It could be due to the fact that ... the Fast Track Ultra ... is optimized for 96/24, whereas another audio interface may work better at a lower sample rate.

Dan Lavry has said as much. An interface designer has to pick a sample rate to design around, and might well choose 96KHz as that optimal rate. I would have expected a prosumer-class device like the Fast Track Ultra, being targeted at hobbyists, would have been optimized for 44.1. But who knows what their reasoning was?

And of course you do get lower latency with higher sample rates, whether it sounds better or not. For me, 5ms is fine. It's no different than being 5 feet away from your amp or monitor on stage. I don't play soft synths in real time, as I prefer hardware synths for tracking and then substituting computer-based samples later on. Latency is therefore never an issue.

But mainly, 44.1 makes sense for me because I use a lot of synthesizers. Whether soft or hard, they are all playing 44.1k or 48k samples. There is no benefit to upsampling them, so I'd just be burning up disk space for nothing.

the point is I guess that humans don't perceive it due to the technology they use to acquire it. The value could be undiscovered.

This is double talk and has no meaning. You did the same before with this..

Just because most of us can not quantify the difference with current playback technology does not mean there is not a difference that has value.

I get the impression you are talking about magic. Some unknown thing that can neither be quantified or perceived. It has no business being introduced into a rational discussion.  

I understand there is a company in Los Angeles that will insert pixie dust into your interface for eight hundred dollars.

Nothing above 7k plz, I just love the Phone Tone

bitflipper


I understand there is a company in Los Angeles that will insert pixie dust into your interface for eight hundred dollars.

I will do it for $300.

The problem here is not just one of a metaphor, nor is the problem one of just being overly finicky with language. It's not a nerdy-technical argument as much as it's a real misunderstanding of how this stuff works.

And if for no other reason, you want to try to grasp it because it may help when you buy your next interface.

  -- Joe "And let the pie-throwing continue" Tabby

rabeach


Audio signals that we sample are not signals that are bandlimited and sampled for infinite time therefore per the theorem cannot be perfectly reconstructed using the nyquist frequency.

But again, we are talking about audio for humans to hear. As far as human perception is concerned, the signal is sufficiently band limited. Whatever the filters do not entirely reject is at such a low level that we can not perceive it either at it's original frequency (which also falls outside of our hearing range) or after it aliases back. And the same goes for the errors introduced by the lack of infinite sample points. Any errors are buried deep below the noise floor where we can't hear them.

UnderTow

rabeach


the point is I guess that humans don't perceive it due to the technology they use to acquire it. The value could be undiscovered.

Rubbish. The technology is not limiting us. For instance, we can go way beyond the frequency abilities of our auditory system. If we can't perceive a difference when having more than twice (88.2Khz) the bandwidth, or four times (176.4Khz) or even more than eight times the bandwith (384Khz sampling rates) than we can actually perceive, there is absolutely no reason whatsoever to believe that we would perceive an even bigger bandwidth increase and believing that is just silly. Don't forget that these are off the shelf products available to anyone. It isn't even at the real limits of our technology. (But no one in their right mind would consider going even further into cookoo land than we already have gone with some of these converters).

UnderTow

can we talk wow and flutter and get this over with !!

I'm still trying to figure out where in the SONAR menus you adjust the bias oscillator.