NEW HIGH PRECISION HARMONIC ANALYSIS METHOD FOR POWER
QUALITY ASSESSMENT
Lucian Mandache
Canada Research Chair CRC-EECPE
Electric Energy Conversion and Power Electronics
Ecole de technologie superieure
email: lmandach@ele.etsmtl.ca
Abstract
The power quality is one of actual major problems in electrical
engineering. Generally, power electronics equipments damage power
quality parameters, disturbing radio communications or the
functionality of other equipments. A rigorous design of most
appropriate filters for power quality improvement is possible only
through a high precision analysis that allows estimating power
quality parameters and the influence of each harmonic component on
the network-drive system. Actual industrial equipments intended to
perform spectral analysis are not appropriate for strongly deformed
signals, with frequent discontinuities, as in power electronics. Our
paper presents a new and accurate method of harmonic analysis that
permits to mitigate most of power quality related problems. The
principle is to estimate intermediate points between the initial samples
given by the available data acquisition system; therefore, the Fourier
coefficients are estimated more precisely using the fast Fourier
transform. As interpolation technique we chose the reconstruction of
the analog signal using an ideal lowpass filter. The excellent results
are validated on a pair of synthesized signals having known harmonic
spectrum.
Keywords: Harmonic analysis; interpolation; lowpass filter.
1 Introduction
Power electronics equipments strongly improve the
performance and efficiency of electrical drives. Unfortunately,
the latter can cause supplementary losses in power systems or
electrical machines (power transformers, motors), breaking
torques in motors; can disturb the radio communications or the
functionality of other equipments. That is due to the level of
current harmonics injected into the utility distribution lines.
A high precision harmonic analysis can help to estimate the
influence of each harmonic component on the network-drive
systems. An appropriate energetic analysis for each harmonic
component and an accurate estimation of power quality
parameters using the most efficient methods permit to mitigate
most of power quality related problems. Consequently, the
most efficient methods to reduce the undesired harmonic
components can be performed.
In order to keep experimentally measured electric signals, a
high precision data acquisition system is therefore needed. The
sampling rate of the data acquisition system limits the order of
the fastest harmonic component that can be estimated. In
addition, a low sampling rate causes a bad accuracy in
0-7803-8886-0/05/$20.00 ©2005 IEEE
CCECE/CCGEI, Saskatoon, May 2005
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Kamal Al-Haddad
Canada Research Chair CRC-EECPE
Electric Energy Conversion and Power Electronics
Ecole de technologie superieure
email: kamal@ele.etsmtl.ca
harmonic decomposition using the classical discrete Fourier
transform (DFT), which is the major tool for spectral
estimation. The resolution of the DFT is 1/N, where N is the
number of data points per cycle; which is quite inadequate and
can cause imprecision for small ratio of N. More accurate
spectral estimation techniques offer the possibility of enhanced
spectral resolution with various amounts of added
computational complexity. These are called collectively high-
resolution spectral estimation techniques [1].
This paper presents a new, more accurate and appropriate
method of harmonic analysis in power electronics equipments.
Its principle is to estimate intermediate points between the
initial samples; therefore, the Fourier coefficients can be
estimated more precisely using the fast Fourier transform
(FFT). As interpolation technique we have chosen the
reconstruction of the analog signal using an ideal lowpass
filter, with the cutoff frequency equal to the Nyquist frequency.
We have developed a dedicated program under MATLAB
environment and have performed harmonic analysis on many
signals. We shall present here the excellent results obtained for a
pair of artificially synthesized signals with a known harmonic
spectrum.
2 Harmonic spectrum
Generally, any signal that conveys information about the
behavior of a physical system can be represented mathematically
as a function of time. The independent variable (time) is always
continuous and the corresponding signal is analog.
A real function that describes an analog signal and is
periodic x (t ) = x (t + T ) , with a finite number of
discontinuities, finite number of extremes and absolutely
integrable in any period, can be expressed as a series of
harmonic functions (Fourier series) as [1-5]:
∞
§ 2π ·
x (t ) = C 0 + ¦ Ck sin¨© k t +γk ¸ (1)
k =1
T ¹
where k is the order of harmonic components. The known
Fourier transform applied to a restriction during one period of
the time function allows to compute the magnitudes C k and
phases γ k . If the time is considered as discrete values (2), the
corresponding analog signal becomes a discrete-time signal
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(3); it is represented as sequences of numbers and can be kept
using hardware medium.
t ∈ {t 0 , t1 , t 2 ,..., t n ,..., t N }
x (t 0 ) ≡ x 0 , x (t1 ) ≡ x1 ,..., x (t n ) ≡ x n ,..., x (t N ) ≡ x N
The discrete-time signals are approximations of analog ones.
In order to hold the experimentally measured electrical signals,
current and/or voltage transducers and data acquisition systems
are therefore needed. The quality of the sampled signal
depends on both the transducers response time and the sampled
rate of the data acquisition system which limit the order of the
fastest harmonic component that can be measured. Generally,
the sampled period is constant, being a characteristic of data
acquisition system (4):
∆t = t n +1 − t n or t n = n∆t , ∀n = 0,..., N − 1
The study of some signals in the frequencies domain is
possible using the DFT applied to its corresponding discrete-
time functions. If the signal is periodic and the sampled
sequence corresponds to one period T = t N − t 0 = 1 f (this
fact is equivalent to a windowing of length T applied to the
time signal [1-4,6,7]), the DFT is expressed as Riemann sum
with N terms starting from the analog Fourier transform:
N −1
X d (k ) ≡ X d (kf ) = ∆t ¦ x(t n )e − j 2π (kf ) tn
n =0
Therefore, the series in expression (1) is truncated in N terms with:
N −1
1 2
C0 = ¦ xn ; Ck = (Re[X d (k )])2 + (Im[X d (k )])2
T n =0 T (6)
Re[X d ( k )]
γ k = arctan
Im[X d ( k )]
The highest harmonic component that can be computed from
N-point per cycle sampling has the frequency f max = k max ⋅ f
named Nyquist frequency, equal to half of sampled rate
f s = 1 ∆t in accordance to the Nyquist theorem [2]:
f max = f s 2
having the maximum order of k max = f s (2 f ) = N 2 .
3 Errors in harmonic decomposition
The sampled signal is an approximation of the analog one;
therefore, the harmonic decomposition computed by the DFT
is an approximation of the real harmonic spectrum of the
analog signal. Applying DFT the errors can be unacceptable in
many cases. One can minimize the errors only by knowing
their sources and their manifestation, as follows:
- Gibbs effect [2-4], due to the signal discontinuities. If the
periodic signal is a continuous time function, it becomes
discontinuities through the time windowing. Discontinuities
have theoretically an infinite number of harmonic components
and the DFT offers only truncated series that is an inaccurate
approximation of discontinuities. A weak sampled rate can
cause important discontinuities for continuous, real, signals;
the corresponding Gibbs effect can be strongly reduced
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through a greater sampled rate. Generally, the data acquisition
system doesn’t permit this fact. The sampled rate can be
(2) increase artificially, using numeric methods to approximating
the time function among the sampled points. The easily
(3) practical method to reduce the Gibbs effect to the time window
edges is to choose the rectangular time window so that the
periodic signal values are zero or near zero at their edges.
- Leakage phenomenon [3,8], if the window length is not an
integer multiple of the period of each harmonic component of
the time varying signal. It consists in parasitic harmonic
components, computed by DFT, which accompany the true
component. In addition, the phenomenon occurs if the time
signal contains harmonic frequencies which are not integer
multiple of the fundamental component, but this situation is
generally not achieved in power electronics applications. In
(4)
order to reduce this phenomenon and improve the precision of
harmonic decomposition, many authors recommend replacing
the rectangular time windowing by other types of functions
[2,6,7]. The merits of a given time window depend on the
application. One may not use such windowing automatically.
The results can be poorer in most cases, depending on the
signal. The authors experience shows that in power electronics
applications the rectangular window with a good estimation of
the fundamental period offers the best results. If the sampled
(5) rate of the used data acquisition system is not an integer
multiple of the fundamental frequency, the window edges are
between samples, where the signal values are not known;
therefore, one can’t use the best windowing. Our solution is to
increase the sampled rate artificially in order to estimate values
of signal between samples, as we are already invoking for
Gibbs phenomenon.
- Aliasing effect [2,9], appears if the signal contains
harmonic components faster than half of the sampled rate
(Nyquist frequency); therefore, the original signal is strongly
distorted through sampling. These components are not
theoretically viewed, but they appear in the harmonic spectrum
computed by DFT as parasitic components (named alias
(7)
components); if the analog signal contains a component of
order N 2 + m , then it appears as order N 2 − m in the DFT
computed spectrum. The best method to minimize this effect is
to eliminate the harmonic components faster than the Nyquist
frequency from the time signal, using analog lowpass filters
before sampling. We performed numeric filtering of sampled
signal in combination with an interpolation method to
minimize both Gibbs and leakage phenomena, as we shall
present lather.
4 Method of Ideal Lowpass Filters (ILF)
We shall perform a processing of the sampled signal using a
digital filter; it is an ideal lowpass filter with the cutoff
frequency equal to the half of sampled rate (fig. 1). It has unity
attenuation for all frequency components less than Nyquist
frequency and allows a high precision interpolation of the
signal between sampled points in order to reconstruct a
continuous-time signal.
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 Sampled signal Analog signal it assures the reduction of all sources of errors in harmonic
xn Lowpass x(t) decomposition based on DFT. The method can be used
filter successfully if adequate hardware computation resources are
available and if the computation time is not critique.
Fig. 1. Sampled signal filtering 25

20

Such filter with a cutoff frequency of f c = 1 ( 2∆t ) has an

15
impulse time-response as:
§π · 10
sin¨ t ¸
sin(2πf c t ) © ∆t ¹ (8)
y (t ) = = 5
2πf c t π
t
∆t 0

having a graphic representation depicted in fig. 2 for a sampled -5

rate of 5kHz and a corresponding sampled time of 0.2ms. The -10

graphic representation was performed for both positive and
1 0.8 0.6 0.4 0.2 0
0 0.2 0.4 0.6 0.8 x 10
1
-3

negative values of time, this fact being necessary further. -3

Time [x10 secs]
1 Fig. 3. Filtering process
The ILF method assures an excellent approximation of the
0.5
analog signal as we show in fig. 4; the figure represents a
synthesized signal that contains a sinusoidal component of 120
0
kHz and a lot of slower components; the sampling rate is 250
kHz, a little greater than two times the analog signal frequency.
-0.5
-1 -0.8 -0.6 -0.4 -0.2 0 0.2 0.4 0.6 0.8 1 Even though the sampled signal is inaccurate approximation of
Time [x10-3 secs] the analog one, the intermediary points (computed through the
Fig. 2. Impulse time-response of an ideal lowpass filter method of lowpass filter) allow an accurate reconstruction of
the original signal. The sampling rate improved artificially by a
In order to obtain the output signal, one must perform the factor of three (two approximations between two samples).
convolution between the input signal (given through N sampled Therefore, using an appropriate improved sampled rate the
points) and the impulse time-response; because the input signal original signal can be exactly estimated.
is discrete, the convolution will be expressed as a sum: Analog signal
Sampled signal
870
ILF approximation
§ t ·
N N sin π ¨ − n ¸
© ∆t ¹ (9)
x ( t ) = x ( n∆t ) ∗ y (t ) =¦ ¦
x ( n∆t ) y (t − n∆t ) = xn
§ t ·
860

n =0 n =0 π¨ − n¸
© ∆t ¹ 850

The equation above represents an ideal digital-to- 840

continuous-time converter; it suppresses all frequency
components faster than cutoff frequency and assures the 830

corresponding minimization of the aliasing effect (as we have

already showed). An image of this procedure is shown in fig. 3 820

for a short sequence of a sampled signal. The sampled signal is 0.0168 0.0168 0.0168 0.0168 0.0168 0.0168 0.0168 0.0168 0.0168
marked by points (six points) and the reconstructed one is
Fig. 4. ILF interpolation starting from the sampled signal
marked as solid line. For each visible sampled point the figure
shows (as dashed and thin lines) the associated term of (9),
already known from fig. 2. 5 Experimental results
In order to find the harmonic spectrum of the filtered signal
in analog form (9) one can use the Fourier transform. This is Let us consider two artificially created signals i(t) and u(t)
only a theoretic assumption, because the calculus effort is synthesized from many ideal harmonic components:
unacceptably high. Practically, equation (9) can be used to § π· § π·
i (t ) = 50 + 500 sin¨120πt + ¸ + 100 sin¨ 3 ⋅ 120πt + ¸ +
compute intermediary points between the initial samples; © 4 ¹ © 6¹
therefore, the Fourier coefficients can be estimated using DFT § π· § π·
+ 150 sin¨ 5 ⋅ 120πt + ¸ + 50 sin¨ 7 ⋅ 120πt + ¸ +
(FFT) and equations (6). © 3¹ © 4¹
The method of ideal lowpass filter offers high precision § π· § π·
+ 200 sin¨11 ⋅ 120πt + ¸ + 25 sin¨ 2000 ⋅ 120πt + ¸ A
results if the number of intermediary points is sufficiently high; © 2¹ © 6¹

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 u(t ) = 50 + 1000 sin(120πt ) + 200 sin (3 ⋅ 120πt ) + 300 sin (5 ⋅ 120πt ) + image of harmonic spectrum (both magnitudes and phases) and
+ 100 sin (7 ⋅ 120πt ) + 300 sin(11 ⋅ 120πt ) + 50 sin (2000 ⋅ 120πt ) V an immediate examination on the recomposed signal. In
The fundamental frequency is 60 Hz and the order of the addition, numerical values of computed harmonics are given in
highest harmonic component is 2000 . According to Nyquist table 1 and table 2. Values of RMS, THD, maxima, peak factor
theorem, the sampling frequency should be at most for both voltage and current are computed. The software has
2 ⋅ 2000 ⋅ 60 = 240 kHz . We chose fs = 250 kHz . the capabilities to compute energy related components such as:
Using our dedicated MATLAB application, based on the active and reactive powers for each harmonic component, as
above presented method, we have obtained results that are well as total active, reactive, apparent and distorting powers.
shown in fig. 5. This graphic representation offers a suggestive
Voltage - recomposed signal / original signal Current - rec omposed signal / original signal
1000
1000

0 0

-1000
-1000
0 0.002 0.004 0.006 0.008 0.01 0.012 0.014 0.016 0.018 0 0.002 0.004 0.006 0.008 0.01 0.012 0.014 0.016 0.018
Time [secs] Time [secs]
Magnitude [% ]

Magnitude [% ]
30 Harm.no.1=100% 40
Harm.no.1=100%
20
20
10
0 0
0 1 3 5 7 11 2000 0 1 3 5 7 11 2000
3 100
Phase [deg.]

Phase [deg.]
2
50
1

0 0
Harmonic order Harmonic order

Fig. 5. Graphic representation of computed voltage and current spectrum

Table 1 - Voltage harmonic decomposition lot of other numeric algorithms are opportunely combined.
Order Magnitude [V] Phase [deg] Percentage [%] The excellent accuracy of this method has the price of a
0 5.0000e+001 0 5.0000e+000 considerable great amount of calculus, required by lowpass
1 1.0000e+003 1.4388e-003 1.0000e+002 filtering process. It increases the total analysis time so that a
3 2.0000e+002 4.3138e-003 2.0000e+001 real-time analysis is not possible.
5 3.0000e+002 7.1958e-003 3.0000e+001
7 9.9994e+001 1.0068e-002 9.9994e+000 Acknowledgement
11 2.9999e+002 1.5836e-002 2.9999e+001 The authors wish to thanks Canada research chair and AUF for
2000 4.9979e+001 2.8800e+000 4.9979e+000 supporting this work
Table 2 - Current harmonic decomposition References
Order Magnitude [A] Phase [deg] Percentage [%]
0 5.0006e+001 0 1.0001e+000 [1] C.H.Chen, SIGNAL PROCESSING HANDBOOK, Marcel Dekker,
1 5.0001e+002 4.5002e+001 1.0000e+002 Inc., New York, 1988
3 1.0000e+002 3.0008e+001 2.0000e+001 [2] A.V. Oppenheim, R.W. Schafer, J.R. Buck, DISCRETE-TIME
5 1.5000e+002 6.0009e+001 3.0000e+001 SIGNAL PROCESSING, Prentice Hall, New Jeresey, 1999
7 5.0000e+001 4.5016e+001 9.9999e+000 [3] P.A. Lynn, W. Fuerst, DIGITAL SIGNAL PROCESSING WITH
11 2.0000e+002 9.0016e+001 4.0000e+001 COMPUTER APPLICATIONS, John Wiley and Sons, West Sussex, 1997
[4] R.W. Ramirez, THE FFT, FUNDAMENTALS AND CONCEPTS,
2000 2.4989e+001 3.2880e+001 4.9978e+000
Englewood Cliffs, N.J., Prentice-Hall, 1985
[5] R.N. Bracewell, THE FOURIER TRANSFORM AND ITS
APPLICATIONS, McGraw-Hill, New York, 1986
6 Conclusions [6] J.A. de la O Serna, “On the use of amplitude shaping pulses as
windows for harmonic analysis”, IEEE Transactions on Inst. and
The presented method assures best conditions for Measurement, Volume: 50, Issue: 6, Dec. 2001, pp. 1556 – 1562
minimization of Gibbs, leakage and aliasing effects through an [7] T.L.J. Ferris, A.J. Grant, “Frequency domain method for
appropriate increasing of sampling rate. It is suitable to process windowing in Fourier analysis”, IEEE Electronics Letters, Volume:
experimentally measured electric signals using a data 28, Issue: 15, 16 July 1992, pp. 1440
acquisition system in power electronics equipments; [8] D. Elliott, K.R. Rao, FAST TRANSFORMS: ALGORITHMS,
ANALYSES, APPLICATIONS, Academic Press, Orlando, 1982
synthesized signals in the previous example are chosen only to
[9] R.B. Randall, B.A. Tech, FREQUENCY ANALYSIS, Bruel and
prove the remarkable accuracy. Kjaer, Danenmark, 1987
The known method of filtering, fast Fourier transform and a

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