@camera.ui/server 0.0.69 → 0.0.71-dev.0

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

@@ -1,19 +0,0 @@
-import { SubscribedPublic } from '@camera.ui/common/utils';
-import { ReplaySubject, Subject } from 'rxjs';
-import { RtpPacket } from 'werift';
-import type { CameraInput, FfmpegOptions, ReturnAudioFFmpegOptions, RTSPConnectionOptions, RTSPSession as RTSPSessionInterface } from '@camera.ui/types';
-import type { CameraDevice } from '../device.js';
-export declare class RTSPSession extends SubscribedPublic implements RTSPSessionInterface {
-    #private;
-    readonly onStarted: ReplaySubject<void>;
-    readonly onError: ReplaySubject<Error>;
-    readonly onCallEnded: ReplaySubject<void>;
-    readonly onVideoRtp: Subject<RtpPacket>;
-    readonly onAudioRtp: Subject<RtpPacket>;
-    constructor(cameraDevice: CameraDevice, source: CameraInput, options?: RTSPConnectionOptions);
-    startStream(): Promise<void>;
-    startTranscoding(ffmpegOptions: FfmpegOptions): Promise<void>;
-    transcodeReturnAudio(ffmpegOptions: ReturnAudioFFmpegOptions): Promise<void>;
-    stop(): void;
-    sendAudioPacket(rtp: RtpPacket): void;
-}

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

@@ -1,196 +0,0 @@
-import { FfmpegProcess, generateAudioStreamInfo, RtpSplitter, transformAudioStreamInfoToFFmpegArgs } from '@camera.ui/common/camera';
-import { SubscribedPublic } from '@camera.ui/common/utils';
-import { RTSPClient } from '@camera.ui/rtsp/rtspClient';
-import { ReplaySubject, Subject } from 'rxjs';
-import { take } from 'rxjs/operators';
-import { RtpPacket } from 'werift';
-export class RTSPSession extends SubscribedPublic {
-    onStarted = new ReplaySubject();
-    onError = new ReplaySubject();
-    onCallEnded = new ReplaySubject(1);
-    onVideoRtp = new Subject();
-    onAudioRtp = new Subject();
-    #audioSplitter = new RtpSplitter();
-    #videoSplitter = new RtpSplitter();
-    #returnAudioSplitter = new RtpSplitter();
-    #hasEnded = false;
-    #cameraDevice;
-    #logger;
-    #source;
-    #client;
-    #url;
-    #ff;
-    constructor(cameraDevice, source, options) {
-        super();
-        this.#cameraDevice = cameraDevice;
-        this.#logger = cameraDevice.logger;
-        this.#source = this.#cameraDevice.sources.find((s) => s.name === source.name);
-        this.#url = options?.url ?? this.#source.generateRTSPUrl();
-        this.#client = new RTSPClient({ url: this.#url }, {
-            // log: (...args) => this.#logger.log(...args),
-            error: (...args) => this.#logger.error(...args),
-            // debug: (...args) => this.#logger.trace(...args),
-            warn: (...args) => this.#logger.warn(...args),
-        });
-        this.#bindConnection();
-    }
-    async startStream() {
-        this.#hasEnded = false;
-        await this.#client.start();
-    }
-    async startTranscoding(ffmpegOptions) {
-        const transcodeAudioStream = ffmpegOptions.audio !== undefined && ffmpegOptions.audio.length > 0;
-        const transcodeVideoStream = ffmpegOptions.video !== undefined && ffmpegOptions.video.length > 0;
-        const loggerPrefix = ffmpegOptions.logPrefix
-            ? `[${ffmpegOptions.logPrefix}] `
-            : transcodeAudioStream && transcodeVideoStream
-                ? '[Transcoding] '
-                : transcodeAudioStream
-                    ? '[Audio] '
-                    : '[Video] ';
-        try {
-            if (!transcodeAudioStream && !transcodeVideoStream) {
-                throw new Error('No transcoding options provided! Can not start transcoding!');
-            }
-            this.#logger.trace(loggerPrefix + 'Start transcoding');
-            const url = this.#source.generateRTSPUrl({
-                video: transcodeVideoStream,
-                audio: transcodeAudioStream,
-                backchannel: false,
-            });
-            this.#ff = new FfmpegProcess({
-                ffmpegArgs: [
-                    '-hide_banner',
-                    ...(ffmpegOptions.input ?? []),
-                    '-rtsp_transport',
-                    'tcp',
-                    '-i',
-                    url,
-                    ...(ffmpegOptions.audio ?? []),
-                    ...(ffmpegOptions.video ?? []),
-                    ...(ffmpegOptions.output ?? []),
-                ],
-                ffmpegPath: ffmpegOptions.ffmpegPath,
-                startedCallback: () => {
-                    this.onStarted.next();
-                },
-                exitCallback: (code, signal) => {
-                    if (code !== 0 && code !== 255) {
-                        this.#callError(new Error(`FFmpeg exited with code ${code} and signal ${signal}`));
-                    }
-                    else {
-                        this.#callEnded();
-                    }
-                },
-                logLabel: loggerPrefix,
-                logger: {
-                    error: (...args) => this.#logger.error(...args),
-                    trace: (...args) => this.#logger.trace(...args),
-                },
-            });
-        }
-        catch (error) {
-            this.#callError(error);
-            throw error;
-        }
-    }
-    async transcodeReturnAudio(ffmpegOptions) {
-        const loggerPrefix = ffmpegOptions.logPrefix ? `[${ffmpegOptions.logPrefix}] ` : '[Return Audio] ';
-        try {
-            await this.startStream();
-            const returnAudioTrack = this.#client.getBackchannelTrack();
-            const returnAudioStreamInfos = returnAudioTrack ? generateAudioStreamInfo([returnAudioTrack.mediaSource]) : [];
-            const returnAudioStream = returnAudioStreamInfos[0];
-            if (!returnAudioStream) {
-                this.stop();
-                throw new Error('No return audio stream found! Can not start talkback!');
-            }
-            const audioOutForwarder = new RtpSplitter();
-            await audioOutForwarder.prepare('udp4');
-            if (!audioOutForwarder.port) {
-                this.stop();
-                throw new Error('No ports available for audio out splitter! Can not start talkback!');
-            }
-            audioOutForwarder.addMessageHandler(({ message }) => {
-                try {
-                    const rtp = RtpPacket.deSerialize(message);
-                    this.#client.sendAudioRtp(rtp);
-                }
-                catch {
-                    //
-                }
-                return null;
-            });
-            this.#logger.log(loggerPrefix.trim(), returnAudioStream);
-            const inputArgs = ffmpegOptions.input ?? [];
-            const returnAudioCodecs = transformAudioStreamInfoToFFmpegArgs(returnAudioStream);
-            const ff = new FfmpegProcess({
-                ffmpegArgs: [
-                    '-hide_banner',
-                    '-protocol_whitelist',
-                    'pipe,udp,rtp,file,crypto',
-                    '-re',
-                    '-i',
-                    ...inputArgs,
-                    ...returnAudioCodecs,
-                    '-flags',
-                    '+global_header',
-                    '-f',
-                    'rtp',
-                    `rtp://127.0.0.1:${audioOutForwarder.port}`,
-                ],
-                ffmpegPath: ffmpegOptions.ffmpegPath,
-                startedCallback: () => {
-                    this.onStarted.next();
-                },
-                exitCallback: () => this.#callEnded(),
-                logLabel: loggerPrefix,
-                logger: {
-                    error: (...args) => this.#logger.error(...args),
-                    trace: (...args) => this.#logger.trace(...args),
-                },
-            });
-            this.onCallEnded.pipe(take(1)).subscribe(() => ff.stop());
-        }
-        catch (error) {
-            this.stop();
-            throw error;
-        }
-    }
-    stop() {
-        this.#callEnded();
-    }
-    sendAudioPacket(rtp) {
-        this.#client.sendAudioRtp(rtp);
-    }
-    #bindConnection() {
-        this.#client.onVideoRtp.subscribe(this.onVideoRtp);
-        this.#client.onAudioRtp.subscribe(this.onAudioRtp);
-        this.#client.onClose.subscribe((error) => {
-            if (error) {
-                this.#callError(error);
-            }
-            else {
-                this.#callEnded();
-            }
-        });
-    }
-    #callError(error) {
-        this.onError.next(error);
-        this.#callEnded();
-    }
-    #callEnded() {
-        if (this.#hasEnded) {
-            return;
-        }
-        this.#hasEnded = true;
-        this.unsubscribe();
-        this.#client.close();
-        this.onCallEnded.next();
-        this.#ff?.stop();
-        this.#audioSplitter.close();
-        this.#videoSplitter.close();
-        this.#returnAudioSplitter.close();
-    }
-}
-//# sourceMappingURL=rtsp-session.js.map

package/dist/server/src/camera/streaming/rtsp-session.js.map

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

@@ -1,44 +0,0 @@
-import { Subscribed } from '@camera.ui/common';
-import { ReplaySubject, Subject } from 'rxjs';
-import type { CameraInput, LoggerService, WebRTCConnectionOptions } from '@camera.ui/types';
-import type { Observable } from 'rxjs';
-import type { RtpPacket } from 'werift';
-import type { CameraDevice } from '../device.js';
-export interface Go2RtcOfferMessage {
-    type: 'webrtc/offer';
-    value: Sdp;
-}
-export interface Go2RtcAnswerMessage {
-    type: 'webrtc/answer';
-    value: Sdp;
-}
-export interface Go2RtcIceCandidateMessage {
-    type: 'webrtc/candidate';
-    value: IceCandiate;
-}
-export type Sdp = string;
-export type IceCandiate = string;
-export interface ErrorMessage {
-    type: 'error';
-    value: string;
-}
-export type Go2RtcIncomingMessage = Go2RtcAnswerMessage | Go2RtcIceCandidateMessage | ErrorMessage;
-export type Go2RtcOutgoingMessage = Go2RtcOfferMessage | Go2RtcIceCandidateMessage;
-export declare class WebrtcConnection extends Subscribed {
-    #private;
-    onCameraConnected: ReplaySubject<void>;
-    onCallAnswered: ReplaySubject<string>;
-    onCallEnded: ReplaySubject<void>;
-    onError: ReplaySubject<void>;
-    onMessage: ReplaySubject<{
-        method: string;
-    }>;
-    onWsOpen: Observable<unknown>;
-    onAudioRtp: Subject<RtpPacket>;
-    onVideoRtp: Subject<RtpPacket>;
-    constructor(cameraDevice: CameraDevice, source: CameraInput, logger: LoggerService, options?: WebRTCConnectionOptions);
-    upgradeConnection(source: CameraInput): Promise<void>;
-    sendAudioPacket(rtp: RtpPacket): void;
-    stop(): void;
-    requestKeyFrame(): void;
-}

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

@@ -1,177 +0,0 @@
-import { Subscribed } from '@camera.ui/common';
-import { ReplaySubject, Subject, fromEvent } from 'rxjs';
-import { concatMap } from 'rxjs/operators';
-import { WebSocket } from 'ws';
-import { WeriftPeerConnection } from './peer-connection.js';
-export class WebrtcConnection extends Subscribed {
-    onCameraConnected;
-    onCallAnswered;
-    onCallEnded;
-    onError;
-    onMessage;
-    onWsOpen;
-    onAudioRtp;
-    onVideoRtp;
-    #onOfferSent;
-    #pc;
-    #ws;
-    #hasEnded = false;
-    #cameraDevice;
-    #source;
-    #logger;
-    #options;
-    get #logPrefix() {
-        return `[${this.#cameraDevice.name}]`;
-    }
-    constructor(cameraDevice, source, logger, options = {}) {
-        super();
-        this.#cameraDevice = cameraDevice;
-        this.#source = source;
-        this.#logger = logger;
-        this.#options = options;
-        this.#initializeConnection(this.#source);
-    }
-    async upgradeConnection(source) {
-        // Close existing WebSocket and PeerConnection
-        this.unsubscribe();
-        this.#ws.close();
-        this.#pc.close();
-        // Reinitialize connection
-        this.#initializeConnection(source);
-    }
-    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?.();
-    }
-    #initializeConnection(source) {
-        const cameraSource = this.#cameraDevice.sources.find((s) => s.name === source.name);
-        const webRtcSource = cameraSource?.urls.ws.webrtc;
-        if (!webRtcSource) {
-            throw new Error(`Invalid source: ${source.name}`);
-        }
-        this.onCameraConnected = new ReplaySubject(1);
-        this.onCallAnswered = new ReplaySubject(1);
-        this.onCallEnded = new ReplaySubject(1);
-        this.onError = new ReplaySubject(1);
-        this.onMessage = new ReplaySubject();
-        this.#onOfferSent = new ReplaySubject(1);
-        this.#ws = new WebSocket(webRtcSource, {
-            rejectUnauthorized: false,
-        });
-        if (this.#options.createPeerConnection) {
-            this.#pc = this.#options.createPeerConnection();
-            this.onAudioRtp = new Subject();
-            this.onVideoRtp = new Subject();
-        }
-        else {
-            this.#pc = new WeriftPeerConnection(this.#cameraDevice, this.#logger);
-            this.onAudioRtp = this.#pc.onAudioRtp;
-            this.onVideoRtp = this.#pc.onVideoRtp;
-        }
-        this.onWsOpen = fromEvent(this.#ws, 'open');
-        this.#setupEventListeners();
-    }
-    #setupEventListeners() {
-        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((iceCandidate) => {
-            this.#sendMessage({
-                type: 'webrtc/candidate',
-                value: iceCandidate.toJSON().candidate || '',
-            });
-        }));
-    }
-    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

@@ -1,19 +0,0 @@
-import { SubscribedPublic } from '@camera.ui/common/utils';
-import { ReplaySubject, Subject } from 'rxjs';
-import { RtpPacket } from 'werift';
-import type { CameraInput, FfmpegOptions, ProbeStream, ReturnAudioFFmpegOptions, WebRTCConnectionOptions, WeriftSession as WeriftSessionInterface } from '@camera.ui/types';
-import type { CameraDevice } from '../device.js';
-export declare class WeriftSession extends SubscribedPublic implements WeriftSessionInterface {
-    #private;
-    readonly onStarted: ReplaySubject<void>;
-    readonly onError: ReplaySubject<Error>;
-    readonly onCallEnded: ReplaySubject<void>;
-    readonly onVideoRtp: Subject<RtpPacket>;
-    readonly onAudioRtp: Subject<RtpPacket>;
-    constructor(cameraDevice: CameraDevice, source: CameraInput, options?: WebRTCConnectionOptions);
-    startTranscoding(ffmpegOptions: FfmpegOptions, streamProbe?: ProbeStream): Promise<void>;
-    transcodeReturnAudio(ffmpegOptions: ReturnAudioFFmpegOptions, streamProbe?: ProbeStream): Promise<void>;
-    stop(): void;
-    sendAudioPacket(rtp: RtpPacket | Buffer): void;
-    requestKeyFrame(): void;
-}