camera.ui3 0.0.30-alpha.0

package/dist/server/src/camera/streaming/peer-connection.js

@@ -0,0 +1,203 @@
+import { ReplaySubject, Subject, interval, merge } from 'rxjs';
+import { MediaStreamTrack, RTCPeerConnection, RTCRtpCodecParameters } from 'werift';
+import { Subscribed } from '../../utils/subscribed.js';
+function convertIceServers(iceServers) {
+    const rtcIceServers = [];
+    iceServers.forEach((iceServer) => {
+        iceServer.urls.forEach((url) => {
+            rtcIceServers.push({
+                urls: url,
+                username: iceServer.username,
+                credential: iceServer.credential,
+            });
+        });
+    });
+    return rtcIceServers;
+}
+export class WeriftPeerConnection extends Subscribed {
+    onAudioRtp = new Subject();
+    onAudioRtcp = new Subject();
+    onVideoRtp = new Subject();
+    onVideoRtcp = new Subject();
+    onIceCandidate = new Subject();
+    onConnectionState = new ReplaySubject(1);
+    returnAudioTrack = new MediaStreamTrack({ kind: 'audio' });
+    onRequestKeyFrame = new Subject();
+    cameraDevice;
+    sourceName;
+    logger;
+    options;
+    pc;
+    get logPrefix() {
+        return `${this.cameraDevice.name} (${this.sourceName})`;
+    }
+    constructor(cameraDevice, options, logger, sourceName) {
+        super();
+        this.cameraDevice = cameraDevice;
+        this.sourceName = sourceName;
+        this.logger = logger;
+        this.options = options;
+        const pc = (this.pc = this.createPeerCoonection());
+        pc.addTransceiver(this.returnAudioTrack, {
+            direction: 'sendonly',
+        });
+        if (!this.options.onlyTalkback) {
+            const audioTransceiver = pc.addTransceiver(this.returnAudioTrack, {
+                direction: 'recvonly',
+            });
+            audioTransceiver.onTrack.subscribe((track) => {
+                track.onReceiveRtp.subscribe((rtp) => {
+                    this.onAudioRtp.next(rtp);
+                });
+                track.onReceiveRtcp.subscribe((rtcp) => {
+                    this.onAudioRtcp.next(rtcp);
+                });
+                track.onReceiveRtp.once(() => {
+                    this.logger.debug(this.logPrefix, 'received first audio packet');
+                });
+            });
+            const videoTransceiver = pc.addTransceiver('video', {
+                direction: 'recvonly',
+            });
+            videoTransceiver.onTrack.subscribe((track) => {
+                track.onReceiveRtp.subscribe((rtp) => {
+                    this.onVideoRtp.next(rtp);
+                });
+                track.onReceiveRtcp.subscribe((rtcp) => {
+                    this.onVideoRtcp.next(rtcp);
+                });
+                track.onReceiveRtp.once(() => {
+                    this.logger.debug(this.logPrefix, 'received first video packet');
+                    this.addSubscriptions(merge(this.onRequestKeyFrame, interval(4000)).subscribe(() => {
+                        videoTransceiver.receiver.sendRtcpPLI(track.ssrc).catch((error) => this.logger.error(this.logPrefix, error));
+                    }));
+                    this.requestKeyFrame();
+                });
+            });
+        }
+        this.pc.onIceCandidate.subscribe((iceCandidate) => {
+            this.onIceCandidate.next(iceCandidate);
+        });
+        pc.iceConnectionStateChange.subscribe(() => {
+            this.logger.debug(this.logPrefix, `iceConnectionStateChange: ${pc.iceConnectionState}`);
+            if (pc.iceConnectionState === 'closed') {
+                this.onConnectionState.next('closed');
+            }
+        });
+        pc.connectionStateChange.subscribe(() => {
+            this.logger.debug(this.logPrefix, `connectionStateChange: ${pc.connectionState}`);
+            this.onConnectionState.next(pc.connectionState);
+        });
+    }
+    async createOffer() {
+        const offer = await this.pc.createOffer();
+        this.pc.setLocalDescription(offer);
+        return offer;
+    }
+    async acceptAnswer(answer) {
+        await this.pc.setRemoteDescription(answer);
+    }
+    addIceCandidate(candidate) {
+        return this.pc.addIceCandidate(candidate);
+    }
+    requestKeyFrame() {
+        this.onRequestKeyFrame.next();
+    }
+    close() {
+        this.pc.close().catch((error) => this.logger.error(this.logPrefix, error));
+        this.unsubscribe();
+    }
+    createPeerCoonection() {
+        const prefferedVideoCodec = this.options.preferredVideoCodecs?.length ? this.options.preferredVideoCodecs : ['vp8', 'vp9', 'h264', 'av1'];
+        const prefferedAudioCodec = this.options.preferredAudioCodecs?.length ? this.options.preferredAudioCodecs : ['opus', 'g722', 'pcmu', 'pcma'];
+        const video = [];
+        const audio = [];
+        if (!this.options.onlyTalkback) {
+            for (const codec of prefferedVideoCodec) {
+                switch (codec) {
+                    case 'vp8':
+                        video.push(new RTCRtpCodecParameters({
+                            mimeType: 'video/VP8',
+                            clockRate: 90000,
+                            payloadType: 100,
+                            rtcpFeedback: [{ type: 'goog-remb' }, { type: 'ccm', parameter: 'fir' }, { type: 'nack' }, { type: 'nack', parameter: 'pli' }],
+                        }));
+                        break;
+                    case 'vp9':
+                        video.push(new RTCRtpCodecParameters({
+                            mimeType: 'video/VP9',
+                            clockRate: 90000,
+                            payloadType: 101,
+                            rtcpFeedback: [{ type: 'goog-remb' }, { type: 'ccm', parameter: 'fir' }, { type: 'nack' }, { type: 'nack', parameter: 'pli' }],
+                        }));
+                        break;
+                    case 'h264':
+                        video.push(new RTCRtpCodecParameters({
+                            mimeType: 'video/H264',
+                            clockRate: 90000,
+                            rtcpFeedback: [{ type: 'transport-cc' }, { type: 'ccm', parameter: 'fir' }, { type: 'nack' }, { type: 'nack', parameter: 'pli' }, { type: 'goog-remb' }],
+                            parameters: 'packetization-mode=1;profile-level-id=640029;level-asymmetry-allowed=1',
+                        }), new RTCRtpCodecParameters({
+                            mimeType: 'video/rtx',
+                            clockRate: 90000,
+                        }));
+                        break;
+                    case 'av1':
+                        video.push(new RTCRtpCodecParameters({
+                            mimeType: 'video/AV1',
+                            clockRate: 90000,
+                            payloadType: 102,
+                            rtcpFeedback: [{ type: 'goog-remb' }, { type: 'ccm', parameter: 'fir' }, { type: 'nack' }, { type: 'nack', parameter: 'pli' }],
+                        }));
+                        break;
+                }
+            }
+        }
+        for (const codec of prefferedAudioCodec) {
+            switch (codec) {
+                case 'opus':
+                    audio.push(new RTCRtpCodecParameters({
+                        mimeType: 'audio/opus',
+                        clockRate: 48000,
+                        payloadType: 101,
+                        channels: 2,
+                        parameters: 'minptime=10;useinbandfec=1',
+                    }));
+                    break;
+                case 'g722':
+                    audio.push(new RTCRtpCodecParameters({
+                        mimeType: 'audio/G722',
+                        clockRate: 8000,
+                        channels: 1,
+                        payloadType: 9,
+                    }));
+                    break;
+                case 'pcmu':
+                    audio.push(new RTCRtpCodecParameters({
+                        mimeType: 'audio/PCMU',
+                        clockRate: 8000,
+                        channels: 1,
+                        payloadType: 0,
+                    }));
+                    break;
+                case 'pcma':
+                    audio.push(new RTCRtpCodecParameters({
+                        mimeType: 'audio/PCMA',
+                        clockRate: 8000,
+                        channels: 1,
+                        payloadType: 8,
+                    }));
+                    break;
+            }
+        }
+        return new RTCPeerConnection({
+            codecs: {
+                video,
+                audio,
+            },
+            iceServers: convertIceServers(this.options.iceServers ?? []),
+            bundlePolicy: 'max-bundle',
+        });
+    }
+}
+//# sourceMappingURL=peer-connection.js.map

package/dist/server/src/camera/streaming/peer-connection.js.map

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package/dist/server/src/camera/streaming/webrtc-connection.d.ts

@@ -0,0 +1,35 @@
+import { ReplaySubject, Subject } from 'rxjs';
+import { Subscribed } from '../../utils/subscribed.js';
+import type { Observable } from 'rxjs';
+import type { RtpPacket } from 'werift';
+import type { CameraDevice } from '../device.js';
+import type { BaseLogger } from '../types.js';
+import type { StreamingConnectionOptions } from './peer-connection.js';
+export declare class WebrtcConnection extends Subscribed {
+    readonly onCameraConnected: ReplaySubject<void>;
+    readonly onCallAnswered: ReplaySubject<string>;
+    readonly onCallEnded: ReplaySubject<void>;
+    readonly onError: ReplaySubject<void>;
+    readonly onMessage: ReplaySubject<{
+        method: string;
+    }>;
+    readonly onWsOpen: Observable<unknown>;
+    readonly onAudioRtp: Subject<RtpPacket>;
+    readonly onVideoRtp: Subject<RtpPacket>;
+    private readonly pc;
+    private readonly ws;
+    private readonly onOfferSent;
+    private hasEnded;
+    private cameraDevice;
+    private sourceName;
+    private logger;
+    private get logPrefix();
+    constructor(cameraDevice: CameraDevice, sourceName: string, logger: BaseLogger, options: StreamingConnectionOptions);
+    sendAudioPacket(rtp: RtpPacket): void;
+    stop(): void;
+    requestKeyFrame(): void;
+    private initiateCall;
+    private handleMessage;
+    private sendMessage;
+    private callEnded;
+}

package/dist/server/src/camera/streaming/webrtc-connection.js

@@ -0,0 +1,170 @@
+import { ReplaySubject, Subject, fromEvent } from 'rxjs';
+import { concatMap } from 'rxjs/operators';
+import { WebSocket } from 'ws';
+import { Subscribed } from '../../utils/subscribed.js';
+import { WeriftPeerConnection } from './peer-connection.js';
+export class WebrtcConnection extends Subscribed {
+    onCameraConnected = new ReplaySubject(1);
+    onCallAnswered = new ReplaySubject(1);
+    onCallEnded = new ReplaySubject(1);
+    onError = new ReplaySubject(1);
+    onMessage = new ReplaySubject();
+    onWsOpen;
+    onAudioRtp;
+    onVideoRtp;
+    pc;
+    ws;
+    onOfferSent = new ReplaySubject(1);
+    hasEnded = false;
+    cameraDevice;
+    sourceName;
+    logger;
+    get logPrefix() {
+        return `${this.cameraDevice.name} (${this.sourceName})`;
+    }
+    constructor(cameraDevice, sourceName, logger, options) {
+        super();
+        this.cameraDevice = cameraDevice;
+        this.logger = logger;
+        const sources = cameraDevice.sources
+            .map((source) => {
+            if (source.name === sourceName) {
+                return { name: source.name, src: source.urls.ws.webrtc };
+            }
+            else if (source.urls.transcoded.aac.name === sourceName) {
+                return { name: source.urls.transcoded.aac.name, src: source.urls.transcoded.aac.ws };
+            }
+            else if (source.urls.transcoded.opus.name === sourceName) {
+                return { name: source.urls.transcoded.opus.name, src: source.urls.transcoded.opus.ws };
+            }
+        })
+            .filter(Boolean);
+        const webRtcSource = sources[0];
+        if (!webRtcSource) {
+            throw new Error(`Invalid source: ${sourceName}`);
+        }
+        this.sourceName = webRtcSource.name;
+        this.ws = new WebSocket(webRtcSource.src);
+        if (options.createPeerConnection) {
+            // we were passed a custom peer connection factory
+            this.pc = options.createPeerConnection();
+            // passing rtp packets is not supported for custom peer connections
+            this.onAudioRtp = new Subject();
+            this.onVideoRtp = new Subject();
+        }
+        else {
+            // no custom peer connection factory, use the werift and pass along rtp packets
+            const pc = new WeriftPeerConnection(this.cameraDevice, options, this.logger, sourceName);
+            this.pc = pc;
+            this.onAudioRtp = pc.onAudioRtp;
+            this.onVideoRtp = pc.onVideoRtp;
+        }
+        this.onWsOpen = fromEvent(this.ws, 'open');
+        const onMessage = fromEvent(this.ws, 'message');
+        const onError = fromEvent(this.ws, 'error');
+        const onClose = fromEvent(this.ws, 'close');
+        this.addSubscriptions(onMessage
+            .pipe(concatMap((event) => {
+            const message = JSON.parse(event.data);
+            this.onMessage.next(message);
+            return this.handleMessage(message).catch((error) => {
+                this.onError.next(error);
+                throw error;
+            });
+        }))
+            .subscribe(), onError.subscribe((error) => {
+            this.logger.error(this.logPrefix, error);
+            this.callEnded();
+        }), onClose.subscribe(() => {
+            this.callEnded();
+        }), this.cameraDevice.onConnected.subscribe((state) => {
+            if (!state) {
+                this.callEnded();
+            }
+        }), this.pc.onConnectionState.subscribe((state) => {
+            if (state === 'failed') {
+                this.logger.error(this.logPrefix, 'Stream connection failed');
+                this.callEnded();
+            }
+            if (state === 'closed') {
+                this.logger.debug(this.logPrefix, 'Stream connection closed');
+                this.callEnded();
+            }
+        }), this.onError.subscribe((error) => {
+            this.logger.error(this.logPrefix, error);
+            this.callEnded();
+        }), this.onWsOpen.subscribe(() => {
+            this.initiateCall().catch((error) => {
+                this.logger.error(this.logPrefix, error);
+                this.callEnded();
+            });
+        }), this.pc.onIceCandidate.subscribe(async (iceCandidate) => {
+            // await firstValueFrom(this.onOfferSent);
+            this.sendMessage({
+                type: 'webrtc/candidate',
+                value: iceCandidate.toJSON().candidate || '',
+            });
+        }));
+    }
+    sendAudioPacket(rtp) {
+        if (this.hasEnded) {
+            return;
+        }
+        if (this.pc instanceof WeriftPeerConnection) {
+            this.pc.returnAudioTrack.writeRtp(rtp);
+        }
+        else {
+            throw new Error('Cannot send audio packets to a custom peer connection implementation');
+        }
+    }
+    stop() {
+        this.callEnded();
+    }
+    requestKeyFrame() {
+        this.pc.requestKeyFrame?.();
+    }
+    async initiateCall() {
+        const { sdp } = await this.pc.createOffer();
+        this.sendMessage({
+            type: 'webrtc/offer',
+            value: sdp,
+        });
+        this.onOfferSent.next();
+    }
+    async handleMessage(message) {
+        switch (message.type) {
+            case 'webrtc/answer':
+                const sdp = message.value;
+                await this.pc.acceptAnswer({ type: 'answer', sdp });
+                this.onCallAnswered.next(sdp);
+                return;
+            case 'webrtc/candidate':
+                await this.pc.addIceCandidate({
+                    candidate: message.value,
+                    sdpMid: '0',
+                });
+                return;
+            case 'error':
+                throw new Error(message.value);
+            default:
+                throw new Error(message);
+        }
+    }
+    sendMessage(message) {
+        if (this.hasEnded) {
+            return;
+        }
+        this.ws.send(JSON.stringify(message));
+    }
+    callEnded() {
+        if (this.hasEnded) {
+            return;
+        }
+        this.hasEnded = true;
+        this.unsubscribe();
+        this.onCallEnded.next();
+        this.pc.close();
+        this.ws.close();
+    }
+}
+//# sourceMappingURL=webrtc-connection.js.map

package/dist/server/src/camera/streaming/webrtc-connection.js.map

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package/dist/server/src/camera/streaming/werift-session.d.ts

@@ -0,0 +1,41 @@
+import { ReplaySubject, Subject } from 'rxjs';
+import { RtpPacket } from 'werift';
+import type { Subscription } from 'rxjs';
+import type { CameraDevice } from '../device.js';
+import type { BaseLogger, FfmpegOptions, SpawnInput } from '../types.js';
+import type { WebrtcConnection } from './webrtc-connection.js';
+declare class Subscribed {
+    private readonly subscriptions;
+    private readonly additionalSubscriptions;
+    addSubscriptions(...subscriptions: Subscription[]): void;
+    addAdditionalSubscriptions(...subscriptions: Subscription[]): void;
+    unsubscribe(): void;
+    unsubscribeAdditional(): void;
+}
+export declare class WeriftSession extends Subscribed {
+    readonly onCallEnded: ReplaySubject<void>;
+    readonly onVideoRtp: Subject<RtpPacket>;
+    readonly onAudioRtp: Subject<RtpPacket>;
+    private readonly audioSplitter;
+    private readonly videoSplitter;
+    private readonly returnAudioSplitter;
+    private readonly onUsingOpus;
+    private hasEnded;
+    private cameraDevice;
+    private connection;
+    private logger;
+    get isUsingOpus(): Promise<boolean>;
+    constructor(cameraDevice: CameraDevice, logger: BaseLogger, connection: WebrtcConnection);
+    reservePort(bufferPorts?: number): Promise<number>;
+    startTranscoding(ffmpegOptions: FfmpegOptions): Promise<void>;
+    transcodeReturnAudio(ffmpegOptions: {
+        ffmpegPath: string;
+        input: SpawnInput[];
+    }): Promise<void>;
+    stop(): void;
+    sendAudioPacket(rtp: RtpPacket): void;
+    requestKeyFrame(): void;
+    private bindToConnection;
+    private callEnded;
+}
+export {};

package/dist/server/src/camera/streaming/werift-session.js

@@ -0,0 +1,179 @@
+import { FfmpegProcess, reservePorts, RtpSplitter } from '@homebridge/camera-utils';
+import { firstValueFrom, ReplaySubject, Subject } from 'rxjs';
+import { concatMap, map, mergeWith, take } from 'rxjs/operators';
+import { RtpPacket } from 'werift';
+function getCleanSdp(sdp, includeVideo) {
+    return sdp
+        .split('\nm=')
+        .slice(1)
+        .map((section) => 'm=' + section)
+        .filter((section) => includeVideo || !section.startsWith('m=video'))
+        .join('\n');
+}
+class Subscribed {
+    subscriptions = [];
+    additionalSubscriptions = [];
+    addSubscriptions(...subscriptions) {
+        this.subscriptions.push(...subscriptions);
+    }
+    addAdditionalSubscriptions(...subscriptions) {
+        this.additionalSubscriptions.push(...subscriptions);
+    }
+    unsubscribe() {
+        this.subscriptions.forEach((subscription) => subscription.unsubscribe());
+    }
+    unsubscribeAdditional() {
+        this.additionalSubscriptions.forEach((subscription) => subscription.unsubscribe());
+    }
+}
+export class WeriftSession extends Subscribed {
+    onCallEnded = new ReplaySubject(1);
+    onVideoRtp = new Subject();
+    onAudioRtp = new Subject();
+    audioSplitter = new RtpSplitter();
+    videoSplitter = new RtpSplitter();
+    returnAudioSplitter = new RtpSplitter();
+    onUsingOpus = new ReplaySubject(1);
+    hasEnded = false;
+    cameraDevice;
+    connection;
+    logger;
+    get isUsingOpus() {
+        return firstValueFrom(this.onUsingOpus.pipe(mergeWith(this.connection.onError.pipe(map((error) => {
+            throw error;
+        })))));
+    }
+    constructor(cameraDevice, logger, connection) {
+        super();
+        this.cameraDevice = cameraDevice;
+        this.logger = logger;
+        this.connection = connection;
+        this.bindToConnection(connection);
+    }
+    async reservePort(bufferPorts = 0) {
+        const ports = await reservePorts({ count: bufferPorts + 1 });
+        return ports[0];
+    }
+    async startTranscoding(ffmpegOptions) {
+        if (this.hasEnded) {
+            return;
+        }
+        const videoPort = await this.reservePort(1);
+        const audioPort = await this.reservePort(1);
+        const transcodeVideoStream = ffmpegOptions.video !== undefined && ffmpegOptions.video.length > 0;
+        const sdp = await Promise.race([firstValueFrom(this.connection.onCallAnswered), firstValueFrom(this.onCallEnded)]);
+        if (!sdp) {
+            this.logger.debug(this.cameraDevice.name, 'No SDP received, ending session');
+            return;
+        }
+        const usingOpus = await this.isUsingOpus;
+        const ffmpegInputArguments = [
+            '-hide_banner',
+            '-protocol_whitelist',
+            'pipe,udp,rtp,file,crypto',
+            ...(usingOpus ? ['-acodec', 'libopus'] : []),
+            '-f',
+            'sdp',
+            ...(ffmpegOptions.input || []),
+            '-i',
+            'pipe:',
+        ];
+        const inputSdp = getCleanSdp(sdp, transcodeVideoStream)
+            .replace(/m=audio \d+/, `m=audio ${audioPort}`)
+            .replace(/m=video \d+/, `m=video ${videoPort}`);
+        const ff = new FfmpegProcess({
+            ffmpegArgs: ffmpegInputArguments.concat(...(ffmpegOptions.audio || ['-acodec', 'aac']), ...(transcodeVideoStream ? ffmpegOptions.video || ['-vcodec', 'copy'] : []), ...(ffmpegOptions.output || [])),
+            ffmpegPath: ffmpegOptions.ffmpegPath,
+            exitCallback: () => this.callEnded(),
+            logger: {
+                info: (message) => this.logger.trace(this.cameraDevice.name, 'Transcode:', message.trim()),
+                error: (message) => this.logger.error(this.cameraDevice.name, 'Transcode:', message.trim()),
+            },
+        });
+        this.addSubscriptions(this.onAudioRtp
+            .pipe(concatMap((rtp) => {
+            return this.audioSplitter.send(rtp.serialize(), {
+                port: audioPort,
+            });
+        }))
+            .subscribe());
+        if (transcodeVideoStream) {
+            this.addSubscriptions(this.onVideoRtp
+                .pipe(concatMap((rtp) => {
+                return this.videoSplitter.send(rtp.serialize(), {
+                    port: videoPort,
+                });
+            }))
+                .subscribe());
+        }
+        this.onCallEnded.pipe(take(1)).subscribe(() => ff.stop());
+        ff.writeStdin(inputSdp);
+        // Request a key frame now that ffmpeg is ready to receive
+        this.requestKeyFrame();
+    }
+    async transcodeReturnAudio(ffmpegOptions) {
+        if (this.hasEnded) {
+            return;
+        }
+        const audioOutForwarder = new RtpSplitter(({ message }) => {
+            const rtp = RtpPacket.deSerialize(message);
+            this.connection.sendAudioPacket(rtp);
+            return null;
+        });
+        const usingOpus = await this.isUsingOpus;
+        const ff = new FfmpegProcess({
+            ffmpegArgs: [
+                '-hide_banner',
+                '-protocol_whitelist',
+                'pipe,udp,rtp,file,crypto',
+                '-re',
+                '-i',
+                ...ffmpegOptions.input,
+                '-acodec',
+                ...(usingOpus ? ['libopus', '-ac', 2, '-ar', '48k'] : ['pcm_mulaw', '-ac', 1, '-ar', '8k']),
+                '-flags',
+                '+global_header',
+                '-f',
+                'rtp',
+                `rtp://127.0.0.1:${await audioOutForwarder.portPromise}`,
+            ],
+            ffmpegPath: ffmpegOptions.ffmpegPath,
+            exitCallback: () => this.callEnded(),
+            logger: {
+                info: (message) => this.logger.trace(this.cameraDevice.name, 'Return Audio:', message),
+                error: (message) => this.logger.error(this.cameraDevice.name, 'Return Audio:', message),
+            },
+        });
+        this.onCallEnded.pipe(take(1)).subscribe(() => ff.stop());
+    }
+    stop() {
+        this.callEnded();
+    }
+    sendAudioPacket(rtp) {
+        if (this.hasEnded) {
+            return;
+        }
+        this.connection.sendAudioPacket(rtp);
+    }
+    requestKeyFrame() {
+        this.connection.requestKeyFrame();
+    }
+    bindToConnection(connection) {
+        this.addSubscriptions(connection.onAudioRtp.subscribe(this.onAudioRtp), connection.onVideoRtp.subscribe(this.onVideoRtp), connection.onCallAnswered.subscribe((sdp) => {
+            this.onUsingOpus.next(sdp.toLocaleLowerCase().includes(' opus/'));
+        }), connection.onCallEnded.subscribe(() => this.callEnded()));
+    }
+    callEnded() {
+        if (this.hasEnded) {
+            return;
+        }
+        this.hasEnded = true;
+        this.unsubscribe();
+        this.onCallEnded.next();
+        this.connection.stop();
+        this.audioSplitter.close();
+        this.videoSplitter.close();
+        this.returnAudioSplitter.close();
+    }
+}
+//# sourceMappingURL=werift-session.js.map