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Diffstat (limited to 'BUG8/BUG8.csd')
| -rw-r--r-- | BUG8/BUG8.csd | 313 |
1 files changed, 313 insertions, 0 deletions
diff --git a/BUG8/BUG8.csd b/BUG8/BUG8.csd new file mode 100644 index 0000000..5f27a86 --- /dev/null +++ b/BUG8/BUG8.csd @@ -0,0 +1,313 @@ +<CsoundSynthesizer> +<CsOptions> +-odac +</CsOptions> +<CsInstruments> +sr = 44100 +kr = 441 +nchnls = 2 +0dbfs = 1 +seed 0 + +#include "wavetables.udo" + +giampfn init -1 +gifreqfn init -1 +gifftsize = 1024 + + +opcode SCReverb, aa, aakkjpo + + setksmps 1 + + ainL, ainR, kFeedBack, kLPFreq, iSRate, iPitchMod, iSkip xin + iSRate = (iSRate <= 0 ? 1 : (sr / iSRate)) + if (iSkip != 0) igoto SkipInit + + afilt1 init 0 + afilt2 init 0 + afilt3 init 0 + afilt4 init 0 + afilt5 init 0 + afilt6 init 0 + afilt7 init 0 + afilt8 init 0 + + ; Delay times chosen to be prime numbers. + ; Works with sr=44100 ONLY. If you wish to + ; use a different delay time, find some new + ; prime numbers that will give roughly the + ; same delay times for the new sampling rate. + ; Or adjust to taste. + + idel1 = 2473.000 / (iSRate * 44100) + idel2 = 2767.000 / (iSRate * 44100) + idel3 = 3217.000 / (iSRate * 44100) + idel4 = 3557.000 / (iSRate * 44100) + idel5 = 3907.000 / (iSRate * 44100) + idel6 = 4127.000 / (iSRate * 44100) + idel7 = 2143.000 / (iSRate * 44100) + idel8 = 1933.000 / (iSRate * 44100) + + ; kFeedBack: + ; gain of reverb. Adjust empirically + ; for desired reverb time. .6 gives + ; a good small \\\"live\\\" room sound, .8 + ; a small hall, .9 a large hall, + ; .99 an enormous stone cavern. + + ; iPitchMod: + ; amount of random pitch modulation + ; for the delay lines. 1 is the \\\"normal\\\" + ; amount, but this may be too high for + ; held pitches such as piano tones. + ; Adjust to taste. + + ; kLPFreq: + ; Cutoff frequency of lowpass filters + ; in feedback loops of delay lines, + ; in Hz. Lower cutoff frequencies results + ; in a sound with more high-frequency + ; damping. + + ; k1-k8 are used to add random pitch modulation to the + ; delay lines. Helps eliminate metallic overtones + ; in the reverb sound. + + iPitchMod = iPitchMod / iSRate + + k1 randi 0.0010 * iPitchMod, 3.100 * iSRate, 0.06 + k2 randi 0.0011 * iPitchMod, 3.500 * iSRate, 0.90 + k3 randi 0.0017 * iPitchMod, 1.110 * iSRate, 0.70 + k4 randi 0.0006 * iPitchMod, 3.973 * iSRate, 0.30 + k5 randi 0.0010 * iPitchMod, 2.341 * iSRate, 0.63 + k6 randi 0.0011 * iPitchMod, 1.897 * iSRate, 0.70 + k7 randi 0.0017 * iPitchMod, 0.891 * iSRate, 0.90 + k8 randi 0.0006 * iPitchMod, 3.221 * iSRate, 0.44 + + ; apj is used to calculate \\\"resultant junction pressure\\\" for + ; the scattering junction of 8 lossless waveguides + ; of equal characteristic impedance. If you wish to + ; add more delay lines, simply add them to the following + ; equation, and replace the .25 by 2/N, where N is the + ; number of delay lines. + + apj = .25 * (afilt1+afilt2+afilt3+afilt4+afilt5+afilt6+afilt7+afilt8) + + adum1 delayr 1 + adel1 deltap3 idel1 + k1 + delayw ainL + apj - afilt1 + + adum2 delayr 1 + adel2 deltap3 idel2 + k2 + delayw ainR + apj - afilt2 + + adum3 delayr 1 + adel3 deltap3 idel3 + k3 + delayw ainL + apj - afilt3 + + adum4 delayr 1 + adel4 deltap3 idel4 + k4 + delayw ainR + apj - afilt4 + + adum5 delayr 1 + adel5 deltap3 idel5 + k5 + delayw ainL + apj - afilt5 + + adum6 delayr 1 + adel6 deltap3 idel6 + k6 + delayw ainR + apj - afilt6 + + adum7 delayr 1 + adel7 deltap3 idel7 + k7 + delayw ainL + apj - afilt7 + + adum8 delayr 1 + adel8 deltap3 idel8 + k8 + delayw ainR + apj - afilt8 + + ; 1st order lowpass filters in feedback + ; loops of delay lines. + + kLPFreq = kLPFreq * iSRate + + afilt1 tone adel1 * kFeedBack, kLPFreq + afilt2 tone adel2 * kFeedBack, kLPFreq + afilt3 tone adel3 * kFeedBack, kLPFreq + afilt4 tone adel4 * kFeedBack, kLPFreq + afilt5 tone adel5 * kFeedBack, kLPFreq + afilt6 tone adel6 * kFeedBack, kLPFreq + afilt7 tone adel7 * kFeedBack, kLPFreq + afilt8 tone adel8 * kFeedBack, kLPFreq + + ; The outputs of the delay lines are summed + ; and sent to the stereo outputs. This could + ; easily be modified for a 4 or 8-channel + ; sound system. + + aoutL = 0.35 * (afilt1 + afilt3 + afilt5 + afilt7) + aoutR = 0.35 * (afilt2 + afilt4 + afilt6 + afilt8) + + SkipInit: + xout aoutL, aoutR + endop + + +instr liner + index = p4 + isinit = p5 + iampscale = (((gifftsize / 2) - index) / (gifftsize / 2)) * 0.1 + isamp = (isinit == 1) ? random(0, iampscale) : tab_i(index, giampfn) + isfreq = (isinit == 1) ? 440 : tab_i(index, gifreqfn) + tabw oscil:k(1, random(0.01, 1), -1, random(0, 1)) * linseg(isamp, p3, random(0, iampscale)), index, giampfn + tabw linseg(isfreq, p3, random(50, 1900)), index, gifreqfn + +endin + +instr rndoise + ipanrange1 = p4 + ipanrange2 = p5 + ifreqmax = random(100, 1000) + if (random(0, 1) > 0.3) then + a1 oscil line(random(0.5, 1), p3, random(0.5, 1)), random(100, ifreqmax) + else + a1 noise line(random(0.5, 1), p3, random(0.5, 1)), random(-1, 1) + endif + + if (random(0, 1) > 0.6) then + a1 *= oscil(line(random(0.5, 1), p3, random(0.5, 1)), random(100, ifreqmax)) + endif + + if (random(0, 1) > 0.9) then + a1 *= noise(line(random(0.5, 1), p3, random(0.5, 1)), random(-1, 1)) + endif + + if (random(0, 1) > 0.6) then + a1 *= oscil(random:k(0.5, 1), random(100, ifreqmax)) + endif + + if (random(0, 1) > 0.9) then + a1 *= noise(random:k(0.5, 1), random(-1, 1)) + endif + + if (random(0, 1) > 0.6) then + a1 taninv a1 * line(random(1, 10), p3, random(1, 10)) + endif + + if (random(0, 1) > 0.6) then + a1 butterhp a1, line(random(20, 2000), p3, random(20, 2000)) + endif + + if (random(0, 1) > 0.6) then + a1 butterlp a1, line(random(500, 16000), p3, random(500, 16000)) + endif + + if (random(0, 1) > 0.5) then + a1 *= oscil(1, random(1, 50)) + else + a1 *= line(random(0, 1), p3, random(0, 1)) + endif + + if (random(0, 1) > 0.6) then + a1 waveset a1, random(1, 20) + endif + + a1 *= 0.5 + aL, aR pan2 a1, ipanrange1, ipanrange2 + outs aL, aR +endin + + +instr trigrndoise + itime = 0 + imax = random(5, 20) + while (itime < p3) do + ilen = (p3 / imax) * random(0.6, 1.2) + event_i "i", "rndoise", itime, ilen, random(0, 1), random(0, 1) + itime += ilen * random(0.8, 1.1) + od +endin + + +instr clicker + aL noise 0.4, 0.4 + aR noise 0.4, 0.4 + kamp linseg 1, p3, 0 + outs aL*kamp, aR*kamp +endin + + +instr clickmaker + idur random 0.001, 0.01 + kmetro metro random(1, 6) + if (kmetro == 1) then + event "i", "clicker", 0, idur + endif +endin + +instr 1 + ktime timeinsts + isize = gifftsize / 2 + giampfn ftgen 0, 0, isize, 7, 0 + gifreqfn ftgen 0, 0, isize, 7, 0 + + itime = 60 + kmetro metro 1/itime + kinit init 1 + if (kmetro == 1) then + kndex = 0 + while (kndex < isize) do + event "i", "liner", 0, itime, kndex, kinit + kndex += 1 + od + if (kinit == 1) then + kinit = 0 + endif + endif + + event_i "i", "clickmaker", 0, 1 + + kmetro2 metro 3 + if (ktime > 5 && kmetro2 == 1 && random:k(0, 1) > 0.95) then + event "i", "trigrndoise", 0, random:k(0.6, 2) + endif + + ; pvsinit not working here.. + f1 pvsanal oscils(1, 440, 0), gifftsize, gifftsize/4, gifftsize, 1 + pvsftr f1, giampfn, gifreqfn + f2L pvscale f1, 0.03 + oscil(0.01, 0.01) + f2R pvscale f1, 0.03 + oscil(0.01, 0.02, -1, 0.26) + + a2L pvsynth f2L + a2R pvsynth f2R + a2L pareq a2L, 200, 0.3, 0.6 + a2R pareq a2R, 200, 0.3, 0.6 + + k2mul = 3 * (abs(oscil(0.7, 0.1)) + 0.2) + a2L *= k2mul + a2R *= k2mul + + ay pvsynth f1 + ay *= abs(oscil(0.7, 0.001, -1, 0)) + ;ay exciter ay, 3000, 20000, 10, 10 + + a1L = ay + a1R = ay + ;a1L, a1R SCReverb ay, ay, 0.9, 10000 + aL = a1L + a2L + aR = a1R + a2R + + outs aL, aR +endin + + + + +</CsInstruments> +<CsScore> + + +i1 0 100 + +</CsScore> +</CsoundSynthesizer>
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