I’m not either, but studied math (and computer science, physics, electric engineering, music technology, …) in 2 universities … and have forgotten most of it … but still know at least some of the basic things.
No. If both conversions and processing is done right with good anti-alias filters there is absolutely no reason to use higher sample rates. The real question is: do you have control over the whole process? Now let’s identify the problematic stages:
- A/D and D/A conversion: are you using non-oversampling converters?
1.a. If yes, definitely use higher sample rates. Analog filters are far from perfect. (OK, there’s no these kind of converters in market anymore … at least I hope so)
1.b If no, can you trust the digital filter of your oversampling converter?
I would love to see versions of plots alexis refers to, but for A/D converters. - Processing. If you process audio in the way which might introduce ultrasonic content, this content can alias into audible range if not oversampling and good anti-alias filters are used. Can you trust the writer of the plugin did his/her homework?
Yes. I record in 88.2kHz, but NOT because I think it is somehow magically superior audio format, but as an insurance against these 2 points. But then, I use my DAW just as a glorified multitrack recorder, which means I have huge amount of “free” CPU cycles to waste. If I used Cubase to create film scores with sh*tload of VSTi:s, I think I would be at 48kHz in no time.
Human hearing. Do we know enough about it? I think we do. This digital audio debate have been going on for 35 years now. If there is something more, scientists should have found it. But can we be 100% sure?