+{
+    bool hasVirtualizer = false;
+    bool hasDownMixer = false;
+    sp<EffectHandle> finalDownMixer;
+    {
+        Mutex::Autolock _l(mLock);
+        sp<EffectChain> chain = getEffectChain_l(AUDIO_SESSION_OUTPUT_STAGE);
+        if (chain != 0) {
+            hasVirtualizer = chain->getEffectFromType_l(FX_IID_VIRTUALIZER_STAGE) != nullptr;
+            hasDownMixer = chain->getEffectFromType_l(EFFECT_UIID_DOWNMIX) != nullptr;
+        }
+
+        finalDownMixer = mFinalDownMixer;
+        mFinalDownMixer.clear();
+    }
+
+    if (hasVirtualizer) {
+        if (finalDownMixer != nullptr) {
+            int32_t ret;
+            finalDownMixer->disable(&ret);
+        }
+        finalDownMixer.clear();
+    } else if (!hasDownMixer) {
+        std::vector<effect_descriptor_t> descriptors;
+        status_t status = mAudioFlinger->mEffectsFactoryHal->getDescriptors(
+                                                        EFFECT_UIID_DOWNMIX, &descriptors);
+        if (status != NO_ERROR) {
+            return;
+        }
+        ALOG_ASSERT(!descriptors.empty(),
+                "%s getDescriptors() returned no error but empty list", __func__);
+
+        finalDownMixer = createEffect_l(nullptr /*client*/, nullptr /*effectClient*/,
+                0 /*priority*/, AUDIO_SESSION_OUTPUT_STAGE, &descriptors[0], nullptr /*enabled*/,
+                &status, false /*pinned*/, false /*probe*/);
+
+        if (finalDownMixer == nullptr || (status != NO_ERROR && status != ALREADY_EXISTS)) {
+            ALOGW("%s error creating downmixer %d", __func__, status);
+            finalDownMixer.clear();
+        } else {
+            int32_t ret;
+            finalDownMixer->enable(&ret);
+        }
+    }
+
+    {
+        Mutex::Autolock _l(mLock);
+        mFinalDownMixer = finalDownMixer;
+    }
+}
+
 
 // ----------------------------------------------------------------------------
 //      Record
@@ -8631,30 +8774,22 @@
     return String8();
 }
 
-void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event event, pid_t pid,
+void AudioFlinger::RecordThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
                                                  audio_port_handle_t portId) {
-    sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
-
-    desc->mIoHandle = mId;
-
+    sp<AudioIoDescriptor> desc;
     switch (event) {
     case AUDIO_INPUT_OPENED:
     case AUDIO_INPUT_REGISTERED:
     case AUDIO_INPUT_CONFIG_CHANGED:
-        desc->mPatch = mPatch;
-        desc->mChannelMask = mChannelMask;
-        desc->mSamplingRate = mSampleRate;
-        desc->mFormat = mFormat;
-        desc->mFrameCount = mFrameCount;
-        desc->mFrameCountHAL = mFrameCount;
-        desc->mLatency = 0;
+        desc = sp<AudioIoDescriptor>::make(mId, mPatch, true /*isInput*/,
+                mSampleRate, mFormat, mChannelMask, mFrameCount, mFrameCount);
         break;
     case AUDIO_CLIENT_STARTED:
-        desc->mPatch = mPatch;
-        desc->mPortId = portId;
+        desc = sp<AudioIoDescriptor>::make(mId, mPatch, portId);
         break;
     case AUDIO_INPUT_CLOSED:
     default:
+        desc = sp<AudioIoDescriptor>::make(mId);
         break;
     }
     mAudioFlinger->ioConfigChanged(event, desc, pid);
@@ -9278,7 +9413,7 @@
     mActiveTracks.add(track);
     sp<EffectChain> chain = getEffectChain_l(mSessionId);
     if (chain != 0) {
-        chain->setStrategy(AudioSystem::getStrategyForStream(streamType()));
+        chain->setStrategy(getStrategyForStream(streamType()));
         chain->incTrackCnt();
         chain->incActiveTrackCnt();
     }
@@ -9497,31 +9632,26 @@
     return String8();
 }
 
-void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event event, pid_t pid,
+void AudioFlinger::MmapThread::ioConfigChanged(audio_io_config_event_t event, pid_t pid,
                                                audio_port_handle_t portId __unused) {
-    sp<AudioIoDescriptor> desc = new AudioIoDescriptor();
-
-    desc->mIoHandle = mId;
-
+    sp<AudioIoDescriptor> desc;
+    bool isInput = false;
     switch (event) {
     case AUDIO_INPUT_OPENED:
     case AUDIO_INPUT_REGISTERED:
     case AUDIO_INPUT_CONFIG_CHANGED:
+        isInput = true;
+        FALLTHROUGH_INTENDED;
     case AUDIO_OUTPUT_OPENED:
     case AUDIO_OUTPUT_REGISTERED:
     case AUDIO_OUTPUT_CONFIG_CHANGED:
-        desc->mPatch = mPatch;
-        desc->mChannelMask = mChannelMask;
-        desc->mSamplingRate = mSampleRate;
-        desc->mFormat = mFormat;
-        desc->mFrameCount = mFrameCount;
-        desc->mFrameCountHAL = mFrameCount;
-        desc->mLatency = 0;
+        desc = sp<AudioIoDescriptor>::make(mId, mPatch, isInput,
+                mSampleRate, mFormat, mChannelMask, mFrameCount, mFrameCount);
         break;
-
     case AUDIO_INPUT_CLOSED:
     case AUDIO_OUTPUT_CLOSED:
     default:
+        desc = sp<AudioIoDescriptor>::make(mId);
         break;
     }
     mAudioFlinger->ioConfigChanged(event, desc, pid);
diff --git a/services/audioflinger/Threads.h b/services/audioflinger/Threads.h
index 16082a9..38e55a3 100644
--- a/services/audioflinger/Threads.h
+++ b/services/audioflinger/Threads.h
@@ -32,6 +32,7 @@
         OFFLOAD,            // Thread class is OffloadThread
         MMAP_PLAYBACK,      // Thread class for MMAP playback stream
         MMAP_CAPTURE,       // Thread class for MMAP capture stream
+        VIRTUALIZER_STAGE,  //
         // If you add any values here, also update ThreadBase::threadTypeToString()
     };
 
@@ -53,7 +54,8 @@
         CFG_EVENT_CREATE_AUDIO_PATCH,
         CFG_EVENT_RELEASE_AUDIO_PATCH,
         CFG_EVENT_UPDATE_OUT_DEVICE,
-        CFG_EVENT_RESIZE_BUFFER
+        CFG_EVENT_RESIZE_BUFFER,
+        CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS
     };
 
     class ConfigEventData: public RefBase {
@@ -87,7 +89,13 @@
     public:
         virtual ~ConfigEvent() {}
 
-        void dump(char *buffer, size_t size) { mData->dump(buffer, size); }
+        void dump(char *buffer, size_t size) {
+            snprintf(buffer, size, "Event type: %d\n", mType);
+            if (mData != nullptr) {
+                snprintf(buffer, size, "Data:\n");
+                mData->dump(buffer, size);
+            }
+        }
 
         const int mType; // event type e.g. CFG_EVENT_IO
         Mutex mLock;     // mutex associated with mCond
@@ -105,22 +113,22 @@
 
     class IoConfigEventData : public ConfigEventData {
     public:
-        IoConfigEventData(audio_io_config_event event, pid_t pid,
+        IoConfigEventData(audio_io_config_event_t event, pid_t pid,
                           audio_port_handle_t portId) :
             mEvent(event), mPid(pid), mPortId(portId) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "IO event: event %d\n", mEvent);
+            snprintf(buffer, size, "- IO event: event %d\n", mEvent);
         }
 
-        const audio_io_config_event mEvent;
+        const audio_io_config_event_t mEvent;
         const pid_t                 mPid;
         const audio_port_handle_t   mPortId;
     };
 
     class IoConfigEvent : public ConfigEvent {
     public:
-        IoConfigEvent(audio_io_config_event event, pid_t pid, audio_port_handle_t portId) :
+        IoConfigEvent(audio_io_config_event_t event, pid_t pid, audio_port_handle_t portId) :
             ConfigEvent(CFG_EVENT_IO) {
             mData = new IoConfigEventData(event, pid, portId);
         }
@@ -133,7 +141,7 @@
             mPid(pid), mTid(tid), mPrio(prio), mForApp(forApp) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Prio event: pid %d, tid %d, prio %d, for app? %d\n",
+            snprintf(buffer, size, "- Prio event: pid %d, tid %d, prio %d, for app? %d\n",
                     mPid, mTid, mPrio, mForApp);
         }
 
@@ -158,7 +166,7 @@
             mKeyValuePairs(keyValuePairs) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "KeyValue: %s\n", mKeyValuePairs.string());
+            snprintf(buffer, size, "- KeyValue: %s\n", mKeyValuePairs.string());
         }
 
         const String8 mKeyValuePairs;
@@ -181,7 +189,7 @@
             mPatch(patch), mHandle(handle) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+            snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
         }
 
         const struct audio_patch mPatch;
@@ -205,7 +213,7 @@
             mHandle(handle) {}
 
         virtual  void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Patch handle: %u\n", mHandle);
+            snprintf(buffer, size, "- Patch handle: %u\n", mHandle);
         }
 
         audio_patch_handle_t mHandle;
@@ -227,7 +235,7 @@
             mOutDevices(outDevices) {}
 
         virtual void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "Devices: %s", android::toString(mOutDevices).c_str());
+            snprintf(buffer, size, "- Devices: %s", android::toString(mOutDevices).c_str());
         }
 
         DeviceDescriptorBaseVector mOutDevices;
@@ -249,7 +257,7 @@
             mMaxSharedAudioHistoryMs(maxSharedAudioHistoryMs) {}
 
         virtual void dump(char *buffer, size_t size) {
-            snprintf(buffer, size, "mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
+            snprintf(buffer, size, "- mMaxSharedAudioHistoryMs: %d", mMaxSharedAudioHistoryMs);
         }
 
         int32_t mMaxSharedAudioHistoryMs;
@@ -265,6 +273,16 @@
         virtual ~ResizeBufferConfigEvent() {}
     };
 
+    class CheckOutputStageEffectsEvent : public ConfigEvent {
+    public:
+        CheckOutputStageEffectsEvent() :
+            ConfigEvent(CFG_EVENT_CHECK_OUTPUT_STAGE_EFFECTS) {
+        }
+
+        virtual ~CheckOutputStageEffectsEvent() {}
+    };
+
+
     class PMDeathRecipient : public IBinder::DeathRecipient {
     public:
         explicit    PMDeathRecipient(const wp<ThreadBase>& thread) : mThread(thread) {}
@@ -290,8 +308,11 @@
                 // dynamic externally-visible
                 uint32_t    sampleRate() const { return mSampleRate; }
                 audio_channel_mask_t channelMask() const { return mChannelMask; }
+    virtual     audio_channel_mask_t mixerChannelMask() const { return mChannelMask; }
+
                 audio_format_t format() const { return mHALFormat; }
                 uint32_t channelCount() const { return mChannelCount; }
+
                 // Called by AudioFlinger::frameCount(audio_io_handle_t output) and effects,
                 // and returns the [normal mix] buffer's frame count.
     virtual     size_t      frameCount() const = 0;
@@ -311,15 +332,15 @@
                                                     status_t& status) = 0;
     virtual     status_t    setParameters(const String8& keyValuePairs);
     virtual     String8     getParameters(const String8& keys) = 0;
-    virtual     void        ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
+    virtual     void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
                                         audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE) = 0;
                 // sendConfigEvent_l() must be called with ThreadBase::mLock held
                 // Can temporarily release the lock if waiting for a reply from
                 // processConfigEvents_l().
                 status_t    sendConfigEvent_l(sp<ConfigEvent>& event);
-                void        sendIoConfigEvent(audio_io_config_event event, pid_t pid = 0,
+                void        sendIoConfigEvent(audio_io_config_event_t event, pid_t pid = 0,
                                               audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
-                void        sendIoConfigEvent_l(audio_io_config_event event, pid_t pid = 0,
+                void        sendIoConfigEvent_l(audio_io_config_event_t event, pid_t pid = 0,
                                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
                 void        sendPrioConfigEvent(pid_t pid, pid_t tid, int32_t prio, bool forApp);
                 void        sendPrioConfigEvent_l(pid_t pid, pid_t tid, int32_t prio, bool forApp);
@@ -330,7 +351,11 @@
                 status_t    sendUpdateOutDeviceConfigEvent(
                                     const DeviceDescriptorBaseVector& outDevices);
                 void        sendResizeBufferConfigEvent_l(int32_t maxSharedAudioHistoryMs);
+                void        sendCheckOutputStageEffectsEvent();
+                void        sendCheckOutputStageEffectsEvent_l();
+
                 void        processConfigEvents_l();
+    virtual     void        setCheckOutputStageEffects() {}
     virtual     void        cacheParameters_l() = 0;
     virtual     status_t    createAudioPatch_l(const struct audio_patch *patch,
                                                audio_patch_handle_t *handle) = 0;
@@ -574,6 +599,8 @@
                                 return INVALID_OPERATION;
                             }
 
+                product_strategy_t getStrategyForStream(audio_stream_type_t stream) const;
+
     virtual     void        dumpInternals_l(int fd __unused, const Vector<String16>& args __unused)
                             { }
     virtual     void        dumpTracks_l(int fd __unused, const Vector<String16>& args __unused) { }
@@ -824,7 +851,8 @@
     static const nsecs_t kMaxNextBufferDelayNs = 100000000;
 
     PlaybackThread(const sp<AudioFlinger>& audioFlinger, AudioStreamOut* output,
-                   audio_io_handle_t id, type_t type, bool systemReady);
+                   audio_io_handle_t id, type_t type, bool systemReady,
+                   audio_config_base_t *mixerConfig = nullptr);
     virtual             ~PlaybackThread();
 
     // Thread virtuals
@@ -881,6 +909,8 @@
                                 mActiveTracks.updatePowerState(this, true /* force */);
                             }
 
+    virtual     void        checkOutputStageEffects() {}
+
                 void        dumpInternals_l(int fd, const Vector<String16>& args) override;
                 void        dumpTracks_l(int fd, const Vector<String16>& args) override;
 
@@ -942,7 +972,7 @@
                                 { return android_atomic_acquire_load(&mSuspended) > 0; }
 
     virtual     String8     getParameters(const String8& keys);
-    virtual     void        ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
+    virtual     void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
                                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
                 status_t    getRenderPosition(uint32_t *halFrames, uint32_t *dspFrames);
                 // Consider also removing and passing an explicit mMainBuffer initialization
@@ -973,6 +1003,10 @@
 
     virtual     size_t      frameCount() const { return mNormalFrameCount; }
 
+                audio_channel_mask_t mixerChannelMask() const override {
+                    return mMixerChannelMask;
+                }
+
                 status_t    getTimestamp_l(AudioTimestamp& timestamp);
 
                 void        addPatchTrack(const sp<PatchTrack>& track);
@@ -1015,6 +1049,9 @@
 
                 PlaybackThread::Track* getTrackById_l(audio_port_handle_t trackId);
 
+                bool hasMixer() const {
+                    return mType == MIXER || mType == DUPLICATING || mType == VIRTUALIZER_STAGE;
+                }
 protected:
     // updated by readOutputParameters_l()
     size_t                          mNormalFrameCount;  // normal mixer and effects
@@ -1101,6 +1138,9 @@
     // haptic playback.
     audio_channel_mask_t            mHapticChannelMask = AUDIO_CHANNEL_NONE;
     uint32_t                        mHapticChannelCount = 0;
+
+    audio_channel_mask_t            mMixerChannelMask = AUDIO_CHANNEL_NONE;
+
 private:
     // mMasterMute is in both PlaybackThread and in AudioFlinger.  When a
     // PlaybackThread needs to find out if master-muted, it checks it's local
@@ -1134,6 +1174,9 @@
 
     // Cache various calculated values, at threadLoop() entry and after a parameter change
     virtual     void        cacheParameters_l();
+                void        setCheckOutputStageEffects() override {
+                                mCheckOutputStageEffects.store(true);
+                            }
 
     virtual     uint32_t    correctLatency_l(uint32_t latency) const;
 
@@ -1314,6 +1357,8 @@
                 // audio patch used by the downstream software patch.
                 // Only used if ThreadBase::mIsMsdDevice is true.
                 struct audio_patch mDownStreamPatch;
+
+                std::atomic_bool mCheckOutputStageEffects{};
 };
 
 class MixerThread : public PlaybackThread {
@@ -1322,7 +1367,8 @@
                 AudioStreamOut* output,
                 audio_io_handle_t id,
                 bool systemReady,
-                type_t type = MIXER);
+                type_t type = MIXER,
+                audio_config_base_t *mixerConfig = nullptr);
     virtual             ~MixerThread();
 
     // Thread virtuals
@@ -1611,6 +1657,24 @@
     }
 };
 
+class VirtualizerStageThread : public MixerThread {
+public:
+    VirtualizerStageThread(const sp<AudioFlinger>& audioFlinger,
+                           AudioStreamOut* output,
+                           audio_io_handle_t id,
+                           bool systemReady,
+                           audio_config_base_t *mixerConfig);
+            ~VirtualizerStageThread() override {}
+
+            bool hasFastMixer() const override { return false; }
+
+protected:
+            void checkOutputStageEffects() override;
+
+private:
+            sp<EffectHandle> mFinalDownMixer;
+};
+
 // record thread
 class RecordThread : public ThreadBase
 {
@@ -1721,7 +1785,7 @@
                                                status_t& status);
     virtual void        cacheParameters_l() {}
     virtual String8     getParameters(const String8& keys);
-    virtual void        ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
+    virtual void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
                                         audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
     virtual status_t    createAudioPatch_l(const struct audio_patch *patch,
                                            audio_patch_handle_t *handle);
@@ -1930,7 +1994,7 @@
     virtual     bool        checkForNewParameter_l(const String8& keyValuePair,
                                                     status_t& status);
     virtual     String8     getParameters(const String8& keys);
-    virtual     void        ioConfigChanged(audio_io_config_event event, pid_t pid = 0,
+    virtual     void        ioConfigChanged(audio_io_config_event_t event, pid_t pid = 0,
                                             audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE);
                 void        readHalParameters_l();
     virtual     void        cacheParameters_l() {}
diff --git a/services/audiopolicy/AudioPolicyInterface.h b/services/audiopolicy/AudioPolicyInterface.h
index 2e49e71..c9cf564 100644
--- a/services/audiopolicy/AudioPolicyInterface.h
+++ b/services/audiopolicy/AudioPolicyInterface.h
@@ -18,6 +18,7 @@
 #define ANDROID_AUDIOPOLICY_INTERFACE_H
 
 #include <media/AudioCommonTypes.h>
+#include <media/AudioContainers.h>
 #include <media/AudioDeviceTypeAddr.h>
 #include <media/AudioSystem.h>
 #include <media/AudioPolicy.h>
@@ -198,7 +199,7 @@
     virtual product_strategy_t getStrategyForStream(audio_stream_type_t stream) = 0;
 
     // return the enabled output devices for the given stream type
-    virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream) = 0;
+    virtual DeviceTypeSet getDevicesForStream(audio_stream_type_t stream) = 0;
 
     // retrieves the list of enabled output devices for the given audio attributes
     virtual status_t getDevicesForAttributes(const audio_attributes_t &attr,
@@ -359,7 +360,8 @@
     // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
     virtual status_t openOutput(audio_module_handle_t module,
                                 audio_io_handle_t *output,
-                                audio_config_t *config,
+                                audio_config_t *halConfig,
+                                audio_config_base_t *mixerConfig,
                                 const sp<DeviceDescriptorBase>& device,
                                 uint32_t *latencyMs,
                                 audio_output_flags_t flags) = 0;
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
index 1f9b535..7c7f02d 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioOutputDescriptor.h
@@ -362,7 +362,8 @@
                            const struct audio_port_config *srcConfig = NULL) const;
     virtual void toAudioPort(struct audio_port_v7 *port) const;
 
-        status_t open(const audio_config_t *config,
+        status_t open(const audio_config_t *halConfig,
+                      const audio_config_base_t *mixerConfig,
                       const DeviceVector &devices,
                       audio_stream_type_t stream,
                       audio_output_flags_t flags,
@@ -423,6 +424,7 @@
     uint32_t mDirectOpenCount; // number of clients using this output (direct outputs only)
     audio_session_t mDirectClientSession; // session id of the direct output client
     bool mPendingReopenToQueryProfiles = false;
+    audio_channel_mask_t mMixerChannelMask = AUDIO_CHANNEL_NONE;
 };
 
 // Audio output driven by an input device directly.
diff --git a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
index cf1f64c..9837336 100644
--- a/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
+++ b/services/audiopolicy/common/managerdefinitions/include/AudioPolicyConfig.h
@@ -202,6 +202,20 @@
             {AUDIO_FORMAT_AC4, {}}};
     }
 
+    //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+    // until then, use DEEP_BUFFER+FAST flag combo to indicate the virtualizer stage output profile
+    void convertVirtualizerStageFlag()
+    {
+        for (const auto& hwModule : mHwModules) {
+            for (const auto& curProfile : hwModule->getOutputProfiles()) {
+                if (curProfile->getFlags()
+                        == (AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER)) {
+                    curProfile->setFlags(AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE);
+                }
+            }
+        }
+    }
+
 private:
     static const constexpr char* const kDefaultEngineLibraryNameSuffix = "default";
 
diff --git a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
index 20b4044..58d05c6 100644
--- a/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
+++ b/services/audiopolicy/common/managerdefinitions/include/DeviceDescriptor.h
@@ -168,6 +168,10 @@
     DeviceVector getDevicesFromDeviceTypeAddrVec(
             const AudioDeviceTypeAddrVector& deviceTypeAddrVector) const;
 
+    // Return the device vector that contains device descriptor whose AudioDeviceTypeAddr appears
+    // in the given AudioDeviceTypeAddrVector
+    AudioDeviceTypeAddrVector toTypeAddrVector() const;
+
     // If there are devices with the given type and the devices to add is not empty,
     // remove all the devices with the given type and add all the devices to add.
     void replaceDevicesByType(audio_devices_t typeToRemove, const DeviceVector &devicesToAdd);
diff --git a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
index 6b08f7c..6c3386f 100644
--- a/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/AudioOutputDescriptor.cpp
@@ -491,7 +491,8 @@
     return true;
 }
 
-status_t SwAudioOutputDescriptor::open(const audio_config_t *config,
+status_t SwAudioOutputDescriptor::open(const audio_config_t *halConfig,
+                                       const audio_config_base_t *mixerConfig,
                                        const DeviceVector &devices,
                                        audio_stream_type_t stream,
                                        audio_output_flags_t flags,
@@ -504,45 +505,62 @@
                         "with the requested devices, all device types: %s",
                         __func__, dumpDeviceTypes(devices.types()).c_str());
 
-    audio_config_t lConfig;
-    if (config == nullptr) {
-        lConfig = AUDIO_CONFIG_INITIALIZER;
-        lConfig.sample_rate = mSamplingRate;
-        lConfig.channel_mask = mChannelMask;
-        lConfig.format = mFormat;
+    audio_config_t lHalConfig;
+    if (halConfig == nullptr) {
+        lHalConfig = AUDIO_CONFIG_INITIALIZER;
+        lHalConfig.sample_rate = mSamplingRate;
+        lHalConfig.channel_mask = mChannelMask;
+        lHalConfig.format = mFormat;
     } else {
-        lConfig = *config;
+        lHalConfig = *halConfig;
     }
 
     // if the selected profile is offloaded and no offload info was specified,
     // create a default one
     if ((mProfile->getFlags() & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
-            lConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
+            lHalConfig.offload_info.format == AUDIO_FORMAT_DEFAULT) {
         flags = (audio_output_flags_t)(flags | AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD);
-        lConfig.offload_info = AUDIO_INFO_INITIALIZER;
-        lConfig.offload_info.sample_rate = lConfig.sample_rate;
-        lConfig.offload_info.channel_mask = lConfig.channel_mask;
-        lConfig.offload_info.format = lConfig.format;
-        lConfig.offload_info.stream_type = stream;
-        lConfig.offload_info.duration_us = -1;
-        lConfig.offload_info.has_video = true; // conservative
-        lConfig.offload_info.is_streaming = true; // likely
-        lConfig.offload_info.encapsulation_mode = lConfig.offload_info.encapsulation_mode;
-        lConfig.offload_info.content_id = lConfig.offload_info.content_id;
-        lConfig.offload_info.sync_id = lConfig.offload_info.sync_id;
+        lHalConfig.offload_info = AUDIO_INFO_INITIALIZER;
+        lHalConfig.offload_info.sample_rate = lHalConfig.sample_rate;
+        lHalConfig.offload_info.channel_mask = lHalConfig.channel_mask;
+        lHalConfig.offload_info.format = lHalConfig.format;
+        lHalConfig.offload_info.stream_type = stream;
+        lHalConfig.offload_info.duration_us = -1;
+        lHalConfig.offload_info.has_video = true; // conservative
+        lHalConfig.offload_info.is_streaming = true; // likely
+        lHalConfig.offload_info.encapsulation_mode = lHalConfig.offload_info.encapsulation_mode;
+        lHalConfig.offload_info.content_id = lHalConfig.offload_info.content_id;
+        lHalConfig.offload_info.sync_id = lHalConfig.offload_info.sync_id;
+    }
+
+    audio_config_base_t lMixerConfig;
+    if (mixerConfig == nullptr) {
+        lMixerConfig = AUDIO_CONFIG_BASE_INITIALIZER;
+        lMixerConfig.sample_rate = lHalConfig.sample_rate;
+        lMixerConfig.channel_mask = lHalConfig.channel_mask;
+        lMixerConfig.format = lHalConfig.format;
+    } else {
+        lMixerConfig = *mixerConfig;
     }
 
     mFlags = (audio_output_flags_t)(mFlags | flags);
 
+    //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+    audio_output_flags_t halFlags = mFlags;
+    if ((mFlags & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0) {
+        halFlags = (audio_output_flags_t)(AUDIO_OUTPUT_FLAG_FAST | AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
+    }
+
     ALOGV("opening output for device %s profile %p name %s",
           mDevices.toString().c_str(), mProfile.get(), mProfile->getName().c_str());
 
     status_t status = mClientInterface->openOutput(mProfile->getModuleHandle(),
                                                    output,
-                                                   &lConfig,
+                                                   &lHalConfig,
+                                                   &lMixerConfig,
                                                    device,
                                                    &mLatency,
-                                                   mFlags);
+                                                   halFlags);
 
     if (status == NO_ERROR) {
         LOG_ALWAYS_FATAL_IF(*output == AUDIO_IO_HANDLE_NONE,
@@ -550,9 +568,10 @@
                             "selected device %s for opening",
                             __FUNCTION__, *output, devices.toString().c_str(),
                             device->toString().c_str());
-        mSamplingRate = lConfig.sample_rate;
-        mChannelMask = lConfig.channel_mask;
-        mFormat = lConfig.format;
+        mSamplingRate = lHalConfig.sample_rate;
+        mChannelMask = lHalConfig.channel_mask;
+        mFormat = lHalConfig.format;
+        mMixerChannelMask = lMixerConfig.channel_mask;
         mId = PolicyAudioPort::getNextUniqueId();
         mIoHandle = *output;
         mProfile->curOpenCount++;
diff --git a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
index a92d31e..1722032 100644
--- a/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/DeviceDescriptor.cpp
@@ -451,6 +451,14 @@
     return devices;
 }
 
+AudioDeviceTypeAddrVector DeviceVector::toTypeAddrVector() const {
+    AudioDeviceTypeAddrVector result;
+    for (const auto& device : *this) {
+        result.push_back(AudioDeviceTypeAddr(device->type(), device->address()));
+    }
+    return result;
+}
+
 void DeviceVector::replaceDevicesByType(
         audio_devices_t typeToRemove, const DeviceVector &devicesToAdd) {
     DeviceVector devicesToRemove = getDevicesFromType(typeToRemove);
diff --git a/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp b/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
index 8c61b90..5986069 100644
--- a/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
+++ b/services/audiopolicy/common/managerdefinitions/src/PolicyAudioPort.cpp
@@ -30,9 +30,9 @@
 // --- PolicyAudioPort class implementation
 void PolicyAudioPort::attach(const sp<HwModule>& module)
 {
+    mModule = module;
     ALOGV("%s: attaching module %s to port %s",
             __FUNCTION__, getModuleName(), asAudioPort()->getName().c_str());
-    mModule = module;
 }
 
 void PolicyAudioPort::detach()
diff --git a/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml b/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml
index 98415b7..22ff954 100644
--- a/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml
+++ b/services/audiopolicy/config/bluetooth_audio_policy_configuration.xml
@@ -22,6 +22,17 @@
                      samplingRates="8000,16000,24000,32000,44100,48000"
                      channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
         </mixPort>
+        <mixPort name="le audio input" role="sink">
+            <profile name="" format="AUDIO_FORMAT_PCM_16_BIT"
+                     samplingRates="8000,16000,24000,32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+            <profile name="" format="AUDIO_FORMAT_PCM_24_BIT_PACKED"
+                     samplingRates="8000,16000,24000,32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+            <profile name="" format="AUDIO_FORMAT_PCM_32_BIT"
+                     samplingRates="8000,16000,24000,32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+        </mixPort>
     </mixPorts>
     <devicePorts>
         <!-- A2DP Audio Ports -->
@@ -49,6 +60,7 @@
         -->
         <devicePort tagName="BLE Headset Out" type="AUDIO_DEVICE_OUT_BLE_HEADSET" role="sink"/>
         <devicePort tagName="BLE Speaker Out" type="AUDIO_DEVICE_OUT_BLE_SPEAKER" role="sink"/>
+        <devicePort tagName="BLE Headset In" type="AUDIO_DEVICE_IN_BLE_HEADSET" role="source"/>
     </devicePorts>
     <routes>
         <route type="mix" sink="BT A2DP Out"
@@ -61,6 +73,8 @@
                sources="hearing aid output"/>
         <route type="mix" sink="BLE Headset Out"
                sources="le audio output"/>
+        <route type="mix" sink="le audio input"
+               sources="BLE Headset In"/>
         <route type="mix" sink="BLE Speaker Out"
                sources="le audio output"/>
     </routes>
diff --git a/services/audiopolicy/config/le_audio_policy_configuration.xml b/services/audiopolicy/config/le_audio_policy_configuration.xml
index a3dc72b..dcdd805 100644
--- a/services/audiopolicy/config/le_audio_policy_configuration.xml
+++ b/services/audiopolicy/config/le_audio_policy_configuration.xml
@@ -7,13 +7,20 @@
                      samplingRates="8000,16000,24000,32000,44100,48000"
                      channelMasks="AUDIO_CHANNEL_OUT_MONO,AUDIO_CHANNEL_OUT_STEREO"/>
         </mixPort>
+        <mixPort name="le audio input" role="sink">
+            <profile name="" format="AUDIO_FORMAT_PCM_16_BIT,AUDIO_FORMAT_PCM_24_BIT,AUDIO_FORMAT_PCM_32_BIT"
+                     samplingRates="8000,16000,24000,32000,44100,48000"
+                     channelMasks="AUDIO_CHANNEL_IN_MONO,AUDIO_CHANNEL_IN_STEREO"/>
+        </mixPort>
     </mixPorts>
     <devicePorts>
         <devicePort tagName="BLE Headset Out" type="AUDIO_DEVICE_OUT_BLE_HEADSET" role="sink"/>
         <devicePort tagName="BLE Speaker Out" type="AUDIO_DEVICE_OUT_BLE_SPEAKER" role="sink"/>
+        <devicePort tagName="BLE Headset In" type="AUDIO_DEVICE_IN_BLE_HEADSET" role="source"/>
     </devicePorts>
     <routes>
         <route type="mix" sink="BLE Headset Out" sources="le audio output"/>
         <route type="mix" sink="BLE Speaker Out" sources="le audio output"/>
+        <route type="mix" sink="le audio input" sources="BLE Headset In"/>
     </routes>
 </module>
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index cc2d8e8..3833a4c 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -246,9 +246,11 @@
                     sp<SwAudioOutputDescriptor> desc = mOutputs.valueFor(output);
                     // close unused outputs after device disconnection or direct outputs that have
                     // been opened by checkOutputsForDevice() to query dynamic parameters
-                    if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE) ||
-                            (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
-                                (desc->mDirectOpenCount == 0))) {
+                    if ((state == AUDIO_POLICY_DEVICE_STATE_UNAVAILABLE)
+                            || (((desc->mFlags & AUDIO_OUTPUT_FLAG_DIRECT) != 0) &&
+                                (desc->mDirectOpenCount == 0))
+                            || (((desc->mFlags & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0) &&
+                                (desc != mVirtualizerStageOutput))) {
                         clearAudioSourcesForOutput(output);
                         closeOutput(output);
                     }
@@ -925,6 +927,36 @@
     return profile;
 }
 
+sp<IOProfile> AudioPolicyManager::getVirtualizerStageOutputProfile(
+        const audio_config_t *config __unused, const AudioDeviceTypeAddrVector &devices,
+        bool forOpening) const
+{
+    for (const auto& hwModule : mHwModules) {
+        for (const auto& curProfile : hwModule->getOutputProfiles()) {
+            if (curProfile->getFlags() != AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) {
+                continue;
+            }
+            // reject profiles not corresponding to a device currently available
+            DeviceVector supportedDevices = curProfile->getSupportedDevices();
+            if (!mAvailableOutputDevices.containsAtLeastOne(supportedDevices)) {
+                continue;
+            }
+            if (!devices.empty()) {
+                if (supportedDevices.getDevicesFromDeviceTypeAddrVec(devices).size()
+                        != devices.size()) {
+                    continue;
+                }
+            }
+            if (forOpening && !curProfile->canOpenNewIo()) {
+                continue;
+            }
+            ALOGV("%s found profile %s", __func__, curProfile->getName().c_str());
+            return curProfile;
+        }
+    }
+    return nullptr;
+}
+
 audio_io_handle_t AudioPolicyManager::getOutput(audio_stream_type_t stream)
 {
     DeviceVector devices = mEngine->getOutputDevicesForStream(stream, false /*fromCache*/);
@@ -1094,7 +1126,7 @@
 
     *output = AUDIO_IO_HANDLE_NONE;
     if (!msdDevices.isEmpty()) {
-        *output = getOutputForDevices(msdDevices, session, *stream, config, flags);
+        *output = getOutputForDevices(msdDevices, session, resultAttr, config, flags);
         if (*output != AUDIO_IO_HANDLE_NONE && setMsdOutputPatches(&outputDevices) == NO_ERROR) {
             ALOGV("%s() Using MSD devices %s instead of devices %s",
                   __func__, msdDevices.toString().c_str(), outputDevices.toString().c_str());
@@ -1103,7 +1135,7 @@
         }
     }
     if (*output == AUDIO_IO_HANDLE_NONE) {
-        *output = getOutputForDevices(outputDevices, session, *stream, config,
+        *output = getOutputForDevices(outputDevices, session, resultAttr, config,
                 flags, resultAttr->flags & AUDIO_FLAG_MUTE_HAPTIC);
     }
     if (*output == AUDIO_IO_HANDLE_NONE) {
@@ -1265,7 +1297,8 @@
     // all MSD patches to prioritize this request over any active output on MSD.
     releaseMsdOutputPatches(devices);
 
-    status_t status = outputDesc->open(config, devices, stream, flags, output);
+    status_t status =
+            outputDesc->open(config, nullptr /* mixerConfig */, devices, stream, flags, output);
 
     // only accept an output with the requested parameters
     if (status != NO_ERROR ||
@@ -1300,7 +1333,7 @@
 audio_io_handle_t AudioPolicyManager::getOutputForDevices(
         const DeviceVector &devices,
         audio_session_t session,
-        audio_stream_type_t stream,
+        const audio_attributes_t *attr,
         const audio_config_t *config,
         audio_output_flags_t *flags,
         bool forceMutingHaptic)
@@ -1322,6 +1355,9 @@
     if ((*flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC) != 0) {
         *flags = (audio_output_flags_t)(*flags | AUDIO_OUTPUT_FLAG_DIRECT);
     }
+
+    audio_stream_type_t stream = mEngine->getStreamTypeForAttributes(*attr);
+
     // only allow deep buffering for music stream type
     if (stream != AUDIO_STREAM_MUSIC) {
         *flags = (audio_output_flags_t)(*flags &~AUDIO_OUTPUT_FLAG_DEEP_BUFFER);
@@ -1341,6 +1377,11 @@
         ALOGV("Set VoIP and Direct output flags for PCM format");
     }
 
+    if (mVirtualizerStageOutput != nullptr
+            && canBeVirtualized(attr, config, devices.toTypeAddrVector())) {
+        return mVirtualizerStageOutput->mIoHandle;
+    }
+
     audio_config_t directConfig = *config;
     directConfig.channel_mask = channelMask;
     status_t status = openDirectOutput(stream, session, &directConfig, *flags, devices, &output);
@@ -2123,8 +2164,9 @@
                                              audio_port_handle_t *portId)
 {
     ALOGV("%s() source %d, sampling rate %d, format %#x, channel mask %#x, session %d, "
-          "flags %#x attributes=%s", __func__, attr->source, config->sample_rate,
-          config->format, config->channel_mask, session, flags, toString(*attr).c_str());
+          "flags %#x attributes=%s requested device ID %d",
+          __func__, attr->source, config->sample_rate, config->format, config->channel_mask,
+          session, flags, toString(*attr).c_str(), *selectedDeviceId);
 
     status_t status = NO_ERROR;
     audio_source_t halInputSource;
@@ -4802,6 +4844,136 @@
     return source;
 }
 
+bool AudioPolicyManager::canBeVirtualized(const audio_attributes_t *attr,
+                                      const audio_config_t *config,
+                                      const AudioDeviceTypeAddrVector &devices)  const
+{
+    // The caller can have the audio attributes criteria ignored by either passing a null ptr or
+    // the AUDIO_ATTRIBUTES_INITIALIZER value.
+    // If attributes are specified, current policy is to only allow virtualization for media
+    // and game usages.
+    if (attr != nullptr && *attr != AUDIO_ATTRIBUTES_INITIALIZER &&
+            attr->usage != AUDIO_USAGE_MEDIA && attr->usage != AUDIO_USAGE_GAME) {
+        return false;
+    }
+
+    // The caller can have the devices criteria ignored by passing and empty vector, and
+    // getVirtualizerStageOutputProfile() will ignore the devices when looking for a match.
+    // Otherwise an output profile supporting a virtualizer stage effect that can be routed
+    // to the specified devices must exist.
+    sp<IOProfile> profile =
+            getVirtualizerStageOutputProfile(config, devices, false /*forOpening*/);
+    if (profile == nullptr) {
+        return false;
+    }
+
+    // The caller can have the audio config criteria ignored by either passing a null ptr or
+    // the AUDIO_CONFIG_INITIALIZER value.
+    // If an audio config is specified, current policy is to only allow virtualization for
+    // 5.1, 7.1and 7.1.4 audio.
+    // If the virtualizer stage output is already opened, only channel masks included in the
+    // virtualizer stage output mixer channel mask are allowed.
+    if (config != nullptr && *config != AUDIO_CONFIG_INITIALIZER) {
+        if (config->channel_mask != AUDIO_CHANNEL_OUT_5POINT1
+                && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1
+                && config->channel_mask != AUDIO_CHANNEL_OUT_7POINT1POINT4) {
+            return false;
+        }
+        if (mVirtualizerStageOutput != nullptr) {
+            if ((config->channel_mask & mVirtualizerStageOutput->mMixerChannelMask)
+                    != config->channel_mask) {
+                return false;
+            }
+        }
+    }
+
+    return true;
+}
+
+void AudioPolicyManager::checkVirtualizerClientRoutes() {
+    std::set<audio_stream_type_t> streamsToInvalidate;
+    for (size_t i = 0; i < mOutputs.size(); i++) {
+        const sp<SwAudioOutputDescriptor>& outputDescriptor = mOutputs[i];
+        for (const sp<TrackClientDescriptor>& client : outputDescriptor->getClientIterable()) {
+            audio_attributes_t attr = client->attributes();
+            DeviceVector devices = mEngine->getOutputDevicesForAttributes(attr, nullptr, false);
+            AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
+            audio_config_base_t clientConfig = client->config();
+            audio_config_t config = audio_config_initializer(&clientConfig);
+            if (canBeVirtualized(&attr, &config, devicesTypeAddress)) {
+                streamsToInvalidate.insert(client->stream());
+            }
+        }
+    }
+
+    for (audio_stream_type_t stream : streamsToInvalidate) {
+        mpClientInterface->invalidateStream(stream);
+    }
+}
+
+status_t AudioPolicyManager::getVirtualizerStageOutput(const audio_config_base_t *mixerConfig,
+                                                        const audio_attributes_t *attr,
+                                                        audio_io_handle_t *output) {
+    *output = AUDIO_IO_HANDLE_NONE;
+
+    if (mVirtualizerStageOutput != nullptr) {
+        return INVALID_OPERATION;
+    }
+
+    DeviceVector devices = mEngine->getOutputDevicesForAttributes(*attr, nullptr, false);
+    AudioDeviceTypeAddrVector devicesTypeAddress = devices.toTypeAddrVector();
+    audio_config_t *configPtr = nullptr;
+    audio_config_t config;
+    if (mixerConfig != nullptr) {
+        config = audio_config_initializer(mixerConfig);
+        configPtr = &config;
+    }
+    if (!canBeVirtualized(attr, configPtr, devicesTypeAddress)) {
+        return BAD_VALUE;
+    }
+
+    sp<IOProfile> profile =
+            getVirtualizerStageOutputProfile(configPtr, devicesTypeAddress, true /*forOpening*/);
+    if (profile == nullptr) {
+        return BAD_VALUE;
+    }
+
+    mVirtualizerStageOutput = new SwAudioOutputDescriptor(profile, mpClientInterface);
+    status_t status = mVirtualizerStageOutput->open(nullptr, mixerConfig, devices,
+                                                    mEngine->getStreamTypeForAttributes(*attr),
+                                                    AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE, output);
+    if (status != NO_ERROR) {
+        ALOGV("%s failed opening output: status %d, output %d", __func__, status, *output);
+        if (*output != AUDIO_IO_HANDLE_NONE) {
+            mVirtualizerStageOutput->close();
+        }
+        mVirtualizerStageOutput.clear();
+        *output = AUDIO_IO_HANDLE_NONE;
+        return status;
+    }
+
+    checkVirtualizerClientRoutes();
+
+    addOutput(*output, mVirtualizerStageOutput);
+    mPreviousOutputs = mOutputs;
+    mpClientInterface->onAudioPortListUpdate();
+
+    ALOGV("%s returns new virtualizer stage output %d", __func__, *output);
+    return NO_ERROR;
+}
+
+status_t AudioPolicyManager::releaseVirtualizerStageOutput(audio_io_handle_t output) {
+    if (mVirtualizerStageOutput == nullptr) {
+        return INVALID_OPERATION;
+    }
+    if (mVirtualizerStageOutput->mIoHandle != output) {
+        return BAD_VALUE;
+    }
+    closeOutput(output);
+    mVirtualizerStageOutput.clear();
+    return NO_ERROR;
+}
+
 // ----------------------------------------------------------------------------
 // AudioPolicyManager
 // ----------------------------------------------------------------------------
@@ -4851,6 +5023,8 @@
         ALOGE("could not load audio policy configuration file, setting defaults");
         getConfig().setDefault();
     }
+    //TODO: b/193496180 use virtualizer stage flag at audio HAL when available
+    getConfig().convertVirtualizerStageFlag();
 }
 
 status_t AudioPolicyManager::initialize() {
@@ -4990,7 +5164,8 @@
             sp<SwAudioOutputDescriptor> outputDesc = new SwAudioOutputDescriptor(outProfile,
                                                                                  mpClientInterface);
             audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-            status_t status = outputDesc->open(nullptr, DeviceVector(supportedDevice),
+            status_t status = outputDesc->open(nullptr /* halConfig */, nullptr /* mixerConfig */,
+                                               DeviceVector(supportedDevice),
                                                AUDIO_STREAM_DEFAULT,
                                                AUDIO_OUTPUT_FLAG_NONE, &output);
             if (status != NO_ERROR) {
@@ -5012,7 +5187,8 @@
                     outProfile->getFlags() & AUDIO_OUTPUT_FLAG_PRIMARY) {
                 mPrimaryOutput = outputDesc;
             }
-            if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0) {
+            if ((outProfile->getFlags() & AUDIO_OUTPUT_FLAG_DIRECT) != 0
+                || (outProfile->getFlags() & AUDIO_OUTPUT_FLAG_VIRTUALIZER_STAGE) != 0 ) {
                 outputDesc->close();
             } else {
                 addOutput(output, outputDesc);
@@ -5917,13 +6093,13 @@
     return (stream1 == stream2);
 }
 
-audio_devices_t AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
+DeviceTypeSet AudioPolicyManager::getDevicesForStream(audio_stream_type_t stream) {
     // By checking the range of stream before calling getStrategy, we avoid
     // getOutputDevicesForStream's behavior for invalid streams.
     // engine's getOutputDevicesForStream would fallback on its default behavior (most probably
     // device for music stream), but we want to return the empty set.
     if (stream < AUDIO_STREAM_MIN || stream >= AUDIO_STREAM_PUBLIC_CNT) {
-        return AUDIO_DEVICE_NONE;
+        return DeviceTypeSet{};
     }
     DeviceVector activeDevices;
     DeviceVector devices;
@@ -5954,8 +6130,7 @@
         devices.merge(mAvailableOutputDevices.getDevicesFromType(AUDIO_DEVICE_OUT_SPEAKER));
         devices.remove(speakerSafeDevices);
     }
-    // FIXME: use DeviceTypeSet when Java layer is ready for it.
-    return deviceTypesToBitMask(devices.types());
+    return devices.types();
 }
 
 status_t AudioPolicyManager::getDevicesForAttributes(
@@ -6995,7 +7170,7 @@
     }
     sp<SwAudioOutputDescriptor> desc = new SwAudioOutputDescriptor(profile, mpClientInterface);
     audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-    status_t status = desc->open(nullptr, devices,
+    status_t status = desc->open(nullptr /* halConfig */, nullptr /* mixerConfig */, devices,
             AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
     if (status != NO_ERROR) {
         return nullptr;
@@ -7025,7 +7200,7 @@
         config.offload_info.channel_mask = config.channel_mask;
         config.offload_info.format = config.format;
 
-        status = desc->open(&config, devices,
+        status = desc->open(&config, nullptr /* mixerConfig */, devices,
                             AUDIO_STREAM_DEFAULT, AUDIO_OUTPUT_FLAG_NONE, &output);
         if (status != NO_ERROR) {
             return nullptr;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 98f96d1..5cafef8 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -193,7 +193,7 @@
         }
 
         // return the enabled output devices for the given stream type
-        virtual audio_devices_t getDevicesForStream(audio_stream_type_t stream);
+        virtual DeviceTypeSet getDevicesForStream(audio_stream_type_t stream);
 
         virtual status_t getDevicesForAttributes(
                 const audio_attributes_t &attributes,
@@ -356,6 +356,16 @@
                     BAD_VALUE : NO_ERROR;
         }
 
+        virtual bool canBeVirtualized(const audio_attributes_t *attr,
+                                      const audio_config_t *config,
+                                      const AudioDeviceTypeAddrVector &devices) const;
+
+        virtual status_t getVirtualizerStageOutput(const audio_config_base_t *config,
+                                                const audio_attributes_t *attr,
+                                                audio_io_handle_t *output);
+
+        virtual status_t releaseVirtualizerStageOutput(audio_io_handle_t output);
+
         bool isCallScreenModeSupported() override;
 
         void onNewAudioModulesAvailable() override;
@@ -797,6 +807,8 @@
         sp<SwAudioOutputDescriptor> mPrimaryOutput;     // primary output descriptor
         // list of descriptors for outputs currently opened
 
+        sp<SwAudioOutputDescriptor> mVirtualizerStageOutput;
+
         SwAudioOutputCollection mOutputs;
         // copy of mOutputs before setDeviceConnectionState() opens new outputs
         // reset to mOutputs when updateDevicesAndOutputs() is called.
@@ -933,7 +945,7 @@
         audio_io_handle_t getOutputForDevices(
                 const DeviceVector &devices,
                 audio_session_t session,
-                audio_stream_type_t stream,
+                const audio_attributes_t *attr,
                 const audio_config_t *config,
                 audio_output_flags_t *flags,
                 bool forceMutingHaptic = false);
@@ -948,6 +960,13 @@
                 audio_output_flags_t flags,
                 const DeviceVector &devices,
                 audio_io_handle_t *output);
+
+        sp<IOProfile> getVirtualizerStageOutputProfile(const audio_config_t *config,
+                                                       const AudioDeviceTypeAddrVector &devices,
+                                                       bool forOpening) const;
+
+        void checkVirtualizerClientRoutes();
+
         /**
          * @brief getInputForDevice selects an input handle for a given input device and
          * requester context
diff --git a/services/audiopolicy/service/AudioPolicyClientImpl.cpp b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
index cd53073..c8db45b 100644
--- a/services/audiopolicy/service/AudioPolicyClientImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyClientImpl.cpp
@@ -40,7 +40,8 @@
 
 status_t AudioPolicyService::AudioPolicyClient::openOutput(audio_module_handle_t module,
                                                            audio_io_handle_t *output,
-                                                           audio_config_t *config,
+                                                           audio_config_t *halConfig,
+                                                           audio_config_base_t *mixerConfig,
                                                            const sp<DeviceDescriptorBase>& device,
                                                            uint32_t *latencyMs,
                                                            audio_output_flags_t flags)
@@ -55,14 +56,18 @@
     media::OpenOutputResponse response;
 
     request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
-    request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+    request.halConfig = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_t_AudioConfig(*halConfig, false /*isInput*/));
+    request.mixerConfig = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_base_t_AudioConfigBase(*mixerConfig, false /*isInput*/));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_DeviceDescriptorBase(device));
     request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_output_flags_t_int32_t_mask(flags));
 
     status_t status = af->openOutput(request, &response);
     if (status == OK) {
         *output = VALUE_OR_RETURN_STATUS(aidl2legacy_int32_t_audio_io_handle_t(response.output));
-        *config = VALUE_OR_RETURN_STATUS(aidl2legacy_AudioConfig_audio_config_t(response.config));
+        *halConfig = VALUE_OR_RETURN_STATUS(
+                aidl2legacy_AudioConfig_audio_config_t(response.config, false /*isInput*/));
         *latencyMs = VALUE_OR_RETURN_STATUS(convertIntegral<uint32_t>(response.latencyMs));
     }
     return status;
@@ -131,7 +136,8 @@
     media::OpenInputRequest request;
     request.module = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_module_handle_t_int32_t(module));
     request.input = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_io_handle_t_int32_t(*input));
-    request.config = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_config_t_AudioConfig(*config));
+    request.config = VALUE_OR_RETURN_STATUS(
+            legacy2aidl_audio_config_t_AudioConfig(*config, true /*isInput*/));
     request.device = VALUE_OR_RETURN_STATUS(legacy2aidl_AudioDeviceTypeAddress(deviceTypeAddr));
     request.source = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_source_t_AudioSourceType(source));
     request.flags = VALUE_OR_RETURN_STATUS(legacy2aidl_audio_input_flags_t_int32_t_mask(flags));
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 9987252..4f920b1 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -99,13 +99,13 @@
         const media::AudioDevice& deviceAidl,
         media::AudioPolicyDeviceState stateAidl,
         const std::string& deviceNameAidl,
-        media::audio::common::AudioFormat encodedFormatAidl) {
+        const media::AudioFormatDescription& encodedFormatAidl) {
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl.type));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl.type));
     audio_policy_dev_state_t state = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioPolicyDeviceState_audio_policy_dev_state_t(stateAidl));
     audio_format_t encodedFormat = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioFormat_audio_format_t(encodedFormatAidl));
+            aidl2legacy_AudioFormatDescription_audio_format_t(encodedFormatAidl));
 
     if (mAudioPolicyManager == NULL) {
         return binderStatusFromStatusT(NO_INIT);
@@ -131,7 +131,7 @@
 Status AudioPolicyService::getDeviceConnectionState(const media::AudioDevice& deviceAidl,
                                                     media::AudioPolicyDeviceState* _aidl_return) {
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl.type));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl.type));
     if (mAudioPolicyManager == NULL) {
         *_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
                 legacy2aidl_audio_policy_dev_state_t_AudioPolicyDeviceState(
@@ -149,11 +149,11 @@
 Status AudioPolicyService::handleDeviceConfigChange(
         const media::AudioDevice& deviceAidl,
         const std::string& deviceNameAidl,
-        media::audio::common::AudioFormat encodedFormatAidl) {
+        const media::AudioFormatDescription& encodedFormatAidl) {
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl.type));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl.type));
     audio_format_t encodedFormat = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioFormat_audio_format_t(encodedFormatAidl));
+            aidl2legacy_AudioFormatDescription_audio_format_t(encodedFormatAidl));
 
     if (mAudioPolicyManager == NULL) {
         return binderStatusFromStatusT(NO_INIT);
@@ -292,7 +292,7 @@
             aidl2legacy_int32_t_audio_session_t(sessionAidl));
     audio_stream_type_t stream = AUDIO_STREAM_DEFAULT;
     audio_config_t config = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioConfig_audio_config_t(configAidl));
+            aidl2legacy_AudioConfig_audio_config_t(configAidl, false /*isInput*/));
     audio_output_flags_t flags = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_int32_t_audio_output_flags_t_mask(flagsAidl));
     audio_port_handle_t selectedDeviceId = VALUE_OR_RETURN_BINDER_STATUS(
@@ -523,7 +523,7 @@
     audio_session_t session = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_int32_t_audio_session_t(sessionAidl));
     audio_config_base_t config = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioConfigBase_audio_config_base_t(configAidl));
+            aidl2legacy_AudioConfigBase_audio_config_base_t(configAidl, true /*isInput*/));
     audio_input_flags_t flags = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_int32_t_audio_input_flags_t_mask(flagsAidl));
     audio_port_handle_t selectedDeviceId = VALUE_OR_RETURN_BINDER_STATUS(
@@ -925,12 +925,13 @@
 }
 
 Status AudioPolicyService::setStreamVolumeIndex(media::AudioStreamType streamAidl,
-                                                int32_t deviceAidl, int32_t indexAidl) {
+                                                const media::AudioDeviceDescription& deviceAidl,
+                                                int32_t indexAidl) {
     audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
     int index = VALUE_OR_RETURN_BINDER_STATUS(convertIntegral<int>(indexAidl));
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl));
 
     if (mAudioPolicyManager == NULL) {
         return binderStatusFromStatusT(NO_INIT);
@@ -949,11 +950,12 @@
 }
 
 Status AudioPolicyService::getStreamVolumeIndex(media::AudioStreamType streamAidl,
-                                                int32_t deviceAidl, int32_t* _aidl_return) {
+                                                const media::AudioDeviceDescription& deviceAidl,
+                                                int32_t* _aidl_return) {
     audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl));
     int index;
 
     if (mAudioPolicyManager == NULL) {
@@ -971,12 +973,13 @@
 }
 
 Status AudioPolicyService::setVolumeIndexForAttributes(
-        const media::AudioAttributesInternal& attrAidl, int32_t deviceAidl, int32_t indexAidl) {
+        const media::AudioAttributesInternal& attrAidl,
+        const media::AudioDeviceDescription& deviceAidl, int32_t indexAidl) {
     audio_attributes_t attributes = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioAttributesInternal_audio_attributes_t(attrAidl));
     int index = VALUE_OR_RETURN_BINDER_STATUS(convertIntegral<int>(indexAidl));
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl));
     RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
             AudioValidator::validateAudioAttributes(attributes, "169572641")));
 
@@ -993,11 +996,12 @@
 }
 
 Status AudioPolicyService::getVolumeIndexForAttributes(
-        const media::AudioAttributesInternal& attrAidl, int32_t deviceAidl, int32_t* _aidl_return) {
+        const media::AudioAttributesInternal& attrAidl,
+        const media::AudioDeviceDescription& deviceAidl, int32_t* _aidl_return) {
     audio_attributes_t attributes = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioAttributesInternal_audio_attributes_t(attrAidl));
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl));
     int index;
     RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
             AudioValidator::validateAudioAttributes(attributes, "169572641")));
@@ -1075,14 +1079,14 @@
 
 //audio policy: use audio_device_t appropriately
 
-Status AudioPolicyService::getDevicesForStream(media::AudioStreamType streamAidl,
-                                               int32_t* _aidl_return) {
+Status AudioPolicyService::getDevicesForStream(
+        media::AudioStreamType streamAidl,
+        std::vector<media::AudioDeviceDescription>* _aidl_return) {
     audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
 
     if (uint32_t(stream) >= AUDIO_STREAM_PUBLIC_CNT) {
-        *_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
-                legacy2aidl_audio_devices_t_int32_t(AUDIO_DEVICE_NONE));
+        *_aidl_return = std::vector<media::AudioDeviceDescription>{};
         return Status::ok();
     }
     if (mAudioPolicyManager == NULL) {
@@ -1091,7 +1095,9 @@
     Mutex::Autolock _l(mLock);
     AutoCallerClear acc;
     *_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
-            legacy2aidl_audio_devices_t_int32_t(mAudioPolicyManager->getDevicesForStream(stream)));
+            convertContainer<std::vector<media::AudioDeviceDescription>>(
+                    mAudioPolicyManager->getDevicesForStream(stream),
+                    legacy2aidl_audio_devices_t_AudioDeviceDescription));
     return Status::ok();
 }
 
@@ -1427,7 +1433,7 @@
         const media::AudioAttributesInternal& attributesAidl,
         bool* _aidl_return) {
     audio_config_base_t config = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioConfigBase_audio_config_base_t(configAidl));
+            aidl2legacy_AudioConfigBase_audio_config_base_t(configAidl, false /*isInput*/));
     audio_attributes_t attributes = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioAttributesInternal_audio_attributes_t(attributesAidl));
     RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
@@ -1602,7 +1608,7 @@
     _aidl_return->ioHandle = VALUE_OR_RETURN_BINDER_STATUS(
             legacy2aidl_audio_io_handle_t_int32_t(ioHandle));
     _aidl_return->device = VALUE_OR_RETURN_BINDER_STATUS(
-            legacy2aidl_audio_devices_t_int32_t(device));
+            legacy2aidl_audio_devices_t_AudioDeviceDescription(device));
     return Status::ok();
 }
 
@@ -1802,13 +1808,14 @@
 }
 
 
-Status AudioPolicyService::getStreamVolumeDB(media::AudioStreamType streamAidl, int32_t indexAidl,
-                                             int32_t deviceAidl, float* _aidl_return) {
+Status AudioPolicyService::getStreamVolumeDB(
+        media::AudioStreamType streamAidl, int32_t indexAidl,
+        const media::AudioDeviceDescription& deviceAidl, float* _aidl_return) {
     audio_stream_type_t stream = VALUE_OR_RETURN_BINDER_STATUS(
             aidl2legacy_AudioStreamType_audio_stream_type_t(streamAidl));
     int index = VALUE_OR_RETURN_BINDER_STATUS(convertIntegral<int>(indexAidl));
     audio_devices_t device = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_int32_t_audio_devices_t(deviceAidl));
+            aidl2legacy_AudioDeviceDescription_audio_devices_t(deviceAidl));
 
     if (mAudioPolicyManager == NULL) {
         return binderStatusFromStatusT(NO_INIT);
@@ -1820,7 +1827,7 @@
 }
 
 Status AudioPolicyService::getSurroundFormats(media::Int* count,
-        std::vector<media::audio::common::AudioFormat>* formats,
+        std::vector<media::AudioFormatDescription>* formats,
         std::vector<bool>* formatsEnabled) {
     unsigned int numSurroundFormats = VALUE_OR_RETURN_BINDER_STATUS(
             convertIntegral<unsigned int>(count->value));
@@ -1842,7 +1849,8 @@
     numSurroundFormatsReq = std::min(numSurroundFormats, numSurroundFormatsReq);
     RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
             convertRange(surroundFormats.get(), surroundFormats.get() + numSurroundFormatsReq,
-                         std::back_inserter(*formats), legacy2aidl_audio_format_t_AudioFormat)));
+                         std::back_inserter(*formats),
+                         legacy2aidl_audio_format_t_AudioFormatDescription)));
     formatsEnabled->insert(
             formatsEnabled->begin(),
             surroundFormatsEnabled.get(),
@@ -1852,7 +1860,7 @@
 }
 
 Status AudioPolicyService::getReportedSurroundFormats(
-        media::Int* count, std::vector<media::audio::common::AudioFormat>* formats) {
+        media::Int* count, std::vector<media::AudioFormatDescription>* formats) {
     unsigned int numSurroundFormats = VALUE_OR_RETURN_BINDER_STATUS(
             convertIntegral<unsigned int>(count->value));
     if (numSurroundFormats > MAX_ITEMS_PER_LIST) {
@@ -1872,13 +1880,14 @@
     numSurroundFormatsReq = std::min(numSurroundFormats, numSurroundFormatsReq);
     RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
             convertRange(surroundFormats.get(), surroundFormats.get() + numSurroundFormatsReq,
-                         std::back_inserter(*formats), legacy2aidl_audio_format_t_AudioFormat)));
+                         std::back_inserter(*formats),
+                         legacy2aidl_audio_format_t_AudioFormatDescription)));
     count->value = VALUE_OR_RETURN_BINDER_STATUS(convertIntegral<uint32_t>(numSurroundFormats));
     return Status::ok();
 }
 
 Status AudioPolicyService::getHwOffloadEncodingFormatsSupportedForA2DP(
-        std::vector<media::audio::common::AudioFormat>* _aidl_return) {
+        std::vector<media::AudioFormatDescription>* _aidl_return) {
     std::vector<audio_format_t> formats;
 
     if (mAudioPolicyManager == NULL) {
@@ -1889,16 +1898,16 @@
     RETURN_IF_BINDER_ERROR(binderStatusFromStatusT(
             mAudioPolicyManager->getHwOffloadEncodingFormatsSupportedForA2DP(&formats)));
     *_aidl_return = VALUE_OR_RETURN_BINDER_STATUS(
-            convertContainer<std::vector<media::audio::common::AudioFormat>>(
+            convertContainer<std::vector<media::AudioFormatDescription>>(
                     formats,
-                    legacy2aidl_audio_format_t_AudioFormat));
+                    legacy2aidl_audio_format_t_AudioFormatDescription));
     return Status::ok();
 }
 
 Status AudioPolicyService::setSurroundFormatEnabled(
-        media::audio::common::AudioFormat audioFormatAidl, bool enabled) {
+        const media::AudioFormatDescription& audioFormatAidl, bool enabled) {
     audio_format_t audioFormat = VALUE_OR_RETURN_BINDER_STATUS(
-            aidl2legacy_AudioFormat_audio_format_t(audioFormatAidl));
+            aidl2legacy_AudioFormatDescription_audio_format_t(audioFormatAidl));
     if (mAudioPolicyManager == NULL) {
         return binderStatusFromStatusT(NO_INIT);
     }
diff --git a/services/audiopolicy/service/AudioPolicyService.cpp b/services/audiopolicy/service/AudioPolicyService.cpp
index 4d0e1f1..7313893 100644
--- a/services/audiopolicy/service/AudioPolicyService.cpp
+++ b/services/audiopolicy/service/AudioPolicyService.cpp
@@ -127,6 +127,7 @@
         loadAudioPolicyManager();
         mAudioPolicyManager = mCreateAudioPolicyManager(mAudioPolicyClient);
     }
+
     // load audio processing modules
     sp<AudioPolicyEffects> audioPolicyEffects = new AudioPolicyEffects();
     sp<UidPolicy> uidPolicy = new UidPolicy(this);
@@ -139,6 +140,8 @@
     }
     uidPolicy->registerSelf();
     sensorPrivacyPolicy->registerSelf();
+
+    AudioSystem::audioPolicyReady();
 }
 
 void AudioPolicyService::unloadAudioPolicyManager()
@@ -443,13 +446,15 @@
             media::RecordClientInfo clientInfoAidl = VALUE_OR_RETURN_STATUS(
                     legacy2aidl_record_client_info_t_RecordClientInfo(*clientInfo));
             media::AudioConfigBase clientConfigAidl = VALUE_OR_RETURN_STATUS(
-                    legacy2aidl_audio_config_base_t_AudioConfigBase(*clientConfig));
+                    legacy2aidl_audio_config_base_t_AudioConfigBase(
+                            *clientConfig, true /*isInput*/));
             std::vector<media::EffectDescriptor> clientEffectsAidl = VALUE_OR_RETURN_STATUS(
                     convertContainer<std::vector<media::EffectDescriptor>>(
                             clientEffects,
                             legacy2aidl_effect_descriptor_t_EffectDescriptor));
             media::AudioConfigBase deviceConfigAidl = VALUE_OR_RETURN_STATUS(
-                    legacy2aidl_audio_config_base_t_AudioConfigBase(*deviceConfig));
+                    legacy2aidl_audio_config_base_t_AudioConfigBase(
+                            *deviceConfig, true /*isInput*/));
             std::vector<media::EffectDescriptor> effectsAidl = VALUE_OR_RETURN_STATUS(
                     convertContainer<std::vector<media::EffectDescriptor>>(
                             effects,
diff --git a/services/audiopolicy/service/AudioPolicyService.h b/services/audiopolicy/service/AudioPolicyService.h
index 3b77ed8..0b76936 100644
--- a/services/audiopolicy/service/AudioPolicyService.h
+++ b/services/audiopolicy/service/AudioPolicyService.h
@@ -71,13 +71,13 @@
             const media::AudioDevice& device,
             media::AudioPolicyDeviceState state,
             const std::string& deviceName,
-            media::audio::common::AudioFormat encodedFormat) override;
+            const media::AudioFormatDescription& encodedFormat) override;
     binder::Status getDeviceConnectionState(const media::AudioDevice& device,
                                             media::AudioPolicyDeviceState* _aidl_return) override;
     binder::Status handleDeviceConfigChange(
             const media::AudioDevice& device,
             const std::string& deviceName,
-            media::audio::common::AudioFormat encodedFormat) override;
+            const media::AudioFormatDescription& encodedFormat) override;
     binder::Status setPhoneState(media::AudioMode state, int32_t uid) override;
     binder::Status setForceUse(media::AudioPolicyForceUse usage,
                                media::AudioPolicyForcedConfig config) override;
@@ -103,22 +103,27 @@
     binder::Status releaseInput(int32_t portId) override;
     binder::Status initStreamVolume(media::AudioStreamType stream, int32_t indexMin,
                                     int32_t indexMax) override;
-    binder::Status setStreamVolumeIndex(media::AudioStreamType stream, int32_t device,
+    binder::Status setStreamVolumeIndex(media::AudioStreamType stream,
+                                        const media::AudioDeviceDescription& device,
                                         int32_t index) override;
-    binder::Status getStreamVolumeIndex(media::AudioStreamType stream, int32_t device,
+    binder::Status getStreamVolumeIndex(media::AudioStreamType stream,
+                                        const media::AudioDeviceDescription& device,
                                         int32_t* _aidl_return) override;
     binder::Status setVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
-                                               int32_t device, int32_t index) override;
+                                               const media::AudioDeviceDescription& device,
+                                               int32_t index) override;
     binder::Status getVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
-                                               int32_t device, int32_t* _aidl_return) override;
+                                               const media::AudioDeviceDescription& device,
+                                               int32_t* _aidl_return) override;
     binder::Status getMaxVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
                                                   int32_t* _aidl_return) override;
     binder::Status getMinVolumeIndexForAttributes(const media::AudioAttributesInternal& attr,
                                                   int32_t* _aidl_return) override;
     binder::Status getStrategyForStream(media::AudioStreamType stream,
                                         int32_t* _aidl_return) override;
-    binder::Status getDevicesForStream(media::AudioStreamType stream,
-                                       int32_t* _aidl_return) override;
+    binder::Status getDevicesForStream(
+            media::AudioStreamType stream,
+            std::vector<media::AudioDeviceDescription>* _aidl_return) override;
     binder::Status getDevicesForAttributes(const media::AudioAttributesEx& attr,
                                            std::vector<media::AudioDevice>* _aidl_return) override;
     binder::Status getOutputForEffect(const media::EffectDescriptor& desc,
@@ -187,16 +192,17 @@
     binder::Status stopAudioSource(int32_t portId) override;
     binder::Status setMasterMono(bool mono) override;
     binder::Status getMasterMono(bool* _aidl_return) override;
-    binder::Status getStreamVolumeDB(media::AudioStreamType stream, int32_t index, int32_t device,
+    binder::Status getStreamVolumeDB(media::AudioStreamType stream, int32_t index,
+                                     const media::AudioDeviceDescription& device,
                                      float* _aidl_return) override;
     binder::Status getSurroundFormats(media::Int* count,
-                                      std::vector<media::audio::common::AudioFormat>* formats,
+                                      std::vector<media::AudioFormatDescription>* formats,
                                       std::vector<bool>* formatsEnabled) override;
     binder::Status getReportedSurroundFormats(
-            media::Int* count, std::vector<media::audio::common::AudioFormat>* formats) override;
+            media::Int* count, std::vector<media::AudioFormatDescription>* formats) override;
     binder::Status getHwOffloadEncodingFormatsSupportedForA2DP(
-            std::vector<media::audio::common::AudioFormat>* _aidl_return) override;
-    binder::Status setSurroundFormatEnabled(media::audio::common::AudioFormat audioFormat,
+            std::vector<media::AudioFormatDescription>* _aidl_return) override;
+    binder::Status setSurroundFormatEnabled(const media::AudioFormatDescription& audioFormat,
                                             bool enabled) override;
     binder::Status setAssistantUid(int32_t uid) override;
     binder::Status setA11yServicesUids(const std::vector<int32_t>& uids) override;
@@ -659,7 +665,8 @@
         // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
         virtual status_t openOutput(audio_module_handle_t module,
                                     audio_io_handle_t *output,
-                                    audio_config_t *config,
+                                    audio_config_t *halConfig,
+                                    audio_config_base_t *mixerConfig,
                                     const sp<DeviceDescriptorBase>& device,
                                     uint32_t *latencyMs,
                                     audio_output_flags_t flags);
diff --git a/services/audiopolicy/tests/Android.bp b/services/audiopolicy/tests/Android.bp
index b296fb0..8fbe8b2 100644
--- a/services/audiopolicy/tests/Android.bp
+++ b/services/audiopolicy/tests/Android.bp
@@ -25,7 +25,7 @@
         "libmedia_helper",
         "libutils",
         "libxml2",
-        "libpermission",
+        "framework-permission-aidl-cpp",
         "libbinder",
     ],
 
diff --git a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
index f7b0565..84b40d2 100644
--- a/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyManagerTestClient.h
@@ -37,7 +37,8 @@
 
     status_t openOutput(audio_module_handle_t module,
                         audio_io_handle_t *output,
-                        audio_config_t * /*config*/,
+                        audio_config_t * /*halConfig*/,
+                        audio_config_base_t * /*mixerConfig*/,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t * /*latencyMs*/,
                         audio_output_flags_t /*flags*/) override {
diff --git a/services/audiopolicy/tests/AudioPolicyTestClient.h b/services/audiopolicy/tests/AudioPolicyTestClient.h
index 1384864..4e0735b 100644
--- a/services/audiopolicy/tests/AudioPolicyTestClient.h
+++ b/services/audiopolicy/tests/AudioPolicyTestClient.h
@@ -30,7 +30,8 @@
     }
     status_t openOutput(audio_module_handle_t /*module*/,
                         audio_io_handle_t* /*output*/,
-                        audio_config_t* /*config*/,
+                        audio_config_t* /*halConfig*/,
+                        audio_config_base_t* /*mixerConfig*/,
                         const sp<DeviceDescriptorBase>& /*device*/,
                         uint32_t* /*latencyMs*/,
                         audio_output_flags_t /*flags*/) override { return NO_INIT; }
diff --git a/services/camera/libcameraservice/CameraService.cpp b/services/camera/libcameraservice/CameraService.cpp
index dc101ff..7b138a6 100644
--- a/services/camera/libcameraservice/CameraService.cpp
+++ b/services/camera/libcameraservice/CameraService.cpp
@@ -1867,6 +1867,27 @@
     CameraServiceProxyWrapper::logOpen(cameraId, facing, clientPackageName,
             effectiveApiLevel, isNdk, openLatencyMs);
 
+    {
+        Mutex::Autolock lock(mInjectionParametersLock);
+        if (cameraId == mInjectionInternalCamId && mInjectionInitPending) {
+            mInjectionInitPending = false;
+            status_t res = NO_ERROR;
+            auto clientDescriptor = mActiveClientManager.get(mInjectionInternalCamId);
+            if (clientDescriptor != nullptr) {
+                BasicClient* baseClientPtr = clientDescriptor->getValue().get();
+                res = baseClientPtr->injectCamera(mInjectionExternalCamId, mCameraProviderManager);
+                if (res != OK) {
+                    mInjectionStatusListener->notifyInjectionError(mInjectionExternalCamId, res);
+                }
+            } else {
+                ALOGE("%s: Internal camera ID = %s 's client does not exist!",
+                        __FUNCTION__, mInjectionInternalCamId.string());
+                res = NO_INIT;
+                mInjectionStatusListener->notifyInjectionError(mInjectionExternalCamId, res);
+            }
+        }
+    }
+
     return ret;
 }
 
@@ -2527,7 +2548,7 @@
         const String16& externalCamId,
         const sp<ICameraInjectionCallback>& callback,
         /*out*/
-        sp<hardware::camera2::ICameraInjectionSession>* cameraInjectionSession) {
+        sp<ICameraInjectionSession>* cameraInjectionSession) {
     ATRACE_CALL();
 
     if (!checkCallingPermission(sCameraInjectExternalCameraPermission)) {
@@ -2544,18 +2565,30 @@
         __FUNCTION__, String8(packageName).string(),
         String8(internalCamId).string(), String8(externalCamId).string());
 
-    binder::Status ret = binder::Status::ok();
-    // TODO: Implement the injection camera function.
-    // ret = internalInjectCamera(...);
-    // if(!ret.isOk()) {
-    //     mInjectionStatusListener->notifyInjectionError(...);
-    //     return ret;
-    // }
-
+    {
+        Mutex::Autolock lock(mInjectionParametersLock);
+        mInjectionInternalCamId = String8(internalCamId);
+        mInjectionExternalCamId = String8(externalCamId);
+        status_t res = NO_ERROR;
+        auto clientDescriptor = mActiveClientManager.get(mInjectionInternalCamId);
+        // If the client already exists, we can directly connect to the camera device through the
+        // client's injectCamera(), otherwise we need to wait until the client is established
+        // (execute connectHelper()) before injecting the camera to the camera device.
+        if (clientDescriptor != nullptr) {
+            mInjectionInitPending = false;
+            BasicClient* baseClientPtr = clientDescriptor->getValue().get();
+            res = baseClientPtr->injectCamera(mInjectionExternalCamId, mCameraProviderManager);
+            if(res != OK) {
+                mInjectionStatusListener->notifyInjectionError(mInjectionExternalCamId, res);
+            }
+        } else {
+            mInjectionInitPending = true;
+        }
+    }
     mInjectionStatusListener->addListener(callback);
     *cameraInjectionSession = new CameraInjectionSession(this);
 
-    return ret;
+    return binder::Status::ok();
 }
 
 void CameraService::removeByClient(const BasicClient* client) {
@@ -3820,22 +3853,62 @@
 }
 
 void CameraService::InjectionStatusListener::notifyInjectionError(
-        int errorCode) {
-    Mutex::Autolock lock(mListenerLock);
+        String8 injectedCamId, status_t err) {
     if (mCameraInjectionCallback == nullptr) {
         ALOGW("InjectionStatusListener: mCameraInjectionCallback == nullptr");
         return;
     }
-    mCameraInjectionCallback->onInjectionError(errorCode);
+
+    switch (err) {
+        case -ENODEV:
+            mCameraInjectionCallback->onInjectionError(
+                    ICameraInjectionCallback::ERROR_INJECTION_SESSION);
+            ALOGE("No camera device with ID \"%s\" currently available!",
+                    injectedCamId.string());
+            break;
+        case -EBUSY:
+            mCameraInjectionCallback->onInjectionError(
+                    ICameraInjectionCallback::ERROR_INJECTION_SESSION);
+            ALOGE("Higher-priority client using camera, ID \"%s\" currently unavailable!",
+                    injectedCamId.string());
+            break;
+        case DEAD_OBJECT:
+            mCameraInjectionCallback->onInjectionError(
+                    ICameraInjectionCallback::ERROR_INJECTION_SESSION);
+            ALOGE("Camera ID \"%s\" object is dead!",
+                    injectedCamId.string());
+            break;
+        case INVALID_OPERATION:
+            mCameraInjectionCallback->onInjectionError(
+                    ICameraInjectionCallback::ERROR_INJECTION_SESSION);
+            ALOGE("Camera ID \"%s\" encountered an operating or internal error!",
+                    injectedCamId.string());
+            break;
+        case UNKNOWN_TRANSACTION:
+            mCameraInjectionCallback->onInjectionError(
+                    ICameraInjectionCallback::ERROR_INJECTION_UNSUPPORTED);
+            ALOGE("Camera ID \"%s\" method doesn't support!",
+                    injectedCamId.string());
+            break;
+        default:
+            mCameraInjectionCallback->onInjectionError(
+                    ICameraInjectionCallback::ERROR_INJECTION_INVALID_ERROR);
+            ALOGE("Unexpected error %s (%d) opening camera \"%s\"!",
+                    strerror(-err), err, injectedCamId.string());
+    }
 }
 
 void CameraService::InjectionStatusListener::binderDied(
         const wp<IBinder>& /*who*/) {
-    Mutex::Autolock lock(mListenerLock);
     ALOGV("InjectionStatusListener: ICameraInjectionCallback has died");
     auto parent = mParent.promote();
     if (parent != nullptr) {
-        parent->stopInjectionImpl();
+        auto clientDescriptor = parent->mActiveClientManager.get(parent->mInjectionInternalCamId);
+        if (clientDescriptor != nullptr) {
+            BasicClient* baseClientPtr = clientDescriptor->getValue().get();
+            baseClientPtr->stopInjection();
+        }
+        parent->clearInjectionParameters();
     }
 }
 
@@ -3851,7 +3924,20 @@
         return STATUS_ERROR(ICameraInjectionCallback::ERROR_INJECTION_SERVICE,
                 "Camera service encountered error");
     }
-    parent->stopInjectionImpl();
+
+    status_t res = NO_ERROR;
+    auto clientDescriptor = parent->mActiveClientManager.get(parent->mInjectionInternalCamId);
+    if (clientDescriptor != nullptr) {
+        BasicClient* baseClientPtr = clientDescriptor->getValue().get();
+        res = baseClientPtr->stopInjection();
+        if (res != OK) {
+            ALOGE("CameraInjectionSession: Failed to stop the injection camera!"
+                " ret != NO_ERROR: %d", res);
+            return STATUS_ERROR(ICameraInjectionCallback::ERROR_INJECTION_SESSION,
+                "Camera session encountered error");
+        }
+    }
+    parent->clearInjectionParameters();
     return binder::Status::ok();
 }
 
@@ -4560,10 +4646,14 @@
     return mode;
 }
 
-void CameraService::stopInjectionImpl() {
+void CameraService::clearInjectionParameters() {
+    {
+        Mutex::Autolock lock(mInjectionParametersLock);
+        mInjectionInitPending = true;
+        mInjectionInternalCamId = "";
+    }
+    mInjectionExternalCamId = "";
     mInjectionStatusListener->removeListener();
-
-    // TODO: Implement the stop injection function.
 }
 
 }; // namespace android
diff --git a/services/camera/libcameraservice/CameraService.h b/services/camera/libcameraservice/CameraService.h
index 9021170..ca184a8 100644
--- a/services/camera/libcameraservice/CameraService.h
+++ b/services/camera/libcameraservice/CameraService.h
@@ -320,6 +320,14 @@
         // Set/reset camera mute
         virtual status_t setCameraMute(bool enabled) = 0;
 
+        // The injection camera session to replace the internal camera
+        // session.
+        virtual status_t injectCamera(const String8& injectedCamId,
+                sp<CameraProviderManager> manager) = 0;
+
+        // Stop the injection camera and restore to internal camera session.
+        virtual status_t stopInjection() = 0;
+
     protected:
         BasicClient(const sp<CameraService>& cameraService,
                 const sp<IBinder>& remoteCallback,
@@ -1191,7 +1199,7 @@
 
             void addListener(const sp<hardware::camera2::ICameraInjectionCallback>& callback);
             void removeListener();
-            void notifyInjectionError(int errorCode);
+            void notifyInjectionError(String8 injectedCamId, status_t err);
 
             // IBinder::DeathRecipient implementation
             virtual void binderDied(const wp<IBinder>& who);
@@ -1218,7 +1226,15 @@
             wp<CameraService> mParent;
     };
 
-    void stopInjectionImpl();
+    void clearInjectionParameters();
+
+    // This is the existing camera id being replaced.
+    String8 mInjectionInternalCamId;
+    // This is the external camera Id replacing the internalId.
+    String8 mInjectionExternalCamId;
+    bool mInjectionInitPending = true;
+    // Guard mInjectionInternalCamId and mInjectionInitPending.
+    Mutex mInjectionParametersLock;
 };
 
 } // namespace android
diff --git a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp
index ef15f2d..652842b 100644
--- a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp
+++ b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.cpp
@@ -330,5 +330,19 @@
                 CaptureResultExtras());
 }
 
+status_t CameraOfflineSessionClient::injectCamera(const String8& injectedCamId,
+            sp<CameraProviderManager> manager) {
+    ALOGV("%s: This client doesn't support the injection camera. injectedCamId: %s providerPtr: %p",
+            __FUNCTION__, injectedCamId.string(), manager.get());
+
+    return OK;
+}
+
+status_t CameraOfflineSessionClient::stopInjection() {
+    ALOGV("%s: This client doesn't support the injection camera.", __FUNCTION__);
+
+    return OK;
+}
+
 // ----------------------------------------------------------------------------
 }; // namespace android
diff --git a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h
index b219a4c..b5238b8 100644
--- a/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h
+++ b/services/camera/libcameraservice/api2/CameraOfflineSessionClient.h
@@ -98,6 +98,9 @@
     void notifyPrepared(int streamId) override;
     void notifyRequestQueueEmpty() override;
     void notifyRepeatingRequestError(long lastFrameNumber) override;
+    status_t injectCamera(const String8& injectedCamId,
+            sp<CameraProviderManager> manager) override;
+    status_t stopInjection() override;
 
 private:
     mutable Mutex mBinderSerializationLock;
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.cpp b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
index 13d044a..1147e23 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.cpp
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.cpp
@@ -413,6 +413,17 @@
     mRemoteCallback.clear();
 }
 
+template <typename TClientBase>
+status_t Camera2ClientBase<TClientBase>::injectCamera(const String8& injectedCamId,
+        sp<CameraProviderManager> manager) {
+    return mDevice->injectCamera(injectedCamId, manager);
+}
+
+template <typename TClientBase>
+status_t Camera2ClientBase<TClientBase>::stopInjection() {
+    return mDevice->stopInjection();
+}
+
 template class Camera2ClientBase<CameraService::Client>;
 template class Camera2ClientBase<CameraDeviceClientBase>;
 
diff --git a/services/camera/libcameraservice/common/Camera2ClientBase.h b/services/camera/libcameraservice/common/Camera2ClientBase.h
index 6246f7b..b593bfa 100644
--- a/services/camera/libcameraservice/common/Camera2ClientBase.h
+++ b/services/camera/libcameraservice/common/Camera2ClientBase.h
@@ -114,6 +114,10 @@
         mutable Mutex mRemoteCallbackLock;
     } mSharedCameraCallbacks;
 
+    status_t      injectCamera(const String8& injectedCamId,
+                               sp<CameraProviderManager> manager) override;
+    status_t      stopInjection() override;
+
 protected:
 
     // The PID provided in the constructor call
diff --git a/services/camera/libcameraservice/common/CameraDeviceBase.h b/services/camera/libcameraservice/common/CameraDeviceBase.h
index 85b0cc2..3c95ed3 100644
--- a/services/camera/libcameraservice/common/CameraDeviceBase.h
+++ b/services/camera/libcameraservice/common/CameraDeviceBase.h
@@ -427,6 +427,18 @@
      */
     void setImageDumpMask(int mask) { mImageDumpMask = mask; }
 
+    /**
+     * The injection camera session to replace the internal camera
+     * session.
+     */
+    virtual status_t injectCamera(const String8& injectedCamId,
+            sp<CameraProviderManager> manager) = 0;
+
+    /**
+     * Stop the injection camera and restore to internal camera session.
+     */
+    virtual status_t stopInjection() = 0;
+
 protected:
     bool mImageDumpMask = 0;
 };
diff --git a/services/camera/libcameraservice/device3/Camera3Device.cpp b/services/camera/libcameraservice/device3/Camera3Device.cpp
index d572d57..877b683 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.cpp
+++ b/services/camera/libcameraservice/device3/Camera3Device.cpp
@@ -2835,17 +2835,28 @@
         mRequestBufferSM.onStreamsConfigured();
     }
 
+    // First call injectCamera() and then run configureStreamsLocked() case:
     // Since the streams configuration of the injection camera is based on the internal camera, we
-    // must wait until the internal camera configure streams before calling injectCamera() to
+    // must wait until the internal camera configure streams before running the injection job to
     // configure the injection streams.
     if (mInjectionMethods->isInjecting()) {
-        ALOGV("%s: Injection camera %s: Start to configure streams.",
+        ALOGD("%s: Injection camera %s: Start to configure streams.",
               __FUNCTION__, mInjectionMethods->getInjectedCamId().string());
         res = mInjectionMethods->injectCamera(config, bufferSizes);
         if (res != OK) {
             ALOGE("Can't finish inject camera process!");
             return res;
         }
+    } else {
+        // First run configureStreamsLocked() and then call injectCamera() case:
+        // If the stream configuration has been completed and camera deive is active, but the
+        // injection camera has not been injected yet, we need to store the stream configuration of
+        // the internal camera (because the stream configuration of the injection camera is based
+        // on the internal camera). When injecting occurs later, this configuration can be used by
+        // the injection camera.
+        ALOGV("%s: The stream configuration is complete and the camera device is active, but the"
+              " injection camera has not been injected yet.", __FUNCTION__);
+        mInjectionMethods->storeInjectionConfig(config, bufferSizes);
     }
 
     return OK;
@@ -6584,6 +6595,13 @@
     ALOGI("%s Injection camera: injectedCamId = %s", __FUNCTION__, injectedCamId.string());
     ATRACE_CALL();
     Mutex::Autolock il(mInterfaceLock);
+    // When the camera device is active, injectCamera() and stopInjection() will call
+    // internalPauseAndWaitLocked() and internalResumeLocked(), and then they will call
+    // mStatusChanged.waitRelative(mLock, timeout) of waitUntilStateThenRelock(). But
+    // mStatusChanged.waitRelative(mLock, timeout)'s parameter: mutex "mLock" must be in the locked
+    // state, so we need to add "Mutex::Autolock l(mLock)" to lock the "mLock" before calling
+    // waitUntilStateThenRelock().
+    Mutex::Autolock l(mLock);
 
     status_t res = NO_ERROR;
     if (mInjectionMethods->isInjecting()) {
@@ -6606,16 +6624,25 @@
         return res;
     }
 
-    camera3::camera_stream_configuration injectionConfig;
-    std::vector<uint32_t> injectionBufferSizes;
-    mInjectionMethods->getInjectionConfig(&injectionConfig, &injectionBufferSizes);
     // When the second display of android is cast to the remote device, and the opened camera is
     // also cast to the second display, in this case, because the camera has configured the streams
     // at this time, we can directly call injectCamera() to replace the internal camera with
     // injection camera.
-    if (mOperatingMode >= 0 && injectionConfig.num_streams > 0
-                && injectionBufferSizes.size() > 0) {
-        ALOGV("%s: The opened camera is directly cast to the remote device.", __FUNCTION__);
+    if (mInjectionMethods->isStreamConfigCompleteButNotInjected()) {
+        ALOGD("%s: The opened camera is directly cast to the remote device.", __FUNCTION__);
+
+        camera3::camera_stream_configuration injectionConfig;
+        std::vector<uint32_t> injectionBufferSizes;
+        mInjectionMethods->getInjectionConfig(&injectionConfig, &injectionBufferSizes);
+        if (mOperatingMode < 0 || injectionConfig.num_streams <= 0
+                    || injectionBufferSizes.size() <= 0) {
+            ALOGE("Failed to inject camera due to abandoned configuration! "
+                    "mOperatingMode: %d injectionConfig.num_streams: %d "
+                    "injectionBufferSizes.size(): %zu", mOperatingMode,
+                    injectionConfig.num_streams, injectionBufferSizes.size());
+            return DEAD_OBJECT;
+        }
+
         res = mInjectionMethods->injectCamera(
                 injectionConfig, injectionBufferSizes);
         if (res != OK) {
@@ -6630,6 +6657,7 @@
 status_t Camera3Device::stopInjection() {
     ALOGI("%s: Injection camera: stopInjection", __FUNCTION__);
     Mutex::Autolock il(mInterfaceLock);
+    Mutex::Autolock l(mLock);
     return mInjectionMethods->stopInjection();
 }
 
diff --git a/services/camera/libcameraservice/device3/Camera3Device.h b/services/camera/libcameraservice/device3/Camera3Device.h
index 733ed15..310ee27 100644
--- a/services/camera/libcameraservice/device3/Camera3Device.h
+++ b/services/camera/libcameraservice/device3/Camera3Device.h
@@ -1371,25 +1371,34 @@
         // when device is IDLE and request thread is paused.
         status_t injectCamera(
                 camera3::camera_stream_configuration& injectionConfig,
-                std::vector<uint32_t>& injectionBufferSizes);
+                const std::vector<uint32_t>& injectionBufferSizes);
 
         // Stop the injection camera and switch back to backup hal interface.
         status_t stopInjection();
 
         bool isInjecting();
 
+        bool isStreamConfigCompleteButNotInjected();
+
         const String8& getInjectedCamId() const;
 
         void getInjectionConfig(/*out*/ camera3::camera_stream_configuration* injectionConfig,
                 /*out*/ std::vector<uint32_t>* injectionBufferSizes);
 
+        // When the streaming configuration is completed and the camera device is active, but the
+        // injection camera has not yet been injected, the streaming configuration of the internal
+        // camera will be stored first.
+        void storeInjectionConfig(
+                const camera3::camera_stream_configuration& injectionConfig,
+                const std::vector<uint32_t>& injectionBufferSizes);
+
       private:
         // Configure the streams of injection camera, it need wait until the
         // output streams are created and configured to the original camera before
         // proceeding.
         status_t injectionConfigureStreams(
                 camera3::camera_stream_configuration& injectionConfig,
-                std::vector<uint32_t>& injectionBufferSizes);
+                const std::vector<uint32_t>& injectionBufferSizes);
 
         // Disconnect the injection camera and delete the hal interface.
         void injectionDisconnectImpl();
@@ -1407,6 +1416,17 @@
         // Generated injection camera hal interface.
         sp<HalInterface> mInjectedCamHalInterface;
 
+        // Backup of the original camera hal result FMQ.
+        std::unique_ptr<ResultMetadataQueue> mBackupResultMetadataQueue;
+
+        // FMQ writes the result for the injection camera. Must be guarded by
+        // mProcessCaptureResultLock.
+        std::unique_ptr<ResultMetadataQueue> mInjectionResultMetadataQueue;
+
+        // The flag indicates that the stream configuration is complete, the camera device is
+        // active, but the injection camera has not yet been injected.
+        bool mIsStreamConfigCompleteButNotInjected = false;
+
         // Copy the configuration of the internal camera.
         camera3::camera_stream_configuration mInjectionConfig;
 
diff --git a/services/camera/libcameraservice/device3/Camera3DeviceInjectionMethods.cpp b/services/camera/libcameraservice/device3/Camera3DeviceInjectionMethods.cpp
index f145dac..7026934 100644
--- a/services/camera/libcameraservice/device3/Camera3DeviceInjectionMethods.cpp
+++ b/services/camera/libcameraservice/device3/Camera3DeviceInjectionMethods.cpp
@@ -86,7 +86,7 @@
         return DEAD_OBJECT;
     }
 
-    std::unique_ptr<ResultMetadataQueue>& resQueue = parent->mResultMetadataQueue;
+    std::unique_ptr<ResultMetadataQueue>& resQueue = mInjectionResultMetadataQueue;
     auto resultQueueRet = session->getCaptureResultMetadataQueue(
         [&resQueue](const auto& descriptor) {
             resQueue = std::make_unique<ResultMetadataQueue>(descriptor);
@@ -127,10 +127,8 @@
 
 status_t Camera3Device::Camera3DeviceInjectionMethods::injectCamera(
         camera3::camera_stream_configuration& injectionConfig,
-        std::vector<uint32_t>& injectionBufferSizes) {
+        const std::vector<uint32_t>& injectionBufferSizes) {
     status_t res = NO_ERROR;
-    mInjectionConfig = injectionConfig;
-    mInjectionBufferSizes = injectionBufferSizes;
 
     if (mInjectedCamHalInterface == nullptr) {
         ALOGE("%s: mInjectedCamHalInterface does not exist!", __FUNCTION__);
@@ -148,7 +146,6 @@
     if (parent->mStatus == STATUS_ACTIVE) {
         ALOGV("%s: Let the device be IDLE and the request thread is paused",
                 __FUNCTION__);
-        parent->mPauseStateNotify = true;
         res = parent->internalPauseAndWaitLocked(maxExpectedDuration);
         if (res != OK) {
             ALOGE("%s: Can't pause captures to inject camera!", __FUNCTION__);
@@ -188,7 +185,7 @@
         ALOGV("%s: Restarting activity to inject camera", __FUNCTION__);
         // Reuse current operating mode and session parameters for new stream
         // config.
-        parent->internalUpdateStatusLocked(STATUS_ACTIVE);
+        parent->internalResumeLocked();
     }
 
     return OK;
@@ -196,6 +193,11 @@
 
 status_t Camera3Device::Camera3DeviceInjectionMethods::stopInjection() {
     status_t res = NO_ERROR;
+    mIsStreamConfigCompleteButNotInjected = false;
+    if (mInjectionConfig.streams != nullptr) {
+        delete [] mInjectionConfig.streams;
+        mInjectionConfig.streams = nullptr;
+    }
 
     sp<Camera3Device> parent = mParent.promote();
     if (parent == nullptr) {
@@ -208,7 +210,6 @@
     if (parent->mStatus == STATUS_ACTIVE) {
         ALOGV("%s: Let the device be IDLE and the request thread is paused",
                 __FUNCTION__);
-        parent->mPauseStateNotify = true;
         res = parent->internalPauseAndWaitLocked(maxExpectedDuration);
         if (res != OK) {
             ALOGE("%s: Can't pause captures to stop injection!", __FUNCTION__);
@@ -229,7 +230,7 @@
         ALOGV("%s: Restarting activity to stop injection", __FUNCTION__);
         // Reuse current operating mode and session parameters for new stream
         // config.
-        parent->internalUpdateStatusLocked(STATUS_ACTIVE);
+        parent->internalResumeLocked();
     }
 
     return OK;
@@ -243,6 +244,10 @@
     }
 }
 
+bool Camera3Device::Camera3DeviceInjectionMethods::isStreamConfigCompleteButNotInjected() {
+    return mIsStreamConfigCompleteButNotInjected;
+}
+
 const String8& Camera3Device::Camera3DeviceInjectionMethods::getInjectedCamId()
         const {
     return mInjectedCamId;
@@ -260,10 +265,26 @@
     *injectionBufferSizes = mInjectionBufferSizes;
 }
 
+void Camera3Device::Camera3DeviceInjectionMethods::storeInjectionConfig(
+        const camera3::camera_stream_configuration& injectionConfig,
+        const std::vector<uint32_t>& injectionBufferSizes) {
+    mIsStreamConfigCompleteButNotInjected = true;
+    if (mInjectionConfig.streams != nullptr) {
+        delete [] mInjectionConfig.streams;
+        mInjectionConfig.streams = nullptr;
+    }
+    mInjectionConfig = injectionConfig;
+    mInjectionConfig.streams =
+        (android::camera3::camera_stream_t **) new camera_stream_t*[injectionConfig.num_streams];
+    for (size_t i = 0; i < injectionConfig.num_streams; i++) {
+        mInjectionConfig.streams[i] = injectionConfig.streams[i];
+    }
+    mInjectionBufferSizes = injectionBufferSizes;
+}
 
 status_t Camera3Device::Camera3DeviceInjectionMethods::injectionConfigureStreams(
         camera3::camera_stream_configuration& injectionConfig,
-        std::vector<uint32_t>& injectionBufferSizes) {
+        const std::vector<uint32_t>& injectionBufferSizes) {
     ATRACE_CALL();
     status_t res = NO_ERROR;
 
@@ -326,7 +347,6 @@
             mInjectedCamId.string());
 
     auto rc = parent->mPreparerThread->resume();
-
     if (rc != OK) {
         ALOGE("%s: Injection camera %s: Preparer thread failed to resume!",
                  __FUNCTION__, mInjectedCamId.string());
@@ -380,10 +400,18 @@
         return INVALID_OPERATION;
     }
 
-    if (keepBackup && mBackupHalInterface == nullptr) {
-        mBackupHalInterface = parent->mInterface;
-    } else if (!keepBackup) {
+    if (keepBackup) {
+        if (mBackupHalInterface == nullptr) {
+            mBackupHalInterface = parent->mInterface;
+        }
+        if (mBackupResultMetadataQueue == nullptr) {
+            mBackupResultMetadataQueue = std::move(parent->mResultMetadataQueue);
+            parent->mResultMetadataQueue = std::move(mInjectionResultMetadataQueue);
+        }
+    } else {
         mBackupHalInterface = nullptr;
+        parent->mResultMetadataQueue = std::move(mBackupResultMetadataQueue);
+        mBackupResultMetadataQueue = nullptr;
     }
     parent->mInterface = newHalInterface;
 
diff --git a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
index 0204d49..2f4d669 100644
--- a/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
+++ b/services/camera/libcameraservice/device3/Camera3IOStreamBase.cpp
@@ -82,7 +82,7 @@
             camera_stream::width, camera_stream::height,
             camera_stream::format, camera_stream::data_space);
     lines.appendFormat("      Max size: %zu\n", mMaxSize);
-    lines.appendFormat("      Combined usage: %" PRIu64 ", max HAL buffers: %d\n",
+    lines.appendFormat("      Combined usage: 0x%" PRIx64 ", max HAL buffers: %d\n",
             mUsage | consumerUsage, camera_stream::max_buffers);
     if (strlen(camera_stream::physical_camera_id) > 0) {
         lines.appendFormat("      Physical camera id: %s\n", camera_stream::physical_camera_id);
diff --git a/services/mediacodec/registrant/Android.bp b/services/mediacodec/registrant/Android.bp
index 696b967..d10e339 100644
--- a/services/mediacodec/registrant/Android.bp
+++ b/services/mediacodec/registrant/Android.bp
@@ -7,7 +7,7 @@
     default_applicable_licenses: ["frameworks_av_services_mediacodec_license"],
 }
 
-cc_library_shared {
+cc_library {
     name: "libmedia_codecserviceregistrant",
     vendor_available: true,
     srcs: [
diff --git a/services/mediacodec/registrant/fuzzer/Android.bp b/services/mediacodec/registrant/fuzzer/Android.bp
new file mode 100644
index 0000000..22c10a8
--- /dev/null
+++ b/services/mediacodec/registrant/fuzzer/Android.bp
@@ -0,0 +1,37 @@
+/*
+ * 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.
+ */
+
+cc_fuzz {
+    name: "codecServiceRegistrant_fuzzer",
+    srcs: [
+        "codecServiceRegistrant_fuzzer.cpp",
+    ],
+    static_libs: [
+        "libmedia_codecserviceregistrant",
+    ],
+    header_libs: [
+        "libmedia_headers",
+    ],
+    defaults: [
+        "libcodec2-hidl-defaults",
+    ],
+    fuzz_config: {
+        cc: [
+            "android-media-fuzzing-reports@google.com",
+        ],
+        componentid: 155276,
+    },
+}
diff --git a/services/mediacodec/registrant/fuzzer/README.md b/services/mediacodec/registrant/fuzzer/README.md
new file mode 100644
index 0000000..0ffa063
--- /dev/null
+++ b/services/mediacodec/registrant/fuzzer/README.md
@@ -0,0 +1,56 @@
+# Fuzzer for libmedia_codecserviceregistrant
+
+## Plugin Design Considerations
+The fuzzer plugin for libmedia_codecserviceregistrant is designed based on the understanding of the library and tries to achieve the following:
+
+##### Maximize code coverage
+The configuration parameters are not hardcoded, but instead selected based on
+incoming data. This ensures more code paths are reached by the fuzzer.
+
+libmedia_codecserviceregistrant supports the following parameters:
+1. C2String (parameter name: `c2String`)
+2. Width (parameter name: `width`)
+3. Height (parameter name: `height`)
+4. SamplingRate (parameter name: `samplingRate`)
+5. Channels (parameter name: `channels`)
+6. Stream (parameter name: `stream`)
+
+| Parameter| Valid Values| Configured Value|
+|------------- |-------------| ----- |
+| `c2String` |`String` | Value obtained from FuzzedDataProvider|
+| `width` |`UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+| `height` |`UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+| `samplingRate` |`UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+| `channels` |`UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+| `stream` |`UINT32_MIN` to `UINT32_MAX` | Value obtained from FuzzedDataProvider|
+
+This also ensures that the plugin is always deterministic for any given input.
+
+##### Maximize utilization of input data
+The plugin feeds the entire input data to the libmedia_codecserviceregistrant module.
+This ensures that the plugin tolerates any kind of input (empty, huge,
+malformed, etc) and doesnt `exit()` on any input and thereby increasing the
+chance of identifying vulnerabilities.
+
+## Build
+
+This describes steps to build codecServiceRegistrant_fuzzer binary.
+
+### Android
+
+#### Steps to build
+Build the fuzzer
+```
+  $ mm -j$(nproc) codecServiceRegistrant_fuzzer
+```
+#### Steps to run
+
+To run on device
+```
+  $ adb sync data
+  $ adb shell /data/fuzz/${TARGET_ARCH}/codecServiceRegistrant_fuzzer/codecServiceRegistrant_fuzzer
+```
+
+## References:
+ * http://llvm.org/docs/LibFuzzer.html
+ * https://github.com/google/oss-fuzz
diff --git a/services/mediacodec/registrant/fuzzer/codecServiceRegistrant_fuzzer.cpp b/services/mediacodec/registrant/fuzzer/codecServiceRegistrant_fuzzer.cpp
new file mode 100644
index 0000000..c8a8d8a
--- /dev/null
+++ b/services/mediacodec/registrant/fuzzer/codecServiceRegistrant_fuzzer.cpp
@@ -0,0 +1,155 @@
+/*
+ * 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.
+ */
+#include "../CodecServiceRegistrant.cpp"
+#include "fuzzer/FuzzedDataProvider.h"
+#include <C2Config.h>
+#include <C2Param.h>
+
+using namespace std;
+
+constexpr char kServiceName[] = "software";
+
+class CodecServiceRegistrantFuzzer {
+public:
+  void process(const uint8_t *data, size_t size);
+  ~CodecServiceRegistrantFuzzer() {
+    delete mH2C2;
+    if (mInputSize) {
+      delete mInputSize;
+    }
+    if (mSampleRateInfo) {
+      delete mSampleRateInfo;
+    }
+    if (mChannelCountInfo) {
+      delete mChannelCountInfo;
+    }
+  }
+
+private:
+  void initH2C2ComponentStore();
+  void invokeH2C2ComponentStore();
+  void invokeConfigSM();
+  void invokeQuerySM();
+  H2C2ComponentStore *mH2C2 = nullptr;
+  C2StreamPictureSizeInfo::input *mInputSize = nullptr;
+  C2StreamSampleRateInfo::output *mSampleRateInfo = nullptr;
+  C2StreamChannelCountInfo::output *mChannelCountInfo = nullptr;
+  C2Param::Index mIndex = C2StreamProfileLevelInfo::output::PARAM_TYPE;
+  C2StreamFrameRateInfo::output mFrameRate;
+  FuzzedDataProvider *mFDP = nullptr;
+};
+
+void CodecServiceRegistrantFuzzer::initH2C2ComponentStore() {
+  using namespace ::android::hardware::media::c2;
+  shared_ptr<C2ComponentStore> store =
+      android::GetCodec2PlatformComponentStore();
+  if (!store) {
+    return;
+  }
+  android::sp<V1_1::IComponentStore> storeV1_1 =
+      new V1_1::utils::ComponentStore(store);
+  if (storeV1_1->registerAsService(string(kServiceName)) != android::OK) {
+    return;
+  }
+  string const preferredStoreName = string(kServiceName);
+  sp<IComponentStore> preferredStore =
+      IComponentStore::getService(preferredStoreName.c_str());
+  mH2C2 = new H2C2ComponentStore(preferredStore);
+}
+
+void CodecServiceRegistrantFuzzer::invokeConfigSM() {
+  vector<C2Param *> configParams;
+  uint32_t width = mFDP->ConsumeIntegral<uint32_t>();
+  uint32_t height = mFDP->ConsumeIntegral<uint32_t>();
+  uint32_t samplingRate = mFDP->ConsumeIntegral<uint32_t>();
+  uint32_t channels = mFDP->ConsumeIntegral<uint32_t>();
+  if (mFDP->ConsumeBool()) {
+    mInputSize = new C2StreamPictureSizeInfo::input(0u, width, height);
+    configParams.push_back(mInputSize);
+  } else {
+    if (mFDP->ConsumeBool()) {
+      mSampleRateInfo = new C2StreamSampleRateInfo::output(0u, samplingRate);
+      configParams.push_back(mSampleRateInfo);
+    }
+    if (mFDP->ConsumeBool()) {
+      mChannelCountInfo = new C2StreamChannelCountInfo::output(0u, channels);
+      configParams.push_back(mChannelCountInfo);
+    }
+  }
+  vector<unique_ptr<C2SettingResult>> failures;
+  mH2C2->config_sm(configParams, &failures);
+}
+
+void CodecServiceRegistrantFuzzer::invokeQuerySM() {
+  vector<C2Param *> stackParams;
+  vector<C2Param::Index> heapParamIndices;
+  if (mFDP->ConsumeBool()) {
+    stackParams = {};
+    heapParamIndices = {};
+  } else {
+    uint32_t stream = mFDP->ConsumeIntegral<uint32_t>();
+    mFrameRate.setStream(stream);
+    stackParams.push_back(&mFrameRate);
+    heapParamIndices.push_back(mIndex);
+  }
+  vector<unique_ptr<C2Param>> heapParams;
+  mH2C2->query_sm(stackParams, heapParamIndices, &heapParams);
+}
+
+void CodecServiceRegistrantFuzzer::invokeH2C2ComponentStore() {
+  initH2C2ComponentStore();
+  shared_ptr<C2Component> component;
+  shared_ptr<C2ComponentInterface> interface;
+  string c2String = mFDP->ConsumeRandomLengthString();
+  mH2C2->createComponent(c2String, &component);
+  mH2C2->createInterface(c2String, &interface);
+  invokeConfigSM();
+  invokeQuerySM();
+
+  vector<shared_ptr<C2ParamDescriptor>> params;
+  mH2C2->querySupportedParams_nb(&params);
+
+  C2StoreIonUsageInfo usageInfo;
+  std::vector<C2FieldSupportedValuesQuery> query = {
+      C2FieldSupportedValuesQuery::Possible(
+          C2ParamField::Make(usageInfo, usageInfo.usage)),
+      C2FieldSupportedValuesQuery::Possible(
+          C2ParamField::Make(usageInfo, usageInfo.capacity)),
+  };
+  mH2C2->querySupportedValues_sm(query);
+
+  mH2C2->getName();
+  mH2C2->getParamReflector();
+  mH2C2->listComponents();
+  shared_ptr<C2GraphicBuffer> src;
+  shared_ptr<C2GraphicBuffer> dst;
+  mH2C2->copyBuffer(src, dst);
+}
+
+void CodecServiceRegistrantFuzzer::process(const uint8_t *data, size_t size) {
+  mFDP = new FuzzedDataProvider(data, size);
+  invokeH2C2ComponentStore();
+  /** RegisterCodecServices is called here to improve code coverage */
+  /** as currently it is not called by codecServiceRegistrant       */
+  RegisterCodecServices();
+  delete mFDP;
+}
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t *data, size_t size) {
+  CodecServiceRegistrantFuzzer codecServiceRegistrantFuzzer;
+  codecServiceRegistrantFuzzer.process(data, size);
+  return 0;
+}
diff --git a/services/mediametrics/Android.bp b/services/mediametrics/Android.bp
index 5989181..0351d2d 100644
--- a/services/mediametrics/Android.bp
+++ b/services/mediametrics/Android.bp
@@ -148,7 +148,8 @@
         "statsd_mediaparser.cpp",
         "statsd_nuplayer.cpp",
         "statsd_recorder.cpp",
-        "StringUtils.cpp"
+        "StringUtils.cpp",
+        "ValidateId.cpp",
     ],
 
     proto: {
diff --git a/services/mediametrics/AudioAnalytics.cpp b/services/mediametrics/AudioAnalytics.cpp
index 11ec993..a0dcb55 100644
--- a/services/mediametrics/AudioAnalytics.cpp
+++ b/services/mediametrics/AudioAnalytics.cpp
@@ -28,6 +28,7 @@
 #include "AudioTypes.h"           // string to int conversions
 #include "MediaMetricsService.h"  // package info
 #include "StringUtils.h"
+#include "ValidateId.h"
 
 #define PROP_AUDIO_ANALYTICS_CLOUD_ENABLED "persist.audio.analytics.cloud.enabled"
 
@@ -562,7 +563,7 @@
         const auto flagsForStats = types::lookup<types::INPUT_FLAG, short_enum_type_t>(flags);
         const auto sourceForStats = types::lookup<types::SOURCE_TYPE, short_enum_type_t>(source);
         // Android S
-        const auto logSessionIdForStats = stringutils::sanitizeLogSessionId(logSessionId);
+        const auto logSessionIdForStats = ValidateId::get()->validateId(logSessionId);
 
         LOG(LOG_LEVEL) << "key:" << key
               << " id:" << id
@@ -717,7 +718,7 @@
                  types::lookup<types::TRACK_TRAITS, short_enum_type_t>(traits);
         const auto usageForStats = types::lookup<types::USAGE, short_enum_type_t>(usage);
         // Android S
-        const auto logSessionIdForStats = stringutils::sanitizeLogSessionId(logSessionId);
+        const auto logSessionIdForStats = ValidateId::get()->validateId(logSessionId);
 
         LOG(LOG_LEVEL) << "key:" << key
               << " id:" << id
diff --git a/services/mediametrics/LruSet.h b/services/mediametrics/LruSet.h
new file mode 100644
index 0000000..1f0ab60
--- /dev/null
+++ b/services/mediametrics/LruSet.h
@@ -0,0 +1,123 @@
+/*
+ * 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.
+ */
+
+#pragma once
+
+#include <list>
+#include <sstream>
+#include <unordered_map>
+
+namespace android::mediametrics {
+
+/**
+ * LruSet keeps a set of the last "Size" elements added or accessed.
+ *
+ * (Lru stands for least-recently-used eviction policy).
+ *
+ * Runs in O(1) time for add, remove, and check.  Internally implemented
+ * with an unordered_map and a list.  In order to remove elements,
+ * a list iterator is stored in the unordered_map
+ * (noting that std::list::erase() contractually
+ * does not affect iterators other than the one erased).
+ */
+
+template <typename T>
+class LruSet {
+    const size_t mMaxSize;
+    std::list<T> mAccessOrder;                 // front is the most recent, back is the oldest.
+    // item T with its access order iterator.
+    std::unordered_map<T, typename std::list<T>::iterator> mMap;
+
+public:
+    /**
+     * Constructs a LruSet which checks whether the element was
+     * accessed or added recently.
+     *
+     * The parameter maxSize is used to cap growth of LruSet;
+     * eviction is based on least recently used LRU.
+     * If maxSize is zero, the LruSet contains no elements
+     * and check() always returns false.
+     *
+     * \param maxSize the maximum number of elements that are tracked.
+     */
+    explicit LruSet(size_t maxSize) : mMaxSize(maxSize) {}
+
+    /**
+     * Returns the number of entries in the LruSet.
+     *
+     * This is a number between 0 and maxSize.
+     */
+    size_t size() const {
+        return mMap.size();
+    }
+
+    /** Clears the container contents. */
+    void clear() {
+        mMap.clear();
+        mAccessOrder.clear();
+    }
+
+    /** Returns a string dump of the last n entries. */
+    std::string dump(size_t n) const {
+        std::stringstream ss;
+        auto it = mAccessOrder.cbegin();
+        for (size_t i = 0; i < n && it != mAccessOrder.cend(); ++i) {
+            ss << *it++ << "\n";
+        }
+        return ss.str();
+    }
+
+    /** Adds a new item to the set. */
+    void add(const T& t) {
+        if (mMaxSize == 0) return;
+        auto it = mMap.find(t);
+        if (it != mMap.end()) { // already exists.
+            mAccessOrder.erase(it->second);  // move-to-front on the chronologically ordered list.
+        } else if (mAccessOrder.size() >= mMaxSize) {
+            const T last = mAccessOrder.back();
+            mAccessOrder.pop_back();
+            mMap.erase(last);
+        }
+        mAccessOrder.push_front(t);
+        mMap[t] = mAccessOrder.begin();
+    }
+
+    /**
+     * Removes an item from the set.
+     *
+     * \param t item to be removed.
+     * \return false if the item doesn't exist.
+     */
+    bool remove(const T& t) {
+        auto it = mMap.find(t);
+        if (it == mMap.end()) return false;
+        mAccessOrder.erase(it->second);
+        mMap.erase(it);
+        return true;
+    }
+
+    /** Returns true if t is present (and moves the access order of t to the front). */
+    bool check(const T& t) { // not const, as it adjusts the least-recently-used order.
+        auto it = mMap.find(t);
+        if (it == mMap.end()) return false;
+        mAccessOrder.erase(it->second);
+        mAccessOrder.push_front(it->first);
+        it->second = mAccessOrder.begin();
+        return true;
+    }
+};
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/MediaMetricsService.cpp b/services/mediametrics/MediaMetricsService.cpp
index 1d64878..35e0ae4 100644
--- a/services/mediametrics/MediaMetricsService.cpp
+++ b/services/mediametrics/MediaMetricsService.cpp
@@ -19,6 +19,7 @@
 #include <utils/Log.h>
 
 #include "MediaMetricsService.h"
+#include "ValidateId.h"
 #include "iface_statsd.h"
 
 #include <pwd.h> //getpwuid
@@ -204,6 +205,15 @@
     // now attach either the item or its dup to a const shared pointer
     std::shared_ptr<const mediametrics::Item> sitem(release ? item : item->dup());
 
+    // register log session ids with singleton.
+    if (startsWith(item->getKey(), "metrics.manager")) {
+        std::string logSessionId;
+        if (item->get("logSessionId", &logSessionId)
+                && mediametrics::stringutils::isLogSessionId(logSessionId.c_str())) {
+            mediametrics::ValidateId::get()->registerId(logSessionId);
+        }
+    }
+
     (void)mAudioAnalytics.submit(sitem, isTrusted);
 
     (void)dump2Statsd(sitem, mStatsdLog);  // failure should be logged in function.
@@ -309,6 +319,9 @@
                 result << "-- some lines may be truncated --\n";
             }
 
+            result << "LogSessionId:\n"
+                   << mediametrics::ValidateId::get()->dump();
+
             // Dump the statsd atoms we sent out.
             result << "Statsd atoms:\n"
                    << mStatsdLog->dumpToString("  " /* prefix */,
diff --git a/services/mediametrics/TransactionLog.h b/services/mediametrics/TransactionLog.h
index 0ca4639..fd42518 100644
--- a/services/mediametrics/TransactionLog.h
+++ b/services/mediametrics/TransactionLog.h
@@ -158,7 +158,7 @@
                 ++it) {
             if (ll <= 0) break;
             if (prefix != nullptr && !startsWith(it->first, prefix)) break;
-            auto [s, l] = dumpMapTimeItem(it->second, ll - 1, sinceNs, prefix);
+            std::tie(s, l) = dumpMapTimeItem(it->second, ll - 1, sinceNs, prefix);
             if (l == 0) continue; // don't show empty groups (due to sinceNs).
             ss << " " << it->first << "\n" << s;
             ll -= l + 1;
diff --git a/services/mediametrics/ValidateId.cpp b/services/mediametrics/ValidateId.cpp
new file mode 100644
index 0000000..0cc8593
--- /dev/null
+++ b/services/mediametrics/ValidateId.cpp
@@ -0,0 +1,65 @@
+/*
+ * 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 "MediaMetricsService"  // not ValidateId
+#include <utils/Log.h>
+
+#include "ValidateId.h"
+
+namespace android::mediametrics {
+
+std::string ValidateId::dump() const
+{
+    std::stringstream ss;
+    ss << "Entries:" << mIdSet.size() << "  InvalidIds:" << mInvalidIds << "\n";
+    ss << mIdSet.dump(10);
+    return ss.str();
+}
+
+void ValidateId::registerId(const std::string& id)
+{
+    if (id.empty()) return;
+    if (!mediametrics::stringutils::isLogSessionId(id.c_str())) {
+        ALOGW("%s: rejecting malformed id %s", __func__, id.c_str());
+        return;
+    }
+    ALOGV("%s: registering %s", __func__, id.c_str());
+    mIdSet.add(id);
+}
+
+const std::string& ValidateId::validateId(const std::string& id)
+{
+    static const std::string empty{};
+    if (id.empty()) return empty;
+
+    // reject because the id is malformed
+    if (!mediametrics::stringutils::isLogSessionId(id.c_str())) {
+        ALOGW("%s: rejecting malformed id %s", __func__, id.c_str());
+        ++mInvalidIds;
+        return empty;
+    }
+
+    // reject because the id is unregistered
+    if (!mIdSet.check(id)) {
+        ALOGW("%s: rejecting unregistered id %s", __func__, id.c_str());
+        ++mInvalidIds;
+        return empty;
+    }
+    return id;
+}
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/ValidateId.h b/services/mediametrics/ValidateId.h
new file mode 100644
index 0000000..166b39a
--- /dev/null
+++ b/services/mediametrics/ValidateId.h
@@ -0,0 +1,80 @@
+/*
+ * 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.
+ */
+
+#pragma once
+
+#include "LruSet.h"
+#include "StringUtils.h"
+#include "Wrap.h"
+
+namespace android::mediametrics {
+
+/*
+ * ValidateId is used to check whether the log session id is properly formed
+ * and has been registered (i.e. from the Java MediaMetricsManagerService).
+ *
+ * The default memory window to track registered ids is set to SINGLETON_LRU_SET_SIZE.
+ *
+ * This class is not thread-safe, but the singleton returned by get() uses LockWrap<>
+ * to ensure thread-safety.
+ */
+class ValidateId {
+    mediametrics::LruSet<std::string> mIdSet;
+    size_t mInvalidIds = 0;  // count invalid ids encountered.
+public:
+    /** Creates a ValidateId object with size memory window. */
+    explicit ValidateId(size_t size) : mIdSet{size} {}
+
+    /** Returns a string dump of recent contents and stats. */
+    std::string dump() const;
+
+    /**
+     * Registers the id string.
+     *
+     * If id string is malformed (not 16 Base64Url chars), it is ignored.
+     * Once registered, calling validateId() will return id (instead of the empty string).
+     * ValidateId may "forget" the id after not encountering it within the past N ids,
+     * where N is the size set in the constructor.
+     *
+     * param id string (from MediaMetricsManagerService).
+     */
+    void registerId(const std::string& id);
+
+    /**
+     * Returns the empty string if id string is malformed (not 16 Base64Url chars)
+     * or if id string has not been seen (in the recent size ids);
+     * otherwise it returns the same id parameter.
+     *
+     * \param id string (to be sent to statsd).
+     */
+    const std::string& validateId(const std::string& id);
+
+    /** Singleton set size */
+    static inline constexpr size_t SINGLETON_LRU_SET_SIZE = 2000;
+
+    using LockedValidateId = mediametrics::LockWrap<ValidateId>;
+    /**
+     * Returns a singleton locked ValidateId object that is thread-safe using LockWrap<>.
+     *
+     * The Singleton ValidateId object is created with size LRU_SET_SIZE (during first call).
+     */
+    static inline LockedValidateId& get() {
+        static LockedValidateId privateSet{SINGLETON_LRU_SET_SIZE};
+        return privateSet;
+    }
+};
+
+} // namespace android::mediametrics
diff --git a/services/mediametrics/fuzzer/mediametrics_service_fuzzer.cpp b/services/mediametrics/fuzzer/mediametrics_service_fuzzer.cpp
index 8b0b479..06ab16e 100644
--- a/services/mediametrics/fuzzer/mediametrics_service_fuzzer.cpp
+++ b/services/mediametrics/fuzzer/mediametrics_service_fuzzer.cpp
@@ -48,6 +48,7 @@
     void invokeAudioAnalytics(const uint8_t *data, size_t size);
     void invokeTimedAction(const uint8_t *data, size_t size);
     void process(const uint8_t *data, size_t size);
+    std::atomic_int mValue = 0;
 };
 
 void MediaMetricsServiceFuzzer::invokeStartsWith(const uint8_t *data, size_t size) {
@@ -342,11 +343,10 @@
 void MediaMetricsServiceFuzzer::invokeTimedAction(const uint8_t *data, size_t size) {
     FuzzedDataProvider fdp = FuzzedDataProvider(data, size);
     android::mediametrics::TimedAction timedAction;
-    std::atomic_int value = 0;
 
     while (fdp.remaining_bytes()) {
         timedAction.postIn(std::chrono::seconds(fdp.ConsumeIntegral<int32_t>()),
-                           [&value] { ++value; });
+                           [this] { ++mValue; });
         timedAction.size();
     }
 }
diff --git a/services/mediametrics/statsd_audiorecord.cpp b/services/mediametrics/statsd_audiorecord.cpp
index 41efcaa..c53b6f3 100644
--- a/services/mediametrics/statsd_audiorecord.cpp
+++ b/services/mediametrics/statsd_audiorecord.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -143,8 +143,7 @@
     // log_session_id (string)
     std::string logSessionId;
     (void)item->getString("android.media.audiorecord.logSessionId", &logSessionId);
-    const auto log_session_id =
-            mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+    const auto log_session_id = mediametrics::ValidateId::get()->validateId(logSessionId);
 
     android::util::BytesField bf_serialized( serialized.c_str(), serialized.size());
     int result = android::util::stats_write(android::util::MEDIAMETRICS_AUDIORECORD_REPORTED,
diff --git a/services/mediametrics/statsd_audiotrack.cpp b/services/mediametrics/statsd_audiotrack.cpp
index 59627ae..707effd 100644
--- a/services/mediametrics/statsd_audiotrack.cpp
+++ b/services/mediametrics/statsd_audiotrack.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -137,8 +137,7 @@
     // log_session_id (string)
     std::string logSessionId;
     (void)item->getString("android.media.audiotrack.logSessionId", &logSessionId);
-    const auto log_session_id =
-            mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+    const auto log_session_id = mediametrics::ValidateId::get()->validateId(logSessionId);
 
     android::util::BytesField bf_serialized( serialized.c_str(), serialized.size());
     int result = android::util::stats_write(android::util::MEDIAMETRICS_AUDIOTRACK_REPORTED,
diff --git a/services/mediametrics/statsd_codec.cpp b/services/mediametrics/statsd_codec.cpp
index 46cbdc8..8581437 100644
--- a/services/mediametrics/statsd_codec.cpp
+++ b/services/mediametrics/statsd_codec.cpp
@@ -34,7 +34,7 @@
 
 #include "cleaner.h"
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -228,7 +228,7 @@
 
     std::string sessionId;
     if (item->getString("android.media.mediacodec.log-session-id", &sessionId)) {
-        sessionId = mediametrics::stringutils::sanitizeLogSessionId(sessionId);
+        sessionId = mediametrics::ValidateId::get()->validateId(sessionId);
         metrics_proto.set_log_session_id(sessionId);
     }
     AStatsEvent_writeString(event, codec.c_str());
diff --git a/services/mediametrics/statsd_extractor.cpp b/services/mediametrics/statsd_extractor.cpp
index bcf2e0a..a8bfeaa 100644
--- a/services/mediametrics/statsd_extractor.cpp
+++ b/services/mediametrics/statsd_extractor.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -86,7 +86,7 @@
 
     std::string log_session_id;
     if (item->getString("android.media.mediaextractor.logSessionId", &log_session_id)) {
-        log_session_id = mediametrics::stringutils::sanitizeLogSessionId(log_session_id);
+        log_session_id = mediametrics::ValidateId::get()->validateId(log_session_id);
         metrics_proto.set_log_session_id(log_session_id);
     }
 
diff --git a/services/mediametrics/statsd_mediaparser.cpp b/services/mediametrics/statsd_mediaparser.cpp
index 921b320..67ca874b 100644
--- a/services/mediametrics/statsd_mediaparser.cpp
+++ b/services/mediametrics/statsd_mediaparser.cpp
@@ -31,7 +31,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/enums/stats/mediametrics/mediametrics.pb.h"
 #include "iface_statsd.h"
 
@@ -81,7 +81,7 @@
 
     std::string logSessionId;
     item->getString("android.media.mediaparser.logSessionId", &logSessionId);
-    logSessionId = mediametrics::stringutils::sanitizeLogSessionId(logSessionId);
+    logSessionId = mediametrics::ValidateId::get()->validateId(logSessionId);
 
     int result = android::util::stats_write(android::util::MEDIAMETRICS_MEDIAPARSER_REPORTED,
                                timestamp_nanos,
diff --git a/services/mediametrics/statsd_recorder.cpp b/services/mediametrics/statsd_recorder.cpp
index b29ad73..5f54a68 100644
--- a/services/mediametrics/statsd_recorder.cpp
+++ b/services/mediametrics/statsd_recorder.cpp
@@ -32,7 +32,7 @@
 #include <statslog.h>
 
 #include "MediaMetricsService.h"
-#include "StringUtils.h"
+#include "ValidateId.h"
 #include "frameworks/proto_logging/stats/message/mediametrics_message.pb.h"
 #include "iface_statsd.h"
 
@@ -59,7 +59,7 @@
     // string kRecorderLogSessionId = "android.media.mediarecorder.log-session-id";
     std::string log_session_id;
     if (item->getString("android.media.mediarecorder.log-session-id", &log_session_id)) {
-        log_session_id = mediametrics::stringutils::sanitizeLogSessionId(log_session_id);
+        log_session_id = mediametrics::ValidateId::get()->validateId(log_session_id);
         metrics_proto.set_log_session_id(log_session_id);
     }
     // string kRecorderAudioMime = "android.media.mediarecorder.audio.mime";
diff --git a/services/mediametrics/tests/mediametrics_tests.cpp b/services/mediametrics/tests/mediametrics_tests.cpp
index 2336d6f..69ec947 100644
--- a/services/mediametrics/tests/mediametrics_tests.cpp
+++ b/services/mediametrics/tests/mediametrics_tests.cpp
@@ -28,6 +28,7 @@
 
 #include "AudioTypes.h"
 #include "StringUtils.h"
+#include "ValidateId.h"
 
 using namespace android;
 
@@ -1127,3 +1128,100 @@
    validId2[3] = '!';
    ASSERT_EQ("", mediametrics::stringutils::sanitizeLogSessionId(validId2));
 }
+
+TEST(mediametrics_tests, LruSet) {
+    constexpr size_t LRU_SET_SIZE = 2;
+    mediametrics::LruSet<std::string> lruSet(LRU_SET_SIZE);
+
+    // test adding a couple strings.
+    lruSet.add("abc");
+    ASSERT_EQ(1u, lruSet.size());
+    ASSERT_TRUE(lruSet.check("abc"));
+    lruSet.add("def");
+    ASSERT_EQ(2u, lruSet.size());
+
+    // now adding the third string causes eviction of the oldest.
+    lruSet.add("ghi");
+    ASSERT_FALSE(lruSet.check("abc"));
+    ASSERT_TRUE(lruSet.check("ghi"));
+    ASSERT_TRUE(lruSet.check("def"));  // "def" is most recent.
+    ASSERT_EQ(2u, lruSet.size());      // "abc" is correctly discarded.
+
+    // adding another string will evict the oldest.
+    lruSet.add("foo");
+    ASSERT_FALSE(lruSet.check("ghi")); // note: "ghi" discarded when "foo" added.
+    ASSERT_TRUE(lruSet.check("foo"));
+    ASSERT_TRUE(lruSet.check("def"));
+
+    // manual removing of a string works, too.
+    ASSERT_TRUE(lruSet.remove("def"));
+    ASSERT_FALSE(lruSet.check("def")); // we manually removed "def".
+    ASSERT_TRUE(lruSet.check("foo"));  // "foo" is still there.
+    ASSERT_EQ(1u, lruSet.size());
+
+    // you can't remove a string that has not been added.
+    ASSERT_FALSE(lruSet.remove("bar")); // Note: "bar" doesn't exist, so remove returns false.
+    ASSERT_EQ(1u, lruSet.size());
+
+    lruSet.add("foo");   // adding "foo" (which already exists) doesn't change size.
+    ASSERT_EQ(1u, lruSet.size());
+    lruSet.add("bar");   // add "bar"
+    ASSERT_EQ(2u, lruSet.size());
+    lruSet.add("glorp"); // add "glorp" evicts "foo".
+    ASSERT_EQ(2u, lruSet.size());
+    ASSERT_TRUE(lruSet.check("bar"));
+    ASSERT_TRUE(lruSet.check("glorp"));
+    ASSERT_FALSE(lruSet.check("foo"));
+}
+
+TEST(mediametrics_tests, LruSet0) {
+    constexpr size_t LRU_SET_SIZE = 0;
+    mediametrics::LruSet<std::string> lruSet(LRU_SET_SIZE);
+
+    lruSet.add("a");
+    ASSERT_EQ(0u, lruSet.size());
+    ASSERT_FALSE(lruSet.check("a"));
+    ASSERT_FALSE(lruSet.remove("a")); // never added.
+    ASSERT_EQ(0u, lruSet.size());
+}
+
+// Returns a 16 Base64Url string representing the decimal representation of value
+// (with leading 0s) e.g. 0000000000000000, 0000000000000001, 0000000000000002, ...
+static std::string generateId(size_t value)
+{
+    char id[16 + 1]; // to be filled with 16 Base64Url chars (and zero termination)
+    char *sptr = id + 16; // start at the end.
+    *sptr-- = 0; // zero terminate.
+    // output the digits from least significant to most significant.
+    while (value) {
+        *sptr-- = value % 10;
+        value /= 10;
+    }
+    // add leading 0's
+    while (sptr > id) {
+        *sptr-- = '0';
+    }
+    return std::string(id);
+}
+
+TEST(mediametrics_tests, ValidateId) {
+    constexpr size_t LRU_SET_SIZE = 3;
+    constexpr size_t IDS = 10;
+    static_assert(IDS > LRU_SET_SIZE);  // IDS must be greater than LRU_SET_SIZE.
+    mediametrics::ValidateId validateId(LRU_SET_SIZE);
+
+
+    // register IDs as integer strings counting from 0.
+    for (size_t i = 0; i < IDS; ++i) {
+        validateId.registerId(generateId(i));
+    }
+
+    // only the last LRU_SET_SIZE exist.
+    for (size_t i = 0; i < IDS - LRU_SET_SIZE; ++i) {
+        ASSERT_EQ("", validateId.validateId(generateId(i)));
+    }
+    for (size_t i = IDS - LRU_SET_SIZE; i < IDS; ++i) {
+        const std::string id = generateId(i);
+        ASSERT_EQ(id, validateId.validateId(id));
+    }
+}
diff --git a/services/oboeservice/AAudioServiceEndpoint.cpp b/services/oboeservice/AAudioServiceEndpoint.cpp
index 13dd3d3..340076e 100644
--- a/services/oboeservice/AAudioServiceEndpoint.cpp
+++ b/services/oboeservice/AAudioServiceEndpoint.cpp
@@ -59,6 +59,7 @@
     result << "    Device Id:            " << getDeviceId() << "\n";
     result << "    Sample Rate:          " << getSampleRate() << "\n";
     result << "    Channel Count:        " << getSamplesPerFrame() << "\n";
+    result << "    Channel Mask:         0x" << std::hex << getChannelMask() << std::dec << "\n";
     result << "    Format:               " << getFormat() << "\n";
     result << "    Frames Per Burst:     " << mFramesPerBurst << "\n";
     result << "    Usage:                " << getUsage() << "\n";
@@ -164,6 +165,10 @@
         configuration.getSamplesPerFrame() != getSamplesPerFrame()) {
         return false;
     }
+    if (configuration.getChannelMask() != AAUDIO_UNSPECIFIED &&
+        configuration.getChannelMask() != getChannelMask()) {
+        return false;
+    }
     return true;
 }
 
diff --git a/services/oboeservice/AAudioServiceEndpointMMAP.cpp b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
index a08098c..35a0890 100644
--- a/services/oboeservice/AAudioServiceEndpointMMAP.cpp
+++ b/services/oboeservice/AAudioServiceEndpointMMAP.cpp
@@ -126,20 +126,15 @@
     }
     config.sample_rate = aaudioSampleRate;
 
-    int32_t aaudioSamplesPerFrame = getSamplesPerFrame();
-
     const aaudio_direction_t direction = getDirection();
 
+    config.channel_mask = AAudio_getChannelMaskForOpen(
+            getChannelMask(), getSamplesPerFrame(), direction == AAUDIO_DIRECTION_INPUT);
+
     if (direction == AAUDIO_DIRECTION_OUTPUT) {
-        config.channel_mask = (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
-                              ? AUDIO_CHANNEL_OUT_STEREO
-                              : audio_channel_out_mask_from_count(aaudioSamplesPerFrame);
         mHardwareTimeOffsetNanos = OUTPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at DAC later
 
     } else if (direction == AAUDIO_DIRECTION_INPUT) {
-        config.channel_mask =  (aaudioSamplesPerFrame == AAUDIO_UNSPECIFIED)
-                               ? AUDIO_CHANNEL_IN_STEREO
-                               : audio_channel_in_mask_from_count(aaudioSamplesPerFrame);
         mHardwareTimeOffsetNanos = INPUT_ESTIMATED_HARDWARE_OFFSET_NANOS; // frames at ADC earlier
 
     } else {
@@ -225,9 +220,9 @@
     }
 
     // Get information about the stream and pass it back to the caller.
-    setSamplesPerFrame((direction == AAUDIO_DIRECTION_OUTPUT)
-                       ? audio_channel_count_from_out_mask(config.channel_mask)
-                       : audio_channel_count_from_in_mask(config.channel_mask));
+    setChannelMask(AAudioConvert_androidToAAudioChannelMask(
+            config.channel_mask, getDirection() == AAUDIO_DIRECTION_INPUT,
+            AAudio_isChannelIndexMask(config.channel_mask)));
 
     // AAudio creates a copy of this FD and retains ownership of the copy.
     // Assume that AudioFlinger will close the original shared_memory_fd.
@@ -247,9 +242,9 @@
     setFormat(config.format);
     setSampleRate(config.sample_rate);
 
-    ALOGD("%s() actual rate = %d, channels = %d"
-          ", deviceId = %d, capacity = %d\n",
-          __func__, getSampleRate(), getSamplesPerFrame(), deviceId, getBufferCapacity());
+    ALOGD("%s() actual rate = %d, channels = %d channelMask = %#x, deviceId = %d, capacity = %d\n",
+          __func__, getSampleRate(), getSamplesPerFrame(), getChannelMask(),
+          deviceId, getBufferCapacity());
 
     ALOGD("%s() format = 0x%08x, frame size = %d, burst size = %d",
           __func__, getFormat(), calculateBytesPerFrame(), mFramesPerBurst);
diff --git a/services/oboeservice/AAudioServiceEndpointShared.cpp b/services/oboeservice/AAudioServiceEndpointShared.cpp
index 5fbcadb..5af0a91 100644
--- a/services/oboeservice/AAudioServiceEndpointShared.cpp
+++ b/services/oboeservice/AAudioServiceEndpointShared.cpp
@@ -78,7 +78,7 @@
     result = mStreamInternal->open(builder);
 
     setSampleRate(mStreamInternal->getSampleRate());
-    setSamplesPerFrame(mStreamInternal->getSamplesPerFrame());
+    setChannelMask(mStreamInternal->getChannelMask());
     setDeviceId(mStreamInternal->getDeviceId());
     setSessionId(mStreamInternal->getSessionId());
     setFormat(AUDIO_FORMAT_PCM_FLOAT); // force for mixer
diff --git a/services/oboeservice/AAudioServiceStreamBase.cpp b/services/oboeservice/AAudioServiceStreamBase.cpp
index 34ddd4d..4ffc127 100644
--- a/services/oboeservice/AAudioServiceStreamBase.cpp
+++ b/services/oboeservice/AAudioServiceStreamBase.cpp
@@ -73,7 +73,8 @@
 }
 
 std::string AAudioServiceStreamBase::dumpHeader() {
-    return std::string("    T   Handle   UId   Port Run State Format Burst Chan Capacity");
+    return std::string(
+            "    T   Handle   UId   Port Run State Format Burst Chan Mask     Capacity");
 }
 
 std::string AAudioServiceStreamBase::dump() const {
@@ -88,6 +89,7 @@
     result << std::setw(7) << getFormat();
     result << std::setw(6) << mFramesPerBurst;
     result << std::setw(5) << getSamplesPerFrame();
+    result << std::setw(8) << std::hex << getChannelMask() << std::dec;
     result << std::setw(9) << getBufferCapacity();
 
     return result.str();
diff --git a/services/oboeservice/AAudioServiceStreamShared.cpp b/services/oboeservice/AAudioServiceStreamShared.cpp
index c665cda..ad06d97 100644
--- a/services/oboeservice/AAudioServiceStreamShared.cpp
+++ b/services/oboeservice/AAudioServiceStreamShared.cpp
@@ -164,11 +164,11 @@
         goto error;
     }
 
-    setSamplesPerFrame(configurationInput.getSamplesPerFrame());
-    if (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) {
-        setSamplesPerFrame(endpoint->getSamplesPerFrame());
+    setChannelMask(configurationInput.getChannelMask());
+    if (getChannelMask() == AAUDIO_UNSPECIFIED) {
+        setChannelMask(endpoint->getChannelMask());
     } else if (getSamplesPerFrame() != endpoint->getSamplesPerFrame()) {
-        ALOGD("%s() mSamplesPerFrame = %d, need %d",
+        ALOGD("%s() mSamplesPerFrame = %#x, need %#x",
               __func__, getSamplesPerFrame(), endpoint->getSamplesPerFrame());
         result = AAUDIO_ERROR_OUT_OF_RANGE;
         goto error;
diff --git a/services/oboeservice/fuzzer/README.md b/services/oboeservice/fuzzer/README.md
index 00b85df..ae7af3eb 100644
--- a/services/oboeservice/fuzzer/README.md
+++ b/services/oboeservice/fuzzer/README.md
@@ -15,7 +15,7 @@
 4. InService
 5. DeviceId
 6. SampleRate
-7. SamplesPerFrame
+7. ChannelMask
 8. Direction
 9. SharingMode
 10. Usage
@@ -31,7 +31,7 @@
 | `InService`   | `bool` | Value obtained from FuzzedDataProvider |
 | `DeviceId`   | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
 | `SampleRate`   | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
-| `SamplesPerFrame` | `INT32_MIN` to `INT32_MAX` | Value obtained from FuzzedDataProvider |
+| `ChannelMask` | `AAUDIO_UNSPECIFIED`, `AAUDIO_CHANNEL_INDEX_MASK_1`, `AAUDIO_CHANNEL_INDEX_MASK_2`, `AAUDIO_CHANNEL_INDEX_MASK_3`, `AAUDIO_CHANNEL_INDEX_MASK_4`, `AAUDIO_CHANNEL_INDEX_MASK_5`, `AAUDIO_CHANNEL_INDEX_MASK_6`, `AAUDIO_CHANNEL_INDEX_MASK_7`, `AAUDIO_CHANNEL_INDEX_MASK_8`, `AAUDIO_CHANNEL_INDEX_MASK_9`, `AAUDIO_CHANNEL_INDEX_MASK_10`, `AAUDIO_CHANNEL_INDEX_MASK_11`, `AAUDIO_CHANNEL_INDEX_MASK_12`, `AAUDIO_CHANNEL_INDEX_MASK_13`, `AAUDIO_CHANNEL_INDEX_MASK_14`, `AAUDIO_CHANNEL_INDEX_MASK_15`, `AAUDIO_CHANNEL_INDEX_MASK_16`, `AAUDIO_CHANNEL_INDEX_MASK_17`, `AAUDIO_CHANNEL_INDEX_MASK_18`, `AAUDIO_CHANNEL_INDEX_MASK_19`, `AAUDIO_CHANNEL_INDEX_MASK_20`, `AAUDIO_CHANNEL_INDEX_MASK_21`, `AAUDIO_CHANNEL_INDEX_MASK_22`, `AAUDIO_CHANNEL_INDEX_MASK_23`, `AAUDIO_CHANNEL_INDEX_MASK_24`, `AAUDIO_CHANNEL_MONO`, `AAUDIO_CHANNEL_STEREO`, `AAUDIO_CHANNEL_FRONT_BACK`, `AAUDIO_CHANNEL_2POINT0POINT2`, `AAUDIO_CHANNEL_2POINT1POINT2`, `AAUDIO_CHANNEL_3POINT0POINT2`, `AAUDIO_CHANNEL_3POINT1POINT2`, `AAUDIO_CHANNEL_5POINT1`, `AAUDIO_CHANNEL_MONO`, `AAUDIO_CHANNEL_STEREO`, `AAUDIO_CHANNEL_2POINT1`, `AAUDIO_CHANNEL_TRI`, `AAUDIO_CHANNEL_TRI_BACK`, `AAUDIO_CHANNEL_3POINT1`, `AAUDIO_CHANNEL_2POINT0POINT2`, `AAUDIO_CHANNEL_2POINT1POINT2`, `AAUDIO_CHANNEL_3POINT0POINT2`, `AAUDIO_CHANNEL_3POINT1POINT2`, `AAUDIO_CHANNEL_QUAD`, `AAUDIO_CHANNEL_QUAD_SIDE`, `AAUDIO_CHANNEL_SURROUND`, `AAUDIO_CHANNEL_PENTA`, `AAUDIO_CHANNEL_5POINT1`, `AAUDIO_CHANNEL_5POINT1_SIDE`, `AAUDIO_CHANNEL_5POINT1POINT2`, `AAUDIO_CHANNEL_5POINT1POINT4`, `AAUDIO_CHANNEL_6POINT1`, `AAUDIO_CHANNEL_7POINT1`, `AAUDIO_CHANNEL_7POINT1POINT2`, `AAUDIO_CHANNEL_7POINT1POINT4`, `AAUDIO_CHANNEL_9POINT1POINT4`, `AAUDIO_CHANNEL_9POINT1POINT6` | Value obtained from FuzzedDataProvider |
 | `Direction` | `AAUDIO_DIRECTION_OUTPUT`, `AAUDIO_DIRECTION_INPUT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
 | `SharingMode` | `AAUDIO_SHARING_MODE_EXCLUSIVE`, `AAUDIO_SHARING_MODE_SHARED` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
 | `Usage` | `AAUDIO_USAGE_MEDIA`, `AAUDIO_USAGE_VOICE_COMMUNICATION`, `AAUDIO_USAGE_VOICE_COMMUNICATION_SIGNALLING`, `AAUDIO_USAGE_ALARM`, `AAUDIO_USAGE_NOTIFICATION`, `AAUDIO_USAGE_NOTIFICATION_RINGTONE`, `AAUDIO_USAGE_NOTIFICATION_EVENT`, `AAUDIO_USAGE_ASSISTANCE_ACCESSIBILITY`, `AAUDIO_USAGE_ASSISTANCE_NAVIGATION_GUIDANCE`, `AAUDIO_USAGE_ASSISTANCE_SONIFICATION`, `AAUDIO_USAGE_GAME`, `AAUDIO_USAGE_ASSISTANT`, `AAUDIO_SYSTEM_USAGE_EMERGENCY`, `AAUDIO_SYSTEM_USAGE_SAFETY`, `AAUDIO_SYSTEM_USAGE_VEHICLE_STATUS`, `AAUDIO_SYSTEM_USAGE_ANNOUNCEMENT` | Value chosen from valid values by obtaining index from FuzzedDataProvider |
diff --git a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
index 4bc661c..17e8d36 100644
--- a/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
+++ b/services/oboeservice/fuzzer/oboeservice_fuzzer.cpp
@@ -68,10 +68,71 @@
     AAUDIO_INPUT_PRESET_UNPROCESSED,       AAUDIO_INPUT_PRESET_VOICE_PERFORMANCE,
 };
 
+aaudio_channel_mask_t kAAudioChannelMasks[] = {
+    AAUDIO_UNSPECIFIED,
+    AAUDIO_CHANNEL_INDEX_MASK_1,
+    AAUDIO_CHANNEL_INDEX_MASK_2,
+    AAUDIO_CHANNEL_INDEX_MASK_3,
+    AAUDIO_CHANNEL_INDEX_MASK_4,
+    AAUDIO_CHANNEL_INDEX_MASK_5,
+    AAUDIO_CHANNEL_INDEX_MASK_6,
+    AAUDIO_CHANNEL_INDEX_MASK_7,
+    AAUDIO_CHANNEL_INDEX_MASK_8,
+    AAUDIO_CHANNEL_INDEX_MASK_9,
+    AAUDIO_CHANNEL_INDEX_MASK_10,
+    AAUDIO_CHANNEL_INDEX_MASK_11,
+    AAUDIO_CHANNEL_INDEX_MASK_12,
+    AAUDIO_CHANNEL_INDEX_MASK_13,
+    AAUDIO_CHANNEL_INDEX_MASK_14,
+    AAUDIO_CHANNEL_INDEX_MASK_15,
+    AAUDIO_CHANNEL_INDEX_MASK_16,
+    AAUDIO_CHANNEL_INDEX_MASK_17,
+    AAUDIO_CHANNEL_INDEX_MASK_18,
+    AAUDIO_CHANNEL_INDEX_MASK_19,
+    AAUDIO_CHANNEL_INDEX_MASK_20,
+    AAUDIO_CHANNEL_INDEX_MASK_21,
+    AAUDIO_CHANNEL_INDEX_MASK_22,
+    AAUDIO_CHANNEL_INDEX_MASK_23,
+    AAUDIO_CHANNEL_INDEX_MASK_24,
+    AAUDIO_CHANNEL_MONO,
+    AAUDIO_CHANNEL_STEREO,
+    AAUDIO_CHANNEL_FRONT_BACK,
+    AAUDIO_CHANNEL_2POINT0POINT2,
+    AAUDIO_CHANNEL_2POINT1POINT2,
+    AAUDIO_CHANNEL_3POINT0POINT2,
+    AAUDIO_CHANNEL_3POINT1POINT2,
+    AAUDIO_CHANNEL_5POINT1,
+    AAUDIO_CHANNEL_MONO,
+    AAUDIO_CHANNEL_STEREO,
+    AAUDIO_CHANNEL_2POINT1,
+    AAUDIO_CHANNEL_TRI,
+    AAUDIO_CHANNEL_TRI_BACK,
+    AAUDIO_CHANNEL_3POINT1,
+    AAUDIO_CHANNEL_2POINT0POINT2,
+    AAUDIO_CHANNEL_2POINT1POINT2,
+    AAUDIO_CHANNEL_3POINT0POINT2,
+    AAUDIO_CHANNEL_3POINT1POINT2,
+    AAUDIO_CHANNEL_QUAD,
+    AAUDIO_CHANNEL_QUAD_SIDE,
+    AAUDIO_CHANNEL_SURROUND,
+    AAUDIO_CHANNEL_PENTA,
+    AAUDIO_CHANNEL_5POINT1,
+    AAUDIO_CHANNEL_5POINT1_SIDE,
+    AAUDIO_CHANNEL_5POINT1POINT2,
+    AAUDIO_CHANNEL_5POINT1POINT4,
+    AAUDIO_CHANNEL_6POINT1,
+    AAUDIO_CHANNEL_7POINT1,
+    AAUDIO_CHANNEL_7POINT1POINT2,
+    AAUDIO_CHANNEL_7POINT1POINT4,
+    AAUDIO_CHANNEL_9POINT1POINT4,
+    AAUDIO_CHANNEL_9POINT1POINT6,
+};
+
 const size_t kNumAAudioFormats = std::size(kAAudioFormats);
 const size_t kNumAAudioUsages = std::size(kAAudioUsages);
 const size_t kNumAAudioContentTypes = std::size(kAAudioContentTypes);
 const size_t kNumAAudioInputPresets = std::size(kAAudioInputPresets);
+const size_t kNumAAudioChannelMasks = std::size(kAAudioChannelMasks);
 
 class FuzzAAudioClient : public virtual RefBase, public AAudioServiceInterface {
    public:
@@ -305,7 +366,11 @@
 
     request.getConfiguration().setDeviceId(fdp.ConsumeIntegral<int32_t>());
     request.getConfiguration().setSampleRate(fdp.ConsumeIntegral<int32_t>());
-    request.getConfiguration().setSamplesPerFrame(fdp.ConsumeIntegral<int32_t>());
+    request.getConfiguration().setChannelMask((aaudio_channel_mask_t)(
+        fdp.ConsumeBool()
+            ? fdp.ConsumeIntegral<int32_t>()
+            : kAAudioChannelMasks[fdp.ConsumeIntegralInRange<int32_t>(
+                    0, kNumAAudioChannelMasks - 1)]));
     request.getConfiguration().setDirection(
         fdp.ConsumeBool() ? fdp.ConsumeIntegral<int32_t>()
                           : (fdp.ConsumeBool() ? AAUDIO_DIRECTION_OUTPUT : AAUDIO_DIRECTION_INPUT));