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Electric Power Systems Research 78 (2008) 14071415

Frequency measurement of distorted signals using Fourier and

zero crossing techniques

Milenko B. Djuric, Zeljko

R. Djurisic
Faculty of Electrical Engineering, University of Belgrade, Bulevar Kralja Aleksandra 73, 11000 Belgrade, Serbia
Received 3 November 2006; received in revised form 26 August 2007; accepted 3 January 2008
Available online 7 March 2008

Abstract
An approach to the design of a digital algorithm for network frequency estimation is proposed. The algorithm is derived by using the Fourier and
zero crossing techniques. The Fourier method is used for digital filtering and the zero crossing technique is applied to the cosine or sine components
of the original signal, which is usually corrupted by higher harmonics. The algorithm showed a very high level of robustness as well as a high
measurement accuracy over a wide range of frequency changes. It can be used for frequency tracking in power networks when higher harmonics
are present in the voltage or current signals. The theoretical basis and practical implementation of the technique are described. The performance
of the developed algorithm has been verified by the computer simulations, and the field and laboratory tests.
2008 Elsevier B.V. All rights reserved.
Keywords: Protective relaying; Frequency measurement; Algorithm; Fourier method; Zero crossing technique

1. Introduction
The frequency of a power network is an important operational
parameter for the safety, stability, and efficiency of the power
system. Reliable frequency measurement is a prerequisite for
the effective power control, load shedding, load restoration, and
system protection. Therefore, there is a need for a fast and accurate estimation of the frequency of the power network by using
voltage waveforms which may be corrupted by the noise and
higher harmonics.
Several digital methods for frequency measurement have
been proposed over the past few decades. The use of the zero
crossing detection and calculation of the number of cycles within
a predetermined time interval [1] is a simple and well-known
method. The discrete Fourier transform (DFT), least squares
error, and Kalman filter are also known signal processing techniques used for frequency measurement [25]. As shown in [6],
the bilinear form approach seems to be a very efficient method
for both small frequency deviation and off-nominal frequency
estimation. An adaptive algorithm for frequency measurement

Corresponding author.
E-mail addresses: mdjuric@etf.bg.ac.yu (M.B. Djuric),

djurisic@etf.bg.ac.yu (Z.R.
Djurisic).
0378-7796/$ see front matter 2008 Elsevier B.V. All rights reserved.
doi:10.1016/j.epsr.2008.01.008

over a wide range is suggested in [7]. For generator protection, where frequency is to be estimated over a wide range, an
algorithm with a variable window length is proposed in [8].
A Newton-type algorithm has been proposed in [9]. Majority of the algorithms used in modern multifunctional digital
relays for frequency estimation and phasor tracking is based
on DFT. Several of these algorithms have appeared in the literature [1014]. The sensitivities of these algorithms to higher
harmonics in the vicinity of the nominal frequency are relatively
small. As the frequency deviates from the nominal value, a periodic error in the frequency measurement [15], proportional to
the frequency deviation, arises. The frequency deviation also
increases the error caused by the presence of higher harmonics
owing to the so called spectrum leakage. An algorithm based
on DFT and Pronys estimation method has been published in
[16]. The algorithms based on Pronys method [17], Newton
optimization technique [9], and expansion to Taylor series [18]
have been tested in [17]. A comparative analysis showed that the
highest demand for processor time was set by Pronys method
and that it was quite sensitive to the DC component and higher
harmonics. The algorithm based on Newton method was less
sensitive to the DC component and harmonic distortions, but it
was quite hardware demanding. The frequency measuring algorithm using complex computer techniques like generic algorithm
[19] or neural networks [20,21] have been developed recently.

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In this paper, a very simple algorithm of acceptable accuracy

is derived. The Fourier algorithm is used for digital filtering to
extract the cosine and sine parts of the fundamental frequency
component. Then, the zero crossing technique is applied. The
non-recursive Fourier algorithm provides the cosine and sine
components of the fundamental harmonic of the signal. These
components are functions of time and have the same frequency
as the fundamental harmonic of the signal. This fact, well known
from the signal processing theory, is used in this paper. The original signal may be corrupted by the noise and higher harmonics
which may provoke untrue zero crossings. In these cases the
zero crossing technique would give false frequency.
The cosine and sine components of the processed signal contain much less higher harmonics and noise compared to the
original signal. Therefore, the zero crossing technique applied
to the cosine or sine components of the original signal gives
estimates the fundamental frequency of the processed signal
with a high accuracy. The derived algorithm is very simple and
relatively modest resources are required for its implementation.
2. Zero crossing technique applied to orthogonal
components of Fourier series
Let us assume the following observation model of the measured signal x(t):
x(t) = C cos(t + ) + R(t),

(1)

where C is the amplitude of the fundamental harmonic, = 2f

the fundamental harmonic angular frequency, f the fundamental
harmonic frequency, the phase of the fundamental harmonic,
and R(t) the part of the signal which contains higher harmonics
and zero-mean white noise.
If the exact value of is unknown, by using discrete Fourier
series for an assumed angular frequency a parameter Ca can
be estimated as follows:
i+m




i+m


T
T
2

a
a
a
a
xn cos
n j xn sin
n
X(i)
=
m
m
m
n=i

n=i

= A(i) + jB(i),
(2)

where C(i)a = A(i)2 + B(i)2 is the estimated amplitude of the
fundamental harmonic in the i-th data window, a the assumed
angular frequency in the Fourier series (relation (2) gives exact

value of C(i) only for a = ), m the number of samples in the

assumed period of the fundamental harmonic, Ta = 2/a , and
xn the n-th sample of the signal.
The assumed signal is sampled by the frequency
fs = m/Ta = 1/Ts , where Ts is the sampling period. It means that
the assumed frequency of the discrete Fourier series is defined
by the sampling period and number of samples in the assumed
period of the fundamental harmonic. The cosine and sine components, A and B, can be calculated for the i-th data window
according to the following relations:
i+m

A(i) =

2
xn cos(n)
m

i+m

and

B(i) =

n=i

2
xn sin(n),
m
n=i

(3)

where = (a Ta )/m = (2)/m.

If data window sweeps along the measured signal, relations
(3) give sets of points corresponding to the periodic time functions A(t) and B(t). If the fundamental signal frequency is equal
to the assumed fundamental frequency of the Fourier series
(Ts m = Ta = T = 1/f), then A(t) and B(t) are the orthogonal cosine
and sine functions of frequency f. If Ta = T, A(t) and B(t) are not
pure sine and cosine waves but their fundamental frequency is f.
For the purpose of frequency estimation it is enough to consider
the cosine or sine components of the original signal.
Fig. 1 shows a 50 Hz fundamental frequency signal which
contains 100% of the first, 30% of the third and 20% of the fifth
harmonic and its cosine component A.
For a practical application of relations (3) the auxiliary vectors of cosine, sine, and signal samples are required. The length
of all vectors is m. The cosine and sine vectors are
2
[cos(), cos(2), cos(3), . . . cos((m 1)), 1]T ,
m
2
SI = [sin(), sin(2), sin(3), . . . sin((m 1)), 0]T
m
(4)

CO =

The vector of signal samples is

SAM = [x1

x2

x3

...

xm ]T .

(5)

By using the auxiliary vectors CO and SI, the calculation in

(3) requires only multiplication and addition, without calculation of trigonometric functions and without division. After the

Fig. 1. The original signal x(t) and its cosine component (A).

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M.B. Djuric, Z.R.

acquisition of a new sample xnew it is necessary to reorder the

vector SAM as follows x1 = x2 , x2 = x3 , . . ., xm = xnew . Therefore,
the moving data window and samples of the processed signal are
considered as scalars. For each data window the corresponding
points of the cosine (A) and sine (B) components are calculated
according to relation (3).
Thus, the new algorithm for frequency estimation can be
subdivided into the following steps:
1. Acquisition of signal sample xn with a defined sampling
period (Ts = Ta /m = 1/mfa ), where Ta = 1/fa is the assumed
period of the fundamental harmonic of the signal (Ta = 0.02 s
for a 50 Hz signal) and m is the number of samples in the
assumed period Ta = 1/fa .
2. Calculation of cosine (A) or sine (B) components by using
relation (3).
3. The number of samples mA contained in the period of the
orthogonal component is determined.
4. Since fa = 1/mTs , the fundamental frequency of the processed
signal is
f =

1
m
= fa
.
mA Ts
mA

(6)

The above algorithm has some limitations. When the frequencies f and fa are different, the period of the orthogonal component
does not contain an integer number of samples. For a high sampling frequency this fact will cause a small error. But for a low
sampling frequency, the error will be significant. Therefore, it is
necessary to modify the previous algorithm. Fig. 2 shows a zero
crossing of the cosine or sine component.
Let us observe the last positive sample Ap in the previous
period and the first negative sample An in the subsequent period.
In the zero crossing area the cosine and sine functions can be
well represented by a linear function and the following equations
are obtained:
Ap
An
=
K
P

and K + P = Ts .

(7)

From (7) it follows:

P =

An
Ts
Ap A n

and K =

Ap
Ts .
Ap A n

(8)

Now we can modify the third step of the previously defined

algorithm.

Fig. 2. Modification of the zero crossing algorithm.

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When the signal passes through zero, from the positive to the
negative half of the period, the number P (P < Ts ) is indicated as
the FIRST sample of the subsequent period. At the next signal
passing through zero, from the positive to the negative half of the
period, the number K (K < Ts ) is indicated as the LAST sample
of the previous period and number P (P < Ts ) is indicated as
the FIRST sample of the subsequent period. All the samples
between An and Ap which correspond to one period are equal
to Ts . The sum P + K, from the same period can be different
from Ts . Therefore, the number of samples mA of one whole
period of the cosine or sine component can be a fraction. This
modification makes the algorithm to be much more accurate.
The accuracy of the algorithm can also be improved by using
the average number of samples of M periods. One register of
length M is needed where the number of samples of the last M
periods (M may be 410) is stored.
MA = [mA1

mA2

mA3

...

mAM ]T .

(9)

After a new number of samples mnew is obtained, register

(9) has to be reordered. The first sample is discarded and the
new sample is added as follows: mA1 = mA2 , mA2 = mA3 , . . .,
mAM = mAnew . The average number of samples of M periods is
mAs =

mA1 + mA2 + + mAM

,
M

(10)

and the frequency of the processed signal is given by

f = ff

m
mAs

(11)

This process is repeated for each new number of samples in

the period of the processed signal. For a practical application of
the modified zero crossing technique two registers, one of length
m and the other of length M, are needed. Therefore, the presented
algorithm requires very small additional memory resources. It
is very simple, accurate and fast.
3. Selection of the orthogonal component
Do both orthogonal components have the same features?
The answer is no. When the actual signal frequency and
the assumed frequency in the Fourier series are not equal,
the well-known leakage effect is present. If the measured
signal is corrupted by higher harmonics, as a consequence of
leakage effect the higher harmonics also appear in the Fourier
components A and B. Higher harmonics in the orthogonal
components A and B decrease the accuracy of the algorithm.
Fig. 3 shows the amplitude characteristics (or filtering features)
of the orthogonal sine B(t) and cosine A(t) components. They
were formed by using the sine and cosine input signals of 100%
amplitude. The frequency was varied from 0 to 500 Hz in 1 Hz
steps. The assumed frequency in (3) was fa = 50 Hz and the
sampling frequency was fs = 1 kHz.
The presence of higher harmonics affects the Fourier components, but the components keep the fundamental period equal
to the fundamental period of the measured signal, irrespective
of the difference between the assumed and true frequency of
the measured signal. Therefore, the error in measuring the

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M.B. Djuric, Z.R.

Fig. 3. Filtering features of the sine (B) and cosine (A) orthogonal components.

frequency by the combined Fourier and zero crossing method

is exclusively related to the error of the zero crossing method.
The higher harmonics deform the shape of the signal, but do
not affect its periodic property. For this reason the zero crossing
technique does not make any measurement error if an integer
number of sampling periods fits in the fundamental period of
the processed component.
If an integer number of sampling periods does not fit in the
fundamental period of the processed component of the measured signal, the presence of higher harmonics increases the
error of the zero crossing method. The cause of the error is that
the higher harmonics impair the linear approximation of the processed signal (A or B) in the vicinity of the zero crossing point.
The distortions of the measured signal brought in by the higher
harmonics cause an increased variation of the first derivative of
the measured signal within the sampling period which increases
the error of the linear approximation of the signal, A(t) or B(t),
at the zero crossing point, relations (7) and (8).
If the assumed frequency in the Fourier expansion coincides
with the frequency of the fundamental harmonic of the measured signal, both sine and cosine filters completely eliminate the
higher harmonics. If the assumed frequency in the Fourier expansion ff differs from the basic frequency of the measured signal
f, then, based on Fig. 3, some leakage of higher harmonics in
the Fourier components A and B will occur. The sine orthogonal
component A has better filtering feature than the cosine component for high frequency components with f > 50 Hz (50 Hz is the
fundamental frequency). The cosine orthogonal component B

has better filtering feature than the sine component for low frequency components with f < 50 Hz (subharmonic components).
These conclusions are explicitly presented in Fig. 3.
4. Performance evaluation of the algorithm through
simulation
The static, dynamic, and noise tests were performed by using
computer-simulated test signals. A comparison of the accuracy
of the results obtained by applying the cosine or sine components
is given.
4.1. The algorithm sensitivity to the presence of higher
harmonics
For testing the efficiency of the proposed technique for estimating the frequency in the presence of harmonics, the input
signal shown in Fig. 1 is used. The fundamental frequency in
the Fourier series was fa = 50 Hz and the sampling frequency
was fs = 1 kHz. The frequency estimates, obtained by the zero
crossing method applied to the original signal and to the cosine
component of the original signal are shown in Fig. 4.
The true values of the fundamental frequency of the
processed signal were obtained by using the proposed algorithm in the frequency range defined by Shannons sampling
theorem, i.e. for the fh < fs /2 = 500 Hz, where fh is the highest harmonic frequency in the signal. The zero crossing
technique applied directly to the signal in the presence

Fig. 4. Frequency estimates of the 50 Hz signal shown in Fig. 1.

 Djurisic / Electric Power Systems Research 78 (2008) 14071415

M.B. Djuric, Z.R.

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Fig. 5. Frequency estimates of a complex periodic signal containing, in addition to the fundamental, the 2nd, 3rd, 5th, and 9th harmonics. t 3 s: V1 = 100, V2 = 5,
V3 = 20, V5 = 30, and V9 = 15; 3 s < t 6 s: V1 = 10, V2 = 0.5, V3 = 2, V5 = 3, and V9 = 1.5; 6 s < t 9 s: V1 = 10, V2 = 5, V3 = 20, V5 = 30, and V9 = 15.

of harmonics resulted in inaccurate estimates of the frequency.

The test has also been carried out by using a complex periodic
signal for different amplitudes of the fundamental harmonic and
higher harmonics. The complex periodic input signal of the basic
frequency 51 Hz contained, besides fundamental, the 2nd, 3rd,
5th, and 9th harmonics. For t 3 s the amplitudes of the fundamental and higher harmonics were V1 = 100, V2 = 5, V3 = 20,
V5 = 30, and V9 = 15; for 3 s < t 6 s: V1 = 10, V2 = 0.5, V3 = 2,
V5 = 3, and V9 = 1.5; and for 6 s < t 9 s: V1 = 10, V2 = 5, V3 = 20,
V5 = 30, and V9 = 15. The fundamental frequency in the Fourier
series was f0 = 50 Hz and the sampling frequency fs = 2 kHz.
The frequency was measured by processing the sine component of the simulated signal and averaging over three periods.
The results are shown in Fig. 5.
The algorithm was not sensitive to step variations of the
amplitude of the measured signal. The presence of harmonics
increased the measurement error, but even under conditions of
a strongly corrupted signal, having the total harmonic distortion
(THD) factor 97%, the error was less than 6 mHz, i.e. less than
0.012%.
4.2. The algorithm sensitivity to the presence of noise
The effect of the presence of noise in the signal was studied
by estimating the frequency of a sinusoidal 50 Hz test signal

with the superimposed additive white zero-mean Gaussian noise.

Fig. 6 shows the input signal corrupted by white noise and its
cosine component.
Since the cosine component is much closer to a pure sinusoidal wave than the original signal, the proposed algorithm
showed a low sensitivity to white noise. The random noise was
selected in order to obtain controlled values of the signal-tonoise ratio (SNR), defined as:

SNR = 20 log


,

(12)

where C is the amplitude of the fundamental harmonic and is

the standard deviation of noise. In practice the SNR of a voltage
signal obtained from a power system ranges between 50 and
70 dB. At this level of noise, very small errors should be expected
with the proposed technique, as shown in Fig. 7.
Fig. 7 shows the results of frequency estimation when
the signal contains white Gaussian noise characterized by
SNR = 30 dB. Fig. 7a shows the non-averaged estimated frequency (M = 1) and Fig. 7b shows the averaged estimated
frequency over five (M = 5) periods of the signal. The fundamental frequency in the Fourier series was fa = 50 Hz and the
sampling frequency 1 kHz. In general, the higher the sampling
frequency the lower the algorithm sensitivity to the presence of
white noise in the signal (see Fig. 7).

Fig. 6. Test signal x(t) corrupted by noise and its cosine component A(t).

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M.B. Djuric, Z.R.

It may also be concluded that the influence of noise on the measurement accuracy is significant for SNR 40 dB. In a power
system under normal operating conditions the level of noise in
the voltage signals is typically SNR > 50 dB. The maximum frequency measurement error caused by the noise corresponding to
SNR = 70 dB and with the sampling frequency of fs = 1000 Hz is
about 3 mHz without averaging (M = 1) and bellow 0.5 mHz if
the averaging was performed over three (M = 3) periods. These
error estimates due to noise should be added to the systematic
error of the zero crossing method if f = ff , in accordance with
the analyses carried out in the preceding sections. The test indicates a good performance of the algorithm even if the measured
signal is weak when the quantization noise may become significant. The minimum required amplitude of the fundamental
harmonic of the measured signal depends on the A/D converter,
i.e. on the level of noise which appears as the quantization error
in the measured signal.
4.3. Dynamic tests

Fig. 7. Frequency estimates of the signal shown in Fig. 6. (a) Non-averaged

estimated frequency and (b) averaged estimated frequency over five (M = 5)
periods of the signal.

In the subsequent test the influence of the level of noise on the

maximum error of frequency measurement for different averaging intervals is analysed. The sine component of the input signal
of the basic frequency of 50 Hz is processed. The test has been
carried out by varying the SNR and averaging interval (M) and
keeping the sampling frequency at fs = 1 kHz. The results are
shown in Fig. 8.
The test shows that the proposed algorithm is applicable even
in the presence of strong noise in the measured signal. From
Fig. 8 one may conclude that a relatively small number of the
periods of averaging (M) can result in a manifold increase of the
measurement accuracy. For practical applications of the algorithm it is sufficient to do averaging over three (M = 3) periods.

The dynamic behavior of the presented algorithm is tested

by using two computer-simulated signals. In the first test, the
frequency of the sinusoidal signal was changed from 50 to 55 Hz
and from 55 to 48 Hz in a stepwise manner. The results are
shown in Fig. 9a. The algorithm response to the step changes of
the frequency is delayed because the process of averaging was
applied to the last five periods. In the second test the amplitude of
the 50 Hz sinusoidal signal was changed from 100 to 200% and
from 200 to 20% in a stepwise manner. The results are shown in
Fig. 9b. In both tests the assumed frequency in the Fourier series
was fa = 50 Hz, and the sampling frequency fs = 1 kHz.
These dynamic tests showed that the dynamic features of the
proposed algorithm were good. In these tests sinusoidal signals
were assumed, therefore, there was no error in measuring the
frequency at steady state.
4.4. Comparison of the results obtained by using the cosine
and sine orthogonal components
To demonstrate the influence of the choice of the orthogonal
component upon algorithm accuracy the algorithm with applied

Fig. 8. Frequency estimation error as function of SNR without averaging (dark) and with averaging over the last three periods (gray).

 Djurisic / Electric Power Systems Research 78 (2008) 14071415

M.B. Djuric, Z.R.

1413

nents were different and both involved some errors, as shown

in Fig. 12. The results shown in Figs. 11 and 12 were obtained
without averaging over the last several periods (M = 1).
The results shown in Fig. 12 indicate that the algorithm with
sine orthogonal component has slightly better accuracy compared to that of the algorithm with cosine component. The
maximum error with the cosine component was 0.036 Hz or
0.066%, whereas the maximum error of the algorithm with the
sine component was 0.025 Hz, or 0.046%. These errors may be
further reduced by using the average value of the last several
periods (M > 1).
5. The eld and laboratory tests of the algorithm
To verify the presented algorithm in practice, the field and
laboratory tests were performed. For the first test a steady-state
voltage from the local distribution network was used. For the
second, the voltage of a single-phase motor-generator laboratory
set was used.
5.1. Frequency estimation of the steady-state distribution
network voltage

Fig. 9. Frequency estimation: (a) stepwise change of the input signal frequency
and (b) stepwise change of the input signal amplitude.

cosine or sine orthogonal component was tested by using the

distorted signal presented in Fig. 10.
For the signal of fundamental frequency f = 50 Hz (Fig. 10),
assuming the frequency in the Fourier series fa = 50 Hz and
sampling frequency fs = 1 kHz, the results for both orthogonal
components were identical and accurate, as shown in Fig. 11.
For the signal of fundamental frequency f = 55 Hz of the
wave shape approximately equal to the signal of Fig. 10, assuming the frequency in the Fourier series fa = 50 Hz and sampling
frequency fs = 1 kHz, the results for the two orthogonal compo-

The steady-state network voltage samples have been acquired

by using Data Acquisition Digital System (12-bit A/D converter with sampling frequency 5000 Hz) installed at the Faculty
of Electrical Engineering of Belgrade University. The input
signal, shown in Fig. 13a, is obviously corrupted by higher
harmonics. The assumed frequency in the Fourier series was
fa = 50 Hz. The results of the frequency estimation are shown in
Fig. 13b.
The actual frequency of the signal, measured by means of a
Hewlett Packard 5234L Electronic Counter, was 49.877 Hz. The
maximum error that occurred in the frequency estimation was
less than 0.003 Hz.
5.2. Frequency estimation of the voltage from a
motor-generator set
The motor-generator set can run with a variable speed. Therefore, the generator frequency was a function of time. The

Fig. 10. The original signal x(t) and its cosine (A) and sine (B) components.

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M.B. Djuric, Z.R.

Fig. 11. Frequency estimates of the 50 Hz signal shown in Fig. 10 by the present algorithm by using either the sine or cosine orthogonal components.

Fig. 12. Frequency estimates of the 55 Hz signal, like the one of Fig. 10, by the present algorithm by using the sine (B) and cosine (A) orthogonal components.

Fig. 13. (a) Steady-state network voltage waves and (b) frequency estimation.

Fig. 14. (a) Variable generator voltage and (b) frequency estimation during the
voltage and frequency transients.

 Djurisic / Electric Power Systems Research 78 (2008) 14071415

M.B. Djuric, Z.R.

changes of the generator voltage and frequency have been caused

by the step changes of the generator load. The generator voltage
changes versus time are shown in Fig. 14a. The results of the
frequency estimation are shown in Fig. 14b.
The transient test has shown that the proposed algorithm had
a good frequency tracking performance.
6. Conclusion
A new approach to the zero crossing technique for the purpose of frequency estimation is presented. This approach gives a
very simple and robust algorithm of an acceptable accuracy. The
Fourier algorithm is used for digital filtering in order to extract
the cosine and sine parts of the fundamental frequency component. Then, the zero crossing technique is applied to the cosine
or sine components of the signal. The derived algorithm requires
modest computer resources for its implementation. Verification
of the proposed algorithm has been carried out by performing the
computer simulations, and several filed and laboratory tests. All
the tests have shown that the presented algorithm was accurate,
applicable over a wide range, robust, and with a good frequency
tracking performance. The results of the tests confirmed that
the algorithm could be a very useful tool in the power system
protection applications.
Acknowledgement
This research was partially supported by the Ministry of Science and Environmental Protection of Serbia,
Project 223001.
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