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Background
Spectrum Analysis
Background
This section will cover the operation and theory of the FFT analyzer,
which is the most commonly used piece of signal analysis
equipment in the vibration field. Many workers think of the FFT
analyzer as a "magic box," into which you put a signal and out of
which comes a spectrum. The assumption usually is that the
spectrum tells the truth -- the box cannot lie. We will see that this
assumption is valid in many cases, but we will also see that we can
be misled, for there are several pitfalls in the process of digital
signal analysis. One of the purposes of this section is to help you
avoid falling into any of the pitfalls, and if you do, how to crawl out
smelling like a rose.
FFT analysis is but one type of digital spectrum analysis, but we will
not concentrate on the other types because they do not apply
directly to the VMS program.
Spectrum Analysis
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The Fourier Series operates on a time signal that is periodic, i.e., a
time signal whose waveform repeats over and over again out to
infinite time. Fourier showed that such a signal is equivalent to a
collection of sine and cosine functions whose frequencies are
multiples of the reciprocal of the period of the time signal. The
rather unexpected result is that any wave shape whatsoever, as
long as it is not infinite in length, can be represented, as the sum of
a collection of harmonic components, and the fundamental
frequency of the harmonic series is 1 divided by the length of the
wave shape. The amplitudes of the various harmonics are called the
Fourier coefficients, and their values can be calculated easily if the
equation for the wave shape is known. They can also be calculated
graphically from the wave shape itself. A certain physics class is
known to have done this with the silhouette of Marilyn Monroe.
They posted the MM coefficients on the bulletin board as an "in"
joke.
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Fourier Coefficients
Fourier Coefficients
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over a wide frequency range and vice versa. This was seen in the
Introduction to Vibration chapter where a spectrum of a short
impulse is shown.
The Discrete Fourier Transform
Neither the Fourier Series nor the Fourier Transform lends itself
easily to calculation by digital computers. To overcome this hurdle,
the so-called Discrete Fourier Transform, or DFT was developed.
Probably the first person to conceive the DFT was Wilhelm
Friederich Gauss, the famous 19th century German mathematician,
although he certainly did not have a digital computer on which to
implement it. The DFT operates on a sampled, or discrete, signal in
the time domain, and generates from this a sampled, or discrete,
spectrum in the frequency domain. The resulting spectrum is an
approximation of the Fourier Series, an approximation in the sense
that information between the samples of the waveform is lost. The
key to the DFT is the existence of the sampled waveform, i.e., the
possibility of representing the waveform by a series of numbers. To
generate this series of numbers from an analog signal, a process of
sampling and analog to digital conversion is required. The sampled
signal is a mathematical representation of the instantaneous signal
level at precisely defined time intervals. It contains no information
about the signal between the actual sample times.
If the sampling rate is high enough to ensure a reasonable
representation of the shape of the signal, the DFT does produce a
spectrum very close to a theoretically true spectrum. This spectrum
is also discrete, and there is no information between the samples,
or "lines" of the spectrum. In theory, there is no limit to the number
of samples that can be used, or the speed of the sampling, but
there are practical limitations we must live with. Most of these
limitations are the result of using a digital computer as the
calculating agent.
The Fast Fourier Transform
In order to adapt the DFT for use with digital computers, the so-
called Fast Fourier Transform (FFT) was developed. The FFT is
simply an algorithm for calculating the DFT in a fast and efficient
manner.
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Cooley and Tukey are credited with the discovery of the FFT in
1967, but it existed much earlier, although without the digital
computers needed to exploit it. The FFT algorithm places certain
limitations on the signal and the resulting spectrum. For instance,
the sampled signal to be transformed must consist of a number of
samples equal to a power of two. Most FFT analyzers allow 512,
1024, 2048, or 4096 samples to be transformed. The frequency
range covered by FFT analysis depends on the number of samples
collected and on the sampling rate, as will be explained shortly.
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It can be seen here that the sampling rate determines the highest
frequency in the signal that can be encoded. The sampled waveform
cannot know anything about what happens in the signal between
the sampled times. Claude Shannon, the developer of the branch of
mathematics called information theory, determined that to encode
all the information in a signal being sampled, the sampling
frequency must be at least double the highest frequency present in
the signal. This fact is sometimes called the Nyquist criterion.
Aliasing
Aliasing
Here the actual signal is represented in black and the sampled
representation of it is in gray. The vertical lines represent the
sampling frequency. Note that if the sampling frequency is the same
as the sampled frequency, each sample is the same size, and the
output of the sampling circuit will be a constant direct voltage --
obviously having no relation to frequency of the input signal.
Now note what happens if the actual signal is higher in frequency
than the sampling frequency. The sampler output looks like a very
low frequency, and again it is not a correct representation of the
actual signal. This phenomenon is called aliasing, and it can lead to
gross errors unless it is avoided. The best way to avoid aliasing is to
pass the input signal through an analog low-pass filter whose cut-
off frequency is less than one-half the sampling frequency. In most
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modern FFT analyzers, the sampling frequency is set to 2.56 times
the filter cut-off frequency. The filter must have a very sharp cut off
characteristic, or roll off, and this means it will also have Phase Shift
that can affect the data if one needs phase information near the
upper end of the frequency span of the analyzer. To avoid this,
select a frequency span so the frequency in question is in the lower
half of the frequency range. This is important in performing
balancing with an FFT analyzer, where phase of the 1X vibration
signal is needed.
Aliasing also occurs in other media, such as motion pictures. For
instance, sometimes in western movies the wagon wheel spokes
may appear stopped, or rotating backward. This is optical aliasing,
for a movie is a sampled representation of the original motion.
Another example of optical aliasing is the stroboscope, which is set
to flash at a rate equal to or near the rotation rate of the object
being observed, making it appear stationary or slowly turning.
Sampling Rules for Digital Signal Analysis
The data path must contain an analog Anti-Aliasing low-pass filter
You must sample at least twice as fast as the highest frequency to
be analyzed
The Frequency Response of the analysis depends on the sampling
frequency
These rules apply to all FFT analysis, and the analyzer automatically
takes care of them. The anti-aliasing filter is internally set to the
appropriate value for each frequency range of the analyzer. The
total sampling time is called the time record length and the nature
of the FFT dictates that the spacing between the frequency
components in the spectrum (also called the frequency resolution)
is 1 divided by the record length. For instance, if the frequency
resolution is one Hz, then the record length is one second, and if
the resolution is 0.1 Hz, then the record length is 10 seconds, etc.
From this it can be seen that in order to perform high resolution
spectrum analysis relatively long times are required to collect the
data. This has nothing to do with the speed of the calculations in
the analyzer; it is simply a natural law of frequency analysis.
Leakage
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The FFT analyzer is a batch processing device; that is it samples the
input signal for a specific time interval collecting the samples in a
buffer, after which it performs the FFT calculation on that "batch"
and displays the resulting spectrum
If a sinusoidal signal waveform is passing through zero level at the
beginning and end of the time record, i.e., if the time record
encompasses exactly an integral number of cycles of the waveform,
the resulting FFT spectrum will consist of a single line with the
correct amplitude and at the correct frequency. If, on the other
hand, the signal level is not at zero at one or both ends of the time
record, truncation of the waveform will occur, resulting in a
discontinuity in the sampled signal. This discontinuity is not handled
well by the FFT process, and the result is a smearing of the
spectrum from a single line into adjacent lines. This is called
"leakage"; it is as if the energy in the signal "leaks" from its proper
location into the adjacent lines.
The shape of the "leaky" spectrum depends on the amount of signal
truncation, and is generally unpredictable for real signals.
Windows
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this. The most common forms of windows and their uses are
considered next.
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dependent upon the frequency range of the analysis. Most FFT
analyzers allow the user to see the time record on the screen, so it
can be assured that this condition is met.
The Hanning Window
The Hanning window, after its inventor whose name was Von Hann,
has the shape of one cycle of a cosine wave with 1 added to it so it
is always positive. The sampled signal values are multiplied by the
Hanning function, and the result is shown in the figure. Note that
the ends of the time record are forced to zero regardless of what
the input signal is doing.
While the Hanning window does a good job of forcing the ends to
zero, it also adds distortion to the wave form being analyzed in the
form of amplitude modulation; i.e., the variation in amplitude of the
signal over the time record. Amplitude Modulation in a wave form
results in sidebands in its spectrum, and in the case of the Hanning
window, these sidebands, or side lobes as they are called,
effectively reduce the frequency resolution of the analyzer by 50%.
It is as if the analyzer frequency "lines" are made wider. In the
illustration here, the curve is the actual filter shape that the FFT
analyzer with Hanning weighting produces. Each line of the FFT
analyzer has the shape of this curve -- only one is shown in the
figure.
If a signal component is at the exact frequency of an FFT line, it will
be read at its correct amplitude, but if it is at a frequency that is
one half of delta F (One half the distance between lines), it will be
read at an amplitude that is too low by 1.4 dB.
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The illustration shows this effect, and also shows the side lobes
created by the Hanning window. The highest-level side lobes are
about 32 dB down from the main lobe.
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frequency and phase content of a transient is intimately connected
with its shape.
The measured level will also be greatly distorted. Even if the
transient were in the center of the Hanning window, the measured
level would be twice as great as the actual level because of the
amplitude correction the analyzer applies when using the Hanning
weighting.
A Hanning weighted signal actually is only half there, the other half
of it having been removed by the windowing. This is not a problem
with a perfectly smooth and continuous signal like a sinusoid, but
most signals we want to analyze, such as machine vibration
signatures are not perfectly smooth. If a small change occurs in the
signal near the beginning or end of the time record, it will either be
analyzed at a much lower level than its true level, or it may be
missed altogether. For this reason, it is a good idea to employ
overlap processing. To do this, two time buffers are required in the
analyzer. For 50% overlap, the sequence of events is as follows:
When the first buffer is half full, i.e., it contains half the samples of
a time record, the second buffer is connected to the data stream
and also begins to collect samples. As soon as the first buffer is full,
the FFT is calculated, and the buffer begins to take data again.
When the second buffer is filled, the FFT is again calculated on its
contents, and the result sent to the spectrum-averaging buffer. This
process continues on until the desired number of averages is
collected.
Overlap Processing
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If the overlap is 2/3, i.e., 66.7%, then the overall time weighting of
the data will be flat, and there is no advantage to using a greater
overlap. Most data collection for machinery analysis uses 50% data
overlap, which provides adequate amplitude accuracy for most
vibration work.
Here is a summary of the relationship between sampling rate,
number of samples, time record length, and frequency resolution
that affect FFT analysis. The sampling rate in samples per second,
times the time record length T in seconds, equals the number of
samples N. In the FFT analyzer, the number of samples N is
constrained to a power of two.
FFT Fundamentals
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The FFT algorithm, operating on N samples of time data produces
N/2 frequency lines. Thus a time record of 512 samples will
generate a spectrum of 256 lines. FFT analyzers generally do not
display the upper spectral lines because of the possibility of their
being contaminated by aliased components. This is because the
anti-aliasing filter is not perfect, and has a finite slope in its cut-off
range. Therefore, a 256 line spectrum will be displayed as a 200
line spectrum, and a 512-line spectrum will be displayed as a 400
line spectrum, etc.
The frequency resolution, DF, is equal to the frequency span divided
by the number of lines, and this is equal to 1/T. Conversely, the
time record length T equals 1/DF. From this it can be seen that as
the frequency resolution increases (smaller DF), the time record
length also increases in proportion. For this reason, to create a
high-resolution spectrum requires a relatively long time to acquire
the data.
The Picket Fence Effect
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This phenomenon is called resolution bias error, or more commonly,
the picket fence effect. In other words, looking at an FFT spectrum
is a little like looking at mountain range through a picket fence.
Averaging
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The average gradually accumulates those portions of the signal that
are synchronized with the trigger, and other parts of the signal,
such as noise, are effectively averaged out. This is the only type of
averaging which actually does reduce noise.
More information on applications of time synchronous can be found
in the next chapter on Machine Vibration Monitoring.
Pitfalls in the FFT
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