Final edit

selkies-project · May 20, 2024 · 10fefe6 · 10fefe6
1 parent f35de4c
commit 10fefe6
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Showing 2 changed files with 9 additions and 9 deletions.
diff --git a/addons/gst-web/src/app.js b/addons/gst-web/src/app.js
@@ -434,7 +434,7 @@ function onBothStreamConnected() {
     audio_webrtc.peerConnection.getReceivers().forEach(receiver => {
         receiver.jitterBufferTarget = receiver.jitterBufferDelayHint = receiver.playoutDelayHint = 0.0;
     });
-    // Start watching stats.
+    // Start watching stats
     var videoBytesReceivedStart = 0;
     var audioBytesReceivedStart = 0;
     var statsStart = new Date().getTime() / 1000;
@@ -446,13 +446,13 @@ function onBothStreamConnected() {
                 if (videoConnected !== "connected" || audioConnected !== "connected") return;
                 var now = new Date().getTime() / 1000;
 
-                // Connection latency in ms.
+                // Connection latency in milliseconds
                 app.connectionLatency = 0;
                 app.connectionVideoLatency = (stats.general.currentRoundTripTime !== null) ? (stats.general.currentRoundTripTime * 1000.0) : (app.serverLatency * 2.0);
                 app.connectionAudioLatency = (audioStats.general.currentRoundTripTime !== null) ? (audioStats.general.currentRoundTripTime * 1000.0) : (app.serverLatency * 2.0);
                 app.connectionLatency += Math.max(app.connectionVideoLatency, app.connectionAudioLatency);
 
-                // Sum of video+audio packets.
+                // Sum of video+audio packets
                 app.connectionPacketsReceived = 0;
                 app.connectionPacketsLost = 0;
 
@@ -462,7 +462,7 @@ function onBothStreamConnected() {
                 app.connectionBytesSent = ((stats.general.bytesSent + audioStats.general.bytesSent) * 1e-6).toFixed(2) + " MBytes";
                 app.connectionAvailableBandwidth = ((parseInt(stats.general.availableReceiveBandwidth) + parseInt(audioStats.general.availableReceiveBandwidth)) / 1e+6).toFixed(2) + " mbps";
 
-                // Video stats.
+                // Video stats
                 app.connectionPacketsReceived += stats.video.packetsReceived;
                 app.connectionPacketsLost += stats.video.packetsLost;
                 app.connectionCodec = stats.video.codecName;
@@ -472,14 +472,14 @@ function onBothStreamConnected() {
                 app.connectionVideoBitrate = (((stats.video.bytesReceived - videoBytesReceivedStart) / (now - statsStart)) * 8 / 1e+6).toFixed(2);
                 videoBytesReceivedStart = stats.video.bytesReceived;
 
-                // Audio stats.
+                // Audio stats
                 app.connectionPacketsReceived += audioStats.audio.packetsReceived;
                 app.connectionPacketsLost += audioStats.audio.packetsLost;
                 app.connectionAudioCodecName = audioStats.audio.codecName;
                 app.connectionAudioBitrate = (((audioStats.audio.bytesReceived - audioBytesReceivedStart) / (now - statsStart)) * 8 / 1e+3).toFixed(2);
                 audioBytesReceivedStart = audioStats.audio.bytesReceived;
 
-                // Latency stats.
+                // Latency stats
                 app.connectionVideoLatency = parseInt(Math.round(app.connectionVideoLatency + (1000.0 * (stats.video.jitterBufferDelay - stats.video.previousJitterBufferDelay) / (stats.video.jitterBufferEmittedCount - stats.video.previousJitterBufferEmittedCount) || 0)));
                 app.connectionAudioLatency = parseInt(Math.round(app.connectionAudioLatency + (1000.0 * (audioStats.audio.jitterBufferDelay - audioStats.audio.previousJitterBufferDelay) / (audioStats.audio.jitterBufferEmittedCount - audioStats.audio.previousJitterBufferEmittedCount) || 0)));
 
@@ -491,7 +491,7 @@ function onBothStreamConnected() {
                 webrtc.sendDataChannelMessage("_stats_video," + JSON.stringify(stats.allReports));
                 webrtc.sendDataChannelMessage("_stats_audio," + JSON.stringify(audioStats.allReports));
 
-                // Stats refresh loop.
+                // Stats refresh loop
                 setTimeout(statsLoop, 1000);
             });
         });

diff --git a/addons/gst-web/src/webrtc.js b/addons/gst-web/src/webrtc.js
@@ -543,7 +543,7 @@ class WebRTCDemo {
         return new Promise(function (resolve, reject) {
             // Statistics API:
             // https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_Statistics_API
-            pc.getStats().then( (stats) => {
+            pc.getStats().then((stats) => {
                 var reports = {
                     transports: {},
                     candidatePairs: {},
@@ -634,7 +634,7 @@ class WebRTCDemo {
                     connectionDetails.data.messagesSent =  dataChannel.messagesSent;
                 }
 
-                // Extract transport stats (RTCTransportStats.selectedCandidatePairId or RTCIceCandidatePairStats.selected).
+                // Extract transport stats (RTCTransportStats.selectedCandidatePairId or RTCIceCandidatePairStats.selected)
                 if (Object.keys(reports.transports).length > 0) {
                     var transport = reports.transports[Object.keys(reports.transports)[0]];
                     connectionDetails.general.bytesReceived = transport.bytesReceived;