diff --git a/services/audioflinger/PatchPanel.cpp b/services/audioflinger/PatchPanel.cpp
index 45dd258..b54b41f 100644
--- a/services/audioflinger/PatchPanel.cpp
+++ b/services/audioflinger/PatchPanel.cpp
@@ -313,12 +313,19 @@
                         patch->sources[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
                         patch->sources[0].flags.input : AUDIO_INPUT_FLAG_NONE;
                 audio_io_handle_t input = AUDIO_IO_HANDLE_NONE;
+                audio_source_t source = AUDIO_SOURCE_MIC;
+                // For telephony patches, propagate voice communication use case to record side
+                if (patch->num_sources == 2
+                        && patch->sources[1].ext.mix.usecase.stream
+                                == AUDIO_STREAM_VOICE_CALL) {
+                    source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+                }
                 sp<ThreadBase> thread = mAudioFlinger.openInput_l(srcModule,
                                                                     &input,
                                                                     &config,
                                                                     device,
                                                                     address,
-                                                                    AUDIO_SOURCE_MIC,
+                                                                    source,
                                                                     flags,
                                                                     outputDevice,
                                                                     outputDeviceAddress);
@@ -516,9 +523,14 @@
     audio_output_flags_t outputFlags = mAudioPatch.sinks[0].config_mask & AUDIO_PORT_CONFIG_FLAGS ?
             mAudioPatch.sinks[0].flags.output : AUDIO_OUTPUT_FLAG_NONE;
     audio_stream_type_t streamType = AUDIO_STREAM_PATCH;
+    audio_source_t source = AUDIO_SOURCE_DEFAULT;
     if (mAudioPatch.num_sources == 2 && mAudioPatch.sources[1].type == AUDIO_PORT_TYPE_MIX) {
         // "reuse one existing output mix" case
         streamType = mAudioPatch.sources[1].ext.mix.usecase.stream;
+        // For telephony patches, propagate voice communication use case to record side
+        if (streamType == AUDIO_STREAM_VOICE_CALL) {
+            source = AUDIO_SOURCE_VOICE_COMMUNICATION;
+        }
     }
     if (mPlayback.thread()->hasFastMixer()) {
         // Create a fast track if the playback thread has fast mixer to get better performance.
@@ -546,7 +558,8 @@
                                                  inChannelMask,
                                                  format,
                                                  frameCount,
-                                                 inputFlags);
+                                                 inputFlags,
+                                                 source);
     } else {
         // use a pseudo LCM between input and output framecount
         int playbackShift = __builtin_ctz(playbackFrameCount);
@@ -566,7 +579,9 @@
                                                  frameCount,
                                                  nullptr,
                                                  (size_t)0 /* bufferSize */,
-                                                 inputFlags);
+                                                 inputFlags,
+                                                 {} /* timeout */,
+                                                 source);
     }
     status = mRecord.checkTrack(tempRecordTrack.get());
     if (status != NO_ERROR) {
diff --git a/services/audioflinger/RecordTracks.h b/services/audioflinger/RecordTracks.h
index e8552c4..daec57e 100644
--- a/services/audioflinger/RecordTracks.h
+++ b/services/audioflinger/RecordTracks.h
@@ -87,6 +87,10 @@
                                     && (flags & AUDIO_INPUT_FLAG_HW_AV_SYNC) == 0;
                         }
 
+            using SinkMetadatas = std::vector<record_track_metadata_v7_t>;
+            using MetadataInserter = std::back_insert_iterator<SinkMetadatas>;
+            virtual void    copyMetadataTo(MetadataInserter& backInserter) const;
+
 private:
     friend class AudioFlinger;  // for mState
 
@@ -134,7 +138,8 @@
                 void *buffer,
                 size_t bufferSize,
                 audio_input_flags_t flags,
-                const Timeout& timeout = {});
+                const Timeout& timeout = {},
+                audio_source_t source = AUDIO_SOURCE_DEFAULT);
     virtual             ~PatchRecord();
 
     virtual Source* getSource() { return nullptr; }
@@ -166,7 +171,8 @@
                         audio_channel_mask_t channelMask,
                         audio_format_t format,
                         size_t frameCount,
-                        audio_input_flags_t flags);
+                        audio_input_flags_t flags,
+                        audio_source_t source = AUDIO_SOURCE_DEFAULT);
 
     Source* getSource() override { return static_cast<Source*>(this); }
 
diff --git a/services/audioflinger/Threads.cpp b/services/audioflinger/Threads.cpp
index e26c9d1..4ac24a9 100644
--- a/services/audioflinger/Threads.cpp
+++ b/services/audioflinger/Threads.cpp
@@ -8775,21 +8775,9 @@
         return; // nothing to do
     }
     StreamInHalInterface::SinkMetadata metadata;
+    auto backInserter = std::back_inserter(metadata.tracks);
     for (const sp<RecordTrack> &track : mActiveTracks) {
-        // Do not forward PatchRecord metadata to audio HAL
-        if (track->isPatchTrack()) {
-            continue;
-        }
-        // No track is invalid as this is called after prepareTrack_l in the same critical section
-        record_track_metadata_v7_t trackMetadata;
-        trackMetadata.base = {
-                .source = track->attributes().source,
-                .gain = 1, // capture tracks do not have volumes
-        };
-        trackMetadata.channel_mask = track->channelMask(),
-        strncpy(trackMetadata.tags, track->attributes().tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
-
-        metadata.tracks.push_back(trackMetadata);
+        track->copyMetadataTo(backInserter);
     }
     mInput->stream->updateSinkMetadata(metadata);
 }
diff --git a/services/audioflinger/Tracks.cpp b/services/audioflinger/Tracks.cpp
index 95ca855..0a1bc81 100644
--- a/services/audioflinger/Tracks.cpp
+++ b/services/audioflinger/Tracks.cpp
@@ -1480,7 +1480,7 @@
         }
     }
 
-    metadata.channel_mask = mChannelMask,
+    metadata.channel_mask = mChannelMask;
     strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
     *backInserter++ = metadata;
 }
@@ -2024,7 +2024,6 @@
 {
     Buffer *pInBuffer;
     Buffer inBuffer;
-    bool outputBufferFull = false;
     inBuffer.frameCount = frames;
     inBuffer.raw = data;
 
@@ -2054,7 +2053,6 @@
                 ALOGV("%s(%d): thread %d no more output buffers; status %d",
                         __func__, mId,
                         (int)mThreadIoHandle, status);
-                outputBufferFull = true;
                 break;
             }
             uint32_t waitTimeMs = (uint32_t)ns2ms(systemTime() - startTime);
@@ -2750,6 +2748,25 @@
     }
 }
 
+void AudioFlinger::RecordThread::RecordTrack::copyMetadataTo(MetadataInserter& backInserter) const
+{
+
+    // Do not forward PatchRecord metadata with unspecified audio source
+    if (mAttr.source == AUDIO_SOURCE_DEFAULT) {
+        return;
+    }
+
+    // No track is invalid as this is called after prepareTrack_l in the same critical section
+    record_track_metadata_v7_t metadata;
+    metadata.base = {
+            .source = mAttr.source,
+            .gain = 1, // capture tracks do not have volumes
+    };
+    metadata.channel_mask = mChannelMask;
+    strncpy(metadata.tags, mAttr.tags, AUDIO_ATTRIBUTES_TAGS_MAX_SIZE);
+
+    *backInserter++ = metadata;
+}
 
 // ----------------------------------------------------------------------------
 #undef LOG_TAG
@@ -2763,9 +2780,10 @@
                                                      void *buffer,
                                                      size_t bufferSize,
                                                      audio_input_flags_t flags,
-                                                     const Timeout& timeout)
+                                                     const Timeout& timeout,
+                                                     audio_source_t source)
     :   RecordTrack(recordThread, NULL,
-                audio_attributes_t{} /* currently unused for patch track */,
+                audio_attributes_t{ .source = source } ,
                 sampleRate, format, channelMask, frameCount,
                 buffer, bufferSize, AUDIO_SESSION_NONE, getpid(),
                 audioServerAttributionSource(getpid()), flags, TYPE_PATCH),
@@ -2876,9 +2894,10 @@
         audio_channel_mask_t channelMask,
         audio_format_t format,
         size_t frameCount,
-        audio_input_flags_t flags)
+        audio_input_flags_t flags,
+        audio_source_t source)
         : PatchRecord(recordThread, sampleRate, channelMask, format, frameCount,
-                nullptr /*buffer*/, 0 /*bufferSize*/, flags),
+                nullptr /*buffer*/, 0 /*bufferSize*/, flags, {} /* timeout */, source),
           mPatchRecordAudioBufferProvider(*this),
           mSinkBuffer(allocAligned(32, mFrameCount * mFrameSize)),
           mStubBuffer(allocAligned(32, mFrameCount * mFrameSize))
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
index f9415fe..49a6159 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.cpp
@@ -788,7 +788,8 @@
     ALOGV("%s between source %s and sink %s", __func__,
             srcDevice->toString().c_str(), sinkDevice->toString().c_str());
     auto callTxSourceClientPortId = PolicyAudioPort::getNextUniqueId();
-    const audio_attributes_t aa = { .source = AUDIO_SOURCE_VOICE_COMMUNICATION };
+    const auto aa = mEngine->getAttributesForStreamType(AUDIO_STREAM_VOICE_CALL);
+
     struct audio_port_config source = {};
     srcDevice->toAudioPortConfig(&source);
     mCallTxSourceClient = new InternalSourceClientDescriptor(
@@ -4534,7 +4535,7 @@
                 // In case of Hw bridge, it is a Work Around. The mixPort used is the one declared
                 // in config XML to reach the sink so that is can be declared as available.
                 audio_io_handle_t output = AUDIO_IO_HANDLE_NONE;
-                sp<SwAudioOutputDescriptor> outputDesc = nullptr;
+                sp<SwAudioOutputDescriptor> outputDesc;
                 if (!sourceDesc->isInternal()) {
                     // take care of dynamic routing for SwOutput selection,
                     audio_attributes_t attributes = sourceDesc->attributes();
@@ -4604,7 +4605,8 @@
                         audio_port_config srcMixPortConfig = {};
                         outputDesc->toAudioPortConfig(&srcMixPortConfig, nullptr);
                         // for volume control, we may need a valid stream
-                        srcMixPortConfig.ext.mix.usecase.stream = !sourceDesc->isInternal() ?
+                        srcMixPortConfig.ext.mix.usecase.stream =
+                            (!sourceDesc->isInternal() || isCallTxAudioSource(sourceDesc)) ?
                                     mEngine->getStreamTypeForAttributes(sourceDesc->attributes()) :
                                     AUDIO_STREAM_PATCH;
                         patchBuilder.addSource(srcMixPortConfig);
diff --git a/services/audiopolicy/managerdefault/AudioPolicyManager.h b/services/audiopolicy/managerdefault/AudioPolicyManager.h
index 87e6974..2c1db79 100644
--- a/services/audiopolicy/managerdefault/AudioPolicyManager.h
+++ b/services/audiopolicy/managerdefault/AudioPolicyManager.h
@@ -639,6 +639,10 @@
             return mCallRxSourceClient != nullptr && source == mCallRxSourceClient;
         }
 
+        bool isCallTxAudioSource(const sp<SourceClientDescriptor> &source) {
+            return mCallTxSourceClient != nullptr && source == mCallTxSourceClient;
+        }
+
         void connectTelephonyRxAudioSource();
 
         void disconnectTelephonyAudioSource(sp<SourceClientDescriptor> &clientDesc);