lib-jitsi-meet/modules/RTC/RTC.js

/* global __filename, APP, module */
var logger = require("jitsi-meet-logger").getLogger(__filename);
var EventEmitter = require("events");
var RTCBrowserType = require("./RTCBrowserType");
var RTCEvents = require("../../service/RTC/RTCEvents.js");
var RTCUtils = require("./RTCUtils.js");
var JitsiTrack = require("./JitsiTrack");
var JitsiLocalTrack = require("./JitsiLocalTrack.js");
var DataChannels = require("./DataChannels");
var JitsiRemoteTrack = require("./JitsiRemoteTrack.js");
var MediaType = require("../../service/RTC/MediaType");
var VideoType = require("../../service/RTC/VideoType");

function createLocalTracks(tracksInfo, options) {
    var newTracks = [];
    var deviceId = null;
    tracksInfo.forEach(function(trackInfo){
        if (trackInfo.type === MediaType.AUDIO) {
          deviceId = options.micDeviceId;
        } else if (trackInfo.videoType === VideoType.CAMERA){
          deviceId = options.cameraDeviceId;
        }
        var localTrack = new JitsiLocalTrack(trackInfo.stream,
            trackInfo.videoType, trackInfo.resolution, deviceId);
        newTracks.push(localTrack);
    });
    return newTracks;
}

function RTC(room, options) {
    this.room = room;
    this.localTracks = [];
    //FIXME: We should support multiple streams per jid.
    this.remoteTracks = {};
    this.localAudio = null;
    this.localVideo = null;
    this.eventEmitter = new EventEmitter();
    var self = this;
    this.options = options || {};
    room.addPresenceListener("videomuted", function (values, from) {
        var videoTrack = self.getRemoteVideoTrack(from);
        // If there is no video track, but we receive it is muted,
        // we need to create a dummy track which we will mute, so we can
        // notify interested about the muting
        if (!videoTrack) {
            videoTrack = self.createRemoteTrack(
                {
                    peerjid: room.roomjid + "/" + from,
                    videoType: VideoType.CAMERA,
                    jitsiTrackType: MediaType.VIDEO
                },
                null, null);
            self.eventEmitter
                .emit(RTCEvents.FAKE_VIDEO_TRACK_CREATED, videoTrack);
        }
        videoTrack.setMute(values.value == "true");
    });
    room.addPresenceListener("audiomuted", function (values, from) {
        var audioTrack = self.getRemoteAudioTrack(from);
        if (audioTrack) {
            audioTrack.setMute(values.value == "true");
        }
    });
    room.addPresenceListener("videoType", function(data, from) {
        var videoTrack = self.getRemoteVideoTrack(from);
        if (videoTrack) {
            videoTrack._setVideoType(data.value);
        }
    });
}

/**
 * Creates the local MediaStreams.
 * @param {Object} [options] optional parameters
 * @param {Array} options.devices the devices that will be requested
 * @param {string} options.resolution resolution constraints
 * @param {bool} options.dontCreateJitsiTrack if <tt>true</tt> objects with the
 * following structure {stream: the Media Stream,
 * type: "audio" or "video", videoType: "camera" or "desktop"}
 * will be returned trough the Promise, otherwise JitsiTrack objects will be
 * returned.
 * @param {string} options.cameraDeviceId
 * @param {string} options.micDeviceId
 * @returns {*} Promise object that will receive the new JitsiTracks
 */

RTC.obtainAudioAndVideoPermissions = function (options) {
    return RTCUtils.obtainAudioAndVideoPermissions(options).then(
        function (tracksInfo) {
            return createLocalTracks(tracksInfo, options);
    });
};

RTC.prototype.onIncommingCall = function(event) {
    if(this.options.config.openSctp)
        this.dataChannels = new DataChannels(event.peerconnection,
            this.eventEmitter);
    for(var i = 0; i < this.localTracks.length; i++)
        if(this.localTracks[i])
        {
            var ssrcInfo = null;
            if(this.localTracks[i].isMuted() &&
                this.localTracks[i].getType() === MediaType.VIDEO) {
                /**
                 * Handles issues when the stream is added before the peerconnection is created.
                 * The peerconnection is created when second participant enters the call. In
                 * that use case the track doesn't have information about it's ssrcs and no
                 * jingle packets are sent. That can cause inconsistant behavior later.
                 *
                 * For example:
                 * If we mute the stream and than second participant enter it's remote SDP won't
                 * include that track. On unmute we are not sending any jingle packets which
                 * will brake the unmute.
                 *
                 * In order to solve issues like the above one here we have to generate the ssrc
                 * information for the track .
                 */
                this.localTracks[i]._setSSRC(
                    this.room.generateNewStreamSSRCInfo());
                ssrcInfo = {
                    mtype: this.localTracks[i].getType(),
                    type: "addMuted",
                    ssrc: this.localTracks[i].ssrc,
                    msid: this.localTracks[i].initialMSID
                }
            }
            this.room.addStream(this.localTracks[i].getOriginalStream(),
                function () {}, ssrcInfo, true);
        }
};

RTC.prototype.selectedEndpoint = function (id) {
    if(this.dataChannels)
        this.dataChannels.handleSelectedEndpointEvent(id);
};

RTC.prototype.pinEndpoint = function (id) {
    if(this.dataChannels)
        this.dataChannels.handlePinnedEndpointEvent(id);
};

RTC.prototype.addListener = function (type, listener) {
    this.eventEmitter.on(type, listener);
};

RTC.prototype.removeListener = function (eventType, listener) {
    this.eventEmitter.removeListener(eventType, listener);
};

RTC.addListener = function (eventType, listener) {
    RTCUtils.addListener(eventType, listener);
};

RTC.removeListener = function (eventType, listener) {
    RTCUtils.removeListener(eventType, listener)
};

RTC.isRTCReady = function () {
    return RTCUtils.isRTCReady();
};

RTC.init = function (options) {
    this.options = options || {};
    return RTCUtils.init(this.options);
};

RTC.getDeviceAvailability = function () {
    return RTCUtils.getDeviceAvailability();
};

RTC.prototype.addLocalTrack = function (track) {
    this.localTracks.push(track);
    track._setRTC(this);

    if (track.isAudioTrack()) {
        this.localAudio = track;
    } else {
        this.localVideo = track;
    }
};

/**
 * Get local video track.
 * @returns {JitsiLocalTrack}
 */
RTC.prototype.getLocalVideoTrack = function () {
    return this.localVideo;
};

/**
 * Gets JitsiRemoteTrack for AUDIO MediaType associated with given MUC nickname
 * (resource part of the JID).
 * @param resource the resource part of the MUC JID
 * @returns {JitsiRemoteTrack|null}
 */
RTC.prototype.getRemoteAudioTrack = function (resource) {
    if (this.remoteTracks[resource])
        return this.remoteTracks[resource][MediaType.AUDIO];
    else
        return null;
};

/**
 * Gets JitsiRemoteTrack for VIDEO MediaType associated with given MUC nickname
 * (resource part of the JID).
 * @param resource the resource part of the MUC JID
 * @returns {JitsiRemoteTrack|null}
 */
RTC.prototype.getRemoteVideoTrack = function (resource) {
    if (this.remoteTracks[resource])
        return this.remoteTracks[resource][MediaType.VIDEO];
    else
        return null;
};

/**
 * Set mute for all local audio streams attached to the conference.
 * @param value the mute value
 * @returns {Promise}
 */
RTC.prototype.setAudioMute = function (value) {
    var mutePromises = [];
    for(var i = 0; i < this.localTracks.length; i++) {
        var track = this.localTracks[i];
        if(track.getType() !== MediaType.AUDIO) {
            continue;
        }
        // this is a Promise
        mutePromises.push(value ? track.mute() : track.unmute());
    }
    // we return a Promise from all Promises so we can wait for their execution
    return Promise.all(mutePromises);
};

RTC.prototype.removeLocalTrack = function (track) {
    var pos = this.localTracks.indexOf(track);
    if (pos === -1) {
        return;
    }

    this.localTracks.splice(pos, 1);

    if (track.isAudioTrack()) {
        this.localAudio = null;
    } else {
        this.localVideo = null;
    }
};

RTC.prototype.createRemoteTrack = function (data, sid, thessrc) {
    var remoteTrack = new JitsiRemoteTrack(this, data, sid, thessrc);
    if(!data.peerjid)
        return;
    var resource = Strophe.getResourceFromJid(data.peerjid);
    if(!this.remoteTracks[resource]) {
        this.remoteTracks[resource] = {};
    }
    var mediaType = remoteTrack.getType();
    if (this.remoteTracks[resource][mediaType]) {
        logger.warn(
            "Overwriting remote track !", resource, mediaType);
    }
    this.remoteTracks[resource][mediaType] = remoteTrack;
    return remoteTrack;
};

RTC.prototype.removeRemoteTrack = function (resource) {
    // FIXME this clears both audio and video tracks!
    if(this.remoteTracks[resource]) {
        delete this.remoteTracks[resource];
    }
};

RTC.getPCConstraints = function () {
    return RTCUtils.pc_constraints;
};

RTC.attachMediaStream =  function (elSelector, stream) {
    return RTCUtils.attachMediaStream(elSelector, stream);
};

RTC.getStreamID = function (stream) {
    return RTCUtils.getStreamID(stream);
};

RTC.getVideoSrc = function (element) {
    return RTCUtils.getVideoSrc(element);
};

/**
 * Returns true if retrieving the the list of input devices is supported and
 * false if not.
 */
RTC.isDeviceListAvailable = function () {
    return RTCUtils.isDeviceListAvailable();
};

/**
 * Returns true if changing the camera / microphone device is supported and
 * false if not.
 */
RTC.isDeviceChangeAvailable = function () {
    return RTCUtils.isDeviceChangeAvailable();
};
/**
 * Allows to receive list of available cameras/microphones.
 * @param {function} callback would receive array of devices as an argument
 */
RTC.enumerateDevices = function (callback) {
    RTCUtils.enumerateDevices(callback);
};

RTC.setVideoSrc = function (element, src) {
    RTCUtils.setVideoSrc(element, src);
};

/**
 * A method to handle stopping of the stream.
 * One point to handle the differences in various implementations.
 * @param mediaStream MediaStream object to stop.
 */
RTC.stopMediaStream = function (mediaStream) {
    RTCUtils.stopMediaStream(mediaStream);
};

/**
 * Returns whether the desktop sharing is enabled or not.
 * @returns {boolean}
 */
RTC.isDesktopSharingEnabled = function () {
    return RTCUtils.isDesktopSharingEnabled();
};

RTC.prototype.dispose = function() {
};

RTC.prototype.switchVideoTracks = function (newStream) {
    this.localVideo.stream = newStream;

    this.localTracks = [];

    //in firefox we have only one stream object
    if (this.localAudio.getOriginalStream() != newStream)
        this.localTracks.push(this.localAudio);
    this.localTracks.push(this.localVideo);
};

RTC.prototype.setAudioLevel = function (resource, audioLevel) {
    if(!resource)
        return;
    var audioTrack = this.getRemoteAudioTrack(resource);
    if(audioTrack) {
        audioTrack.setAudioLevel(audioLevel);
    }
};

/**
 * Searches in localTracks(session stores ssrc for audio and video) and
 * remoteTracks for the ssrc and returns the corresponding resource.
 * @param ssrc the ssrc to check.
 */
RTC.prototype.getResourceBySSRC = function (ssrc) {
    if((this.localVideo && ssrc == this.localVideo.getSSRC())
        || (this.localAudio && ssrc == this.localAudio.getSSRC())) {
        return Strophe.getResourceFromJid(this.room.myroomjid);
    }

    var self = this;
    var resultResource = null;
    Object.keys(this.remoteTracks).forEach(function (resource) {
        var audioTrack = self.getRemoteAudioTrack(resource);
        var videoTrack = self.getRemoteVideoTrack(resource);
        if((audioTrack && audioTrack.getSSRC() == ssrc) ||
            (videoTrack && videoTrack.getSSRC() == ssrc)) {
            resultResource = resource;
        }
    });

    return resultResource;
};

module.exports = RTC;