56 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO.
1, JANUARY 2016
Power System Frequency Estimation Method
in the Presence of Harmonics
Hui Xue, Member, IEEE, Maohai Wang, Rengang Yang, and Yan Zhang
Abstract— A frequency estimation method with good resistance
to harmonics is proposed in this paper. In the proposed method,
a new sequence is constructed using the summation of the samples
in the original sampling sequence. The amplitude ratio of the
new sequence and the original sequence is used to calculate
the fundamental frequency deviation. The effects of negative
fundamental component, harmonics, and interharmonics are
studied. A preprocessing procedure is also proposed to eliminate
the effects of the interharmonics with frequency near the funda-
mental frequency. Theoretical analysis and simulations validate
that the proposed method has good immunity to harmonics. Due
to its low required processing effort and high precision, this
method is a good candidate for online frequency estimation in
the presence of harmonics.
Index Terms— Discrete Fourier transform (DFT), frequency
domain analysis, frequency estimation, harmonic distortion,
sequences.
I. I NTRODUCTION
P OWER system frequency is an important operating
indicator for the stability, efficiency, and safety of a
power system. In particular, power system frequency is the
required quantity for the synchronization between power grid
and distributed generation systems. Therefore, for the reliable
operation of systems, it is essential to estimate the power
system frequency quickly and accurately.
The wide uses of power electronics in distributed generation
systems have allowed flexible control of the power transforms.
However, those devices are generators of harmonics, which are
corrupting the purity of the 60-/50-Hz sine waves that should,
in theory, be the only frequency component in the power
systems. In addition, many industrial customers are creating
harmonics using power electronics equipment, arc furnaces,
and so on [1]. Therefore, fast and accurate frequency estima-
tion in the presence of harmonics is an essential requisite to
the control and protection of distributed generation systems.
Various techniques have been developed to estimate power
system frequency in the presence of harmonics. Popular
Manuscript received April 15, 2015; revised August 6, 2015; accepted
August 9, 2015. Date of publication September 25, 2015; date of current
version December 7, 2015. This work was supported in part by the National
Natural Science Foundation of China under Grant 51207160 and Grant
51477173 and in part by the Program for New Century Excellent Talents under
Grant NCET-13-0565. The Associate Editor coordinating the review process
was Dr. Edoardo Fiorucci.
H. Xue, R. Yang, and Y. Zhang are with the College of Information and
Electrical Engineering, China Agricultural University, Beijing 100083, China
(e-mail: xuehui@cau.edu.cn).
M. Wang is with North China Grid Company, Ltd., Beijing 100053, China
(e-mail: wmhmw@tom.com).
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/TIM.2015.2477157
0018-9456 © 2015 IEEE. Personal use is permitted, but republication/redistribution requires IEEE permission.
See http://www.ieee.org/publications_standards/publications/rights/index.html for more information.
methods include the discrete Fourier transform (DFT)-based
method [1]–[5], phase-locked loop (PLL) method [6]–[9],
adaptive notch filter method [10]–[12], least mean square
method [13]–[15], Kalman filter method [16], Taylor
method [17], zero crossing method [1], Prony method [18],
Newton method [19], demodulation method [20], and so on.
Among these methods, the DFT-based method [1]–[5] is
widely used due to its easy implementation and high precision.
The DFT-based method assumes the fundamental frequency
as f 0 and calculates the phasors of two consecutive data
windows using DFT, then the phase angle difference of the
two phasors is used to calculate the frequency deviation
value  f between the actual frequency and the assumed
frequency f 0 , and finally the actual frequency can be estimated
as f 0 + f . However, inherent errors caused by harmonics still
exist in the DFT-based method. In this paper, an accurate fre-
quency estimation method with better immunity to harmonics
is proposed.
The proposed method also assumes the fundamental
frequency as f 0 , then a new sequence is constructed using the
summation of the samples of the original sampling sequence,
and the amplitude ratio of the new sequence and the original
sequence is used to calculate the frequency deviation value.
The effects of negative fundamental component, harmon-
ics, and interharmonics are studied. To eliminate the effects
of the interharmonics with frequency near the fundamental
frequency, a preprocessing procedure is also proposed. The-
oretical analysis and experiments validate that the proposed
method has good immunity to harmonics. Due to its good
resistance to harmonics and low required processing effort,
the proposed method is a good candidate for online frequency
estimation under harmonic-polluted situations.
The organization of this paper is as follows. The pro-
posed method is introduced in Section II. The effects of
interharmonics and the preprocessing procedure are presented
in Section III. Simulations are presented in Section IV. Tests
with actual signals are presented in Section V, and the
conclusion is given in Section VI.
II. P ROPOSED M ETHOD
A. Basic Principle of the Proposed Method
Suppose a pure exponential signal as
x(n) = A1 e j φ1 e j 2π f 1n/ f s (1)
where A1 and φ1 are the magnitude and phase angle of the
exponential component, respectively, f s is the sampling fre-
quency, and f1 is the actual frequency. The goal is to estimate
the actual frequency f1 .
XUE et al.: POWER SYSTEM FREQUENCY ESTIMATION METHOD IN THE PRESENCE OF HARMONICS 57

In practice, the frequency deviation  f is usually estimated phase angles. Therefore, the sign of r can be obtained as
before the estimation of f 1 .  f is defined as  f = f 1 − f 0 , follows.
where f 0 = f s /N. f0 is the assumed frequency, and N is the If |φ[x1 (n)/x  (n)]| < π/2 [φ(x) is the phase angle of x],
number of sample points in each assumed cycle. which means that the phasors x1 (n) and x  (n) have the same
To estimate  f , a new sequence x1(n) is constructed using phase angles, then r is positive; if |φ[x1 (n)/x  (n)]| > π/2,
the summation of x(n) as follows: which means that the phasors x1 (n) and x  (n) have the
opposite phase angles, then r is negative.

n+N−1 
n+N−1
x1(n) = x(m) = A1 e j φ1 e j 2π f 1m/ f s . (2) The above deduction shows that for a pure exponential
m=n m=n signal with positive fundamental frequency, the frequency
deviation can be obtained using (7). In the following, the effect
The right-hand side of (2) is a geometrical sequence with the
of negative frequency and harmonics on (7) will be studied.
common ratio of e j 2π f 1/ f s . Equation (2) can be expressed as
follows using the summation formula of geometrical sequence:
B. Effect of Negative Frequency Component
1 − e j 2π N f 1 / f s
x1(n) = A1 e j φ1 e j 2π f 1n/ fs . (3) Power system signals are always real signals, which can
1 − e j 2π f 1/ f s be expressed as the summation of two complex exponen-
Comparing (1) and (3), the ratio of x1(n) and x(n), denoted tial signals with positive frequency and negative frequency,
by r in this paper, can be expressed as follows: respectively, as follows:
1 − e j 2π N f1 / f s x(n) = 2 A1 cos(2π f 1 n/ f s + φ1 )
r = x1(n)/x(n) = . (4)
1 − e j 2π f 1/ f s = A1 e j φ1 e j 2π f 1n/ f s + A1 e j −φ1 e j −2π f 1n/ fs . (9)
Using f1 = f0 +  f in (4), then Then, x1(n) can be expressed as
1 − e j 2π N f 0 / f s e j 2π N f / f s 1 − e j 2π N f / f s 
n+N−1 
n+N−1
r= = . (5)
1 − e j 2π( f 0+ f )/ f s 1 − e j 2π( f 0+ f )/ f s x1(n) = x(m) = A1 e j φ1 e j 2π f 1m/ f s
m=n m=n
In (5), the values of 2π N f/ fs and 2π( f 0 +  f )/ f s are
much smaller than 1, then (5) can be simplified as 
n+N−1
+ A1 e j −φ1 e− j 2π f 1 m/ f s . (10)
2π N f/ fs N f m=n
r≈ = . (6)
2π( f 0 +  f )/ f s f0 +  f Equation (10) can be expressed as follows using the sum-
Equation (6) shows that the value of r is related to the fre- mation formula of geometrical sequence:
quency deviation  f . Therefore, the frequency deviation  f x1(n)
can be calculated based on (6) as follows:
1 − e j 2π N f1 / f s
f0r = A1 e j φ1 e j 2π f 1n/ fs
f = . (7) 1 − e j 2π f 1/ f s
N −r 1 − e j 2π N(− f1 )/ f s
+ A1e j −φ1 e− j 2π f 1n/ fs
Equation (7) is the basic idea of the proposed frequency 1 − e j 2π(− f 1)/ f s
estimation method in this paper. If the values of f 0 and N are N f −N f
fixed,  f is related only to the value of r . ≈ A1 e j φ1 e j 2π f 1n/ fs + A1e j −φ1 e− j 2π f 1n/ f s
f0 + f −(f 0 + f )
The value of r in (7) can be obtained using any two corre- N f
sponding samples in the constructed sequence x1(n) and the = x(n) . (11)
f0 +  f
original sequence x(n), as shown in (4). However, the value
of r obtained using only two samples will be greatly affected Then, the value of r can be obtained as
by noise. To eliminate the effect of noise, we use the ratio of N f
r = x1(n)/x(n) ≈ . (12)
two phasors, and the absolute value of r can be calculated as f0 +  f
follows: Equation (12) is the same as (6). It shows that the negative
|r | = |x1(n)/x  (n)| (8) frequency exponential component hardly affects the relation-
ship between r and  f . In other words, the precision of (7) is
where x1 (n) and x  (n) are two phasors of the con- not affected by the addition of negative frequency component.
structed
n−N+1 and original sequence, respectively: x1 (n) =
x1(m)W m−n
, x  (n) = n−N+1
x(m)W Nm−n , and
m=n N m=n C. Effect of Harmonics
W N = e− j 2π/N . The sign |x| means the absolute value of x.
Power system signals usually contain harmonic components
Equation (8) gives the absolute value of r, and the sign of r
due to the application of power electronic devices, nonlinear
can be determined by the comparison between the phase angles
loads, and so on. The effect of harmonics is studied in the
of x1 (n) and x  (n). Equation (4) shows that if r is positive,
following. Suppose a kth harmonic component is added to
then x1(n) and x(n) have the same sign, which means that the
the signal in (1), and the expression of the signal becomes as
phasors x1 (n) and x  (n) have the same phase angles; if r is
follows:
negative, then x1(n) and x(n) have the opposite sign, which
means that the phasors x1 (n) and x  (n) have the opposite x(n) = A1 e j φ1 e j 2π f 1n/ f s + Ak e j φk e j 2πk f1 n/ fs (13)
58 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO. 1, JANUARY 2016

where Ak and φk are the magnitude and phase angle of the

kth harmonic component, respectively, and k is the harmonic
order. Here, the value of k is assumed as k  N, which means
that only the low-order harmonics are considered here. There
are two reasons for this assumption. First, the magnitude of
harmonics usually decreases with the increase in the harmonic
order; therefore, the magnitude of the high-order harmonic is
usually much less than that of the low-order harmonic. Second,
one cycle DFT will be applied in the following procedure to
calculate the magnitude of the sequence. Note that one cycle
DFT can be considered as a bandpass filter, which will decay
the high-order harmonics greatly. Therefore, the effect of the
high-order harmonics is reduced even further. Thus, only the Fig. 1. Frequency response of the FIR filter h1(n).
low-order harmonics are considered here.
Then, x1(n) can be constructed and expressed as follows: are all 1 s. The frequency response of the FIR filter can be

n+N−1 
n+N−1 expressed as follows:
x1(n) = x(m) = A1 e j φ1 e j 2π f 1m/ f s
H 1( f ) = N sin c(π f / f 0 )e j [−π( f /f 0)×(1−1/N)] (19)
m=n m=n

n+N−1 where H 1( f ) is the frequency response of h1(n), f 0 = f s /N,
+ Ak e j φk e j 2πmk f1 / f s . (14) and f s is the sampling frequency. A section of the normalized
m=n amplitude and phase angle of H 1( f ) is plotted in Fig. 1.
Equation (14) can be expressed as follows using the sum- The spectrum of the filtered signal is the multiplication of the
mation formula of geometrical sequence: spectrum of x(n) and the frequency response of the FIR filter
as follows:
1 − e j 2π N f 1 / f s
x1(n) = A1 e j φ1 e j 2π f 1n/ fs X1( f ) = X ( f )H 1( f )
1 − e j 2π f 1/ f s (20)
1 − e j 2π Nk f1 / f s where X1( f ) and X ( f ) are the spectra of x1(n) and x(n),
+ Ak e j φk e j 2πk f1 n/ fs . (15)
1 − e j 2π f 1k/ fs respectively.
In (15), since k  N, (1 − e j 2π Nk f/ fs /1 − e j 2π f k/ fs ) ≈ Equation (20) shows that after the filtering process in (18),
(N f / f 0 +  f ), and (15) can be simplified as the filter gain for the component with a frequency of f
is H 1( f ). Thus, the filter gains for different frequency
x1(n) components can be analyzed from the expression of H 1( f )
N f k N f in (19). For the positive fundamental component in the signal,
≈ A1 e j φ1 e j 2π f 1n/ f s + Ak e j φk e j 2πk f1 n/ fs
f 0 + f k( f 0 + f ) f = f 1 = f0 +  f , then
N f
= x(n) . (16) H 1( f 1) = N sin c(π f 1 / f 0 ) × e j [−π( f 1/ f 0 )×(1−1/N)]
f0 +  f
= −N sin(π f / f 0 )/(π( f + f 0 )/ f 0 )
From (16), the value of r can be obtained as
× e j [−π( f 0+ f )/ f 0 ×(1−1/N)] ≈ N f/( f0 +  f ).
N f
r = x1(n)/x(n) ≈ . (17) (21)
f0 +  f
For the negative fundamental component in the signal,
Equation (17) is the same as (6). It shows that the addition of
f = − f 1 , then
harmonics hardly affects the relationship between r and  f .
In other words, the precision of (7) will not be affected by H 1(− f 1) = N sin c(−π f 1 / f 0 ) × e j [π f 1/ f 0 ×(1−1/N)]
the addition of harmonics. Equation (7) is the basic idea of = −N sin(−π f / f 0 )/(−π( f + f 0 )/ f 0 )
the proposed method, and (7) is obtained using the deduction
in the time domain. An analysis in the frequency domain to × e j [π( f 0+ f )/ f 0 (1−1/N)] ≈ N f/( f0 +  f ).
validate the above deduction is given in the following. (22)
For the kth harmonic component in the signal,
D. Frequency Analysis of the Proposed Method f = f k = k f 0 + k f , then
Equation (2) can be considered to filter the original signal H 1( f k ) = N sin c(π f k / f 0 ) × e j [πk f1 / f 0 (1−1/N)]
with a finite-impulse response (FIR) filter h1(n) as follows:
= (−1)k N sin(kπ f / f 0 )/(πk( f + f 0 )/ f 0 )

n+N−1
× e j [−πk( f 0 + f )/ f 0 (1−1/N)] ≈ N f/( f0 +  f ).
x1(n) = x(m) = x(n) ∗ h1(n) (18)
m=n (23)
where h1(n) is the impulse response of an FIR filter. The Equations (21)–(23) show that after the filtering process
length of the FIR filter is N, and the coefficients of the filter in (18), the gains for the positive fundamental frequency,
XUE et al.: POWER SYSTEM FREQUENCY ESTIMATION METHOD IN THE PRESENCE OF HARMONICS
Fig. 2. Absolute values of the approximate and the theoretical r (the red
line represents the approximation value, and the blue line is for the theoretical
value).
negative fundamental frequency, and harmonic components
are almost the same. This means that after the FIR filter
process in (18), all of these components have a similar gain as
N f/( f0 + f ). Therefore, the addition of negative frequency
and harmonic components will not affect the relationship
between r and  f , or in other words, the precision of (7)
will not be affected by the addition of negative frequency and
harmonic components.
E. Large Frequency Deviation
In the design of power analyzer or instrument, large fre-
quency deviation should be considered, for example, the Inter-
national Electro technical Commission (IEC) standard recom-
mended the accuracy requirement for the frequency estimation
in the range 42.5–57.5 Hz for the 50-Hz system. Therefore,
the performance of the proposed method under large frequency
deviation situations is analyzed in the following.
For the proposed method described in Section II-A, the
approximation is used only in the transformation from
(5) to (6), where the assumption of small frequency
deviation is used, and (N f / f 0 ) +  f is used to approx-
imate (1 − e j 2π N f / f s/1 − e j 2π( f 0+ f )/ f s ). Fig. 2 is given
to analyze the approximation error. In Fig. 2, the hori-
zontal axis is the value of  f in hertz, and the vertical
axis denotes the absolute values of the approximation
r = (N f )/( f 0 +  f ) (red line) and the theoretical
r = (1 − e j 2π N f / f s /1 − e j 2π( f 0+ f )/ f s ) (blue line).
In Fig. 2, f s is set as 3200 Hz, f 0 is set as 50 Hz, and
N = f s / f 0 = 64.
Fig. 2 shows that the two waveforms are almost overlap
when  f is small, and the difference between the two wave-
forms increases with the increase in the absolute value of  f .
It means that the approximation error is small when  f is
small, and the error increases with the increase in  f . The
above deduction shows that the proposed method has high
precision when  f is small, which is similar to the DFT-based
method.
An example is given in the following to validate our
deduction above. The proposed method is compared
with the DFT-based method [3]. The sampled signal is
x(t) = sin(2π f 1 t) + 0.1 cos(6π f 1 t) + 0.1 cos(10π f 1 t), where
f 1 is the fundamental frequency and the value of f 1 is set
59
Fig. 3. Frequency estimation errors (the red line is for the DFT-based method,
and the blue line is for the proposed method).
from 48 to 52 Hz to study the effects of different frequency
deviations. The frequency estimation errors (FE) are given
in Fig. 3. In Fig. 3, the horizontal axis is the actual fundamen-
tal frequency in hertz, and the vertical axis is the frequency
estimation errors in hertz.
Fig. 3 shows that for both methods, the precision of fre-
quency estimation is high when the frequency deviation is
quite small, and the frequency estimation errors increase with
the increase in frequency deviation. Fig. 3 also shows that the
error of the proposed method increases much slower than that
of the DFT-based method when  f is small; for example,
the error of the proposed method is less than 0.01 Hz when
 f = ±2 Hz while that of the DFT-based method is higher
than 0.05 Hz. It means that the proposed method is less
sensitive to frequency deviations than the DFT-based method
when  f is small, for example, in the range [−2, 2 Hz]. This
conclusion is consistent with the analysis based on Fig. 2.
Fig. 3 shows that the frequency estimation error of the
proposed method is increased with the increase in frequency
deviation. To solve this problem and improve frequency esti-
mation precision under large frequency deviation situations,
an iterative process, similar to the iterative process of the
DFT-based method, can be applied to the proposed method
as follows. For example, in the beginning, the sample freque-
(1)
ncy f s is set as 3200 Hz, the assumed frequency f 0 is set
(1) (1)
as 50 Hz, and N = f s / f 0 = 64. If the estimated frequency
deviation using (2)–(8) is  f (1) = 5 Hz, then the estimated
(2) (1)
frequency is f 1 = f 0 + f (1) = 55 Hz, and the value of N
is adjusted as N (2) = || f s / f 1(2) || = 58, where the sign ||x||
means the integer closest to x. The assumed frequency is set
(2)
as f0 = f s /N (2) = 57.1429, then recalculate the frequency
deviation  f (2) using (2)–(8), and the estimated frequency is
(3) (2)
f 1 = f 0 +  f (2) = 57.1429 +  f (2) . The process can be
(n) (n−1)
continued until the estimated f 1 and f1 have the same
values. Therefore, the precision of the proposed method can
be improved by adjusting the value of N and f0 , which is
similar to the DFT-based method.
F. Data Window Length and Computational Burden
For the proposed method, the data window length needed for
frequency estimation is determined by (4) and (8). In the above
deduction, the length of the FIR filter is N samples in (4),
60 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO. 1, JANUARY 2016

Fig. 4. Filter process.

which means that to obtain one value of x1(n), N samples of

x(n) are needed. The length of the FIR filter is also N samples
in (8), which means to obtain one value of x1 (n), N samples
of x1(n) are needed. Therefore, the data window length needed
for frequency estimation is 2N − 1 samples. However, the
actual value of N is not known beforehand and will be preset
as N = fs / f 0 ( f0 = 50 for the 50-Hz systems and f 0 = 60 Fig. 5. Frequency response of the FIR filters.
for 60 Hz systems; f s is the sampling frequency) in practice.
Therefore, a longer data window of 3N is sampled and stored
to consider the large frequency deviation situation.
The computational burden of the proposed method is mainly
on the construction of x1(n) in (4) and two phasor calculations
in (8), which are totally about N 2 real additions, 2N complex
additions, and 2N complex multiplications. Therefore, the
proposed method is easy to be implemented in practice.

III. E FFECTS OF I NTERHARMONICS AND THE

P REPROCESSING P ROCEDURE
A. Effects of Interharmonics
The time variable characters of power system may give forth
to interharmonic components in the voltage and current sig-
nals. The effects of interharmonics are analyzed in this section,
and a preprocessing procedure is proposed to eliminate the
effects of interharmonics with frequency near the fundamental
frequency.
Equation (2) can be considered to filter the original signal Fig. 6. Frequency estimation errors (the red curve is for the DFT-based
method, and the blue curve is for the proposed method).
with an FIR filter h1(n) as shown in (18). The phasor
calculation in (8) can be considered as another filter process
as follows:
Fig. 5 shows that in the normalized amplitude

n−N+1

x1 (n) = x1(m)W Nm−n = x1(n) ∗ h2(n) (24) of H 1( f ) and H 2( f ), the sidelobes in the range [−∞, − f 0 ]
m=n and [2 f 0 , ∞] are rather low. Therefore, the interharmonic
components with frequency among this range will be
where h2(n) is the impulse response of the FIR filter. The
decreased greatly, and will hardly affect the frequency
length of the FIR filter is N, and the coefficients of the filter
estimation precision. However, the interharmonics in the
are h2(n) = W NN−n . The frequency response of such a filter is
range [− f 0 , 2 f 0 ], especially those in the range [0, f 0 ], will
similar to Fig. 1 except for a right shift of 1 in the horizontal
greatly affect the precision of the proposed method.
axis and can be expressed as follows:
An example is given in the following to validate our
H 2( f ) = N sin c[π( f / f 0 − 1)] × e− j (π×(1−1/N)×( f /f 0−1)] deduction above. The proposed method is compared with the
(25) DFT-based method [3]. The model of the test signal is as
follows: x(t) = sin(2π f 1 t) + 0.01 cos(2π f i t), where f 1 is
where H 2( f ) is the frequency response of h2(n). Thus, x1 (n) the fundamental frequency and f 1 = 50.2 Hz. f i is the
can be considered as a another new sequence obtained by frequency of the interharmonic, and the value of f i is set
filtering the original signal with two filters as shown in Fig. 4. from 0 to 400 Hz to study the effects of interharmonics with
The spectrum of x1 (n) is the multiplication of spectrum of different frequencies. The frequency estimation errors (FE) are
x(n) and the frequency response of the two filters as follows: shown in Fig. 6. In Fig. 6, the horizontal axis is the frequency
of the interharmonic in hertz, and the vertical axis represents
X1 ( f ) = X ( f )H 1( f )H 2( f ). (26)
the frequency estimation errors in hertz. Fig. 6 shows that if the
The normalized amplitude of H 1( f ) and H 2( f ) and their frequency of the interharmonic is higher than 100 Hz, the error
multiplications are plotted in Fig. 5. of the proposed method is much lower than the DFT-based
XUE et al.: POWER SYSTEM FREQUENCY ESTIMATION METHOD IN THE PRESENCE OF HARMONICS 61

method; if the frequency of the interharmonic is in the

range [50, 100 Hz], the errors of both methods are comparable;
if the frequency of the interharmonic is lower than 50 Hz, the
error of the proposed method is higher than the DFT-based
method. Therefore, the proposed method is immune to inter-
harmonics with a frequency higher than 100 Hz. This conclu-
sion is consistent with the analysis based on Fig. 5.

B. Preprocessing Procedure
As regards the frequency synchronization, it is known that
the main problems arise in the presence of interharmonics
with frequency near the fundamental frequency. However,
Figs. 5 and 6 show that the proposed method is sensitive
to the interharmonics with frequency near the fundamental
frequency. A further analysis shows that the proposed method
is based on the signal model with a constant amplitude as
shown in (1). However, the interharmonics with frequency
near the fundamental frequency will give forth to amplitude Fig. 7. Test waveform.
variations. Our analysis shows that if the amplitude variations
are eliminated, the frequency estimation errors can be reduced.
To eliminate the amplitude variation of the sampled signal, the first valid local maximum x(m0) and then search the
a preprocessing procedure is proposed in the following. neighborhood of x(m0 + N), find the second valid local
The idea of the preprocessing procedure is to extract the maximum x(m1) and then search for the neighborhood
upper envelop of the sampled signal and divide the sampled of x(m1 + N), and find the third valid local maximum,
signal using this envelop. The preprocessing procedure has until all the valid local maxima are found.
three steps. 2) Step 2: Get the upper envelop of the sampled signal
1) Step 1: Find the valid local maxima of x(n). The valid using the cubic spline data interpolation on the valid
local maxima are defined as follows. If x(m) is a local maxima obtained in Step 1. This upper envelop can
valid local maximum, it must satisfy the following three be considered as the amplitude of the sampled signal.
conditions. 3) Step 3: Divide the sampled signal using its upper
a) Condition 1 envelop, and the amplitude variations of the sampled
[x(m) > x(m − 1)] & [x(m) > x(m + 1)]. signal can be greatly reduced.
The preprocessing procedure also belongs to batch signal
b) Condition 2
processing techniques where a block of data is required [21].
x(m) = max[x(m − K ), x(m − K +1), . . . , Our experiments show that block data of 4–5 nominal cycles
x(m), . . . , (m + K )] are accurate enough for the extraction of the upper envelop
and the elimination of the amplitude variations.
where K is a constant.
To validate the efficiency of the proposed procedure,
c) Condition 3 a simulation is given in the following. The model of the
The valid local maxima should be approximately test signal is as follows: x(t) = [1 + 0.1 sin(2π2t) +
equally separated. 0.1 sin(2π4t)] sin(2π f 1 t)+0.1 sin(6π f 1 t)+0.1 sin(10π f 1 t)+
Condition 1 is used to find the local maxima. However, 0.1 sin(18π f 1 t), where f 1 is the fundamental frequency and
not all local maxima are valid to get the upper envelop, f 1 = 50.2 Hz. There is a 0.2-Hz frequency deviation for
for example, the maxima caused by noise and harmonics. the fundamental component, and there are 10% third, fourth,
Therefore, conditions 2 and 3 are proposed to eliminate fifth, and ninth harmonics. There are also two amplitude
such invalid local maxima. Condition 2 is used to assure modulations with frequencies of 2 and 4 Hz in the test signal.
that x(m) is the maximum point in the data window Therefore, there will be 50.2 ± 2 and 50.2 ± 4 Hz interhar-
centered as x(m) and with the length of 2K + 1. The monics existing in the test signal, and these interharmonics
value of K determines the length of the data window and will give forth to amplitude variations in the test signal. The
can be set as smaller than N/2, where N is the number test signal x(n) in the duration of 1 s is shown in Fig. 7. Fig. 7
of sample points in each fundamental cycle, for example, shows that there are apparent amplitude variations in the test
K can be set as K = N/2 − 10. Condition 3 is used to signal.
estimate the location of the next valid local maxima. For The data block for the preprocessing procedure is set as
example, if the first valid local maximum is x(m0), then four nominal cycles, or 0.08 s. It means that the preprocessing
the second valid local maximum will be located in the procedure is performed based on a 0.08-s data block, and
neighborhood of x(m0 + N). Therefore, the valid local then a frequency estimation is obtained using the data after
maxima can be recursively found as follows. First, find the preprocessing procedure. The time label of the estimated
62 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO. 1, JANUARY 2016

Fig. 10. Frequency estimations with the preprocessing procedure.

Fig. 8. First data block and its upper envelop obtained using the preprocessing
procedure (the blue line is for the data block, and the red line is for the
extracted upper envelop).
in Fig. 9, which are exactly 3N data samples, are used for
the frequency estimation to avoid the end-effect errors of the
cubic spline data interpolation.
Fig. 10 shows the measured fundamental frequency with
the preprocessing procedure in the duration of 1 s. Fig. 10
shows that the estimated frequencies have high precision with
a maximum error of less than ±0.013 Hz in the duration
of 1 s. Figs. 7–10 show that with the preprocessing procedure,
the proposed method can give accurate frequency estimation
in the presence of interharmonics with frequency near the
fundamental frequency.

C. Computational Burden of the Preprocessing Procedure

There are three steps in the preprocessing procedure, and
the computational burden of the preprocessing procedure is
mainly on the cubic spline data interpolation in Step 2. The
cubic spline data interpolation function is defined as follows.
Suppose there are interpolation nodes in [a, b], a = x 1 <
x 2 < · · · < x k = b, with the corresponding values of
y1 , y2 , . . . , yk . If the function S(x) satisfies S(x j ) = y j
Fig. 9. Data after the preprocessing procedure.
( j = 1, 2, . . . , k), S(x) is a polynomial not higher than three
times in [x j , x j +1] ( j = 1, 2, . . . , k − 1), and S(x) has a
continuous secondary derivative in [a, b], and then S(x) is a
cubic spline interpolation function. S(x) can be obtained using
frequency is set as the middle of the data block. Then shift
the following equation:
the data window by one sample and obtain the next frequency  
estimation. (x j +1 − x)3 (x − x j )3 Mj 2
S(x) = Mj + M j +1 + y j − hj
Fig. 8 shows the test signal in the first data block, that is, 6h j 6h j 6
in the time range [0, 0.08 s]. There are apparent amplitude  
x j +1 − x M j +1 2 x − x j
variations in the test signal in the time range [0, 0.08 s]. × + y j +1 − hj
hj 6 hj
The upper envelop obtained using the preprocessing procedure
(27)
is also shown in Fig. 8. Fig. 8 shows that the obtained
upper envelop approximates to the amplitude variations very where x is in [x j , x j +1], h j = x j +1 − x j , and M j can be
well. Fig. 9 shows the signal obtained after the preprocessing obtained by solving the three moment equations as follows:
procedure. Fig. 9 shows that the amplitude variation has been ⎧
⎪
⎪ μ1 M0 + 2M1 + λ1 M2 = d1
greatly reduced. Then, the data in Fig. 9 are used to estimate ⎪
⎨μ M + 2M + λ M = d
the frequency. Note that there are 4N data samples in Fig. 9, 2 1 2 2 3 2
(28)
⎪
⎪ ···
and only 3N samples are needed for a frequency estimation ⎪
⎩
as shown in Section II-F. Therefore, the data after t = 0.02 s μk−1 Mk−2 + 2Mk−1 + λk−1 Mk = dk−1
XUE et al.: POWER SYSTEM FREQUENCY ESTIMATION METHOD IN THE PRESENCE OF HARMONICS 63

where μ j = h j −1 /(h j −1 + h j ), λ j = h j /(h j −1 + h j ),

d j = 6((y j +1 − y j /h j ) − (y j − y j −1/h j −1 ))(1/h j −1 + h j ),
and γ j = 1 − μ j .
The computational burden of (27) and (28) is analyzed in
the following. In (27), the values of (M j /6h j ), (M j +1 /6h j ),
(y j − (M j /6)h 2j )(1/h j ), and (y j +1 − (M j +1 /6)h 2j )(1/h j ) can
be precalculated and saved as constants. Therefore, to obtain
one value of S(x), the computational burden is about ten
real multiplications and five real additions. As mentioned
above, block data of 4 (/5) nominal cycles are needed for the
preprocessing procedure, which means that there are 4N (/5N)
data in S(x), where N denotes the sample points in every
nominal cycle. Therefore, the total computational burden of
(27) is about 40N (/50N) real multiplications and 20N (/25N)
real additions. In (28), it is well known that the three moment
equations shown in (28) can be solved using chasing method.
The computational burden is about 5k − 4 multiplications,
where k is the number of interpolation nodes. The number
of interpolation nodes in our experiments is usually small, for
example, k = 3–5. Therefore, the computational burden of
(28) is quite small.
The above analysis shows that the computational burden of
the cubic spline data interpolation in Step 2 is about
40N (/50N) real multiplications and 20N (/25N) real addi-
tions. What is more, with a data block of 4N (/5N) samples,
the computational burden of Step 1 of the preprocessing
procedure is less than 4N (/5N) additions and that of Step 3 is
4N (/5N) multiplications. Therefore, the total computational
burden of the preprocessing procedure is about 44N (/55N)
real multiplications and 24N (/30N) real additions.

D. Flowchart of the Proposed Method

The flowchart of the proposed method with the preprocess-
ing procedure is given in Fig. 11. In the flowchart,
the sign abs(x) means the absolute value of x and the
sign ||x|| means the integer closet to x. Similar to the
DFT-based method, the proposed method belongs to batch
signal processing techniques where a block of data is required
for the measurement [21]. Therefore, a block of data is
sampled and stored before the preprocessing procedure. Our
above analysis shows that block data of 4–5 nominal cycles
are accurate enough for the preprocessing procedure, thus a
data window of 4N (/5N) is sampled and stored at first.
In the flowchart, there is a condition to judge whether it
is necessary to perform Steps 2 and 3 in the preprocessing
procedure. The reasons are as follows. In the preprocess-
ing procedure, the computational burden is mainly on the
cubic spline data interpolation in Step 2. The function of
Steps 2 and 3 is to get the upper envelop of the sampled signal
and thus eliminate the amplitude variations of the sampled
signal. If there is no amplitude variation in the sampled signal,
Steps 2 and 3 are not necessary. Therefore, a condition to judge
Fig. 11. Flowchart of the proposed method.
whether Steps 2 and 3 are necessary is added in the flowchart.
The parameter D is defined as where x(m i ), i = 0, 1, 2, k − 1, is the
 valid local maxima
k−1 obtained in Step 1 and x(m) = (1/k) k−1 i=0 x(m i ). The value
[x(m i ) − x(m)]2
D = i=0 k−1 (29) of D is related to the amplitude variations of the sampled
i=0 [x(m i )]
2
signal; in other words, the smaller the value of D, the smaller
64 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO. 1, JANUARY 2016

the amplitude variations. Therefore, if the value of D is small

enough, for example, D < 5e − 3, it means that there is no
amplitude variation of the sampled signal, and there is no need
to perform Steps 2 and 3 in the preprocessing procedure. The
condition is helpful to reduce the computational burden of the
preprocessing procedure if there are no amplitude variations
in the sampled signal.
Note that the data window for the preprocessing procedure
is set as 4N, while only 3N samples are needed for frequency
estimation as mentioned in Section II-F. Therefore, in the
flowchart, we set x(n) = x(n + N), n = 0, 1, . . . , 3N − 1,
after the preprocessing procedure. It means that the first N data
are discarded and only the following 3N data are used after
the preprocessing procedure. Discarding the first N data is
helpful to avoid the end-effect errors of the cubic spline
data interpolation and improve the frequency measurement
Fig. 12. Frequency estimation errors of case 1.
accuracy.

IV. S IMULATIONS
Simulations are given to compare the proposed method
and the DFT-based method [3]. In Section IV-A, steady-state
verifications are performed, where the fundamental frequency
has a constant value. In Section IV-B, the frequency variations
are studied, where the linear and sinusoidal fundamental
frequency variations are studied.

A. Steady-State Verifications
Standard IEC 61000-4-30 provides the requirements for
the frequency synchronization, in both reference working
condition and in the presence of disturbances. The esti-
mation uncertainty for class A instrument is as follows:
the maximum uncertainty is 10 mHz in the range ±15%
of the nominal frequency: 50 ± 7.5 and 60 ± 9 Hz.
Case 1 is performed to test the estimation uncertainty of
Fig. 13. Frequency estimation errors of case 2.
the 50-Hz system. Case 2 is performed to test the esti-
mation uncertainty of the 60-Hz system. Case 3 is per-
the proposed method has better immunity to the negative
formed to test the estimation uncertainty in the presence
fundamental frequency component.
of harmonics and interharmonics specified in the uncer-
Case 2: The model of the test signal is as follows:
tainty steady-state verification in standard IEC 61000-4-30.
x(t) = cos(2π f 1 t), where f 1 is the fundamental frequency
Case 4 is performed to test the effects of noise. Case 5 is
and f 1 is set as from 51 to 69 Hz with a step of 0.1 Hz to
performed to test the methods’ performance in the presence
test the measurement uncertainty in the range ±15% of the
of harmonics/interharmonics with variable amplitudes.
nominal frequency of 60-Hz systems. The sample frequency
Case 1: The model of the test signal is as follows:
f s is set as 3840 Hz.
x(t) = cos(2π f 1 t), where f 1 is the fundamental frequency
The frequency estimation errors (FE) are given in Fig. 13.
and f1 is set as from 42.5 to 57.5 Hz with a step of 0.1 Hz
Fig. 13 shows that both methods satisfy the class A require-
to test the measurement uncertainty in the range ±15% of the
ments of IEC 61000-4-30, and the maximum uncertainty
nominal frequency of 50-Hz systems. The sample frequency
is less than 10 MHz in the range ±15% of the nominal
f s is set as 3200 Hz.
frequency: 60 ± 7.5 Hz. Fig. 13 also shows that the errors
The frequency estimation errors (FE) are given in Fig. 12.
of the proposed method are much less than those of the
Fig. 12 shows that both methods satisfy the class A require-
DFT-based method.
ments of IEC 61000-4-30, and the maximum uncertainty
Case 3: The model of the test signal is as follows:
is less than 10 MHz in the range ±15% of the nominal
frequency: 50 ± 7.5 Hz. Fig. 12 also shows that the errors x(t) = cos(ω1 t) + 0.1 cos(3ω1 t) + 0.05 cos(5ω1 t + π)
of the proposed method are much less than those of the
+ 0.1 cos(7ω1 t) + 0.05 cos(13ω1 t)
DFT-based method. An error analysis shows that the errors
of the DFT-based method are mainly caused by the negative + 0.05 cos(25ω1 t) + 0.05 cos(29ω1 t)
fundamental frequency component in the test signal, while + 0.01 cos(2π175t) + 0.01 cos(2π375t)
XUE et al.: POWER SYSTEM FREQUENCY ESTIMATION METHOD IN THE PRESENCE OF HARMONICS
Fig. 14. Frequency estimation errors of case 3.
Fig. 15. Frequency estimation errors of case 4.
where ω1 = 2π f 1 , f 1 is the fundamental frequency, and
f 1 = 49 Hz with a frequency deviation of 1 Hz. There
are 10% 3rd and 7th harmonics, 5% 5th, 13th, 25th, and
29th harmonics, and two interharmonics with frequencies
of 175 and 375 Hz. The parameters of the harmonics and
interharmonics are set according to the uncertainty steady-state
verification in standard IEC 61000-4-30.
The frequency estimation errors (FE) in the duration of 1 s
are given in Fig. 14. Fig. 14 shows that under the harmonic
and interharmonic disturbances, the maximum error of the pro-
posed method is about 10 mHz, while that of the DFT-based
method is more than 40 mHz. This simulation shows that
the proposed method has better immunity in the presence
of harmonics and interharmonics specified in the uncertainty
steady-state verification in standard IEC 61000-4-30.
Case 4: To test the methods’ performance under noise
environment, 60-dB white noise is added to the test signal
in Case 3. The frequency estimation errors (FE) are given
in Fig. 15. Fig. 15 shows that both methods have good
resistance to noise. The maximum error of the proposed
method is about 15 mHz and that of the DFT-based method
is about 48 mHz.
65
Fig. 16. Frequency estimation errors of case 5.
Case 5: To test the methods performance in the presence of
harmonics/interharmonics with variable amplitude. The model
of the test signal is
x(t) = cos(ω1 t) + 0.1 cos(7ω1 t) + 0.05 cos(13ω1 t)
+ 0.05 cos(25ω1 t) + 0.05 cos(29ω1 t)
+ 0.1[1 + 0.1 cos(2π1t)] cos(3ω1 t)
+ 0.05[1 + 0.1 cos(2π2t)] cos(5ω1 t + π)
+ [1 + 0.1 cos(2π1t)]0.01 cos(2π175t)
+ [1 + 0.1 cos(2π1t)]0.01 cos(2π375t).
The test model is similar to that in Case 3. The only
difference is that in Case 3, the amplitudes of harmonics
and interharmonics are set up with constant values, while
in Case 5, the amplitudes of the third and fifth harmonics
and the two interharmonics vary with time. The frequency
estimation errors (FE) are given in Fig. 16. Fig. 16 shows that
the proposed method can provide more accurate frequency
estimation than the DFT-based method in the presence of
harmonics/interharmonics with variable amplitudes.
B. Frequency Variations
In power system, the transient state of the system will
give forth to frequency variations. The linear and sinusoidal
frequency variations are studied in the following.
Case 6: The model  of the test signal is as follows:
x(t) = cos(2π f1 t) + k=5,7,11,13 0.5 sin(2πk f 1 t), where the
fundamental frequency linearly changes from 49 to 51 Hz
in the 1-s duration. The frequency tracking results are given
in Fig. 17.
Fig. 17 shows that the DFT-based method has large oscil-
lation errors in the frequency tracking, while the proposed
method tracks the linear change of fundamental frequency
precisely.
Case 7: The model of the test signal is the same as that
in Case 6, and the fundamental frequency has a sinusoidal
variation with a frequency of 2 Hz and an amplitude of 0.2 Hz.
The frequency tracking results are given in Fig. 18.
66 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO. 1, JANUARY 2016
Fig. 17. Frequency tracking results (red graph is for the DFT-based method,
and blue graph is for the proposed method).
Fig. 18. Frequency tracking results (the red curve is for the DFT-based
method, and the blue curve is for the proposed method).
Fig. 18 shows that both methods can track the sinusoidal
variation of the fundamental frequency; however, there are
large oscillation errors in the DFT-based method, while the
proposed method can track the sinusoidal variation of funda-
mental frequency precisely.
V. T ESTS W ITH ACTUAL S IGNALS
In order to evaluate the performance of the proposed
method, tests with actual signals are performed. The actual
signals are voltage signals of a microgrid and the connected
power grid during the transition from the islanding mode to the
grid-connected mode. First, the microgrid worked at islanding
mode, and then the grid-connected control signal was given
at time instant t = 0.115 s, and the microgrid successfully
connected to the power grid after about 0.05 s, and worked at
grid-connected mode.
The sampled voltage signals of the microgrid and the power
grid are given in Fig. 19. The sample frequency is set as
51 200 Hz. Note that it is clearly shown in Fig. 19 that
before the grid-connected process, the microgrid works at
the islanding mode, and the voltage of the microgrid is not
synchronous with that of the power grid. After the grid-
connected process, the microgrid works at grid-connected
Fig. 19. Sampled voltage signals (the red line is for the microgrid, and the
blue line is for the connected power grid).
mode, and the voltage of the micorgrid is synchronous with
that of the power grid. In this experiment, the proposed method
is compared with the DFT-based method [3] and the PLL
method [9].
Note that the DFT-based method and the proposed method
belong to batch signal processing techniques where a block
of data is required for the estimations [21]. The response
time of the batch signal processing is mainly determined by
the time delay for sampling and storing the data blocks, and
the precision of frequency estimation methods is also related
to the length of data blocks. In other words, the longer the
data blocks, the slower the response time, and the higher
the estimation precision. For a fair comparison, the length
of the data block is set as 80 ms for the DFT-based method
and the proposed method, and the time label of the frequency
estimation is set as the middle of the data block.
On the other hand, the PLL belongs to the recursive analysis
techniques, which track the time trajectories of the estimated
parameters [21]. The convergence time and frequency esti-
mation precision are usually two contradictory factors for
recursive analysis techniques: the faster the convergence time,
the worse the estimation precision. Our experiments also show
that the postfilter is needed in the PLL structure to smooth
the output frequencies. Therefore, a second-order low-pass
filter with a cutoff frequency of 25 Hz is added in the PLL
implementation as the postfilter for the frequency estimation.
The addition of the postfilter introduces some delay, but
smoothes the output frequency significantly.
The frequency estimations using the DFT-based method
and the proposed method are given in Fig. 20. The fre-
quency estimations using PLL with/without postfilter are given
in Fig. 21. Fig. 21 shows that the frequency estimations
using PLL without postfilter have large oscillations and cannot
satisfy the frequency estimation requirements as specified in
IEC 61000-4-30. Therefore, only the frequency estimation
using PLL with postfilter is shown in Fig. 22. Some com-
parisons are given based on Figs. 20 and 22 as follows.
1) The DFT-based method and the proposed method belong
to batch signal processing techniques where a block of
XUE et al.: POWER SYSTEM FREQUENCY ESTIMATION METHOD IN THE PRESENCE OF HARMONICS
Fig. 20. Frequency estimations (the red line is for the microgrid, and the
blue line is for the connected power grid).
Fig. 21. Frequency estimations of the microgrid and the connected
power grid.
data is required for the estimation. In this experiment,
the length of the data block is set as 80 ms, and the time
label of the frequency estimation is set as the middle of
the data block. Therefore, there will be no output in the
first 0.04 s, and a steady output is obtained after that.
2) The PLL belongs to the recursive analysis techniques,
which track the time trajectories of the estimated
parameters. The response time of the recursive signal
processing techniques is mainly determined by the con-
vergence time of the PLL. Fig. 22 shows that after about
0.1 s, a steady output is obtained with the proper setting
of the parameters of the PLL structure.
3) For the grid-connected transient, the frequency of the
microgrid changes from about 50.25 to about 50.02 Hz.
All of the three mentioned methods can track these
frequency changes. The overshoots during the transient
are less than 1 Hz, and the response time to follow the
transient is similar, that is, from about 0.16 to 0.2 s.
67
Fig. 22. Frequency estimations using PLL with postfilter.
Fig. 23. Frequency estimations of the microgrid.
Figs. 20 and 22 show that all of the three methods give
similar estimations and can track the frequency change of the
microgrid during the grid-connected transient. Figs. 20 and 22
show that all of the methods have similar stationary accuracy
and dynamic response characters. All the above analyses
show that the proposed method can give feasible frequency
estimations, as well as the DFT-based method and the
PLL method.
A further comparison is given in Fig. 23. Fig. 23 shows
the estimated frequency of the microgrid using the
three mentioned methods. To give a clear comparison in the
steady state, the axis of Fig. 23 is set in the range [0.04, 0.3 s]
for the horizontal axis and [49.99, 50.3 Hz] for the vertical
axis. Fig. 23 shows that the estimations obtained using the
PLL have apparently larger oscillations than the DFT-based
method and the proposed method in the steady state.
To further compare the DFT-based method and the proposed
method, a zoomed-in view of Fig. 23 is shown in Fig. 24.
The axis of Fig. 24 is set in the range [0.04, 0.3 s] for the
horizontal axis and [49.99, 50.3 Hz] for the vertical axis.
68 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 65, NO. 1, JANUARY 2016
Fig. 24. Zoomed-in view of Fig. 23.
Fig. 25. Zoomed-in view of Fig. 19.
Fig. 24 shows that the frequency estimations obtained using
the DFT-based method have many high-frequency oscillations,
while those obtained using the proposed method provide
more stable estimations. Our analysis shows that the rea-
son for the oscillations in the DFT-based estimations is the
high-frequency harmonics/interharmonics in the voltage of
the microgrid before the grid-connected process, and these
high-frequency harmonics/interharmonics give forth to the
oscillation errors in the DFT-based method. Fig. 25 shows
a zoomed-in view of Fig. 19. Fig. 25 clearly shows that there
are high-frequency harmonics/interharmonics in the voltage of
the microgrid before the grid-connected process. The effects of
these harmonics/interharmonics are also shown in Fig. 21(a).
In Fig. 21(a), there are apparent larger oscillations in the
red line (the estimations obtained using the PLL without the
postfilter) before the grid-connected process, and these oscil-
lations are caused by the high-frequency harmonics and inter-
harmonics in the test signal. Compared with the DFT-based
method and PLL method, the proposed method can provide
more stable estimations in the presence of high-frequency
harmonics and interharmonics. This experiment validates that
the proposed method can provide stable and feasible frequency
estimations for practical applications.
VI. C ONCLUSION
A novel frequency estimation method is proposed in this
paper. A new sequence is constructed using the summation
of the samples of the original sequence, and the amplitude
ratio of the new and original sequences is used to calculate the
frequency deviation value. The effects of negative fundamental
component, harmonics, and interharmonics are studied. A pre-
processing procedure is also proposed to eliminate the effects
of the interharmonics with frequency near the fundamental
frequency. Theoretical analysis and simulations validate that
the proposed method has good immunity to harmonics. Due to
its good resistance to harmonics and low required processing
effort, the proposed method is a good candidate for online
frequency estimation under harmonic-polluted situations.
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Hui Xue (M’14) received the B.S. and M.S. degrees from the Electric
Power Department, Shandong University, Jinan, China, in 1995 and 1998,
respectively, and the Ph.D. degree from the College of Information and
Electrical Engineering, China Agricultural University, Beijing, China, in 2003.
She is currently an Associate Professor with the College of Information
and Electrical Engineering, China Agriculture University. Her current research
interests include power quality and harmonic analysis.
69
Maohai Wang received the B.S. degree from North China Electric Power
University, Beijing, China, in 1997, the M.S. degree from Wuhan University,
Wuhan, China, in 2002, and the Ph.D. degree from Tsinghua University,
Beijing, in 2006.
He is currently a Senior Engineer with the Power Dispatch Control Center,
North China Power Grid Company, Ltd., Beijing. His current research interests
include phasor measurement unit and wide area measurements.
Rengang Yang received the Ph.D. degree from Tsinghua University, Beijing,
China, in 1993.
He is currently a Professor with the College of Information and Electrical
Engineering, China Agricultural University, Beijing. His current research
interests include power quality monitoring, harmonic analysis, and reactive
power compensation.
Yan Zhang received the B.S. degree from China Agricultural University,
Beijing, China, in 2014, where she is currently pursuing the master’s degree
with the College of Information and Electrical Engineering.
Her current research interests include power quality monitoring and har-
monic analysis.