fix: disable WebRTC burst and add signal level monitoring to player

Author: Datanoise

relay/transcode.go

@@ -130,7 +130,7 @@ func (tm *TranscoderManager) performTranscode(ctx context.Context, inst *Transco
 	// 2. Decode (assuming MP3 input for now as standard)
 	decoder, err := mp3.NewDecoder(reader)
 	if err != nil {
-		logrus.WithError(err).Error("Failed to initialize decoder")
+		logrus.WithError(err).Errorf("Transcoder %s: Failed to initialize decoder for input %s", inst.Config.Name, inst.Config.InputMount)
 		return
 	}
 

relay/webrtc.go

@@ -99,14 +99,15 @@ func (wm *WebRTCManager) streamToTrack(pc *webrtc.PeerConnection, track *webrtc.
 	})
 
 	id := fmt.Sprintf("webrtc-%d", time.Now().UnixNano())
-	offset, signal := stream.Subscribe(id, 64*1024)
+	// For WebRTC we start at the current head (no burst)
+	offset, signal := stream.Subscribe(id, 0)
 	defer stream.Unsubscribe(id)
 
 	// Seek to next "OggS" magic
 	syncBuf := make([]byte, 16384)
 	foundSync := false
 	var searched int64 = 0
-	for !foundSync && searched < 512*1024 { // Safeguard: stop after 512KB
+	for !foundSync && searched < 512*1024 {
 		select {
 		case <-ctx.Done():
 			return
@@ -148,6 +149,8 @@ func (wm *WebRTCManager) streamToTrack(pc *webrtc.PeerConnection, track *webrtc.
 		return
 	}
 
+	logrus.Infof("WebRTC: Beginning packet transmission for %s", stream.MountName)
+	sentCount := 0
 	for {
 		packet, err := opusReader.ReadAudioPacket()
 		if err != nil {
@@ -163,5 +166,9 @@ func (wm *WebRTCManager) streamToTrack(pc *webrtc.PeerConnection, track *webrtc.
 		}); err != nil {
 			return
 		}
+		sentCount++
+		if sentCount%100 == 0 {
+			logrus.Debugf("WebRTC: Sent 100 packets to %s", stream.MountName)
+		}
 	}
 }

server/server.go

@@ -1824,7 +1824,9 @@ func (s *Server) handleWebRTCOffer(w http.ResponseWriter, r *http.Request) {
 
 	answer, err := s.WebRTCM.HandleOffer(mount, offer)
 	if err != nil {
-		http.Error(w, err.Error(), http.StatusInternalServerError)
+		w.Header().Set("Content-Type", "application/json")
+		w.WriteHeader(http.StatusBadRequest)
+		json.NewEncoder(w).Encode(map[string]string{"error": err.Error()})
 		return
 	}
 

server/templates/webrtc_player.html

@@ -150,7 +150,8 @@ <h1>{{.Stream.Name}}</h1>
             });
 
             if (!response.ok) {
-                const err = await response.text();
+                const data = await response.json();
+                const err = data.error || "Unknown signaling error";
                 stateEl.textContent = "Error: " + err;
                 log('ERROR', `Signaling failed: ${err}`);
                 btn.disabled = false;
@@ -174,17 +175,30 @@ <h1>{{.Stream.Name}}</h1>
             const bufferLength = analyzer.frequencyBinCount;
             const dataArray = new Uint8Array(bufferLength);
 
+            let lastLevelReport = 0;
+
             function draw() {
                 requestAnimationFrame(draw);
                 analyzer.getByteFrequencyData(dataArray);
                 cctx.clearRect(0, 0, canvas.width, canvas.height);
+                
+                let sum = 0;
                 const barWidth = (canvas.width / bufferLength) * 2.5;
                 let x = 0;
                 for (let i = 0; i < bufferLength; i++) {
                     const barHeight = dataArray[i] / 2;
                     cctx.fillStyle = '#38bdf8';
                     cctx.fillRect(x, canvas.height - barHeight, barWidth, barHeight);
                     x += barWidth + 1;
+                    sum += dataArray[i];
+                }
+
+                // Report signal level every 2 seconds
+                const avg = sum / bufferLength;
+                const now = Date.now();
+                if (now - lastLevelReport > 2000) {
+                    log('AUDIO', `Signal Level: ${Math.round(avg)}`);
+                    lastLevelReport = now;
                 }
             }
             draw();