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(a)
1.0
0.9
0.8
0.7
%f %r
0.6
0.5
0.4
0.3
0.2
0.1
0
1 2 3 4 5 6 7 8 9 10 11 12 13 14 15 16 17 18 19 20 21 22 23

(b)
40
35
30
UTHD (%)

25
20
15
10
5
16:00 18:00 20:00 22:00 00:00 02:00 04:00 06:00 08:00
Hour:Minute

Figure 7.8 (a) Spectrum of a waveform represented in %f (fundamental) and %r (r.m.s. value);
(b) example of THD characteristic in industrial network

7.3 MEASUREMENTS

The main analysis tool used by measurement devices is the Fourier transform (FT) and its
modifications and improvements. This transform can be applied to an arbitrary function,
both periodic and non-periodic. The result of this transform is the spectrum in the frequency
domain which in the case of a non-periodic function is continuous. Its special case is a
periodic function whose spectrum is discrete and its lines are components: the fundamental
and harmonic.
The discrete Fourier transform (DFT) is a digital application of the classical Fourier
transform. The analogue signal to be analyzed is sampled, A/D converted (sampled, quan-
tized and then coded) and the results of sampling are stored. In practical applications the
signal is analyzed in a limited time interval (measurement window of duration Tw ) using
a limited number of samples (M) of the actual signal. Results of the DFT depend on the
choice of the Tw and M values. The inverse of Tw is known as the fundamental Fourier
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196 VOLTAGE AND CURRENT HARMONICS

frequency (fF ) (DFT resolution). The DFT is applied to the measured signal within the
measurement window. The signal outside the window is not processed and is assumed to
be identical to the waveform inside the window, i.e. the signal is assumed to be M-samples
periodic. In this way, the real signal is substituted with a virtual one, which is periodic
with a period equal to the window width. To ensure that the results of the DFT applied to
the functions which are considered periodic will be the same as those obtained from the
Fourier analysis of an infinite length signal, the duration TW of the time window should be
an integer multiple of the fundamental period, i.e. TW = NT1 ; this requires the sampling rate
fS to be an integer multiple of the basic DFT frequency fs = Mfb = M/NT1 .
Sampling synchronization is of key importance. The loss of synchronization can alter
the spectrum by adding extra components (lines) and changing the amplitudes of actual
components. Before DFT processing, the samples obtained in the time window TW are often
weighted by multiplying them with a special symmetrical function (windowing function).
However, for periodic signals and synchronous sampling, it is preferable to use a rectangular
weighting window which multiplies each sample by unity.
In practical applications, when equipment and software limitations require the number
of samples M to be no greater than a certain maximum number, the measurement time
is limited. A measurement time different from the fundamental Fourier period leads to a
discontinuity between the signal at the beginning and the end of the measuring window. This
issue gives rise to errors in identification of the components known as spectrum leakage.
A possible solution is the use of the weighted time window to a time-varying signal before
DFT (or fast FT, FFT) analysis. In common practice two kinds of measuring windows are
used: rectangular and Hanning windows.
The FFT is a special algorithm allowing computation times to be shortened. It requires
that the number of signal samples (M) be an integer power of 2 (M = 2i ). In other words,
it requires that the ratio of the sampling rate and basic DFT frequency be expressed
by an integer power of 2. Considering the capabilities of state-of-the art digital signal
processors, sine and cosine tables used in the DFT could be a helpful modification of the
algorithm.
If the supply system voltage is analyzed, the component with fundamental frequency
is that of the largest amplitude. It is not always the first line in the spectrum obtained
from DFT processing of a time function. Where the current is analyzed, the fundamental
frequency component is not necessarily the one with the largest amplitude.
Most instruments for measurements in the frequency domain work correctly when
only integer harmonic components are present in the measured signal. These instruments
feature a phase-locked loop aiming to synchronize measurements with the fundamental
component and sample the signal during one or several cycles in order to analyze it using
the FFT. Due to this phase-locked loop, the single-cycle samples provide an accurate
representation of the waveform spectrum only when it does not contain interharmonics. If
other non-harmonic frequencies (in relation to the measuring period) are present and/or the
sampled waveform is not periodic in this time interval, the interpretation of results becomes
difficult.
Because of the contemporaneous presence of both harmonic and interharmonic compo-
nents, the Fourier frequency (the greatest common divisor of all component frequencies
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contained in the signal) is different from the supply voltage fundamental frequency and
usually very small. This leads to two problems:

1. The minimum sampling time may be long and the number of samples large.
2. It is difficult to predict the fundamental Fourier frequency because not all the components
of the signal are known

This can be easily understood with the following examples.

The signal to be analyzed has a fundamental component (50 Hz) and an interharmonic
(71.2 Hz) and harmonic (2500 Hz). The fundamental Fourier frequency is 0.2 Hz, much
lower than the frequency of the fundamental. The corresponding period is 5 s making the
permissible minimum sampling time equal to 5 s. If the sampling rate is 10 kHz, which is
practically the minimum applicable value resulting from the Nyquist criterion, the minimum
required number of samples M is 50 000. Without the interharmonic component, the
minimum time measurement would be 20 ms and the number of samples would be 200.
Another example can be a remote control signal with frequency 175 Hz superimposed
on the sinusoidal supply voltage signal frequency 50 Hz. The superposition yields a periodic
voltage with a period of 40 ms and DFT resolution 25 Hz. Classical Fourier analysis of this
voltage yields a 25 Hz fundamental component with zero amplitude and two components
with non-zero amplitudes: the second harmonic (50 Hz) with amplitude equal to the supply
voltage; and the seventh harmonic (175 Hz) with amplitude equal to that of the remote
control signal.
The greatest difficulty associated with sampling a continuous signal is the problem of
ambiguity. The essence of this problem is illustrated in Figure 7.9. It follows from the figure
that the same set of sampled data may describe several waveforms, indistinguishable by
measuring equipment. The principle of frequency analysis is the representation of an arbitrary
waveform by the sum of a series of sinusoidal signals. Such a method of presentation
allows the quantitative analysis of the problem of ambiguity. For this purpose, consider the
waveform shown in Figure 7.10.
A signal xt is sampled in equal intervals of time h, determining the instants of
sampling, for which values of the measured signal are indicated in the figure. Assume that
the function xt is sinusoidal with frequency f0 . The same points could also represent

Other possible waveforms

Sample time

Figure 7.9 Ambiguity

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198 VOLTAGE AND CURRENT HARMONICS

f2 = 7.0 f1 = 3.0 x(t) f0 = 2.0

Sample time

Figure 7.10 Analysis of ambiguity

sinusoids with frequencies f1 or f2 . These various frequencies are obviously associated with
the sampling period. The frequency f0 is referred to as the fundamental frequency.
It could be stated, without presentation of the mathematical proof, that the range of
frequencies for which the effect of ambiguity does not occur extends from f0 = 0 to f0 = fN ,
where fN , the maximum frequency, is referred to as the Nyquist frequency. It determines
the limit frequency of data sampling, the so-called Shannon limit, beyond which a unique
reconstruction of a continuous signal is not possible. Thus, if the signal being analyzed does
not contain any component frequencies greater than fN , then the minimum sampling rate
necessary to allow the sampled signal to represent the real signal is given as fS ≥ 2fN , or,
because f ≥ 1/h, then fN ≥ 1/2h.
This is the so-called sampling theorem. It follows that, for a given spectrum of frequen-
cies, the components situated between f0 = 0 and f0 = fN can be considered separately.
If the signal contains components of frequencies f > fN , these components will not be
distinguished. Therefore it is necessary to limit the bandwidth of the measured signal to
reduce a direct consequence of the ambiguity during its sampling. That implies the need
to filter the signal to be measured through a low-pass filter before sampling, in order to
eliminate all frequencies greater than fN .
The precise computing of harmonics is a difficult task and it often leads to a ‘blurring’
of the spectrum, even with synchronous sampling; for this reason the so-called grouping was
introduced [22]. A given harmonic is then assessed using not only its spectral line, but also
a group of lines around the harmonic being sought. The concept of grouping is particularly
useful for the assessment of interharmonics. It allows only the spectral components to be
determined in 5 Hz intervals, thus finding for example the 278 Hz interharmonic is not
possible, since such a spectral line does not exists. The energy of this component will
be distributed over several adjacent spectral lines. The approximate value of the sought
interharmonic can be given by the value of this component group or subgroup.
The concept of grouping is illustrated in Figure 7.11 and Figure 7.12. Values of groups
and subgroups are determined from the following relationships.
For harmonics, the group is

 2
 Ck−5  4
C2
Ggn = + 2
Ck+i + k+5 (7.3)
2 i=−4 2
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Harmonic Interharmonic
group group
n+2 n+4
C

DFT output
Harmonic n n+1 n+2 n+3 n+4 n+5 n+6
order

Figure 7.11 Illustration of harmonic and interharmonic groups [22] (Reproduced from
current-carrying capacity and related overcurrent protection (Revision of
section 523”), IEC TC 64 WG 2)

Harmonic Interharmonic
subgroup centred subgroup
n+2 n+4

DFT output

Harmonic n n+1 n+2 n+3 n+4 n+5 n+6

order

Figure 7.12 Illustration of a harmonic subgroup and an interharmonic centered subgroup [22]
(Reproduced from current-carrying capacity and related overcurrent protection (Revision
of section 523”), IEC TC 64 WG 2)

and the subgroup is



 1
Gsgn = 2
Ck+i (7.4)
i=−1

where Ggn , Gsgn are the values of the harmonic group or subgroup, respectively, and Ck+i
is the r.m.s. value of the ith line with respect to the harmonic of order n.
For interharmonics, the group is


 9 2
Gign = Ck+i (7.5)
i=1

and the subgroup is



 8
Gisgn = 2
Ck+i (7.6)
i=2

where Gign , Gisgn are the values of the interharmonic group or subgroup, respectively, and
Ck+i is the r.m.s. value of the ith line with respect to the harmonic of order n.
Detailed information on harmonics measurements can be found in standard IEC
61000-4-7 [22].