International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190

ICRTIET-2014 Conference Proceeding, 30th -31st August 2014

Improving the Convergence Time of Adaptive Notch Filters to

Harmonic Detection
Vipin chauhan# , Pankaj Bansal*
Department of Electronics Engineering

ABSTRACT
As a kernel technique of active power filter (APF),
harmonic detection and its tracking method are
investigated in this paper. Firstly, according to APF
discussed here, an adaptive notch filter is presented
and the LMS algorithm is examined to meet the
crucial requirements of the filters tracking behavior
under time-varying condition. Secondly, an upgraded
method is emphasized on the basis of the traditional
LMS algorithm, the variable step-size method is
analyzed with both its merits and demerits, in order to
guarantee the convergence rate and steady-state
performance, a correct- max mending method is
introduced as the major solution to improve the
performance of variable step-size, so as to obtain good
tracking reliability as well as rapidity. For better
understanding, the simulation results are given out and
illustrated, with the friendly toolbox of Matlab.
Index Terms Adaptive filtering, harmonics, LMS
algorithm, variable step-size

I. INTRODUCTION
Harmonics, as the major issue of the power quality
problems, concerns both power manufacturers and
customers, becomes a great challenge to suppress it
under corresponding criterions. Recently, APF is
efficiently designed and widely applied to many
rigorous
occasions
to
suppress
harmonics
automatically.
Along with the development of semiconductorelectron industry, a lot of power electronic devices
and computer loads are being used widely in power
transmission and distribution systems. Not only are
these types of loads a source of additional harmonics,
but they also have a high sensitivity to non-sinusoidal
waveforms. Harmonic pollution has become very
serious problem for both utilities and customers. What
an ideal public power network provides is the reliable
and clean fundamental frequency sinusoidal electric

power. There are many ways to avoid and overcome

the harmonics problem, the passive power filters were
used conventionally in power systems to suppress
harmonic and improve power factor. With the
advancement of power electronic apparatus (GRT,
GTO, IGBT etc.), microcomputer, and control
technology, APF is getting practical use. Compared
with the passive power filter, it possesses an obvious
advantage of controllability and responsibility. Most
harmonic detecting methods applied are realized by
analog circuit, it costs more and because of the special
sensitivity to frequency and temperature of the analog
components, it may lead to inaccurate estimates of
harmonic magnitudes, and lead to bad controllability
of APF. The digital methods should be adopted to
save investment and enhance reliability.
Fig. 1 shows the basic elements of active power filter.
The APF uses a power converter to supply the power
line with the current harmonics injected by the nonlinear load. The power supplied, from the utilities
point of view, is in the form of current of equal
magnitude but opposite phase to the injected
harmonics. Current injection is performed so as to
make the distorted signal sinusoidal. Current injection
refers to the controlled connection of a dc source of
current. The dc source is connected through an
inverter. The inverter, through injection of current,
supplies the utility with harmonic content of opposite
phase to cancel out the distortions [1]-[3]. The filter
may also be used for real-time compensation of
harmonic changes that occur. The system is only
delayed by the length of the cycle sampled and the
time to compute the content of the harmonics. APF
applies itself to estimate and compensate the total
harmonics synchronously, so in this paper, the
estimation of harmonics is outspreaded and the
improvement of key algorithm is emphasized.
However, the realization of the hardware will be
overleaped, as an application supporter; this article
attends to improve design of APF.

Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India

91

International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
ICRTIET-2014 Conference Proceeding, 30th -31st August 2014

correlation function of input signals nor the inverse

matrix. But it suffers from problems of misadjustment
and adaptive convergence [4].

II. THE PRINCIPLE OF ADAPTIVE NOTCH

FILTER AND THE TRADITIONAL LMS
ALGORITHM
A. Adaptive notch filter
In power system, the current can be described as

Respectively, the fundamental current is denoted by

i1 (t ) while the harmonics can be written as

Eliminating the fundamental components, the problem

arisen is how to extract the sinusoidal waveform;
apparently, it is tractable with adaptive notch filter. In
next paragraph how the adaptive notch filter works in
power system will be explained and its principle and
relative algorithm will be given.
The main structure of adaptive notch filter is shown
in Fig. 2, fundamental signal is regarded as the input
interference signal among i(t ) , ih (t ) as the desired
signal, the system fundamental voltage u(t) and its
phase-shifting signal as the reference input signal.
Here, i ( n) , xi (n) , (i=1, 2) represent the discrete
values of i(t ) , u(t). Since the reference input signal
has been processed by adaptive filter, the output signal
y(n) will track the variation of fundamental signal, and
y(n) approximates both amplitude and phase of
fundamental signal , then the desired harmonics can be
obtained by subtracting y(n) from i(n).
B. LMS algorithm and its performance
The least-mean-square (LMS) algorithm was proposed
by Widrow of Stanford University in 1960, it is based
on single-frequency adaptive filter to eliminate the
fundamental component from the distorted waveform,
and applied widely for it needs neither to compute the

The coefficients of the adaptive notch filter are

acquired using the LMS algorithm, and the
performance of the LMS algorithm should be
considered when it is used to update the coefficients.
1) LMS algorithm:
The LMS algorithm can be summarized as

w(n) is the coefficient of FIR filter, x(n) is the input

sequence of u(t), the error sequence {e(n)} can be
formed by taking the difference between y(n) which is
adaptation step-size. As a step-size parameter, plays
an important role later. The above adaptive notch filter
is initialized by setting x(0)=y(0)=0 .
2) Stability:
The LMS adaptive notch filter converges in the meansquare sense, that is, the transient MSE (Mean Square
Error) dies out, if the adaptation step-size satisfies
the condition

Where trR is the trace of the input correlation matrix

and K is a constant that depends weakly on the
statistics of the input data vector. In addition, this
condition ensures that on average the LMS adaptive
notch filter converges to the optimum filter. We stress
that in most practical applications, where the
independence assumption does not hold, the step-size
should be much smaller than k/trR. Therefore, the
exact value of K is not important in practice.
3) Tracking Performance of adaptive algorithms:
Tracking of a time-varying system is an important
problem in many areas of application. Considering, for

Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India

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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
ICRTIET-2014 Conference Proceeding, 30th -31st August 2014

example, a power line in which the harmonic

characteristics (amplitude, frequency) may change
with time for various reasons. If we want to
incorporate an adaptive notch filter in such a system,
then clearly the notch filter must monitor the changing
impulse response of the current pass so that it can
generate an accurate replica of the harmonics. This
will require the adaptive algorithm of a notch filter to
possess an acceptable tracking capability.

estimation using error signals, it can adjust the stepsize effectively and ensure rapid dynamic response
rate; furthermore, it decreases the interference of
uncorrelated noise signal updating the step-size, even
if under strong interferences, the system still hold
considerably small steady-state error.
The step-size recursive formula is as follows

4) Rate of convergence
As is intuitively expected, the smaller the step-size
and the farther the initial coefficients are from their
optimum settings, the more iteration it takes for the
LMS algorithm to converge.

Where p (n) is called as the average estimation of

error signal e (n), 0<<1 namely forgetting factor,
which is used to regulate the width of time window so
as to control the influence of former signals on present
states. The larger of earlier datum is, the weaker its
influence is. The sliding window width should be
efficiently small by choosing properly to ensure
effective memory. (1) is the average estimation
coefficients; 0<<1, mainly controls the
convergence rate of coefficient w(n); affects
significantly both convergence rate and steady-state
error, in order to ensure acceptable misadjustment, the
value of should be small enough. The choice of
and may refer to [6].
The Equation (8) will adjust the magnitude of while
maintaining its gradient, so as to satisfy the tradeoff.
The rest recursive formulas are same with the
traditional LMS algorithm.
To ensure stability of the algorithm, the step size
must be chosen in certain range. So the step-size will
be limited during the recursive process as follows

III. TWO IMPROVED METHODS OF LMS

ALGORITHM
A. Speed versus quality of adaptation
Discussion above shows that there is a tradeoff
between rate of convergence (speed of adaptation) and
steady-state excess MSE (quality of adaptation, or
accuracy of the adaptive filter). The first requirement
for an adaptive filter is stability, which is ensured by
choosing to satisfy (6), within this range, decreasing
to reduce the desired level of misadjustment will
decrease the speed of convergence; contrarily, if is
increased to increase the speed of convergence; this
results in an increase in misadjustment. This tradeoff
between speed of convergence and misadjustment is a
fundamental feature of the LMS algorithm.
B. Introduction of two new improved methods
1) Variable step-size LMS algorithm
The step-size parameter controls the rate of
convergence of the algorithm to the optimum solution,
so it also controls the dynamic response rate of the
adaptive filter. A large value of leads to large stepsize adjustments and thus to rapid convergence;
however, if is chosen too large the algorithm will
become unstable. While a small value of results in
slower convergence, its self-learning time is too long
even though the output waveform may eventually be
consistent with the ideal fundamental waveform. Both
solutions have their own advantages and
shortcomings. To ensure the self-learning ability of
the adaptive filter and at the same time to avoid
distortion of the filtered waveform, the variable
convergence rate method can be adopted.
[5] introduced a new variable step-size LMS algorithm
which controls the step-size updating by average

The choice of max must ensure the stability of the

algorithm, as a rule max should approach the critical
stable step-size of the traditional LMS algorithm. The
choice of min
should give attention to the
requirements of both steady-state misadjustment and
convergence rate, commonly a relatively small value.
2) Correct- max LMS algorithm
The variable step-size LMS algorithm based notch
filter is capable of separating and extracting a desired
sinusoidal component of a given signal if the
characteristics of the signal (amplitude, frequency) are
stable or change smoothly. However, when the case
gets worse, the performance may also vary with time
abruptly. The larger the initial step-size of the
algorithm is chosen, the more inoperative it becomes

Divya Jyoti College of Engineering & Technology, Modinagar, Ghaziabad (U.P.), India

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International Journal of Advanced Engineering Research and Technology (IJAERT), ISSN: 23488190
ICRTIET-2014 Conference Proceeding, 30th -31st August 2014

if any change in the amplitude and frequency of the

input signal occurred.
Obviously, the fixed initial step-size of variable stepsize LMS does not offer any better solution to
compromise between speed and quality, the ideal
solution is an correct- max LMS which is capable of
turning the initial step -size into an automated value
by judging its ceasing point, and thereby
accommodating sharp frequency and magnitude
variations and tracking the harmonics in the input
signal.

IV. COMPARISON OF TRACKING

PERFORMANCE
In order to compare the performances of algorithms
above-mentioned, a simulation calculation is carried
out. The first 10 sinusoidal periods are shown in Fig.
3. Axes x stands for time in 0.5ms, axes y stands for
the amplitude of the tracking signal in A.
When the adaptive notch filter is applied to the sharp
time-varying current, the tracking performances for
the LMS algorithm and improved LMS algorithms are
given by Fig. 3(a) and Fig. 3(b), (c), while the
waveform of real harmonics is given by Fig. 3(d). A
general comparison is easy to make, it has been
shown that: In Fig. 3(a), the LMS algorithm leads to
severe distortion compared with Fig. 3(d), it is
incapable in tracking sharp time varying harmonics. In
Fig. 3(b), the variable step-size LMS method proved
an abortion in tracking and hence the results cannot be
adopted. In Fig. 3(c), the correct- max LMS algorithm
corrects the former mistake and tracks the varying
coefficients nicely with expected speed and quality.

V. CONCLUSIONS
In this paper the improved harmonic tracking method
applied to APF is discussed, by choosing the adaptive
notch filter, we basically analyzed its adaptive
filtering algorithmsLMS algorithm, aiming at
estimating the sharp time-varying harmonics
efficiently, two improved methods named variable
step-size LMS algorithm and correct- max LMS
algorithm are introduced, a novel correct- max LMS
proposed can efficiently make up for the deficiency of
the variable step-size LMS which may abort in
tracking. With the powerful tool Matlab, we compared
their tracking performances, and verified that the new
proposed method is especially suitable in sharp timevarying systems.
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[3]
Sundar G,Sankaran ,On ways to improve
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[4] Dimitris G. Manolakis , Vinay K.Ingle ,Stephen
M.Kogon ,Staistical and Adaptive Signal Processing,
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[5] Li Hui Yi-ping,A Novel Adaptive Harmonic
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