2013 International Conference on Power, Energy and Control (ICPEC)
ESTIMATION OF HARMONICS AND
INTERHARMONICS USING PHASE-
LOCKED LOOP
Sharmitha D
Department of EEE
Adhiyamaan College of Engineering
Hosur 635 109
Tamilnadu, India
Email: sarmie88@gmail.com
Abstract-In recent times, the parameters of power systems are
highly distorted, due to the increased application of nonlinear
loads. The monitoring of harmonic and interharmonic
distortion is an essential issue for delivering high quality power.
The proposed harmonic and interharmonic estimation
algorithm employs a filter bank and multirate signal processing.
An enhanced phase-locked loop is used to estimate the
parameters of the harmonics. The filter bank consists of a set of
bandpass filters. The filters extract each of the harmonic and
interharmoniccomponents. Further, the noise level of the
components is also reduced. The extracted components are
downsampled. In the last stage, the enhanced phase-locked loop
estimates the amplitude, phase and frequency of the
decomposed signal. The bandpass filters are centered at the
estimated frequency. The estimation time and accuracy of this
method compares favorably with alternate methods.
Index Terms-Power quality, filter bank, multirate digital signal
processing, enhanced phase-locked loop, time-varying
harmonics and interharmonics.
I. INTRODUCTION
Due to the increased application of nonlinear loads, the
power quality (PQ) of power systems is affected. The current
harmonics may produce voltage drops on the transmission
lines or feeder’s impedance and, hence, it leads to voltage
disturbance at the load terminals. The reasons for the time-
variance of the voltage and current harmonics arethe constant
changes in the configuration of the system and the load
characteristics, the wide usage of distributed resources and
the possibility of connecting islanded microgrids. The
extraction and detection of the harmonics on the power lines
become difficult due to their time-varying nature. Hence,
accurate estimation of the parameters of the time-varying
harmonics is inevitable to maintain the PQ under control [1]-
[3].
The proposed Enhanced Phase-Locked Loop (EPLL)-
based harmonic estimation algorithm estimates the
parameters of the harmonics and interharmonics. The EPLL
is used as the primary tool for harmonic and interharmonic
estimation under time-varying conditions [4]-[6]. In [7],
several of these sections are connected in cascade. Each
section is tuned to estimate a single harmonic. Typically, the
978-1-4673-6030-2/13/$31.00 ©2013 IEEE
Sundararajan D
Department of EEE
Adhiyamaan College of Engineering
Hosur 635 109
Tamilnadu, India
Email:d_sundararajan@yahoo.com
convergence time is about 18 cycles. In the case of large
frequency deviation, the convergence time for higher
harmonics can be more than 100 cycles. A novel
multiratefilter bank approach for the extraction of harmonic
and interharmonic is proposed in [8]. The method employs
bandpassfilterswith a narrowpassbandalong with
downsamplersand EPLL as an estimator. The computational
complexity of estimating the parameters of time-varying
harmonics is considerably reduced.
In this paper, we propose a new EPLL-based harmonic
estimator that combines the advantages of [7] and [8]. One
feature of the proposed method is the downsampling of the
high frequency harmonics, which results in only one setting
of the EPLL parameters. Further, the sampling rate and
hence, the convergence time are reduced. The system works
in a lower frequency range due to downsampling.The
proposed system is more suitable for implementation in
hardware and software with fixed-point arithmetic, which
results in a reduced cost.
The organization of this paper is as follows. The
fundamentals of digital filter bank are presented in Section II.
The concepts of downsampling and aliasing in digital
multirate processing are described in Section III. The
recursive equations of the EPLL method and the proposed
harmonic estimator are presented in Section IV. The
proposed system which is a combination of filter bank,
downsampler and EPLL is described in Section V. A brief
description of the Short-Time Fourier Transform (STFT) is
given in Section VI. In Section VII, the simulation results are
presented and compared. The conclusion is presented in
Section VIII.
II. DIGITAL FILTER BANK
An analysis digital filter bank is an interconnection of a set
of digital bandpass filters with a common input [9]-[11]. The
analysis filter bank,shown in Fig. 1, decomposes the input
signal x n into a setofcomponents corresponding to different
bands of the spectrum of the signal. The bandpass filter
separates the harmonic and interharmoniccomponentsof the
distorted power system signal. Further, the bandpass
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Fig. 1. Analysis filter bank.
Fig. 2. Magnitude of the frequency responsesof digital bandpass filters.
Fig. 3. Block diagram representation of the downsampler.
filters reduce the noise level of the components. Fig. 2
showsthe frequency responses of the bandpass filters.The
first response is centered at 0.1 radians. Similarly, the other
responses are centered at their respective harmonic
frequencies. In this figure, π corresponds to normalized
frequency.For example, if the sampling frequency is 1000
cycles/sample, then π corresponds to 500 cycles/sample. The
Fig. 4.The real and apparent frequency due to aliasing effect with N=6400
sampleswith a downsampling factor of M=16. Frequency components with
index 0 to 200 have true representations. Frequency components with a
higher index, have alias representations with corresponding apparent
frequency.
passband of each filter lies in the neighbourhoodof its center
frequency.
III. DOWNSAMPLING
A downsampler discards (M-1) samples of every set of M
consecutive samples of the signal, where M is
thedownsampling factor.Therefore, the sampling rate of the
downsampled signal is 1/M times of that of the original
signal.The block diagram representation of the downsampler
is shown in Fig. 3.The outputy n of the downsampler is
given by
yn x nM (1)
where x n is a input sequence.
Fig. 4 showsthe real and apparent frequency due to aliasing
effect with N=6400 samples with a downsampling factor of
M=16. Therefore, the new sampling rate is 6400/16=400
cycles/sample. This implies that frequency from indices 0 to
200 only will have unambiguous representation. The higher-
indexed frequencies will have aliased representations with
apparent frequencies. For example, consider a frequency
component with 150 cycles/sample.As the frequency is less
than fs/M = 200,it has anunaliased representation. However, a
frequency component with frequency 250 cycles/sample is
folded back into the range 0 to 200 [12]. Therefore, it is also
represented with an apparent frequency of 150 cycles/sample,
as shown in Fig. 4.
IV. PROPOSED EPLLHARMONIC ESTIMATOR
EachEPLLis preset to work in the neighbourhood of a
harmonic frequency and it estimates the amplitude, frequency
and phase of the corresponding sinusoidal component. It also
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tracks the parameters of the input signal in a neighbourhood. 3. The structure is highly immune to internal
The major advantage of the proposed harmonic estimator is parameter variations and noise.
the good estimation of the parameters of the signal even in
non-stationary and noisy environments.The block diagram of
the EPLL structure is shown in Fig. 5.The upper and lower
branches of the EPLL circuit respectively, estimatethe
amplitude A t , the frequency and phaseφ t of the
preselected harmonic/interharmonic of their input signal. The
required arithmetic operations of the EPLL are three
multiplications, two additions, two trigonometric
functionalevaluations, three integrations, and three gain
constants.
The EPLL receives input signalu t and simultaneously,
provides the errorsignal e t , the outputy t , and the
corresponding amplitude A t , frequency ω t , and phase
angle θ t . Speed and accuracy of the EPLL structure are
controlled by means of three independent internal parameters
μ1, μ2, and μ3. Parameter μ1controls the amplitude with its
time constant approximated as τA=1/μ1. The other two
parameters μ2 and μ3, control the frequency and phase
respectively. The structure of the EPLLis simple. Further, it
provides a good estimation of the error signal e t , the
outputy t and the corresponding amplitude A t , frequency
ω t and phase angle θ t even in noisy environments. Note
that φ t ω t θ t .
Thethreeequations governing the estimation of the
Fig. 5. EPLL structure
parameters,which can be deduced from Fig. 5, are

A t µ e t sin φ t (5)

ω t µ e t cos φ t (6)

φ t ω t µ ω t (7)

On the top of A t , ω t , and φ t , the dot indicates the

respective derivatives, where, for example, A t . The
error signal is e t u t y t .The input signal is .
The preselected harmonic/interharmonic component isy t
A t sin φ t and A t , ω t and φ t are the amplitude,
frequency, and phase.
The change in each of the variables, the amplitude,
frequency andphase, are mutually dependent.The parameters
μ1, μ2, and μ3have to be selected so that there is a
compromise between convergence and overshoot. The EPLL
provides the following advantages:

1. The extracted outputy t not only closely

resemblesto the input signal, but also in
synchronization with the desired component of the
input; i.e., the phase differenceδ t is constant
between y t and u t .
2. Online estimation of the amplitude A t , total phase
φ t , constant phase δ t and their corresponding
time-derivatives, of the
preselectedharmonic/interharmonic component of Fig. 6.Proposed structure of harmonic estimation.
the input signal are provided.

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TABLE I
THE REAL AND APPARENT FREQUENCIES FOR VARIOUS XSTFT e , n xn mwme
DOWNSAMPLING FACTORS.
(8)
Frequency(Hz)
k Mk
wherew[n] is an appropriately selected windowed sequence
Real Apparent
[9]. The window enables the selection of a portion of the
yk vk
sequence x n .It is understood that the spectral characteristics
1 16 50 50
of the selected section are stationary within a given tolerance.
2 8 100 100 If w[n] =1, the window becomes a rectangular window. As
3 16 150 150 the use of a window function makes the selected sequence
finite, the existence of the selected sequence is guaranteed.
4 12 200 200
As the window is centered at different time instants, the
5 16 250 150 STFT is a function of two variables n and ω. For a segment
6 14 300 158.57 of a sinusoidal sequence, the STFT is virtually the same as
the Discrete Fourier Transform (DFT).
7 16 350 50
8 14 400 58.57 VII. SIMULATION RESULTS
9 16 450 50
The simulation results of the proposed system
10 11 500 88.18 withvariations in the parameter of the input signal are
11 16 550 150
presented in this section. The fundamental frequency is 50
Hz. For better visualization of results 10% deviation in the
12 12 600 70 amplitude, up to1 Hz change in the fundamental frequency
13 16 650 150 and 20% change in the phase of the input signal are
considered. A set of values of convergence factors μ1=200,
14 15 700 154
μ2=300, and μ3=2 are chosen as the basis for the parameter
15 16 750 50 settings of the EPLL. The integrators are set to
zeroinitialconditions.Theamplitude, frequency and phase of
all the harmonics can be estimated.
V. PROPOSEDSTRUCTURE Fig. 7 shows thesinusoidalinput signal. At t= 0 msec, the
frequency is 50 Hz with amplitude one andphase 60 degrees.
A block diagram of the proposed EPLL-based harmonic At t = 280 msec, the amplitude is reduced to 0.5.At t=500
estimation algorithm is shown in Fig. 6. The first stage is a msec, the amplitude is restored to one. At t=750msec,the
digital filter bank which is an interconnection of a set of amplitude is again reducedto 0.5, the frequency is changed to
digital bandpass filters with a common input. The bandpass 45 Hz and phase to 30degrees.
filter separates the harmonic and interharmonic components Fig. 8 shows that the obtained values for the amplitude,
of the distorted power system signal and also reduces the which show that the amplitude variation of the input signal is
noise level. The second stage is the set of downsamplers used tracked quite closely. It can be seen that the step change in
to downsample the input signal.The advantage of the use of frequency at t =750msec causes ripples in amplitude. This
downsamplers are the reduction of the computational also illustrates that an unbiased steady state is obtained after
complexity and the adequacy of single setting of the EPLL almost onecycle.It is noted that increasing the value of μ1
parameters.The last stage is made up of a set of EPLL increases the speed of convergence, however it also creates
estimators for estimating the amplitude, frequency and phase oscillations in the response.
of the input. Note that D/A and A/D interfacesare required in Fig. 9 shows the estimated values of the phase. It can be
using an analog EPLL in a digital system. An alternate seen from the figure that the phase angle jumps at t=750
solution is to use a digital EPLL.Table I shows the realand msec. The phase angle deviation in the neighbourhood is
apparent frequencies for various downsampling factors with trackedclosely by the EPLL within a very short time.Fig.
k=1,2,...,15, where k is the number of bandpass filters in the 10shows the estimation of the phase angle. The solid line
filter bank. The columnsyk and vkshow the true and the shows the actual phase of the input signal without the phase
apparent frequency values before and after downsampling change and dashed line shows the change in phase. The phase
with a downsampling factor Mk. jump following a phase change at t = 750 msec is not clear in
VI. SHORT-TIME FOURIER TRANSFORM Fig. 9, whereas in Fig. 10, the dotted line shows the change
in phase clearly. The frequency is also estimated with the
The short-time (windowed or time dependent) Fourier slope change. Fig. 11 shows the EPLL output. It can be seen
transform (STFT), of a sequence x n is defined by the that the variations of the input signal is tracked by EPLL
equation quite closely. The convergence and speed of the EPLL is
very fast.

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Fig. 10. Estimation of the phase angle.

Fig. 7. Input signal to the EPLL.

Fig. 8. Estimation of the amplitude. Fig. 11. Output of the EPLL.

Fig. 9. Estimation of the phase. Fig. 12.STFT of the input signal.

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Fig. 12 shows a part of the magnitude of the STFT of the [6] M. Karimi-Ghartemani, S. A. Khajehoddin, P. K. Jain, and A. Bakhshai,
“Derivation and Design of In-Loop Filters in Phase-Locked Loop
input signal. There are 10000 samples in the input signal. It is Systems”, IEEE Trans. Inst. Measurement, vol. 61, no. 4, pp. 930–939,
well known that DFT of a single sinusoid with N samples Apr. 2012.
yields a symmetric spectrum with magnitude N/2. The [7] M. Karimi-Ghartemani and M. R. Iravani, “Measurement of harmonics/
magnitude of the spectral value is 3000. The magnitude of interharmonics of time-varying frequencies,” IEEE Trans.Power Del.,
the sinusoidal input signal is (3000+3000)/10000=0.6, vol. 20, no. 1, pp. 23–31, Jan. 2005.
whichis inaccurate. Due to leakage, the STFT output is [8] J. R. Carvalho, P. H. Gomes, C. A. Duque, M. V. Ribeiro, A.
blurred. The S.Cerqueira, and T. L. Baldwin, “A PLL-Based Multirate Structure for
Time-Varying Power Systems Harmonic/Interharmonic Estimation,”
EPLL tracks the time-varying input more precisely compared IEEE Trans.Power Del., vol. 24, no. 4, pp. 1789–1800, Oct. 2009.
to the STFT. [9] S. K. Mitra, Digital Signal Processing: A Computer-based Approach,
3rd ed. New York: McGraw-Hill, 2006, pp. 383–385.
VIII. CONCLUSION [10] D. Sundararajan,A Practical Approach to Signal and Systems, New
York: Wiley 2008.
This paperpresentsa harmonic estimator, which is a [11] D. Sundararajan, Digital Signal Processing: Theory And Practice,
combination of multirate digital signal processing and Singapore: World scientific, 2003.
EPLL.Thecombinationis highly efficient for power quality [12] D. Sundararajan, The Discrete Fourier Transform: Theory, Algorithms
monitoring applications. For non-stationary signals, the And Applications, Singapore: World scientific, 2001.
STFT is the method, if the signal is stationary over the
window width. When these conditions fail, alternate methods
have to be applied for the accurate estimation of the
parameters of a signal. In the proposed method, bandpass
filters are used to isolate the harmonics and also to reduce the
noise. The downsampling of the filtered signal allows the
single setting of EPLL parameters. Further, the device can
operate at a lowersampling rate resulting in the reduced cost
of the device. EPLL structure is different from that of the
conventional PLL in that additional circuits are incorporated
for the extraction of amplitude, frequency and phase of the
input signal. The advantages of the proposed method include:
(1) The simplicity of the structure and its robustness to
disturbances and setting of its internal
parameters,(2)Adaptivity to arbitrarilyvarying conditions and
(3)Ability to work in noisy environments.
The conclusion, which is evident by the simulation results
shown, is that the combination ofbandpass filters,
downsamplersand EPLL provides an efficient system for the
extraction oftheparameters oftheharmonics and
interharmonics, which are located in a close
neighbourhood.Theproposed system can be easily
implemented in hardware.Asmootheroutput can be obtained
by the use of lowpass filter in the loops of the
EPLL.Theproposed system can be easily adapted to three-
phase power systems.

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