Subtractive synthesis is, at least conceptually, the converse of additive synthesis in that instead of building complex sound through the addition of simple cellular materials such as sine waves, subtractive synthesis begins with a complex sound source, such as white noise or a recorded sample, or a rich waveform, such as a sawtooth or pulse, and proceeds to refine that sound by removing partials or entire sections of the frequency spectrum through the use of audio filters.
The creation of dynamic spectra (an arduous task in additive synthesis) is relatively simple in subtractive synthesis as all that will be required will be to modulate a few parameters pertaining to any filters being used. Working with the intricate precision that is possible with additive synthesis may not be as easy with subtractive synthesis but sounds can be created much more instinctively than is possible with additive or FM synthesis.
The first example represents perhaps the classical idea of subtractive synthesis: a simple two oscillator synth filtered using a single resonant lowpass filter. Many of the ideas used in this example have been inspired by the design of the Minimoog synthesizer (1970) and other similar instruments.
Each oscillator can describe either a sawtooth, PWM waveform (i.e. square - pulse etc.) or white noise and each oscillator can be transposed in octaves or in cents with respect to a fundamental pitch. The two oscillators are mixed and then passed through a 4-pole / 24dB per octave resonant lowpass filter. The opcode 'moogladder' is chosen on account of its authentic vintage character. The cutoff frequency of the filter is modulated using an ADSR-style (attack-decay-sustain-release) envelope facilitating the creation of dynamic, evolving spectra. Finally the sound output of the filter is shaped by an ADSR amplitude envelope.
As this instrument is suggestive of a performance instrument controlled via MIDI, this has been partially implemented. Through the use of Csound's MIDI interoperability opcode, mididefault, the instrument can be operated from the score or from a MIDI keyboard. If a MIDI note is received, suitable default p-field values are substituted for the missing p-fields. MIDI controller 1 can be used to control the global cutoff frequency for the filter.
A schematic for this instrument is shown below:
<CsoundSynthesizer>
<CsOptions>
-odevaudio -b512 -Ma
</CsOptions>
<CsInstruments>
sr = 44100
ksmps = 4
nchnls = 2
0dbfs = 1
initc7 1,1,0.8 ;set initial controller position
prealloc 1, 10
instr 1
iNum notnum ;read in midi note number
iCF ctrl7 1,1,0.1,14 ;read in midi controller 1
; set up default p-field values for midi activated notes
mididefault iNum, p4 ;pitch (note number)
mididefault 0.3, p5 ;amplitude 1
mididefault 2, p6 ;type 1
mididefault 0.5, p7 ;pulse width 1
mididefault 0, p8 ;octave disp. 1
mididefault 0, p9 ;tuning disp. 1
mididefault 0.3, p10 ;amplitude 2
mididefault 1, p11 ;type 2
mididefault 0.5, p12 ;pulse width 2
mididefault -1, p13 ;octave displacement 2
mididefault 20, p14 ;tuning disp. 2
mididefault iCF, p15 ;filter cutoff freq
mididefault 0.01, p16 ;filter env. attack time
mididefault 1, p17 ;filter env. decay time
mididefault 0.01, p18 ;filter env. sustain level
mididefault 0.1, p19 ;filter release time
mididefault 0.3, p20 ;filter resonance
mididefault 0.01, p21 ;amp. env. attack
mididefault 0.1, p22 ;amp. env. decay.
mididefault 1, p23 ;amp. env. sustain
mididefault 0.01, p24 ;amp. env. release
; asign p-fields to variables
iCPS = cpsmidinn(p4) ;convert from note number to cps
kAmp1 = p5
iType1 = p6
kPW1 = p7
kOct1 = octave(p8) ;convert from octave displacement to multiplier
kTune1 = cent(p9) ;convert from cents displacement to multiplier
kAmp2 = p10
iType2 = p11
kPW2 = p12
kOct2 = octave(p13)
kTune2 = cent(p14)
iCF = p15
iFAtt = p16
iFDec = p17
iFSus = p18
iFRel = p19
kRes = p20
iAAtt = p21
iADec = p22
iASus = p23
iARel = p24
;oscillator 1
if iType1==1||iType1==2 then ;if type is sawtooth or square...
iMode1 = (iType1=1?0:2) ;...derive vco2 'mode' from waveform type
aSig1 vco2 kAmp1,iCPS*kOct1*kTune1,iMode1,kPW1;VCO audio oscillator
else ;otherwise...
aSig1 noise kAmp1, 0.5 ;...generate white noise
endif
;oscillator 2 - identical to oscillator 1
if iType2==1||iType2==2 then
iMode2 = (iType2=1?0:2)
aSig2 vco2 kAmp2,iCPS*kOct2*kTune2,iMode2,kPW2
else
aSig2 noise kAmp2,0.5
endif
;mix oscillators
aMix sum aSig1,aSig2
;lowpass filter
kFiltEnv expsegr 0.0001,iFAtt,iCPS*iCF,iFDec,iCPS*iCF*iFSus,iFRel,0.0001
aOut moogladder aMix, kFiltEnv, kRes
;amplitude envelope
aAmpEnv expsegr 0.0001,iAAtt,1,iADec,iASus,iARel,0.0001
aOut = aOut*aAmpEnv
outs aOut,aOut
endin
</CsInstruments>
<CsScore>
;p4 = oscillator frequency
;oscillator 1
;p5 = amplitude
;p6 = type (1=sawtooth,2=square-PWM,3=noise)
;p7 = PWM (square wave only)
;p8 = octave displacement
;p9 = tuning displacement (cents)
;oscillator 2
;p10 = amplitude
;p11 = type (1=sawtooth,2=square-PWM,3=noise)
;p12 = pwm (square wave only)
;p13 = octave displacement
;p14 = tuning displacement (cents)
;global filter envelope
;p15 = cutoff
;p16 = attack time
;p17 = decay time
;p18 = sustain level (fraction of cutoff)
;p19 = release time
;p20 = resonance
;global amplitude envelope
;p21 = attack time
;p22 = decay time
;p23 = sustain level
;p24 = release time
; p1 p2 p3 p4 p5 p6 p7 p8 p9 p10 p11 p12 p13 p14 p15 p16 p17 p18 p19 p20 p21 p22 p23 p24
i 1 0 1 50 0 2 .5 0 -5 0 2 0.5 0 5 12 .01 2 .01 .1 0 .005 .01 1 .05
i 1 + 1 50 .2 2 .5 0 -5 .2 2 0.5 0 5 1 .01 1 .1 .1 .5 .005 .01 1 .05
i 1 + 1 50 .2 2 .5 0 -8 .2 2 0.5 0 8 3 .01 1 .1 .1 .5 .005 .01 1 .05
i 1 + 1 50 .2 2 .5 0 -8 .2 2 0.5 -1 8 7 .01 1 .1 .1 .5 .005 .01 1 .05
i 1 + 3 50 .2 1 .5 0 -10 .2 1 0.5 -2 10 40 .01 3 .001 .1 .5 .005 .01 1 .05
i 1 + 10 50 1 2 .01 -2 0 .2 3 0.5 0 0 40 5 5 .001 1.5 .1 .005 .01 1 .05
f 0 3600
e
</CsScore>
</CsoundSynthesizer
The next example makes extensive use of bandpass filters arranged in parallel to filter white noise. The bandpass filter bandwidths are narrowed to the point where almost pure tones are audible. The crucial difference is that the noise source always induces instability in the amplitude and frequency of tones produced - it is this quality that makes this sort of subtractive synthesis sound much more organic than an additive synthesis equivalent. If the bandwidths are widened then more of the characteristic of the noise source comes through and the tone becomes 'airier' and less distinct; if the bandwidths are narrowed the resonating tones become clearer and steadier. By varying the bandwidths interesting metamorphoses of the resultant sound are possible.
22 reson filters are used for the bandpass filters on account of their ability to ring and resonate as their bandwidth narrows. Another reason for this choice is the relative CPU economy of the reson filter, a not inconsiderable concern as so many of them are used. The frequency ratios between the 22 parallel filters are derived from analysis of a hand bell, the data was found in the appendix of the Csound manual here.
In addition to the white noise as a source, noise impulses are also used as a sound source (via the 'mpulse' opcode). The instrument will automatically and randomly slowly crossfade between these two sound sources.
A lowpass and highpass filter are inserted in series before the parallel bandpass filters to shape the frequency spectrum of the source sound. Csound's butterworth filters butlp and buthp are chosen for this task on account of their steep cutoff slopes and lack of ripple at the cutoff point.
The outputs of the reson filters are sent alternately to the left and right outputs in order to create a broad stereo effect.
This example makes extensive use of the 'rspline' opcode, a generator of random spline functions, to slowly undulate the many input parameters. The orchestra is self generative in that instrument 1 repeatedly triggers note events in instrument 2 and the extensive use of random functions means that the results will continually evolve as the orchestra is allowed to perform.
A flow diagram for this instrument is shown below:
<CsoundSynthesizer>
<CsOptions>
-odevaudio -b512 -dm0
</CsOptions>
<CsInstruments>
;Example written by Iain McCurdy
sr = 44100
ksmps = 16
nchnls = 2
0dbfs = 1
instr 1 ; triggers notes in instrument 2 with randomised p-fields
krate randomi 0.2,0.4,0.1 ;rate of note generation
ktrig metro krate ;triggers used by schedkwhen
koct random 5,12 ;fundemental pitch of synth note
kdur random 15,30 ;duration of note
schedkwhen ktrig,0,0,2,0,kdur,cpsoct(koct) ;trigger a note in instrument 2
endin
instr 2 ; subtractive synthesis instrument
aNoise pinkish 1 ;a noise source sound: pink noise
kGap rspline 0.3,0.05,0.2,2 ;time gap between impulses
aPulse mpulse 15, kGap ;a train of impulses
kCFade rspline 0,1,0.1,1 ;crossfade point between noise and impulses
aInput ntrpol aPulse,aNoise,kCFade ;implement crossfade
; cutoff frequencies for low and highpass filters
kLPF_CF rspline 13,8,0.1,0.4
kHPF_CF rspline 5,10,0.1,0.4
; filter input sound with low and highpass filters in series -
; - done twice per filter in order to sharpen cutoff slopes
aInput butlp aInput, cpsoct(kLPF_CF)
aInput butlp aInput, cpsoct(kLPF_CF)
aInput buthp aInput, cpsoct(kHPF_CF)
aInput buthp aInput, cpsoct(kHPF_CF)
kcf rspline p4*1.05,p4*0.95,0.01,0.1 ; fundemental
; bandwidth for each filter is created individually as a random spline function
kbw1 rspline 0.00001,10,0.2,1
kbw2 rspline 0.00001,10,0.2,1
kbw3 rspline 0.00001,10,0.2,1
kbw4 rspline 0.00001,10,0.2,1
kbw5 rspline 0.00001,10,0.2,1
kbw6 rspline 0.00001,10,0.2,1
kbw7 rspline 0.00001,10,0.2,1
kbw8 rspline 0.00001,10,0.2,1
kbw9 rspline 0.00001,10,0.2,1
kbw10 rspline 0.00001,10,0.2,1
kbw11 rspline 0.00001,10,0.2,1
kbw12 rspline 0.00001,10,0.2,1
kbw13 rspline 0.00001,10,0.2,1
kbw14 rspline 0.00001,10,0.2,1
kbw15 rspline 0.00001,10,0.2,1
kbw16 rspline 0.00001,10,0.2,1
kbw17 rspline 0.00001,10,0.2,1
kbw18 rspline 0.00001,10,0.2,1
kbw19 rspline 0.00001,10,0.2,1
kbw20 rspline 0.00001,10,0.2,1
kbw21 rspline 0.00001,10,0.2,1
kbw22 rspline 0.00001,10,0.2,1
imode = 0 ; amplitude balancing method used by the reson filters
a1 reson aInput, kcf*1, kbw1, imode
a2 reson aInput, kcf*1.0019054878049, kbw2, imode
a3 reson aInput, kcf*1.7936737804878, kbw3, imode
a4 reson aInput, kcf*1.8009908536585, kbw4, imode
a5 reson aInput, kcf*2.5201981707317, kbw5, imode
a6 reson aInput, kcf*2.5224085365854, kbw6, imode
a7 reson aInput, kcf*2.9907012195122, kbw7, imode
a8 reson aInput, kcf*2.9940548780488, kbw8, imode
a9 reson aInput, kcf*3.7855182926829, kbw9, imode
a10 reson aInput, kcf*3.8061737804878, kbw10,imode
a11 reson aInput, kcf*4.5689024390244, kbw11,imode
a12 reson aInput, kcf*4.5754573170732, kbw12,imode
a13 reson aInput, kcf*5.0296493902439, kbw13,imode
a14 reson aInput, kcf*5.0455030487805, kbw14,imode
a15 reson aInput, kcf*6.0759908536585, kbw15,imode
a16 reson aInput, kcf*5.9094512195122, kbw16,imode
a17 reson aInput, kcf*6.4124237804878, kbw17,imode
a18 reson aInput, kcf*6.4430640243902, kbw18,imode
a19 reson aInput, kcf*7.0826219512195, kbw19,imode
a20 reson aInput, kcf*7.0923780487805, kbw20,imode
a21 reson aInput, kcf*7.3188262195122, kbw21,imode
a22 reson aInput, kcf*7.5551829268293, kbw22,imode
; amplitude control for each filter output
kAmp1 rspline 0, 1, 0.3, 1
kAmp2 rspline 0, 1, 0.3, 1
kAmp3 rspline 0, 1, 0.3, 1
kAmp4 rspline 0, 1, 0.3, 1
kAmp5 rspline 0, 1, 0.3, 1
kAmp6 rspline 0, 1, 0.3, 1
kAmp7 rspline 0, 1, 0.3, 1
kAmp8 rspline 0, 1, 0.3, 1
kAmp9 rspline 0, 1, 0.3, 1
kAmp10 rspline 0, 1, 0.3, 1
kAmp11 rspline 0, 1, 0.3, 1
kAmp12 rspline 0, 1, 0.3, 1
kAmp13 rspline 0, 1, 0.3, 1
kAmp14 rspline 0, 1, 0.3, 1
kAmp15 rspline 0, 1, 0.3, 1
kAmp16 rspline 0, 1, 0.3, 1
kAmp17 rspline 0, 1, 0.3, 1
kAmp18 rspline 0, 1, 0.3, 1
kAmp19 rspline 0, 1, 0.3, 1
kAmp20 rspline 0, 1, 0.3, 1
kAmp21 rspline 0, 1, 0.3, 1
kAmp22 rspline 0, 1, 0.3, 1
; left and right channel mixes are created using alternate filter outputs.
; This shall create a stereo effect.
aMixL sum a1*kAmp1,a3*kAmp3,a5*kAmp5,a7*kAmp7,a9*kAmp9,a11*kAmp11,\
a13*kAmp13,a15*kAmp15,a17*kAmp17,a19*kAmp19,a21*kAmp21,
aMixR sum a2*kAmp2,a4*kAmp4,a6*kAmp6,a8*kAmp8,a10*kAmp10,a12*kAmp12,\
a14*kAmp14,a16*kAmp16,a18*kAmp18,a20*kAmp20,a22*kAmp22
kEnv linseg 0, p3*0.5, 1,p3*0.5,0,1,0 ; global amplitude envelope
outs (aMixL*kEnv*0.00002), (aMixR*kEnv*0.00002) ; audio sent to outputs
endin
</CsInstruments>
<CsScore>
i 1 0 3600 ; instrument 1 (note generator) plays for 1 hour
e
</CsScore>
</CsoundSynthesizer>
The final example in this section uses precisely tuned bandpass filters, to simulate the sound of the human voice expressing vowel sounds. Spectral resonances in this context are often referred to as 'formants'. Five formants are used to simulate the effect of the human mouth and head as a resonating (and therefore filtering) body. The filter data for simulating the vowel sounds A,E,I,O and U as expressed by a bass, tenor, counter-tenor, alto and soprano voice were found in the appendix of the Csound manual here. Bandwidth and intensity (dB) information is also needed to accurately simulate the various vowel sounds.
reson filters are again used but butbp and others could be equally valid choices.
Data is stored in GEN07 linear break point function tables, as this data is read by k-rate line functions we can interpolate and therefore morph between different vowel sounds during a note.
The source sound for the filters comes from either a pink noise generator or a pulse waveform. The pink noise source could be used if the emulation is to be that of just the breath whereas the pulse waveform provides a decent approximation of the human vocal chords buzzing. This instrument can however morph continuously between these two sources.
A flow diagram for this instrument is shown below:
<CsoundSynthesizer>
<CsOptions>
-odevaudio -b512 -dm0
</CsOptions>
<CsInstruments>
;example by Iain McCurdy
sr = 44100
ksmps = 16
nchnls = 2
0dbfs = 1
instr 1
kFund expon p4,p3,p5 ; fundemental
kVow line p6,p3,p7 ; vowel select
kBW line p8,p3,p9 ; bandwidth factor
iVoice = p10 ; voice select
kSrc line p11,p3,p12 ; source mix
aNoise pinkish 3 ; pink noise
aVCO vco2 1.2,kFund,2,0.02 ; pulse tone
aInput ntrpol aVCO,aNoise,kSrc ; input mix
; read formant cutoff frequenies from tables
kCF1 table kVow,1+(iVoice*15),1
kCF2 table kVow,2+(iVoice*15),1
kCF3 table kVow,3+(iVoice*15),1
kCF4 table kVow,4+(iVoice*15),1
kCF5 table kVow,5+(iVoice*15),1
; read formant intensity values from tables
kDB1 table kVow,6+(iVoice*15),1
kDB2 table kVow,7+(iVoice*15),1
kDB3 table kVow,8+(iVoice*15),1
kDB4 table kVow,9+(iVoice*15),1
kDB5 table kVow,10+(iVoice*15),1
; read formant bandwidths from tables
kBW1 table kVow,11+(iVoice*15),1
kBW2 table kVow,12+(iVoice*15),1
kBW3 table kVow,13+(iVoice*15),1
kBW4 table kVow,14+(iVoice*15),1
kBW5 table kVow,15+(iVoice*15),1
; create resonant formants byt filtering source sound
aForm1 reson aInput, kCF1, kBW1*kBW, 1 ; formant 1
aForm2 reson aInput, kCF2, kBW2*kBW, 1 ; formant 2
aForm3 reson aInput, kCF3, kBW3*kBW, 1 ; formant 3
aForm4 reson aInput, kCF4, kBW4*kBW, 1 ; formant 4
aForm5 reson aInput, kCF5, kBW5*kBW, 1 ; formant 5
; formants are mixed and multiplied both by intensity values derived from tables and by the on-screen gain controls for each formant
aMix sum aForm1*ampdbfs(kDB1),aForm2*ampdbfs(kDB2),aForm3*ampdbfs(kDB3),aForm4*ampdbfs(kDB4),aForm5*ampdbfs(kDB5)
kEnv linseg 0,3,1,p3-6,1,3,0 ; an amplitude envelope
outs aMix*kEnv, aMix*kEnv ; send audio to outputs
endin
</CsInstruments>
<CsScore>
f 0 3600 ;DUMMY SCORE EVENT - PERMITS REALTIME PERFORMANCE FOR UP TO 1 HOUR
;FUNCTION TABLES STORING FORMANT DATA FOR EACH OF THE FIVE VOICE TYPES REPRESENTED
;BASS
f 1 0 32768 -7 600 10922 400 10922 250 10924 350 ;FREQ
f 2 0 32768 -7 1040 10922 1620 10922 1750 10924 600 ;FREQ
f 3 0 32768 -7 2250 10922 2400 10922 2600 10924 2400 ;FREQ
f 4 0 32768 -7 2450 10922 2800 10922 3050 10924 2675 ;FREQ
f 5 0 32768 -7 2750 10922 3100 10922 3340 10924 2950 ;FREQ
f 6 0 32768 -7 0 10922 0 10922 0 10924 0 ;dB
f 7 0 32768 -7 -7 10922 -12 10922 -30 10924 -20 ;dB
f 8 0 32768 -7 -9 10922 -9 10922 -16 10924 -32 ;dB
f 9 0 32768 -7 -9 10922 -12 10922 -22 10924 -28 ;dB
f 10 0 32768 -7 -20 10922 -18 10922 -28 10924 -36 ;dB
f 11 0 32768 -7 60 10922 40 10922 60 10924 40 ;BAND WIDTH
f 12 0 32768 -7 70 10922 80 10922 90 10924 80 ;BAND WIDTH
f 13 0 32768 -7 110 10922 100 10922 100 10924 100 ;BAND WIDTH
f 14 0 32768 -7 120 10922 120 10922 120 10924 120 ;BAND WIDTH
f 15 0 32768 -7 130 10922 120 10922 120 10924 120 ;BAND WIDTH
;TENOR
f 16 0 32768 -7 650 8192 400 8192 290 8192 400 8192 350 ;FREQ
f 17 0 32768 -7 1080 8192 1700 8192 1870 8192 800 8192 600 ;FREQ
f 18 0 32768 -7 2650 8192 2600 8192 2800 8192 2600 8192 2700 ;FREQ
f 19 0 32768 -7 2900 8192 3200 8192 3250 8192 2800 8192 2900 ;FREQ
f 20 0 32768 -7 3250 8192 3580 8192 3540 8192 3000 8192 3300 ;FREQ
f 21 0 32768 -7 0 8192 0 8192 0 8192 0 8192 0 ;dB
f 22 0 32768 -7 -6 8192 -14 8192 -15 8192 -10 8192 -20 ;dB
f 23 0 32768 -7 -7 8192 -12 8192 -18 8192 -12 8192 -17 ;dB
f 24 0 32768 -7 -8 8192 -14 8192 -20 8192 -12 8192 -14 ;dB
f 25 0 32768 -7 -22 8192 -20 8192 -30 8192 -26 8192 -26 ;dB
f 26 0 32768 -7 80 8192 70 8192 40 8192 40 8192 40 ;BAND WIDTH
f 27 0 32768 -7 90 8192 80 8192 90 8192 80 8192 60 ;BAND WIDTH
f 28 0 32768 -7 120 8192 100 8192 100 8192 100 8192 100 ;BAND WIDTH
f 29 0 32768 -7 130 8192 120 8192 120 8192 120 8192 120 ;BAND WIDTH
f 30 0 32768 -7 140 8192 120 8192 120 8192 120 8192 120 ;BAND WIDTH
;COUNTER TENOR
f 31 0 32768 -7 660 8192 440 8192 270 8192 430 8192 370 ;FREQ
f 32 0 32768 -7 1120 8192 1800 8192 1850 8192 820 8192 630 ;FREQ
f 33 0 32768 -7 2750 8192 2700 8192 2900 8192 2700 8192 2750 ;FREQ
f 34 0 32768 -7 3000 8192 3000 8192 3350 8192 3000 8192 3000 ;FREQ
f 35 0 32768 -7 3350 8192 3300 8192 3590 8192 3300 8192 3400 ;FREQ
f 36 0 32768 -7 0 8192 0 8192 0 8192 0 8192 0 ;dB
f 37 0 32768 -7 -6 8192 -14 8192 -24 8192 -10 8192 -20 ;dB
f 38 0 32768 -7 -23 8192 -18 8192 -24 8192 -26 8192 -23 ;dB
f 39 0 32768 -7 -24 8192 -20 8192 -36 8192 -22 8192 -30 ;dB
f 40 0 32768 -7 -38 8192 -20 8192 -36 8192 -34 8192 -30 ;dB
f 41 0 32768 -7 80 8192 70 8192 40 8192 40 8192 40 ;BAND WIDTH
f 42 0 32768 -7 90 8192 80 8192 90 8192 80 8192 60 ;BAND WIDTH
f 43 0 32768 -7 120 8192 100 8192 100 8192 100 8192 100 ;BAND WIDTH
f 44 0 32768 -7 130 8192 120 8192 120 8192 120 8192 120 ;BAND WIDTH
f 45 0 32768 -7 140 8192 120 8192 120 8192 120 8192 120 ;BAND WIDTH
;ALTO
f 46 0 32768 -7 800 8192 400 8192 350 8192 450 8192 325 ;FREQ
f 47 0 32768 -7 1150 8192 1600 8192 1700 8192 800 8192 700 ;FREQ
f 48 0 32768 -7 2800 8192 2700 8192 2700 8192 2830 8192 2530 ;FREQ
f 49 0 32768 -7 3500 8192 3300 8192 3700 8192 3500 8192 2500 ;FREQ
f 50 0 32768 -7 4950 8192 4950 8192 4950 8192 4950 8192 4950 ;FREQ
f 51 0 32768 -7 0 8192 0 8192 0 8192 0 8192 0 ;dB
f 52 0 32768 -7 -4 8192 -24 8192 -20 8192 -9 8192 -12 ;dB
f 53 0 32768 -7 -20 8192 -30 8192 -30 8192 -16 8192 -30 ;dB
f 54 0 32768 -7 -36 8192 -35 8192 -36 8192 -28 8192 -40 ;dB
f 55 0 32768 -7 -60 8192 -60 8192 -60 8192 -55 8192 -64 ;dB
f 56 0 32768 -7 50 8192 60 8192 50 8192 70 8192 50 ;BAND WIDTH
f 57 0 32768 -7 60 8192 80 8192 100 8192 80 8192 60 ;BAND WIDTH
f 58 0 32768 -7 170 8192 120 8192 120 8192 100 8192 170 ;BAND WIDTH
f 59 0 32768 -7 180 8192 150 8192 150 8192 130 8192 180 ;BAND WIDTH
f 60 0 32768 -7 200 8192 200 8192 200 8192 135 8192 200 ;BAND WIDTH
;SOPRANO
f 61 0 32768 -7 800 8192 350 8192 270 8192 450 8192 325 ;FREQ
f 62 0 32768 -7 1150 8192 2000 8192 2140 8192 800 8192 700 ;FREQ
f 63 0 32768 -7 2900 8192 2800 8192 2950 8192 2830 8192 2700 ;FREQ
f 64 0 32768 -7 3900 8192 3600 8192 3900 8192 3800 8192 3800 ;FREQ
f 65 0 32768 -7 4950 8192 4950 8192 4950 8192 4950 8192 4950 ;FREQ
f 66 0 32768 -7 0 8192 0 8192 0 8192 0 8192 0 ;dB
f 67 0 32768 -7 -6 8192 -20 8192 -12 8192 -11 8192 -16 ;dB
f 68 0 32768 -7 -32 8192 -15 8192 -26 8192 -22 8192 -35 ;dB
f 69 0 32768 -7 -20 8192 -40 8192 -26 8192 -22 8192 -40 ;dB
f 70 0 32768 -7 -50 8192 -56 8192 -44 8192 -50 8192 -60 ;dB
f 71 0 32768 -7 80 8192 60 8192 60 8192 70 8192 50 ;BAND WIDTH
f 72 0 32768 -7 90 8192 90 8192 90 8192 80 8192 60 ;BAND WIDTH
f 73 0 32768 -7 120 8192 100 8192 100 8192 100 8192 170 ;BAND WIDTH
f 74 0 32768 -7 130 8192 150 8192 120 8192 130 8192 180 ;BAND WIDTH
f 75 0 32768 -7 140 8192 200 8192 120 8192 135 8192 200 ;BAND WIDTH
; p4 = fundemental begin value (c.p.s.)
; p5 = fundemental end value
; p6 = vowel begin value (0 - 1 : a e i o u)
; p7 = vowel end value
; p8 = bandwidth factor begin (suggested range 0 - 2)
; p9 = bandwidth factor end
; p10 = voice (0=bass; 1=tenor; 2=counter_tenor; 3=alto; 4=soprano)
; p11 = input source begin (0 - 1 : VCO - noise)
; p12 = input source end
; p4 p5 p6 p7 p8 p9 p10 p11 p12
i 1 0 10 50 100 0 1 2 0 0 0 0
i 1 8 . 78 77 1 0 1 0 1 0 0
i 1 16 . 150 118 0 1 1 0 2 1 1
i 1 24 . 200 220 1 0 0.2 0 3 1 0
i 1 32 . 400 800 0 1 0.2 0 4 0 1
e
</CsScore>
</CsoundSynthesizer>
Hopefully these examples have demonstrated the strengths of subtractive synthesis in its simplicity, intuitive operation and its ability to create organic sounding timbres. Further research could explore Csound's other filter opcodes including vcomb, wguide1, wguide2 and the more esoteric phaser1, phaser2 and resony.