Re: [Audacity-devel] Classic Filters status

[Federico M. <fm...@fc...>]

Norm,

> - badly named: I agree, that's why I called it Scientific Filters when 
> I submitted it. There's ample justification for this name: 
> CoolEditPro/Audition calls the same effect Scientific Filters, and 
> Matlab (certainly a major scientific application) uses these same 
> filters (butter, cheby1 and cheby2) as its workhorse filters

I think this had been already discussed. Scientific Filters is just a 
fantasy name. These filters are no more (and no less) scientific that 
any other, including fft filters implemented in the EQ effect.

> - the filters are "unorthodox": Certainly not - as I pointed out, 
> CEP/Audition and Matlab use these same filters (with the same 
> behaviour near the Nyquist frequency), as do many other audio 
> applications (eg. Logic, Goldwave and Wavosaur). Yes, the high 
> frequency response is different from the other filters in Audacity, 
> but it's good to have choices.

I think that when somebody said that, he was meaning that low-order 
Butterworth or Chebyshev are not truely so, but have discrepancies at 
frequencies close to the Nyquist limit, as you mentioon here:

> - I thought I saw that someone had complained the first order lowpass 
> is different from the other orders. It's not, they're all completely 
> consistent. I'll explain further if anyone still thinks this is an issue.

It is impossible that an IIR filter can match exactly the amplitude and 
phase frequency response along the full Nyquist range because the 
bilinear transformation on which it lies (I love this pun) is only an 
approximation that fails close to the Nyquist limit. May be there is a 
more close transformation (and of course there may be ad-hoc 
corrections), but bilinear is a standard conversion from analog filrters 
to some digital "equivalent".

> - documentation is inadequate: OK. CEP's Scientific Filter 
> documentation would be a good place to start. And I provided a fair 
> amount of commentary on the filter when I first submitted it. I'd be 
> glad to provide more - just let me know exactly what needs to be 
> addressed.

Possibly, some hints to its applications, for instance simulation of 
analog filters, particularly their transient response, perhaps emulation 
of some vintage effects that may have used these filters, simulation of 
analog antialiasing filters. Another, very low-latency real-time 
filtering, but I don't know if the idea was to implement real-time 
operation.

> - doesn't work properly on tracks with multiple sample rates (same 
> problem as Equalization has, I think): True, and I don't think there's 
> a simple consistent way to make it work, as the filter response varies 
> with sample rate. 

There is, provided the track sample rate is known (and I believe it is 
known), since there is a question of applying a scale factor for the 
frequency arguments that depends on the sample rate.

> - Chebyshev filters are too uncommon to be used here: Certainly 
> they're less common than Butterworth but it's far from unheard of to 
> find them in audio gear, both analog and digital. I've got lots of 
> examples if anyone cares.

Please, comment on them.

Regards,

Federico