By

Imtiyaz Mohiuddin
10361A0434

In charge
Dr.M.Narayana
Electronics and Communications Engineering
SECTION OUTLINE:
 Introduction to Digital filter design
 Introduction to FIR Filter
 Design of FIR Filter using WINDOW Techniques
 Introduction to IIR Filter
 Design of IIR Filter using Approximation Method
 MATLAB Code of Designed Filters
Introduction:
 A digital filter is a system that performs mathematical
operations on a sampled, discrete-time signal to reduce or
enhance certain aspects of that signal.
 In digital signal processing, there are two important types
of systems:
 Digital filters: perform signal filtering in the time
domain
 Spectrum analyzers: provide signal representation in the
frequency domain
Digital Filter:

Sampling
frequency
fS
Analog xn yn
x(t) anti- A D Analog y(t)
aliasing D Digital Filter A smoothing
C C filter
filter
Preliminaries:
 The design of a digital filter is carried out in three steps:
 Specifications: they are determined by the applications
 Approximations: once the specification are defined, we use various
concepts and mathematics that we studied so far to come up with a
filter description that approximates the given set of specifications.
(in detail)
 Implementation: The product of the above step is a filter
description in the form of either a difference equation, or a system
function H(z), or an impulse response h(n). From this description
we implement the filter in hardware or through software on a
computer.
Classification:
 Digital filters are classified into one of two basic forms,
according to how they respond to a unit impulse:
 Finite impulse response
 Infinite impulse response
Finite Impulse Response:
 In signal processing, a finite impulse response
(FIR) filter is a filter whose impulse response is
of finite duration, because it settles to zero in finite
time.
 FIR digital filters use only current and past input
samples, and none of the filter's previous output
samples, to obtain a current output sample value
The transfer function is given by
N 1
H ( z )   h(n).z n
n 0

 The length of Impulse Response is N

 All poles are at Z=0. .
 Zeros can be placed anywhere on the z-plane
Filter Design by Windowing

 Simplest way of designing FIR filters

    h ne
 
Hd e j

n  
d
 j n
hd n 
1

2  
Hd 
e j
e jn
d

 Start with ideal frequency response

h n 0  n  M
hn   d
 0 else
 Choose ideal frequency response as desired response
 Most ideal impulse responses are of infinite length
Commonly used windows
Rectangular: 1
2n
N 1
N n 
 2n  2
Bartlett: 1  cos 
 N 
Hamming:  2n 
0.54  0.46 cos 
 N 
2n   4n 
Blackman: 0.42  0.5 cos   0.08 cos 
 N   N 
 2
J 0  1   
2 n
Kaiser:    J 0 ( )

  N  1  
 Kaiser window
 Kaiser window

β Transition Min. stop attn

width (Hz) dB
2.12 1.5/N 30

4.54 2.9/N 50

6.76 4.3/N 70

8.96 5.7/N 90
 Rectangular Window
 Narrowest main lob
– 4/(M+1)
– Sharpest transitions at
discontinuities in frequency
1 0  n  M
wn  
0 else

 Large side lobs

– Large oscillation around
discontinuities
– -13 dB
Simplest window possible
 Hamming Window
 Medium main lob
– 8/M
  2n 
0.54  0.46 cos  0nM
wn    M 

 0 else
 Good side lobs
– -41 dB
– Simpler than
Blackman
 Kaiser Window
 Parameterized equation forming a set
of windows

 Parameter to change main-lob width

and side-lob area trade-off

  2
 I0  1   n  M / 2  

  M / 2  
wn    
 0nM
 I0(.) represents zeroth-order
 I0 Bessel
modified  function of 1st kind

 0 else
MATLAB CODE:
 %Design of LPF&HPF using rectangular,hamming and kaiser windows
 clc;clear all;close all;
 rp=input('enter attenuation in pass band');
 rs=input('enter attenuation in stop band');
 fp=input('enter pass band frequency');
 fs=input('enter stop band frequency');
 Fs=input('enter sampling frequency');
 wp=2*pi*fp/Fs;
 ws=2*pi*fs/Fs;
 %formula for FIR filter
 num=-20*log10(sqrt(rp*rs))-13;
 den=14.6*(fs-fp)/Fs;
 n=ceil(num/den);
 disp('order of filter is \n');
 disp(n);
 disp('press any key to continue');
 pause;
 n1=n+1;
 %For even order
 if(rem(n,2)~=0)
 n1=n;
 end
 %LPF
 %LPF
 s1=input('enter the value for window 0-rectangularLPF 1-kaiserLPF 2-hammingLPF 3-
rectangularHPF 4-kaiserHPF 5-hammingHPF');
 switch(s1);
 case 0
 y=rectwin(n1);
 [b,a]=fir1(n,wp,'low',y);
 freqz(b,a,512);
 case 1
 y=kaiser(n1);
 [b,a]=fir1(n,wp,'low',y);
 freqz(b,a,512);
 case 2
 y=hamming(n1);
 [b,a]=fir1(n,wp,'low',y);
 freqz(b,a,512);
 case 3
 y=rectwin(n1);
 [b,a]=fir1(n,wp,'high',y);
 freqz(b,a,512);
 case 4
 y=kaiser(n1);
 [b,a]=fir1(n,wp,'high',y);
 freqz(b,a,512);
 case 5
 y=hamming(n1);
 [b,a]=fir1(n,wp,'high',y);
 freqz(b,a,512);
 end
Pros & Cons:
FIR filters have the following
advantages: FIR filters have the following
disadvantages:
 Exactly linear phase is possible
• Higher filter order than IIR
 Always stable, even when quantized filters
 Design methods are generally linear • Corresponding greater
delays
 Efficient hardware realizations

 Startup transients have finite

duration
Infinite Impulse Response Filter:
 IIR systems have an impulse response function that is
non-zero over an infinite length of time. This is in
contrast to finite impulse response (FIR) filters, which
have fixed-duration impulse responses
 IIR filters may be implemented as
either analog or digital filters
Cont..
 While designing a digital IIR filter , an analog filter
(e.g. Chebyshev filter, Butterworth filter) is first
designed and then is converted to a digital filter by
applying discretization techniques such as Bilinear
transform or Impulse invariance.
Discretization techniques
Chebyshev Filter:
 Chebyshev filters are analog or digital filters having a
steeper roll-off and more passband ripple (type I) or
stopband ripple (type II)
 Chebyshev filters have the property that they
minimize the error between the idealized and the
actual filter characteristic over the range of the filter,
but with ripples in the passband
Cont..
 Type-1 Chebyshev Filter

 Type-2 Chebyshev Filter:

Butterworth filter
 The Butterworth filter is a type of signal processing
filter designed to have as flat a frequency response as
possible in the pass band. It is also referred to as a
maximally flat magnitude filter
 MATLAB Prototype Filter Design Commands

 [B,A] = BUTTER(N,Wn)
 [B,A] = CHEBY1(N,R,Wn)
 [B,A] = CHEBY2(N,R,Wn)
 [B,A] = ELLIP(N,Rp,Rs,Wn)
– N = filter order
– R = pass band ripple (cheby1) or stop-band ripple (cheby2) in
dB. (Rp and Rs respectively for the elliptic filter)
– Wn = cut-off frequency (radians/sec for analog filters or
normalized digital frequencies for digital filters)
– [B,A] = filter coefficients, s-domain (analog filter) or z-domain
(digital filter)
Design Example
 Filter Specifications:
 Butterworth response
 Pass-band edges = 400 Hz and 600 Hz
 Stop-band edges = 300 Hz and 700 Hz
 Pass-band ripple = 1 dB
 Stop-band attenuation = -20 dB
 Sampling Frequency = 2000 Hz
Design Example Results
Magnitude Response
1.4

1.2

1
Magnitude

0.8
Band Edges
0.6
(-1dB and -20 dB)
0.4

0.2

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9
Frequency (kHz)
Design Example
Chebyshev II High-Pass Filter
 Filter specifications:
 Chebyshev II response (stop-band ripple)
 Pass-band edge = 1000 Hz
 Stop-band edge = 900 Hz
 Pass-band ripple = 1 dB
 Stop-band attenuation = -40 dB
 Sampling frequency = 8 kHz
MATLAB Code for Design Example
>> fs=8000;
>> Wp=[2*1000/fs]; % Pass-band edge normalized digital frequency
>> Ws=[2*900/fs]; % Stop-band edge normalized digital frequency
>> [N,Wn]=cheb2ord(Wp,Ws,1,40); % The “order” command
>> [B,A]=cheby2(N,40,Wn,'high');
% cheby2 is the “filter” command. In this command
% the syntax requires the stop-band attenuation
% as the second parameter
>> fvtool(B,A)
Design Example Results

Magnitude Response (dB)

20

-20
Magnitude (dB)

-40

-60

-80

-100
0 0.5 1 1.5 2 2.5 3 3.5
Frequency (kHz)
MATLAB CODE:
 %Design of IIR filters
 fp1=input('enter pass band frequency');
 fs1=input('enter stop band frequency');
 Fs1=input('enter sampling frequency');
 wp1=fp1/Fs1;
 ws1=fs1/Fs1;
 [n1,wn1]=buttord(wp1,ws1,2,60);
 [x,y]=butter(n1,wn1,'low');
 figure;
 freqz(x,y,512);
 [n1,wn1]=buttord(wp1,ws1,2,60);
 [x,y]=butter(n1,wn1,'high');
 figure;
 freqz(x,y,512);
 [n1,wn1]=cheb1ord(wp1,ws1,2,60);
 [x,y]=cheby1(n1,3,wn1,'low');
 figure;
 freqz(x,y,512);
 [n1,wn1]=cheb1ord(wp1,ws1,2,60);
 [x,y]=cheby1(n1,3,wn1,'high');
 figure;
 freqz(x,y,512);
Summary of IIR Filter:

 IIR filters can be design by pole-zero location

– Digital oscillators: poles on the unit circle
– Notch filters: zeros on the unit circle with nearby poles to
control notch width
 Classic analog filters can be designed using the
bilinear transformation
 IIR filters have the advantage of smaller filter order for
a given frequency response.
 IIR filters have the disadvantages of possible instability
due to coefficient quantization effects and non-linear
phase response.
References:
 “Design of IIR Filter” by K.S Chandra, M.Tech, IIT-Bombay, Jan-2006
 “Digital Filter Design” by Prof. A.G. Constantinides, University of
Auckland, 2006
 “FIR Filter Design”, Gao Xinbo,School of E.E., Xidian Univ.
xbgao@ieee.org
 “FIR Filter by Windowing”- The lab Book Pages.com
 “Digital Signal Processing”, Prof.Ramesh Babu, Pondicherry Govt.
College, TataMcgraw-Hill publication.
 Wikipedia.org