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Anti-aliasing filter used in monitoring systems of
electrical equipmentTopic number: .
Do not indicate authors' names!
Abstract - Anti-aliasing filters are an important element indata acquisition systems. Mounted upstream of the analogue digital converter, they serve to limit the frequency spectrum ofthe input signal, to avoid overlapping signal spectra of the signalresulting from sampling, known as the refolding The obtainedcircuit has advantages of modularization design, strongtransplantation, and has widely use value and applicationprospect.
I. GENERALNOTIONSOFDATAACQUISITIONSYSTEMS
Data acquisition is the process of sampling signals that
measure real world physical conditions and converting the
resulting samples into digital numeric values that can be
manipulated by a computer. Data acquisition systems
(abbreviated with the acronym DAS or DAQ) typically
convert analog waveforms into digital values for processing.
Therefore a data acquisition system, is designed to measure
or generate physical signals, and for this purpose contains in
its structure a data acquisition board.
Acquisition board is often considered the entire system.
These include in sensors that convert physical parameters to
electrical signals, their construction conditioning signalequipment to convert sensor signals into a form that can be
converted to digital values; analog to digital converters
(CAN) and Digital Analogue (CNA), inputs and outputs (I /
O) Analog and Digital, count (count, time clock generators)
and DMA (Direct memory Access) memory units to transmit
data directly to the computer without using microprocessor.
The most important operation is the analog digital
conversion, made with one or more circuits, depending on the
type of application). Another often encountered operations in
such systems are sampling and temporary storing of taken
samples.
The sampling frequency is determined by: the frequencyspectrum of the input signals; speed of the A / D converter;
accuracy of required processing.
Information processing may consist of simple operations
(comparisons) to complicated mathematical processing
(integration, differentiation, mediation, Fourier transform
calculation, etc..).
II. THE SHANNONSAMPLINGTHEOREM
To analyze and synthesize a signal in the digital domain,
analog signals must be converted into digital signals through
the acquisition of "samples" at certain regular intervals.
This process, called sampling is accomplished by
"questioning" the analog signal at regular periods of time,
denoted Te (where Te is called the sampling period is theinverse of the sampling frequency, fe (fe = 1/Te)).
In mathematical form, a sample of a signal is obtained by
multiplying the signal with a unit step pulse.
To avoid loss of information in the process of sampling an
analog signal, the sampling frequency must be at least two
times the maximum frequency of the sampled signal.The consequences of sampling at a lower frequency that the
highest frequency of the signal, generates a phenomena
known as refolding (aliasing). This concept results from
misidentification of the required frequency for signal
reconstruction.
The Shannon sampling theorem introduced restrictions on
the frequency of sampled signal in time, f (t) as follows: To
reconstruct a signal f (t), it is necessary sampling at a rate
twice as high as the maximum signal frequency.
If Shannon's theorem is not respected, it may cause
interference with other frequencies that are not in the band of
interest regarding the analyzed signal(aliasing, mentionedabove).
The same unwanted phenomenon is met if a digital signal
contains frequencies higher than the Nyquist frequency
(where the Nyquist frequency is half the sampling frequency
fN=fe/2 ).In fact, the spectrum of an acquired signal is infinite width.
Although Shannon's sampling theorem respects, the
phenomenon of aliasing occurs (overlapping power spectrum
of the signal). For this it is necessary to filter the source
signal by placing a low-pass filter (filter guard) upstream of
the sampling device.
III. THEORETICALCONCEPTSABOUTFILTERS
In signal processing, a filter is a device or process that
removes from asignalsome unwanted component or feature,
this means removing somefrequenciesand not others in order
to suppress interfering signals and reduce background noise.
The main specific size filters are cut off frequency - the
frequency at which the amplitude of the output signal
represents 0.707 of the input signal amplitude, and band pass
- band between zero and cutoff frequency.
http://en.wikipedia.org/wiki/Signal_processinghttp://en.wikipedia.org/wiki/Signal_processinghttp://en.wikipedia.org/wiki/Signal_(electronics)http://en.wikipedia.org/wiki/Signal_(electronics)http://en.wikipedia.org/wiki/Signal_(electronics)http://en.wikipedia.org/wiki/Frequencyhttp://en.wikipedia.org/wiki/Frequencyhttp://en.wikipedia.org/wiki/Frequencyhttp://en.wikipedia.org/wiki/Signal_noisehttp://en.wikipedia.org/wiki/Signal_processinghttp://en.wikipedia.org/wiki/Signal_(electronics)http://en.wikipedia.org/wiki/Frequencyhttp://en.wikipedia.org/wiki/Signal_noise -
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The frequency response can be classified into a number ofdifferentbandformsdescribing which frequencies the filter
passes (thepassband) and which it rejects (thestopband):
low-pass filter low frequencies are passed, high frequencies
are attenuated.High-pass filter high frequencies are passed,
low frequencies are attenuated. Band-pass filter only
frequencies in a frequency band are passed. Band-stop
filteror band-reject filter only frequencies in a frequencyband are attenuated. Notch filter rejects just one specific
frequency - an extreme band-stop filter. Comb filter has
multiple regularly spaced narrow passbands giving the
bandform the appearance of a comb.All-pass filter all
frequencies are passed, but the phase of the output is
modified.
Depending on the design of analog filters, they can be
classified: Passive filters consisting only of passive circuit
elements (resistors, coils, capacitors). The consequence of
such structures is that the amplitude of the signal at the output
of the filter may be greater than the amplitude of the input
signal (maximum value of the transfer function can not be
over-unit).Active filterwhich is a combination of passive andactive elements of the filter circuit (most of the time, the
operational amplifiers). The presence of active elements
ensure amplification of signals with frequencies in the
passband and improved filter characteristics.
Order filter is important for several reasons. It is directly
related to the number of components of the filter, and thus the
cost, size and complexity of the filter. Therefore, higher order
filters are more expensive, take up more space and are more
difficult to create. The main advantage of a high order filter is
that it will have greater attenuation slope from lower order
filters.
First and second order filters are building floors for higherorder filters.Even order filters contain two floors of the order
and odd order filters contain a floor plus first order.
Usually the amount of attenuation is expressed in dB. The
most common measure of attenuation is "dB / decade."
Maximum attenuation rate is 20 dB / decade for all pole filter,
low-pass filters and high-pass.
IV. MONITORINGELECTRICALDRIVESUSING PC DATAACQUISITIONSYSTEMSWITHANTIALISINGFILTERS
Real signals of voltage or current may contain higher
frequency spectral components that the Nyquist frequency,known as refolding effect, usually causing inaccurate
measurements.
The most simple and effective way of avoiding this
phenomena is to filter the original signal using a low-pass
type filter characterized by a cut off frequency equal to the
Nyquist frequency.This filter is called antialiasing filter and it
requires its attenuation outside the pass band more than
50dB.
An anti-aliasing filter has two properties defining it. One is
the cutoff frequency, the point at which frequencies above it
are attenuated, the other attribute of a filter is its slope,
measured in dB per octave.
One more notable aspect of anti-aliasing filters is that the
steeper their slope, the more distortion is present near the
cutoff frequency. This is due to a property of physics that
applies only to analog filters. Thus it is better to increase your
sampling rate to allow for a higher Nyquist frequency so that
your anti-aliasing filter's slope can be shallower.The problem which arises in the implementation of this
kind of filter is that the data acquisition interface allows
software selection of the sampling frequency in a wide range
of values. So the cutoff frequency of antialiasing filter should
modify in order to eliminate higher frequencies.
One of the techniques used to prevent the refolding is
oversampling and analog-to-digital conversion in
combination with anti aliasing analog filter. It is additionally
used digital filters and decimation operations along with
keeping requirements sampling theorem.
In order to determine corrective action, we must directly
address the two conditions necessary for exact signal
reconstruction. The first solution calls for low-pass filteringbefore sampling.This method, known as prefiltering,
bandlimits the signal to levels below fmax, thereby eliminating
the offending high frequencies. Notice that the frequency at
which the signal is to be sampled imposes limits on the
allowable bandwidth.
The second solution is to point-sample at a higher
frequency. In doing so, the replicated spectra are spaced
farther apart, thereby separating the overlapping spectra tails.
This approach theoretically implies sampling at a resolution
determined by the highest frequencies present in the signal.
In practice, antialiasing is performed through a
combination of these two approaches. That is, the samplingfrequency is increased so as to reduce the amount of
bandlimiting to a minimum.
http://en.wikipedia.org/wiki/File:Bandform_template.svghttp://en.wikipedia.org/wiki/File:Bandform_template.svghttp://en.wikipedia.org/wiki/File:Bandform_template.svghttp://en.wikipedia.org/wiki/Passbandhttp://en.wikipedia.org/wiki/Stopbandhttp://en.wikipedia.org/wiki/Stopbandhttp://en.wikipedia.org/wiki/Low-pass_filterhttp://en.wikipedia.org/wiki/Low-pass_filterhttp://en.wikipedia.org/wiki/High-pass_filterhttp://en.wikipedia.org/wiki/Band-pass_filterhttp://en.wikipedia.org/wiki/Band-pass_filterhttp://en.wikipedia.org/wiki/Band-stop_filterhttp://en.wikipedia.org/wiki/Band-stop_filterhttp://en.wikipedia.org/wiki/Notch_filterhttp://en.wikipedia.org/wiki/Notch_filterhttp://en.wikipedia.org/wiki/Comb_filterhttp://en.wikipedia.org/wiki/Comb_filterhttp://en.wikipedia.org/wiki/All-pass_filterhttp://en.wikipedia.org/wiki/All-pass_filterhttp://en.wikipedia.org/wiki/All-pass_filterhttp://en.wikipedia.org/wiki/File:Bandform_template.svghttp://en.wikipedia.org/wiki/Passbandhttp://en.wikipedia.org/wiki/Stopbandhttp://en.wikipedia.org/wiki/Low-pass_filterhttp://en.wikipedia.org/wiki/High-pass_filterhttp://en.wikipedia.org/wiki/Band-pass_filterhttp://en.wikipedia.org/wiki/Band-stop_filterhttp://en.wikipedia.org/wiki/Band-stop_filterhttp://en.wikipedia.org/wiki/Notch_filterhttp://en.wikipedia.org/wiki/Comb_filterhttp://en.wikipedia.org/wiki/All-pass_filter -
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V. NUMERICALSIMULATIONSANDRESULTS
ACKNOWLEDGMENT
The preferred spelling of the word acknowledgment in
America is without an e after the g. Try to avoid the
stilted expression, One of us (R. B. G.) thanks Instead,
try R.B.G. thanks
REFERENCES
[1] G. Eason, B. Noble, and I.N. Sneddon, On certain integrals ofLipschitz-Hankel type involving products of Bessel functions, Phil.Trans. Roy. Soc. London, vol. A247, pp. 529-551, April 1955.
[2] J. Clerk Maxwell,A Treatise on Electricity and Magnetism, 3rd ed., vol.2. Oxford: Clarendon, 1892, pp.68-73.
[3] I.S. Jacobs and C.P. Bean, Fine particles, thin films and exchangeanisotropy, in Magnetism, vol. III, G.T. Rado and H. Suhl, Eds. NewYork: Academic, 1963, pp. 271-350.
[4] K. Elissa, Title of paper if known, unpublished.[5] R. Nicole, Title of paper with only first word capitalized, J. Name
Stand. Abbrev., in press.[6] Y. Yorozu, M. Hirano, K. Oka, and Y. Tagawa, Electron spectroscopy
studies on magneto-optical media and plastic substrate interface,IEEETransl. J. Magn. Japan, vol. 2, pp. 740-741, August 1987 [Digests 9th
Annual Conf. Magnetics Japan, p. 301, 1982].
[7] M. Young, The Technical Writers Handbook. Mill Valley, CA:University Science, 1989.