Re: [Audacity-devel] Classic Filters status

[Steve t. F. <ste...@gm...>]

On 14 November 2014 06:43, Norm C <n_b...@ho...> wrote:

> Federico Miyara wrote
> > 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.
>
> Fantasy maybe, but the many thousands of CEP/Audition users will recognize
> it immediately. And I do think it's better than Classic Filters, but
> whatever, I'm easy.
>
>
> >> - 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:
>
> All orders have that issue. It's just more obvious on the lower ones.
>
>
> >> - 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".
>
> Indeed it is.
>
>
> >> - 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.
>
> Good points, except I'm not sure what the relevance of real-time operation
> is...?
>
>
> >> - 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.
>
> Of course, there's no problem applying the filter to multiple tracks
> correctly; the consistency problem is that the UI displays a frequency
> response graph which (with multiple partly-overlapping lines) will be
> confusing to users, or (with a single line) wrong.
>
> >> - 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.
>
> Chebyshev filters have long been used in crossovers (esp for subwoofers),
> low-pass filtering for BBD devices (eg. in chorus/flanger/vibrato gear),
> low-pass filtering for downsampling in digital audio gear, radar signal
> processing etc. etc. A quick google search will show that there's much
> interest in calculators for Cheby designs, and they are discussed in all
> the
> major signal processing and audio engineering handbooks, plus you'll find
> them in app notes from the big DSP chip vendors (TI, ADI...). They're
> certainly not obscure.
>

What would be typical applications of Chebyshev type I and type II filters
for end users of Audacity?

Steve



>
> Anyway, I'd be happy to put in the effort to document these filters
> satifactorily, and to fix the multiple-sample-rate issue... just need some
> indication that no one has some whopping big objection to reinstating the
> effect.
>
> Norm
>
>
>
>
> --
> View this message in context:
> http://audacity.238276.n2.nabble.com/Classic-Filters-status-tp7565152p7565203.html
> Sent from the audacity-devel mailing list archive at Nabble.com.
>
>
> ------------------------------------------------------------------------------
> Comprehensive Server Monitoring with Site24x7.
> Monitor 10 servers for $9/Month.
> Get alerted through email, SMS, voice calls or mobile push notifications.
> Take corrective actions from your mobile device.
>
> http://pubads.g.doubleclick.net/gampad/clk?id=154624111&iu=/4140/ostg.clktrk
> _______________________________________________
> audacity-devel mailing list
> aud...@li...
> https://lists.sourceforge.net/lists/listinfo/audacity-devel
>