import Logger from '@jitsi/logger';
import * as JitsiConferenceErrors from './JitsiConferenceErrors';
import * as JitsiConferenceEvents from './JitsiConferenceEvents';
import JitsiConnection from './JitsiConnection';
import * as JitsiConnectionErrors from './JitsiConnectionErrors';
import * as JitsiConnectionEvents from './JitsiConnectionEvents';
import JitsiMediaDevices from './JitsiMediaDevices';
import * as JitsiMediaDevicesEvents from './JitsiMediaDevicesEvents';
import JitsiTrackError from './JitsiTrackError';
import * as JitsiTrackErrors from './JitsiTrackErrors';
import * as JitsiTrackEvents from './JitsiTrackEvents';
import * as JitsiTranscriptionStatus from './JitsiTranscriptionStatus';
import RTC from './modules/RTC/RTC';
import RTCStats from './modules/RTCStats/RTCStats';
import browser from './modules/browser';
import NetworkInfo from './modules/connectivity/NetworkInfo';
import { TrackStreamingStatus } from './modules/connectivity/TrackStreamingStatus';
import getActiveAudioDevice from './modules/detection/ActiveDeviceDetector';
import * as DetectionEvents from './modules/detection/DetectionEvents';
import TrackVADEmitter from './modules/detection/TrackVADEmitter';
import FeatureFlags from './modules/flags/FeatureFlags';
import ProxyConnectionService
from './modules/proxyconnection/ProxyConnectionService';
import recordingConstants from './modules/recording/recordingConstants';
import Settings from './modules/settings/Settings';
import LocalStatsCollector from './modules/statistics/LocalStatsCollector';
import Statistics from './modules/statistics/statistics';
import ScriptUtil from './modules/util/ScriptUtil';
import * as VideoSIPGWConstants from './modules/videosipgw/VideoSIPGWConstants';
import AudioMixer from './modules/webaudio/AudioMixer';
import { MediaType } from './service/RTC/MediaType';
import * as ConnectionQualityEvents
from './service/connectivity/ConnectionQualityEvents';
import * as E2ePingEvents from './service/e2eping/E2ePingEvents';
import { createGetUserMediaEvent } from './service/statistics/AnalyticsEvents';
import * as RTCStatsEvents from './modules/RTCStats/RTCStatsEvents';
import { VideoType } from './service/RTC/VideoType';
const logger = Logger.getLogger(__filename);
/**
* The amount of time to wait until firing
* {@link JitsiMediaDevicesEvents.PERMISSION_PROMPT_IS_SHOWN} event.
*/
const USER_MEDIA_SLOW_PROMISE_TIMEOUT = 1000;
/**
* Indicates whether GUM has been executed or not.
*/
let hasGUMExecuted = false;
/**
* Extracts from an 'options' objects with a specific format (TODO what IS the
* format?) the attributes which are to be logged in analytics events.
*
* @param options gum options (???)
* @returns {*} the attributes to attach to analytics events.
*/
function getAnalyticsAttributesFromOptions(options) {
const attributes: any = {};
attributes['audio_requested'] = options.devices.includes('audio');
attributes['video_requested'] = options.devices.includes('video');
attributes['screen_sharing_requested'] = options.devices.includes('desktop');
if (attributes.video_requested) {
attributes.resolution = options.resolution;
}
return attributes;
}
interface ICreateLocalTrackOptions {
cameraDeviceId?: string;
devices?: any[];
firePermissionPromptIsShownEvent?: boolean;
fireSlowPromiseEvent?: boolean;
micDeviceId?: string;
resolution?: string;
}
interface IJitsiMeetJSOptions {
enableAnalyticsLogging?: boolean;
enableWindowOnErrorHandler?: boolean;
externalStorage?: Storage;
flags?: {
runInLiteMode?: boolean;
ssrcRewritingEnabled?: boolean;
}
}
interface ICreateLocalTrackFromMediaStreamOptions {
stream: MediaStream,
sourceType: string,
mediaType: MediaType,
videoType?: VideoType
}
/**
* The public API of the Jitsi Meet library (a.k.a. {@code JitsiMeetJS}).
*/
export default {
version: '{#COMMIT_HASH#}',
JitsiConnection,
/**
* {@code ProxyConnectionService} is used to connect a remote peer to a
* local Jitsi participant without going through a Jitsi conference. It is
* currently used for room integration development, specifically wireless
* screensharing. Its API is experimental and will likely change; usage of
* it is advised against.
*/
ProxyConnectionService,
constants: {
recording: recordingConstants,
sipVideoGW: VideoSIPGWConstants,
transcriptionStatus: JitsiTranscriptionStatus,
trackStreamingStatus: TrackStreamingStatus
},
events: {
conference: JitsiConferenceEvents,
connection: JitsiConnectionEvents,
detection: DetectionEvents,
track: JitsiTrackEvents,
mediaDevices: JitsiMediaDevicesEvents,
connectionQuality: ConnectionQualityEvents,
e2eping: E2ePingEvents,
rtcstats: RTCStatsEvents
},
errors: {
conference: JitsiConferenceErrors,
connection: JitsiConnectionErrors,
track: JitsiTrackErrors
},
errorTypes: {
JitsiTrackError
},
logLevels: Logger.levels,
mediaDevices: JitsiMediaDevices as unknown,
analytics: Statistics.analytics as unknown,
init(options: IJitsiMeetJSOptions = {}) {
Logger.setLogLevel(Logger.levels.INFO);
// @ts-ignore
logger.info(`This appears to be ${browser.getName()}, ver: ${browser.getVersion()}`);
Settings.init(options.externalStorage);
Statistics.init(options);
// Initialize global window.connectionTimes
// FIXME do not use 'window'
if (!window.connectionTimes) {
window.connectionTimes = {};
}
if (options.enableAnalyticsLogging !== true) {
logger.warn('Analytics disabled, disposing.');
this.analytics.dispose();
}
return RTC.init(options);
},
/**
* Returns whether the desktop sharing is enabled or not.
*
* @returns {boolean}
*/
isDesktopSharingEnabled() {
return RTC.isDesktopSharingEnabled();
},
/**
* Returns whether the current execution environment supports WebRTC (for
* use within this library).
*
* @returns {boolean} {@code true} if WebRTC is supported in the current
* execution environment (for use within this library); {@code false},
* otherwise.
*/
isWebRtcSupported() {
return RTC.isWebRtcSupported();
},
setLogLevel(level) {
Logger.setLogLevel(level);
},
/**
* Expose rtcstats to the public API.
*/
rtcstats: {
/**
* Sends identity data to the rtcstats server. This data is used
* to identify the specifics of a particular client, it can be any object
* and will show in the generated rtcstats dump under "identity" entries.
*
* @param {Object} identityData - Identity data to send.
* @returns {void}
*/
sendIdentityEntry(identityData) {
RTCStats.sendIdentity(identityData);
},
/**
* Sends a stats entry to rtcstats server.
* @param {string} statsType - The type of stats to send.
* @param {Object} data - The stats data to send.
*/
sendStatsEntry(statsType, data) {
RTCStats.sendStatsEntry(statsType, null, data);
},
/**
* Events generated by rtcstats, such as PeerConnections state,
* and websocket connection state.
*
* @param {RTCStatsEvents} event - The event name.
* @param {function} handler - The event handler.
*/
on(event, handler) {
RTCStats.events.on(event, handler);
}
},
/**
* Sets the log level to the Logger instance with given id.
*
* @param {Logger.levels} level the logging level to be set
* @param {string} id the logger id to which new logging level will be set.
* Usually it's the name of the JavaScript source file including the path
* ex. "modules/xmpp/ChatRoom.js"
*/
setLogLevelById(level, id) {
Logger.setLogLevelById(level, id);
},
/**
* Registers new global logger transport to the library logging framework.
*
* @param globalTransport
* @see Logger.addGlobalTransport
*/
addGlobalLogTransport(globalTransport) {
Logger.addGlobalTransport(globalTransport);
},
/**
* Removes global logging transport from the library logging framework.
*
* @param globalTransport
* @see Logger.removeGlobalTransport
*/
removeGlobalLogTransport(globalTransport) {
Logger.removeGlobalTransport(globalTransport);
},
/**
* Sets global options which will be used by all loggers. Changing these
* works even after other loggers are created.
*
* @param options
* @see Logger.setGlobalOptions
*/
setGlobalLogOptions(options) {
Logger.setGlobalOptions(options);
},
/**
* Creates local media tracks.
*
* @param options Object with properties / settings specifying the tracks
* which should be created. should be created or some additional
* configurations about resolution for example.
* @param {Array} options.effects optional effects array for the track
* @param {boolean} options.firePermissionPromptIsShownEvent - if event
* JitsiMediaDevicesEvents.PERMISSION_PROMPT_IS_SHOWN should be fired
* @param {boolean} options.fireSlowPromiseEvent - if event
* JitsiMediaDevicesEvents.USER_MEDIA_SLOW_PROMISE_TIMEOUT should be fired
* @param {Array} options.devices the devices that will be requested
* @param {string} options.resolution resolution constraints
* @param {string} options.cameraDeviceId
* @param {string} options.micDeviceId
*
* @returns {Promise.<{Array.}, JitsiConferenceError>} A promise
* that returns an array of created JitsiTracks if resolved, or a
* JitsiConferenceError if rejected.
*/
createLocalTracks(options: ICreateLocalTrackOptions = {}) {
let promiseFulfilled = false;
const { firePermissionPromptIsShownEvent, fireSlowPromiseEvent, ...restOptions } = options;
if (firePermissionPromptIsShownEvent && !RTC.arePermissionsGrantedForAvailableDevices()) {
// @ts-ignore
JitsiMediaDevices.emit(JitsiMediaDevicesEvents.PERMISSION_PROMPT_IS_SHOWN, browser.getName());
} else if (fireSlowPromiseEvent) {
window.setTimeout(() => {
if (!promiseFulfilled) {
JitsiMediaDevices.emit(JitsiMediaDevicesEvents.SLOW_GET_USER_MEDIA);
}
}, USER_MEDIA_SLOW_PROMISE_TIMEOUT);
}
let isFirstGUM = false;
let startTS = window.performance.now();
if (!window.connectionTimes) {
window.connectionTimes = {};
}
if (!hasGUMExecuted) {
hasGUMExecuted = true;
isFirstGUM = true;
window.connectionTimes['firstObtainPermissions.start'] = startTS;
}
window.connectionTimes['obtainPermissions.start'] = startTS;
return RTC.obtainAudioAndVideoPermissions(restOptions)
.then(tracks => {
promiseFulfilled = true;
let endTS = window.performance.now();
window.connectionTimes['obtainPermissions.end'] = endTS;
if (isFirstGUM) {
window.connectionTimes['firstObtainPermissions.end'] = endTS;
}
Statistics.sendAnalytics(
createGetUserMediaEvent(
'success',
getAnalyticsAttributesFromOptions(restOptions)));
if (this.isCollectingLocalStats()) {
for (let i = 0; i < tracks.length; i++) {
const track = tracks[i];
if (track.getType() === MediaType.AUDIO) {
Statistics.startLocalStats(track,
track.setAudioLevel.bind(track));
}
}
}
// set real device ids
const currentlyAvailableMediaDevices
= RTC.getCurrentlyAvailableMediaDevices();
if (currentlyAvailableMediaDevices) {
for (let i = 0; i < tracks.length; i++) {
const track = tracks[i];
track._setRealDeviceIdFromDeviceList(
currentlyAvailableMediaDevices);
}
}
return tracks;
})
.catch(error => {
promiseFulfilled = true;
if (error.name === JitsiTrackErrors.SCREENSHARING_USER_CANCELED) {
Statistics.sendAnalytics(
createGetUserMediaEvent(
'warning',
{
reason: 'extension install user canceled'
}));
} else if (error.name === JitsiTrackErrors.NOT_FOUND) {
const attributes
= getAnalyticsAttributesFromOptions(options);
attributes.reason = 'device not found';
attributes.devices = error.gum.devices.join('.');
Statistics.sendAnalytics(
createGetUserMediaEvent('error', attributes));
} else {
const attributes
= getAnalyticsAttributesFromOptions(options);
attributes.reason = error.name;
Statistics.sendAnalytics(
createGetUserMediaEvent('error', attributes));
}
let endTS = window.performance.now();
window.connectionTimes['obtainPermissions.end'] = endTS;
if (isFirstGUM) {
window.connectionTimes['firstObtainPermissions.end'] = endTS;
}
return Promise.reject(error);
});
},
/**
* Manually create JitsiLocalTrack's from the provided track info, by exposing the RTC method
*
* @param {Array} tracksInfo - array of track information
* @returns {Array} - created local tracks
*/
createLocalTracksFromMediaStreams(tracksInfo) {
return RTC.createLocalTracks(tracksInfo.map((trackInfo) => {
const tracks = trackInfo.stream.getTracks()
.filter(track => track.kind === trackInfo.mediaType);
if (!tracks || tracks.length === 0) {
throw new JitsiTrackError(JitsiTrackErrors.TRACK_NO_STREAM_TRACKS_FOUND, null, null);
}
if (tracks.length > 1) {
throw new JitsiTrackError(JitsiTrackErrors.TRACK_TOO_MANY_TRACKS_IN_STREAM, null, null);
}
trackInfo.track = tracks[0];
return trackInfo;
}));
},
/**
* Create a TrackVADEmitter service that connects an audio track to an VAD (voice activity detection) processor in
* order to obtain VAD scores for individual PCM audio samples.
* @param {string} localAudioDeviceId - The target local audio device.
* @param {number} sampleRate - Sample rate at which the emitter will operate. Possible values 256, 512, 1024,
* 4096, 8192, 16384. Passing other values will default to closes neighbor.
* I.e. Providing a value of 4096 means that the emitter will process 4096 PCM samples at a time, higher values mean
* longer calls, lowers values mean more calls but shorter.
* @param {Object} vadProcessor - VAD Processors that does the actual compute on a PCM sample.The processor needs
* to implement the following functions:
* - getSampleLength() - Returns the sample size accepted by calculateAudioFrameVAD.
* - getRequiredPCMFrequency() - Returns the PCM frequency at which the processor operates.
* i.e. (16KHz, 44.1 KHz etc.)
* - calculateAudioFrameVAD(pcmSample) - Process a 32 float pcm sample of getSampleLength size.
* @returns {Promise}
*/
createTrackVADEmitter(localAudioDeviceId, sampleRate, vadProcessor) {
return TrackVADEmitter.create(localAudioDeviceId, sampleRate, vadProcessor);
},
/**
* Create AudioMixer, which is essentially a wrapper over web audio ChannelMergerNode. It essentially allows the
* user to mix multiple MediaStreams into a single one.
*
* @returns {AudioMixer}
*/
createAudioMixer() {
return new AudioMixer();
},
/**
* Go through all audio devices on the system and return one that is active, i.e. has audio signal.
*
* @returns Promise