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音视频通话功能完成电话铃声

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liqi
2025-12-03 18:02:26 +08:00
parent 3a98cee2b7
commit 6ddea9d988
1 changed files with 99 additions and 26 deletions
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125
src/composables/useWebRTC.ts
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@@ -15,15 +15,18 @@ export interface CallState {
statusText: string
id: string | null
muted: boolean
remoteMuted: boolean // 新增：对方是否静音
camOff: boolean
remoteCamOff: boolean // 新增：对方是否关闭摄像头
remoteCamOff: boolean
duration: number
startTime: number | null
}
// 铃声资源 (使用在线资源作为示例)
const RINGTONE_INCOMING = 'https://assets.mixkit.co/active_storage/sfx/2860/2860-64-bit.mp3' // 电话铃声
const RINGTONE_DIALING = 'https://assets.mixkit.co/active_storage/sfx/2864/2864-64-bit.mp3' // 嘟嘟声
// 使用 Base64 数据 URI 避免 403 和跨域问题
// 简短的电话铃声 (Incoming)
const RINGTONE_INCOMING_BASE64 = 'data:audio/mp3;base64,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'
// 简短的嘟嘟拨号音 (Dialing)
const RINGTONE_DIALING_BASE64 = 'data:audio/wav;base64,UklGRl9vT1BXQVZFZm10IBAAAAABAAEAQB8AAEAfAAABAAgAZGF0YU'
export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string) => void) {
const toastStore = useToastStore()
@@ -37,6 +40,7 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
statusText: '',
id: null,
muted: false,
remoteMuted: false, // 初始化
camOff: false,
remoteCamOff: false,
duration: 0,
@@ -51,39 +55,102 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
const pendingCandidates: RTCIceCandidate[] = []
let currentReceiverUserId = ''
// 音频对象
const audioIncoming = new Audio(RINGTONE_INCOMING)
const audioDialing = new Audio(RINGTONE_DIALING)
// 音频对象初始化 (使用 Base64)
const audioIncoming = new Audio(RINGTONE_INCOMING_BASE64)
const audioDialing = new Audio(RINGTONE_DIALING_BASE64) // 使用 Base64 或简单的音频
// 由于 Base64 字符串较短，需要循环播放
audioIncoming.loop = true
audioDialing.loop = true
// 简单的嘟嘟声生成器 (如果 Base64 不工作作为备选，但通常 Base64 更可靠)
// 这里直接使用 Audio Context 生成嘟嘟声可能更专业，但为了兼容性先用 Audio 标签
// 注意：上面的 Base64 只是示例片段，为了确保有声音，建议用真实的短音频 Base64
// --- 铃声控制 ---
function playRingtone(type: 'incoming' | 'dialing') {
stopRingtone() // 先停止当前的
const audio = type === 'incoming' ? audioIncoming : audioDialing
// 如果是拨号音，我们可以用 Web Audio API 生成一个标准的嘟嘟声，这样不需要加载文件
if (type === 'dialing') {
playOscillatorTone()
return
}
const audio = audioIncoming
audio.currentTime = 0
audio.play().catch(e => console.warn('Autoplay prevented:', e))
}
// 使用 Web Audio API 生成拨号音 (避免资源 403 问题)
let oscCtx: AudioContext | null = null
let oscillator: OscillatorNode | null = null
let gainNode: GainNode | null = null
function playOscillatorTone() {
try {
const AudioContext = window.AudioContext || (window as any).webkitAudioContext
if (!AudioContext) return
oscCtx = new AudioContext()
oscillator = oscCtx.createOscillator()
gainNode = oscCtx.createGain()
oscillator.type = 'sine'
oscillator.frequency.setValueAtTime(440, oscCtx.currentTime) // 440Hz 标准音
// 模拟嘟-嘟-嘟的效果
gainNode.gain.setValueAtTime(0.1, oscCtx.currentTime)
oscillator.connect(gainNode)
gainNode.connect(oscCtx.destination)
oscillator.start()
// 简单的循环效果
const pulse = () => {
if(!gainNode || !oscCtx) return
gainNode.gain.setValueAtTime(0.1, oscCtx.currentTime)
setTimeout(() => {
if(gainNode && oscCtx) gainNode.gain.setValueAtTime(0, oscCtx.currentTime)
}, 800)
setTimeout(pulse, 2000)
}
pulse()
} catch(e) { console.error(e) }
}
function stopRingtone() {
audioIncoming.pause()
audioDialing.pause()
audioIncoming.currentTime = 0
audioDialing.currentTime = 0
// 停止 Web Audio API 的声音
if (oscillator) {
try { oscillator.stop(); oscillator.disconnect() } catch(e){}
oscillator = null
}
if (gainNode) {
try { gainNode.disconnect() } catch(e){}
gainNode = null
}
if (oscCtx) {
try { oscCtx.close() } catch(e){}
oscCtx = null
}
}
// --- 系统通知 ---
function sendSystemNotification(title: string, body: string) {
if (!('Notification' in window)) return
if (Notification.permission === 'granted') {
new Notification(title, { body, icon: '/favicon.ico' })
} else if (Notification.permission !== 'denied') {
// 再次尝试请求权限 (如果是用户触发的操作中调用)
if (Notification.permission !== 'granted' && Notification.permission !== 'denied') {
Notification.requestPermission().then(permission => {
if (permission === 'granted') {
new Notification(title, { body, icon: '/favicon.ico' })
}
})
} else if (Notification.permission === 'granted') {
new Notification(title, { body, icon: '/favicon.ico' })
}
}
@@ -179,17 +246,17 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
messageApi.sendMessage(payload).catch(console.error)
}
// --- 摄像头开关同步信令 ---
function sendSyncState(type: 'cam-toggle', value: boolean) {
// --- 状态同步信令 (摄像头/麦克风) ---
function sendSyncState(type: 'cam-toggle' | 'mic-toggle', value: boolean) {
if (call.status !== 'connected') return
const payload = {
sender_client_id: wsManager.getClientId() || '',
receiver_user_id: currentReceiverUserId,
room_id: '',
message_type: 6, // 借用信令通道
message_type: 6,
content: JSON.stringify({ action: type, value }),
call_id: call.id,
call_status: 'sync_state' as any, // 自定义状态
call_status: 'sync_state' as any,
extra: JSON.stringify({ type: call.type })
}
messageApi.sendMessage(payload).catch(console.error)
@@ -231,12 +298,14 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
call.duration = 0
call.camOff = false
call.remoteCamOff = false
call.remoteMuted = false // 重置
remoteStream.value = null
try {
// 立即获取本地媒体流，确保自己能看到自己
await initMedia(type === 'video')
await createPC()
playRingtone('dialing') // 播放拨号音
playRingtone('dialing')
sendSignal('invite', undefined, receiverUserId)
} catch (error: any) {
toastStore.error(error.message || '无法启动通话')
@@ -246,12 +315,13 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
async function acceptCall(senderUserId?: string) {
isCaller.value = false
stopRingtone() // 停止来电铃声
stopRingtone()
if (senderUserId) currentReceiverUserId = senderUserId
call.status = 'connected'
call.statusText = '正在连接...'
call.camOff = false
call.remoteCamOff = false
call.remoteMuted = false
try {
await initMedia(call.type === 'video')
@@ -291,10 +361,12 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
if (extra.type) call.type = extra.type
// 处理自定义状态同步信令
// 处理状态同步
if (signal === 'sync_state') {
if (content.action === 'cam-toggle') {
call.remoteCamOff = content.value
} else if (content.action === 'mic-toggle') {
call.remoteMuted = content.value
}
return
}
@@ -308,14 +380,15 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
call.status = 'incoming'
call.statusText = `邀请你进行${call.type === 'video' ? '视频' : '语音'}通话`
call.remoteCamOff = false
call.remoteMuted = false
playRingtone('incoming') // 播放来电铃声
sendSystemNotification('新来电', `收到来自 ${message.sender_user_id} 的${call.type === 'video' ? '视频' : '语音'}通话邀请`) // 触发系统通知
playRingtone('incoming')
sendSystemNotification('新来电', `收到来自 ${message.sender_user_id} 的${call.type === 'video' ? '视频' : '语音'}通话邀请`)
if (onIncomingCall) onIncomingCall(message.sender_user_id)
} else if (signal === 'accepted') {
stopRingtone() // 对方接听，停止拨号音
stopRingtone()
call.status = 'connected'
call.statusText = '通话中'
startCallTimer()
@@ -324,7 +397,7 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
sendSignal('offer', offer)
} else if (signal === 'offer') {
stopRingtone() // 建立连接，停止可能存在的铃声
stopRingtone()
if (!pc) {
await initMedia(call.type === 'video')
await createPC()
@@ -365,13 +438,13 @@ export function useWebRTC(userId: string, onIncomingCall?: (senderUserId: string
function toggleMute() {
call.muted = !call.muted
sendSyncState('mic-toggle', call.muted) // 同步
if (localStream.value) localStream.value.getAudioTracks().forEach(t => t.enabled = !call.muted)
}
function toggleCamera() {
call.camOff = !call.camOff
// 同步状态给对方
sendSyncState('cam-toggle', call.camOff)
sendSyncState('cam-toggle', call.camOff) // 同步
if (localStream.value) localStream.value.getVideoTracks().forEach(t => t.enabled = !call.camOff)
}