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Chapter 7

Filter Design
Techniques

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Outline
Digital Signal Processing

7.0 Introduction
7.1 Design of Discrete Time IIR Filters
7.2 Design of FIR Filters

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7.0 Introduction
Digital Signal Processing

Definition of Filter

Filter is a system that passes certain


frequency components and totally rejects
all others, but in a broader context any
system that modifies certain frequencies
relative to others is called a filter.

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The Design of Filter
Digital Signal Processing

1 The specification of the desired


properties of the system.
2 The approximation of the
specification using a causal
discrete-time system.
3 The realization of the system.

In this chapter, we focus on the


second step.

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The relationship between specifications
of the discrete-time filter and the
Digital Signal Processing
effective continuous-time filter

When a discrete-time filter is to be


used for discrete-time processing of
continuous-time filter and the effective
continuous-time filter are typically given
in the frequency domain.

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Digital Signal Processing

If a effective continuous-time
system has the frequency response.

Basic system for discrete-time filtering of


continuous-time signals.

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Digital Signal Processing

In such cases, it is straightforward to


convert from specifications on the effective
continuous-time filter through the relation
ω = ΩT.That is, H(ejω) is specified over one
period by the equation :

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Digital Signal Processing
Example

Consider a discrete-time filter that is to be


used to lowpass filter a continuous-time
signal using the basic configuration.
Specifically, we want the overall system to
have the following properties when the
sampling rate is 104 samples/s (T=10-4 s) :
(1) The gain |Heff(jΩ)| should be within
∓0.01 (0.086dB) of unity (zero dB) in the
frequency band 0 ≤ Ω ≤ 2π(2000).

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Digital Signal Processing

(2) The gain should be no greater than 0.001


(-60dB) in the frequency band 2π(3000) ≤ Ω

Such a set of lowpasss pecifications on


|Heff(jΩ)| can be depicted where the limits of
tolerable approximation error are indicated
by the shaded horizontal lines.

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Digital Signal Processing

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7.1 Design of Discrete-time IIR Filters
form Continuous-Time Filters
Digital Signal Processing

The Transformation of a continuous-time


filter into a discrete-time filter meeting
prescribed specifications. The Reasons for
Using this Method:
- The art of continuous-time IIR filter design
has developed and many results can be used.
- Many continuous-time IIR filter design
methods have relatively simple closed form
design formulas, therefore it is easy to carry
out.

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Digital Signal Processing

- The standard approximation methods for


continuous-time IIR filters can not be
directly used in discrete-time filter design.
3. Processes of design:
- Specifications transformation;
- Continuous-time filter design;
- Mapping continuous-time filter into
discrete-time filter (From s-plane to z-
plane) .

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7.1.1 Filter Design by Impulse Invariance
Digital Signal Processing

If hc(t) is the impulse response of continuous-


time filter, and hc(nTd) is equally spaced samples
of it.

The frequency response :

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Digital Signal Processing

If the continuous-time filter is bandlimited,


so that

then

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Digital Signal Processing

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Digital Signal Processing
Example

Assume that the specifications for the


designed discrete-time filter are shown in
next slide with ,δ1 = 0.10875, δ2= 0.17783,
ωp = 0.2π and ωs = 0.3π. Τhe maximum gain
in stopband is -15dB (20log10 0.17783), The
maximum deviation of 1dB below 0dB gain
in passband (20log10(1) – 20log10(1-0.10875)
=-1 dB). In this case the band pass tolerance
is between 1- δ1 and 1.

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The impulse invariance transformation


from CT to DT :

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Digital Signal Processing
Example

Consider the design of a lowpass discrete-


time filter by applying impulse invariance to
an appropriate Butterworth continuous-time
filter. The specifications for the discrete-time
filter are :

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Digital Signal Processing

Choose Td =1 so that ω=Ω

Continuous-time Butterworth filter with


magnitude function |Hc(jΩ)|

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Digital Signal Processing

Let Ωp = 0.2π and Ωs = 0.3π

The magnitude squared function of a


Btterworth filter

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Digital Signal Processing

So that the filter design process consists


of determining the parameters N and Ωc to
meet the desired specification.

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Digital Signal Processing

Since N must be integer N=6 substuting N=6


in equation slide 26. We have Ωc = 0.7032

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Find the poles :

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7.1.2 Bilinear Transformation
Digital Signal Processing

In order to avoid the aliasing in impulse


invariance, we introduce another method of
transformation bilinear transformation, which
use an algebraic transform between the
variables s and z. This transform is

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Digital Signal Processing

In the transformation, -∞ ≤ Ω ≤ ∞ maps


onto -π ≤ ω ≤ π ,the transformation
between the continuous-time and discrete-
time frequency variables must be
nonlinear. Therefore the use of this
technique is restricted to the situation
where the corresponding warping of the
frequency axis is acceptable.

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Digital Signal Processing

To develop the properties of the algebraic


transformationWe solve for z to obtain :

Substituting s = σ+jΩ, we obtain :

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Digital Signal Processing

If σ<0 then |z|<1 for any value of Ω.


Similarly, If σ>0 then |z|>1for all Ω.That is
if a pole of Hc(s) is in the left−half s-plane,
its image in the z-plane will be inside the
unit circle. Therefore causal stable
continuous-time filters map into causal
stable discrete-time filters.

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Digital Signal Processing

To show that the jΩ-axis of the s-plane


maps onto the unit circle, we substitute
s=jΩ :

It is clear that |z| =1 for all value of s on


the jΩ-axis

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Digital Signal Processing

To derive the relationship between the


variables ω and Ω, we substituting z= ejω.

or

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Equating real and imaginary parts on


both sides leads to the relations σ=0

or

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Digital Signal Processing

The bilinear transformation avoids the


problem of aliasing encountered with the use of
impulse invariance because it maps the entire
imaginary axis of the s-plane onto the unit circle
in the z-plane. The price paid for this, however,
is the nonlinear compression the frequency axis.

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Digital Signal Processing

If we transform a lowpass filter


from continuous-time form into
discrete-time form, the warping of
bilinear transformation can be
demonstrated in next slide.

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Digital Signal Processing

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If the critical frequencies (such as the


passband and stopband edge frequencies)
of continuous-time filter are prewaped
according the equation

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Digital Signal Processing

then when the continuous-time filter is


transformed to the discrete-time filter the
discrete-time filter will meet the desired
specifications.

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Digital Signal Processing
Example

Consider the specification on the discrete-


time filter :

Using the bilinear transformation, the


critical frequencies of the discrete-time filter
must be prewarped to the corresponding
continuous-time frequency

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Digital Signal Processing

For convenience we choose Td=1, since


Butterworth filter has a monotonic magnitude
response, so from above equations we obtain :

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Digital Signal Processing

The form of the magnitude-squared


function for the Butterworth filter is :

Solving for N and Ωc, we obtain :

and

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Digital Signal Processing

log [1/0.1782­1/1/0.892­1]
N=
2 log [ tan 0.15/ tan 0.1]

The result are N = 5.30466, and take N=6,


substituting N = 6 and Ωc = 0. 7662.

For this value of Ωc, the passband


specifications are exceeded and the stopband
specifications are met exactly.

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Digital Signal Processing

In the s-plane, the 12 poles are uniformly


distributed in angle on a circle of radius
0.76622.

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Digital Signal Processing

The system function of the continuous-


time filter by selecting the left-plane poles is

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Digital Signal Processing

The magnitude, log magnitude, and group


delay of the frequency response of the
discrete-time filter are shown in next slides
At ω=0.2π, the log magnitude is -0.56dB,
and at ω=0.3π, log magnitude is exactly -15dB.

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From above example, we know Nth-order


Butterworth filter has the following form

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Homework
Digital Signal Processing

We wish to design a lowpass digital filter


to meet the specifications :

δ1 = 0.01, δ2 = 0.001, ωp = 0.4π and ωs = 0.6π.

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Digital Signal Processing

1. Butterworth filter design by impulse


invariance.
2. Butterworth filter design by bilinear
transformation
3. Chebyshev filter design by bilinear
transformation

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