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3288 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 58, NO.

9, SEPTEMBER 2009

An Improved Taylor Method for Frequency


Measurement in Power Systems
Zoran Salcic, Senior Member, IEEE, Sing Kiong Nguang, Senior Member, IEEE, and Yanzhen Wu

Abstract—The frequency of a power system is an important [12]–[14]. Recently, a symmetry detection technique has been
operational parameter for the safety, stability, and efficiency of proposed [15].
the power system. Accurate and fast measurement of frequency Among the aforementioned methods, the Taylor algorithm
is a goal required in many applications, for example, for load
shedding. An improved Taylor approximation method (ITAM) al- is known to be a simple but efficient algorithm, with low
gorithm is proposed, and a comparison of the new algorithm with computation requirements and being suitable for hardware
the conventional Taylor approximation method (TAM) algorithm implementation. However, there are two factors limiting its
is presented in this paper under the assumption of the presence of a further application. One is the sampling rate. The algorithm
direct-current (dc) offset component, higher order harmonics, and is restricted to the sampling rate at a specific value. When
random noise in the power system signal. The proposed ITAM has
the characteristics of low computational and simple hardware re- increasing the sampling rate, larger measurement errors will
quirements, as well as more accurate measurement performance. be obtained [12], [13]. The other one is that the algorithm is
Another key feature of the algorithm is a good tracing ability for a bit rough. The proposed Taylor algorithm in [12] and [14]
the frequency change and measurement of rate of change of the only uses the first two terms of Taylor series expansion, so
frequency of a power system. larger measurement errors are also obtained when the system
Index Terms—Field-programmable gate array (FPGA), fre- frequency has a larger deviation from the nominal frequency.
quency measurement, power system, real time. This means that this method is not capable of providing mea-
surements over a wide range of frequencies. It is mainly limited
I. I NTRODUCTION to ±5 Hz of the nominal frequency of a power system.
This paper presents a new frequency measurement algorithm
T HE FREQUENCY of a power system is an important op-
erational parameter for its safety, stability, and efficiency.
Frequency reflects the dynamic energy balance between the
based on the Taylor method suitable for digital implementation,
which we call the improved Taylor approximation method
load and the generated power. It is essential to maintain the (ITAM), that provides accurate and fast measurement results
frequency at its nominal value (or as close to it as possible), over a wide range of frequencies where the input signals contain
which is 50 Hz in New Zealand and many other countries and significant levels of harmonics, dc offset, and random noise.
60 Hz in the U.S. and Canada. Simulation results confirm the high reliability and accuracy of
The deviation of the frequency from its nominal value could the proposed algorithm. The ITAM algorithm is suitable for
lead to harmful consequences to the equipment in the power digital implementation. A prototype of a real-time frequency
system. Therefore, reliable frequency measurement is a pre- measurement system (FMS) based on field-programmable gate
requisite for effective power control, load shedding, load resto- arrays (FPGAs) is then implemented.
ration, and generator protection.
Prior to 1979, the frequency measuring devices used were II. P ROPOSED A LGORITHM B ASED ON THE T AYLOR
based on electromechanical and analog solid-state devices, and M ETHOD FOR A P URE S INUSOID
they were very reliable [1]. The watershed changes in such
Assume that a power system voltage signal is a pure sinusoid
devices were introduced by microprocessor technology. The
sampled at intervals of length T
use of digital techniques has been well developed since then.
A large class of frequency measurement algorithms for power vk = V sin(2π f kT + θ) (1)
systems is based upon extracting information about the voltage
waveform for making measurement decisions, including the where V is the peak value of the voltage, θ is a phase angle, f
well-known zero-crossing technique [2], discrete Fourier is the frequency, and k is the sample number.
transform [3], [4], least-square-error technique [5], [6], Kalman Consider the expression
filtering [7], [8], phasor demodulation [9], [10], Newton
algorithm [11], Prony algorithm [3], and Taylor method
2
Dk = vk−1 − vk vk−2 . (2)
By substituting (1) into (2), we obtain
Manuscript received February 13, 2006; revised October 15, 2008. First
published July 21, 2009; current version published August 12, 2009.
The authors are with the Department of Electrical and Computer Engineer-
Dk = V 2 sin2 (2π f T ). (3)
ing, University of Auckland, 1142 Auckland, New Zealand (e-mail: z.salcic@
aucland.ac.nz). That is
Color versions of one or more of the figures in this paper are available online  
1 v 2
at http://ieeexplore.ieee.org. sin(2π f T ) = k−1 − vk vk−2 . (4)
Digital Object Identifier 10.1109/TIM.2009.2022377 V
0018-9456/$26.00 © 2009 IEEE
SALCIC et al.: IMPROVED TAYLOR METHOD FOR FREQUENCY MEASUREMENT IN POWER SYSTEMS 3289

For simplification, let frequency calculation step. Equation (18) provides the basis for
a simple algorithm to estimate the power system frequency.
x = sin(2π f T ) (5)
III. A LGORITHM I MMUNITY TO DC O FFSET ,
then
H ARMONICS , AND N OISE
1
f= arcsin(x). (6) Practical signals in a power system are not pure sinusoidal
2π T waveforms. They are contaminated by dc offset, harmonics,
Suppose the nominal frequency of the power system is f0 ; and random noise since there are many nonlinear devices in the
then, using Taylor series expansion for (6) at f = f0 , we have power networks.

∞
1 (n)
f = f0 + f0 (x − x0 )n (7) A. Power System Model
n=1
n!
It should be noted that there are mainly odd harmonics
where present in a power system. In practice, it is enough to consider
only the third- and fifth-order harmonics for the power system
x0 = sin(2πf T )|f =f0 = sin(2πf0 T ) (8) model. Therefore, the power system model can be simplified as

(n) dn [arcsin(x)]  v(t) = V0 + V sin(ω0 t + θ) + V3 sin(3ω0 t + θ3 )
f0 =  . (9)
dxn x=x0
+ V5 sin(5ω0 t + θ5 ) + rand(t) (19)
Specifically
where
 −1 V0 value of the dc component;

f0 = 1 − x02 (10) f0 fundamental frequency;
 −3 ω0 fundamental angular frequency (equal to
 2πf0 );
f0 = x0 1 − x0 2 (11)
 −5 V , V3 , and V5 amplitudes of the fundamental signal, third-

   order harmonic, and fifth-order harmonic,
f0 = 1 + 2x02
1 − x0
2 (12) respectively;
 −7 θ, θ3, and θ5 phase angles of the fundamental signal,
(4)  
f0 = 9x + 6x0 3
1 − x02 (13) third-order harmonic, and fifth-order har-
monic, respectively, in radians;
and so forth. t time in seconds;
Let F1 , F2 , F3 , and F4 be the result of the first-, second-, rand random noise with zero-mean Gaussian
third-, and fourth-order derivatives of the frequency multiplied distribution.
by the constant 1/2πT , respectively. Then, we obtain The random noise standard deviation is selected to have a
prescribed value of the SNR, defined as
1
F1 = · f0 (14) V
2πT SNR = 20 log √ (dB) (20)
1 2σ
F2 = · f0 = x0 F13 (15)
2πT where σ is the noise standard deviation.
1  
F3 = · f0 = 1 + 2x20 F15 (16)
2πT B. Influence of an Impure Sinusoid Input Signal and Prefilter
1 (4)  
F4 = · f0 = 9x + 6x30 F17 . (17) For simplification, assume that the input signal has only
2πT
dc offset and a third-harmonic component and is sampled at
Obviously, the more terms we use, the less calculation errors intervals of length Ts
will be obtained in terms of (9). In this algorithm, four to five
terms are recommended, and the first to fourth derivatives are vk = V0 + V sin(ω Ts k + θ) + V3 sin(3ωTs k + θ3 ) (21)
given in (10)–(13). For the convenience of implementation, the where
frequency can be expressed by k sample number;
f system frequency;
f≈f0 +(x−x0 )(F1+(x−x0 )(F2 +(x−x0 )(F3 +(x−x0 )F4))).
ω angular frequency (equal to 2πf );
(18)
V0 value of the dc component;
As can be seen, for a pure sinusoid input signal, only three V amplitude of the fundamental signal;
consecutive samples are needed for the frequency estimations. θ phase angle of the fundamental signal in radians;
As (4) requires either a constant or known value of peak voltage V3 amplitude of the third harmonic;
V , it is important that this value is known in advance for each θ3 phase angle of the third harmonic in radians.
3290 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 58, NO. 9, SEPTEMBER 2009

Fig. 2. Comparison between TAM and ITAM algorithm.

Fig. 1. Frequency response of the bandpass FIR filter.

According to (2), Dk then becomes

Dk = V 2 sin2 (ω Ts 
) + V32 
sin2 (3ω Ts )
ω Ts
+ 4V0 V sin2 sin (ω Ts (k − 1) + θ)
2
 
3ω Ts
+ 4V0 V3 sin2 sin (3ω Ts (k − 1) + θ3 )
2
+ 2V V3 sin2 (2ω Ts ) cos (2ω Ts (k−1)−θ+θ3 )
−2V V3 sin2 (ω Ts ) cos (4ω Ts (k−1)+θ+θ3 ) . (22)
As can be seen from (22), two kinds of source are contribut-
ing to the errors: the constant term and the periodic terms. Thus,
a finite-impulse-response (FIR) bandpass filter for filtering out
those unwanted components in the system signal is used, and Fig. 3. Comparison result of ITAM with TAM. Sampling rate f s is 2 kHz.
the running average over N samples, as shown in the following
estimated frequency result in the ITAM algorithm, while the
equation, is employed to reduce the contribution of the periodic
TAM algorithm has the opposite tendency, i.e., increasing the
terms:
sampling rate results in larger errors.
1 
N The results of the comparison of the proposed ITAM algo-
f¯k = fi (23) rithm with the TAM algorithm are also shown in Figs. 3 and 4,
N i=1
where the system signal consists of dc offset, third- to eleventh-
where N equals the number of input signal samples in one odd-order higher harmonics, and random noise.
period of a power signal and, in the hardware implementation Fig. 3 demonstrates the comparison results of the two algo-
presented in Section IV, was 40. rithms when using the same FIR bandpass filter shown in Fig. 1
Fig. 1 shows the frequency response of the bandpass fifty- and the sampling rate is f s = 2 kHz. The input signal in this
sixth-order FIR filter in the proposed algorithm. case is

v(t) = 0.2 + sin(ω t + 2.5) + 0.3 sin(ω t + 2.5)


IV. S IMULATION R ESULTS
+ 0.3 sin(3ω t + 0.35) + 0.2 sin(5ω t + 1.3)
As an ITAM, the proposed algorithm was investigated in + 0.15 sin(7ω t + 0.3) + 0.1 sin(9ω t + 0.1)
Matlab and compared with the Taylor approximation method + 0.08 sin(11ω t+1.23)+rand(SNR = 21 dB). (24)
(TAM). Fig. 2 shows the comparison results between the ITAM
algorithm and the TAM algorithm in terms of different sam- Obviously, the smaller errors of the ITAM algorithm over a
pling rates over a frequency range from 15 to 75 Hz under wide frequency range from 35 to 70 Hz are of interest for us. As
the assumption of a pure sinusoidal power system voltage a higher sampling rate used in the TAM algorithm will result in
signal. At the same sampling rate condition, the estimated larger measurement errors, another simulation is done that uses
frequency of the ITAM algorithm is far more accurate than that the filter mentioned in [12] and [13] for TAM with a sampling
of the TAM algorithm over a wide range of frequencies. On rate of only 400 Hz, while the proposed ITAM algorithm uses
the other hand, the increase of the sampling rate improves the the FIR bandpass filter shown in Fig. 1 with the sampling rate
SALCIC et al.: IMPROVED TAYLOR METHOD FOR FREQUENCY MEASUREMENT IN POWER SYSTEMS 3291

TABLE I
COMPARISON OF COMPUTATIONAL COMPLEXITY
FOR T HREE A LGORITHMS

a comparison of the computational complexity and timing


requirements of the ITAM algorithm with those of the TAM
algorithm [13] and the reference-point (RP) algorithm [15]. An
FPGA was used to implement the algorithm and the whole
FMS, while the other two methods required the use of a proces-
sor to implement a part of the corresponding algorithms in
Fig. 4. Comparison result of ITAM (f s = 2 kHz) with TAM (f s = 400 Hz) software.
where the input signal is the same as in Fig. 3. The overall design of an FPGA-implemented FMS is pre-
sented in Fig. 6. The system comprises three modules: the
FIR filter, the ITAM algorithm block, and the result storage,
display, and serial transmission block. Fixed-point number rep-
resentation and arithmetic has been used to reduce the resource
requirements for the system.
The FIR filter module receives 16-bit input sample values,
generates a 32-bit-wide data output, and contains a multiply and
accumulate unit, memory, which stores Matlab precomputed
FIR coefficients, and a control unit. Small embedded memory
blocks (EABs) have been used in the design to store input
samples, the intermediate results of computation, and constants.
In the rest of this section, we present in more detail the
implementation of the ITAM algorithm block, which is split
into a cascade of three subblocks.
In the ITAM algorithm module, the ITAM algorithm imple-
mentation is split into three units. They are called the ITAM_1,
ITAM_2, and ITAM_3 units, respectively. These three units are
Fig. 5. Estimated frequency when the fundamental frequency of the input connected in cascade.
signal is changed stepwise. Fig. 7 represents the flowchart for the frequency measure-
ment algorithm, which reflects its hardware design. To ensure
kept at 2 kHz. The comparison results are shown in Fig. 4.
that there is no overflow in each computation, data in all steps
Although the measurement results show an improvement for the
of the frequency algorithm were scaled by 211 . On the other
TAM algorithm, its estimated errors are still higher than those
hand, due to the multiplication operation t, the coefficients F1 ,
of ITAM, even in a narrow frequency range, e.g., ± 10 Hz of
F2 , and F3 must be multiplied by 2, 22 , and 23 , respectively,
the nominal frequency of the power system f0 = 50 Hz (from
shown in Table II.
40 Hz on the lower boundary to 60 Hz on the upper boundary).
ITAM_1 Unit: In the ITAM_1 unit, presented in Fig. 8,
Fig. 5 illustrates a power system signal as presented by
the computations include multiplication, subtraction, absolute
(19). The fundamental frequency of the system instantaneously
value calculation, and square-root manipulation. The square-
changes from 50 to 48 Hz at t = 0.5 s and stays at 48 Hz
root block has been implemented in a precomputed lookup
afterward. A satisfactory fast tracing of the system frequency
table. Other operations are implemented using standard library
change can be noticed. The algorithm takes approximately
blocks.
40 ms to settle down the change. The results show that the
ITAM_2 Unit: The main purpose of the ITAM_2 unit is to
algorithm is stable under dynamic power system conditions and
implement the computation of the estimated frequency f as
yet is unaffected by fast transient.
follows:

V. I MPLEMENTATION f = f0 + F1 ∗ y + F2 ∗ y 2 + F3 ∗ y 3 (25)

The ITAM algorithm has low computation requirements. where y = x − x0 , obtained from the ITAM_1 unit.
It requires the calculation of a square root, four additions/ Rewrite it into a recursive form
subtractions, and six multiplications between two consecutive
samples of the filtered power signal waveform. Table I shows f = f0 + y ∗ (F1 + y ∗ (F2 + y ∗ F3 )) . (26)
3292 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 58, NO. 9, SEPTEMBER 2009

Fig. 6. Top-level schema of the frequency measurement system.


TABLE II
TAYLOR SERIES EXPANSION COEFFICIENTS

Fig. 7. Flowchart for the ITAM algorithm implementation.

Then, a simple multiplier can fulfill (26) in four clock cycles


of the ITAM_2 unit, which is illustrated in Fig. 9.
ITAM_3 Unit: The ITAM_3 unit performs two major tasks:
1) the averaging of the results obtained from the ITAM_2 unit
and 2) the computation of the rate of change of the frequency.
The frequency measurement results are the average value out
of every 40 consecutive results obtained from the ITAM_2 unit.
To avoid using a divider to calculate the average value, we use a Fig. 8. Structure of ITAM_1 unit design.
multiplier instead. The constant operand is the reciprocal of 40.
A five-cell 32-bit first-in–first-out memory is used to store The very high speed integrated circuit hardware description
five consecutive frequency measurement results that are used to language design of the FMS was synthesized for the Altera
compute the rate of change of frequency. APEX20K200E device, which has the smallest logic element
SALCIC et al.: IMPROVED TAYLOR METHOD FOR FREQUENCY MEASUREMENT IN POWER SYSTEMS 3293

FPGA chip. Simulation results have shown that the algorithm


has an advantage over the Taylor method [12] algorithm in these
aspects: it has no constraint in terms of the sampling rate, higher
accuracy, wider range of frequencies measurement, and lower
complexity. Another key feature of the algorithm is a good
tracing ability for the frequency change in a power system.

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Fig. 10. Test result of measuring the frequency of the continuous contami- [13] H. Tao and I. F. Morrison, “The measurement of power system frequency
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narios as described in this paper. An example of a measurement
of frequency where the synthesized power system input signal
was contaminated with dc offset, higher harmonics, and random Zoran Salcic (SM’97) received the B.E., M.E., and Ph.D. degrees in electrical
engineering from the University of Sarajevo, Sarajevo, Bosnia, in 1972, 1974,
noise as in (24) is shown in Fig. 10. The maximum error is and 1976, respectively. He did most of his Ph.D. research at the City University
0.015 Hz. of New York (CUNY), New York.
He is currently a Professor of computer systems engineering with the
Department of Electrical and Computer Engineering, University of Auckland,
VI. C ONCLUSION Auckland, New Zealand. He is the author of more than 200 journal and confer-
ence papers in the areas of complex digital systems design, custom-computing
This paper has described a new algorithm based on the Taylor machines, reconfigurable systems, FPGAs, processor and computer system
method for fast and accurate frequency measurement in a power architectures, embedded systems and their implementation, design automation
system in the presence of dc offset, higher harmonics, and tools for embedded systems, hardware–software codesign, new computing ar-
chitectures, and models of computation for heterogeneous embedded systems.
random noise. The algorithm is relatively simple and is suitable Dr. Salcic is a Fellow of the Royal Society New Zealand (Academy of
for real-time frequency measurement and implementation on an Science).
3294 IEEE TRANSACTIONS ON INSTRUMENTATION AND MEASUREMENT, VOL. 58, NO. 9, SEPTEMBER 2009

Sing Kiong Nguang (M’97–SM’00) received the B.E. degree in electrical Yanzhen Wu received the M.E. degree from Hunan University, Changsha,
engineering (with first-class honors) and the Ph.D. degree from the University China, in 1993 and the Ph.D. degree from South China University of Tech-
of Newcastle, Callaghan, Australia, in 1992 and 1995, respectively. nology, Guangzhou, China, in 1998.
He is currently is an Associate Professor with the Department of Electrical During 2001 and 2003, she did research with the Department of Electrical
and Computer Engineering, University of Auckland, Auckland, New Zealand. and Computer Engineering, University of Auckland, Auckland, New Zealand.
He is the author of more than 100 journal papers and more than 80 conference She is currently a Research Fellow with the Department of Electrical and
papers/presentations on nonlinear control design, nonlinear H-infinity control Computer Engineering, University of Auckland.
systems, nonlinear time-delay systems, nonlinear sampled-data systems, bio-
medical systems modeling, fuzzy modeling and control, biological systems
modeling and control, and food and bioproduct processing.

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