772 IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 22, NO.

2, APRIL 2007

A New Power System Digital Harmonic Analyzer

Josif J. Tomić, Miodrag D. Kušljević, and Vladimir V. Vujičić, Member, IEEE

Abstract—In this paper, new digital instruments measuring

power-quality indicators and harmonic analyzers are developed.
A new technique for simultaneous local system frequency and
amplitudes of the fundamental and higher harmonics estimation
from either a voltage or current signal is presented. The structure
consists of three decoupled modules: the first one for an adaptive
filter of input signal, the second one for frequency estimation, and
the third one for harmonic amplitude estimation. A very suitable
algorithm for frequency and harmonic amplitude estimation is
obtained. This technique provides accurate frequency estimates
with error in the range of 0.002 Hz and amplitude estimates with
=
error in the range of 0.03% for SNR 60 dB in about 25 ms. The
theoretical basis and practical implementation of the technique
are described. To demonstrate the performance of the developed
algorithm, computer simulated data records are processed. Data
of the distribution power system voltage signals are also collected
in the laboratory environment and are processed in a newly devel-
oped digital PC-based harmonic analyzer. It has been found that
the proposed method really meets the need of offline applications.
Even more, by using the parallel computation algorithms, this
method should meet the need of online applications and should be
more practical.
Index Terms—Adaptive filtering, amplitude estimation, com-
puter relaying, digital filters, finite-impulse-response (FIR) filter,
frequency estimation, frequency measurement, harmonic anal-
ysis, harmonic analyzer, harmonics, power quality (PQ), recursive
algorithm.
I. INTRODUCTION
ITH the increasing use of nonlinear loads in power sys-
W tems, harmonic distortion becomes more serious. With
the progress of industry, power-electronic equipment is widely
used in power systems, but the nonlinear characteristics of this
equipment have also produced serious harmonic pollution. In
addition, many ill effects (i.e., worse power quality (PQ) for end
users, more loss in transmission lines, overheating of machines,
and malfunction of relays and breakers) are due to harmonic pol-
lution [18]. The detailed frequency analysis method has been
required to overcome these problems. It goes without saying
that harmonic analysis is a very important subject in power
systems. Regarding harmonic analysis, several algorithms have
been proposed [3]–[6], and fast Fourier transform (FFT) is the
most widely used computation algorithm for harmonic analysis.
However, the leakage effect, picket-fence effect, and aliasing ef-
fect make FFT suffer from specific restrictions. Some methods
[6]–[8] have also been provided to improve these drawbacks.
Manuscript received December 22, 2005; revised June 20, 2006. Paper no.
TPWRD-00713-2005.
J. J. Tomić and V. V. Vujičić are with the University of Novi Sad, Novi Sad
21000, Serbia.
M. D. Kušljević is with Termoelektro Enel AD, Belgrade 11060, Serbia.
Color versions of one or more of the figures in this paper are available online
at http://ieeexplore.ieee.org.
Digital Object Identifier 10.1109/TPWRD.2007.893372
0885-8977/$25.00 © 2007 IEEE
Fig. 1. Block diagram of the estimation algorithm.
The FFT does not consider time information. It assumes that
the analyzed signal is stationary and, hence, this technique is
not suitable for studying time-varying harmonics. The derived
results will contain errors if the harmonics vary within the FFT
time window. As an improved method of Fourier transform,
short-time Fourier transform (STFT) is introduced to overcome
this deficiency. However, STFT still inherits the constraints
of uniform time and frequency resolutions, which have to be
traded against each other, thereby restraining the accuracy of
the method. Therefore, the time–frequency analysis using the
wavelet transform (WT) is proposed [4], [5]. Applying the
WT instead of Fourier transform makes it possible to catch a
phenomenon from both sides of the time and frequency.
A Newton-type numerical algorithm that considers the
system frequency also as an unknown signal-model parameter
to be estimated has been proposed [9], [10]. This approach
solves the problem of sensitivity to frequency variations. With
the introduction of power frequency in the vector of unknown
model parameters, the signal model becomes nonlinear, so
strategies of nonlinear estimation are used.
This paper describes a simple algorithm with decoupled
linear models for frequency and amplitude of the fundamental
and higher frequency estimation. The proposed harmonic anal-
ysis method is composed of three different parts: an accurate
frequency estimation algorithm [16], an accurate amplitude
estimation algorithm [17], and a finite-impulse-response (FIR)
adaptive comb (cascade) filter [17], [18].
The block diagram of the algorithm is given in Fig. 1. Four
parts could be noticed. The first part is prefiltering using in the
order of denoising and antialiasing. The other three parts are the
adaptive filtering, the frequency estimation, and the amplitude
estimation of the basic and higher harmonics.
Data from power system voltage signals of the distri-
bution system are collected in the laboratory environment
by the analog-to-digital converter (ADC) card (ADC 16-b,
250-kSamples/s data) and processed in the PC-based harmonic
analyzer. The FIR filters and estimation algorithm developed
and tested using MATLAB are being involved and used in
TOMIĆ et al.: A NEW POWER SYSTEM DIGITAL HARMONIC ANALYZER 773

LabVIEW software. Both the simulation and the application

of the measurement algorithm were made using LabVIEW
software.
The organization of this paper is as follows: mathematical
models and proposed algorithms are described in Section II, the
simulation results are presented in Section III, the configura-
tion of the system is presented in Section IV, and the experi-
mental results are in Section V. Finally, the conclusion is given
in Section VI.

II. MATHEMATICAL MODELS AND PROPOSED ALGORITHMS

A. Filtering of Harmonics with Online Adaptation

Input signal filtering in algorithms for measurement of the
power system electrical parameters is very important. Filters are
used to minimize the noise effect and to eliminate the presence
of all harmonics except the measured one. Since the frequency is Fig. 2. Frequency response of filter subsections and complete filter for the first
not constant, filter parameters have to be online adapted during harmonic.
frequency estimation. The frequency response of the filters must
have nulls at the harmonic frequencies that are expected to be
present in the signal except the measured one which has to have the th harmonic. denotes the maximum integer part
unity gain. If the frequency of the power system is not constant, of the . It is equal to the number of cascades.
coefficients of filter have to be calculated during measurement. In general, the transfer function of the filter for the th har-
An efficient method for online designing of digital filters of si- monic is given as follows:
nusoidal signals is used in this paper [17], [18]. This method
describes closed forms for calculating filter coefficients and re-
(3)
quires modest computations. The algorithm proposed in [16]
was used for frequency estimation, but also another algorithm
could be used successfully. The chosen model is linear and a For filtering of a dc component, filter (4) could be used
linear algorithm for parameter estimation is used. Therefore, the
derived algorithm is very simple. Data window lengths can also
be changed during measurement in dependence on frequency (4)
changes.
In general, the idea is to search the second-order subsec- where , eliminates frequency
tions that eliminate the dc component and all harmonic fre-
quencies except the measured one for which has to have unity
gain. The complete filter can be realized as cascade of all these
subsections.
In the next examples, a sampling frequency (12
The second-order subsection that eliminates the dc compo-
samples per period ) was selected.
nent and frequency and has a unity gain at the frequency of
The frequency responses of subsections for dc and each
the th harmonic is given by the following transfer
higher harmonics and the response of complete filters given by
function:
(3) for , , and (4) for dc are shown in Figs. 2–4,
respectively.
(1) The frequency responses of filters for the first harmonic for
different fundamental frequency are given in Fig. 5.
where , , B. Frequency Estimation
, is sampling frequency, , and
, , and is the fundamental In an electric power system, an increase or decrease in fre-
frequency (frequency of the first harmonic). quency occurs due to disturbances in the power system. Large
The module that eliminates the harmonic and has unity blocks of load are being connected or disconnected, or large
gain at the frequency is shown as follows: sources of generation are going offline. Frequency variations are
much more likely to occur for the loads that are supplied by a
generator isolated from the utility system (islands).
(2)
Several digital methods for measuring the value of the fre-
quency have been proposed for the last 30–40 years. Various
where gain numerical algorithms for power measurements are sensitive to
, is used to adjust gain for frequency variations. Typical examples are algorithms based on
774 IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 22, NO. 2, APRIL 2007

Fig. 5. Frequency responses of the first harmonic filter for different funda-
Fig. 3. Frequency response of filter subsections and complete filter for the third
mental frequencies.
harmonic.

A technique for frequency estimation that provides accurate

estimation in about 25 ms in a wide range of frequency changes
requires modest computations as presented in [15]. An iterative
procedure for frequency estimation and filters adjusting is pre-
sented. The algorithm shows advantages in accuracy and con-
vergence rate in comparison to the previous methods.
A novel recursive method for the measurement of a local
system frequency where three consecutive samples are used to
update the associated weights recursively has been described in
[16]. The proposed technique is suitable for frequency estima-
tion in a wide range of frequency variations.
For real-time use, most of the aforementioned methods have
a tradeoff between accuracy and speed.
The algorithm for frequency estimation from [16] has been
used and will be described shortly.
Let us assume the following observation model of the mea-
sured signal (arbitrary voltage or current), digitized at the mea-
Fig. 4. Frequency response of filter subsections and complete filter for dc.
surement location:

(5)
the FFT, or algorithms based on the least mean squares tech-
nique (LMS) and on the assumption that system frequency is where
known in advance and constant (50 or 60 Hz).
Conventional methods assume that the power system voltage instantaneous signal value;
waveform is purely sinusoidal and, therefore, the time between amplitude;
two zero crossings is an indication of system frequency. Use of radian frequency;
zero crossing detection and calculation of the number of cycles instantaneous time.
that occur in a predetermined time interval [11] is a simple Assuming the signal is uniformly sampled at frequency
and well-known methodology. Discrete Fourier transforma- , the above equations have the following digital form:
tion, least error squares, and Kalman filter are known signal
processing techniques, used for the frequency measurement of
(6)
power system signals [12]–[14]. A nonlinear filter based on the
extended Kalman filter principle to estimate the frequency of
Three consecutive samples are connected with the following
the signal corrupted with white noise has been proposed in [14].
equation [14], [16]:
In this paper, the proposed method deals with the measurement
of the parameters of power system signals which is usually
perturbed with noise and high disturbances. (7)
TOMIĆ et al.: A NEW POWER SYSTEM DIGITAL HARMONIC ANALYZER 775

Fig. 6. Estimations for y(t) = sin(2 ft) where f = 50 Hz for t < 0 s and f = 49.8 Hz for t > 0 s with SNR = 60 dB.

or The system (10) is overdetermined (i.e., the number of equa-

tions is much larger then the number of the variables). There-
fore, the recursive methods as the least mean square method and
(8) maximum likelihood method can be used.
The can be calculated using the following formula:
where

(12)
(9)

Substituting in (8), we get the

that is
following linear equations system:

(10) (13)
where (14)

where is the forgetting factor.

(11) The part (13) must be calculated in each iteration but the part
(14) has to be calculated only in order to form the output signal.
It can be noticed that the algorithm is very simple and suitable
and is the forgetting factor. for implementation.
776 IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 22, NO. 2, APRIL 2007

Fig. 7. Frequency and amplitude estimation for y(t) = sin(2 ft) where f = 50 + sin(10 t).

In the iterative procedure (13), the is assumed to be a con- adaptability. This technique provides accurate estimation with
stant. As has been shown in [16], an adaptation of the forgetting error in the range of 0.002 Hz for SNR 60 dB in about 25 ms
factor can be a convenient way to improve performance of and requires modest computations.
the estimating procedure. We can define a covariance of the es-
timation error as follows: C. Harmonic Amplitudes Estimation
The presence of harmonics and decaying dc can also distort
(15) the basic relation (5) derived from the sinusoid model of the
signal. In the presence of harmonics, the equation will be mod-
Using the form of matrices , , and given in (11), we obtain
ified as
(16)
(19)
Forgetting factor can be calculated as follows:

(17) where
instantaneous signal value;
or harmonic order;
highest frequency component of the signal;
(18) dc value;
amplitude of the th harmonic;
where and are chosen values.
phase of the th harmonic.
The algorithm showed a very high level of robustness as well
as high measurement accuracy over a wide range of frequency We assume that the voltage signal is filtered so that desired
changes. The algorithm convergence provided fast response and harmonic is passed and all others harmonics are canceled. If
TOMIĆ et al.: A NEW POWER SYSTEM DIGITAL HARMONIC ANALYZER 777

Fig. 8. Maximum estimation errors for noisy input signals.

the th harmonic is passed, the filtered output signal has the If we assume that is known from (8), the
following form: can be calculated easily using trigonometric for-
mulas. Equation (24) has the same form as (8) and can be
(20) solved in the same way using the recursive algorithm (12) [i.e.,
Performing a square of from (20), it follows: (13) and (14)].
For dc estimation, the following model can be used:
(21)
(25)
The model (21) is very similar to the mentioned model (6). In
a similar way, we can derive the algorithm for the th harmonic where
amplitude estimation For the and , the relation (21)
can be written as follows:
(22) III. PERFORMANCE EVALUATION THROUGH SIMULATION
(23) The algorithm presented in the previous sections is tested by
Equations (21)–(23) yield the following: means of the input data obtained through the computer simula-
tion. The static, dynamic, and noise tests were performed using
(24) computer-simulated test signals.
The time responses of the algorithm proposed in this paper
where
with the presence of an additive white zero-mean Gaussian noise
are shown in Fig. 6. As shown in Fig. 6, the faster convergence
is achieved as the value decreases. The better accuracy was
778 IEEE TRANSACTIONS ON POWER DELIVERY, VOL. 22, NO. 2, APRIL 2007

obtained as the value increases. A compromise between ac-

curacy and convergence for the presented example should be
achieved for the adaptation of during the estimation proce-
dure. In this way, we have obtained a technique that provides ac-
curate frequency estimation with error in the range of 0.002 Hz
and amplitude estimation with error in the range of 0.03% for
SNR 60 dB, in about 25 ms. It can be also noted that proposed
algorithm has much less estimation error in comparison to the
FFT algorithm.
The ability of the frequency estimation over a wide
range of frequency changes is investigated using the si-
nusoidal test signal with the following time dependence:
. The estimation is computed for the
proposed algorithm with various values of and for the FFT
algorithm and shown in Fig. 7. The good dynamic responses
of the proposed algorithm can be noticed. It can be noted
that the proposed algorithm has much less estimation error in
comparison to the FFT algorithm.
The effect of presence of noise in the signals was studied by
estimating the frequency and harmonic amplitudes of signals Fig. 9. Screenshot of a LabView project to visualize the result.
that contain noise. A sinusoidal 50-Hz input test signal with
the superimposed additive white zero-mean Gaussian noise was
used as input for the test. The random noise was selected to ob-
tain a prescribed value of the signal-to-noise ratio (SNR), de-
fined as , where is the amplitude of
the signal fundamental harmonics and is the noise standard
deviation. Fig. 8. shows the maximum errors observed in fre-
quency and harmonic amplitude estimates when input signals of
30, 50, and 70 Hz having an SNR of 40, 50, 60, 70, and 80 dB
were used. It should be noted that, in practice, the SNR of the
voltage signal obtained from a power system ranges between 50
and 70 dB. At this level of noise, very little errors are expected
with the proposed technique as is depicted in Fig. 8.

IV. SYSTEM CONFIGURATION

New digital instruments measuring PQ indicators and the har-
monic analyzer are being developed.
In the following section of this paper, a brief description of a
system configuration is presented. The FIR filter and estimation
algorithm developed and tested using MATLAB are being using
in LabVIEW software. Both the simulation and the applica-
tion of the measurement algorithm were made using LabVIEW
software (Fig. 9). This approach has two important advantages:
1) flexibility to implement the algorithm and 2) easy to display
and simulate the results against drawbacks as slow environment
and high cost.
The setup consists of PC [750 MHz, 256-MB random-access
memory (RAM)] with an add-on data-acquisition card (in the
motherboard).
Data collected from the power system voltage signals of
the distribution system are collected in the laboratory envi-
ronment by an ADC card. The ADC card has the following
characteristics:
• 16 analog inputs at 16 b, 250 kSamples/s;
• 2 analog outputs at 16 b, 833 kSamples/s (6- full-scale
settling time);
Fig. 10. Laboratory setup.
• programmable input range ( 10, 5, 1, 0.2 V) per
channel;
• 24 TTL/CMOS digital I/O lines (eight hardware timed at
1 MHz);
• Two 32-b, 80-MHz counter/timers;
• digital triggering;
• 6 DMA channels for fast data throughput.
V. EXPERIMENTAL RESULTS
A. Setup
The proposed algorithm has been tested in the laboratory
to establish its feasibility in a real-time environment. The test
signal has been derived from a function generator.
Fig. 10 shows the outline of the setup. The signal is derived
from the signal generator though an unshielded ribbon cable.
TOMIĆ et al.: A NEW POWER SYSTEM DIGITAL HARMONIC ANALYZER
Fig. 11. Frequency estimation from voltage data for (a) f = 800 Hz. (b) f = 1600 Hz.
Fig. 12. First harmonic amplitude estimation from voltage data for (a)
Fig. 13. Higher harmonic amplitude estimations from voltage data for (a)
The measurement is deliberately made crude by inserting mul-
tiple loops in the connecting wire to catch extraneous noise [14].
Options are also made available to increase the noise in the sam-
pling process. The frequency of the signal is first accurately
measured and the FFT analysis is performed by a signal spectral
analyzer.
B. Results
Frequency and harmonic amplitudes are estimated from the
function generator at a sampling rate of 800 Hz and 1600 Hz.
779
f = 800 Hz. (b) f = 1600 Hz.
f = 800 Hz. (b) f = 1600 Hz.
Figs. 11–13 show the performance of the algorithm at a typical
situation where the voltage signals are distorted. The peak-to-
peak variation in the frequency estimation is 0.004 Hz and the
peak-to-peak variation in the amplitudes estimation is 0.06%
in the steady state. Better accuracy was achieved at a sampling
frequency of 1600 Hz. The presence of the third harmonic in the
range of 0.2% should be noted. Similar accuracy was obtained
in the whole range of frequency variation from 40 to 60 Hz. It
should be noted that the estimation process has started at 30 ms
(after buffering input data necessary for FIR processing).
780
VI. CONCLUSION
In this paper, new digital instruments measuring power-
quality indicators and harmonic analyzers are presented. The
design and implementation of a novel, very accurate recursive
method for the measurement of frequency and harmonics
amplitudes of a power system have been used. The proposed
technique is suitable for measuring of these parameters in a
wide range of frequency variations. This method is easy to
implement and very flexible. The development, the simulation,
and the application of the measurement algorithm were made
using MATLAB/LabVIEW software because of the flexibility
to implement the algorithm and that it is easy to display.
The simulations and experimental results have shown that the
proposed technique provides accurate estimation. This method
really meets the need of offline applications. Even more, by
using parallel computation algorithms, this method should meet
the need of online applications and should be more practical.
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Josif J. Tomić was born in Kovin, Serbia, in 1962.
He received the B.Sc. and M.Sc. degrees in electrical
measurement from the University of Novi Sad, Novi
Sad, Serbia, in 1990 and 2004, respectively.
Currently, he is a Research Assistant of measure-
ment techniques at the University of Novi Sad. His re-
search interests are in the area of microprocessor and
DSP-based measurements and control applications.
Miodrag D. Kušljević was born in Šule, Pljevlja,
Montenegro, in 1962. He received the B.Sc. degree
from the Defense Technical Engineering Fac-
ulty—Missile Systems Department, Zagreb, Croatia,
in 1986 and the M.Sc. degree from the Faculty of
Electrical Engineering and Computing, Zagreb, in
1990.
From 1990 to 1991, he was a Research As-
sistant with the Defense Technical Engineering
Faculty—Missile Systems Department, Zagreb. In
1992, he joined Termoelektro Enel AD, Belgrade,
Serbia, where he is currently Technical Manager. His areas of research interests
are algorithms for digital signal processing and algorithms for measurement,
control, and protection in power systems.
Vladimir V. Vujičić (M’95) was born in Novi Sad,
Serbia, in 1947. He received the B.Sc. degree from
the University of Belgrade, Belgrade, Serbia, in 1974,
and the M.Sc. and Ph.D. degrees in electrical engi-
neering from the University of Novi Sad, Novi Sad,
Serbia, in 1983 and 1992, respectively.
Currently, he is a Full Professor and Head of the
Chair for Measurement Techniques with the Univer-
sity of Novi Sad, where he has been since 1975. His
research interests are mainly various analog-to-dig-
ital conversion techniques and corresponding fast
processing techniques in the field of electrical measurements.