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PostPosted: Thu Jan 24, 2019 10:03 pm 

Joined: Thu Jan 24, 2019 9:58 pm
Posts: 3
I have become a fan of Stereo Tool and I now process a lot of old 80s/90s CD albums with Stereo Tool to bring out the sparkle. However, I have run into a bit of an issue - some albums have quite big differences in "average DB" between tracks. So if I run Stereo Tool on several tracks in sequence, sometimes there will be a noticeable dip or rise in volume at the beginning of a track. I am converting primarily Jazz music.

Is there some way to prevent this? I have found Bojcha's Dyanrider preset fairly good at gently managing the volume (thanks Bojcha!), but not always, and other presets seem to suffer from this, even if they are otherwise better sounding. I thought of maybe adding a test-tone between the tracks as I process them that is generated at that tracks average DB or something like that, but just wondering is there a better way? Or some way to get Stereo Tool to look ahead more when applying compression.

Thanks!


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PostPosted: Fri Jan 25, 2019 12:57 pm 
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Joined: Tue Sep 15, 2015 12:22 pm
Posts: 257
If you don't want large differences in level and you don't want to heavily compress your audio, I would suggest normalizing the tracks in advance. If you're processing in real-time, then look for a playback program that supports level normalization. If you're processing files, have a look at WatchCat (available from our Downloads page) which supports RMS normalisation before processing.

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PostPosted: Fri Jan 25, 2019 7:08 pm 

Joined: Thu Jan 24, 2019 9:58 pm
Posts: 3
Thanks for your reply!

However I did manage to figure out another solution, my main issue was just at the beginning of tracks, so I thought, why not make it do the start of the track to calibrate it, then do the actual track?

Using FFMPEG and Powershell, I generate a 10 second snippet of each track to process (with 500ms of silence at the end to prevent any buffer issues). Here is my Powershell script:
Code:
ls *.flac|%{
    $outputTitle = "CalibrationTrack" + $_.name.Substring(0,2)
    $outputFile = $_.name.Substring(0,2).Insert(2," !.flac")
    .\ffmpeg -i $_ -metadata title=$outputTitle -metadata artist="" -metadata album="" -t 00:00:10 -af "volume=0:enable='between(t,9.5,10)'" $outputFile
 }
Put ffmpeg.exe and this script in a new file called Script.ps1 in the same folder as the audio tracks

It expects you to have a folder of files like this:
01 First Track.flac
02 Another Track.flac
03 A Track.flac

When you run the PS1 script it will generate files that go inbetween like this:
01 !.flac
01 First Track.flac
02 !.flac
02 Another Track.flac
03 !.flac
03 A Track.flac

So when you drag the folder into Winamp, it will use alphabetical sorting so these 10 second "calibration tracks" will appear before each track.

I'm using Winamp's Disk Writer plugin with %title% as the output format, since FFMPEG tags the files as "CalibrationTrack" you can just delete these files from the output folder.

Hope this helps someone!


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PostPosted: Mon Jan 28, 2019 9:30 pm 

Joined: Thu Jan 24, 2019 9:58 pm
Posts: 3
OK now I feel a bit stupid - I found simply turning off the AGC in the processing section yields the exact kind of result I wanted :lol: - no need for calibrating like this.


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