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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. * Exposes the private helper for converting a WebRTC MediaStream to a
  179. * JitsiLocalTrack.
  180. *
  181. * @param {Array<Object>} tracksInfo
  182. * @returns {Array<JitsiLocalTrack>}
  183. */
  184. static newCreateLocalTracks(tracksInfo) {
  185. return _newCreateLocalTracks(tracksInfo);
  186. }
  187. /**
  188. * Creates the local MediaStreams.
  189. * @param {object} [options] Optional parameters.
  190. * @param {array} options.devices The devices that will be requested.
  191. * @param {string} options.resolution Resolution constraints.
  192. * @param {string} options.cameraDeviceId
  193. * @param {string} options.micDeviceId
  194. * @returns {*} Promise object that will receive the new JitsiTracks
  195. */
  196. static obtainAudioAndVideoPermissions(options) {
  197. const usesNewGumFlow = browser.usesNewGumFlow();
  198. const obtainMediaPromise = usesNewGumFlow
  199. ? RTCUtils.newObtainAudioAndVideoPermissions(options)
  200. : RTCUtils.obtainAudioAndVideoPermissions(options);
  201. return obtainMediaPromise.then(
  202. tracksInfo => {
  203. const tracks = usesNewGumFlow
  204. ? _newCreateLocalTracks(tracksInfo)
  205. : createLocalTracks(tracksInfo, options);
  206. return tracks.some(track => !track._isReceivingData())
  207. ? Promise.reject(
  208. new JitsiTrackError(
  209. JitsiTrackErrors.NO_DATA_FROM_SOURCE))
  210. : tracks;
  211. });
  212. }
  213. /**
  214. * Initializes the bridge channel of this instance.
  215. * At least one of both, peerconnection or wsUrl parameters, must be
  216. * given.
  217. * @param {RTCPeerConnection} [peerconnection] WebRTC peer connection
  218. * instance.
  219. * @param {string} [wsUrl] WebSocket URL.
  220. */
  221. initializeBridgeChannel(peerconnection, wsUrl) {
  222. this._channel = new BridgeChannel(
  223. peerconnection, wsUrl, this.eventEmitter);
  224. this._channelOpenListener = () => {
  225. // Mark that channel as opened.
  226. this._channelOpen = true;
  227. // When the channel becomes available, tell the bridge about
  228. // video selections so that it can do adaptive simulcast,
  229. // we want the notification to trigger even if userJid
  230. // is undefined, or null.
  231. try {
  232. this._channel.sendPinnedEndpointMessage(
  233. this._pinnedEndpoint);
  234. this._channel.sendSelectedEndpointsMessage(
  235. this._selectedEndpoints);
  236. if (typeof this._maxFrameHeight !== 'undefined') {
  237. this._channel.sendReceiverVideoConstraintMessage(
  238. this._maxFrameHeight);
  239. }
  240. } catch (error) {
  241. GlobalOnErrorHandler.callErrorHandler(error);
  242. logger.error(
  243. `Cannot send selected(${this._selectedEndpoint})`
  244. + `pinned(${this._pinnedEndpoint})`
  245. + `frameHeight(${this._maxFrameHeight}) endpoint message`,
  246. error);
  247. }
  248. this.removeListener(RTCEvents.DATA_CHANNEL_OPEN,
  249. this._channelOpenListener);
  250. this._channelOpenListener = null;
  251. // If setLastN was invoked before the bridge channel completed
  252. // opening, apply the specified value now that the channel
  253. // is open. NOTE that -1 is the default value assumed by both
  254. // RTC module and the JVB.
  255. if (this._lastN !== -1) {
  256. this._channel.sendSetLastNMessage(this._lastN);
  257. }
  258. };
  259. this.addListener(RTCEvents.DATA_CHANNEL_OPEN,
  260. this._channelOpenListener);
  261. // Add Last N change listener.
  262. this.addListener(RTCEvents.LASTN_ENDPOINT_CHANGED,
  263. this._lastNChangeListener);
  264. }
  265. /**
  266. * Receives events when Last N had changed.
  267. * @param {array} lastNEndpoints The new Last N endpoints.
  268. * @private
  269. */
  270. _onLastNChanged(lastNEndpoints = []) {
  271. const oldLastNEndpoints = this._lastNEndpoints || [];
  272. let leavingLastNEndpoints = [];
  273. let enteringLastNEndpoints = [];
  274. this._lastNEndpoints = lastNEndpoints;
  275. leavingLastNEndpoints = oldLastNEndpoints.filter(
  276. id => !this.isInLastN(id));
  277. enteringLastNEndpoints = lastNEndpoints.filter(
  278. id => oldLastNEndpoints.indexOf(id) === -1);
  279. this.conference.eventEmitter.emit(
  280. JitsiConferenceEvents.LAST_N_ENDPOINTS_CHANGED,
  281. leavingLastNEndpoints,
  282. enteringLastNEndpoints);
  283. }
  284. /**
  285. * Should be called when current media session ends and after the
  286. * PeerConnection has been closed using PeerConnection.close() method.
  287. */
  288. onCallEnded() {
  289. if (this._channel) {
  290. // The BridgeChannel is not explicitly closed as the PeerConnection
  291. // is closed on call ended which triggers datachannel onclose
  292. // events. If using a WebSocket, the channel must be closed since
  293. // it is not managed by the PeerConnection.
  294. // The reference is cleared to disable any logic related to the
  295. // channel.
  296. if (this._channel && this._channel.mode === 'websocket') {
  297. this._channel.close();
  298. }
  299. this._channel = null;
  300. this._channelOpen = false;
  301. }
  302. }
  303. /**
  304. * Sets the maximum video size the local participant should receive from
  305. * remote participants. Will cache the value and send it through the channel
  306. * once it is created.
  307. *
  308. * @param {number} maxFrameHeightPixels the maximum frame height, in pixels,
  309. * this receiver is willing to receive.
  310. * @returns {void}
  311. */
  312. setReceiverVideoConstraint(maxFrameHeight) {
  313. this._maxFrameHeight = maxFrameHeight;
  314. if (this._channel && this._channelOpen) {
  315. this._channel.sendReceiverVideoConstraintMessage(maxFrameHeight);
  316. }
  317. }
  318. /**
  319. * Elects the participants with the given ids to be the selected
  320. * participants in order to always receive video for this participant (even
  321. * when last n is enabled). If there is no channel we store it and send it
  322. * through the channel once it is created.
  323. *
  324. * @param {Array<string>} ids - The user ids.
  325. * @throws NetworkError or InvalidStateError or Error if the operation
  326. * fails.
  327. * @returns {void}
  328. */
  329. selectEndpoints(ids) {
  330. this._selectedEndpoints = ids;
  331. if (this._channel && this._channelOpen) {
  332. this._channel.sendSelectedEndpointsMessage(ids);
  333. }
  334. }
  335. /**
  336. * Elects the participant with the given id to be the pinned participant in
  337. * order to always receive video for this participant (even when last n is
  338. * enabled).
  339. * @param {stirng} id The user id.
  340. * @throws NetworkError or InvalidStateError or Error if the operation
  341. * fails.
  342. */
  343. pinEndpoint(id) {
  344. // Cache the value if channel is missing, till we open it.
  345. this._pinnedEndpoint = id;
  346. if (this._channel && this._channelOpen) {
  347. this._channel.sendPinnedEndpointMessage(id);
  348. }
  349. }
  350. /**
  351. *
  352. * @param eventType
  353. * @param listener
  354. */
  355. static addListener(eventType, listener) {
  356. RTCUtils.addListener(eventType, listener);
  357. }
  358. /**
  359. *
  360. * @param eventType
  361. * @param listener
  362. */
  363. static removeListener(eventType, listener) {
  364. RTCUtils.removeListener(eventType, listener);
  365. }
  366. /**
  367. *
  368. * @param options
  369. */
  370. static init(options = {}) {
  371. this.options = options;
  372. return RTCUtils.init(this.options);
  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. // FIXME: We should rename iceConfig to pcConfig.
  403. if (browser.supportsSdpSemantics()) {
  404. iceConfig.sdpSemantics = 'plan-b';
  405. }
  406. peerConnectionIdCounter = safeCounterIncrement(peerConnectionIdCounter);
  407. const newConnection
  408. = new TraceablePeerConnection(
  409. this,
  410. peerConnectionIdCounter,
  411. signaling,
  412. iceConfig, pcConstraints,
  413. isP2P, options);
  414. this.peerConnections.set(newConnection.id, newConnection);
  415. return newConnection;
  416. }
  417. /* eslint-enable max-params */
  418. /**
  419. * Removed given peer connection from this RTC module instance.
  420. * @param {TraceablePeerConnection} traceablePeerConnection
  421. * @return {boolean} <tt>true</tt> if the given peer connection was removed
  422. * successfully or <tt>false</tt> if there was no peer connection mapped in
  423. * this RTC instance.
  424. */
  425. _removePeerConnection(traceablePeerConnection) {
  426. const id = traceablePeerConnection.id;
  427. if (this.peerConnections.has(id)) {
  428. // NOTE Remote tracks are not removed here.
  429. this.peerConnections.delete(id);
  430. return true;
  431. }
  432. return false;
  433. }
  434. /**
  435. *
  436. * @param track
  437. */
  438. addLocalTrack(track) {
  439. if (!track) {
  440. throw new Error('track must not be null nor undefined');
  441. }
  442. this.localTracks.push(track);
  443. track.conference = this.conference;
  444. }
  445. /**
  446. * Returns the current value for "lastN" - the amount of videos are going
  447. * to be delivered. When set to -1 for unlimited or all available videos.
  448. * @return {number}
  449. */
  450. getLastN() {
  451. return this._lastN;
  452. }
  453. /**
  454. * Get local video track.
  455. * @returns {JitsiLocalTrack|undefined}
  456. */
  457. getLocalVideoTrack() {
  458. const localVideo = this.getLocalTracks(MediaType.VIDEO);
  459. return localVideo.length ? localVideo[0] : undefined;
  460. }
  461. /**
  462. * Get local audio track.
  463. * @returns {JitsiLocalTrack|undefined}
  464. */
  465. getLocalAudioTrack() {
  466. const localAudio = this.getLocalTracks(MediaType.AUDIO);
  467. return localAudio.length ? localAudio[0] : undefined;
  468. }
  469. /**
  470. * Returns the local tracks of the given media type, or all local tracks if
  471. * no specific type is given.
  472. * @param {MediaType} [mediaType] Optional media type filter.
  473. * (audio or video).
  474. */
  475. getLocalTracks(mediaType) {
  476. let tracks = this.localTracks.slice();
  477. if (mediaType !== undefined) {
  478. tracks = tracks.filter(
  479. track => track.getType() === mediaType);
  480. }
  481. return tracks;
  482. }
  483. /**
  484. * Obtains all remote tracks currently known to this RTC module instance.
  485. * @param {MediaType} [mediaType] The remote tracks will be filtered
  486. * by their media type if this argument is specified.
  487. * @return {Array<JitsiRemoteTrack>}
  488. */
  489. getRemoteTracks(mediaType) {
  490. let remoteTracks = [];
  491. for (const tpc of this.peerConnections.values()) {
  492. const pcRemoteTracks = tpc.getRemoteTracks(undefined, mediaType);
  493. if (pcRemoteTracks) {
  494. remoteTracks = remoteTracks.concat(pcRemoteTracks);
  495. }
  496. }
  497. return remoteTracks;
  498. }
  499. /**
  500. * Set mute for all local audio streams attached to the conference.
  501. * @param value The mute value.
  502. * @returns {Promise}
  503. */
  504. setAudioMute(value) {
  505. const mutePromises = [];
  506. this.getLocalTracks(MediaType.AUDIO).forEach(audioTrack => {
  507. // this is a Promise
  508. mutePromises.push(value ? audioTrack.mute() : audioTrack.unmute());
  509. });
  510. // We return a Promise from all Promises so we can wait for their
  511. // execution.
  512. return Promise.all(mutePromises);
  513. }
  514. /**
  515. *
  516. * @param track
  517. */
  518. removeLocalTrack(track) {
  519. const pos = this.localTracks.indexOf(track);
  520. if (pos === -1) {
  521. return;
  522. }
  523. this.localTracks.splice(pos, 1);
  524. }
  525. /**
  526. * Removes all JitsiRemoteTracks associated with given MUC nickname
  527. * (resource part of the JID). Returns array of removed tracks.
  528. *
  529. * @param {string} Owner The resource part of the MUC JID.
  530. * @returns {JitsiRemoteTrack[]}
  531. */
  532. removeRemoteTracks(owner) {
  533. let removedTracks = [];
  534. for (const tpc of this.peerConnections.values()) {
  535. const pcRemovedTracks = tpc.removeRemoteTracks(owner);
  536. removedTracks = removedTracks.concat(pcRemovedTracks);
  537. }
  538. logger.debug(
  539. `Removed remote tracks for ${owner}`
  540. + ` count: ${removedTracks.length}`);
  541. return removedTracks;
  542. }
  543. /**
  544. *
  545. */
  546. static getPCConstraints(isP2P) {
  547. const pcConstraints
  548. = isP2P ? RTCUtils.p2pPcConstraints : RTCUtils.pcConstraints;
  549. if (!pcConstraints) {
  550. return {};
  551. }
  552. return JSON.parse(JSON.stringify(pcConstraints));
  553. }
  554. /**
  555. *
  556. * @param elSelector
  557. * @param stream
  558. */
  559. static attachMediaStream(elSelector, stream) {
  560. return RTCUtils.attachMediaStream(elSelector, stream);
  561. }
  562. /**
  563. * Returns the id of the given stream.
  564. * @param {MediaStream} stream
  565. */
  566. static getStreamID(stream) {
  567. return RTCUtils.getStreamID(stream);
  568. }
  569. /**
  570. * Returns the id of the given track.
  571. * @param {MediaStreamTrack} track
  572. */
  573. static getTrackID(track) {
  574. return RTCUtils.getTrackID(track);
  575. }
  576. /**
  577. * Returns true if retrieving the the list of input devices is supported
  578. * and false if not.
  579. */
  580. static isDeviceListAvailable() {
  581. return RTCUtils.isDeviceListAvailable();
  582. }
  583. /**
  584. * Returns true if changing the input (camera / microphone) or output
  585. * (audio) device is supported and false if not.
  586. * @param {string} [deviceType] Type of device to change. Default is
  587. * undefined or 'input', 'output' - for audio output device change.
  588. * @returns {boolean} true if available, false otherwise.
  589. */
  590. static isDeviceChangeAvailable(deviceType) {
  591. return RTCUtils.isDeviceChangeAvailable(deviceType);
  592. }
  593. /**
  594. * Returns whether the current execution environment supports WebRTC (for
  595. * use within this library).
  596. *
  597. * @returns {boolean} {@code true} if WebRTC is supported in the current
  598. * execution environment (for use within this library); {@code false},
  599. * otherwise.
  600. */
  601. static isWebRtcSupported() {
  602. return browser.isSupported();
  603. }
  604. /**
  605. * Returns currently used audio output device id, '' stands for default
  606. * device
  607. * @returns {string}
  608. */
  609. static getAudioOutputDevice() {
  610. return RTCUtils.getAudioOutputDevice();
  611. }
  612. /**
  613. * Returns list of available media devices if its obtained, otherwise an
  614. * empty array is returned/
  615. * @returns {array} list of available media devices.
  616. */
  617. static getCurrentlyAvailableMediaDevices() {
  618. return RTCUtils.getCurrentlyAvailableMediaDevices();
  619. }
  620. /**
  621. * Returns event data for device to be reported to stats.
  622. * @returns {MediaDeviceInfo} device.
  623. */
  624. static getEventDataForActiveDevice(device) {
  625. return RTCUtils.getEventDataForActiveDevice(device);
  626. }
  627. /**
  628. * Sets current audio output device.
  629. * @param {string} deviceId Id of 'audiooutput' device from
  630. * navigator.mediaDevices.enumerateDevices().
  631. * @returns {Promise} resolves when audio output is changed, is rejected
  632. * otherwise
  633. */
  634. static setAudioOutputDevice(deviceId) {
  635. return RTCUtils.setAudioOutputDevice(deviceId);
  636. }
  637. /**
  638. * Returns <tt>true<tt/> if given WebRTC MediaStream is considered a valid
  639. * "user" stream which means that it's not a "receive only" stream nor a
  640. * "mixed" JVB stream.
  641. *
  642. * Clients that implement Unified Plan, such as Firefox use recvonly
  643. * "streams/channels/tracks" for receiving remote stream/tracks, as opposed
  644. * to Plan B where there are only 3 channels: audio, video and data.
  645. *
  646. * @param {MediaStream} stream The WebRTC MediaStream instance.
  647. * @returns {boolean}
  648. */
  649. static isUserStream(stream) {
  650. return RTC.isUserStreamById(RTCUtils.getStreamID(stream));
  651. }
  652. /**
  653. * Returns <tt>true<tt/> if a WebRTC MediaStream identified by given stream
  654. * ID is considered a valid "user" stream which means that it's not a
  655. * "receive only" stream nor a "mixed" JVB stream.
  656. *
  657. * Clients that implement Unified Plan, such as Firefox use recvonly
  658. * "streams/channels/tracks" for receiving remote stream/tracks, as opposed
  659. * to Plan B where there are only 3 channels: audio, video and data.
  660. *
  661. * @param {string} streamId The id of WebRTC MediaStream.
  662. * @returns {boolean}
  663. */
  664. static isUserStreamById(streamId) {
  665. return streamId && streamId !== 'mixedmslabel'
  666. && streamId !== 'default';
  667. }
  668. /**
  669. * Allows to receive list of available cameras/microphones.
  670. * @param {function} callback Would receive array of devices as an
  671. * argument.
  672. */
  673. static enumerateDevices(callback) {
  674. RTCUtils.enumerateDevices(callback);
  675. }
  676. /**
  677. * A method to handle stopping of the stream.
  678. * One point to handle the differences in various implementations.
  679. * @param {MediaStream} mediaStream MediaStream object to stop.
  680. */
  681. static stopMediaStream(mediaStream) {
  682. RTCUtils.stopMediaStream(mediaStream);
  683. }
  684. /**
  685. * Returns whether the desktop sharing is enabled or not.
  686. * @returns {boolean}
  687. */
  688. static isDesktopSharingEnabled() {
  689. return RTCUtils.isDesktopSharingEnabled();
  690. }
  691. /**
  692. * Closes the currently opened bridge channel.
  693. */
  694. closeBridgeChannel() {
  695. if (this._channel) {
  696. this._channel.close();
  697. this._channelOpen = false;
  698. this.removeListener(RTCEvents.LASTN_ENDPOINT_CHANGED,
  699. this._lastNChangeListener);
  700. }
  701. }
  702. /* eslint-disable max-params */
  703. /**
  704. *
  705. * @param {TraceablePeerConnection} tpc
  706. * @param {number} ssrc
  707. * @param {number} audioLevel
  708. * @param {boolean} isLocal
  709. */
  710. setAudioLevel(tpc, ssrc, audioLevel, isLocal) {
  711. const track = tpc.getTrackBySSRC(ssrc);
  712. if (!track) {
  713. return;
  714. } else if (!track.isAudioTrack()) {
  715. logger.warn(`Received audio level for non-audio track: ${ssrc}`);
  716. return;
  717. } else if (track.isLocal() !== isLocal) {
  718. logger.error(
  719. `${track} was expected to ${isLocal ? 'be' : 'not be'} local`);
  720. }
  721. track.setAudioLevel(audioLevel, tpc);
  722. }
  723. /* eslint-enable max-params */
  724. /**
  725. * Sends message via the bridge channel.
  726. * @param {string} to The id of the endpoint that should receive the
  727. * message. If "" the message will be sent to all participants.
  728. * @param {object} payload The payload of the message.
  729. * @throws NetworkError or InvalidStateError or Error if the operation
  730. * fails or there is no data channel created.
  731. */
  732. sendChannelMessage(to, payload) {
  733. if (this._channel) {
  734. this._channel.sendMessage(to, payload);
  735. } else {
  736. throw new Error('Channel support is disabled!');
  737. }
  738. }
  739. /**
  740. * Selects a new value for "lastN". The requested amount of videos are going
  741. * to be delivered after the value is in effect. Set to -1 for unlimited or
  742. * all available videos.
  743. * @param {number} value the new value for lastN.
  744. */
  745. setLastN(value) {
  746. if (this._lastN !== value) {
  747. this._lastN = value;
  748. if (this._channel && this._channelOpen) {
  749. this._channel.sendSetLastNMessage(value);
  750. }
  751. this.eventEmitter.emit(RTCEvents.LASTN_VALUE_CHANGED, value);
  752. }
  753. }
  754. /**
  755. * Indicates if the endpoint id is currently included in the last N.
  756. * @param {string} id The endpoint id that we check for last N.
  757. * @returns {boolean} true if the endpoint id is in the last N or if we
  758. * don't have bridge channel support, otherwise we return false.
  759. */
  760. isInLastN(id) {
  761. return !this._lastNEndpoints // lastNEndpoints not initialised yet.
  762. || this._lastNEndpoints.indexOf(id) > -1;
  763. }
  764. /**
  765. * Updates the target audio output device for all remote audio tracks.
  766. *
  767. * @param {string} deviceId - The device id of the audio ouput device to
  768. * use for all remote tracks.
  769. * @private
  770. * @returns {void}
  771. */
  772. _updateAudioOutputForAudioTracks(deviceId) {
  773. const remoteAudioTracks = this.getRemoteTracks(MediaType.AUDIO);
  774. for (const track of remoteAudioTracks) {
  775. track.setAudioOutput(deviceId);
  776. }
  777. }
  778. }