FFT ANALYSIS

SOLUTION USER MANUAL

FFT V21-3

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1. Table of contents
1. Table of contents 2

2. What is frequency analysis? 4

3. Fourier transform 5
3.1 Fourier series 5
3.2 Discrete Fourier transform 6
3.3 FFT - Fast Fourier transform 8
3.4 Properties of Fourier transform 10

4. Windowing functions 13

5. Fourier transformation errors 18

5.1 Amplitude error (picket-fence effect) 18

6. Aliasing 23
6.1 Aliasing example 24

7. Averaging of the signal 27

7.1 Overlapping 28

8. Representation of different signals in the FFT 29

8.1 Triangle, rectangular signal 29
8.2 Impulse signal 30
8.3 White noise 30
8.4 Beating (two closely spaced signals) 31
8.5 Amplitude modulated signal 31

9. FFT analysis in Dewesoft 32

9.1 Output spectra 33
9.2 Amplitude function 33
9.3 Amplitude format 33
9.4 Frequency weighting 34
9.5 Cross spectrum 35
9.6 Output quantity and axis unit 36
9.7 Calculation type 36
9.8 Averaging type 36
9.9 Window functions 37
9.10 Resolution settings 37
9.11 DC cutoff frequency 37
9.12 Overlap 38
9.13 Output scalars 38
9.14 Harmonics 38

10. FFT markers 39

10.1 Free marker 39
10.2 MAX marker 40

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10.3 RMS marker 44

10.4 Sideband marker 45
10.5 Harmonic marker 47
10.6 Damping marker 49
10.7 Kinematic cursor 53
10.8 Interpolation 56

11. Warranty information 59

11.1 Calibration 59
11.2 Support 59
11.3 Service/repair 59
11.4 Restricted Rights 59
11.5 Printing History 60
11.6 Copyright 60
11.7 Trademarks 60

12. Safety instructions 60

12.1 Safety symbols in the manual 60
12.2 General Safety Instructions 60
12.2.1 Environmental Considerations 61
12.2.2 Product End-of-Life Handling 61
12.2.3 System and Components Recycling 61
12.2.4. General safety and hazard warnings for all Dewesoft systems 61

13. Documentation version history 64

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2. What is frequency analysis?

For cyclical processes, such as rotation, oscillations, or waves, frequency is defined as a number of cycles
per unit of time. For counts per unit of time, the SI unit for frequency is hertz (Hz); 1 Hz means that an
event repeats once per second.

The time period (T) is the duration of one cycle and is the reciprocal of the frequency (f):

1
𝑇 = 𝑓

Frequency analysis is just another way of looking at the same data. Instead of observing the data in the
time domain, with some not very difficult, yet inventive mathematics, frequency analysis decomposes
time data in the series of sinus waves.

We can also say that frequency analysis checks the presence of certain fixed frequencies.

The image below shows the signal, which consists of three sine waves with the frequencies of 0.5 Hz, 1
Hz, and 2 Hz, and then on the right side the decomposed signal.

Just to make those sine waves better visible, let us show them in a nicer way. On the x-axis, there are
frequencies and on the y-axis, there are amplitudes of the sine waves.

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And this is really what the frequency analysis is all about: showing the signal as the sum of sinus signals.
And understanding how that works, helps us to overcome problems that it brings with it.

3. Fourier transform
The mathematician Fourier proved that any continuous function could be produced as an infinite sum
of sine and cosine waves. His result has far-reaching implications for the reproduction and synthesis of
sound.

A pure sine wave can be converted into sound by a loudspeaker and will be perceived to be a steady,
pure tone of a single pitch. The sounds from orchestral instruments usually consist of a fundamental
and a complement of harmonics, which can be considered to be a superposition of sine waves of a
fundamental frequency f and integer multiples of that frequency.

Fourier analysis of a periodic function refers to the extraction of the series of sines and cosines which
when superimposed will reproduce the function. This analysis can be expressed as a Fourier series.

3.1 Fourier series

Any periodic waveform can be decomposed into a series of sine and cosine waves:
∞ ∞
2π𝑛𝑡 2π𝑛𝑡
𝑓(𝑡) = 𝑎0 + ∑ 𝑎𝑛 · 𝑐𝑜𝑠( 𝑇
) + ∑ 𝑏𝑛 · 𝑠𝑖𝑛( 𝑇
)
𝑛=0 𝑛=0

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where a0, an, and bn are Fourier coefficients:

𝑇/2
1
𝑎0 = 𝑇
∫ 𝑓(𝑡)𝑑𝑡
−𝑇/2
𝑇/2
2 2π𝑛𝑡
𝑎𝑛 = 𝑇
∫ 𝑓(𝑡) 𝑐𝑜𝑠( 𝑇
)𝑑𝑡
−𝑇/2
𝑇/2
2 2π𝑛𝑡
𝑏𝑛 = 𝑇
∫ 𝑓(𝑡) 𝑠𝑖𝑛( 𝑇
)𝑑𝑡
−𝑇/2

3.2 Discrete Fourier transform

For discrete data, the computational basis of spectral analysis is the discrete Fourier transform (DFT). The
DFT transforms time-based or space-based data into frequency-based data.

The DFT of a vector x of length n is another vector y of length n:

𝑛−1
𝑗𝑝
𝑦𝑝+1 = ∑ ω 𝑥𝑗+1
𝑗=0
where w is a complex nth root of unity:
−2π𝑖/𝑛
ω= 𝑒
We used i for the imaginary unit and p and j for indices that run from 0 to n-1. The indices p+1 and j+1 run
from 1 to n.

Data in the vector x are assumed to be separated by a constant interval in time or space, dt = 1/fs or ds =
1/fs, where fs is the sampling frequency. The DFT y is complex-valued. The absolute value of y at index p+1
measures the amount of the frequency (f = p(fs / n)) present in the data.

The first element of y, corresponding to zero frequency, is the sum of the data in x. This DC component is
often removed from y so that it does not obscure the positive frequency content of the data.

An example of this is the square wave in the picture below. A square wave is composed of an infinite
summation of sinusoidal waves.

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Let's think about how the equation for discrete Fourier transform works:
𝑁−1
2π𝑘𝑛 2π𝑘𝑛
𝑥(𝑘ω0) = ∑ 𝑥(𝑛) · (𝑐𝑜𝑠( 𝑁
) + 𝑖 · 𝑠𝑖𝑛( 𝑁
))
𝑛
To check the presence of a certain sine wave in a data sample, the equation does the following:
1. Multiplies the signal with a sine wave of that frequency which we want to extract. The image
below shows the signal (black line), which consists only of a sine wave with 50 Hz. We try to
extract the 36 Hz on the left side and 50 Hz on the right side (they are shown as blue lines). Light
blue filled wave shows multiplied values
2. Multiplied values are summed together and this is the main trick. If there is a component in a
signal like in the right picture the multiplication of positive signal parts and extraction of sine
waves gives the positive result. Also, the multiplication of negative signal parts and negative
extraction sine waves gives positive results (observe the right image). In this case, the sum of the
multiplied sine waves will be nonzero and will show the amplitude of the 50 Hz part of the signal.

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In the case of 36 Hz, there are both positive and negative sides of multiplication values and the
sum will be (almost, as we will see further on) zero.
3. And that's it. That sum gives the estimate of the presence of frequencies in the signal. We check
sine and cosine to get also phase shift (in the worst case if the phase shift would be 90 deg, the
sum of sine functions would always give zero).

The principle shown above can extract basically any frequency from the sine wave, but it has one
disadvantage - it is awfully slow. The next important step in the usage of DFT was the FFT algorithm -
this analysis reduces the number of calculations by rearranging the data. The disadvantage is only that
the data samples must be of length, which is the power of two (like 256, 512, 1024 and so on). Apart from
that, the result is practically the same as for the DFT.

3.3 FFT - Fast Fourier transform

Fast Fourier transform is a mathematical method for transforming a function of time into a function of
frequency.

It is described as transforming from the time domain to the frequency domain.

The Fast Fourier transform (FFT) is a development of the Discrete Fourier transform (DFT) which
removes duplicate terms in the mathematical algorithm to reduce the number of mathematical
operations performed. In this way, it is possible to use large numbers of samples without compromising
the speed of the transformation. The FFT reduces computation by a factor of N/(log2(N)).

FFT computes the DFT and produces exactly the same result as evaluating the DFT; the most important
difference is that an FFT is much faster!

Let x0, ...., xN-1 be complex numbers. We have already seen that DFT is defined by the formula:

𝑁−1 𝑛
−𝑖2π𝑘 𝑁
𝑋𝑘 = ∑ 𝑥𝑛𝑒 , 𝑘 = 0,..., 𝑁 − 1
𝑛=0

Evaluating this definition directly requires N2 operations: there are N outputs of Xk, and each output
requires a sum of N terms. An FFT is any method to compute the same results in N log(N) operations. All
known FFT algorithms require N log(N) operations.

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To illustrate the savings of an FFT, consider the count of complex multiplications and additions.

Evaluating the DFT's sums directly involves N2 complex multiplications and N(N-1) complex additions.
The FFT algorithm can compute the same result with only (N/2)log2(N) complex multiplications and
N/log2(N) complex additions.

DFT FFT

Complex multiplications N2 (N/2)log2(N)

Complex additions N(N-1) N/log2(N)

In practice, actual performance on modern computers is usually dominated by factors other than the
speed of arithmetic operations and the analysis is a complicated subject, but the overall improvement
from N2 to N log2(N) remains.

In the image below, you can see original data of a signal in the time domain (units in seconds [s]), and
data after Fast Fourier transformation in the frequency domain (units in hertz [Hz]).

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Once you know the harmonic content of a signal from Fourier analysis, you have the capability of
synthesizing that signal from a series of pure tone generators by properly adjusting their amplitudes
and phases and adding them together. This is called Fourier synthesis.

3.4 Properties of Fourier transform

In the formula below, we can see a typical FFT screen. The maximum frequency of the FFT is half of the
signal sampling frequency (in this case the sample rate was 22000 samples/sec), but in the upper region
the results are never reliable, so the sampling result should be set to:

𝑆𝑎𝑚𝑝𝑙𝑒𝑅𝑎𝑡𝑒 = 𝑀𝑎𝑥𝑖𝑚𝑢𝑚𝑆𝑖𝑔𝑛𝑎𝑙𝐹𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦 · 2 · 1, 25
1.25 is the absolute minimum factor for getting the right values also in the upper region of the FFT. This
is the equation another way around the famous Nyquist criteria, which says that maximal signal
frequency adequately presented in the digitized wave is the half of the sampling rate.

The result of FFT is a set of amplitudes of certain frequencies. The number of lines in the set is
user-selectable, but they only change the resolution of the FFT. Line resolution is a change in frequency
between two frequency lines, which are extracted from the signal and is calculated with the equation:

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𝑆𝑎𝑚𝑝𝑙𝑒𝑅𝑎𝑡𝑒/2
𝐿𝑖𝑛𝑒𝑅𝑒𝑠𝑜𝑙𝑢𝑡𝑖𝑜𝑛 = 𝑁𝑢𝑚𝑏𝑒𝑟𝑂𝑓𝐿𝑖𝑛𝑒𝑠

So the question is: why not always use the maximum number of available frequency lines, which gives
more exact results? The answer is simple: because with larger frequency lines it takes more time to
calculate FFT.

2·𝑁𝑢𝑚𝑏𝑒𝑟𝑂𝑓𝐿𝑖𝑛𝑒𝑠
𝑇𝑖𝑚𝑒𝑇𝑜𝐶𝑎𝑙𝑐𝑢𝑙𝑎𝑡𝑒 = 𝑆𝑎𝑚𝑝𝑙𝑒𝑅𝑎𝑡𝑒

Just for fun we can also combine the equations above and we get:

1
𝐿𝑖𝑛𝑒𝑅𝑒𝑠𝑜𝑙𝑢𝑡𝑖𝑜𝑛 = 𝑇𝑖𝑚𝑒𝑇𝑜𝐶𝑎𝑙𝑐𝑢𝑙𝑎𝑡𝑒

Let's look at the equations above and make a list for the 22 kHz sample rate:

Number of lines Line resolution [Hz] Calculation time [s]

512 21,5 0,046

1024 10,75 0,093

4096 2,685 0,372

16384 0,67 1,49

So the number of lines combined with the sample rate also defines the speed of the FFT when
non-stationary signals are applied. With more lines, FFT will appear slower and changes in signal will not
be shown that rapidly.

Different amplitude scales of FFT can reveal more about the signal if used correctly. Linear amplitude
scale gives the best view of maximum peaks in the signal, a logarithmic amplitude scale can show more
invisible peaks and signal noise but gives a worse comparison of high and low peaks. Scale in dB gives
the best estimation of signal noise if 0 dB is maximum measurable value and is also used in noise
measurements, where the dB scaling is actually the result since the human ear has logarithmic
sensitivity to noise.

FFT results displayed on a linear scale

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FFT results displayed on a logarithmic scale

X scale can be either linear or logarithmic. Linear scaling is the correct representation of the
mathematical transformation and usually gives the best information for analysis. Sometimes, like in the
example shown in the picture above, it is nice to see the x-axis in logarithmic values since most
interesting frequencies are in a lower region. We have to know that just to set the x scale to logarithmic
does not enhance the results in the lower region, so the resolution will be better in the upper region
since there are more frequency lines available there.

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Linear frequency scale

Logarithmic frequency scale

If we use another technique, called CPB (constant percentage bandwidth) or octave analysis, this will
give us the same resolution in all regions when the x-axis is logarithmic. This is achieved by the fact that
upper region lines cover wider frequency ranges than the lower one.

The resolution of the bands is defined by 1/n description, where n is the number of bands in one octave.
The most widely used is the 1/3 octave analysis, which is the standard for noise measurements. 1/12 and
even better 1/24 octave analysis already gives good resolution also for signal analysis.

4. Windowing functions
If the sine wave is not on the frequency line, we get high amplitude values on both sides of the main
band. The amplitudes are really high (with no window, it is about 10% of the original values for about 10

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neighbor lines). If there is another sine wave in the signal in this region, which is lower than this 10%, it
will be completely hidden by the leakage effect.

This is a phenomenon that occurs because the FFT algorithm can only be applied to periodic signals so
the sampled input signal is 'periodized'. If the sampled signal is not periodic, or an integer number of
periods is not sampled, discontinuities occur in the periodic signal processed by the FFT, causing the
energy contained in the signal to 'leak' from the signal frequency bin into adjacent frequency bins. This
leakage causes amplitude errors in the frequency spectrum.

As a result of the amplitude errors caused by spectral leakage, small frequency peaks occur close to
larger ones.

Window functions are used to reduce the effects of spectral leakage. Windowing is used to assign a
weighting coefficient to each of the input samples, reducing those samples that cause spectral leakage.
In effect, samples at the beginning and at the end of the sampling period are reduced to zero so that
the discontinuities in the periodized sampled signal are removed.

In the picture below we can see the effect of windowing in a signal (Periodized signal with
discontinuities on the left and Discontinuities "ironed out" by windowing on the right side).

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In the picture below we can see a spectrum of a signal without spectral leakage, spectrum with spectral
leakage, and spectrum with windowing.

Spectrum display without spectral leakage

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Small frequency peak obscured as a result of leakage

Adjacent peak spectrum display with windowing, smaller frequency peak is no longer obscured.

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Windows are characterized by a number of properties as shown in the picture below.

The shape of the window's main lobe is defined by the -3 dB and -6 dB main lobe width. These are
defined as the width of the main lobe, in frequency bins, where the window response becomes
respectively 3 dB or 6 dB less than the main lobe peak gain. The width of the main lobe of the frequency
spectrum is important, as it affects the frequency resolution of the window (ability to distinguish
between closely spaced frequency components). As the main lobe narrows, frequency resolution
increases. However, with this narrowing of the main lobe, the window energy spreads into the side lobes,
increasing the spectral leakage. Therefore, a compromise between frequency resolution and spectral
leakage must be reached.

The maximum sidelobe level is defined as the level, in decibels, of the maximum side lobe, relative to the
main lobe peak gain.

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Sidelobe roll-off rate is the rate of decay of frequency of the side-lobe peaks, in decibels per decade.
The choice of the window depends upon the frequency content of the signal. A popular choice is the
Hanning window.

This window has quite a narrow main lobe, therefore, good frequency resolution and reasonable side
lobe suppression making it suitable for many applications.

Blackman-Harris window has excellent sideband rejection with an acceptably narrow main lobe.

5. Fourier transformation errors

The theoretical discrete Fourier transformation (DFT) has absolutely no error. The only problem is that
the sum goes from minus infinity to plus infinity. Because we live in a fast-paced world we don't have
the time to wait that long so we run into problems.

5.1 Amplitude error (picket-fence effect)

The sum can produce "non-null' results even when the signal does not correspond to the frequencies
extracted from the signal. The pure frequencies are because of that 'leaked' over neighbor frequencies.
For the same reason, if the frequency does not fall exactly on the frequency line, the amplitudes seem to
be lower. This is called the "picket fence" effect.
Let's look at the picture below - 10 Hz and 12 Hz are the exact frequency lines. In the example, there are
10 Hz and 12 Hz sine waves marked as black, which are transformed correctly, and there are also
frequencies in between which have lower amplitudes. Maximum amplitude error can go up to 35% of
the correct value.

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For amplitude errors, a bunch of people tried to minimize that problem. Those were Hamming, Hanning,
Blackman, Harris, and others. They have created an assortment of functions, which try to correct the
errors. Window functions are multiplied with the original time-domain signal and because they are
usually 0 at the beginning and the end, sine waves could also be in-between lines or phase-shifted and
they will leak less over neighbor frequencies.

The picture below shows some of these functions in the time domain.

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And here is the most common question to FFT: what are the differences between windows and when to
use certain windows?

The rule of thumb is that when we want a pure transformation with no window's side effects (for
advanced calculations), we should use a Rectangular window (which is equal to no window).
For general-purpose, Hanning or Hamming are commonly used because they provide a good
compromise between fall-off and amplitude error (maximum of 15%). This comes from the fact that old
frequency analyzers didn't have that many possibilities in terms of frequency lines and these two
windows have a narrow sideband.

When a more dynamic range is necessary (we want to see very small signals among large ones),
Blackman or Kaiser's window is a better choice because sidebands are 10 times lower than with the
Hanning window. However, the sideband width is wider. Here it comes to the point - if more lines in FFT
are chosen, we can use these windows and still larger sidebands had no real disadvantage.

If correct amplitudes are needed, we should use the flat-top window. The amplitudes would be wrong
by only a fraction (as low as 1%). Of course, there is a penalty - neighbor frequencies are also very high
(sideband width is high). This window is most suitable for calibration. But here it is the same: with
modern equipment with lots of lines, this is no longer that much of a problem.

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Window characteristics (maximum amplitude error, sideband width, highest sideband attenuation, a
sideband slope attenuation) are best described in the picture below. We have already discussed the
maximum amplitude error: it is an error of amplitude if the sine waves do not fall on the frequency line.
Windows try to eliminate this problem and because of that, they widen the first band. The sine waves
are no longer on one line in FFT but spread along several lines. The ability to recognize small sine waves
among larger ones is determined by the highest sideband attenuation and the sideband slope
attenuation. These two values determine the leakage of the FFT and that's nicely seen in the picture
below. For example, if there is a signal with a frequency of 30 Hz and an amplitude of 0.0001, we would
never see it because the 10.5 Hz signal has bigger leakage than the requested frequency signal. But if
the rectangular window is used, we would never even see the signal with an amplitude of 0.01.

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For different kinds of windows, the table below shows the values of all window properties. This is a
numerical representation of the above-mentioned rules.

Window type Maximum Width of the first Highest sideband Sideband slope
amplitude error band [line] [%] [dB/decade)
[%]

Rectangular 36 2 22 -20

Hanning 15 2 2,5 -60

Hamming 18 2 0,7 -20

Blackman 12 3 0,12 -40

Flat top 0,02 5 0,04 -20

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The image below shows the zoomed FFT of a pure sine wave, which fits the frequency line exactly. The
Abscissa axis shows the value of the line. In normal FFT, only values of the 0, 1, 2, etc. are calculated, so
only those values are shown in the FFT. We can see the width of the first sideband, the highest sideband,
and the sideband attenuation very clearly.

If a sine wave signal frequency falls between two lines, we see only the values of 0.5, 1.5, 2.5..., which
always produce higher sidebands. This is best seen if we take a function generator, set the frequency to
an exact frequency line, set the amplitude scaling to logarithmic and the FFT will look fantastic. No
leakage, exact amplitude. Now switch the frequency from the function generator to the one between
two lines in the FFT and the result will be just terrible: large amplitude errors, huge leakage.

There is one more trick with windows: if we are sure that all the frequencies will fall on their frequency
lines, a rectangular window will give us the best result. For example, to measure the harmonics of the
power line (50 Hz in Europe or higher), choose 6400 or 9600 sample/sec sampling rate, so that the line
resolution will give exactly 50, 100, 150 Hz... FFT lines, then choose a rectangular window and observe the
perfect result in the Y log scale.

6. Aliasing
The other problem comes from the fact of the signal conditioning. If simple A/D converters are used, the
sampling frequency must be at least twice higher than the maximum frequency of the signal. This is
called the Nyquist theorem (Aliasing effect). The image below shows the reason for it. Vertical lines
represent samples taken with A/D converter and the black line is the original signal. But if we look at the
orange line, which is the signal from the A/D converter, the signal is totally wrong because too few
samples per period were taken to correctly represent the signal.

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Of course, the problem above is not an FFT problem, but it is very important to know how to correctly
identify the cause of the error. And sometimes there are some lines in FFT, which can be only explained
in terms of aliases. In FFT, if we change the frequency to the ranges above the maximum frequency
limit, that line will not disappear but will bounce back and will show a fake frequency.

6.1 Aliasing example

To see that effect, a function generator and Dewesoft SIRIUS HS (high speed) with no anti-aliasing filter
are used and the online FFT analyser perfectly shows the problem.
The signal is sampled with 1 kHz.

On the upper left side of the screen, we can see the FFT of the signal recognized by hardware with no
anti-aliasing filter. On the upper right side, there is a picture of a function generator, with the output
frequency in red rectangular.

The first output frequency from a function generator was 400 Hz. Also, the frequency detected by our
hardware was 400 Hz.

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The second output frequency was 500 Hz(exactly half of our sampling rate). We can see that the
hardware with no anti-aliasing filter detects a frequency of 0 Hz. This is because of the Nyquist theorem,
which is described above.

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The third output frequency was 600 Hz. We can clearly see that the signal above 500 Hz bounces back.
Our hardware detected a signal with a frequency of 400 Hz.

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For the problem of aliasing, there is not much to be done in the FFT domain. Actually, there is absolutely
nothing we can do when the samples have already been taken. So the first thing to do would be to
choose the A/D board which has anti-aliasing filters in the front, the second thing to do would be to use
external filters or we can simply set the sampling rate to more than twice the maximum frequency
present in the signal.

7. Averaging of the signal

To enhance the result, we can use averaging of the signal in the frequency domain. Averaging means
that we calculate many FFTs during the time and average frequency lines.
There are many ways to average the signal, but the most important are linear, exponential and peak
hold average.

● linear averaging - each FFT counts the same in the results

● exponential averaging - FFTs becomes less and less important with time
● peak hold average - only maximum results are stored and shown

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There is one more thing about the averaging: loss of information. When averaging is used with window
functions, we could lose some data due to the window multiplication effects.

In the image above, there is one example where the signal only consists of one pulse. If we average the
result, use the window function and we are unlucky, the signal will fall in the region where the window
sets the values to zero, and in the resulting FFT, we will never see this pulse.

7.1 Overlapping
That's why there is a procedure called overlapping which overcomes this problem. It no longer calculates
averages one after another but takes some part of the time signal, which is already calculated and uses
it again for calculation. There could be any number for overlap, but usually, there is 25%, 50%, 66.7%, and
75% overlapping.

50% overlapping means that the calculation will take half of the old data. Now all data will be for sure
shown in the resulting FFT.

With 66.7% and higher overlapping, every sample in the time domain will count exactly the same in the
frequency domain, so if it's possible, we should use this value for overlapping to get mathematically
correct results.

What does a 'real-time' frequency analyzer mean? It means that it is able to calculate and show data
with 66.7% overlapping and, therefore, has no data loss.

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8. Representation of different signals in the FFT

All signals that are periodic in time but are not pure sine waves, produce base harmonic components as
well as additional higher harmonics. More the signal is not like a sine, the higher the harmonics are.

A harmonic of a wave is a component frequency of the signal that is an integer multiple of the
fundamental frequency f, the harmonics have frequencies 2f, 3f, 4f,. . .. The harmonics have the property
that they are all periodic at the fundamental frequency. If the fundamental frequency (first harmonic) is
25 Hz, the frequencies of the next harmonic are 50 Hz (second harmonic), 75 Hz (third harmonic), 100 Hz
(fourth harmonic), etc.

8.1 Triangle, rectangular signal

On the left side, in the picture below we can see a Triangle signal in the time domain and on the right
side is the Triangle signal in the frequency domain.

On the left side, in the picture below we can see a Rectangular signal in the time domain, and on the
right side is the Rectangular signal in the frequency domain.

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8.2 Impulse signal

An impulse is quite an interesting thing - it cannot be described as a sum of sine waves. Or in other
words: it is shown equally on all of the frequency lines. That's the reason why we use it as the basic
excitation principle to get frequency responses of the system. The other ones are swept sine and noise,
but this is already a part of another story - dual-channel frequency analysis.
On the left side, in the picture below we can see the Impulse signal in the time domain, and on the right
side is the Impulse signal in the frequency domain.

8.3 White noise

The theory says that white noise consists of all frequencies. That's why the infinite frequency spectrum
of the white noise is the straight line. The shorter the samples are, the more different amplitudes for
certain frequencies we get in the noise level. To get a fixed noise line averaging must be used. The
picture below shows an already averaged FFT of the white noise.
On the left side, in the picture below we can see White noise in the time domain, and on the right side is
White noise in the frequency domain.

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8.4 Beating (two closely spaced signals)

Beating in the time domain is somehow hidden and looks like one frequency with changing
amplitudes. Only FFT will reveal two frequency lines if a high enough line resolution is chosen. The
difference between the two frequencies is the modulation frequency shown in the time domain.
On the left side, in the picture below we can see a Beating signal in a time domain, and on the right side
is the NBeating signal in the frequency domain.

8.5 Amplitude modulated signal

The amplitude modulated (AM) signal is shown as two sideband frequencies. The difference between
the base frequency and the sideband frequency is the modulated frequency (10 Hz, in this case) also
seen clearly in the time domain. The rule here is the same as with beating - to reveal the modulation; we
should choose high enough line resolution. In fact, the time signal, which is the base for the FFT
calculation, should show some modulation peaks. When windowing is used (we know that the
baseband could be even 4 lines wide) and the main band, which is always the highest, covers the
modulation with low line resolution, time signal should show at least 16 or 32 modulation peaks that the
modulation is shown in the FFT.

On the left side, in the picture below we can see Amplitude modulated signal in the time domain, and
on the right side is Amplitude modulated signal in the frequency domain.

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9. FFT analysis in Dewesoft

In Dewesoft, we add a new FFT analysis module by selecting the + button and then the FFT Analysis..

When we add a new FFT analysis module the following setup appears:

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9.1 Output spectra

● Complex - outputs are amplitude, phase, imaginary and real part of the signal.
● Amplitude - output is the amplitude of the signal.

9.2 Amplitude function

In Amplitude function we can choose from:

● Linear - is the pure signal amplitude [V].

● Power - calculated as RMS value squared [V*V].
● PSD - calculated as RMS squared, divided by the line resolution and sqrt(2), used for checking the
noise [V*V/Hz].
● ESD - energy distribution for each unit frequency.
● ASD - acceleration spectral density.

9.3 Amplitude format

The Amplitude type display section defines display in Y-amplitude axis.

In Amplitude type we can choose from:

● Peak - is the pure signal amplitude [V].

● RMS - is the RMS amplitude, calculated as Amplitude/sqrt(2) [V].
● Power - Peak-peak - is the signal's peak-peak amplitude [V].

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9.4 Frequency weighting

As a standard, FFT analyzer uses Linear Weighting. For sound analysis, special FFT weighting can be set.
As opposed to the sound module in math, where the weightings will be calculated in the time domain,
this will calculate the sound weighting in the frequency domain.

● Linear weighting - is linear at all frequencies and it has the same effect on all measured values.
● A weighting - A-weighting is applied to instrument-measured sound levels in an effort to account
for the relative loudness perceived by the human ear, as the ear is less sensitive to low audio
frequencies.
● B weighting - B-weighting is used for intermediate levels and is similar to A, except for the fact
that low-frequency attenuation is a lot less extreme though still significant (-10 dB at 60 Hz). This
is the best weighting to use for musical listening purposes.
● C weighting - C-weighting is similar to A and B as far as the high frequencies are concerned. In
the low-frequency range, it hardly provides attenuation. This weighting is used for high-level
noise.
● D weighting - D-weighting was specifically designed for use when measuring high-level aircraft
noise. The large peak in the D-weighting curve is not a feature of the equal-loudness contours
but reflects the fact that humans hear random noise differently from pure tones, an effect that is
particularly pronounced around6 kHz

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9.5 Cross spectrum

Cross-spectrum is used as part of a frequency domain analysis of the cross-correlation between two
time series. Cross Spectrum is used for calculation of power cross-spectrum, phase spectrum and
coherence spectrum.

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9.6 Output quantity and axis unit

Frequency integration/derivation is possible directly in an FFT analyser. If the input quantity is
acceleration and we select output quantity as displacement, the FFT analyser will perform double
integration in frequency domain.

The output axis using can be selected from:

● Hz
● RPM - the X axis will have RPM units ([Hz] * 60).
● CPM - count per minute

The output unit can be selected from:

● Acceleration - m/s^2, g
● Velocity - m/s, mm/s, km/h, mph, knot, ips
● Displacement - m, km, cm, ft, ...

9.7 Averaging mode

● No averaging
● Block based - uses blocks to calculate FFT spectrum.
● Overall - gives only one averaged FFT spectrum at the end of the measurement. It will average all
the blocks in the signal and the output will be only one FFT for the whole measurement.

9.8 Averaging type

When Block cased or Overall mode are selected, there are 4 different averaging types available:

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● Linear average - linear average calculates the arithmetic mean of all the values. Linear average
gives the same weight to each amplitude value in the average.
● Energy average - The energy average is the default average type for stationary measurements.
The energy average calculates the averages of the squared values of all values. For this reason,
sometimes the term RMS (root mean square) average is used.
● Energy exp. average - The energy exponential averages the values and weights the acquisitions
taken later in time more heavily than the earlier acquisitions.
● Maximum - maximum value average takes the maximum value at the spectral line. Often this is
referred to as peak hold averaging. The maximum value average is useful for measuring the
worst case vibration.

9.9 Window functions

Windowing functions are described in chapter 4 of this manual.

9.10 Resolution settings

The resolution of the measurement results in the frequency domain can be defined in the number of
Lines, delta frequency Df [Hz] or with Block duration [s].

Block size is calculated from the selected number of lines and the defined sample rate.

9.11 DC cutoff frequency

To remove DC or low-frequency components, select from a drop-down list the DC cutoff filter - lower
limit.

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9.12 Overlap
Overlap defines the percentage of time signal that has already been calculated and it is used again for
calculation (example: 50% overlapping means that the calculation will take half of the old data).

More information about overlap can be found in chapter 7.1 in this manual.

9.13 Output scalars

The option output scalars will output a scalar (a number) that will be calculated in a selected frequency
range.

● RMS
● Peak
● Peak-Peak

For example, if RMS is selected as the output scalar and the whole frequency range is selected, the
output channel is Overall RMS value.

9.14 Harmonics
THD - Total Harmonic Distortion is a useful technique to analyze any non‑linear behavior of a system.

The frequency type can be from:

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● spectrum,
● fixed frequency or
● from input channel.

10. FFT markers

To learn more about processing markers please take a look at the manual.

The 2D graph can display values of the currently selected point with the crosshair cursor. When clicking
on such a point with the left mouse button, the marker line will be added showing the x-axis value on
the x-axis and showing the y-axis value of a certain point above the marked point. All points can be
removed by pressing the right mouse button and selecting the Delete selected marker.

Let's make a square wave with a frequency of 200 Hz in Dewesoft math and put the signal into the FFT
analyzer.

When we go to measure and FFT graph, we can see that the square wave is composed of a sum of sine
waves with different frequencies. We can see those frequencies as peaks in the FFT graph, but now we
would like to know the exact position of those peaks.

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10.1 Free marker

Free markers can be freely added. The marker shows us the frequency of the peak at which it stands
and its amplitude.

With the Show marker table selected you can see the table of markers - its ID, type, channel, color, its
frequency (X-axis), and its amplitude (Y-axis). You can select if you want markers to be visible or not, you
can also edit and remove them.

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10.2 MAX marker

Max marker finds the highest amplitude in the spectrum. Move the mouse to the FFT graph and select
icon for Max marker. When we select Max marker and click on the FFT graph with the left mouse button
the following setup opens.

First, we select the FFT curve to which the marker is related. The position is calculated by the program.

Interpolation - estimating frequency and amplitude

Depending on the selected window type, the frequency component (actual peak) can fall in between
two adjacent lines.

In the example below, we have a signal with a frequency of 256.5 Hz and an amplitude of 1. The
frequency resolution in our case is 2 Hz. When we add a free marker on the peak (non-interpolated), we
see that the marker is at 256 Hz and has an amplitude of 0,97 (because the amplitude is split between
two peaks).

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If we want to get the exact value, we have to interpolate the peak. To get the right interpolation, at least
three lines on each side (left and right) have to have a smaller value than the peak. Now, the frequency
of the peak is in the exact position. Also, the amplitude has the right value.

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It is possible to estimate the actual frequency and amplitude to a greater resolution than given by delta
frequency (df). Dewesoft uses a weighted average of the values around a detected peak to calculate
exact frequency and amplitude values.

Also, if two or more frequency peaks are within six lines of each other, they contribute to inflating the
estimated powers and skewing the actual frequencies. But anyway, if two peaks are that close, they are
probably already interfering with one another because of the spectral leakage.

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Number of peaks

Then we select a number of peaks we want to find. If that number is 1 program will only find the peak
with maximum amplitude. If that number is 3 it will find also the peak with the third-highest amplitude.
The picture below shows the max marker with 3 number of peaks selected.

10.3 RMS marker

RMS marker will sum up all the FFT lines in the selected band and calculate the RMS value. RMS marker
calculates RMS value of the channel between cursors or between defined areas.

The RMS value of the channel between cursors can also be adjusted by dragging the cursor with a
mouse. RMS will be calculated automatically if the area changes.

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10.4 Sideband marker

The sideband marker monitors the modulated frequencies to the left and right from the selected
centerline.

Let's generate an amplitude modulated signal with a carrier frequency of 1000 Hz and the baseband
signal with a frequency of 100 Hz.

Sideband markers have a center marker and several equally spaced sideband markers. By selecting the
center marker, you can drag the sideband markers to different positions while still maintaining the
individual sideband space.

Each sideband cursor can be selected and moved to a different frequency hence changing the
individual ratio of the sidebands with respect to that of the center cursor.

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The sideband marker draws markers around the selected peak. We have to define the number of bands
(for how many bands in each direction we want to see drawn lines) and Delta (distance between bands
in Hz). For example, the selected position is 1000 Hz, a number of bands are 1, and Delta frequency is 100.

We can see that the central position is at 1000 Hz and we have one band in each direction. So the line on
the left side is at 900 Hz and the line on the right side is at 1100 Hz. Distance between the lines can be
defined by the user, in our example, it was 100 Hz.

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10.5 Harmonic marker

The harmonic marker is a great help when identifying the fundamentals of the frequency.

The harmonic marker can be enabled at any frequency. The harmonic marker will mark the harmonics
of the selected frequency. The base marker of the harmonic marker can be selected and moved to any
other frequency with the harmonics updated live.

Monitoring harmonics is very important in the order tracking analysis. An example was made with a blue
toy in the picture below (an accelerometer was attached to the machine). We run the machine to 3000
RPMs and measure vibrations in the process.

Move the mouse to the FFT graph and select the icon for a Harmonic marker. Then select the base
frequency with the mouse and add a harmonic marker with the click on the left button. We select the
first peak at 21.97 Hz.

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If we select the Number of harmonics as 3, we will see lines at 21.77 Hz, 43.54 Hz (2 x 21.77 Hz), and at 65.31
Hz (3 x 21.77 Hz). And the theoretical harmonics also nicely match our measurement results - the first
three harmonics are nicely seen.

You can also pick and drag the fundamental frequency through the FFT spectrum. Harmonics will
automatically follow.

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10.6 Damping marker

Damping markers are best to use in modal testing when we want to find out how our transfer curve is
damped. We select it when we are interested in the quality factor, damping ratio, or attenuation rate of a
selected peak.

Damping factor type can be selected from the following options:

Q factor

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The Q (quality) factor of the damped system is defined as:

𝑓𝐶 𝑓𝐶
𝑄= 𝑓2−𝑓1
= ∆𝑓

The higher the Q, the narrower, and 'sharper' the peak is.

Damping ratio

Damping ratio and quality factor Q are related through the equation:

1
ζ = 2𝑄

Attenuation ratio

Attenuation is the gradual loss in intensity of any kind of flux through a medium. It is usually measured
in units of decibels per unit length of the medium.

In the picture below we can see a transfer curve of a beam. On each of the peaks, we attach a damping
factor and in the marker table we can see the quality factor (Q), which tells us how much the transfer
curve is damped.

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If the Damping factor type is chosen as a Damping ratio, the result is Zeta for each peak.

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If the Damping factor type is chosen as Attenuation, the result is the attenuation ratio for each peak.

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10.7 Kinematic cursor

Kinematic cursors are used to identify the bearing frequencies and bearing faults.

Each bearing database includes bearing data (what is the base of the component (cage, rolling element,
outer race, and inner race) at 1 Hz and at which frequency has the component a peak in the frequency
domain).

To add a new bearing go to the Kinematic cursor editor.

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In the Kinematic cursor editor, add a new bearing or select it from the existing database, selecting the
Append bearing option.

To use the Kinematic cursor we have to add the Envelope detection math channel.

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Calculation type defines the principle of calculation.

● Filtering - uses filter procedure for envelope calculation. Filtering is a standard procedure for
calculating envelopes used also in other implementations.
● Peak detection - uses the procedure of detecting peak values in the signal. Peak detection is a
procedure which calculates amplitudes more accurately than filtering.
● Bandpass - checkbox enables or disables the first stage of calculation - band pass filtering.

Channel calculated with Envelope detection math must be now set as the input channel to the FFT
analyzer.

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The frequency spectrum from FFT is displayed on a 2D graph. To add a Kinematic marker on a 2D graph,
right click on the spectrum and Add Kinematic marker.

After adding a Kinematic marker, the setup appears. Users must select which bearing is used and what
is the frequency of rotation (from manual position or from channel value).

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As a result, kinematic markers indicate the frequency of the bearing fault in the spectrum.

Please note that in order to use Kinematic marker you need to have a valid license for FFT
analyser or a DSA license.

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10.8 Interpolation

The FFT lines are responsible for the frequency resolution. The higher the FFT lines value, the better the
resolution. This line resolution depends on the sampling rate and the number of lines chosen for the
FFT. So if we want to have fast response on the FFT, we choose fewer lines, but we will have a lower
frequency resolution. If we want to see the exact frequency, we set a higher line resolution.
If our peak falls between frequency lines, the frequency will not be exact.

Because harmonics are multipliers of the fundamental frequency, the error will increase at every higher
harmonic.

Non interpolated peak and the error that increases with harmonics.

If we mark the interpolate peak options, our markers will be interpolated in frequency and in amplitude!

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11. Warranty information

Notice
The information contained in this document is subject to change without notice.

Note:
Dewesoft d.o.o. shall not be liable for any errors contained in this document. Dewesoft MAKES NO
WARRANTIES OF ANY KIND WITH REGARD TO THIS DOCUMENT, WHETHER EXPRESS OR IMPLIED.
DEWESOFT SPECIFICALLY DISCLAIMS THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS
FOR A PARTICULAR PURPOSE. Dewesoft shall not be liable for any direct, indirect, special, incidental, or
consequential damages, whether based on contract, tort, or any other legal theory, in connection with
the furnishing of this document or the use of the information in this document.

The copy of the specific warranty terms applicable to your Dewesoft product and replacement parts can
be obtained from your local sales and service office. To find a local dealer for your country, please visit
https://dewesoft.com/support/distributors.

11.1 Calibration
Every instrument needs to be calibrated at regular intervals. The standard norm across nearly every
industry is annual calibration. Before your Dewesoft data acquisition system is delivered, it is calibrated.
Detailed calibration reports for your Dewesoft system can be requested. We retain them for at least one
year, after system delivery.

11.2 Support
Dewesoft has a team of people ready to assist you if you have any questions or any technical difficulties
regarding the system. For any support please contact your local distributor first or Dewesoft directly.

Dewesoft d.o.o.
Gabrsko 11a
1420 Trbovlje Slovenia

Europe Tel.: +386 356 25 300

Web: http://www.dewesoft.com
Email: [email protected]
The telephone hotline is available Monday to Friday from 07:00 to 16:00 CET (GMT +1:00)

11.3 Service/repair
The team of Dewesoft also performs any kinds of repairs to your system to assure a safe and proper
operation in the future. For information regarding service and repairs please contact your local
distributor first or Dewesoft directly on https://dewesoft.com/support/rma-service.

11.4 Restricted Rights

Use Slovenian law for duplication or disclosure. Dewesoft d.o.o. Gabrsko 11a, 1420 Trbovlje, Slovenia /
Europe.

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11.5 Printing History

Version 2.1.0, Revision 2021 Released 2021 Last changed: 24. June 2021 at 15:00.

11.6 Copyright
Copyright © 2015-2021 Dewesoft d.o.o. This document contains information which is protected by
copyright. All rights are reserved. Reproduction, adaptation, or translation without prior written
permission is prohibited, except as allowed under the copyright laws. All trademarks and registered
trademarks are acknowledged to be the property of their owners.

11.7 Trademarks
We take pride in our products and we take care that all key products and technologies are registered as
trademarks all over the world. The Dewesoft name is a registered trademark. Product families
(KRYPTON, SIRIUS, DSI, DS-NET) and technologies (DualCoreADC, SuperCounter, GrandView) are
registered trademarks as well. When used as the logo or as part of any graphic material, the registered
trademark sign is used as a part of the logo. When used in text representing the company, product or
technology name, the ® sign is not used. The Dewesoft triangle logo is a registered trademark but the ®
sign is not used in the visual representation of the triangle logo.

12. Safety instructions

Your safety is our primary concern! Please be safe!

12.1 Safety symbols in the manual

Warning
Calls attention to a procedure, practice, or condition that could cause the body injury or death

Caution
Calls attention to a procedure, practice, or condition that could possibly cause damage to
equipment or permanent loss of data.

12.2 General Safety Instructions

Warning

The following general safety precautions must be observed during all phases of operation, service, and
repair of this product. Failure to comply with these precautions or with specific warnings elsewhere in
this manual violates safety standards of design, manufacture, and intended use of the product.
Dewesoft d.o.o. assumes no liability for the customer’s failure to comply with these requirements.

All accessories shown in this document are available as an option and will not be shipped as standard
parts.

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12.2.1 Environmental Considerations

Information about the environmental impact of the product.

12.2.2 Product End-of-Life Handling

Observe the following guidelines when recycling a Dewesoft system:

12.2.3 System and Components Recycling

Production of these components required the extraction and use of natural resources. The substances
contained in the system could be harmful to your health and to the environment if the system is
improperly handled at its end of life! Please recycle this product in an appropriate way to avoid
unnecessary pollution of the environment and to keep natural resources.

This symbol indicates that this system complies with the European Union’s requirements
according to Directive 2002/96/EC on waste electrical and electronic equipment (WEEE).
Please find further information about recycling on the Dewesoft web site www.dewesoft.com

Restriction of Hazardous Substances

This product has been classified as Monitoring and Control equipment and is outside the
scope of the 2002/95/EC RoHS Directive. However, we take care of our environment and the product is
lead-free.

12.2.4. General safety and hazard warnings for all Dewesoft systems
Safety of the operator and the unit depend on following these rules.
● Use this system under the terms of the specifications only to avoid any possible danger.
● Read your manual before operating the system.
● Observe local laws when using the instrument.
● DO NOT touch internal wiring!
● DO NOT use higher supply voltage than specified!
● Use only original plugs and cables for harnessing.
● You may not connect higher voltages than rated to any connectors.
● The power cable and connector serve as Power-Breaker. The cable must not exceed 3 meters, the
disconnect function must be possible without tools.
● Maintenance must be executed by qualified staff only.
● During the use of the system, it might be possible to access other parts of a more comprehensive
system. Please read and follow the safety instructions provided in the manuals of all other
components regarding warning and security advice for using the system.
● With this product, only use the power cable delivered or defined for the host country.
● DO NOT connect or disconnect sensors, probes or test leads, as these parts are connected to a
voltage supply unit.
● Ground the equipment: For Safety Class 1 equipment (equipment having a protective earth
terminal), a non-interruptible safety earth ground must be provided from the mains power
source to the product input wiring terminals.
● Please note the characteristics and indicators on the system to avoid fire or electric shocks.
Before connecting the system, please read the corresponding specifications in the product
manual carefully.

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● The inputs must not, unless otherwise noted (CATx identification), be connected to the main
circuit of category II, III and IV.
● The power cord separates the system from the power supply. Do not block the power cord, since
it has to be accessible for the users.
● DO NOT use the system if equipment covers or shields are removed.
● If you assume the system is damaged, get it examined by authorized personnel only.
● Adverse environmental conditions are Moisture or high humidity Dust, flammable gases, fumes
or dissolver Thunderstorm or thunderstorm conditions (except assembly PNA) Electrostatic
fields, etc.
● The measurement category can be adjusted depending on module configuration.
● Any other use than described above may damage your system and is attended with dangers like
short-circuiting, fire or electric shocks.
● The whole system must not be changed, rebuilt or opened.
● DO NOT operate damaged equipment: Whenever it is possible that the safety protection features
built into this product have been impaired, either through physical damage, excessive moisture,
or any other reason, REMOVE POWER and do not use the product until the safe operation can be
verified by service-trained personnel. If necessary, return the product to Dewesoft sales and
service office for service and repair to ensure that safety features are maintained.
● If you assume a more riskless use is not provided anymore, the system has to be rendered
inoperative and should be protected against inadvertent operation. It is assumed that a more
riskless operation is not possible anymore if the system is damaged obviously or causes strange
noises. The system does not work anymore. The system has been exposed to long storage in
adverse environments. The system has been exposed to heavy shipment strain.
● Warranty void if damages caused by disregarding this manual. For consequential damages, NO
liability will be assumed!
● Warranty void if damage to property or persons caused by improper use or disregarding the
safety instructions.
● Unauthorized changing or rebuilding the system is prohibited due to safety and permission
reasons (CE).
● Be careful with voltages >25 VAC or >35 VDC! These voltages are already high enough in order to
get a perilous electric shock by touching the wiring.
● The product heats during operation. Make sure there is adequate ventilation. Ventilation slots
must not be covered!
● Only fuses of the specified type and nominal current may be used. The use of patched fuses is
prohibited.
● Prevent using metal bare wires! Risk of short circuit and fire hazard!
● DO NOT use the system before, during or shortly after a thunderstorm (risk of lightning and high
energy over-voltage). An advanced range of application under certain conditions is allowed with
therefore designed products only. For details please refer to the specifications.
● Make sure that your hands, shoes, clothes, the floor, the system or measuring leads, integrated
circuits and so on, are dry.
● DO NOT use the system in rooms with flammable gases, fumes or dust or in adverse
environmental conditions.
● Avoid operation in the immediate vicinity of high magnetic or electromagnetic fields,
transmitting antennas or high-frequency generators, for exact values please refer to enclosed
specifications.
● Use measurement leads or measurement accessories aligned with the specification of the
system only. Fire hazard in case of overload!

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● Do not switch on the system after transporting it from a cold into a warm room and vice versa.
The thereby created condensation may damage your system. Acclimatise the system unpowered
to room temperature.
● Do not disassemble the system! There is a high risk of getting a perilous electric shock.
Capacitors still might be charged, even if the system has been removed from the power supply.
● The electrical installations and equipment in industrial facilities must be observed by the security
regulations and insurance institutions.
● The use of the measuring system in schools and other training facilities must be observed by
skilled personnel.
● The measuring systems are not designed for use in humans and animals.
● Please contact a professional if you have doubts about the method of operation, safety or the
connection of the system.
● Please be careful with the product. Shocks, hits and dropping it from already- lower level may
damage your system.
● Please also consider the detailed technical reference manual as well as the security advice of the
connected systems.
● This product has left the factory in safety-related flawlessness and in proper condition. In order to
maintain this condition and guarantee safety use, the user has to consider the security advice
and warnings in this manual.

EN 61326-3-1:2008
IEC 61326-1 applies to this part of IEC 61326 but is limited to systems and equipment for industrial
applications intended to perform safety functions as defined in IEC 61508 with SIL 1-3.

The electromagnetic environments encompassed by this product family standard are industrial, both
indoor and outdoor, as described for industrial locations in IEC 61000-6-2 or defined in 3.7 of IEC 61326-1.

Equipment and systems intended for use in other electromagnetic environments, for example, in the
process industry or in environments with potentially explosive atmospheres, are excluded from the
scope of this product family standard, IEC 61326-3-1.

Devices and systems according to IEC 61508 or IEC 61511 which are considered as “operationally
well-tried”, are excluded from the scope of IEC 61326-3-1.

Fire-alarm and safety-alarm systems, intended for the protection of buildings, are excluded from the
scope of IEC 61326-3-1.

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13. Documentation version history

Last modified: Friday 30 September 2021

Version Date Notes

[dd.mm.yyyy]

V21-1 29.05.21 ☑ initial revision

V21-2 30.09.21 ☑ updated screenshots of the UI, updated description of averaging

modes and types

V21-3 29.10.21 ☑ Updated description of Kinematic marker

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