[WIP] Fix: Audio Latency issues: move audio to a dedicated media stream

For more details please see: https://groups.google.com/g/discuss-webrtc/c/ZvAHvkHsb0E

internal/audio/adaptive_buffer.go

@@ -294,7 +294,7 @@ func (abm *AdaptiveBufferManager) ActivateGracefulDegradation(level int) {
 		atomic.StoreInt64(&abm.currentInputBufferSize, minSize)
 		atomic.StoreInt64(&abm.currentOutputBufferSize, minSize)
 
-		abm.logger.Error().
+		abm.logger.Warn().
 			Int("level", level).
 			Int64("buffer_size", minSize).
 			Msg("Activated severe graceful degradation - emergency mode")

internal/audio/core_config_constants.go

@@ -530,7 +530,7 @@ func DefaultAudioConfig() *AudioConfigConstants {
 		// Graceful Degradation Configuration
 		CongestionMildReductionFactor:     0.75,            // Buffer reduction factor for mild congestion (0.75)
 		CongestionModerateReductionFactor: 0.5,             // Buffer reduction factor for moderate congestion (0.5)
-		CongestionThresholdMultiplier:     10.0,            // Multiplier for congestion threshold calculations (increased to reduce false emergency mode triggers)
+		CongestionThresholdMultiplier:     36.0,            // Multiplier for congestion threshold calculations (increased to reduce false emergency mode triggers)
 		CongestionRecoveryTimeout:         5 * time.Second, // Timeout for congestion recovery (5 seconds)
 
 		// Buffer Pool Cache Configuration

ui/src/components/WebRTCVideo.tsx

@@ -345,8 +345,13 @@ export default function WebRTCVideo({ microphone }: WebRTCVideoProps) {
 
       peerConnection.addEventListener(
         "track",
-        (e: RTCTrackEvent) => {
-          addStreamToVideoElm(e.streams[0]);
+        (_e: RTCTrackEvent) => {
+          // The combined MediaStream is now managed in the main component
+          // We'll use the mediaStream from the store instead of individual track streams
+          const { mediaStream } = useRTCStore.getState();
+          if (mediaStream) {
+            addStreamToVideoElm(mediaStream);
+          }
         },
         { signal },
       );

ui/src/routes/devices.$id.tsx

@@ -475,8 +475,27 @@ export default function KvmIdRoute() {
       }
     };
 
-    pc.ontrack = function (event) {
-      setMediaStream(event.streams[0]);
+    pc.ontrack = function (event: RTCTrackEvent) {
+      // Handle separate MediaStreams for audio and video tracks
+      const track = event.track;
+      const streams = event.streams;
+      
+      if (streams && streams.length > 0) {
+        // Get existing MediaStream or create a new one
+        const existingStream = useRTCStore.getState().mediaStream;
+        let combinedStream: MediaStream;
+        
+        if (existingStream) {
+          combinedStream = existingStream;
+          // Add the new track to the existing stream
+          combinedStream.addTrack(track);
+        } else {
+          // Create a new MediaStream with the track
+          combinedStream = new MediaStream([track]);
+        }
+        
+        setMediaStream(combinedStream);
+      }
     };
 
     setTransceiver(pc.addTransceiver("video", { direction: "recvonly" }));

webrtc.go

@@ -233,13 +233,13 @@ func newSession(config SessionConfig) (*Session, error) {
 		}
 	})
 
-	session.VideoTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264}, "video", "kvm")
+	session.VideoTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeH264}, "video", "kvm-video")
 	if err != nil {
 		scopedLogger.Warn().Err(err).Msg("Failed to create VideoTrack")
 		return nil, err
 	}
 
-	session.AudioTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "kvm")
+	session.AudioTrack, err = webrtc.NewTrackLocalStaticSample(webrtc.RTPCodecCapability{MimeType: webrtc.MimeTypeOpus}, "audio", "kvm-audio")
 	if err != nil {
 		scopedLogger.Warn().Err(err).Msg("Failed to add VideoTrack to PeerConnection")
 		return nil, err