676 IEEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-34, NO.

7, JULY 1986

A Transformation for Digital Simulation of Analog

Filters
FLOYD M. GARDNER, FELLOW, IEEE

Abstract-One method of simulatingan analog filter on a digital permittheactionof H&) tobesimulatedona digital
computer is to transform the continuous-time differential equation of the computer.
filter into a corresponding discrete-time difference equation. This paper The standard signal processing texts [ 2 ] , [3] describe two
describes the useful properties of the step-invariant transformation. useful transformations that have had widespread application:
the bilinear transformation and the impulse-invariant transfor-
mation. Each has its attractive features, but each has charac-
I. INTRODUCTION teristicsthat,arenotwellsuitedforsimulationofdigital
communications signals.
C OMMUNICATIONS systems are
studiedbymeansofdigitalsimulations.Examplesand
further references are found in [l].
increasingly
being
BiIinear Transformation
Typical programs partition the system into separate blocks, Thismethodistheonepresentedmostfavorablyinthe
and routines are written for each type of block. A filter is a literature. key It is defined as
block in anysystem; anysimulationprogrammusthave
methods for dealing with filters.
It is possible to simulate a filter in the time domainor in the
frequencydomain;bothapproachesarefound in extant
programs. This paper treats only time-domain simulation.
In essence, the computer simulates an analog filter in the where T, is the sampling interval.
time domain by transforming the continuous-time differential Attractive features include the following.
equation of the filter into a discrete-time difference equation. Mapping from the s-plane to the z-plane is one-to-one.
No uniquely satisfactoly transformation exists. The digital The entire lefthalf of the s-plane is mapped onto theunit
simulation necessarily is an approximation to the exact analog disk in the z-plane.
performance; any simulation inevitably introduces some dis- Response aliasing (to be encountered below) is avoided.
tortion.Ourobjectiveis to findacomparativelysimple Unattractive features include the following.
transformationthatcan be expectedtoprovideagood The frequency scale in the frequency domain is warped.
approximation to analog operations. In consequence of frequency warping, waveforms are not
Thestandardtextsondigitalsignalprocessing [ 2 ] , [ 3 ] well preserved between analog and digital representations.
discuss several common transformations in detail. However, a The last is serious
a deficiency when simulationof
subtle distinction needs to be made. The common transforma- waveforms is the overriding goal.
Presumably, the bilinear transformation can simulate wave-
tionsprovidemethodsofderivinggood digital filtersfrom
known, good analogfilters.Thesimulationproblem is to forms as accurately as may be desired by taking Ts sufficiently
derive a digital filter that closelymimicsaspecificanalog small. But that requires extra computing effort; it would be
filter, irrespective of whether the analog filter is good or not. preferable to
have atransformationthatbetter
preserves
The common transformations do not necessarily lead to good waveforms with larger values for T, than those required with
simulations as opposed to good filters. the binlinear transformation.
In this paper we first review the common transformations Impulse-Invariant Transformation
(bilinearandimpulse-invariant)andpointupboththeir
favorable and awkward properties in the context of a simula- Better waveform preservation is achieved by requiring that
tion of a digital communications system. Next, we derive a the response of the digital filter to a unit sampleidenticalbe to
transformation based upon invariance of the sampled response the sampled response of the analog filter to a unit impulse.
to a unit rectangular pulse. (Afterwards it was discovered that Specifically, let h,(t) and h&) represent the inverse Laplace
derivation via step-function invariance leads to the identical andz-transforms,respectively, of ET&) and H&). Then
transformation.)Lastly,weillustratecharacteristicsofthe impulse invariance defines the digital filter via
transformation by means of numerous examples.
We conclude that the proposed transformation is attractive
for many(althoughnotall)practicalfiltersanddeserves
consideration in preference to the common transformations. Attractive features include the following.
The frequency scale is not warped.
11. COMMON TRANSFORMATIONS Therefore, waveforms tend to be preserved.
Given an analog filter specified as a transfer functionH&), Unattractive features include the following.
wewant to findadigitaltransferfunctionHd(z)that will Multiplestripsofthe left-half s-plane all overlay onc
another on the unit diskin the z-plane. Mapping is not one-to
one.
Paper approved by the Editor for Computer-Aided Design of Communica-
tions Systems of the IEEE Communications Society. Manuscript received In consequence of the overlapping mapping, frequenc:
September 16, 1985;revised January 28, 1986. Thiswork was supported by a response is aliased.
contract from Lockheed Missiles and Space Company, Sunnyvale, CA. Gain becomes unduly large if T, is small.
The author is at 1755 University Avenue, Palo Alto, CA 94301. The literature states that only low-pass and bandpass filter
IEEE Log Number 8609079. can be treated by the impulse-invariant transformation, an'

0090-6778/86/0700-0676$01.OO 0 1986 IEEE

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GARDNER: TRANSFORMATION FOR DIGITAL SIMULATION OF ANALOG FILTERS 677

that these must be adequately band limited. Aliasing with high- Its sampled version is
pass or band-elimination filters is described as intolerable.
In fact, the situation is even worse; the impulse-invariant x ( n ) = 1, Osn<M
transformation collapses entirely for certain classes of filters,
including high-pass and band-elimination filters. The collapse =0, n<O or n z M . (6)
may be seen as follows. Advantages of this choice are as follows.
Represent the analog filter by a partial-fraction expansion A rectangular pulseof this description actually appearsin
many practical communications systems. In such systems, use
of a rectangular pulsein the definition of the transformation is
the most realistic and accurate choice that can be made.
The resultingtransformationissimpleandreadilyap-
where ck is the residue of the kth pole s k . The expansion of (3) plied. Lessimportant,butconvenient,isthe fact that the
requires that all poles be simple and that the degree of the simple pulse shape leads to relatively simple mathematics in
numerator of H,(s) not exceed the degreeof the denominator. the derivation of the transformation, as will be shown below.
(Repeated poles can be handled in a simulation by cascading Disadvantages are somewhat more nebulous.
networks of identical, simple poles or by modification of (3).) If the actual system being simulated uses a different pulse
For the case of co = 0, the impulse-invariant transformation shape (e.g., a band-limited Nyquist pulse), it is not evident
produces a digital transfer function that a transformation based upon a rectangular pulse leads to
the most accurate simulation.
A rectangular pulse is not band limited, so there must
inevitably be aliasing in a sampled simulation. Aliasing must
always be taken into account in a sampled system; its relation
to the proposed transformation will bediscussed in a later
However, if co # 0, then the analog impulse response contains section.
an impulse term coS(t), which cannot be represented by any On balance, it was judged that the clear advantages of the
finite digital sample. In particular, the unit sample (definedas rectangular-pulse definition outweigh the as-yet-unevaluated
u l ( n ) = 1 , n = 0; and u l ( n ) = 0, n # 0) isnotavalid disadvantages;theremainderofthispaperdealswiththe
representation of the unit impulse, which is a singular functiontransformationresulting frominvariance in response tothe
defined only through its integral [4].The underlying premise rectangular symbol pulse.
of impulse invariance fails if co # 0.
Derivation of Transformation
111. SYMBOL PULSE INVARIANCE
Response of the analog filter to the square pulse x ( t ) is
Properties of theestablishedtransformationsprovidea given by the convolution integrals
guide to desirable characteristics of a different transformation
moresuitableforsimulationpurposes.Suchcharacteristics
include the following.
Waveforms should be preserved insofar as possible.
Gain should be preserved in the transformation.
The transformation should not collapse for any common-
place filter type.
= 1:
h,(t- r ) dr, t? T=MTs
The mathematics should be simple (a quality offered by
both of the established transformations). *,
A stable analog filter should transform to a stable digital =
Jo
1.
h , ( t - r ) dr, OstlT=MTs
filter. (Both established transformations preserve stability.)
Aliasing should be minimal. = 0, t<O. (7)
The proposed transformation offers most of these desirable
features. (The last line expresses causality of the filter. Causality will be
assumed henceforth.)
Definition If all poles of the filter are simple (an assumption that can be
In asynchronousPAMcommunicationssystem,thedata removed,butisappliedforeaseofexplanation),thenthe
stream consists of a sequence of pulses of identical waveshape impulse response of the filter can be expanded in accordance
x ( t ) , spaced at uniform intervals of duration T. Each pulse with the partial-fraction representation of (3) in the form
carries information on one data symbol by meansof the pulse
amplitude. Simulation is concerned with the response y ( t ) of K
the filter to the pulse waveform x ( t ) . h,(t) = co6(t) +
ckeskk'u(t). (8)
The waveforms are sampled at intervals T, = T / M , where k= 1
M is an integer. Samples of waveforms are represented by
This representation of h,(t) is substituted into the convolution
x(n7'') = x(n) for the input and y ( n ) for the output. We want
integrals and the output y ( t ) is sampled at times t = nT, to
a transformation suchthat the output y ( n ) of the digital filter in
give
response to the sampled pulse x(n) is identical to the sampled
output y(nTs)of the analog filterin response to the continuous
pulse x ( t ) .
What pulse shape should be chosen forx(t)? One attractive
candidate is the time-limited, rectangular pulse with duration
of exactly one symbol interval.
That
is,
theproposed
transformation is based upon the pulse
x(t)=1, Ost< T
=0, t < O or t r T. (5)

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678 IEEE TRANSACTIONS
COMMUNICATIONS,
ON VOL.
NO. COM-34, I. JULY 1986

Take the z-transform of y ( n ) to obtain y(n = 0) should be zero, and that constraint is imposed
by the
z-' in the numerators.
It is readily shown that (15) and ( 3 ) have the same dc gain.
That is.

so that gain is preserved. There is no gain magnification such

After some manipulation, (10) can be put into the form
as arises in the impulse-invariant transformation.
Consider a complex pole in the s-plane, located at sk =
- a k + j D k , where ( Y k and @ k are real, nonnegative numbers.
Then the transformed pole in the z-plane is located at Zk =
e-akTsejokTs. As noted earlier, a pole in the left half of the s-

plane maps into the interior of the unit circle of the z-plane, so
thetransformationpreservesstability.However, if &Ts is
large enough, the z-plane pole can be aliased to small angles
correspondingtosmallvaluesof PkT,. Thatis,thepole
position can be aliased unless T, is selected sufficiently small.
To avoid aliasing of poles it is necessary that

6k Ts< (17)
Take the z-transform of x ( n )
for the largest P k to be included in the simulation.
M- I 2-M- 1 The same constraint applies in the impulse-invariant trans-
X(z)= z-fl=- formation.
n=O 2-1- 1
Examples
and divide it into Y(z) to obtain TableIshowsdigitaltransferfunctionsresultingfrom
symbol-pulse-invariant transformations of a number of simple
Hd(z)= y ( z ) / x ( z ) analog transfer functions of first and second order. Inspection
of these examples provides additional insight into the charac-
teristics of the transformation.
Comments follow on several of the examples.
High-Pass, First-Order(B): The digital filter has null a at z
Since = 1, correspondingtothedc null oftheanalogfilter.
Therefore, the symbol-pulse-invariant transformationis usable
withhigh-passfilters, in contrasttothefailureofimpulse
invariance with the same filter.
The bandpass filter ( F ) also has a dc null in the analog
response, which also transforms to anull at z = 1 in the digital
we finally obtain the desired digital transfer function filter.
I I All-Pass, First-Order(C): Since the digital pole is at e-bTs,
the corresponding zero for digital all-pass response should be
+ bT
at z = e ,. In fact, the zero is transformed to 2 - e-bTs.If
I 1 bT, 4 1, then e
bT
* 1 + bT,and 2 - e-bTS5 1 + bT,, so
the zero would be nearly in the correct position.
which is the transformation that we have been seeking. Suppose that bT, is not small. For a numerical example, let
e-bTS= 0.4 so that e+", = 2.5. However, 2 - e-bTS= 1.6,
Properties of Transformation which is the actual position of the zero. That is considerably
Compare the symbol-pulse-invariant transformation of (15) removed from 2.5, so the transformation does not preserve an
to the impulse-invariant transformation of (4). Temporarily all-pass characteristic.
assume co = 0 so that impulse invariance is properly defined. All-Pass, Second-Order: Similarcommentsapplytothe
The poles for both approaches are identical. Since impulse second-order all-pass network ( G ) . For T, sufficiently small,
invariance maps a stable analog filter into a stable digital filter, the zeros are nearly in the correct location. As a numerical
the samestability preservation will be provided by the symbol-exampleforlarger T,, assume that aTs = @Ts= d 2 . For
pulse invariant transformation. those parameters, the di ita1 poles lie at kj0.208 while the
+j 1.9,
Residuesof (15) are not thesameasresidues of (4). zerosareat 0.935 e- . Thezeroshavemigratedto the
Therefore, the two transformations will notyieldthesame interioroftheunitcircleforthischoiceofparameters.
zeros in their two transfer functions. However, if lskTsl 4 1, Response of the digital filter will not be nearly all-pass.
then 1 - eSkTsf - skTs andthenumeratorofeachpartial Low-Pass,Second-Order, All-Poles (D): Although thc
fraction in (15) reduces to ckTSz-l.Thus,thezeros will analog filter has no finite zeros, the transformationintroduce5
closely approach the zeros of the impulse-invariant transfor- a real zero falling someplace on the interval ( - 1, 0) in the z.
mation for small T, and co = 0. plane. If T, is small, the zero approaches z = - 1, therebJ
A 2 - l is present in the numerator of each partial fraction. providing better attenuation in the digital filter than does thc
Thisissimplyadelayandcouldbe neglected if co = 0. analog filter at the same frequency.
Nonetheless, the delay is a valid representation of a filter with For a numerical example, parameters of aT, = 0.1 and 07
morepolesthanzeros;response of suchafiltertoany = d 8 were arbitrarily assumed and amplitude responses o
nonsingular input must be zero at time t = 0 + . Therefore, theanaloganddigitalfilterswerecalculated.Resultsan

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GARDNER: TRANSFORMATION FOR DIGITAL SIMULATION OF ANALOG FILTERS 679
TABLE 1
EXAMPLES OF SYMBOL-PULSE INVARIANT TRANSFORMATIONS

A . Low Pass -
b
( f i r s to r d e r ) stb

B. High Pass -
S
(firstorder) s+b

C . Allpass b- s
-
(firstorder) s+b

0. Low Pass
(second-order,
a l lp o l e s )

E. Low Pass
(second-order,
one z e r o )

F. Bandpass
(second-order)

G. Allpass
(second o r d e r )

H . Null
(second o r d e r )

I . Lead-Lag -
s+a
s+b

J . Integrator a/ s

plotted in Fig. 1 ; the ordinate is normalized to unity at zero Frequency response for thesetparameters is piotted in Fig. 2
frequency. (Of course, the frequency response of the digital for both the analog and digital filters.It is apparent,that Tshas
filter is periodic in l / T s ; periodic repetition is not shown in been chosen overly large in this example, since the low-pass
Fig. I . ) nature of the filter doesnot appear withiti the Nyquist interval.
The chosen parameters lead to appreciable peaking in the Nonetheless, the frequency responses qfthe analog and digital
response. It was thought that peaking would bring out versions are nearly identical at low frequencies and show only
discrepanciesbetweenthedigitalandanalogfilters. Atlow moderate discrepancies at higher frequencies.
frequenciesthere isnot sufficientdifferencebetweenthe Null filter (H): The analog filter has .a null at w = ( a 2 +
responses to, be seen on the plotted scale. At thepeak P 2 ) there is a pair of zeros on the imaginary axis of the s-
frequency (which calculated to be identical in both filters), the plane. Unforiunately, those zeros do no$ transform to the unit
peak response differs by about 0.05 dB. circle of the z-plane. For assumed parameter valuesof aTs =
Discrepanciesbetweenthetwofilterresponses begin to PT, = 7d2, the poles are at z = k j 0.208, but the zeros areat
showupathigherfrequencies, with thedigitalresponse z = 0.68 e*Jl.26. The null will not beverydeepand it is
showing more attenuation than the analog. The effect of the greatly shifted from its analog-domain frequency of T / ( ~ ) ” ~ T ,
negative, real zero is apparent. = 2.22/Ts.
Low-Pass, Second-Order, One Zero (E): The zero of the Integrator: Equation (15) is indeterminate if sk = 0. BY
analog filter transforms to a finite zero in the digital filter. using h,(t) = au(t) in the derivation of the transformation, we
Location of the digital-filter zero is a complicated function of obtain H&) as shown on line J of Table I. Notation u ( t )
the three parameters of the filter; it has not been possible to denotes the unit-step function.
gain much insight into the range of the zero’s position, other
than to recognize that it must be real. IV. CONCLUSIONS
Numerical valuesof aT, = aT, = PT, = n / 2 were A symbol-pulse invariant transformation has been shown to
assigned. (These would be considered large values.) The poles be
a useful tool for simulation of
analog
filters in a
lie at z = kj0.208 while the zero is at z = 0.208. communications link. This new transformation is a modifica-

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 680 IEEE TRANSACTIONS ON COMMUNICATIONS, VOL. COM-34, NO. l , JULY 1986

function H d ( z ) givenin (15).’ Derivation via symbol-pulse

invariance is helpful because:
one gains confidence that the method is a valid approxi-
mation (as opposed to steps);
0 0 no convergence difficulties are encountered in obtaining

the digital transformation of an integrator.

The method transforms a stable analog filter into a stable
digital filter (as does the impulse-invariant method) and avoids
-10 thegainmagnification for small T, encounteredwiththe
m
U
impulse-invariant transformation.
W
The sampling interval T, must be takensufficiently small so
VI
c as to avoid aliasing of complex poles. (The same restriction
W
; -20 should also be appliedto the impulse-invariant transformation.
CL
aJ
No such restriction is necessary with the bilinear transforma-
U tion.)
-
.-*
3

5a -30
Effects of thenewtransformationonall-polefilters,
filterswithlow-frequencyrealzerosappeartobewell
or
W .. behaved.However,becausezeros are notmapped in any
>
.-
I-’
m
c
‘. straightforward manner, filters with complex, high-frequency
2 zeros do not fareaswell.Examplesshowedthat all-pass
-40

0 0.1 0.2

Frequency, f T S
0.3 0.4
1 0.5
analog filtersare not transformed to all-passdigital filters, and
that analog null filters may transformto digital null filters with
poor nulls thatareseriouslyshifted

ACKNOWLEDGMENT
in frequency. If zero-
position is important, it may be preferable to use the matched
z-transformation of [3, p. 2241.

I want to thank A . Weiss of Lockheed who painstakingly

Fig. 1. Amplitude response of low-pass filter. (All poles: cxT, = 0; PT, =
~18.) checked my calculationsandtherebysavedmefromthe
embarrassment of publishing erroneous results.
rn
REFERENCES
[l] Special Issue on Computer-Aided Modeling, Analysis, and Design of
Communication Systems, IEEE J. Select. Areas Commun., vol.
SAC-2, Jan. 1984.
[2] A. V. Oppenheim and R. W . Schafer, Digital Signal Processing.
Englewood Cliffs, NJ: Prentice-Hall, 1975, ch. 5.
[3] L. R. Rabiner and B. Gold, Theory andApplication of Digital Signal
Processing. Englewood Cliffs, NJ: Prentice-Hall, 1975,ch.4.
[4] A.Papoulis, The Fourier Integral. New York: McGraw-Hill, 1962,
5 0 0.1 0.2 0.3 0.4 0.5 Appendix I .

*
DI

Frequency, f T S

Fig. 2. Amplitude response of low-pass filter. (One zero: UT, = aT, = PT,
= T12.) Floyd M. Gardder (S’49-A’54-SM’58-F’80), for a photograph and biogra-
phy, see p. 429 of the May 1986 issue of this TRANSACTIONS.
tion
of thewell-knownimpulse-invariancemethod.The
modifications permit simulation of high-pass and band rejec-
tion filters (an impossibility with the impulse-invariant trans- ’ A reviewer has commented that the identity between step invariance and
formation). Simulation is perfect for systems using rectangular symbol-pulse invariance should have been obvious. He arguesthat the filter is
data pulses. a linear, time-invariant operation and that the pulse is the sum of two steps.
Actually, further analysis has revealed that a step-invariant Therefore, the filter pulse response is the sum of the responses to two steps,
transformationyieldsidenticallythesamedigitaltransfer applied individually.

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