@camera.ui/server 0.0.49 → 0.0.50-internal.1

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

@@ -0,0 +1,177 @@
+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

@@ -0,0 +1,19 @@
+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;
+}

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

@@ -0,0 +1,246 @@
+import { FfmpegProcess, reservePorts, RtpSplitter, transformAudioStreamInfoToFFmpegArgs } from '@camera.ui/common/camera';
+import { getCleanSdp, isWebRtcCompatibleAudioCodec, isWebRtcCompatibleVideoCodec, SubscribedPublic } from '@camera.ui/common/utils';
+import { firstValueFrom, ReplaySubject, Subject } from 'rxjs';
+import { take, tap } from 'rxjs/operators';
+import { RtpPacket } from 'werift';
+import { WebrtcConnection } from './webrtc-connection.js';
+export class WeriftSession 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;
+    #webrtcConnection;
+    constructor(cameraDevice, source, options = {}) {
+        super();
+        this.#cameraDevice = cameraDevice;
+        this.#logger = cameraDevice.logger;
+        this.#source = this.#cameraDevice.sources.find((s) => s.name === source.name);
+        this.#webrtcConnection = new WebrtcConnection(this.#cameraDevice, source, this.#logger, options);
+        this.#bindConnection();
+    }
+    async startTranscoding(ffmpegOptions, streamProbe) {
+        if (this.#hasEnded) {
+            return;
+        }
+        const transcodeAudioStream = ffmpegOptions.audio !== undefined && ffmpegOptions.audio.length > 0;
+        const transcodeVideoStream = ffmpegOptions.video !== undefined && ffmpegOptions.video.length > 0;
+        const loggerPrefix = (ffmpegOptions.logPrefix ?? (transcodeAudioStream && transcodeVideoStream)) ? 'Transcoding' : transcodeAudioStream ? 'Audio' : 'Video';
+        try {
+            if (!transcodeAudioStream && !transcodeVideoStream) {
+                throw new Error('No transcoding options provided! Can not start transcoding!');
+            }
+            const sdp = await Promise.race([firstValueFrom(this.#webrtcConnection.onCallAnswered), firstValueFrom(this.onCallEnded)]);
+            if (!sdp) {
+                throw new Error('No SDP received! Can not start transcoding!');
+            }
+            const probeStream = streamProbe ?? (await this.#source.probeStream({ video: true, audio: ['pcma', 'opus'] }));
+            if (!probeStream) {
+                throw new Error('Could not probe stream! Can not start transcoding!');
+            }
+            const audioStream = probeStream.audio.filter((audio) => audio.direction === 'sendonly' && isWebRtcCompatibleAudioCodec(audio.codec))[0];
+            const videoStream = probeStream.video.filter((video) => video.direction === 'sendonly' && isWebRtcCompatibleVideoCodec(video.codec))[0];
+            if (!audioStream && !videoStream) {
+                throw new Error('No sendonly audio or video streams 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 or video splitter! Can not start transcoding!');
+            }
+            const videoPort = await this.#reservePort(1);
+            const audioPort = await this.#reservePort(1);
+            const audioInputArgs = audioStream && transcodeAudioStream && audioStream ? ['-acodec', audioStream.ffmpegCodec] : [];
+            const videoInputArgs = videoStream && transcodeVideoStream && videoStream ? ['-vcodec', videoStream.ffmpegCodec] : [];
+            const audioOutputArgs = audioStream && transcodeAudioStream ? (ffmpegOptions.audio ?? ['-acodec', audioStream.ffmpegCodec]) : [];
+            const videoOutputArgs = videoStream && transcodeVideoStream ? (ffmpegOptions.video ?? ['-vcodec', videoStream.ffmpegCodec]) : [];
+            const inputArgs = ffmpegOptions.input ?? [];
+            const outputArgs = ffmpegOptions.output ?? [];
+            const inputSdp = getCleanSdp(probeStream.sdp, transcodeVideoStream, 'sendonly', true)
+                .replace(/m=audio \d+/, `m=audio ${audioPort}`)
+                .replace(/m=video \d+/, `m=video ${videoPort}`);
+            this.#logger.trace('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);
+            this.requestKeyFrame();
+        }
+        catch (error) {
+            this.stop();
+            throw error;
+        }
+    }
+    async transcodeReturnAudio(ffmpegOptions, streamProbe) {
+        if (this.#hasEnded) {
+            return;
+        }
+        const loggerPrefix = ffmpegOptions.logPrefix ?? 'Return Audio';
+        try {
+            const probeStream = streamProbe ?? (await this.#source.probeStream({ microphone: true }));
+            if (!probeStream) {
+                this.stop();
+                throw new Error('Could not probe stream! Can not start transcoding!');
+            }
+            const audioStreams = probeStream.audio.filter((audio) => audio.direction === 'recvonly');
+            const returnAudioStream = audioStreams[0];
+            if (!returnAudioStream) {
+                this.stop();
+                throw new Error('No recvonly audio stream found! Can not start transcoding!');
+            }
+            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 transcoding!');
+            }
+            audioOutForwarder.addMessageHandler(({ message }) => {
+                const rtp = RtpPacket.deSerialize(message);
+                this.#webrtcConnection.sendAudioPacket(rtp);
+                return null;
+            });
+            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: (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),
+                },
+            });
+            this.onCallEnded.pipe(take(1)).subscribe(() => ff.stop());
+        }
+        catch (error) {
+            this.stop();
+            throw error;
+        }
+    }
+    stop() {
+        this.#callEnded();
+    }
+    sendAudioPacket(rtp) {
+        if (this.#hasEnded) {
+            return;
+        }
+        this.#webrtcConnection.sendAudioPacket(Buffer.isBuffer(rtp) ? RtpPacket.deSerialize(rtp) : rtp);
+    }
+    requestKeyFrame() {
+        this.#webrtcConnection.requestKeyFrame();
+    }
+    async #reservePort(bufferPorts = 0) {
+        const ports = await reservePorts({ count: bufferPorts + 1 });
+        return ports[0];
+    }
+    #handleVideo(rtp) {
+        this.onVideoRtp.next(rtp);
+    }
+    #handleAudio(rtp) {
+        this.onAudioRtp.next(rtp);
+    }
+    #bindConnection() {
+        this.addSubscriptions(this.#webrtcConnection.onVideoRtp.subscribe(this.#handleVideo.bind(this)), this.#webrtcConnection.onAudioRtp.subscribe(this.#handleAudio.bind(this)), this.#webrtcConnection.onCallEnded.subscribe(() => this.#callEnded()));
+    }
+    #callError(error) {
+        this.onError.next(error);
+    }
+    #callEnded() {
+        if (this.#hasEnded) {
+            return;
+        }
+        this.#hasEnded = true;
+        this.unsubscribe();
+        this.onCallEnded.next();
+        this.#webrtcConnection.stop();
+        this.#audioSplitter.close();
+        this.#videoSplitter.close();
+        this.#returnAudioSplitter.close();
+    }
+}
+//# sourceMappingURL=werift-session.js.map

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

@@ -0,0 +1 @@
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package/dist/server/src/camera/types.d.ts

@@ -1,11 +1,17 @@
-import type { FrameData, FrameMetadata, MotionState, SetValues, StateValues } from '@camera.ui/types';
-export interface FrameState extends MotionState {
-    frameData: FrameData;
-    metadata: FrameMetadata;
+import type { AudioSetEvent, AudioState, BatterySetEvent, BatteryState, CameraDevice, CameraDeviceCapability, DoorbellSetEvent, DoorbellState, LightSetEvent, LightState, MotionSetEvent, MotionState, ObjectSetEvent, ObjectState, ProbeConfig, ProbeStream, PTZInterface, SirenSetEvent, SirenState, SnapshotInterface, StreamingInterface } from '@camera.ui/types';
+import type { RefreshedStates } from '../rpc/types.js';
+import type { FunctionPropertyNames } from '../types.js';
+export type States = LightState | MotionState | AudioState | ObjectState | DoorbellState | SirenState | BatteryState;
+export type Sets = LightSetEvent | MotionSetEvent | AudioSetEvent | ObjectSetEvent | DoorbellSetEvent | SirenSetEvent | BatterySetEvent;
+export interface CameraDeviceInterface extends Omit<Pick<CameraDevice, FunctionPropertyNames<CameraDevice>>, 'extend' | 'addCapabilities' | 'removeCapabilities' | 'getFrames' | 'onStateChange' | 'onPropertyChange' | 'onCapabilitiesChange'> {
+    refreshStates(): Promise<RefreshedStates>;
+    snapshot(sourceId: string, forceNew?: boolean): Promise<ArrayBuffer | undefined>;
+    probeStream(sourceId: string, probeConfig?: ProbeConfig, refresh?: boolean): Promise<ProbeStream | undefined>;
+    addPluginCapabilities(pluginId: string, capabilities: CameraDeviceCapability[]): void;
+    removePluginCapabilities(pluginId: string, capabilities: CameraDeviceCapability[]): void;
+}
+export type CameraInterfaces = PTZInterface & SnapshotInterface & StreamingInterface;
+export interface PluginCapabilities {
+    pluginId: string;
+    capabilities: CameraDeviceCapability[];
 }
-export type CameraStateCallbacks = {
-    [K in keyof StateValues]: (event: StateValues[K]) => void;
-};
-export type StateValue<T extends keyof StateValues> = StateValues[T];
-export type SetValue<T extends keyof SetValues> = SetValues[T];
-export type OnChangeCallback<T extends keyof StateValues> = (newEvent: StateValues[T], oldEvent: StateValues[T]) => void;

package/dist/server/src/camera/utils.d.ts

@@ -0,0 +1,9 @@
+import type { AudioStreamInfo, VideoStreamInfo } from '@camera.ui/types';
+import type { Go2RTCProducer } from '../api/go2rtc/types.js';
+export declare function generateAudioStreamInfo(producers: Go2RTCProducer[]): AudioStreamInfo[];
+export declare function generateVideoStreamInfo(producers: Go2RTCProducer[]): VideoStreamInfo[];
+export declare function extractSdpInfo(producers: Go2RTCProducer[]): {
+    video: Record<string, Record<string, any>>;
+    audio: Record<string, Record<string, any>>;
+};
+export declare function createProfileLevelId(profile: string, level: number): string | undefined;