@fluxerjs/voice 1.0.3

package/README.md

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+# @fluxerjs/voice
+
+Voice for Fluxer bots. Join channels and play WebM/Opus audio.
+
+## Install
+
+```bash
+pnpm add @fluxerjs/voice @fluxerjs/core
+```
+
+## Usage
+
+```javascript
+import { getVoiceManager } from '@fluxerjs/voice';
+
+const voiceManager = getVoiceManager(client);
+const connection = await voiceManager.join(channel);
+await connection.play(streamUrl);
+
+connection.stop();
+voiceManager.leave(guildId);
+```
+
+Use yt-dlp to get stream URLs from YouTube. For LiveKit, listen for `serverLeave` to reconnect.

package/dist/index.d.mts

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+import { EventEmitter } from 'events';
+import { Client, VoiceChannel } from '@fluxerjs/core';
+import { GatewayVoiceServerUpdateDispatchData, GatewayVoiceStateUpdateDispatchData } from '@fluxerjs/types';
+
+interface VoiceConnectionEvents {
+    ready: [];
+    error: [err: Error];
+    disconnect: [];
+}
+declare class VoiceConnection extends EventEmitter {
+    readonly client: Client;
+    readonly channel: VoiceChannel;
+    readonly guildId: string;
+    private _sessionId;
+    private _token;
+    private _endpoint;
+    private _userId;
+    private voiceWs;
+    private udpSocket;
+    private ssrc;
+    private secretKey;
+    private heartbeatInterval;
+    private sequence;
+    private timestamp;
+    private _playing;
+    private _destroyed;
+    private currentStream;
+    private remoteUdpAddress;
+    private remoteUdpPort;
+    private audioPacketQueue;
+    private pacingInterval;
+    constructor(client: Client, channel: VoiceChannel, userId: string);
+    get sessionId(): string | null;
+    get playing(): boolean;
+    /** Called when we have both server update and state update. */
+    connect(server: GatewayVoiceServerUpdateDispatchData, state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
+    private getWebSocketConstructor;
+    private sendVoiceOp;
+    private setupUDP;
+    /**
+     * Play a stream of raw Opus packets
+     * Uses the same queue and 20ms pacing as play(). Use this for local files (MP3 → PCM → Opus) or other Opus sources.
+     */
+    playOpus(stream: NodeJS.ReadableStream): void;
+    /**
+     * Play a direct WebM/Opus URL or stream. Fetches the URL (if string), demuxes with prism-media WebmDemuxer,
+     * and sends Opus packets to the voice connection. No FFmpeg or encoding; input must be WebM with Opus.
+     */
+    play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
+    private sendAudioFrame;
+    stop(): void;
+    disconnect(): void;
+    destroy(): void;
+}
+
+/** LiveKit-specific: emitted when server sends leave (token expiry, server policy, etc.). Emitted before disconnect. */
+type LiveKitRtcConnectionEvents = VoiceConnectionEvents & {
+    serverLeave: [];
+};
+declare class LiveKitRtcConnection extends EventEmitter {
+    readonly client: Client;
+    readonly channel: VoiceChannel;
+    readonly guildId: string;
+    private _playing;
+    private _destroyed;
+    private room;
+    private audioSource;
+    private audioTrack;
+    private currentStream;
+    private lastServerEndpoint;
+    private lastServerToken;
+    private _disconnectEmitted;
+    constructor(client: Client, channel: VoiceChannel, _userId: string);
+    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). */
+    isSameServer(endpoint: string | null, token: string): boolean;
+    playOpus(_stream: NodeJS.ReadableStream): void;
+    connect(server: GatewayVoiceServerUpdateDispatchData, _state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
+    play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
+    stop(): void;
+    disconnect(): void;
+    destroy(): void;
+}
+declare module 'events' {
+    interface LiveKitRtcConnection {
+        on<E extends keyof LiveKitRtcConnectionEvents>(event: E, listener: (...args: LiveKitRtcConnectionEvents[E]) => void): this;
+        emit<E extends keyof LiveKitRtcConnectionEvents>(event: E, ...args: LiveKitRtcConnectionEvents[E]): boolean;
+    }
+}
+
+/** Maps guild_id -> user_id -> channel_id (null if not in voice). */
+type VoiceStateMap = Map<string, Map<string, string | null>>;
+interface VoiceManagerOptions {
+    /** Gateway shard ID to use for voice (default 0). */
+    shardId?: number;
+}
+declare class VoiceManager extends EventEmitter {
+    readonly client: Client;
+    private readonly connections;
+    /** 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. */
+    getVoiceChannelId(guildId: string, userId: string): string | null;
+    private handleVoiceStateUpdate;
+    private handleVoiceServerUpdate;
+    private registerConnection;
+    private tryCompletePending;
+    /** Join a voice channel. Resolves when the connection is ready. */
+    join(channel: VoiceChannel): Promise<VoiceConnection | LiveKitRtcConnection>;
+    /** Leave a guild's voice channel. */
+    leave(guildId: string): void;
+    getConnection(guildId: string): VoiceConnection | LiveKitRtcConnection | undefined;
+}
+
+/** 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).
+ */
+declare function joinVoiceChannel(client: Client, channel: VoiceChannel, options?: {
+    shardId?: number;
+}): Promise<VoiceConnectionLike>;
+/**
+ * Get or create the VoiceManager for this client.
+ */
+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 };

package/dist/index.d.ts

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+import { EventEmitter } from 'events';
+import { Client, VoiceChannel } from '@fluxerjs/core';
+import { GatewayVoiceServerUpdateDispatchData, GatewayVoiceStateUpdateDispatchData } from '@fluxerjs/types';
+
+interface VoiceConnectionEvents {
+    ready: [];
+    error: [err: Error];
+    disconnect: [];
+}
+declare class VoiceConnection extends EventEmitter {
+    readonly client: Client;
+    readonly channel: VoiceChannel;
+    readonly guildId: string;
+    private _sessionId;
+    private _token;
+    private _endpoint;
+    private _userId;
+    private voiceWs;
+    private udpSocket;
+    private ssrc;
+    private secretKey;
+    private heartbeatInterval;
+    private sequence;
+    private timestamp;
+    private _playing;
+    private _destroyed;
+    private currentStream;
+    private remoteUdpAddress;
+    private remoteUdpPort;
+    private audioPacketQueue;
+    private pacingInterval;
+    constructor(client: Client, channel: VoiceChannel, userId: string);
+    get sessionId(): string | null;
+    get playing(): boolean;
+    /** Called when we have both server update and state update. */
+    connect(server: GatewayVoiceServerUpdateDispatchData, state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
+    private getWebSocketConstructor;
+    private sendVoiceOp;
+    private setupUDP;
+    /**
+     * Play a stream of raw Opus packets
+     * Uses the same queue and 20ms pacing as play(). Use this for local files (MP3 → PCM → Opus) or other Opus sources.
+     */
+    playOpus(stream: NodeJS.ReadableStream): void;
+    /**
+     * Play a direct WebM/Opus URL or stream. Fetches the URL (if string), demuxes with prism-media WebmDemuxer,
+     * and sends Opus packets to the voice connection. No FFmpeg or encoding; input must be WebM with Opus.
+     */
+    play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
+    private sendAudioFrame;
+    stop(): void;
+    disconnect(): void;
+    destroy(): void;
+}
+
+/** LiveKit-specific: emitted when server sends leave (token expiry, server policy, etc.). Emitted before disconnect. */
+type LiveKitRtcConnectionEvents = VoiceConnectionEvents & {
+    serverLeave: [];
+};
+declare class LiveKitRtcConnection extends EventEmitter {
+    readonly client: Client;
+    readonly channel: VoiceChannel;
+    readonly guildId: string;
+    private _playing;
+    private _destroyed;
+    private room;
+    private audioSource;
+    private audioTrack;
+    private currentStream;
+    private lastServerEndpoint;
+    private lastServerToken;
+    private _disconnectEmitted;
+    constructor(client: Client, channel: VoiceChannel, _userId: string);
+    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). */
+    isSameServer(endpoint: string | null, token: string): boolean;
+    playOpus(_stream: NodeJS.ReadableStream): void;
+    connect(server: GatewayVoiceServerUpdateDispatchData, _state: GatewayVoiceStateUpdateDispatchData): Promise<void>;
+    play(urlOrStream: string | NodeJS.ReadableStream): Promise<void>;
+    stop(): void;
+    disconnect(): void;
+    destroy(): void;
+}
+declare module 'events' {
+    interface LiveKitRtcConnection {
+        on<E extends keyof LiveKitRtcConnectionEvents>(event: E, listener: (...args: LiveKitRtcConnectionEvents[E]) => void): this;
+        emit<E extends keyof LiveKitRtcConnectionEvents>(event: E, ...args: LiveKitRtcConnectionEvents[E]): boolean;
+    }
+}
+
+/** Maps guild_id -> user_id -> channel_id (null if not in voice). */
+type VoiceStateMap = Map<string, Map<string, string | null>>;
+interface VoiceManagerOptions {
+    /** Gateway shard ID to use for voice (default 0). */
+    shardId?: number;
+}
+declare class VoiceManager extends EventEmitter {
+    readonly client: Client;
+    private readonly connections;
+    /** 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. */
+    getVoiceChannelId(guildId: string, userId: string): string | null;
+    private handleVoiceStateUpdate;
+    private handleVoiceServerUpdate;
+    private registerConnection;
+    private tryCompletePending;
+    /** Join a voice channel. Resolves when the connection is ready. */
+    join(channel: VoiceChannel): Promise<VoiceConnection | LiveKitRtcConnection>;
+    /** Leave a guild's voice channel. */
+    leave(guildId: string): void;
+    getConnection(guildId: string): VoiceConnection | LiveKitRtcConnection | undefined;
+}
+
+/** 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).
+ */
+declare function joinVoiceChannel(client: Client, channel: VoiceChannel, options?: {
+    shardId?: number;
+}): Promise<VoiceConnectionLike>;
+/**
+ * Get or create the VoiceManager for this client.
+ */
+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 };