349 lines
16 KiB
HTML
349 lines
16 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 chrome://webrtc-internals/</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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// 读取本地音乐列表(本地缓存)
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const store = new IndexedDB('musicDatabase', 1, 'musicObjectStore')
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await store.open()
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const list = await store.getAll()
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console.log('本地音乐列表:', list)
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// 初始化音乐列表(加入本地缓存)
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const musicList = new MusicList({ list })
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musicList.on('remove', item => {
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console.log('移除音乐', item)
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store.delete(item.id)
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})
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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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musicList.on('load', async item => {
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await new Promise((resolve) => {
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console.log('加载音乐', item)
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// 建立一个专用信道, 用于接收音乐数据(接收方已经准备好摘要信息)
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var buffer = new ArrayBuffer(0)
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var count = 0
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clientList.setChannel(`music-data-${item.id}`, {
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onmessage: async (event, client) => {
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buffer = appendBuffer(buffer, event.data)
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console.log('收到音乐数据 chunk', count, buffer.byteLength)
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count++
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if (buffer.byteLength >= item.size) {
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console.log('音乐数据接收完毕')
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item.arrayBuffer = buffer
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//// 尝试直接播放音乐
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//const audioContext = new AudioContext()
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//audioContext.decodeAudioData(item.arrayBuffer, async audioBuffer => {
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// const mediaStreamDestination = audioContext.createMediaStreamDestination()
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// mediaStreamDestination.stream.getAudioTracks().forEach(function (track) {
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// client.webrtc.addTrack(track, mediaStreamDestination.stream)
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// })
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// const 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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//})
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resolve()
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}
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}
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})
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// 要求对方从指定信道发送音乐数据
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clientList.send('base', JSON.stringify({ type: 'get_music_data', id: item.id, channel: `music-data-${item.id}` }))
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//store.put(item) // 只有在like时才保存到本地
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})
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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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// 缓冲分片发送
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const CHUNK_SIZE = 1024 * 128 // 每个块的大小为128KB
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const THRESHOLD = 1024 * 1024 // 缓冲区的阈值为1MB
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const DELAY = 100 // 延迟500ms
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// 将两个ArrayBuffer合并成一个
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function appendBuffer(buffer1, buffer2) {
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const tmp = new Uint8Array(buffer1.byteLength + buffer2.byteLength)
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tmp.set(new Uint8Array(buffer1), 0)
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tmp.set(new Uint8Array(buffer2), buffer1.byteLength)
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return tmp.buffer
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}
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// 只有一个基本信道, 用于交换和调度信息
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clientList.setChannel('base', {
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onopen: async event => {
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// 要求对方发送音乐列表
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clientList.send('base', JSON.stringify({ type: 'get_music_list' }))
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},
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onmessage: async (event, client) => {
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const { type, id, channel, list } = JSON.parse(event.data)
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console.log('收到 base 基本信道数据:', type, id, channel)
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if (type === 'get_music_list') {
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console.log('发送音乐列表:', musicList.list)
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clientList.send('base', JSON.stringify({
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type: 'set_music_list',
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list: musicList.list.map(({ id, name, size, type }) => ({ id, name, size, type }))
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}))
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return
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}
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if (type === 'set_music_list') {
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console.log('接收音乐列表:', event)
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// 将音乐列表添加到本地
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const ids = musicList.list.map(item => item.id)
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list.filter(item => !ids.includes(item.id)).forEach(item => {
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musicList.add(item)
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})
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return
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}
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if (type === 'get_music_data') {
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// 建立一个信道, 用于传输音乐数据(接收方已经准备好摘要信息)
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console.log('建立一个信道, 用于传输音乐数据', musicList.list)
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musicList.list.filter(item => item.id === id).forEach(item => {
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const ch = client.webrtc.createDataChannel(channel, { reliable: true })
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ch.onopen = async event => {
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console.log(`打开 ${channel} 信道, 传输音乐数据`)
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// 将音乐数据分成多个小块，并逐个发送
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async function sendChunk(dataChannel, data, index = 0, buffer = new ArrayBuffer(0)) {
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while (index < data.byteLength) {
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if (dataChannel.bufferedAmount <= THRESHOLD) {
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const chunk = data.slice(index, index + CHUNK_SIZE)
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dataChannel.send(chunk)
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index += CHUNK_SIZE
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buffer = appendBuffer(buffer, chunk)
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}
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await new Promise((resolve) => setTimeout(resolve, DELAY))
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}
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return buffer
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}
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await sendChunk(ch, item.arrayBuffer)
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console.log(`发送 ${channel} 信道数据结束`)
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}
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})
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return
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}
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console.log('未知类型:', type)
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}
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})
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// like对方的条目时亮起(双方高亮)(本地缓存)(可由对比缓存实现)
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// ban对方的条目时灰掉(也禁止对方播放)(并保持ban表)(由插件实现)
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// 只需要在注册时拉取列表, 播放时才需要拉取音乐数据
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</script>
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<!--script type="module">
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// webRTC 传递音乐(分别传输文件和操作事件能更流畅)
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const music = async function () {
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const clients = [] // 客户端列表
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// 对端设备
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const ul = document.createElement('ul')
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document.body.appendChild(ul)
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const protocol = window.location.protocol === 'https:' ? 'wss' : 'ws'
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const host = window.location.host
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const ws = new WebSocket(`${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> |