138 lines
4.0 KiB
JavaScript
138 lines
4.0 KiB
JavaScript
/* global require, APP */
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/**
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* Created by hristo on 8/4/14.
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*/
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var RTPStats = require("./RTPStatsCollector.js");
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var EventEmitter = require("events");
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var StreamEventTypes = require("../../service/RTC/StreamEventTypes.js");
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var XMPPEvents = require("../../service/xmpp/XMPPEvents");
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var CallStats = require("./CallStats");
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var RTCEvents = require("../../service/RTC/RTCEvents");
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var StatisticsEvents = require("../../service/statistics/Events");
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var eventEmitter = new EventEmitter();
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var rtpStats = null;
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function stopRemote() {
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if (rtpStats) {
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rtpStats.stop();
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eventEmitter.emit(StatisticsEvents.STOP);
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rtpStats = null;
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}
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}
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function startRemoteStats (peerconnection) {
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if (rtpStats) {
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rtpStats.stop();
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}
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rtpStats = new RTPStats(peerconnection, 200, 2000, eventEmitter);
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rtpStats.start();
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}
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function onDisposeConference(onUnload) {
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CallStats.sendTerminateEvent();
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stopRemote();
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if (onUnload) {
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eventEmitter.removeAllListeners();
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}
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}
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export default {
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/**
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* Indicates that this audio level is for local jid.
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* @type {string}
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*/
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LOCAL_JID: 'local',
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addListener: function(type, listener) {
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eventEmitter.on(type, listener);
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},
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removeListener: function (type, listener) {
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eventEmitter.removeListener(type, listener);
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},
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stop: function () {
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stopRemote();
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if (eventEmitter) {
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eventEmitter.removeAllListeners();
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}
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},
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onAudioMute (mute) {
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CallStats.sendMuteEvent(mute, "audio");
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},
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onVideoMute (mute) {
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CallStats.sendMuteEvent(mute, "video");
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},
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onGetUserMediaFailed (e) {
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CallStats.sendGetUserMediaFailed(e);
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},
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start: function () {
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const xmpp = APP.conference._room.xmpp;
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xmpp.addListener(
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XMPPEvents.DISPOSE_CONFERENCE,
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onDisposeConference
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);
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//FIXME: we may want to change CALL INCOMING event to
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// onnegotiationneeded
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xmpp.addListener(XMPPEvents.CALL_INCOMING, function (event) {
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startRemoteStats(event.peerconnection);
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// CallStats.init(event);
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});
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xmpp.addListener(
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XMPPEvents.PEERCONNECTION_READY,
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function (session) {
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CallStats.init(session);
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}
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);
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xmpp.addListener(XMPPEvents.CONFERENCE_SETUP_FAILED, function () {
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CallStats.sendSetupFailedEvent();
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});
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xmpp.addListener(RTCEvents.CREATE_OFFER_FAILED, function (e, pc) {
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CallStats.sendCreateOfferFailed(e, pc);
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});
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xmpp.addListener(RTCEvents.CREATE_ANSWER_FAILED, function (e, pc) {
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CallStats.sendCreateAnswerFailed(e, pc);
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});
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xmpp.addListener(
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RTCEvents.SET_LOCAL_DESCRIPTION_FAILED,
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function (e, pc) {
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CallStats.sendSetLocalDescFailed(e, pc);
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}
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);
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xmpp.addListener(
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RTCEvents.SET_REMOTE_DESCRIPTION_FAILED,
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function (e, pc) {
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CallStats.sendSetRemoteDescFailed(e, pc);
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}
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);
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xmpp.addListener(
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RTCEvents.ADD_ICE_CANDIDATE_FAILED,
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function (e, pc) {
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CallStats.sendAddIceCandidateFailed(e, pc);
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}
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);
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},
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/**
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* FIXME:
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* Currently used by torture. If we are removing this, torture needs to
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* be fixed also.
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*
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* Obtains audio level reported in the stats for specified peer.
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* @param peerJid full MUC jid of the user for whom we want to obtain last
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* audio level.
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* @param ssrc the SSRC of audio stream for which we want to obtain audio
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* level.
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* @returns {*} a float form 0 to 1 that represents current audio level or
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* <tt>null</tt> if for any reason the value is not available
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* at this time.
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*/
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getPeerSSRCAudioLevel: function (peerJid, ssrc) {
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var peerStats = rtpStats.jid2stats[peerJid];
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return peerStats ? peerStats.ssrc2AudioLevel[ssrc] : null;
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}
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};
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