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• 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

• 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.

• Tools for investigating video corruption — ffmpeg / libavcodec

  11 juillet 2013, par Gopherkhan

  In my current work I'm trying to encode some images to h264 video using the FFMPEG's C library. The resulting video plays fine in VLC, but has no preview image. The video can play in VLC and Mplayer on ubuntu, but won't play on Mac or PC (in fact, it causes a "VTDecoderXPCService quit unexpectedly" error on Mac).

  If I run the resulting file through FFMPEG using the command line, the resulting file has a preview image, and plays correctly everywhere.

  Apparently the file that I get out of the program is corrupt in some weird place, but I don't have any output during my compilation or run to indicate where. I can't share my code at the moment (work code isn't open source yet :-( ), but I have tried a number of things :

  1. Writing only header and trailer data (av_write_trailer) and no frames
  2. writing frames only minus the trailer (using avcodec_encode_video2 and av_write_frame)
  3. Adjusting our time_base and frame pts values to encode only one frame per second
  4. Removing all variable frame rate code
  5. Numerous other variants that I won't bother you with here

  In creating my project, I've also followed the following tutorials :

  • Dr. Anger's FFMPEG tutorial
  • FFmpeg's Own Tutorials
  • An older tutorial from FFmpeg

  And consulted the deprecated ffmpeg functions list

  And compiled FFMPEG on ubuntu according to the official doc

  But every run of the program runs into the exact same problem.

  My question is, is there anything obvious that causes a programmatic run of FFMpeg to differ from a console run (e.g., an incomplete finalization, some threading issues, etc.) ? Like some obvious reason that a console run could repair a corrupted file ? Or is there a decent tool/method for inspecting a video file and finding the point of corruption ?