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• Normalizing audio of several wav snippets with ffmpeg

  17 mai 2023, par dick_kickem

  I searched the site and I figured that maybe ffmpeg-normalize could be part of my solution but I'm really not sure.

  


  In my free time me and my friends create quizzes for people to solve. One of these is a music quiz, where you hear audio snippets and have to guess the artist and song title. A few years back I did most of them using Audacity, which means recording snippets from existing videos, inserting fade in and fade out to every snippet, putting announcements like "Number x" before the x-th song and also making sure that all songs are fairly of equal loudeness (-6.0 dB).

  


  The result in Audacity looked like this.

  


  

  


  Lazy as I am, I learned about ffmpeg and wrote a script, which does all these steps for me. The script is mostly written in bash. I use some audio files where I extract the audio in wav-format, add a fade in and fade out and then I try to set the volume to -6.0 dB as with Audacity. The part where this happens looks like this :

  


  ...[some code before]...

# write the audio details of temmp.wav into the "info" file
ffmpeg -i temp.wav -filter:a volumedetect -f null - 2> info

#check out the max volume of temp.wav
max_vol=$(grep "max_volume" info | cut -d ' ' -f5)

# determine the difference between the max volume and -6
vol_diff=$(expr "-6-($max_vol)" | bc -l)

# change temp.wav loudness by the determined difference
ffmpeg -y -i temp.wav -filter:a "volume=$vol_diff$db" $count.wav

...[some code after]...


  


  I do this with all snippets, leaving me with usually ten snippets in the format 1.wav, 2.wav and so on. Lastly, I concatenate them all with announcements in the form nr1.wav, 1.wav, nr2.wav, 2.wav and so on. Overall this works really great for me. Yet, the loudness is not quite as equal as in Audacity. Here is a screenshot of a generated music quiz using the described script (not the same music snippets as the example before) :

  


  

  


  You can see some peaks pointing out. It's not bad and in fact, it works for me in most cases, but it's not like what I used to have with Audacity. Does anyone have any idea how to fix that to make it more equal ?

  


  Thank you in advance and kind regards

  


• Create audio file with many audio chunks (WAV) using ffmpeg or sox

  28 juin 2017, par Jacopo Vichi

  I’m searching on a method that can help me creating an audio file (of known length) composed by mixing some audio chunks at specific time position.

  For example :
  I want to create song.wav with duration of 5 minutes, composed by :
   chunk1.wav (time 0:02)
   chunk3.wav (time 0.50)
   chunk2.wav (time 1:20)

  I think I can create 5 minutes of silence then mixing the wave files but I don’t know how.
  I try SOX but no solution found yet
  I’d like also FFMPEG based solution

• Anomalie #3566 : Restaurer une révision ne restaure pas toujours tous les champs modifiés

  10 octobre 2016, par marcimat ☺☮☯♫

  Je confirme que ça remarche (et parfaitement avec la nouvelle fonction) en remettant l’appel à auteur_associer() après le pipeline post_edition.
  Le problème se situe donc bien à ce niveau. Il va falloir trouver comment permettre de dire à Révisions de lever le flag plus tôt que post_insertion… avant donc le passage à pre_insertion_liens ;

  ou réussir à lui faire lever ce flag dans pre_insertion_liens en transmettant un paramètre indiquant que c’est le premier lien ici (mais ça ferait passer ce paramètre de auteur_associer() jusqu’au pipeline dans un grand nombre de fonction. Pas idéal non plus.

  Un pipeline trig_insertion() juste après l’obtention de l’identifiant ? …