@abrar71/lib-jitsi-meet 0.0.0

package/dist/esm/modules/RTC/RTC.js

@@ -0,0 +1,658 @@
+import { getLogger } from '@jitsi/logger';
+import { cloneDeep, isEqual } from 'lodash-es';
+import { JitsiConferenceEvents } from '../../JitsiConferenceEvents';
+import { MediaType } from '../../service/RTC/MediaType';
+import { RTCEvents } from '../../service/RTC/RTCEvents';
+import { VideoType } from '../../service/RTC/VideoType';
+import browser from '../browser';
+import Listenable from '../util/Listenable';
+import { safeCounterIncrement } from '../util/MathUtil';
+import BridgeChannel from './BridgeChannel';
+import JitsiLocalTrack from './JitsiLocalTrack';
+import RTCUtils from './RTCUtils';
+import TraceablePeerConnection from './TraceablePeerConnection';
+const logger = getLogger('rtc:RTC');
+/**
+ * The counter used to generated id numbers assigned to peer connections
+ * @type {number}
+ */
+let peerConnectionIdCounter = 0;
+/**
+ * The counter used to generate id number for the local
+ * <code>MediaStreamTrack</code>s.
+ * @type {number}
+ */
+let rtcTrackIdCounter = 0;
+/**
+ * Creates {@code JitsiLocalTrack} instances from the passed in meta information
+ * about MedieaTracks.
+ *
+ * @param {Object[]} mediaStreamMetaData - An array of meta information with
+ * MediaTrack instances. Each can look like:
+ * {{
+ *     stream: MediaStream instance that holds a track with audio or video,
+ *     track: MediaTrack within the MediaStream,
+ *     videoType: "camera" or "desktop" or falsy,
+ *     sourceId: ID of the desktopsharing source,
+ *     sourceType: The desktopsharing source type,
+ *     effects: Array of effect types
+ * }}
+ */
+function _createLocalTracks(mediaStreamMetaData = []) {
+    return mediaStreamMetaData.map(metaData => {
+        const { constraints, sourceId, sourceType, stream, track, videoType, effects } = metaData;
+        const { deviceId, facingMode } = track.getSettings();
+        // FIXME Move rtcTrackIdCounter to a static method in JitsiLocalTrack
+        // so RTC does not need to handle ID management. This move would be
+        // safer to do once the old createLocalTracks is removed.
+        rtcTrackIdCounter = safeCounterIncrement(rtcTrackIdCounter);
+        return new JitsiLocalTrack({
+            constraints,
+            deviceId,
+            effects,
+            facingMode,
+            mediaType: track.kind,
+            rtcId: rtcTrackIdCounter,
+            sourceId,
+            sourceType,
+            stream,
+            track,
+            videoType: videoType || null
+        });
+    });
+}
+/**
+ *
+ */
+export default class RTC extends Listenable {
+    /**
+     *
+     * @param conference
+     * @param options
+     */
+    constructor(conference, options = {}) {
+        super();
+        this.conference = conference;
+        /**
+         * A map of active <tt>TraceablePeerConnection</tt>.
+         * @type {Map.<number, TraceablePeerConnection>}
+         */
+        this.peerConnections = new Map();
+        this.localTracks = [];
+        this.options = options;
+        // BridgeChannel instance.
+        // @private
+        // @type {BridgeChannel}
+        this._channel = null;
+        /**
+         * The value specified to the last invocation of setLastN before the
+         * channel completed opening. If non-null, the value will be sent
+         * through a channel (once) as soon as it opens and will then be
+         * discarded.
+         * @private
+         * @type {number}
+         */
+        this._lastN = undefined;
+        /**
+         * Defines the forwarded sources list. It can be null or an array once initialised with a channel forwarded
+         * sources event.
+         *
+         * @type {Array<string>|null}
+         * @private
+         */
+        this._forwardedSources = null;
+        // The forwarded sources change listener.
+        this._forwardedSourcesChangeListener = this._onForwardedSourcesChanged.bind(this);
+        this._onDeviceListChanged = this._onDeviceListChanged.bind(this);
+        this._updateAudioOutputForAudioTracks = this._updateAudioOutputForAudioTracks.bind(this);
+        // Switch audio output device on all remote audio tracks. Local audio
+        // tracks handle this event by themselves.
+        if (RTCUtils.isDeviceChangeAvailable('output')) {
+            RTCUtils.addListener(RTCEvents.AUDIO_OUTPUT_DEVICE_CHANGED, this._updateAudioOutputForAudioTracks);
+            RTCUtils.addListener(RTCEvents.DEVICE_LIST_CHANGED, this._onDeviceListChanged);
+        }
+    }
+    /**
+     * Removes any listeners and stored state from this {@code RTC} instance.
+     *
+     * @returns {void}
+     */
+    destroy() {
+        RTCUtils.removeListener(RTCEvents.AUDIO_OUTPUT_DEVICE_CHANGED, this._updateAudioOutputForAudioTracks);
+        RTCUtils.removeListener(RTCEvents.DEVICE_LIST_CHANGED, this._onDeviceListChanged);
+        if (this._channelOpenListener) {
+            this.removeListener(RTCEvents.DATA_CHANNEL_OPEN, this._channelOpenListener);
+        }
+    }
+    /**
+     * Exposes the private helper for converting a WebRTC MediaStream to a
+     * JitsiLocalTrack.
+     *
+     * @param {Array<Object>} tracksInfo
+     * @returns {Array<JitsiLocalTrack>}
+     */
+    static createLocalTracks(tracksInfo) {
+        return _createLocalTracks(tracksInfo);
+    }
+    /**
+     * Creates the local MediaStreams.
+     * @param {object} [options] Optional parameters.
+     * @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 object that will receive the new JitsiTracks
+     */
+    static obtainAudioAndVideoPermissions(options) {
+        return RTCUtils.obtainAudioAndVideoPermissions(options)
+            .then(tracksInfo => _createLocalTracks(tracksInfo));
+    }
+    /**
+     * Initializes the bridge channel of this instance.
+     * At least one of both, peerconnection or wsUrl parameters, must be
+     * given.
+     * @param {RTCPeerConnection} [peerconnection] WebRTC peer connection
+     * instance.
+     * @param {string} [wsUrl] WebSocket URL.
+     */
+    initializeBridgeChannel(peerconnection, wsUrl) {
+        this._channel = new BridgeChannel(peerconnection, wsUrl, this.eventEmitter, this.conference);
+        this._channelOpenListener = () => {
+            const logError = (error, msgType, value) => {
+                logger.error(`Cannot send ${msgType}(${JSON.stringify(value)}) endpoint message`, error);
+            };
+            // When the channel becomes available, tell the bridge about video selections so that it can do adaptive
+            // simulcast, we want the notification to trigger even if userJid is undefined, or null.
+            if (this._receiverVideoConstraints) {
+                try {
+                    this._channel.sendReceiverVideoConstraintsMessage(this._receiverVideoConstraints);
+                }
+                catch (error) {
+                    logError(error, 'ReceiverVideoConstraints', this._receiverVideoConstraints);
+                }
+            }
+            if (typeof this._lastN !== 'undefined' && this._lastN !== -1) {
+                try {
+                    this._channel.sendSetLastNMessage(this._lastN);
+                }
+                catch (error) {
+                    logError(error, 'LastNChangedEvent', this._lastN);
+                }
+            }
+        };
+        this.addListener(RTCEvents.DATA_CHANNEL_OPEN, this._channelOpenListener);
+        // Add forwarded sources change listener.
+        this.addListener(RTCEvents.FORWARDED_SOURCES_CHANGED, this._forwardedSourcesChangeListener);
+    }
+    /**
+     * Callback invoked when the list of known audio and video devices has
+     * been updated. Attempts to update the known available audio output
+     * devices.
+     *
+     * @private
+     * @returns {void}
+     */
+    _onDeviceListChanged() {
+        this._updateAudioOutputForAudioTracks(RTCUtils.getAudioOutputDevice());
+    }
+    /**
+     * Receives events when forwarded sources had changed.
+     *
+     * @param {array} forwardedSources The new forwarded sources.
+     * @private
+     */
+    _onForwardedSourcesChanged(forwardedSources = []) {
+        const oldForwardedSources = this._forwardedSources || [];
+        let leavingForwardedSources = [];
+        let enteringForwardedSources = [];
+        const timestamp = Date.now();
+        this._forwardedSources = forwardedSources;
+        leavingForwardedSources = oldForwardedSources.filter(sourceName => !this.isInForwardedSources(sourceName));
+        enteringForwardedSources = forwardedSources.filter(sourceName => oldForwardedSources.indexOf(sourceName) === -1);
+        logger.debug(`Fowarded sources changed leaving=${leavingForwardedSources}, entering=`
+            + `${enteringForwardedSources} at ${timestamp}`);
+        this.conference.eventEmitter.emit(JitsiConferenceEvents.FORWARDED_SOURCES_CHANGED, leavingForwardedSources, enteringForwardedSources, timestamp);
+    }
+    /**
+     * Should be called when current media session ends and after the
+     * PeerConnection has been closed using PeerConnection.close() method.
+     */
+    onCallEnded() {
+        if (this._channel) {
+            // The BridgeChannel is not explicitly closed as the PeerConnection
+            // is closed on call ended which triggers datachannel onclose
+            // events. If using a WebSocket, the channel must be closed since
+            // it is not managed by the PeerConnection.
+            // The reference is cleared to disable any logic related to the
+            // channel.
+            if (this._channel && this._channel.mode === 'websocket') {
+                this._channel.close();
+            }
+            this._channel = null;
+        }
+    }
+    /**
+     * Sets the capture frame rate to be used for desktop tracks.
+     *
+     * @param {number} maxFps framerate to be used for desktop track capture.
+     */
+    setDesktopSharingFrameRate(maxFps) {
+        RTCUtils.setDesktopSharingFrameRate(maxFps);
+    }
+    /**
+     * Sets the receiver video constraints that determine how bitrate is allocated to each of the video streams
+     * requested from the bridge. The constraints are cached and sent through the bridge channel once the channel
+     * is established.
+     * @param {*} constraints
+     */
+    setReceiverVideoConstraints(constraints) {
+        if (isEqual(this._receiverVideoConstraints, constraints)) {
+            return;
+        }
+        this._receiverVideoConstraints = cloneDeep(constraints);
+        if (this._channel && this._channel.isOpen()) {
+            this._channel.sendReceiverVideoConstraintsMessage(constraints);
+        }
+    }
+    /**
+     * Sends the track's  video type to the JVB.
+     * @param {SourceName} sourceName - the track's source name.
+     * @param {BridgeVideoType} videoType - the track's video type.
+     */
+    sendSourceVideoType(sourceName, videoType) {
+        if (this._channel && this._channel.isOpen()) {
+            this._channel.sendSourceVideoTypeMessage(sourceName, videoType);
+        }
+    }
+    /**
+     *
+     * @param eventType
+     * @param listener
+     */
+    static addListener(eventType, listener) {
+        RTCUtils.addListener(eventType, listener);
+    }
+    /**
+     *
+     * @param eventType
+     * @param listener
+     */
+    static removeListener(eventType, listener) {
+        RTCUtils.removeListener(eventType, listener);
+    }
+    /**
+     *
+     * @param options
+     */
+    static init(options = {}) {
+        this.options = options;
+        return RTCUtils.init(this.options);
+    }
+    /* eslint-disable max-params */
+    /**
+     * Creates new <tt>TraceablePeerConnection</tt>
+     * @param {SignalingLayer} signaling The signaling layer that will provide information about the media or
+     * participants which is not carried over SDP.
+     * @param {object} pcConfig The {@code RTCConfiguration} to use for the WebRTC peer connection.
+     * @param {boolean} isP2P Indicates whether or not the new TPC will be used in a peer to peer type of session.
+     * @param {object} options The config options.
+     * @param {Object} options.audioQuality - Quality settings to applied on the outbound audio stream.
+     * @param {boolean} options.capScreenshareBitrate if set to true, lower layers will be disabled for screenshare.
+     * @param {Array<CodecMimeType>} options.codecSettings - codec settings to be applied for video streams.
+     * @param {boolean} options.disableSimulcast if set to 'true' will disable the simulcast.
+     * @param {boolean} options.disableRtx if set to 'true' will disable the RTX.
+     * @param {boolean} options.enableInsertableStreams set to true when the insertable streams constraints is to be
+     * enabled on the PeerConnection.
+     * @param {boolean} options.forceTurnRelay If set to true, the browser will generate only Relay ICE candidates.
+     * @param {boolean} options.startSilent If set to 'true' no audio will be sent or received.
+     * @param {Object} options.videoQuality - Quality settings to applied on the outbound video streams.
+     * @return {TraceablePeerConnection}
+     */
+    createPeerConnection(signaling, pcConfig, isP2P, options) {
+        const pcConstraints = {};
+        if (options.enableInsertableStreams) {
+            logger.debug('E2EE - setting insertable streams constraints');
+            pcConfig.encodedInsertableStreams = true;
+        }
+        if (options.forceTurnRelay) {
+            pcConfig.iceTransportPolicy = 'relay';
+        }
+        // Set the RTCBundlePolicy to max-bundle so that only one set of ice candidates is generated.
+        // The default policy generates separate ice candidates for audio and video connections.
+        // This change is necessary for Unified plan to work properly on Chrome and Safari.
+        pcConfig.bundlePolicy = 'max-bundle';
+        peerConnectionIdCounter = safeCounterIncrement(peerConnectionIdCounter);
+        const newConnection = new TraceablePeerConnection(this, peerConnectionIdCounter, signaling, pcConfig, pcConstraints, isP2P, options);
+        this.peerConnections.set(newConnection.id, newConnection);
+        return newConnection;
+    }
+    /* eslint-enable max-params */
+    /**
+     * Removed given peer connection from this RTC module instance.
+     * @param {TraceablePeerConnection} traceablePeerConnection
+     * @return {boolean} <tt>true</tt> if the given peer connection was removed
+     * successfully or <tt>false</tt> if there was no peer connection mapped in
+     * this RTC instance.
+     */
+    _removePeerConnection(traceablePeerConnection) {
+        const id = traceablePeerConnection.id;
+        if (this.peerConnections.has(id)) {
+            // NOTE Remote tracks are not removed here.
+            this.peerConnections.delete(id);
+            return true;
+        }
+        return false;
+    }
+    /**
+     *
+     * @param track
+     */
+    addLocalTrack(track) {
+        if (!track) {
+            throw new Error('track must not be null nor undefined');
+        }
+        this.localTracks.push(track);
+        track.conference = this.conference;
+    }
+    /**
+     * Get forwarded sources list.
+     * @returns {Array<string>|null}
+     */
+    getForwardedSources() {
+        return this._forwardedSources;
+    }
+    /**
+     * Get local video track.
+     * @returns {JitsiLocalTrack|undefined}
+     */
+    getLocalVideoTrack() {
+        const localVideo = this.getLocalTracks(MediaType.VIDEO);
+        return localVideo.length ? localVideo[0] : undefined;
+    }
+    /**
+     * Returns all the local video tracks.
+     * @returns {Array<JitsiLocalTrack>}
+     */
+    getLocalVideoTracks() {
+        return this.getLocalTracks(MediaType.VIDEO);
+    }
+    /**
+     * Get local audio track.
+     * @returns {JitsiLocalTrack|undefined}
+     */
+    getLocalAudioTrack() {
+        const localAudio = this.getLocalTracks(MediaType.AUDIO);
+        return localAudio.length ? localAudio[0] : undefined;
+    }
+    /**
+     * Returns the endpoint id for the local user.
+     * @returns {string}
+     */
+    getLocalEndpointId() {
+        return this.conference.myUserId();
+    }
+    /**
+     * Returns the local tracks of the given media type, or all local tracks if
+     * no specific type is given.
+     * @param {MediaType} [mediaType] Optional media type filter.
+     * (audio or video).
+     */
+    getLocalTracks(mediaType) {
+        if (!mediaType) {
+            return this.localTracks.slice();
+        }
+        return this.localTracks.filter(track => track.getType() === mediaType);
+    }
+    /**
+     * Obtains all remote tracks currently known to this RTC module instance.
+     * @param {MediaType} [mediaType] The remote tracks will be filtered
+     *      by their media type if this argument is specified.
+     * @return {Array<JitsiRemoteTrack>}
+     */
+    getRemoteTracks(mediaType) {
+        let remoteTracks = [];
+        for (const tpc of this.peerConnections.values()) {
+            const pcRemoteTracks = tpc.getRemoteTracks(undefined, mediaType);
+            if (pcRemoteTracks) {
+                remoteTracks = remoteTracks.concat(pcRemoteTracks);
+            }
+        }
+        return remoteTracks;
+    }
+    /**
+     * Set mute for all local audio streams attached to the conference.
+     * @returns {Promise}
+     */
+    setAudioMute() {
+        const mutePromises = [];
+        this.getLocalTracks(MediaType.AUDIO).forEach(audioTrack => {
+            mutePromises.push(audioTrack.mute());
+        });
+        return Promise.all(mutePromises);
+    }
+    /**
+    * Set mute for all local video streams attached to the conference.
+    * @returns {Promise}
+    */
+    setVideoMute() {
+        const mutePromises = [];
+        const tracks = this.localTracks.filter(track => track.getType() === MediaType.VIDEO
+            && track.getVideoType() === VideoType.CAMERA);
+        tracks.forEach(track => mutePromises.push(track.mute()));
+        return Promise.all(mutePromises);
+    }
+    /**
+    * Set mute for all local desktop video streams attached to the conference.
+    * @returns {Promise}
+    */
+    setDesktopMute() {
+        const mutePromises = [];
+        const tracks = this.localTracks.filter(track => track.getType() === MediaType.VIDEO
+            && track.getVideoType() === VideoType.DESKTOP);
+        tracks.forEach(track => mutePromises.push(track.mute()));
+        return Promise.all(mutePromises);
+    }
+    /**
+     *
+     * @param track
+     */
+    removeLocalTrack(track) {
+        const pos = this.localTracks.indexOf(track);
+        if (pos === -1) {
+            return;
+        }
+        this.localTracks.splice(pos, 1);
+    }
+    /**
+     *
+     * @param elSelector
+     * @param stream
+     */
+    static attachMediaStream(elSelector, stream) {
+        return RTCUtils.attachMediaStream(elSelector, stream);
+    }
+    /**
+     * Returns true if changing the input (camera / microphone) or output
+     * (audio) device is supported and false if not.
+     * @param {string} [deviceType] Type of device to change. Default is
+     *      undefined or 'input', 'output' - for audio output device change.
+     * @returns {boolean} true if available, false otherwise.
+     */
+    static isDeviceChangeAvailable(deviceType) {
+        return RTCUtils.isDeviceChangeAvailable(deviceType);
+    }
+    /**
+     * 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.
+     */
+    static isWebRtcSupported() {
+        return browser.isSupported();
+    }
+    /**
+     * Returns currently used audio output device id, '' stands for default
+     * device
+     * @returns {string}
+     */
+    static getAudioOutputDevice() {
+        return RTCUtils.getAudioOutputDevice();
+    }
+    /**
+     * Returns list of available media devices if its obtained, otherwise an
+     * empty array is returned/
+     * @returns {array} list of available media devices.
+     */
+    static getCurrentlyAvailableMediaDevices() {
+        return RTCUtils.getCurrentlyAvailableMediaDevices();
+    }
+    /**
+     * Returns event data for device to be reported to stats.
+     * @returns {MediaDeviceInfo} device.
+     */
+    static getEventDataForActiveDevice(device) {
+        return RTCUtils.getEventDataForActiveDevice(device);
+    }
+    /**
+     * Sets current audio output device.
+     * @param {string} deviceId Id of 'audiooutput' device from
+     *      navigator.mediaDevices.enumerateDevices().
+     * @returns {Promise} resolves when audio output is changed, is rejected
+     *      otherwise
+     */
+    static setAudioOutputDevice(deviceId) {
+        return RTCUtils.setAudioOutputDevice(deviceId);
+    }
+    /**
+     * Allows to receive list of available cameras/microphones.
+     * @param {function} callback Would receive array of devices as an
+     *      argument.
+     */
+    static enumerateDevices(callback) {
+        RTCUtils.enumerateDevices(callback);
+    }
+    /**
+     * A method to handle stopping of the stream.
+     * One point to handle the differences in various implementations.
+     * @param {MediaStream} mediaStream MediaStream object to stop.
+     */
+    static stopMediaStream(mediaStream) {
+        RTCUtils.stopMediaStream(mediaStream);
+    }
+    /**
+     * Returns whether the desktop sharing is enabled or not.
+     * @returns {boolean}
+     */
+    static isDesktopSharingEnabled() {
+        return RTCUtils.isDesktopSharingEnabled();
+    }
+    /**
+     * Closes the currently opened bridge channel.
+     */
+    closeBridgeChannel() {
+        if (this._channel) {
+            this._channel.close();
+            this._channel = null;
+        }
+    }
+    /* eslint-disable max-params */
+    /**
+     *
+     * @param {TraceablePeerConnection} tpc
+     * @param {number} ssrc
+     * @param {number} audioLevel
+     * @param {boolean} isLocal
+     */
+    setAudioLevel(tpc, ssrc, audioLevel, isLocal) {
+        const track = tpc.getTrackBySSRC(ssrc);
+        if (!track) {
+            return;
+        }
+        else if (!track.isAudioTrack()) {
+            logger.warn(`Received audio level for non-audio track: ${ssrc}`);
+            return;
+        }
+        else if (track.isLocal() !== isLocal) {
+            logger.error(`${track} was expected to ${isLocal ? 'be' : 'not be'} local`);
+        }
+        track.setAudioLevel(audioLevel, tpc);
+    }
+    /**
+     * Sends message via the bridge channel.
+     * @param {string} to The id of the endpoint that should receive the
+     *      message. If "" the message will be sent to all participants.
+     * @param {object} payload The payload of the message.
+     * @throws NetworkError or InvalidStateError or Error if the operation
+     * fails or there is no data channel created.
+     */
+    sendChannelMessage(to, payload) {
+        if (this._channel) {
+            this._channel.sendMessage(to, payload);
+        }
+        else {
+            throw new Error('BridgeChannel has not been initialized yet');
+        }
+    }
+    /**
+     * Sends the local stats via the bridge channel.
+     * @param {Object} payload The payload of the message.
+     * @throws NetworkError/InvalidStateError/Error if the operation fails or if there is no data channel created.
+     */
+    sendEndpointStatsMessage(payload) {
+        if (this._channel && this._channel.isOpen()) {
+            this._channel.sendEndpointStatsMessage(payload);
+        }
+    }
+    /**
+     * Sends a receiver audio subscription message.
+     * @param {*} message
+     */
+    sendReceiverAudioSubscriptionMessage(message) {
+        if (this._channel && this._channel.isOpen()) {
+            this._channel.sendReceiverAudioSubscriptionMessage(message);
+        }
+    }
+    /**
+     * Selects a new value for "lastN". The requested amount of videos are going
+     * to be delivered after the value is in effect. Set to -1 for unlimited or
+     * all available videos.
+     * @param {number} value the new value for lastN.
+     */
+    setLastN(value) {
+        if (this._lastN !== value) {
+            this._lastN = value;
+            if (this._channel && this._channel.isOpen()) {
+                this._channel.sendSetLastNMessage(value);
+            }
+            this.eventEmitter.emit(RTCEvents.LASTN_VALUE_CHANGED, value);
+        }
+    }
+    /**
+     * Indicates if the source name is currently included in the forwarded sources.
+     *
+     * @param {string} sourceName The source name that we check for forwarded sources.
+     * @returns {boolean} true if the source name is in the forwarded sources or if we don't have bridge channel
+     * support, otherwise we return false.
+     */
+    isInForwardedSources(sourceName) {
+        return !this._forwardedSources // forwardedSources not initialised yet.
+            || this._forwardedSources.indexOf(sourceName) > -1;
+    }
+    /**
+     * Updates the target audio output device for all remote audio tracks.
+     *
+     * @param {string} deviceId - The device id of the audio ouput device to
+     * use for all remote tracks.
+     * @private
+     * @returns {void}
+     */
+    _updateAudioOutputForAudioTracks(deviceId) {
+        const remoteAudioTracks = this.getRemoteTracks(MediaType.AUDIO);
+        for (const track of remoteAudioTracks) {
+            track.setAudioOutput(deviceId);
+        }
+    }
+}
+//# sourceMappingURL=RTC.js.map

package/dist/esm/modules/RTC/RTC.js.map

@@ -0,0 +1 @@
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