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• Connect a remote Ip camera as a Webrtc client

  5 avril 2017, par idosh

  I have 2 cameras :

  • An internal webcam embedded in my laptop.
  • A remote IP camera that is connected to my laptop through Wifi (transmits TCP, raw H264 data - no container). I’m getting the stream using node.js.

  My goal is to create a Webrtc network and connect the remote camera as another client.

  I’m trying to figure out possible solutions :

  • My naive thinking was that I would stream the remote camera payload to the browser. But as I came to understand the browser can’t handle the stream without a container. Fair enough. But I don’t understand why it does handle the video stream that arrives from my internal camera (from the navigator.getUserMedia() function). what’s the difference between the two streams ? why can’t I mimic the stream from the remote camera as the input ?
  • To bypass this problem I thought about creating a virtual camera using Manycam (or Manycam like app). To accomplish that I need to convert my TCP stream into an RTP stream (in order to feed Manycam). Though I did saw some info in ffmpeg command line, I couldn’t find info in their node.js api package "fluent-ffmpeg". Is it possible to do it using fluent-ffmpeg ? Or only using the command line tool ? Would it require another rtp server in the middle such as this one ?.
  • Third option I read about is using node.js as a client in Webrtc. I saw it was implemented in "simple-peer". I tried it out using their co-work with socket.io (socket.io-p2p). unfortunately I couldn’t get it to work / : When i’m trying to create a socket/peer in the server - it throws errors, as it expect options that are only available on the client-side (like window, location, etc.). Am I doing something wrong ? maybe there is more suitable framework for this matter ?
  • Forth option is to use a streaming server in the middle such as Kurnto. From my understanding it receives rtp as an input and transmits it as a webrtc client. I feel it’s the most excessive option, but maybe it’s not so bad (I have to admit that I haven’t investigate this option yet).

  any thoughts ?

  thanks !

• FFMPEG - How to wait until all blobs are written before finishing ffmpeg process when getting them from media recorder API

  7 novembre 2020, par Caio Nakai

  I'm using media recorder API to record a video from user's screen and sending the blobs through web socket to a nodejs server. The nodejs server is using the blobs to create a webm video file, the video is being created fine but with a delay, after the user clicks on the stop recording button it stops the media recorder api, however the server didn't finish the processing of all blobs (at least that's what I think it's happening) and then when I check the video file generated the last few seconds of the recording are missing I wonder if there's an way to solve this. Any help is appreciated :)

  


  This is the front-end code that sends the blobs to the nodejs server

  


    const startScreenCapture = async () => {
  try {
    let screenStream;
    videoElem = document.getElementById("myscreen");
    screenStream = await navigator.mediaDevices.getDisplayMedia(
      displayMediaOptions
    );

    const recorderOptions = {
      mimeType: "video/webm;codecs=vp9",
      videoBitsPerSecond: 3 * 1024 * 1024,
    };

    screenMediaRecorder = new MediaRecorder(screenStream, recorderOptions);
    screenMediaRecorder.start(1); // 1000 - the number of milliseconds to record into each Blob
    screenMediaRecorder.ondataavailable = (event) => {
      console.debug("Got blob data:", event.data);
      console.log("Camera stream: ", event.data);
      if (event.data && event.data.size > 0) {
        socket.emit("screen_stream", event.data);
      }
    };

    videoElem.srcObject = screenStream;
    // console.log("Screen stream", screenStream);
    // socket.emit("screen_stream", screenStream);
  } catch (err) {
    console.error("Error: " + err);
  }
};

const stopCapture = (evt) => {
  let tracks = videoElem.srcObject.getTracks();

  tracks.forEach((track) => track.stop());
  videoElem.srcObject = null;
  screenMediaRecorder.stop();
  socket.emit("stop_screen");
  socket.close();
};


  


  This is the nodejs back-end that handle the blobs and generates the videofile

  


    const ffmpeg2 = child_process.spawn("ffmpeg", [
    "-i",
    "-",
    "-c:v",
    "copy",
    "-c:a",
    "copy",
    "screen.webm",
  ]);


  socket.on("screen_stream", (msg) => {
    console.log("Writing screen blob! ");
    ffmpeg2.stdin.write(msg);
  });

  socket.on("stop_screen", () => {
    console.log("Stop recording..");
  });


  


• Updated(reproducible) - Gaps when recording using MediaRecorder API(audio/webm opus)

  25 mars 2019, par Jack Juiceson

  ----- UPDATE HAS BEEN ADDED BELOW -----

  I have an issue with MediaRecorder API (https://www.w3.org/TR/mediastream-recording/#mediarecorder-api).

  I’m using it to record the speech from the web page(Chrome was used in this case) and save it as chunks.
  I need to be able to play it while and after it is recorded, so it’s important to keep those chunks.

  Here is the code which is recording data :

  navigator.mediaDevices.getUserMedia({ audio: true, video: false }).then(function(stream) {
 recorder = new MediaRecorder(stream, { mimeType: 'audio/webm; codecs="opus"' })
 recorder.ondataavailable = function(e) {
   // Read blob from `e.data`, decode64 and send to sever;
 }
 recorder.start(1000)
})

  The issue is that the WebM file which I get when I concatenate all the parts is corrupted(rarely) !. I can play it as WebM, but when I try to convert it(ffmpeg) to something else, it gives me a file with shifted timings.

  For example. I’m trying to convert a file which has duration 00:36:27.78 to wav, but I get a file with duration 00:36:26.04, which is 1.74s less.

  At the beginning of file - the audio is the same, but after about 10min WebM file plays with a small delay.

  After some research, I found out that it also does not play correctly with the browser’s MediaSource API, which I use for playing the chunks. I tried 2 ways of playing those chunks :

  In a case when I just merge all the parts into a single blob - it works fine.
  In case when I add them via the sourceBuffer object, it has some gaps (i can see them by inspecting buffered property).
  697.196 - 697.528 ( 330ms)
  996.198 - 996.754 ( 550ms)
  1597.16 - 1597.531 ( 370ms)
  1896.893 - 1897.183 ( 290ms)

  Those gaps are 1.55s in total and they are exactly in the places where the desync between wav & webm files start. Unfortunately, the file where it is reproducible cannot be shared because it’s customer’s private data and I was not able to reproduce such issue on different media yet.

  What can be the cause for such an issue ?

  ----- UPDATE -----
  I was able to reproduce the issue on https://jsfiddle.net/96uj34nf/4/

  In order to see the problem, click on the "Print buffer zones" button and it will display time ranges. You can see that there are two gaps :
  0 - 136.349, 141.388 - 195.439, 197.57 - 198.589

  1. 136.349 - 141.388
  2. 195.439 - 197.57

  So, as you can see there are 5 and 2 second gaps. Would be happy if someone could shed some light on why it is happening or how to avoid this issue.

  Thank you