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JitsiConference.js
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JitsiConference.js
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/* global __filename, $, Promise */
import { Strophe } from 'strophe.js';
import {
ACTION_JINGLE_RESTART,
ACTION_JINGLE_SI_RECEIVED,
ACTION_JINGLE_TERMINATE,
ACTION_P2P_ESTABLISHED,
ACTION_P2P_FAILED,
ACTION_P2P_SWITCH_TO_JVB,
ICE_ESTABLISHMENT_DURATION_DIFF,
createJingleEvent,
createP2PEvent
} from './service/statistics/AnalyticsEvents';
import AvgRTPStatsReporter from './modules/statistics/AvgRTPStatsReporter';
import ComponentsVersions from './modules/version/ComponentsVersions';
import ConnectionQuality from './modules/connectivity/ConnectionQuality';
import { getLogger } from 'jitsi-meet-logger';
import GlobalOnErrorHandler from './modules/util/GlobalOnErrorHandler';
import EventEmitter from 'events';
import authenticateAndUpgradeRole from './authenticateAndUpgradeRole';
import * as JitsiConferenceErrors from './JitsiConferenceErrors';
import JitsiConferenceEventManager from './JitsiConferenceEventManager';
import * as JitsiConferenceEvents from './JitsiConferenceEvents';
import JitsiDTMFManager from './modules/DTMF/JitsiDTMFManager';
import JitsiParticipant from './JitsiParticipant';
import JitsiTrackError from './JitsiTrackError';
import * as JitsiTrackErrors from './JitsiTrackErrors';
import * as JitsiTrackEvents from './JitsiTrackEvents';
import Jvb121EventGenerator from './modules/event/Jvb121EventGenerator';
import * as MediaType from './service/RTC/MediaType';
import ParticipantConnectionStatusHandler
from './modules/connectivity/ParticipantConnectionStatus';
import P2PDominantSpeakerDetection from './modules/P2PDominantSpeakerDetection';
import RTC from './modules/RTC/RTC';
import RTCBrowserType from './modules/RTC/RTCBrowserType';
import * as RTCEvents from './service/RTC/RTCEvents';
import Statistics from './modules/statistics/statistics';
import TalkMutedDetection from './modules/TalkMutedDetection';
import Transcriber from './modules/transcription/transcriber';
import VideoType from './service/RTC/VideoType';
import VideoSIPGW from './modules/videosipgw/VideoSIPGW';
import * as VideoSIPGWConstants from './modules/videosipgw/VideoSIPGWConstants';
import * as XMPPEvents from './service/xmpp/XMPPEvents';
import SpeakerStatsCollector from './modules/statistics/SpeakerStatsCollector';
const logger = getLogger(__filename);
/**
* Creates a JitsiConference object with the given name and properties.
* Note: this constructor is not a part of the public API (objects should be
* created using JitsiConnection.createConference).
* @param options.config properties / settings related to the conference that
* will be created.
* @param options.name the name of the conference
* @param options.connection the JitsiConnection object for this
* JitsiConference.
* @param {number} [options.config.avgRtpStatsN=15] how many samples are to be
* collected by {@link AvgRTPStatsReporter}, before arithmetic mean is
* calculated and submitted to the analytics module.
* @param {boolean} [options.config.p2p.enabled] when set to <tt>true</tt>
* the peer to peer mode will be enabled. It means that when there are only 2
* participants in the conference an attempt to make direct connection will be
* made. If the connection succeeds the conference will stop sending data
* through the JVB connection and will use the direct one instead.
* @param {number} [options.config.p2p.backToP2PDelay=5] a delay given in
* seconds, before the conference switches back to P2P, after the 3rd
* participant has left the room.
* @param {number} [options.config.channelLastN=-1] The requested amount of
* videos are going to be delivered after the value is in effect. Set to -1 for
* unlimited or all available videos.
* @param {number} [options.config.forceJVB121Ratio]
* "Math.random() < forceJVB121Ratio" will determine whether a 2 people
* conference should be moved to the JVB instead of P2P. The decision is made on
* the responder side, after ICE succeeds on the P2P connection.
* @param {*} [options.config.openBridgeChannel] Which kind of communication to
* open with the videobridge. Values can be "datachannel", "websocket", true
* (treat it as "datachannel"), undefined (treat it as "datachannel") and false
* (don't open any channel).
* @constructor
*
* FIXME Make all methods which are called from lib-internal classes
* to non-public (use _). To name a few:
* {@link JitsiConference.onLocalRoleChanged}
* {@link JitsiConference.onUserRoleChanged}
* {@link JitsiConference.onMemberLeft}
* and so on...
*/
export default function JitsiConference(options) {
if (!options.name || options.name.toLowerCase() !== options.name) {
const errmsg
= 'Invalid conference name (no conference name passed or it '
+ 'contains invalid characters like capital letters)!';
logger.error(errmsg);
throw new Error(errmsg);
}
this.eventEmitter = new EventEmitter();
this.options = options;
this.eventManager = new JitsiConferenceEventManager(this);
this.participants = {};
this._init(options);
this.componentsVersions = new ComponentsVersions(this);
/**
* Jingle session instance for the JVB connection.
* @type {JingleSessionPC}
*/
this.jvbJingleSession = null;
this.lastDominantSpeaker = null;
this.dtmfManager = null;
this.somebodySupportsDTMF = false;
this.authEnabled = false;
this.startAudioMuted = false;
this.startVideoMuted = false;
this.startMutedPolicy = {
audio: false,
video: false
};
this.availableDevices = {
audio: undefined,
video: undefined
};
this.isMutedByFocus = false;
// Flag indicates if the 'onCallEnded' method was ever called on this
// instance. Used to log extra analytics event for debugging purpose.
// We need to know if the potential issue happened before or after
// the restart.
this.wasStopped = false;
/**
* The object which monitors local and remote connection statistics (e.g.
* sending bitrate) and calculates a number which represents the connection
* quality.
*/
this.connectionQuality
= new ConnectionQuality(this, this.eventEmitter, options);
/**
* Reports average RTP statistics to the analytics module.
* @type {AvgRTPStatsReporter}
*/
this.avgRtpStatsReporter
= new AvgRTPStatsReporter(this, options.config.avgRtpStatsN || 15);
/**
* Indicates whether the connection is interrupted or not.
*/
this.isJvbConnectionInterrupted = false;
/**
* The object which tracks active speaker times
*/
this.speakerStatsCollector = new SpeakerStatsCollector(this);
/* P2P related fields below: */
/**
* Stores reference to deferred start P2P task. It's created when 3rd
* participant leaves the room in order to avoid ping pong effect (it
* could be just a page reload).
* @type {number|null}
*/
this.deferredStartP2PTask = null;
const delay
= parseInt(options.config.p2p && options.config.p2p.backToP2PDelay, 10);
/**
* A delay given in seconds, before the conference switches back to P2P
* after the 3rd participant has left.
* @type {number}
*/
this.backToP2PDelay = isNaN(delay) ? 5 : delay;
logger.info(`backToP2PDelay: ${this.backToP2PDelay}`);
/**
* If set to <tt>true</tt> it means the P2P ICE is no longer connected.
* When <tt>false</tt> it means that P2P ICE (media) connection is up
* and running.
* @type {boolean}
*/
this.isP2PConnectionInterrupted = false;
/**
* Flag set to <tt>true</tt> when P2P session has been established
* (ICE has been connected) and this conference is currently in the peer to
* peer mode (P2P connection is the active one).
* @type {boolean}
*/
this.p2p = false;
/**
* A JingleSession for the direct peer to peer connection.
* @type {JingleSessionPC}
*/
this.p2pJingleSession = null;
this.videoSIPGWHandler = new VideoSIPGW(this.room);
}
// FIXME convert JitsiConference to ES6 - ASAP !
JitsiConference.prototype.constructor = JitsiConference;
/**
* Initializes the conference object properties
* @param options {object}
* @param options.connection {JitsiConnection} overrides this.connection
*/
JitsiConference.prototype._init = function(options = {}) {
// Override connection and xmpp properties (Useful if the connection
// reloaded)
if (options.connection) {
this.connection = options.connection;
this.xmpp = this.connection.xmpp;
// Setup XMPP events only if we have new connection object.
this.eventManager.setupXMPPListeners();
}
const { config } = this.options;
this.room = this.xmpp.createRoom(this.options.name, config);
// Connection interrupted/restored listeners
this._onIceConnectionInterrupted
= this._onIceConnectionInterrupted.bind(this);
this.room.addListener(
XMPPEvents.CONNECTION_INTERRUPTED, this._onIceConnectionInterrupted);
this._onIceConnectionRestored = this._onIceConnectionRestored.bind(this);
this.room.addListener(
XMPPEvents.CONNECTION_RESTORED, this._onIceConnectionRestored);
this._onIceConnectionEstablished
= this._onIceConnectionEstablished.bind(this);
this.room.addListener(
XMPPEvents.CONNECTION_ESTABLISHED, this._onIceConnectionEstablished);
this.room.updateDeviceAvailability(RTC.getDeviceAvailability());
if (!this.rtc) {
this.rtc = new RTC(this, options);
this.eventManager.setupRTCListeners();
}
this.participantConnectionStatus
= new ParticipantConnectionStatusHandler(
this.rtc,
this,
{
// Both these options are not public API, leaving it here only
// as an entry point through config for tuning up purposes.
// Default values should be adjusted as soon as optimal values
// are discovered.
rtcMuteTimeout: config._peerConnStatusRtcMuteTimeout,
outOfLastNTimeout: config._peerConnStatusOutOfLastNTimeout
});
this.participantConnectionStatus.init();
if (!this.statistics) {
// XXX The property location on the global variable window is not
// defined in all execution environments (e.g. react-native). While
// jitsi-meet may polyfill it when executing on react-native, it is
// better for the cross-platform support to not require window.location
// especially when there is a worthy alternative (as demonstrated
// bellow).
const windowLocation = window.location;
let callStatsAliasName = this.myUserId();
if (config.enableDisplayNameInStats && config.displayName) {
callStatsAliasName = config.displayName;
}
this.statistics = new Statistics(this.xmpp, {
callStatsAliasName,
callStatsConfIDNamespace:
config.callStatsConfIDNamespace
|| (windowLocation && windowLocation.hostname)
|| (config.hosts && config.hosts.domain),
callStatsCustomScriptUrl: config.callStatsCustomScriptUrl,
callStatsID: config.callStatsID,
callStatsSecret: config.callStatsSecret,
roomName: this.options.name,
swapUserNameAndAlias: config.enableStatsID,
applicationName: config.applicationName,
getWiFiStatsMethod: config.getWiFiStatsMethod
});
}
this.eventManager.setupChatRoomListeners();
// Always add listeners because on reload we are executing leave and the
// listeners are removed from statistics module.
this.eventManager.setupStatisticsListeners();
if (config.enableTalkWhileMuted) {
// eslint-disable-next-line no-new
new TalkMutedDetection(
this,
() =>
this.eventEmitter.emit(JitsiConferenceEvents.TALK_WHILE_MUTED));
}
if ('channelLastN' in config) {
this.setLastN(config.channelLastN);
}
/**
* Emits {@link JitsiConferenceEvents.JVB121_STATUS}.
* @type {Jvb121EventGenerator}
*/
this.jvb121Status = new Jvb121EventGenerator(this);
// creates dominant speaker detection that works only in p2p mode
this.p2pDominantSpeakerDetection = new P2PDominantSpeakerDetection(this);
};
/**
* Joins the conference.
* @param password {string} the password
*/
JitsiConference.prototype.join = function(password) {
if (this.room) {
this.room.join(password);
}
};
/**
* Authenticates and upgrades the role of the local participant/user.
*
* @returns {Object} A <tt>thenable</tt> which (1) settles when the process of
* authenticating and upgrading the role of the local participant/user finishes
* and (2) has a <tt>cancel</tt> method that allows the caller to interrupt the
* process.
*/
JitsiConference.prototype.authenticateAndUpgradeRole = function(...args) {
return authenticateAndUpgradeRole.apply(this, args);
};
/**
* Check if joined to the conference.
*/
JitsiConference.prototype.isJoined = function() {
return this.room && this.room.joined;
};
/**
* Tells whether or not the P2P mode is enabled in the configuration.
* @return {boolean}
*/
JitsiConference.prototype.isP2PEnabled = function() {
return Boolean(this.options.config.p2p && this.options.config.p2p.enabled)
// FIXME: remove once we have a default config template. -saghul
|| typeof this.options.config.p2p === 'undefined';
};
/**
* When in P2P test mode, the conference will not automatically switch to P2P
* when there 2 participants.
* @return {boolean}
*/
JitsiConference.prototype.isP2PTestModeEnabled = function() {
return Boolean(this.options.config.testing
&& this.options.config.testing.p2pTestMode);
};
/**
* Leaves the conference.
* @returns {Promise}
*/
JitsiConference.prototype.leave = function() {
if (this.participantConnectionStatus) {
this.participantConnectionStatus.dispose();
this.participantConnectionStatus = null;
}
if (this.avgRtpStatsReporter) {
this.avgRtpStatsReporter.dispose();
this.avgRtpStatsReporter = null;
}
this.getLocalTracks().forEach(track => this.onLocalTrackRemoved(track));
this.rtc.closeBridgeChannel();
if (this.statistics) {
this.statistics.dispose();
}
// Close both JVb and P2P JingleSessions
if (this.jvbJingleSession) {
this.jvbJingleSession.close();
this.jvbJingleSession = null;
}
if (this.p2pJingleSession) {
this.p2pJingleSession.close();
this.p2pJingleSession = null;
}
// leave the conference
if (this.room) {
const room = this.room;
// Unregister connection state listeners
room.removeListener(
XMPPEvents.CONNECTION_INTERRUPTED,
this._onIceConnectionInterrupted);
room.removeListener(
XMPPEvents.CONNECTION_RESTORED,
this._onIceConnectionRestored);
room.removeListener(
XMPPEvents.CONNECTION_ESTABLISHED,
this._onIceConnectionEstablished);
this.room = null;
return room.leave().catch(() => {
// remove all participants because currently the conference won't
// be usable anyway. This is done on success automatically by the
// ChatRoom instance.
this.getParticipants().forEach(
participant => this.onMemberLeft(participant.getJid()));
});
}
// If this.room == null we are calling second time leave().
return Promise.reject(
new Error('The conference is has been already left'));
};
/**
* Returns name of this conference.
*/
JitsiConference.prototype.getName = function() {
return this.options.name;
};
/**
* Check if authentication is enabled for this conference.
*/
JitsiConference.prototype.isAuthEnabled = function() {
return this.authEnabled;
};
/**
* Check if user is logged in.
*/
JitsiConference.prototype.isLoggedIn = function() {
return Boolean(this.authIdentity);
};
/**
* Get authorized login.
*/
JitsiConference.prototype.getAuthLogin = function() {
return this.authIdentity;
};
/**
* Check if external authentication is enabled for this conference.
*/
JitsiConference.prototype.isExternalAuthEnabled = function() {
return this.room && this.room.moderator.isExternalAuthEnabled();
};
/**
* Get url for external authentication.
* @param {boolean} [urlForPopup] if true then return url for login popup,
* else url of login page.
* @returns {Promise}
*/
JitsiConference.prototype.getExternalAuthUrl = function(urlForPopup) {
return new Promise((resolve, reject) => {
if (!this.isExternalAuthEnabled()) {
reject();
return;
}
if (urlForPopup) {
this.room.moderator.getPopupLoginUrl(resolve, reject);
} else {
this.room.moderator.getLoginUrl(resolve, reject);
}
});
};
/**
* 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 (audio or video).
*/
JitsiConference.prototype.getLocalTracks = function(mediaType) {
let tracks = [];
if (this.rtc) {
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.
* @param eventId the event ID.
* @param handler handler for the event.
*
* Note: consider adding eventing functionality by extending an EventEmitter
* impl, instead of rolling ourselves
*/
JitsiConference.prototype.on = function(eventId, handler) {
if (this.eventEmitter) {
this.eventEmitter.on(eventId, handler);
}
};
/**
* Removes event listener
* @param eventId the event ID.
* @param [handler] optional, the specific handler to unbind
*
* Note: consider adding eventing functionality by extending an EventEmitter
* impl, instead of rolling ourselves
*/
JitsiConference.prototype.off = function(eventId, handler) {
if (this.eventEmitter) {
this.eventEmitter.removeListener(eventId, handler);
}
};
// Common aliases for event emitter
JitsiConference.prototype.addEventListener = JitsiConference.prototype.on;
JitsiConference.prototype.removeEventListener = JitsiConference.prototype.off;
/**
* Receives notifications from other participants about commands / custom events
* (sent by sendCommand or sendCommandOnce methods).
* @param command {String} the name of the command
* @param handler {Function} handler for the command
*/
JitsiConference.prototype.addCommandListener = function(command, handler) {
if (this.room) {
this.room.addPresenceListener(command, handler);
}
};
/**
* Removes command listener
* @param command {String} the name of the command
*/
JitsiConference.prototype.removeCommandListener = function(command) {
if (this.room) {
this.room.removePresenceListener(command);
}
};
/**
* Sends text message to the other participants in the conference
* @param message the text message.
*/
JitsiConference.prototype.sendTextMessage = function(message) {
if (this.room) {
this.room.sendMessage(message);
}
};
/**
* Send private text message to another participant of the conference
* @param message the text message.
*/
JitsiConference.prototype.sendPrivateTextMessage = function(id, message) {
if (this.room) {
this.room.sendPrivateMessage(id, message);
}
};
/**
* Send presence command.
* @param name {String} the name of the command.
* @param values {Object} with keys and values that will be sent.
**/
JitsiConference.prototype.sendCommand = function(name, values) {
if (this.room) {
this.room.addToPresence(name, values);
this.room.sendPresence();
}
};
/**
* Send presence command one time.
* @param name {String} the name of the command.
* @param values {Object} with keys and values that will be sent.
**/
JitsiConference.prototype.sendCommandOnce = function(name, values) {
this.sendCommand(name, values);
this.removeCommand(name);
};
/**
* Removes presence command.
* @param name {String} the name of the command.
**/
JitsiConference.prototype.removeCommand = function(name) {
if (this.room) {
this.room.removeFromPresence(name);
}
};
/**
* Sets the display name for this conference.
* @param name the display name to set
*/
JitsiConference.prototype.setDisplayName = function(name) {
if (this.room) {
// remove previously set nickname
this.room.removeFromPresence('nick');
this.room.addToPresence('nick', {
attributes: { xmlns: 'http://jabber.org/protocol/nick' },
value: name
});
this.room.sendPresence();
}
};
/**
* Set new subject for this conference. (available only for moderator)
* @param {string} subject new subject
*/
JitsiConference.prototype.setSubject = function(subject) {
if (this.room && this.isModerator()) {
this.room.setSubject(subject);
}
};
/**
* Get a transcriber object for all current participants in this conference
* @return {Transcriber} the transcriber object
*/
JitsiConference.prototype.getTranscriber = function() {
if (this.transcriber === undefined) {
this.transcriber = new Transcriber();
// add all existing local audio tracks to the transcriber
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) {
this.transcriber.addTrack(remoteTrack);
}
}
return this.transcriber;
};
/**
* Returns the transcription status.
*
* @returns {String} "on" or "off".
*/
JitsiConference.prototype.getTranscriptionStatus = function() {
return this.room.transcriptionStatus;
};
/**
* Adds JitsiLocalTrack object to the conference.
* @param track the JitsiLocalTrack object.
* @returns {Promise<JitsiLocalTrack>}
* @throws {Error} if the specified track is a video track and there is already
* another video track in the conference.
*/
JitsiConference.prototype.addTrack = function(track) {
if (track.isVideoTrack()) {
// Ensure there's exactly 1 local video track in the conference.
const localVideoTrack = this.rtc.getLocalVideoTrack();
if (localVideoTrack) {
// Don't be excessively harsh and severe if the API client happens
// to attempt to add the same local video track twice.
if (track === localVideoTrack) {
return Promise.resolve(track);
}
return Promise.reject(new Error(
'cannot add second video track to the conference'));
}
}
return this.replaceTrack(null, track);
};
/**
* Fires TRACK_AUDIO_LEVEL_CHANGED change conference event (for local tracks).
* @param {number} audioLevel the audio level
* @param {TraceablePeerConnection} [tpc]
*/
JitsiConference.prototype._fireAudioLevelChangeEvent = function(
audioLevel,
tpc) {
const activeTpc = this.getActivePeerConnection();
// There will be no TraceablePeerConnection if audio levels do not come from
// a peerconnection. LocalStatsCollector.js measures audio levels using Web
// Audio Analyser API and emits local audio levels events through
// JitsiTrack.setAudioLevel, but does not provide TPC instance which is
// optional.
if (!tpc || activeTpc === tpc) {
this.eventEmitter.emit(
JitsiConferenceEvents.TRACK_AUDIO_LEVEL_CHANGED,
this.myUserId(), audioLevel);
}
};
/**
* Fires TRACK_MUTE_CHANGED change conference event.
* @param track the JitsiTrack object related to the event.
*/
JitsiConference.prototype._fireMuteChangeEvent = function(track) {
// check if track was muted by focus and now is unmuted by user
if (this.isMutedByFocus && track.isAudioTrack() && !track.isMuted()) {
this.isMutedByFocus = false;
// unmute local user on server
this.room.muteParticipant(this.room.myroomjid, false);
}
this.eventEmitter.emit(JitsiConferenceEvents.TRACK_MUTE_CHANGED, track);
};
/**
* Clear JitsiLocalTrack properties and listeners.
* @param track the JitsiLocalTrack object.
*/
JitsiConference.prototype.onLocalTrackRemoved = function(track) {
track._setConference(null);
this.rtc.removeLocalTrack(track);
track.removeEventListener(JitsiTrackEvents.TRACK_MUTE_CHANGED,
track.muteHandler);
track.removeEventListener(JitsiTrackEvents.TRACK_AUDIO_LEVEL_CHANGED,
track.audioLevelHandler);
// send event for stopping screen sharing
// FIXME: we assume we have only one screen sharing track
// if we change this we need to fix this check
if (track.isVideoTrack() && track.videoType === VideoType.DESKTOP) {
this.statistics.sendScreenSharingEvent(false);
}
this.eventEmitter.emit(JitsiConferenceEvents.TRACK_REMOVED, track);
};
/**
* Removes JitsiLocalTrack from the conference and performs
* a new offer/answer cycle.
* @param {JitsiLocalTrack} track
* @returns {Promise}
*/
JitsiConference.prototype.removeTrack = function(track) {
return this.replaceTrack(track, null);
};
/**
* Replaces oldTrack with newTrack and performs a single offer/answer
* cycle after both operations are done. Either oldTrack or newTrack
* can be null; replacing a valid 'oldTrack' with a null 'newTrack'
* effectively just removes 'oldTrack'
* @param {JitsiLocalTrack} oldTrack the current stream in use to be replaced
* @param {JitsiLocalTrack} newTrack the new stream to use
* @returns {Promise} resolves when the replacement is finished
*/
JitsiConference.prototype.replaceTrack = function(oldTrack, newTrack) {
// First do the removal of the oldTrack at the JitsiConference level
if (oldTrack) {
if (oldTrack.disposed) {
return Promise.reject(
new JitsiTrackError(JitsiTrackErrors.TRACK_IS_DISPOSED));
}
}
if (newTrack) {
if (newTrack.disposed) {
return Promise.reject(
new JitsiTrackError(JitsiTrackErrors.TRACK_IS_DISPOSED));
}
}
// Now replace the stream at the lower levels
return this._doReplaceTrack(oldTrack, newTrack)
.then(() => {
if (oldTrack) {
this.onLocalTrackRemoved(oldTrack);
}
if (newTrack) {
// Now handle the addition of the newTrack at the
// JitsiConference level
this._setupNewTrack(newTrack);
}
return Promise.resolve();
}, error => Promise.reject(new Error(error)));
};
/**
* Replaces the tracks at the lower level by going through the Jingle session
* and WebRTC peer connection. The method will resolve immediately if there is
* currently no JingleSession started.
* @param {JitsiLocalTrack|null} oldTrack the track to be removed during
* the process or <tt>null</t> if the method should act as "add track"
* @param {JitsiLocalTrack|null} newTrack the new track to be added or
* <tt>null</tt> if the method should act as "remove track"
* @return {Promise} resolved when the process is done or rejected with a string
* which describes the error.
* @private
*/
JitsiConference.prototype._doReplaceTrack = function(oldTrack, newTrack) {
const replaceTrackPromises = [];
if (this.jvbJingleSession) {
replaceTrackPromises.push(
this.jvbJingleSession.replaceTrack(oldTrack, newTrack));
} else {
logger.info('_doReplaceTrack - no JVB JingleSession');
}
if (this.p2pJingleSession) {
replaceTrackPromises.push(
this.p2pJingleSession.replaceTrack(oldTrack, newTrack));
} else {
logger.info('_doReplaceTrack - no P2P JingleSession');
}
return Promise.all(replaceTrackPromises);
};
/**
* Operations related to creating a new track
* @param {JitsiLocalTrack} newTrack the new track being created
*/
JitsiConference.prototype._setupNewTrack = function(newTrack) {
if (newTrack.isAudioTrack() || (newTrack.isVideoTrack()
&& newTrack.videoType !== VideoType.DESKTOP)) {
// Report active device to statistics
const devices = RTC.getCurrentlyAvailableMediaDevices();
const device
= devices.find(
d =>
d.kind === `${newTrack.getTrack().kind}input`
&& d.label === newTrack.getTrack().label);
if (device) {
Statistics.sendActiveDeviceListEvent(
RTC.getEventDataForActiveDevice(device));
}
}
if (newTrack.isVideoTrack()) {
this.removeCommand('videoType');
this.sendCommand('videoType', {
value: newTrack.videoType,
attributes: {
xmlns: 'http://jitsi.org/jitmeet/video'
}
});
}
this.rtc.addLocalTrack(newTrack);
// ensure that we're sharing proper "is muted" state
if (newTrack.isAudioTrack()) {
this.room.setAudioMute(newTrack.isMuted());
} else {
this.room.setVideoMute(newTrack.isMuted());
}
newTrack.muteHandler = this._fireMuteChangeEvent.bind(this, newTrack);
newTrack.audioLevelHandler = this._fireAudioLevelChangeEvent.bind(this);
newTrack.addEventListener(
JitsiTrackEvents.TRACK_MUTE_CHANGED,
newTrack.muteHandler);
newTrack.addEventListener(
JitsiTrackEvents.TRACK_AUDIO_LEVEL_CHANGED,
newTrack.audioLevelHandler);
newTrack._setConference(this);
this.eventEmitter.emit(JitsiConferenceEvents.TRACK_ADDED, newTrack);
};
/**
* Method called by the {@link JitsiLocalTrack} (a video one) in order to add
* back the underlying WebRTC MediaStream to the PeerConnection (which has
* removed on video mute).
* @param {JitsiLocalTrack} track the local track that will be added as part of
* the unmute operation.
* @return {Promise} resolved when the process is done or rejected with a string
* which describes the error.
*/
JitsiConference.prototype._addLocalTrackAsUnmute = function(track) {
const addAsUnmutePromises = [];
if (this.jvbJingleSession) {
addAsUnmutePromises.push(this.jvbJingleSession.addTrackAsUnmute(track));
} else {
logger.info(
'Add local MediaStream as unmute -'
+ ' no JVB Jingle session started yet');
}
if (this.p2pJingleSession) {
addAsUnmutePromises.push(this.p2pJingleSession.addTrackAsUnmute(track));
} else {
logger.info(
'Add local MediaStream as unmute -'
+ ' no P2P Jingle session started yet');
}
return Promise.all(addAsUnmutePromises);
};
/**
* Method called by the {@link JitsiLocalTrack} (a video one) in order to remove
* the underlying WebRTC MediaStream from the PeerConnection. The purpose of
* that is to stop sending any data and turn off the HW camera device.
* @param {JitsiLocalTrack} track the local track that will be removed.
* @return {Promise}
*/
JitsiConference.prototype._removeLocalTrackAsMute = function(track) {
const removeAsMutePromises = [];
if (this.jvbJingleSession) {
removeAsMutePromises.push(
this.jvbJingleSession.removeTrackAsMute(track));
} else {
logger.info(
'Remove local MediaStream - no JVB JingleSession started yet');
}
if (this.p2pJingleSession) {
removeAsMutePromises.push(
this.p2pJingleSession.removeTrackAsMute(track));
} else {
logger.info(
'Remove local MediaStream - no P2P JingleSession started yet');
}
return Promise.all(removeAsMutePromises);
};
/**
* Get role of the local user.
* @returns {string} user role: 'moderator' or 'none'
*/
JitsiConference.prototype.getRole = function() {
return this.room.role;
};
/**
* Check if local user is moderator.
* @returns {boolean|null} true if local user is moderator, false otherwise. If
* we're no longer in the conference room then <tt>null</tt> is returned.
*/
JitsiConference.prototype.isModerator = function() {
return this.room ? this.room.isModerator() : null;
};
/**
* Set password for the room.
* @param {string} password new password for the room.
* @returns {Promise}
*/
JitsiConference.prototype.lock = function(password) {
if (!this.isModerator()) {
return Promise.reject();
}
return new Promise((resolve, reject) => {
this.room.lockRoom(
password || '',
() => resolve(),