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RTC.js 27KB

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