@fluxerjs/voice 1.0.5 → 1.0.7

package/dist/index.d.mts

@@ -7,6 +7,7 @@ interface VoiceConnectionEvents {
     error: [err: Error];
     disconnect: [];
 }
+/** Voice connection using Discord's UDP-based protocol. Emits `ready`, `error`, `disconnect`. */
 declare class VoiceConnection extends EventEmitter {
     readonly client: Client;
     readonly channel: VoiceChannel;
@@ -30,7 +31,9 @@ declare class VoiceConnection extends EventEmitter {
     private audioPacketQueue;
     private pacingInterval;
     constructor(client: Client, channel: VoiceChannel, userId: string);
+    /** Discord voice session ID. */
     get sessionId(): string | null;
+    /** Whether audio is currently playing. */
     get playing(): boolean;
     /** Called when we have both server update and state update. */
     connect(server: GatewayVoiceServerUpdateDispatchData, state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
@@ -48,42 +51,152 @@ declare class VoiceConnection extends EventEmitter {
      */
     play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
     private sendAudioFrame;
+    /** Stop playback and clear the queue. */
     stop(): void;
+    /** Disconnect from voice (closes WebSocket and UDP). */
     disconnect(): void;
+    /** Disconnect and remove all listeners. */
     destroy(): void;
 }
 
 /** LiveKit-specific: emitted when server sends leave (token expiry, server policy, etc.). Emitted before disconnect. */
 type LiveKitRtcConnectionEvents = VoiceConnectionEvents & {
     serverLeave: [];
+    /** Emitted when voice state should be synced (self_stream/self_video). VoiceManager listens. */
+    requestVoiceStateSync: [payload: {
+        self_stream?: boolean;
+        self_video?: boolean;
+    }];
 };
+/**
+ * Options for video playback via {@link LiveKitRtcConnection.playVideo}.
+ *
+ * @property source - Track source hint sent to LiveKit. Use `'camera'` for typical video streams
+ *   (default) or `'screenshare'` for screen-share-style content. Affects how clients may display the track.
+ * @property loop - When true (default), loops the video continuously to keep the stream live. Required for
+ *   LiveKit: "stream it continuously for the benefit of participants joining after the initial frame."
+ * @property useFFmpeg - When true, use FFmpeg subprocess for decoding instead of node-webcodecs.
+ *   Recommended when node-webcodecs causes libc++abi crashes on macOS. Requires ffmpeg in PATH.
+ *   Also set via FLUXER_VIDEO_FFMPEG=1 env.
+ * @property videoBitrate - Max video bitrate in bps (default: 2_500_000). Higher values improve quality.
+ * @property maxFramerate - Max framerate for encoding (default: 60).
+ * @property width - Output width (default: source). FFmpeg path only.
+ * @property height - Output height (default: source). FFmpeg path only.
+ */
+interface VideoPlayOptions {
+    /** Track source hint - camera or screenshare (default: camera). */
+    source?: 'camera' | 'screenshare';
+    /** Loop video to keep stream continuously live (default: true). */
+    loop?: boolean;
+    /** Use FFmpeg for decoding (avoids node-webcodecs; requires ffmpeg in PATH). */
+    useFFmpeg?: boolean;
+    /** Max video bitrate in bps for encoding (default: 2_500_000). */
+    videoBitrate?: number;
+    /** Max framerate for encoding (default: 60). */
+    maxFramerate?: number;
+    /** Output width for resolution override (FFmpeg path). */
+    width?: number;
+    /** Output height for resolution override (FFmpeg path). */
+    height?: number;
+}
+/**
+ * Voice connection using LiveKit RTC. Used when Fluxer routes voice to LiveKit.
+ *
+ * Supports both audio playback ({@link play}) and video streaming ({@link playVideo}) to voice channels.
+ * Video uses node-webcodecs for decoding (no ffmpeg subprocess). Audio uses prism-media WebM demuxer.
+ *
+ * @emits ready - When connected to the LiveKit room and ready for playback
+ * @emits disconnect - When disconnected from the room
+ * @emits serverLeave - When LiveKit server signals leave (e.g. token expiry), before disconnect
+ * @emits error - On connection, playback, or decoding errors
+ */
 declare class LiveKitRtcConnection extends EventEmitter {
     readonly client: Client;
     readonly channel: VoiceChannel;
     readonly guildId: string;
     private _playing;
+    private _playingVideo;
     private _destroyed;
     private room;
     private audioSource;
     private audioTrack;
+    private videoSource;
+    private videoTrack;
     private currentStream;
+    private currentVideoStream;
+    private _videoCleanup;
     private lastServerEndpoint;
     private lastServerToken;
     private _disconnectEmitted;
+    /**
+     * @param client - The Fluxer client instance
+     * @param channel - The voice channel to connect to
+     * @param _userId - The user ID (reserved for future use)
+     */
     constructor(client: Client, channel: VoiceChannel, _userId: string);
+    /** Whether audio is currently playing. */
     get playing(): boolean;
     private debug;
     private audioDebug;
     private emitDisconnect;
     /** Returns true if the LiveKit room is connected and not destroyed. */
     isConnected(): boolean;
-    /** Returns true if we're already connected to the given server (skip migration). */
+    /**
+     * Returns true if we're already connected to the given server (skip migration).
+     * @param endpoint - Voice server endpoint from the gateway
+     * @param token - Voice server token
+     */
     isSameServer(endpoint: string | null, token: string): boolean;
     playOpus(_stream: NodeJS.ReadableStream): void;
+    /**
+     * Connect to the LiveKit room using voice server and state from the gateway.
+     * Called internally by VoiceManager; typically not used directly.
+     *
+     * @param server - Voice server update data (endpoint, token)
+     * @param _state - Voice state update data (session, channel)
+     */
     connect(server: GatewayVoiceServerUpdateDispatchData, _state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
+    /** Whether a video track is currently playing in the voice channel. */
+    get playingVideo(): boolean;
+    /**
+     * Play video from an MP4 URL or buffer. Streams decoded frames to the LiveKit room as a video track.
+     * Uses node-webcodecs for decoding (no ffmpeg). Supports H.264 (avc1) and H.265 (hvc1/hev1) codecs.
+     *
+     * @param urlOrBuffer - Video source: HTTP(S) URL to an MP4 file, or raw ArrayBuffer/Uint8Array of MP4 data
+     * @param options - Optional playback options (see {@link VideoPlayOptions})
+     * @emits error - On fetch failure, missing video track, or decode errors
+     *
+     * @example
+     * ```ts
+     * const conn = await voiceManager.join(channel);
+     * if (conn instanceof LiveKitRtcConnection && conn.isConnected()) {
+     *   await conn.playVideo('https://example.com/video.mp4', { source: 'camera' });
+     * }
+     * ```
+     */
+    playVideo(urlOrBuffer: string | ArrayBuffer | Uint8Array, options?: VideoPlayOptions): Promise<void>;
+    /**
+     * FFmpeg-based video playback. Bypasses node-webcodecs to avoid libc++abi crashes on macOS.
+     * Requires ffmpeg and ffprobe in PATH. URL input only.
+     */
+    private playVideoFFmpeg;
+    /**
+     * Play audio from a WebM/Opus URL or readable stream. Publishes to the LiveKit room as an audio track.
+     *
+     * @param urlOrStream - Audio source: HTTP(S) URL to a WebM/Opus file, or a Node.js ReadableStream
+     * @emits error - On fetch failure or decode errors
+     */
     play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
+    /**
+     * Stop video playback and unpublish the video track from the LiveKit room.
+     * Safe to call even when no video is playing.
+     */
+    stopVideo(): void;
+    /** Stop playback and clear both audio and video tracks. */
     stop(): void;
+    /** Disconnect from the LiveKit room and stop all playback. */
     disconnect(): void;
+    /** Disconnect from the room and remove all event listeners. */
     destroy(): void;
 }
 declare module 'events' {
@@ -95,46 +208,94 @@ declare module 'events' {
 
 /** Maps guild_id -> user_id -> channel_id (null if not in voice). */
 type VoiceStateMap = Map<string, Map<string, string | null>>;
+/**
+ * Options for creating a VoiceManager.
+ *
+ * @property shardId - Gateway shard ID to use for voice connections (default 0).
+ *   Use when the client runs multiple shards and you need to target a specific one.
+ */
 interface VoiceManagerOptions {
     /** Gateway shard ID to use for voice (default 0). */
     shardId?: number;
 }
+/** Manages voice connections. Use `getVoiceManager(client)` to obtain an instance. */
 declare class VoiceManager extends EventEmitter {
     readonly client: Client;
     private readonly connections;
+    /** guild_id -> connection_id (from VoiceServerUpdate; required for voice state updates when in channel) */
+    private readonly connectionIds;
     /** guild_id -> user_id -> channel_id */
     readonly voiceStates: VoiceStateMap;
     private readonly pending;
     private readonly shardId;
     constructor(client: Client, options?: VoiceManagerOptions);
     private handleVoiceStatesSync;
-    /** Get the voice channel ID the user is in, or null. */
+    /**
+     * Get the voice channel ID the user is currently in, or null if not in voice.
+     * @param guildId - Guild ID to look up
+     * @param userId - User ID to look up
+     */
     getVoiceChannelId(guildId: string, userId: string): string | null;
     private handleVoiceStateUpdate;
     private handleVoiceServerUpdate;
+    private storeConnectionId;
     private registerConnection;
+    /** Upload a placeholder stream preview so the preview URL returns 200 instead of 404. */
+    private uploadStreamPreview;
     private tryCompletePending;
-    /** Join a voice channel. Resolves when the connection is ready. */
+    /**
+     * Join a voice channel. Resolves when the connection is ready.
+     * @param channel - The voice channel to join
+     * @returns The voice connection (LiveKitRtcConnection when Fluxer uses LiveKit)
+     */
     join(channel: VoiceChannel): Promise<VoiceConnection | LiveKitRtcConnection>;
-    /** Leave a guild's voice channel. */
+    /**
+     * Leave a guild's voice channel and disconnect.
+     * @param guildId - Guild ID to leave
+     */
     leave(guildId: string): void;
+    /**
+     * Get the active voice connection for a guild, if any.
+     * @param guildId - Guild ID to look up
+     */
     getConnection(guildId: string): VoiceConnection | LiveKitRtcConnection | undefined;
+    /**
+     * Update voice state (e.g. self_stream, self_video) while in a channel.
+     * Sends a VoiceStateUpdate to the gateway so the server and clients see the change.
+     * Requires connection_id (from VoiceServerUpdate); without it, the gateway would treat
+     * the update as a new join and trigger a new VoiceServerUpdate, causing connection loops.
+     * @param guildId - Guild ID
+     * @param partial - Partial voice state to update (self_stream, self_video, self_mute, self_deaf)
+     */
+    updateVoiceState(guildId: string, partial: {
+        self_stream?: boolean;
+        self_video?: boolean;
+        self_mute?: boolean;
+        self_deaf?: boolean;
+    }): void;
 }
 
 /** Union of connection types (Discord-style or LiveKit). */
 type VoiceConnectionLike = VoiceConnection | LiveKitRtcConnection;
 /**
- * Create a voice manager and join a channel in one call.
- * Uses the default shard (0).
+ * Create a voice manager and join a voice channel in one call.
+ *
+ * @param client - The Fluxer client instance
+ * @param channel - The voice channel to join
+ * @param options - Optional options; `shardId` for the gateway shard to use (default 0)
+ * @returns The voice connection (LiveKitRtcConnection when using LiveKit)
  */
 declare function joinVoiceChannel(client: Client, channel: VoiceChannel, options?: {
     shardId?: number;
 }): Promise<VoiceConnectionLike>;
 /**
  * Get or create the VoiceManager for this client.
+ *
+ * @param client - The Fluxer client instance
+ * @param options - Optional options; `shardId` for the gateway shard to use (default 0)
  */
 declare function getVoiceManager(client: Client, options?: {
     shardId?: number;
 }): VoiceManager;
 
-export { LiveKitRtcConnection, type LiveKitRtcConnectionEvents, VoiceConnection, type VoiceConnectionEvents, type VoiceConnectionLike, VoiceManager, type VoiceManagerOptions, type VoiceStateMap, getVoiceManager, joinVoiceChannel };
+export { LiveKitRtcConnection, type LiveKitRtcConnectionEvents, type VideoPlayOptions, VoiceConnection, type VoiceConnectionEvents, type VoiceConnectionLike, VoiceManager, type VoiceManagerOptions, type VoiceStateMap, getVoiceManager, joinVoiceChannel };

package/dist/index.d.ts

@@ -7,6 +7,7 @@ interface VoiceConnectionEvents {
     error: [err: Error];
     disconnect: [];
 }
+/** Voice connection using Discord's UDP-based protocol. Emits `ready`, `error`, `disconnect`. */
 declare class VoiceConnection extends EventEmitter {
     readonly client: Client;
     readonly channel: VoiceChannel;
@@ -30,7 +31,9 @@ declare class VoiceConnection extends EventEmitter {
     private audioPacketQueue;
     private pacingInterval;
     constructor(client: Client, channel: VoiceChannel, userId: string);
+    /** Discord voice session ID. */
     get sessionId(): string | null;
+    /** Whether audio is currently playing. */
     get playing(): boolean;
     /** Called when we have both server update and state update. */
     connect(server: GatewayVoiceServerUpdateDispatchData, state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
@@ -48,42 +51,152 @@ declare class VoiceConnection extends EventEmitter {
      */
     play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
     private sendAudioFrame;
+    /** Stop playback and clear the queue. */
     stop(): void;
+    /** Disconnect from voice (closes WebSocket and UDP). */
     disconnect(): void;
+    /** Disconnect and remove all listeners. */
     destroy(): void;
 }
 
 /** LiveKit-specific: emitted when server sends leave (token expiry, server policy, etc.). Emitted before disconnect. */
 type LiveKitRtcConnectionEvents = VoiceConnectionEvents & {
     serverLeave: [];
+    /** Emitted when voice state should be synced (self_stream/self_video). VoiceManager listens. */
+    requestVoiceStateSync: [payload: {
+        self_stream?: boolean;
+        self_video?: boolean;
+    }];
 };
+/**
+ * Options for video playback via {@link LiveKitRtcConnection.playVideo}.
+ *
+ * @property source - Track source hint sent to LiveKit. Use `'camera'` for typical video streams
+ *   (default) or `'screenshare'` for screen-share-style content. Affects how clients may display the track.
+ * @property loop - When true (default), loops the video continuously to keep the stream live. Required for
+ *   LiveKit: "stream it continuously for the benefit of participants joining after the initial frame."
+ * @property useFFmpeg - When true, use FFmpeg subprocess for decoding instead of node-webcodecs.
+ *   Recommended when node-webcodecs causes libc++abi crashes on macOS. Requires ffmpeg in PATH.
+ *   Also set via FLUXER_VIDEO_FFMPEG=1 env.
+ * @property videoBitrate - Max video bitrate in bps (default: 2_500_000). Higher values improve quality.
+ * @property maxFramerate - Max framerate for encoding (default: 60).
+ * @property width - Output width (default: source). FFmpeg path only.
+ * @property height - Output height (default: source). FFmpeg path only.
+ */
+interface VideoPlayOptions {
+    /** Track source hint - camera or screenshare (default: camera). */
+    source?: 'camera' | 'screenshare';
+    /** Loop video to keep stream continuously live (default: true). */
+    loop?: boolean;
+    /** Use FFmpeg for decoding (avoids node-webcodecs; requires ffmpeg in PATH). */
+    useFFmpeg?: boolean;
+    /** Max video bitrate in bps for encoding (default: 2_500_000). */
+    videoBitrate?: number;
+    /** Max framerate for encoding (default: 60). */
+    maxFramerate?: number;
+    /** Output width for resolution override (FFmpeg path). */
+    width?: number;
+    /** Output height for resolution override (FFmpeg path). */
+    height?: number;
+}
+/**
+ * Voice connection using LiveKit RTC. Used when Fluxer routes voice to LiveKit.
+ *
+ * Supports both audio playback ({@link play}) and video streaming ({@link playVideo}) to voice channels.
+ * Video uses node-webcodecs for decoding (no ffmpeg subprocess). Audio uses prism-media WebM demuxer.
+ *
+ * @emits ready - When connected to the LiveKit room and ready for playback
+ * @emits disconnect - When disconnected from the room
+ * @emits serverLeave - When LiveKit server signals leave (e.g. token expiry), before disconnect
+ * @emits error - On connection, playback, or decoding errors
+ */
 declare class LiveKitRtcConnection extends EventEmitter {
     readonly client: Client;
     readonly channel: VoiceChannel;
     readonly guildId: string;
     private _playing;
+    private _playingVideo;
     private _destroyed;
     private room;
     private audioSource;
     private audioTrack;
+    private videoSource;
+    private videoTrack;
     private currentStream;
+    private currentVideoStream;
+    private _videoCleanup;
     private lastServerEndpoint;
     private lastServerToken;
     private _disconnectEmitted;
+    /**
+     * @param client - The Fluxer client instance
+     * @param channel - The voice channel to connect to
+     * @param _userId - The user ID (reserved for future use)
+     */
     constructor(client: Client, channel: VoiceChannel, _userId: string);
+    /** Whether audio is currently playing. */
     get playing(): boolean;
     private debug;
     private audioDebug;
     private emitDisconnect;
     /** Returns true if the LiveKit room is connected and not destroyed. */
     isConnected(): boolean;
-    /** Returns true if we're already connected to the given server (skip migration). */
+    /**
+     * Returns true if we're already connected to the given server (skip migration).
+     * @param endpoint - Voice server endpoint from the gateway
+     * @param token - Voice server token
+     */
     isSameServer(endpoint: string | null, token: string): boolean;
     playOpus(_stream: NodeJS.ReadableStream): void;
+    /**
+     * Connect to the LiveKit room using voice server and state from the gateway.
+     * Called internally by VoiceManager; typically not used directly.
+     *
+     * @param server - Voice server update data (endpoint, token)
+     * @param _state - Voice state update data (session, channel)
+     */
     connect(server: GatewayVoiceServerUpdateDispatchData, _state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
+    /** Whether a video track is currently playing in the voice channel. */
+    get playingVideo(): boolean;
+    /**
+     * Play video from an MP4 URL or buffer. Streams decoded frames to the LiveKit room as a video track.
+     * Uses node-webcodecs for decoding (no ffmpeg). Supports H.264 (avc1) and H.265 (hvc1/hev1) codecs.
+     *
+     * @param urlOrBuffer - Video source: HTTP(S) URL to an MP4 file, or raw ArrayBuffer/Uint8Array of MP4 data
+     * @param options - Optional playback options (see {@link VideoPlayOptions})
+     * @emits error - On fetch failure, missing video track, or decode errors
+     *
+     * @example
+     * ```ts
+     * const conn = await voiceManager.join(channel);
+     * if (conn instanceof LiveKitRtcConnection && conn.isConnected()) {
+     *   await conn.playVideo('https://example.com/video.mp4', { source: 'camera' });
+     * }
+     * ```
+     */
+    playVideo(urlOrBuffer: string | ArrayBuffer | Uint8Array, options?: VideoPlayOptions): Promise<void>;
+    /**
+     * FFmpeg-based video playback. Bypasses node-webcodecs to avoid libc++abi crashes on macOS.
+     * Requires ffmpeg and ffprobe in PATH. URL input only.
+     */
+    private playVideoFFmpeg;
+    /**
+     * Play audio from a WebM/Opus URL or readable stream. Publishes to the LiveKit room as an audio track.
+     *
+     * @param urlOrStream - Audio source: HTTP(S) URL to a WebM/Opus file, or a Node.js ReadableStream
+     * @emits error - On fetch failure or decode errors
+     */
     play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
+    /**
+     * Stop video playback and unpublish the video track from the LiveKit room.
+     * Safe to call even when no video is playing.
+     */
+    stopVideo(): void;
+    /** Stop playback and clear both audio and video tracks. */
     stop(): void;
+    /** Disconnect from the LiveKit room and stop all playback. */
     disconnect(): void;
+    /** Disconnect from the room and remove all event listeners. */
     destroy(): void;
 }
 declare module 'events' {
@@ -95,46 +208,94 @@ declare module 'events' {
 
 /** Maps guild_id -> user_id -> channel_id (null if not in voice). */
 type VoiceStateMap = Map<string, Map<string, string | null>>;
+/**
+ * Options for creating a VoiceManager.
+ *
+ * @property shardId - Gateway shard ID to use for voice connections (default 0).
+ *   Use when the client runs multiple shards and you need to target a specific one.
+ */
 interface VoiceManagerOptions {
     /** Gateway shard ID to use for voice (default 0). */
     shardId?: number;
 }
+/** Manages voice connections. Use `getVoiceManager(client)` to obtain an instance. */
 declare class VoiceManager extends EventEmitter {
     readonly client: Client;
     private readonly connections;
+    /** guild_id -> connection_id (from VoiceServerUpdate; required for voice state updates when in channel) */
+    private readonly connectionIds;
     /** guild_id -> user_id -> channel_id */
     readonly voiceStates: VoiceStateMap;
     private readonly pending;
     private readonly shardId;
     constructor(client: Client, options?: VoiceManagerOptions);
     private handleVoiceStatesSync;
-    /** Get the voice channel ID the user is in, or null. */
+    /**
+     * Get the voice channel ID the user is currently in, or null if not in voice.
+     * @param guildId - Guild ID to look up
+     * @param userId - User ID to look up
+     */
     getVoiceChannelId(guildId: string, userId: string): string | null;
     private handleVoiceStateUpdate;
     private handleVoiceServerUpdate;
+    private storeConnectionId;
     private registerConnection;
+    /** Upload a placeholder stream preview so the preview URL returns 200 instead of 404. */
+    private uploadStreamPreview;
     private tryCompletePending;
-    /** Join a voice channel. Resolves when the connection is ready. */
+    /**
+     * Join a voice channel. Resolves when the connection is ready.
+     * @param channel - The voice channel to join
+     * @returns The voice connection (LiveKitRtcConnection when Fluxer uses LiveKit)
+     */
     join(channel: VoiceChannel): Promise<VoiceConnection | LiveKitRtcConnection>;
-    /** Leave a guild's voice channel. */
+    /**
+     * Leave a guild's voice channel and disconnect.
+     * @param guildId - Guild ID to leave
+     */
     leave(guildId: string): void;
+    /**
+     * Get the active voice connection for a guild, if any.
+     * @param guildId - Guild ID to look up
+     */
     getConnection(guildId: string): VoiceConnection | LiveKitRtcConnection | undefined;
+    /**
+     * Update voice state (e.g. self_stream, self_video) while in a channel.
+     * Sends a VoiceStateUpdate to the gateway so the server and clients see the change.
+     * Requires connection_id (from VoiceServerUpdate); without it, the gateway would treat
+     * the update as a new join and trigger a new VoiceServerUpdate, causing connection loops.
+     * @param guildId - Guild ID
+     * @param partial - Partial voice state to update (self_stream, self_video, self_mute, self_deaf)
+     */
+    updateVoiceState(guildId: string, partial: {
+        self_stream?: boolean;
+        self_video?: boolean;
+        self_mute?: boolean;
+        self_deaf?: boolean;
+    }): void;
 }
 
 /** Union of connection types (Discord-style or LiveKit). */
 type VoiceConnectionLike = VoiceConnection | LiveKitRtcConnection;
 /**
- * Create a voice manager and join a channel in one call.
- * Uses the default shard (0).
+ * Create a voice manager and join a voice channel in one call.
+ *
+ * @param client - The Fluxer client instance
+ * @param channel - The voice channel to join
+ * @param options - Optional options; `shardId` for the gateway shard to use (default 0)
+ * @returns The voice connection (LiveKitRtcConnection when using LiveKit)
  */
 declare function joinVoiceChannel(client: Client, channel: VoiceChannel, options?: {
     shardId?: number;
 }): Promise<VoiceConnectionLike>;
 /**
  * Get or create the VoiceManager for this client.
+ *
+ * @param client - The Fluxer client instance
+ * @param options - Optional options; `shardId` for the gateway shard to use (default 0)
  */
 declare function getVoiceManager(client: Client, options?: {
     shardId?: number;
 }): VoiceManager;
 
-export { LiveKitRtcConnection, type LiveKitRtcConnectionEvents, VoiceConnection, type VoiceConnectionEvents, type VoiceConnectionLike, VoiceManager, type VoiceManagerOptions, type VoiceStateMap, getVoiceManager, joinVoiceChannel };
+export { LiveKitRtcConnection, type LiveKitRtcConnectionEvents, type VideoPlayOptions, VoiceConnection, type VoiceConnectionEvents, type VoiceConnectionLike, VoiceManager, type VoiceManagerOptions, type VoiceStateMap, getVoiceManager, joinVoiceChannel };