Sorry I couldn't resist:Funkybot's Evil Twin wrote: Anyway, this section was the reason I'm replying. I'm curious as to how you're testing this. I'm using Plugin Doctor and not seeing anytning like that. In fact, the aliasing performance is quite good. Try this:
1. Load up Plugin Doctor if you have it
2. Load the VCL-25A
3. Go to the Harmonic Analysis tab of Plugin Doctor
4. In VCL-25A, set the input gain knob to 12 o'clock (value of 10.0)
5. Adjust the threshold to 12 o'clock (50)
6. Now go to the Plugin Doctor settings and play with different input levels and frequencies for the harmonic analysis. I'll usually start with a signal as close to 1k as possible with an -18 or -12 db signal, then check higher frequencies (17.5k just randomly became my go-to) to really test for foldover.
Results: pretty fantastic aliasing performance throughout.
I'm really just curious why you're seeing terrible aliasing results and I'm not.
Thank you so much for your elaborations, Funkybot's Evil Twin!
This is a perfect example of why it's a bad idea to try to judge a digital processor's aliasing performance merely by means of measuring its Nyquist gain...there is so much more to it than this.
Let me give an example: I can come up with interpolation/AA filter designs that have like 90 dB of stop-band attenuation around Nyquist, but bluntly put, they'll either have like 1 dB of passband ripple, 512 samples of latency or non-linear phase. And, what does that even mean in terms of how that would actually perform with a given non-linear chain? Correct: Not much, since it's always a trade-off between aliasing/audio transparency/CPU-load/latency for a given setting anyway.
Or here's another one: Imagine a static harmonic spectrum is implemented by means of e.g. Chebychev polynomials - that's not what I do here, but lets consider this for a minute. Converting your polynomial monomial basis you can design and apply your AA filters specifically for each power of the signal because the maximum bandwidth extension is known beforehand. E.g. x^2 will double the bandwidth so you have to apply a halfband filter before squaring your signal etc.. In this case you get around oversampling althogether and depending on the quality of the filters you use the response around Nyquist will reasonably flat.
It's just not a black and white sort of problem and I'm aware of the constraints. This may start a long discussion, I know, but from my POV I'd like to keep it at that
Best,
Ray
