FILTERS

Mrs. LINI MATHEW

Associate Professor
Electrical Engineering Department

NITTTR, Chandigarh

TYPES OF FILTERS
Four types of mathematical functions to
achieve the required approximations in the
response characteristics of filters (Recursive
Type)
Butterworth Approximation Function

Chebyshev Approximation Function

Elliptical Approximation Function
Bessels Approximation Function

BUTTERWORTH APPROXIMATION
The transfer function of a first order LPF(RC) is

1
1
H() =
=
1 + jCR 1 + j c

Squared Magnitude Form

H( )

H( ) =

1 + c 2n

1 + c 2

represents a
Butterworth polynomial
of order n

BUTTERWORTH APPROXIMATION
nth order filter n stages are cascaded
The cut off becomes sharper as the order of the
filter is increased.

The pass band gain remains more or less constant

and flat up to the cut off frequency.
H() in dB

20 log H( ) = 20 log
= -10log 1 + c 2n ]

[ (

1 + c 2n

DESIGN PROCEDURE-BUTTERWORTH FILTER

To design a filter the following information is
required:
(i) Pass band gain required, H1()

(ii) Frequency upto which the pass band gain

must remain more or less constant, 1
(iii) Amount of attenuation required, H2()
(iv) Frequency from which the attenuation must
start, 2

DESIGN PROCEDURE-BUTTERWORTH FILTER

Step I

H( ) =

1 + c 2n

At = 1

H1( ) =

At = 2

1 + 1 c 2n

H2 ( ) =

1 + 2 c 2n

Solve these two eqns to get n and C

Step II Determination of the poles of H(s)

Poles of the Butterworth polynomial lie on a circle
whose radius is C
Number of Butterworth poles = 2n

DESIGN PROCEDURE-BUTTERWORTH FILTER

Angle between the poles, = 360/2n
Location of the poles (i) If n is even, then the
location of the first pole is at /2 from the
x-axis in the counter clockwise direction.
Location of the subsequent poles are /2 + ,
/2 + 2, /2 + 3, ......
(ii) If n is odd, then the location of the first
pole is on the x-axis.
Location of the subsequent poles are , 2,....

with angle measured in the counter clockwise

direction.

DESIGN PROCEDURE-BUTTERWORTH FILTER

Step III Determination of the valid poles of
H(s)
Poles that lie on the left half of s-plane alone
are stable poles.
Poles that lie in between 90o and 270o alone are
valid poles.
If is the angle of a valid pole wrt x-axis,
then the pole and its conjugate are located at
C (cos j sin )

DESIGN PROCEDURE-BUTTERWORTH FILTER

Step IV To find the expression for H(s)
2
2

c
c
H(s) =
=
(s + a + jb )( s + a - jb )
s 2 + 2 c s + 2
c

where a+jb and a-jb are the poles and is

the damping factor.
Step V Determination of the filter components
R & C

Step VI Determination of the amplifier

elements R1, R2, R3, R4 etc.

DESIGN OF DIGITAL IIR FILTERS

To design a desired digital filter, first develop
the expression for H(s) based on the procedures
in the analog filter design, and then modify it in
an appropriate fashion to suit the digital domain.
IMPULSE INVARIANT DESIGN
Convert H(s) to h(t) by inverse Laplace transform
Convert h(t) to H(z) by direct Z transform
Using H(z) construct the required digital filter
BILINEAR TRANSFORMATION
Convert H(s) to H(z) directly by using the
expression
2(1 - z-1)
s=

T (1 + z-1)

DESIGN OF BUTTERWORTH IIR FILTERS

For the design of digital filters, the idea
of sampling has to be incorporated.
Step I Normalization of frequencies
f1
1 = 2
fs

f2
2 = 2
fs

Step II Determination of n and C

H( ) =

1 + c 2n

Step III Determination of valid poles B1,

B2, ....

DESIGN OF BUTTERWORTH IIR FILTERS

Step IV Finding the expression for H(s) in the
analog domain

c2
H(s) =
(s + a + jb )(s + a - jb )
Taking the inverse Laplace transform
h(t) = L-1[H(s)] and H(z) = Z-1[h(t)]

CHEBYSHEV DIGITAL IIR FILTERS

Two types:
(i)Type I (regular) Chebyshev Filter
Passband containing ripples, and stopband
containing no ripples
(ii) Type II (inverse) Chebyshev Filter
Passband containing ripples, and stopband
also containing ripples

H()

1+

2
Cn
2 2
Cn

DESIGN OF CHEBYSHEV LOW PASS FILTERS

Chebyshev design makes use of the Chebyshev
polynomial
1
2
H() =
2 2
1 + Cn
where is the amount of ripple in the
magnitude

Cn is the Chebyshev constant

Cn = cosh[n cosh-1 ]

if /c 1

Cn = cos[n cos -1 ]

if /c 1

DESIGN OF CHEBYSHEV LOW PASS FILTERS

For the Chebyshev design 1=c
Specifications of the desired Chebyshev LPF:
(i) Pass band gain required, H1()
(ii) Frequency upto which the pass band gain
must remain more or less steady, 1=c

(iii) Amount of attenuation required, H2()

(iv) Frequency from which the attenuation must
start, 2

DESIGN OF CHEBYSHEV LOW PASS FILTERS

Step I Determination of n
using the Chebyshev Polynomial & Chebyshev
constant

Step II Determination of poles of H(s) as per

the following rules
(i) From the given specifications, determine the
valid Butterworth poles ie. B1 = a+jb B2 = c+jd
....
1
(ii) Determine the factor k from k = sinh -1 (1 )
n

DESIGN OF CHEBYSHEV LOW PASS FILTERS

(iii) Find tanh(k) and cosh(k)
(iv) Multiply the real part (a,c,...) of the poles
with tanh(k). This is called normalisation. Thus
the normalised (modified) Butterworth poles
are B1 = a tanh(k) +jb,
B2 = c tanh(k) +jd
(v) Now, denormalise the poles by multiplying with
cosh(k) to get the desired Chebyshev poles.
The Chebyshev poles are
C1 = {cosh(k) [a tanh(k) +jb]}
C2 = {cosh(k) [c tanh(k) +jd]}

DESIGN OF CHEBYSHEV LOW PASS FILTERS

Step III Determination of transfer function

H(s)
P
H(s) =
(s - C1)(s - C2 )(s - C3 )(s - C4 )

where P={|C1||C4|}{|C2||C3|}

DIGITAL FILTER DESIGN USING BILINEAR

TRANSFORMATION
A condition to be satisfied before using bilinear
transformation
2
= tan ( 2)
T
Step I Normalization of frequencies
Step II Conversion of digital frequencies to
analog frequencies
Step III Determination of n and c
Step IV Determination of the valid poles and
formulating H(s) and H(z)

DESIGN OF FILTERS USING

WINDOW FUNCTIONS
There are infinite number of coefficients in a
Fourier Series Expansion.
The abrupt termination of the Fourier series
coefficients to a finite value produces sharp
transients or introduce ripples in the frequency
response characteristic H(). This is due to the
non-uniform convergence of the Fourier Series at
a discontinuity.
The oscillatory behaviour near the band edge of
the filter is called Gibbs Phenomenon

DESIGN OF FILTERS USING

WINDOW FUNCTIONS
Window functions are mathematical functions that
are designed to have tapering characteristics
When the impulse response h(n), derived from
the given transfer function H(), is multiplied by
an appropriate window function w(n), we get a
modified impulse function h(n), which shows
gradually decreasing filter coefficients.
These filter coefficients will ensure the absence
of the Gibbs Phenomenon from the operating
regions of the filter.

Low pass filter

designed with
rectangular
window

Low pass filter

designed with
Hamming window

Low pass filter

designed with
Blackman window

Low pass filter

designed with
Kaiser window

WINDOW FUNCTIONS FILTER DESIGN

WINDOW FUNCTIONS FILTER DESIGN

DESIGN OF DIGITAL FIR FILTERS

FIR filters are designed by assuming that the
magnitude of transfer function H() is unity.
The phase of H() is not taken as unity

|H()|=1 and H() = ejn

here is constant, hence FIR filters are called
constant phase filters
We do not use any type of frequency
transformation techniques, in the design of FIR
filters

DESIGN OF DIGITAL FIR FILTERS

USING THE FOURIER SERIES
METHOD
Step I Normalisation of cut off frequency
fc
c = 2
fs
Step II Fixing the transfer function to be used
H() = 1 - /2 / 2
= 0 elsew here
H( ) = e - j0n - /2 / 2
= 0 elsew here

Step III Determining the impulse response of

the filter h(n) = sinc(n/2)

DESIGN OF DIGITAL FIR FILTERS

USING THE FOURIER SERIES
METHOD
Step IV Determining the coefficients of the
impulse response sequence
ie. h(0), h(1), h(2), ....
Step V Determining the transfer function
back from the impulse response
sequence ie.H(z) is determined
Step VI Implementation of the filter

FREQUENCY SAMPLING (FOURIER

TRANSFORMATION) METHOD
In this method, we make use of the
theory of DFT for determining the filter
coefficients
The impulse response h(n) is determined
N
using IDFT
1
h(n) =

N +1

H(k )e j2 kn / N+1

k =0

N is the order of the filter & L=N+1 is

the length of the filter

FREQUENCY SAMPLING (FOURIER

TRANSFORMATION) METHOD
We assume the transfer function
H(k)=1

H(k) is a periodic function with sample

values at k=0,1,2,....N. These sample
values are to be determined first from
the transfer characteristics of the
filter
After this h(n) should be computed
Take the z transform H(z)

FREQUENCY RESPONSE
CHARACTERISTICS OF LPF

Fourier transformation of a signal produces

infinite frequency bands in the positive and
negative frequency axes
In frequency sampling method, the period
for the computation of h(n) is considered to
be 0 to 2

DESIGN STEPS IN THE

FREQUENCY SAMPLING METHOD
Step I Normalisation of frequency
fc
c = 2
fs
Step II Fixing the transfer function to be used

H(k ) = 1
Step III Determination of the locations and
amplitudes of the samples
We have to convert the limits from the factors
related to (0 to 2) to numbers related to k.
ie. interval between adjacent samples =2/N+1
samples are located at k = 2k/N+1

DESIGN STEPS IN THE

FREQUENCY SAMPLING METHOD
Step IV Determination of the values of
impulse response
Step V Determination of the transfer
function H(z)

THANKS