[PD] [lop~] coefficient calculation

[Chris Clepper]

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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