Hi everyone,
I hope I’m posting in the right place to ask this kind of question. I think it’s probably a beginner problem.
I’m interested in building audio-reactive experiments, but I’m still struggling to process the audio signal the way I’d like.
My main questions are:
How can I normalize an audio signal between -1 and 1 in the most proper and efficient way, without compressing it or losing information?
How can I make the signal move in a smoother way, either between 0 and 1 or between -1 and 1, while still using the full range?
Here’s my current approach:
I’ve been using a little trick I found in Elekktronaut tutorials:
Audio File In → Math CHOP to combine the channels → Envelope CHOP → then divide the original signal by the envelope using another Math CHOP.
This normalizes the sound between -1 and 1.
I also add an Audio Spectrum CHOP to my network if I want a signal between 0 and 1.
My main problem is that the signal still feels a bit harsh, and I’d like it to move in a softer, smoother way. If I use a Lag CHOP, it tends to stay too high and doesn’t really move through the full range of the normalized value.
So I’m wondering how more experienced people would approach this kind of problem.
Thank you in advance for any advice. I hope my broken English is clear enough to understand 
). So I think i’ll normalize it in ableton before using it.