Originally Posted by BradLyons
Neither agreeing or disagreeing with Nyquist---what I'm referring to is the initial sampling rate of raw recordings, which is somewhere around 2.6Ghz to 2.8Ghz. Someday we'll be playing back at those sample rates as a standard. Look at VSTi's and how higher sampling rates have less latency than lower sampling rates, again it has to do with less down-sampling. Believe me I'm not the end-all expert on this, but it's what I've learned and studied for many years and has proven itself to be true, at least in my experience.
I had to re-read your post like 10 times to make sure I am comprehending it. By Raw Recording, do you mean Analog Signal? Perhaps though, what's good for software instruments might not be the same for analog signals/analog to digital/digital to analog converters.
I was really interested in your theory above that there was more than 96Khz bandwidth when using a 96khz sampling rate, which would of course be disagreeing with the math. Was that not what you meant? If you meant something else, please accept my apologies for mis reading it.
In my understanding, [please correct me if I am wrong] we need to sample at twice the highest frequency we want to accurately represent. Otherwise we will end up with Aliasing Distortion within that spectrum. This is what the "Anti-Aliasing" filter does. It prevents mis-representations from spinning back into reality. When the Converters are switched into higher sampling rates, this steep brick wall filter is pushed out of the audible_spectrum.
I see you are talking about the "theoretical discrete sampling rate of continuos time", which is not really something I care to discuss. But what does interest me, is your statement above. Having spent some time studying the basics of Nyquist, and learning how this sampling theorem works with ADA converter implementation, I am quite interested in what you meant by that statement.