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• Workflow and data format for sending MediaRecorder output to express server

  30 avril 2021, par Max

  I've been trying to figure this out for a while but got lost between different ways of sending files and different data formats.

  


  I am recording the stream of a canvas animation with MediaRecorder. As far as I understand this returns a blob with the video in binary format. Now I want to send this data to my express server and convert it to an h264 encoded mp4 file. My first impulse was to use ffmpeg on the server. Unfortunately I'm struggling with the details of the implementation. I am unsure on how to best transmit the data and in what format and how to feed it to ffmpeg.

  


  This is what I have on the client side :

  


  // Get stream from element
stream = element.captureStream(30)

// Create media recorder with stream
const recorder = new MediaRecorder(stream)

// Save to file
recorder.ondataavailable = ({ data }) => {
                
    const formData = new FormData()
    formData.append("file", data)

    const options = {
        method: "POST",
        body: formData,
    }

    fetch("http://localhost:3001/api/blob_to_mp4", options).then(
        (res) => {
            console.log(res)
        }
    )
}


  


  And this is what I have on the server side :

  


  "use strict";

const express = require("express");
const cors = require("cors");
const ffmpeg = require("fluent-ffmpeg");
const fs = require("fs");

const port = process.env.PORT || 3001;
const app = express();
var command = ffmpeg();

app.use(cors());
app.use(express.urlencoded({ extended: true }));
app.use(express.json());

app.post("/api/blob_to_mp4", function (req, res) {
  var data = Buffer.from("");

  // Add data
  req.on("data", function (chunk) {
    data = Buffer.concat([data, chunk]);
  });

  // Full data available
  req.on("end", () => {
    req.rawBody = data;
  });

  res.send("hello world");
});

app.listen(port);
console.log(`Server running on port ${port}`);


  


• Cannot convert webm video and audio to mp4 [closed]

  19 décembre 2019, par HHH837

  I am trying to record video and audio in canvas using RecordRTC and converting it to mp4 but getting below error. Please suggest me the command for ffmpeg to fix the issue.

  Received command: -i video.webm -i audio.webm -c:v mpeg4 -c:a aac -b:v 6400k -b:a 4800k -strict experimental output.mp4.  Processing with 268435456 bits.
  
  configuration: --cc=emcc --enable-cross-compile --target-os=none --arch=x86_32 --cpu=generic --disable-ffplay --disable-ffprobe --disable-ffserver --disable-a`enter code here`sm --disable-doc --disable-devices --disable-pthreads --disable-w32threads --disable-network --disable-hwaccels --disable-parsers --disable-bsfs --disable-debug --disable-zlib --disable-protocols --disable-indevs --disable-outdevs --enable-protocol=file --enable-pic --enable-small
  
  libavutil      52. 52.100 / 52. 52.100
  
  libavcodec     55. 41.100 / 55. 41.100
  
  libavformat    55. 21.100 / 55. 21.100
  
  libavdevice    55.  5.100 / 55.  5.100
  
  libavfilter     3. 90.102 /  3. 90.102
  
  libswscale      2.  5.101 /  2.  5.101
  
  libswresample   0. 17.104 /  0. 17.104
  
  [vp8 @ 0xcba330] Warning: not compiled with thread support, using thread emulation
  
  Input #0, matroska,webm, from 'video.webm':
  
  Duration: N/A, start: 0.000000, bitrate: N/A
  
  Stream #0:0(eng): Video: vp8, yuv420p, 300x400, SAR 1:1 DAR 3:4, 1k tbr, 1k tbn, 1k tbc (default)
  
  Metadata:
  
  alpha_mode      : 1
  
  Guessed Channel Layout for  Input Stream #1.0 : stereo
  
  Input #1, matroska,webm, from 'audio.webm':
  
  Duration: N/A, start: 0.000000, bitrate: N/A
  
  Stream #1:0(eng): Audio: opus, 48000 Hz, stereo (default)
  
  [graph 1 input from stream 1:0 @ 0xccf4c0] Invalid sample format (null)
  
  Error opening filters!
  
  Finished processing (took 3688ms)
  
  {"type":"done","data":[{"name":"output.mp4","data":{}}],"time":3688}
  
  {"name":"output.mp4","data":{}}

• WebRTC books – a brief review

  1er janvier 2014, par silvia

  I just finished reading Rob Manson’s awesome book “Getting Started with WebRTC” and I can highly recommend it for any Web developer who is interested in WebRTC.

  Rob explains very clearly how to create your first video, audio or data peer-connection using WebRTC in current Google Chrome or Firefox (I think it also now applies to Opera, though that wasn’t the case when his book was published). He makes available example code, so you can replicate it in your own Web application easily, including the setup of a signalling server. He also points out that you need a ICE (STUN/TURN) server to punch through firewalls and gives recommendations for what software is available, but stops short of explaining how to set them up.

  Rob’s focus is very much on the features required in a typical Web application :

  • video calls
  • audio calls
  • text chats
  • file sharing

  In fact, he provides the most in-depth demo of how to set up a good file sharing interface I have come across.

  Rob then also extends his introduction to WebRTC to two key application areas : education and team communication. His recommendations are spot on and required reading for anyone developing applications in these spaces.

  —

  Before Rob’s book, I have also read Alan Johnson and Dan Burnett’s “WebRTC” book on APIs and RTCWEB protocols of the HTML5 Real-Time Web.

  Alan and Dan’s book was written more than a year ago and explains that state of standardisation at that time. It’s probably a little out-dated now, but it still gives you good foundations on why some decisions were made the way they are and what are contentious issues (some of which still remain). If you really want to understand what happens behind the scenes when you call certain functions in the WebRTC APIs of browsers, then this is for you.

  Alan and Dan’s book explains in more details than Rob’s book how IP addresses of communication partners are found, how firewall holepunching works, how sessions get negotiated, and how the standards process works. It’s probably less useful to a Web developer who just wants to implement video call functionality into their Web application, though if something goes wrong you may find yourself digging into the details of SDP, SRTP, DTLS, and other cryptic abbreviations of protocols that all need to work together to get a WebRTC call working.

  —

  Overall, both books are worthwhile and cover different aspects of WebRTC that you will stumble across if you are directly dealing with WebRTC code.