bob — Runge-Kutte numerical simulation of the Moog analog resonant filter.
Bob is based on bob~ by Miller Puckette in Pure Data. The design is based on the papers by Tim Stilson, Timothy E. Stinchcombe, and Antti Huovilainen.
iosamps -- number of times of oversampling used in the filtering process. This will determine the maximum sharpness of the filter resonance (Q). More oversampling allows higher Qs, less oversampling will limit the resonance. The default is 2 times (iosamps=0).
istor -- initial disposition of internal data space. Since filtering incorporates a feedback loop of previous output, the initial status of the storage space used is significant. A zero value will clear the space; a non-zero value will allow previous information to remain. The default value is 0.
asig -- input signal
xcf -- filter cutoff frequency
xres -- filter resonance. Nominally, a value of 4 should be the limit of stability -- above that, the filter oscillates.
xsat -- saturation. This parameter determines at what signal level the "transistors" in the model saturate. The maximum output amplitude is about 2/3 of that value.
According to bob~ manual page, "By default bob~ does one step of 4th-order Runge-Kutte integration per audio sample. This works OK for resonant/cutoff frequencies up to about 1/2 Nyquist. To improve accuracy and/or to extend the range of the filter to higher cutoff frequencies you can oversample by any factor - but note that computation time rises accordingly. At high cutoff frequencies/resonance values the RK approximation can go unstable. You can combat this by raising the oversampling factor. The saturation parameter determines at what signal level the "transistors" in the model saturate. The maximum output amplitude is about 2/3 of that value".
Here is an example of the bob opcode. Play bob.csd
Example 106. Example of the bob opcode.
See the sections Real-time Audio and Command Line Flags for more information on using command line flags.
<CsoundSynthesizer> <CsOptions> ; Select audio/midi flags here according to platform -odac ;;;realtime audio out ;-iadc ;;;uncomment -iadc if realtime audio input is needed too ; For Non-realtime ouput leave only the line below: ; -o bob.wav -W ;;; for file output any platform </CsOptions> <CsInstruments> sr = 44100 ksmps = 32 nchnls = 2 0dbfs = 1 ga1 init 0 ga2 init 0 seed 777 instr 7 a0 vco2 1, cpsmidinn(p4) ; a saw wave to be filtered k1 jspline 0.7, 0.7, 1/0.7; add some jitter to filter resonance kA linseg 0, p3/17, 1, (17-2)*p3/17, 1, p3/17, 0; overal amplitude envelope k7 linseg 70, p3/3, 700, p3/3, 700, p3/3, 70; filter cutoff envelope k77 expseg sqrt(7), p3/2, 7.7, p3/2, sqrt(7); filter resonance modulation a1 bob a0 * kA, k7 - k1 * 70, k77 + k1, 3; PD bob~ ported a1 /= 7; some normalization to avoid overload aL = p5 * a1; and then panning aR = a1 * (1 - p5) outs aL, aR ga1 += aL/7; AUX send ga2 += aR/7 endin instr 77 ; What a sound without some good reverb?? a1,a2 reverbsc ga1,ga2, 0.97+.1/7, 7777 outs a1, a2 ga1 = 0 ga2 = 0 endin </CsInstruments> <CsScore> i 7 0 30 70 0.7; Perfect 7th i 7 0 30 77 0.7 i 77 0 40 ; In text editor try Find and Replace to substitute all 7 numbers for 3...8 ) ; all 8s is a bit too high, all 3s is a bit too low, 4 sounds great! ; Do such replacement in initial 7 file, otherwise avoid accidental changes in the header... ; Tired of numerology? Try this: ; i 7 0 30 60 0.6 ; i 7 0 32 45 0.4 e </CsScore> </CsoundSynthesizer>