feat(stats): Get audio levels for the top 5 speakers only.

Capture the audio levels only for the top 5 speakers as RTCRtpReceiver#getSynchronizationSources can be expensive when we have too many audio receivers in the call.
Also, capture the audio levels for track that are unmuted if RTCRtpReceiver#getSynchronizationSources is not supported.
Switch Safari to using getStats since its reporting errorneous values, i.e., 0.000001 as audio level for all remote audio tracks.

JitsiConferenceEventManager.js

@@ -5,6 +5,7 @@ import { Strophe } from 'strophe.js';
 
 import * as JitsiConferenceErrors from './JitsiConferenceErrors';
 import * as JitsiConferenceEvents from './JitsiConferenceEvents';
+import { SPEAKERS_AUDIO_LEVELS } from './modules/statistics/constants';
 import Statistics from './modules/statistics/statistics';
 import EventEmitterForwarder from './modules/util/EventEmitterForwarder';
 import * as MediaType from './service/RTC/MediaType';
@@ -505,6 +506,20 @@ JitsiConferenceEventManager.prototype.setupRTCListeners = function() {
                 conference.eventEmitter.emit(
                     JitsiConferenceEvents.DOMINANT_SPEAKER_CHANGED, dominant, previous);
 
+                if (previous && previous.length) {
+                    const speakerList = previous.slice(0);
+
+                    // Add the dominant speaker to the top of the list (exclude self).
+                    if (conference.myUserId !== dominant) {
+                        speakerList.splice(0, 0, dominant);
+                    }
+
+                    // Trim the list to the top 5 speakers only.
+                    if (speakerList.length > SPEAKERS_AUDIO_LEVELS) {
+                        speakerList.splice(SPEAKERS_AUDIO_LEVELS, speakerList.length - SPEAKERS_AUDIO_LEVELS);
+                    }
+                    conference.statistics && conference.statistics.setSpeakerList(speakerList);
+                }
                 if (conference.statistics && conference.myUserId() === dominant) {
                     // We are the new dominant speaker.
                     conference.statistics.sendDominantSpeakerEvent(conference.room.roomjid);

modules/RTC/TraceablePeerConnection.js

@@ -468,6 +468,32 @@ TraceablePeerConnection.prototype._getDesiredMediaDirection = function(
     return 'inactive';
 };
 
+/**
+ * Returns the list of RTCRtpReceivers created for the source of the given media type associated with
+ * the set of remote endpoints specified.
+ * @param {Array<string>} endpoints list of the endpoints
+ * @param {string} mediaType 'audio' or 'video'
+ * @returns {Array<RTCRtpReceiver>} list of receivers created by the peerconnection.
+ */
+TraceablePeerConnection.prototype._getReceiversByEndpointIds = function(endpoints, mediaType) {
+    let remoteTracks = [];
+    let receivers = [];
+
+    for (const endpoint of endpoints) {
+        remoteTracks = remoteTracks.concat(this.getRemoteTracks(endpoint, mediaType));
+    }
+
+    // Get the ids of the MediaStreamTracks associated with each of these remote tracks.
+    const remoteTrackIds = remoteTracks.map(remote => remote.track?.id);
+
+    receivers = this.peerconnection.getReceivers()
+        .filter(receiver => receiver.track
+            && receiver.track.kind === mediaType
+            && remoteTrackIds.find(trackId => trackId === receiver.track.id));
+
+    return receivers;
+};
+
 /**
  * Tells whether or not this TPC instance is using Simulcast.
  * @return {boolean} <tt>true</tt> if simulcast is enabled and active or
@@ -526,16 +552,17 @@ TraceablePeerConnection.prototype._peerMutedChanged = function(
 };
 
 /**
- * Obtains audio levels of the remote audio tracks by getting the source
- * information on the RTCRtpReceivers. The information relevant to the ssrc
- * is updated each time a RTP packet constaining the ssrc is received.
- * @returns {Object} containing ssrc and audio level information as a
- * key-value pair.
+ * Obtains audio levels of the remote audio tracks by getting the source information on the RTCRtpReceivers.
+ * The information relevant to the ssrc is updated each time a RTP packet constaining the ssrc is received.
+ * @param {Array<string>} speakerList list of endpoint ids for which audio levels are to be gathered.
+ * @returns {Object} containing ssrc and audio level information as a key-value pair.
  */
-TraceablePeerConnection.prototype.getAudioLevels = function() {
+TraceablePeerConnection.prototype.getAudioLevels = function(speakerList = []) {
     const audioLevels = {};
-    const audioReceivers = this.peerconnection.getReceivers()
-        .filter(receiver => receiver.track && receiver.track.kind === MediaType.AUDIO);
+    const audioReceivers = speakerList.length
+        ? this._getReceiversByEndpointIds(speakerList, MediaType.AUDIO)
+        : this.peerconnection.getReceivers()
+            .filter(receiver => receiver.track && receiver.track.kind === MediaType.AUDIO && receiver.track.enabled);
 
     audioReceivers.forEach(remote => {
         const ssrc = remote.getSynchronizationSources();

modules/browser/BrowserCapabilities.js

@@ -182,7 +182,11 @@ export default class BrowserCapabilities extends BrowserDetection {
      */
     supportsReceiverStats() {
         return typeof window.RTCRtpReceiver !== 'undefined'
-            && Object.keys(RTCRtpReceiver.prototype).indexOf('getSynchronizationSources') > -1;
+            && Object.keys(RTCRtpReceiver.prototype).indexOf('getSynchronizationSources') > -1
+
+            // Disable this on Safari because it is reporting 0.000001 as the audio levels for all
+            // remote audio tracks.
+            && !this.isWebKitBased();
     }
 
     /**

modules/statistics/RTPStatsCollector.js

@@ -260,6 +260,7 @@ export default function StatsCollector(
     // Updates stats interval
     this.audioLevelsIntervalMilis = audioLevelsInterval;
 
+    this.speakerList = [];
     this.statsIntervalId = null;
     this.statsIntervalMilis = statsInterval;
 
@@ -270,7 +271,15 @@ export default function StatsCollector(
     this.ssrc2stats = new Map();
 }
 
-/* eslint-enable max-params */
+/**
+ * Set the list of the remote speakers for which audio levels are to be calculated.
+ *
+ * @param {Array<string>} speakerList - Endpoint ids.
+ * @returns {void}
+ */
+StatsCollector.prototype.setSpeakerList = function(speakerList) {
+    this.speakerList = speakerList;
+};
 
 /**
  * Stops stats updates.
@@ -308,7 +317,7 @@ StatsCollector.prototype.start = function(startAudioLevelStats) {
         this.audioLevelsIntervalId = setInterval(
             () => {
                 if (browser.supportsReceiverStats()) {
-                    const audioLevels = this.peerconnection.getAudioLevels();
+                    const audioLevels = this.peerconnection.getAudioLevels(this.speakerList);
 
                     for (const ssrc in audioLevels) {
                         if (audioLevels.hasOwnProperty(ssrc)) {

modules/statistics/constants.js

@@ -1 +1,8 @@
 export const CALLSTATS_SCRIPT_URL = 'https://api.callstats.io/static/callstats-ws.min.js';
+
+/**
+ * The number of remote speakers for which the audio levels will be calculated using
+ * RTCRtpReceiver#getSynchronizationSources. Limit the number of endpoints to save cpu on the client as this API call
+ * is known to take longer to execute when there are many audio receivers.
+ */
+export const SPEAKERS_AUDIO_LEVELS = 5;

modules/statistics/statistics.js

@@ -345,6 +345,20 @@ Statistics.prototype.removeLongTasksStatsListener = function(listener) {
     this.eventEmitter.removeListener(StatisticsEvents.LONG_TASKS_STATS, listener);
 };
 
+/**
+ * Updates the list of speakers for which the audio levels are to be calculated. This is needed for the jvb pc only.
+ *
+ * @param {Array<string>} speakerList The list of remote endpoint ids.
+ * @returns {void}
+ */
+Statistics.prototype.setSpeakerList = function(speakerList) {
+    for (const rtpStats of Array.from(this.rtpStatsMap.values())) {
+        if (!rtpStats.peerconnection.isP2P) {
+            rtpStats.setSpeakerList(speakerList);
+        }
+    }
+};
+
 Statistics.prototype.dispose = function() {
     try {
         // NOTE Before reading this please see the comment in stopCallStats...