Last post by wcs13 -
Thanks zeremy. Maybe I could implement this. Here's my current issues :
1. AFAIK, in foobar, .tags files are always named "!.tags", and I haven't found any setting to name them otherwise. If you're suggesting that all the .tags files (past and future) are named "[filename].tags", then how could I achieve this ? (if there's a foobar m-TAGS setting, please let me know)
2. I always (well, 99% of the time) have only one video file per folder. But a lot of my movies include bonus (featurettes) in a specific "featurettes" folder. One single movie can have more than 20 featurettes (deleted scenes, gag reel, interviews, etc.). I can't imagine having them all in separate folders. But I could imagine having 20 .tags files named after every one them. So yes, that's a good argument for naming the .tags files "[filename].tags".
3. I have already spend hundreds of hours to get to my current config, so I can't just delete all my .tags files and start over. Whatever solution will need to take into account the existing "!.tags" files, even if they're deleted/replaced afterwards.
This plugin is really interesting. But there might be a problem for me: Quieter passages of a musical piece get also normalized after a few seconds and get (almost) as loud as the other, louder parts. I think it would be better, if this plugin would only focus on the peak volume of an audio file in order to preserve the dynamics of the music.
Last post by [JAZ] -
I believe he means that he will be receiving analog audio on the repeaters, but that he wants to compress them at the repeater, and send it with broadband to the cloud, where they would be able to listen to it in realtime, hence why the latency is important.
What he wanted to describe is the kind of audio that will need to be compressed so that the codec would not choke with that.
If that is true, the part about analog transmission of the encoded audio is not relevant, but Opus is still a good candiate
Last post by saratoga -
Spectrograms are useful for looking at analog equipment, but for a perceptual encoder they can't tell you anything. The "denoise" you see is just your imagination. If you were making changes based on that, then your changes are random. That is a bad way to do things.
If you couldn't ABX the original file then what you are doing is pointless; you can't improve on something that doesn't have any audible flaws. You need to find something that can be improved first.
If you want to learn about audio codecs, I suggest starting at lower bitrates, maybe 100 kbps, and then finding files where you can hear a problem. Then you can try tuning. However, this is going to be immensely more difficult then you're assuming, and will require you to have a much deeper understanding of how perceptual encoders actually work.