ljm_j1/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");
var GlobalOnErrorHandler = require("../util/GlobalOnErrorHandler");

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

    // Switch audio output device on all remote audio tracks. Local audio tracks
    // handle this event by themselves.
    if (RTCUtils.isDeviceChangeAvailable('output')) {
        RTCUtils.addListener(RTCEvents.AUDIO_OUTPUT_DEVICE_CHANGED,
            function (deviceId) {
                for (var key in self.remoteTracks) {
                    if (self.remoteTracks.hasOwnProperty(key)
                        && self.remoteTracks[key].audio) {
                        self.remoteTracks[key].audio.setAudioOutput(deviceId);
                    }
                }
            });
    }
}

/**
 * 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);
    // Add local Tracks to the ChatRoom
    this.localTracks.forEach(function(localTrack) {
        var ssrcInfo = null;
        if(localTrack.isVideoTrack() && localTrack.isMuted()) {
            /**
             * 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 inconsistent 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 .
             */
            localTrack._setSSRC(
                this.room.generateNewStreamSSRCInfo());
            ssrcInfo = {
                mtype: localTrack.getType(),
                type: "addMuted",
                ssrc: localTrack.ssrc,
                msid: localTrack.initialMSID
            };
        }
        try {
            this.room.addStream(
                localTrack.getOriginalStream(), function () {}, function () {},
                ssrcInfo, true);
        } catch(e) {
            GlobalOnErrorHandler.callErrorHandler(e);
            logger.error(e);
        }
    }.bind(this));
};

RTC.prototype.selectEndpoint = function (id) {
    if(this.dataChannels)
        this.dataChannels.sendSelectedEndpointMessage(id);
};

RTC.prototype.pinEndpoint = function (id) {
    if(this.dataChannels)
        this.dataChannels.sendPinnedEndpointMessage(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) {
    if (!track)
        throw new Error('track must not be null nor undefined');

    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;
    }
};

/**
 * Initializes a new JitsiRemoteTrack instance with the data provided by (a)
 * ChatRoom to XMPPEvents.REMOTE_TRACK_ADDED.
 *
 * @param {Object} event the data provided by (a) ChatRoom to
 * XMPPEvents.REMOTE_TRACK_ADDED to (a)
 */
RTC.prototype.createRemoteTrack = function (event) {
    var ownerJid = event.owner;
    var remoteTrack = new JitsiRemoteTrack(
        this,  ownerJid, event.stream,    event.track,
        event.mediaType, event.videoType, event.ssrc, event.muted);
    var resource = Strophe.getResourceFromJid(ownerJid);
    var remoteTracks
        = this.remoteTracks[resource] || (this.remoteTracks[resource] = {});
    var mediaType = remoteTrack.getType();
    if (remoteTracks[mediaType]) {
        logger.warn("Overwriting remote track!", resource, mediaType);
    }
    remoteTracks[mediaType] = remoteTrack;
    return remoteTrack;
};

/**
 * Removes all JitsiRemoteTracks associated with given MUC nickname (resource
 * part of the JID).
 * @param resource the resource part of the MUC JID
 * @returns {JitsiRemoteTrack|null}
 */
RTC.prototype.removeRemoteTracks = function (resource) {
    var remoteTracks = this.remoteTracks[resource];

    if(remoteTracks) {
        remoteTracks['audio'] && remoteTracks['audio'].dispose();
        remoteTracks['video'] && remoteTracks['video'].dispose();
        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);
};

/**
 * 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 input (camera / microphone) or output
 * (audio) device is supported and false if not.
 * @params {string} [deviceType] - type of device to change. Default is
 *      undefined or 'input', 'output' - for audio output device change.
 * @returns {boolean} true if available, false otherwise.
 */
RTC.isDeviceChangeAvailable = function (deviceType) {
    return RTCUtils.isDeviceChangeAvailable(deviceType);
};

/**
 * Returns currently used audio output device id, '' stands for default
 * device
 * @returns {string}
 */
RTC.getAudioOutputDevice = function () {
    return RTCUtils.getAudioOutputDevice();
};

/**
 * Returns list of available media devices if its obtained, otherwise an
 * empty array is returned/
 * @returns {Array} list of available media devices.
 */
RTC.getCurrentlyAvailableMediaDevices = function () {
    return RTCUtils.getCurrentlyAvailableMediaDevices();
};

/**
 * Returns event data for device to be reported to stats.
 * @returns {MediaDeviceInfo} device.
 */
RTC.getEventDataForActiveDevice = function (device) {
    return RTCUtils.getEventDataForActiveDevice(device);
};

/**
 * Sets current audio output device.
 * @param {string} deviceId - id of 'audiooutput' device from
 *      navigator.mediaDevices.enumerateDevices()
 * @returns {Promise} - resolves when audio output is changed, is rejected
 *      otherwise
 */
RTC.setAudioOutputDevice = function (deviceId) {
    return RTCUtils.setAudioOutputDevice(deviceId);
};

/**
 * Returns <tt>true<tt/> if given WebRTC MediaStream is considered a valid
 * "user" stream which means that it's not a "receive only" stream nor a "mixed"
 * JVB stream.
 *
 * Clients that implement Unified Plan, such as Firefox use recvonly
 * "streams/channels/tracks" for receiving remote stream/tracks, as opposed to
 * Plan B where there are only 3 channels: audio, video and data.
 *
 * @param stream WebRTC MediaStream instance
 * @returns {boolean}
 */
RTC.isUserStream = function (stream) {
    var streamId = RTCUtils.getStreamID(stream);
    return streamId && streamId !== "mixedmslabel" && streamId !== "default";
};

/**
 * 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);
};

/**
 * 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();
};

/**
 * Closes all currently opened data channels.
 */
RTC.prototype.closeAllDataChannels = function () {
    this.dataChannels.closeAllChannels();
};

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

/*
 //FIXME Never used, but probably *should* be used for switching
 //      between camera and screen, but has to be adjusted to work with tracks.
 //      Current when switching to desktop we can see recv-only being advertised
 //      because we do remove and add.
 //
 //      Leaving it commented out, in order to not forget about FF specific
 //      thing
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 track = this.getRemoteTrackBySSRC(ssrc);
    return track? track.getParticipantId() : null;
};

/**
 * Searches in remoteTracks for the ssrc and returns the corresponding track.
 * @param ssrc the ssrc to check.
 */
RTC.prototype.getRemoteTrackBySSRC = function (ssrc) {
    for (var resource in this.remoteTracks) {
        var track = this.getRemoteAudioTrack(resource);
        if(track && track.getSSRC() == ssrc) {
            return track;
        }
        track = this.getRemoteVideoTrack(resource);
        if(track && track.getSSRC() == ssrc) {
            return track;
        }
    }
    return null;
};

module.exports = RTC;