245 lines
10 KiB
Markdown
245 lines
10 KiB
Markdown
# webRTC
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webrtc 实现的 p2p 信道
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- [x] P2P通信
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- [ ] 分离出主要功能, 作为库或桁架使用
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- [x] 音乐播放
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- [x] 请求到单个目标防止接收到重复分片数据
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- [x] 主机记录各自曲目列表以供查询
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- [x] 播放时高亮显示
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- [x] 合并操作按钮
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- [x] 响应列表时不再广播
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- [x] 对方退出时清除其列表
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- [x] 稳定通信
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- [x] 分片请求时立即播放
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- [ ] 上锁防止连续重复加载同一个造成分片混乱
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- [x] 使用单独的状态标识音乐是否缓存
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- [x] 取消本地存储时不直接移除列表
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- [x] 分片下载过程与播放控制分离
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- [x] 分片播放时支持wav
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- [ ] 分片播放时支持flac
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- [ ] 取消本地存储时检查是否移除(其它成员可能有同一曲)
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- [ ] 成员列表刷新时播放被重置BUG
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- [ ] 集群分发
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- [ ] 下载加速
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- [ ] 即时通讯
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- 能获取所有在线设备列表
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- 随机连接至四个设备, 且按效率扩展收缩
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- 将数据拆解同时向多台设备分发, 对端接收后再次分发
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- 需要确保全部设备获得全部数据, 每台设备至少一半不重复
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- 五色
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- 单向链
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- 固定填位(矩阵)
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```txt
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[a1, b1, c1, d1, e1]
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[a2, b2, c2, d2, e2]
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[a3, b3, c3, d3, e3]
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```
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备用代码片段
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```html
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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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``` |