Javascript 调用recordAudio.stopRecording（）时套接字连接断开

我正在尝试从浏览器麦克风录制音频，并将其发送回节点服务器，以便使用谷歌云STT服务将其转换为文本

一切正常，但问题是，当我第一次录制音频时，它工作正常录制的音频文件被发送到服务器进行处理，但当我第二次尝试录制时，套接字连接断开，浏览器重新连接到服务器以建立新的套接字连接

以下是客户端代码：

const socket = io();
const $recordBtn = document.getElementById("record-audio-btn");
const $stopRecrdingBtn = document.getElementById("stop-recording-btn");
// $stopRecrdingBtn.disabled = true;

socket.on("message", ({ msg, _id }) => {
  console.log("msg & _id >> ", msg, _id);
});

// for record audio with recordRTC
let recordAudio;

$recordBtn.onclick = () => {
  $recordBtn.disabled = true;
  // $stopRecrdingBtn.disabled = false;

  navigator.mediaDevices
    .getUserMedia({ audio: { echoCancellation: true } })
    .then((stream) => {      
      recordAudio = RecordRTC(stream, {
        type: "audio",
        mimeType: "audio/webm",
        sampleRate: 44100,
        desiredSampleRate: 16000,

        recorderType: StereoAudioRecorder,
        numberOfAudioChannels: 1, // using monoAudio Channel as backend req it

        // get intervals based blobs
        // value in milliseconds
        timeSlice: 1000,
      });
      recordAudio.startRecording();
      console.log("********** Audio Rec started **************");
    })
    .catch((err) => console.log("ERROR: >", err));
};

$stopRecrdingBtn.onclick = () => {
  $recordBtn.disabled = false;

  console.log("************ stopped ***********");
  console.log("recordAudio >> ", recordAudio);
  recordAudio.stopRecording(() => {
    // after stopping the audio, get the audio data
    recordAudio.getDataURL(function (audioDataURL) {
      var files = {
        audio: {
          type: recordAudio.getBlob().type || "audio/wav",
          dataURL: audioDataURL,
        },
      };
      // send the audio file to the server
      console.log("files >> ", files);

      socket.emit("message-transcribe", files);
      console.log("files Passed to backend for processing!");
    });
  });
};

• 现在在服务器端，我只是将音频文件写入本地机器

服务器代码

const path = require("path");
const express = require("express");
const http = require("http");
const fs = require("fs");
const socketio = require("socket.io");
// const socketStream = require("socket.io-stream");
const app = express();
const PORT = process.env.PORT || 3000;

const publicDirPath = path.join(__dirname, "../public");
app.use(express.static(publicDirPath));

const server = http.createServer(app);
const io = socketio(server); // passing raw http server to socket.io


app.get("/", (req, res) => {
  res.render("index");
});

io.on("connection", (client) => {
  console.log("New websocket connection!...");

  client.emit("message", {
    msg: "New WebSocket connection...",
    _id: client.id,
  });


  client.on("message-transcribe", (data) => {

    // we get the dataURL which was sent from the client
    const dataURL = data.audio.dataURL.split(",").pop();

    // we will convert it to a Buffer
    let fileBuffer = Buffer.from(dataURL, "base64");

    console.log("fileBuffer >> ", fileBuffer);
    
    // write audio file to local machine using fs.writeFileSync()
    fs.writeFileSync("testAudio.wav", fileBuffer);
  });

  client.on("disconnect", (reason) => {
    console.log(`\nDisconnected....\nReason=${reason}\n`);
  });
});


server.listen(PORT, () => {
  console.log(`Server Up & Listening on PORT: ${PORT}`);
});

编辑：我试图调试这个问题，我想，可能是因为传输错误，我看到如果我录制的音频文件高达700kb-800kb，一切正常，如果文件大小超过该值，我会出现传输错误和套接字断开

尝试发送大小超过700-800kb的base-64编码音频文件时发出

New websocket connection!...

// after initiating file tranfer from browser to server
Disconnected....
Reason=transport error // generated from client.on("disconnect", callback(reason))

New websocket connection!...

---

感谢您的帮助

因此，在深入研究socket.io文档之后，我发现如果

maxHttpBufferSize

超过

1e6

（默认值）以避免Dos，套接字连接就会断开

---

我希望它能帮助任何有这个问题的人。

所以在深入研究socket.io文档之后，我发现如果

maxHttpBufferSize

超过

1e6

（默认值），套接字连接会断开，以避免Dos

我希望它能帮助任何有这个问题的人