Lecture 9

FIR and IIR Filter design using Matlab

2007/11/16

Prof. C.M Kyung

 FIR and IIR Filter

 GOAL
 Linear-Time-Invariant (LTI) system and Impulse response

 z-Transform

 Characteristics of FIR and IIR filters

 Design procedure of FIR and IIR filters

 FIR and IIR Filter

 LTI System
 Input (x(t) or x[n]) and Output (y(t) or y[n]) is defined first
 For a given system, it can be either LTI or non-LTI depending on how we
define the input and output.

 Linearity
 For arbitrary x1[n] and x2 [n] , the output of the system for x[n]  x1[n]  x2 [n]
is the sum the output for x1[n] and x2 [n] . (superposition)

i.e. output{x1[n]  x2[n]}  output{x1[n]}  output{x2[n]}

 Time-invariance
 Time-shift in input results in time-shift in output by same amount for time-
invariant systems.
i.e. y[n]  output{x[n]}  y[n  m]  output{x[n  m]}
 FIR and IIR Filter

 Impulse Response
 Definition
 Impulse response is the output of the system when impulse signal
 (t ) or  [n] is applied as the input of the system.

 Importance
 Impulse response “fully describes” the system if the system is LTI.
– Why and how?
– Why the impulse response CANNOT fully describe a non-LTI system?

 Fourier transform of the impulse response shows the characteristic of the

system is frequency domain.
 FIR and IIR Filter

 z-Transform
 Definition 
 The z-transform of a sequence x[n ] is defined as X ( z )  {x[n]}   x[n]z
n  
n

 Example
For x[n]  a u[n] ,
n

  
1
X ( z )  {x[n]}   x[n]z
n  
n
  a u[n]z
n  
n n
  (az 1 ) n 
n 0 1  az 1 ,
| z || a |

 Note
 z-transform is reduced to discrete-time Fourier transform (DTFT) if z is
substituted by e j.
 This means that z-transform on the unit-circle on the complex plain is same as
DTFT.
 Laplace Transform CTFT ~ z-Transform  DTFT
 FIR and IIR Filter

 Ideal frequency-selective filter

 A filter whose frequency response is unity over a certain frequency
range and zero for other frequencies.

 Frequency response of an ideal low-pass filter

Ideal Low pass filter

0.8
Frequency Response

0.6

0.4

0.2

-0.2
-3 -2 -1 0 1 2 3
frequency

 However, an ideal low-pass filter is noncausal.

 FIR and IIR Filter
 FIR / IIR filter
 Definition
 If the length of the impulse response is finite, the filter is an FIR (finite impulse
response) filter. Otherwise, the filter is an IIR (infinite impulse response) filter.

 FIR
 Inherently BIBO (bounded-input, bounded-output) stable
 Nonzero pole does not exist in its transfer function
 Easy to implement
 Can be designed to have linear phase property

 IIR
 Sometimes unstable
 Nonzero pole exists in its transfer function
 Lower filter order than a corresponding FIR filter
 Usually have nonlinear phase property
 FIR and IIR Filter

 Filter Design Procedure

 Design continuous-time IIR filter
 Obtain desired H (s ) using Butterworth, Chebyshev methods

 Convert it to discrete-time IIR filter using impulse invariance

j 
 Impulse invariance : h[n]  Td hc (nTd )  H (e )  H c ( j ), |  | 
Td
if H c ( j)  0, |  |  / Td

 Obtain discrete-time FIR filter by windowing the IIR filter

 Windowing : hFIR[n]  hIIR [n]w[n]
 Commonly used windows : rectangular, Bartlett, Hanning, Hamming,
Blackman, Kaiser, …
 However, windowing does not give the optimum solution and other approaches
can be used.
 FIR and IIR Filter

 Frequency response of various filters

 FIR and IIR Filter

 Problem Statements

 Design several types of FIR and IIR filters

 IIR – butterworth, chebyshev type1, chebyshev type 2, …
 FIR – using different windows ( Hamming, Hanning, Bartlett, … )

 Remove the noise in acoustic signal using the filters

 What are the differences between the filters ?

 Understand the effect of sampling frequency on the sampled signal

distortion (aliasing)
 FIR and IIR Filter

 Experiment Requirements

 PC
 Matlab software (with signal processing toolbox)
 FIR and IIR Filter
 References

 Fundamentals of Signal & System using the web and matlab

- Edward W. Kamen, Bonnie S. Heck

 Discrete-Time Signal Processing

- Alan V. Oppenheim, Ronald W. Schafer

 http://www.mathworks.com/access/helpdesk/help/toolbox/signal/fi
lterde.html