/**
 * WebRTC 连接管理模块
 * 
 * 功能：
 * - 管理 RTCPeerConnection 生命周期
 * - 处理 SDP 交换
 * - 处理 ICE 候选收集
 * - 音频流管理
 */

export type ConnectionState = 'idle' | 'connecting' | 'connected' | 'disconnected' | 'failed'

export interface WebRTCConfig {
  iceServers?: RTCIceServer[]
  onLocalStream?: (stream: MediaStream) => void
  onRemoteStream?: (stream: MediaStream) => void
  onConnectionStateChange?: (state: ConnectionState) => void
  onIceCandidate?: (candidate: RTCIceCandidate) => void
  onError?: (error: Error) => void
}

// 默认 ICE 服务器配置
const DEFAULT_ICE_SERVERS: RTCIceServer[] = [
  { urls: 'stun:stun.l.google.com:19302' },
  { urls: 'stun:stun1.l.google.com:19302' },
  { urls: 'stun:stun2.l.google.com:19302' }
]

export class WebRTCManager {
  private peerConnection: RTCPeerConnection | null = null
  private localStream: MediaStream | null = null
  private remoteStream: MediaStream | null = null
  private config: WebRTCConfig
  private connectionState: ConnectionState = 'idle'
  private pendingIceCandidates: RTCIceCandidate[] = []

  constructor(config: WebRTCConfig = {}) {
    this.config = {
      iceServers: DEFAULT_ICE_SERVERS,
      ...config
    }
  }

  /**
   * 获取当前连接状态
   */
  getConnectionState(): ConnectionState {
    return this.connectionState
  }

  /**
   * 获取本地音频流
   */
  getLocalStream(): MediaStream | null {
    return this.localStream
  }

  /**
   * 获取远程音频流
   */
  getRemoteStream(): MediaStream | null {
    return this.remoteStream
  }

  /**
   * 初始化本地音频流
   */
  async initLocalStream(): Promise<MediaStream> {
    try {
      this.localStream = await navigator.mediaDevices.getUserMedia({
        audio: {
          echoCancellation: true,
          noiseSuppression: true,
          autoGainControl: true
        },
        video: false
      })
      
      this.config.onLocalStream?.(this.localStream)
      return this.localStream
    } catch (error) {
      const err = error instanceof Error ? error : new Error('获取麦克风权限失败')
      this.config.onError?.(err)
      throw err
    }
  }

  /**
   * 创建 PeerConnection
   */
  private createPeerConnection(): RTCPeerConnection {
    const pc = new RTCPeerConnection({
      iceServers: this.config.iceServers
    })

    // 监听 ICE 候选
    pc.onicecandidate = (event) => {
      if (event.candidate) {
        console.log('[WebRTC] ICE candidate:', event.candidate.candidate?.substring(0, 50))
        this.config.onIceCandidate?.(event.candidate)
      }
    }

    // 监听连接状态变化
    pc.onconnectionstatechange = () => {
      console.log('[WebRTC] Connection state:', pc.connectionState)
      this.updateConnectionState(pc.connectionState)
    }

    // 监听 ICE 连接状态
    pc.oniceconnectionstatechange = () => {
      console.log('[WebRTC] ICE connection state:', pc.iceConnectionState)
      if (pc.iceConnectionState === 'failed') {
        this.updateConnectionState('failed')
      }
    }

    // 监听远程流
    pc.ontrack = (event) => {
      console.log('[WebRTC] Remote track received')
      if (event.streams && event.streams[0]) {
        this.remoteStream = event.streams[0]
        this.config.onRemoteStream?.(this.remoteStream)
      }
    }

    return pc
  }

  /**
   * 更新连接状态
   */
  private updateConnectionState(state: RTCPeerConnectionState | string) {
    const stateMap: Record<string, ConnectionState> = {
      'new': 'connecting',
      'connecting': 'connecting',
      'connected': 'connected',
      'disconnected': 'disconnected',
      'failed': 'failed',
      'closed': 'disconnected'
    }
    
    this.connectionState = stateMap[state] || 'idle'
    this.config.onConnectionStateChange?.(this.connectionState)
  }

  /**
   * 创建 Offer（发起方调用）
   */
  async createOffer(): Promise<RTCSessionDescriptionInit> {
    if (!this.localStream) {
      await this.initLocalStream()
    }

    this.peerConnection = this.createPeerConnection()
    this.updateConnectionState('connecting')

    // 添加本地音频轨道
    this.localStream!.getTracks().forEach(track => {
      this.peerConnection!.addTrack(track, this.localStream!)
    })

    // 创建 Offer
    const offer = await this.peerConnection.createOffer()
    await this.peerConnection.setLocalDescription(offer)

    console.log('[WebRTC] Offer created')
    return offer
  }

  /**
   * 处理 Offer（接收方调用）
   */
  async handleOffer(offer: RTCSessionDescriptionInit): Promise<RTCSessionDescriptionInit> {
    if (!this.localStream) {
      await this.initLocalStream()
    }

    this.peerConnection = this.createPeerConnection()
    this.updateConnectionState('connecting')

    // 添加本地音频轨道
    this.localStream!.getTracks().forEach(track => {
      this.peerConnection!.addTrack(track, this.localStream!)
    })

    // 设置远程描述
    await this.peerConnection.setRemoteDescription(new RTCSessionDescription(offer))

    // 处理等待中的 ICE 候选
    for (const candidate of this.pendingIceCandidates) {
      await this.peerConnection.addIceCandidate(candidate)
    }
    this.pendingIceCandidates = []

    // 创建 Answer
    const answer = await this.peerConnection.createAnswer()
    await this.peerConnection.setLocalDescription(answer)

    console.log('[WebRTC] Answer created')
    return answer
  }

  /**
   * 处理 Answer（发起方调用）
   */
  async handleAnswer(answer: RTCSessionDescriptionInit): Promise<void> {
    if (!this.peerConnection) {
      throw new Error('PeerConnection not initialized')
    }

    await this.peerConnection.setRemoteDescription(new RTCSessionDescription(answer))

    // 处理等待中的 ICE 候选
    for (const candidate of this.pendingIceCandidates) {
      await this.peerConnection.addIceCandidate(candidate)
    }
    this.pendingIceCandidates = []

    console.log('[WebRTC] Answer handled')
  }

  /**
   * 添加 ICE 候选
   */
  async addIceCandidate(candidate: RTCIceCandidateInit): Promise<void> {
    const iceCandidate = new RTCIceCandidate(candidate)

    if (this.peerConnection && this.peerConnection.remoteDescription) {
      await this.peerConnection.addIceCandidate(iceCandidate)
      console.log('[WebRTC] ICE candidate added')
    } else {
      // 如果远程描述还没设置，先缓存候选
      this.pendingIceCandidates.push(iceCandidate)
      console.log('[WebRTC] ICE candidate queued')
    }
  }

  /**
   * 静音/取消静音本地音频
   */
  setMuted(muted: boolean): void {
    if (this.localStream) {
      this.localStream.getAudioTracks().forEach(track => {
        track.enabled = !muted
      })
    }
  }

  /**
   * 检查是否静音
   */
  isMuted(): boolean {
    if (this.localStream) {
      const audioTrack = this.localStream.getAudioTracks()[0]
      return audioTrack ? !audioTrack.enabled : true
    }
    return true
  }

  /**
   * 获取音频音量级别（用于音量指示器）
   */
  async getAudioLevel(stream: MediaStream): Promise<number> {
    return new Promise((resolve) => {
      const audioContext = new AudioContext()
      const analyser = audioContext.createAnalyser()
      const source = audioContext.createMediaStreamSource(stream)
      
      source.connect(analyser)
      analyser.fftSize = 256
      
      const dataArray = new Uint8Array(analyser.frequencyBinCount)
      analyser.getByteFrequencyData(dataArray)
      
      // 计算平均音量
      const average = dataArray.reduce((a, b) => a + b, 0) / dataArray.length
      
      audioContext.close()
      resolve(average / 255) // 归一化到 0-1
    })
  }

  /**
   * 关闭连接
   */
  close(): void {
    console.log('[WebRTC] Closing connection')

    // 停止本地流
    if (this.localStream) {
      this.localStream.getTracks().forEach(track => track.stop())
      this.localStream = null
    }

    // 停止远程流
    if (this.remoteStream) {
      this.remoteStream.getTracks().forEach(track => track.stop())
      this.remoteStream = null
    }

    // 关闭 PeerConnection
    if (this.peerConnection) {
      this.peerConnection.close()
      this.peerConnection = null
    }

    this.pendingIceCandidates = []
    this.updateConnectionState('disconnected')
  }

  /**
   * 重置管理器
   */
  reset(): void {
    this.close()
    this.connectionState = 'idle'
  }
}

// 导出单例工厂函数
export function createWebRTCManager(config?: WebRTCConfig): WebRTCManager {
  return new WebRTCManager(config)
}