teleportxr 1.0.86 → 1.0.88

package/client/geometry_service.js

@@ -126,7 +126,7 @@ class GeometryService {
 		// MAY not be in this set:
 		this.streamedNodes.delete(uid);
 		// TODO: now reduce the counts for all the dependent resources.
-		console.log("Unstreaming node ", node_uid," for client ", this.clientID);
+		console.log("Unstreaming node ", uid," for client ", this.clientID);
 	}
 	StreamOrUnstream(resourceMap, uid, diff) {
 		// exclude "undefined"

package/connections/webrtcconnection.js

@@ -431,7 +431,6 @@ class WebRtcConnection extends EventEmitter
 
         this.videoDataChannel = this.createDataChannel("video",20);
         this.tagDataChannel = this.createDataChannel("video_tags",40);
-        this.audioToClientDataChannel = this.createDataChannel("audio_server_to_client",60);
         this.geometryDataChannel = this.createDataChannel("geometry_unframed",80);
         this.reliableDataChannel = this.createDataChannel("reliable",100);
         this.unreliableDataChannel = this.createDataChannel("unreliable",120,false);
@@ -539,12 +538,34 @@ class WebRtcConnection extends EventEmitter
             return;
         }
         if (this._sceneAudioInterval) {
-            this.options.clearTimeout(this._sceneAudioInterval); // not a timeout, but in case
+            clearInterval(this._sceneAudioInterval);
+            this._sceneAudioInterval = null;
         }
         // Drive a 10 ms tick. setInterval is sufficient: RTCAudioSource buffers
         // internally, and a few ms of jitter is absorbed by the WebRTC jitter buffer.
+        this._sceneAudioTickCount = 0;
         this._sceneAudioInterval = setInterval(() => {
-            if (this._sceneAudioStreamer) this._sceneAudioStreamer.tick();
+            if (!this._sceneAudioStreamer) return;
+            this._sceneAudioStreamer.tick();
+            this._sceneAudioTickCount++;
+            // Every ~5 s, log tick count + outbound RTP stats so we can confirm
+            // libwebrtc is actually transmitting audio packets.
+            if ((this._sceneAudioTickCount % 500) === 0) {
+                const n = this._sceneAudioTickCount;
+                if (typeof this.peerConnection.getStats === 'function') {
+                    Promise.resolve(this.peerConnection.getStats()).then((report) => {
+                        let summary = 'no outbound-rtp/audio stats';
+                        try {
+                            const entries = (typeof report.values === 'function') ? Array.from(report.values()) : Object.values(report);
+                            const outRtp = entries.find((s) => s && s.type === 'outbound-rtp' && (s.kind === 'audio' || s.mediaType === 'audio'));
+                            if (outRtp) summary = 'packetsSent='+outRtp.packetsSent+' bytesSent='+outRtp.bytesSent;
+                        } catch (e) { summary = 'stats parse error: '+e.message; }
+                        console.log('SceneAudioStreamer tick #'+n+' — '+summary);
+                    }).catch((e) => { console.log('SceneAudioStreamer tick #'+n+' — getStats failed: '+e.message); });
+                } else {
+                    console.log('SceneAudioStreamer tick #'+n+' — getStats unavailable');
+                }
+            }
         }, sound.FRAME_MS);
         console.log('startSceneAudio: streaming '+loaded+' source(s) at '+sound.SAMPLE_RATE+' Hz mono');
     }
@@ -563,7 +584,6 @@ class WebRtcConnection extends EventEmitter
 	{
         //.videoDataChannel = .("video",20);
         //.tagDataChannel = .("video_tags",40);
-        //.audioToClientDataChannel = .("audio_server_to_client",60);
         //.geometryDataChannel =l("geometry_unframed",80);
         //.reliableDataChannel =l("reliable",100);
         //.unreliableDataChannelnel("unreliable",120,false);

package/package.json

@@ -16,7 +16,7 @@
   },
   "name": "teleportxr",
   "description": "Teleport Spatial Server on node.js",
-  "version": "1.0.86",
+  "version": "1.0.88",
   "repository": {
     "type": "git",
     "url": "git+https://github.com/simul/teleport-nodejs.git"