nl-um-vue-ts/src/composables/useWebRTC.ts

import { reactive, shallowRef } from 'vue'
import * as systemApi from '@/api/modules/system'
import * as messageApi from '@/api/modules/message'
import { wsManager } from '@/api/websocket'
import { useToastStore } from '@/stores/toast'
import type { ChatMessage } from '@/types/api'
import type { CallStatus } from '@/types/message'

export interface CallState {
  active: boolean
  minimized: boolean
  type: 'audio' | 'video'
  status: 'idle' | 'outgoing' | 'incoming' | 'connected'
  statusText: string
  id: string | null
  muted: boolean
  camOff: boolean
  duration: number
  startTime: number | null
}

export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string) => void) {
  const toastStore = useToastStore()

  const call = reactive<CallState>({
    active: false,
    minimized: false,
    type: 'video',
    status: 'idle',
    statusText: '',
    id: null,
    muted: false,
    camOff: false,
    duration: 0,
    startTime: null,
  })

  // 使用 shallowRef 存储 MediaStream，避免 Vue 进行深层代理导致的性能问题
  const localStream = shallowRef<MediaStream | null>(null)
  const remoteStream = shallowRef<MediaStream | null>(null)

  let durationTimer: number | null = null
  let pc: RTCPeerConnection | null = null
  const pendingCandidates: RTCIceCandidate[] = []
  let currentReceiverUserId = ''

  /**
   * 初始化媒体流
   */
  async function initMedia(videoEnabled: boolean): Promise<void> {
    try {
      // 停止之前的流
      if (localStream.value) {
        localStream.value.getTracks().forEach(t => t.stop())
      }

      const constraints: MediaStreamConstraints = {
        video: videoEnabled ? {
          width: { ideal: 1280 },
          height: { ideal: 720 },
          facingMode: 'user', // 优先使用前置摄像头
        } : false,
        audio: {
          echoCancellation: true,
          noiseSuppression: true,
          autoGainControl: true,
        },
      }

      const stream = await navigator.mediaDevices.getUserMedia(constraints)
      localStream.value = stream // 赋值给响应式对象，视图会自动更新
      console.log('[WebRTC] Local media stream obtained')
    } catch (error) {
      console.error('Failed to get media:', error)
      throw new Error('无法获取摄像头或麦克风权限')
    }
  }

  /**
   * 获取并增强 ICE Servers 配置
   */
  async function getIceServers() {
    const servers: RTCIceServer[] = []
    try {
      const apiServers = await systemApi.getIceServers(userId)
      if (apiServers && apiServers.length > 0) {
        servers.push(...apiServers.map((s: any) => ({
          urls: s.urls,
          username: s.username,
          credential: s.credential,
        })))
      }
    } catch (error) {
      console.warn('Failed to fetch ICE servers from API')
    }
    // 添加公共 STUN 服务器作为兜底
    servers.push({ urls: 'stun:stun.l.google.com:19302' })
    servers.push({ urls: 'stun:global.stun.twilio.com:3478' })
    return servers
  }

  /**
   * 创建 PeerConnection
   */
  async function createPC(): Promise<void> {
    const iceServers = await getIceServers()

    // 关闭旧连接
    if (pc) pc.close()

    pc = new RTCPeerConnection({
      iceServers,
      iceCandidatePoolSize: 10
    })

    // 添加本地轨道
    if (localStream.value) {
      localStream.value.getTracks().forEach((track) => {
        pc!.addTrack(track, localStream.value!)
      })
    } else {
      console.warn('[WebRTC] No local stream to add to PC')
    }

    // 监听远程轨道
    pc.ontrack = (e) => {
      console.log('[WebRTC] Received remote track', e.streams)
      if (e.streams && e.streams[0]) {
        remoteStream.value = e.streams[0] // 赋值给响应式对象
      }
    }

    // 监听 ICE 候选
    pc.onicecandidate = (e) => {
      if (e.candidate && call.id) {
        sendSignal('candidate', e.candidate)
      }
    }

    // 监听连接状态
    pc.onconnectionstatechange = () => {
      console.log('[WebRTC] Connection state:', pc?.connectionState)
      if (pc?.connectionState === 'disconnected' || pc?.connectionState === 'failed') {
        toastStore.error('通话连接中断')
      }
    }
  }

  /**
   * 发送信令 (统一封装)
   */
  function sendSignal(status: CallStatus, data?: any, receiverUserId?: string) {
    if (!call.id) return
    const targetUserId = receiverUserId || currentReceiverUserId
    if (!targetUserId) return

    const payload = {
      sender_client_id: wsManager.getClientId() || '',
      receiver_user_id: targetUserId,
      room_id: '',
      message_type: 6, // Signaling Message
      content: JSON.stringify(data || {}),
      call_id: call.id,
      call_status: status,
      extra: JSON.stringify({ type: call.type }),
    }

    messageApi.sendMessage(payload).catch(console.error)
  }

  // --- 计时器逻辑 ---
  function startCallTimer() {
    stopCallTimer()
    call.startTime = Date.now()
    call.duration = 0
    durationTimer = window.setInterval(() => {
      if (call.startTime) call.duration = Math.floor((Date.now() - call.startTime) / 1000)
    }, 1000)
  }

  function stopCallTimer() {
    if (durationTimer) {
      clearInterval(durationTimer)
      durationTimer = null
    }
    call.startTime = null
    call.duration = 0
  }

  function formatDuration(seconds: number): string {
    const m = Math.floor(seconds / 60).toString().padStart(2, '0')
    const s = (seconds % 60).toString().padStart(2, '0')
    return `${m}:${s}`
  }

  // --- 通话控制 ---

  async function startCall(type: 'audio' | 'video', receiverUserId: string) {
    currentReceiverUserId = receiverUserId
    call.type = type
    call.id = Date.now().toString()
    call.active = true
    call.minimized = false
    call.status = 'outgoing'
    call.statusText = '正在呼叫...'
    call.duration = 0
    remoteStream.value = null // 重置远程流

    try {
      await initMedia(type === 'video')
      await createPC()
      sendSignal('invite', undefined, receiverUserId)
    } catch (error: any) {
      toastStore.error(error.message || '无法启动通话')
      endCall()
    }
  }

  async function acceptCall(senderUserId?: string) {
    if (senderUserId) currentReceiverUserId = senderUserId
    call.status = 'connected'
    call.statusText = '正在连接...'

    try {
      await initMedia(call.type === 'video')
      await createPC()
      // 发送 accepted，等待发起方创建 Offer
      sendSignal('accepted', undefined, currentReceiverUserId)
    } catch (error) {
      endCall()
    }
  }

  function endCall() {
    sendSignal('hangup')
    closeCall()
  }

  function closeCall() {
    call.active = false
    call.status = 'idle'
    call.statusText = ''
    call.id = null
    stopCallTimer()

    if (pc) {
      pc.close()
      pc = null
    }

    // 停止本地流
    if (localStream.value) {
      localStream.value.getTracks().forEach(t => t.stop())
      localStream.value = null
    }
    remoteStream.value = null
  }

  // --- 信令处理 ---

  async function handleSignaling(message: ChatMessage) {
    const signal = message.call_status as CallStatus
    const content = message.content ? JSON.parse(message.content) : {}
    const extra = message.extra ? (typeof message.extra === 'string' ? JSON.parse(message.extra) : message.extra) : {}

    // 更新通话类型
    if (extra.type) call.type = extra.type

    if (signal === 'invite') {
      currentReceiverUserId = message.sender_user_id
      call.id = message.call_id
      call.active = true
      call.minimized = false
      call.status = 'incoming'
      call.statusText = `邀请你进行${call.type === 'video' ? '视频' : '语音'}通话`
      if (onIncomingCall) onIncomingCall(message.sender_user_id)

    } else if (signal === 'accepted') {
      // 对方已接受，作为发起方，创建 Offer
      call.status = 'connected'
      call.statusText = '通话中'
      startCallTimer()

      const offer = await pc!.createOffer()
      await pc!.setLocalDescription(offer)
      sendSignal('offer', offer)

    } else if (signal === 'offer') {
      // 接收 Offer，创建 Answer
      if (!pc) {
        await initMedia(call.type === 'video')
        await createPC()
      }
      await pc!.setRemoteDescription(content)
      // 处理缓冲的 Candidates
      processPendingCandidates()

      const answer = await pc!.createAnswer()
      await pc!.setLocalDescription(answer)
      sendSignal('answer', answer, message.sender_user_id)

      call.status = 'connected'
      call.statusText = '通话中'
      startCallTimer()

    } else if (signal === 'answer') {
      // 接收 Answer
      await pc!.setRemoteDescription(content)
      processPendingCandidates()

    } else if (signal === 'candidate') {
      if (pc && pc.remoteDescription) {
        await pc.addIceCandidate(content)
      } else {
        pendingCandidates.push(content)
      }

    } else if (['hangup', 'ended', 'answered_elsewhere'].includes(signal)) {
      closeCall()
      if (signal === 'answered_elsewhere') toastStore.info('已在其他设备接听')
    }
  }

  async function processPendingCandidates() {
    while (pendingCandidates.length > 0) {
      const candidate = pendingCandidates.shift()
      if (candidate && pc) {
        await pc.addIceCandidate(candidate)
      }
    }
  }

  // --- 媒体控制 ---

  function toggleMute() {
    call.muted = !call.muted
    if (localStream.value) {
      localStream.value.getAudioTracks().forEach(t => t.enabled = !call.muted)
    }
  }

  function toggleCamera() {
    call.camOff = !call.camOff
    if (localStream.value) {
      localStream.value.getVideoTracks().forEach(t => t.enabled = !call.camOff)
    }
  }

  return {
    call,
    localStream, // 暴露流
    remoteStream, // 暴露流
    startCall,
    acceptCall,
    endCall,
    handleSignaling,
    toggleMute,
    toggleCamera,
    formatDuration,
  }
}