Lecture 11 - FIR Filter Design

ELEN 5346/4304 DSP and Filter Design Fall 2008
1
Lecture 11: LTI FIR filter design
Instructor:
Dr. Gleb V.
Tcheslavski
Contact:
gleb@ee.lamar.edu
Office Hours: Room
2030
Class web site:
http://ee.lamar.edu/gleb/
dsp/index.htm
2
Preliminary considerations
1
0
( )
M
k
k
k
H z h z
=
=
FIR filter:
unit-pulse response transfer function
(11.2.1)
Here M-1 is the filter order.
M is the number of filters coefficients. Assuming that M is odd:
1
1
0
0
0
0
( ) ; 1
M
k
k
M
k
k
k
k
b z
H z h z a
a
=
=
= =
BTW, using our

rational notation:
(11.2.2)
1 2 ( 2) ( 1)
0 1 2 2 1
3
1 1 1
1
2
2 2 2
1
1
0
2
2
( ) ...
M M
M M
M
M M M
M
k k
M k k
M
k
k
H z h h z h z h z h z
z h h z h z

+
=
=
= + + + + +
(
(
= + +
(
(

(11.2.3)
3
3 3
1
1 1
2 2
2
2 2
1 1
0 0
2
3
1
1 1
2
2
2 2
1 1
0
2
( )
M M
M
M M
k
k
M k M k
k k
M
M
M M
k
k
M k M k
k
H z z h h z h z
z h h z h z

(

(

= =
(

(

=
(
(
= + +
(
(

(
| |
(
= + +
|
( |
\ .
(

For an FIR filter to have a linear phase:

*
1 k M k
h h

=
That means an even symmetry of the coefficients.
(11.3.1)
(11.3.2)
Remember: FIR filters are always stable (no poles other than at the origin)!
4
Since the filter is stable:
3
1
1 1
2
2
2 2
1
3
1
2
2
1
0
0
2
2
( )
1
2 cos
2
( )
k k
M
M
M M
j k
j j
j
M
M
j
M k k
k
j h j h
M k
k
k
j
c
H e
M
e
e h h
h h
e e e e
e
k h
R
e
e e
e
e
e
e
(

(
Z Z
=
=

=
(
| |
(
(
| |
(
+ + |
( |
\ .
(
+ + Z
|
(
(

\ .
(
=
Here is a real function of frequency. ( )

j
c
R e
e
(11.4.1)
( ) ( )
1
2
M
j
j j
c
H e e R e
e
e e
=
As a result of symmetry:
(11.4.2)
5
Therefore:
( )
( )
( )
1
0
2
1
0
2
j
c
j
j
c
M
if R e
H e
M
if R e
e
e
e
e
e t
>
Z =

<
Generalized linear phase

We can use to design ANY amplitude of frequency response: LPF,
HPF, BPF,
( )
j
c
R e
e
When changes sign, the phase undergoes an abrupt change of 180
0

( )
j
c
R e
e
In M is even:
2
2
0
1
( ) 2 cos
2
M
j
c k k
k
M
R e h k h
e
e
=
| |
(
= +Z
|
(

\ .
(11.5.1)
(11.5.2)
6
*
1 k M k
h h

=
3
2
0
1
( ) 2 sin ;
2
M
j
s k k
k
M
R e h k h M is odd
e
e
=
| |
(
= +Z
|
(

\ .
An FIR filter will also have a linear phase in the case of odd symmetry of
filter coefficients (antisymmetric impulse response), i.e.:
It can be shown that
2
2
0
1
( ) 2 sin ;
2
M
j
s k k
k
M
R e h k h M is even
e
e
=
| |
(
= +Z
|
(

\ .
(11.6.1)
(11.6.2)
(11.6.3)
( ) ( )
1
2
M
j
j j
s
H e e jR e
e
e e
(
(

=
Therefore:
(11.6.4)
BTW:
( )
( ) ( )
H j differentiator
H j sign Hilbert transformer
O = O
O = O
(11.6.5)
(11.6.6)
7
Thus:
( )
( )
( )
1
0
2 2
3 1
0
2 2
j
s
j
j
s
M
if R e
H e
M
if R e
e
e
e
t
e
t
e
>
Z =

<
Generalized linear phase

(11.7.1)
Summarizing, a unit-pulse response of a GLP FIR filter must satisfy:
*
1 k M k
h h

=
(11.7.2)
1
0
( )
M
k
k
k
H z h z
=
=
(11.7.3)
( )
( )
( )
1
2
, , , ,...
,
j
M
j
c
j
j
s
R e LPF HPF BPF BSF
H e e
jR e differentiators Hilbert transformers
e
e
e
e
(
(11.7.4)
We only need to specify (M-1)/2 (odd M) or M/2 unique coefficients (even M).
8
Combining (11.7.2) and (11.7.3), we arrive at:
( ) { }
( )
1
( 1
1 1
* * ( 1)
1
0 0 0
)
1
( 1) * *
0
*
( 1
1
)
M M M
k k m M
k M k m
k k m
M
M m
m
m
M
H z h z h z m M k h z
z h z z H z

= = =

=

= = = = =
= =

(11.8.1)
0
j
has a zero at z z re
u
= =
*
0
1 1
j
has a zero at z e
z r
u
= =
-1 -0.5 0 0.5 1
-1
-0.5
0
0.5
1
4
Real Part
I
m
a
g
i
n
a
r
y

P
a
r
t
Example: if M = 5:
1 2 3 4
0 1 2 3 4
( ) H z h h z h z h z h z

= + + + +
Has two pairs of reciprocal zeros.
firls(4,[0 0.25],[1 0]);
9
There are four types of real GLP FIR filters:
indicate structural zeros
Type I:
symmetric h
n
,
odd M
(number of
coefficients)
even order.
Type II:
symmetric h
n
,
even M
Type III:
antisymmetric
h
n
, odd M
Type IV:
antisymmetric
h
n
, even M
10
1. For M = 4, symmetrical pulse response ( Type II FIR):
{ }
1 2 3 1 2 3
0 1 2 3 0 1 1 0
( ) H z h h z h z h z even symmetry h h z h z h z

= + + + = = + + +
1 1 0
1
0
( ) 0
z
h h H h z h
=
= +
2. For M = 4, antisymmetrical pulse response ( Type IV FIR):
{ }
1 2 3 1 2 3
0 1 2 3 0 1 1 0
( ) H z h h z h z h z odd symmetry h h z h z h z

= + + + = = +
1 1 0
1
0
( ) 0
z
h H h h z h
=
= +
3. For M = 5, antisymmetrical pulse response ( Type III FIR):
{ }
1 2 3 4 1 2 3 4
0 1 2 3 4 0 1 2 1 0
( ) H z h h z h z h z h z odd symmetry h h z h z h z h z

= + + + + = = + +
{ }
0 1 2 1 0 2
1
0 ( ) 0
z
h h h h h H z h
=
= + + = =
{ }
1
0 1 2 1 0 2
0 0 ( )
z
h h h H z h h h
=
= + = = +
(11.10.1)
(11.10.2)
(11.10.3)
T
h
i
s

i
s

w
h
e
r
e

s
t
r
u
c
t
u
r
a
l

z
e
r
o
s

c
o
m
e

f
r
o
m

11
As a result of structural zeros
Type II FIR HPF would have a zero at t - specs will be violated!
Type III FIR HPF would have a zero at t - specs will be violated!
Type IV FIR LPF would have a zero at 0 - specs will be violated!

If something like tis happens, increase or decrease the FIR order to change
the type of your filter.

We need to be careful with selection of filter orders!
12
1. Least-square error minimization
The desired magnitude response (something we specify):
( )
,
j j n
d d n
DTFT
n
H e h e
e e
=

(11.12.1)
Filter coefficients could be found as:
( )
,
1
2
j j n
d n d
h H e e d
t
e e
t
e
t

=
}
Example: Ideal LPF
( ) ( )
,
2 sin
1
2
s
2
i
1
n
2
c
c c
c
j n j n
c j n
L
c
P
c
F
c
n
j n
e e
h e
n
n
d
j n
e
e e
e
e
e
e
t t t
e
e
t e
= = = =
}
(11.12.2)
(11.12.3)
Not causal, infinite length!
We need to preserve the main features while making the filter causal.
13
The filter specified by (11.12.3) has an infinite impulse response. Therefore, we
need to truncate it at some point to make an FIR filter. As a criterion for such
truncation, we need to minimize the approximation error (the difference between
the desired and the truncated frequency responses).
The magnitude of the frequency response of the truncated filter:
( )
,
U
j j n
t t n
n L
H e h e
e e
=
=
Where L and U are points at which the pulse response was truncated.
Therefore, the objective is to find a finite-duration impulse response sequence,
whose DTFT would approximate the desired frequency response.
(11.13.1)
We need to minimize the least-square integral error of approximation.
14
In particular:
( ) ( )
2
1
min
2
j j
R t d
H e H e d
t
e e
t
e
t

u =
}
1
2 2 2
, , , ,
2
1
, ,
U L
t n d n d R t n d n n d n
n L n n U n
h h h h h h

= = = +
=
u = = + +

Apparently, to minimize u
R
(LS error), we select h
t,n
= h
d,n
for n = LU.
Moreover, GLP requires symmetry of filter coefficients, therefore L = -U.
The best finite length approximation of the ideal infinite length impulse response
in the LS error sense is obtained by truncation.
To make the resulting FIR causal, we need to shift it by eL radians.
(11.14.1)
(11.14.2)
15
Relating the point of truncation to the FIR filter order: L = (M 1)/2
,
1
sin
2
1
2
c
c
LP FIR n
c
M
n
h
M
n
e
e
t
e
| |
(
|
(

\ .
=
(

Note: filter length M (number of filter coefficients) can be both even or odd.
(11.15.1)
-50 0 50
-0.1
0
0.1
0.2
0.3
n
h
n
0 20 40 60 80 100
-0.1
0
0.1
0.2
0.3
n
h
n
0 0.2 0.4 0.6 0.8 1
-4000
-2000
0
Normalized Frequency (t rad/sample)
P
h
a
s
e

(
d
e
g
r
e
e
s
)
0 0.2 0.4 0.6 0.8 1
-100
-50
0
Normalized Frequency (t rad/sample)
M
a
g
n
i
t
u
d
e

(
d
B
)
Non-causal Causal
16
2. Windowing effects
We can view the truncation of the infinite impulse response of an ideal filter as
windowing. In frequency domain, this operation is equivalent to convolution of
a frequency response of the window function with the frequency response of
an ideal filter.
, , t n d n n
h h w =
( ) ( ) ( )
( )
1
2
j j j
t d
H e H e W e d
t
e e
t

t

=
}
from the Modulation theorem:
-1 -0.5 0 0.5 1
-0.2
0
0.2
0.4
0.6
0.8
1
1.2
e/t

|H
t
(e)|
|W(e)|
|H
d
(e)|
(11.16.1)
(11.16.2)
Gibbs phenomenon
(comes from truncated
Fourier series)
Ripples of equal magnitude in PB and SB
Transition band another window artifact
W
R
:
17
2. Windowing effects: window types
As discussed previously, different window functions can be used
Window MLW (4t/M) PSL (dB) TB (2t/M) Max SB
ripple (dB)
Rectangular ~ 0.9 -13 0.9 -21
Hanning ~ 2 -31 3.1 -44
Hamming ~ 2 -41 3.3 -53
Blackman ~ 3 -57 5.5 -74
Window properties Filter properties
Fixed windows:
18
1, 0,1,... 1
R
n
W n M = =
Rectangular:
Hanning:
Hamming:
2
1 2 2
1 cos sin , 0,1,... 1
2 1 1
N
n
n n
W n M
M M
t t | |
( (
= ~ =
|
( (

\ .
2
0.54 0.46cos , 0,1,... 1
1
M
n
n
W n M
M
t
(
= =
(

2 4
0.42 0.5cos 0.08cos , 0,1,... 1
1 1
B
n
n n
W n M
M M
t t
( (
= + =
( (

Blackman:
(11.18.1)
(11.18.2)
(11.18.3)
(11.18.4)
19
( )
2
0
0
2
1 1
1
, 0,1,... 1
K
n
n
I
M
W n M
I
|
|
| |
(
|

(
|

\ .
= =
Kaiser window:
(11.19.1)
( )
2
0
1
2
( ) 1
!
m
m
x
I x
m
=
(
= + (
(

Where: (11.19.2)
Normally, 15-20 terms in the summation are sufficient.
Note: if | = 0, W
n
K
= W
n
R
.
Adjustable windows:
20
| TB (2t/M) Max SB ripple (dB)
2 1.5 -29
3 2.0 -37
4 2.6 -45
5 3.2 -54
7 4.5 -72
8 5.1 -81
Properties:
Kaiser window is the most frequently used for the FIR filter design.
Hanning
Hamming
Blackman
21
2. Windowing: experimental design
For the given transition band:
2
s p
f
e e
t
=
Stop-band attenuation:
20lg
s
A o =
Order estimate:
7.95
1
14.36
A
M
f
~ +
Adjustable parameter:
0.4
0.1102 ( 8.7) 50
0.5842 ( 21) 0.07886 ( 21) 21 50
0 21
A A dB
A A A
A
|
>
~ + < <
<
(11.21.1)
(11.21.2)
(11.21.3)
(11.21.4)
M controls transition band, | changes ripples.

Note: in practice, ripples in SB and PB are approximately equal. If this does not
hold, need to select minimum of o
s
, o
p
.
22
2. Windowing: experimental design
Example 11.1: design an FIR LPF using a Kaiser window for the following specs:
0.3 ; 0.5 ; 40 min
p s
A dB SB attenuation e t e t = = =
The transition band:
0.5 0.3
0.1
2 2
s p
f
e e
t t
t t

= = =
7.95 40 7.95
1 1 23.32
14.36 14.36 0.1
A
M
f

~ + = + ~

Order:
0.4
0.5842 ( 21) 0.07886 ( 21) 3.4 A A | ~ + ~ Parameter:
We select M = 24, | = 4. The filter is given by (11.15.1) with the cutoff frequency:
0.4
2
p s
c
e e
e t
+
= =
The resulting filter is a type II FIR which is ok for LPF.
,
sin
;
c
t n n
n
h w L n L
n
e
t
= s s
The transfer function:
23
3. Frequency sampling
The desired frequency response
( )
1
,
0
M
j j n
d d n
DTFT
n
H e h e
e e
=
=

can be specified by its frequency samples at equally spaced (discrete) frequencies:
( )
2
k
k
M
t
e o = +
Where
(11.23.1)
(11.23.2)
1
0,1,... ,
2
2
0,1,... ,
2
M
M odd
k
M
M even
o = 0:
0
0.5
no offset
offset by M
o
t

(11.23.3)
(11.23.4)
o = 0.5:
24
The sampled frequency response will be:
( )
2
1
( )
, ,
0
, 0,1,... 1
k
M
j k n
j
M
d k d d n
n
H H e h e k M
t
o
e
o
e e
+
+
=
=
= = =
Evaluating IDFT:
2
1
( )
,
0
1
, 0,1,... 1
M
j k n
M
n d k
k
h H e n M
M
t
o
o
+
+
=
= =
Therefore, we can compute the filter coefficients from the specified M

frequency samples.
(11.24.1)
(11.24.2)
Since h
n
is real:
*
k M k
H H
o o +
=
Since h
n
is symmetric, we only need to specify either (M+1)/2 (M is odd) or
M/2 (M is even) frequency samples to determine the pulse response.
(11.24.3)
25
We can rewrite the FIR filter frequency response (11.7.4) as:
( )
( )
( )
1
2
1
1
2 2
1
,
, V
M
j
j
r n M n
j
M
j
j
r n M n
H e e symmetric h h types
H e
H e e antisymmetric h h types
e
e
e
t
e
e

| |
+
|
\ .

= I, II
= III, I
2 ; 0,1,... 1
k
k
k M
M
o
e t
+
= =
Sampled at the frequencies
the response becomes:
| |
( )
2 1
2 2
2
M
j k
M
k r
H H k e
M
t t
| o
o
t
o
( | |
+
| (
\ .
+
| |
= +
|
\ .
(11.25.1)
(11.25.2)
(11.25.3)
26
Here:
0
n
n
symmetric h
antisymmetric h
|

=

1
For simplicity, we specify the following set of real frequency samples:

( ) | |
2
1 ; 0,1,... 1
k
k r
G H k k M
M
o
t
o
+
| |
+ =
|
\ .
Therefore:
( )
2 1
2 2
M
j k
M j k
k k
H G e e
t t
| o
t
o o
( | |
+
| (
\ .
+ +
=
0 0
; 4 ...
1 2
different cases o |

= =

1

(11.26.1)
(11.26.2)
(11.26.3)
27
Case 1:
0
0 : 2
n
k
symmetric h
no offset k M
|
o e t
=
= =
2 1
2
2
2
M
M k
j k
j k j
j k j k M
M M
k
k
M
k k
j
k
G e e G H G e e e e
t
t
t t
t t
| |
|
\ .
= = =
2 2 2 2
1 1 1
0 0 0 1
2 ( ) 2
( )
0
1 1
2 1 2 1
2
0
1
1 1 1
1
1
k k k
M M U M
j kn j j kn j j kn j j kn
M M M M M M M
k k k k
k k k k U
k M k
U U
j j kn j j M k n
M M M M
k M k
k k
U
j k n j k n
M
n
M
k k
k
H e G e e G e e G e e
M M M
G G e e G e e
M
G G e G e
M
h
t t t t t t t
t t t t
t t

= = = = +
= =
| |
+ +
|
\ .
=
(
= = = +
(

(
= + +
(

= + +

{ }
2 2
0
1
1
2
1
1 2 1
2 cos ; 0,1,...
2
1
1
;
U
k
k
U
j j n j j n
k
G G
e
k n n M
M M
e e e
t t t t
t
| |
|
=
=
\ .
(

| |
= + + =
`
(
= = =
(
( |
\ .
(11.27.1)
(11.27.2)
28
( )
2
1
k
k r
G H k
M
t
| |
=
|
\ .
Since h
n
is real:
*
k M k
H H

=
| |
| |
2 1 2 1
*
2 2
2 2
M M
j k j M k
M M
r r
H k e H M k e
M M
t t
t t

| | | |
=
| |
\ . \ .
Since H
r
is real:
| |
| |
| | | |
( )
( ) | |
( )
( )
2 1 2 1
2 2
2 2
1
2 2
2 2
2 2 2
2 2
1 1
M M
j k j M k
M M
r r
M
j j
j M
r r r
k k
j M k j k j
r r
H k e H M k e
M M
H k H M k e e H M k e
M M M
H k H M k e e e
M M
t t
t t
t
t t t
t t
t t t
t t

| | | |
=
| |
\ . \ .
| | | | | |
= =
| | |
\ . \ . \ .
| | | |
=
| |
\ . \ .
(11.28.1)
(11.28.2)
(11.28.3)
(11.28.4)
29
Therefore:
k M k
G G

= (11.29.1)
The number of coefficients (frequency samples) to be specified:
1
;
2
1;
2
M
odd M
U
M
even M
- forced zero at z = -1
(11.29.2)
( ) { }
0
j
n
H e DFT h
e
= ,
Since the unit-pulse response is already evaluated, we can estimate the
corresponding frequency response as its zero-padded DFT:
and check whether the specifications are satisfied.
30
Case 2:
0
1 2 : 2 ( 0.5)
n
k
symmetric h
ffset k M
|
o e t
=
= o = +
( )
0.5
2 1
1
2
k
k r
G H k
M
t
+
| |
(
= +
|
(

\ .
0.5
0
2 2 1 1
2sin ; 0,1,... 1
2 2
U
n k
k
h G k n n M
M M
t
+
=
(

| || |
= + + =
`
( | |
\ .\ .
)

1
;
2
1;
2
M
odd M
U
M
even M
(11.30.1)
(11.30.2)
(11.30.3)
The desired frequency samples:
31
Case 3:
1
0 : 2
n
k
antisymmetric h
no ffset k M
|
o e t
=
= o =
( )
0.5
2
1
k
k r
k
G H
M
t
+
| |
=
|
\ .
( )
( ) 2 2
1
2
1
1 2 1
1 2 sin ;
2
M
n
n M k
k
h G G k n M is even
M M
t
+
=
(

| |
= +
` (
|
\ .
)

(11.31.1)
(11.31.2)
( ) 1 2
1
2 2 1
sin ;
2
M
n k
k
k
h G n M is odd
M M
t
=
(

| |
= +
` (
|
\ .
)

(11.31.3)
32
0.5
0
2 2 1 1
cos ; 0,1,... 1
2 2
U
n k
k
h G k n n M
M M
t
+
=
(

| || |
= + + =
`
( | |
\ .\ .
)

Case 4:
| |
1
0.5 : 2 0.5
n
k
antisymmetric h
ffset k M
|
o e t
=
= o = +
( )
| |
0.5
2 0.5
1
k
k r
k
G H
M
t
+
| | +
=
|
\ .
(11.32.1)
3
;
2
1;
2
M
odd M
U
M
even M
(11.32.2)
(11.32.3)
33
0 0.1 0.2 0.3 0.4 0.5
0
0.2
0.4
0.6
0.8
1
1.2
fractional frequency
|
H
(
e
)
|

actual
specified
We specify the desired response in the
SB and the PB only
The stopband attenuation in this case
(only SB and PB samples are given) is
approximately -20 dB.
0 0.1 0.2 0.3 0.4 0.5
0
0.2
0.4
0.6
0.8
1
1.2
fractional frequency
|
H
(
e
)
|

actual
specified
Alternatively, we can add transition sample(s)
and make the frequency response smoother.
The stopband attenuation would be:
for one TB sample: approximately -40 dB.
for two TB sample: approximately -60 dB.
for three TB sample: approximately -80 dB.
34
The transition band sample(s) optimized experimentally are:
# of TB samples Sample(s) value(s) SB attenuation, dB
T1 0.3789795 if o = 0
0.3570496 if o = 0.5
~ 40
T2 [0.1065 0.5886] ~ 60
T3 [0.025779 0.251635 0.723071]
[0.030957 0.27557 0.744348]
Note: only one of them is correct.
~ 80
T4 [0.006061 0.09324 0.4082 0.82097] ~ 100
Effects of adding TB sample(s) are increased SB attenuation and wider TB.
These samples must be optimized for the situation.
Selection of the offset o can help too.
Note: we can specify any desired frequency response at the samples but not in
between! Ultimately, we can view all frequency samples as the TB samples,
which leads to an optimal (equiripple) design.
35
4. Optimal equiripple design
We are interested in a LP filter satisfying the following requirements:
( )
( )
1 1 ;
;
j
p r p
j
s r s
H e PB
H e SB
e
e
o o
o o
s s +
s s
Considering four FIR types:
Type 1: symmetric pulse response, odd M.
1 n M n
h h

=
The real-valued frequency response:
( )
1 3
2
0
2
0
1
2
1 1
2 co
2
s
2
c s o
M
j
r
M
k
M
n
k n
H e a k
M M
h h n k n
e
e e
= =

| |
= + = =
`
|
\ .
=
)

1
2
1
2
, 0
1
2 , 1, 2,...
2
M
k
M
k
h k
a
M
h k
=

=
(11.35.1)
(11.35.2)
(11.35.3)
(11.35.4)
36
Type 2: symmetric pulse response, even M.
1 n M n
h h

=
( )
2
2 2
0 1
1 2 1
2 cos cos
2 2 2
M M
j
r n k
n k
M M
H e h n k n b k
e
e e
= =

| | | |
= = = =
`
| |
\ . ) \ .

2
1
2 , 1, 2,...
2
k M
k
M
b h k
= =
(11.36.1)
(11.36.2)
(11.36.3)
( )
1
2
0
cos cos
2
M
j
r k
k
H e b k
e
e
e
=
=

0 1
1
1
2 2
0.5
2 ; 1, 2,... 2
2
2
k k k
M M
b b
M
b b b k
b b
=
= =
=
Or:
Where:
(11.36.4)
(11.36.5)
37
Type 3: antisymmetric pulse response, odd M.
( )
3 1
2 2
0 1
1 1
2 sin sin
2 2
M M
j
r n k
n k
M M
H e h n k n c k
e
e e

= =

| |
= = = =
`
|
\ . )

1
2
1
2 , 1, 2,...
2
k M
k
M
c h k
= =
(11.37.1)
(11.37.2)
(11.37.3)
( )
3
2
0
sin cos
M
j
r k
k
H e c k
e
e e
=
=

( )
3 1
2 2
1 1
0 2 1
2 ; 2 5 2
0.5
M M
k k k
c c
c c c k M
c c c

+
=
= s s
+ =
Or:
Where:
(11.37.4)
(11.37.5)
1 n M n
h h

=
38
Type 4: antisymmetric pulse response, even M.
( )
2
2 2
0 1
1
2 sin sin
2 2
M M
j
r n k
n k
M M
H e h n k n d k
e
e e
= =
| |
= = = =
`
|
\ . )

2
2 , 1, 2,...
2
k M
k
M
d h k
= =
(11.38.1)
(11.38.2)
(11.38.3)
( )
2
2
0
sin cos
2
M
j
r k
k
H e d k
e
e
e
=
=

( )
3
2 2
1
0 1 1
2
2 ; 2 3 2
0.5
M M
k k k
d d
d d d k M
d d d
=
= s s
=
Or:
Where:
(11.38.4)
(11.38.5)
1 n M n
h h

=
39
Therefore, we can express the frequency response for all FIR types as:
( ) ( ) ( )
j j j
r
H e Q e P e
e e e
=
Where:
Filter type Q(e
je
) P(e
je
)
1: M odd 1
2: M even
3: M odd
4: M even
1 n M n
h h

=
1 n M n
h h

=
1 n M n
h h

=
1 n M n
h h

=
cos
2
e
sin
2
e
sine
( ) 1 2
0
cos
M
k
k
a k e
( ) 2 2
0
cos
M
k
k
b k e
( ) 3 2
0
cos
M
k
k
c k e
( ) 2 2
0
cos
M
k
k
d k e
(11.39.1)
( )
0
cos
L
j
k
k
P e k
e
o e
=
=
(11.39.2)
40
The desired frequency response:
( )
1
0
j
dr
in PB
H e
in SB
e

We can compute and compare it to . Adjusting P(e) and

selecting filter type, we can obtain the desired frequency characteristic.
( )
j
dr
H e
e
( )
j
r
H e
e
We introduce the weighting function on the approximation error:
( )
1
j
s p
in SB
W e
in PB
e
o o

If o
s
< o
p
bigger error in the passband is allowed.
(11.40.1)
(11.40.2)
41
The weighted approximation error:
( ) ( ) ( ) ( ) ( ) ( ) ( ) ( )
( ) ( )
( )
( )
( ) ( ) ( ) ( )

j j j j j j j
dr r dr
j
dr
j j j
j
j
j j j
dr
W e H e H e W e H e Q e P e
H e
W e Q e P e
E e
W e H e P e
Q e
e e e e e e e
e
e e e
e
e
e e e
( (
= =

(
( =
(
(

(11.41.1)
Where the modified weighting function and the modified desired frequency
response are:
( ) ( ) ( )
j j j
W e W e Q e
e e e
=
( )
( )
( )
j
dr
j
dr
j
H e
H e
Q e
e
e
e
=
(11.41.2)
(11.41.3)
42
For the given error function E(e
je
), the Chebyshev (mini-max)
approximation problem is to determine the filter parameters o
k
that
minimize the maximum absolute error over the frequency bands of interest:
{ }
( )
{ }
( ) ( )
0

min max min max cos
k k
L
j j j
dr k
over over S S
k
E e W e H e k
e e e
o o e e
o e
e e
=
(
(
(
=
(
(

(11.42.1)
Usually, the frequency bands of interest are specified as:
S PB SB =
(11.42.2)
a disjoint union of the passband and the stopband: 0 - e
p
and e
s
- t (for a LPF).
That is we ignore the transition band. More precisely speaking, error over the
TB will not be optimized.
The solution of this problem can be found via the alternation theorem.
43
The alternation theorem
A necessary and sufficient conditions for to be
unique and best approximation to in S is that the error function
E(e
je
) exhibit at least L + 2 extremal frequencies in S. That is, there must
exist at least L + 2 frequencies {e
i
}, such that:
( )
0
cos
L
j
k
k
P e k
e
o e
=
=
( )
j
dr
H e
e
1 2 2
...
L
e e e
+
< < <
( ) ( )
max , 1, 2,..., 2
i
j j
S
E e E e i L
e e
ee
= = +
( ) ( )
1 i i
j j
E e E e
e e
+
=
I.e., error alternates in sign between two successive extremal frequencies.
(11.43.1)
(11.43.2)
(11.43.3)
44
4. Optimal equiripple design: Example
Example 11.2: design a LPF. Since both the desired frequency response H
dr

and the weighting function W are piecewise constants:
( )
( ) ( ) ( )
{ }
( )
0
j j
r
j j j
dr r
dE e dH e
d
W e H e H e
d d d
e e
e e e
e e e
(
= = =

Consequently, the frequencies {e
i
} that correspond to the peaks of the error
function E(e
je
) also correspond to the peaks of H
r
(e
je
), i.e., where the
frequency response meets the specified error tolerance. Since H
r
(e
je
) is a
polynomial of degree L:
( ) ( ) ( )
0 0 0 0
cos cos cos
L L k L
n k
j
r k k nk k
k k n k
H e k
e
o e o | e o e
= = = =
= = =

(11.44.1)
(11.44.2)
H
r
(e
je
) has at most L-1 local minima and maxima and, therefore, at most L+1
extremal frequencies (bandages) since we add e = 0 and e = t. Furthermore, the
band-edge frequencies e
p
and e
s
are also extrema of the error function.
Therefore, E(e
je
) has at most L+3 extremal frequencies.
45
However, from the alternation theorem, there are at least L+2 extremal
frequencies in E(e
je
). Thus the error function for the LPF design will have
either L+2 or L+3 (extra ripple filters) extremal frequencies.
At the specified extremal frequencies e
n
:
( ) ( ) ( )
( )

1 ; 0,1,... 1
n n n
n
j j j
dr
W e H e P e n L
e e e
o
(
= = +

Here o represents the maximum value of the error function E(e
je
).
alternations
( )
( )
( )
( )
1

0,1,... 1
n n
n
n
j j
dr
j
P e H e n L
W e
e e
e
o
+ = ; = +
( )
( )
( )
0
1

cos 0,1,... 1
n
n
n
L
j
k n dr
j
k
k H e n L
W e
e
e
o
o e
=
+ = ; = +
or alternatively:
(11.45.1)
(11.45.2)
(11.45.3)
46
We can consider the {o
k
} and o as the unknown (to be determined)
design parameters for the given extremal frequencies. Therefore:
( )
( )
( )
( )
( )
( )
( )
0 0
1 1
1 1
0
0 0 0
1
1 1 1
1
1 1 1

1 cos cos 2 cos 1

1 cos cos 2 cos 1

1 cos cos 2 cos 1
L
L L
j j
dr
j j
dr
L
L
j j
dr L L L
L W e H e
L W e H e
H e L W e
e e
e e
e e
o
e e e
o
e e e
o
e e e
o
+
+ +
+
+ + +
( ( (
( ( (
( ( (
( ( = (
( ( (
( ( (
( ( (

(11.46.1)
Note: initially, both the design parameters and the extremal frequencies are
unknown. The above system can be solved by the iterative algorithm (Remez
exchange algorithm):
1) Guess the set of L + 1 extremal frequencies;
2) Solve for {o
k
} and o;
3) Determine the error function as in (11.39.1);
4) Determine the new set of L + 1 extremal frequencies and go to 2)
Keep running until E(e
je
) s o - convergence.
47
Alternatively, we can compute o analytically [Rabiner 75]:
( ) ( ) ( )
( ) ( ) ( )
0 1 1
0 1 1
0 1 1
1
0 1 1

...
( 1)
...

L
L
j j j
dr dr L dr
L
L
j j j
H e H e H e
W e W e W e
e e e
e e e

o

+
+
+
+
+
+ + +
=
+ +
where:
1
0
1
cos cos
L
k
n
k n
n k
e e
+
=
=
=
[
(11.47.1)
(11.47.2)
Therefore, the initial guess of the L + 2 extremal frequencies allows us to
compute the maximum value of the error o. Thus:
( )
0
, cos
L
j k
k
k
P e x x
e
o e
=
= =
(11.47.3)
48
Since we know that the polynomial at the points x
n
= cos e
n
has the values:
( ) ( )
( )
( 1)
, 0,1,... 1
n n
n
n
j j
dr
j
P e H e n L
W e
e e
e
o
= = +
the Lagrange interpolation formula can be used, which leads to
( )
( )
( )
( )
0
0
k
L
j
k k
j
k
L
k k
k
P e x x
P e
x x
e
e
|
|
=
=
(

=
(

Here:
cos
cos
k k
x
x
e
e
=
=
0
1
L
k
n
k n
n k
x x
|
=
=
=
[
(11.48.1)
(11.48.2)
(11.48.3)
(11.48.4)
(11.48.5)
49
Having the solution for P, we can compute the error function:
( ) ( ) ( ) ( )

j j j j
dr
E e W e H e P e
e e e e
(
=

on a dense set of frequency points. Usually, 16M frequency points are sufficient.
If the error exceeds the estimated tolerance o, we select a new set of
frequencies corresponding to the L+2 largest peaks of error function and the
procedure starts from (11.45.1). New set of critical frequencies will lead to
increased o. As a result, the algorithm produces the optimal solution with equal
ripples in the PB and the SB for the given M.
The order of the filter can be estimated as:
( ) ( )
20lg 13
1
14.6 2
s p
s p
M
o o
e e t

= +
(11.49.1)
(11.49.2)
50
Summary
Historically, the window-based algorithm was the first proposed method
for FIR design. Its major disadvantage is the lack of precise control of
the critical frequencies, such as e
s
and e
p
. These values, in general,
depend on the type of the window and the filter length M.

The frequency sampling method is attractive since it specifies an
arbitrary frequency response at the uniformly spaced frequencies and
the transition band is a multiple of 2t/M. However, no control for the
response in between the samples.

The Chebyshev approximation method provides a total control of the
filter specifications and may lead to an equiripple design. This way, the
approximation error is spread evenly across the PB and SB, which
leads to an optimal filter. This method is usually preferred.
51
Remarks and conclusions
Since orders of FIR filters can be quite high, these filters can introduce a
considerable group delay, which is approximately M/2.
The following Matlab functions are handy for FIR design:
fir1 FIR filter design using the windowing method
fir2 FIR filter design via the frequency sampling method
firpm Parks-McClellan optimal equiripple FIR design
firpmord - Parks-McClellan optimal equiripple FIR order estimator
52
Appendix: the ideal transfer functions
LPF:
( )
,
sin
,
c
LP n
n
h n
n
e
t
= < <
HPF:
( ) ,
1 , 0
sin
, 0
c
HP n
c
n
h
n
n
n
e t
e
t
=
=

>
( ) ( )
2 1
,
sin sin
, 0
c c
BP n
n n
h n
n n
e e
t t
= >
( )
( ) ( )
2 1
,
1 2
1 , 0
sin sin
, 0
c c
BS n
c c
n
h
n n
n
n n
e e
t
e e
t t

=
>
BPF:
BSF:
(11.52.1)
(11.52.2)
(11.52.3)
(11.52.4)

Lecture 11 - FIR Filter Design

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Lecture 11 - FIR Filter Design

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ELEN 5346/4304 DSP and Filter Design Fall 2008

BTW, using our

For an FIR filter to have a linear phase:

Here is a real function of frequency. ( )

Generalized linear phase

Generalized linear phase

Therefore, we can compute the filter coefficients from the specified M

For simplicity, we specify the following set of real frequency samples:

We can compute and compare it to . Adjusting P(e) and

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