HTML5 Video 與 WebRTC

HTML5 Video

1.有關canvas串流

參考此 https://github.com/EasonWang01/Node.js-stream-video/tree/master/Desktop/Node.js-stream-video-master

但其原理為使用canvas擷取影像，並使用websocket傳遞canvas資料，所以只有影像，這種做法client端用來顯示的 dataURL會持續改變，所以雖然是串流，但畫面會有閃爍問題

2.從HTML5錄製影片並下載

教學

https://developers.google.com/web/updates/2016/01/mediarecorder

Source code

https://github.com/webrtc/samples/blob/gh-pages/src/content/getusermedia/record/js/main.js

其原理為使用 navigator.mediaDevices.getUserMedia存取網頁攝影機後使用new MediaRecorder錄製

而MediaRecorder接到資料後要存入blob

const haveLoadedMetadata = stream => {
      const video = document.querySelector("#localVideo");
      video.srcObject = stream;
      video.play();
      return new Promise(resolve => video.onloadedmetadata = () => resolve(stream)); 
    };
    var constraints = { audio: true, video: { width: 400, height: 200 } };

    navigator.mediaDevices
      .getUserMedia(constraints)
      .then(mediaStream => haveLoadedMetadata(mediaStream))
      .then((mediaStream) => {
          var options = { mimeType: "video/webm; codecs=vp9" };
          const recorder = new MediaRecorder(mediaStream, options);
          recorder.ondataavailable = (e) => {
            console.log(e) //這裡記得要呼叫 recorder.stop() 才會有  ondataavailable
          }
          recorder.start();
      setTimeout(() => {
        recorder.stop()
      }, 2000);
     })
    .catch(function (err) {
      console.log(err.name + ": " + err.message);
    });
    
    
//或是可以用第三方模組 msr
import MediaStreamRecorder from "msr";
var multiStreamRecorder = new MediaStreamRecorder.MultiStreamRecorder(
   mediaStream // from getUserMedia
);
multiStreamRecorder.ondataavailable = function (blob) {
  // POST/PUT "Blob" using FormData/XHR2
  ws.send(blob.video);
};
multiStreamRecorder.start(3000);

之後再把blob轉格式

var superBuffer = new Blob(blob1, {type: 'video/webm'});

最後轉為可用在url的型態

window.URL.createObjectURL(superBuffer)

把他放到video的src即可

可參考mediaSource 模式：
https://github.com/webrtc/samples/blob/gh-pages/src/content/getusermedia/record/js/main.js

3. getUserMedia streaming with WebSocket

後來想到可以使用將影片擷取10秒一格並分開連續傳送給client達到串流的效果，但一樣因為最後要在前端將video.srcObject 改為blob，只要更改video src都會造成畫面閃爍

client

import React, { useEffect } from "react";
import MediaStreamRecorder from "msr";
import "./App.css";

const ws = new WebSocket("ws://localhost:3003");

function App() {
  useEffect(() => {
    const video = document.querySelector("#clientVideo");
    video.onloadedmetadata = function (e) {
      video.play();
    };
    ws.onopen = () => {
      console.log("open");
    };
    ws.onmessage = (msg) => {
      if (msg.data instanceof Blob) {
        const video = document.querySelector("#clientVideo");
        video.src = window.URL.createObjectURL(msg.data);
      }
    };
  });
  const send = () => {
    var constraints = { audio: true, video: { width: 400, height: 200 } };

    navigator.mediaDevices
      .getUserMedia(constraints)
      .then((mediaStream) => {
        const video = document.querySelector("#localVideo");
        video.srcObject = mediaStream;
        video.onloadedmetadata = function (e) {
          video.play();
        };

        var multiStreamRecorder = new MediaStreamRecorder.MultiStreamRecorder(
          mediaStream
        );
        multiStreamRecorder.ondataavailable = function (blob) {
          // POST/PUT "Blob" using FormData/XHR2
          ws.send(blob.video);
        };
        multiStreamRecorder.start(7000);
      })
      .catch(function (err) {
        console.log(err.name + ": " + err.message);
      });
  };
  return (
    <div className="App">
      <video id="localVideo"></video>
      <button onClick={() => send()}>send stream</button>
      <div style={{ width: 400, border: "1px solid black", margin: "0 auto" }}>
        <div>client</div>
        <video id="clientVideo" autoPlay></video>
      </div>
    </div>
  );
}

export default App;

server

const WebSocket = require('ws');

const wss = new WebSocket.Server({ port: 3003 });

wss.on('connection', function connection(ws) {
  ws.on('message', (message) => {
    try {
      console.log(message);
      console.log(Buffer.isBuffer(message))
      ws.send(message);
    } catch(err) {
      console.log(err)
    }
  });
  ws.send('something');
});

但因為video src 每次更新後畫面會閃爍

4. 之後有了mediaSource API

https://developer.mozilla.org/zh-TW/docs/Web/API/MediaSource

MediaSource 串流範例

video.src = URL.createObjectURL(mediaSource); 後mediaSource.addEventListener("sourceopen") 才會觸發

https://stackoverflow.com/a/52379544

Async & 加上 websocket 版本:

socket.io 版本參照bitbucket

import React, { useEffect } from "react";
import "./App.css";
const ws = new WebSocket("ws://localhost:3003");

function App() {
  const start = () => {
    ws.onopen = () => {
      console.log("open");
    };
    ws.onmessage = async (msg) => {
      console.log(msg.data)
      const remoteBuf = await msg.data.arrayBuffer();
      // TODO receive remoteBuf
    };
    (async function() {
      const devices = await navigator.mediaDevices.enumerateDevices();
      console.table(devices);

      const stream = await navigator.mediaDevices.getUserMedia({
        video: true,
        audio: true,
      });

      const rec = new MediaRecorder(stream, {
        mimeType: 'video/webm; codecs="opus,vp8"',
      });

      const ms = new MediaSource();

      const video = document.querySelector("#video");

      //video.srcObject = ms;
      video.src = URL.createObjectURL(ms);
      video.volume = 0;
      video.controls = true;
      video.autoplay = true;

      await new Promise((resolve, reject) => {
        ms.addEventListener("sourceopen", () => resolve(), { once: true });
      });

      const sb = ms.addSourceBuffer(rec.mimeType);

      rec.ondataavailable = ({ data }) => {
        (async function () {
          try {
            if (data.size === 0) {
              console.warn("empty recorder data");
              throw new Error("empty recorder data");
            }

            const buf = await data.arrayBuffer();
            sb.appendBuffer(buf);
            ws.send(buf);

            if (video.buffered.length > 1) {
              console.warn("MSE buffered has a gap!");
              throw new Error("MSE buffered has a gap!");
            }
          } catch (err) {
            console.error(err);
          }
        })();
      };

      rec.start(1000);
      console.info("start");
    })();
  };

  return (
    <div id="container">
      <video id="video"></video>
      <button onClick={() => start()}>start</button>
    </div>
  );
}

export default App;

WebRTC串流

WebRTC 可用範例：

WebRTC samples

Debug

chrome://webrtc-internals

名詞:

https://blog.mozilla.com.tw/posts/3261/webrtc-相關縮寫名詞簡介blog.mozilla.com.tw

WebRTC 流程：

https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API/Signaling_and_video_calling

範例:

https://shanetully.com/2014/09/a-dead-simple-webrtc-example/

https://github.com/shanet/WebRTC-Example

以上兩個為很好且簡單的範例，上面是文章下面是程式碼。

Client 過程

1. 初始化連線: 
```
new RTCPeerConnection({
  iceServers: [
    { urls: "stun:stun.stunprotocol.org:3478" },
    { urls: "stun:stun.l.google.com:19302" },
  ]
})
```
並加入本地影像 
```
peerConnection.addStream(localStream);
```

2. 發起人利用 createOffer 產生 description 後設定 setLocalDescription 後傳送 SDP 給其他人: 
```
peerConnection.createOffer().then(createdDescription);
```

```
function createdDescription(description) {
  peerConnection
    .setLocalDescription(description)
    .then(function () {
      serverConnection.send(
        JSON.stringify({ sdp: peerConnection.localDescription, uuid: uuid })
      );
    })
}
```

3. 其他人收到對方的 sdp 後呼叫 setRemoteDescription，
之後用 createAnswer 產生 description 後也用 createdDescription 傳給發起人
```
  if (signal.sdp) {
    peerConnection
      .setRemoteDescription(new RTCSessionDescription(signal.sdp))
      .then(function () {
        if (signal.sdp.type === "offer") {
          peerConnection
            .createAnswer()
            .then(createdDescription)
        }
      })
  }
```  

4. 在本地蒐集到ice後傳送給另一個peer: onicecandidate 並且傳送 serverConnection.send
(ICE candidate可能接收到多個)
https://developer.mozilla.org/en-US/docs/Web/API/RTCIceCandidate
```
peerConnection.onicecandidate = gotIceCandidate;
serverConnection.send(JSON.stringify({ ice: event.candidate, uuid: uuid }));
```

4. 另一個peer接到ice後把ice加入： peerConnection.addIceCandidate
```
if (signal.ice) {
    peerConnection
      .addIceCandidate(new RTCIceCandidate(signal.ice))
}
```

5. 接收到視訊：peerConnection.ontrack = gotRemoteStream

6. 顯示遠端視訊： 
```
function gotRemoteStream(event) {
  remoteVideo.srcObject = event.streams[0];
}
```

Server

單純廣播所有接收到的訊息給連線的client

const wss = new WebSocketServer({server: httpsServer});

wss.on('connection', function(ws) {
  ws.on('message', function(message) {
    // Broadcast any received message to all clients
    console.log('received: %s', message);
    wss.broadcast(message);
  });
});

wss.broadcast = function(data) {
  this.clients.forEach(function(client) {
    if(client.readyState === WebSocket.OPEN) {
      client.send(data);
    }
  });
};

video.src vs srcObject

Older versions of the Media Source specification required using createObjectURL() to create an object URL then setting src to that URL. Now you can just set srcObject to the MediaStream directly.
https://developer.mozilla.org/en-US/docs/Web/API/HTMLMediaElement/srcObject

可能錯誤

Failed to execute 'appendBuffer' on 'SourceBuffer': This SourceBuffer has been removed from the parent media source

解法：將recorder.start(2000); 錄影間隔毫秒縮短即可

Video 屬性

video.autoplay = true; 等同於

    video.onloadedmetadata = function (e) {
      video.play();
    };

WebRTC 錄製

GitHub - muaz-khan/RecordRTC: RecordRTC is WebRTC JavaScript library for audio/video as well as screen activity recording. It supports Chrome, Firefox, Opera, Android, and Microsoft Edge. Platforms: Linux, Mac and Windows.GitHub

https://stackoverflow.com/questions/16571044/how-to-record-webcam-and-audio-using-webrtc-and-a-server-based-peer-connection

React Native WebRTC

https://github.com/react-native-webrtc/react-native-webrtc

0.6 版本以後需要配置如下，不然 Android 啟動後會 crash

https://github.com/react-native-webrtc/react-native-webrtc/issues/885#issuecomment-723116643

PreviousHTML5 audio NextWebRTC 進階

Last updated 4 years ago