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• Using FFmpeg within a python application : ffmpeg tool or libav* libraries ?

  5 septembre 2017, par user2457666

  I am working on a python project that uses ffmpeg as part of its core functionality. Essentially the functionality from ffmpeg that I use boils down to these two commands :

  ffmpeg -i udp:// -qscale:v 2 -vf "fps=30" sttest%04d.jpg
  
  ffmpeg -i udp:// -map data-re -codec copy -f data out.bin

  Pretty simple stuff.

  I am trying to create a self-contained program (which uses the above ffmpeg functionality) that can easily be installed on any particular system without relying on that system having the necessary dependencies, as hopefully I would package those dependencies with the program itself.

  With that in mind, would it be best to use the libav* libraries to perform this functionality from within the program ? Or would a wrapper (ffmpy) for the ffmpeg command line tool be a better option ? My current thinking on the drawbacks of each is that using the libraries may be the best practice, but it seems overly complex to have to learn how to use them (and potentially learn C, which I’ve never learned, in the process) just to do those two basic things I mentioned above. The libraries overall are a bit of a bit of a black box to me and don’t have very much documentation. But the problem with using a wrapper for ffmpeg would be that it essentially relies on calling a subprocess, which seems somewhat sloppy. Although I’m not sure why I feel so viscerally opposed to subprocesses.

• How much the duration of an mp3 file given by Ffprobe is precise ?

  24 septembre 2024, par Julien Larget-Piet

  I have an mp3 file at a sample rate value of 44100, let's name it a.mp3.

  


  So i get the duration of a.mp3 with the following ffprobe command :

  


  ffprobe -v error -show_entries format=duration -of default=noprint_wrappers=1:nokey=1 a.mp3

  


  I get 7.752 seconds.

  


  Because the sample rate is per second, in theory the total amount of samples should be,

  


  44100 * 7.752 = 341863.2

  


  Let's round it to 341863

  


  But using python library ffmpegio with the following code, i get a total amount of sample equal to 340560.

  


  with ffmpegio.open(file, 'ra', blocksize = 16, sample_fmt = 'dbl') as file_opened:

        for i, indata in enumerate(file_opened):

            do some stuff
    print(i * 16)


  


  So there is a non-negligible difference between 341863 and 340560.

  


  I think it comes from the duration value given by Ffmpeg.
What do you thin ?

  


  I search in ffmpegio documentation, but found nothing that trigered my attention :

  


  https://python-ffmpegio.github.io/python-ffmpegio/

  


  Thanks.

  


  Search for answers about the precision of the ffprobe command.

  


• Fixing audio/video out of sync when editing recorded broadcast

  27 décembre 2019, par sba

  I have recorded broadcast-ed material using a DVB-T tuner.

  That produces several TS files per recording (most likely to keep filesize within FAT32 limitations).

  I have concatenated each recording into a single TS file using :

  ffmpeg.exe -f concat -i $filelist -c copy -y $outputfile

  From there, I need to perform edits to remove commercials, strip extra stuff from the start and end, and optionally extract each episode into a separate file.

  I own Pinnacle Studio 23 Ultimate, that doesn’t support TS as input so I’m converting TS to MP4 using something like :

  ffmpeg -y -i foo.ts -crf 18 -ac 2 -r 25 -c:v libx265 -ar 48000 -b:a 192k -c:a aac -aspect 16:9 bar.mp4

  (I’ve tried several ways/options, including using HandBrake for that conversion).

  What happens is that in the resulting edited material, audio and video are out of sync. Can be in the whole file, or only in some sections.

  This could be linked to glitches (missing frames...) in the original recording.

  But when I play the original single-file TS, or the MP4 version, in any player such as VLC, audio and video are properly aligned. So these players are able to deal with the aforementioned glitches and "re-align" the audio and video streams.

  How can I "rewrite" the whole input file in such a way that audio and video are "fully synchronized" so that editing will be possible ?

  Thanks in advance for your help.