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  1. /* global __filename */
  2. import { getLogger } from 'jitsi-meet-logger';
  3. import BridgeChannel from './BridgeChannel';
  4. import GlobalOnErrorHandler from '../util/GlobalOnErrorHandler';
  5. import * as JitsiConferenceEvents from '../../JitsiConferenceEvents';
  6. import JitsiLocalTrack from './JitsiLocalTrack';
  7. import JitsiTrackError from '../../JitsiTrackError';
  8. import * as JitsiTrackErrors from '../../JitsiTrackErrors';
  9. import Listenable from '../util/Listenable';
  10. import { safeCounterIncrement } from '../util/MathUtil';
  11. import * as MediaType from '../../service/RTC/MediaType';
  12. import browser from '../browser';
  13. import RTCEvents from '../../service/RTC/RTCEvents';
  14. import RTCUtils from './RTCUtils';
  15. import Statistics from '../statistics/statistics';
  16. import TraceablePeerConnection from './TraceablePeerConnection';
  17. import VideoType from '../../service/RTC/VideoType';
  18. const logger = getLogger(__filename);
  19. /**
  20. * The counter used to generated id numbers assigned to peer connections
  21. * @type {number}
  22. */
  23. let peerConnectionIdCounter = 0;
  24. /**
  25. * The counter used to generate id number for the local
  26. * <code>MediaStreamTrack</code>s.
  27. * @type {number}
  28. */
  29. let rtcTrackIdCounter = 0;
  30. /**
  31. *
  32. * @param tracksInfo
  33. * @param options
  34. */
  35. function createLocalTracks(tracksInfo, options) {
  36. const newTracks = [];
  37. let deviceId = null;
  38. tracksInfo.forEach(trackInfo => {
  39. if (trackInfo.mediaType === MediaType.AUDIO) {
  40. deviceId = options.micDeviceId;
  41. } else if (trackInfo.videoType === VideoType.CAMERA) {
  42. deviceId = options.cameraDeviceId;
  43. }
  44. rtcTrackIdCounter = safeCounterIncrement(rtcTrackIdCounter);
  45. const localTrack = new JitsiLocalTrack({
  46. ...trackInfo,
  47. deviceId,
  48. facingMode: options.facingMode,
  49. rtcId: rtcTrackIdCounter
  50. });
  51. newTracks.push(localTrack);
  52. });
  53. return newTracks;
  54. }
  55. /**
  56. * Creates {@code JitsiLocalTrack} instances from the passed in meta information
  57. * about MedieaTracks.
  58. *
  59. * @param {Object[]} mediaStreamMetaData - An array of meta information with
  60. * MediaTrack instances. Each can look like:
  61. * {{
  62. * stream: MediaStream instance that holds a track with audio or video,
  63. * track: MediaTrack within the MediaStream,
  64. * videoType: "camera" or "desktop" or falsy,
  65. * sourceId: ID of the desktopsharing source,
  66. * sourceType: The desktopsharing source type
  67. * }}
  68. */
  69. function _newCreateLocalTracks(mediaStreamMetaData = []) {
  70. return mediaStreamMetaData.map(metaData => {
  71. const {
  72. sourceId,
  73. sourceType,
  74. stream,
  75. track,
  76. videoType
  77. } = metaData;
  78. const { deviceId, facingMode } = track.getSettings();
  79. // FIXME Move rtcTrackIdCounter to a static method in JitsiLocalTrack
  80. // so RTC does not need to handle ID management. This move would be
  81. // safer to do once the old createLocalTracks is removed.
  82. rtcTrackIdCounter = safeCounterIncrement(rtcTrackIdCounter);
  83. return new JitsiLocalTrack({
  84. deviceId,
  85. facingMode,
  86. mediaType: track.kind,
  87. rtcId: rtcTrackIdCounter,
  88. sourceId,
  89. sourceType,
  90. stream,
  91. track,
  92. videoType: videoType || null
  93. });
  94. });
  95. }
  96. /**
  97. *
  98. */
  99. export default class RTC extends Listenable {
  100. /**
  101. *
  102. * @param conference
  103. * @param options
  104. */
  105. constructor(conference, options = {}) {
  106. super();
  107. this.conference = conference;
  108. /**
  109. * A map of active <tt>TraceablePeerConnection</tt>.
  110. * @type {Map.<number, TraceablePeerConnection>}
  111. */
  112. this.peerConnections = new Map();
  113. this.localTracks = [];
  114. this.options = options;
  115. // BridgeChannel instance.
  116. // @private
  117. // @type {BridgeChannel}
  118. this._channel = null;
  119. // A flag whether we had received that the channel had opened we can
  120. // get this flag out of sync if for some reason channel got closed
  121. // from server, a desired behaviour so we can see errors when this
  122. // happen.
  123. // @private
  124. // @type {boolean}
  125. this._channelOpen = false;
  126. /**
  127. * The value specified to the last invocation of setLastN before the
  128. * channel completed opening. If non-null, the value will be sent
  129. * through a channel (once) as soon as it opens and will then be
  130. * discarded.
  131. * @private
  132. * @type {number}
  133. */
  134. this._lastN = -1;
  135. /**
  136. * Defines the last N endpoints list. It can be null or an array once
  137. * initialised with a channel last N event.
  138. * @type {Array<string>|null}
  139. * @private
  140. */
  141. this._lastNEndpoints = null;
  142. /**
  143. * The number representing the maximum video height the local client
  144. * should receive from the bridge.
  145. *
  146. * @type {number|undefined}
  147. * @private
  148. */
  149. this._maxFrameHeight = undefined;
  150. /**
  151. * The endpoint ID of currently pinned participant or <tt>null</tt> if
  152. * no user is pinned.
  153. * @type {string|null}
  154. * @private
  155. */
  156. this._pinnedEndpoint = null;
  157. /**
  158. * The endpoint IDs of currently selected participants.
  159. *
  160. * @type {Array}
  161. * @private
  162. */
  163. this._selectedEndpoints = [];
  164. // The last N change listener.
  165. this._lastNChangeListener = this._onLastNChanged.bind(this);
  166. // Switch audio output device on all remote audio tracks. Local audio
  167. // tracks handle this event by themselves.
  168. if (RTCUtils.isDeviceChangeAvailable('output')) {
  169. RTCUtils.addListener(RTCEvents.AUDIO_OUTPUT_DEVICE_CHANGED,
  170. deviceId => this._updateAudioOutputForAudioTracks(deviceId));
  171. RTCUtils.addListener(
  172. RTCEvents.DEVICE_LIST_CHANGED,
  173. () => this._updateAudioOutputForAudioTracks(
  174. RTCUtils.getAudioOutputDevice()));
  175. }
  176. }
  177. /**
  178. * Creates the local MediaStreams.
  179. * @param {object} [options] Optional parameters.
  180. * @param {array} options.devices The devices that will be requested.
  181. * @param {string} options.resolution Resolution constraints.
  182. * @param {bool} options.dontCreateJitsiTrack If <tt>true</tt> objects with
  183. * the following structure {stream: the Media Stream, type: "audio" or
  184. * "video", videoType: "camera" or "desktop"} will be returned trough
  185. * the Promise, otherwise JitsiTrack objects will be returned.
  186. * @param {string} options.cameraDeviceId
  187. * @param {string} options.micDeviceId
  188. * @returns {*} Promise object that will receive the new JitsiTracks
  189. */
  190. static obtainAudioAndVideoPermissions(options) {
  191. const usesNewGumFlow = browser.usesNewGumFlow();
  192. const obtainMediaPromise = usesNewGumFlow
  193. ? RTCUtils.newObtainAudioAndVideoPermissions(options)
  194. : RTCUtils.obtainAudioAndVideoPermissions(options);
  195. return obtainMediaPromise.then(
  196. tracksInfo => {
  197. const tracks = usesNewGumFlow
  198. ? _newCreateLocalTracks(tracksInfo)
  199. : createLocalTracks(tracksInfo, options);
  200. return tracks.some(track => !track._isReceivingData())
  201. ? Promise.reject(
  202. new JitsiTrackError(
  203. JitsiTrackErrors.NO_DATA_FROM_SOURCE))
  204. : tracks;
  205. });
  206. }
  207. /**
  208. * Initializes the bridge channel of this instance.
  209. * At least one of both, peerconnection or wsUrl parameters, must be
  210. * given.
  211. * @param {RTCPeerConnection} [peerconnection] WebRTC peer connection
  212. * instance.
  213. * @param {string} [wsUrl] WebSocket URL.
  214. */
  215. initializeBridgeChannel(peerconnection, wsUrl) {
  216. this._channel = new BridgeChannel(
  217. peerconnection, wsUrl, this.eventEmitter);
  218. this._channelOpenListener = () => {
  219. // Mark that channel as opened.
  220. this._channelOpen = true;
  221. // When the channel becomes available, tell the bridge about
  222. // video selections so that it can do adaptive simulcast,
  223. // we want the notification to trigger even if userJid
  224. // is undefined, or null.
  225. try {
  226. this._channel.sendPinnedEndpointMessage(
  227. this._pinnedEndpoint);
  228. this._channel.sendSelectedEndpointsMessage(
  229. this._selectedEndpoints);
  230. if (typeof this._maxFrameHeight !== 'undefined') {
  231. this._channel.sendReceiverVideoConstraintMessage(
  232. this._maxFrameHeight);
  233. }
  234. } catch (error) {
  235. GlobalOnErrorHandler.callErrorHandler(error);
  236. logger.error(
  237. `Cannot send selected(${this._selectedEndpoint})`
  238. + `pinned(${this._pinnedEndpoint})`
  239. + `frameHeight(${this._maxFrameHeight}) endpoint message`,
  240. error);
  241. }
  242. this.removeListener(RTCEvents.DATA_CHANNEL_OPEN,
  243. this._channelOpenListener);
  244. this._channelOpenListener = null;
  245. // If setLastN was invoked before the bridge channel completed
  246. // opening, apply the specified value now that the channel
  247. // is open. NOTE that -1 is the default value assumed by both
  248. // RTC module and the JVB.
  249. if (this._lastN !== -1) {
  250. this._channel.sendSetLastNMessage(this._lastN);
  251. }
  252. };
  253. this.addListener(RTCEvents.DATA_CHANNEL_OPEN,
  254. this._channelOpenListener);
  255. // Add Last N change listener.
  256. this.addListener(RTCEvents.LASTN_ENDPOINT_CHANGED,
  257. this._lastNChangeListener);
  258. }
  259. /**
  260. * Receives events when Last N had changed.
  261. * @param {array} lastNEndpoints The new Last N endpoints.
  262. * @private
  263. */
  264. _onLastNChanged(lastNEndpoints = []) {
  265. const oldLastNEndpoints = this._lastNEndpoints || [];
  266. let leavingLastNEndpoints = [];
  267. let enteringLastNEndpoints = [];
  268. this._lastNEndpoints = lastNEndpoints;
  269. leavingLastNEndpoints = oldLastNEndpoints.filter(
  270. id => !this.isInLastN(id));
  271. enteringLastNEndpoints = lastNEndpoints.filter(
  272. id => oldLastNEndpoints.indexOf(id) === -1);
  273. this.conference.eventEmitter.emit(
  274. JitsiConferenceEvents.LAST_N_ENDPOINTS_CHANGED,
  275. leavingLastNEndpoints,
  276. enteringLastNEndpoints);
  277. }
  278. /**
  279. * Should be called when current media session ends and after the
  280. * PeerConnection has been closed using PeerConnection.close() method.
  281. */
  282. onCallEnded() {
  283. if (this._channel) {
  284. // The BridgeChannel is not explicitly closed as the PeerConnection
  285. // is closed on call ended which triggers datachannel onclose
  286. // events. If using a WebSocket, the channel must be closed since
  287. // it is not managed by the PeerConnection.
  288. // The reference is cleared to disable any logic related to the
  289. // channel.
  290. if (this._channel && this._channel.mode === 'websocket') {
  291. this._channel.close();
  292. }
  293. this._channel = null;
  294. this._channelOpen = false;
  295. }
  296. }
  297. /**
  298. * Sets the maximum video size the local participant should receive from
  299. * remote participants. Will cache the value and send it through the channel
  300. * once it is created.
  301. *
  302. * @param {number} maxFrameHeightPixels the maximum frame height, in pixels,
  303. * this receiver is willing to receive.
  304. * @returns {void}
  305. */
  306. setReceiverVideoConstraint(maxFrameHeight) {
  307. this._maxFrameHeight = maxFrameHeight;
  308. if (this._channel && this._channelOpen) {
  309. this._channel.sendReceiverVideoConstraintMessage(maxFrameHeight);
  310. }
  311. }
  312. /**
  313. * Elects the participants with the given ids to be the selected
  314. * participants in order to always receive video for this participant (even
  315. * when last n is enabled). If there is no channel we store it and send it
  316. * through the channel once it is created.
  317. *
  318. * @param {Array<string>} ids - The user ids.
  319. * @throws NetworkError or InvalidStateError or Error if the operation
  320. * fails.
  321. * @returns {void}
  322. */
  323. selectEndpoints(ids) {
  324. this._selectedEndpoints = ids;
  325. if (this._channel && this._channelOpen) {
  326. this._channel.sendSelectedEndpointsMessage(ids);
  327. }
  328. }
  329. /**
  330. * Elects the participant with the given id to be the pinned participant in
  331. * order to always receive video for this participant (even when last n is
  332. * enabled).
  333. * @param {stirng} id The user id.
  334. * @throws NetworkError or InvalidStateError or Error if the operation
  335. * fails.
  336. */
  337. pinEndpoint(id) {
  338. // Cache the value if channel is missing, till we open it.
  339. this._pinnedEndpoint = id;
  340. if (this._channel && this._channelOpen) {
  341. this._channel.sendPinnedEndpointMessage(id);
  342. }
  343. }
  344. /**
  345. *
  346. * @param eventType
  347. * @param listener
  348. */
  349. static addListener(eventType, listener) {
  350. RTCUtils.addListener(eventType, listener);
  351. }
  352. /**
  353. *
  354. * @param eventType
  355. * @param listener
  356. */
  357. static removeListener(eventType, listener) {
  358. RTCUtils.removeListener(eventType, listener);
  359. }
  360. /**
  361. *
  362. * @param options
  363. */
  364. static init(options = {}) {
  365. this.options = options;
  366. return RTCUtils.init(this.options);
  367. }
  368. /**
  369. *
  370. */
  371. static getDeviceAvailability() {
  372. return RTCUtils.getDeviceAvailability();
  373. }
  374. /* eslint-disable max-params */
  375. /**
  376. * Creates new <tt>TraceablePeerConnection</tt>
  377. * @param {SignalingLayer} signaling The signaling layer that will
  378. * provide information about the media or participants which is not
  379. * carried over SDP.
  380. * @param {object} iceConfig An object describing the ICE config like
  381. * defined in the WebRTC specification.
  382. * @param {boolean} isP2P Indicates whether or not the new TPC will be used
  383. * in a peer to peer type of session.
  384. * @param {object} options The config options.
  385. * @param {boolean} options.disableSimulcast If set to 'true' will disable
  386. * the simulcast.
  387. * @param {boolean} options.disableRtx If set to 'true' will disable the
  388. * RTX.
  389. * @param {boolean} options.disableH264 If set to 'true' H264 will be
  390. * disabled by removing it from the SDP.
  391. * @param {boolean} options.preferH264 If set to 'true' H264 will be
  392. * preferred over other video codecs.
  393. * @return {TraceablePeerConnection}
  394. */
  395. createPeerConnection(signaling, iceConfig, isP2P, options) {
  396. const pcConstraints = RTC.getPCConstraints(isP2P);
  397. if (typeof options.abtestSuspendVideo !== 'undefined') {
  398. RTCUtils.setSuspendVideo(pcConstraints, options.abtestSuspendVideo);
  399. Statistics.analytics.addPermanentProperties(
  400. { abtestSuspendVideo: options.abtestSuspendVideo });
  401. }
  402. peerConnectionIdCounter = safeCounterIncrement(peerConnectionIdCounter);
  403. const newConnection
  404. = new TraceablePeerConnection(
  405. this,
  406. peerConnectionIdCounter,
  407. signaling,
  408. iceConfig, pcConstraints,
  409. isP2P, options);
  410. this.peerConnections.set(newConnection.id, newConnection);
  411. return newConnection;
  412. }
  413. /* eslint-enable max-params */
  414. /**
  415. * Removed given peer connection from this RTC module instance.
  416. * @param {TraceablePeerConnection} traceablePeerConnection
  417. * @return {boolean} <tt>true</tt> if the given peer connection was removed
  418. * successfully or <tt>false</tt> if there was no peer connection mapped in
  419. * this RTC instance.
  420. */
  421. _removePeerConnection(traceablePeerConnection) {
  422. const id = traceablePeerConnection.id;
  423. if (this.peerConnections.has(id)) {
  424. // NOTE Remote tracks are not removed here.
  425. this.peerConnections.delete(id);
  426. return true;
  427. }
  428. return false;
  429. }
  430. /**
  431. *
  432. * @param track
  433. */
  434. addLocalTrack(track) {
  435. if (!track) {
  436. throw new Error('track must not be null nor undefined');
  437. }
  438. this.localTracks.push(track);
  439. track.conference = this.conference;
  440. }
  441. /**
  442. * Returns the current value for "lastN" - the amount of videos are going
  443. * to be delivered. When set to -1 for unlimited or all available videos.
  444. * @return {number}
  445. */
  446. getLastN() {
  447. return this._lastN;
  448. }
  449. /**
  450. * Get local video track.
  451. * @returns {JitsiLocalTrack|undefined}
  452. */
  453. getLocalVideoTrack() {
  454. const localVideo = this.getLocalTracks(MediaType.VIDEO);
  455. return localVideo.length ? localVideo[0] : undefined;
  456. }
  457. /**
  458. * Get local audio track.
  459. * @returns {JitsiLocalTrack|undefined}
  460. */
  461. getLocalAudioTrack() {
  462. const localAudio = this.getLocalTracks(MediaType.AUDIO);
  463. return localAudio.length ? localAudio[0] : undefined;
  464. }
  465. /**
  466. * Returns the local tracks of the given media type, or all local tracks if
  467. * no specific type is given.
  468. * @param {MediaType} [mediaType] Optional media type filter.
  469. * (audio or video).
  470. */
  471. getLocalTracks(mediaType) {
  472. let tracks = this.localTracks.slice();
  473. if (mediaType !== undefined) {
  474. tracks = tracks.filter(
  475. track => track.getType() === mediaType);
  476. }
  477. return tracks;
  478. }
  479. /**
  480. * Obtains all remote tracks currently known to this RTC module instance.
  481. * @param {MediaType} [mediaType] The remote tracks will be filtered
  482. * by their media type if this argument is specified.
  483. * @return {Array<JitsiRemoteTrack>}
  484. */
  485. getRemoteTracks(mediaType) {
  486. let remoteTracks = [];
  487. for (const tpc of this.peerConnections.values()) {
  488. const pcRemoteTracks = tpc.getRemoteTracks(undefined, mediaType);
  489. if (pcRemoteTracks) {
  490. remoteTracks = remoteTracks.concat(pcRemoteTracks);
  491. }
  492. }
  493. return remoteTracks;
  494. }
  495. /**
  496. * Set mute for all local audio streams attached to the conference.
  497. * @param value The mute value.
  498. * @returns {Promise}
  499. */
  500. setAudioMute(value) {
  501. const mutePromises = [];
  502. this.getLocalTracks(MediaType.AUDIO).forEach(audioTrack => {
  503. // this is a Promise
  504. mutePromises.push(value ? audioTrack.mute() : audioTrack.unmute());
  505. });
  506. // We return a Promise from all Promises so we can wait for their
  507. // execution.
  508. return Promise.all(mutePromises);
  509. }
  510. /**
  511. *
  512. * @param track
  513. */
  514. removeLocalTrack(track) {
  515. const pos = this.localTracks.indexOf(track);
  516. if (pos === -1) {
  517. return;
  518. }
  519. this.localTracks.splice(pos, 1);
  520. }
  521. /**
  522. * Removes all JitsiRemoteTracks associated with given MUC nickname
  523. * (resource part of the JID). Returns array of removed tracks.
  524. *
  525. * @param {string} Owner The resource part of the MUC JID.
  526. * @returns {JitsiRemoteTrack[]}
  527. */
  528. removeRemoteTracks(owner) {
  529. let removedTracks = [];
  530. for (const tpc of this.peerConnections.values()) {
  531. const pcRemovedTracks = tpc.removeRemoteTracks(owner);
  532. removedTracks = removedTracks.concat(pcRemovedTracks);
  533. }
  534. logger.debug(
  535. `Removed remote tracks for ${owner}`
  536. + ` count: ${removedTracks.length}`);
  537. return removedTracks;
  538. }
  539. /**
  540. *
  541. */
  542. static getPCConstraints(isP2P) {
  543. const pcConstraints
  544. = isP2P ? RTCUtils.p2pPcConstraints : RTCUtils.pcConstraints;
  545. if (!pcConstraints) {
  546. return {};
  547. }
  548. return JSON.parse(JSON.stringify(pcConstraints));
  549. }
  550. /**
  551. *
  552. * @param elSelector
  553. * @param stream
  554. */
  555. static attachMediaStream(elSelector, stream) {
  556. return RTCUtils.attachMediaStream(elSelector, stream);
  557. }
  558. /**
  559. * Returns the id of the given stream.
  560. * @param {MediaStream} stream
  561. */
  562. static getStreamID(stream) {
  563. return RTCUtils.getStreamID(stream);
  564. }
  565. /**
  566. * Returns the id of the given track.
  567. * @param {MediaStreamTrack} track
  568. */
  569. static getTrackID(track) {
  570. return RTCUtils.getTrackID(track);
  571. }
  572. /**
  573. * Returns true if retrieving the the list of input devices is supported
  574. * and false if not.
  575. */
  576. static isDeviceListAvailable() {
  577. return RTCUtils.isDeviceListAvailable();
  578. }
  579. /**
  580. * Returns true if changing the input (camera / microphone) or output
  581. * (audio) device is supported and false if not.
  582. * @param {string} [deviceType] Type of device to change. Default is
  583. * undefined or 'input', 'output' - for audio output device change.
  584. * @returns {boolean} true if available, false otherwise.
  585. */
  586. static isDeviceChangeAvailable(deviceType) {
  587. return RTCUtils.isDeviceChangeAvailable(deviceType);
  588. }
  589. /**
  590. * Returns whether the current execution environment supports WebRTC (for
  591. * use within this library).
  592. *
  593. * @returns {boolean} {@code true} if WebRTC is supported in the current
  594. * execution environment (for use within this library); {@code false},
  595. * otherwise.
  596. */
  597. static isWebRtcSupported() {
  598. return browser.isSupported();
  599. }
  600. /**
  601. * Returns currently used audio output device id, '' stands for default
  602. * device
  603. * @returns {string}
  604. */
  605. static getAudioOutputDevice() {
  606. return RTCUtils.getAudioOutputDevice();
  607. }
  608. /**
  609. * Returns list of available media devices if its obtained, otherwise an
  610. * empty array is returned/
  611. * @returns {array} list of available media devices.
  612. */
  613. static getCurrentlyAvailableMediaDevices() {
  614. return RTCUtils.getCurrentlyAvailableMediaDevices();
  615. }
  616. /**
  617. * Returns event data for device to be reported to stats.
  618. * @returns {MediaDeviceInfo} device.
  619. */
  620. static getEventDataForActiveDevice(device) {
  621. return RTCUtils.getEventDataForActiveDevice(device);
  622. }
  623. /**
  624. * Sets current audio output device.
  625. * @param {string} deviceId Id of 'audiooutput' device from
  626. * navigator.mediaDevices.enumerateDevices().
  627. * @returns {Promise} resolves when audio output is changed, is rejected
  628. * otherwise
  629. */
  630. static setAudioOutputDevice(deviceId) {
  631. return RTCUtils.setAudioOutputDevice(deviceId);
  632. }
  633. /**
  634. * Returns <tt>true<tt/> if given WebRTC MediaStream is considered a valid
  635. * "user" stream which means that it's not a "receive only" stream nor a
  636. * "mixed" JVB stream.
  637. *
  638. * Clients that implement Unified Plan, such as Firefox use recvonly
  639. * "streams/channels/tracks" for receiving remote stream/tracks, as opposed
  640. * to Plan B where there are only 3 channels: audio, video and data.
  641. *
  642. * @param {MediaStream} stream The WebRTC MediaStream instance.
  643. * @returns {boolean}
  644. */
  645. static isUserStream(stream) {
  646. return RTC.isUserStreamById(RTCUtils.getStreamID(stream));
  647. }
  648. /**
  649. * Returns <tt>true<tt/> if a WebRTC MediaStream identified by given stream
  650. * ID is considered a valid "user" stream which means that it's not a
  651. * "receive only" stream nor a "mixed" JVB stream.
  652. *
  653. * Clients that implement Unified Plan, such as Firefox use recvonly
  654. * "streams/channels/tracks" for receiving remote stream/tracks, as opposed
  655. * to Plan B where there are only 3 channels: audio, video and data.
  656. *
  657. * @param {string} streamId The id of WebRTC MediaStream.
  658. * @returns {boolean}
  659. */
  660. static isUserStreamById(streamId) {
  661. return streamId && streamId !== 'mixedmslabel'
  662. && streamId !== 'default';
  663. }
  664. /**
  665. * Allows to receive list of available cameras/microphones.
  666. * @param {function} callback Would receive array of devices as an
  667. * argument.
  668. */
  669. static enumerateDevices(callback) {
  670. RTCUtils.enumerateDevices(callback);
  671. }
  672. /**
  673. * A method to handle stopping of the stream.
  674. * One point to handle the differences in various implementations.
  675. * @param {MediaStream} mediaStream MediaStream object to stop.
  676. */
  677. static stopMediaStream(mediaStream) {
  678. RTCUtils.stopMediaStream(mediaStream);
  679. }
  680. /**
  681. * Returns whether the desktop sharing is enabled or not.
  682. * @returns {boolean}
  683. */
  684. static isDesktopSharingEnabled() {
  685. return RTCUtils.isDesktopSharingEnabled();
  686. }
  687. /**
  688. * Closes the currently opened bridge channel.
  689. */
  690. closeBridgeChannel() {
  691. if (this._channel) {
  692. this._channel.close();
  693. this._channelOpen = false;
  694. this.removeListener(RTCEvents.LASTN_ENDPOINT_CHANGED,
  695. this._lastNChangeListener);
  696. }
  697. }
  698. /* eslint-disable max-params */
  699. /**
  700. *
  701. * @param {TraceablePeerConnection} tpc
  702. * @param {number} ssrc
  703. * @param {number} audioLevel
  704. * @param {boolean} isLocal
  705. */
  706. setAudioLevel(tpc, ssrc, audioLevel, isLocal) {
  707. const track = tpc.getTrackBySSRC(ssrc);
  708. if (!track) {
  709. return;
  710. } else if (!track.isAudioTrack()) {
  711. logger.warn(`Received audio level for non-audio track: ${ssrc}`);
  712. return;
  713. } else if (track.isLocal() !== isLocal) {
  714. logger.error(
  715. `${track} was expected to ${isLocal ? 'be' : 'not be'} local`);
  716. }
  717. track.setAudioLevel(audioLevel, tpc);
  718. }
  719. /* eslint-enable max-params */
  720. /**
  721. * Sends message via the bridge channel.
  722. * @param {string} to The id of the endpoint that should receive the
  723. * message. If "" the message will be sent to all participants.
  724. * @param {object} payload The payload of the message.
  725. * @throws NetworkError or InvalidStateError or Error if the operation
  726. * fails or there is no data channel created.
  727. */
  728. sendChannelMessage(to, payload) {
  729. if (this._channel) {
  730. this._channel.sendMessage(to, payload);
  731. } else {
  732. throw new Error('Channel support is disabled!');
  733. }
  734. }
  735. /**
  736. * Selects a new value for "lastN". The requested amount of videos are going
  737. * to be delivered after the value is in effect. Set to -1 for unlimited or
  738. * all available videos.
  739. * @param {number} value the new value for lastN.
  740. */
  741. setLastN(value) {
  742. if (this._lastN !== value) {
  743. this._lastN = value;
  744. if (this._channel && this._channelOpen) {
  745. this._channel.sendSetLastNMessage(value);
  746. }
  747. this.eventEmitter.emit(RTCEvents.LASTN_VALUE_CHANGED, value);
  748. }
  749. }
  750. /**
  751. * Indicates if the endpoint id is currently included in the last N.
  752. * @param {string} id The endpoint id that we check for last N.
  753. * @returns {boolean} true if the endpoint id is in the last N or if we
  754. * don't have bridge channel support, otherwise we return false.
  755. */
  756. isInLastN(id) {
  757. return !this._lastNEndpoints // lastNEndpoints not initialised yet.
  758. || this._lastNEndpoints.indexOf(id) > -1;
  759. }
  760. /**
  761. * Updates the target audio output device for all remote audio tracks.
  762. *
  763. * @param {string} deviceId - The device id of the audio ouput device to
  764. * use for all remote tracks.
  765. * @private
  766. * @returns {void}
  767. */
  768. _updateAudioOutputForAudioTracks(deviceId) {
  769. const remoteAudioTracks = this.getRemoteTracks(MediaType.AUDIO);
  770. for (const track of remoteAudioTracks) {
  771. track.setAudioOutput(deviceId);
  772. }
  773. }
  774. }