Extensions of the Karplus-Strong Plucked-String Algorithm

Author(s): David A. Jaffe and Julius O. Smith

Source: Computer Music Journal , Summer, 1983, Vol. 7, No. 2 (Summer, 1983), pp. 56-
69
Published by: The MIT Press

Stable URL: https://www.jstor.org/stable/3680063

JSTOR is a not-for-profit service that helps scholars, researchers, and students discover, use, and build upon a wide
range of content in a trusted digital archive. We use information technology and tools to increase productivity and
facilitate new forms of scholarship. For more information about JSTOR, please contact [email protected].

Your use of the JSTOR archive indicates your acceptance of the Terms & Conditions of Use, available at
https://about.jstor.org/terms

The MIT Press is collaborating with JSTOR to digitize, preserve and extend access to Computer
Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
David A. Jaffe and Julius 0. Smith
Center for Computer Research in Music and
Extensions of the
Acoustics (CCRMA)
Stanford University Karplus-Strong
Stanford, California 94305
Plucked-String
Algorithm

Introduction The String-Simulation Algorithm

In 1960, an efficient computational model for vi- The Karplus-Strong plucked-string algorithm is pre-
brating strings, based on physical resonating, was sented in this issue of Computer Music Journal.
proposed by McIntyre and Woodhouse (1960). This From our point of view, the algorithm consists of
model plays a crucial role in their recent work on a high-order digital filter, which represents the
bowed strings (McIntyre, Schumacher, and Wood- string; and a short noise burst, which represents
house 1981; 1983), and methods for calibrating the the "pluck." The digital filter is given by the dif-
model to recorded data have been developed (Smith ference equation
1983).
Independently, in 1978, Alex Strong devised an Yn = Yn
Xn += Yn-N
Yn-N+ YN+
+ yn-(N+1) (1)
2
efficient special case of the McIntyre-Woodhouse
string model that produces remarkably rich and re- where x. is the input signal amplitude at sample n,
alistic timbres despite its simplicity (Karplus and y, is the output amplitude at sample n, and N is
Strong 1983). Since then, Strong and Kevin Karplus the (approximate) desired pitch period of the note in
have explored several variations and refinements of samples. The noise burst is defined by
the algorithm, with an emphasis on small-system
Au, n = 0, 1, 2,...,N- 1
implementations. We have found that the Karplus- xn = 0, n N,
Strong algorithm can be used with equally impres-
sive results on fast, high-power equipment. The where A is the desired amplitude, and un, [-1,
availability of multiplies, for example, allows sev- the output of a random-number generator. The o
eral modifications and extensions that increase its put yn is taken beginning at time n = N in our
usefulness and flexibility. These extensions are de-implementation.
scribed in this paper. The developments were moti-
vated by musical needs that arose during the
composition of May All Your Children Be Acrobats Analysis of the String Simulator
(1981) for computer-generated tape, eight guitars,
and voice and Silicon Valley Breakdown (1982) for Before proceeding to practical extensions of the
four-channel, computer-generated tape, both writ-gorithm, we will describe the theory on which
ten by David Jaffe. Our theoretical approach and many of them are based. Various concepts from
the extensions based on it have also been applied todigital filter theory are employed. For a tutorial i
the McIntyre-Woodhouse algorithm (Smith 1983). troduction to digital filter theory, see the works
Smith (1982b) and Steiglitz (1974).
The input-output relation of Eq. (1) may be ex-
David A. Jaffe is also affiliated with the Music Department at
Stanford University, and Julius O. Smith is also affiliated with
the Electrical Engineering Department there.
1. In some situations, the sound more closely resembles a str
Computer Music Journal, Vol. 7, No. 2, struck with a hammer or mallet than one plucked with a pic
Summer 1983, 0148-9267/83/020056-14 $04.00/0, but we will always use the term pluck when referring to the
? 1983 Massachusetts Institute of Technology. excitation.

56 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
 Fig. 1. Block diagram for
the basic string simulator.

pressed differently by means of delay-operator nota-

tion. We define the unit-sample delay operator d by
the relation

dkx A- Xn-k,

where x, is an arbitrary signal, and k is an integer.

(The symbol A means "is defined as.") Thus, multi-
plying a signal by dk delays the signal in time by k
samples. In these terms, Eq. (1) becomes

yn = dNy + dN+lyn
tor is the following set of difference equations:

Vn = Yn-N
=x+n L+ d 2
V= n + Vn-
Solving for yn yields 2

Xn1 yn = xn + Wn
Yn XI+d' (2)
1- dN The frequency response of a digital filter is de-
2
fined as the transfer function evaluated at z =

We can convert linear delay-operator equations e's = cos(oT,) + j sin(o Ts), where T, is the sam-
immediately to z-transform equations by replacing pling period in seconds (Ts is the inverse of the sam-
each time signal with its z-transform, and replacing pling rate fj, o = 27rf is radian frequency, f is
d with z-'. It is customary to denote a time signal frequency in Hz, and j = \-7. The frequency re-
in lowercase letters (e.g., x) and the corresponding sponse of the string simulator is then
z-transform in uppercase letters (e.g., X(z)). The 1
H(e,Ts) =
transfer function of a (linear, time-invariant) digital He ') 1 - Ha eiw sHb(e s)'
filter is the z-transform of the output signal divided
where
by the z-transform of the input. The transfer func-
tion of the string simulator is then found to be 1 + e- 'T T
H|(e 2Ts) = e e-' 2cos(coT,/2)
H(Z) AY(z)
Y = X_ 1
H X(z)x 1 l+z - = e-fTscos(rfTs)
2
Hb(es s ) = e- iN = e- if.NT
1
1 - Ha(z)Hb(z)' In this paper it is necessary to consider the am-
plitude response and phase delay of the feedback fil-
where ters separately. The amplitude response is defined
as the magnitude of the frequency response, and
1 + z-
Ha(z) Q 22 it gives the gain of the filter as a function of fre-
quency. The phase delay is defined as minus the
Hb(z) z-N complex angle of the frequency response divided
by radian frequency, and it gives the time delay
This form of the description is shown in Fig. 1. The(in seconds) experienced by a sinusoid at each
feedback loop consists of a length N delay line frequency.
Hb(z) in series with a two-point average Ha(z). Cor- The amplitude response of each component filter
responding to this breakdown of the string simula- is given by

Jaffe and Smith 57

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
 _ __

Ga(f) A IHa(eiTs)I = cos(oT,/2)l = Icos(T7fT)l NTs + Ts/2 sec. Experience shows this to corre-
spond well with perceived pitch.
Gb(f) A IHblee ws)I 1.
Thus the delay line Hb is lossless, and the two-
point average Ha exhibits a gain that decreases with Decay of "Harmonics"
frequency according to the first quadrant of a co-
sine. We will assume hereafter that all frequencies Since the signal is only quasi-periodic, it does not
are restricted to the Nyquist limit, that is, If l consist of discrete sinusoids. Essentially, we have
fs/2. In this range, we have (cos(i7fTj) = cos(1rfTs).
many narrow "bands" of energy decaying to zero at
It is convenient to define phase delay in units different rates. When these energy bands are cen-
consisting of samples rather than seconds. The tered at frequencies that are an integer multiple of a
phase delays of Ha and Hb in samples are given by lowest frequency, they will be referred to as har-
monics. When the frequency components are not
Pa(f) - / H(e'2) =1
"- ~oT, 2' necessarily uniformly spaced, the term partial will
be used to emphasize the possibility of inharmo-
nicity. Consider, then, a partial at frequency f Hz
Pb(f) A - / Hb(e'S) = N.
PfT, circulating in the loop. On each pass through the
loop, it suffers an attenuation equal to the loop-
(Lz denotes the complex angle of z). The two-point
amplitude response, Ga(f)Gb(f) = cos(IrfTs); that is,
average has a phase delay equal to half a sample,
and the delay line has a phase delay equal to its one period's attenuation = cos(rfTs).
length.
Since the round-trip time in the loop equals N +
Since the total loop consists of Ha and Hb in se-
1/2 samples, the number of trips through the loop
ries, the loop gain and effective loop length are
after n samples (nTs sec) is equal to n/(N + 1/2) =
loop gain = G(f) Gb(f) = cos(rfT,), tfS/(N + 1/2). Thus the attenuation factor at time
and t = nT, is given by
tf5
loop length = Pa(f) + Pb(f) = N + 1/2 (samples)
af(t) A [cos(7rfTs)] N + 2. (3)
for each sinusoidal frequency f Hz. For example, an initial partial amplitude A at time
In synthesizing a single plucked-string note, we 0 becomes amplitude Aaf(t) at time t seconds,
feed in N samples of white noise at amplitude A where f is the frequency of the partial.
and listen to the output immediately afterward. It The time constant of an exponential decay is tra-
is equivalent to initialize the delay line Hb with ditionally defined as the time when the amplitude
sealed random numbers at time 0 and employ no has decayed to l/e - 0.37 times its initial value.
input signal. Since the two-point average Ha is con- The time constant at frequency f is found by equat-
stantly changing the contents of the loop, the out- ing Eq. (3) to e-'/f and solving for rf, which gives
put signal is not periodic. It is close to periodic,
however, and we use the term period in this loose Zs

sense. Each period of the synthetic string sound

corresponds to the contents of the delay line at a -t ( )
particular time, and each period equals a somewhat
InTf =
af(t) In cos(rTrfT) = - n
lowpass version of the previous period. More pre- For audio, it is normally more useful to define the
cisely, a running two-point average of the samples time constant of decay as the time it takes to decay
comprising one period gives the next period in the -60 db, or to 0.001 times the initial value. In this
output waveform. Since the effective loop length is case, we equate Eq. (3) to 0.001 and solve for t. This
N + 1/2 samples, the period is best defined to be value of t is often called t6o. Conversion from f to

58 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
 Fig. 2. Spectral evolution
during the first 16 periods.

t60(/) is accomplished by

t6o(f) = ln(1000),rf 6.91 f. (5)

For example, if a sinusoid at frequency f Hz has

amplitude A at time 0, then at time t,o(f) it has am-
plitude Aaf(t6o (f)) = A/1000, or it is 60 db below its
starting level.
The above analysis describes the attenuation due
to "propagation" around the loop. It does not, how-
ever, incorporate the fact that sinusoids that do
not "fit" in the loop are quickly destroyed by self-
interference. This situation is analogous to making
an actual string vibrate. Any signal may be "fed
into" the string, but after the input ceases, the re-
maining energy quickly assumes a quasi-periodic
nature. Thus, even though the loop is initialized
with random numbers, after a very short time the monic amplitudes, since
primary frequencies present in the loop are those from one harmonic amp
that have an integral number of periods in N + 1/2 In certain extensions to
samples. These frequencies are all multiples of the other than a two-point
attenuation factor of th
frequency whose period exactly matches the loop
length N + 1/2. This lowest frequency provides the onds is approximately
fundamental, or pitch frequency, of the note: tf,

ak(t) = G,(fk )N + Plfk), (10)

A (6)fs
where we require Ga(fl) 1 for stability. The phase
(N+2)
2 12 T~ N4+-
delay Pa(fk) of Ha may be used to create inharmonic
Setting f to the spectra (Smith 1983). series
harmonic The spectrum is beginn
harmonic
only when Pa,fk) is the same at all harmonic
f, = la, E k fs
frequencies fk. k = 2, ... N
Similarly, when Ha is more general, the time con-
N+ 22
stant of decay for each harmonic becomes
gives the decay factor at time t for the kth harmonic
to be T N + P,(fk) t seconds. (11)
Tk = - f In G(2rfkT) econ

ak(t) = [costlrfkTs)]ft. (8)Having provided an analytic vocabulary, we now

proceed to a detailed examination of our additions
Similarly, the time contrast per harmonic is to the Karplus-Strong algorithm.
given by

_-t 1
k In ak(t) f In Tuning
cos(rfkT,) (secon
The fact that the delay-line
Figure 2 shows the spectral length N must be an
evaluation during
integer causes having
first 16 periods of a note tuning problems. Sincea the funda-
period of
samples. A 128 length mental
Fastfrequency is f, = f/(N + 1/2), the
Fourier allowed
Transform
pitches are quantized,
was computed every other especially at highEach
period. frequency. curve
For large
figure is interpreted as thevalues ofenvelope
N (low pitches), the difference
of the ha

Jaffe and Smith 59

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
 Fig. 3. Desired pitch versus
resulting pitch for a 50-
KHz sampling rate.

whole string is now

- 20- H(z) A
) 1 - Ha(z)Hb(z)Hc(z)
The filter H, is a first-order allpass filter, and as
a -/ such it has a constant amplitude response. Indeed,
the amplitude response is simply

4 10- / Gc(f) - IHc(e s)l= 1 + Ce -7T| 1

N

The use of an allpass filter ensures that no modi-

0
CO
O - fication of the decay rate will take place. The loop
gain is G,(f)Gb(f)G,(f) = cos(rTfTs) as before.
We will select the phase delay of H, so as to tune
5 10 15 20 25 fl to the precise desired frequency. This requires
only the ability to
Desired frequency select phase delays between 0
(KHz)
and T, sec, or one sample's worth.
The phase delay of the first-order allpass H, is
between the pitch at N and N + 1 is ve
given by
However, for high pitches, N and N +
different pitches and tuning becomes
PC(f) A _ /Hc (e's)
3 shows the distortion in coTT frequency fo
pling rate 50 KHz. For lower sampling
-1 C + e-'"T
curve is identical in form, and the dist
coTs 1 + Ce-'T
at proportionately lower frequencies.
The key to a solution L(1 for + Ce-"T) _ (C + e-'"T)
this problem
expression for loop length coTs in coT terms of
The fundamental frequency is given by
= 1tan(- C sinlT)
_ = fs
f N+Pa(fl)'
oT tanI -sin(cT)
where Pa(f) = 1/2 when the two-point average is oT,s tn C + cos( oTj (13)
)
used for Ha. To make up the difference between f
When the arguments to the arc tangent above have
and the desired frequency, we need to introduce
magnitude less than unity, we can use the power-
into the feedback loop a filter that can contribute a
series expansion (Abramowitz and Stegun 1966),
small delay without altering the loop gain. The fil-
ter we introduce has the difference equation X3 XS X7
tan-'(x) = x -- + + ., Ix< 1.
Yn = CXn + Xn-1 - CYn- (12)
and transfer function Thus we can approximate the low-fr
delay by
H,(z)A
1 + Cz-' Pcf ) = sin(oT,) C sin(co T
cT,()C + cos(oT,)) oT,(1 + C cos(oTj))
where C is the only coefficient to be set. For sta-
1 C 1-C
bility, we must have ICI < 1. It can be shown that C+ 1 +C 1+C'
(14)
when the input x, is bounded by 1, the output is
bounded by 21CI + 1. The transfer function of the A plot of the exact phase delay is given in Fig. 4 for

60 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
Fig. 4. Phase delay for the
fine-tuning allpass filter
H,(z) = (C + z-')/
(1 +C z-).

2- and the delay Pc(fL) required from the allpass filter

become

N A Floor(P, - P(f,) - ) (15)

Pc(fl) A Pi - N - P,(f),
c :
where s > 0 is the offset that shifts Pc(fl) into the
4

cn range [e, 1 +e], and Pa(f,) is the delay in samples du

-'1 _ ...-~

to the filter Ha. In the simple case where Ha is a

two-point average, Pa(f,) = 1/2.
C
We next solve for the filter coefficient in Eq. (13)
as a function of Pc(f,). Taking the tangent of both
sides, and using an identity for the tangent of a
difference leads to the quadratic equation in C,

n
j
C2 sin(co1TJP,(f,) + cw,Tj + 2C sin(co,TPc(fA)) +
sin(co,TP,(fl) - coTj) = 0,
I. I I I ? I

0 Frequency F,/2
where co A_ 27rf. The solution is found, after some
manipulation (Mont-Reynaud 1982), to be
17 values of C equally spaced between -0.999 and
0.999. Note that delays between 0 and 1 sample can -sin(ol TsP(fL)) + sin(olTs)
be provided somewhat uniformly across the fre- sin(wol TsP(f ) + coTj )

quency axis. A delay of 0 samples corresponds to We have introduced an extra root by producing a
C = 1, where the pole and zero of Hc(z) cancel to quadratic equation. The previous approximation
give Hc(z) 1. However, pole-zero cancellation on Eq. (14) indicates that the + sign should be taken.
the unit circle is not a good thing in practice, since Therefore, the final solution is
round-off errors may yield an unstable filter. There-
fore, it is preferable to shift the range of one-sample - sin(c, Tj - sin(o, TsP,c(f))
delay control to the region s < Pc < (1 + e) for sin(cl TsPc(fl) + c Ts)
some small nonnegative e (0 < < 1). It is best not
to shift very far, since the phase-delay curves are -sin( co T - o, TsPc(fl)
less flat in the region beyond one sample's delay. (16)
Note that the delay curves below the one-sample = i( TT,+co, TsPff,
level in Fig. 4 correspond to slightly flattened up-
per partials, while the delay curves above the one- which can be approximated, at low frequencies, by
sample level correspond to slightly sharpened upper 1 - Pc(f,)
partials. The timbre change due to slight systemat- (17)
i + Pc(AM)
ic shifting of the upper partials, of an amount less
than one sample period, was found to be hardly no- Although this technique provides a precise funda-
ticeable. It may even be desirable as a source of sub- mental frequency, it does not guarantee an in-tune
tle timbral variation. In any case, it is important to percept, since the perceived pitch does not always
get the perceived pitch right. coincide with the fundamental frequency. An ad-
To tune the instrument precisely to a desired ditional mapping onto a perceptual tuning dimen-
fundamental frequency f, let P1 equal fs/f, the real sion may be needed for the very high notes. In our
value for the period of the first partial, in samples, case, it was found that tuning the octaves slightly
which would give perfect tuning. Then we desire stretched, as is done in piano tuning, gives a more
N + Pa(fi) + P,(fl) = P. The integer buffer length N satisfying in-tune percept.

61
Jaffe and Smith

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
Decay-time Alteration rable in duration with notes from a real
the loss factor operative, the decay-time
The basic algorithm naturally results in a shorter for the fundamental frequency becomes
decay time for high pitches than for low pitches, 1
reflecting the behavior of real strings. This is due r(p) = - f, In lp cos(rTf, Tj)I ' (19)
to two effects. First, higher frequencies are more
attenuated by the two-point average Ha; second, Decay shortening produces a damped version of the
higher pitch means more trips through the atten- algorithm, analogous to substitution of a soft mate-
uating loop in a given time. This may be seen in rial for the bridge of a string instrument.
Eq. (3).
Unfortunately, the range of decay times between
the high and low pitches is too extreme. The high- Decay Stretching
pitched notes die away so fast that only a click is
perceived, while the low-pitched notes last for an To stretch the decay, the feedback average (Ha) can
unnaturally long time. In addition, the decay time be changed to a two-point weighted average. This
of real strings varies with many factors such as ten- reduces the amount of loss at high frequencies.
sion, length, thickness, and material. Consequently, Thus, we replace H(z}) by
we have found it useful to add a means for altering
H,(z, S) = (1 - S) + Sz-, (20)
the note duration. The ability to control decay time
is essential for a realistic simulation as well as for where S, the stretching factor, is between 0 and 1.
musical flexibility. The gain of this filter is
On systems that have separate control of the
G,(f, S) = (1 - S) + Se-' it
sampling rate of each voice, sampling-rate change
can be used to control decay time. On other sys- = V ((1 - S) + S cos oTJ)2 + (S sin(w)T,))2
tems, however, it is necessary to use other methods
= V (1 - S)2 + S2 + 2S(1 - S) cos wTr
to alter decay time.

(21)

Decay Shortening With S = 1/2, Ha(z, S) reduces to the previous case

Ha(z). For stability of the overall string, we must have
To shorten the decay time, a loss factor p can be in- 0 < S < 1. If S = 0 or 1, the frequency-dependent
term disappears, and the gain response is unity for
troduced in the feedback loop. With the loss factor,
all f; in this case, the initial white-noise burst cir-
the difference Eq. (1) for the string becomes
culates forever in the loop, producing harmonics
y,=fx,,+ p (18)
Yn = X -N + Yn-(N+)18 that never decay. At intermediate values, 0 < S < 1,
the effective note duration (t60) is finite, and it is
The amplitude envelope of a sinusoidminimum for S = 1/2. The amplitude trajectory
at frequency
and the
f, previously given by Eq. (8), is now proportional decay-time constanttofor each partial can be
obtained by substituting Eq. (21) into Eqs. (10) and
a,(t, p) = |p cos(TrfTs)jfl = IPIM'taf(t).
(11), respectively.
Thus all partials are affected equally: For the the relative
greatest control, both the uniform-loss
decay rates are unchanged. method and the weighted two-point-average method
Note that p cannot be used to lengthen may be used thefor decay-time
decay alteration. The result-
ing
time, since the amplitude at 0 Hz would increasedecay time is then a function of loss factor p
exponentially. In general, we must have Ipl - 1S. if
and stretch factor Karplus and Strong (1983) de-
scribe a
the string is to be stable. Thus p is used to shortenmethod of decay stretching that uses no
multiplies.
the low-pitched notes to make them more compa-

62 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
Fig. 5. Phase delay for the
decay-stretching, one-zero
filter Ha(z, S) = (1 - S) +
S z-'.

Note that for S > 0 the phase delay is qu

over most of the frequency axis. Another
of interest is that since Ga(z, S) = Ga(z,
2/
we may choose the case that yields the be
-s /4 - / delay curve for the fine-tuning allpass H,
Since the tuning calculation needs to b
only once per note, the precise form of E
can be used for each new frequency witho
computational expense.

Dynamics

The loudness of the signal output by the

0 Frequency F,/2 is a function of the amplitude of the inpu
burst. However, this is an unsatisfactory
simulating the timbral effect of dynamic
occurs in the case of a real string instrum
effect of varying initial amplitude gives t
Effect of Decay Stretching on Tuning
sion of a change more in the distance bet
listener and the apparent source than in d
Changing S changes the effective loop length as a
Since strings plucked hard have more ene
function of frequency, since it changes the phase
higher partials than strings plucked light
delay of the overall loop. We must therefore com-
nonlinearities becoming important), the d
pute Pa(f,) for use in Eq. (15) when fine tuning with
simulation is based on modeling this diffe
the allpass filter H,. The phase delay of the weighted
spectral balance. We therefore change the
two-point average is given by
spectral bandwidth of a note to modulate
ent intensity.
Pa(f, S) coT,
- ?Ha(e'TS, S) The bandwidth is controlled by means of a one-
pole, lowpass filter applied to the initial noise burst
((L - S) + Se-' T) (before it is fed into the string). This filter will be
coT,
referred to as the dynamics filter. The difference
_ tan-' -Ssin(oT) ,2
equation of the dynamics filter is
- , tan- (1 - S)+ S cos(ouT) ) (22)
yn = (1 - R)xn + Ryn-,
and for low frequencies, relative to the sampling
and its transfer function is
rate, we may use the approximation
1-R
P(f, S) T -S sin(oTs) Hd(z) = 1 - Rz-' (23)
Pa( f T,({1 - S) + ScoTs cos()oT,)
where R is a real number between 0 and 1, com-
- S, O < S 1.
puted as a function of fundamental frequency fl and
the algorithm,
For S = 1/2, we have the basic string desired dynamic level L. When a series of notes
and the phase delay of Ha is 1/2, as given f,
at pitch is played
by thewhile R is moved gradually to-
ward 1,
above approximation. For other values of S the a diminuendo is approximated in terms of
both decreasing
approximation is always precise at f = 0. Figure 5 loudness and spectral bandwidth
reduction.
shows the true phase-delay curves of H,(z, S) as S
is stepped uniformly through 17 values Wefromdefine 0 thetodynamic
1. level L as a bandwidth be-

Jaffe and Smith 63

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
 Fig. 6. Frequency response
of H(z) = (1 - R)
(1 - R z-l) computed at
dynamic level L = 100 for
six values of f,.

tween 0 and f/2. If L is small, the spectrum is more

lowpass filtered, corresponding to a softer dynamic
level. Conversely, large L gives a bright spectrum
corresponding to louder notes. It is not sufficient to
o
-- 25 -25t1
use a fixed lowpass filter for all pitches, since low-
pitched notes would then be louder than high-pitched
notes. Rather, for a given dynamic level, R must be
changed with pitch to yield a uniform perceived -50 = 100 Hz
loudness. While this is a difficult problem in gen-
eral, a good approximation is obtained by varying R 50123I ....
so that the amplitude of the fundamental frequency 0 1 2 3 4
is constant.
fi fm Frequency (KHz) f.
It remains to be shown how R is computed for a
given pitch fl and dynamic level L. The main steps
are as follows. First, a one-pole, lowpass filter is de-
signed having bandwidth L. Second, the gain of this
Now, for any desired fundamental freque
filter at a "middle" frequency is computed. Third,
is computed so as to provide gain Gd(fl) = G
the dynamics filter is computed as a one-pole, low-
words, all fundamental frequencies are ma
pass filter having this gain at the desired fundamen-
the same amplitude. The value of R is fou
tal f,. The remainder of this section gives the equa-
solving
tions needed for these steps.
The reference frequency fm is chosen as the loga- l - R
rithmic middle (geometric mean) of the range to GL
GLR |I1 - Re-i2fI Ts
=
be used (a function of the particular musical con-
Squaring both sides of this
text and the sampling frequency):
resulting quadratic polynom
fm= e(l/2)(log(f)+ log(fl)) = /-f
R= R 1 - GL cos(2rfr Ts)
where f, is the upper pitch limit (< f/2), and fi is 1 - GL
the lower pitch limit.
The one-pole lowpass filter having bandwidth L + 2GL sinr V' T 1rf
- G2 COS2( rTf, Ts)
V G-
1- T1 L )
is given by
We use whichever value is < 1 in magnitude to en-
HL(Z) A I - RL sure stability.
L 1 - RZ-'
A family of frequency-response curves for Hd is
where shown in Fig. 6 for six fundamental frequencies in
octave steps from f, = 100 Hz to fl = 3,200 Hz. The
RL A e- LTs.
dynamic level in each case is L = 100 Hz. A verti-
cal line is drawn to each curve at the fundamental
The substitution RL = e- LT is a somewhat stan-
frequency to which it applies. The reference fre-
dard approximate formula for mapping bandwidth
quency fm is set to 282.84 Hz (the geometric mean
to pole radius.
of f, = 20 Hz and fu = f,/2), and the sampling rate is
The gain of the lowpass filter HL at the reference
fs = 8,000 Hz.
frequency is defined as
To add to the effect of simulated dynamics, it is
GL A IHL(e"7sfms)
sometimes helpful to do a bit of decay shortening
on the low soft notes, using a loss factor p as de-
1 - RL scribed previously. It is also possible to simulate
11 - RLe-'21rmTs' the spectral characteristics of soft notes by simply

64 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
turning the output of the algorithm on late, after cay shortening, is set to a relatively small value in
some of the high-frequency energy has died away. the last few milliseconds of the note. The duration
This latter method has the effect of diminishing the of the decay is dependent on the loss factor chosen.
attack noise, corresponding to a milder pluck. For A loss factor close to 1 simulates a string being
other decay-softening techniques, see the section damped with a soft material such as the flesh of the
entitled Varying Attack Characteristics. finger, while a smaller loss factor simulates damp-
ing with a hard material, such as a pick. It is useful
to compute p as a function of a desired t60. Substi-
Rests and the Ends of Notes tuting Eq. (19) into Eq. (5) and solving for p yields

In the case of a real string instrument, the player p(t60) = . A

e- l/fT t6o
cot6 s(= ICOSfT)I' = ln(
often plays two notes on the same string, playing
the second note before the first has died away. The For pitches above about 3 KHz
basic algorithm handles this without a problem foreven at the onset of notes. Onset clicks can also
all but very high pitches, since the discontinuity atoccur when a pianissimo setting of the dynamics
the beginning of the new note is perceived as a newfilter is used, since the masking effect of the on-
pluck rather than as a click. (With a real instrument,set noise burst is weakened by the dynamic filter.
there is a short time when the pick has muted the To alleviate this problem, it may be necessary to
vibration for the previous note but has not yet set multiply the output of the algorithm by an expo-
the string again into vibration, but this brief silencenential or linear envelope, rather than switching it
was not found to be necessary for realism in the on abruptly.
synthetic model.) A problem arises, however, when
there is a rest after a note. If the output of the algo-
rithm is abruptly turned off (replaced by zeros), a Glissandi and Slurs
discontinuity in the waveform results, causing a
click. A slur can be simulated by changing the order of
Even if the note is allowed to decay for a very the filter without reexciting it; that is, by chang-
long time, a note turned off abruptly may cause a ing the delay buffer length N. The result is analo-
click because the feedback loop has unity gain at gous to a performer's refretting a string without re-
0 Hz. To see this, note that Ga(O) = cos(O) = 1. Thus plucking it (what guitarists call the "hammer-on"
the final value of the waveform is the mean of the and "pull-off" technique). A rapid alternation of as-
initial noise burst. Since the pseudo-random-number cending and descending pitch changes gives a good
sequence used to initially excite the filter has a left-hand-trill effect.
mean of zero only in special cases or when an in- If the buffer length is gradually changed over
finite number of samples is taken, the 0 Hz compo-time, a crude glissando results. For low pitches and
nent can be significant. For uniform pseudorandomhigh sampling rates, it can be quite smooth, but for
white noise at amplitude A, the statistical variance higher pitches and lower sampling rates, the pitch
is A2/3, which implies a variance in sample meanquantization resulting from the integer-delay lengths
over N samples equal to A2/(3N). Thus the stan- becomes noticeable. In the upper range, the effect
dard deviation of the mean in a length of N sam- is similar to a glissando on a fretted instrument,
ples is A/\V3N. As N decreases, the probability of where the pitch changes in discrete steps, and it
having a large amount of energy at 0 Hz becomes can be musically useful. Of course, the synthetic
greater. Karplus and Strong describe a technique quantization is not in semitone intervals.
they call "dithering" to handle this problem, but A perfectly smooth glissando can be created by
this technique was not economical in our context.ramping C, the tuning coefficient, during the time
Our solution to the discontinuity problem is as between buffer-length changes. This technique can
follows: the loss factor p, as was discussed for de- also be used to create vibrato.

Jaffe and Smith 65

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
Sympathetic String Simulation that a loss factor p, such as was introduced with
reference to decay shortening, be used to provide
In a real string instrument, sympathetic vibrations energy dissipation.
of other open strings, as well as resonances in the The effect of several sympathetic strings can be
instrument body, give each pitch in the range of the created simply by a bank of parallel sympathetic
instrument an individual character and thus give strings, as defined above, each tuned to a different
the instrument as a whole a distinctive identifi- frequency. The resulting overall string-transfer func-
able character. In comparison, the basic form of the tion (omitting fine tuning and decay alteration for
Karplus-Strong algorithm, like many instrument- clarity) is then
simulation algorithms, has an excessive homogene-
ity of character throughout its range. One way to H(z) = l+z
remedy this situation is to feed the output of the 1-2 2
string into a body resonator. This technique has
produced impressive results, but its discussion is M -i
beyond the scope of this paper (see Smith 1983). (1
Another way to combat unnatural uniformity is i "il-p."I z-) - p
to create the effect of an instrument with sym-
pathetic strings, using a modified version of the where y is the fractional part of the plucked-string
basic algorithm. signal sent to the sympathetic strings, M is the
Just as a sympathetic string is set into motion bynumber of sympathetic strings, pi is the loss factor
the vibration of another string, the illusion of a for the ith sympathetic string, Ni = fJ/fi where f, is
sympathetically vibrating string can be created by the fundamental frequency of the ith sympathetic
exciting one copy of the string simulator by a small string, and fs/N is the pitch of the plucked string
percentage of the output from another (plucked) and hence of the played note.
string, tuned to a different pitch. The sympathetic-string version of the algorithm
In the discussion that follows, the algorithm thatis also helpful in creating a stereo or quadraphonic
is excited with the noise burst is referred to as the image. By distributing around the room the output
plucked string and the algorithm that is excited of several banks of differently tuned sympathetic
only by the plucked string is referred to as the sym-strings, all fed with the same plucked string, the
pathetic string. All partials of the plucked string effect of being inside a huge guitar can be created.
that do not coincide with those of the sympathetic Attractive musical results have been created by
string will be highly attenuated. Thus the sympa- replacing the plucked string with another computer
thetic string acts as a bank of very narrow bandpassinstrument, so that the bank of sympathetic string
filters with center frequencies at the partial fre- is used as a "reverberator." One can achieve the
quencies of the sympathetic string. The partials of effect, for example, of a clarinet being played into
the plucked string that will strongly resonate are an open grand piano with the pedal down.
those for which

fi = fA,
Simulation of a Moving Pick
where fk is the frequency of the kth partial of the
sympathetically resonating string, and f, is the fre- An effective means of simulating pick position is to
quency of the jth partial of the plucked string. introduce zeros uniformly distributed over the spec-
A problem can arise after several successive noise trum of the noise burst. This can quite accurately
bursts have excited the plucked string. The repeated simulate the effect of plucking a string at varying
reintroduction of energy into the sympathetic string distances from the bridge. The noise burst is fil-
may cause it to overflow. Therefore, it is essential tered with a comb filter, H,, having the difference

66 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
equation ner, the steady state fuses into a single note. In a
real mandolin, the strings are never perfectly tuned,
Yn = x, - Xn-,,N
and the beating effect of the slightly mistuned strings
where /, is the fraction of the string between the is a strong recognition cue. A better mandolin sim-
bridge and pluck point. When /u = 1/2, the even ulation simply uses two parallel forms of the algo-
harmonics are removed, and the effect is that of rithm, differing in pitch by a few cents and excited
plucking a string at its midpoint. Similarly, whenat slightly different times.
A = 1/10, every tenth harmonic is suppressed, and
the effect is like plucking a tenth of the way up the
Use of Other Filters in Feedback Loop
string. With /A = 1/N, the filter approximates a dif-
ferentiator, creating a sharp sul ponticello sound.
For the theory behind the simulation of pick loca-The use of filters other than a one-zero for Ha will
tion, see Smith's paper (1982a). give a different decay characteristic and, in turn, a
different timbre. However, care must be taken that
the amplitude response G, does not reach unity near
Varying the Character and Number of Attacksany partial frequency. Energy at any frequency fk
for which Ga(fk) = 1 will never decay, and if G,(fk)
Since the attack is very important in perception of > 1, the amplitude will grow exponentially until
timbre, it is advantageous to be able to alter its overflow.
character. To give a noticeably more noisy attack, For example, a one-pole filter with the pole be-
approximating the sound of a snap or "Bart6k pizzi- tween 0 and 1 gives a tone with the same attack
cato," the duration of the noise burst x, can simplyand a more mellow decay than with a one-zero fil-
be increased from tx = NTs to some tx > NTs. Simi- ter; that is, the higher partials decay more rapidly.
larly, the attack can be subdued by making 0 < tx Placing
< the pole at z = Q and normalizing the peak-
NT,, though for very small tx the pluck illusion (asamplitude response of the one-pole filter to unity
well as loudness) fades. yield the transfer function
A variety of other methods can soften the attack.
The string can be excited with a rich harmonic - IQI
H() = 1 - Ql"
spectrum rather than a noise spectrum, or with
some mixture of the two, with the sum of their am-
where IQI < 1 is required for stability of the one-
plitudes not exceeding 1. Another possibility is to
pole filter. The transfer function of the whole string
lowpass filter the noise burst. Yet another way tobecomes
soften the attack is to turn on the output of the
algorithm late, after some of the high-frequency en- 1
H(z) =
ergy has been filtered out. H 1z-z 1-IQI
A realistic simulation of the up-and-down pick- 1 - Qz-
ing pattern characteristic of a mandolin tremolo
1 - Qz-'
has been created by using a one-pole, lowpass filter,
1 - Qz- - (1 - IQI)z-
H,, to mellow the "up" picks while using the stan-
dard unfiltered noise burst for the "down" picks. Thus the difference equation is
A crude simulation of instruments having multi-
Yn = Xn - Qx,n- + QYn-1 + (1 - IQI)yn-N
ple strings tuned in unison, such as the mandolin
or bazooki, can be created by simply exciting the This version of the algorithm, with 0 < Q < 1 is
string with two successive noise bursts separated byuseful for pitches in the lower half of the range. In
a short amount of time, on the order of .05 sec. the upper range, the notes die away too fast to be of
While multiple attacks can be achieved in this man- use. This is because the one-pole, lowpass filter,
with Q > 0, filters out the high-frequency energy

Jaffe and Smith 67

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
Fig. 7. Block diagram con-
taining algorithm exten-
sions discussed in this
paper.

To sympathetic strings fk k _ kf,s(k)

Po + Pg(fk) s(l)
0 n)
fsl)s(_) pI
Pg(fk) = f1s(k)
X(n) HeHfO 0 Ha
where Po is the length of the loop in the absence of
the allpass Hg (typically P + 1/2). Methods for de-
Hg H,
signing allpass filters with prescribed phase delay
are reviewed by Smith (1983).

Summary
in the loop much more drastically than does the
one-zero, lowpass filter. A technique similar to that
Figure 7 shows a block diagram of the string simu-
used in dynamics simulation could be used to com-
lator with some of our revisions, where Ha is the
pensate for this trend.
feedback lowpass filter, Hb is the delay line, H, is
the allpass filter used for tuning, Hd is the lowpass
filter used to simulate dynamics, He is the comb fil-
Simulation of Stiff Strings
ter that simulates pick position, Hf is the filter that
simulates the difference between "up" and "down"
The spectral components of the basic algorithm
picking, and Hg is the allpass filter used to simulate
that have significant amplitude are almost perfectly
harmonic after the attack noise has been filtered string stiffness.
The simulator provides a high degree of flexibil-
away, corresponding, in the real world, to a per-
ity that begins to approach that of a skilled player
fectly flexible string. But since real strings always
performing on a real musical instrument. Many as-
have some degree of stiffness, it is desirable to alter
pects of a real string instrument have been simu-
the spectrum of the algorithm accordingly. The the-
lated. Pitch can be precisely specified, and articula-
ory of stiff strings (Morse 1976) indicates that stiff-
tion can be finely tuned. An expressive vocabulary
ness creates a stretching of the partials according to
is provided by a wide variety of performance nu-
the approximate formula
ances, including such "left-hand" techniques as
glissandi, slurs, and trills, as well as such "right-
fk = kfo 1 + 1 + k2 8
Tr 282) A kfos(k),
hand" techniques as variation in dynamic level,
4
pick position, and attack characteristics. These pa-
k = 1, 2,..., k2 < 2 rameters were found to be sufficient to create shaped
musical phrases. Furthermore, parameters such as
where fo is near the fundamental frequency,sustain and k time, body resonance, string flexibility,
is the partial number. The parameter 8 has been bridge and pick hardness, and degree of sympathetic
called the coefficient of inharmonicity; if 8string = 0, excitation, which, in the case of real instru-
then perfect harmonicity results. ments, are usually fixed at the time of instrument
This effect can be created, in principle, by construction,
intro- are available as performance parame-
ducing an allpass filter Hg(z) in the string loop ters. It is important to point out that this variety is
(Al-
len 1982) much as was done for the fine tuning at no time
of achieved at the expense of the integrity
pitch. The phase delay in samples desired for of an
the all-
basic sound. Rather, as is the case with a real
pass filter inserted in the feedback loop of a musicalharmonic instrument, the diversity exists within the
string simulator tuned to fl is given by solving bounds of a clearly defined sound domain.

68 Computer Music Journal

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms
 I

Conclusion damentals of Bowed String Dynamics." Acustica 43(2):

93-108.

The algorithm originated by Karplus and Strong McIntyre, M. E., R. T. Schumacher, and J. Woodhouse.
and extended by the methods outlined here has 1981. "Aperiodicity in Bowed-String Motion." Acusti
49(1): 13-32.
proven very useful as a computer instrument. In
McIntyre, M. E., R. T. Schumacher, and J. Woodhouse.
the process of composing May All Your Children
press. "On the Oscillations of Musical Instruments."
Be Acrobats and Silicon Valley Breakdown, it was Journal of the Acoustical Society of America.
found to be sufficiently flexible to allow for a wide Mont-Reynaud, B. 1982. Private communication.
range of musical expression and sufficiently idio- Morse, P. M. 1976. Vibration and Sound. New York:
syncratic to maintain a characteristic identity. We American Institute of Physics for the Acoustical So-
expect that new refinements of the algorithm will ciety of America. (Originally published in two editio
continue to arise. [1936 and 1948].)
Smith, J. 0. 1982a. "Synthesis of Bowed Strings." Pape
presented at the Acoustical Society of America Con-
References ference, Chicago, Illinois. (Reprints available upon
request.)
Smith, J. 0. 1982b. "Introduction to Digital Filters."
Abramowitz, M., and I. A. Stegun, eds. 1966. Handbook
Typescript. (Copies available upon request.)
of Mathematical Functions. Washington, D.C.: Na-
tional Bureau of Standards. Smith, J. 0. 1983. "Techniques for Digital Filter Design
and System Identification with Application to the Vio-
Allen, J. B. 1982. Private communication.
lin." Ph.D. Diss., Electrical Engineering Department,
Karplus, K., and A. Strong. 1983. "Digital Synthesis of
Stanford University.
Plucked-String and Drum Timbres." Computer Music
Steiglitz, K. 1974. An Introduction to Discrete Systems.
Journal 7(2): 43- 55.
New York: Wiley.
McIntyre, M. E., and J. Woodhouse. 1960. "On the Fun-

69
Jaffe and Smith

This content downloaded from

132.187.245.121 on Thu, 07 Nov 2024 14:32:11 UTC
All use subject to https://about.jstor.org/terms