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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