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of Signal Analysis
Chapter 1 Introduction 4
Appendix B Bibliography 66
Index 67
3
Chapter 1
Introduction
4
Chapter 2
The Time, Frequency and
Modal Domains:
5
not affect the results appreciably. Figure 2.3
Also the deflection of the mass Simplified
must be large enough to give a oscillograph
operation.
usable result, otherwise a me-
chanical lever system to amplify
the motion would have to be
added with its attendant mass
and friction.
6
Section 2: The Frequency Figure 2.5
Domain Any real
waveform
can be
It was shown over one hundred produced
years ago by Baron Jean Baptiste by adding
Fourier that any waveform that sine waves
together.
exists in the real world can be
generated by adding up sine
waves. We have illustrated this in
Figure 2.5 for a simple waveform
composed of two sine waves. By
picking the amplitudes, frequen-
Figure 2.6
cies and phases of these sine
The relationship
waves correctly, we can generate between the time
a waveform identical to our and frequency
domains.
desired signal.
a) Three
dimensional
Conversely, we can break down coordinates
showing time,
our real world signal into these
frequency and
same sine waves. It can be shown amplitude
that this combination of sine b) Time
domain view
waves is unique; any real world
c) Frequency
signal can be represented by only domain view
one combination of sine waves.
7
The Need for Decibels
Since one of the major uses of the frequency
domain is to resolve small signals in the
presence of large ones, let us now address Figure 2.8
the problem of how we can see both large The relation-
and small signals on our display ship between
simultaneously. decibels, power
and voltage.
Suppose we wish to measure a distortion
component that is 0.1% of the signal. If we set
the fundamental to full scale on a four inch
(10 cm) screen, the harmonic would be only
four thousandths of an inch. (.1mm) tall.
Obviously, we could barely see such a signal,
much less measure it accurately. Yet many
analyzers are available with the ability to
measure signals even smaller than this.
8
It is very important to understand Figure 2.7
a) Time Domain - small signal not visible
that we have neither gained nor Small signals
are not hidden
lost information, we are just in the frequency
representing it differently. We domain.
are looking at the same three-
dimensional graph from different
angles. This different perspective
can be very useful.
9
So we see that the frequency Figure 2.10
domain is not at all uncommon. Frequency
We are just not used to seeing it spectrum ex-
amples.
in graphical form. But this graphi-
cal presentation is really not any
stranger than saying that the
temperature changed with time
like the displacement of a line
on a graph.
Spectrum Examples
10
Network Analysis Figure 2.11
One-port
If the frequency domain were network
analysis
restricted to the analysis of signal examples.
spectrums, it would certainly not
be such a common engineering
tool. However, the frequency
domain is also widely used in
analyzing the behavior of net-
works (network analysis) and
in design work.
11
Two-port analysis gives the re- Figure 2.12
sponse at a second port due to an Two-port
input at the first port. We are gen- network
analysis.
erally interested in the transmis-
sion and rejection of signals and
in insuring the integrity of signal
transmission. The concept of two-
port analysis can be extended to
any number of inputs and outputs.
This is called N-port analysis, a
subject we will use in modal
analysis later in this chapter.
θ2
θ1
θ1
θ2
12
When we say a network is linear, Figure 2.16
we mean it behaves like the net- A positioning
work in Figure 2.13. Suppose one system.
input causes an output A and a
second input applied at the same ∑
port causes an output B. If we
apply both inputs at the same
time to a linear network, the
output will be the sum of the
individual outputs, A + B.
13
Analysis of Linear Networks Figure 2.17
Linear network
As we have seen, many systems response to a
sine wave input.
are designed to be reasonably lin-
ear to meet design specifications.
This has a fortuitous side benefit
when attempting to analyze
networks*.
14
network by the input, we get a Figure 2.19
normalized result called the fre- Three classes
quency response of the network. of frequency
response.
As shown in Figure 2.18, the fre-
quency response is the gain (or
loss) and phase shift of the net-
work as a function of frequency.
Because the network is linear, the
frequency response is indepen-
dent of the input amplitude; the
frequency response is a property
of a linear network, not depen-
dent on the stimulus.
15
In Figure 2.20, we put a square Figure 2.20
wave into a bandpass filter. We Bandpass filter
recall from Figure 2.10 that a response to a
square wave
square wave is composed of input.
harmonically related sine waves.
The frequency response of our
example network is shown in
Figure 2.20b. Because the filter is
narrow, it will pass only one com-
ponent of the square wave. There-
fore, the steady-state response of
this bandpass filter is a sine wave.
Transient Response
16
information necessary to predict Figure 2.22
the transient response of the net- Parallel filter
work to any signal. analyzer.
17
How many filters should we use Figure 2.23
to cover the desired spectrum? Simplified
Here we have a trade-off. We swept spectrum
analyzer.
would like to be able to see
closely spaced spectral lines, so
we should have a large number
of filters. However, each filter is
expensive and becomes more ex-
pensive as it becomes narrower,
so the cost of the analyzer goes
up as we improve its resolution.
Typical audio parallel-filter ana-
lyzers balance these demands
with 32 filters, each covering
1/3 of an octave.
18
spectrum analyzers. In addition, Figure 2.25
two channel Dynamic Signal Gain-phase
Analyzers are in many ways better meter
operation.
network analyzers than the ones
we will introduce next.
Network Analyzers
19
not come without their price, Figure 2.27
however, as tracking filters and a The vibration
dedicated source must be added of a tuning fork.
to the simpler and less costly
gain-phase meter.
Section 4:
The Modal Domain
20
Each of these peaks, large and Figure 2.29
small, corresponds to a vibration Reducing the
mode of the tuning fork. For in- second harmonic
by damping the
stance, we might expect for this second vibration
simple example that the major mode.
tone is caused by the vibration
mode shown in Figure 2.28a. The
second harmonic might be caused
by a vibration like Figure 2.28b
21
We see that the sharp peaks Figure 2.31
(resonances) all occur at the The relationship
same frequencies independent between the
frequency and
of where they are measured on the modal
the structure. Likewise we would domains.
find by measuring the width of
each resonance that the damping
(or Q) of each resonance is inde-
pendent of position. The only
parameter that varies as we move
from point to point along the
structure is the relative height
of resonances.* By connecting
the peaks of the resonances of a
given mode, we trace out the
mode shape of that mode.
22
Section 5: Figure 2.32
Instrumentation for Curve fitting
the Modal Domain removes
measurement
noise.
There are many ways that the
modes of vibration can be deter-
mined. In our simple tuning fork
example we could guess what the
modes were. In simple structures
like drums and plates it is pos-
sible to write an equation for the
modes of vibration. However, in
almost any real problem, the
solution can neither be guessed
nor solved analytically because
the structure is too complicated.
In these cases it is necessary to
measure the response of the
structure and determine the
modes.
Figure 2.33
There are two basic techniques Single mode
excitation
for determining the modes of modal analysis.
vibration in complicated struc-
tures; 1) exciting only one mode
at a time, and 2) computing the
modes of vibration from the total
vibration.
23
In more realistic, three dimen- Figure 2.34
sional problems, it is necessary to Measured mode
add many more shakers to ensure shape.
that only one mode is excited.
The difficulties and expense of
testing with many shakers has
limited the application of this tra-
ditional modal analysis technique.
24
Chapter 3
Understanding Dynamic
Signal Analysis
We saw in the previous chapter Figure 3.1
that the Dynamic Signal Analyzer The FFT samples
has the speed advantages of paral- in both the time
and frequency
lel-filter analyzers without their domains.
low resolution limitations. In
addition, it is the only type of
analyzer that works in all three
domains. In this chapter we will
develop a fuller understanding of
this important analyzer family,
Dynamic Signal Analyzers. We
begin by presenting the properties
of the Fast Fourier Transform
(FFT) upon which Dynamic Sig-
nal Analyzers are based. No proof
of these properties is given, but
heuristic arguments as to their
validity are used where appropri-
ate. We then show how these FFT
properties cause some undesir-
able characteristics in spectrum
analysis like aliasing and leakage.
Having demonstrated a potential
difficulty with the FFT, we then
show what solutions are used to Figure 3.2
make practical Dynamic Signal A time record
Analyzers. Developing this basic is N equally
spaced samples
knowledge of FFT characteristics of the input.
makes it simple to get good
results with a Dynamic Signal
Analyzer in a wide range of
measurement problems.
25
Time Records Figure 3.3
The FFT works
A time record is defined to be on blocks
of data.
N consecutive, equally spaced
samples of the input. Because it
makes our transform algorithm
simpler and much faster, N is
restricted to be a multiple of 2,
for instance 1024.
26
How Many Lines are There? Figure 3.5
The relationship
We stated earlier that the time between the time
and frequency
record has N equally spaced domains.
samples. Another property of
the FFT is that it transforms
these time domain samples to
N/2 equally spaced lines in the
frequency domain. We only get
half as many lines because each
frequency line actually contains
two pieces of information, ampli-
tude and phase. The meaning of
this is most easily seen if we look
again at the relationship between
the time and frequency domain.
27
In addition, there is a frequency Figure 3.7
line at zero Hertz, DC. This is Lowest frequency
merely the average of the input resolvable by the
FFT.
over the time record. It is rarely
used in spectrum or network
analysis. But, we have now estab-
lished the spacing between these
two lines and hence every line; it
is the reciprocal of the time
record.
Figure 3.9
Frequency range
of Dynamic Signal
Analyzers is
determined by
sample rate.
28
Section 2*: Figure 3.10
Sampling and Digitizing Block diagram
of dynamic
Signal Analyzer.
Recall that the input to our
Dynamic Signal Analyzer is a
continuous analog voltage. This
voltage might be from an elec-
tronic circuit or could be the
Figure 3.11
output of a transducer and be The Sampler
proportional to current, power, and ADC must
pressure, acceleration or any not introduce
errors.
number of other inputs. Recall
also that the FFT requires digi-
tized samples of the input for its
digital calculations. Therefore, we
need to add a sampler and analog
to digital converter (ADC) to our
FFT processor to make a spec-
trum analyzer. We show this basic specifications, but the ADC for A Simple Data Logging
block diagram in Figure 3.10. a Dynamic Signal Analyzer must Example of Aliasing
be much faster than typical fast
For the analyzer to have the high DVMs. A fast DVM might take a Let us look at a simple data log-
accuracy needed for many mea- thousand readings per second, but ging example to see what aliasing
surements, the sampler and ADC in a typical Dynamic Signal Ana- is and how it can be avoided. Con-
must be quite good. The sampler lyzer the ADC must take at least sider the example for recording
must sample the input at exactly a hundred thousand readings temperature shown in Figure 3.12.
the correct time and must accu- per second. A thermocouple is connected to a
rately hold the input voltage digital voltmeter which is in turn
measured at this time until the Section 3: Aliasing connected to a printer. The sys-
ADC has finished its conversion. tem is set up to print the tempera-
The ADC must have high resolu- The reason an FFT spectrum ture every second. What would
tion and linearity. For 70 dB of analyzer needs so many samples we expect for an output?
dynamic range the ADC must per second is to avoid a problem
have at least 12 bits of resolution called aliasing. Aliasing is a If we were measuring the tem-
and one half least significant bit potential problem in any sampled perature of a room which only
linearity. data system. It is often over- changes slowly, we would expect
looked, sometimes with every reading to be almost the
A good Digital Voltmeter (DVM) disastrous results. same as the previous one. In fact,
will typically exceed these we are sampling much more often
than necessary to determine the
temperature of the room with
time. If we plotted the results of
this thought experiment, we
Figure 3.12
would expect to see results like
A simple
sampled Figure 3.13.
data system.
30
This minimum sample rate Figure 3.16
requirement is known as the A frequency
Nyquist Criterion. It is easy to see domain view
of how to avoid
in the time domain that a sam- aliasing - sample
pling frequency exactly twice the at greater than
input frequency would not always twice the highest
input frequency.
be enough. It is less obvious that
slightly more than two samples
in each period is sufficient infor-
mation. It certainly would not be
enough to give a high quality time
display. Yet we saw in Figure 3.16
that meeting the Nyquist Criterion
of a sample rate greater than
twice the maximum input fre-
quency is sufficient to avoid
aliasing and preserve all the Figure 3.17
information in the input signal. Nyquist
criterion in the
time domain.
The Need for an
Anti-Alias Filter
31
off and finite rejection of undes- Figure 3.19
ired signals. Large input signals Block diagrams
of analog and
which are not well attenuated in digital filtering.
the transition band could still
alias into the desired input fre-
quency range. To avoid this, the
sampling frequency is raised to
twice the highest frequency of the
transition band. This guarantees
that any signals which could alias
are well attentuated by the stop
band of the filter. Typically, this
means that the sample rate is now
two and a half to four times the
maximum desired input fre-
quency. Therefore, a 25 kHz FFT
Spectrum Analyzer can require an
ADC that runs at 100 kHz as we
stated without proof in Section 2
of this Chapter*.
This would mean at least twelve analog anti-alias filter, always
The Need for More anti-alias filters would be required provides aliasing protection. It is
Than One Anti-Alias Filter for each channel. called digital filtering because it
filters the input signal after we
Recall from Section 1 of this Each of these filters must have have sampled and digitized it. To
Chapter, that due to the proper- very good performance. It is de- see how this works, let us look at
ties of the FFT we must vary the sirable that their transition bands Figure 3.19.
sample rate to vary the frequency be as narrow as possible so that
span of our analyzer. To reduce as many lines as possible are free In the analog case we already
the frequency span, we must from alias products. Additionally, discussed, we had to use a new
reduce the sample rate. From in a two channel analyzer, each filter every time we changed the
our considerations of aliasing, filter pair must be well matched sample rate of the Analog to Digi-
we now realize that we must for accurate network analysis tal Converter (ADC). When using
also reduce the anti-alias filter measurements. These two points digital filtering, the ADC sample
frequency by the same amount. unfortunately mean that each of rate is left constant at the rate
the filters is expensive. Taken needed for the highest frequency
Since a Dynamic Signal Analyzer together they can add signifi- span of the analyzer. This means
is a very versatile instrument used cantly to the price of the analyzer. we need not change our anti-alias
in a wide range of applications, it Some manufacturers dont have a filter. To get the reduced sample
is desirable to have a wide range low enough frequency anti-alias rate and filtering we need for the
of frequency spans available. filter on the lowest frequency narrower frequency spans, we
Typical instruments have a mini- spans to save some of this ex- follow the ADC with a digital
mum span of 1 Hertz and a maxi- pense. (The lowest frequency filter.
mum of tens to hundreds of filters cost the most of all.) But
kilohertz. This four decade range as we have seen, this can lead to This digital filter is known as
typically needs to be covered with problems like our case of the a decimating filter. It not only
at least three spans per decade. missing temperature. filters the digital representation
of the signal to the desired fre-
Digital Filtering quency span, it also reduces the
* Unfortunately, because the spacing of the sample rate at its output to the
FFT lines depends on the sample rate,
increasing the sample rate decreases the Fortunately, there is an alterna- rate needed for that frequency
number of lines that are in the desired tive which is cheaper and when span. Because this filter is digital,
frequency range. Therefore, to avoid used in conjunction with a single there are no manufacturing varia-
aliasing problems Dynamic Signal Analyzer
have only .25N to .4N lines instead of N/2
lines.
32
tions, aging or drift in the filter. Figure 3.20
Therefore, in a two channel ana- High resolution
lyzer the filters in each channel measurements
with Band
are identical. It is easy to design Selectable
a single digital filter to work on Analysis.
many frequency spans so the need
for multiple filters per channel is
avoided. All these factors taken
together mean that digital filtering
is much less expensive than
analog anti-aliasing filtering.
Section 4:
Band Selectable Analysis
34
Figure 3.25
should be very different. Since Actual FFT results.
sharp phenomena in one domain a) b)
are spread out in the other
domain, we would expect the
spectrum of our sine wave to be
spread out through the frequency
domain.
35
Figure 3.26
The effect of
windowing in the
time domain.
What is Windowing?
36
Figure 3.28
Windowing reduces
leakage but does
not eliminate it.
a ) Leakage free measurement - input periodic b ) Windowed measurement - input not
in time record periodic in time record
The Hanning Window
Figure 3.34
Reduced
resolution
of the flat-top
window.
39
Analyzers call the uniform, Figure 3.37
Hanning and flattop functions Frequency
windows and other analyzers response
measurements
call those functions pass-band with a sine
shapes. Use whichever terminol- wave stimulus.
ogy is easier for the problem at
hand as they are completely inter-
changeable, just as the time and
frequency domains are completely
equivalent.
Section 6:
Network Stimulus
40
gether. Although this sounds Figure 3.39
very expensive, just such a Random noise
source can be easily generated as a stimulus.
digitally. It is called a pseudo-
random noise or periodic ran-
dom noise source.
41
Figure 3.41
Nonlinear transfer function.
a) Pseudo-random noise stimulus b) Random noise stimulus
Section 7: Averaging The standard technique in statis- exist between the spectra. This
tics to improve the estimates of averaging technique is very valu-
To make it as easy as possible to a value is to average. When we able for determining the average
develop an understanding of Dy- watch a noisy reading on a Dy- power in any of the filters of our
namic Signal Analyzers we have namic Signal Analyzer, we can Dynamic Signal Analyzers. The
almost exclusively used examples guess the average value. But be- more averages we take, the better
with deterministic signals, i.e., cause the Dynamic Signal Ana- our estimate of the power level.
signals with no noise. However, lyzer contains digital computation
as the real world is rarely so capability we can have it compute In Figure 3.43, we show RMS
obliging, the desired signal often this average value for us. Two averaged spectra of random noise,
must be measured in the presence kinds of averaging are available, digital data and human voices.
of significant noise. At other RMS (or power averaging) and Each of these examples is a
times the signals we are trying linear averaging. fairly random process, but when
to measure are more like noise averaged we can see the basic
themselves. Common examples RMS Averaging properties of its spectrum.
that are somewhat noise-like in-
clude speech, music, digital data, When we watch the magnitude If we want to measure a small
seismic data and mechanical vi- of the spectrum and attempt to signal in the presence of noise,
brations. Because of these two guess the average value of the RMS averaging will give us a good
common conditions, we must de- spectrum component, we are do- estimate of the signal plus noise.
velop techniques both to measure ing a crude RMS* average. We are We can not improve the signal to
signals in the presence of noise trying to determine the average noise ratio with RMS averaging;
and to measure the noise itself. magnitude of the signal, ignoring we can only make more accurate
any phase difference that may estimates of the total signal plus
noise power.
* RMS stands for root-mean-square and
is calculated by squaring all the values,
adding the squares together, dividing by
the number of measurements (mean) and
taking the square root of the result.
43
Linear Averaging Figure 3.44
Linear averaging.
However, there is a technique for
a) Single record, no averaging b) Single record, no averaging
improving the signal to noise ratio
of a measurement, called linear
averaging. It can be used if a
trigger signal which is synchro-
nous with the periodic part of the
spectrum is available. Of course,
the need for a synchronizing sig-
nal is somewhat restrictive, al-
though there are numerous
situations in which one is avail-
able. In network analysis prob-
lems the stimulus signal itself c c) 128 linear averages d) 128 linear averages
an often be used as a synchroniz-
ing signal.
44
Section 8: buffer fills, then we are said to be Real Time Bandwidth
Real Time Bandwidth operating in real time. Requirements
Until now we have ignored the To see what this means, let us How wide a real time bandwidth
fact that it will take a finite time look at the case where the FFT is needed in a Dynamic Signal
to compute the FFT of our time computation takes longer than the Analyzer? Let us examine a few
record. In fact, if we could com- time to fill the time record. The typical measurements to get a
pute the transform in less time case is illustrated in Figure 3.48. feeling for the considerations
than our sampling period we Although the buffer is full, we involved.
could continue to ignore this have not finished the last trans-
computational time. Figure 3.45 form, so we will have to stop tak- Adjusting Devices
shows that under this condition ing data. When the transform is
we could get a new frequency finished, we can transfer the time If we are measuring the spectrum
spectrum with every sample. As record to the FFT and begin to or frequency response of a device
we have seen from the section on take another time record. This which we are adjusting, we need
aliasing, this could result in far means that we missed some input to watch the spectrum change in
more spectrums every second data and so we are said to be not what might be called psychologi-
than we could possibly compre- operating in real time. cal real time. A new spectrum
hend. Worse, because of the com- every few tenths of a second is
plexity of the FFT algorithm, it Recall that the time record is not sufficiently fast to allow an opera-
would take a very fast and very constant but deliberately varied to tor to watch adjustments in what
expensive computer to generate change the frequency span of the he would consider to be real
spectrums this rapidly. analyzer. For wide frequency time. However, if the response
spans the time record is shorter. time of the device under test is
A reasonable alternative is to add Therefore, as we increase the fre- long, the speed of the analyzer is
a time record buffer to the block quency span of the analyzer, we immaterial. We will have to wait
diagram of our analyzer. In Figure eventually reach a span where the for the device to respond to the
3.47 we can see that this allows time record is equal to the FFT changes before the spectrum will
us to compute the frequency spec- computation time. This frequency be valid, no matter how many
trum of the previous time record span is called the real time band- spectrums we generate in that
while gathering the current time width. For frequency spans at and time. This is what makes adjust-
record. If we can compute the below the real time bandwidth, ing lightly damped (high Q)
transform before the time record the analyzer does not miss any resonances tedious.
data.
45
RMS Averaging Figure 3.49
RMS averaging
A second case of interest in time.
determining real time bandwidth
requirements is measurements
that require RMS averaging. We
might be interested in determin-
ing the spectrum distribution of
the noise itself or in reducing the
variation of a signal contaminated
by noise. There is no requirement
in averaging that the records must
be consecutive with no gaps*. Figure 3.50
Transient
Therefore, a small real time band- analysis.
width will not affect the accuracy
of the results.
46
lost as the analyzer updates the Figure 3.51
display with the spectrum of the Understanding
latest time record. A special dis- overlap
processing.
play which can show more than
one spectrum (waterfall dis-
play), mass memory, a high speed
link to a computer or a high speed
facsimile recorder is needed. The
output device must be able to
record a spectrum every time
record or information will be lost.
47
RMS Averaging Figure 3.52
RMS averaging
Overlap processing can give speed improve-
ments with
dramatic reductions in the time overlap
to compute RMS averages with a processing.
given variance. Recall that win-
dow functions reduce the effects
of leakage by weighting the ends
of the time record to zero. Over-
lapping eliminates most or all of
the time that would be wasted
taking this data. Because some
overlapped data is used twice,
more averages must be taken to
get a given variance than in the
non-overlapped case. Figure 3.52
shows the improvements that can Section 10: Summary We then added several features to
be expected by overlapping. our basic Dynamic Signal Ana-
In this chapter we have developed lyzer to enhance its capabilities.
Transients the basic properties of Dynamic Band Selectable Analysis allows
Signal Analyzers. We found that us to make high resolution mea-
For transients shorter than the many properties could be under- surements even at high frequen-
time record, overlap processing stood by considering what hap- cies. Averaging gives more
is useless. For transients longer pens when we transform a finite, accurate measurements when
than the time record the real time sampled time record. The length noise is present and even allows
bandwidth of the analyzer and of this record determines how us to improve the signal to noise
spectrum recorder is usually a closely our filters can be spaced ratio when we can use linear aver-
limitation. If it is not, overlap in the frequency domain and the aging. Finally, we incorporated a
processing allows more spectra to number of samples determines noise source in our analyzer to act
be generated from the transient, the number of filters in the fre- as a stimulus for transfer function
usually improving resolution of quency domain. We also found measurements.
resulting plots. that unless we filtered the input
we could have errors due to
aliasing and that finite time
records could cause a problem
called leakage which we mini-
mized by windowing.
48
Chapter 4
Using Dynamic
Signal Analyzers
In Chapters 2 & 3, we developed Figure 4.1
an understanding of the time, Harmonic distortion
frequency and modal domains of an Audio Oscillator -
Flattop window used.
and how Dynamic Signal Analyz- a) Logarithmic amplitude scale b) Linear amplitude scale
ers operate. In this chapter we
show how to use Dynamic Signal
Analyzers in a wide variety of
measurement situations. We
introduce the measurement
functions of Dynamic Signal
Analyzers as we need them for
each measurement situation.
49
important in communications and Figure 4.3 a) Block diagram of phase noise measurement
radar. One measure of this is Phase Noise
called phase noise. It is often Measurement.
measured by the technique shown
in Figure 4.3a. This mixes down
and cancels the oscillator carrier
leaving only the phase noise
sidebands. It is therefore possible
to measure the phase noise far
below the carrier level since the
carrier does not limit the range
of our measurement. Figure 4.3b
shows the close-in phase noise of
a 20 MHz synthesizer. Here, since b ) Phase noise of a frequency synthesizer -
we are measuring noise, we use RMS averaging and Hanning window used for noise measurements
RMS averaging and the Hanning
window.
50
In addition, phase noise and might look like Figure 4.4. It is lines is crucial to our capability to
other noise measurements are obviously much more compli- diagnose why the vibration levels
often normalized to the power cated than the relatively clean of a rotating machine are exces-
that would be measured in a 1 Hz spectrum of the electronic oscilla- sive. The actions we would take
wide square filter. This measure- tor we looked at previously. There to correct an excessive vibration
ment is called a power spectral is also a great deal of random at 120 Hz are quite different if it is
density and is often provided on noise; stray vibrations from caused by a loose stator pole
Dynamic Signal Analyzers. It sim- sources other than our motor that rather than an imbalanced rotor.
ply changes the presentation on the accelerometer picks up. The
the display to this desired form; effects of this stray vibration have Since the bearings are the most
the data is exactly the same in been minimized in Figure 4.4b unreliable part of most rotating
Figures 4.3b and 4.3c, but the RMS averaging. machines, we would also like to
latter is in the more conventional check our spectrum for indica-
presentation. In Figure 4.5, we have used the tions of bearing failure. Any defect
Band Selectable Analysis capabil- in a bearing, say a spalling on the
Rotating Machinery ity of our analyzer to zoom-in outer face of a ball bearing, will
Characterization and separate the vibration of the cause a small vibration to occur
stator at 120 Hz from the vibration each time a ball passes it. This
A rotating machine can be thought caused by the rotor imbalance will produce a characteristic fre-
of as a mechanical oscillator.* only a few tenths of a Hertz lower quency in the vibration called the
Therefore, many of the measure- in frequency.** This ability to passing frequency. The frequency
ments we made for an electronic resolve closely spaced spectrum
oscillator are also important in
characterizing rotating machinery.
Figure 4.4
To characterize a rotating ma- Spectrum of
chine we must first change its electrical motor
vibration.
mechanical vibration into an elec-
trical signal. This is often done by
mounting an accelerometer on a
bearing housing where the vibra-
tion generated by shaft imbalance
and bearing imperfections will be
the highest. A typical spectrum
* Or, if you prefer, electronic oscillators can ** The rotor in an AC induction motor always
be viewed as rotating machines which can runs at a slightly lower frequency than the
go at millions of RPMs. excitation, an effect called slippage.
51
domain is ideal for separating this Figure 4.7
small vibration from all the other Desktop
frequencies present. This means computer
system for
that we can detect impending monitoring
bearing failures and schedule rotating
a shutdown long before they machinery
vibration.
become the loudly squealing
problem that signals an immedi-
ate shutdown is necessary.
52
Electronic Filter Figure 4.9
Characterization Test setup
to measure
frequency
In Section 6 of the last chapter, response
we developed most of the prin- of filter.
ciples we need to characterize a
low frequency electronic filter. We
show the test setup we might use
in Figure 4.9. Because the filter is
linear we can use pseudo-random
noise as the stimulus for very fast
test times. The uniform window is
used because the pseudo-random Figure 4.10
Frequency response
noise is periodic in the time of electronic filter using
record.* No averaging is needed PRN and uniform window.
since the signal is periodic and a) Frequency response magnitude b) Frequency response magnitude and phase
reasonably large. We should be
careful, as in the single channel
case, to set the input sensitivity
for both channels to the most
sensitive position which does not
overload the analog to digital
converters.
Structural Frequency
Response
53
we can use pseudo-random noise Figure 4.12 a) Impact stimulus
as a test stimulus. But we might Frequency
also desire to use true random response of a
linear network
noise, swept-sine or an impulse is independent
(hammer blow) as the stimulus. of the stimulus
In Figure 4.12 we show each of used.
these measurements and the fre-
quency responses. As we can see,
the results are all the same.
eration measurement. Displace- parts of a mechanical structure. but has not completely eliminated
ment is a double integration of However, there are times when the 1 KHz interference.* If we did
acceleration. Many Dynamic Sig- the components we wish to test not know of the interference, we
nal Analyzers have the capability can not be isolated from other would think that this filter has an
of integrating a trace by simply disturbances. For instance, in additional resonance at 1 KHz.
pushing a button. Therefore, we electronics we might be trying to But Dynamic Signal Analyzers can
can easily generate all the com- measure the frequency response often make an additional mea-
mon mechanical measurements of a switching power supply surement that is not available
without the need of many which has a very large component with traditional network analyz-
expensive transducers. at the switching frequency. Or ers called coherence. Coherence
we might try to measure the fre- measures the power in the re-
Coherence quency response of part of a sponse channel that is caused by
machine while other machines the power in the reference chan-
Up to this point, we have been are creating severe vibration. nel. It is the output power that is
measuring networks which we coherent with the input power.
have been able to isolate from the In Figure 4.15 we have simulated
rest of the world. That is, the only these situations by adding noise Figure 4.17 shows the same fre-
stimulus to the network is what and a 1 KHz signal to the output of quency response magnitude from
we apply and the only response an electronic filter. The measured Figure 4.16 and its coherence.
is that caused by this controlled frequency response is shown in The coherence goes from 1
stimulus. This situation is often Figure 4.16. RMS averaging has (all the output power at that
encountered in testing compo- reduced the noise contribution, frequency is caused by the input)
nents, e.g., electric filters or
56
be large. If the time shift between Figure 4.21 a) Time record of random noise
the two waveforms is made large Auto correlation
however, the waveforms appear of random noise.
dissimilar and the final sum is
small.
57
Figure 4.23
Auto correlation
of periodic
waveforms.
58
Figure 4.25 a) Transmitted signal, a swept-frequendy sine wave
Simulated radar
cross correlation.
Cross Correlation
If auto-correlation is concerned
with the similarity between a
signal and a time shifted version
of itself, then it is reasonable to
suppose that the same technique
could be used to measure the c ) Result of cross correlation the transmitted and received signals.
similarity between two non-iden- Distance from left edge to peak represents tranmission delay.
tical waveforms. This is called
the cross correlation function. If
the same signal is present in both
waveforms, it will be reinforced
in the cross correlation function,
while any uncorrelated noise will
be reduced. In many network
analysis problems, the stimulus
can be cross correlated with the
response to reduce the effects of
noise. Radar, active sonar, room
acoustics and transmission path
delays all are network analysis
problems where the stimulus can
be measured and used to remove
contaminating noise from the
response by cross correlation.* Figure 4.26
Cross correlation
shows multiple
transmission
paths.
59
Section 3: Modal Domain apart that high frequency bend- measurements should be made in
Measurements ing modes in the structure can these critical areas as well.
not be measured accurately. To
In Section 1 we learned how to decide on a reasonable number Once we have decided on
make frequency domain measure- of measurement points, take a where the measurements should
ments of mechanical structures few trial frequency response be taken, we number these mea-
with Dynamic Signal Analyzers. measurements of the structure surement points (the order can be
Let us now analyze the behavior to determine the highest signifi- arbitrary) and enter the coordi-
of a simple mechanical structure cant resonant frequencies nates of each point into our modal
to understand how to make mea- present. The wave length can analyzer. This is necessary so that
surements in the modal domain. be determined empirically by the analyzer can correlate the
We will test a simple metal plate changing the distance between measurements we make with a
shown in Figure 4.27. The plate the stimulus and the sensor position on the structure to
is freely suspended using rubber until a full 360° phase shift has compute the mode shapes.
cords in order to isolate it from occurred from the original mea-
any object which would alter its surement point. Measurement The next decision we must make
properties. point spacing should be approxi- is what signal we should use for a
mately one-quarter or less of stimulus. Our plate example is a
The first decision we must make this wavelength. linear structure as it has no loose
in analyzing this structure is how rivet joints, non-linear damping
many measurements to make and Measurement points can be materials, or other non-linearities.
where to make them on the struc- spaced uniformly over the Therefore, we know that we can
ture. There are no firm rules for structure using this guideline, use any of the stimuli described in
this decision; good engineering but it may be desirable to Chapter 3, Section 6. In this case,
judgment must be exercised in- modify this procedure slightly. an impulse would be a particularly
stead. Measuring too many points Few structures are as uniform as good test signal. We could supply
make the calculations unnecessar- this simple plate example,* but the impulse by hitting the struc-
ily complex and time consuming. complicated structures are made ture with a hammer equipped with
Measuring too few points can of simpler, more uniform parts. a force transducer. This is prob-
cause spatial aliasing; i.e., the The behavior of the structure ably the easiest way to excite the
measurement points are so far at the junction of these parts structure as a shaker and its asso-
is often of great interest, so ciated driver are not required. As
60
we saw in the last chapter, how- Figure 4.29
ever, if the structure were non- A frequency
response of
linear, then random noise would the plate.
be a good test signal. To supply
random noise to the structure we
would need to use a shaker. To
keep our example more general,
we will use random noise as a
stimulus.
61
just determined, the analyzer can Figure 4.30
calculate the mode shapes by Mode shapes
curve fitting the responses of of a rectangular
plate.
each point with the measured
resonances. In Figure 4.30 we
show several mode shapes of our
simple rectangular plate. These
mode shapes can be animated on
the display to show the relative
motion of the various parts of the
structure. The graphs in Figure
4.30, however, only show the
maximum deflection.
Section 4: Summary
62
Appendix A
The Fourier Transform: A
Mathematical Background
64
If we RMS average this result to
try to eliminate the noise, we As before, we will assume a mea-
find the S xSn terms approach surement condition like Figure A.2.
zero because S x, and S n, are Then, as we have shown before,
uncorrelated. However, the |Sn|2
term remains as an error and so Gxy= G xx|H|2+SxHSn*+ Sx*H*Sn+|S|2
we get
Gxy=GxxH+SnSx*
Gyy |Sn|2
= |H| 2 + As we average, the cross terms SnSx,
Gxx Gxx approach zero, assuming that the
Therefore if we try to measure signal and the noise are not related.
|H|2 by this single channel tech- So the coherence becomes
niques, our value will be high by
the noise to signal ratio. (HGyx)2
γ 2(f) =
Gxx(|H|2Gxx+|Sn|2)
If instead we average the cross
power spectrum we will eliminate
|H|2Gxx
this noise error. Using the same γ 2(f) =
example, |H|2Gxx+Sn2
65
Appendix B
Bibliography
That is, the auto correlation can Bendat, Julius S. and Piersol, Potter, R. and Lortscher, J.,
be found by taking a signal and Allan G.,Random Data: Analysis What in the World is Overlap
multiplying it by the same signal and Measurement Procedures , Processing, Hewlett-Packard
displaced by a timeτ and averag- Wiley-Interscience, Santa Clara DSA/Laser Division
ing the product ouer all time. New York, 1971. Update, Sept. 1978.
However, most Dynamic Signal
Analyzers compute this quantity Bendat, Julius S. and Piersol, Ramse , K.A.,Effective Measure-
by taking advantage of its dual inAllan G.,Engineering Applica- ments for Structural Dynamics
the frequency domain. It can be tions of Correlation and Spectral Testing , Part 1, Sound and
shown that Analysis , Wiley-lnterscience, Vibration Magazine,
-1
R (τ)=F [S (f)S *(f)] New York, 1980. November 1975, pp. 24-35.
xx x x
where F-1
is the inverse Fourier Bracewell, R.,The Fourier Roth, P.,Effective Measurements
Transform and xSis the Fourier Transform and its Applications , Using Digital Signal Analysis ,
Transform of x(t) McGraw-Hill, 1965. IEEE Spectrum, April 1971,
pp. 62-70.
Since both techniques yield the Cooley, J.W. and Tukey, J.W.,
same answer, the latter is usuallyAn Algorithm for the Machine Roth, P.,Digital Fourier Analy-
chosen for Dynamic Signal Calculation of Complex Fourier sis , Hewlett-Packard Journal,
Anlyzer since the Frequency Series , Mathematics of Computa- June 1970.
Transform algorithm is already in tion, Vo. 19, No. 90, p. 297,
the instrument and the results canApril 1965. Welch, Peter D.,The Use of
be computed faster because less Fast Fourier Transform for the
multiplications are required. Otnes, R.K. and Enochson, L., Estimation of Power Spectra:
Digital Time Series Analysis , A Method Based on Time
Cross Correlation , Rxy(τ), is a John Wiley, 1972. Averaging Over Short, Modified
comparison of two signals as a Periodograms , IEEE Transac-
function of a time shift between McKinney, W.,Band Selectable tions on Audio and Electro-
them. It is defined as: Fourier Analysis , Hewlett- acoustics, Vol. AU-15, No. 2,
Packard Journal, April 1975, June 1967, pp. 70-73.
⌠ pp. 20-24.
Rxy(τ)= lim 1 ⌡ x(t)y(t+τ)dt
T→∞ T Τ
Rxy(τ)=F -1[Gxy]
h(t) = F-1[H(f)]
66
Index
67
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