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M Tech Microelectronics,

WILP
Prof. Kranthi Kumar Palavalasa.
BITS Pilani Assistant Professor
Pilani Campus Department of EEE (WILP)
BITS Pilani
Pilani Campus

ES ZG573/ EEE ZG573/ MEL ZG573 :


Digital Signal Processing

Lecture 13
Implementation of
Discrete-Time Systems
Introduction
Design of IIR filters from analog filters
Frequency transformations

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Introduction
the realization of linear time-invariant discrete time systems in either software or
hardware.
particular importance are the cascade, parallel. and lattice structures, which exhibit
robustness in finite-word-length implementations. An analog filter can be described by its
system function.

where {αk} and {βk} are the filter coefficients.

The Laplace transform

The analog filter having the rational system function H(s), can be described by the linear
constant-coefficient differential equation

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


STRUCTURES FOR THE REALIZATION OF DISCRETE-TIME
SYSTEMS
consider the important class of linear time-invariant discrete-time systems characterized
by the general linear constant-coefficient difference equation.

linear time-invariant discrete-time systems are also characterized by the rational system
function

the various methods of implementing (7.1.1) or (7.1.2) in either hardware, or in software


on a programmable digital computer.

Copyright © 2005. Shi Ping


BITS Pilani, Deemed to be University under Section 3 CUC
of UGC Act, 1956
the major factors that influence our choice of a specific realization are computational
complexity, memory requirements, and finite-word-length effects in the computations
Computational complexity refers to the number of arithmetic operations (multiplications,
divisions, and additions) required to compute an output value y(n) for the system.
Memory requirements refers to the number of memory locations required to store the
system parameters, past inputs. past outputs. and any intermediate computed values.
Finite-word-lengrh effecrs or finite-precision effects refer to the quantization effects that
are inherent in any digital implementation of the system. either in hardware or in
software.

Other factors, such as whether the structure or the realization lends itself to parallel
processing, or whether the computations can be pipelined. may play a role in our
selection of the specific implementation.

Copyright © 2005. Shi Ping


BITS Pilani, Deemed to be University under Section 3 CUC
of UGC Act, 1956
STRUCTURES FOR FIR
SYSTEMS
In general, an FIR system is described by the difference equation.

or, equivalently, by the system function

Furthermore, the unit sample response of the FIR system is identical to the coefficients
{bk},that is,

Copyright © 2005. Shi Ping


BITS Pilani, Deemed to be University under Section 3 CUC
of UGC Act, 1956
Direct-Form Structure

The direct-form realization follows immediately from the non-recursive difference


equation given by or, equivalently, by the convolution summation

this structure requires M - 1 memory locations for storing the M - 1 previous inputs,
and has a complexity of M multiplications and M - 1 additions per output point.

Direct-form realization of FIR system


the structure in Fig. 7.1, resembles a tapped delay line or a transversal system
BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956
When the FIR system has linear phase, as described in Section 8.2, the unit
sample response of the system satisfies either the symmetry or asymmetry
condition.

For such a system the number of multiplications is reduced from M to M/2 for M
even and to (M- 1)/2 for M odd.

Direct-
form
realization
of linear-
phase FIR
system

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Cascade-Form Structures

The cascade realization follows naturally from the system function


factor H(z) into second-order FIR systems so that

Where

The cascade-form realization along with


the basic second-order section
are shown in Fig.

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Frequency-Sampling Structures

This mapping has the desirable


property that a stable analog
filter is transformed into a
stable digital filter.

It is not possible, for example,


to transform a high pass analog
filter into a corresponding high
pass digital filter.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Example 1

Convert the analog band pass filter with system function into a digital IIR filter by use
of the backward difference for the derivative

Solution Substitution for s equivalent into Ha(s) yields

For example, if T =0.1, the poles are located at

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Example 2

Convert the analog band pass filter in Example 1 into a digital IIR filter by use of
the mapping

Solution: By substituting for S in H(s), we obtain

This mapping has introduced two additional poles in the conversion from H a(s) to
H(z). .
As a consequence, the digital filter is significantly more complex than the analog
filter. This is a major drawback to the mapping given above.

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IIR Filter Design by lmpulse lnvariance

Objective is to design an IIR filter having a unit sample response h(n) that is the
sampled version of the impulse response of the analog filter.

Implications:
Continuous time signal xa(t) with spectrum Xa(F) is sampled at a rate Fs = 1/T
samples per second.

The spectrum of the sampled signal is the periodic repetition of the scaled spectrum
FsXa(F)with period Fs.

Where f = F/Fs is the normalized frequency.


Aliasing occurs if the sampling rate Fs is less than twice the highest frequency
contained in Xa(F).

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


An analog filter with frequency response Ha(F), the digital filter with unit sample
response h(n)= ha(nT) has the frequency response.

or, equivalently,

Frequency response Ha(Ω) of the analog


filter
and
frequency response of the corresponding
digital filter with aliasing.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


The digital filter with frequency response H(w) has the frequency response
characteristics of the corresponding analog filter.
If the sampling interval T is selected sufficiently small to completely avoid or
at least minimize the effects of aliasing.

It is also clear that the impulse invariance method is inappropriate for designing high
pass filters due the to spectrum aliasing that results from the sampling process.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


The mapping of points between the z-plane and the s-plane implied by the sampling
process,
The z-transform of h(n) to the Laplace transform of ha(t).

Consider the mapping of points from the s-plane to the z-plane


σ < 0 implies that 0 < r < 1; the LHP in s is
mapped inside the unit circle
σ > 0 implies that r > 1, RHP in s is mapped
outside the unit circle
σ = 0, we have r = 1.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Example 3

Convert the analog filter with system function into a digital IIR filter by means of the
impulse invariance method

Solution We note that the analog filter has a zero at s = -0.1 and a pair of complex
conjugate poles at

The partial-fraction expansion of Ha(s)

Frequency response of digital filter in Example


BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956
IIR Filter Design by the Bilinear Transformation

The IIR filter design techniques described have a severe limitation


that they are appropriate only for low pass filters and a limited class of band
pass filters.

A mapping from the s-plane to the z-plane, called the bilinear transformation
overcomes the limitation of the other two design

The BiIinear transformation is a conformal mapping that transforms the jΩ-axis into
the unit circle in the z-plane only once, thus avoiding aliasing of frequency
components.

All points in the LHP of s are mapped inside the unit circle in the z-plane
All points in the RHP of s are mapped into corresponding points outside the unit
circle in the z-plane.

The bilinear transformation can be linked to the trapezoidal formula for numerical
integration.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Consider an analog Iinear filter with system function

This system is also characterized by the differential equation

Integrate the derivative and approximate the integral by the trapezoidal formula.

The approximation of the integral by the trapezoidal formula at t = nT and to = nT - T


yields

Now the differential equation evaluated at t = nT yields

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We use () to substitute for the derivative in () and thus obtain a difference equation for
the equivalent discrete-time system.
With y(n) = y ( n T ) and x ( n ) = x ( n T ) , we obtain the result

The Z-transform of this difference equation is

the system function of the equivalent digital filter is

or, equivalently,

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Clearly, the mapping from the s-plane to the z-plane is

This is called the bilinear transformation


Although our derivation of the bilinear transformation was performed for a
first-order differential equation, it holds, in general, for an Nth-order differential
equation.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Characteristics of the bilinear transformation

Consequently,

if r < 1, then α < 0: the LHP in s maps into the inside of the unit circle
if r > 1, then α > 0: the RHP in s maps into the outside of the unit circle.
When r =1, then α = 0

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


The relationship between the frequency variables in the two domains is illustrated in
Fig.

the entire range in Ω is mapped only


once into the range – π ≤ ω ≤ π.

the mapping is highly nonlinear

Frequency compression or frequency


warping, due to the nonlinearity of the
arctangent function.

the bilinear transformation maps the point


s = ∞ into the point z = -1. => the single-
pole lowpass filter in which has a zero at s
= ∞, results in a digital filter that has a zero
at z =-1.

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Example 4

Solution

the desired mapping is

The resulting digital filter has the system function

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


This filter has poles at

and zeros at

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Example 5
Design a single-pole lowpass digital filter with a 3-dB bandwidth of 0.2π, using the
bilinear transformation applied to the analog filter

Thus the analog filter has the system function

This represents our filter design in the analog domain

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


apply the bilinear transformation to convert the analog filter into the desired digital filter.

The frequency response of the digital filter is

At ω =0, H (0)=1, and at ω =0.2 π we have IH(0.2π)| = 0.707, which is the desired
response.

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FREQUENCY TRANSFORMATIONS

Frequency Transformations in the Analog Domain

The transformation that accompIishes this is

The system function of the high pass filter is

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Equivalently, we can accomplish the same result in a single step by means of the
transformation

where

Thus we obtain

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which leads to

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BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956
Example 6
Transform the single-pole lowpass Butterworth filter with system function

The resulting filter has a zero at s = 0 and poles at

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Frequency Transformations in the
Digital Domain

It is clear that we must have |g(w)l =1 for all ω.That is, the mapping must be
all-pass. Hence it is of the form

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


Convert the single-pole lowpass Butterworth filter with system function

Substitution into H(z) yields

BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956


BITS Pilani, Deemed to be University under Section 3 of UGC Act, 1956

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