So, I got through recording my entire book (yay!), following the Honorable Koz’s method of EQ>Normalize to 3.2>Compress>Normalize to 3.2…
The files met ACX requirements. Life was good. Then, by happenstance I recorded the last chapter speaking slightly closer to the microphone. It sounded noticeably better. Like someone was actually there, instead of listening to a recorded voice, if that makes any sense…
BEFORE, my raw tracks were consistently around -9db before normalizing. The occasional exclamation or stress would get up to -5db or so, and then the normalizing would pull everything up.
NOW, my raw tracks are dancing around -3db in every sentence. The waveform looks more dynamic. Almost every other sentence, though, I go over -3.2db.
If I were to just blindly normalize to -3.2db, I would likely be pushing everything down a couple decibels, as I surely have a -1db peak in there somewhere. That’s no good - I want to keep the quieter portions as is.
So, I’ve been manually finding peaks over -3.2db (using the ACX Check plugin), and normalizing only that word to keep it under the threshold. The result sounds great, but holy moly is it time consuming.
My question then, is if there is a better way to do this? I suppose compression with a -4db threshold and 3:1 compression would work in theory? But I’m afraid to try that and then compress AGAIN later. Side note - I actually met ACX standards before compression and normalization on one of the tracks.
Related, is there a way to have Audacity highlight or somehow show you all instances of the sound going over a set amount? This would save me tons of time.
Or, am I terribly wrong with all of this and I should just shut up and go home?