@evolve.ui/core-components 1.0.0

package/WebRTCPlayer-C-sTtskR.mjs.map

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+{"version":3,"file":"WebRTCPlayer-C-sTtskR.mjs","names":[],"sources":["../src/WebRTCPlayer/types.ts","../src/WebRTCPlayer/adaptor/NativeWebRTCAdaptor.ts","../src/WebRTCPlayer/hooks/useWebRTCStream.ts","../src/WebRTCPlayer/WebRTCPlayer.tsx"],"sourcesContent":["/**\n * Configuration constants for WebRTC player\n */\nexport const DEFAULT_WS_PORT = 5080;\nexport const DEFAULT_WSS_PORT = 5443;\nexport const DEFAULT_PATH = '/WebRTCAppEE/websocket';\n// Interval between reconnect attempts. The reconnect loop retries indefinitely\n// (no max-attempt cap) by design: a live stream can resume minutes after\n// dropping, so the player keeps attempting until the stream returns or the\n// component unmounts.\nexport const RECONNECT_INTERVAL = 5000; // 5 seconds\nexport const PING_INTERVAL = 30000; // 30 seconds\n// Cap on how long to wait for a host-supplied fetchStreamId() before falling back\n// to the current ID, so a hung callback can't permanently block reconnection.\n// INVARIANT: must stay strictly below RECONNECT_INTERVAL. The reconnect loop\n// awaits fetchStreamId() on each tick, so a timeout >= the interval would let the\n// next tick start while the previous one is still awaiting, producing overlapping\n// reconnect attempts.\nexport const FETCH_STREAM_ID_TIMEOUT = 3000; // 3 seconds\n\n/**\n * WebRTC connection state\n */\nexport type WebRTCConnectionState = 'idle' | 'connecting' | 'connected' | 'playing' | 'error' | 'reconnecting';\n\n/**\n * Error tokens emitted by NativeWebRTCAdaptor via its callbackError channel.\n * Exported so the adaptor (producer) and the hook's friendly-message map\n * (consumer) reference the same literals and can't silently drift apart — a\n * rename here is a compile error on both sides rather than a silent fallthrough.\n */\nexport const WebRTCErrorToken = {\n    failedToCreateAnswer: 'Failed to create answer',\n    failedToSetLocalDescription: 'Failed to set local description',\n    failedToSetRemoteDescription: 'Failed to set remote description',\n    iceConnectionFailed: 'IceConnectionFailed',\n    peerConnectionFailed: 'PeerConnectionFailed',\n    webSocketError: 'WebSocket error',\n    webSocketNotConnected: 'WebSocketNotConnected',\n    webSocketNotSupported: 'WebSocketNotSupported',\n} as const;\n\nexport type WebRTCErrorToken = (typeof WebRTCErrorToken)[keyof typeof WebRTCErrorToken];\n\n/**\n * WebRTC adaptor callback information types\n */\nexport type WebRTCCallbackInfo =\n    | 'initialized'\n    | 'play_started'\n    | 'play_finished'\n    | 'publish_started'\n    | 'publish_finished'\n    | 'closed'\n    | 'ice_connection_state_changed'\n    | 'updated_stats'\n    | 'data_received';\n\n/**\n * Configuration for WebRTC player\n */\nexport type WebRTCConfig = {\n    /** Whether to enable debug logging */\n    debug?: boolean;\n    /**\n     * Optional callback invoked before each reconnect attempt to obtain a fresh\n     * stream ID. A stream that restarts on the server is often issued a new ID,\n     * so reconnecting with the original `streamId` would target a dead stream.\n     * Return the latest ID (sync or async); return a falsy value to keep the\n     * current one. Errors are swallowed and the current ID is reused.\n     */\n    fetchStreamId?: () => string | Promise<string>;\n    /** WebRTC server hostname */\n    hostname: string;\n    /** Stream ID from Ant Media Server */\n    streamId: string;\n    /** WebSocket port for non-SSL connections */\n    wsPort?: number;\n    /** WebSocket port for SSL connections */\n    wssPort?: number;\n};\n\n/**\n * WebRTC player state returned by the hook\n */\nexport type WebRTCPlayerState = {\n    /** Current connection state */\n    connectionState: WebRTCConnectionState;\n    /** Error message if connection failed */\n    error: string | null;\n    /** Whether the stream is currently loading/connecting */\n    loading: boolean;\n    /** Reference to attach to video element */\n    videoRef: React.RefObject<HTMLVideoElement>;\n};\n","/* eslint-disable no-console, @typescript-eslint/no-unnecessary-condition */\n/**\n * Native WebRTC Adaptor for Ant Media Server\n *\n * This is a lightweight TypeScript implementation for WebRTC streaming from Ant Media Server.\n * It uses only native browser APIs (WebSocket, RTCPeerConnection) without external dependencies.\n *\n * This adaptor only supports PLAY mode (consuming streams), not publishing.\n *\n * Note: Console statements are intentionally enabled in this file for WebRTC debugging,\n * as they are crucial for troubleshooting connection issues in production environments.\n */\n\nimport { WebRTCErrorToken } from '../types';\n\n/**\n * WebSocket message types from Ant Media Server\n */\ntype WebSocketCommand =\n    | 'start'\n    | 'takeCandidate'\n    | 'takeConfiguration'\n    | 'stop'\n    | 'error'\n    | 'notification'\n    | 'streamInformation'\n    | 'pong';\n\n/**\n * WebSocket message structure\n */\ntype WebSocketMessage = {\n    candidate?: string;\n    command: WebSocketCommand;\n    definition?: string;\n    label?: number;\n    sdp?: string;\n    streamId?: string;\n    type?: RTCSdpType;\n};\n\n/**\n * Configuration for the WebRTC adaptor\n */\nexport type NativeWebRTCAdaptorConfig = {\n    /** Callback for status updates */\n    callback: (info: string, obj?: unknown) => void;\n    /** Callback for errors */\n    callbackError: (error: string, errorObject?: unknown) => void;\n    /** Enable debug logging */\n    debug?: boolean;\n    /** ICE server configuration */\n    peerconnection_config?: RTCConfiguration;\n    /** ID of the video element to attach the stream */\n    remoteVideoId?: string;\n    /** SDP constraints */\n    sdp_constraints?: RTCOfferOptions;\n    /** WebSocket URL for Ant Media Server */\n    websocket_url: string;\n};\n\n/**\n * WebSocket adaptor for managing the connection to Ant Media Server\n */\nclass WebSocketAdaptor {\n    private wsConn: WebSocket;\n    private connected = false;\n    private callback: (info: string, obj?: unknown) => void;\n    private callbackError: (error: string, errorObject?: unknown) => void;\n    private closePeerConnection: (streamId: string) => void;\n    private debug: boolean;\n    private takeCandidate: (streamId: string, label: number, candidate: string) => void;\n    private takeConfiguration: (streamId: string, sdp: string, type: RTCSdpType) => void;\n\n    constructor(config: {\n        callback: (info: string, obj?: unknown) => void;\n        callbackError: (error: string, errorObject?: unknown) => void;\n        closePeerConnection: (streamId: string) => void;\n        debug: boolean;\n        takeCandidate: (streamId: string, label: number, candidate: string) => void;\n        takeConfiguration: (streamId: string, sdp: string, type: RTCSdpType) => void;\n        websocket_url: string;\n    }) {\n        this.callback = config.callback;\n        this.callbackError = config.callbackError;\n        this.closePeerConnection = config.closePeerConnection;\n        this.debug = config.debug;\n        this.takeCandidate = config.takeCandidate;\n        this.takeConfiguration = config.takeConfiguration;\n\n        this.wsConn = new WebSocket(config.websocket_url);\n        this.setupEventHandlers();\n    }\n\n    /**\n     * Setup WebSocket event handlers\n     */\n    private setupEventHandlers(): void {\n        this.wsConn.onopen = () => {\n            if (this.debug) {\n                console.log('[NativeWebRTC] WebSocket connected');\n            }\n            this.connected = true;\n            this.callback('initialized');\n        };\n\n        this.wsConn.onmessage = ({ data }) => {\n            let obj: WebSocketMessage;\n            try {\n                obj = JSON.parse(data) as WebSocketMessage;\n            } catch (error) {\n                // Malformed/hostile frame — drop it rather than throwing out of the handler.\n                if (this.debug) {\n                    console.error('[NativeWebRTC] Failed to parse WebSocket message:', error);\n                }\n                return;\n            }\n            if (this.debug) {\n                console.log('[NativeWebRTC] WebSocket message:', obj);\n            }\n\n            switch (obj.command) {\n                case 'start':\n                    // 'start' command is for publishing mode - we only support play mode\n                    if (this.debug) {\n                        console.log('[NativeWebRTC] Start command received (ignored in play-only mode)');\n                    }\n                    break;\n\n                case 'takeCandidate':\n                    if (obj.streamId && obj.label !== undefined && obj.candidate) {\n                        if (this.debug) {\n                            console.log(`[NativeWebRTC] Received ICE candidate for stream ${obj.streamId}`);\n                        }\n                        this.takeCandidate(obj.streamId, obj.label, obj.candidate);\n                    }\n                    break;\n\n                case 'takeConfiguration':\n                    if (obj.streamId && obj.sdp && obj.type) {\n                        if (this.debug) {\n                            console.log(\n                                `[NativeWebRTC] Received remote description for stream ${obj.streamId}, type: ${obj.type}`\n                            );\n                        }\n                        this.takeConfiguration(obj.streamId, obj.sdp, obj.type);\n                    }\n                    break;\n\n                case 'stop':\n                    if (this.debug) {\n                        console.log('[NativeWebRTC] Stop command received');\n                    }\n                    if (obj.streamId) {\n                        this.closePeerConnection(obj.streamId);\n                    }\n                    break;\n\n                case 'error':\n                    if (this.debug) {\n                        console.error('[NativeWebRTC] Error from server:', obj.definition);\n                    }\n                    this.callbackError(obj.definition || 'Unknown error');\n                    break;\n\n                case 'notification':\n                    this.callback(obj.definition || 'notification', obj);\n                    if (obj.definition === 'play_finished' || obj.definition === 'publish_finished') {\n                        if (obj.streamId) {\n                            this.closePeerConnection(obj.streamId);\n                        }\n                    }\n                    break;\n\n                case 'streamInformation':\n                    this.callback('streamInformation', obj);\n                    break;\n\n                case 'pong':\n                    if (this.debug) {\n                        console.log('[NativeWebRTC] Pong received');\n                    }\n                    break;\n\n                default:\n                    if (this.debug) {\n                        console.log('[NativeWebRTC] Unhandled message:', obj);\n                    }\n            }\n        };\n\n        this.wsConn.onerror = (error) => {\n            console.error('[NativeWebRTC] WebSocket error:', error);\n            this.callbackError(WebRTCErrorToken.webSocketError);\n        };\n\n        this.wsConn.onclose = (event) => {\n            this.connected = false;\n            if (this.debug) {\n                console.log('[NativeWebRTC] WebSocket connection closed');\n            }\n            this.callback('closed', event);\n        };\n    }\n\n    /**\n     * Send a message through the WebSocket\n     */\n    send(text: string): void {\n        if (this.wsConn.readyState !== WebSocket.OPEN) {\n            this.callbackError(WebRTCErrorToken.webSocketNotConnected);\n            return;\n        }\n        this.wsConn.send(text);\n    }\n\n    /**\n     * Close the WebSocket connection\n     */\n    close(): void {\n        this.wsConn.close();\n    }\n\n    /**\n     * Check if WebSocket is connected\n     */\n    isConnected(): boolean {\n        return this.connected;\n    }\n}\n\n/**\n * Native WebRTC Adaptor for Ant Media Server (Play Mode Only)\n */\nexport class NativeWebRTCAdaptor {\n    private callback: (info: string, obj?: unknown) => void;\n    private callbackError: (error: string, errorObject?: unknown) => void;\n    private debug: boolean;\n    private websocket_url: string;\n    private peerconnection_config: RTCConfiguration;\n    private sdp_constraints: RTCOfferOptions;\n    private remoteVideoId?: string;\n    private webSocketAdaptor: WebSocketAdaptor | null = null;\n    private remotePeerConnection: Record<string, RTCPeerConnection> = {};\n    private playStreamId: string[] = [];\n    private pingInterval: ReturnType<typeof setInterval> | null = null;\n\n    constructor(config: NativeWebRTCAdaptorConfig) {\n        this.callback = config.callback;\n        this.callbackError = config.callbackError;\n        this.debug = config.debug || false;\n        this.websocket_url = config.websocket_url;\n        this.peerconnection_config = config.peerconnection_config || {\n            iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n        };\n        this.sdp_constraints = config.sdp_constraints || {\n            offerToReceiveAudio: true,\n            offerToReceiveVideo: true,\n        };\n        this.remoteVideoId = config.remoteVideoId;\n\n        this.init();\n    }\n\n    /**\n     * Initialize the WebSocket connection\n     */\n    private init(): void {\n        if (!('WebSocket' in window)) {\n            console.error('[NativeWebRTC] WebSocket not supported');\n            this.callbackError(WebRTCErrorToken.webSocketNotSupported);\n            return;\n        }\n\n        if (!this.webSocketAdaptor?.isConnected()) {\n            this.webSocketAdaptor = new WebSocketAdaptor({\n                callback: this.callback,\n                callbackError: this.callbackError,\n                closePeerConnection: this.closePeerConnection,\n                debug: this.debug,\n                takeCandidate: this.takeCandidate,\n                takeConfiguration: this.takeConfiguration,\n                websocket_url: this.websocket_url,\n            });\n        }\n    }\n\n    /**\n     * Start playing a stream\n     */\n    play(streamId: string): void {\n        this.playStreamId.push(streamId);\n        const jsCmd = { command: 'play', streamId };\n        if (this.debug) {\n            console.log(`[NativeWebRTC] Sending play command for stream: ${streamId}`);\n        }\n        this.webSocketAdaptor?.send(JSON.stringify(jsCmd));\n    }\n\n    /**\n     * Stop playing a stream\n     */\n    stop(streamId: string): void {\n        this.closePeerConnection(streamId);\n        const jsCmd = { command: 'stop', streamId };\n        if (this.debug) {\n            console.log(`[NativeWebRTC] Sending stop command for stream: ${streamId}`);\n        }\n        this.webSocketAdaptor?.send(JSON.stringify(jsCmd));\n    }\n\n    /**\n     * Send a ping to keep the connection alive\n     */\n    ping(streamId: string): void {\n        const jsCmd = { command: 'ping', streamId };\n        if (this.debug) {\n            console.log('[NativeWebRTC] Sending ping');\n        }\n        this.webSocketAdaptor?.send(JSON.stringify(jsCmd));\n    }\n\n    /**\n     * Start sending periodic pings\n     */\n    startPinging(streamId: string): void {\n        this.pingInterval = setInterval(() => {\n            this.ping(streamId);\n        }, 30000); // 30 seconds\n    }\n\n    /**\n     * Stop sending pings\n     */\n    stopPinging(): void {\n        if (this.pingInterval) {\n            clearInterval(this.pingInterval);\n            this.pingInterval = null;\n        }\n    }\n\n    /**\n     * Leave the stream and close connections\n     */\n    leave(streamId: string): void {\n        this.stopPinging();\n        const jsCmd = { command: 'leave', streamId };\n        if (this.debug) {\n            console.log(`[NativeWebRTC] Leaving stream: ${streamId}`);\n        }\n        this.webSocketAdaptor?.send(JSON.stringify(jsCmd));\n        this.closePeerConnection(streamId);\n        this.webSocketAdaptor?.close();\n    }\n\n    /**\n     * Close the WebSocket connection\n     */\n    closeWebSocket(): void {\n        this.webSocketAdaptor?.close();\n    }\n\n    /**\n     * Initialize peer connection for a stream\n     */\n    private initPeerConnection = (streamId: string): void => {\n        // eslint-disable-next-line @typescript-eslint/no-unnecessary-condition\n        if (this.remotePeerConnection[streamId]) {\n            return;\n        }\n\n        if (this.debug) {\n            console.log(`[NativeWebRTC] Initializing peer connection for stream: ${streamId}`);\n        }\n\n        const peerConnection = new RTCPeerConnection(this.peerconnection_config);\n        this.remotePeerConnection[streamId] = peerConnection;\n\n        // Handle ICE candidates\n        peerConnection.onicecandidate = (event) => {\n            this.iceCandidateReceived(event, streamId);\n        };\n\n        // Handle incoming tracks\n        peerConnection.ontrack = (event) => {\n            this.onTrack(event, streamId);\n        };\n\n        // Detect media-path failures. The WebSocket can stay open while the\n        // peer connection dies (e.g. the network drops mid-stream), which would\n        // otherwise freeze the video with no signal. Surfacing 'failed' as an\n        // error lets the consumer trigger reconnection. 'disconnected' is\n        // intentionally not treated as fatal — it is often transient and may\n        // recover on its own.\n        peerConnection.onconnectionstatechange = () => {\n            if (peerConnection.connectionState === 'failed') {\n                if (this.debug) {\n                    console.log('[NativeWebRTC] Peer connection failed');\n                }\n                this.callbackError(WebRTCErrorToken.peerConnectionFailed);\n            }\n        };\n\n        peerConnection.oniceconnectionstatechange = () => {\n            if (peerConnection.iceConnectionState === 'failed') {\n                if (this.debug) {\n                    console.log('[NativeWebRTC] ICE connection failed');\n                }\n                this.callbackError(WebRTCErrorToken.iceConnectionFailed);\n            }\n        };\n\n        if (this.debug) {\n            console.log(`[NativeWebRTC] Peer connection created for stream: ${streamId}`);\n        }\n    };\n\n    /**\n     * Close peer connection for a stream\n     */\n    private closePeerConnection = (streamId: string): void => {\n        const peerConnection = this.remotePeerConnection[streamId];\n        // eslint-disable-next-line @typescript-eslint/no-unnecessary-condition\n        if (peerConnection && peerConnection.signalingState !== 'closed') {\n            if (this.debug) {\n                console.log(`[NativeWebRTC] Closing peer connection for stream: ${streamId}`);\n            }\n            peerConnection.close();\n            delete this.remotePeerConnection[streamId];\n\n            const playStreamIndex = this.playStreamId.indexOf(streamId);\n            if (playStreamIndex !== -1) {\n                this.playStreamId.splice(playStreamIndex, 1);\n            }\n        }\n    };\n\n    /**\n     * Handle incoming track (video/audio stream)\n     */\n    private onTrack = (event: RTCTrackEvent, streamId: string): void => {\n        if (this.debug) {\n            console.log(`[NativeWebRTC] Track received for stream: ${streamId}`);\n        }\n\n        const remoteVideo = this.remoteVideoId ? document.getElementById(this.remoteVideoId) : null;\n\n        if (remoteVideo && remoteVideo instanceof HTMLVideoElement) {\n            if (remoteVideo.srcObject !== event.streams[0]) {\n                remoteVideo.srcObject = event.streams[0];\n                if (this.debug) {\n                    console.log('[NativeWebRTC] Remote stream attached to video element');\n                }\n            }\n        } else {\n            // If no video element specified, pass the stream through callback\n            const dataObj = {\n                streamId,\n                track: event.streams[0],\n            };\n            this.callback('newStreamAvailable', dataObj);\n        }\n    };\n\n    /**\n     * Handle ICE candidate\n     */\n    private iceCandidateReceived = (event: RTCPeerConnectionIceEvent, streamId: string): void => {\n        if (event.candidate) {\n            const jsCmd = {\n                candidate: event.candidate.candidate,\n                command: 'takeCandidate',\n                id: event.candidate.sdpMid,\n                label: event.candidate.sdpMLineIndex,\n                streamId,\n            };\n\n            if (this.debug) {\n                console.log(`[NativeWebRTC] Sending ICE candidate for stream: ${streamId}`);\n            }\n\n            this.webSocketAdaptor?.send(JSON.stringify(jsCmd));\n        }\n    };\n\n    /**\n     * Handle remote configuration (SDP)\n     */\n    private takeConfiguration = (streamId: string, sdp: string, type: RTCSdpType): void => {\n        this.initPeerConnection(streamId);\n        const peerConnection = this.remotePeerConnection[streamId];\n\n        if (!peerConnection) {\n            console.error('[NativeWebRTC] Peer connection not found');\n            return;\n        }\n\n        peerConnection\n            .setRemoteDescription(new RTCSessionDescription({ sdp, type }))\n            .then(() => {\n                if (this.debug) {\n                    console.log(`[NativeWebRTC] Remote description set successfully for stream: ${streamId}`);\n                }\n\n                if (type === 'offer') {\n                    // Create answer for the offer\n                    if (this.debug) {\n                        console.log(`[NativeWebRTC] Creating answer for stream: ${streamId}`);\n                    }\n                    peerConnection\n                        .createAnswer(this.sdp_constraints)\n                        .then((description) => {\n                            this.gotDescription(description, streamId);\n                        })\n                        .catch((error) => {\n                            console.error('[NativeWebRTC] Failed to create answer:', error);\n                            this.callbackError(WebRTCErrorToken.failedToCreateAnswer, error);\n                        });\n                }\n            })\n            .catch((error) => {\n                console.error('[NativeWebRTC] Failed to set remote description:', error);\n                this.callbackError(WebRTCErrorToken.failedToSetRemoteDescription, error);\n            });\n    };\n\n    /**\n     * Handle local description (SDP)\n     */\n    private gotDescription = (description: RTCSessionDescriptionInit, streamId: string): void => {\n        const peerConnection = this.remotePeerConnection[streamId];\n\n        if (!peerConnection) {\n            console.error('[NativeWebRTC] Peer connection not found');\n            return;\n        }\n\n        peerConnection\n            .setLocalDescription(description)\n            .then(() => {\n                if (this.debug) {\n                    console.log(`[NativeWebRTC] Local description set successfully for stream: ${streamId}`);\n                }\n\n                const jsCmd = {\n                    command: 'takeConfiguration',\n                    sdp: description.sdp,\n                    streamId,\n                    type: description.type,\n                };\n\n                this.webSocketAdaptor?.send(JSON.stringify(jsCmd));\n            })\n            .catch((error) => {\n                console.error('[NativeWebRTC] Failed to set local description:', error);\n                this.callbackError(WebRTCErrorToken.failedToSetLocalDescription, error);\n            });\n    };\n\n    /**\n     * Handle ICE candidate from server\n     */\n    private takeCandidate = (streamId: string, label: number, candidateSdp: string): void => {\n        const candidate = new RTCIceCandidate({\n            candidate: candidateSdp,\n            sdpMLineIndex: label,\n        });\n\n        this.initPeerConnection(streamId);\n        const peerConnection = this.remotePeerConnection[streamId];\n\n        if (!peerConnection) {\n            console.error('[NativeWebRTC] Peer connection not found');\n            return;\n        }\n\n        peerConnection\n            .addIceCandidate(candidate)\n            .then(() => {\n                if (this.debug) {\n                    console.log(`[NativeWebRTC] ICE candidate added for stream: ${streamId}`);\n                }\n            })\n            .catch((error) => {\n                console.error(`[NativeWebRTC] Failed to add ICE candidate for stream ${streamId}:`, error);\n            });\n    };\n}\n","import { useCallback, useEffect, useRef, useState } from 'react';\n\nimport { NativeWebRTCAdaptor } from '../adaptor/NativeWebRTCAdaptor';\nimport {\n    DEFAULT_PATH,\n    DEFAULT_WS_PORT,\n    DEFAULT_WSS_PORT,\n    FETCH_STREAM_ID_TIMEOUT,\n    PING_INTERVAL,\n    RECONNECT_INTERVAL,\n    WebRTCConfig,\n    WebRTCConnectionState,\n    WebRTCErrorToken,\n    WebRTCPlayerState,\n} from '../types';\n\n/**\n * Races a value/promise against a timeout, resolving to `undefined` if the\n * timeout wins. Used to prevent a hung host-supplied fetchStreamId() from\n * permanently blocking the reconnect loop.\n */\nconst withTimeout = async <T>(value: Promise<T> | T, ms: number): Promise<T | undefined> => {\n    let timer: ReturnType<typeof setTimeout> | undefined;\n    const valuePromise = Promise.resolve(value);\n    // If the timeout wins the race, the value promise is left dangling. Attach a\n    // no-op catch so a late rejection doesn't surface as an unhandledRejection.\n    // (An early rejection still propagates through Promise.race to the caller.)\n    valuePromise.catch(() => undefined);\n    try {\n        return await Promise.race([\n            valuePromise,\n            new Promise<undefined>((resolve) => {\n                timer = setTimeout(() => {\n                    resolve(undefined);\n                }, ms);\n            }),\n        ]);\n    } finally {\n        if (timer) {\n            clearTimeout(timer);\n        }\n    }\n};\n\n/**\n * Maps adaptor error tokens to user-facing copy so raw identifiers like\n * `IceConnectionFailed` never reach the error overlay. Keyed off the shared\n * WebRTCErrorToken constants so this map can't drift from what the adaptor emits.\n */\nconst FRIENDLY_ERROR_MESSAGES = new Map<string, string>([\n    [WebRTCErrorToken.failedToCreateAnswer, 'Connection lost'],\n    [WebRTCErrorToken.failedToSetLocalDescription, 'Connection lost'],\n    [WebRTCErrorToken.failedToSetRemoteDescription, 'Connection lost'],\n    [WebRTCErrorToken.iceConnectionFailed, 'Connection lost'],\n    [WebRTCErrorToken.peerConnectionFailed, 'Connection lost'],\n    [WebRTCErrorToken.webSocketError, 'Connection lost'],\n    [WebRTCErrorToken.webSocketNotConnected, 'Connection lost'],\n    [WebRTCErrorToken.webSocketNotSupported, 'Your browser does not support live streaming'],\n]);\n\nconst toFriendlyError = (raw: string): string => FRIENDLY_ERROR_MESSAGES.get(raw) ?? raw;\n\n/**\n * Custom hook for managing WebRTC stream connections to Ant Media Server\n *\n * This hook handles the complete lifecycle of a WebRTC stream connection including:\n * - WebSocket connection management\n * - RTCPeerConnection setup and negotiation\n * - Stream state management (loading, playing, error, reconnecting)\n * - Automatic reconnection on connection failure\n * - Proper cleanup on unmount\n *\n * @param config - Configuration object for the WebRTC connection\n * @returns WebRTC player state including video ref, loading state, playing state, error, and mute controls\n *\n * @example\n * ```tsx\n * const { videoRef, loading, playing, error } = useWebRTCStream({\n *   streamId: 'my-stream-id',\n *   hostname: 'media.example.com'\n * });\n *\n * return (\n *   <video ref={videoRef} autoPlay playsInline />\n * );\n * ```\n */\nexport const useWebRTCStream = ({\n    debug = false,\n    fetchStreamId,\n    hostname,\n    streamId,\n    wsPort = DEFAULT_WS_PORT,\n    wssPort = DEFAULT_WSS_PORT,\n}: WebRTCConfig): WebRTCPlayerState => {\n    const videoRef = useRef<HTMLVideoElement>(null);\n    const [loading, setLoading] = useState(true);\n    const [error, setError] = useState<string | null>(null);\n    const [connectionState, setConnectionState] = useState<WebRTCConnectionState>('idle');\n\n    const webRTCAdaptorRef = useRef<NativeWebRTCAdaptor | null>(null);\n    const reconnectIntervalRef = useRef<ReturnType<typeof setInterval> | null>(null);\n    const pingIntervalRef = useRef<ReturnType<typeof setInterval> | null>(null);\n    const isReconnectingRef = useRef(false);\n    // True while the component is mounted; lets async init bail after an await if\n    // the component has since unmounted (avoids setState/adaptor build post-teardown).\n    const isMountedRef = useRef(true);\n    // Monotonic token identifying the current connection attempt. Each\n    // initializeWebRTC() call claims a fresh token; if a newer attempt starts\n    // (reconnect tick, streamId prop change), the older one becomes stale and\n    // bails at the next checkpoint instead of building an adaptor or mutating\n    // shared state — preventing overlapping adaptors and stale-id connects.\n    const initTokenRef = useRef(0);\n    // Holds the latest initializeWebRTC so the reconnect loop never calls a stale\n    // closure (and so handleError/handleCallback don't need to depend on it).\n    /* istanbul ignore next -- @preserve: the default initializer is never invoked; an effect overwrites this ref on first commit, and no reconnect (the only caller) can fire before the mount effect creates an adaptor. */\n    const initializeWebRTCRef = useRef<() => void | Promise<void>>(() => undefined);\n    // The stream ID the adaptor is currently using. Defaults to the streamId prop\n    // and is refreshed by fetchStreamId() before each reconnect attempt.\n    const activeStreamIdRef = useRef(streamId);\n    // Latest fetchStreamId, held in a ref so the reconnect path always calls the\n    // current one without needing it in any dependency array.\n    const fetchStreamIdRef = useRef(fetchStreamId);\n    fetchStreamIdRef.current = fetchStreamId;\n\n    /**\n     * Generates the WebSocket URL based on the current page protocol\n     */\n    const getWebSocketUrl = useCallback(() => {\n        const isSecure = window.location.protocol === 'https:';\n        const protocol = isSecure ? 'wss' : 'ws';\n        const port = isSecure ? wssPort : wsPort;\n        const url = `${protocol}://${hostname}:${port}${DEFAULT_PATH}`;\n\n        if (debug) {\n            // eslint-disable-next-line no-console\n            console.log('[WebRTC] Generated WebSocket URL:', {\n                hostname,\n                isSecure,\n                location: window.location.protocol,\n                port,\n                protocol,\n                url,\n            });\n        }\n\n        return url;\n    }, [hostname, wsPort, wssPort, debug]);\n\n    /**\n     * Clears reconnection interval\n     */\n    const stopReconnecting = useCallback(() => {\n        if (reconnectIntervalRef.current) {\n            clearInterval(reconnectIntervalRef.current);\n            reconnectIntervalRef.current = null;\n        }\n        isReconnectingRef.current = false;\n    }, []);\n\n    /**\n     * Starts the reconnection loop if one is not already running.\n     *\n     * Any failure signal (WebSocket error, WebSocket close, or the stream\n     * finishing) funnels through here so that a dropped connection always\n     * triggers recovery. The loop is idempotent — guarded by isReconnectingRef —\n     * and is cleared by stopReconnecting() once the stream re-initializes or\n     * starts playing again.\n     *\n     * The loop retries indefinitely (no max-attempt cap) by design: a live stream\n     * can resume minutes after dropping, so we keep attempting every\n     * RECONNECT_INTERVAL until it returns or the component unmounts.\n     */\n    const startReconnecting = useCallback(() => {\n        if (isReconnectingRef.current) {\n            return;\n        }\n\n        if (debug) {\n            // eslint-disable-next-line no-console\n            console.log(`[WebRTC] Starting reconnection attempts (every ${RECONNECT_INTERVAL}ms)`);\n        }\n\n        isReconnectingRef.current = true;\n        reconnectIntervalRef.current = setInterval(() => {\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] 🔄 Attempting to reconnect...');\n            }\n            // initializeWebRTC sets the 'reconnecting' state while isReconnectingRef is true.\n            void initializeWebRTCRef.current();\n        }, RECONNECT_INTERVAL);\n    }, [debug]);\n\n    /**\n     * Starts ping interval to keep connection alive\n     */\n    const startPinging = useCallback(() => {\n        if (pingIntervalRef.current) {\n            clearInterval(pingIntervalRef.current);\n        }\n\n        pingIntervalRef.current = setInterval(() => {\n            const activeStreamId = activeStreamIdRef.current;\n            if (webRTCAdaptorRef.current && activeStreamId) {\n                try {\n                    webRTCAdaptorRef.current.ping(activeStreamId);\n                } catch (err) {\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.error('Error sending ping:', err);\n                    }\n                }\n            }\n        }, PING_INTERVAL);\n    }, [debug]);\n\n    /**\n     * Stops ping interval\n     */\n    const stopPinging = useCallback(() => {\n        if (pingIntervalRef.current) {\n            clearInterval(pingIntervalRef.current);\n            pingIntervalRef.current = null;\n        }\n    }, []);\n\n    /**\n     * Handles WebRTC adaptor callbacks\n     */\n    const handleCallback = useCallback(\n        (info: string, obj?: unknown) => {\n            // A late callback can arrive after unmount (e.g. closeWebSocket() during\n            // cleanup triggers onclose). Ignore it so we don't setState or re-arm the\n            // reconnect interval on a torn-down component.\n            if (!isMountedRef.current) {\n                return;\n            }\n\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log(`[WebRTC] Callback: ${info}`, obj);\n            }\n\n            switch (info) {\n                case 'initialized':\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.log('[WebRTC] ✅ Initialized - WebSocket connected, starting playback');\n                    }\n                    setConnectionState('connected');\n                    setLoading(true);\n                    stopReconnecting();\n                    startPinging();\n                    // Start playing the stream using the ID the adaptor connected with\n                    if (webRTCAdaptorRef.current && activeStreamIdRef.current) {\n                        if (debug) {\n                            // eslint-disable-next-line no-console\n                            console.log(`[WebRTC] Calling play() for streamId: ${activeStreamIdRef.current}`);\n                        }\n                        webRTCAdaptorRef.current.play(activeStreamIdRef.current);\n                    }\n                    break;\n\n                case 'play_started':\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.log('[WebRTC] ✅ Play started - Stream is now playing');\n                    }\n                    setConnectionState('playing');\n                    setLoading(false);\n                    // setPlaying(true);\n                    setError(null);\n                    stopReconnecting();\n                    break;\n\n                case 'play_finished':\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.log('[WebRTC] ⚠️ Play finished - Stream ended');\n                    }\n                    setConnectionState('error');\n                    setError('Stream ended');\n                    // A finished stream is treated as a recoverable failure: the\n                    // stream may simply have restarted on the server. Attempt to\n                    // reconnect rather than leaving the player dead.\n                    startReconnecting();\n                    break;\n\n                case 'closed':\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.log('[WebRTC] ❌ Connection closed');\n                    }\n                    setConnectionState('error');\n                    setError('Connection closed');\n                    // A dropped WebSocket is reported via callback('closed'), NOT\n                    // callbackError, so the reconnect loop must be started here too\n                    // — otherwise a network failure leaves the player stuck.\n                    startReconnecting();\n                    break;\n\n                case 'publish_started':\n                case 'publish_finished':\n                case 'ice_connection_state_changed':\n                case 'updated_stats':\n                case 'data_received':\n                default:\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.log(`[WebRTC] Unhandled callback: ${info}`, obj);\n                    }\n            }\n        },\n        [debug, startPinging, stopReconnecting, startReconnecting]\n    );\n\n    /**\n     * Handles WebRTC adaptor errors\n     */\n    const handleError = useCallback(\n        (errorMessage: string, errorObject?: unknown) => {\n            // Ignore late errors from an adaptor that outlived the component.\n            if (!isMountedRef.current) {\n                return;\n            }\n\n            // eslint-disable-next-line no-console\n            console.error('[WebRTC] ❌ Error:', errorMessage, errorObject);\n            setConnectionState('error');\n            setLoading(false);\n            setError(toFriendlyError(errorMessage));\n\n            // Start reconnection attempts if not already reconnecting\n            startReconnecting();\n        },\n        [startReconnecting]\n    );\n\n    /**\n     * Initializes the WebRTC adaptor\n     */\n    const initializeWebRTC = useCallback(async () => {\n        if (!activeStreamIdRef.current) {\n            // eslint-disable-next-line no-console\n            console.error('[WebRTC] ❌ No stream ID provided');\n            setError('No stream ID provided');\n            return;\n        }\n\n        // Claim this attempt. Any later attempt bumps the token, marking this one\n        // stale. `isCurrent()` is checked after every await so a superseded or\n        // unmounted attempt bails before mutating shared state / building an adaptor.\n        const token = ++initTokenRef.current;\n        const isCurrent = () => token === initTokenRef.current && isMountedRef.current;\n\n        // Clean up existing adaptor (using the ID it was created with, before we\n        // potentially swap in a freshly fetched one below).\n        if (webRTCAdaptorRef.current) {\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] Cleaning up existing adaptor');\n            }\n            try {\n                stopPinging();\n                webRTCAdaptorRef.current.stop(activeStreamIdRef.current);\n                webRTCAdaptorRef.current.closeWebSocket();\n            } catch (err) {\n                // eslint-disable-next-line no-console\n                console.error('[WebRTC] Error cleaning up previous adaptor:', err);\n            }\n            webRTCAdaptorRef.current = null;\n        }\n\n        // Preserve the 'reconnecting' state across the rebuild so the UI can\n        // surface it; a first-time connect uses 'connecting'.\n        const reconnecting = isReconnectingRef.current;\n        setConnectionState(reconnecting ? 'reconnecting' : 'connecting');\n        // On a reconnect tick keep `loading` false: the error overlay (and its\n        // \"Attempting to reconnect…\" caption) is gated behind `!loading`, and the\n        // failure handler that started the loop has already set `error`, so leaving\n        // loading false is what makes the reconnecting UI visible. Only a first-time\n        // connect shows the loading spinner.\n        setLoading(!reconnecting);\n\n        // On reconnect, give the host a chance to supply a fresh stream ID — a\n        // restarted stream is usually issued a new ID, so reusing the old one\n        // would just reconnect to a dead stream. Bounded by a timeout so a hung\n        // callback can't wedge the reconnect loop.\n        if (isReconnectingRef.current && fetchStreamIdRef.current) {\n            try {\n                const freshStreamId = await withTimeout(fetchStreamIdRef.current(), FETCH_STREAM_ID_TIMEOUT);\n                // A newer attempt (e.g. a streamId prop change) may have superseded\n                // us during the await. Don't clobber its ID or continue this build.\n                if (!isCurrent()) {\n                    return;\n                }\n                if (freshStreamId) {\n                    if (debug && freshStreamId !== activeStreamIdRef.current) {\n                        // eslint-disable-next-line no-console\n                        console.log(`[WebRTC] Refreshed stream ID: ${activeStreamIdRef.current} -> ${freshStreamId}`);\n                    }\n                    activeStreamIdRef.current = freshStreamId;\n                }\n            } catch (err) {\n                // eslint-disable-next-line no-console\n                console.error('[WebRTC] fetchStreamId failed, reusing current stream ID:', err);\n                if (!isCurrent()) {\n                    return;\n                }\n            }\n        }\n\n        // Final checkpoint before building: bail if unmounted or superseded.\n        if (!isCurrent()) {\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] Initialization superseded or unmounted, aborting');\n            }\n            return;\n        }\n\n        const activeStreamId = activeStreamIdRef.current;\n\n        if (debug) {\n            // eslint-disable-next-line no-console\n            console.log(`[WebRTC] 🚀 Initializing WebRTC for streamId: ${activeStreamId}`);\n        }\n\n        try {\n            const websocketUrl = getWebSocketUrl();\n\n            // Ensure video element has an ID before creating adaptor\n            if (videoRef.current && !videoRef.current.id) {\n                videoRef.current.id = 'webrtc-video-' + activeStreamId;\n            }\n            const videoId = videoRef.current?.id || 'webrtc-video';\n\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] Creating WebRTCAdaptor with config:', {\n                    isPlayMode: true,\n                    remoteVideoId: videoId,\n                    streamId: activeStreamId,\n                    websocket_url: websocketUrl,\n                });\n            }\n\n            webRTCAdaptorRef.current = new NativeWebRTCAdaptor({\n                callback: handleCallback,\n                callbackError: handleError,\n                debug,\n                peerconnection_config: {\n                    iceServers: [{ urls: 'stun:stun.l.google.com:19302' }],\n                },\n                remoteVideoId: videoId,\n                sdp_constraints: {\n                    offerToReceiveAudio: true,\n                    offerToReceiveVideo: true,\n                },\n                websocket_url: websocketUrl,\n            });\n\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] ✅ WebRTCAdaptor created, waiting for connection...');\n            }\n        } catch (err) {\n            // eslint-disable-next-line no-console\n            console.error('[WebRTC] ❌ Failed to initialize WebRTC:', err);\n            handleError('Failed to initialize WebRTC', err);\n        }\n    }, [debug, getWebSocketUrl, handleCallback, handleError, stopPinging]);\n\n    /**\n     * Track mount status so async init can bail after an await once the component\n     * has unmounted. Kept separate from the streamId-scoped cleanup effect below\n     * so a prop change never flips this to unmounted.\n     */\n    useEffect(() => {\n        isMountedRef.current = true;\n        return () => {\n            isMountedRef.current = false;\n        };\n    }, []);\n\n    /**\n     * Keep the ref pointed at the latest initializeWebRTC so the reconnect loop\n     * always reconnects with the current streamId/config rather than a closure\n     * captured when the loop first started.\n     */\n    useEffect(() => {\n        initializeWebRTCRef.current = initializeWebRTC;\n    }, [initializeWebRTC]);\n\n    /**\n     * Initialize WebRTC connection on mount or when streamId changes\n     */\n    useEffect(() => {\n        if (streamId) {\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] StreamId changed or component mounted, scheduling initialization');\n            }\n            // A streamId prop change is authoritative and supersedes any ID\n            // previously fetched during reconnection.\n            activeStreamIdRef.current = streamId;\n            // Small delay to ensure video element is mounted\n            const timeoutId = setTimeout(() => {\n                void initializeWebRTC();\n            }, 100);\n\n            return () => {\n                clearTimeout(timeoutId);\n            };\n        }\n        return undefined;\n    }, [streamId, initializeWebRTC, debug]);\n\n    /**\n     * Cleanup on unmount\n     */\n    useEffect(() => {\n        return () => {\n            if (debug) {\n                // eslint-disable-next-line no-console\n                console.log('[WebRTC] 🧹 Cleaning up on unmount');\n            }\n            stopReconnecting();\n            stopPinging();\n\n            if (webRTCAdaptorRef.current && activeStreamIdRef.current) {\n                try {\n                    webRTCAdaptorRef.current.stop(activeStreamIdRef.current);\n                    webRTCAdaptorRef.current.closeWebSocket();\n                    if (debug) {\n                        // eslint-disable-next-line no-console\n                        console.log('[WebRTC] ✅ Cleanup complete');\n                    }\n                } catch (err) {\n                    // eslint-disable-next-line no-console\n                    console.error('[WebRTC] Error during cleanup:', err);\n                }\n            }\n\n            webRTCAdaptorRef.current = null;\n        };\n    }, [streamId, debug, stopReconnecting, stopPinging]);\n\n    /**\n     * Assign the stream to the video element when it's available\n     */\n    useEffect(() => {\n        if (videoRef.current && webRTCAdaptorRef.current) {\n            const remoteVideo = videoRef.current;\n\n            // The Ant Media adaptor will set the srcObject automatically\n            // We just need to ensure the video element has the correct ID\n            if (!remoteVideo.id) {\n                remoteVideo.id = 'webrtc-video-' + streamId;\n            }\n        }\n    }, [streamId]);\n\n    return {\n        connectionState,\n        error,\n        loading,\n        videoRef: videoRef as React.RefObject<HTMLVideoElement>,\n    };\n};\n","import { cn } from '@evolve.ui/theme-system';\nimport React, { useEffect, useMemo, useState } from 'react';\n\nimport { Button } from '../Button/Button';\nimport { IconButton } from '../IconButton/IconButton';\nimport { ArrowSquareOut } from '../Icons';\nimport { LoadingSpinner } from '../LoadingSpinner/LoadingSpinner';\nimport { BodySecondary, Caption, H5 } from '../Typography/Text/Text';\nimport { useWebRTCStream } from './hooks/useWebRTCStream';\nimport { DEFAULT_WS_PORT, DEFAULT_WSS_PORT } from './types';\n\nexport type WebRTCPlayerProps = {\n    /** Allow fullscreen mode (default: true) */\n    allowFullscreen?: boolean;\n    /** Allow picture-in-picture mode (default: true) */\n    allowPictureInPicture?: boolean;\n    /** Additional CSS classes for the container */\n    className?: string;\n    /** Test ID for testing */\n    'data-testid'?: string;\n    /** Enable debug logging */\n    debug?: boolean;\n    /**\n     * Optional callback invoked before each reconnect attempt to obtain a fresh\n     * stream ID. A stream that restarts on the server is usually issued a new ID,\n     * so reconnecting with the original `streamId` would target a dead stream.\n     * Return the latest ID (sync or async); return a falsy value to keep the\n     * current one.\n     */\n    fetchStreamId?: () => string | Promise<string>;\n    /** WebRTC server hostname */\n    hostname: string;\n    /** Callback when user clicks the \"Contact Support\" button on error */\n    onContactSupportOnError?: () => void;\n    /** Callback when stream encounters an error */\n    onError?: (error: string) => void;\n    /** Callback when user enters or exits fullscreen mode */\n    onFullscreenToggle?: (isFullscreen: boolean) => void;\n    /** Callback when user clicks \"open in new window\" button */\n    onOpenInNewWindow?: () => void;\n    /** Callback when video is paused */\n    onPause?: () => void;\n    /** Callback when user enters or exits picture-in-picture mode */\n    onPipToggle?: (isPip: boolean) => void;\n    /** Callback when stream starts playing */\n    onPlay?: () => void;\n    /** Callback when mute state changes */\n    onToggleMute?: (isMuted: boolean) => void;\n    /** Callback when volume changes */\n    onVolumeChange?: (volume: number) => void;\n    /** Optional placeholder image URL to display before stream starts (defaults to black background) */\n    placeholderUrl?: string;\n    /**\n     * Show native HTML5 video controls bar (default: true).\n     * Set to `false` for compact surfaces where the host app renders custom controls (e.g. mute, expand).\n     */\n    showControls?: boolean;\n    /** Stream ID from Ant Media Server */\n    streamId: string;\n    /** WebSocket port for non-SSL connections (defaults to 5080) */\n    wsPort?: number;\n    /** WebSocket port for SSL connections (defaults to 5443) */\n    wssPort?: number;\n};\n\n/**\n * WebRTC video player component for streaming from Ant Media Server\n *\n * This component provides a simple interface for playing live WebRTC streams from\n * Ant Media Server. It handles connection management, error recovery, and provides\n * a clean UI with loading states and error messages.\n *\n * Features:\n * - Automatic connection to Ant Media Server\n * - Loading spinner while connecting\n * - Error state display with automatic reconnection\n * - Native HTML5 video controls with volume, fullscreen, and picture-in-picture\n * - Responsive container with aspect ratio preservation\n * - Configurable hostname and ports for different server setups\n *\n * @example\n * ```tsx\n * <WebRTCPlayer\n *   hostname=\"media.example.com\"\n *   streamId=\"my-stream-id\"\n *   onPlay={() => console.log('Stream started')}\n *   onError={(error) => console.error('Stream error:', error)}\n * />\n * ```\n *\n * @example Custom server configuration\n * ```tsx\n * <WebRTCPlayer\n *   streamId=\"my-stream-id\"\n *   hostname=\"custom-server.com\"\n *   wsPort={8080}\n *   wssPort={8443}\n * />\n * ```\n */\nexport const WebRTCPlayer = ({\n    allowFullscreen = true,\n    allowPictureInPicture = true,\n    className,\n    'data-testid': testid = 'webrtc-player',\n    debug = false,\n    fetchStreamId,\n    hostname,\n    onContactSupportOnError,\n    onError,\n    onFullscreenToggle,\n    onOpenInNewWindow,\n    onPause,\n    onPipToggle,\n    onPlay,\n    onToggleMute,\n    onVolumeChange,\n    placeholderUrl,\n    showControls = true,\n    streamId,\n    wsPort = DEFAULT_WS_PORT,\n    wssPort = DEFAULT_WSS_PORT,\n}: WebRTCPlayerProps) => {\n    const { connectionState, error, loading, videoRef } = useWebRTCStream({\n        debug,\n        fetchStreamId,\n        hostname,\n        streamId,\n        wsPort,\n        wssPort,\n    });\n\n    const [showContent, setShowContent] = useState(false);\n    const hasUserInteractedRef = React.useRef(false);\n\n    const controlsList = useMemo(\n        () => [!allowFullscreen && 'nofullscreen', 'nodownload'].filter(Boolean).join(' ') || undefined,\n        [allowFullscreen]\n    );\n\n    // Track when user has hovered over the video\n    const handleUserInteraction = () => {\n        hasUserInteractedRef.current = true;\n    };\n\n    // Trigger fade-in animation when loading completes\n    useEffect(() => {\n        if (!loading) {\n            // Small delay to ensure smooth transition\n            const timer = setTimeout(() => setShowContent(true), 50);\n            return () => clearTimeout(timer);\n        }\n        setShowContent(false);\n        return undefined;\n    }, [loading]);\n\n    // Call onError callback when an error occurs\n    useEffect(() => {\n        if (error && onError) {\n            onError(error);\n        }\n    }, [error, onError]);\n\n    // Listen for fullscreen changes\n    useEffect(() => {\n        if (!onFullscreenToggle || !allowFullscreen) {\n            return;\n        }\n\n        const handleFullscreenChange = () => {\n            if (!hasUserInteractedRef.current) {\n                return;\n            }\n            const isFullscreen = Boolean(document.fullscreenElement);\n            onFullscreenToggle(isFullscreen);\n        };\n\n        document.addEventListener('fullscreenchange', handleFullscreenChange);\n        return () => {\n            document.removeEventListener('fullscreenchange', handleFullscreenChange);\n        };\n    }, [allowFullscreen, onFullscreenToggle]);\n\n    // Listen for picture-in-picture changes\n    useEffect(() => {\n        const video = videoRef.current;\n        if (!onPipToggle || !allowPictureInPicture || !Boolean(video)) {\n            return;\n        }\n\n        const handleEnterPip = () => {\n            if (!hasUserInteractedRef.current) {\n                return;\n            }\n            onPipToggle(true);\n        };\n        const handleLeavePip = () => {\n            if (!hasUserInteractedRef.current) {\n                return;\n            }\n            onPipToggle(false);\n        };\n\n        video.addEventListener('enterpictureinpicture', handleEnterPip);\n        video.addEventListener('leavepictureinpicture', handleLeavePip);\n\n        return () => {\n            video.removeEventListener('enterpictureinpicture', handleEnterPip);\n            video.removeEventListener('leavepictureinpicture', handleLeavePip);\n        };\n    }, [allowPictureInPicture, onPipToggle, videoRef]);\n\n    return (\n        <div\n            className={cn(\n                'relative w-full overflow-hidden rounded-container bg-background-inverse',\n                'aspect-video', // 16:9 aspect ratio\n                className\n            )}\n            data-testid={testid}\n        >\n            {/* eslint-disable-next-line jsx-a11y/media-has-caption */}\n            <video\n                aria-label=\"Live video stream\"\n                autoPlay\n                className={cn(\n                    'h-full w-full object-cover transition-opacity duration-300',\n                    showContent && !error ? 'opacity-100' : 'opacity-0'\n                )}\n                controls={showControls}\n                controlsList={controlsList}\n                disablePictureInPicture={!allowPictureInPicture}\n                muted\n                onMouseEnter={handleUserInteraction}\n                onPause={() => {\n                    if (!hasUserInteractedRef.current) {\n                        return;\n                    }\n                    onPause?.();\n                }}\n                onPlay={() => {\n                    if (!hasUserInteractedRef.current) {\n                        return;\n                    }\n                    onPlay?.();\n                }}\n                onTouchStart={handleUserInteraction}\n                onVolumeChange={(e: React.SyntheticEvent<HTMLVideoElement>) => {\n                    if (!hasUserInteractedRef.current) {\n                        return;\n                    }\n                    const video = e.currentTarget;\n                    onVolumeChange?.(video.volume);\n                    onToggleMute?.(video.muted);\n                }}\n                playsInline\n                poster={placeholderUrl}\n                ref={videoRef}\n            >\n                <track kind=\"captions\" />\n            </video>\n\n            {onOpenInNewWindow && !error && !loading && (\n                <IconButton\n                    aria-label=\"Open in new window\"\n                    className=\"absolute right-xs top-xs z-10\"\n                    data-testid={`${testid}-open-in-new-window`}\n                    icon={ArrowSquareOut}\n                    onClick={onOpenInNewWindow}\n                    shape=\"circle\"\n                    size=\"sm\"\n                    style=\"inverse\"\n                    variant=\"transparent\"\n                />\n            )}\n\n            {/* Loading overlay */}\n            {loading && (\n                <div\n                    className=\"absolute inset-0 flex items-center justify-center\"\n                    data-testid={`${testid}-loading`}\n                >\n                    <LoadingSpinner\n                        color=\"inverse\"\n                        size=\"md\"\n                    />\n                </div>\n            )}\n\n            {/* Error overlay */}\n            {error && !loading && (\n                <div\n                    className={cn(\n                        'absolute inset-0 flex w-full flex-col items-center justify-center p-sm transition-opacity duration-300',\n                        showContent ? 'opacity-100' : 'opacity-0'\n                    )}\n                    data-testid={`${testid}-error`}\n                >\n                    <div className=\"flex w-full flex-col items-center justify-center gap-xs text-center\">\n                        <H5 color=\"inverse\">Connection Error</H5>\n                        <BodySecondary color=\"inverse\">{error}</BodySecondary>\n                        {onContactSupportOnError && (\n                            <Button\n                                onClick={onContactSupportOnError}\n                                size=\"sm\"\n                                style=\"inverse\"\n                                variant=\"outlined\"\n                            >\n                                Contact Support\n                            </Button>\n                        )}\n                        {connectionState === 'reconnecting' && (\n                            <Caption\n                                className=\"mt-sm\"\n                                color=\"inverse\"\n                            >\n                                Attempting to reconnect...\n                            </Caption>\n                        )}\n                    </div>\n                </div>\n            )}\n        </div>\n    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