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Recursive Filters:
analog digital
s-plane z-plane
H a ( s) H d ( z)
y (nT ) y (nT T )
y (nT )
T
approximation of “s”
take the z-Transform of both sides:
Y ( z) z 1Y ( z) 1 z 1
Z y (nT ) Y ( z)
T T
1 z 1
s
T
Example:
2
take the analog filter with transfer function H a ( s) and discretize it
s 1
with a sampling frequency Fs 5Hz .
By Euler’s approximation
2 0.333 z
Hd ( z )
1 z 1 z 0.833
1
0.2
analog digital
s-plane z-plane
Problem with Euler Approximation:
it maps the whole stable region of the s-plane into a subset of the stable region in the z-plane
s-plane z-plane
s
since
1
s
1 1 1 1
z T
2 1 sT 2 2 s 1 1 1
T T
T
z
1
2
1
2
if Re[s]<0.
Bilinear Transformation. It is based on the relationship
nT T 2
area ABCD
C
Take the z-Transform of both sides:
B
1 T
Y ( z) z Y ( z) Y1 ( z) z 1Y1 ( z)
2
A D
nT-T nT
which yields the bilinear transformation:
2
2 1 z 1 s
s z T
T 1 z 1 2
s
T
Main Property of the Bilinear Transformation: it preserves the stability regions.
s-plane z-plane
since: s
2
s
T
| z| 1 2 2
2
s T T
T
Mapping of Frequency with the Bilinear Transformation.
Magnitude: j
2
j
T
| e j | 1
2
2 2
j T T
T
Phase:
2
j
phase T
( ) 2 j
2
j
T 2
1 T
T
where tg
2 j
T
2tg
1
2
2
tg
T 2
See the meaning of this:
T
2tg 1
2
2
tg
T 2
c c
0
c 0 c
Example: we want to design a digital low pass filter with a bandwith Fo 8kHz
and a sampling frequency Fs 24 kHz . Use the Bilinear Transformation.
Solution:
• Step 1: specs in the digital freq. domain c (2 )(8 / 24) 2 / 3 rad
• Step 2: specs of the analog filter to be digitized:
c
c 2 Fs tg (2)(24 103 ) tg ( / 3) 8313
. 103 rad / sec
2
or equivalently Fc 13.23kHz
• Step 3: design an analog low pass filter (more later) with a bandwith Fc 13.23kHz ;
• Step 4: apply Bilinear Transformation to obtain desired digital filter.
Design of Analog Filters
| H ()|2
Specifications: 1
1 2
1
2
2 2
p c s
Butterworth:
1 1
| H ()|
2
2N 2N
1 1
2
c p
1
2 s
2N 2 2 N
2
s
1 2 log
p p
Poles of Butterworth Filter:
1 1
| H ()| H ( s) H ( s) s j
2
H ( s) H ( s)
2N
s 2 N
1 1 2
c
s c
j
which yields the poles as solutions
k 2
s2
1 j ( 2 k 1)
N
j j
2 1 s jce 2N
c e 2 e 2N
, k 0,...,2 N 1
c
N=2
and choose the N poles in the stable region.
+ +
poles
+ +
s-plane
Example: design a low pass filter, Butterworth, with 3dB bandwith of 500Hz and 40dB
attenuation at 1000Hz.
Solution: c 1000 ; s 2000 ,
1 1
2 N 0.01
2
s
2N
1 2
1
c
| H ( )|dB
log(104 1)
N 6.64 N 7
2 log(2)
poles at
sk 1000 e j ( / 2( 2 k 1) /14) , k 0,...,6
Chebychev Filters.
Based on Chebychev Polynomials: TN ( x ) cos( Nt ) x cos( t )
T0 ( x ) cos(0) 1, T3 ( x )
T1 ( x ) cos(t ) x cos( t ) x ,
T2 ( x ) cos(2t ) 2 cos2 (t ) 1 x cos( t ) 2 x 2 1,
TN 1 ( x ) 2 xTN ( x ) TN 1 ( x )
x
Property of Chenychev Polynomials:
within the interval 1 x 1 Chebychev polynomials have least maximum deviation
from 0 compared to polynomials of the same degree and same highest order coefficient
P( x) 4 x 3 ax 2 bx c
T3 ( x) 4 x 3 3x
x
Why? Suppose there exists P( x) 4 x 3 ax 2 bx c with smaller deviation then T3 ( x )
P T3 1 P T3 1
T3 ( x )
B D
P( x ) T3 ( x )
A
C
P( x )
D
A C
P T3 1 P T3 1 root
B root root
So: you cannot find a P(x) which does better (in terms of deviation from 0) then the
Chebychev polynomial.
Chebychev Filter:
1
| H ()|
2
1 TN
2 2
p
1
Since TN (1) 1 (easy to show from the definition), then | H ( p )|2
1 2
| H ()|2 1
1 2
p
Design of Chebychev Filters:
Formulas are tedious to derive. Just give the results:
2 1
1
1 1 N
r1,2
2
2
k (2 k 1) , k 0,..., N 1
2 2N
s-plane
Example: design a Chebychev low pass filter with the following specs:
• passband Fp 500 Hz, with a 1dB ripple,
• stopband Fs 1000 Hz, with attenuation of at least 40dB.
N
log 1 1 (1 /
2
2
2
2 2
2
log( / ) ( / ) 1
s p s p
2
c c c c
same value at c
c 2 c
H H H ( jc )
jc j
Low Pass to Band Pass:
c c
u l l u
s2 l u
The tranformation s c
s(u l )
maps
l 2 l u
s j l c j c
j l ( u l )
u 2 l u
s j u c j c
j u ( u l )
Low Pass to Band Stop
c c
u l l u
s( u l )
s c 2
s cl
How to make the transformation:
Consider the transfer function
K ( s z1 )( s z2 )...( s zm )
H ( s) , mn
( s p1 )( s p2 )...( s pn )
K F ( s) z1 ... F ( s) zm
then with s F ( s) HT ( s) H F ( s)
we obtain
F ( s) p ... F ( s) p
1 n
F ( s) zk , k 1,..., m
with zeros and poles solutions of F ( s) p j , j 1,..., n
also n-m extra zeros at s where F ( s)
IIR filter design using Matlab
In Matlab there are three functions for each class of filters (Butterworth, Chebytchev1, Chebytchev2):
BUTTAP CHEB1AP CHEB2AP Poles and Zeros of Analog Prototype Filter
BUTTER CHEBY1 CHEBY2 Numerator and Denominator from N and c
BUTTORD CHEBY1ORD CHEBY2ORD N and c from specifications
Example. We want to design an IIR Digital Filter with the following specifications:
Pass Band 0 to 4kHz, with 1dB ripple;
Stop Band > 8kHz with at least 40 dB attenuation
Sampling frequency 40kHz
Type of Filter: Butterworth.
Using Matlab:
>> [N, fc]=butterord(fp, fs, Rp, Rs); % fp, fs=passband and stopband freq relative to Fs/2
>> [B, A]=butter(N, fc); % B, A vectors of numerator and denominator coefficients.
In our case:
[N, fc]=butterord(4/20, 8/20, 1, 40), would yield N=7, fc=0.2291;
[B, A]=butter(7, 0.2291), would yield the transfer function B(z)/A(z).
Let’s verify these numbers:
Step 1: specifications in the digital frequency domain:
4
p 2 rad band pass freq
40 5
8 2
s 2 rad stop band freq
40 5
Step 2: specifications for analog filter from the transformation 2 Fs tan
2
p 2(40 10 ) tan 26 103 rad / sec
3
10
s 2(40 103 ) tan 58 103 rad / sec
5
2 1
Step 3: choose (say) Butterworh Filter H ( ) 2N
1 2
p
1
10 log 10 2N
40
s
1 0.5
2
p
s 58
with yields N=7
p 26
Step 5: finally the cutoff frequency, from the equation
2 1 1
H ( c )
c
14
2
2
1 0.5
26 103