1152 lines
40 KiB
TypeScript
1152 lines
40 KiB
TypeScript
import { reactive, shallowRef, ref } from 'vue'
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import flvjs from 'flv.js'
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import * as messageApi from '@/api/modules/message'
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import * as systemApi from '@/api/modules/system'
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import * as callApi from '@/api/modules/call'
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import { wsManager } from '@/api/websocket'
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import { useToastStore } from '@/stores/toast'
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import { useChatStore } from '@/stores/chat'
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import { useAuthStore } from '@/stores/auth'
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import { isRTMPMode, isAutoMode, shouldUseRTMP, parsePlatform, type Platform } from '@/config/call'
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import type { ChatMessage, Contact, ICEServerConfig } from '@/types/api'
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import type { CallStatus } from '@/types/message'
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export interface CallState {
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active: boolean
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minimized: boolean
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type: 'audio' | 'video'
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status: 'idle' | 'outgoing' | 'incoming' | 'connected'
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statusText: string
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id: string | null
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muted: boolean
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remoteMuted: boolean
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camOff: boolean
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remoteCamOff: boolean
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duration: number
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startTime: number | null
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// 小程序流信息
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miniprogramStreams: FlvStreamInfo[]
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// 来电者信息（用于来电显示）
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callerName?: string
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callerAvatar?: string
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callerId?: string
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}
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// FLV 流信息（来自小程序端）
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export interface FlvStreamInfo {
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userId: string
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flvUrl: string
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player?: flvjs.Player
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}
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const RINGTONE_INCOMING_BASE64 = 'data:audio/mp3;base64,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'
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export function useWebRTC(
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userId: string,
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onIncomingCall?: (senderUserId: string) => void,
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getRoomId?: (receiverUserId?: string) => string
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) {
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const toastStore = useToastStore()
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const chatStore = useChatStore()
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const authStore = useAuthStore()
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const call = reactive<CallState>({
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active: false,
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minimized: false,
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type: 'video',
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status: 'idle',
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statusText: '',
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id: null,
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muted: false,
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remoteMuted: false,
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camOff: false,
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remoteCamOff: false,
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duration: 0,
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startTime: null,
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miniprogramStreams: [],
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// 来电者信息
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callerName: undefined,
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callerAvatar: undefined,
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callerId: undefined,
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})
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const isCaller = ref(false)
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const localStream = shallowRef<MediaStream | null>(null)
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const remoteStream = shallowRef<MediaStream | null>(null)
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let durationTimer: number | null = null
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let pc: RTCPeerConnection | null = null
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const pendingCandidates: RTCIceCandidate[] = []
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let currentReceiverUserId = ''
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let currentRoomId = ''
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// Auto 模式：跟踪对方平台
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let remotePlatform: Platform = 'unknown'
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const useRTMP = ref(false) // 当前通话是否使用 RTMP 模式
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// RTMP 模式相关
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let wsPushSocket: WebSocket | null = null
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let mediaRecorder: MediaRecorder | null = null
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const wsPushUrl = ref<string>('')
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const audioIncoming = new Audio(RINGTONE_INCOMING_BASE64)
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audioIncoming.loop = true
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// --- 信令处理 (修复版) ---
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async function handleSignaling(message: ChatMessage) {
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try {
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const content = message.content ? JSON.parse(message.content) : {}
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// 【关键修复】如果是群聊信令，直接忽略！
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// 通过判断是否存在 callRoomId 或 participantIds 来识别
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if (content.callRoomId || content.participantIds) {
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console.log('[WebRTC] 忽略群聊信令')
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return
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}
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const signal = message.call_status as any
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const extra = message.extra ? (typeof message.extra === 'string' ? JSON.parse(message.extra) : message.extra) : {}
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console.log('[WebRTC] 📨 收到信令:', signal, {
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from: message.sender_user_id,
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room_id: message.room_id,
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call_id: message.call_id,
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content: content,
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extra: extra
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})
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if (extra.type) call.type = extra.type
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if (signal === 'sync_state') {
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if (content.action === 'cam-toggle') call.remoteCamOff = content.value
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else if (content.action === 'mic-toggle') call.remoteMuted = content.value
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return
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}
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if (signal === 'invite') {
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// Auto 模式：检测对方平台，决定使用 WebRTC 还是 RTMP
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remotePlatform = parsePlatform(extra.platform)
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useRTMP.value = shouldUseRTMP(remotePlatform)
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console.log('[WebRTC] 📞 收到来电邀请:', {
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from: message.sender_user_id,
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room_id: message.room_id,
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call_id: message.call_id,
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type: extra.type,
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platform: extra.platform,
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remotePlatform,
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useRTMP: useRTMP.value,
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senderName: extra.senderName,
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senderAvatar: extra.senderAvatar
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})
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if (call.active) {
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console.log('[WebRTC] ⚠️ 当前有通话，忽略新来电')
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return
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}
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isCaller.value = false
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currentReceiverUserId = message.sender_user_id
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if (message.room_id) currentRoomId = message.room_id
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call.id = message.call_id
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call.active = true
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call.minimized = false
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call.status = 'incoming'
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call.statusText = `邀请你通话`
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// 设置来电者信息
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call.callerId = message.sender_user_id
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// 优先从信令 extra 获取来电者信息（小程序端会发送）
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if (extra.senderName) {
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call.callerName = extra.senderName
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call.callerAvatar = extra.senderAvatar
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} else {
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// 否则从本地联系人列表查找
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const contact = chatStore.contacts.find(c =>
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c.contact_user_id === message.sender_user_id ||
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c.user_id === message.sender_user_id
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)
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if (contact?.user) {
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call.callerName = contact.remark_name || contact.user.name
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call.callerAvatar = contact.user.avatar
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}
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}
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console.log('[WebRTC] 👤 来电者信息:', { name: call.callerName, avatar: call.callerAvatar })
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console.log('[WebRTC] ✅ 设置来电状态, roomId:', currentRoomId, 'mode:', useRTMP.value ? 'RTMP' : 'WebRTC')
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playRingtone('incoming')
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if (onIncomingCall) onIncomingCall(message.sender_user_id)
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} else if (signal === 'accepted') {
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// Auto 模式：从 accepted 信令中检测对方平台，决定最终使用的模式
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const previousMode = useRTMP.value
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if (extra.platform) {
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remotePlatform = parsePlatform(extra.platform)
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useRTMP.value = shouldUseRTMP(remotePlatform)
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console.log('[WebRTC] 📱 对方平台:', extra.platform, '→', remotePlatform,
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'| 模式:', useRTMP.value ? 'RTMP' : 'WebRTC',
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previousMode !== useRTMP.value ? '(已切换)' : '')
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} else {
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// 如果没有 platform 信息，默认认为对方是 h5，使用 WebRTC
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remotePlatform = 'h5'
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useRTMP.value = false
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console.log('[WebRTC] ⚠️ 对方未提供平台信息，默认为 h5，使用 WebRTC')
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}
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console.log('[WebRTC] ✅ 对方已接听, 最终模式:', useRTMP.value ? 'RTMP' : 'WebRTC')
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stopRingtone()
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call.status = 'connected'
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call.statusText = '通话中'
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startCallTimer()
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// 加入房间获取小程序用户的 FLV 流地址（RTMP 模式下也会启动 WebSocket 推流）
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console.log('[WebRTC] 📡 准备加入房间, roomId:', currentRoomId, 'mode:', useRTMP.value ? 'RTMP' : 'WebRTC')
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await joinCallRoomAndGetFlvUrls()
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// WebRTC 模式：创建 Offer（只有 useRTMP 为 false 时才使用 WebRTC）
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if (!useRTMP.value) {
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console.log('[WebRTC] WebRTC 模式：创建 Offer')
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if (!pc) await createPC()
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const offer = await pc!.createOffer()
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await pc!.setLocalDescription(offer)
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sendSignal('offer', offer)
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} else {
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console.log('[WebRTC] RTMP 模式：等待 FLV 流')
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}
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} else if (signal === 'offer') {
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// WebRTC 模式才处理 offer（RTMP 模式不使用 WebRTC 信令）
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if (!useRTMP.value) {
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stopRingtone()
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if (!pc) {
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await initMedia(call.type === 'video')
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await createPC()
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}
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await pc!.setRemoteDescription(content)
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processPendingCandidates()
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const answer = await pc!.createAnswer()
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await pc!.setLocalDescription(answer)
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sendSignal('answer', answer, message.sender_user_id)
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call.status = 'connected'
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call.statusText = '通话中'
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startCallTimer()
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}
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} else if (signal === 'answer') {
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// WebRTC 模式才处理 answer（RTMP 模式不使用 WebRTC 信令）
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if (!useRTMP.value && pc) {
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await pc.setRemoteDescription(content)
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processPendingCandidates()
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}
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} else if (signal === 'candidate') {
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// WebRTC 模式才处理 candidate（RTMP 模式不使用 WebRTC 信令）
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if (!useRTMP.value) {
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if (pc && pc.remoteDescription) await pc.addIceCandidate(content)
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else pendingCandidates.push(content)
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}
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} else if (signal === 'participant_joined') {
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// 新参与者加入房间（来自小程序端）
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console.log('[WebRTC] 🆕 收到 participant_joined 信令:', {
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user_id: content.user_id,
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platform: content.platform,
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flv_url: content.flv_url,
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pull_url: content.pull_url
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})
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// 如果有 FLV 地址，添加或更新小程序流列表
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if (content.flv_url && content.user_id) {
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const existingIndex = call.miniprogramStreams.findIndex(s => s.userId === content.user_id)
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if (existingIndex >= 0) {
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// 更新现有流的地址（重要：当用户重新加入时使用新的流地址）
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const oldFlvUrl = call.miniprogramStreams[existingIndex].flvUrl
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call.miniprogramStreams[existingIndex].flvUrl = content.flv_url
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console.log('[WebRTC] 🔄 更新小程序流地址:', content.user_id)
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console.log('[WebRTC] 旧地址:', oldFlvUrl)
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console.log('[WebRTC] 新地址:', content.flv_url)
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} else {
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// 添加新的小程序流
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call.miniprogramStreams.push({
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userId: content.user_id,
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flvUrl: content.flv_url
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})
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console.log('[WebRTC] ✅ 添加小程序流:', content.user_id, content.flv_url)
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}
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console.log('[WebRTC] 📺 当前小程序流列表:', call.miniprogramStreams)
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} else {
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console.log('[WebRTC] ⚠️ participant_joined 没有 FLV 地址')
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}
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// 如果当前没有小程序流，尝试重新获取房间信息
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if (call.miniprogramStreams.length === 0 && currentRoomId) {
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console.log('[WebRTC] 📡 没有小程序流，尝试刷新房间信息...')
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refreshRoomStreams()
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}
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} else if (['hangup', 'ended', 'answered_elsewhere'].includes(signal)) {
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if (isCaller.value) {
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if (call.status === 'outgoing') sendSummaryMessage('rejected')
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else if (call.status === 'connected') sendSummaryMessage('connected')
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}
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closeCall()
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if (signal === 'answered_elsewhere') toastStore.info('已在其他设备接听')
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}
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} catch (error) {
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console.error('Error handling signaling:', error)
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}
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}
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// ... (其余辅助函数保持不变，为节省篇幅只展示关键变动，但你需要完整代码，下面补充完整辅助函数) ...
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function playRingtone(type: 'incoming' | 'dialing') {
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stopRingtone()
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const playPromise = audioIncoming.play()
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if (playPromise !== undefined) playPromise.catch(e => {})
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}
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function stopRingtone() {
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audioIncoming.pause()
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audioIncoming.currentTime = 0
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}
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function getSafeRoomId(targetUserId?: string): string | null {
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if (currentRoomId) return currentRoomId
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if (targetUserId && getRoomId) {
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const rid = getRoomId(targetUserId)
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if (rid) return rid
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}
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if (targetUserId && userId) {
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return [userId, targetUserId].sort().join('_')
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}
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return null
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}
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async function sendSummaryMessage(reason: 'connected' | 'cancelled' | 'rejected' | 'busy') {
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const roomId = getSafeRoomId(currentReceiverUserId)
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if (!roomId) return
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// 通话类型说明
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const callTypeText = call.type === 'video' ? '[视频通话]' : '[语音通话]'
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// 根据原因生成系统消息内容
|
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let content = ''
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switch (reason) {
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case 'connected':
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content = `${callTypeText} 通话时长 ${formatDuration(call.duration)}`
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break
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case 'cancelled':
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content = `${callTypeText} 已取消`
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break
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case 'rejected':
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content = `${callTypeText} 对方未接听`
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break
|
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case 'busy':
|
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content = `${callTypeText} 对方忙`
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break
|
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}
|
||
|
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// 构造通话结果的 extra 数据
|
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const extraData = {
|
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call_type: call.type,
|
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call_reason: reason,
|
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call_duration: call.duration,
|
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call_id: call.id
|
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}
|
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|
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try {
|
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// 发送系统消息 (message_type = 4)
|
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const payload = {
|
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sender_client_id: wsManager.getClientId() || '',
|
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receiver_user_id: currentReceiverUserId,
|
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room_id: roomId,
|
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message_type: 4, // 系统消息
|
||
content: content,
|
||
extra: JSON.stringify(extraData)
|
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}
|
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await messageApi.sendMessage(payload)
|
||
|
||
// 同时添加到本地消息列表
|
||
const systemMessage: ChatMessage = {
|
||
id: Date.now(),
|
||
room_id: roomId,
|
||
sender_user_id: userId,
|
||
message_type: 4,
|
||
content: content,
|
||
extra: extraData,
|
||
created_at: new Date().toISOString(),
|
||
updated_at: new Date().toISOString()
|
||
}
|
||
chatStore.addMessage(roomId, systemMessage)
|
||
} catch (error) {
|
||
console.error('发送通话结束消息失败:', error)
|
||
}
|
||
}
|
||
|
||
// ICE 服务器缓存
|
||
let cachedIceServers: RTCIceServer[] | null = null
|
||
|
||
/**
|
||
* 获取 ICE 服务器配置（优先从后端 API 获取，失败时使用备用配置）
|
||
*/
|
||
async function getIceServers(): Promise<RTCIceServer[]> {
|
||
// 如果已有缓存，直接使用
|
||
if (cachedIceServers) {
|
||
return cachedIceServers
|
||
}
|
||
|
||
try {
|
||
const servers = await systemApi.getIceServers(userId)
|
||
if (servers && servers.length > 0) {
|
||
// 转换为 RTCIceServer 格式
|
||
cachedIceServers = servers.map((s: ICEServerConfig) => ({
|
||
urls: s.urls,
|
||
username: s.username,
|
||
credential: s.credential
|
||
}))
|
||
return cachedIceServers
|
||
}
|
||
} catch (error) {
|
||
console.warn('获取 ICE 服务器配置失败，使用备用配置:', error)
|
||
}
|
||
|
||
// 备用配置：公共 STUN 服务器
|
||
cachedIceServers = [
|
||
{ urls: 'stun:stun.l.google.com:19302' },
|
||
{ urls: 'stun:stun1.l.google.com:19302' }
|
||
]
|
||
return cachedIceServers
|
||
}
|
||
|
||
// --- WebRTC ---
|
||
async function initMedia(videoEnabled: boolean): Promise<void> {
|
||
try {
|
||
if (localStream.value) {
|
||
localStream.value.getTracks().forEach(t => t.stop())
|
||
localStream.value = null
|
||
}
|
||
const constraints: MediaStreamConstraints = {
|
||
video: videoEnabled ? { width: { ideal: 640 }, height: { ideal: 480 } } : false,
|
||
audio: { echoCancellation: true, noiseSuppression: true },
|
||
}
|
||
const stream = await navigator.mediaDevices.getUserMedia(constraints)
|
||
localStream.value = stream
|
||
} catch (error: any) {
|
||
throw new Error('无法获取设备权限')
|
||
}
|
||
}
|
||
|
||
async function createPC(): Promise<void> {
|
||
const iceServers = await getIceServers()
|
||
if (pc) { pc.close(); pc = null }
|
||
pc = new RTCPeerConnection({ iceServers })
|
||
|
||
pc.oniceconnectionstatechange = () => {
|
||
if (pc?.iceConnectionState === 'failed') {
|
||
call.statusText = '连接失败'
|
||
endCall()
|
||
} else if (pc?.iceConnectionState === 'connected') {
|
||
call.statusText = '通话中'
|
||
}
|
||
}
|
||
|
||
if (localStream.value) {
|
||
localStream.value.getTracks().forEach((track) => pc!.addTrack(track, localStream.value!))
|
||
}
|
||
|
||
pc.ontrack = (e) => {
|
||
if (e.streams && e.streams[0]) {
|
||
remoteStream.value = e.streams[0]
|
||
}
|
||
}
|
||
|
||
pc.onicecandidate = (e) => {
|
||
if (e.candidate && call.id) sendSignal('candidate', e.candidate)
|
||
}
|
||
}
|
||
|
||
function sendSignal(status: CallStatus, data?: any, receiverUserId?: string) {
|
||
if (!call.id) {
|
||
console.warn('[WebRTC] ⚠️ sendSignal: call.id 为空，跳过')
|
||
return
|
||
}
|
||
const targetUserId = receiverUserId || currentReceiverUserId
|
||
const roomId = getSafeRoomId(targetUserId)
|
||
if (!roomId) {
|
||
console.warn('[WebRTC] ⚠️ sendSignal: roomId 为空，跳过')
|
||
return
|
||
}
|
||
|
||
console.log('[WebRTC] 📤 发送信令:', status, { to: targetUserId, roomId, callId: call.id })
|
||
|
||
// 获取当前用户信息，发送给对方（让对方显示来电者信息）
|
||
const currentUser = authStore.user
|
||
|
||
const payload = {
|
||
sender_client_id: wsManager.getClientId() || '',
|
||
receiver_user_id: targetUserId,
|
||
room_id: roomId,
|
||
message_type: 6,
|
||
content: JSON.stringify(data || {}),
|
||
call_id: call.id,
|
||
call_status: status,
|
||
// 包含平台信息和发送者信息，让对方显示来电者昵称和头像
|
||
extra: JSON.stringify({
|
||
type: call.type,
|
||
platform: 'web',
|
||
senderName: currentUser?.name,
|
||
senderAvatar: currentUser?.avatar
|
||
}),
|
||
}
|
||
messageApi.sendMessage(payload).catch(err => {
|
||
console.error('[WebRTC] ❌ 发送信令失败:', err)
|
||
})
|
||
}
|
||
|
||
function sendSyncState(type: 'cam-toggle' | 'mic-toggle', value: boolean) {
|
||
if (call.status !== 'connected') return
|
||
const roomId = getSafeRoomId(currentReceiverUserId)
|
||
if (!roomId) return
|
||
const payload = {
|
||
sender_client_id: wsManager.getClientId() || '',
|
||
receiver_user_id: currentReceiverUserId,
|
||
room_id: roomId,
|
||
message_type: 6,
|
||
content: JSON.stringify({ action: type, value }),
|
||
call_id: call.id,
|
||
call_status: 'sync_state' as any,
|
||
extra: JSON.stringify({ type: call.type })
|
||
}
|
||
messageApi.sendMessage(payload).catch(console.error)
|
||
}
|
||
|
||
async function startCall(type: 'audio' | 'video', receiverUserId: string, roomId?: string, contact?: Contact) {
|
||
console.log('[WebRTC] 📞 发起通话:', { type, receiverUserId, roomId, contactRoomId: contact?.room_id })
|
||
|
||
if (roomId) currentRoomId = roomId
|
||
else if (contact?.room_id) currentRoomId = contact.room_id
|
||
else if (getRoomId) {
|
||
const found = getRoomId(receiverUserId)
|
||
if (found) currentRoomId = found
|
||
}
|
||
|
||
if (!currentRoomId) {
|
||
console.error('[WebRTC] ❌ 缺少房间信息')
|
||
toastStore.error('无法建立通话连接：缺少房间信息')
|
||
return false
|
||
}
|
||
|
||
// Auto 模式：发起通话时，默认使用 WebRTC，收到 accepted 后根据对方平台调整
|
||
// 如果对方是小程序，会在 accepted 处理时切换到 RTMP
|
||
// 固定模式：使用配置的模式
|
||
if (isAutoMode()) {
|
||
// Auto 模式下，发起方先假设使用 WebRTC（Web-to-Web 最优）
|
||
// 收到 accepted 后，如果对方是小程序才切换到 RTMP
|
||
useRTMP.value = false
|
||
console.log('[WebRTC] ✅ 房间ID:', currentRoomId, 'Auto 模式：发起通话，默认 WebRTC，等待对方平台信息')
|
||
} else {
|
||
useRTMP.value = isRTMPMode()
|
||
console.log('[WebRTC] ✅ 房间ID:', currentRoomId, '固定模式:', useRTMP.value ? 'RTMP' : 'WebRTC')
|
||
}
|
||
|
||
isCaller.value = true
|
||
currentReceiverUserId = receiverUserId
|
||
remotePlatform = 'unknown' // 重置远程平台
|
||
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
|
||
call.camOff = false
|
||
call.remoteCamOff = false
|
||
call.remoteMuted = false
|
||
remoteStream.value = null
|
||
|
||
try {
|
||
console.log('[WebRTC] 🎥 初始化媒体设备...')
|
||
await initMedia(type === 'video')
|
||
console.log('[WebRTC] ✅ 媒体设备初始化成功')
|
||
|
||
// 根据当前模式选择不同的推流方式
|
||
if (useRTMP.value) {
|
||
console.log('[WebRTC] 📡 使用 RTMP 模式（auto 模式下可能在 accepted 后切换）')
|
||
// RTMP 模式：加入房间后会自动开始 WebSocket 推流
|
||
} else {
|
||
console.log('[WebRTC] 📡 使用 WebRTC 模式')
|
||
await createPC()
|
||
}
|
||
|
||
playRingtone('dialing')
|
||
console.log('[WebRTC] 📤 发送 invite 信令给:', receiverUserId)
|
||
sendSignal('invite', undefined, receiverUserId)
|
||
return true
|
||
} catch (error: any) {
|
||
console.error('[WebRTC] ❌ 发起通话失败:', error)
|
||
toastStore.error(error.message || '无法启动通话')
|
||
closeCall()
|
||
return false
|
||
}
|
||
}
|
||
|
||
async function acceptCall(senderUserId?: string) {
|
||
// 接听时 useRTMP 已经在 invite 处理时设置好了
|
||
console.log('[WebRTC] 📞 接听来电:', { senderUserId, currentRoomId, mode: useRTMP.value ? 'RTMP' : 'WebRTC' })
|
||
|
||
isCaller.value = false
|
||
stopRingtone()
|
||
if (senderUserId) currentReceiverUserId = senderUserId
|
||
call.status = 'connected'
|
||
call.statusText = '正在连接...'
|
||
call.camOff = false
|
||
call.remoteCamOff = false
|
||
|
||
try {
|
||
console.log('[WebRTC] 🎥 初始化媒体设备...')
|
||
await initMedia(call.type === 'video')
|
||
console.log('[WebRTC] ✅ 媒体设备初始化成功')
|
||
|
||
// 根据模式选择不同的推流方式（useRTMP 在收到 invite 时已确定）
|
||
if (useRTMP.value) {
|
||
console.log('[WebRTC] 📡 接听：使用 RTMP 模式，准备加入房间')
|
||
// RTMP 模式：加入房间后会自动开始 WebSocket 推流
|
||
await joinCallRoomAndGetFlvUrls()
|
||
} else {
|
||
console.log('[WebRTC] 📡 接听：使用 WebRTC 模式')
|
||
await createPC()
|
||
}
|
||
|
||
console.log('[WebRTC] 📤 发送 accepted 信令给:', currentReceiverUserId)
|
||
sendSignal('accepted', undefined, currentReceiverUserId)
|
||
} catch (error) {
|
||
console.error('[WebRTC] ❌ 接听失败:', error)
|
||
endCall()
|
||
}
|
||
}
|
||
|
||
function endCall() {
|
||
stopRingtone()
|
||
// 只有发起方发送通话结束系统消息，避免重复
|
||
if (isCaller.value) {
|
||
if (call.status === 'outgoing') {
|
||
sendSummaryMessage('cancelled') // 呼出状态主动取消
|
||
} else if (call.status === 'connected') {
|
||
sendSummaryMessage('connected') // 通话中主动挂断
|
||
}
|
||
}
|
||
// 被叫方不发送消息，由发起方统一发送
|
||
sendSignal('hangup')
|
||
// 离开通话房间
|
||
leaveCallRoom()
|
||
closeCall()
|
||
}
|
||
|
||
function closeCall() {
|
||
stopRingtone()
|
||
currentRoomId = ''
|
||
call.active = false
|
||
call.status = 'idle'
|
||
call.statusText = ''
|
||
call.id = null
|
||
// 清除来电者信息
|
||
call.callerName = undefined
|
||
call.callerAvatar = undefined
|
||
call.callerId = undefined
|
||
stopCallTimer()
|
||
|
||
// 停止 WebSocket 推流 (RTMP 模式)
|
||
stopWebSocketPush()
|
||
|
||
if (pc) {
|
||
pc.close()
|
||
pc = null
|
||
}
|
||
if (localStream.value) {
|
||
localStream.value.getTracks().forEach(t => t.stop())
|
||
localStream.value = null
|
||
}
|
||
remoteStream.value = null
|
||
// 清理 FLV 播放器
|
||
destroyFlvPlayers()
|
||
call.miniprogramStreams = []
|
||
}
|
||
|
||
/**
|
||
* 加入通话房间并获取小程序用户的 FLV 流地址
|
||
*/
|
||
async function joinCallRoomAndGetFlvUrls(): Promise<void> {
|
||
console.log('[WebRTC] 🚪 准备加入房间:', { roomId: currentRoomId, userId })
|
||
|
||
if (!currentRoomId) {
|
||
console.error('[WebRTC] ❌ joinCallRoomAndGetFlvUrls: currentRoomId 为空，跳过')
|
||
return
|
||
}
|
||
|
||
try {
|
||
console.log('[WebRTC] 📤 发送加入房间请求...')
|
||
const response = await callApi.joinCallRoom({
|
||
room_id: currentRoomId,
|
||
user_id: userId,
|
||
platform: 'web'
|
||
})
|
||
|
||
console.log('[WebRTC] ✅ 加入房间响应:', JSON.stringify(response, null, 2))
|
||
|
||
// 检查是否有小程序用户的 FLV 流
|
||
console.log('[WebRTC] 📺 检查 FLV 拉流地址:', response.flv_pull_urls)
|
||
if (response.flv_pull_urls && response.flv_pull_urls.length > 0) {
|
||
console.log('[WebRTC] ✅ 检测到小程序用户流:', response.flv_pull_urls.length, '个')
|
||
call.miniprogramStreams = response.flv_pull_urls.map(p => ({
|
||
userId: p.user_id,
|
||
flvUrl: p.flv_url || ''
|
||
})).filter(s => s.flvUrl)
|
||
console.log('[WebRTC] 📺 已添加小程序流:', call.miniprogramStreams)
|
||
} else {
|
||
console.log('[WebRTC] ⚠️ 没有检测到小程序用户流')
|
||
}
|
||
|
||
// 检查参与者列表
|
||
console.log('[WebRTC] 👥 房间参与者:', response.participants)
|
||
|
||
// RTMP 模式：获取 WebSocket 推流地址并开始推流
|
||
if (useRTMP.value && response.ws_push_url) {
|
||
wsPushUrl.value = response.ws_push_url
|
||
console.log('[WebRTC] RTMP 模式，WebSocket 推流地址:', response.ws_push_url)
|
||
|
||
// 确保 localStream 存在再推流，增加重试机制
|
||
const tryStartPush = (retryCount = 0) => {
|
||
if (localStream.value) {
|
||
console.log('[WebRTC] ✅ localStream 就绪，开始 WebSocket 推流')
|
||
startWebSocketPush(response.ws_push_url!)
|
||
} else if (retryCount < 10) {
|
||
console.warn(`[WebRTC] ⚠️ localStream 未就绪，500ms 后重试 (${retryCount + 1}/10)`)
|
||
setTimeout(() => tryStartPush(retryCount + 1), 500)
|
||
} else {
|
||
console.error('[WebRTC] ❌ localStream 始终未就绪，放弃推流')
|
||
}
|
||
}
|
||
tryStartPush()
|
||
}
|
||
|
||
// 关键修复：通知房间内其他参与者"我已加入"
|
||
// 发送 participant_joined 信令，让小程序端知道可以拉取我的流
|
||
sendParticipantJoinedSignal(response)
|
||
|
||
} catch (error) {
|
||
console.error('[WebRTC] 加入房间失败:', error)
|
||
}
|
||
}
|
||
|
||
/**
|
||
* 发送 participant_joined 信令通知其他参与者
|
||
*/
|
||
function sendParticipantJoinedSignal(response: callApi.JoinCallRoomResponse) {
|
||
console.log('[WebRTC] 📤 准备发送 participant_joined 信令:', { callId: call.id, roomId: currentRoomId, to: currentReceiverUserId })
|
||
|
||
if (!call.id || !currentRoomId) {
|
||
console.warn('[WebRTC] ⚠️ sendParticipantJoinedSignal: call.id 或 roomId 为空，跳过')
|
||
return
|
||
}
|
||
|
||
// 构建参与者信息，包含自己的 FLV 地址供小程序拉流
|
||
const participantInfo = {
|
||
user_id: userId,
|
||
platform: 'web',
|
||
// Web 端的 FLV 地址（供小程序拉流）
|
||
pull_url: response.ws_push_url || '', // WebSocket 推流地址（仅供参考）
|
||
flv_url: response.self_flv_url || '', // HTTP-FLV 地址（供小程序 live-player 播放）
|
||
}
|
||
|
||
console.log('[WebRTC] 📤 发送 participant_joined 信令:', participantInfo, 'to:', currentReceiverUserId)
|
||
|
||
const payload = {
|
||
sender_client_id: wsManager.getClientId() || '',
|
||
receiver_user_id: currentReceiverUserId,
|
||
room_id: currentRoomId,
|
||
message_type: 6,
|
||
content: JSON.stringify(participantInfo),
|
||
call_id: call.id,
|
||
call_status: 'participant_joined' as any,
|
||
extra: JSON.stringify({ type: call.type, platform: 'web' }),
|
||
}
|
||
messageApi.sendMessage(payload).then(() => {
|
||
console.log('[WebRTC] ✅ participant_joined 信令发送成功')
|
||
}).catch(err => {
|
||
console.error('[WebRTC] ❌ participant_joined 信令发送失败:', err)
|
||
})
|
||
}
|
||
|
||
/**
|
||
* 刷新房间流信息（当收到 participant_joined 但没有流地址时）
|
||
*/
|
||
async function refreshRoomStreams() {
|
||
if (!currentRoomId) return
|
||
|
||
try {
|
||
const response = await callApi.joinCallRoom({
|
||
room_id: currentRoomId,
|
||
user_id: userId,
|
||
platform: 'web'
|
||
})
|
||
|
||
console.log('[WebRTC] 刷新房间流信息:', response)
|
||
|
||
// 更新小程序流列表
|
||
if (response.flv_pull_urls && response.flv_pull_urls.length > 0) {
|
||
for (const p of response.flv_pull_urls) {
|
||
const exists = call.miniprogramStreams.some(s => s.userId === p.user_id)
|
||
if (!exists && p.flv_url) {
|
||
call.miniprogramStreams.push({
|
||
userId: p.user_id,
|
||
flvUrl: p.flv_url
|
||
})
|
||
console.log('[WebRTC] 刷新后添加小程序流:', p.user_id, p.flv_url)
|
||
}
|
||
}
|
||
}
|
||
} catch (error) {
|
||
console.error('[WebRTC] 刷新房间流信息失败:', error)
|
||
}
|
||
}
|
||
|
||
/**
|
||
* 开始 WebSocket 推流 (RTMP 模式)
|
||
* 使用 MediaRecorder 编码本地流，通过 WebSocket 发送到服务器
|
||
*/
|
||
function startWebSocketPush(wsUrl: string): void {
|
||
if (!localStream.value) {
|
||
console.error('[WebRTC] 无法推流：本地流不存在')
|
||
return
|
||
}
|
||
|
||
// 关闭已有连接
|
||
stopWebSocketPush()
|
||
|
||
console.log('[WebRTC] 开始 WebSocket 推流:', wsUrl)
|
||
|
||
// 创建 WebSocket 连接
|
||
wsPushSocket = new WebSocket(wsUrl)
|
||
|
||
wsPushSocket.onopen = () => {
|
||
console.log('[WebRTC] WebSocket 连接已建立')
|
||
// 连接成功后开始录制
|
||
startMediaRecorder()
|
||
}
|
||
|
||
wsPushSocket.onclose = (event) => {
|
||
console.log('[WebRTC] WebSocket 连接已关闭:', event.code, event.reason)
|
||
}
|
||
|
||
wsPushSocket.onerror = (error) => {
|
||
console.error('[WebRTC] WebSocket 错误:', error)
|
||
toastStore.error('推流连接失败')
|
||
}
|
||
|
||
wsPushSocket.onmessage = (event) => {
|
||
// 处理服务器消息（如有需要）
|
||
console.log('[WebRTC] WebSocket 收到消息:', event.data)
|
||
}
|
||
}
|
||
|
||
/**
|
||
* 开始 MediaRecorder 录制
|
||
* 优先使用 H.264 编码，因为 H.264 可以直接封装到 FLV，无需转码
|
||
* 如果浏览器不支持 H.264，则降级到 VP8/VP9（需要服务器端 FFmpeg 转码）
|
||
*/
|
||
function startMediaRecorder(): void {
|
||
if (!localStream.value || !wsPushSocket) {
|
||
return
|
||
}
|
||
|
||
// 检查浏览器支持的 MIME 类型
|
||
// 优先级：H.264 > VP8 > VP9 > 默认
|
||
// H.264 可以直接重封装为 FLV，VP8/VP9 需要 FFmpeg 转码
|
||
const mimeTypes = [
|
||
'video/webm;codecs=h264,opus', // 优先：H.264 可直接封装 FLV
|
||
'video/webm;codecs=avc1,opus', // 备选：H.264 另一种写法
|
||
'video/webm;codecs=vp8,opus', // 降级：需要 FFmpeg 转码
|
||
'video/webm;codecs=vp9,opus', // 降级：需要 FFmpeg 转码
|
||
'video/webm', // 最后兜底
|
||
]
|
||
|
||
let selectedMimeType = ''
|
||
let selectedCodec = 'unknown'
|
||
for (const mimeType of mimeTypes) {
|
||
if (MediaRecorder.isTypeSupported(mimeType)) {
|
||
selectedMimeType = mimeType
|
||
// 解析出视频编解码器
|
||
if (mimeType.includes('h264') || mimeType.includes('avc1')) {
|
||
selectedCodec = 'h264'
|
||
} else if (mimeType.includes('vp9')) {
|
||
selectedCodec = 'vp9'
|
||
} else if (mimeType.includes('vp8')) {
|
||
selectedCodec = 'vp8'
|
||
}
|
||
break
|
||
}
|
||
}
|
||
|
||
if (!selectedMimeType) {
|
||
console.error('[WebRTC] 浏览器不支持 WebM 录制')
|
||
toastStore.error('您的浏览器不支持视频推流')
|
||
return
|
||
}
|
||
|
||
console.log('[WebRTC] 使用编码格式:', selectedMimeType, '视频编解码器:', selectedCodec)
|
||
|
||
// 如果不是 H.264，提示用户可能需要转码
|
||
if (selectedCodec !== 'h264') {
|
||
console.warn('[WebRTC] ⚠️ 浏览器不支持 H.264 编码，使用', selectedCodec, '（需要服务器转码）')
|
||
}
|
||
|
||
try {
|
||
mediaRecorder = new MediaRecorder(localStream.value, {
|
||
mimeType: selectedMimeType,
|
||
videoBitsPerSecond: 500000, // 500kbps
|
||
audioBitsPerSecond: 64000, // 64kbps
|
||
})
|
||
|
||
// 首次发送编解码器信息给服务器
|
||
let codecInfoSent = false
|
||
|
||
mediaRecorder.ondataavailable = (event) => {
|
||
if (event.data.size > 0 && wsPushSocket?.readyState === WebSocket.OPEN) {
|
||
// 首次发送时，先发送编解码器信息（JSON 格式）
|
||
if (!codecInfoSent) {
|
||
const codecInfo = JSON.stringify({
|
||
type: 'codec_info',
|
||
video_codec: selectedCodec,
|
||
audio_codec: 'opus',
|
||
mime_type: selectedMimeType
|
||
})
|
||
wsPushSocket.send(codecInfo)
|
||
codecInfoSent = true
|
||
console.log('[WebRTC] 已发送编解码器信息:', codecInfo)
|
||
}
|
||
wsPushSocket.send(event.data)
|
||
}
|
||
}
|
||
|
||
mediaRecorder.onerror = (error) => {
|
||
console.error('[WebRTC] MediaRecorder 错误:', error)
|
||
}
|
||
|
||
mediaRecorder.onstart = () => {
|
||
console.log('[WebRTC] MediaRecorder 开始录制, 编解码器:', selectedCodec)
|
||
}
|
||
|
||
mediaRecorder.onstop = () => {
|
||
console.log('[WebRTC] MediaRecorder 停止录制')
|
||
}
|
||
|
||
// 每 100ms 发送一次数据
|
||
mediaRecorder.start(100)
|
||
|
||
} catch (error) {
|
||
console.error('[WebRTC] 创建 MediaRecorder 失败:', error)
|
||
toastStore.error('初始化推流失败')
|
||
}
|
||
}
|
||
|
||
/**
|
||
* 停止 WebSocket 推流
|
||
*/
|
||
function stopWebSocketPush(): void {
|
||
// 停止 MediaRecorder
|
||
if (mediaRecorder) {
|
||
try {
|
||
if (mediaRecorder.state !== 'inactive') {
|
||
mediaRecorder.stop()
|
||
}
|
||
} catch (e) {
|
||
// 忽略错误
|
||
}
|
||
mediaRecorder = null
|
||
}
|
||
|
||
// 关闭 WebSocket
|
||
if (wsPushSocket) {
|
||
try {
|
||
wsPushSocket.close()
|
||
} catch (e) {
|
||
// 忽略错误
|
||
}
|
||
wsPushSocket = null
|
||
}
|
||
|
||
wsPushUrl.value = ''
|
||
console.log('[WebRTC] WebSocket 推流已停止')
|
||
}
|
||
|
||
/**
|
||
* 离开通话房间
|
||
*/
|
||
async function leaveCallRoom(): Promise<void> {
|
||
if (!currentRoomId) return
|
||
|
||
try {
|
||
await callApi.leaveCallRoom({
|
||
room_id: currentRoomId,
|
||
user_id: userId
|
||
})
|
||
} catch (error) {
|
||
console.error('[WebRTC] 离开房间失败:', error)
|
||
}
|
||
}
|
||
|
||
/**
|
||
* 初始化 FLV 播放器
|
||
* @param videoElement 视频元素
|
||
* @param streamIndex 流索引
|
||
*/
|
||
function initFlvPlayer(videoElement: HTMLVideoElement, streamIndex: number = 0): flvjs.Player | null {
|
||
console.log('[WebRTC] 🎬 初始化 FLV 播放器, streamIndex:', streamIndex, '当前流列表:', call.miniprogramStreams)
|
||
|
||
if (!flvjs.isSupported()) {
|
||
console.error('[WebRTC] ❌ 浏览器不支持 flv.js')
|
||
toastStore.error('您的浏览器不支持 FLV 播放')
|
||
return null
|
||
}
|
||
|
||
const stream = call.miniprogramStreams[streamIndex]
|
||
if (!stream || !stream.flvUrl) {
|
||
console.error('[WebRTC] ❌ 没有可用的 FLV 流, stream:', stream)
|
||
return null
|
||
}
|
||
|
||
// 如果已有播放器，先完整销毁（必须按顺序调用 pause, unload, detachMediaElement, destroy）
|
||
if (stream.player) {
|
||
console.log('[WebRTC] 🔄 销毁已有的 FLV 播放器')
|
||
try {
|
||
stream.player.pause()
|
||
stream.player.unload()
|
||
stream.player.detachMediaElement()
|
||
stream.player.destroy()
|
||
} catch (e) {
|
||
console.warn('[WebRTC] 销毁 FLV 播放器时出错:', e)
|
||
}
|
||
stream.player = undefined
|
||
}
|
||
|
||
console.log('[WebRTC] 🎬 创建 FLV 播放器:', stream.flvUrl)
|
||
|
||
const player = flvjs.createPlayer({
|
||
type: 'flv',
|
||
url: stream.flvUrl,
|
||
isLive: true,
|
||
hasAudio: true,
|
||
hasVideo: true,
|
||
}, {
|
||
enableWorker: false,
|
||
enableStashBuffer: false,
|
||
stashInitialSize: 128,
|
||
lazyLoad: false,
|
||
})
|
||
|
||
player.attachMediaElement(videoElement)
|
||
player.load()
|
||
player.play()
|
||
|
||
// 监听错误
|
||
player.on(flvjs.Events.ERROR, (errorType, errorDetail, errorInfo) => {
|
||
console.error('[WebRTC] FLV 播放错误:', errorType, errorDetail, errorInfo)
|
||
if (errorType === flvjs.ErrorTypes.NETWORK_ERROR) {
|
||
toastStore.error('网络错误，无法播放小程序视频流')
|
||
} else if (errorType === flvjs.ErrorTypes.MEDIA_ERROR) {
|
||
// 处理媒体错误，包括编解码器不支持
|
||
if (errorDetail === 'CodecUnsupported') {
|
||
console.error('[WebRTC] 音视频编解码器不支持，小程序端需要配置 audio-codec="aac"')
|
||
toastStore.error('视频格式不兼容，请让小程序端更新后重试')
|
||
} else {
|
||
toastStore.error('视频播放失败: ' + errorDetail)
|
||
}
|
||
}
|
||
})
|
||
|
||
// 保存播放器引用
|
||
stream.player = player
|
||
|
||
return player
|
||
}
|
||
|
||
/**
|
||
* 销毁所有 FLV 播放器
|
||
*/
|
||
function destroyFlvPlayers(): void {
|
||
for (const stream of call.miniprogramStreams) {
|
||
if (stream.player) {
|
||
try {
|
||
stream.player.pause()
|
||
stream.player.unload()
|
||
stream.player.detachMediaElement()
|
||
stream.player.destroy()
|
||
} catch (e) {
|
||
console.error('[WebRTC] 销毁 FLV 播放器失败:', e)
|
||
}
|
||
stream.player = undefined
|
||
}
|
||
}
|
||
}
|
||
|
||
/**
|
||
* 检查是否有小程序流需要播放
|
||
*/
|
||
function hasMiniprogramStreams(): boolean {
|
||
return call.miniprogramStreams.length > 0
|
||
}
|
||
|
||
async function processPendingCandidates() {
|
||
while (pendingCandidates.length > 0) {
|
||
const c = pendingCandidates.shift()
|
||
if (c && pc) await pc.addIceCandidate(c).catch(e => {})
|
||
}
|
||
}
|
||
|
||
function startCallTimer() {
|
||
stopCallTimer()
|
||
call.startTime = Date.now()
|
||
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')
|
||
const h = Math.floor(seconds / 3600)
|
||
return h > 0 ? `${h.toString().padStart(2, '0')}:${m}:${s}` : `${m}:${s}`
|
||
}
|
||
|
||
function toggleMute() {
|
||
call.muted = !call.muted
|
||
sendSyncState('mic-toggle', call.muted)
|
||
if (localStream.value) localStream.value.getAudioTracks().forEach(t => t.enabled = !call.muted)
|
||
}
|
||
|
||
function toggleCamera() {
|
||
call.camOff = !call.camOff
|
||
sendSyncState('cam-toggle', call.camOff)
|
||
if (localStream.value) localStream.value.getVideoTracks().forEach(t => t.enabled = !call.camOff)
|
||
}
|
||
|
||
return {
|
||
call,
|
||
localStream,
|
||
remoteStream,
|
||
startCall,
|
||
acceptCall,
|
||
endCall,
|
||
handleSignaling,
|
||
toggleMute,
|
||
toggleCamera,
|
||
formatDuration,
|
||
// FLV 播放相关（用于小程序流）
|
||
initFlvPlayer,
|
||
destroyFlvPlayers,
|
||
hasMiniprogramStreams,
|
||
}
|
||
}
|