Chapter 5

Adaptive harmonic control:

tuning in the frequency domain

S. M. Veres and T. Meurers

School of Engineering Sciences, University of Southampton
Southampton, UK, Email:[email protected]

This chapter discusses adaptive control based on control at individual harmon-

ics for cancellation of periodic disturbances. It is shown that the original idea,
which has been around for a long time, can be developed much further to pro-
duce robust and highly adaptable controllers. First, a review is given of the
most accessible literature, after which frequency-selective RLS and LMS meth-
ods are presented. Simulations illustrate the effectiveness of the method.

5.1 Introduction

As in some applications the plant can be considered linear, a fundamental idea

is to separate the control problem at each relevant harmonic of the disturbance
if that is dominated by a discrete set of frequencies. Then, by linearity there
is no interaction between the control solutions found at different harmonics.
This idea can be described as follows for a single tonal active vibration or
sound control problem. At frequency u denote the detection signal by X(OJ),
the control signal by u(u), the additive disturbance by d(uj) and the error
signal by e(uj). Figure 5.1 describes the problem of producing u(u) so that
e(cj) is as near to zero as possible.
Let CJI, CJ2, ..., cjn be a finite set of frequencies at which x or d have
relevant harmonic components. The objective is to design a W such that

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 98 Active Sound and Vibration Control

Detection d(G>)
Signal
Controller Plant
y(co)
am
X((O) U{(O)
W(JG))

Figure 5.1 Block diagram for the harmonic control problem

e = d + GWx = 0 at these frequencies. At each frequency the harmonic

components of the signals, and also the response of the controller and the
plant dynamics, can be characterised by single complex numbers, which will
be denoted by x(uk), u(u)k), d(uk), e(uk) and W(jwk), G(ju)k), respectively.
Assuming that there is no feedback from the actuator to the detection, the
component of the error signal can be expressed as:
e(cjfc) = d(u)k) + W{juk)G(juk)x{uk), k = 1,2,..., n
Figure 5.1 shows the block diagram of the harmonic control system. For
cancellation the controller has to satisfy:

W(jojk) = — (5.1)

Control of periodic sound at individual harmonics has been around for some
time and Reference [219] gives a summary of the subject. William Conover
built an electronic system to achieve this cancellation for the harmonics of
electrical transformers [58] in 1956. To satisfy these conditions in Conover's
work the transfer function W is physically realised by bandpass filters, phase
shifters and amplifiers. Figure 5.2 shows a block diagram of Conover's sys-
tem. The analogue circuits offered little flexibility in the selected frequencies,
the amplifiers (amplitude control) and phase shifters were tuned manually.
Later digital control solutions were sought when computer technology made
that a possibility. Various adaptive digital feedforward control methods have
been proposed, among them the most widely used today is the filtered-x LMS
method.

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 Adaptive harmonic control: tuning in the frequency domain 99

Band-pass Phase Amplitude

filter at CO i shifter control

Band-pass Phase Amplitude

—•»> shifter control
filter atC02
i f ]

Reference Addition Actuator

pickup circuit amplifier

Band-pass Phase Amplitude

filter at con shifter control

Figure 5.2 Block diagram of Conover's harmonic controller

This chapter presents a study of adaptive frequency selective solutions

to periodic disturbance cancellation. The control schemes suggested are pure
feedback control schemes, no detection signal is needed, as these are frequently
unavailable in practical problems. Achievable performance is excellent com-
pared to pure linear feedback solutions and the advantages can be summarised
as follows:

(i) For periodic disturbances with a nearly discrete spectrum the perfor-
mance is far better than that which is achievable by linear feedback
systems.

(ii) A high degree of adaptability for dynamical changes of the plant is

achieved.

(iii) Stability is ensured despite a the high level of adaptation.

Compared to standard LMS-type feedforward control an obvious advantage is

that there is no need for a detection signal which is strongly correlated with
the disturbance. The only disadvantage of the scheme presented is that the
disturbance has to be dominated by a discrete spectrum (possibly slowly time
varying). This is a requirement that is satisfied by a large number of practical
vibration and sound control problems, hence the usefulness of the approach.

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 100 Active Sound and Vibration Control

5.2 Problem formulation

T h e plant t o b e controlled is a linear time-varying two-input/two-output sys-

tem as presented in Figure 5.3: It is described by stable 2 x 2 transfer

Figure 5.3 Block diagram of the basic control scheme

functions:
(5.2)
so that at each time instant t:

y(t) = £ w *u(t - i ) , * > 0; y = [yl y*]T, u = [ul u2]T (5.3)

and in matrix form we can also write:

(5.4)

where
Gk{q (5.5)
'-
The following situation is practically important:
(i) u1 is an unmeasured excitation of the vibration,
(ii) u2 is a control input.

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 Adaptive harmonic control: tuning in the frequency domain 101

(iii) y1 is a measurable output which is to be regulated to zero.

(iv) y2 is a measurable output.
The regulation problem of yl to zero, described under conditions (i)—(iv), is
difficult or in general impossible to solve using mainstream adaptive control
theory because of the unmeasured ti1. In the following an RLS and an LMS
approach will be looked at to remedy this situation. To solve these problems
some simplifying assumptions will be made and the ideas of harmonic control
will be used.

FSF

FSF ^
Control

FSF

KiJK

Figure 5.4 Frequency selective filtering (FSF) at a set of frequencies

(i) Regulation of y1 to zero is to be achieved at a given set of frequencies

uuu2,".,wn € [0,2TT]

(ii) The plant is slowly time varying so that ||G t - Gt+i||i < c, t > 0 with
i > 0 known a priori.
(iii) The outputs are measurable with a given accuracy p > 0.
A possible scheme to achieve this is as follows. Each output is led through
a frequency selective filter (FSF) for each frequency CJI,CO2, ... ,cj n as shown

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 102 Active Sound and Vibration Control

in the block diagram in Figure 5.4. The signals coining from the filters are
processed by the controller and a suitable input is synthesised. The main
result of this section is dealing with the problem of how to compute a suitable
control input in the frequency domain.
First, some of the surrounding details will be clarified. The frequency
selective filters are of the form:

where r < 1, r ~ 1. The advantage of frequency selective filters is that the

settling time of the filters can typically be smaller than the length of an FFT
batch which could be used instead. Typically, the settling time of a satisfactory
FSF can be around 40 ~ 60 sampling periods and FFTs should be based on
256 samples to have similar accuracy.
Notations:
), i = 1,2,... ,n
will be used for the steady-state complex amplitudes of the sine waves:
Hi(q)y\ Hi(q)y2, Hi(q)u\ Hi{q)u\ i = 1,2,..., n
respectively. Then (A) means that the control objective is to achieve:

The control signal will be computed as a linear combination of phase-shifted

sine-waves (an alternative method could be based on a robust HQO controller
but there is more danger of instability in that case because of possibly false
uncertainty assessment of the frequency selective model used). Let the infinite
time horizon be split into periods of TV sampling instants, so that the kth.
period is denoted by:

In many practical systems it might be realistic to assume that there is an

average transfer function Gfc(^~1) valid over T{k) such that:
|| G *(e-^) - Gt(e-**)ll < v, t€ T(k)
with a priori known v > 0, where ||G|| denotes max|G; m |. Similarly, for the
filtered signals:
Hi(q)y\ Hi(q)y2, Hi(q)ul, Hi{q)u\ i = 1,2,... ,n,

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 Adaptive harmonic control: tuning in the frequency domain 103

the respective average complex amplitudes over time period T(k) will be de-
noted by:

yi(eM), VIW"% «i(e*"), «2(e*"), « = 1,2,..., n

uKe?"*), i = 1,2,..., n, actually represent the components of the control input
so that u\ is defined as the sum of sine waves with complex amplitudes ul(e?Wi),
i = 1,2,..., n during period T(k). This operation is clearly not linear filtering.
For further use let the components of Gk(e^Ui)[ul(ejUi) 1]T be denoted by:

5.3 A frequency selective RLS solution

Let Xfc, dfc and e* denote complex numbers representing the harmonic content
of the signals y2, Gllu\ and yi, during time period T(k), as described above at
given frequency UJ and indicated in Figure 5.6 . Assume that G12 changes slowly
and good estimates g^ can be calculated for its complex gain at frequency w.
(Let's postpone discussion of the estimability of G12 for a little while.) First a
simple filtered-x algorithm can be introduced for each frequency CJ of interest.
Introduce r* = x^g*. For any complex number c the notation c = c1 4- c*j
will be used. Then the equation:

dk + nk = ek (5.6)

can be rewritten in the form:

e\ = rlw\ - r2wl + d\ + nj, e\ = r 2 ^ + r1^^ + dk + n\ (5.7)

={ H kN< t < kN+N+l }

N sampling periods N sampling periods N sampling periods N sampling periods t

-I 1 j : 1 1 j
k k+l k+2 fc+3
Figure 5.5 Periodic sectioning of the time scale for compensator design and
frequency response estimation

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 104 Active Sound and Vibration Control

w* g*

RLS
\

Figure 5.6 Block diagram of the RLS solution at a single frequency

for the output errors e* which are measured with error/noise n\ and n\. To
find a suitable controller gain w^ represented by the coefficients wk and w%,
the following exponentially weighted criterion can be optimised:

by the usual RLS algorithm with a forgetting factor A < 1. This can be carried
out in two stages for each k > 1:

Kk = = (I~ Kk(j>l)Pk
(5.8)
where the notations:

9k = Pk =
-rl
are used and Kk, wk, Pk are only intermediate variables in an updating step.
Hence Wk, Pk are computed recursively with initial conditions dependent on
a priori knowledge of the plant. The following simulation illustrates how this
algorithm works.

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 Adaptive harmonic control: tuning in the frequency domain 105

Let:
r, _ r / > * • % _ [ 1-2 - 0.2j 0.6-0.05
u-u{e> ) - [ 0 8 + 0Aj 0.9-O.lj
where gk is time-varying gain, gk = c(cos0.02A:+jsin0.02fc) and c = 0.15, the
dynamics are time varying.
Non-controlled Output (200 periods)

Real parts

Controlled Output (200 periods)

0.05 -

-0.05 -

-0.1
-0.02

Figure 5.7 Complex amplitudes of the controlled and uncontrolled outputs for
200 periods: (a) Non-controlled output, (b) Controlled output

The forgetting factor is set to A = 0.7 and the detection signal is a fixed
constant rk = 1, therefore no measured detection is used. As the forgetting
factor is low, reasonable performance is achieved as shown in Figures 5.7 and
5.8. For a plant model the fixed and incorrect:

r - 0.9 + 0.4j 0.94 - O . l j

is used.

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 106 Active Sound and Vibration Control

The simulation illustrates that the method can work as the complex ampli-
tudes of the controlled output are small in Figures 5.7 and 5.8. Conditions of
stability with regard to G and xk are difficult to establish. The frequency selec-
tive LMS solution presented in the next Section easily lends itself to stability
conditions.

Non-controlled Output Non-controlled Output

50 100 150 200 50 100 150 200

Controlled Output Controlled Output

-20

-60
200 200

Figure 5.8 Amplitudes (dB) and phase (°) of uncontrolled outputs for 200
periods: (a) Non-controlled output, (b) Controlled output

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 Adaptive harmonic control: tuning in the frequency domain 107

Change of G(1,2) during 200 periods

0.15

0.05 -

i -0.05 -

-0.1 -

-0.2
0.45

Figure 5.9 Change of the complex gain G12 during 200 periods

5.4 A frequency selective LMS solution

An adaptive law will be derived for the input complex amplitude at a single
frequency and then its stability will be analysed in view of the uncertainty of
the transfer function of the plant. It will be proved that the adaptation law is
convergent under large relative uncertainty of the plant transfer function.
Let the input complex amplitude during time period T(k) be denoted by
u>k = u>r +uiJ- Similarly, the disturbance and output amplitudes will be denoted
by dk = dj + dij and yk = y* + yfj, respectively. Then, at a single frequency
the model of the plant equation will be:

Vk dk

where g is a complex number representing the plant transfer at the given

frequency.

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 108 Active Sound and Vibration Control

The objective will be to bring yk to zero and therefore an instantaneous

criterion function for control will be:

The output can be rewritten as:

Vr + ViJ = 9rUkr + 9^ + dkr + JigrVfl + ftfi* +
or in matrix form as:
yk = Gu* + d r
where

The control law will be designed to move the control signal u*;+i in the negative
gradient direction of the criterion c&. The gradient of the criterion function
can be calculated as:

which finally gives:

Vcfc = -2G T yjfe
Note that for the computation of the gradient only an estimate is available as
G is not known. With a fi > 0 step size this will define an adapted control
signal as:
Ufc+it/u* - f*GTyk (5.9)
Let the estimated and the actual transfer matrices of the plant be denoted by:

* -«• 1 and G 0 ^ f § ' $

ft Sr J [9i 9r
respectively. The actual output will be obtained from the equation:
y* = G0Ufc + d + ii* (5.10)
where n* is the measurement noise of the complex amplitude of the output.
Substituting eqn. (5.10) into eqn. (5.9) gives that:
ufc+1 = (I - nGTG0)uk + ^G T d + nGTnk (5.11)

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 Adaptive harmonic control: tuning in the frequency domain 109

Lemma 5.4.1 (a) The adaptive law eqn (5.9) will be stable if the matrix:

has eigenvalues with modulus less than 1.

(b) If the adaptive law eqn (5.9) is stable and n*. = 0 , then

lim I!* = Go ld

where u = G0"1d is the ideal control input to eliminate the vibration of the
output.

Proof: (a) is obvious from linear system theory. For (b) the steady state of the
given linear stable system can be calculated as:

[ql - (I - /iG T G 0 )]-VG T (d + n,),, = 1 = GjTM + Go'n*

which gives the desired result.
The most interesting case, practically, is to find out how small the step size
jj, will have to be defined to ensure stability under a given G and its relative
error. Let's denote the estimated complex gain associated with G and Go by:

8^9r+J9i ^dgo^9r+J9i
respectively. Then the following theorem holds.

Theorem 5.4.1 the control law eqn. (5.9) will be stable under any relative
error less than 5 > 0 of the estimate g, (i.e. for |g0 — g| < <J|g[i, if and only
if S < 1 and the condition

<512)

is satisfied.

Proof:firstof all note that:

CTC _[ 9r 9i] \ 9°r -9i] _ I" 9r9°r+9i9! ~9r9? + 9i9°r

^ ^ - " l - a 9r\ [SS 9°r J " 1-arf+ft.tf fttf + fctf
Introducing the notations a^grg® + fag? and bd= - grg? +fag?will give that:

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 110 Active Sound and Vibration Control

with eigenvalues 1 — fj,(a ± bj). To prove the theorem one has therefore to
prove that |1 — /x(a ± bj)\ < 1 for any relative error < 8 if and only if:

(513)

By definition it can also be seen that a + bj = ggj holds. Define the unit
disc C(l) = {z € C | |1 - z\ < 1}. In view of the relative error specification
the true transfer can be written as g0 = g + tflglre*" with some r € [0,1] and
u € [0, 2TT]. The uncertainty set of a + bj under maximum relative error 8 of
g is:
p u € [0, 2TT], r € [0,1]} (5.14)
Hence |1 - /z(a ± bj)\ < 1 will be satisfied under any relative error < 8 of g if
and only if D(8) C C(l). It can, however, be easily shown that D(8) C C(l)
if and only if eqn. (5.12) holds. To see this notice that:

^ ) = {/x|g|2 + ^ | g | 2 r e ^ | U;E[0,2TT], r e [0,1]} (5.15)

and the shaded disk D(8) with radius 8\g\2 will be contained in C(l) if and
only if:
/x|g|2 + fy|g|2 < 2 and 8 < 1 (5.16)
which is equivalent to eqn. (5.12).

5.5 Simulation example

In this Section a combination of the above frequency-domain LMS and time-
domain FSF will be used to illustrate the advantages and difficulties which
appear in this harmonic control approach.
The simulated plant is a twelfth-order linear dynamical system with pole
locations and Bode plots shown in Figure 5.10. Its transfer function is approx-
imately:

where the coefficients are rounded to four decimal points.

The output of this system is affected by a periodic disturbance shown in
Figure 5.11 together with its estimated spectrum. The simulated disturbance
had harmonic components at the frequencies 1.5Hz, 5Hz, 7Hz, 4Hz, lOHz,
11.5Hz and an additive white noise of amplitude 0.01.

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 Adaptive harmonic control: tuning in the frequency domain 111

Poles of the simulated plant in the complex unit circle

frequency (rad/s)

Figure 5.10 Poles and Bode plot of the plant simulated. Resonant and lightly
damped modes can be observed around 2 Hz, 3.5 Hz and 7 Hz

Disregarding the ripples, the estimate of the spectrum of the disturbance

has three high peaks at 1.5022Hz, 5.05Hz, 6.9951Hz and three low peaks at
4.019Hz, 10.077 Hz and 11.554Hz. The control system will aim to cancel only
three components of the disturbance, those around 1.5Hz, 5.5Hz and 7Hz.
As the precise frequencies may not be known in a practical situation, the
control scheme will use an online estimation of the relevant frequencies of the
disturbance.
The frequency selective adaptation is based on segmentation of the time axis
into equal periods T(k) of N = 600 samples. During each period T(k) the
output is filtered through three FSFs concentrated at the estimated three
most relevant frequencies of the disturbance. The feedback input to the plant
is a mix of three sine waves with constant phase and amplitude during each
T(k). The frequency selective RLS adaptive system described by the recursive

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 112 Active Sound and Vibration Control

The output disturbance (300 samples of the time signal)

Estimated spectrum of the disturbance by Hanning windOW

50

-100

-150
— •- ——,
•^ • i r -

-200

-250

_ann : : : : : : : i
102
frequency (Hz)

Figure 5.11 A section of the time signal of the disturbance and the estimate
of its spectrum by Hanning window, based on 500 samples

eqns (5.8) is applied at each frequency to determine suitable amplitudes and

phases of the sine waves, based on the adaptation of w* by eqns (5.8). As
the results of the method are very sensitive to errors in the frequencies of
the harmonic components of the disturbance, a frequency estimation is also
carried out at the end of each T(k). The frequency of the FSF outputs is
estimated by nonlinear optimisation of the squared-sum error function on the
basis of the last 100 samples during each period T(k), which ensures that the
FSF settled by that time during the previous 500 samples of T(k). To filter
out the high-frequency noise of the estimates, a low-pass filter is also applied
to each LS frequency estimate fy. i = 1,2,3 to obtain smothered estimates by:

= 0.98/'+0.02/*, 2 = 1,2,3 (5.17)

is then used during time period T(k + 1) to set the frequency of the

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 Adaptive harmonic control: tuning in the frequency domain 113

Output during periods 2-3

200 400 600 800 1000 1200

Output during periods 28-30

0.5

-0-5 H

200
'IT IT IT IT IT II IT IT IT IT IT I1'1 I'll I111! I
400 600 800 1000 1200 1400 1600 1800

Figure 5.12 Time signals of the output during periods T(2) — T(3) and
T(28) - T(30)

respective controller sinewave. The length of the time-domain simulation is 29

periods of T(k), k = 2,3, ...,30. Figure 5.12 shows the reduced vibration of
the output during periods T(28), T(29), T(30) compared with the vibration
during periods T(2) and T(3) displayed on the top plot. Figure 5.13 displays
the complex gains of the control sinewaves during the periods from T(2) to
T(30). The bottom plot in Figure 5.13 shows the amplitude of the three output
harmonic components, which were obtained by estimation of the amplitude
and phase of the sinewave outputs of the three FSF filters concentrated around
1.5Hz, 5.5Hz and 7Hz. This figure shows that the harmonic components of
output y at frequencies 1.5Hz, 5.5Hz and 7Hz goes to near zero after 30 periods.
The sinewave phase and amplitude estimation during T(k) is carried out by
ordinary least-squares fitting of a sinewave to the last 100 output samples
of each FSF applied to y during T(k). The top plot in Figure 5.13 shows
the convergence of the complex amplitudes of the three sinewave components

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 114 Active Sound and Vibration Control

Complex gains of control inputs at three frequencies

Complex gains of outputs at three frequencies

Figure 5.13 The complex gains of the control inputs and the amplitudes of
the estimated output harmonic components during time periods
T(2)-r(30)

of the control inputs, the computation of which was based on the FSF-RLS
method of eqns (5.8) applied at each main harmonic component of disturbance
d.

For the sake of completeness, and to illustrate the size of the control input
used, Figure 5.14 shows the control input during time periods T(2) — T(3) and
T(28) - T(30). Note that in Figure 5.12 the output during periods T(28) -
T(30) still contains the harmonic components of the disturbance at frequencies
4Hz, lOHz and 11 Hz which were not intended to be cancelled by the controller.
Hence, the controller achieved only about 20 dB reduction in the vibration of
the output y.

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 Adaptive harmonic control: tuning in the frequency domain 115

Control input during periods 2-3

200 400 600 800 1000 1200 1400

Control input during periods 28-30

-20

-40
200 400 600 800 1000 1200 1400 1600 1800 2000

Figure 5.14 Time signals of the control input during periods T(2) - T(3) and
T(28) - T(30)

5.6 Conclusions

The basic schemes of frequency selective RLS (FS-RLS) and LMS (FS-LMS)
methods have been described for periodic disturbance compensation. For the
FS-LMS method a stability robustness theorem was given. Advantages of the
methods were pointed out as (i) no principle limits of performance as in linear
feedback control, (ii) high degree of adaptivity and (iii) high degree of stability
robustness. A disadvantage is that the scheme only works for disturbances
dominated by a discrete spectrum, although that can be slowly time varying.
This introductory research into the topic has some shortcomings which
future research will resolve:

• In the methods presented the discrete spectrum of the disturbances was

assumed to be known. Preliminary automatic spectrum analysis of the

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 116 Active Sound and Vibration Control

disturbed output can be used to set the frequencies of interest. Sensi-

tivity to estimation errors in the disturbance frequencies is high. How
to use the best estimates and sensitivity analysis should be the topic
of future research. Detection of changes in an online system is of great
practical interest.
• The efFect of white noise disturbances should be analysed more precisely
although the simulation shown indicates practically acceptable low sen-
sitivity.
• The effect of nonlinear dynamics added to the basically nonlinear dy-
namics is a practically relevant question.
• Best numerical implementation can be investigated, parallel computa-
tional architectures can be made use of and large numbers of frequency
points can be handled while performance is monitored and control action
is supervised.
• Extension to multi-input multi-output systems is a future task which is
likely to result in a stable and highly adaptive control scheme to numer-
ous applications where the disturbance vector is dominated by a discrete
spectrum.

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