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• Stream Stops Working Using FFMPEG Inside a Node Child_Process Spawn

  14 septembre 2020, par Brad Spencer

  I'm using a video stream on a raspberry pi with raspivid and ffmpeg in a node app. Using them in the terminal (without using node) will stream for hours, but when I use them in a node child_process (I spawn 2, one for each) it works great for a little over 3 minutes and then the stream stops. The child_processes are still running and I'm not seeing any errors.

  


  The gist of my code :

  


  let camera = spawn('raspivid', args)
let ffmpeg = spawn('ffmpeg', args)  

camera.stdout.on('data', (data) => {
  ffmpeg.stdin.write(data)
})


  


  Any ideas why it is stopping after 3 minutes ? Thanks !

  


• How to use AVCodecContext::skip_frame ?

  28 mai 2012, par Denis Loh

  I am developing a Video over IP telephone client which is using ffmpeg libavcodec to decode h.264 video streams. I am facing the issue that audio and video are getting asynch during high network load (due a parallel upload or something similar) or short network interruptions.

  I get the following message by ffmpeg : Increasing reorder buffer to 16.

  This message comes very seldom and when it comes, I get a high delay of approx. 1 second after the appropriate audio. I googled hours and days and only found that the decoding performance is much too low to decode the frames in time. However, the CPU load is not that high and the issue occurs at any frame rate and any frame resolution. Always with the same effect.

  The only thing I believe could solve the problem is to tell the decoder to speed up by dropping some framess.

  Can this be done by using AVCodecContext::skip_frame ? If so, how does it work ? How do I avoid the message above or better how do I react on that ?

  Thank you for your help !

• How to get accurate time information from ffmpeg audio outputs ?

  11 février 2016, par user2192778

  I want to figure out the best way to have accurate (down to millisecond) time information encoded into a web stream audio recording. So, if I wanted to know what was streaming at exactly 07:57:25 yesterday, I could retrieve that information using my program code. How can I do this in my ffmpeg function ?

  So far I have a python script that calls the following ffmpeg function every hour :

  ffmpeg -i http://webstreamurl.mp3 -t 01:30:00 outputname.mp3

  It will record 1.5 hours of my stream every hour. This leaves me with many 1.5 hour-long clips which together (due to some overlap) will give me audio at every moment.

  I don’t know how I will sync these clips with their stream times, however. Is there a way to put timestamps in the audio and have python read it ? Or is there a way to name these output files such that python could calculate the times itself ? Any other way ?