Lecroy 93XXC-OM-E21 User Manual

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The WP02 Spectral Analysis package, with FFT (Fast Fourier Transform), reveals signal characteristics not visible in t he time domain and adds the power of frequency domain analysis to your oscilloscope. FFT converts a time domain waveform into frequency domain spectra similar to those of a spectrum analyzer, but with important differences and added benefits.
:K\8VH))7" For a large class of signals, greater insight can be gained by
looking at spectral representation rather than time description. Signals encountered in the frequency response of amplifiers, oscillator phase noise and those in mechanic al vibration analysis
— to mention just som e applications — are easier to observe in the frequency domain.
If sampling is done at a rate f ast enough to f aithfully approxim ate the original waveform (usually five times the highest frequency component in the signal), the resulting discrete data series will uniquely describe the analog signal.
This is of particular value when dealing with transient signals because, unlike FFT, conventional swept spectrum analyzers cannot handle them.
7KHRU\%HKLQG))7 Spectral analysis theory assumes that the signal for
transformation be of infinite duration. Since no physical signal can meet this condition, a useful assumption for reconciling theory and practice is to view the signal as consisting of an infinite series of replica of itself . Thes e replica are m ultiplied by a rectangular window (the display grid) that is zero outside of the observation grid.
À
For an explanation of FFT terms: see the Glossary
on page C–17
À
Using FFT Functions: see page C–9
À
FFT Algorithms: page C–14
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Figure C–1 shows spectra of a swept triangular wave. Discontinuities at the edges of the wave produce leakage, an effect clearly visible in Trace A, which was computed with a rectangular window, but less pronounced in the Von Hann window in Trace B (
explanations
first harmonic.
). Histogramming in Trace C tracks the spread of the
see below for leakage and window-type
Figure C–1
Slicing the waveform in the fashion described above is tantamount to diluting the spectral energy in an infinite number of side lobes, which correspond to multiples of the frequency resolution ∆f ( T determines the frequency resolution of the FFT (∆f=1/T). Whereas the sampling period and the record length set the maximum frequency span that can be obtained (f
Fig. C–2
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). The observation window or capture time
=∆f*N/2).
Nyq
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Figure C–2
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An FFT operation on an N point time-dom ain signal may thus be compared to passing the signal through a comb f ilter consisting of a bank of N/2 filters. All the filter s have the same shape and width and are centered at N/2 discrete frequencies. Each filter collects the signal energy that falls into the immediate neighborhood of its center frequency. Thus it can be said that there are N/2 frequency bins. The distance in Hz between the center frequencies of two neighboring bins is always the same:
f.
3RZHU'HQVLW\6SHFWUXP Because of the linear scale used to show magnitudes, lower
amplitude components are often hidden by larger com ponents . In addition to the functions offering magnitude and phase representations, the FFT option offers power density and power
spectrum density functions, selec ted from the “FFT result” m enu shown in the figures. These latter functions are even better suited for characterizing spectra. The power spectr um (V
2
) is the square of the magnitude spectrum (0 dB m corresponds to voltage equivalent to 1 mW into 50 Ω.) This is the representation
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of choice for signals containing isolated peaks — periodic signals, for instance.
2
The power density spectrum (V divided by the equivalent noise bandwidth of the filter in Hz associated with the FFT calculation. This is best employed for characterizing broad-band signals such as noise.
0HPRU\IRU))7 The amount of acquisition memory available will determine the
maximum range (Nyquist frequency) over which signal components can be observed. Consider the problem of determining the length of the observation window and the size of the acquisition buffer if a Nyquist rate of 500 MHz and a resolution of 10 kHz are required. To obtain a resolution of 10 kHz, the acquisition time must be at least:
T = 1/∆f = 1/10 kHz = 100 µs.
For a digital oscilloscope with a memory of 100 k, the highest frequency that can be analyzed is:
∆f ×
N/2 = 10 kHz × 100 k/2 = 500 MHz.
/Hz) is the power spectrum
))73LWIDOOVWR$YRLG Take care to ensure that signals are correctly acquired: im proper
waveform positioning within the observation window produces a distorted spectrum. T he most com mon distortions can be trac ed to insufficient sampling, edge discontinuities, windowing or the “picket fence” effect.
Because the FFT acts like a bank of bandpass filters c entered at multiples of the frequency resolution, components that are not exact multiples of that frequency will fall within two consecutive filters. This results in an attenuation of the true amplitude of these components.
3LFNHW)HQFHDQG6FDOORS The highest point in the spectrum c an be 3.92 dB lower when the
source frequency is halfway between two discrete frequencies. This variation in spectrum magnitude is the picket fence effect. And the corresponding attenuation loss is ref erred to as scallop loss. LeCroy scopes automatically correc t for the scallop effect, ensuring that the magnitude of the spectra lines correspond to their true values in the time domain.
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If a signal contains a frequency component above Nyquist, the spectrum will be aliased, meaning that the frequencies will be folded back and spurious. Spotting aliased frequencies is of ten difficult, as the aliases may ride on top of real harmonics. A simple way of checking is to modify the sample rate and verify whether the frequency distribution changes.
/HDNDJH FFT assumes that the signal contained within the time grid is
replicated endlessly outside the observation window. Therefore if the signal contains discontinuities at its edges, pseudo­frequencies will appear in the spectral dom ain, distorting the real spectrum. W hen the start and end phas e of the signal differ, the signal frequency falls within two frequency cells broadening the spectrum.
This effect is illustrated in Figure C–1. Bec ause the display does not contain an integral number of periods, the spectrum displayed in Trace B does not reveal sharp frequency components. Intermediate components exhibit a lower and broader peak. The broadening of the base, stretching out in many neighboring bins, is termed leakage. Cures for this are to ensure that an integral number of periods is contained within the display grid or that no discontinuities appear at the edges. Another is to use a window function to smooth the edges of the signal.
&KRRVLQJD:LQGRZ The choice of a spectral window is dictated by the signal’s
characteristics. Weighting functions control the filter response shape and affect noise bandwidth as well as side-lobe levels. Ideally, the main lobe should be as narrow and flat as possible to effectively discriminate all spectral components, while all side lobes should be infinitely attenuated.
Chosen from the “with window” menu, the window type defines the bandwidth and shape of the equivalent filter to be used in the FFT processing.
In the same way as one would choose a particular camera lens for taking a picture, s ome experimenting is generally necessary to determine which window is most suitable. However, the following general guidelines should help (
window types
).
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see page C–11 for
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Rectangular windows provide the highest frequency resolution and are thus useful for estim ating the type of harm onics pres ent in the signal. Because the rectangular window decays as a sinx/x function in the spectral domain, slight attenuation will be induced.
Alternative functions with less attenuation — Flat-top and Blackman-Harr is — provide maximum amplitude at the expense of frequency resolution. Whereas, Hamming and von Hann are good for general purpose use with continuous waveforms.
,PSURYLQJ'\QDPLF5DQJH Enhanced resolution (
technique that can potentially provide for three additional bits (18 dBs) if the signal noise is uniformly distributed (white). Low pass filtering should be considered when high frequency components are irrelevant. A distinct advantage of this technique is that it works for both repetitive and transient signals. The SNR increase is conditioned by the cut-off frequency of the Eres low pass filter and the noise shape (frequency distribution).
LeCroy digital oscilloscopes employ FIR digital filters so that a constant phase shift is maintained. The phase information is therefore not distorted by the filtering action.
6SHFWUDO3RZHU$YHUDJLQJ Even greater dynamic-range im provem ent is obtained on signals
showing periodicity. Moreover, the range can be increased without sacrificing frequency response. T he LeCroy oscilloscope being used is equipped with accumulation buffers 32 bits wide to prevent overflows.
Spectral power averaging is useful when the signal var ies in tim e and the mean power of the signal needs to be estim ated. T ypical applications include noise and pseudo- random noise. Whereas time averaging ignores phase information, spectral averaging tracks magnitude as well as phase information. It is thus a superior estimator. And the improvement is typically proportional to the square root of the number of averages. For instance, averaging white noise at full scale over 10 sweeps yields a typical improvement of nearly 20 dBs.
see Appendix B
) uses a low pass f iltering
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Spectral power averaging is the technique of choice when determining the frequency response of passive network s suc h as filters. Figures 3 and 4 show the transfer functions of a low pass filter with a 3 dB cutoff o1 11 MHz obtained by exciting the filter with a white noise source ( (
Fig. C–4
The choice of method is governed by the availability of an adequate generating source.
The spectra of single tim e-domain waveforms can be computed and displayed to obtain power averages obtained over as many as 50 000 spectra.
). Both techniques give substantially the same results.
Fig. C–3
) and a sine swept generator
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Figure C–3
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Figure C–4
2YHUDOO« Because of its versatility, FFT analysis has become a popular
analysis tool. However, some care must be taken with it. In most instances, incorrect positioning of the signal within the display grid will significantly alter the spectrum. Eff ects such as leakage and aliasing that distort the spectrum must be understood if meaningful conclusions are to be arrived at when using FFT.
An effective way to reduce these effects is to maximize the acquisition record length. Record length directly conditions the effective sampling rate of the scope and therefore determines the frequency resolution and span at which spectral analysis can be carried out.
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Select “FFT” from the “Math Type” menu (
for a full description of math and waveform processing menus
running from zero to the Nyquist frequency are shown at the right-hand edge of the trace. The frequency scale factors (Hz/div) are in a 1–2–5 sequence.
The processing equation is displayed at the bottom of the screen, together with the three key parameters that characterize an FFT spectrum. These are:
1. Transform Size N (number of input points)
2. Nyquist frequency (= ½ sample rate), and
3. Frequency Increment, ∆f, between two successive points of
These parameters are related as:
Where: ∆f = 1/T, and where T is the duration of the input waveform record (10 ∗ time/div). The num ber of output points is equal to N/2.
). Spectra displayed with a linear frequency axis
the spectrum.
Nyquist frequency = ∆f ∗ N/2.
see Chapter 10
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over the entire source time-domain waveform. This limits the number of points used for FFT processing. If the input waveform contains more points than the selected
maximum (in “for Math use max points”, they are decimated before FFT processing. But if it has fewer, all points are used.
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FFT spectra are computed
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The following selections can be made using the “FFT result” menu.
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Measured with respect to a cosine whose maximum occurs at the left-hand edge of the screen, at which point it has 0 °. Similarly, a positive-going sine starting at the left-hand edge of the screen has a –90 ° phase. (Displayed in degrees.)
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The signal power normalized to the bandwidth of the equivalent filter associated with the FFT calculation. The power density is suitable for characterizing broad-band noise. (It is displayed on a logarithmic vertical axis calibrated in dBm.)
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The signal power (or magnitude) represented on a logarithmic vertical scale: 0 dBm corresponds to the voltage (0.316 V peak ) which is equivalent to 1 mW into 50 Ω. The power spectrum is suitable for characterizing spectra which contain isolated peaks. (dBm.)
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The peak signal am plitude represented on a linear scale. ( Same units as input signal.)
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These represent the complex result of the FFT processing. (Same units as input signal.)
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:LQGRZV Chosen using the “with window” menu, the window type defines the
bandwidth and shape of the filter to be used in the FFT processing
see the table on page C–17 for these filters’ parameters
( “AC” is selected from the same m enu, the DC com ponent of the input signal is forced to zero prior to the FFT processing. This improves the amplitude r esolution, especially when the input has a large DC component.
Window Type Applications and Limitations
Normally used when the signal is transient — completely contained in the time-domain window — or known to have a fundamental
Rectangular
Hanning (Von Hann)
Hamming
Flat Top
Blackman–Harris
frequency component that is an integer multiple of the fundamental frequency of the window. Signals other than these types will show varying amounts of spectral leakage and scallop loss, corrected by selecting another type of window.
Reduce leakage and improve amplitude accuracy. However, frequency resolution is also reduced.
Reduce leakage and improve amplitude accuracy. However, frequency resolution is also reduced.
This window provides excellent amplitude accuracy with moderate reduction of leakage, but also at the loss of frequency resolution.
It reduces the leakage to a minim um, but again along with reduc ed frequency resolution.
). When
))73RZHU$YHUDJH A function can be defined as the power average of FFT spectra
computed by another function (
“FFTAVG” from the “Math Type” Menu, and “Power Spect” from “FFT Result”.
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see page C–6
). Choose
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$GGLWLRQDO3URFHVVLQJ Other waveform processing functions, such as Averaging and
Arithmetic, can be applied to waveform s before FFT processing is performed. Tim e-domain averaging prior to FFT, for ex ample, can be used if a stable trigger is available to reduce random noise in the signal.
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To increase the FFT frequency range, the Nyquist frequency, raise the effective sampling frequency by increasing the maximum number of points or using a faster time base.
À
To increase the FFT frequency resolution, increase the length of the time-domain waveform record by using a slower time base.
0HPRU\6WDWXV When FFT is used, the field beneath the griddisplays
parameters of the waveform descriptor, including number of points, horizontal and vertical scale factors and units.
8VLQJ&XUVRUVZLWK))7 For reading the amplitude and frequency of a data point, the
Absolute Time cursor can be m oved into the frequency domain by going beyond the right-hand edge of a time-domain waveform.
The Relative Time cursors can be moved into the frequency domain to simultaneously indicate the frequency difference and the amplitude difference between two points on each frequency­domain trace.
The Absolute Voltage cursor reads the absolute value of a point in a spectrum in the appropriate units, and the cursors indicate the difference between two levels on each trace
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Relative Voltage
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(UURU0HVVDJHV One of these FFT-related error messages may be displayed at
the top of the screen.
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Message Meaning
“Incompatible input record type”
“Horizontal units don't match” FFT of a frequency-domain waveform is not available.
“FFT source data zero filled” If there are invalid data points in the source waveform (at
“FFT source data over/underflow” The source waveform data has been clipped in
“Circular computation” A function definition is circular (i.e. the function is its own
FFT power average is defined only on a function defined as FFT.
the beginning or at the end of the record), these are replaced by zeros before FFT processing.
amplitude, either in the acquisition — gain too high or inappropriate offset — or in previous processing. The resulting FFT contains harmonic components which would not be present in the unclipped waveform. The settings defining the acquisition or processing should be changed to eliminate the over/underflow condition.
source, indirectly via another function or expansion). One of the definitions should be changed.
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A summary of the algorithms used in the oscilloscope’s FFT computation is given here in the form of seven steps:
1. If the maximum number of points is smaller than the source
number of points, the source waveform data are decimated prior to the FFT. These decim ated data extend over the full length of the source waveform. The resulting sampling interval and the actual transform size selected provide the
frequency scale factor in a 1–2–5 sequence.
2. The data are multiplied by the selected window function.
3. FFT is computed, using a fast implementation of the DFT
(Discrete Fourier Transform):
=−
1
kN
1
where:
X
nk
N
x
is a complex array whose real part is the m odified
k
=
k
0
xW
source time domain wavef orm, and whose imaginary part is
X
is the resulting complex frequency-domain waveform ;
0;
n
2
; and
N
W
=
is the number of points in
The generalized FFT algorithm, as im plemented her e, works
not
on N, which need
be a power of 2.
nk
,
x
and
X
k
.
n
X
4. The resulting com plex vector
is divided by the coherent
n
gain of the window function, in order to compensate for the loss of the signal energy due to windowing. This compensation provides accurate amplitude values for isolated spectrum peaks.
X
5. The real part of
is symmetric around the Nyquist
n
frequency, i.e.
Rn = R
N-n
,
while the imaginary part is asymmetric, i.e.
In = –I
N-n
.
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The energy of the signal at a frequency n is distributed equally between the first and the second halves of the spec trum; the energy at frequency 0 is completely contained in the 0 term.
The first half of the spectrum (Re, Im) , from 0 to the Nyquist frequency is kept for further processing and doubled in amplitude:
R’n = 2  R
= 2  I
I’
n
n
n
0 ≤ n < N/2 0 ≤ n < N/2.
6. The resultant waveform is computed for the spectrum type
selected.
If “Real”, “Imaginary”, or “Real + Imaginary” is selected, no further computation is needed. The appropriate part of the
R’
or
I’
or
R’
+
jI’
complex result is given as the re sult (
n
n
, as
n
n
defined above). If “Magnitude” is selected, the magnitude of the complex
vector is computed as:
MRI
nnn
22
=+’’
.
Steps 1–6 lead to the following result: An AC sine wave of amplitude 1.0 V with an integral number of
periods Np in the time window, transformed with the rectangular window, results in a fundamental peak of 1.0 V m agnitude in the spectrum at frequency Np × ∆f. However, a DC component of 1.0 V, transformed with the rectangular window, results in a peak of
2.0 V magnitude at 0 Hz. The waveforms for the other available spectrum types are
computed as follows:
Phase: angle = arctan (
angle = 0
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In/R
n
)
Mn > M Mn ≤ M
min min
.
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Where
M
is the minimum magnitude, fixed at about 0.001 of
min
the full scale at any gain setting, below which the angle is not well defined.
dBm Power Spectrum:
The
2
where
M
n
dBm PS
M
ref
10 20
log log
10
= 0.316 V (that is, 0 dBm is defined as a sine wave of
M
ref
2
M
n
10
M
ref
0.316 V peak or 0.224 V RMS, giving 1.0 mW into 50Ω). The dBm Power Spectrum is the same as dBm Magnitude, as
suggested in the above formula. dBm Power Density:
dBmPD dBmPS ENBW f
where
ENBW
corresponding to the selected window, and
=−× ×1010log
()
is the equivalent noise bandwidth of the filter
f
is the current
frequency resolution (bin width).
7. The FFT Power Average takes the complex frequency-
R’n
and
I’n
domain data
for each spectrum generated in Step
5, and computes the square of the magnitude:
2
2
M
n
= R’
2
+ I’
n
,
n
then sums
2
M
and counts the accumulated spectra. The
n
total is normalized by the number of spectra and converted to the selected result type using the same formulae as are us ed for the Fourier Transform.
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*ORVVDU\
Defines the terms frequently used in FFT spectrum analysis and relates them to the oscilloscope.
$OLDVLQJ If the input signal to a sampling acquisition system contains
components whose frequency is greater than the Nyquist frequency (half the sampling fr equency), ther e will be less than two samples per signal period. The r esult is that the contribution of these components to the sampled waveform is indistinguishable from that of components below the Nyquist frequency. This is aliasing.
The timebase and transform -size should be selected so that the resulting Nyquist frequency is higher than the highest significant component in the time-domain record.
&RKHUHQW*DLQ The normalized coherent gain of a filter corresponding to each
window function is 1.0 (0 dB) for a rectangular window and less than 1.0 for other windows. It defines the loss of signal energy due to the multiplication by the window function. This loss is compensated in the oscilloscope. This table lists the values for the implemented windows.
:LQGRZ)UHTXHQF\'RPDLQ3DUDPHWHUV
Highest Side
Window Type
Rectangular –13 3.92 1.0 0.0 von Hann Hamming –43 1.78 1.37 –5.35 Flat Top
Blackman–Harris
Lobe
(dB)
–32 1.42 1.5 – 6.02
–44 0.01 2.96 –11.05 –67 1.13 1.71 –7.53
Scallop Loss
(dB)
ENBW
(bins)
Coherent Gain
(dB)
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(1%: Equivalent Noise BandWidth (ENBW) is the bandwidth of a
rectangular filter (sam e gain at the center frequency), equivalent to a filter associated with each frequency bin, which would collect the same power from a white noise signal. In the table on the previous page, the ENBW is listed for each window function implemented and is given in bins.
)LOWHUV Computing an N-point FFT is equivalent to passing the time-
domain input signal through N/2 filters and plotting their outputs against the frequency. The spacing of f ilters is ∆f = 1/T while the bandwidth depends on the window function used (see Frequency bins).
)UHTXHQF\ELQV The FFT algorithm takes a discrete source waveform, defined
over N points, and computes N complex Fourier coefficients, which are interpreted as harmonic components of the input signal.
For a real source waveform (imaginary part equals 0), there are only N/2 independent harmonic components.
An FFT corresponds to analyzing the input signal with a bank of N/2 filters, all having the same shape and width, and centered at N/2 discrete frequencies. Each filter collects the signal energy that falls into the immediate neighborhood of its center
frequency, and thus it can be said that there are N/2 “frequency bins”.
The distance in hertz between the center frequencies of two neighboring bins is always:
f = 1/T, where T is the duration of the time-domain record in seconds. The width of the main lobe of the filter centered at each bin
depends on the window function used. The rectangular window has a nominal width at 1.0 bin. Other windows have wider main lobes (
see table
).
)UHTXHQF\5DQJH The range of frequencies computed and displayed is 0 Hz
(displayed at the left-hand edge of the screen) to the Nyquist frequency (at the rightmost edge of the trace).
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)UHTXHQF\5HVROXWLRQ In a simple sense, the frequency resolution is equal to the bin
width ∆f. That is, if the input signal changes its frequency by ∆f, the corresponding spectrum peak will be displaced by ∆f. For smaller changes of frequency, only the shape of the peak will change.
However, the effective frequency resolution (i.e. the ability to resolve two signals whose frequencies are alm ost the same) is further limited by the use of window functions. T he ENBW value of all windows other than the rectangular is greater than ∆f and the bin width.
the implemented windows.
/HDNDJH In the power spectrum of a sine wave with an integral num ber of
periods in the (rectangular) time window (i.e. the source frequency equals one of the bin frequencies), the spectrum contains a sharp component whose value accurately reflects the source waveform’s amplitude. For intermediate input frequencies this spectral component has a lower and broader peak.
The broadening of the base of the peak , str etching out into m any neighboring bins is termed side lobes of the filter associated with each frequency bin.
The filter side lobes and the r esulting leakage are reduced when one of the available window functions is applied. The best
reduction is provided by the Blackman–Harris and Flat Top windows. However, this reduction is offset by a broadening of the main lobe of the filter.
The table on page C–17 lists the ENBW values for
leakage
. It is due to the relatively high
1XPEHURI3RLQWV FFT is computed over the number of points (Transform Size)
whose upper bounds are the source number of points, and by the maximum number of points selected in the menu. FFT generates spectra of N/2 output points.
1\TXLVW)UHTXHQF\ The Nyquist frequency is equal to one half of the effective
sampling frequency (after the decimation): ∆f × N/2.
3LFNHW)HQFH(IIHFW If a sine wave has a whole number of periods in the time dom ain
record, the power spectrum obtained with a rectangular window will have a sharp peak, corresponding exactly to the frequency and amplitude of the sine wave. Otherwise the spectrum peak with a rectangular window will be lower and broader.
The highest point in the power spectrum can be 3.92 dB lower (1.57 times) when the source f requency is halfway between two
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discrete bin frequencies. This variation of the spectrum magnitude is called the
picket fence effect
scallop loss). All window functions compensate this loss to some extent, but
the best compensation is obtained with the Flat Top window.
2
3RZHU6SHFWUXP The power spectrum (V
) is the square of the magnitude
spectrum. The power spectrum is displayed on the dBm scale, with 0 dBm
corresponding to:
2
= (0.316 Vpeak)2,
Vref
where Vref is the peak value of the sinusoidal voltage, which is equivalent to 1 mW into 50 Ω.
2
3RZHU'HQVLW\6SHFWUXP The power density spectrum (V
/Hz) is the power spectrum divided by the equivalent noise bandwidth of the filter in hertz. The power density spectrum is displayed on the dBm scale, with
2
0 dBm corresponding to (Vref
/Hz).
6DPSOLQJ)UHTXHQF\ The time-domain records are acquired at sampling frequencies
dependent on the selected time base. Before the FFT computation, the time-domain record may be decimated. If the selected maximum number of points is lower than the source number of points, the effective sampling frequency is reduced. The effective sampling frequency equals twice the Nyquist frequency.
(the loss is called the
6FDOORS/RVV Loss associated with the picket fence effect. :LQGRZ)XQFWLRQV All available window functions belong to the sum of cosines
family with one to three non-zero cosine terms:
=−
1
mM
Wa
=
km
=
0
m
where:
M = 3
is the maximum number of terms, coefficients of the terms, decimated source waveform, and
k
2
p
 
N
is the number of points of the
mkN
N
k
is the time index.
≤<
0cos
a
are the
m
,
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The following table lists the coefficients functions seen in the time domain are symmetric around the point k = N/2.
&RHIILFLHQWV2I:LQGRZ)XQFWLRQV
Window Type a0 a1 a2
Rectangular 1.0 0.0 0.0 von Hann 0.5 –0.5 0.0 Hamming 0.54 –0.46 0.0 Flat-Top 0.281 –0.521 0.198 Blackman-Harris 0.423 –0.497 0.079
$SSHQGL[&5HIHUHQFHV Bergland, G.D.,
IEEE Spectrum, July 1969, pp. 41–52. A general introduction to FFT theory and applications.
am.
The window
A Guided Tour of the Fast Fourier Transform
,
Brigham, E.O.,
The Fast Fourier Transform
, Prentice Hall, Inc., Englewood Cliffs, N.J., 1974. Theory, applications and implementation of FFT. Includes discussion of FFT algorithms for N not a power of 2.
Harris, F.J.,
the Discrete Fourier Transform
On the Use of Windows for Har monic Analysis with
, Proceedings of the IEEE, vol. 66, No. 1, January 1978, pp. 51–83. Classic paper on window functions and their figures of mer it, with many examples of windows.
Ramirez, R.W.,
The FFT Fundamentals and Concepts
, Prentice Hall, Inc., Englewood Cliffs, N.J., 1985. Practice-oriented, many examples of applications.
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