I am moving from software engineering to audio production. I read a good deal about audio editing (mostly from Transom.org) and I want to produce audio for both radio at -24 LUFS and podcasts at -16 LUFS. The problem is that the loudness range of my recordings is too large: with loudness normalization of -16 LUFS, the peaks go close to 0dB. At the same time, I'm not sure how to use a compressor, as the FFMPEG help page mentions:
The right compression is the key to reach a professional sound and is the high art of mixing and mastering. Because of its complex settings it may take a long time to get the right feeling for this kind of effect.
My recordings are a superposition of background noise at -66 dB with either a performance voice at -20 dB and peaks at -11 dB or background talking while I direct the voice actor. I used FFMPEG's linear loudness normalization:
ffmpeg -i /path/to/input.wav -af loudnorm=I=-16:TP=-1:LRA=11:measured_I=-27.2:measured_TP=-14.4:measured_LRA=0.1:measured_thresh=-37.7:offset=-0.7:linear=true:print_format=summary output.wav
where the measured values come from a first pass (see here). I want a linear normalization to avoid increasing the loudness of background talk.
I tried to set
I=-16:TP=-9 (loudness of -16, peaks below -9) and the voice sounds saturated.
I believe that I need some affine transformation: any linear transformation will boil down to a different loudness normalization. And I understand from the graph in Audacity that the compressor is exactly an affine transformation.
All in all, I am looking for entry-level compressor settings of FFMPEG's
acompressor filter that will squash the range of the voice actor and produce sound at -16 LUFS that sounds natural and peaks away from 0 dB. I would like to apply these settings to around 1 hour of recordings and think that this will help me in mastering this high art of mastering and mixing.
Is that reasonable? Or is the only way to use compressor trial-and-error and much listening to the audio?