Hello list, long time again that I sneaked in here... but still working with faust and it is generally a nice experience, thanks everybody!
I am currently looking into multimode filters; we are considering building something (analog) with a multimode filter based on the design found in http://www.soundsemiconductor.com/downloads/AN701.pdf p.19 however with a feedback path to control filter resonance (as outlined on p.16) I build a preliminary emulation in faust like so [1]. However, I am not happy with the dynamics of the filter when changing its parameters. I was thinking to use the fi.svf.lp filter as a building block instead but it is (AFAICS, it is not mentioned in the doc string...) not a 1st order design (which is crucial here, partly for consistency with the to be emulated analog circuitry, partly because the phase shift should be consistent to be able to deal with the resonance feedback). I would be very much interested in other approaches, especially one's that are closely related to the analog filter topology and its numerical stability when applying dynamics to the filter freq and q. any help much appreciated! Till (I am also searching for a gentle introduction to filter theory, i.e. explanations of Laplace / z-transforms, transfer functions, (virtual) analog filter design, ... I am OK in general maths but filter theory escapes me and most of the literature I find online assumes to much preliminary knowledge or dives too fast into mathematical deeps for me to follow...) [1] a parametric multimode filter inspired by Semiconductor's AN701 #### Parameters + a_amp: amplitude of original signal added in the mixing stage * `< 0` — use table * `> 0` — use this amplitude + freq; filter frequency + q; Q-factor + idx; indicates one of the 45 modes of the filter based on Sound Semiconductor's AN701 [ 4LP, 3LP1HP, 3LP1AP, 3LP, 2LP2HP, 2LP1HP1AP, 2LP1HP, 2LP2AP, 2LP1AP, 2LP, 2LP1NT, 1LP3HP, 1LP2HP1AP, 1LP2HP, 1LP1HP2AP, 1LP1HP1AP, 1LP1HP, 1LP1HP1NT, 1LP3AP, 1LP2AP, 1LP1AP, 1LP1AP1NT, 1LP, 1LP1NT, 4HP, 3HP1AP, 3HP, 2HP2AP, 2NP1AP, 2HP, 2HP1NT, 1HP3AP, 1HP2AP, 1HP1AP, 1HP1AP1NT, 1HP, 1HP1NT, 4AP1, 3AP, 2AP, 2AP1NT, 1AP, 1AP1NT, 1NT, 2NT ] #### Usage process = _ : multimode(freq, q, idx, a_amp) * amp <: _,_ with { freq = hslider("[0]freq[scale:exp]",100,10,20000,0.001); q = hslider("[1]Q",0,0,15,0.001); idx = hslider("[2]mode",10, 0,44,1); amp = hslider("outamp",1, 0,4,0.001); a_amp = hslider("a_amp",-0.001, -0.001,1.0001,0.001); }; ``` multimode(freq, q, idx, a_amp) = _ : mix ~ _ : !, _ with { a(fbck, x) = x - (q * fbck); node = _ : lp(freq) * ba.db2linear(-3) * (-1) with { lp(freq) = fi.lowpass(1, freq); }; b(fbck, x) = a(fbck, x) : node; c(fbck, x) = b(fbck, x) : node; d(fbck, x) = c(fbck, x) : node; e(fbck, x) = d(fbck, x) : node; // factors for filter modes according to Sound Semiconductor's AN701, 45 states a_table = waveform{ 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.16666666666667, 0.11111111111111, 0.33333333333333, 1.0, 0.2, 0.5, 0.14285714285714, 0.35, 0.125, 0.33333333333333, 1.0, 1.0, 0.25, 0.75, 0.083333333333333, 0.25, 1.0, 0.5, 0.16666666666667, 0.5, 0.125 }; b_table = waveform{ 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.0, 0.33333333333333, 0.2, 0.125, 0.125, 0.33333333333333, 1.0, 0.25, 0.083333333333333, 0.25, 0.5, 0.16666666666667, 1.0, 0.5, 0.66666666666667, 0.55555555555556, 1.0, 0.0625, 0.8, 1.0, 0.57142857142857, 0.05, 0.625, 1.0, 0.066666666666667, 1.0, 0.75, 1.0, 0.5, 1.0, 0.0625, 1.0, 0.66666666666667, 1.0, 0.5 }; c_table = waveform{ 0.0, 0.0, 0.0, 0.0, 0.5, 0.33333333333333, 1.0, 0.25, 0.5, 1.0, 0.5, 1.0, 0.8, 0.625, 0.625, 1.0, 1.0, 0.75, 0.5, 1.0, 1.0, 0.66666666666667, 0.0, 1.0, 1.0, 1.0, 1.0, 0.1875, 1.0, 0.5, 1.0, 0.4, 1.0, 0.66666666666667, 0.0, 0.0, 1.0, 0.125, 1.0, 1.0, 0.25, 0.0, 1.0, 1.0, 1.0 }; d_table = waveform{ 0.0, 1.0, 0.5, 1.0, 1.0, 1.0, 1.0, 1.0, 1.0, 0.0, 1.0, 1.0, 1.0, 1.0, 1.0, 0.66666666666667, 0.0, 1.0, 1.0, 1.0, 0.0, 1.0, 0.0, 1.0, 0.66666666666667, 0.77777777777778, 0.33333333333333, 0.75, 0.4, 0.0, 0.85714285714286, 1.0, 0.5, 0.0, 0.66666666666667, 0.0, 0.5, 0.083333333333333, 0.66666666666667, 0.0, 1.0, 0.0, 0.66666666666667, 0.0, 1.0 }; e_table = waveform{ 1.0, 1.0, 1.0, 0.0, 0.5, 0.66666666666667, 0.0, 1.0, 0.0, 0.0, 1.0, 0.33333333333333, 0.4, 0.5, 0.5, 0.0, 0.0, 0.5, 0.66666666666667, 0.0, 0.0, 0.66666666666667, 0.0, 0.0, 0.16666666666667, 0.22222222222222, 0.0, 0.25, 0.0, 0.0, 0.28571428571429, 0.4, 0.0, 0.0, 0.26666666666667, 0.0, 0.0, 0.66666666666667, 0.0, 0.0, 0.5, 0.0, 0.0, 0.0, 0.5}; fac(table, i) = table, int(i): rdtable : si.smoo; fac_custom_amp(table, i, a_amp) = select2(a_amp >= 0, fac(table, i), a_amp); mix(fbck, x) = e(fbck, x), ( a(fbck, x) * fac_custom_amp(a_table, idx, a_amp) + b(fbck, x) * fac(b_table, idx) + c(fbck, x) * fac(c_table, idx) + d(fbck, x) * fac(d_table, idx) + e(fbck, x) * fac(e_table, idx) ); }; ``` -- Till Bovermann https://tai-studio.org | http://lfsaw.de | https://www.instagram.com/_lfsaw/ _______________________________________________ Faudiostream-users mailing list Faudiostream-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/faudiostream-users
