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Oscillator and Filter

Laboratory of Acoustics and Audio Signal

Processing

Algorithms for Virtual

P.O. Box 3000, FI-02015 TKK, Espoo, Finland

www.acoustics.hut.fi/ Analog Synthesis

vesa.valimaki@tkk.fi

antti.huovilainen@acoustics.hut.fi

Virtual analog synthesis refers to computational derstood, because imitating analog electronics with

methods that imitate the sound production prin- digital processing is not as easy as it may seem. One

ciples used in electronic music synthesizers of the problem is aliasing caused by sampling of analog

1960s and 1970s. In practice, it means digital sub- waveforms that have rapid changes. The spectra of

tractive synthesis. In this paper, we introduce new such waveforms contain infinitely high frequencies,

methods to generate digital versions of classical and the signals are thus not band-limited. Another

analog waveforms with reduced aliasing. We also difficulty is that analog filters do not obey simple

propose modifications to the digital nonlinear linear theory. With high signal levels they generate

model of the Moog ladder filter. These virtual ana- distortion. This does not naturally occur in digital

log synthesis techniques enable the production of processing, but it must be designed and implemented

retro sounds with modern computers. on purpose (Rossum 1992; Huovilainen 2004).

In this paper, we discuss new versions of oscilla-

tor and resonant filtering algorithms that can sound

Introduction like old analog synthesizers. Computationally very

efficient oscillator algorithms not requiring wave-

Virtual analog synthesis refers to computational tables and having reduced aliasing distortion are

simulation of the sound generation principles of proposed for classical waveforms used in subtrac-

analog synthesizers of the 1960s and 1970s. In prac- tive synthesis. These algorithms are modifications

tice, it means digital subtractive synthesis. The ba- and extensions of the digital sawtooth waveform

sic principle in subtractive synthesis is, first, to algorithm based on the differentiated parabolic

generate a signal with rich spectral content, and wave (DPW) proposed recently by Vlimki (2005).

then to filter that signal with a time-varying reso- A new digital resonant filter structure is also pro-

nant filter. posed for subtractive synthesis. It is a modified ver-

Virtual analog synthesis became a popular com- sion of the nonlinear digital Moog ladder filter

mercial term around 1995, when Clavia introduced introduced previously by Huovilainen (2004). The

the Nord Lead 1 synthesizer, which was marketed new structure reduces the computational cost of

as an analog-sounding digital synthesizer that uses the nonlinear digital Moog filter by using a single

no samples (Clavia 2002). Instead, all sounds were nonlinearity instead of five nonlinear functions in-

generated by simulating analog subtractive synthe- side filter sections. The new digital Moog filter

sis. Previously, the Roland D-50 synthesizer of the structure also decouples fairly well the cutoff and

late 1980s worked in a similar way, although it con- the resonance parameters and offers several response

tained sampled sounds. An early example of an at- types by selecting a weighted sum of different out-

tempt to design a digital synthesizer that sounds put points.

analog was Synergy (Kaplan 1981).

Design and implementation of digital subtractive

synthesis are more demanding than is generally un- Analog Subtractive Synthesis

This paper is a revised and extended version of the paper New The electronic music modules introduced by Rob-

approaches to digital subtractive synthesis that was published ert A. Moog in the mid-1960s are one of the most

at the 2005 International Computer Music Conference, important innovations in music technology (Moog

Barcelona, Spain, September 2005.

1965). A few years later, his company introduced

Computer Music Journal, 30:2, pp. 1931, Summer 2006 products where the various modules, such as oscil-

2006 Massachusetts Institute of Technology. lators, filters, and amplifiers, were integrated into a

Figure 1. A typical block Figure 2. The block dia- Figure 3. Matlab code

diagram of subtractive gram of the DPW algo- showing the implementa-

synthesis as it was imple- rithm (Vlimki 2005). tion of the DPW sawtooth

mented in the Prophet 5 algorithm. The variable

synthesizer in the late delta must be initialized to

1970s. the ratio of the fundamen-

tal frequency to the sam-

cepted that some aliasing occurs, but an at-

tempt is made to attenuate it sufficiently.

single portable unit. Subtractive synthesis was the methods. Three approaches are currently available:

main principle used in these instruments. The Min- oversampling of trivial waveforms (Chamberlin

imoog was one of the most popular analog synthe- 1985, pp. 423424), the distortion and filtering of

sizers in the 1970s. sine waves (Lane et al. 1997), and the differentiated

The Prophet 5 synthesizer introduced by Sequen- parabolic waveform (DPW) (Vlimki 2005). We dis-

tial Circuits in 1979 has microprocessor controlled cuss and extend the latter method in the following.

electronics, but it is still an analog synthesizer. Its

block diagram, shown in Figure 1, is today a classic

example of the subtractive synthesis principle. It in- DPW Algorithm and the Sawtooth Waveform

cludes two oscillators, a resonant low-pass filter,

and two envelope generators (ADSR in Figure 1 The sawtooth waveform is a common source signal

stands for attack time, decay time, sustain level, in subtractive synthesis. Vlimki (2005) found out

and release time). There are a couple of alternative that a digital signal that closely resembles the saw-

waveforms available together with a noise source. tooth wavebut with attenuated aliasing distor-

tioncan be produced by differentiating a piecewise

parabolic waveform. The simplest version of the

Digital Oscillators DPW algorithm generates the sawtooth waveform

in four stages, as illustrated in Figure 2. First the

The sharp corners of geometric waveforms, such as trivial sawtooth waveform is generated using a

the sawtooth or the square wave, cause aliasing be- bipolar modulo counter, then the waveform is

cause such signals are not band-limited. Three dif- raised to the second power, the signal is differenti-

ferent classes of methods are known to avoid this ated with a first difference filter with transfer func-

problem: tion HD(z) = 1 z 1, and, finally, the obtained

waveform is scaled by factor

1. Band-limited methods that generate har-

c = fs/[4f0(1 f0/fs)], (1)

monics only below the Nyquist limit, such

as additive synthesis (Moore 1990, pp. 270 where f0 is the fundamental frequency of the saw-

271; Chaudhary 1998) and its variants, e.g., tooth signal and fs is the sampling rate. This scaling

wavetable synthesis (Burk 2004) and the dis- factor may well be replaced with c = fs/4f0, which

crete summation formulae (Winham and gives a slightly too small gain for high fundamental

Steiglitz 1970; Moorer 1976; Moore 1990, frequencies, but is accurate within 1 dB up to 4 kHz,

pp. 271273; Lazzaro and Wawrzynek 2004); when the sampling rate of 44.1 kHz is used.

2. Quasi-band-limited methods in which alias- Figure 3 shows how the basic DPW algorithm can

ing is low and its level can be adjusted by de- be programmed using Matlab. Note that the bipolar

sign to save computational costs, such as in modulo counter is slightly different from a phase

the BLIT (Stilson and Smith 1996a) and the accumulator used in conventional wavetable syn-

MinBLEP (Brandt 2001) techniques; thesis, because counting starts from 1 instead of 0.

pling frequency, f0/fs. The Figure 4. Waveform and and the sampling rate is disturbance. These and

variable c is the scaling co- spectrum of (a, b) the triv- 44.1 kHz. In (b), (d), and other spectra presented in

efficient explained in text. ial sawtooth waveform, (c, (f), the desired spectral this paper have been com-

The comment after the per- d) the squared sawtooth components (2793.8 Hz, puted from a 1 s signal seg-

cent sign explains the pur- wave, and (e, f) the differ- 5587.6 Hz, 8381 Hz, . . .) ment with a 65536-point

pose of each line of code. entiated parabolic wave- are circled (o), while the FFT using a Chebyshev

form. The fundamental rest of the spectral compo- window function that

frequency is 2793.8 Hz nents are caused by alias- has a 120-dB side-lobe

(MIDI note number 101), ing and are heard as attenuation.

Figure 3

(a) (b)

(c) (d)

(e) (f)

Figure 4

The waveform produced by the bipolar modulo Raising the signal to the second power modifies

counter resembles the sawtooth waveform, as seen the waveform so that it now consists of parabola seg-

in Figure 4a, but it sounds badly distorted. The rea- ments, which form the unipolar, non-negative signal

son is that its spectrum decays slowly, about 6 dB shown in Figure 4c. The spectrum of this waveform

per octave. When it is sampled, the spectral compo- decays at about 12 dB per octave, and this is why

nents above the Nyquist limit are mirrored down to aliasing is reduced in Figure 4d (Vlimki 2005). Fi-

the audible frequencies. This is clearly seen in Fig- nally, when the piecewise parabolic signal is differ-

ure 4b, where the desired harmonics are indicated entiated and scaled, the signal again looks like the

by circles and the rest of the peaks are aliased im- sawtooth waveform, see Figure 4e, but the aliased

ages. This signal is called the trivial sawtooth wave. components are suppressed, as seen in Figure 4f.

Figure 5. Magnitude re- point average filter (dash- Figure 6. (a) The waveform

sponses of (a) the ideal dif- dot line), (b) the ideal dif- and (b) the spectrum of the

ferentiator (dashed line), ferentiator (dashed line), signal obtained with the

the first-order FIR differen- and the averaged differen- averaged differentiator.

tiator (solid line), the two- tiator (solid line).

(a) (a)

(b)

(b)

sponding spectrum (see Figure 6b) has decayed

faster at high frequencies than that in Figure 4f. Lis-

A remaining problem is that at high frequencies tening confirms that this attenuation of the highest

the level of aliased components is close to that of octave suppresses the aliased components and

the harmonics. This may lead in some cases to slightly improves the sound quality.

beating. A solution to alleviate this is to replace the

differentiator with its averaged version HD(z) = (1

z 2)/2 = (1 z 1) (1 + z 1)/2, which does not spoil the Pulse Waveform

simplicity of the algorithm. This filter is a combina-

tion of a two-point average filter and the first- Another common waveform used in subtractive

difference filter. Figure 5a compares the magnitude synthesis is the rectangular pulse waveform. A

responses of the first-difference filter (1 z 1)/2 and special case is the square wave, which is a sym-

the ideal differentiator, and also shows the magni- metric rectangular wave with a duty cycle of 50%.

tude response of the two-point average filter. It can A rectangular waveform can be produced by sub-

be seen that the response of the simple differentia- tracting two sawtooth waves with a proper phase

tor is very close to the ideal at low frequencies. The shift. There are two different principles to imple-

error exceeds 1 dB above 11 kHz. Figure 5b gives the ment this, as shown in Figure 7. Any sawtooth

magnitude responses of the ideal differentiator and waveform generator principle can be used. We

the proposed averaged differentiator, which is an consider the use of the DPW algorithm described

FIR comb filter. The proposed filter is very similar above.

to the ideal differentiator at low frequencies with Figure 7a uses two sawtooth waveform generators

less than a 1-dB deviation below 5 kHz. At high fre- with a phase shift to generate a pulse waveform.

quencies, much attenuation is achieved, which is This two-oscillator method allows smooth pulse-

necessary to suppress the potential beating artifacts. width modulation, since the phase shift can be con-

The waveform and spectrum resulting from the tinuously controlled by offsetting the counter in

use of the proposed averaged differentiator are one of the oscillators. The computational cost of

shown in Figure 6. It is seen that in the discrete- this method is simply twice that of a single saw-

time waveform in Figure 6a the transitions from the tooth generator.

maximum value (near +1) to the minimum value Alternatively, as suggested by Lowenfels (2003),

Figure 7. Two alternative outputs and (b) a combina- Figure 8. Comparison of

ways for implementing a tion of a sawtooth wave- rectangular waveforms: (a)

rectangular pulse genera- generator and an FIR comb The trivial and (b) DPW-

tor: (a) a difference of two filter. based square waveforms;

sawtooth wave generator (c) the trivial and (d) DPW-

based rectangular pulse

wave with 20% duty cycle.

(a) (a)

(b)

(b)

(c)

sawtooth wave to produce a pulse wave, as depicted

in Figure 7b. This basic implementation requires

some additional memory and one addition per

sample. However, only phase shifts corresponding (d)

to an integral number of samples are available. Con-

sequently, not just any pulse-width can be imple-

mented accurately, and real-time modulation of the

pulse-width will not sound smooth but will be con-

taminated by zipper noise.

An interpolated delay line can be used in the

structure of Figure 7b to enable continuous control

of the phase shift for pulse-width modulation. In ence of two DPW sawtooth generators with an ap-

practice, a delay line is then cascaded with a frac- propriate phase shift. Their spectra are shown in

tional delay filter, which produces the decimal part Figure 9. The trivial square waveor a pulse wave

of the delay-line length. A first-order all-pass filter with 50% duty cycledisplayed in Figure 8a suffers

or a low-order Lagrange interpolation FIR filter from heavy aliasing, as can be seen in Figure 9a,

(Laakso et al. 1996) is a good alternative for imple- where everything else but the four odd partials of

menting the required fractional delay. However, the the fundamental are images caused by aliasing. The

computational cost of the FIR comb filter with an DPW-based square wave shown in Figure 8b has a

interpolated delay line is approximately the same as much cleaner spectrum, as seen in Figure 9b: just a

that of two DPW-type sawtooth waveform genera- fraction of aliased components are visible above the

tors. For some hardware systems, such as signal 60-dB level, and all aliased components at frequen-

processors, the need for additional delay-line mem- cies below 10 kHz have been suppressed consider-

ory may be inconvenient because of the very lim- ably. However, at high frequencies the level of some

ited on-chip memory. Thus, the two-oscillator first-generation images have not been reduced at all;

method shown in Figure 7a is a more attractive real- compare, for example, the levels of the spectral peak

ization scheme for pulse-width modulation. at 19 kHz in Figures 9a and 9b.

Figure 8 shows examples of trivial pulse wave- As another example, the trivial and DPW-based

forms and the same waveforms generated as a differ- rectangular pulse waves with a duty cycle of 20% are

Figure 9. The spectra of the Figure 10. The triangular Figure 11. Signals from in-

waveforms in Figure 8. waveform generation termediate phases of the

based on a piecewise para- proposed DPW-based trian-

bolic signal. gular waveform algorithm:

(a) The bipolar modulo

counter, (b) the squared

sawtooth (filled circles) and

the square (open circles),

(a)

Figure 10

(b) (a)

(c) (b)

(d) (c)

ment is obtained: aliasing is reduced much at low and

mid frequencies, but not that much at high frequencies

around 20 kHz (see Figures 9c and 9d). The DC com-

ponent (i.e., the average signal level) is also smaller

in the DPW-based signal than in the trivial one.

Figure 11

Triangular Waveform

ity for subtractive synthesis. It is realized with a

The triangular waveform is also a typical source sig- bipolar modulo counter that toggles its direction of

nal in subtractive synthesis. In discrete-time trian- counting every time it reaches the value +1 or 1.

gular waveform generation, aliasing is not as serious It is possible to devise a high-quality algorithm to

a problem as it is in the cases of the sawtooth and produce a triangular waveform with reduced alias-

the pulse waveforms. This stems from the fact that ing. Lane et al. (1997) proposed one such algorithm,

the spectrum of the triangular waveform falls off at which is based on filtering a nonlinearly distorted

the rate of about 12 dB per octave, and not 6 dB sine wave. We show how the triangular waveform

per octave. For this reason, a trivial triangular wave- can be produced by employing the bipolar piecewise

form generator may provide a sufficient sound qual- parabolic waveform.

(c) the bipolar piecewise par- Figure 12. (a) The trivial Figure 13. The spectra of spectral components, or

abolic, and (d) the approxi- triangular, (b) the bipolar the waveforms shown in odd harmonics of the fun-

mated triangular signals. piecewise parabolic, and Figure 12: (a) the trivial tri- damental (2793.8 Hz,

The fundamental frequency (c) the alias-free triangular angular, (b) the bipolar 8381.4 Hz, 13969 Hz, . . .),

of the obtained triangular waveforms. The funda- piecewise parabolic, and are circled (o).

signal is 2793.8 Hz and that mental frequency is (c) the alias-free triangular

of the modulo counter is 2793.8 Hz. waveforms. The desired

two times higher, 5587.6 Hz.

(a) (a)

(b) (b)

(c) (c)

Figure 12 Figure 13

The block diagram of the proposed algorithm is spectra of the trivial and DPW-based triangular

shown in Figure 10. A bipolar modulo counter is waveforms. It is seen in Figures 12a and 12c that the

used, where the fundamental frequency is twice difference between these two waveforms is micro-

that of the desired triangular signal. This initial scopic, apart from the initial transient in Figure 12c,

waveform, which is equivalent to the trivial saw- which lasts for one sample when the first-order FIR

tooth signal, is shown in Figure 11a. After squaring differentiator is used. Nevertheless, the spectra

the initial waveform, it must be turned upside given in Figure 13a and Figure 13c are quite differ-

down by subtracting it from one (see Figure 11b). ent: the level of the aliased components is much re-

The signal is modulated by a (trivial) square wave duced at low frequencies in Figure 13c. Figure 12b

with the fundamental frequency 2f0 (see Figure 11b, shows the bipolar parabolic signal, whose spectrum

open circles) to produce a piecewise parabolic wave- is given in Figure 13b. It can be noticed that the

form of the desired type (see Figure 11c). The square spectrum of this signal decays fast with frequency,

waveform must be phase-locked to the bipolar mod- about 18 dB per octave. This is why aliasing is less

ulo counter; this is easy by toggling the sign of the severe than in the case of the trivial triangular

square waveform when the counter reaches +1. Fi- wave. It can also be noted that the triangular wave

nally, the resulting bipolar piecewise parabolic approximation (see Figure 12c) is perfectly symmet-

waveform is differentiated and scaled to obtain a tri- ric above and below the zero level. Thus, its spectrum

angular waveform with reduced aliasing, which is (see Figure 13c) is free of even harmonics, which are

presented in Figure 11d. not well suppressed in some other triangular wave-

Figures 12 and 13 compare the wave shapes and form approximations.

Figure 14. The compromise

one-pole filter section.

A resonant filter used in computer music differs

from traditional IIR filters in mainly three ways: the

parameters are changed at a rapid rate, the order is

usually predetermined instead of matching to cer-

tain stop-band attenuation specification, and a con-

trollable resonant peak is introduced near the cutoff

frequency. nance. To realize the filter, a unit delay has to be in-

A perfect digital resonant filter then fulfills the serted in the feedback path, but this couples the

following criteria. cutoff and the resonance controls. Various ways of

compensation have been examined, with the com-

1. Coefficient update should be fast. To avoid

promise version (Stilson and Smith 1996b) being

clicks, the coefficients should be updated on

the most attractive. The compromise version in-

a per-sample basis (Rossum 1992).

serts a zero at z = 0.3 inside each one-pole filter

2. The filter cutoff and resonance parameters

section, thus mostly decoupling the resonance and

should be decoupled. Change in one should

the cutoff parameters. The modified one-pole filter

not affect the other.

structure is shown in Figure 14.

3. The filter should stay unconditionally stable

The coefficient g determines the cutoff frequency.

as long as parameters are inside the allowed

For very low frequencies, it can be trivially derived

range.

from the analog prototype filter as g 2fc/fs, where

4. The filter should have a response similar to

fc is the desired cutoff frequency and fs is the sam-

an existing analog resonant filter. Some ana-

pling rate. This, however, leads to the actual cutoff

log filters have a characteristic sound that

frequencies increasingly deviating from the desired.

should be emulated, if possible.

Due to the transfer-function zero at z = 0.3 and the

5. The filter should be capable of self-oscillation.

unit delay in the feedback path, the resonant fre-

A number of filters trying to meet the criteria have quency cannot be easily predicted. Figure 15a com-

been developed. We take a closer look at the Moog pares three ways of calculating coefficient g: the

low-pass filter (Moog 1965) here. impulse invariant, the bilinear, and the direct linear

method.

All of the methods deviate from the desired tun-

Digital Moog Filter ing at high cutoff frequencies. Some form of com-

pensating this deviation must therefore be used. A

The Moog ladder filter (Moog 1965) can be consid- least-squares fit of a fourth-order polynomial to

ered the first musical filter. It features independent compensate the directly tuned section gives the fol-

voltage control of both the cutoff frequency fc and lowing equation for parameter g:

the resonance Cres while also having a characteristic

g = 0.9892wc 0.4342wc2 + 0.1381wc3 0.0202wc4 (2)

sound of its own. The filter consists of four identi-

cal one-pole low-pass sections (implemented with where wc = 2fc/fs is the cutoff frequency in radians

an innovative transistor ladder circuit) in series per sample. The tuning error, see Figure 15a, is then

with a global negative feedback to produce a reso- below 15 cents within the 0 to 14-kHz range of the

nant peak near the cutoff frequency. cutoff frequency at a 44.1-kHz sampling rate. This

A digital model of the Moog filter was first pre- is accurate enough so that the filter can be tuned to

sented by Stilson and Smith (1996b). As in the ana- a multiple of a tones fundamental frequency to em-

log prototype, it has four one-pole filters in series, phasize an individual harmonic or, when self-

and a global feedback is used to produce the reso- oscillating, to produce an additional harmonic.

Figure 15. (a) Deviation in scaling, and the polyno- Figure 16. Improved Moog- can be used to select the

tuning of the resonance fre- mial method for calculat- style filter. Each block la- type of output (low-pass,

quency in the compromise ing the filter cutoff beled as LP contains the high-pass, band-pass, or

Moog filter using the coefficient. (b) Deviation one-pole filter structure notch filter or one of their

scaled impulse invariant in feedback amount from shown in Figure 14. Coeffi- combinations).

transform, the bilinear z- ideal. cients A, B, C, D, and E

transform, the direct linear

(a)

and Smith 1996b) is certainly useful and solves the

problem of fast coefficient update, it becomes un-

stable at very large resonance values and it cannot

self-oscillate. Furthermore, it does not emulate the

(b) characteristic distortion produced by the original

transistor ladder circuit. Huovilainen has developed

an improved model that simulates the ladder cir-

cuit by inserting nonlinearities inside the one-pole

filter sections (Huovilainen 2004). This improved

model emulates more closely the characteristic

sound of the original Moog filter and is also capable

of self-oscillation. A disadvantage is the need for

five hyperbolic tangent (tanh) function evaluations

per sample and for oversampling by a factor of at

least two.

We propose an alternative extended model pre-

sented in Figure 16. The embedded nonlinearities

within sections are replaced by a single nonlinear-

ity, thus greatly reducing the computational cost of

the filter. We have used the tanh function for the

Figure 15b shows the error in the amount of feed- nonlinearity, but any smoothly saturating function

back, too. This error can similarly be compensated may be used. There is a difference in the sound

by a suitable polynomial. Using compared to the full nonlinear Moog filter model

(Huovilainen 2004), but this new model can emu-

Gres = Cres (1.0029 + 0.0526wc 0.0926wc2 + late most of the behavior, such as self-oscillation.

0.0218wc3), (3) Its output is also always bounded.

The traditional Moog filter and similar cascaded

where 0 Cres 1, the error is less than 1% at fre- one-pole filters suffer from a decreasing pass-band

quencies below 17 kHz. This polynomial has been gain as the resonance is increased, because the reso-

obtained by fitting in the least-square sense a third- nance is produced with a global negative feedback.

order polynomial to the inverse function of the feed- If a fraction of the input signal is subtracted from

back deviation curve shown in Figure 15b. The the feedback signal before multiplying by the reso-

corrected behavior is also displayed in Figure 15b. nance parameter, the pass-band gain change can be

Figure 17. Comparison of

output waveforms of the

simplified (see Figure 16)

and the full (Huovilainen

2004) nonlinear Moog filter

models using a sinusoidal

input for two different

choices of cutoff frequency.

controlled (Curtis Electromusic Specialties 1984). A both the input waveform and the distortion func-

value of 1.0 for the Gcomp parameter in Figure 16 tion are symmetric. The difference almost disap-

keeps the pass-band gain constant. This, however, pears when the resonance parameter is increased.

results in a large increase of the output amplitude This can be attributed to the negative feedback em-

as the resonance is increased. To maintain the over- ployed to produce the resonance peak. As soon as

all level approximately constant, the value of the the pass-band gain compensation is used, the differ-

Gcomp parameter should be set to 0.5, resulting in a ence is restored. The simplified Moog model of

6-dB pass-band gain decrease at the maximum reso- Figure 16 therefore seems best suited for uses where

nance (compared to a 12-dB decrease in the original a large overdrive is not demanded. The simple non-

Moog model). linearity still limits the amplitude when a very

Figures 17 and 18 compare the two models for high resonance amount or the self-oscillation

210-Hz sinusoidal and sawtooth inputs using mod- mode is used.

erate overdrive. The input peak-to-peak amplitude Another improvement to the original Moog model

is 2.0 for the full model and 1.0 for the simplified is the addition of various frequency response modes

model in order to visually match their behavior. besides the original 24-dB/oct low-pass filter mode.

The output signals have been normalized to ease This can be easily achieved by mixing the outputs of

the comparison. The difference is clearly visible the individual sections with different weights. The

in the sine wave when the cutoff is much above the concept was pioneered in the Oberheim Xpander

input frequency. For the sawtooth wave the differ- and Matrix-12 synthesizers (Oberheim 1984), but it

ence is smaller. An interesting effect is the asym- was not widely used due to the large number of re-

metric output shape in the full model even though quired components and the need for precision resis-

Figure 18. Comparison of wave input with (left) no

output waveforms of the resonance, (center) with

simplified (see Figure 16) 50% resonance, and (right)

and the full (Huovilainen with 50% resonance com-

2004) nonlinear Moog filter bined with a 6-dB pass-

models using a sawtooth band gain compensation.

Table 1. Examples of Weighting Coefficient Values Figure 19 shows examples of four-pole low-pass,

for Typical Magnitude Response Types That Can Be two-pole low-pass, and two-pole high-pass filter re-

Obtained with the Filter Structure of Figure 16. sponses produced by the filter structure of Figure

16. More examples of responses as well as the equa-

Filter Type A B C D E

tions for deriving the coefficients are available else-

Two-pole low-pass 0 0 1 0 0 where (Oberheim 1984). The filter response shapes

Four-pole low-pass 0 0 0 0 1 remain relatively constant independent of the cutoff

Two-pole band-pass 0 2 2 0 0 frequency, but the amplitude falls as the cutoff fre-

Four-pole band-pass 0 0 4 8 4 quency is increased. The drop depends on the re-

Two-pole high-pass 1 2 1 0 0 sponse used but can be readily compensated using a

Four-pole high-pass 1 4 6 4 1

polynomial gain correction. For most responses, a

simple linear correction will be sufficient.

and a large number of different low-pass, band-pass, Conclusions

high-pass, and notch filter responses and their com-

binations can be easily realized. Table 1 shows We discussed new oscillator and resonant filter al-

weighting coefficient values for common filter gorithms. The DPW oscillator algorithm generates

types. Morphing between these modes is possible an approximate sawtooth waveform that has re-

by changing the coefficients at runtime, thus allow- duced aliasing with respect to the trivial sawtooth

ing interesting modulation possibilities. waveform (i.e., the modulo counter output). This re-

Figure 19. Example filter

responses with a 1-kHz

and a 10-kHz cutoff fre-

quencies.

cently proposed method is probably the simplest Valkonen of VLSI Solution Oy (Tampere, Finland)

useful technique for this purpose, because only the for helpful discussions.

trivial sawtooth is simpler, which is practically use-

less due to its heavy aliasing. In this paper we pro-

posed an alternative differentiator for the DPW References

algorithm that suppresses high frequencies to avoid

audible artifacts. We introduced methods to pro- Brandt, E. 2001. Hard Sync without Aliasing. Proceed-

duce pulse and triangular waveforms based on the ings of the 2001 International Computer Music Confer-

parabolic waveform. ence. San Fransisco: International Computer Music

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