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3860 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 60, NO.

12, DECEMBER 2011


Simultaneous Reactive-Power and Frequency
Estimations Using Simple Recursive WLS
Algorithm and Adaptive Filtering
Miodrag D. Kuljevi c and Predrag D. Poljak
AbstractA new simple approach to the design of digital algo-
rithm for simultaneous reactive-power and frequency estimations
of local system is presented. The algorithm is derived using the
weighted-least-square method. During the algorithm derivation,
a pure sinusoidal voltage model was assumed. Cascade nite-
impulse-response (FIR) comb digital lters are used to minimize
the noise effect and to eliminate the presence of harmonics effect.
The most important point of this paper is the mathematical model
that transforms the problem of estimation into an overdetermined
set of linear equations. The investigation was simplied because
the total similarity to the state of the problem of the active-power
and frequency estimations was noticed. The only difference is the
adaptive phase shifter applied to the voltage signal. In addition,
coefcient-sensitivity problems of the large-order FIR comb cas-
cade structure are overridden by using a multirate (decimation)
digital signal processing technique. Even more, by using anti-
aliasing lters, the parameter estimation accuracy is improved.
The effectiveness of the proposed techniques is demonstrated by
both simulation and experimental results. The algorithm shows
a very high level of robustness, as well as high measurement
accuracy over a wide range of frequency changes.
Index TermsCascaded-integrator-comb (CIC) lter, digital
phase shifter, downsampling, nite-impulse-response (FIR) lter,
frequency estimation, reactive-power estimation, recursive algo-
rithm, weighted least square (WLS).
I. INTRODUCTION
R
EACTIVE-POWER measurement is of essential signif-
icance for management, design, and planning of elec-
tropower systems. The energy transfer from a utility to its
customers is conducted at the system frequency. In system
management, the rst-harmonic active- and reactive-energy
generation, ow, and absorption is of interest [1]. Energy
relations at this frequency are of primary importance; the rest
of the frequencies can be treated as noise or pollution. If such
an approach is taken, then the rst-harmonic reactive power
appears to be the most appropriate indicator.
Manuscript received December 16, 2008; revised March 1, 2010; accepted
March 11, 2010. Date of publication June 23, 2011; date of current version
November 9, 2011.
M. D. Kuljevi c is with the Termoelektro Enel AD, 11060 Belgrade, Serbia
(e-mail: Miodrag.Kusljevic@te-enel.co.rs).
P. D. Poljak is with the School of Electrical Engineering and Computer
Science of Applied Studies, 11000 Belgrade, Serbia (e-mail: Predrag.Poljak@
viser.edu.rs).
Digital Object Identier 10.1109/TIM.2011.2144690
In addition, the amount and the complexity of household
electrical equipment have tremendously increased over last
decades. Electronic ballast lighting, computer monitors, and
air conditioners are disadvantages for energy distributors that
bill energy based only on the active power. With the applica-
tion of nonlinear loads to power lines, the active energy no
longer represents the total energy delivered. As response to
improve billing, the measuring of the reactive power is gaining
interest [2].
During the last fewdecades, the reactive-power measurement
problem has been very seriously studied [1][6]. The problem
of power dening in systems of alternating current is still open.
Measurements in nonsinusoidal conditions are in the center of
investigation, and a lot of efforts have been submitted in order
to dene the term reactive power in new way. In fact, in the
presence of harmonics in voltages and/or currents, the conven-
tional denition of reactive power is not meaningful anymore.
As a consequence, the current standards refer to sinusoidal
cases, and they do not give a reference denition for reactive
energy (or power) in nonsinusoidal conditions. Similarly, they
do not give any specic accuracy requirements and related test
conditions in the presence of harmonic distortion. The only
standard on this topic is the IEEE Standard 14592010 [7],
which does not introduce a specic denition of reactive power
under nonsinusoidal conditions. In this standard, the concept
is based on splitting power into its fundamental and remaining
terms. This approach of separating into fundamental and har-
monic parts can be observed for the most important quantities,
being normalized as indicators of power quality. Thus, the
traditional billing quantities are emphasized, being dened as
the fundamental active, reactive, and apparent powers and the
related power factor [8].
In this paper, the simple algorithm for simultaneous esti-
mation of the fundamental reactive power and frequency is
prescribed. It is based on the algorithm for the simultaneous
estimation of the active power and frequency, which is given
in [9]. The only difference is the phase shifter applied to
the voltage signal. Since the frequency of the signal is being
estimated during measurement, it is not necessary to perform
frequency-insensitive phase shifting. It is easier to adjust the
phase shifter parameter in accordance to the actual frequency.
In addition, implementation problems connected with the coef-
cient sensitivity of large-order nite-impulse-response (FIR)
comb lters are overridden by using a multirate (decimation)
digital signal processing technique. Even more, by using these
0018-9456/$26.00 2011 IEEE
KULJEVI

C AND POLJAK: SIMULTANEOUS REACTIVE-POWER AND FREQUENCY ESTIMATIONS 3861
Fig. 1. Block diagram of the simultaneous estimation of the frequency and the reactive power.
antialiasing lters, the accuracy of the reactive-power and fre-
quency estimations is improved. Simulated and experimental
results in the presence of different noise levels and different
values of the reduction and decimation factors, and different
phase and amplitude values of the harmonics showed that the
implemented method allows better results compared with the
fast-Fourier-transform (FFT)-based method.
II. REACTIVE-POWER DEFINITION
The reactive power Q for sinusoidal voltage and current, i.e.,
v(t) =V sin (t +
v
) (1)
i(t) =I sin (t +
i
) (2)
is dened by IEEE norms [10] as the product of the voltage and
current root-mean-square values and the sine of the phase angle
between them, i.e.,
Q = V
RMS
I
RMS
sin (
v

i
) =
V I
2
sin () (3)
where the + and signs point to the load and generator
powers, respectively. For the case of multiphase sinusoidal
systems, there is no unique denition for the total reactive
power. According to the IEEE [10], the total reactive power in a
multiphase system is an algebraic sum of single-phase powers.
The active power P is a mean value in the time of the instant
power p(t) = v(t)i(t). For sinusoidal signals v(t) and i(t), it
follows that
p(t) =V sin (t +
v
)I sin (t +
i
)
=
V I
2
cos (
v

i
)
V I
2
cos (2t +
v
+
i
) (4)
P =
1
T
0
T
0
_
0
p(t)dt =
V I
2
cos (
v

i
) (5)
where T
0
= 2/. From (3)(5), it is shown that the measure-
ment of the reactive power of sinusoidal signals v(t) and i(t)
could be reduced to the measurement of the active power of
signals v

(t) and i

(t), i.e.,
v

(t) =V sin (t +

v
) (6)
i

(t) =I sin (t +

i
) (7)
where the phases of the voltage and the current are changed
such that an additional phase shift of /2 is introduced, i.e.,

i
=
v

i
(/2).
According to (4) and (5), the active power of signals v

(t)
and i

(t) amounts to
P

=
V I
2
cos (

i
) (8)
which is equal to the reactive power Q of signals v(t) and i(t)
according to denition (3). This result enables the measurement
of the fundamental reactive energy by conventional meters of
the active energy in such a way that an additional phase shift of
/2 is added between the voltage and the current prior to the
introduction of the signals into the active-energy meter.
III. PROPOSED ALGORITHM
The general block diagram of the proposed algorithm is
shown in Fig. 1. As shown, the proposed technique employs
two decoupled parts. The rst part consists of the cascade
of antialiasing cascaded-integrator-comb (CIC) and adaptive-
bandpass FIR lters to lter out the fundamental components
of voltage and current signals. The second part is used to esti-
mate the fundamental reactive-power component of the power
signals and their frequency.
A. Simultaneous Frequency and Reactive-Power Estimations
Let us suppose that the discrete voltage and current signals
v

(t) and i

(t), which are previously ltered out and dened


by (6) and (7), are uniformly sampled at frequency
s
= 2/T
(
S
is an angular sampling frequency). The following discrete
signals are obtained:
v

n
=V sin (nT +

v
) (9)
i

n
=I sin (nT +

i
) . (10)
By multiplying voltage and current signals dened in (9) and
(10), respectively, it follows
p

n
= v

n
i

n
= QS cos (2nT +

) (11)
where
S =V I/2
Q =S cos

= S sin

i
=

2

v
+

i
.
The three consecutive samples of signal p

n
, which were
sampled at instants n 2, n 1, and n, are connected with the
3862 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 60, NO. 12, DECEMBER 2011
following equation, as well as the samples of the active-power
signal p
n
in [9]:
a
T
n
x
n
= b
n
(12)
where
x
n
=
_
x
1
x
2
_
n
=
_
x
1,n
x
2,n
_
=
_
Q(1 cos (2T))
cos (2T)
_
a
n
=
_
2
2p

n1
_
b
n
=p

n
+p

n2
.
By substituting n = 1, 2, . . . , k (k > 2) in (12), we get the
following linear equation system:
Ax = b (13)
where
b =
_

_
W
k1
b
1
W
k2
b
2
. . .
b
k
_

_ A =
_

_
W
k1
a
T
1
W
k2
a
T
2
. . .
a
T
k
_

_. (14)
The weighting (forgetting) factor (0 < W < 1) is involved
to ensure higher impact on the newer samples.
The solution to (13), at instant k, based on the least-square
error method is given by
x
k
=
_
A
T
k
A
k
_
1
A
T
k
b
k
. (15)
By solving (15), the components of vector x
k
can be calcu-
lated using (16) and (17), shown at the bottom of this page.
To implement (16) and (17) in a recursive fashion, fth
accumulators are introduced, i.e.,
Acc_1
k
=W
2
Acc_1
k1
+ 1
Acc_p

k
=W
2
Acc_p

k1
+p

k
Acc_p
2
k
=W
2
Acc_p
2
k1
+p
2
k
Acc_b
k
=W
2
Acc_b
k1
+b
k
Acc_p

b
k
=W
2
Acc_p

b
k1
+p

k
b
k
(18)
x
1,k
=
1
2
Acc_p

k
Acc_b
k
+ Acc_1
k
Acc_p

b
k
Acc_1
k
Acc_p
2
k
(Acc_p

k
)
2
(19)
x
2,k
=
1
2
Acc_p
2
k
Acc_b
k
Acc_p

k
Acc_p

b
k
Acc_1
k
Acc_p
2
k
(Acc_p

k
)
2
. (20)
The results are weighted-least-mean-square estimations of
the quantities at instant k, which are based on the measurement
until instant k.
Having obtained vector x
k
, the frequency and the reactive
power are then derived by
f
k
= arccos (x
2,k
)/(4RT) (21)
Q
k
=x
1,k
/(1 x
2,k
). (22)
Equations (21) and (22) have to be only calculated in order to
form output signals. It can be noticed that the algorithm is very
simple and suitable for implementation.
The forgetting factor ensures the higher impact of new data
and the weakening of older data. The algorithm has the possi-
bility of trading speed for accuracy by specifying the forgetting
factor W
2
. The estimation can be either fast and less accurate
(for smaller W
2
) or slow and more accurate (for higher W
2
).
For W
2
= 0, the estimation is the fastest and is only based
on the last three signal samples, but in this case, the accuracy
is least. As have been shown in [9] and [11], a compromise
between accuracy and convergence for the presented example
should be achieved for the adaptation of W
2
during the estima-
tion procedure.
Let us dene a value of the residual error at the kth iteration,
which can be calculated as
e
k
= b
k
A
k
x
k
. (23)
We can dene a covariance of the estimation error as follows:
R
k
= e
T
k
e
k
= (b
k
A
k
x
k
)
T
(b
k
A
k
x
k
). (24)
Using the form of matrices b, A, and x that is given in (13)
and (14), we obtain
R
k
= W
2
k1
R
k1
+
_
b
k
a
T
k
x
k
_
2
. (25)
The forgetting factor W
2
can be calculated as follows
[9], [11]:
W
2
k
= W
min
+
W
max
W
min
1 +|R
k
/R
0
|
p
(26)
or [5]
W
2
k
= W
min
+ (W
max
W
min
)e
|R
k
/R
0
|
p
(27)
where W
min
, W
max
, R
0
, and p are the chosen values. These
parameters have to be heuristically selected, depending on the
x
1,k
=
1
2

i=0
W
2(ki)
p

i
k

i=0
W
2(ki)
b
i
+
k

i=0
W
2(ki)
k

i=0
W
2(ki)
p

i
b
i
k

i=0
W
2(ki)
k

i=0
W
2(ki)
p
2
i

_
k

i=0
W
2(ki)
p

i
_
2
(16)
x
2,k
=
1
2
k

i=0
W
2(ki)
p
2
i
k

i=0
W
2(ki)
b
i

i=0
W
2(ki)
p

i
k

i=0
W
2(ki)
p

i
b
i
k

i=0
W
2(ki)
k

i=0
W
2(ki)
p
2
i

_
k

i=0
W
2(ki)
p

i
_
2
(17)
KULJEVI

C AND POLJAK: SIMULTANEOUS REACTIVE-POWER AND FREQUENCY ESTIMATIONS 3863
signal parameters dynamic. W
min
and W
max
bound an interval
in which W
2
can take place during tuning (W
min
W
2

W
max
). Smaller W
min
allows faster but, at the same time, more
oscillatory estimation. Higher W
max
allows more accurate but,
at the same time, slower estimation. Smaller R
0
and higher p
provide faster but more oscillatory estimation.
Bad data corresponding to poor frequency and reactive-
power estimates is identied and ignored by median lters.
In contrast, the output of a linear lter is calculated from a
linear combination of data samples, including bad data. Another
advantage in median ltering is that there is no need to specify
the cutoff frequency [12].
B. Adaptive Filtering of Power System Signals
Both the troubling synchronization to the signal fundamental
frequency and the use of restricting window functions could
be avoided using adaptive digital FIR lters for sinusoidal
signals [11]. The FIR digital lters are also used to process
input voltage and current signals to minimize the noise effect
and are not affected by the presence of harmonics. It means
that the frequency response of the lters must have nulls at
the harmonic frequencies that are expected to be present in
the signal. Thus, the estimation of the frequency requires the
design of new lters at each iteration. The method proposed
in [11] uses closed forms for calculating lter coefcients. The
complete lter can be realized as a cascade of the second-order
subsections that eliminate direct-current (dc) component and all
harmonic frequencies, except the measured one for which has
to have unity gain.
The second-order subsection that eliminates the dc compo-
nent and frequency
s
/2 = /T and has unity gain at the
fundamental frequency is given by the following z-domain
transfer function [11]:
H
0
(z) =
1 z
2
|1 z
2
1
|
(28)
where |1 z
2
1
| = 2 sin (T) [11]. The subsection that rejects
harmonic
m
= m and has unity gain at the fundamental
frequency is shown as follows:
H
m
(z) =
1 2 cos (
m
T)z
1
+z
2

1 2 cos (
m
T)z
1
1
+z
2
1

, m = 2, 3, . . . , M
(29)
where |12 cos (
m
T)z
1
1
+z
2
1
| =2| cos (T)cos (mT)|
[11], where M denotes the maximum integer part of
S
/(2).
It is equal to the number of the subsections in cascade.
The transfer function of the complete lter is given as
follows:
H
1
(z) = H
0
(z)
M

m=2
H
m
(z). (30)
It is shown that a calculation of the lter coefcients can be
easily implemented if the fundamental frequency is known.
Particularly convenient are the algorithms for the frequency
measurement where either cos (T) or cos (2T) is directly
Fig. 2. Frequency responses of the rst harmonic lter for different funda-
mental frequencies and the sampling frequency of fs = 800 Hz.
estimated, such is that in the previously described algorithm. In
this case, cos (mT) can be easily calculated using trigonomet-
ric formulas.
The frequency responses of the lters for the fundamental
harmonic for different fundamental frequencies and a sam-
pling frequency of f
s
= 800 Hz (16 samples per period; T
0
=
1/50 = 0.02 s) are given in Fig. 2.
C. Adaptive Phase Shifter
The measurement of the reactive power under sinusoidal
conditions, according to the IEEE denition [10], requires the
introduction of a phase shift of /2 exactly between the voltage
and current signals. This phase shift should be equal to /2
exactly and should be independent on the frequency of the
input signals. [13] describes a digital realization of a circuit
for an additional phase shift of /2 between input signals,
with a constant amplication nondependent on the frequency.
A versatile method for high-precision frequency-insensitive
quadrature phase shifting is described in [14].
A phase shift corrector with the following transfer function
can perform a phase shift of /2 on the fundamental fre-
quency [6], [11]:
H
PHSH
_
z,

2
_
=
cos (T) +z
1
sin (T)
. (31)
The cascade connection of the phase shift corrector, i.e.,
(31) and lter (30), gives a new lter that, together with lter
(30), forms an orthogonal pair for the fundamental frequency
[11], i.e.,
H
2
(z) = H
PHSH
_
z,

2
_
H
0
(z)
M

m=2
H
m
(z). (32)
D. Decimation Filtering
In the case of signals with a low signal-to-noise ratio (SNR),
the accuracy of the algorithm can be further improved by
the oversampling of the input signal combined with high-
order antialiasing lters and decimation. Each doubling of the
sampling frequency will lower the in-band noise by 3 dB and
will increase the resolution of the measurement by 1/2 bit [15].
In addition, implementation problems regarding coefcient
sensitivity for large-order FIR cascade comb lters are present.
On the other hand, the low-pass lter (LPF) cannot eliminate
all harmonic components unless a high-order LPF is selected.
3864 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 60, NO. 12, DECEMBER 2011
Fig. 3. Frequency responses of the CIC lters for different numbers of stages
in cascade and fs = 6400 Hz and D = 8.
In the case of choosing a high-order LPF, the transient response
of the frequency estimation method decreases, which is not
desired. Fortunately, good properties could be achieved by the
cascade connection of the antialiasing LPF with the cutoff
frequency that is high enough to ensure fast response (up to
816 harmonics) and the reduced-order FIR comb lter that is
applied on the decimated signal. This way, both coefcient sen-
sitivity problems will be avoided, and the parameter estimation
accuracy will be improved.
The CIC lters are computationally efcient implementa-
tions of narrow-band LPF and are often embedded in hardware
implementations of decimation and interpolation in modern
communications systems. The CIC lters were introduced to
the signal processing community about three decades ago,
but their application possibilities have grown in the recent
years. Improvements in chip technology, the increased use of
polyphase ltering techniques, advances in deltasigma con-
verter implementations, and the signicant growth in wireless
communications have all spurred much interest in CIC lters
[15], [16].
The CIC lters originate from the notion of a recursive
running-sum lter, which is itself an efcient form of a
nonrecursive moving averager. A z-domain H
CIC
(z) transfer
function of the CIC lter for the D-point moving-average
process is
H
CIC
(z) =
1
D
1 z
D
1 z
1
. (33)
The most common method to improve the CIC lter an-
tialiasing and image-reject attenuation is by increasing order
S of the CIC lter using multiple stages. For S CIC stages
in cascade, the overall frequency magnitude response is the
product of their individual responses, or [16]

H
CIC,Sthorder
(e
j2f
)

= |sin (fD)/ sin (f)|


S
. (34)
Fig. 3 shows the frequency magnitude responses of the rst-,
second- and third-order (S = 1, 2, 3) CIC decimating lters.
Notice the increased attenuation of the third-order CIC lter
compared with the rst-order CIC lter in Fig. 3.
The CIC lters create errors in magnitude and phase spec-
trum responses, and they have drooping passband gains and
wide transition regions. In order to eliminate the side effect of
the proposed CIC lter, correction factors were provided.
Fig. 4. Estimations for y(t) = sin (2ft), where f = 50 Hz for t < 0 s and
f = 49.8 Hz for t > 0 s, with SNR = 60 dB.
IV. PERFORMANCE EVALUATION THROUGH SIMULATION
The proposed algorithm was evaluated using simulated volt-
age and current signals with dc components, high harmonic
content, and superimposed noise, which were sampled at
6.4 kHz (the standard EN 50160 [17] considers the rst
40 harmonic components, and hence, a sampling frequency of
greater than 4 kS/s must be used). The algorithm parameters
used were W
min
= 0.5, W
max
= 0.999, R
0
= 10
6
, and p =
2. For the antialiasing lter H
CIC
(z), D = 8 and S = 3 were
chosen. The reduction factor R = 4 was selected. A oating-
point arithmetic was applied.
An input frequency-modulated sinusoidal test signal (step
frequency change from 50 to 49.8 Hz at t = 0 s) is processed.
White noise with an SNR of 60 dB was added to sinusoidal
signals. The SNR is dened as SNR = 20 log(A/

2), where
A is the magnitude of the signal fundamental harmonics and
is the noise standard deviation. The results obtained conrm the
good dynamic response of the algorithm for the frequency step
change, as well as the accuracy of parameter estimation. We
have a technique that provides accurate estimates for SNR =
60 dB with the frequency estimation error in the range of
0.002 Hz and the reactive-power estimation errors in the range
of 0.03% with respect to the fundamental apparent power, in
about 25 ms (see Fig. 4). A comparison is made between the
proposed and FFT-based techniques. Note that sliding frames
of the incoming samples for computing the FFT are collected.
A total of 128 input samples (one fundamental period) are used
to generate the magnitudes of 128 complex FFT values. It can
be noted that the proposed algorithm has much less estimation
error compared with the FFT-based algorithm.
The ability of the frequency and reactive-power estimations
over a wide range of frequency changes is investigated using
a sinusoidal test signal with the time dependence of f(t) =
50 + 0.5 sin (10t). This situation exceeds the real power-line
frequency deviation and rate of change. The estimation is
shown in Fig. 5. The good dynamic responses and the high
KULJEVI

C AND POLJAK: SIMULTANEOUS REACTIVE-POWER AND FREQUENCY ESTIMATIONS 3865
Fig. 5. Frequency and reactive-power estimations for y(t) = sin (2ft),
where f(t) = 50 + 0.5 sin (10t) and SNR = 60 dB.
Fig. 6. Maximum estimation errors for noisy input signals.
accuracy of measurement can be noticed. The arguments and
the conclusions of the preceding frequency-step-change case
are conrmed again in this experiment.
The effect of the noise presence in the signals was studied by
estimating the frequency and the reactive power of signals that
contain noise. A sinusoidal 50-Hz input test signal with the su-
perimposed additive white zero-mean Gaussian noise was used
as input for the test. The random noise was selected to obtain
a prescribed value of the SNR. The maximum reactive-power
estimation errors in terms of the SNR in the range 4070 dB,
when the decimation factors were 2, 4, and 8 (the sampling
frequencies were 3200, 6400, and 12800 Hz, respectively) so
that the reduced sampling frequency was always 1600 Hz, are
shown in Fig. 6. As shown, the uncertainty of the reactive-
power estimation smoothly increases with a decrease in the
Fig. 7. Maximum estimation errors for the harmonic presence.
SNR, but very little error is expected for the noise level that
is usually present in practice. The maximum error for R = 8,
as shown in Fig. 6, is about 0.28% with a low SNR of 40 dB.
The estimation method with R = 2 gave a maximumerror of up
to about 0.50% for waveforms in a similar noise environment.
This comparison indicates better performance of the proposed
method for a higher decimation factor.
To demonstrate the effectiveness of the proposed technique
in estimating the frequency and the reactive power in the
presence of harmonics, an input signal having a fundamental
frequency of 50 Hz, a third-harmonic component in the range
from 0% to 20%, and a fth-harmonic component equal to
half of the third component have been used. Fig. 7 shows
the maximum errors for the proposed technique when the
decimation factors of 2, 4, and 8 (the sampling frequencies of
3200, 6400, and 12 800 Hz, respectively) having 0%, 5%, 10%,
15%, and 20% third harmonic were used. The errors of estimate
values are nearly zero despite the presence of the harmonic
component. It can be noted that the proposed algorithm has
very small reactive-power estimation errors (30 ppm). The
errors of estimate values are nearly zero despite the presence
of harmonic components. As shown, the performance of the
proposed method does not depend on the number of the present
phase signals and the level of the harmonic disturbances.
V. EXPERIMENTAL RESULTS
The proposed algorithm has been tested by means of the
rapid prototyping system that had been developed. The com-
puter controls the experimental procedure using the LabView
software package. The environment of LabView is very friendly
for developing programs using graphical programming lan-
guage. LabView offers analysis and mathematical routines that
natively work together with data acquisition functions and
display capabilities so that they can be easily built into
any application. In addition, LabView offers analysis routines
3866 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 60, NO. 12, DECEMBER 2011
TABLE I
TEST CONDITIONS. (a) FUNDAMENTAL VOLTAGE: V
1
= 230 V; PHASE = 0

VOLTAGE HARMONIC CONTENT.


(b) FUNDAMENTAL CURRENT: I
1
= 5 A; PHASE = 47.11

CURRENT HARMONIC CONTENT [6]


for point-by-point execution; these routines are specically
designed to meet the needs of inline analysis in real-time
applications.
The implemented digital data acquisition system uses a stan-
dard data acquisition card for the Peripheral Component Inter-
connect (PCI) bus, i.e., PCI-6014 from National Instruments.
Its main characteristics are 16 analog inputs, 16-bit resolution,
200-kS/s sampling rate, and high-speed direct-memory-access
channel.
A known harmonic content in voltages and currents has
been generated, and the related power quantities have been
calculated. The generated signals have been converted by a
16-bit digital-to-analog converter. Then, they have been sam-
pled by the 16-bit analog-to-digital converter of the instrument
with the sampling frequency fs = 6.4 kHz. No transducers have
been used to evaluate the uncertainties only related to the
instrument, because the main contribution to the uncertainty
of the instrument is due to the transducers [6]. In addition,
the main point of this paper is the measurement algorithm.
Measurements have been performed in various cases, where
the fundamental frequency of the generated signals has been
changed in the range of 3070 Hz. The setup is designed to
create different test signals with various harmonic contents,
starting from pure sinusoidal test signals and ending with
harmonic-rich square signals. The testing of the instrument
was also performed by applying the fundamental frequency
plus higher harmonic components in accordance with Table I,
which was taken over from [6]. The amplitude and the phase
displacement of the voltage and the current are reported for the
TABLE II
MAXIMUM MEASUREMENT ERRORS WITHOUT TRANSDUCERS
ve equally spaced points from the sinusoidal condition to the
maximum harmonic content.
In Table II, maximum errors for the measured power quanti-
ties and for each considered testing signals are reported. The
results show that the achieved uncertainty is very low and
rather below standard limits of 1% [18] and in accordance with
those obtained by simulations. It should be noted that these
uncertainties are minor, in comparison with the ones owned by
transducers [6].
VI. CONCLUSION
The design and the implementation of a novel recursive
method for the measurement of the reactive power and the
local system frequency have been described in this paper. The
KULJEVI

C AND POLJAK: SIMULTANEOUS REACTIVE-POWER AND FREQUENCY ESTIMATIONS 3867
chosen model is linear, and the linear-algorithm weighted least
square for parameter estimation has been used. Therefore, the
derived algorithm is very simple and requires modest resources
for implementation. The proposed technique is suitable for
measurement for a wide range of frequency changes. It is useful
in designing microprocessor-based devices that need accurate
and fast measuring of power system parameters.
The results of the computer simulation have been quoted
to conrm the validity and the performance of the proposed
algorithm. The obtained results have conrmed a good dynamic
response of the algorithm, as well as the accuracy of reactive-
power and frequency estimations. A measurement algorithm
has been shown to be accurate within 0.002 Hz for the fre-
quency and 0.03% for the reactive power (with an SNR of
60 dB). The algorithm calculates a new estimate of quantities at
every sampling interval irrespective of the phase of the input
signal. The algorithm is capable of accurately tracking fre-
quency under dynamic power system conditions and provides
high-speed measurement exhibiting a delay of only 25 ms. This
algorithm has shown immunity to the presence of harmonics.
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Miodrag D. Kuljevi c was born in ule, Pljevlja,
Montenegro, in 1962. He received the B.Sc. degree
from the Defense Technical Engineering Faculty,
Zagreb, Croatia, in 1986, the M.Sc. degree from the
University of Zagreb, Zagreb, in 1990, and the Ph.D.
degree from the University of Novi Sad, Novi Sad,
Serbia, in 2006.
From 1990 to 1991, he was a Research Assis-
tant with the Defense Technical Engineering Fac-
ulty, Missile Systems Department. Since 1992, he
has been with the Termoelektro Enel AD, Belgrade,
Serbia, where he is currently the Technical Manager. His areas of interest
are algorithms for digital signal processing and algorithms for measurement,
automatic control, and protection in power systems.
Predrag D. Poljak was born in Sisak, Croatia, in
1976. He received both B.Sc. and M.Sc. in electrical
engineering from the University of Novi Sad, Novi
Sad, Serbia, in 2008.
He is currently an Expert Associate with the De-
partment of Automatic and Vehicle Control Systems,
School of Electrical Engineering and Computer Sci-
ence of Applied Studies, Belgrade, Serbia. His re-
search interests include digital signal processing,
design and analysis of the virtual instrumentation,
and automotive and power system applications.

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