Digital Signal Processing: IIR Filters

DIGITAL
SIGNAL
PROCESSING
Part 3
Digital Signal Processing
A.S.Kayhan
IIR Filters:
Infinite impulse response (IIR) filters have rational transfer
M
functions as
k
H (z)
bk z
ak z k
k 0
N
k 0
Some IIR filter types are: Butterworth, Chebyshev,

Elliptical.
IIR filters have to be stable.
IIR filters may not have linear-phase.
A.S.Kayhan
Filter Specifications (Low-pass):
20 log
10
( )
Design analog filter transform to Digital filter
A.S.Kayhan
Analog Butterworth Filters:

Approximate ideal LPF with magnitude-squared response
H ( )
K ,

0,
c
c
by the following rational function:

H ( )
K 1 a 1 2 a n 1 2 n 2
,
1 b1 2 b n 2 n
bn 0
|H()|2 must be even function of , and denominator

degree must be higher than degree of numerator (lowpass).
A.S.Kayhan
For maximal flatness at the origin, =0, The first 2n-1

derivatives of |H()|2 must be zero. This requires
i 1, 2 , , n 1 .
a i bi ,
H ( )
K 1 b1 2 b n 1 2 n 2
,
1 b1 2 b n 2 n
bn 0
Maximal flatness also at , =, requires

i 1,2 , , n 1 .
a i bi 0,
Then, we have
2
H ( )
K
1 bn
2n
A.S.Kayhan
Let |H(=0)|2 = 1. Then

H ( )
1
1 bn
2n
At the half-power frequency

H ( )
1
1
2
1 bn
2n
o
bn
2n
Then, we have the Butterworth response

H ( )
1

1
o
2n
A.S.Kayhan
Consider the filter specifications:

20 log
10 log
10
/ 10
10
10
H ( ) dB .
2n
2n
dB .
To find n:
Using max and p
10
max
/ 10
10
/ 10
1/ 2n
Using min and s
p
o
2n
10
min
/ 10
s
o
2n
A.S.Kayhan
Dividing these equations
10
10
min
max
/ 10
/ 10

1

1
s
p
2n
Taking logarithm, we get the filter order as:
10 min / 10 1
log
10 max / 10 1
2 log s
p
A.S.Kayhan
We can normalize the frequency so that o = 1. Then

2
H ( )
1
1
2n
To find the filter poles, we let =s/j,

H (s) H ( s)
1
1
2n
1 (s / j)
1 ( 1) n s 2 n
then, set the denominator to zero

1 ( 1) n s 2 n 0
( 1) n s 2 n 1
If n is even, then
s 2n 1 e
sk e
1 2 k
j (
)
2n
j ( 2 k )
, k 0 ,1 ,..., 2 n 1 .
, k 0 ,1 ,..., 2 n 1 .
A.S.Kayhan
If n is odd, then
s 2n 1 e
sk e
k
n
j 2k
, k 0 ,1 ,..., 2 n 1 .
, k 0 ,1 ,..., 2 n 1 .
Note that all the poles lie on a circle with radius 1, because
the frequency is normalized. (If we want to design analog
filter we need to correct this).
Also, choose the poles in the left-half plane to get a stable
filter.
A.S.Kayhan
Example: Design specs. for a Butterworth LP filter:

At f = 2000 Hz, max= 3dB
At f = 3000 Hz, min= 10dB
Then, the filter order n is
n 2.7154 n = 3.
Poles are at
sk e
k
3
, k 0 ,1 ,..., 5 .
1
( s 1 )( s 2 s 1 )
1
.
1
3
1
3
( s 1 )( s ( j
))( s ( j
))
2
2
2
2
H (s)
A.S.Kayhan
Analog Chebyshev Filters:

We can generalize Butterworth response as:
H ( )
1
1
.
n
1 F n ( ) 2
1 ( ) 2
Fn() is a function of . Now, consider:

x cos( ) cos 1 ( x )
y n cos( n ) cos( n cos
( x ))
This is the Chebyshev polynomial of order n. When

x 1 , C n ( x ) cosh( n cosh
( x ))
When x 1 , Cn(x) 0.
A.S.Kayhan
Cn(x) has zeros in -1< x < 1 :

C1(x)
C2(x)
C3(x)
C4(x)
C5(x)
-1
-1
0
x
-1
1 -1
n 1
0
x
-1
-1
0
x
-1
1 -1
(x) 2 x C n (x) C
-1
1 -1
0
x
n 1
0
x
( x ),
C o ( x ) 1, C 1 ( x ) x .
A.S.Kayhan
Use Cn(.) in :
H ( )
1
1 2 C n2 ( )
1 , C n ( ) cosh( n cosh
1 , C n ( ) cos( n cos
2 1 ,
( ))
( ))
is known as the ripple factor.
A.S.Kayhan
Behaviour at =0:
C n2 ( ) 0 , H ( 0 )
C n2 ( ) 1 , H ( 0 )
1,
if n is odd
1
,
(1 2 )
if n is even
Behaviour at =1:
C n2 ( ) 1 ,
for all n
H ( 1)
1
(1 2 )
A.S.Kayhan
The attenuation is
10 log 1 2 C n2 ( )
10
max/ 10
When 2 C n2 ( ) 1 3 . 01 dB .
This defines the half-power frequency (3dB cut-off) hp.
Then,
1
cosh( n cosh
1
1
cosh 1 ( hp )
cosh 1 ( )
n
1
1
hp cosh(
cosh 1 ( )).
n
C n (
hp
hp
))
hp
1 ).
A.S.Kayhan
The specifications for a Chebyshev filter are max, min, s

(p=1). Bandpass is between 0 1 rad/s.
To find n:
min 10 log 1 2 C n2 ( s )
10 min
With
cosh( n cosh
Finally,
cosh
n
/ 10
10
1
1 2 C n2 ( s )
max/ 10
( s ))
10
/ 10
min
cosh
10
10
1
10
min
/ 10
max
/ 10
max
1
1
/ 10
( s )
A.S.Kayhan
To find the filter poles, we let =s/j,

H (s) H ( s)
then
1
1 C n2 ( s / j )
2
1 2 C n2 ( s / j ) 0 C n ( s / j ) j
Let
With
cos( x )
cos( w ) cos( u jv ) s / j
cos 1 ( s / j ) w
e
jx
e
2
jx
, cos( jx )
C n ( s / j ) cos( n cos
ex ex
cosh( x )
2
( s / j )) cos( nw )
C n ( s / j ) cos( nu ) cosh( nv ) j sin( nu ) sinh( nv )

A.S.Kayhan
With
cos( nu ) cosh( nv ) 0
C n (s / j) j
then
sin( nu ) sinh( nv )
cosh(nv) can never be zero, therefore cos(nu) must be

zero. It is possible for
3 5
,
,
,
2n 2n 2n
( 2 k 1 ), k 0 ,1 , , 2 n 1 .
2n
uk
uk
A.S.Kayhan
For these values of u, sin(nu) = 1, then

vk
Remember that
then
1
sinh
n
cos
1
) a.
( s / j ) w u jv

2 k 1 ja
s k j cos( w k ) j cos
2n
The poles are :

s k k j k , k 0 ,1 , , 2 n 1
2k 1
) sinh( a )
2n
2k 1
cos(
) cosh( a )
2n
sin(
Choose left
half poles.
A.S.Kayhan
10
Discrete Time Filter Design(IIR):

There are 3 approches:
1-Sampling (Impulse invariance)
2-Bilinear transformation
3-Optimal procedures
Impulse invariance method:

We can sample the impulse response hc(t) of an analog filter
with desired specifications as (Td is sampling interval):
then
h [ n ] T d h c nT
H ( )
H c(
k ).
Td
Td
If
then
A.S.Kayhan
/ Td
H c ( ) 0 ,
),
.
Td
Analog and digital frequencies have a linear relation:
Td .
H ( ) H c (
Consider the transfer function of a system expressed as

N
H c (s)
k 1
Ak
,
s sk
Then, the impulse response is

N
hc (t )
Ak e skt ,
t 0.
k 1
A.S.Kayhan
11
The impulse response of the discrete time filter is

h n T d h c ( nT
N
d
T d A k e s k nT d u n
k 1
h n
T d ( A k e s k T d ) n u n .
k 1
Transfer (or system) function of the discrete time filter is

N
H (z)
k 1
Poles are
Td Ak
.
1 e sk T d z 1
s s k z e skTd .
If Hc(s) is stable (k < 0), then H(z) is also stable (|z| < 1).
A.S.Kayhan
Example: Transfer function of analog filter is

1
( s 1 )( s 2 s 1 )
1
1
1
1
j
j
1
2
2
12
12
.
s 1
1
3
1
3
s ( j
)
s ( j
)
2
2
2
2
H c (s)
then
Td
1 z 1e Td
1
1
1
Td ( j
)
Td ( j
2
2
12
H (z)
1 z 1 e
Td (
1
3
j
)
2
2
1 z 1e
Td (
1
)
12
.
1
3
2
A.S.Kayhan
12
Bilinear Transformation:
Transformation needed to convert an analog filter to a
discrete time filter must have following properties:
1- j axis of the s-plane must be mapped onto the unit
circle of the z-plane,
2- stable analog filters must be tranformed into stable
discrete time filters (left hand plane of the s-plane must be
mapped into inside the unit circle of the z-plane).
Following BT satisfies these conditions:
1 z 1
s K
1 z 1
1 s / K
.
1 s / K
Let
s j ,
z re
A.S.Kayhan
then
r
(1 / K ) 2 ( / K ) 2
.
(1 / K ) 2 ( / K ) 2
Therefore
0
( LH plane ) r 1 ( inside
( j axis ) r 1 ( the u.c. )
( RH plane ) r 1 ( outside
the u.c. )
the u.c. )
A.S.Kayhan
13
Now, let
then
s j ,
z e
1 e j
e j / 2 ( e j / 2 e
j K
K j / 2
1 e j
e
( e j / 2 e
sin( / 2 )
j Kj
jK tan( / 2 ).
cos( / 2 )
or
A.S.Kayhan
j / 2
j / 2
)
.
)
K tan( / 2 ).
A.S.Kayhan
14
Observations:
1- According to Taylor series expansion of tan(), for small
K / 2 3 / 24 K / 2 .
2- For high frequencies, the relation is nonlinear causing a
distortion called warping effect. Therefore, BT is usually used
for the design of LPF to avoid this.
A.S.Kayhan
Discrete Butterworth Filter Design:

We convert a frequency normalized analog Butterworth filter
to a discrete filter using the BT.
Normalized half power frequecy HP=1 is mapped into the
discrete half power frequency HP. To do that, we set K to
K b ( HP 1) / tan(HP / 2) cot( HP / 2).
A.S.Kayhan
15
Then, we use
K b tan(0.5 ) tan(0.5 ) / tan(0.5HP ).
in
H
( )
1
1
2n
And get the discrete Butterworth Low Pass Filter as:

H
( )
tan( 0 . 5 )
1
tan( 0 . 5 HP )
2n
Using the design specifications, we find the filter order as
min / 10
1
log 10
max / 10
10
2 log tan( 0 . 5 s )
tan( 0 . 5 p )
A.S.Kayhan
The half-power freq. is
HP
2 tan
tan( 0 . 5
10
max
/ 10
1/ 2n
K b cot(0.5 HP ).
Example: Consider the second order analog filter

H
(s)
1
(s2
2 s 1)
Applying the BT, we get (Kb = 1)

( z 1) 2
H 2(z)
0 . 2929
( z 2 0 . 1716 )
A.S.Kayhan
16
Remarks:
1- Given the specs.:
a) find the filter order n
b) obtain the analog filter H(s)(using LHP poles)
c) calculate HP and Kb
d) use biliear transformation to find H(z).
2- H(z) is BIBO stable, because H(s) is stable.
3- Applying the BT to high order filters may be tedious.
Therefore, first express H(s) as product or some of first
and second order functions, then apply the BT.
A.S.Kayhan
Example: Design specs. for a LP digital Butterworth filter:

At f = 0 Hz, 1= 18dB
At f = 2250 Hz, 2= 21dB
At f = 2500 Hz, 3= 27dB
Sampling freq. fsmp= 9000 Hz
(ej0)= 18dB
fp
max = 2 1=3dB
fs
p 2
, s 2
min = 3 1=9dB
f smp
f smp
Filter order n 5.52,n = 6, Kb = 1 and a gain G for = 18dB
H 6(z)
(z2
0 . 0037 ( z 1 ) 6
0 . 59 )( z 2 0 . 17 )( z 2 0 . 0 . 02 )
A.S.Kayhan
17
Discrete Chebyshev Filter Design:

We find the constant K by transforming (normalized
passband freq.) P=1 into the discrete passband frequency
P. We get
K c 1 / tan(0.5P ) tan(0.5 ) / tan(0.5 P )
And get the discrete Chebyshev Low Pass Filter as:

H ( )
1 C
2
n
1
(tan( 0 . 5 ) / tan( 0 . 5
))
Using C1(1)=1, we let p ,
max 10 log 1
10 min
Also using s ,
We find the filter order
10
max
min
10
1
cosh 1 [tan( 0 . 5 s ) / tan( 0 . 5 p )]
cosh
/ 10
A.S.Kayhan
min
/ 10
max
/ 10
A.S.Kayhan
18
Example: Design specs. for a LP digital Chebyshev filter:

At f = 0 Hz, 1= 0dB
At f = 2250 Hz, max= 3dB
At f = 2500 Hz, min= 10dB
f smp
Filter order n = 3,
H 3(z)
fs
f smp
Kc = 1
(z2
0 . 09 ( z 1 ) 3
0 . 15 z 0 . 72 )( z 0 . 54 )
A.S.Kayhan

At = 2/5 rad, max= 1dB
At = /2 rad, min= 9dB
Filter order n = 3
H 3(z)
(z2
0 . 736 ( z 1 ) 3
0 . 505 z 0 . 619 )( z 0 . 472 )
Example: Design specs. for a LP digital Butterworth filter:

At = /2 rad, max= 3dB
= 0 rad, = 0dB
At = 5/9 rad, min= 10dB
Filter order n = 7,
H
(z)
z(z2
Kb = 1
0 . 01656 ( z 1 ) 7
0 . 636 )( z 2 0 . 232 )( z 2 0 . 052 )
A.S.Kayhan
19
Frequency Transformations:
The objective is to obtain transfer functions of other types of
dicrete filters from already available prototype Low Pass
Filters using tranformation functions g(z) as
H (z) H
LP
( g ( z ))

At f = 0 Hz, = 0dB
At f = 2250 Hz, 2= 21dB
At f = 2500 Hz, 3= 27dB
LPF
(z)
z3
A.S.Kayhan
0 . 09 ( z 1 ) 3
0 . 691 z 2 0 . 802 z 0 . 39
A.S.Kayhan
20
Now, we want a HPF with cutt-off at 3.6kHz, we use

following transformation
( z 1 0 . 5095 )
1
z
HPF
(z)
z3
(1 0 . 5095 z 1 )
0 . 0066 ( z 3 3 z 2 3 z 1 )
2 . 361 z 2 2 . 102 z 0 . 6884
A.S.Kayhan
Design of FIR Filters by Windowing:

Ideal frequency response and corresponding impulse
response functions are H ( e j ), h n .
d
Generally hd[n] is infinitely long. To obtain a causal practical

FIR filter, we can truncate it as
,0 n M
h n
h n d
, otherwise.
0
In a general form we can write it as

h n h d n w n
Where w[n] is window function.

A.S.Kayhan
21
In frequency domain, we have

H (e
1
)
2
(e
)W ( e
)d .
A.S.Kayhan
Some commonly used window functions are Rectangular,

Bartlett, Hamming, Hanning, Blackman, Kaiser.
Some important factors in choosing windows are:
Main lobe width and Peak side lobe level. For rectangular:
A.S.Kayhan
22
Main lobe width and transition region; Peak side lobe level
and oscillations of filter are related.
A.S.Kayhan
A.S.Kayhan
23
A.S.Kayhan
A.S.Kayhan
24
Discrete Time Fourier Series (DTFS):

~
Consider a periodic sequence x n with period N:

~
x n x n N .
Which can be represented by a Fourier series as:

N 1
1
x n
N
X [ k ]e
2
nk
N
k0
The DTFS coefficients are obtained as:

N 1 ~
x n e
X [k ]
2
nk
N
n0
Let
A.S.Kayhan
2
N
The DTFS analysis and synthesis equations are:

N 1 ~
x n W
X [k ]
nk
N
n0
x n
1
N
N 1
X [k ] W
nk
N
k 0
Example: Consider the periodic impulse train:

~
x n
[ n rN ]
The DTFS
~
X [k ]
N 1
n W
nk
N
for all k .
n0
A.S.Kayhan
25
Properties:
1- Linearity: It is linear.
2- Shift of a sequence: If
x n X [ k ]
x n m W
then
Similarly,
nl
N
km
N
X [k ]
x n X [ k l ]
3- Periodic convolution: Consider two periodic

sequences with period N
~
x 1 n X 1 [ k ]
x 2 n X 2 [ k ]
Then,
~
X 3[ k ] X 1[ k ] X 2 [k ]
x 3 n
N 1 ~
x 1 [ n m ] x 2 [ m ].
m0
A.S.Kayhan
Example: Consider periodic convolution of two sequences:
A.S.Kayhan
26
Sampling the Fourier Transform:

Consider signal x[n] with DTFT X() and assume by
sampling X() , we get
~
X [ k ] X ( ) |
2

k
N
X
k
N
X [ k ] could be the sequence of DTFS coefficients
of a periodic signal which may be obtained as

1
x n
N
~
N 1
X [k ] W
nk
N
k 0
A.S.Kayhan
Substituting,
X
x m
e j m ,
X k X ( ) | 2
then,
x n x n *
n rN
x n
rN .
A.S.Kayhan
27
If N is too small, then aliasing occurs in the time domain.
If there is no aliasing, we can recover x[n].
A.S.Kayhan
Discrete Fourier Tranform (DFT):

DFT is obtained by taking samples of the DTFT.
Remember DTFT is defined as:
X (e
x [ n ]e
j n
We take samples of X(ej) at uniform intervals as:

X [k ] X (e
We define:
W
2
k
N
j
, k 0 ,1 , N 1 .
2
N
A.S.Kayhan
28
Then the DFT is defined as (analysis equation):

N
X [k ]
x[ n ] W
kn
N
n0
for k=0,1,...,N-1.
The inverse DFT is defined as (synthesis equation):
x[ n ]
1
N
X [k ] W
kn
N
n0
for n=0,1,...,N-1.
A.S.Kayhan
A.S.Kayhan
Example:
N=5
29
N=10
Properties:
1- Linearity: DFT is a linear operation.
2- Circular shift:
x n m
N , 0
n N 1 e
A.S.Kayhan
2
km
N
X [ k ].
A.S.Kayhan
30
3- Circular convolution:
x 3 n x 1 n
x 2 n
N 1
x 1 [ n m
] x2[m ]
m 0
Example:
A.S.Kayhan
A.S.Kayhan
Example:
N=L
31
N=2L
A.S.Kayhan
4- Multiplication(Modulation):
x 3 n x 1 n x 2 n X
k .
Linear Convolution Using the DFT:

Since there are efficient algorithms to take DFT, like FFT,
we can use it instead of direct convolution as:
1. Compute N point DFTs, X 1 k and X 2 k .
2. Multiply them to get X 3 k X 1 k X 2 k .
3. Compute the inverse DFT, to get:
x 3 n x 1 n
x 2 n
A.S.Kayhan
32
Linear Convolution of Finite Length Signals:

Consider two sequences x1[n] of length L and x2[n] of
length P, and x3[n] = x1[n]* x2[n].
Observe that x3[n] will be of length (L+P-1).
Circular Convolution as Linear Convolution with

Aliasing :
Consider again x1[n] of length L and x2[n] of length P,
and x3[n] = x1[n]* x2[n].
X 3 (e
) X 1 (e
) X 2 (e
A.S.Kayhan
Taking N samples
X
k X 2 k .
Now, taking the inverse DFT, we have

x 3 n rN , 0 n N 1
x 3 p n r

0,
otherwise.
x 3 p n x 1 n
x 2 n
This circular convolution is identical to the linear

convolution corresponding to X1(ej) X2(ej),
if N (L+P-1).
A.S.Kayhan
33
Example:
A.S.Kayhan
A.S.Kayhan
Example:
34
A.S.Kayhan
LTI Systems Using the DFT :

Consider x[n] of length L and h[n] of length P, and y[n] =
x[n]* h[n] will be of length (L+P-1).
To obtain the result of linear convolution using the DFT,
we must use N ( (L+P-1))-point DFTs. For that x[n]
and h[n] must be augmented with zeros (zero-padding).
Overlap-Add Method :
Consider h[n] of length P, and length of x[n] is much
grater than P.
We can write x[n] as combination of length L segments:
x n rL , 0 n L 1
x r n
,
0
,
otherwise
x n
x r n rL
A.S.Kayhan
35
Since the system is LTI then

y n
y r n rL ,
y r n x r n * h n
Each output segment yr[n] can be obtained using (L+P-1)

point DFT.
Example:
A.S.Kayhan
A.S.Kayhan
36
Overlap-Save Method :
Consider again h[n] of length P, and length of x[n] is much
grater than P.
In this method L-point circular convolution (or DFT) is
used. Since the first (P-1) points will be incorrect,
input segments must overlap.
We can write each L sample segment of x[n] as :
x r n x n r ( L P 1 ) ( P 1 ) ,
y n
0 n L 1
y r n r ( L P 1 ) ( P 1 ) ,
y r n y rp n , P 1 n L 1
A.S.Kayhan
A.S.Kayhan
Example:
37
Efficient Computation of the DFT :

Remember the definition of the DFT and inverse DFT
N 1
x[ n ]
N 1
X [k ] W
kn
N
X [k ]
n0
x[n ] W
kn
N
n0
N2 comlex multiplications and N(N-1) additions or 4N2 real

mult. and 4N(N-2) real additions necessary:
Re{ x [ n ] } Re{ W Nkn }
kn
Im{ x [ n ] } Im{ W N }
Re{ x [ n ] } Im{ W Nkn }
j Im{ x [ n ] } Re{ W kn }
N
X [k ]
n0
A.S.Kayhan
To improve the efficiency, we can use some properties as:

1) W
2) W
k(N n)
N
k(N n)
N
W
W
kn
N
kn
N
(W
W
kn
N
)*
(k N )n
N
Example:
Re{ x [ n ] } Re{ W
kn
N
} Re{ x [ N n ] } Re{ W
Re{ x [ n ] } Re{ x [ N n ] } Re{ W
kn
N
k(N n)
N
}.
A.S.Kayhan
38
Fast Fourier Transform(FFT):

Fast Fourier Transform algorithms are used to implement
DFT efficiently to reduce the number of multiplication
and addition operations.
Two main algorithms are:
Decimation in time and Decimation in frequency. Both
these algorithms require that N=2m.
The number of operations using either of these will
require (N/2)log2N complex multiplications and Nlog2N
additions.
Direct implementation of DFT requires N2 complex
multiplications and additions. For N=1024= 210
N2=1048576
but
Nlog2N=10240.
A.S.Kayhan
Decimation in time FFT Algorithm:

Assume N=2m
N 1
X [k ]
kn
N
x[n ] W
, k 0 ,1 , , N 1
n0
Which can be written as

X [k ]
x[ n ] W
kn
N
n even
x[n ] W
kn
N
n odd
With n = 2r (even) and n = 2r+1 (odd)

N / 2 1
X [k ]
N / 2 1
x [ 2 r ] (W
2
N
) rk W
k
N
r0
x [ 2 r ] (W
rk
N / 2
r0
G k
2
N
) rk
r0
N / 2 1
x [ 2 r 1 ] (W
N / 2 1
) W
k
N
x[ 2 r 1] W
rk
N / 2
r0
k
N
k .
A.S.Kayhan
39
X [ k ] G k
k
N
k .
Requires N+2(N/2)2 multiplications

A.S.Kayhan
Since G[k] and H[k] requires 2(m-1)-point DFTs, each one can
be similarly decomposed until we reach 2-point DFTs.
A.S.Kayhan
40
A.S.Kayhan
A.S.Kayhan
Butterfly:
(in place comp.)
Bit reversed
order
41
Decimation in frequency FFT Algorithm:

Again assume N=2m
N 1
X [k ]
x[n ] W
kn
N
, k 0 ,1 , , N 1
n0
then
N 1
X [2 r ]
x[ n ] W
2 rn
N
r 0 ,1 , , ( N / 2 1 )
n0
N / 2 1
X [2 r ]
N 1
x[ n ] W
2 rn
N
n0
2 rn
N
N / 2 1
x[ n ] W
2 rn
N
n0
N / 2 1
X [2r ]
x[ n ] W
nN /2
N / 2 1
X [2 r ]
x[ n N / 2 ] W
2r(n N /2)
N
n0
N / 2 1
( x [ n ] x [ n N / 2 ]) W
rn
N /2
n0
rn
N /2
n0
Similarly for
g [n ] W
A.S.Kayhan
0 r ( N / 2 1)
N / 2 1
X [ 2 r 1]
( x [ n ] x [ n N / 2 ]) W
n
N
rn
N /2
n0
N / 2 1
h [ n ] W Nn W
rn
N /2
n0
A.S.Kayhan
42
Procedure is repeated until 2-point DFTs
A.S.Kayhan
Computation of Inverse DFT:

x[ n ]
1
N
X [ k ]W
kn
N
, n 0 ,1 , , N 1
n0
X [k ]
x[ n ] W
kn
N
, k 0 ,1 , , N 1
n0
1 N
1
x [n]
X * [ k ]W Nkn
DFT
N n0
N
*
1
x[ n ]
DFT X * [ k ]
N
*
[k ]
A.S.Kayhan
43
End of Part 3
A.S.Kayhan
44

Digital Signal Processing: IIR Filters

Diunggah oleh

Informasi Dokumen

Judul Asli

Hak Cipta

Format Tersedia

Bagikan dokumen Ini

Bagikan atau Tanam Dokumen

Opsi Berbagi

Apakah menurut Anda dokumen ini bermanfaat?

Apakah konten ini tidak pantas?

Hak Cipta:

Format Tersedia

Digital Signal Processing: IIR Filters

Diunggah oleh

Hak Cipta:

Format Tersedia

DIGITAL

Some IIR filter types are: Butterworth, Chebyshev,

Digital Signal Processing

Filter Specifications (Low-pass):

Design analog filter transform to Digital filter

Digital Signal Processing

Analog Butterworth Filters:

by the following rational function:

|H()|2 must be even function of , and denominator

For maximal flatness at the origin, =0, The first 2n-1

Maximal flatness also at , =, requires

Digital Signal Processing

Let |H(=0)|2 = 1. Then

At the half-power frequency

Then, we have the Butterworth response

Digital Signal Processing

Consider the filter specifications:

Using min and s

Digital Signal Processing

Dividing these equations

Taking logarithm, we get the filter order as:

Digital Signal Processing

We can normalize the frequency so that o = 1. Then

To find the filter poles, we let =s/j,

then, set the denominator to zero

Digital Signal Processing

Example: Design specs. for a Butterworth LP filter:

Digital Signal Processing

Analog Chebyshev Filters:

Fn() is a function of . Now, consider:

This is the Chebyshev polynomial of order n. When

Cn(x) has zeros in -1< x < 1 :

Digital Signal Processing

is known as the ripple factor.

Digital Signal Processing

Digital Signal Processing

Digital Signal Processing

The specifications for a Chebyshev filter are max, min, s

Digital Signal Processing

To find the filter poles, we let =s/j,

C n ( s / j ) cos( nu ) cosh( nv ) j sin( nu ) sinh( nv )

cosh(nv) can never be zero, therefore cos(nu) must be

Digital Signal Processing

For these values of u, sin(nu) = 1, then

The poles are :

Digital Signal Processing

Discrete Time Filter Design(IIR):

Impulse invariance method:

Digital Signal Processing

Consider the transfer function of a system expressed as

Then, the impulse response is

Digital Signal Processing

The impulse response of the discrete time filter is

Transfer (or system) function of the discrete time filter is

Example: Transfer function of analog filter is

Digital Signal Processing

Digital Signal Processing

( j axis ) r 1 ( the u.c. )

Digital Signal Processing

Digital Signal Processing

Digital Signal Processing

Discrete Butterworth Filter Design:

Digital Signal Processing