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• 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

• 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 !

• Revision 36958 : Petite mise à jour de base de donnée... Pixelformat n’est pas un int mais ...

  5 avril 2010, par kent1@… — Log

  Petite mise à jour de base de donnée... Pixelformat n’est pas un int mais un string
  On ne refile pas de valeurs hallucinantes à l’encodeur non plus