[PD] [lop~] coefficient calculation

[Joe White]

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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>


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