@agentuity/frontend 1.0.0 → 1.0.2

package/dist/webrtc-manager.d.ts

@@ -0,0 +1,420 @@
+import type { WebRTCConnectionState, WebRTCDisconnectReason, DataChannelConfig, DataChannelState, ConnectionQualitySummary, RecordingOptions, RecordingHandle, TrackSource as CoreTrackSource } from '@agentuity/core';
+/**
+ * Track source interface extended for browser environment.
+ */
+export interface TrackSource extends Omit<CoreTrackSource, 'getStream'> {
+    getStream(): Promise<MediaStream>;
+}
+/**
+ * User media (camera/microphone) track source.
+ */
+export declare class UserMediaSource implements TrackSource {
+    private constraints;
+    readonly type: "user-media";
+    private stream;
+    constructor(constraints?: MediaStreamConstraints);
+    getStream(): Promise<MediaStream>;
+    stop(): void;
+}
+/**
+ * Display media (screen share) track source.
+ */
+export declare class DisplayMediaSource implements TrackSource {
+    private constraints;
+    readonly type: "display-media";
+    private stream;
+    constructor(constraints?: DisplayMediaStreamOptions);
+    getStream(): Promise<MediaStream>;
+    stop(): void;
+}
+/**
+ * Custom stream track source - wraps a user-provided MediaStream.
+ */
+export declare class CustomStreamSource implements TrackSource {
+    private stream;
+    readonly type: "custom";
+    constructor(stream: MediaStream);
+    getStream(): Promise<MediaStream>;
+    stop(): void;
+}
+/**
+ * Callbacks for WebRTC client state changes and events.
+ * All callbacks are optional - only subscribe to events you care about.
+ */
+export interface WebRTCClientCallbacks {
+    /**
+     * Called on every state transition.
+     */
+    onStateChange?: (from: WebRTCConnectionState, to: WebRTCConnectionState, reason?: string) => void;
+    /**
+     * Called when connected to at least one peer.
+     */
+    onConnect?: () => void;
+    /**
+     * Called when disconnected from all peers.
+     */
+    onDisconnect?: (reason: WebRTCDisconnectReason) => void;
+    /**
+     * Called when local media stream is acquired.
+     */
+    onLocalStream?: (stream: MediaStream) => void;
+    /**
+     * Called when a remote media stream is received.
+     */
+    onRemoteStream?: (peerId: string, stream: MediaStream) => void;
+    /**
+     * Called when a new track is added to a stream.
+     */
+    onTrackAdded?: (peerId: string, track: MediaStreamTrack, stream: MediaStream) => void;
+    /**
+     * Called when a track is removed from a stream.
+     */
+    onTrackRemoved?: (peerId: string, track: MediaStreamTrack) => void;
+    /**
+     * Called when a peer joins the room.
+     */
+    onPeerJoined?: (peerId: string) => void;
+    /**
+     * Called when a peer leaves the room.
+     */
+    onPeerLeft?: (peerId: string) => void;
+    /**
+     * Called when negotiation starts with a peer.
+     */
+    onNegotiationStart?: (peerId: string) => void;
+    /**
+     * Called when negotiation completes with a peer.
+     */
+    onNegotiationComplete?: (peerId: string) => void;
+    /**
+     * Called for each ICE candidate generated.
+     */
+    onIceCandidate?: (peerId: string, candidate: RTCIceCandidateInit) => void;
+    /**
+     * Called when ICE connection state changes for a peer.
+     */
+    onIceStateChange?: (peerId: string, state: string) => void;
+    /**
+     * Called when an error occurs.
+     */
+    onError?: (error: Error, state: WebRTCConnectionState) => void;
+    /**
+     * Called when a data channel is opened.
+     */
+    onDataChannelOpen?: (peerId: string, label: string) => void;
+    /**
+     * Called when a data channel is closed.
+     */
+    onDataChannelClose?: (peerId: string, label: string) => void;
+    /**
+     * Called when a message is received on a data channel.
+     *
+     * **Note:** String messages are automatically parsed as JSON if valid.
+     * - If the message is valid JSON, `data` will be the parsed object/array/value
+     * - If the message is not valid JSON, `data` will be the raw string
+     * - Binary messages (ArrayBuffer) are passed through unchanged
+     *
+     * To distinguish between parsed JSON and raw strings, check the type:
+     * ```ts
+     * onDataChannelMessage: (peerId, label, data) => {
+     *   if (typeof data === 'string') {
+     *     // Raw string (failed JSON parse)
+     *   } else if (data instanceof ArrayBuffer) {
+     *     // Binary data
+     *   } else {
+     *     // Parsed JSON object/array/primitive
+     *   }
+     * }
+     * ```
+     */
+    onDataChannelMessage?: (peerId: string, label: string, data: string | ArrayBuffer | unknown) => void;
+    /**
+     * Called when a data channel error occurs.
+     */
+    onDataChannelError?: (peerId: string, label: string, error: Error) => void;
+    /**
+     * Called when screen sharing starts.
+     */
+    onScreenShareStart?: () => void;
+    /**
+     * Called when screen sharing stops.
+     */
+    onScreenShareStop?: () => void;
+    /**
+     * Called when a reconnect attempt is scheduled.
+     */
+    onReconnecting?: (attempt: number) => void;
+    /**
+     * Called after a successful reconnection.
+     */
+    onReconnected?: () => void;
+    /**
+     * Called when reconnect attempts are exhausted.
+     */
+    onReconnectFailed?: () => void;
+}
+/**
+ * Options for WebRTCManager
+ */
+export interface WebRTCManagerOptions {
+    /** WebSocket signaling URL */
+    signalUrl: string;
+    /** Room ID to join */
+    roomId: string;
+    /** Whether this peer is "polite" in perfect negotiation (default: auto-determined) */
+    polite?: boolean;
+    /** ICE servers configuration */
+    iceServers?: RTCIceServer[];
+    /**
+     * Media source configuration.
+     * - `false`: Data-only mode (no media)
+     * - `MediaStreamConstraints`: Use getUserMedia with these constraints
+     * - `TrackSource`: Use a custom track source
+     * Default: { video: true, audio: true }
+     */
+    media?: MediaStreamConstraints | TrackSource | false;
+    /**
+     * Data channels to create when connection is established.
+     * Only the offerer (late joiner) creates channels; the answerer receives them.
+     */
+    dataChannels?: DataChannelConfig[];
+    /**
+     * Callbacks for state changes and events.
+     */
+    callbacks?: WebRTCClientCallbacks;
+    /**
+     * Whether to auto-reconnect on WebSocket/ICE failures (default: true)
+     */
+    autoReconnect?: boolean;
+    /**
+     * Maximum reconnection attempts before giving up (default: 5)
+     */
+    maxReconnectAttempts?: number;
+    /**
+     * Connection timeout in ms for connecting/negotiating (default: 30000)
+     */
+    connectionTimeout?: number;
+    /**
+     * ICE gathering timeout in ms (default: 10000)
+     */
+    iceGatheringTimeout?: number;
+}
+/**
+ * WebRTC manager state
+ */
+export interface WebRTCManagerState {
+    state: WebRTCConnectionState;
+    peerId: string | null;
+    remotePeerIds: string[];
+    isAudioMuted: boolean;
+    isVideoMuted: boolean;
+    isScreenSharing: boolean;
+}
+/**
+ * Framework-agnostic WebRTC connection manager with multi-peer mesh networking,
+ * perfect negotiation, media/data channel handling, and screen sharing.
+ *
+ * Uses an explicit state machine for connection lifecycle:
+ * - idle: No resources allocated, ready to connect
+ * - connecting: Acquiring media + opening WebSocket
+ * - signaling: In room, waiting for peer(s)
+ * - negotiating: SDP/ICE exchange in progress with at least one peer
+ * - connected: At least one peer is connected
+ *
+ * @example
+ * ```ts
+ * const manager = new WebRTCManager({
+ *   signalUrl: 'wss://example.com/call/signal',
+ *   roomId: 'my-room',
+ *   callbacks: {
+ *     onStateChange: (from, to, reason) => console.log(`${from} → ${to}`, reason),
+ *     onConnect: () => console.log('Connected!'),
+ *     onRemoteStream: (peerId, stream) => { remoteVideos[peerId].srcObject = stream; },
+ *   },
+ * });
+ *
+ * await manager.connect();
+ * ```
+ */
+export declare class WebRTCManager {
+    private ws;
+    private localStream;
+    private trackSource;
+    private previousVideoTrack;
+    private peerId;
+    private peers;
+    private isAudioMuted;
+    private isVideoMuted;
+    private isScreenSharing;
+    private _state;
+    private isConnecting;
+    private basePolite;
+    private options;
+    private callbacks;
+    private reconnectManager;
+    private isReconnecting;
+    private intentionalClose;
+    private connectionTimeoutId;
+    private recordings;
+    constructor(options: WebRTCManagerOptions);
+    /**
+     * Current connection state
+     */
+    get state(): WebRTCConnectionState;
+    /**
+     * Get current manager state
+     */
+    getState(): WebRTCManagerState;
+    /**
+     * Get local media stream
+     */
+    getLocalStream(): MediaStream | null;
+    /**
+     * Get remote media streams keyed by peer ID
+     */
+    getRemoteStreams(): Map<string, MediaStream>;
+    /**
+     * Get a specific peer's remote stream
+     */
+    getRemoteStream(peerId: string): MediaStream | null;
+    /**
+     * Whether this manager is in data-only mode (no media streams).
+     */
+    get isDataOnly(): boolean;
+    /**
+     * Get connected peer count
+     */
+    get peerCount(): number;
+    private setState;
+    private mapToDisconnectReason;
+    private handleStateTimeouts;
+    private startConnectionTimeout;
+    private clearConnectionTimeout;
+    private handleTimeout;
+    private shouldAutoReconnect;
+    private updateConnectionState;
+    private send;
+    /**
+     * Connect to the signaling server and start the call
+     */
+    connect(): Promise<void>;
+    private ensureLocalStream;
+    private openWebSocket;
+    private handleConnectionLoss;
+    private scheduleReconnect;
+    private reconnect;
+    private handleSignalingMessage;
+    private createPeerSession;
+    private createOffer;
+    private handleRemoteSDP;
+    private isOffererFor;
+    private handleRemoteICE;
+    private closePeerSession;
+    private cleanupPeerSessions;
+    private scheduleIceGatheringTimeout;
+    private setupDataChannel;
+    /**
+     * Create a new data channel to all connected peers.
+     */
+    createDataChannel(config: DataChannelConfig): Map<string, RTCDataChannel>;
+    /**
+     * Get a data channel by label from a specific peer.
+     */
+    getDataChannel(peerId: string, label: string): RTCDataChannel | undefined;
+    /**
+     * Get all open data channel labels.
+     */
+    getDataChannelLabels(): string[];
+    /**
+     * Get the state of a data channel for a specific peer.
+     */
+    getDataChannelState(peerId: string, label: string): DataChannelState | null;
+    /**
+     * Send a string message to all peers on a data channel.
+     */
+    sendString(label: string, data: string): boolean;
+    /**
+     * Send a string message to a specific peer.
+     */
+    sendStringTo(peerId: string, label: string, data: string): boolean;
+    /**
+     * Send binary data to all peers on a data channel.
+     */
+    sendBinary(label: string, data: ArrayBuffer | Uint8Array): boolean;
+    /**
+     * Send binary data to a specific peer.
+     */
+    sendBinaryTo(peerId: string, label: string, data: ArrayBuffer | Uint8Array): boolean;
+    /**
+     * Send JSON data to all peers on a data channel.
+     */
+    sendJSON(label: string, data: unknown): boolean;
+    /**
+     * Send JSON data to a specific peer.
+     */
+    sendJSONTo(peerId: string, label: string, data: unknown): boolean;
+    /**
+     * Close a specific data channel on all peers.
+     */
+    closeDataChannel(label: string): boolean;
+    /**
+     * Mute or unmute audio
+     */
+    muteAudio(muted: boolean): void;
+    /**
+     * Mute or unmute video
+     */
+    muteVideo(muted: boolean): void;
+    /**
+     * Start screen sharing, replacing the current video track.
+     * @param options - Display media constraints
+     */
+    startScreenShare(options?: DisplayMediaStreamOptions): Promise<void>;
+    /**
+     * Stop screen sharing and restore the previous video track.
+     */
+    stopScreenShare(): Promise<void>;
+    /**
+     * Get raw stats for a peer connection.
+     */
+    getRawStats(peerId: string): Promise<RTCStatsReport | null>;
+    /**
+     * Get raw stats for all peer connections.
+     */
+    getAllRawStats(): Promise<Map<string, RTCStatsReport>>;
+    /**
+     * Get a normalized quality summary for a peer connection.
+     */
+    getQualitySummary(peerId: string): Promise<ConnectionQualitySummary | null>;
+    /**
+     * Get quality summaries for all peer connections.
+     */
+    getAllQualitySummaries(): Promise<Map<string, ConnectionQualitySummary>>;
+    private parseStatsToSummary;
+    private findMatchingReport;
+    private getStatReport;
+    /**
+     * Start recording a stream.
+     * @param streamId - 'local' for local stream, or a peer ID for remote stream
+     * @param options - Recording options
+     */
+    startRecording(streamId: string, options?: RecordingOptions): RecordingHandle | null;
+    /**
+     * Check if a stream is being recorded.
+     */
+    isRecording(streamId: string): boolean;
+    /**
+     * Stop all recordings and return the blobs.
+     */
+    stopAllRecordings(): Promise<Map<string, Blob>>;
+    private selectMimeType;
+    /**
+     * End the call and disconnect from all peers
+     */
+    hangup(): void;
+    /**
+     * Clean up all resources
+     */
+    dispose(): void;
+}
+//# sourceMappingURL=webrtc-manager.d.ts.map

package/dist/webrtc-manager.d.ts.map

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