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FFT Spectral Analysis

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):

What is frequency analysis?

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.

Rec
1
sine(0,5) [-]

Rec
2.2326

t[s]
-1

+
2.5 5.0 7.5 9.9 12.4
1.1163

=
sine(1) [-]
signal [-]
0

t[s]
-1

+
2.5 5.0 7.5 9.9 12.4
-1.1163

1
sine(2) [-]
t[s]
-2.2326

2.5 5.0 7.5 9.9 12.4

t[s]
-1

2.5 5.0 7.5 9.9 12.4

1
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.

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

2
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.

Fourier series
Any periodic waveform can be decomposed into a series of sine and cosine waves:

where a0, an, and bn are Fourier coefficients:

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:

3
where w is a complex nth root of unity:

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.

Let's think about how the equation for discrete Fourier transform works:

4
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 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. 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.

5
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 duplicated 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:

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.

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. FFT algorithm can compute the same
result with only (N/2)log2(N) complex multiplications and Nlog2(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.

On 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]).

6
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.

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Properties of Fourier transform
In the image 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:

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 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:

8
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.

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

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.

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X scale can be either linear or logarithmic. Linear scaling is the correct representation of the mathematic 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.

10
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.

11
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 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 occurring close to larger ones.

“Leakage” from a sinusoid (rectangular window)

0
MAIN LOBE
SIDELOBE LEVEL
-10 S ID
ELO
BES

-20
deciels

-30

-40

-50

-60
-4 -3 -2 -1 0 1 2 3 4 5 6 7
DFT bins

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.

12
In the picture below we can see the effect of windowing in a signal.

Image 12: Periodized signal with discontinuities Image 13: Discontinuities "ironed out" by windowing

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

0 0 0

-10 -10 -10

-20 -20 -20

-30 -30 -30

-40 -40 -40

gain (dB)

gain (dB)

gain (dB)

-50 -50 -50

-60 -60 -60

-70 -70 -70

-80 -80 -80

-90 -90 -90

-100 -100 -100

0 10 20 30 40 50 60 0 10 20 30 40 50 60 0 10 20 30 40 50 60
frequency bin frequency bin frequency bin

Image 16: Adjacent peak spectrum

Image 14: Spectrum display without Image 15: Small frequency peak
display with windowing, smaller
spectral leakage obscured as a result of leakage
frequency peak is no longer obscured.

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

13
 Amplitude
(dB)
Main Lobe Peak Amplitude

-3 dB

-6 dB

Sidelobe rollof rate

-3dB Main -6dB Main

Lobe width Lobe width

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.

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.

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15
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.

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.

0,9

0,8

0,7

0,6

0,5

0,4

0,3

0,2

0,1

0
0 4 8 12 16 20
AMPLITUDE ERROR AND LEAKAGE OF THE FFT (NO WINDOW)

16
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.

Hamming
Hanning 0
Flattop

-1
WINDOWS IN THE TIME DOMAIN

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
17
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.

1 1

0,9 0,9

0,8 0,8

0,7 0,7

0,6 0,6

0,5 0,5

0,4 0,4

0,3 0,3

0,2 0,2

0,1 0,1

0 0
0 4 8 12 18 20 0 4 8 12 18 20
HANNING WINDOW FLAT TOP WINDOW

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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 1

Amplitude
error

0,1

Highest
sideband

0,01

0,001

Sideband
slope

0,0001

First band width

10-⁵
0 10 20 30 40 50

WINDOW PROPERTIES DESCRIPTION

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.

Maximum amplitude Width of the first band Sideband slope

Window type Highest sideband [%]
error [%] [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

The image below shows zoomed FFT of a pure sine wave, which fits the frequency line exactly. 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.

19
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 exact 50, 100, 150 Hz... FFT lines, then
choose a rectangular window and observe the perfect result in the Y log scale.

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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.

ALIASING

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.

Aliasing example
To see that effect, a function generator and Dewesoft SIRIUS HS (high speed) with no anti-aliasing filter are used and the

21
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.

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.

22
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.

23
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.

24
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

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.

SIGNAL

WINDOWING AND AVERAGING WITH

0% OVERLAP

WINDOWING AND AVERAGING WITH

50% OVERLAP

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.

Overlapping
That's why there is a procedure called overlapping which overcomes this problem. It no longer calculates averages one after
25
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.

Real-time frequency analyzer

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.

26
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 harmonics are 50 Hz (second
harmonic), 75 Hz (third harmonic), 100 Hz (fourth harmonic), etc.

1.) Triangle, rectangular

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.

27
2.) Impulse
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.

3.) White noise

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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.

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.

29
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.

30
31
FFT analysis module in Dewesoft
In Dewesoft, we add a new FFT analysis module by selecting the + button and then the FFT Analyser.

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

32
Output spectrums

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

Calculation type

Block history calculation type uses blocks to calculate the FFT spectrum.

For example, let's set the FFT resolution to 1024 lines.

33
The FFTs are acquired shot by shot and put into the buffer. For every 1024 lines, a new FFT will be calculated and shown on
the 2D graph.

With the block history calculation type, we can average more blocks together.

Let's say we want to average 2 blocks of the signal. Each new FFT will be calculated for two blocks together.

34
FFT spectrum can also be observed on the 3D graph.

Overall (Averaged) calculation type 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.

35
Calculation parameters

36
Window functions were already described on previous pages.

Resolution can be defined in the number of Lines or with the delta frequency Df (Hz).

The FFT lines are responsible for the frequency resolution. The higher the FFT lines value, the better the resolution - but also
longer calculation time.

The 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. The simple rule is: if it takes 1 second to acquire the data from which the FFT is
calculated, the resulting FFT will have a 1 Hz line resolution. If we acquire data for 2 seconds then the line resolution will be 0.5
Hz.

Example: The sampling rate has been set to 10000 samples/sec and the resolution of 1024 FFT lines. These settings allow an
FFT analysis of up to 5000 Hz (half the sampling rate). Now you divide the max analyses frequency by the FFT lines (5000 Hz /
1024 lines). The result is 4.88 Hz per line resolution (mentioned in the selection line).

Amplitude type

37
The Amplitude type display section defines display in the Y-amplitude axis.

From the Amplitude scaling type display drop-down list, we can select different types of amplitude scaling of the FFT. The
basic setting is Amplitude (Auto), which shows for pure sine wave the amplitude of the sine.

Real - is the pure signal amplitude [V]

RMS - is the RMS amplitude, calculated as Amplitude/sqrt(2) [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]
RMS SD - calculated as RMS value, divided by the square root of line resolution - also used for checking the noise
[V/sqrt(Hz)]
Peak-peak - is the difference in amplitude from signals negative peak and positive peak [V]

DC cutoff

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

Overlap
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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).

When the window type is used, we have to use an overlap otherwise some of the data will be ignored. Therefore, the use of
overlap is highly recommended.

Weighting

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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

Averaging type
If we chose the Calculation type as Overall (Averaged), we have to select also the Averaging type.

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

FFT visual control

There is another option on how to get an FFT on a signal. During the measurement add an FFT preview widget. Click the
Design button and then add an FFT preview widget by clicking on the icon.

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The FFT visual control can display the position and amplitude of maximum peaks, RMS values, or marked peaks.

Difference between FFT analysis module and FFT visual

control
How FFTs collect data:

Visual FFT always takes the values left and right in an equal amount from a position of the yellow cursor.
FFT analysis module takes the values in the block from where you can see the timestamps. The start is on the first
stamp and the end is on the second stamp.

Visual FFT is more dynamic to get quick look wherever you put yellow cursor and Math FFT gives you exact block so you know
from where to where some FFT is exactly calculated.

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Example of measurement with FFT analyser
For a measurement example, we used a blue toy with an electromotor and an encoder. An accelerometer was placed on the
housing of the toy. When we run the machine up to 3000 RPMs the machine vibrates.

To observe the behavior of the machine we add an FFT analyzer. The input signal is an accelerometer signal that is attached to
the rotating machine.

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Before we run the machine, let's add a visual control 3D graph.

Select the design button and add a 3D graph widget.

The next step is to select the channel that will be shown in the graph. In our example, it was the signal from the FFT analyzer.

When we run the machine, we clearly see the first harmonic. On 2D graph, we will see amplitude [m/s2] plotted against
frequency [Hz].

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If we want to change the view from 2D to 3D we just select different projection types.

Now we have added also a time domain to our graph - if we take a look at a signal from the FFT analyzer in a 3D graph, we can
see how the harmonics are evolving in time.

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FFT markers

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