diff --git a/media/codec2/components/aac/C2SoftAacDec.cpp b/media/codec2/components/aac/C2SoftAacDec.cpp
index f39620e..9884733 100644
--- a/media/codec2/components/aac/C2SoftAacDec.cpp
+++ b/media/codec2/components/aac/C2SoftAacDec.cpp
@@ -89,11 +89,18 @@
         addParameter(
                 DefineParam(mChannelCount, C2_PARAMKEY_CHANNEL_COUNT)
                 .withDefault(new C2StreamChannelCountInfo::output(0u, 1))
-                .withFields({C2F(mChannelCount, value).inRange(1, 8)})
+                .withFields({C2F(mChannelCount, value).inRange(1, MAX_CHANNEL_COUNT)})
                 .withSetter(Setter<decltype(*mChannelCount)>::StrictValueWithNoDeps)
                 .build());
 
         addParameter(
+                DefineParam(mMaxChannelCount, C2_PARAMKEY_MAX_CHANNEL_COUNT)
+                .withDefault(new C2StreamMaxChannelCountInfo::input(0u, MAX_CHANNEL_COUNT))
+                .withFields({C2F(mMaxChannelCount, value).inRange(1, MAX_CHANNEL_COUNT)})
+                .withSetter(Setter<decltype(*mMaxChannelCount)>::StrictValueWithNoDeps)
+                .build());
+
+        addParameter(
                 DefineParam(mBitrate, C2_PARAMKEY_BITRATE)
                 .withDefault(new C2StreamBitrateInfo::input(0u, 64000))
                 .withFields({C2F(mBitrate, value).inRange(8000, 960000)})
@@ -225,6 +232,7 @@
     int32_t getDrcAttenuationFactor() const { return mDrcAttenuationFactor->value * 127. + 0.5; }
     int32_t getDrcEffectType() const { return mDrcEffectType->value; }
     int32_t getDrcAlbumMode() const { return mDrcAlbumMode->value; }
+    u_int32_t getMaxChannelCount() const { return mMaxChannelCount->value; }
     int32_t getDrcOutputLoudness() const { return (mDrcOutputLoudness->value <= 0 ? -mDrcOutputLoudness->value * 4. + 0.5 : -1); }
 
 private:
@@ -241,6 +249,7 @@
     std::shared_ptr<C2StreamDrcAttenuationFactorTuning::input> mDrcAttenuationFactor;
     std::shared_ptr<C2StreamDrcEffectTypeTuning::input> mDrcEffectType;
     std::shared_ptr<C2StreamDrcAlbumModeTuning::input> mDrcAlbumMode;
+    std::shared_ptr<C2StreamMaxChannelCountInfo::input> mMaxChannelCount;
     std::shared_ptr<C2StreamDrcOutputLoudnessTuning::output> mDrcOutputLoudness;
     // TODO Add : C2StreamAacSbrModeTuning
 };
@@ -366,9 +375,10 @@
     ALOGV("AAC decoder using MPEG-D DRC album mode %d", albumMode);
     aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_ALBUM_MODE, albumMode);
 
-    // By default, the decoder creates a 5.1 channel downmix signal.
-    // For seven and eight channel input streams, enable 6.1 and 7.1 channel output
-    aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, -1);
+    // AAC_PCM_MAX_OUTPUT_CHANNELS
+    u_int32_t maxChannelCount = mIntf->getMaxChannelCount();
+    ALOGV("AAC decoder using maximum output channel count %d", maxChannelCount);
+    aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, maxChannelCount);
 
     return status;
 }
@@ -707,6 +717,11 @@
         ALOGV("AAC decoder using MPEG-D DRC album mode %d", albumMode);
         aacDecoder_SetParam(mAACDecoder, AAC_UNIDRC_ALBUM_MODE, albumMode);
 
+        // AAC_PCM_MAX_OUTPUT_CHANNELS
+        int32_t maxChannelCount = mIntf->getMaxChannelCount();
+        ALOGV("AAC decoder using maximum output channel count %d", maxChannelCount);
+        aacDecoder_SetParam(mAACDecoder, AAC_PCM_MAX_OUTPUT_CHANNELS, maxChannelCount);
+
         mDrcWrap.update();
 
         UINT inBufferUsedLength = inBufferLength[0] - bytesValid[0];
@@ -847,6 +862,51 @@
                     ALOGE("Getting output loudness failed");
                 }
             }
+
+            // update config with values used for decoding:
+            //    Album mode, target reference level, DRC effect type, DRC attenuation and boost
+            //    factor, DRC compression mode, encoder target level and max channel count
+            // with input values as they were not modified by decoder
+
+            C2StreamDrcAttenuationFactorTuning::input currentAttenuationFactor(0u,
+                    (C2FloatValue) (attenuationFactor/127.));
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentAttenuationFactor));
+
+            C2StreamDrcBoostFactorTuning::input currentBoostFactor(0u,
+                    (C2FloatValue) (boostFactor/127.));
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentBoostFactor));
+
+            C2StreamDrcCompressionModeTuning::input currentCompressMode(0u,
+                    (C2Config::drc_compression_mode_t) compressMode);
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentCompressMode));
+
+            C2StreamDrcEncodedTargetLevelTuning::input currentEncodedTargetLevel(0u,
+                    (C2FloatValue) (encTargetLevel*-0.25));
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentEncodedTargetLevel));
+
+            C2StreamDrcAlbumModeTuning::input currentAlbumMode(0u,
+                    (C2Config::drc_album_mode_t) albumMode);
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentAlbumMode));
+
+            C2StreamDrcTargetReferenceLevelTuning::input currentTargetRefLevel(0u,
+                    (float) (targetRefLevel*-0.25));
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentTargetRefLevel));
+
+            C2StreamDrcEffectTypeTuning::input currentEffectype(0u,
+                    (C2Config::drc_effect_type_t) effectType);
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentEffectype));
+
+            C2StreamMaxChannelCountInfo::input currentMaxChannelCnt(0u, maxChannelCount);
+            work->worklets.front()->output.configUpdate.push_back(
+                    C2Param::Copy(currentMaxChannelCnt));
+
         } while (decoderErr == AAC_DEC_OK);
     }
 
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.cpp b/media/codec2/sfplugin/CCodecBufferChannel.cpp
index c07f3dc..0626c8d 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.cpp
+++ b/media/codec2/sfplugin/CCodecBufferChannel.cpp
@@ -618,13 +618,14 @@
 }
 
 void CCodecBufferChannel::feedInputBufferIfAvailableInternal() {
-    if (mInputMetEos ||
-           mOutput.lock()->buffers->hasPending() ||
-           mPipelineWatcher.lock()->pipelineFull()) {
+    if (mInputMetEos || mPipelineWatcher.lock()->pipelineFull()) {
         return;
-    } else {
+    }
+    {
         Mutexed<Output>::Locked output(mOutput);
-        if (!output->buffers || output->buffers->numClientBuffers() >= output->numSlots) {
+        if (!output->buffers ||
+                output->buffers->hasPending() ||
+                output->buffers->numClientBuffers() >= output->numSlots) {
             return;
         }
     }
@@ -731,6 +732,9 @@
     std::shared_ptr<const C2StreamHdr10PlusInfo::output> hdr10PlusInfo =
         std::static_pointer_cast<const C2StreamHdr10PlusInfo::output>(
                 c2Buffer->getInfo(C2StreamHdr10PlusInfo::output::PARAM_TYPE));
+    if (hdr10PlusInfo && hdr10PlusInfo->flexCount() == 0) {
+        hdr10PlusInfo.reset();
+    }
 
     {
         Mutexed<OutputSurface>::Locked output(mOutputSurface);
@@ -782,7 +786,7 @@
                     .maxLuminance = hdrStaticInfo->mastering.maxLuminance,
                     .minLuminance = hdrStaticInfo->mastering.minLuminance,
                 };
-                hdr.validTypes = HdrMetadata::SMPTE2086;
+                hdr.validTypes |= HdrMetadata::SMPTE2086;
                 hdr.smpte2086 = smpte2086_meta;
             }
             // If the content light level fields are 0, do not use them, it
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 4925ea4..3fd3fc3 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -33,10 +33,8 @@
     path: "aidl",
 }
 
-aidl_interface {
-    name: "resourcemanager_aidl_interface",
-    unstable: true,
-    local_include_dir: "aidl",
+filegroup {
+    name: "resourcemanager_aidl",
     srcs: [
         "aidl/android/media/IResourceManagerClient.aidl",
         "aidl/android/media/IResourceManagerService.aidl",
@@ -45,6 +43,16 @@
         "aidl/android/media/MediaResourceParcel.aidl",
         "aidl/android/media/MediaResourcePolicyParcel.aidl",
     ],
+    path: "aidl",
+}
+
+aidl_interface {
+    name: "resourcemanager_aidl_interface",
+    unstable: true,
+    local_include_dir: "aidl",
+    srcs: [
+        ":resourcemanager_aidl",
+    ],
 }
 
 cc_library_shared {
diff --git a/services/audioflinger/AudioStreamOut.cpp b/services/audioflinger/AudioStreamOut.cpp
index d13cb8f..7e06096 100644
--- a/services/audioflinger/AudioStreamOut.cpp
+++ b/services/audioflinger/AudioStreamOut.cpp
@@ -164,6 +164,10 @@
         stream = outStream;
         mHalFormatHasProportionalFrames = audio_has_proportional_frames(config->format);
         status = stream->getFrameSize(&mHalFrameSize);
+        LOG_ALWAYS_FATAL_IF(status != OK, "Error retrieving frame size from HAL: %d", status);
+        LOG_ALWAYS_FATAL_IF(mHalFrameSize <= 0, "Error frame size was %zu but must be greater than"
+                " zero", mHalFrameSize);
+
     }
 
     return status;
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index 024168b..5930117 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -6062,10 +6062,6 @@
     bool trackPaused = false;
     bool trackStopped = false;
 
-    if ((mType == DIRECT) && audio_is_linear_pcm(mFormat) && !usesHwAvSync()) {
-        return !mStandby;
-    }
-
     // do not put the HAL in standby when paused. AwesomePlayer clear the offloaded AudioTrack
     // after a timeout and we will enter standby then.
     if (mTracks.size() > 0) {
@@ -8436,13 +8432,14 @@
     }
     result = mInput->stream->getFrameSize(&mFrameSize);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
+    LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
+            mFrameSize);
     result = mInput->stream->getBufferSize(&mBufferSize);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);
     mFrameCount = mBufferSize / mFrameSize;
-    ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%lld, "
-            "mBufferSize=%lld, mFrameCount=%lld",
-            this, mChannelCount, mFormat, (long long)mFrameSize, (long long)mBufferSize,
-            (long long)mFrameCount);
+    ALOGV("%p RecordThread params: mChannelCount=%u, mFormat=%#x, mFrameSize=%zu, "
+            "mBufferSize=%zu, mFrameCount=%zu",
+            this, mChannelCount, mFormat, mFrameSize, mBufferSize, mFrameCount);
     // This is the formula for calculating the temporary buffer size.
     // With 7 HAL buffers, we can guarantee ability to down-sample the input by ratio of 6:1 to
     // 1 full output buffer, regardless of the alignment of the available input.
@@ -9018,6 +9015,8 @@
     LOG_ALWAYS_FATAL_IF(!audio_is_linear_pcm(mFormat), "HAL format %#x is not linear pcm", mFormat);
     result = mHalStream->getFrameSize(&mFrameSize);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving frame size from HAL: %d", result);
+    LOG_ALWAYS_FATAL_IF(mFrameSize <= 0, "Error frame size was %zu but must be greater than zero",
+            mFrameSize);
     result = mHalStream->getBufferSize(&mBufferSize);
     LOG_ALWAYS_FATAL_IF(result != OK, "Error retrieving buffer size from HAL: %d", result);
     mFrameCount = mBufferSize / mFrameSize;
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index 3730c34..c5c13e9 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -5770,15 +5770,6 @@
     DeviceVector filteredDevices = outputDesc->filterSupportedDevices(devices);
     DeviceVector prevDevices = outputDesc->devices();
 
-    // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
-    // output profile or if new device is not supported AND previous device(s) is(are) still
-    // available (otherwise reset device must be done on the output)
-    if (!devices.isEmpty() && filteredDevices.isEmpty() &&
-            !mAvailableOutputDevices.filter(prevDevices).empty()) {
-        ALOGV("%s: unsupported device %s for output", __func__, devices.toString().c_str());
-        return 0;
-    }
-
     ALOGV("setOutputDevices() prevDevice %s", prevDevices.toString().c_str());
 
     if (!filteredDevices.isEmpty()) {
@@ -5793,6 +5784,17 @@
         muteWaitMs = 0;
     }
 
+    // no need to proceed if new device is not AUDIO_DEVICE_NONE and not supported by current
+    // output profile or if new device is not supported AND previous device(s) is(are) still
+    // available (otherwise reset device must be done on the output)
+    if (!devices.isEmpty() && filteredDevices.isEmpty() &&
+            !mAvailableOutputDevices.filter(prevDevices).empty()) {
+        ALOGV("%s: unsupported device %s for output", __func__, devices.toString().c_str());
+        // restore previous device after evaluating strategy mute state
+        outputDesc->setDevices(prevDevices);
+        return muteWaitMs;
+    }
+
     // Do not change the routing if:
     //      the requested device is AUDIO_DEVICE_NONE
     //      OR the requested device is the same as current device
diff --git a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
index 9577160..34d07b6 100644
--- a/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
+++ b/services/audiopolicy/service/AudioPolicyInterfaceImpl.cpp
@@ -593,7 +593,7 @@
     }
 
     // including successes gets very verbose
-    // but once we cut over to westworld, log them all.
+    // but once we cut over to statsd, log them all.
     if (status != NO_ERROR) {
 
         static constexpr char kAudioPolicy[] = "audiopolicy";
diff --git a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
index 1a0881f..a63f402 100644
--- a/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
+++ b/services/camera/libcameraservice/api2/HeicCompositeStream.cpp
@@ -620,7 +620,8 @@
         if (mPendingInputFrames.find(mAppSegmentFrameNumbers.front()) == mPendingInputFrames.end()) {
             ALOGE("%s: mPendingInputFrames doesn't contain frameNumber %" PRId64, __FUNCTION__,
                     mAppSegmentFrameNumbers.front());
-            mInputYuvBuffers.erase(it);
+            mInputAppSegmentBuffers.erase(it);
+            mAppSegmentFrameNumbers.pop();
             continue;
         }
 
@@ -664,6 +665,7 @@
             ALOGE("%s: mPendingInputFrames doesn't contain frameNumber %" PRId64, __FUNCTION__,
                     mMainImageFrameNumbers.front());
             mInputYuvBuffers.erase(it);
+            mMainImageFrameNumbers.pop();
             continue;
         }
 
diff --git a/services/mediametrics/AudioPowerUsage.cpp b/services/mediametrics/AudioPowerUsage.cpp
index c441110..cca6b41 100644
--- a/services/mediametrics/AudioPowerUsage.cpp
+++ b/services/mediametrics/AudioPowerUsage.cpp
@@ -34,7 +34,7 @@
 #define PROP_AUDIO_METRICS_DISABLED "persist.media.audio_metrics.power_usage_disabled"
 #define AUDIO_METRICS_DISABLED_DEFAULT (false)
 
-// property to set how long to send audio power use metrics data to westworld, default is 24hrs
+// property to set how long to send audio power use metrics data to statsd, default is 24hrs
 #define PROP_AUDIO_METRICS_INTERVAL_HR "persist.media.audio_metrics.interval_hr"
 #define INTERVAL_HR_DEFAULT (24)
 
diff --git a/services/mediametrics/MediaMetricsService.cpp b/services/mediametrics/MediaMetricsService.cpp
index d682fed..48e766e 100644
--- a/services/mediametrics/MediaMetricsService.cpp
+++ b/services/mediametrics/MediaMetricsService.cpp
@@ -185,7 +185,7 @@
     }
 
     if (!isTrusted || item->getTimestamp() == 0) {
-        // Westworld logs two times for events: ElapsedRealTimeNs (BOOTTIME) and
+        // Statsd logs two times for events: ElapsedRealTimeNs (BOOTTIME) and
         // WallClockTimeNs (REALTIME), but currently logs REALTIME to cloud.
         //
         // For consistency and correlation with other logging mechanisms
diff --git a/services/mediametrics/MediaMetricsService.h b/services/mediametrics/MediaMetricsService.h
index d152264..792b7f0 100644
--- a/services/mediametrics/MediaMetricsService.h
+++ b/services/mediametrics/MediaMetricsService.h
@@ -65,7 +65,7 @@
     static nsecs_t roundTime(nsecs_t timeNs);
 
     /**
-     * Returns true if we should use uid for package name when uploading to WestWorld.
+     * Returns true if we should use uid for package name when uploading to statsd.
      */
     static bool useUidForPackage(const std::string& package, const std::string& installer);