[PD] [lop~] coefficient calculation

Sat Jul 19 05:57:53 CEST 2014

It's been noted that [vcf~] can be obtained with a [cpole~] object - though
I didn't do it yet as I find it a bit hard to get to the coeficients from
the vcf~ code.

Anyway, if you can get it with a [cpole~] object, it means you could do it
with [biquad~] coeficients, right? I suppose so, but then, how exactly? I
mean, if I have the coeficients of [cpole~], how do I get to [biquad~]'s?

maybe this will help getting biquad's coefficients from [vcf~]'s
parameters... maybe not, I don't know, I need help :)

thanks

2014-07-19 0:44 GMT-03:00 Alexandre Torres Porres <porres at gmail.com>:

> hi all, I've been working on filter patches for my courses and I'm still
> failing ti get biquad coeficients from the [vcf~] code. Maybe anyone out
> there could help?
>
> I wanted this to plot the frequency response in realtime...
>
> the [vcf~] filters aren't in the audio Audio-EQ-Cookbook, and the code
> looks a bit too complicated
>
> thanks
>
>
> 2014-05-26 10:23 GMT-03:00 Joe White <white.joe4 at gmail.com>:
>
> 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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>>
>>
>> --
>> Follow me on Twitter @diplojocus
>>
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