DSP Lect 11 Iir Design

Discrete Time Systems
Lecture # 11
IIR Filter Design
Filter Design Specifications
Problem
The filter design process
Analysis
Performance
Constraints
Magnitude Response
Phase response
Cost/complexity
Design
FIR/IIR
Subtype
order
Dr Sajjad Zaidi - DSP
Transfer
function
Implementation
Platform
structure
Performance Constraints
.. in terms of magnitude response:
Performance Constraints
Smallest
Passband
Ripple
Best Filter
Narrowest
Transition
Band
Min Stopband
Attenuation
Greatest
improving one usually worsens others
But: increasing filter order (i.e. cost)

improves all three measures
Passband Ripple
Assume peak passband gain = 1

then minimum passband gain =
Or, ripple
Stopband Ripple
Peak passband gain is A larger than peak stopband gain

Hence, minimum stopband attenuation
Filter Type Choice FIR vs IIR

FIR
IIR
No Feedback
(just zeros)
Feedback
(poles & zeros)
Always stable
May be unstable
Can be linear phase
Difficult to control phase
High order
(20 2000)
Typ < 1/10

Order of FIR (4-20)
Unrelated to CT filtering Derive from analog

prototype
FIR vs. IIR

If you care about computational cost
Use low-complexity IIR
Computation no object LP FIR
If you care about phase response

Use linear-phase FIR
Phase unimportant go with simple IIR)
INFINITE IMPULSE RESPONSE FILTERS
IIR Filter Design

IIR filters are directly related to
analog filters (continuous time)
via a mapping of H(s) (Continuous Time)
to H(z) (Discrete Time)
It preserves many properties
Analog filter design is sophisticated

decades of pre-DSP signal processing
Design IIR filters via analog prototype

hence, need to learn some CT filter design
10
Analog Filter Design

Decades of analysis of transistor-based
filters sophisticated, well understood
Basic choices:
ripples vs. flatness in stop and/or passband
more ripples narrower transition band
11
CT Transfer Functions
Analog systems: s-transform (Laplace)
12
Butterworth Filters
Maximally flat in both pass & stop bands
13
Butterworth Filters
14
Butterworth Filters
How to meet design specifications?
15
Butterworth Filters
16
Butterworth Example
Design a Butterworth filter
with 1 dB cutoff at 1kHz and
a minimum attenuation of 40
dB at 5 kHz
17
Steps in Filter Design

The general filter design problem can be briefly stated as
follows.
Given some ideal frequency response, D(f), find a realizable IIR or
FIR digital filter whose frequency response, H(f), approximates D(f)
The realizable filter is found by optimizing some measure

of the filters performance, e.g
Minimizing the filter order (IIR) or the filter length (FIR),
Minimizing the width of the transition bands,
Reducing the passband error and/or stopband error
Setting up the specifications for the general filter design

problem will define these parameters and show which
trade-offs are possible.
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Butterworth Example
Order N = 4 will satisfy constraints;
What are c and filter coefficients?
from a lookup table, -1dB = 0.845 when c = 1
So c = 1000/0.845 = 1.184 kHz
from a table, get normalized coefficients for

N = 4, scale by 1184
Or, use Matlab:

[b,a] = butter(N,c,s);
19
Analog Filter Types Summary
20
21
22
Steps in Filter Designing

Following are the steps to design a filter
1.
2.
3.
4.
5.
6.
Create the Design Specifications

Obtain Specs Derived from Analog Filtering
Specifying an Error Measure
Select the Filter Type and Order
Design the Filter
Realize the Designed Filter
23
Infinite Input Response Filter

The impulse response h(n) has an infinite number of nonzero samples (infinite length)
As an example, for a general IIR filter, h(n)0 only for No<n<,
where No is a non-negative integer
The frequency response H() is a rational function, i.e., a

ratio of two finite-degree polynomials in ej of the form
where No is an integer constant.
The order of an IIR filter is equal to N, which is the degree

of the denominator ;
Usually the degree of numerator M is not greater than N
24

Remember
The order N also determines the number of previous output
samples that need to be stored and then fed back to compute the
current output sample.
Therefore, IIR systems are also known as feedback systems.
The filter coefficients {bn} and {an} in the Eq correspond to the
unknown parameters of the design.
25

Designing an IIR filter amounts to finding the rational
function
The H( ) should best approximates the design

specifications.
In the frequency domain, this is done by computing the
optimal (relative to some criteria) coefficients {bn} and {an}
The filter order N is usually fixed, but can also be
considered as a free parameter to be optimized.
26

Filter transfer function, denoted by H(z), is the z-transform
of h(n) and is useful for studying the stability of the system.
For LSI filters, stability implies that a bounded input to the
filter will always result in a bounded output.
For IIR filters, H(z) is a rational function in the complex
variable z and is given by
27

Because the impulse response is infinitely long,
convolution can no longer be used to implement the IIR
filters.
Instead, IIR filters are efficiently implemented using
feedback difference equations
The noise characteristics of an IIR filter can be a major
consideration when doing an implementation, especially in
fixed-point arithmetic.
Coefficient quantization degrades the actual filter response
from that designed by high-precision software.
More critical is round-off noise sensitivity which can be
amplified by the feedback loops in the filter.
28
Infinite Input Response Filter - Pros

Compared to FIR filters,
IIR filters can achieve the desired design specifications with a relatively low order (as few
as 4 to 6 poles).
Fewer unknown parameters need to be computed and stored, which might

lead to a lower design and implementation complexity.
However, the phase response of IIR filters is never linear, which leads to the
use of all pass filters to compensate the group delay, and
Thus raises the order of the filter and
The complexity of the design process.
IIR filters are commonly designed by using closed-form design formulas

corresponding to classical filter types
The order-estimating formulas for IIR filters are exact since they are derived from
the mathematical properties of the classical prototypes.
These formulas are very useful to obtain the IIR filter order needed to satisfy the
desired design specifications.
29
Designing the Filter

IIR filters are Commonly designed by
Transforming the digital design specifications in to analog design
specifications
Performing the filter design in the analog domain.
Transforming the analog filter into a digital filter using a suitable
transformation.
Popular Method are

Bilinear Transformation method
1.
2.
3.
4.
Butterworth,
Chebyshev I
Chebyshev II
Elliptic
Digital-only IIR design methods
30
Example IIR Filter Design

Question Design a digital low-pass Butterworth filter with a 3dB cut-off frequency
of 2kHz and minimum attenuation of 30dB at 4.25kHz for a sampling rate of 10kHz.
31
32
Realizing the Designed Filter

Realizing the designed digital filter corresponds to
computing the output of the filter in response to any given
input.
For LSI filters, this is simplified by the fact that the input
and output signals are related through a simple convolution
operation in the time/space domain.
If x(n) is the input, y(n) the corresponding output, and h(n)
the impulse response of the LSI filter, then this relation is
given by
where N1,N2 are the indices of the 1st and last non-zero samples of h(n).
33
Realizing the Designed Filter

In the frequency (Fourier transform) domain, the
convolution relation (11.14) corresponds to multiplication of
the respective Fourier transforms:
where X(), H(), and Y() are the DTFT of x(n), h(n), and y(n), respectively.
But The variable is continuous and, therefore, it cannot be

implemented in practice
An implementable version is obtained by using the Discrete
Fourier Transform (DFT), which is a sampled version of the
DTFT and which consists of samples of the DTFT
evaluated at the points
N is the size of the DFT and
2 k
k 0,K , N 1 corresponds to the number of

sample points within the period 2
N DFT
DFT
34
Simple IIR Lowpass

IIR feedback, zeros and poles,
conditional stability, h[n] less useful
1
1 z
H LP [ z ] K
1 z 1
scale to make
gain = 1 at = 0
pole-zero
diagram
K = (1 - )/2
frequency
response
Dr Sajjad Zaidi
- DSP
FR on
log-log axes
35
Simple IIR Lowpass

1
1 z
H LP [ z ] K
1 z 1
Cutoff freq. c from
36
Simple IIR Highpass
1 z 1
H HP [ z ] K
1
1 z
Pass =
HHP(-1)=1
K=(1+)/2
37
Highpass and Lowpass

Consider lowpass filter:
then
However
(unless H(ejw) is pure real not for IIR)
38
Simple IIR Bandpass
39
Simple Filter Example

Task
Design a second order IIR bandpass filter with c=0.4,
3dB bandwidth of 0.1
40
Simple IIR Bandstop
41
Cascading Filter
Repeating a filter (cascade connection) makes its
characteristics more abrupt:
Repeated roots in z-plane:
42
Cascading Filter
Cascade systems are higher order
e.g. longer (finite) impulse response:
In general, cascade filters will not be optimal for a given

order
43
Linear-phase Filters
|H(ej)| alone can hide phase distortion
differing delays for adjacent frequencies can mangle the signal
Prefer filters with a flat phase response

e.g. () = 0
zero phase filter
A filter with constant delay p = D at all freqs has

() = D
linear phase
Linear phase can shift to zero phase
44
Time reversal filtering

u[n]
v[n]
x[n]
H(z)
Time
reversal
H(z)
w[n
]
Time
reversal
y[n]
Achieves zero-phase result

Not causal! Need whole signal first
45
Some Filter Types

We have seen the basics of filters and a range of
simple examples
Now look at a couple of other classes:
Comb filters - multiple pass/stop bands
Allpass filters - only modify signal phase
46
Comb Filter
Replace all system delays z-1 with longer delays z-L
System that behaves the same at a longer timescale
47
Comb Filter
Parent filter impulse response h[n] becomes comb filter
output
g[n] = {h[0] 0 0 0 0 h[1] 0 0 0 0 h[2]..}
Thus,
Thus frequency response:
High-pass response
pass = /L, 3/L, 5/L...
cut = 0, 2/L, 4/L...
48
Allpass Filter
Allpass filter has |A(ej)|2 = K
i.e. spectral energy is not changed
Phase response is nonzero (else

trivial)
phase correction or special effects
e.g.
Allpass has special form of system
fn
AM(z) has poles where DM() = 0

AM(z) has zeros = 1/
49
Good Luck
50

DSP Lect 11 Iir Design

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Discrete Time Systems

Filter Design Specifications

The filter design process

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

But: increasing filter order (i.e. cost)

Dr Sajjad Zaidi - DSP

Assume peak passband gain = 1

Dr Sajjad Zaidi - DSP

Peak passband gain is A larger than peak stopband gain

Dr Sajjad Zaidi - DSP

Filter Type Choice FIR vs IIR

Can be linear phase

Difficult to control phase

Typ < 1/10

Unrelated to CT filtering Derive from analog

Dr Sajjad Zaidi - DSP

FIR vs. IIR

If you care about phase response

Dr Sajjad Zaidi - DSP

INFINITE IMPULSE RESPONSE FILTERS

Dr Sajjad Zaidi - DSP

IIR Filter Design

It preserves many properties

Analog filter design is sophisticated

Design IIR filters via analog prototype

Dr Sajjad Zaidi - DSP

Analog Filter Design

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Steps in Filter Design

The realizable filter is found by optimizing some measure

Setting up the specifications for the general filter design

Dr Sajjad Zaidi - DSP

from a table, get normalized coefficients for

Or, use Matlab:

Dr Sajjad Zaidi - DSP

Analog Filter Types Summary

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Dr Sajjad Zaidi - DSP

Steps in Filter Designing

Create the Design Specifications

Dr Sajjad Zaidi - DSP

Infinite Input Response Filter

The frequency response H() is a rational function, i.e., a

where No is an integer constant.

The order of an IIR filter is equal to N, which is the degree

Dr Sajjad Zaidi - DSP

Infinite Input Response Filter

Dr Sajjad Zaidi - DSP

Infinite Input Response Filter

The H( ) should best approximates the design

Dr Sajjad Zaidi - DSP

Infinite Input Response Filter

Dr Sajjad Zaidi - DSP

Infinite Input Response Filter

Dr Sajjad Zaidi - DSP