@camera.ui/server 0.0.49 → 0.0.50-internal.0

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

@@ -0,0 +1,213 @@
+import { Subscribed } from '@camera.ui/common';
+import { ReplaySubject, Subject, interval, merge } from 'rxjs';
+import { MediaStreamTrack, RTCPeerConnection, RTCRtpCodecParameters } from 'werift';
+export 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;
+    #logger;
+    #pc;
+    get #logPrefix() {
+        return `[${this.#cameraDevice.name}]`;
+    }
+    constructor(cameraDevice, logger) {
+        super();
+        this.#cameraDevice = cameraDevice;
+        this.#logger = logger;
+        const pc = (this.#pc = this.#createPeerCoonection());
+        pc.addTransceiver(this.returnAudioTrack, {
+            direction: 'sendonly',
+        });
+        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(() => {
+                    if (track.ssrc) {
+                        videoTransceiver.receiver.sendRtcpPLI(track.ssrc).catch((error) => this.#logger.error(this.#logPrefix, error));
+                    }
+                }));
+                this.requestKeyFrame();
+            });
+        });
+        this.#pc.onIceCandidate.subscribe((iceCandidate) => {
+            if (iceCandidate && this.#isLocalCandidate(iceCandidate.candidate)) {
+                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);
+    }
+    async addIceCandidate(candidate) {
+        if (this.#isLocalCandidate(candidate.candidate)) {
+            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 = ['vp8', 'vp9', 'h264', 'av1'];
+        const prefferedAudioCodec = ['opus', 'g722', 'pcmu', 'pcma'];
+        const video = [];
+        const audio = [];
+        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/H265',
+                        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,
+            },
+            bundlePolicy: 'max-bundle',
+            iceAdditionalHostAddresses: ['127.0.0.1'],
+            iceUseIpv4: true,
+        });
+    }
+    #isLocalCandidate(candidateString) {
+        const parts = candidateString.split(' ');
+        const ip = parts[4];
+        const type = parts[7];
+        return ip === '127.0.0.1' || type === 'host';
+    }
+}
+//# sourceMappingURL=peer-connection.js.map

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

@@ -0,0 +1 @@
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package/dist/server/src/camera/streaming/rtsp-session.d.ts

@@ -0,0 +1,19 @@
+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

@@ -0,0 +1,253 @@
+import { FfmpegProcess, generateAudioStreamInfo, generateSdp, generateVideoStreamInfo, reservePorts, RtpSplitter, transformAudioStreamInfoToFFmpegArgs, } from '@camera.ui/common/camera';
+import { getCleanSdp, SubscribedPublic } from '@camera.ui/common/utils';
+import { RTSPClient } from '@camera.ui/rtsp/rtspClient';
+import { ReplaySubject, Subject } from 'rxjs';
+import { take, tap } 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;
+    constructor(cameraDevice, source, options) {
+        super();
+        this.#cameraDevice = cameraDevice;
+        this.#logger = cameraDevice.logger;
+        this.#source = this.#cameraDevice.sources.find((s) => s.name === source.name);
+        const url = options?.url ?? this.#source.generateRTSPUrl();
+        this.#client = new RTSPClient({ 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!');
+            }
+            await this.startStream();
+            const audioTrack = this.#client.getAudioTrack();
+            const videoTrack = this.#client.getVideoTrack();
+            if (!audioTrack && !videoTrack) {
+                throw new Error('No audio or video tracks found! Can not start transcoding!');
+            }
+            await this.#audioSplitter.prepare('udp4');
+            await this.#videoSplitter.prepare('udp4');
+            if (!this.#audioSplitter.port || !this.#videoSplitter.port) {
+                throw new Error('No ports available for audio and video splitter! Can not start transcoding!');
+            }
+            const videoPort = await this.#reservePort(1);
+            const audioPort = await this.#reservePort(1);
+            const videoStreamInfos = videoTrack ? generateVideoStreamInfo([videoTrack.mediaSource]) : [];
+            const audioStreamInfos = audioTrack ? generateAudioStreamInfo([audioTrack.mediaSource]) : [];
+            const parsedSdp = generateSdp(videoStreamInfos, audioStreamInfos);
+            const audioStream = audioStreamInfos[0];
+            const videoStream = videoStreamInfos[0];
+            const audioInputArgs = audioTrack && transcodeAudioStream && audioStream ? ['-acodec', audioStream.ffmpegCodec] : [];
+            const videoInputArgs = videoTrack && transcodeVideoStream && videoStream ? ['-vcodec', videoStream.ffmpegCodec] : [];
+            const audioOutputArgs = audioTrack && transcodeAudioStream ? (ffmpegOptions.audio ?? ['-acodec', audioStream.ffmpegCodec]) : [];
+            const videoOutputArgs = videoTrack && transcodeVideoStream ? (ffmpegOptions.video ?? ['-vcodec', videoStream.ffmpegCodec]) : [];
+            const inputArgs = ffmpegOptions.input ?? [];
+            const outputArgs = ffmpegOptions.output ?? [];
+            const inputSdp = getCleanSdp(parsedSdp, transcodeVideoStream, 'recvonly')
+                .replace(/m=audio \d+/, `m=audio ${audioPort}`)
+                .replace(/m=video \d+/, `m=video ${videoPort}`);
+            this.#logger.trace(loggerPrefix + 'Start transcoding with SDP:\n', inputSdp);
+            const ff = new FfmpegProcess({
+                ffmpegArgs: [
+                    '-hide_banner',
+                    '-protocol_whitelist',
+                    'pipe,udp,rtp,file,crypto',
+                    '-reorder_queue_size',
+                    '0',
+                    ...audioInputArgs,
+                    ...videoInputArgs,
+                    '-f',
+                    'sdp',
+                    ...inputArgs,
+                    '-i',
+                    'pipe:',
+                    ...audioOutputArgs,
+                    ...videoOutputArgs,
+                    ...outputArgs,
+                ],
+                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 ${signal}`));
+                    }
+                    else {
+                        this.#callEnded();
+                    }
+                },
+                logLabel: loggerPrefix,
+                logger: {
+                    error: (...args) => this.#logger.error(...args),
+                    trace: (...args) => this.#logger.trace(...args),
+                },
+            });
+            if (transcodeAudioStream) {
+                this.addSubscriptions(this.onAudioRtp
+                    .pipe(tap((rtp) => {
+                    return this.#audioSplitter.send(rtp.serialize(), {
+                        port: audioPort,
+                        address: '127.0.0.1',
+                    });
+                }))
+                    .subscribe());
+            }
+            if (transcodeVideoStream) {
+                this.addSubscriptions(this.onVideoRtp
+                    .pipe(tap((rtp) => {
+                    return this.#videoSplitter.send(rtp.serialize(), {
+                        port: videoPort,
+                        address: '127.0.0.1',
+                    });
+                }))
+                    .subscribe());
+            }
+            this.onCallEnded.pipe(take(1)).subscribe(() => ff.stop());
+            ff.writeStdin(inputSdp);
+        }
+        catch (error) {
+            this.stop();
+            throw error;
+        }
+    }
+    async transcodeReturnAudio(ffmpegOptions) {
+        if (this.#hasEnded) {
+            return;
+        }
+        const loggerPrefix = ffmpegOptions.logPrefix ? `[${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) {
+        if (this.#hasEnded) {
+            return;
+        }
+        this.#client.sendAudioRtp(rtp);
+    }
+    async #reservePort(bufferPorts = 0) {
+        const ports = await reservePorts({ count: bufferPorts + 1 });
+        return ports[0];
+    }
+    #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.#audioSplitter.close();
+        this.#videoSplitter.close();
+        this.#returnAudioSplitter.close();
+    }
+}
+//# sourceMappingURL=rtsp-session.js.map

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

@@ -0,0 +1 @@
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package/dist/server/src/camera/streaming/webrtc-connection.d.ts

@@ -0,0 +1,44 @@
+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;
+}