Filters

Geo-physical Data Processing

Submitted to: Mr. Muyassar Hussain

Submitted by: Shoaib Rathore
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Contents
What is Filtering?.................................................................................................3
Objective:...............................................................................................................3
Filter Classification:.............................................................................................3
Design Examples for Passive and Active Filters:..........................................4
Passive R, L, C Filter Design:...........................................................................4
Active Filter Design:......................................................................................5
Discrete-Time Filters:...................................................................................7
Digital Filter Design Process:.............................................................................7
FIR Filter Design:.................................................................................................9
Windowed FIR Filters:.......................................................................................10
Design of Narrowband FIR Filters:................................................................11
Ant aliasing and Smoothing Filters:................................................................11
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What is Filtering?

A filter can be defined as a signal processing system whose output signal, usually
called the response,
Differs from the input signal, called the excitation, such that the output signal has
some prescribed properties.

In more practical terms an electric filter is a device designed to suppress, pass, or

separate a group of signals from a mixture of signals according to the
specifications in a particular application

Objective:
The Objective areas of filtering are manifold, for example to band-limit signals
before sampling to reduce aliasing, to eliminate unwanted noise in communication
systems, to resolve signals into their frequency components, to convert discrete-
time signals into continuous-time signals, to demodulate signals, etc. Filters are
generally classified into three broad classes: continuous-time, sampled data, and
discrete-time filters depending on the type of signal being processed by the filter.
Therefore, the concept of signals is fundamental in the design of filters.

Filter Classification:

Filters are commonly classified according to the filter function they perform. The
basic functions are:
low-pass, high-pass, band pass, and band stop.

If a filter passes frequencies from zero to its cutoff frequency Wc and stops all
frequencies higher than the cutoff frequencies, then this filter type is called an ideal
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low pass filter. In contrast, an idea high-pass filter stops all frequencies below its
cutoff frequency and passes all frequencies above it. Frequencies extending
fromW1toW2are passed by an ideal Band pass filter, while all other frequencies
are stopped. An ideal band stop filter stops frequencies fromW1toW2and passes all
other frequencies.

The Figure depicts the magnitude functions of the four basic ideal filter types.
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FIGURE: The magnitude function of an ideal filter is 1 in the pass band and 0 in the stop band
as shown for
(a) low-pass, (b) high-pass, (c) band pass, and (d) stop band filters.

Design Examples for Passive and Active Filters:

Passive R, L, C Filter Design:

The simplest and most commonly used passive filter is the simple, first-order (N =
1) R–C filter shown in Figure below. Its transfer function is that of a first-order
Butterworth low-pass filter. The transfer function and –3 dB Wc are
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FIGURE: A passive first-order RC filter can serve as an ant aliasing filter or to minimize high-
frequency noise.

FIGURE: A passive filter can have the symbolic representation of a doubly terminated filter.

FIGURE: Even and odd N passive all-pole filter networks can be realized by several circuit
configurations (Nodd, above; N even, below).

Active Filter Design:

Active filters are widely used and commercially available with cutoff frequencies
from mill hertz to
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Megahertz. The characteristics that make them the implementation of choice for
several applications are
Small size for low-frequency filters because they do not use inductors; precision
realization of theoretical
transfer functions by use of precision resistors and capacitors; high input
impedance that is easy to drive
and for many circuit configurations the source impedance does not effect the
transfer function; low output impedance that can drive loads without effecting the
transfer function and can drive the transient, switched-capacitive, loads of the input
stages of A/D converters and low (N+THD) performance for pre- A/D ant aliasing
applications (as low as –100 dBc).

The most commonly used two-pole active filter circuits are the Sallen and Key
low-pass resonator, the multiple feedback band pass, and the state variable
implementation as shown in Figure a, b, and c. In the analyses that follow, the
more commonly used circuits are used in their simplest form.

The Sallen and Key circuit of Figure (a) is used primarily for its simplicity. Its
component count is
the minimum possible for a two-pole active filter. It cannot generate stop band
zeros and therefore is
limited in its use to monotonic roll-off transfer functions such as Butterworth and
Bessel filters. Other
limitations are that the phase shift of the amplifier reduces the Q of the section and
the capacitor ratio
becomes large for high-Q circuits. The amplifier is used in a follower configuration
and therefore is
subjected to a large common mode input signal swing which is not the best
condition for low distortion
performance. It is recommended to use this circuit for a section Q < 10 and to use
an amplifier whose
gain bandwidth product is greater than 100 fp.
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FIGURE: Second-order active filters can be realized by common filter circuits: (A) Sallen and
Key low-pass,
(B) multiple feedback bandpass, (C) state variable.

Discrete-Time Filters:

A digital filter is a circuit or a computer program that computes a discrete output

sequence from a discrete input sequence. Digital filters belong to the class of
discrete-time LTI (linear time invariant) systems, which are characterized by the
properties of causality, recursibility, and stability, and may be characterized in the
time domain by their impulse response and in the transform domain by their
transfer function. The most general case of a discrete-time LTI system with the
input sequence denoted by x(kT) and the resulting output sequence y(kT) can be
described by a set of linear difference equations with constant coefficients.
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Digital Filter Design Process:

The digital filter design procedure consists of the following basic steps:

1. Determine the desired response. The desired response is normally specified in

the frequency
domain in terms of the desired magnitude response and/or the desired phase
response.

2. Select a class of filters (e.g., linear-phase FIR filters or IIR filters) to

approximate the desired
response.

3. Select the best member in the filter class.

FIGURE: The difference equation of a digital filter can be realized by a direct-form I

implementation that uses
separate delay paths for the X and Y summations.
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FIGURE: A direct-form II implementation of the difference equations minimizes the number of

delay elements.

4. Implement the best filter using a general-purpose computer, a DSP, or a custom

hardware chip.

5. Analyze the filter performance to determine whether the filter satisfies all the
given criteria.

FIR Filter Design:

In many digital signal-processing applications, FIR filters are generally preferred

over their IIR counterparts, because they offer a number of advantages compared
with their IIR equivalents. Some of the good properties of FIR filters are a direct
consequence of their no recursive structure. First, FIR filters are inherently stable
and free of limit cycle oscillations under finite-word length conditions. In addition,
they exhibit a very low sensitivity to variations in the filter coefficients. Second,
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the design of FIR filters with exactly linear phase (constant group delay) vs.
frequency behavior can be accomplished easily. This property is useful in many
application areas, such as speech processing, phase delay equalization, image
processing, etc.

FIGURE: The sequence of the delays and summations can be varied to produce alternative
direct-form
implementations

The accuracy of an FIR approximation is described by the following parameters:

dP passband ripple

ds stopband attenuation

Dw transition bandwidth

These quantities are depicted in Figure below for a prototype low-pass filter. dp
and ds characterize the
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permissible errors in the passband and in the stopband, respectively. Usually, the
passband ripple and
stopband attenuation are given in decibels, in which case their values are related to
the parameters dp
and ds.

FIGURE 22.12 Tolerance limits must be defined for an FIR low-pass filter magnitude response

Windowed FIR Filters:

Several design techniques can be employed to synthesize linear-phase FIR filters.

The simplest implementation is based on windowing, which commonly begins by
specifying the ideal frequency response and expanding it in a Fourier series and
then truncating and smoothing the ideal impulse response by means of a window
function. The truncation results in large ripples before and after the discontinuity of
the ideal frequency response known as the Gibbs phenomena, which can be
reduced by using a window function that tapers smoothly at both ends. Filters
designed in this way possess equal pass band ripple and stop band attenuation.
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Design of Narrowband FIR Filters:

When using conventional techniques to design FIR filters with especially narrow
bandwidths, the resulting
filter lengths may be very high. FIR filters with long filter lengths often require
lengthy design and
implementation times, and are more susceptible to numerical inaccuracy. In some
cases, conventional
filter design techniques, such as the Parks–McClellan algorithm, may fail the
design altogether. A very
efficient algorithm called the interpolated finite impulse response (IFIR) filter
design technique can be
employed to design narrowband FIR filters. Using this technique produces
narrowband filters that require
far fewer coefficients than those filters designed by the direct application of the
Parks–McClellan algorithm.

Ant aliasing and Smoothing Filters:

In this section two practical application areas of filters in the analog conditioning
stage of a data
acquisition system are discussed. A block diagram of a typical data acquisition
system is shown in Figure, consisting of an ant aliasing filter before the analog-
to-digital converter (ADC) and a smoothing filter after the digital-to-analog
converter (DAC).

FIGURE: A data acquisition system with continuous time inputs and outputs uses antialias
prefiltering, an A/D
converter, digital signal processing, a D/A converter, and an output smoothing filter.
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Bibliography

Ant aliasing filter: Ant aliasing filters remove any frequency elements above the
Nyquist frequency. They
are employed before the sampling operation is conducted to prevent aliasing in the
sampled version
of the continuous-time signal.

Band pass filter: A filter whose pass band extends from a lower cutoff frequency
to an upper cutoff
frequency. All frequencies outside this range are stopped.

Equiripple: Characteristic of a frequency response function whose magnitude

exhibits equal maxima
and minima in the pass band.

Finite impulse response (FIR) filter: A filter whose response to a unit impulse
function is of finite length,
i.e., identically zero outside a finite interval.

High-pass filter: A filter that passes all frequencies above its cutoff frequency and
stops all frequencies
below it.

Ideal filter: An ideal filter passes all frequencies within its pass band with no
attenuation and rejects all
frequencies in its stop band with infinite attenuation. There are five basic types of
ideal filters: low pass,
high-pass, band pass, stop band, and all-pass.

Infinite impulse response (IIR) filter: A filter whose response to a unit impulse
function is of infinite
length, i.e., nonzero for infinite number of samples.

Low-pass filter: A filter that attenuates the power of any signals with frequencies
above its defined cutoff frequency.
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Pass band: The range of frequencies of a filter up to the cutoff frequency.

Stop band: The range of frequencies of a filter above the cutoff frequency.

Transition region: The range of frequencies of a filter between a pass band and a
stop band.

References

1. S. Mitra and J. Kaiser, Handbook for Digital Signal Processing, New York:
John Wiley & Sons, 1993.

2. E. Christian and E. Eisenmann, Filter Design Tables and Graphs, New York:
John Wiley & Sons, 1966.

3. A. Antoniou, Digital Filter: Analysis and Design, New York: McGraw-Hill,

1979.

4. R. Saal, Handbuch zum Filterentwurf [Handbook of filter design], Frankfurt:

Allgemeine Elektricitäts- Gesellschaft AEG-Telefunken, 1979.

5. G. Temes and S. Mitra, Modern Filter Theory and Design, New York: John
Wiley & Sons, 1973.

6. E. Cunningham, Digital Filtering, New York: John Wiley & Sons, 1995.
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