235 lines
10 KiB
HTML
235 lines
10 KiB
HTML
<!DOCTYPE html>
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<html lang="en">
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<head>
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<meta charset="UTF-8">
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<title>webRTC</title>
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</head>
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<body>
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<div>
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<h1>webRTC</h1>
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<p>选择音乐使频道内所有设备同步播放</p>
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</div>
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<script type="module">
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import IndexedDB from './database.js'
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import MusicList from './music.js'
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import ClientList from './client.js'
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// 初始化音乐列表
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const musicList = new MusicList()
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musicList.on('play', item => {
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console.log('播放音乐', item)
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})
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// 初始化客户端列表
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const clientList = new ClientList()
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// 先拉取所有对方的音乐列表, 对比去重, 拉取音乐数据
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// 主动发送自己的音乐列表, 对比去重, 发送音乐数据
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console.log('musicList.list:', musicList)
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clientList.send('music-list', '23333')
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// webRTC 传递音乐(分别传输文件和操作事件能更流畅)
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const music = async function () {
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const clients = [] // 客户端列表
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// 对端设备
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const ul = document.createElement('ul')
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document.body.appendChild(ul)
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const protocol = window.location.protocol === 'https:' ? 'wss' : 'ws'
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const host = window.location.host
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const ws = new WebSocket(`${protocol}://${host}/webrtc/music`)
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const pc = new RTCPeerConnection()
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var audioSource = null
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// 监听音乐列表播放事件
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musicList.on('play', async item => {
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audioSource?.stop() // 先停止可能在播放的音乐
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console.log('播放音乐', item.arrayBuffer)
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// 复制一份 item.arrayBuffer
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const arrayBuffer = item.arrayBuffer.slice(0)
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// 传输音乐文件向远程端
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const audioContext = new AudioContext()
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audioContext.decodeAudioData(arrayBuffer, async audioBuffer => {
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// 将音乐流添加到 RTCPeerConnection
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const mediaStreamDestination = audioContext.createMediaStreamDestination()
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mediaStreamDestination.stream.getAudioTracks().forEach(function (track) {
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pc.addTrack(track, mediaStreamDestination.stream)
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})
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// 播放音乐(远程)
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audioSource = audioContext.createBufferSource()
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audioSource.buffer = audioBuffer
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audioSource.connect(mediaStreamDestination)
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audioSource.start()
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// 创建SDP offer并将其设置为本地描述, 发送给指定的远程端
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const id = clients[0].id
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await pc.setLocalDescription(await pc.createOffer()) // 设置本地描述为 offer
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ws.send(JSON.stringify({ id, offer: pc.localDescription })) // 发送给远程终端 offer
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})
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})
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// 监听音乐列表停止事件
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musicList.on('stop', async () => {
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audioSource?.stop()
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audioSource = null
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})
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// 监听 ICE 候选事件
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pc.onicecandidate = event => {
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if (event.candidate) {
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const id = clients[0].id
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ws.send(JSON.stringify({ id, candidate: event.candidate })) // 发送 ICE 候选到远程终端
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}
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}
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// 监听远程流事件
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pc.ontrack = function (event) {
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console.log('pc ontrack:', event)
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const audio = document.createElement('audio')
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audio.srcObject = event.streams[0]
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audio.play()
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}
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ws.onmessage = async (event) => {
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const data = JSON.parse(event.data)
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if (data.type === 'push') {
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console.log('收到 type:push 将设备增加', data.id)
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clients.push({ id: data.id, channel: data.channel })
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const li = document.createElement('li')
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li.innerText = `id:${data.id} channel:${data.channel}`
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li.id = data.id
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li.onclick = async () => {
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console.log('点击设备', data.id)
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// 清理所有选中状态
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clients.forEach(client => {
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const li = document.getElementById(client.id)
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if (data.id === client.id) {
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li.style.backgroundColor = 'red'
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console.log('设置选中状态', data.id)
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return
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}
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li.style.backgroundColor = 'transparent'
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console.log('清理选中状态', client.id)
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})
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}
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ul.appendChild(li)
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return
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}
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if (data.type === 'pull') {
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console.log('收到 type:pull 将设备删除', data.id)
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const index = clients.findIndex(client => client.id === data.id)
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if (index !== -1) {
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clients.splice(index, 1)
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const li = document.getElementById(data.id)
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li.remove()
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}
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return
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}
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if (data.type === 'error') {
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console.log('收到 type:error 没什么可操作的', data.id)
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return
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}
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if (data.offer) {
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const id = clients[0].id
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console.log('收到 offer 并将其设置为远程描述', data.offer)
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await pc.setRemoteDescription(new RTCSessionDescription(data.offer)) // 设置远程描述为 offer
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await pc.setLocalDescription(await pc.createAnswer()) // 设置本地描述为 answer
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ws.send(JSON.stringify({ id, answer: pc.localDescription })) // 发送给远程终端 answer
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return
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}
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if (data.answer) {
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console.log('收到 answer 并将其设置为远程描述', data.answer)
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await pc.setRemoteDescription(new RTCSessionDescription(data.answer))
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return
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}
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if (data.candidate) {
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console.log('收到 candidate 并将其添加到远程端', data.candidate)
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await pc.addIceCandidate(new RTCIceCandidate(data.candidate))
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return
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}
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}
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}
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//music()
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</script>
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<!--script type="module">
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// 创建 RTCPeerConnection
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const pc = new RTCPeerConnection()
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// webSocket 连接服务器
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const protocol = window.location.protocol === 'https:' ? 'wss' : 'ws'
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const host = window.location.host
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const ws = new WebSocket(`${protocol}://${host}/webrtc/default`)
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ws.onopen = function () {
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console.log('video ws open')
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}
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ws.onmessage = function (event) {
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const data = JSON.parse(event.data)
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console.log('ws message:', data)
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if (data.offer) {
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console.log('收到 offer 并将其设置为远程描述')
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pc.setRemoteDescription(new RTCSessionDescription(data.offer))
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// 创建SDP answer并将其设置为本地描述, 发送给远程端
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pc.createAnswer().then(function (answer) {
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pc.setLocalDescription(answer);
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ws.send(JSON.stringify({ answer }))
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})
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return
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}
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if (data.answer) {
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console.log('收到 answer 并将其设置为远程描述')
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pc.setRemoteDescription(new RTCSessionDescription(data.answer))
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return
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}
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if (data.candidate) {
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console.log('收到 candidate 并将其添加到远程端')
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pc.addIceCandidate(new RTCIceCandidate(data.candidate))
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}
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}
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ws.onclose = function () {
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console.log('ws close')
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}
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setTimeout(() => {
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// 获取本地视频流
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navigator.mediaDevices.getUserMedia({ audio: false, video: true }).then(stream => {
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// 创建本地视频元素
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const localVideo = document.createElement('video');
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localVideo.srcObject = stream;
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localVideo.autoplay = true;
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localVideo.muted = true;
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document.body.appendChild(localVideo);
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// 添加本地视频流到 RTCPeerConnection
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stream.getTracks().forEach(function (track) {
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pc.addTrack(track, stream);
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});
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// 监听 ICE candidate 事件
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pc.onicecandidate = function (event) {
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if (event.candidate) {
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// 发送 ICE candidate 到远程端
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ws.send(JSON.stringify({ candidate: event.candidate }))
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}
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};
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// 监听远程视频流事件
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pc.ontrack = function (event) {
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// 创建远程视频元素
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var remoteVideo = document.createElement('video');
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remoteVideo.srcObject = event.streams[0];
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remoteVideo.autoplay = true;
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document.body.appendChild(remoteVideo);
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}
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// 创建SDP offer并将其设置为本地描述, 发送给远程端
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pc.createOffer().then(function (offer) {
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pc.setLocalDescription(offer);
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ws.send(JSON.stringify({ offer }))
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})
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}).catch(error => {
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console.log(error)
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})
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}, 1000)
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</script-->
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</body>
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</html> |