[PD] [lop~] coefficient calculation

[Frank Barknecht]

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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