The biggest proof was when I first recorded a drumset at 176k. I could not believe how awesome it sounded.
@vangrieg - I don't know, but I think a lot of oversampling would increase the latency substantially. Currently, it seems the latency is just that of the AD/DA converters.
And that means no oversampling.
How are you controlling for placebo there? Are you doing A/B/X, double-blind?...There absolutely was a difference then and there is one now. I have yet to not prove it even to the most skeptical if using decent equipment. The biggest proof was when I first recorded a drumset at 176k. I could not believe how awesome it sounded. It was astounding. Every nuance was clear and present...
The Fractal approach makes perfect sense, given they can work at incredible fidelity at 48Khz, and you can just adjust it to fit your audio project after the fact.
I don’t grok how you can love Fractal products and simultaneously claim that ultra-high sampling rates are the only way you’ll experience audio.
Who would convert something into analog just to change the sample rate within a project? That's a horrible way to do it (although it would achieve the desired outcome).Sigh...
You may use other equipment. Fixed sampling rate creates interoperability problems. You may be forced to use multiple DA/AD conversions, that's worse than resampling within the unit. Nobody outside of Fractal world does projects in 48k by default. And so on.
Yeah, I get that. But should we limit everyone's presets so that we know they'll work at 384Khz, which takes a lot more processing power than 48Khz?The question isn’t “how do I record Axe into a project with a sample rate other than 48k”. There are ways to do it, and there are pros and cons to each of them. The question is “please make it possible to switch sample rate in Axe FX”. There are multiple reasons why this will be convenient, and not just for recording.
Helix Rack for an example of a modeler that can serve as the center of your workstation.
I'm 28 years old, and I don't here any differences between 48 kHz and 96 kHz processed material. But I do remember something about filter slopes getting worse near the Nyquist frequency.Sampling rate affects only one thing: the highest frequency that can be accurately reproduced. That's it.
With modern anti-aliasing filters, 48 KHz sampling can accurately reproduce just a little over 20 KHz, which is the highest frequency that the best human eats can hear. Anything above that is inaudible, which makes it a waste of resources in an audio system.
Well that’s not proof. Scientific tests do not confirm that, or that people can actually distinguish between a properly encoded 320 kbps mp3 and uncompressed wav or even dsd. A lot of mp3s (or m4as, doesn’t matter), are not encoded properly, so there’s clipping whole decoding and that’s where the noticeable difference comes from.