add send jitter metric

Author: vpalmisano

scripts/get-user-media.js

@@ -154,7 +154,7 @@ if (navigator.getUserMedia) {
 if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {
   const nativeGetUserMedia = navigator.mediaDevices.getUserMedia.bind(navigator.mediaDevices)
   navigator.mediaDevices.getUserMedia = async function (constraints, ...args) {
-    log(`getUserMedia:`, constraints)
+    log(`getUserMedia:`, JSON.stringify(constraints, null, 2))
     try {
       overrideGetUserMedia(constraints)
     } catch (err) {
@@ -192,7 +192,7 @@ if (navigator.mediaDevices && navigator.mediaDevices.getUserMedia) {
 if (navigator.mediaDevices && navigator.mediaDevices.getDisplayMedia) {
   const nativeGetDisplayMedia = navigator.mediaDevices.getDisplayMedia.bind(navigator.mediaDevices)
   navigator.mediaDevices.getDisplayMedia = async function (constraints, ...args) {
-    log(`getDisplayMedia:`, constraints)
+    log(`getDisplayMedia:`, JSON.stringify(constraints, null, 2))
     let stopFakeScreenshare = null
     if (window.PARAMS?.fakeScreenshare) {
       stopFakeScreenshare = await webrtcperf.setupFakeScreenshare(window.PARAMS?.fakeScreenshare)

scripts/peer-connection-stats.js

@@ -127,6 +127,7 @@ async function getPeerConnectionStats(id, pc, now, raw = false, verbose = false)
               s.packetsLost = remoteInboundRtpStreamStats.packetsLost
               s.totalRoundTripTime = remoteInboundRtpStreamStats.totalRoundTripTime
               s.roundTripTimeMeasurements = remoteInboundRtpStreamStats.roundTripTimeMeasurements
+              s.jitter = remoteInboundRtpStreamStats.jitter
             }
             const {
               kind,
@@ -146,6 +147,7 @@ async function getPeerConnectionStats(id, pc, now, raw = false, verbose = false)
               nackCount,
               totalRoundTripTime,
               roundTripTimeMeasurements,
+              jitter,
               totalEncodeTime,
               totalPacketSendDelay,
             } = s
@@ -165,6 +167,7 @@ async function getPeerConnectionStats(id, pc, now, raw = false, verbose = false)
               pliCountReceived: pliCount,
               totalRoundTripTime,
               roundTripTimeMeasurements,
+              jitter,
               totalEncodeTime,
               totalPacketSendDelay,
               qualityLimitationResolutionChanges,
@@ -201,6 +204,7 @@ async function getPeerConnectionStats(id, pc, now, raw = false, verbose = false)
               'pliCountReceived',
               'totalRoundTripTime',
               'roundTripTimeMeasurements',
+              'jitter',
               'totalEncodeTime',
               'totalPacketSendDelay',
             ].forEach(prop => maxOptional(values.outboundRtp, outboundRtp, prop))

scripts/screenshare.js

@@ -39,7 +39,7 @@ webrtcperf.setupFakeScreenshare = ({
   wrapper.setAttribute('id', 'webrtcperf-fake-screenshare')
   wrapper.setAttribute(
     'style',
-    `position: fixed; top: 0; left: 0; width: ${width}px; height: ${height}px; z-index: -1; background-color: black; isolation: isolate; transform-style: flat;`,
+    `all: unset; position: fixed; top: 0; left: 0; width: ${width}px; height: ${height}px; z-index: -1; background-color: black; isolation: isolate; transform-style: flat;`,
   )
   document.body.appendChild(wrapper)
   window.GET_DISPLAY_MEDIA_CROP = '#webrtcperf-fake-screenshare'
@@ -60,7 +60,7 @@ webrtcperf.setupFakeScreenshare = ({
       img.setAttribute('src', `https://picsum.photos/seed/${i + 1}/${width}/${height}`)
       img.setAttribute(
         'style',
-        `position: absolute; width: ${width}px; height: ${height}px; transform: translateX(100%); opacity: 0;`,
+        `all: unset; position: absolute; width: ${width}px; height: ${height}px; transform: translateX(100%); opacity: 0;`,
       )
       wrapper.appendChild(img)
       slidesElements.push(img)

src/rtcstats.ts

@@ -110,6 +110,8 @@ export enum RtcStatsMetricNames {
   audioSentNackCountRecv = 'audioSentNackCountRecv',
   /** The sent audio round trip time. */
   audioSentRoundTripTime = 'audioSentRoundTripTime',
+  /** The sent audio jitter. */
+  audioSentJitter = 'audioSentJitter',
   /** The audio RTC transport round trip time. */
   audioSentTransportRoundTripTime = 'audioSentTransportRoundTripTime',
   /** The sent audio encoding max bitrate. */
@@ -161,6 +163,8 @@ export enum RtcStatsMetricNames {
   videoSentRoundTripTime = 'videoSentRoundTripTime',
   /** The transport send video round trip time. */
   videoSentTransportRoundTripTime = 'videoSentTransportRoundTripTime',
+  /** The video sent jitter. */
+  videoSentJitter = 'videoSentJitter',
   /** The sent video RTX packets. */
   videoSentRetransmittedPackets = 'videoSentRetransmittedPackets',
 
@@ -208,6 +212,8 @@ export enum RtcStatsMetricNames {
   screenSentRoundTripTime = 'screenSentRoundTripTime',
   /** The transport sent screen round trip time. */
   screenSentTransportRoundTripTime = 'screenSentTransportRoundTripTime',
+  /** The screen sent jitter. */
+  screenSentJitter = 'screenSentJitter',
   /** The screen audio RTX packets. */
   screenSentRetransmittedPackets = 'screenSentRetransmittedPackets',
 
@@ -439,6 +445,7 @@ export function updateRtcStats(
       setStats(stats, (prefix + 'SentNackCountRecv') as RtcStatsMetricNames, key, outboundRtp.nackCount)
       //setStats(stats, prefix + 'SentPacketsLostCount', key, outboundRtp.packetsLost)
       setStats(stats, (prefix + 'SentRoundTripTime') as RtcStatsMetricNames, key, outboundRtp.roundTripTime)
+      setStats(stats, (prefix + 'SentJitter') as RtcStatsMetricNames, key, outboundRtp.jitter)
       setStats(
         stats,
         (prefix + 'SentTransportRoundTripTime') as RtcStatsMetricNames,