import * as _vueuse_core0 from "@vueuse/core";
import * as vue from "vue";

//#region src/use-web-rtc.d.ts
/**
 * WebRTC 配置选项
 * 继承自 RTCConfiguration，支持自定义 WebSocket 协议
 */
type WebRTCOptions = RTCConfiguration & {
  /**
   * WebSocket 子协议列表，用于指定连接时使用的协议
   */
  protocols?: string[];
};
/**
 * WebRTC 连接状态
 * - pending: 初始状态，等待注册
 * - ready: 已注册，可以发起连接
 * - connected: 已建立连接
 * - closed: 连接已关闭
 */
type Status = 'pending' | 'ready' | 'connected' | 'closed';
/**
 * WebSocket 信令消息类型定义
 * 用于 WebRTC 连接建立过程中的信令交换
 */
type HandleEvent = {
  /**
   * 注册消息：服务器分配唯一标识符
   */
  'register': {
    payload: {
      /**
       * 分配给客户端的唯一标识符
       */
      key: string;
    };
  };
  /**
   * Offer 消息：发起连接请求
   */
  'offer': {
    payload: {
      /**
       * 目标客户端标识符
       */
      key: string;
      /**
       * SDP offer 描述
       */
      desc: RTCSessionDescriptionInit;
    };
  };
  /**
   * Answer 消息：响应连接请求
   */
  'answer': {
    payload: {
      /**
       * 发起方客户端标识符
       */
      key: string;
      /**
       * SDP answer 描述
       */
      desc: RTCSessionDescriptionInit;
    };
  };
  /**
   * Answer-ok 消息：确认 answer 已接收
   */
  'answer-ok': {
    payload: {
      /**
       * 目标客户端标识符
       */
      key: string;
    };
  };
  /**
   * ICE candidate 消息：交换网络候选地址
   */
  'ice-candidate': {
    payload: {
      /**
       * ICE 候选者信息
       */
      candidate: RTCIceCandidate;
    };
  };
};
/**
 * WebRTC 连接管理 Composable
 *
 * 提供完整的 WebRTC 连接管理功能，包括：
 * - 通过 WebSocket 进行信令交换
 * - 支持数据通道和媒体流连接
 * - 实时状态监控和事件通知
 * - 自动清理和资源释放
 *
 * @param url WebSocket 信令服务器地址
 * @param options WebRTC 配置选项，继承 RTCConfiguration
 * @returns WebRTC 连接管理对象
 *
 * @example
 * ```ts
 * const { connect, onConnection, status } = useWebRTC('ws://localhost:8080')
 *
 * onConnection((event) => {
 *   const channel = event.channel
 *   channel.onmessage = (e) => console.log('收到消息:', e.data)
 * })
 *
 * const dataChannel = await connect('remote-peer-id', 'chat-channel')
 * dataChannel.send('Hello!')
 * ```
 *
 * @example
 * ```ts
 * const { connectStream, onConnectionStream, status } = useWebRTC('ws://localhost:8080')
 *
 * const stream = await navigator.mediaDevices.getUserMedia({ video: true, audio: true })
 * await connectStream('remote-peer-id', stream)
 *
 * onConnectionStream((event) => {
 *   const remoteStream = event.streams[0]
 *   videoElement.srcObject = remoteStream
 * })
 * ```
 *
 * @example
 * ```ts
 * const { connect, status } = useWebRTC('ws://localhost:8080', {
 *   iceServers: [
 *     { urls: 'stun:stun.l.google.com:19302' },
 *     {
 *       urls: 'turn:your-turn-server.com:3478',
 *       username: 'username',
 *       credential: 'password'
 *     }
 *   ],
 *   iceTransportPolicy: 'relay'
 * })
 * ```
 */
declare function useWebRTC(url: string | URL, options?: WebRTCOptions): {
  id: vue.Ref<string | undefined, string | undefined>;
  connected: vue.Ref<string[], string[]>;
  status: vue.Ref<Status, Status>;
  peer: RTCPeerConnection;
  signalingState: vue.Ref<RTCSignalingState, RTCSignalingState>;
  connectionState: vue.Ref<RTCPeerConnectionState, RTCPeerConnectionState>;
  connect: (id: string, label?: string) => Promise<RTCDataChannel>;
  connectStream: (id: string, stream: MediaStream) => Promise<RTCPeerConnection>;
  onReady: _vueuse_core0.EventHookOn<void>;
  onConnection: _vueuse_core0.EventHookOn<RTCDataChannelEvent>;
  onConnectionStream: _vueuse_core0.EventHookOn<RTCTrackEvent>;
};
/**
 * useWebRTC 函数的返回类型
 */
type UseWebRTCReturns = ReturnType<typeof useWebRTC>;
//#endregion
export { HandleEvent, Status, UseWebRTCReturns, WebRTCOptions, useWebRTC };