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

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