ref(RTC): cleanup local track access

JitsiConference.js

@@ -262,17 +262,27 @@ JitsiConference.prototype.getExternalAuthUrl = function (urlForPopup) {
 JitsiConference.prototype.getLocalTracks = function (mediaType) {
     let tracks = [];
     if (this.rtc) {
-        tracks = this.rtc.localTracks.slice();
-    }
-    if (mediaType !== undefined) {
-        tracks = tracks.filter(
-            (track) => {
-                return track && track.getType && track.getType() === mediaType;
-            });
+        tracks = this.rtc.getLocalTracks(mediaType);
     }
     return tracks;
 };
 
+/**
+ * Obtains local audio track.
+ * @return {JitsiLocalTrack|null}
+ */
+JitsiConference.prototype.getLocalAudioTrack = function () {
+    return this.rtc ? this.rtc.getLocalAudioTrack() : null;
+};
+
+/**
+ * Obtains local video track.
+ * @return {JitsiLocalTrack|null}
+ */
+JitsiConference.prototype.getLocalVideoTrack = function () {
+    return this.rtc ? this.rtc.getLocalVideoTrack() : null;
+};
+
 /**
  * Attaches a handler for events(For example - "participant joined".) in the conference. All possible event are defined
  * in JitsiConferenceEvents.
@@ -394,12 +404,10 @@ JitsiConference.prototype.getTranscriber = function(){
     if (this.transcriber === undefined){
         this.transcriber = new Transcriber();
         //add all existing local audio tracks to the transcriber
-        // FIXME accessing localTracks field directly
-        this.rtc.localTracks.forEach(function (localTrack) {
-            if (localTrack.isAudioTrack()){
-                this.transcriber.addTrack(localTrack);
-            }
-        }.bind(this));
+        const localAudioTracks = this.getLocalTracks(MediaType.AUDIO);
+        for (const localAudio of localAudioTracks) {
+            this.transcriber.addTrack(localAudio);
+        }
         //and all remote audio tracks
         const remoteAudioTracks = this.rtc.getRemoteTracks(MediaType.AUDIO);
         for (const remoteTrack of remoteAudioTracks){
@@ -1028,7 +1036,7 @@ function (jingleSession, jingleOffer, now) {
 
     this.rtc.initializeDataChannels(jingleSession.peerconnection);
     // Add local Tracks to the ChatRoom
-    this.rtc.localTracks.forEach(function(localTrack) {
+    this.getLocalTracks().forEach(function(localTrack) {
         var ssrcInfo = null;
         /**
          * We don't do this for Firefox because, on Firefox, we keep the
@@ -1124,7 +1132,7 @@ JitsiConference.prototype.onCallEnded
     // will learn what their SSRC from the new PeerConnection which will be
     // created on incoming call event.
     var self = this;
-    this.rtc.localTracks.forEach(function(localTrack) {
+    this.getLocalTracks().forEach(function(localTrack) {
         // Reset SSRC as it will no longer be valid
         localTrack._setSSRC(null);
         // Bind the handler to fetch new SSRC, it will un register itself once
@@ -1177,21 +1185,25 @@ JitsiConference.prototype.myUserId = function () {
 };
 
 JitsiConference.prototype.sendTones = function (tones, duration, pause) {
+    // FIXME P2P 'dtmfManager' must be cleared, after switching jingleSessions
     if (!this.dtmfManager) {
-        var connection = this.xmpp.connection.jingle.activecall.peerconnection;
-        if (!connection) {
-            logger.warn("cannot sendTones: no conneciton");
+        if (!this.jingleSession) {
+            logger.warn("cannot sendTones: no jingle session");
             return;
         }
 
-        var tracks = this.getLocalTracks().filter(function (track) {
-            return track.isAudioTrack();
-        });
-        if (!tracks.length) {
+        const peerConnection = this.jingleSession.peerconnection;
+        if (!peerConnection) {
+            logger.warn("cannot sendTones: no peer connection");
+            return;
+        }
+
+        const localAudio = this.getLocalAudioTrack();
+        if (!localAudio) {
             logger.warn("cannot sendTones: no local audio stream");
             return;
         }
-        this.dtmfManager = new JitsiDTMFManager(tracks[0], connection);
+        this.dtmfManager = new JitsiDTMFManager(localAudio, peerConnection);
     }
 
     this.dtmfManager.sendTones(tones, duration, pause);

JitsiConferenceEventManager.js

@@ -573,9 +573,9 @@ JitsiConferenceEventManager.prototype.setupStatisticsListeners = function () {
     });
 
     conference.statistics.addByteSentStatsListener(function (stats) {
-        conference.getLocalTracks().forEach(function (track) {
-            var ssrc = track.getSSRC();
-            if(!track.isAudioTrack() || !ssrc || !stats.hasOwnProperty(ssrc))
+        conference.getLocalTracks(MediaType.AUDIO).forEach(function (track) {
+            const ssrc = track.getSSRC();
+            if (!ssrc || !stats.hasOwnProperty(ssrc))
                 return;
 
             track._setByteSent(stats[ssrc]);

modules/RTC/JitsiTrack.js

@@ -308,15 +308,6 @@ JitsiTrack.prototype.dispose = function () {
 JitsiTrack.prototype.isScreenSharing = function() {
 };
 
-/**
- * FIXME remove hack in SDP.js and this method
- * Returns id of the track.
- * @returns {string|null} id of the track or null if this is fake track.
- */
-JitsiTrack.prototype._getId = function () {
-    return this.getTrackId();
-};
-
 /**
  * Returns id of the track.
  * @returns {string|null} id of the track or null if this is fake track.

modules/RTC/RTC.js

@@ -55,8 +55,6 @@ export default class RTC extends Listenable {
         this.localTracks = [];
         //FIXME: We should support multiple streams per jid.
         this.remoteTracks = {};
-        this.localAudio = null;
-        this.localVideo = null;
         this.options = options;
         // A flag whether we had received that the data channel had opened
         // we can get this flag out of sync if for some reason data channel got
@@ -259,20 +257,39 @@ export default class RTC extends Listenable {
         this.localTracks.push(track);
 
         track.conference = this.conference;
-
-        if (track.isAudioTrack()) {
-            this.localAudio = track;
-        } else {
-            this.localVideo = track;
-        }
     }
 
     /**
      * Get local video track.
-     * @returns {JitsiLocalTrack}
+     * @returns {JitsiLocalTrack|undefined}
      */
     getLocalVideoTrack () {
-        return this.localVideo;
+        const localVideo = this.getLocalTracks(MediaType.VIDEO);
+        return localVideo.length ? localVideo[0] : undefined;
+    }
+
+    /**
+     * Get local audio track.
+     * @returns {JitsiLocalTrack|undefined}
+     */
+    getLocalAudioTrack () {
+        const localAudio = this.getLocalTracks(MediaType.AUDIO);
+        return localAudio.length ? localAudio[0] : undefined;
+    }
+
+    /**
+     * Returns the local tracks of the given media type, or all local tracks if
+     * no specific type is given.
+     * @param {MediaType} [mediaType] optional media type filter
+     * (audio or video).
+     */
+    getLocalTracks (mediaType) {
+        let tracks = this.localTracks.slice();
+        if (mediaType !== undefined) {
+            tracks = tracks.filter(
+                (track) => { return track.getType() === mediaType; });
+        }
+        return tracks;
     }
 
     /**
@@ -344,32 +361,22 @@ export default class RTC extends Listenable {
      * @returns {Promise}
      */
     setAudioMute (value) {
-        var mutePromises = [];
-        for(var i = 0; i < this.localTracks.length; i++) {
-            var track = this.localTracks[i];
-            if(track.getType() !== MediaType.AUDIO) {
-                continue;
-            }
+        const mutePromises = [];
+        this.getLocalTracks(MediaType.AUDIO).forEach(function(audioTrack){
             // this is a Promise
-            mutePromises.push(value ? track.mute() : track.unmute());
-        }
+            mutePromises.push(value ? audioTrack.mute() : audioTrack.unmute());
+        });
         // we return a Promise from all Promises so we can wait for their execution
         return Promise.all(mutePromises);
     }
 
     removeLocalTrack (track) {
-        var pos = this.localTracks.indexOf(track);
+        const pos = this.localTracks.indexOf(track);
         if (pos === -1) {
             return;
         }
 
         this.localTracks.splice(pos, 1);
-
-        if (track.isAudioTrack()) {
-            this.localAudio = null;
-        } else {
-            this.localVideo = null;
-        }
     }
 
     /**
@@ -636,13 +643,13 @@ export default class RTC extends Listenable {
      * @param ssrc the ssrc to check.
      */
     getResourceBySSRC (ssrc) {
-        if((this.localVideo && ssrc == this.localVideo.getSSRC())
-            || (this.localAudio && ssrc == this.localAudio.getSSRC())) {
+        if (this.getLocalTracks().find(
+                localTrack => { return localTrack.getSSRC() == ssrc; })) {
             return this.conference.myUserId();
         }
 
-        var track = this.getRemoteTrackBySSRC(ssrc);
-        return track? track.getParticipantId() : null;
+        const track = this.getRemoteTrackBySSRC(ssrc);
+        return track ? track.getParticipantId() : null;
     }
 
     /**

modules/connectivity/ConnectionQuality.js

@@ -5,7 +5,6 @@ import {getLogger} from "jitsi-meet-logger";
 import RTCBrowserType from "../RTC/RTCBrowserType";
 
 var XMPPEvents = require('../../service/xmpp/XMPPEvents');
-var MediaType = require('../../service/RTC/MediaType');
 var VideoType = require('../../service/RTC/VideoType');
 var Resolutions = require("../../service/RTC/Resolutions");
 
@@ -359,8 +358,7 @@ export default class ConnectionQuality {
         // about the resolution, and they look at their local rendered
         // resolution instead. Consider removing this.
         let localVideoTrack
-            = this._conference.getLocalTracks(MediaType.VIDEO)
-                .find(track => track.isVideoTrack());
+            = this._conference.getLocalVideoTrack();
         if (localVideoTrack && localVideoTrack.resolution) {
             data.resolution = localVideoTrack.resolution;
         }
@@ -390,9 +388,8 @@ export default class ConnectionQuality {
         let key;
         let updateLocalConnectionQuality
             = !this._conference.isConnectionInterrupted();
-        let localVideoTrack =
-                this._conference.getLocalTracks(MediaType.VIDEO)
-                    .find(track => track.isVideoTrack());
+        let localVideoTrack
+            = this._conference.getLocalVideoTrack();
         let videoType = localVideoTrack ? localVideoTrack.videoType : undefined;
         let isMuted = localVideoTrack ? localVideoTrack.isMuted() : true;
         let resolution = localVideoTrack ? localVideoTrack.resolution : null;

modules/xmpp/SDP.js

@@ -251,12 +251,14 @@ SDP.prototype.toJingle = function (elem, thecreator) {
                     elem.c('parameter');
                     elem.attrs({name: "cname", value:Math.random().toString(36).substring(7)});
                     elem.up();
-                    var msid = null;
-                    if(mline.media == "audio") {
-                        // FIXME what is this ? global APP.RTC in SDP ?
-                        msid = APP.RTC.localAudio._getId();
-                    } else {
-                        msid = APP.RTC.localVideo._getId();
+                    // FIXME what case does this code handle ? remove ???
+                    let msid = null;
+                    // FIXME what is this ? global APP.RTC in SDP ?
+                    const localTrack = APP.RTC.getLocalTracks(mline.media);
+                    if (localTrack) {
+                        // FIXME before this changes the track id was accessed,
+                        // but msid stands for the stream id, makes no sense ?
+                        msid = localTrack.getTrackId();
                     }
                     if(msid != null) {
                         msid = SDPUtil.filter_special_chars(msid);