[PD] [lop~] coefficient calculation

[Joe White]

Ahh yes of course thanks Frank!

Have you guys checked out this paper on 'High-Order Digital Parametric
Equalizer Design <http://www.aes.org/e-lib/browse.cfm?elib=13397>'?
Apparently it reduces the need to cascade filter implementations to achieve
high orders.

Cheers,
Joe


On 24 May 2014 09:53, Frank Barknecht <fbar at footils.org> wrote:

> Hi Joe,
>
> versions of these calculations without [expr] are also part of the
> rj-library as u_lowpass, u_lowpassq etc. These have been taken straight
> from the Audio-EQ-Cookbook.
>
> Ciao
> --
> Frank
>
> On Fri, May 23, 2014 at 12:06:45PM +0100, Joe White wrote:
> > Thanks for the abstractions Chris. Am I correct in thinking the licensing
> > issues for [expr] have been resolved now?
> >
> > Cheers,
> > Joe
> >
> >
> > On 21 May 2014 23:22, Chris Clepper <cgclepper at gmail.com> wrote:
> >
> > > On Wed, May 21, 2014 at 5:31 PM, Joe White <white.joe4 at gmail.com>
> wrote:
> > >
> > >>
> > >>
> > >> Is it intentional to not a bank of go-to filters? [biquad~] is the
> next
> > >> one I would go to, but generating your own coefficients isn't that...
> err..
> > >> efficient when you're wanting some that just 'works' :)
> > >>
> > >>
> > > Attached are a set of abstractions wrapping most of the 'Audio EQ
> > > Cookbook' formulae around biquad~.  It would be nice for Pd to include
> > > something like this.
> > >
> > > The only drawback to [biquad~] is it doesn't take audio rate
> coefficients.
> > >  There are of course externals that do audio rate for cutoff, Q, etc.
> > >
> > > Chris
> > >
> > >
> > >>
> > >>
> > >> On 21 May 2014 17:31, Miller Puckette <msp at ucsd.edu> wrote:
> > >>
> > >>> Hi Joe -
> > >>>
> > >>> That code is an approximation that works well for low cutoff
> > >>> frequencies but badly for high ones.  (I should probably warn
> > >>> about this in the help window... that'll go on my dolist)
> > >>>
> > >>> cheers
> > >>> M
> > >>>
> > >>>
> > >>> On Fri, May 16, 2014 at 12:58:31PM +0100, Joe White wrote:
> > >>> > Hi,
> > >>> >
> > >>> > I've been looking at the [lop~] implementation (Pd-0.45-4) and
> noticed
> > >>> > something that seem weird to me.
> > >>> >
> > >>> > In d_filter, line 176:
> > >>> >
> > >>> > static void siglop_ft1(t_siglop *x, t_floatarg f)
> > >>> > {
> > >>> >     if (f < 0) f = 0;
> > >>> >     x->x_hz = f;
> > >>> >     x->x_ctl->c_coef = f * (2 * 3.14159) / x->x_sr;
> > >>> >     if (x->x_ctl->c_coef > 1)
> > >>> >         x->x_ctl->c_coef = 1;
> > >>> >     else if (x->x_ctl->c_coef < 0)
> > >>> >         x->x_ctl->c_coef = 0;
> > >>> > }
> > >>> >
> > >>> >
> > >>> > Is it correct that for:
> > >>> >
> > >>> > y[n] = x[n] * a + y[n-1] * b
> > >>> >
> > >>> > *a = 2π * Fc / Fs*
> > >>> > b = 1.0 - a
> > >>> >
> > >>> > where Fc is the cut-off frequency and Fs the sampling frequency.
> > >>> >
> > >>> > I appreciate the a coefficient is bounded afterwards but wouldn't
> that
> > >>> mean
> > >>> > that Fc values greater than Fs / 2π will have no impact on the
> sound
> > >>> being
> > >>> > processed.
> > >>> >
> > >>> > For example if Fs is 44100, then Fc values above ~7020Hz will not
> > >>> affect
> > >>> > the filter.
> > >>> >
> > >>> > Have I missed something crucial or could this a bug in the code?
> > >>> >
> > >>> > The simple IIR filter described in
> > >>> > http://en.wikipedia.org/wiki/Low-pass_filter suggests that the
> actual
> > >>> > coefficient calculation should be more like:
> > >>> >
> > >>> > a = 2π*Fc / (2π*Fc + Fs)
> > >>> >
> > >>> > Looking forward to understand this more!
> > >>> >
> > >>> > Cheers,
> > >>> > Joe
> > >>> >
> > >>> > --
> > >>> > Follow me on Twitter @diplojocus
> > >>>
> > >>> > _______________________________________________
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> > >>>
> > >>>
> > >>
> > >>
> > >> --
> > >> Follow me on Twitter @diplojocus
> > >>
> > >> _______________________________________________
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> > >>
> > >
> >
> >
> > --
> > Follow me on Twitter @diplojocus
>
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>


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