AAS 07-340

ENHANCEMENTS OF REPETITIVE CONTROL USING

SPECIALIZED FIR ZERO-PHASE FILTER DESIGNS

Jiangcheng Bao1 and Richard W. Longman2

Repetitive control can cancel the effects of periodic disturbances on a feedback

system, e.g. it can be used on a satellite active vibration mount to cancel the
effects on fine pointing equipment of jitter from imbalance in a momentum
wheel. To produce stability robustness to residual modes, real-time zero-phase
low-pass filtering must be used in practice that cuts off the learning process
above some frequency. Improved methods of designing such filters are
presented, using quadratic programming with inequality constraints, allowing
one to use a higher cutoff. Also, digital control often makes use of anti-aliasing
filters. In the repetitive control problem this can cause unwanted distortions of
the signal. It is shown here how one can employ a non-causal zero-phase FIR
digital filter for this purpose that avoids the unwanted distortions.

INTRODUCTION

Many control systems are subject to periodic disturbances. Control systems that do the fine
pointing of instruments on board satellites in many cases are subject to jitter caused by slight imbalance in
a momentum wheel, a cryo pump, or in reaction wheels, or control moment gyros. Repetitive control (RC)
theory presents methods of adjusting the command to feedback control systems, learning over time what
command to give the control system such in theory it totally cancels the effects of the period disturbance on
the output. Reference [1] develops very effective methods of designing FIR compensators in order to
stabilize the learning process. Reference [2] extends the methods to handle multiple unrelated periods, as
would be needed in the case of 3 reaction wheels, or 4 CMGs. The design process can start with an
analytical model, or it can start from experimental frequency response data [3]. To robustify designs one
can introduce a zero-phase low-pass filter, as in the Q filter of [4,5], or the zero-phase Butterworth filter in
ILC in [6] and [7]. The Q filter in [4,5] is a very simple and hence far from ideal filter, and [8] investigates
designing improved high order FIR zero-phase low-pass filters for RC by methods similar to those in [1]. It
is the purpose of this paper to further improve such filter design specifically for the needs of RC.
Without such a frequency cutoff, RC aims for zero error at the fundamental frequency and all
harmonics up to Nyquist frequency. This is an unusual requirement in feedback control design, and it
results in the need for rather accurate modeling up to Nyquist frequency. Phase errors in the model of 90
degrees are usually enough to destabilize the RC system, and one missing pole can supply such a phase
error. The instability can be very slow, because the response at high frequency is very small, but high
frequency components of the signal eventually accumulate and the instability becomes evident. One set of
experiments done at Xerox Corporation did not exhibit evidence of instability until roughly 2000 periods of
the repetitive process, but nevertheless with a long enough run the system was seen to be unstable. The FIR
filter is used to cutoff the learning process above the frequency when the true system and the model start to
differ by something like 90 degree phase difference. Since our model is imperfect and we do not know
what the real model is, the adjustment of the cutoff must be made in hardware.

1
Doctoral Candidate, Department of Mechanical Engineering, Columbia University, New York, NY 10027
2
Professor of Mechanical Engineering, and Professor of Civil Engineering and Engineering Mechanics, Columbia
University, New York, NY 10027
 The FIR filter has two special requirements, it must operate in real time, and it must be zero-phase,
which means it is a noncausal filter. This is possible because the filter is operating on older data. Reference
[3] shows how to design such filters in a straightforward way. But FIR filters have ripple in the passband,
and in the stopband. In practice, the passband ripple can necessitate the use of a much lower cutoff
frequency of the learning than would be possible with an ideal filter. It is this difficulty that motivates much
of the present work. One may also want as high a cutoff as possible based on the result in [9] for ILC that
showed the design cutoff needed to produce stability is often higher than the resulting effective frequency
cutoff on the disturbance in the sensitivity transfer function, a phenomenon which is studied here for RC.
With this motivation, this work develops a method based on quadratic programming with inequality
constraints to design filters that have no ripple going above unity in the passband. The design parameters of
the filters are studied and tuned for the RC objective. A transition band is introduced to improve the
performance in the passband and in the stopband. Examples are given in which the design allows a much
higher cutoff frequency.
The use of high sample rates is studied for its effect on the filter design and performance. Often
people will choose to use the fastest possible sample rate consistent with real time computation
requirements. It is shown that a fast sample rate with a cutoff frequency at a correspondingly low percent of
Nyquist frequency aggravates the design and requires extra attention. As the sample rate is increased, the
number of terms in the FIR filter, and the real time computation load, must increase in order to maintain the
same level of filter performance.
A second use of the design process developed here is also studied. In many implementations of
digital control it is important to use an anti-aliasing filter. Normally, one considers that this must be an
analog filter appearing just before the analog to digital converter, aiming to significantly decrease any
energy above Nyquist frequency that will be folded by the sampling to be erroneously interpreted signals
below Nyquist frequency. As such, it must be a causal filter and it will introduce attenuation and phase lag
in the measured feedback information. This is a serious issue in repetitive control, since it can only create
zero error in following some measured signal, and these measurements now have distortion compared to
the output one wants to control. Here we present a method of handling this problem, making use of the
same FIR design process. We make use of a fast analog to digital converter, apply the anti-aliasing filter as
a digital non-causal FIR filter, and then downsample. The use of a digital filter operating at a high sample
rate was done in the satellite vibration mount experiments in reference [10]. In the repetitive control
situation, the filtering is done on data from the previous period of the periodic disturbance. And this means
we have the opportunity to use a real-time, zero-phase FIR filter for anti-aliasing, an unusual opportunity in
digital feedback control.

MATHEMATICAL BACKGROUND

A typical repetitive control system is shown in Figure 1. In the diagram, G (z) is the transfer function
of the closed loop feedback control system, YD (z) is the desired trajectory for the system with a period of p
time steps (which includes constant trajectories as a special case), and V (z) is a periodic output disturbance
also with a p time step period. Wherever a periodic disturbance enters the feedback control system, there is
!
an equivalent periodic disturbance that one can add to the output of the transfer function x(k) , or X (z) in
!
the z-transform version, to form the system output y(k) or Y (z) , and this is v(k) , or V (z) . Hence
!
Y (z) = G (z)U (z) + V (z) = X (z) + V (z) (1)
! !
In Fig. 1, F (z) is a compensator applied!to the error
! ! H (z)!is a low pass filter designed to
E(z) , whereas
stop the learning process above some cutoff frequency. The repetitive control system aims at obtaining
!
perfect tracking of the periodic command YD (z) or eliminating the influence of the periodic disturbance
V (z) on the output, or both. The simplest form of repetitive control adjusts the command u(k) to the
! ! !
feedback control system every time step according to
! u(k) = u(k " p) + # e(k " p + 1) (2)
! !

!
 where " is the repetitive control gain, and e(k) = y d (k) " y(k) represents the error, the desired output
minus the actual output. The one time step added to the argument in e(k " p + 1) reflects the delay of the
system between the time step when an input is changed and the time step it first influences the sampled
output.
!
Running equation (2) recursively!makes a sum of all errors observed in the past for the current phase of
the periodic disturbance, which is essentially a discrete time !equivalent of an integral. In practice, in order
to stabilize the repetitive control process, we multiply the gain by a compensator, F (z) , which is designed
to include the one time step shift in the error, and we introduce a zero-phase low-pass filter H (z) that
operates on the signal before it is applied to the system, cutting out high frequencies where the model is no
longer accurate enough to produce a stable design. The z-domain repetitive control law takes the form
!
"p !
U (z) = z H (z)[U(z) + #F (z)E(z)]
$ #H (z)F (z) ' (3)
=& p )E(z)
% z " H (z) (

Setting H (z) = 1 and F (z) = z in equation (3) produces equation (2).

The sensitivity transfer function S(z) from desired trajectory YD (z) and disturbance V (z) to the
!
corresponding error E(z) , and the associated difference equation whose solution is the error history, are
! given by !
! S(z) = E(z) z p " H (z) !
= p ! (4)
! YD (z) " V (z) z " H (z)[1" #F (z)G (z)]
{z p " H (z)[1" #F (z)G (z)]}E(z) = (z p " H (z))(YD (z) " V (z)) (5)

Once equation (5) is cleared

! of fractions, one can interpret the powers of z times transformed variables as
time shifted values of these variables and create a difference equation. Concerning stability, we want all
!
roots of the characteristic polynomial in the curly brackets on the left to lie inside the unit circle. Direct
application of Nyquist stability criterion to the repetitive control loop is very difficult because of the p poles
on the unit circle, but with appropriate modifications one can reformulate the problem to avoid this
difficulty [11]. Nevertheless, since p is usually a large number, one needs to be careful with the numerical
computations. On the other hand, [11] proves that if

H (z)[1 " #G (e i$T )F (e i$T )] < 1 (6)

for all " up to Nyquist frequency ( T is the sample time interval), then the Nyquist stability condition is
satisfied, making (6) a sufficient condition for stability. Because p is normally a large number, the
!
difference between this sufficient condition for stability and the necessary and sufficient Nyquist stability
! condition is usually insignificant
!
[12].
Reference [1] develops a method of designing the compensator F (z) to make the square bracket term
in (6) as small as possible in the sense of minimizing the sum of the squares of the values summed over
frequencies. An FIR filter is generated that approximates the inverse of the system frequency response, and
it has the simple form of a linear combination of measured errors at chosen time steps in the previous
period !
m"1 m"2 0
F (z) = a1z + a2z + L+ a m z + La n"1z "(n"m"1) + a n z "(n"m)
(7)
= (a1z n"1 + a 2 z n"2 + L+ a m z n"m + L+ a n"1z1 + a n z 0 ) / z (n"m )

The z 0 term corresponds to the error one period back, e(k " p) , and terms are included both forward and
backward from this time. Experience indicates that after picking the number of errors included, n, then m
!

! !
 should be chosen to equal 1+ n / 2 when n is even, and 1+ (n + 1) / 2 when n is odd, for systems with a one
time step delay through G (z) . The coefficients in (7) are chosen to minimize

N
! !
i# j T i# j T i# j T i# j T
!
JF = $[1" G (e )F (e )]W j [1 " G (e )F (e )] * (8)
j= 0

The asterisk indicates the complex conjugate, the W j are weighting functions that are chosen as unity in
the present study, and the sum is taken over a chosen number of equally spaced frequencies from zero to
!
Nyquist. One adjusts the values of n and m in order to make the square bracket term in (6) as small as
possible. If one gets it to be less than one in magnitude at all frequencies based on the system model used,
then one has a stable design for that model. ! The cutoff H (z) is not needed based on the model, but is likely
to be needed in applications due to model error at high frequency. One may also make a design that does
not keep the magnitude of the square bracket term less than one based on the model, and use the cutoff to
stabilize the analytical design.
!
Now consider the forcing function in difference equation (5). If there is no cutoff filter, then H (z) = 1,
and the forcing function becomes zero because both the command and the periodic disturbance are of
period p time steps. Then if the system is stable, the error will become zero for all frequencies of period p.
If H (z) is an ideal cutoff filter, equal to one in the passband, and equal to zero in the stopband, the forcing
function is zero for all frequencies of period p below the cutoff, and the components ! above the cutoff
p p
remain unaltered by the factor z " H (z) = z . Nevertheless, disturbances in this region can still be
amplified when going from forcing function to the resulting particular solution to the difference equation,
!
and are predicted by the sensitivity transfer function (4) which will obey the waterbed effect.
!
PERFORMANCE OBJECTIVES IN DESIGNING THE ZERO-PHASE LOW-PASS FILTER

There are many characteristics of the behavior of the low pass filter that are of importance in the
performance of the resulting repetitive control system, as listed here.

(1) The filter must operate in real time.

(2) However, it must also be zero phase. Normal causal filters like a Butterworth filter do a reasonable job
of remaining near unity magnitude response in the passband, but the phase change can be very large.
Examining the right hand side of (5), if H (z) were of magnitude one at some frequency, but the phase
angle was 180 degrees, then the z p " H (z) would double the input disturbance in the forcing function
instead of eliminating it. Normally zero phase filters cannot be causal, and hence could not operate in real
time, but in repetitive control, we are filtering a signal from roughly p time steps back, so a non causal filter
!
can still be used. Reference [8] chose to design an FIR filter of the same form as equation (7)
!
n
H (z) = a n z n + a n"1z n"1 + L+ a 0 z 0 + L+ a"(n"1)z "(n"1) + a"n z "n = #a z k
k
(9)
k="n

except that the number of forward gains must be equal to the number of backward gains, the number of
gains must be odd given by 2n +1, and they must satisfy a"k = a k , for k = 1,2,K,n . This symmetry of the
!
coefficients creates zero phase in the passband (and alternatively 180 deg and zero going throught the
stopband peaks). Ideally, the filter should be equal to one below the cutoff, and zero above, so that [8]
chose to minimize
! ! !
jc N H "1
i# j T i# j T i# j T i# j T
JH = $ [1 " H (e )]W j [1 " H (e )]* + $[H (e )]V j [H (e )]* (10)
j= 0 j= j c +1

!
The first sum is over the passband which ends with the frequency " j with subscript j c , and the second
sum is over the stopband. Weights W j ,V j can be adjusted to change the relative importance of each. The
purpose of this paper is to improve this design process.
(3) The zero phase filter can have the purpose of cutting off ! the learning process
! above some frequency in
order to not ask the hardware to work very hard trying to eliminate error components far above the
bandwidth of the system.!But very often the more urgent objective is to stabilize the repetitive control
process by making equation (6) satisfied. In this case, one is likely to have to make the cutoff frequency
choice based on hardware tests, turning down the cutoff until stable behavior is achieved. It will be
achieved once the compensator F (z) which was designed based on our model of G (z) makes (6) satisfied
when the G (z) in the equation represents the real world system behavior. FIR filters as in (9) are not ideal.
They have ripple in the passband, meaning that the magnitude response oscillates above and below one,
with the amplitude increasing until the response starts its decay toward the stopband. As the frequency
! !
increases toward that frequency for which 1" G (z)F (z) goes above unity, having a ripple in H (z) that
!
makes its magnitude go above unity, will force the designer to use an unnecessarily low cutoff frequency.
We seek to eliminate this difficulty by asking that the gains chosen in (9) minimize (10) subject to the
inequality constraints that the magnitude of H (z) is always less than or equal to unity. This will allow us to
! !
have the highest possible cutoff frequency that maintains stability.
(4) In addition to this inequality constraint within the passband introduced for stability, we are also very
much interested in having the magnitude stay as close to unity as possible throughout the passband. Our
!
aim is to very precisely follow periodic commands, or very accurately eliminate periodic disturbances
within the passband. Any error in the magnitude of the output of this zero-phase filter will cause z p " H (z)
to no longer be the precise difference of periodic signals from one period to the next, and hence the
difference equation (5) will have a non zero forcing function and the solution to the stable difference
equation will converge to this non zero particular solution. The final error level reached when eliminating
errors within the passband can only be as perfect as the filter is in the passband. ! The weights in the
objective function can be adjusted with this and other aims in mind.
(5) The above point applies to all frequencies in the passband, but one frequency that is often of particular
importance is DC. Hence, we seek coefficients in the FIR filter (9) that minimize (10) subject to the
inequality constraints above, and in addition subject to the equality constraint that the DC gain of the filter
is unity. Mathematically, this requires that the sum of all coefficients in (9) equals one.
(6) There is also ripple in the stopband, which in the magnitude plot takes the form of a series of decaying
bumps with sharp cusps at the junction between one bump and the next. The phase in the passband has
been made zero by the choices above, and this zero phase is maintained until the magnitude response first
decays to zero output. As the frequency goes above this point into the first bump, the sign of the filter
output changes, and the phase angle flips by 180 degrees. The forcing function on the right of difference
equation (5) would just be the periodic input functions themselves if H (z) were a perfect zero, but for
frequencies within the first bump, the amplitude of the forcing function is increased by the height of the
bump. The same is true of all odd numbered bumps going up in frequency in the stopband. On the other
hand, the forcing function is decreased in amplitude for frequencies within the even numbered bumps.
Nevertheless, in general one would like to eliminate the ripple ! as much as possible in the stopband. The
relative importance of this objective versus other objectives can dictate the compromise one must make in
designing the filter, and is dependent on the error environment of the physical problem.
(7) The design used in [8] in equation (10) has one summation for the stopband frequencies, and one
summation for the passband frequencies, without any gap between the two frequency sets. Here we
introduce a transition band by replacing the j = j c + 1 in the second summation in (10) by some larger
number, so the frequencies within the gap introduced will not be addressed by either summation. Very
often applications of FIR low pass filters have the primary object of producing a very sharp cutoff. The
objective here may be rather different. Note that for all frequencies below the first bump in the stopband,
!
the phase is zero. This means that the factor z p " H (z) in the forcing function of equation (5) will be
attenuating the amplitude of the forcing function until the first bump of the stopband is reached. Hence, a
wide transition band may actually be of help in this application. Of course, it is not really the amplitude of
the forcing function that matters, it is the resulting particular solution, and numerical studies below can
suggest that with model error one may!want to make compromises on this issue.
 THE QUADRATIC PROGRAMMING PROBLEM

The above discussion produces a numerical optimization problem in quadratic programming.

Mathematical Statement of the Problem: Given the sequence of numbers e(k) to filter, the filtered
values are given by

f (k) = a n e(k " n) + a n"1e(k " n + 1) + L + a 0e(k) + a1e(k + 1) + L + a n e(k + n) (11)

!
T
The unknowns: x = [ a 0 2a1 L 2a n ]
T
!
Coefficient matrix: b j = [1 cos " j L cos n" j ]
Normalized frequencies: " j = j# / N ; j = 0,1, 2,K, N $ 1
! frequencies: j = 0,1,K, j (want filter to be equal to 1 for these j)
Passband p
Stopband !frequencies: j = j s ,K, N " 1 ( j s " j p + 1 , want filter to have zero output at these j)
Transition band!frequencies: j p + 1 " j " j s # 1 (there is no transition band unless j s > j p + 1 )
Optimization !Criterion:
!%' p )
j N #1
!
2' %1 )
! $ T
J = min&" [1 # b j x] +
x '
2
$ [b j x] * = min& x T Px + c T x *
T

'+ x (2 ! +
(12)
( j= 0 j= j s

Without constraints this is a linear problem, just differentiate, set to zero, and solve the linear equations.
The " adjusts the relative importance of passing the desired frequencies without changing their amplitudes,
!
versus the importance of successfully stopping the remaining frequencies.

Constraints:
!
bTj x " 1 j = 0,1,K, j c
(13)
a 0 + 2a1 + L + 2a n = 1

The first ensures that there is no amplification of the frequency components in the passband, and the second
says that the DC or constant component comes through the filter unchanged.
!
Quadratic Programming

The numerical solution of this quadratic programming problem can be quite efficient. There are
many numerical algorithms that appear rather different but have very strong similarities [13]. An outline of
an effective approach used by Matlab is as follows. One employs an active set strategy. First one needs a
feasible solution. To obtain it, one defines a linear programming problem that subtracts the same slack
variable from the left hand side of all inequality constraints, and then seeks to minimize the slack variable.
One can use the Simplex algorithm to solve such a problem. One makes the slack variable large enough so
all inequalities are satisfied, and then one minimizes it. The result is that one (or more) of the inequality
constraints is now an equality constraint, and the rest of the constraints are satisfied as inequalities.
Now that we have a feasible starting solution, one can start addressing the QP problem. Take the
minimization subject to the real equality constraint, and whichever of the inequality constraints is equality
from above (which is the initial active set). Get a solution for the linear equations that result using the
Karush Kuhn Tucker (KKT) conditions. Plug the solution into the other inequality conditions that are not
on the boundary. Examine which ones are violated and see how much. Go the minimum distance in that
direction to the first of these boundaries. Add this new inequality to the active set. Repeat, as long as the
new direction that you calculate doesn’t include any new inequality constraints (or you have the number of
equality conditions equal to the number of variables to adjust, in which case there is no minimization
possible).
 Then check if you can omit one of the equalities that are active, currently being treated as equality.
Check the adjoint variables. All must be of the right sign based on the KKT conditions. If all fulfill the sign
condition, you have found the solution. If not, pick one of these that has the wrong sign and start over on
the QP computation above without this equality. Only take out one at a time. There are special strategies to
decide what order to take them out. One can prove that under normal conditions, this produces a finite
procedure, but it can be “np hard” meaning the number of steps may grow exponentially with some
measure of the information contained in the problem, e.g. the number of inequalities and number of
variables, etc. However, solution is very fast in most cases, and only in extreme cases does one see the
exponential dependence. To handle such cases there are interior point methods that grow in a polynomial
way with the size of the problem. Our experience here is that the algorithm gives solutions very quickly
unless we ask for filters with a very large number of gains.
In the following sections, numerical studies are performed to investigate the use of the design
method presented here, and to investigate the ability of adjusting the filter parameters to address the issues
of interest in the repetitive control problem

ADJUSTING THE FILTER PARAMETERS

Notation: Various symbols are used in the sequel: w d is the desired frequency interval for the
transition band in the optimization criterion (12) expressed as a percent of Nyquist frequency, w e is the
effective transition band as a percent of Nyquist going from the last frequency when the filter magnitude
output leaves +1 until the magnitude first reaches 0, f d is the frequency in Hz of the desired cutoff which
!
is the center of the frequency range of w d , f e is the effective cutoff frequency in Hz defined as the
!
frequency when the magnitude response crosses 0.5 on its way from the passband to the stopband (0.5 is
used because we are using a zero-phase filter, and 0.707 is used when we consider any nonzero phase
!
transfer function cutoff), f N is Nyquist frequency in Hz, f p is the highest frequency in the passband when
! !
the magnitude output is unity, and f s is the first frequency in the stopband when the output reaches zero.
The frequency range from zero to Nyquist in (12) is divided into 1000 evenly spaced frequencies,
N = 1000 . For this
! section the Nyquist frequency is 100 Hz.
!
Adjusting Filter Weights: Figure 2 uses no transition band and weight " = 1, and one sees
!
substantial ripple in the passband. By going to the much larger weight " = 5000 produces Fig. 3, which
greatly improves the passband, but amplifies the ripple in the stopband. When the problem in Fig. 2 is
! addressed using the quadratic programming algorithm with inequality constraints introduced, one gets Fig.
! above unity, as desired. In a
4. One sees that it is effective at keeping the ripple in the stopband from going
!
later section this property is shown to allow substantially higher cutoff of the learning process in RC. The
quadratic programming algorithm is used for all the following results.
Introducing a Transition Band: There is always a transition band whether one asks for one in
(12) or not. Figures 5 through 8 investigate how the requested band and the resulting effective band are
related and how introducing such a band influences the ripple in the stopband and in the passband.
Allowing the width of this band to equal 5% of Nyquist frequency gives the results in Fig. 5 which
produces substantial improvement in ripple in both the passband and the stopband. Increasing the transition
band to 10% and then 15% can produce additional substantial improvement, as shown in Figs. 6 and 7.
Figure 8 studies the relationship between the specified transition band width wd and the effective width
we . Table 1 summarizes the properties of these different designs. The maximum dip depth in the passband
decreases from 0.0691 to 0.00005378 when one relaxes the transition band from 0% to 20%. Table 1 also
shows that the actual transition band width is always slightly larger then the specified one. This is
especially the case with a smaller specified transition band: one gets a 6.51% transition band width when
asking for 0%, and get 8.62% when asks for 5%. On the other hand, the larger the transition band one
specifies, the closer the actual transition band width will be to the specified width. There is also a
relationship between the chosen weight " and the effective cutoff frequency. Increasing " makes the plot
stay closer to +1 in the passband, and at the same time delays the effective cutoff frequency as shown in
Fig. 9. If a specific cutoff is needed, one might need to keep this effect in mind when specifying the filter
parameters.
! !
CREATING FILTERS FOR A CUTOFF AT A LOW PERCENT OF NYQUIST FREQUENCY

Sometimes people designing digital systems will simply use the fastest possible sample rate
consistent with the computation time needed per time step. This can create a Nyquist frequency that is very
high compared to the bandwidth of the control system, and hence high compared to the frequencies when
the model and the real world may start to differ sufficiently that a cutoff is needed. Note that
mathematically, everything in the optimization criterion (12) can be normalized by Nyquist frequency, so
the optimized filter gains are the same for all filters that have the same percent of Nyquist for the cutoff, no
matter what the Nyquist frequency is. Figures 10 through 14 study the situation when the desired cutoff is a
small percent of the Nyquist frequency. Nyquist frequency has been increased to 1000Hz for this section
only. Figure 10 asks for a cutoff at 5% Nyquist with a 3% transition band. One observes that the passband
has very poor performance. The magnitude response starts down immediately from zero frequency. As
mentioned above, accuracy of the magnitude response in the passband is very important, and this represents
a very poor design. Figure 11 examines the possible benefit of increasing the width of the transition band,
and it again has very poor performance in the passband, although the ripple in the stopband is much
improved. Figure 12 studies the result of increasing " to 1000, and the performance is improved. Figure 13
increases the number of gains in the fitler to 101 from 51, and Fig. 14 further increases to 401 gains, and
the result is getting close to ideal performance. It appears that the faster the sample frequency is compared
to the cutoff frequency needed, the larger the number of gains needed in the filter. And this number of gains
!
determines the computation load per time step.

EXAMINING THE DIFFERENCE BETWEEN THE FILTER CUTOFF AND THE EFFECTIVE
RC CUTOFF

Reference [5] studies ILC and examines the difference between the filter cutoff frequency that is
specified in order to satisfy stability condition (6), and the resulting effective cutoff of the learning. The
cutoff filter in that case is a 5th order Butterworth filter done in zero phase producing an effective cutoff of
a 10th order filter. When the filter cutoff needed was well above the bandwidth of the system, the effective
cutoff could be very substantially below that of the filter, meaning that one loses the ability to eliminate
errors in the frequency interval between these cutoffs. We examine this phenomenon for RC using the FIR
filter designs developed here. Figures 15 and 16 study the robustness of the control design with this new
zero-phase low pass filter, when applied to a system whose transfer function we pick to correspond to fifth
order low-pass Butterworth filter (since it is not zero phase, the cutoff is at the 0.707 level). We find the
cutoff frequency of the sensitivity transfer function by plotting it only at addressed frequencies, and then
find the frequency where the magnitude is 0.707. Figures 15 and 16 are nearly linear implying that the
effective cutoff of the learning in the repetitive control system is very nearly the same as the filter cutoff.
These results are much better than those reported in [5], perhaps because the FIR filter design here has a
substantially faster cutoff than a 10th order roll off when using the Butterworth zero-phase filter in ILC.

STUDYING THE USE OF IMPROVED FILTER DESIGNS

Designing an RC System from Noisy Frequency Response Data: Reference [3] studies the design
of repetitive control compensators by the method of equation (8), using experimental frequency response
from noisy input-output data. As the frequency goes up, the signals approach the noise level making it hard
to have a good model all the way to Nyquist frequency. Figures 17 through 24 study designing a cutoff
filter for an RC system with a seven gain compensator designed from noisy data in [3] and using " = 1.
Five thousand data points were used at 200Hz sample rate, using a white Gaussian known input function
with unit standard deviation. These inputs are fed through a zero order hold into a third order system,
having unity DC gain, a real root with a break frequency at 1.4Hz, and a complex conjugate pair with
!
damping ratio of 0.5, and undamped natural frequency 5.9Hz. The root mean square of the simulated output
was computed, and white Gaussian noise was added to each sampled output measurement to produce a
signal to noise ratio of 100. Optimization Method 1 refers to designs using the original optimization
criterion (10). Optimization Method 2 uses the quadratic programming approach developed here.
 If the cutoff filter were perfect, the compensator designed from noisy data would require a true
cutoff at 63Hz. Figure 18 shows how the ripple in the passband has make the plot of the left hand side of
equation (6) go above unity making the design unstable when we simply design for this cutoff with " = 1.
Figure 17 shows how the cutoff had to artificially be reduced to 47.5Hz before stability could be achieved.
Figure 19 shows the improvement obtained by emphasizing the passband in the choice of " . The stopband
ripple is enlarged, but one can raise the cutoff to where it is needed. It can be difficult in some of the plots
!
of this type to distinguish the three curves. The plot of H (z) is the low pass filter and stays around +1 until
its cutoff and then descends. The plot of 1" #F (z)G (z) is small at low frequencies ! and near +1 at high
frequencies, and H (z)(1" #F (z)G (z) is low at low frequencies, and aims toward +1 as the frequency goes
up, until H (z) cuts it down. Figure 20 shows ! how the frequency components of the error change with
periods as the repetitive control system
! is running. The error below about 50Hz is nearly zero, and there is
improvement with repetitions or periods up to 63Hz, and the error is stabilized and not learning above this
! Figure 21 gives the sensitivity transfer function for all frequencies of period p time steps, and
frequency.
!
one sees that the behavior is not so flat in the passband as the frequencies get close to the cutoff.
Figure 22 applies Optimization Method 2, showing how it is able to eliminate apparent ripple in
the stopband by using a wider transition band. The sensitivity transfer function ripples in the passband are
slightly improved at the expense of a peak in the stopband, as shown in Figure 23. Increasing the weight on
the passband aiming to decrease the ripple in this band is effective as shown in Fig. 24, but at the expense
of a larger peak in the stopband. To understand this peak, note that all three curves are near +1 around
63Hz. Note also that for each addressed frequency of period p time steps, z p is equal to +1. These together
imply that the magnitude of the sensitivity transfer function (4) is near to zero divided by zero, making the
resulting amplitude ill conditioned.
Figures 25 and 26 consider designing the compensator from the true model using Optimization
Method 1. Note that Fig. 26 exhibits substantial ripple in the stopband! magnitude of the sensitivity transfer
function. To understand this, observe that the term | 1 " ! F ( z )G ( z ) | in (4) becomes small, of magnitude 0.1
as shown in Fig. 25, and when multiplied by the small value of H (z) in the stopband, it becomes very
small. As a result the denominator of the sensitivity transfer function becomes near unity, and the
numerator looks like 1" H (z) , which oscillates around +1. Hence, the stopband ripples come through
more or less unmodified in the sensitivity transfer function.
! When Optimization Method 2 is applied
instead as shown in Fig. 27, there the ripples in the stopband are very small because the filter itself has only
small ripples there.
!
Designing RC with a Linear Phase Lead Compensator with a Cutoff: Reference [1] promotes
the use of a linear phase lead compensator as a simple design method that allows one to raise the cutoff
frequency substantially, simply by adjusting one parameter, the phase lead " in the simple compensator
F (z) = "z # . The " is set to unity, and ! = 9 which allows the highest possible cutoff frequency in (6).
Figure 28 shows that when Optimization Method 1 is used with " = 1, an artificially low cutoff of 7.5Hz is
required for the RC system to be stable, even though | 1 ! " F ( z )G ( z ) !|< 1 is satisfied for all frequencies up to

! 22Hz. By ! emphasizing the passband, stability can be achieved with a cutoff of 22Hz using Optimization
Method 1, at the cost of enlarged stopband ripples as shown
! in Fig. 29. However, the new zero-phase filter
design using Optimization Method 2 produces stable control up to a cutoff of 22Hz without causing any
visible ripples in the stopband, as shown in Figure 30.

USING A DIGITAL FIR ZERO-PHASE FILTER FOR ANTI-ALIASING

Now consider the use of the new filter design method as a way to introduce an anti-aliasing filter
into a repetitive control system (see, e.g. [14] for a discussion of anti-aliasing filters). In some applications
it is the error signal that is sampled, as in Fig. 31, and in other systems it is the feedback measurement that
is sampled, as in Fig. 32. Normally, the anti-aliasing filter must be an analog filter that appears just before
the analog to digital converter, but using such a filter will compromise the performance of a repetitive
control system. We suggest a method to address this issue, preserving the ability of RC to get to zero error
in the passband and simultaneously have the benefit of anti-aliasing. The approach is illustrated by a
specific example. We suggest that one use a high sample rate, 2400Hz in the examples here, and apply a
digital anti-aliasing filter (AAF) R(z) to this sampled signal using a cutoff frequency equal to the desired
cutoff, chosen here as 240Hz. Note that this filter can be operating on old data, allowing one to use a zero-
phase anti-aliasing filter, something not possible in routine feedback digital control. Then one
downsamples, using only every other time step of the filtered signal. This corresponds to a sample rate of
1200Hz and a final Nyquist! frequency of 600Hz. The repetitive controller operates on this signal, meaning
that we apply another zero-phase low-pass filter (LPF), H (z) , again with the 240Hz desired cutoff. Figures
33 through 45 study this process. To use this method the anti-aliasing filter needs to filter with a cutoff at a
substantially smaller percent of Nyquist than the filter H (z) . If this presents a problem, one could raise the
percent Nyquist somewhat on the anti-aliasing filter. A passband weight of " = 1000 is used in designing
!
this filter because, any error in the passband amplitude directly translates into a nonzero deterministic error
in the performance of the repetitive controller once it has converged.
Figures 33 and 34 show the AAF and ! the LPF (both with 51 gains). Each was designed aiming for
identical 60Hz transition bands, meaning that the specified transition ! bands were 5% for the AAF and 10%
for the LPF. In Fig. 33, f p = 193 and f s = 298 , while in Fig. 34 f p = 203.5 and f s = 276.5, so the
effective transition bands do not match. Of course, one could iteratively adjust parameters to get them to
match. The AAF has larger ripple in the stopband because of the smaller percent Nyquist for the desired
cutoff. The most important part of this ripple is between 1200Hz and 960Hz, which corresponds to
frequencies that are! folded into ! !
the passband seen by the RC!system. The remaining part of the stopband
ripple will be further attenuated by the stopband of the LPF. Figure 35 is an arbitrarily chosen signal for be
filtered, given in terms of its frequency content.
We will follow the different parts of this signal as it goes through the AAF, down sampling, and
then the LPF. There are four separate paths for different parts of the signal. The first part of the signal is the
frequency range between 0 and 1200Hz, which is Nyquist frequency for the initial 2400Hz sample rate.
Call this signal A. First we follow this part of the signal through the filtering process, and then we return to
the folded part of the signal shown folded about the 1200Hz line in Fig. 36, and we call this signal B.
(i) Signal A goes through the AAF to produce Fig. 37, where the vertical line indicates the
intended cutoff. This signal is down sampled, producing two parts. The frequency range up to 600Hz is
sent through the LPF to produce Fig. 41. This is the only part of the signal that has no folding in any stage
of the process. Figure 39 shows the part of Fig. 37 above 600Hz, which is folded as the result of
downsampling. Then the LPF is applied to this part of the signal to result in Fig. 42.
(ii) Signal B goes through the AAF to produce Fig. 38 where the vertical line is the desired cutoff.
The part of this signal below 600Hz is then sent through the LPF after down sampling to produce Fig. 43.
And the part of this signal above 600Hz is folded due to the downsampling as shown in Fig. 40, and then it
goes through the LPF to produce Fig. 44.
Figure 41 is the desired result that has no aliasing. The actual result is the sum of Figs. 41, 42, 43,
and 44. This result is to be compared to the result of not using an anti-aliasing filter, and simply applying
the LPF to the signal operating at the final 1200Hz sample rate with 600Hz Nyquist. These results are
plotted in Fig. 45. The decay above the cutoff is slower with the AAF than without. This might be
eliminated by matching the effective transition bands of the AAF and the LPF. Below the desired 240Hz
cutoff, the result with the AAF is much closer to the original signal, giving the desired result.
Figures 46 through 48 examine an example motivated by the application of repetitive control to
the Thomas Jefferson 8 GEV electron beam accelerator. Since the direct current in the electromagnets
controlling the beam comes from rectified AC current, their magnetic strength has fluctuations that disturb
the beam focus at 60Hz and many harmonics. To create an example of the use of the above anti-aliasing
design, consider disturbances at 60, 180, 300, 360, 420, 660, 720, 960, and 1080Hz, all of which were
visible in the beam position histories. Pick amplitudes of 237, 55, 37, 18, 13, 13, 6, 5, and 8 for sinusoids at
these frequencies, respectively, as shown in Fig. 46. We arbitrarily consider applying repetitive control at a
sample rate of 1200Hz to a continuous time system with transfer function a /(s + a) where a = 300" , fed
by a zero order hold. The design process of equation (8) produces the system inverse as a compensator. The
design cutoff at 240Hz aims to eliminate the two largest peaks at 60 and 180Hz, and leave the remaining
frequencies unaddressed. Figure 47 shows the result when R(z) is set to one so that no anti-aliasing filter is
!
! and Fig. 48 shows
used but we are still able to see the output at the higher sample rate, the result when the
anti-aliasing filter is introduced. Both figures show the results examined with a 2400Hz sample rate, so
they see the error remaining at intermediate time steps between those addressed by the repetitive controller.
!
In Fig. 48, all frequencies above the 240Hz cutoff remain essentially unaffected by the repetitive control.
However, there are two new frequencies introduced, at 1020 and 1140Hz. These must be components seen
at the 2400Hz sample rate, that result from canceling 180 and 60Hz signals with a zero order hold at
1200Hz sample rate, with nonzero error at the intermediate steps. In Fig. 47, when no anti-aliasing filter is
used, we see four peaks below the cutoff frequency remaining after convergence of the repetitive control
system. This demonstrates the effectiveness of the anti-aliasing process. Two of these signals are images of
the 960 and 1080Hz disturbances folded to 120 and 240Hz, and the repetitive controller produced signals of
these folded frequencies to cancel the folded signals at the sample times, introducing error at the lower
frequencies, and failing to cancel the actual higher frequency disturbances.

CONCLUSIONS

(1) A practical method of reducing passband ripple peaks below one in the frequency response of a zero-
phase FIR filter has been presented. This is of practical importance because it allows one to raise the cutoff
frequency to the maximum value allowed by the model mismatch with the true world model.
(2) The resulting quadratic programming problem is easy to solve and gives answers quickly.
(3) It was noted that in contrast to the usual purposes of low pass filters, it can be quite acceptable to
introduce a wide transition band. It is shown how such a band can assist the filter to have extra small ripple
in the passband and in the stopband, and the phase does not flip 180 degrees until the magnitude output
reaches zero for the first time in the stopband.
(4) Before introducing the transition band, the relative weight given to the passband and the stopband
defined a needed tradeoff between the importance of ripple in each band. Introducing the wider transition
band makes this tradeoff less important.
(5) The final error level reached by a repetitive control system in the passband can only be as good as the
filter is in this frequency range. Hence, a filter with very little magnitude error in this range is very
important for high precision control. It is seen that one can produce very good accuracy by choice of
weights and transition band width.
(6) The design of the low pass filter becomes more difficult when the cutoff frequency is a small percent of
Nyquist frequency. Methods are studied to address this problem. This is of importance when one chooses to
have a high sample rate for improved fidelity of representation of signals, even though the system dynamics
do not require it. It is also important in the anti-aliasing application.
(7) Anti-aliasing filters are often beneficial or needed in digital control systems to limit the effects of
folding energy above Nyquist into energy below Nyquist that is acted on inappropriately by the digital
controller. Normally this filter must be causal operating in real time on current data. In RC applications the
presence of such a filter can seriously deteriorate performance. The RC can only aim for zero error in the
filtered signal, so any distortions of amplitude and phase come through as error. It is shown here how one
can use a zero-phase non-causal digital FIR low-pass filter as an anti-aliasing filter. The fact that we are
operating on old data means the filter no longer needs to be causal, allowing it to be zero phase. This forms
what can be a very important application of the filter design methodology presented here.

REFERENCES

1. B. Panomruttanarug and R. W. Longman, “Repetitive Controller Design Using Optimization in the

Frequency Domain,” Proceedings of the 2004 AIAA/AAS Astrodynamics Specialist Conference,
Providence, RI, August 2004.
2. R. W. Longman, J. W. Yeol, and Y. S. Ryu, “Improved Methods to Cancel Multiple Unrelated
Periodic Disturbances by Repetitive Control,” Advances in the Astronautical Sciences, Vol. 123, 2006,
pp. 199-218.
3. B. Panomruttanarug and R. W. Longman, “Designing Optimized FIR Repetitive Controllers from
Noisy Frequency Response Data,” Advances in the Astronautical Sciences, to appear.
4. K. K. Chew, M. Tomizuka, “Digital Control of Repetitive Errors in Disk Drive Systems,” IEEE
Control Systems Magazine, Vol. 10, No. 1, January 1990, pp. 16-19.
5. H. S. Lee and M. Tomizuka, “Robust motion contoller design for high-accuracy positioning systems,”
IEEE Transactions on Industrial Electronics, Vol. 43, Feb, 1996, pp. 48-55.
6. H. Elci, M. Phan, R. W. Longman, J.-N. Juang, and R. Ugoletti, “Experiments in the use of Learning
Control for Maximum Precision Robot Trajectory Tracking,” Proceedings of the 1994 Conference on
Information Sciences and Systems, Princeton, NJ, March 1994, pp. 951-958.
7. M. Norrlof, S. Gunnarsson, “Experimental Comparison of Some Classical Iterative Learning Control
Algorithms,” IEEE Transactions on Robotics and Automation, 2002, Vol. 18, No. 4, pp. 636-641.
8. B. Panomruttanarug and R. W. Longman, “Frequency Based Optimal Design of FIR Zero-Phase
Filters and Compensators for Robust Repetitive Control,” Advances in the Astronautical Sciences, Vol.
123, 2006, pp. 219-238.
9. R. W. Longman and W. Kang, “Issues in Robustification of Iterative Learning Control using a Zero-
Phase Filter Cutoff,” Proceedings of the 2007 AAS/AIAA Spaceflight Mechanics Conference, Sedona,
AZ, to appear.
10. S. G. Edwards, B. N. Agrawal, M. Q. Phan, and R. W. Longman, “Disturbance Identification and
Rejection Experiments on an Ultra Quiet Platform,” Advances in the Astronautical Sciences, Vol. 103,
1999, pp. 633-651.
11. R. W. Longman, “Iterative Learning Control and Repetitive Control for Engineering Practice,”
International Journal of Control, Special Issue on Iterative Learning Control, Vol. 73, No. 10, July
2000, pp. 930-954.
12. S. Songschon and R. W. Longman, “Comparison of the Stability Boundary and the Frequency
Response Stability Condition in Learning and Repetitive Control,” International Journal of Applied
Mathematics and Computer Science, Vol. 13, No. 2, 2003, pp. 169-177.
13. J. Nocedal and S. Wright, Numerical Optimization, Springer-Verlag, New York, 1999.
14. S. Elliot, Signal Processing for Active Control, Academic Press, New York, 2001.

Table 1 Width of Transition Band

n=51, f d =30Hz, α = 1, with unit DC gain and inequality constraints
Desired transition band 0% 5% 8% 10% 15% 20%
wd
Effective cutoff f e 29.92 30.00 30.05 30.08 30.16 30.26
Freq starts down f p 26.02 25.14 24.16 23.42 21.50 19.56
Freq hit zero f s 32.53 33.63 34.8 35.66 37.95 40.34
Effective transition 6.51% 8.49% 10.64% 12.24% 16.45% 20.78%
band we
Number of dips in 3 3 3 4 4 3
passband
Passband ripple (max 0.0691 / 2.516e-2 / 8.748e-3 / 4.071e-3 / 5.1715e-4 / 5.378e-5 /
dip depth / location) 22.63 22.05 21.36 20.80 19.19 17.30
Stopband bump 0.0850 / 3.013e-2 / 1.051e-2 / 4.936e-3 / 6.989e-4 / 9.585e-5 /
(max height / location) 34.10 34.90 35.86 36.60 38.67 40.91

V(z)
YD(z) E(z) φ H ( z)F ( z) U(z) X(z) Y(z)
+ G(z) + +
z p − H ( z)
-

Figure 1 Repetitive control system block diagram.

 1 1

0.8 0.8

Magnitude

Magnitude
0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 20 40 60 80 100 0 20 40 60 80 100
Frequency (Hz) Frequency (Hz)
Figure 2 A 51-gain filter, α =1, 30% Figure 3 Same as Fig. 2 but with
cutoff, no transition band. α =5000.

1 1

0.8 0.8
Magnitude

Magnitude
0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 20 40 60 80 100 0 20 40 60 80 100
Frequency (Hz) Frequency (Hz)
Figure 4 Introducing passband inequality Figure 5 Fig. 4 with a 5% transition band.
constraints in Fig. 2 with DC gain
constraint.

1 1

0.8 0.8
Magnitude

Magnitude

0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 20 40 60 80 100 0 20 40 60 80 100
Frequency (Hz) Frequency (Hz)
Figure 6 Fig. 4 with a 10% transition band. Figure 7 Fig. 4 with a 15% transition band.
 7 0.025

d
Error in Transition Band | w - w |/ w
d
6

N
0.02

Error in Cutoff | f - f |/ f
e
5

d
e
4 0.015

3 0.01
2
0.005
1

0 0 0
0 5 10 15 20 10 10
2
10
4
Desired Transition Band w Passband Weight α
d
Figure 8 Actual transition band versus Figure 9 Difference between actual and
specified transition band. specified cutoff frequency versus passband
weight, with a transition band of 10%.

1 1

0.8 0.8
Magnitude

Magnitude
0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 200 400 600 800 1000 0 20 40 60 80 100
Frequency (Hz) Frequency (Hz)
Figure 10 5% cutoff design, 51 gains, Figure 11 Modify Fig. 10 for 10%
α =1, 3% transition band, DC gain and transition band.
inequality constraints included.

1 1

0.8 0.8
Magnitude

Magnitude

0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 200 400 600 800 1000 0 200 400 600 800 1000
Frequency (Hz) Frequency (Hz)
Figure 12 Modify Fig. 10 by increasing α Figure 13 Modifying Fig. 10 by increasing
to 1000. the number of gains to 101.
 100
1

80

Effective cutoff of S(z)

0.8

Magnitude
60
0.6

0.4 40

0.2 20

0 0
0 200 400 600 800 1000 0 20 40 60 80 100
Frequency (Hz) Effective cutoff of H(z)
Figure 14 Modifying Fig. 10 by increasing Figure 15 Sensitivity transfer function
number of gains to 401. S (z ) cutoff versus filter H (z ) cutoff when
G (z ) cutoff fixed at 34.15Hz.

100

80
Effective cutoff of S(z)

60

40

20

0
0 20 40 60 80 100
Effective cutoff of G(z)
Figure 16 Sensitivity transfer function Figure 17 Artificially low required cutoff
S (z ) cutoff versus G (z ) cutoff when due to passband ripples, using
H ( z ) cutoff fixed at 30.764Hz. Optimization Method 1, α =1, cut off at
47.5Hz.

Figure 18 Instability due to ripples when Figure 19 Optimization Method 1,

cutoff is at 63Hz. α =5000, 63Hz cutoff.
 1

0.8

Magnitude
0.6

0.4

0.2

0
0 20 40 60 80 100
Frequency (Hz)
Figure 21 Sensitivity transfer function for
Figure 20 FFT of error for Fig. 19 at Fig. 19 at addressed frequencies.
different repetition numbers.

0.8

Magnitude
0.6

0.4

0.2

0
0 20 40 60 80 100
Frequency (Hz)
Figure 22 Fig. 19 for Optimization Method Figure 23 Sensitivity transfer function for
2, design cutoff at 71.8Hz, α =1, 10% Fig. 22
transition band, 51 gains.

2.5

2
Magnitude

1.5

0.5

0
0 20 40 60 80 100
Frequency (Hz)
Figure 24 Increasing α to 1000 in Fig. 23. Figure 25 Optimization Method 1 with
compensator designed from true model,
63Hz cutoff, α =5000.
 1.4
1
1.2
0.8
1

Magnitude
Magnitude
0.8 0.6

0.6
0.4
0.4
0.2
0.2

0 0
0 20 40 60 80 100 0 20 40 60 80 100
Frequency (Hz) Frequency (Hz)
Figure 26 Sensitivity transfer function for Figure 27 Optimization Method 2,
Fig. 25. compensator designed from true model,
10% transition band, α =1, 51 gains,
71.8Hz cutoff.

Figure 28 Optimization Method 1, α =1, Figure 29 Optimization Method 1,

7.5Hz cutoff. α =5000, 22Hz cutoff.

Figure 30 Optimization Method 2, α =1,

22Hz cutoff, 10% transition band.
 V(s)
YD(s) E0(s) E0(z) E1(z) E(z) U(z)
+ Down φ H ( z)F (z) D/A + + Y(s)
A/D R(z) Sampling ZOH G(s)
z p − H ( z)
-
2,400Hz Anti-aliasing 1,200Hz

Y(s)
Figure 31 Anti-aliasing filter applied to error signal.

V(s)
YD(z) + E(z) φ H ( z)F ( z) U(z) D/A + + Y(s)
G(s)
z − H ( z)
p ZOH
- 1,200Hz

Down A/D
Sampling R(z) 2,400Hz
Y(s)
Anti-aliasing
Figure 32 Anti-aliasing filter applied to sensor measurement.

1 1

0.8 0.8
Magnitude

0.6 Magnitude 0.6

0.4 0.4

0.2 0.2

0 0
0 200 400 600 800 1000 1200 0 200 400 600
Frequency (Hz) Frequency (Hz)
Figure 33 Magnitude frequency response Figure 34 Magnitude frequency response
of anti-aliasing filer R(z ) , α =1000. of cutoff filter H (z ) , α =1000.

1 1

0.8 0.8
Magnitude

Magnitude

0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 400 800 1200 1600 2000 2400 0 400 800 1200 1600 2000 2400
Frequency (Hz) Frequency (Hz)
Figure 35 Magnitude frequency content of Figure 36 Fig. 35 folded at 1200Hz
signal to be sampled and filtered. Nyquist frequency.
 1 1

0.8 0.8

Magnitude

Magnitude
0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 200 400 600 800 1000 1200 0 200 400 600 800 1000 1200
Frequency (Hz) Frequency (Hz)
Figure 37 Signal of Fig. 35 after going Figure 38 Fig. 36 after going through anti-
through anti-aliasing filter. aliasing filter.

-3 -4
x 10 x 10
2

4
Magnitude

Magnitude
1
2

0 0
0 100 200 300 400 500 600 0 100 200 300 400 500 600
Frequency (Hz) Frequency (Hz)
Figure 39 Fig. 37 after folding from down Figure 40 Fig. 38 after folding from down
sampling to 600Hz. sampling to 600Hz.

1 1

0.8 0.8
Magnitude

Magnitude

0.6 0.6

0.4 0.4

0.2 0.2

0 0
0 100 200 300 400 500 600 0 100 200 300 400 500 600
Frequency (Hz) Frequency (Hz)
Figure 41 Fig. 37 after passing through Figure 42 Fig. 38 after passing through
cutoff filter. cutoff filter.
 -3 -4
x 10 x 10
1

Magnitude

Magnitude
0.5
2

0 0
0 100 200 300 400 500 600 0 100 200 300 400 500 600
Frequency (Hz) Frequency (Hz)
Figure 43 Fig. 39 after passing through Figure 44 Fig. 40 after passing through
cutoff filter. cutoff filter.

1 250
original signal up to 240Hz
R and H filtered

Magnitude of Disturbance
0.8 200
H filtered
Magnitude

0.6 150

0.4 100

0.2 50

0 0
0 100 200 300 400 500 600 0 200 400 600 800 1000 1200
Frequency (Hz) Frequency (Hz)
Figure 45 Comparison of original signal Figure 46 Example frequency spectrum of
up to cutoff (Fig. 35), sum of Figs. 41 a disturbance signal.
through 44 for signal passing through
R(z ) and H (z ) , and corresponding result
passing through H (z ) only.

40 40

30 30
Magnitude of Error

Magnitude of Error

20 20

10 10

0 0
0 200 400 600 800 1000 1200 0 200 400 600 800 1000 1200
Frequency (Hz) Frequency (Hz)
Figure 47 Spectrum of steady state error Figure 48 Same as Fig. 47 with anti-
from simulation with system Fig. 31, with aliasing filter introduced.
no anti-aliasing filter.