[LAU] jack/oversampling

[Ralf Mardorf]

On Sun, 2014-03-16 at 14:39 -0400, Gene Heskett wrote:
> On Sunday 16 March 2014 14:25:14 Ralf Mardorf did opine:
> 
> > On Sun, 2014-03-16 at 08:58 -0700, Len Ovens wrote:
> > > I would mix the project  at 48k or 96k
> > 
> > Why 96 KHz? 48 KHz doesn't cause any issues, but already provides best
> > sound quality.
> > 
> That I think is a personal call Ralf, primarily because at 48 Khz, your 
> anti-aliasing filters had better be very very good brick walls by the time 
> you get above 24Khz in input content just for the aliasing control.  And 
> aliasing noise, once introduced, cannot be removed by any known math 
> function that does not have precise knowledge of the phasing (aka group 
> delay) of the original signal.
> 
> And those very good brick wall filters _will_ have a considerable group 
> delay.  IMO doing the sampling at 240K, then doing a weighted sum shift (5 
> stage shift) to decimate the data down to 48K, should result in dropping 
> the alias caused noise floor by several db.  Just bring lots of expensive 
> hardware to do that.

Are we talking about reality or golden ears?

I disagree with Fons regarding to the 44.1 KHz are as good as 48 KHz (in
theory he might be right), but at least as long as there aren't software
or hardware issues, _we_ are unable to hear a difference between 48 KHz
and > 48 KHz.

Analog audio quality is something different Gene. Is you claim that > 48
KHz you get that analog thingy, I'm missing for digital recordings?

My RME card can do 192 KHz, I never tested it, but I will compare
recordings ASAP, likely after August this year I have got the time to do
it.