Harmonic Parameter Estimation of a Power Signal
using FT-RLS Algorithm
Santosh Kumar Singh1 Member IEEE
Electrical Engineering Department
National Institute of Technology Silchar, India
Email- santoshkrsingh.nits@gmail.com
Abstract— Many algorithms have been proposed for harmonic
estimation to improve the power quality performance, but till
today it is still a challenge for accurate estimation. In this paper
an adaptive filtering algorithm called Fast Transverse Recursive
Least Square (FT-RLS) is applied for the first time for
estimating harmonic parameters. The algorithm has been
considered to estimate the amplitudes, phases and frequency in
case of time varying power signals containing harmonics in the
presence of White Gaussian Noise in MATLAB simulating
environment. Also comparison results with two recently
proposed algorithms such as Forgetting Factor Recursive Least
Square (FF-RLS) and Recursive Least Square (RLS) are
presented to show the effectiveness of the proposed FT-RLS
algorithm.
Index Terms-- Extended Least Mean Square (ELMS), Fast
Transverse Recursive Least Square (FT-RLS), Forgetting
Factor Recursive Least Square (FF-RLS), Least Mean
Square (LMS), Recursive Least Square (RLS)
I. INTRODUCTION
The presence of nonlinear loads and the increasing use of
power electronic devices during the last decades contribute to
change the characteristics of voltage and current waveforms
from ideal sinusoidal to non sinusoidal in power systems. Thus
destroying the power quality and generating harmonics in the
signal. Under these conditions advanced signal processing
techniques are required for accurate estimation of harmonic
parameters such as amplitude, phase and frequency. Thus
accurate computation of these parameters is also a challenging
problem in power system.
The accurate estimation of supply frequency and voltage
during its variation with time is very essential. Variations in
system frequency from its normal value indicate the
occurrence of a corrective action for its restoration to its
original value. In this context a large number of numerical
methods are available for frequency estimation from the
digitized samples of the system voltage [1-3]. Conventional
method assumes that the nature of the voltage waveform is
required to be purely sinusoidal and therefore the time
between two zero crossings is an indication of system
frequency. The obtained signal parameters can be employed
for revenue purposes, electrical grid characterization and
978-1-4673-6487-4/14/$31.00 ©2014 IEEE
Arup Kumar Goswami2 Member IEEE
Nidul Sinha3 Senior Member IEEE
Electrical Engineering Department
National Institute of Technology Silchar, India
active compensation of power system disturbances.
However various adaptive filtering approaches has
been applied to characterize harmonic signal parameters
such as Least Mean Square (LMS)[3-4], Extended
Least Mean Square (ELMS)[5-6], Recursive Least
Square (RLS)[7-8], Forgetting Factor Recursive Least
Square (FFRLS)[9], Normalized Least Mean Square
(NLMS) [10-11] etc but each of them has several
limitations in terms of inaccuracies due to the presence
of noise, harmonics and other changing parameters such
as change in fault inception angle, change in fault resistance
etc in the signal.
Keeping this in mind, three iterative adaptive filtering
algorithms such as Forgetting Factor Recursive Least Square
(FF-RLS)[9],Recursive Least Square (RLS) and a novel Fast
Transverse Recursive Least Square (FT-RLS) along with
their comparisons are presented in this paper for fast and
accurate estimation of nominal and off-nominal power system
frequency of a harmonic signal. During comparison we
observe that the proposed FT-RLS algorithm outperforms on
FF-RLS and RLS algorithm.
The paper is organized as follows. Section I describes the
introduction part of the recursive adaptive algorithms used for
harmonic estimation. Section II describes the theoretical
background which formulates the harmonic estimation
algorithms based on FT-RLS. Section III discusses the FT-
RLS algorithm implementation. Section IV describes the
signal modeling part for harmonic estimation whereas Section
V describes the simulation test and Estimation performance
of the proposed FT-RLS algorithm through MATLAB
simulations. Finally, test result and simulation performance is
explained in Section VI whereas conclusion is briefly
explained in Section VII.
II. THEORETICAL BACKGROUND
The general set up of an adaptive-filtering environment is
illustrated in Fig.1.
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Fig-1: Adaptive Filtering Structure
Where k is the iteration number, x (k ) denotes the input
signal, y (k ) is the adaptive-filter output signal, and d (k )
defines the desired signal. The error signal e( k ) is calculated
as d ( k ) − y (k ) . The error signal is then used to form a
performance (or objective) function that is required by the
adaptation algorithm in order to determine the appropriate
updating of the filter coefficients. The minimization of the
objective function implies that the adaptive-filter output
signal is matching the desired signal in some sense. The
algorithm is the procedure used to adjust the adaptive filter
coefficients in order to minimize a prescribed criterion. The
algorithm is determined by defining the search method (or
minimization algorithm), the objective function, and the error
signal nature. The choice of the algorithm determines several
crucial aspects of the overall adaptive process, such as the
existence of Sub-optimal solutions, biased optimal solution,
and computational complexity. The basic objective of the
adaptive filter is to set its parameter e( k ) , in such a way that
its output tries to minimize a meaningful objective function
involving the reference signal.
III. FT-RLS ALGORITHM
The complete recursion for efficiently implementing the
FT-RLS adaptive filtering algorithm for the estimation of
errors is given in Table-1 below in Eq-1 to 13.
A ( n ) =Optimal Forward Predictor Coefficient Error
f ( n ) =Estimation Error of the Forward Predictor
α ( n ) = Least Square Value of Forward Prediction Error
rf ( n ) = Optimal Coefficient for Forward Predictor
k L +1 ( n ) =Augmented Gain Vector
rb (n) =Optimal Coefficient for Backward Predictor
ψ ( n ) = Posterior Backward Prediction Vector
L = Successive Samples of one Signal
K (n) = Gain Vector
ε ( n ) = Priori Estimation Error
W ( n ) = Optimal Least Square Coefficient Vector
e ( n ) = Error Vector
d ( n ) = Desired Signal Vector
X L +1 ( k ) = Augmented Input Vector
G ( n ) = Optimal RLS Backward Predictor Coefficient
Vector
IV. SIGNAL MODELING
Let x(k ) be an input signal buried in white Gaussian
noise μk represented by the following structure:
K
x(kTs ) = ∑ Ak sin(ωk kTs + φk ) + μk (14)
k =1
So for the purpose of estimating the signal in parametric form
can be written as:

x(kTs ) = [sin ωk Ts cos ωk Ts ] ⎡⎣α β ] + μ k

T
(15)
Now the signal can be modeled as,
x ( kTs ) = xk * [ wk ] + μk
T
(16)

Where xk = [ sin ωkTS cosωkTs ] and wk = [α β ] which are

input and weight vector matrix of the signal.
Where α ,β parameters to be estimated given by:
α = Ak cos φk
(17)
β = Ak sin φk
So the actual required parameters are amplitude ( Ak ),
phase ( φk ) and frequency ( f k ) which can be obtained
Where, using,
φ ( n ) = Priori Forward Prediction Errors Ak = α 2 + β 2 (18)
x ( n ) =Input Vector of the Signal

158
 ⎛β ⎞ Recursive Least Square (RLS) and Fast Transverse Recursive
φk = tan −1 ⎜ ⎟ (19) Least Square (FT-RLS). Among these algorithms, the
⎝α ⎠ performance of proposed FT-RLS algorithm is found to be
superior for estimating harmonic parameters than FF-RLS
1 ⎡ −1 ⎧ x ( kTs ) − μk ⎫ ⎤
fk = ⎢sin ⎨ ⎬ − φk ⎥ (20) and RLS.
2π kTs ⎢⎣ ⎩ Ak ⎭ ⎥⎦

Fig-3: Amplitude plot for the generated signal corrupted with

Noise at 20 dB SNR

Amplitude, Phase & Frequency plot for Fundamental

Harmonic (50 Hz) Signal

Fig-2: Flowchart for Algorithm Performance and

Estimation Fig-4: Amplitude (Estimated) plot for Harmonic signal
corrupted with Noise using FF-RLS, RLS & FT-RLS
V. SIMULATION TEST
To evaluate the performance of the proposed Fast
Transverse Recursive Least Square (FT-RLS) algorithm for
harmonic analysis in estimating the harmonic amplitude,
phase & frequency a discretized signal having a fundamental
frequency of 50 Hz is generated using MATLAB. The power
signal consists of fundamental harmonics is given in equation
(21). This type of signal is typically present in industrial load Fig-5: Amplitude (Error) plot for Harmonic signal corrupted
comprising of power electronic devices and arc furnaces. with Noise using FF-RLS, RLS & FT-RLS
x(t ) = 1.25sin(ωt + 70D ) + μk (21)
The aforementioned signal is corrupted by random noise
μk = 0.05 randn having normal distribution with zero mean
and unity variance. The amplitude given is in per unit (p.u)
value. The data are sorted for amplitude, phase and frequency
as per algorithm given in Fig-2. The harmonics are the integer
multiples of the fundamental frequency. The presence of
distortion in the sinusoidal signal obtained from the original Fig-6: Amplitude (MSE) plot for Harmonic signal corrupted
wave gives the measurement of THD. Sampling frequency with Noise using FF-RLS, RLS & FT-RLS
(Fs) = 2000 Hz, Number of iterations (Samples) (K=N=100),
and length of the signal to be tested as 500 samples/10 cycles
at frequency F= 50 Hz confirming windowing 200-ms in
practice & signal to noise ratio is taken 20dB for the
generation of the signal given in Fig-5. The parameters for
amplitude, phase and frequency are estimated one by one
using three different adaptive filtering algorithms such as
Fig-7: Phase (Estimated) plot for Harmonic signal corrupted
Forgetting Factor Recursive Least Square (FF-RLS), with Noise using FF-RLS, RLS & FT-RLS

159

Fig-8: Phase (Error) plot for Harmonic signal corrupted with
Noise using FF-RLS, RLS & FT-RLS
Fig-9: Phase (MSE) plot for Harmonic signal corrupted with
Noise using FF-RLS, RLS & FT-RLS
Fig-10: Frequency (Estimated) plot for Harmonic signal
corrupted with Noise using RLS & FT-RLS
Fig-11: Frequency (Error) plot for Harmonic signal corrupted
with Noise using RLS & FT-RLS
Fig-12: Frequency (MSE) plot for Harmonic signal corrupted
with Noise using RLS & FT-RLS
VI. RESULT & DISCUSSION
To improve the performance in the adaptive filtering
algorithms, many extensions and modifications are found in
the literature [9-11]. The enhanced performances gained
through these modifications are less deviation, faster
convergence and recursive calculations. The initial
covariance matrix is taken as αI, where I is the identity matrix
and α= 100. Here the system structure matrix as is taken in
both algorithms such as FF-RLS, RLS & FT-RLS is a (1×2)
matrix of fundamental harmonic signal. Estimation of
160
frequency is done after the estimation of amplitude and phase
of the signal by using Eq-20.
Fig-1 represents the Adaptive Filtering Strategy which
clearly describes the functional operation of any adaptive
algorithms whereas Fig-2 describes the flow chart for
Algorithm Performance and Estimation. Fig-3 shows the
Amplitude plot for the harmonic signal corrupted with noise
and without noise.
Fig-4, Fig-5 & Fig-6 represents the plot for Estimated
Amplitude, Error in Amplitude and Mean Square Error (MSE)
of Amplitude for harmonic signal using three different
adaptive algorithms such as FF-RLS, RLS & FT-RLS
respectively. The performance of proposed FT-RLS algorithm
can be observed clearly. The Amplitude error and mean square
error (MSE) estimation in proposed FT-RLS is less than that
of FF-RLS & RLS.
Fig-7, Fig-8 & Fig-9 represents the plot for Estimated
Phase, Error in Phase and Mean Square Error (MSE) of Phase
for harmonic signal using three different adaptive algorithms
such as FF-RLS, RLS & FT-RLS respectively. The Phase
error and mean square error (MSE) for Phase estimation in
proposed FT-RLS is less than that of FF-RLS & RLS. Overall,
the performance of the proposed FT-RLS algorithm is found to
be better.
Fig-10, Fig-11 & Fig-12 represents the plot for Estimated
Frequency, Error in Frequency and Mean Square Error (MSE)
of Frequency for harmonic signal using three different
adaptive algorithms such as FF-RLS, RLS & FT-RLS
respectively. The Frequency error and mean square error
(MSE) for Frequency estimation in proposed FT-RLS is less
than that of FF-RLS & RLS. Overall, the performance of the
proposed FT-RLS algorithm is found to be better.
Table-2: Performance of FF-RLS, RLS & FT-RLS Algorithm
Table-2. Shows a comparative assessment among these
three algorithms, from this table we conclude that three
estimation errors of all the parameters (Amplitude, Phase &
frequency) and computational time are reduced in case of FT-
RLS Algorithm for harmonic signal. So this algorithm
outperforms on FF-RLS & RLS algorithm.
VII. CONCLUSION
This paper presents the estimation of amplitude, phase and
frequency of a harmonic signal corrupted with white
Gaussian Noise by various recursive estimation techniques
such as FF-RLS, RLS and proposed FT-RLS. However
choice of the covariance matrix of the input signal is very
crucial at the initial instant for both algorithms. Improper
choice of covariance matrix leads to more computational time
with more estimation error. But at the same time these
recursive algorithms are very simple by the representation of
the parametric form of the signal. The computational time is
less due to the simplicity of these algorithms and estimation
error is also less. The overall performance of the proposed
FT-RLS algorithm is found to be one of the effective
recursive algorithm as compared to recently proposed FF-
RLS and RLS algorithm for power system harmonic
estimation.
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Santosh Kumar Singh received
B.Tech & M.Tech degree in electrical
engineering from West Bengal
University of Technology (WBUT),
Kolkata, India, in 2009 & 2011
respectively. He is currently a Ph.D
research scholar in electrical
engineering department at NIT Silchar,Assam, India. His
research interest includes system identification and
application of signal processing and soft computing
techniques for power quality analysis.
Arup Kumar Goswami received
B.Tech degree in electrical
engineering from National Institute of
Technology (NIT) Kurukshetra,the
M.Tech degree in electrical
engineering from Birla Institute of
Technology (BIT) Meshra, Ranchi and the Ph.D degree in
electrical engineering from Indian Institute of Technology
Roorkee (IIT-Roorkee), India. He is currently an Assistant
Professor of electrical engineering department at National
Institute of Technology Silchar (NIT Silchar). His research
includes Power Quality, Power System
Reliability,Congestion Management, Deregulated Power
Systems, Application of Power Electronics in Power Systems,
Power System Protection etc.
Nidul Sinha received B.E degree in
electrical engineering from National
Institute of Technology (NIT) Agartala,
former Tripura engineering college in
1984 , the M.Tech degree in electrical
engineering from Indian Institute of
Technology Delhi (IIT-Delhi),India in 1989, and the Ph.D
degree in electrical engineering from Jadavpur University
(JU), Kolkata, India. He is currently a Professor of electrical
engineering department at National Institute of Technology
Silchar (NIT Silchar). His research includes Power system
optimization, Automatic generation control under
conventional and deregulated environment, Application of
soft computing techniques, technical forecasting, Case base
reasoning, intelligent instrumentation. He is a senior member
of IEEE and serves as a reviewer for IEEE
TRANSACTIONS ON POWER SYSTEMS and POWER
DELIVERY and many other reputed journals.