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• FFmpeg RTP_Mpegts over RTP protocol

  7 mars 2020, par Nicolò

  I’m tryin to implement a client/server application based on FFmpeg. Unfortunately RTP_MPEGTS isn’t documented in the official FFmpeg Documentation - Formats.
  Anyway i found inspiration from this old thread.

  Server Side

  (1) Capture mic audio as input. (2)Encode it as pcm 8khz mono and (3) send it locally as RTP_MPEGTS format over rtp protocol.

  ffmpeg -f avfoundation -i none:2  -ar 8000 -acodec pcm_u8 -ac 1 -f rtp_mpegts rtp://127.0.0.1:41954

  • This works, but on initiation it alerts "[mpegts @ 0x7fda13024600] frame size not set"

  Client Side (on the same machine)

  (1) Receive rtp audio stream input (2) write it in a file or playback.

  ffmpeg -i rtp://127.0.0.1:41954 -vcodec copy -y "output.wav"

  • I’m using -vcodec copy because i’ve already verified it in another rtp stream in which -acodec copy didn’t work.

  • This stuck and while closing with Ctrl+C shortcut it prints :

    Input #0, rtp, from 'rtp://127.0.0.1:41954':
    
    Duration: N/A, start: 8.956122, bitrate: N/A
    
    Program 1 
    
    Metadata:
    
      service_name    : Service01
    
      service_provider: FFmpeg
    
    Stream #0:0: Data: bin_data ([6][0][0][0] / 0x0006)
    
    Output #0, wav, to 'output.wav':
    
    Output file #0 does not contain any stream

  ---

  1. I don’t understand if the client didn’t receive any stream, or it cannot write rtp packets into "output.wav" file. (Client or server problem ?)

  2. In the old thread is explained a workaround. On server could run 2 ffmpeg instance :
     One produces "tmp.ts" file due to mpegts, and the other takes "tmp.ts" as input and streams it over rtp. Is it possibile ?

  3. Is there any better way to do implement this client/server with the lowest latency possible ?

  ---

  Thanks for any help provided.

• using node-fluent-ffmpeg to transcode with ffmpeg on windows not working

  25 juillet 2015, par jansmolders86

  I’m trying to use the module node-fluent-ffmpeg (https://github.com/schaermu/node-fluent-ffmpeg) to transcode and stream a videofile. Since I’m on a Windows machine, I first downloaded FFMpeg from the official site (http://ffmpeg.zeranoe.com/builds/). Then I extracted the files in the folder C :/FFmpeg and added the path to the system path (to the bin folder to be precise). I checked if FFmpeg worked by typing in the command prompt : ffmpeg -version. And it gave a successful response.

  After that I went ahead and copied/altered the following code from the module (https://github.com/schaermu/node-fluent-ffmpeg/blob/master/examples/express-stream.js) :

  app.get('/video/:filename', function(req, res) {
  
  res.contentType('avi');
  
  console.log('Setting up stream')
  
  
  
  var stream = 'c:/temp/' + req.params.filename
  
  var proc = new ffmpeg({ source: configfileResults.moviepath + req.params.filename, nolog: true, timeout: 120, })
  
      .usingPreset('divx')
  
      .withAspect('4:3')
  
      .withSize('640x480')
  
      .writeToStream(res, function(retcode, error){
  
          if (!error){
  
              console.log('file has been converted succesfully',retcode);
  
          }else{
  
              console.log('file conversion error',error);
  
          }
  
      });
  
  });

  I’ve properly setup the client with flowplayer and tried to get it running but
  nothing happens. I checked the console and it said :

  file conversion error timeout

  After that I increased the timeout but somehow, It only starts when I reload the page. But of course immediately stops because of the page reload. Do I need to make a separate node server just for the transcoding of files ? Or is there some sort of event I need to trigger ?

  I’m probably missing something simple but I can’t seem to get it to work.
  Hopefully someone can point out what I’ve missed.

  Thanks

• Set RTSP/UDP buffer size in FFmpeg/LibAV

  9 décembre 2016, par chuckleplant

  Note : I’m aware ffmpeg and libav are different libraries. This is a problem common to both.

  Disclaimer : Duplicate of SO question marked as answered but actually didn’t give a proper solution.

  ---

  Insufficient UDP buffer size causes broken streams for several high resolution video streams. In LibAV/FFMPEG it’s possible to set the udp buffer size for udp urls (udp ://...) by appending some options (buffer_size) to it.

  However, for RTSP urls this is not supported.

  These are the only solutions I’ve found :

  • Rebuilding ffmpeg/libav changing the UDP_MAX_PKT_SIZE in the udp.c source file.
  • Using a nasty hack to find and modify the required value, by casting some private structs.
  • Using a different decoding library (proposed solution to aforementioned related SO question).

  None of these is actually a solution. From what I found it should be possible to use the API’s AVOptions to find and set this value. Or else, the AVDictionary.

  It’s very difficult to find how to set these throughout the documentation of either libav or ffmpeg.

  Update :

  The following patches have been submited to Libav tackling this topic, thanks to Libav developer @lu_zero :

  • Add a buffer_size option
  • Map the urloptions to AVOptions

  Which should offer a hint on how to implement those, still these are not yet available through the official stable API.