1026 IEEE Transactions on Power Systems, Vol. 13, No.

3, August 1998

A NEW DIGITAL FILTER FOR PHASOR COMPUTATION

PART I: THEORY

Josk A. de la 0 Hector J. Altuve Ismael Diaz

Universidad Autonoma de Nuevo Leon
Facultad de Ingenieria Mecanica y Electrica
Programa Doctoral en Ingenieria Electrica
Apdo. Postal 113-F, Ciudad Universitaria
66450 San Nicolas de 10s Garza, Nuevo Leon, M6xico
Phone: (52-8) 329-4020 ext. 5773 Fax: (52-8) 376-4514, 332-0904
email: jdelao@ccr.dsi.uanl.mx

Abstract: A new pair of orthogonal filters for phasor
computation is presented. They have excellent time-frequency
characteristics for fault location and measurement, Their
impulse responses are obtained applying the inverse Fourier
transform to a single-lobe function with a strong stopband.
From this process, a new window emerges. Without side lobes, it
overcomes the temporal barriers imposed by the rectangular
window, implicit in digital Fourier filtering. Its length is not
restricted to a multiple of one cycle, and it can be adjusted to
cover totally the available samples of the fault, extracting the
fundamental component with growing precision. On the other
hand, its sampling frequency can be reduced to twice the
fundamental frequency. In particular, at this minimum
sampling frequency, the digital Cosine filter rejects the even
harmonics and the aperiodic component.
1. INTRODUCTION
Today’s digital relays are microprocessor-based systems
receiving information in real time from the electric power
system. They sample the input waveforms to perform
functions such as protection, control, measurement and
monitoring. For these functions, digital filtering algorithms
are required to estimate some of the components of the
signals.
The protection function normally requires the extraction
of the fundamental component in a very short time, about one
PE-176-PWRS-16-09-1997 A paper recommended and approved by
the IEEE Power System Engineering Committee of the IEEE Power
Engineering Society for publication in the IEEE Transactions on Power
Systems. Manuscript submitted May 27, 1997; made available for
printing September 30, 1997.
period of the fundamental frequency [1,2,3]. As a
consequence of the fault-induced transient, the input signals
can be highly contaminated with an aperiodic exponential
component, high-frequency damped oscillations, and other
components [1,2,3]. The estimation of the fundamental
component is also required in the fault location function, but
in this case the information is accumulated during the total
time of the fault, about 3 to 5 cycles [2,3]. Finally, the
measurement function of voltage and current, contaminated
with harmonics generated by the non-linearities of the
elements of the power system, requires the estimation of rms
values of the fundamental frequency [4].
In the last three decades, the development of digital
filtering algorithms for power system protection has received
considerable attention from researchers. In most of the
modem digital relays Fourier filtering algorithms with a
rectangular window of one cycle are used. Up to now,
algorithms based in the solution of the differential equation
of the protected line for direct estimation of its parameters
[2,3], have had limited application.
As a rule, the same filtering algorithm is used for the
protection and fault location functions, changing only the
window length. However, these functions have different
requirements for accuracy and processing speed. It could
then be convenient to use different filtering algorithms for
both functions.
In this paper, a new digital filtering algorithm is
proposed. This technique extracts accurately the fundamental
component of fault voltage and current signals, using a
window with a scalar length and practically without side
lobes. The first section presents the characteristics and
limitations of the digital Fourier filtering. This is the starting
point of our research. In the next section, the time-frequency
characteristics of the new digital filters are introduced. From
this process, an interesting window is obtained. In contrast
with the classical windows proposed in the literature, its side
lobes are negligible. In consequence, Fourier filters using this
window will have a tight stop band outside the fundamental
component.

0885-8950/98/$10.00 0 1997 IEEE

 1027

11. CONVENTIONAL FILTERING

FOLJER
Conventional Fourier filters are obtained restricting the
sine and cosine waves tlo a domain defined by a rectangular
window of duration T. Defining this window as
lobe width is . As it is known, thls width is inversely
proportional to the length of the window. Then, if the
duration of the window is c cycles, vidh c being a positive
integer, the side-lobe width is fdc. It means to have c side
lobes in the inter-harmonic'intervals.

v(t) = I-. 1
T
---<t<-
T
2 k-.
According to Eqs. (4) and ( 5 ) , the frequency responses of
the orthogonal filters combine two replicas of Vcr) tuned at
If zero gain is required at the harmonics, the window
length must be a multiple of one cycle. Indeed, for a window
shorter than one cycle, the width of the main lobe of Vcr) is
then the impulse response of the pair of Fourier filters arc:: grater than 2f0, which is the separation between the replicas,
and no zero gain at the harmonics would result in the
hc ( t ) = v ( t ) coJ(27fol) (2) superposition. On the other hand, the zero crossings of the
replicas coincide with harmonics only when the window
h,(t) = jv(f)nn(2~f~t) (3) length is a multiple of one cycle.
where fo is the fundamental frequency. The corresponciing A final comment about the superpositions of Figs. 1 and
frequency responses are : 2. Apparently, the maximum gain is one, and it coincides
with the fundamental frequency. A close- up of the right
1
[ soI]
'-I
owv
H , (Si = - V ( / + So + V ( f - (4)
2
0.5
H, ( fi = -[2
1
-V(f + soI + L7(f- soI]
H4u)
where the function:
V ( f ) = Tsinc(xTf) (6) -5.3
-8-6 -4 -2 0 2 4 6 8

is the Cardinal sine function of amplitude T, resulting from U

the Fourier transform of the rectangular window v(t). Fig. 1. Frequency response of the Cosine filter
It should be noted that HCfJ and H,m are the
convolutions of the Fourier Transform of the window VQI
with those of the cosine and imaginary sine waves
respectively.
The functions Hccf) and H&I, for a rectangular window
of one cycle (T=l/Jo), are shown in Figs. 1 and 2. Their gains
were adjusted to 1 a t p h ,and the frequency was normalized
byfo ( ~ = f l f o ) .
It can be observed that the filters do eliminate the L ~~

-8-6 -4 -2 0 2 4 6 8
harmonics, because their gain is zero at frequencies being
U
multiples of the fundamental. However, both filters exhibit
leakage in the inter-harmonic bands. This leakage is due to Fig. 2 Frequency response of the Sine filter.
the side lobes of the rectangular window.
The Fourier filter is composed by the former ones:
h, ( t ) = h,(t) + hs(t) = ('7)

= v(t){cos(2?2f0t)+ jsin( 27fot))

and its Fourier transform is:

HF(f I = H,(f) + H , ( f ) = V(f - f o ) (8)

-0.S
-8-6 -4 -2 0 2 4 6 8
This replicates Vy>at fo,,as it appears in Fig. 3 for a full-
U
cycle rectangular window, The main-lobe width doubles that
of the side lobes. For a window length of one cycle, the side- Fig 3 Frequency response of the Fourier filter.
1028
1.2
1.1
W
-
u )
1
W u )
......
0.9
0.8
0.5 0.75 1 1.25 1.5
U The purpose of our research was to enhance the
Fig. 4. Close-up of the top of the Cosine and Sine filters
summit of both filters is shown in Fig. 4. It can be observed
that their maximum gains are really greater than one, and
they are not located at the fundamental frequency ( u = l ) .
T h s asymmetric gain will introduce a distortion in the
estimation of the phasor when the electrical power system
deviates from the fundamental frequency. In this case,
instead of having a circular trajectory, the phasor will follow
an elliptic one.
Finally, let us consider the modification of the frequency
characteristics introduced by the sampling process inherent
to digital filters. The sequences of the impulse responses are
obtained sampling the functions in Eq. ( 2 ) and (3):
h,[kl = h,(kc); h, [kl = h , ( h ) (9)
where r is the sampling period. The Fourier transform of a
sequence h [ k ] , obtained by sampling a continuous function
h(t),is:
1 *
c H(f-;)
g ( f ) = -n=-cc
r
(10)
where H ( f ) is the Fourier transform of h(t), and E ( f ) is a
repetitive function in frequency, formed by superposed
replicas of Hen, with amplitude 1/r, and tuned at multiples
of the sampling frequency l/z.
As we can see in Figs. 1 and 2, the frequency response of
the continuous filters extends even ahead of the eighth
harmonic. If interference between replicas is to be avoided,
the sampling frequency needs then to be greater than at least
16 fo. Or, if the overlapping of the replicas is tolerated,
providing that zero gain at harmonics is preserved, the
sampling frequency needs to be a multiple of the
fundamental. That means that the sampling period needs to
be a fraction of one cycle. Obviously, the sampling frequency
of the digital cofilters must match that of the signal to be
filtered, whch in turn depends on its spectral content. The
former condition considers only the internal characteristics
of the filters.
111. NEWPROPOSED
FILTER
To obtain the new filters, a functional approximation of
digital filters selective in frequency was followed. The details
of t h s methodology are presented in [5,6,7,8]. Basically, it
consists in defining a family of functions, which determine
in frequency the borders of the approximation to an ideal
lowpass filter. Each family specifies a type of filter.
Buttenvorth, Chebyshev and Elliptical filters arise from this
method.
properties of the conventional Fourier filter. In particular,
the elimination of the lateral inter-harmonic leakage, and the
correction of the asymmetric summits. Both goals are
fulfilled by the following family of functions:
J
2<-<0
fo
where S is a nonnegative real number that controls the wide
of the lobe. The higher its value , the narrower the lobe. This
is defined with the maximum possible width, so that the
replicas forming the filters do not overlap. Fig. 5 shows
HocT, s), for 6=2; t h ~ sis the first value for whch the lobe
starts and ends with zero slope. Without side lobes, the
symmetrical convexity at f=fo of the Sine and Cosine filters
will be preserved.
In contrast with the classical filters proposed in the
literature[%],and listed above, the family of functions in (1 1)
does not have ripples in the stop bands. It is then convenient
to give a name to the filters issued from this family of
functions. The name single-lobe filter is proposed for this
new family of filters. In the following development, where
the equations are solved only for 6=2, the function H,cfT2)
will be denoted H,v) for simplicity.
The impulse response of H & ) is
7r
h, ( t ) = sin2 (-f)exp(j2ntf)df
0
2fo
Integrating, we obtain the Eq. (13):
+J
4nf0t + 2 n
+J
413fv' - 2 %
 1024,
The Cardinal sine and cosine functions can be defined as:
I I I l / ) , l I
sen(x ) 1 - cos(x) (14)
sinc(x) = ~ ; COSC(X)
=
X X

where sinc(O)=l and cosc(O)=O. Separating real and

imagmaq parts, Eqs. (15 ) and (16) are obtained:

[slnc(4?fot) 1 -4-3 -2 -1 0 1 2 3 4
U

Fie. 5 . Frequency response of the filter Ho(u.2).

1
h,(tJ = i f 0

12. 1
--(cosc(4?fof + 2.n) + cosc(4nfot - 2 4
1 (I6’

These are the continuous impulse responses of the

orthogonal filters proposed. These functions can be
simplified as follows:
h, ( t ) = o ( ~ ) c o s ( ~ x ~ ~ ~ ) (17) -2-1.5 - 1 -0.5 0 05 1 1.5 2
ut
hi ( t ) = j o ( t ) sen(2xf0t) (18) Fig 6. Window o(t).

where o(t) is a new window given by:

rslnc(2 1

Comparing the Eqs. (17) and (18) with ( 2 ) and (3), it can
be concluded that the filters are obtained applying a new
window to the cosine and sine functions. Fig. 6 shows this
-Ot
-1201 I
I
I I I I I\ , I
-2-1.5 -1 Q . 5 0 0.5 1 1.5 2
window in normalized time (ut=&). The zero crossings of
U
o(t) are at lfo, 1.5fo, 2/9b, 2.5f0, etc. After elfo,
the values
are very small, and for practical applications, the window Fig. 7. Fourier transform of the single-lobe window.
length can be limited to only two cycles, without impairing The Fourier transforms of h,(t) and h,(t) are equivalent to
their main frequency characteristic. T h s length is the those of the h,(q and h,(t) in Eqs (4) and ( 5 ) respectively.
minimum one, because the lobe was defined as the widest. They are obtained changing V f l by Om, where O m is the
On the other hand, Fig. 7 shows the spectral Fourier transform of o(t). Figs. 8 and 9 show these
characteristic of the window, with the gain &splayed in dB transforms for a two-cycle window. Widening the window
(2010g10(0y)) as a function of the normalized frequency narrows the lobe, and improves the precision of phasor
(u=f/fo).It can be observed the absence of side lobes. This estimation. As the new filters do not have side lobes, its
property is very attractive. In contrast with the windows duration is not anymore conditioned to be a multiple of one
proposed in the literature 181, such as the rectangular, cycle. We can then adjust it to cover the whole signal
Hamming or Hanning windows, t h ~ sone does not have side information available, thus obtaining the phasor with the
lobes. Furthermore, absolute values are needless, because the best precision.
new window is nonnegative. This is an interesting property The dlscrete version of the function o(t) (Eq.(19)) is
for spectral estimation. Given the importance of these
obtained by sampling it with a sampling period T, and
characteristics, the name of single-lobe window of order two
multiplying by the factor z
is proposed.
o[k] = 7 o(kr),
1030
so that its Fourier transform
W 0.5 I I I I\ I
c O(+/
&f)= n=-W z
(21)

will be an infinite repetition of replicas of O m , tuned at

multiples of the sampling frequency (UT).
Taking M samples over the span of the window, we
obtain:
r 4n 1 -05
-4 -2 0
I
2 4
4 U
o [ k ]= -
4n 4n Fig 9 Frequency response of the smgle-lobe filter Sme type
k + n) + sine(- k -
M M

for -A4 / 2 2 k I hi / 2,and o[k]=Ooutside this interval.

CONSIDERATIONS
IV. PRACTICAL
The digital single-lobe filters are obtained from (17) and
As we can see in Fig. 10, the coefficients of the digital
(lS), and represented in Fig. 10, for .r=1/(16&). The Fourier
filters cover two cycles. In protection, where a short window
transforms of these sequences are formed by the
must be applied (virtually one cycle), it is impossible to
superposition of an infinity of replicas of Figs. 8 and 9 tuned
obtain single-lobe filters with zero gain in the harmonics
at multiples of the sampling frequency.
(includingf=O). Shortening the window widens the lobe, and
As we can see, without side lobes, the replicas can be the replicas of the Cosine and Sine filters will overlap.
brought near, using lower sampling frequencies, without Fig. 11 displays the frequency response of the Cosine
side-lobe interference between them. Interesting digital (dotted line) and Sine (continuous line) filters with a full-
filters result from this. They were not possible before, cycle single-lobe window. It can be observed the infiltration
because of the side-lobe interference of the classical of the dc component and the second harmonic. Then, these
windows. An interesting case is the digital Cosine filter at filters cannot be used directly for protection.
the minimum sampling frequency 2h. This filter has only
three coeficients, because, at this sampling frequency, most
of its samples coincide with the zero crossings of the
window. To match the frequency of the filtered signal its 0.1
inter-sample intervals must be padded with zeros. This filter ?(k)
rejects fully the even harmonics and strongly attenuates the 0
hi( k)
aperiodic component, and its operation takes only one cycle. 0
-0 1
From its output, which contains only odd harmonics, the
fundamental component can be obtained filtering the -02
remaining non-fundamental harmonics. -16 -8 0 8 16
k

Fig 10 Coefficients of the single-lobe filters Cosine (f) and Sine

(0)

0.375

Hpcu) 0.25

0.125

0 -1.5
-4 -2 0 2 4 -4-3 -2 -1 0 1 2 3 4
U
U

Fig. 8 Frequency response of the single-lobe filter Cosine type. Fig 11. Frequent), response of the one-cycle single-lobe filters

V. CONCLUSIONS
The filters proposed in this paper are superior to the
conventional Fourier filiters in that they are free from side
lobes. However, their minimum possible length is greater
than that of the filters having a rectangular window.
In measurement and fault location functions, the
available data correspond to several cycles. Then the single-
lobe filters offer good time-frequency characteristics for the
extraction of the harmonics and the aperiodx component.
The best estimation of the phasor is obtained elongating the
window to cover the whole information on the event.
Finally, a bank of filters could be constructed by tuning
each filter at the corresponding frequency for harmonic
analysis.
VI. REFERENCES
E.O. Schweitzer and Daqing Hou, "Filtering for protective relays," 1gh
Annual Western Protective Relay Conference, Spokane, Washington,
October 1992.
A.G. Phadke, J.S. Thorp, Computer Relaying for Power Systems, New
York John Wiley and Sons, 1788.
Microprocessor Relays and Protection Systems, IEEE Tutorial Course,
Publication Number 88EH0269- 1-PWR.
M. Kezunovic, E. Soljanin, B. Perunicic and S. Levi, "New approach to
the design of digital algorithms for electric power measurements, IEEE
"
Transactions on Power Delivery, vol. 6, No. 2, April 1771, pp. 116-
523.
E.A. Guillemin, Synthesis ofpassive Networks, New York: John Wiley
and Sons, 1757.
R.W. Daniels, Approximation Methods for Electronic Filter Design,
New York: Mc Graw Hill, 1774.
H. Lam, Analog and Digital Filters: Design and Realization,
Engleewood Cliffs, NJ: Prentice Hall, 1979.
H V. Oppenheim and R.W. Schafer, Discrete Time Signal Processing,
New Jersey: Prentice Hall, 1987.
1031
VII. BIOGRAPHIES
Jose Antonio de la 0 Serna was born in San Pedro,
Coahuila, MBxico in 1753 He received the Ph D.
degree from the Ecole Nationale Suphieure de
TBIBcommunications, Pans, France, in 1782 Between
1982 and 1786 he was a professor in the Instituto
Tecnolbgico y de Estudios Superiores de Monterrey,
Mexico In 1987, he joined the Ph. D Program in
Electrical Engineering at the Universidad Autonoma de
Nuevo Leon, where he was a member of the Doctoral
Committee. From 1988 to 1993 he joined the Electrical
Department of the Polytechnical School in Yaoundb,
Cameroon In 1974 he retumed to the Ph D Program in
Electrical Engineering at the UANL, where he is
member of the staff
Hector J. Altuve Ferrer (SM '95) was bom in 1947
in Cuba. He graduated in 1767 as an Electrical Engineer
at the Universidad Central de las Villas, Cuba He
received his Ph D. degree in 1781 from Kiev
Polytechnic Institute, USSR From 1767 to 1973 he was
a professor of the Electrical Engineering Faculty of
Universidad Central de las Villas. From 1970 to 1993
he was also a visiting professor of the Ph. D Program in
Electrical Engineering at the Universidad Autonoma de
Nuevo Leon Since 1773, Dr. Altuve is a member of the
Ph D Program staff His research area is power system
protection and control
Ismael Dim Verduzco was bom in Zacapu,
Michoacin, Mexico, in 1969. He graduated as an
Industrial Engineer specialized in Electricity at Instituto
Tecnologico de Morelia in 1792 He is at present a
student of the Ph D Program in Electrical Engineering
at the Universidad Autonoma de Nuevo Leon His
research area is power system protection