import EventEmitter from 'eventemitter3';
import OutgoingCall from './call/outgoing-call';
import IncomingCall from './call/incoming-call';
import Offer from './signalings/offer';

const DEFAULT_RTCCONF = {
  iceServers: [
    {
      urls: [
        'stun:stun.l.google.com:19302',
        'stun:stun1.l.google.com:19302',
        'stun:stun2.l.google.com:19302',
        'stun:stun3.l.google.com:19302',
        'stun:stun4.l.google.com:19302',
        'stun:stun.ekiga.net',
        'stun:stun.ideasip.com',
        'stun:stun.rixtelecom.se',
        'stun:stun.schlund.de',
        'stun:stun.stunprotocol.org:3478',
        'stun:stun.voiparound.com',
        'stun:stun.voipbuster.com',
        'stun:stun.voipstunt.com',
        'stun:stun.voxgratia.org',
      ],
    },
  ],
};

export default class WebRTCClient extends EventEmitter {
  /**
   * 无法直接实例化，请使用 {@link createWebRTCClient Realtime#createWebRTCClient} 创建新的 WebRTCClient
   */
  constructor(id, options) {
    if (typeof id !== 'string') {
      throw new TypeError('id is not a string');
    }
    super();
    /** @type {string} */
    this.id = id;
    this.options = {
      RTCConfiguration: DEFAULT_RTCCONF,
      ...options,
    };
  }

  _open(realtime, clientOptions) {
    return realtime
      .createIMClient(this.id, clientOptions, 'webrtc')
      .then(imClient => {
        this._imClient = imClient;
        this.id = imClient.id;
        imClient.on('message', (message, conversation) => {
          if (message instanceof Offer) {
            return this._handleOffer(message, conversation);
          }
          return false;
        });
        /**
         * 用户在其他客户端登录，当前客户端被服务端强行下线。详见文档「单点登录」章节。
         * @event WebRTCClient#conflict
         */
        imClient.on('conflict', (...payload) =>
          this.emit('conflict', ...payload)
        );
        return this;
      });
  }

  /**
   * 关闭客户端
   * @return {Promise}
   */
  close() {
    return this._imClient.close();
  }

  /**
   * 呼叫另一个用户
   * @param  {string} targetId 用户 ID
   * @param  {MediaStream} stream 本地流媒体，参见 {@link https://developer.mozilla.org/en-US/docs/Web/API/Media_Streams_API MediaStream}
   * @return {Promise.<OutgoingCall>} 呼出通话
   */
  call(targetId, stream) {
    if (typeof targetId !== 'string') {
      throw new TypeError('target id is not a string');
    }
    if (!stream) {
      throw new TypeError('a MediaStream instance is required to make a call');
    }
    return this._imClient.ping([targetId]).then(onlineClients => {
      if (!onlineClients.length) {
        throw new Error(`Call failed as ${targetId} is not online`);
      }
      const outgoingCall = new OutgoingCall(
        targetId,
        null,
        this.options.RTCConfiguration
      );
      const promise = new Promise(resolve => {
        outgoingCall._peerConnection.onnegotiationneeded = resolve;
      });
      outgoingCall._peerConnection.addStream(stream);
      return promise
        .then(() =>
          Promise.all([
            this._imClient.createConversation({
              members: [targetId],
              unique: true,
            }),
            outgoingCall._peerConnection
              .createOffer()
              .then(localDescription => {
                outgoingCall._peerConnection.setLocalDescription(
                  localDescription
                );
              }),
          ])
        )
        .then(([conversation]) => {
          outgoingCall._setConversation(conversation);
          return conversation.send(
            new Offer(outgoingCall._peerConnection.localDescription)
          );
        })
        .then(() => outgoingCall);
    });
  }

  _handleOffer(offer, conversation) {
    const incomingCall = new IncomingCall(
      offer,
      conversation,
      this.options.RTCConfiguration
    );
    /**
     * 收到其他用户的呼叫
     * @event WebRTCClient#call
     * @param {incomingCall} incomingCall 呼入通话
     */
    this.emit('call', incomingCall);
  }
}