New Time-Domain Based Power System Frequency Estim Algorithm

A new time-domain based power system frequency
estimation algorithm
Heresh Seyedi
1
*
,y
and Majid Sanaye-Pasand
2
1
Faculty of Electrical and Computer Engineering, University of Tabriz, Iran
2
Control and Intelligent Processing Center of Excellence, School of ECE, University of Tehran, Iran
SUMMARY
Frequency stability is one of the most important features of power systems. It is closely dependent on the
value of system frequency. In the present day modern power systems, fast and accurate frequency estimation
has become quite vital. Any mistake in accurate estimation of frequency may threaten the frequency stability
or at least lead to system operation problems. In this paper, a new time-domain based power system
frequency estimation algorithm is proposed. Different abnormal events such as frequency decline due to
load-generation mismatch are modeled and ability of the new frequency measurement technique to track the
frequency is tested. Performance of the new method is evaluated and compared with an existing frequency
estimation algorithm. Simulation results conrm advantages of the new approach over the existing method.
Copyright # 2011 John Wiley & Sons, Ltd.
key words: power system frequency measurement; frequency estimation algorithms; frequency tracking;
wavelet transform
1. INTRODUCTION
The fundamental frequency is one of the most important and sensitive parameters of a power system.
Frequency stability of power systems is closely dependent on the value of system frequency. Any
imbalance between the system load and generation directly affects the frequency and leads to
frequency deviations. Grid frequency decline due to shortage of energy supply or disconnection of
a generation plant should be detected by under-frequency load shedding relays. It is necessary to
maintain frequency at its nominal value or as close as possible to that value within a narrow limit.
Deviation of frequency from its rated value is an indication of imbalance between real power
generation and load demand. Any mistake in accurate estimation of frequency could cause inadequate
load shedding by frequency relays, which would eventually be resulted in a major grid collapse [13].
Various methods have been proposed for frequency estimation. These methods may be divided
in two groups [4,5]:
Time-domain based methods: In these methods frequency calculation is performed in time
domain. For example, zero-crossing algorithm is a time-domain based method.
Frequency-domain based methods: In these methods frequency calculation is performed in
frequency domain. For example, Fourier algorithm is a frequency-domain based method.
Most of the modern digital protective relays, using frequency-domain based methods, are based on
digital phasor estimation algorithms [5]. When power system frequency is different from its rated
value, accuracy of the phasor estimation algorithms is decreased. This could cause incorrect operation
of the load shedding relays.
EUROPEAN TRANSACTIONS ON ELECTRICAL POWER
Euro. Trans. Electr. Power (2011)
Published online in Wiley Online Library (wileyonlinelibrary.com). DOI: 10.1002/etep.572
*Correspondence to: Heresh Seyedi, Faculty of Electrical and Computer Engineering, University of Tabriz, Iran.
y
E-mail: hseyedi@tabrizu.ac.ir
Copyright # 2011 John Wiley & Sons, Ltd.
One of the commonly used frequency estimation techniques is Zero Crossing (ZC) method [510].
In this time-domain based method the number of voltage samples in half/one cycle is used to estimate
power frequency. Sensitivity to noise and harmonics is the major shortcoming of this approach.
Meanwhile, this drawback may be resolved to some extent by using appropriate ltering techniques.
Least Error Squares (LES) is another important frequency estimation technique [11,12]. The LES
technique provides a general tool for solving a set of equations.
Orthogonal lters method is another suitable approach for frequency estimation. In this method the
voltage signal is decomposed using two orthogonal lters. The obtained orthogonal components are
used in frequency estimation [1315].
In References [8] and [16], a Newton type algorithm for frequency estimation has been presented. In
this method a combination of NewtonRaphson and LES methods is used. In addition to frequency, the
proposed algorithm estimates magnitude, phase and DC component of the signal.
Fourier algorithm, which is based on the Fourier series, is a conventional method for phasor
computation by virtue of low computational complexity and satisfactory dynamic response. This
algorithmextracts the phasor of a specic signal component fromvoltage or current waveformusing an
orthogonal band-pass lter bank with a rectangular window [17]. Various frequency estimation
techniques, based on Fourier algorithm, are mentioned in the literature [18,19].
All of the above mentioned methods involve a trade off among the accuracy of estimation, speed of
convergence, robustness to noise and sampling rate. Therefore, in evaluating various methods one should
consider howmuch an algorithmsatises the mentioned requirements. In the present day power systems, fast
and accurate frequency estimation has become even more important due to the increasing level of harmonics
and distortions. An acceptable frequency measurement scheme should be immune to noise and distortions
which are present in the power system signals used for estimation and should be able to track the system
frequency. Although frequency tracking over a wide range is indispensable during start-up and shutdown of
generators in power plants, in this paper we focus on tracking near nominal value of frequency.
In a different study, the above-mentioned algorithms were simulated and their performances were
compared. Advantages and shortcomings of these algorithms were investigated and the algorithm
which provided the best performance was selected for comparison [4,20].
In this paper a new time-domain based method is proposed for frequency estimation. Pre-ltering
and de-noising, before frequency estimation, are important features which make the algorithm robust
and immune to practical contaminated signals of power systems. In this approach just ve consecutive
samples of the voltage signal are required for calculation of frequency. The method is faster than
existing methods and its sensitivity to noise and harmonics is reduced using proper ltering techniques.
Speed and accuracy of this method are tested and compared with a well-known existing frequency
estimation technique. Simulation results conrm acceptable performance of the new method and its
advantages over the conventional techniques.
In Section 2 of this paper several existing frequency estimation methods are introduced. In Section 3
the proposed technique of this work is presented. In Section 4 signal de-noising by Wavelet Transform
is introduced and nally in Section 5 simulation results are presented.
2. EXISTING FREQUENCY ESTIMATION TECHNIQUES
2.1. Zero crossing method
ZC method is one of the commonly used frequency estimation techniques. It may be implemented
in two different ways:
Using the number of cycles in a specied time span [7,8].
Using the number of voltage samples in half/one cycle [7,9].
2.2. Least error squares algorithm
LES is a general tool for solving a nonlinear set of equations, when the number of unknowns is less than
the number of equations. It is based on the Taylor expansion around initial estimated value of the real
frequency [1,11].
Copyright # 2011 John Wiley & Sons, Ltd. Euro. Trans. Electr. Power (2011)
DOI: 10.1002/etep
H. SEYEDI AND M. SANAYE-PASAND
2.3. Orthogonal lters
In this method the signal is decomposed into orthogonal components. Let x
1
(t) and x
2
(t) be the
orthogonal components of x(t).
x
1
t X sin2pft f
x
2
t X cos2pft f
(1)
x
0
1
t 2pf X cos2pft f
x
0
2
t 2pf X sin2pft f
(2)
Combining Equations (1) and (2), frequency is calculated according to
f
x
2
t x
0
1
tx
1
t x
0
2
t
2px
2
1
t x
2
2
t
(3)
The discrete-time form of Equation (3) may be written as the following [14]:
f
x
2
n x
0
1
nx
1
n x
0
2
n
2px
2
1
n x
2
2
n
(4)
2.4. Fourier algorithm
Various frequency estimation algorithms based on the Fourier transform have been proposed in the
literature. In this section the smart Fourier method of References [18] and [19] is introduced. In this
method the voltage signal is assumed to contain the mth order harmonic, in addition to the fundamental
frequency
xt X
1
cosvt
1
X
2
cosmvt
2
(5)
In Equation (5), v is dened as
v 2pf
0
Df (6)
where f
0
and Df are the rated frequency and frequency deviation, respectively.
The complete method of calculation, based on Fourier algorithm, is described in Appendix I.
2.5. Performance evaluation
The algorithms introduced in this section were simulated and their performances were investigated.
Through various comparative studies it was found that the smart Fourier algorithm provides the best
performance [4,20]. Therefore, this algorithm is selected in this paper and its performance is compared
with performance of the new proposed algorithm.
3. THE NEW FREQUENCY ESTIMATION METHOD
In the new frequency estimation technique proposed in this paper, continuous and discrete forms of
voltage signal are presented by Equations (7) and (8), respectively.
xt X cos vt f (7)
xn X cos nvDt f (8)
DOI: 10.1002/etep
TIME-DOMAIN BASED FREQUENCY ESTIMATION ALGORITHM
where Dt is the sampling period. Rewriting Equation (8) for four other samples, Equations (9)(14)
are obtained:
xn1 X cos nvDtvDt f (9)
xn 1 X cos nvDt vDt f (10)
xn 1xn1 2 X sinvDt sin nvDt f (11)
xn2 X cos nvDt2vDt f (12)
xn 2 X cos nvDt 2vDt f (13)
xn 2xn2 2 X sin2vDt sin nvDt f (14)
Dividing Equation (14) by Equation (11), Equation (15) is obtained.
xn 2xn2
xn 1xn1
2 cosvDt (15)
If in Equation (15) n is replaced by k 2, Equation (16) is achieved.
xkxk4
xk1xk3
2 cosvDt (16)
Therefore, frequency may be estimated by Equation (17) or (18)
f
1
2pDt
cos
1
0:5
xkxk4
xk1xk3
(17)
f
N f
n
2p
cos
1
0:5
xkxk4
xk1xk3
(18)
where f
n
and N are the nominal frequency and the number of samples per cycle, respectively.
Equation (18) implies that in the new approach, frequency is calculated using the current
sample and four previous samples of the voltage signal. Therefore, the new algorithm is
expected to be faster than the Fourier and LES methods which typically make use of a one cycle
sampling window. However, short window algorithms are more or less sensitive to harmonic and
noise contents of the source signal. Thus, some essential measures must be adopted in this regard as
follows:
The source signal must be de-noised before being used by the algorithm. Signal de-noising may
be realized using the wavelet transform. Application of wavelet for signal de-noising will be
described in Section 4.
High frequency components of the signal must be removed, using a low-pass lter. A low pass
Butterworth lter has been applied in this work.
Once frequency calculation is complete, some corrective stages are applied as well. It is
worth mentioning that the rate of frequency change is typically less than 2 Hz/second, though
in very rare situations it may reach, at most, 4 Hz/second. Thus, the rst corrective stage is to exclude
frequency samples involving unreasonable rates of frequency change. After excluding invalid
samples, the calculated frequency passes through an averaging lter to remove possible remaining
roughness. Final output of the algorithm is the average of three calculated frequencies for the
three phases. Using all three-phase voltage signals for frequency estimation increases accuracy
and robustness of the algorithm. Diagram presented in Figure 1 illustrates different stages of the
algorithm.
DOI: 10.1002/etep
4. WAVELET APPLICATION FOR SIGNAL DE-NOISING
All of the frequency estimation techniques, introduced in this paper, are more or less sensitive to noise.
Therefore, the de-noising block in Figure 1 could signicantly enhance performance of the frequency
estimation algorithm. In this section wavelet transform, an effective tool for signal de-noising is
introduced and its performance is illustrated by an example. Wavelet transform is to some extent
similar to Fourier transform. Meanwhile, wavelet transform has some advantageous features which
could make it more suitable than Fourier transform in various applications. The most important
advantage of wavelet over Fourier transform is the ability of wavelet to maintain time-domain
information of the signal. Using a variable window size, wavelet is capable of detecting both high and
low frequency contents of the signal for any arbitrary instant of time [21].
4.1. Continuous wavelet transform
The continuous wavelet transform is dened as follows [21,22]:
Cscale; position
Z
1
1
f tCscale; position;tdt (19)
In Equation (19), C is the wavelet function which is a function of scale and position. Dependence
on the scale makes the transform sensitive to frequency. On the other hand, dependence on the position
makes it capable of preserving time-domain information.
4.2. Discrete wavelet transform
Calculation of wavelet transform for all possible scales is time consuming and not practical. Instead,
if the scales are based on powers of 2, it turns out that results are much more efcient and as accurate.
An efcient wavelet transform based on this concept was proposed by Mallat [22,23]. In this method
the signal is decomposed into approximation and detail sections. The former contains the important
low frequency content of the signal, while the latter mostly contains high frequency content and noise.
4.3. Signal de-noising
Wavelet transform is widely used in many elds as image processing, signal de-noising and etc.
Wavelet threshold-based de-noising is a well-known application described as follows.
In this method rst the discrete wavelet transform of the noisy signal is performed. Then all detail
coefcients, which are produced during decomposition stage, are passed through a thresholding
function. The most common choices for the thresholding function are hard-thresholding and soft-
thresholding functions [24].
Figure 1. Diagram of the new algorithm.
DOI: 10.1002/etep
The hard-thresholding function chooses all wavelet coefcients that are greater than the given
threshold l and sets the others to zero, as described by Equation (20).
f
h
x
x if x j j l
0 otherwise
8
<
:
(20)
The threshold margin, l, is chosen according to the signal energy and noise variance. If a wavelet
coefcient is greater than l, it is assumed to be a part of the original signal. Otherwise, it is considered
to be due to noise. Discontinuity of f
h
x at x j j l could cause abrupt changes in the de-noised signal.
The soft-thresholding function has a somewhat different rule from the hard-thresholding function.
It shrinks the wavelet coefcients by l toward zero as the following:
f
s
x
xl if x l
0 if x j j < l
x l if x l
8
<
:
(21)
Soft-thresholding function is generally preferred to the hard-thresholding function due to its better
performance and smaller errors.
Once the decomposed signal is de-noised, it should be assembled back into the original signal
without loss of information. It means that, the original signal should be reconstructed from the
coefcients of approximations and details. This process is called reconstruction. While decomposing
a signal involves ltering and down-sampling, the wavelet reconstruction process consists of up-
sampling and ltering. Up-sampling is the process of lengthening a signal component by inserting
zeros between samples [24,25].
Figure 2 depicts an example of noise reduction [22]. In this case, 5-level decomposition is applied.
The original and de-noised signals are depicted in the gure. In this example, the Haar wavelet is used
for signal de-noising.
4.4. Computational burden
4.4.1. Method complexity. According to Equation (18), assuming that N f
n
=2p is once calculated
and stored, the new algorithm requires just 2 subtractions, 2 multiplications and 1 inverse cosine
operations. In addition, some simple averaging modules are used which impose very low burden
on the algorithm. This is, obviously, much simpler than Orthogonal lters, LES and Fourier algorithms,
introduced in Section 2. The proposed method includes a de-noising stage as well. The wavelet
Figure 2. An example of noise reduction using wavelet.
DOI: 10.1002/etep
de-noising stage is implemented using Finite Impulse Response (FIR) lters. These lters do not
impose high burden on the algorithm as they are implemented using few multiplications and additions.
4.4.2. Delay of the de-noising lter. The wavelet de-noising lter introduces a time delay to the
frequency estimation algorithm. In order to enhance the quality of the de-noised signal, it is required
either to increase the number of decomposition stages or to apply a higher level wavelet function. Both
of these approaches would essentially increase the delay. Therefore, a compromise between quality
of the de-noised signal and speed of the ltering process should be made. According to the simulations,
in the case of noisy power system signals, generally three stages of decomposition are sufcient.
Thus, using the db wavelet function, delay of less than ve samples is achievable. For the
32 60 1920 Hz sampling frequency used in this work, the maximum associated delay would be
2.6 milliseconds. Obviously, the benets, achieved from using the de-noising lter, deserves this small
delay.
4.4.3. Delay of the low-pass lter. A second order Butterworth lter has been used as the low
pass lter. This will result in about 2.3 milliseconds delay, according to the phase response of the lter.
Meanwhile, it should be considered that the low-pass lter is not necessarily an exclusive part of the
new method. To implement any frequency estimation algorithm in a digital relay, an analog anti-
aliasing lter must be used. In addition, according to the simulation results of the next section, the
proposed algorithm in contrast to some other methods is inherently immune to harmonic content of
the input signal. Therefore, the 2.3 milliseconds delay of the low pass lter is similar for different
frequency estimation algorithms.
4.4.4. Delay of the frequency calculation block. The frequency calculation block of Figure 1,
according to Equation (15), requires two previous samples of input signal as well as two future
samples. Since the future samples are not accessible instantly, two samples delay is introduced to
the algorithm. For the 1920 Hz sampling frequency used in this work, the delay would be about
1 millisecond.
4.4.5. Delay of the averaging block. The averaging block results in four samples delay, which is
equal to about 2 milliseconds, assuming 1920 Hz sampling frequency.
Therefore, total delay of the algorithm, including the low pass lter, is 7.9 milliseconds. Excluding
the low pass lter delay, the delay would be 5.6 milliseconds. As the frequency-domain based
algorithms such as the LES and Fourier algorithms generally apply a whole-cycle window, the new
algorithm is much faster than the previous methods. In Section 5, performance of the new algorithm
is compared with that of the Fourier method.
5. SIMULATION RESULTS
Various simulation studies were performed and performance of the proposed frequency estimation
algorithm was studied. Some of the obtained results are presented in this section.
5.1. Case study 1: sudden change of voltage magnitude
In this case a sinusoidal signal with constant 60 Hz frequency is applied to the proposed algorithm.
The initial voltage magnitude is 1 per unit. At t 4 seconds, the magnitude suddenly increases to 1.5
per unit, according to Equation (22).
vt
cos2p 60 t 0 < t < 4
1:5 cos2p 60 t t 4
(22)
Figure 3 depicts output of the frequency estimation algorithm which indicates that the algorithm
is not sensitive to abrupt change of voltage magnitude.
DOI: 10.1002/etep
5.2. Case study 2: sudden change of voltage angle
In this case again a sinusoidal signal with constant 60 Hz frequency is applied to the proposed
algorithm. The initial angle of voltage is p/6 radians. At t 4 seconds, the phase angle suddenly
changes to p/3 radians, according to Equation (23).
vt
cos2p 60 t
p
6
0 < t < 4
cos2p 60 t
p
3
t 4
(23)
Output of the frequency estimation algorithm for this case is similar to Figure 3. It indicates that
the algorithm is not sensitive to abrupt change of voltage signal phase angle. It provides the correct
estimation before and after the signal change.
5.3. Case study 3: sudden change of frequency
In this case, a 60 Hz sinusoidal signal is applied to the proposed algorithm. At t 4 seconds, the
frequency suddenly changes to 60.2 Hz, according to Equation (24).
vt
cos2p 60 t 0 < t < 4
cos2p 60:2 t t 4
(24)
Figure 4 shows output of the frequency estimation algorithm. It indicates that the algorithm is
capable of accurately tracking abrupt changes of frequency. The proposed algorithm could correctly
estimate frequency within 0.25 second delay. The delay is due to the omitting unreasonable frequency
samples block of Figure 1. This block does not allow abrupt changes in the estimated frequency.
In this simulation, which is to some extent unrealistic, frequency is suddenly changed by 0.2 Hz.
While in practice this phenomenon could never occur. Therefore, in the case of more practical signals,
the algorithm would be much faster than this case. These cases will also be simulated in this paper
(case studies 79).
5.4. Case study 4: simultaneous change of magnitude and phase angle
In this case the initial voltage magnitude and phase angle are 1 per unit and p/6 radians, respectively.
At t 4 seconds the magnitude suddenly increases to 1.5 per unit and the phase angle changes to p/3
Figure 3. Estimated frequency for sudden change of voltage magnitude.
DOI: 10.1002/etep
radians, according to Equation (25).
vt
cos2p 60 t
p
6
0 < t < 4
1:5 cos2p 60 t
p
3
t 4
(25)
Output of the frequency estimation algorithmfor this case is also similar to Figure 3, which indicates
satisfactory performance of the proposed algorithm.
5.5. Case study 5: application of a noisy signal
In this case a signal contaminated with a random white noise is applied to the frequency estimation
algorithm. In this case study, the signal-to-noise ratio is 20 dB. No sensitivity to noise is observed in
the output of the algorithm, as shown in Figure 5. This immunity to noise is obtained, mainly, by the
omitting unreasonable frequency samples block of Figure 1. This block does not allow any abrupt
change of the estimated frequency.
5.6. Case study 6: application of a harmonic-distorted signal
In this case, a signal distorted with a high amount of third harmonic, is applied to the frequency
estimation algorithm, according to Equation (26).
vt cos2p 60 t 0:1 cos2p 180 t (26)
Figure 5. Estimated frequency for a noisy signal.
Figure 4. Estimated frequency for sudden change of frequency.
DOI: 10.1002/etep
Almost, no sensitivity to harmonic is observed in the output of the algorithm, as shown in Figure 6.
This immunity to harmonics may be described as follows:
The low pass lter block in Figure 1, effectively suppresses the harmonic content of the input
signal.
The algorithm is inherently immune to harmonics due to application of a de-noising stage. In
another simulation case, the same distorted signal as Equation (26) was applied to the algorithm
while the low pass lter was removed and a very similar result was obtained. The estimated
frequency was still 60 Hz.
5.7. Case study 7: 10 per cent generator outage in the IEEE 14 bus benchmark
In this section performance of the new algorithm is compared with performance of the smart Fourier
algorithm, introduced in Section 2.4. The IEEE 14 bus benchmark, as depicted in Figure 7, is used in
this study [26].
Figure 6. Estimated frequency for a harmonic-distorted signal.
Figure 7. Diagram of the IEEE 14 bus benchmark.
DOI: 10.1002/etep
For the event studied in this subsection, at t 3 seconds, 10 per cent of total system generating
capacity is lost. In Figure 8 estimated frequencies of both algorithms are depicted along with the actual
frequency. The actual frequency is obtained, using the angular speed of a nearby generator.
In this case there is no major difference between the estimated frequencies of the newand the Fourier
algorithms. Both of the algorithms are able to track the frequency fast and accurately.
In this case, at t 3 seconds, 30 per cent of the total system generating capacity is lost. The results are
depicted in Figures 9 and 10.
It should be considered that operation of under-frequency load shedding relays generally occurs
between 59.5 and 57.5 Hz. Therefore, proper frequency estimation in this region is essentially
imperative. In Figure 10 this important region is aggrandized and in Figure 11 relative measurement
errors are depicted to get a clear insight into the responses of algorithms. The results of Figures 10
and 11 indicate that in the above-mentioned important region, the new frequency estimation algorithm
is much more accurate than the Fourier method. While the Fourier algorithm results in about 0.15 per
cent error for this event, the error of the new proposed method in most part of the studied region is very
small. The proposed method is able to track the frequency quite accurately. In addition, it provides a
much simpler method, compared with the Fourier algorithm.
Figure 8. Responses of the new and the Fourier algorithms for case study 7.
DOI: 10.1002/etep
Figure 10. Magnication of the important region of Figure 9.
Figure 11. Errors of the new and the Fourier algorithms for case study 8.
DOI: 10.1002/etep
In this event, at t 3seconds, 50 per cent of the IEEE14 bus systemgeneration is lost. This case is simulated
in order to evaluate the performance of new algorithm for a considerably large disturbance. In Figure 12,
estimated frequencies of both algorithms are depicted along with the actual frequency. In Figure 13, the post-
disturbance region of Figure 12 is aggrandized to get a clear insight into the responses of algorithms.
Figure 14 shows relative errors of measurement for both algorithms. According to this gure,
maximum error of the new method is 0.5 per cent, compared with 0.8 per cent of the Fourier algorithm.
The higher error could result in mal-operation of under frequency load shedding relays. It could result
in unnecessary load shed and/or late operation of under frequency relays. The latter might threaten
frequency stability of the system.
The case studies of this paper indicate that the proposed frequency estimation algorithm is accurate
and robust. It is not sensitive to abrupt changes of the signal waveform. Compared with the Fourier
algorithm, the new approach is faster and more accurate.
6. CONCLUSIONS
In this paper a new frequency estimation algorithm was proposed. This new method is a time-domain
based short-window algorithm which is capable of tracking frequency changes within an acceptable
Figure 13. Magnication of the post-disturbance region of Figure 12.
Figure 14. Errors of the new and the Fourier algorithms for case study 9.
DOI: 10.1002/etep
delay. The new algorithm uses some pre-ltering and post-ltering stages to reduce its sensitivity to
harmonics and noises. Signal de-noising before frequency estimation, is one of the key elements of this
new method. Performance of the new algorithm was rst analyzed by applying several abruptly
changing ideal sinusoidal waveforms. Afterwards several generator outage scenarios were simulated
on the IEEE 14 bus benchmark. In this case performance of the new method was compared with the
Fourier algorithm. Results of simulation studies conrm fast and accurate response of the new
algorithm. It is more accurate than the Fourier algorithm, particularly for large disturbances. Moreover,
it provides a much simpler method compared with the Fourier algorithm.
7. LIST OF ABBREVIATIONS
LES Least Error Squares
ZC Zero Crossing
DC Direct Current
ACKNOWLEDGEMENTS
This work was nancially supported by the University of Tehran under the grant number 8101064/1/01.
7. APPENDIX I: FOURIER ALGORITHM
Using Equation (5), xt may be expressed in terms of x
1
and x
2
, phasors of the fundamental and
mth order harmonic components, respectively:
xt
x
1
e
jvt
x
1
e
j vt
2

x
2
e
jmvt
x
2
e
jmvt
2
(A1)
where

indicates complex conjugate.
The discrete form of x(t) and its discrete Fourier transform are presented by Equations (A2) and
(A3), respectively
xk X
1
cosv
k
f
0
N

1
X
2
cosmv
k
f
0
N

2
(A2)
x
^
r

2
N
X
N-1
k0
xk r exp-j
2pk
N
(A3)
Combining Equations (6), (A1), and (A3), after a fairly long manipulations and simplications, x
^
r
is expressed as:
x
^
r
A
r
B
r
C
r
D
r
(A4)
where A
r
, B
r
, C
r
, and D
r
are dened as follows:
A
r

x
1
N
sin
Nu
1
2

sin
u
1
2
exp j
p
f
0
N
Df 2r N1 2f
0
r

(A5)
B
r

x
1
N
sin
Nu
2
2

sin
u
2
2
exp j
p
f
0
N
Df 2r N1 2f
0
r N1

(A6)
DOI: 10.1002/etep
C
r

x
2
N
sin
Nu
3
2

sin
u
3
2
exp j
p
f
0
N
mDf 2r N1 f
0
2mr mNmN 1

(A7)
D
r

x
2
N
sin
Nu
4
2

sin
u
4
2
exp j
p
f
0
N
Df 2mr N1 f
0
2mr mNm N1

(A8)
In Equations (A5)(A8), u
1
, u
2
, u
3
, and u
4
are dened as:
u
1

2pDf
f
0
N
(A9)
u
2

2p2
Df
f
0
N
(A10)
u
3

2pm1
mDf
f
0
N
(A11)
u
4

2pm1
mDf
f
0
N
(A12)
In this stage, four new parameters a, y
^
r
, u
^
r
, and z
^
r
are dened
a exp j
p
f
0
N
2Df 2f
0

(A13)
y
^
r
a
m
x
^
r1
x
^
r
(A14)
u
^
r
a
m
y
^
r1
y
^
r
(A15)
z
^
r
a u
^
r1
u
^
r
(A16)
Using Equations (A14)(A16), after some manipulations, a is calculated as:
a z
^
r2
z
^
r1
a z
^
r1
z
^
r
A
r1
A
r
a (A17)
Expanding Equation (A17), a is calculated using numerical method. Once a is calculated, the
estimated frequency is obtained as follows:
f f
0
Df cos
-1
Rea
f
0
N
2p
(A18)
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DOI: 10.1002/etep

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