N ~ log10 [(10- 0.1Kl - 1)/(10- 0.

1 ~ - 1)]

where,

A = ill (n.. - ill)/[ - ili + ill n..]

B =il2 (ilu - ill)/[il~ - ill ilu]

5.1.5 ChebyshevFilters.
There are two types of chebyshev fllters. Type-I filters are all-pole ftlters that
has equiripple behaviour in the passband and a monotonIC runctIon In the stopband. '
The second type (Type-II) filters contain both oles and zeros and has a monotonic -
behaviour in the rpassband and an equiripple behaviour in the stop an. e zeros
'Of thIs type of tJlters lie on the imaginary axis In the s-plane.
~ The type-I -normalizedcltebyshev lowpass filter"s magnitud~quared frequency
response is given by,
(5.11)
I =
HN (j il) 12 1/(1 H2 T~ (il»
where ( is the filter error parameter defIned as,
. (2= 100.1Ai'- 1; ~ - Attenuation in passband
and TJ\ (il) is the Nth order chebyshev polynomial. The chebyshev polynomials can
be defined as,
cos (K cos- 1 il) for! il I ~ 1.
T (il) = (5.12)
N { cosh (N cosh-1 il) for i il I> 1
It can easily be seen from Eqn (5.12) that,
I
HN (j il)
1
2 is
1+£
~ at il = 0 for even
Nand 1 at il = 0 for odd N. These two cases of magnitude squared frequency response
of the type-I chebyshev filter are given in Fig 5.3.

11-1" (iQ)I' 11-1" 00)1'

lQ, ,0 1 U. 0
Odd-N Even-N

Fig. 5.3 Magnitude squared frequency response for the normalized type-I
,- chebyshev filter of odd.N and even-No
 '?

From Fig 5.3 and Eqn (5.11), we can observe the following properties.

1. ! HN GQ) 2 oscillates between 1 and 1/(1 + ~h within the passband, the so

1

I. 2 .
called equiripple, and has a value of 1/(1 + ( ) at Q = 1, the so called cutoff
frequency.

2. I HN (j Q) 12 is monotonic outside the passband. The stopband begins at

~ with I
. (j Q) 12 =-t-
HN A.

The normalized transfer function HN (S) of the chebyshev filter can be obtained
from Eqn (5.11). That is, we can find the poles of HN (S). HN ( - S) and select the
left half plane poles to form HN (S). The poles of HN (S) . HN ( - S) are given by roots
of the denominator of Eqn (5.11). We can show that the roots of,

1 + (2 T~ (S/j) = O.

are located on an ellipse in the S-plane and defined as 8K = ak + j Qk, where,

. ( l . - 11 \ . 2K - 1 'I
aK=sl
00
lN'SI 00 1 -
()
sm (~
,~1 )
17t
(5.13)

1 . - l' 1 ( 2K - 1
QK =cosh N' sm h( EJ cos l 2N ) 7t
for k= I, 2,..., 2N.

We now form the normalized transfer function H!\ (8) by using the LHP poles of
HN (8). HN ( - 8), as,
r--- - -
. Ho
HN(S)=~ (5.14)

TI (S - Pi)
i= 1

where Pi's are SK's on LHP and we choose the constant Ho as,
N
10- 0.05 ~ . TI ( - Pi) for even -N
i=1
Ho= N (5.15)

TI (-Pi) for odd - N

i=1

so as to achieve zero minimum passband attenuation, that is, which makes Ho equal
to 1 for odd-N and 1/~ for even - N.
Computation of N:
It is essential to get a closed form formula for the order N, in terms of given
filter specifications. Let us derive an equation for N in the case of normalized lowpass
specifications.
For normalized case, the cutoff frequency no =1 rad/sec. Let us choose the
passband edge frequency .01 as equal to the cutoff frequency.
Therefore,

TN (0) = cosh (N cosh- 10) for .01 ~ 0 ~ ".

The normalized loss function, A (j .0) in dB is given by,
. 2 2 1/2
A.9 .0) = 20 log [1 +E TN (.0)]
22
= 10 log [1 + E TN (O)J

At the stopband edge (critical) frequency, 02, the loss f~ction is,
A (j .02)= As ' the stopband attenuation. "-

2 2 -
That is, As = 10 log [1 + E TN (02) ] .

= 10 log { 1 + (10°.1 A" -1) [cosh (N cosh - 102)]2 }

Therefore, from the above relationship, we get an expression for 02 as,

02 = cosh ( ~ cosh- 1 [(10°.1A"- 1)/(10°.1,\, - 1)]1/2 )

Then the ratio of 02 to 01 can be rearranged to obtain,

N = cosh-1 [(10°.1 A, - 1)/(10°.1 A,.- 1)]112

cosh-1 (02/01)

If N is not an integer in the above equation, we choose the next larger integer. So
we reVl'1'ite the equajion as follows:
N2: cosh-1 [(10°.1 A, -1)/(10°.1 '\, -1)]112 (5.16)
cosh- 1 (02/01)

Note: The values of sinh-1 (x) and cosh-1 (x) can be evaluated using the following
identities. -

sinh-1 (x) =In (x + -.Jx2 + 1)

cosh - 1 (x) =in (x + -.Jx2- 1 )

 To design a lowpass filter with unnoimalized transfer function, first we have to
identify the normalized one by using the given specification using Eqns (5.14) and
(5.16) and then do the lowpass-to-Iowpass transformation.
To design other types of chebyshev filters we can proceed as in the Butterworth
fllter case. The required formulas to find the order N of chebyshev polynomial to
form the transfer function of the prototype filter are listed below. I

Highpass filters:
h
-1 0.1A ° 1A
N <:cos [(10 '-
-1 1)/(10' P - 1)]1/2
cosh (il1/il2)
Bandpass filters:

N <:cosh-1 [(10°.1 A, -1)/(10°.1 ~ -1)]112

cosh - 1 [min <I A I, I B I)]

where the parameters A and B are defined as in the Butterworth bandpass filter
case.
In the ciesign of chebyshev bandstop filter, to fmd the order N, the above given
formula for bandpass filters can be used, but the parameters A and B are defined
as in the Buttervvorth bandstop filter case.

5.2.~igital IIR Filter Design

5.2.1. Introduction

e most general form of digital IIR filte.. transfer function is given by,
M
2: biZ-i

H(Z)=i~O (5.17)

.2:i=O aiZ-i

In constrast
--=----. to FIR fllters, stable, realizable IIR filters cannot achieve an exactly
linear phase characteristics. Since, to obtain linear phase required H (Z) H (Z- 1), =
which in the case 01 llK !liters, implies for every pole strictly inside the unit circle
in H (Z) there had to be a mirror-image pole outside the unit circle-there by making
the filter unstable. Therefore the filter design problem for IIR filters always involves
approximation of both magnitude and phase response specifications.
When an TIR filter is determined strictly in terms of a magnitude approximation,
where the phase is completely disregarded, it is convenient to consider designs in
terms of the magnitude squared function, which is defined as,

I H (ei~ 12 = I H (Z)" H (r1) !z=i'

The poles and zeros of a magnitude squared function are distributed with
mirror-image symmetry with respect to the unit circle in the Z-plane. The poles of
H Z are uniquely determined from the magnitude squared fu.nction ~
those I .
mside the unit circle for a stable IIR fi ter. e zeros may lie anywhere in the
-Z:-plane, but,"li" we choose zeros ot magnitude s uared function lying InsIde or on the
unit circle in e -p ane, en e resu ting filter will be a minimum ase filter
(that is, lesser pnase dIstortIon filter. e pro em of designing IIR filters is nothing

~ filter's response approximates a desired response. -

2ut fmdm~ tne niter coenlClent Djs and ai'S of Eqn (5.17) such that some aspects of

A method for designing IIR digital filters is _direct closed form digital design "

in either the frequency or the time domain. Beginning with the desired response of
-the filter, one can often decide where to place poles and zeros at appropriate positions
in the Z-plane to approximate this response directly. Among the techniques that fall
into the category of direct digital design are magnitude squared function design and
time domain design.

Another method by which IIR filters are often designed is by using Optimation
Procedures to place poles and zeros at appropriate positions in the Z-plane to
approXImate in some sense the desired response. This design procedure does not
generally yield closed form expressions for pole and zero positions (that is, filter
coefficients) as a function of the desired response. This optimation procedure is used
to determine the filter coefficients that minimize some error criterion, subject to the
appropriate design equations. Using this iterative procedure either the error
eventually reaches a minimum value or a specified maximum number of iterations
is performed and the procedure terminates. The minimum mean squared error design
is one of the important methods based on optimization procedure.

The most popular technique for designing IIR filters is to digitize an analog
filter that satisfies the design specifications. The technique of designing an
appropriate analog filter and digitizing the resulting transfer function to give a
digital fIlter is most useful for designing standard filters such as lowpass, highpass,
bandpass and bandstop filters where a considerable amount knowledge on such analog
filter is available in the literature. This approach is highly preferred, because:
1. The art of analog filter design is highly advanced and since useful results
can be achieved, it is advantageous to utilize the design procedures already
developed for analog filters.
 2. Many useful analog design methods have relatively simple closed form design
formulas. Therefore, digital filter design methods based on such analog design
formulas are rather simple to compute.

3. In many applications it is of interest to use a digital filter to simulate the

performance of an analog linear time-invariant filt::!r.

There are four most widely used procedures for mapping the analog transfer
function (digitizing) into digital transfer function. They are,
1. Method of mapping of differentials.

2. Matcl1ed z-transform technique.

3. Impulse invariant transformation.

4. Bilinear transformation.

Mapping of differentials is the simplest way of digitizing the continuous system.

The method is to replace differentials in the differential equations of the continuous
syi;tem with finite differences in order to obtain a difference equation that
approximates the giveu differencial equation. This simplest replacement that can be
made is to use a forward or backward difference to replace a first' differential. The
attractive feature of replacing differentials by simple. difference is that rational
transfer functions in S become rational transfer functions in Z by using simple
substitutions -
like 8 = (2 l)/T, 8 = (1 :- Z- l)/T, etc, where T is the sampling period.
This mapping has the desirable property, that a stable analog filter is transformed
into a stable digital f1lter. However, the possible location of the poles of the digital
filter are confined to relatively small frequencies and as a consequence, the mappmg
is restricted to the design of lowpass filters, and bandpass filters having relatively
small resonant frequencies. That is, this mapping method does not adaquately
preserve the filter characteristics.

Another method of converting an analog fllter into an equivalent digital filter

is to map the poles and zeros of H (8) directly into poles and zeros in the Z-plane.
This mapping has the property that an 8-plane pole (zero) at 8 =- a maps to a
Z-plane pole (zero) at Z = e- aT where T is the sampling period. Thus each factor of
the form (8 + a) in H (S) is mapped into the factor (1 - e- aT Z- \ This mapping is
called the matched Z-transformation To preserve the frequency characteristic of
the analog filter, the sampling interval in the matched Z-transformation must be
properly selected to yield the pole and zero locations at the equivalent positions in
the Z-piane. Although the matched Z-transformation is easy to apply, there are many
case when it is not a suitable mapping. For example, if the analog system has zeros
.with ce~fL.fi:.e.qu.e.ncies greater than half of the sanlPling- freqmmcy, the matched
Z-transformation is unsuitable. In general, use of impulse invariant or bilinear
transformation is to be preferred over the matched Z-transformation.
 1

I
1
i
1
I
(5.18)
H(ei°T)=* L
1=--
Haun+jlQ,.)
---

~here ~ =-2T7tis the ra..cIian_samPli?[ frequency_of ~he-~gital s!~tem. This mappin~

f~om the S-plane -to the
-- Z-plane
- -is shown
- -- in---Fig ~.
jn
------------------------.S-plane Z-plane
37tIT

7tIT

-7tIT

}E[C-------.

Fig. 5.4. The mapping from the S-plane to the Z-plane corresponding
to impulse invarian~ transformation.

Each horizontal strip of the S-plane of width4 2'; is mapped into the entire
Z-plane. The left half the strip maps into the interior of the unit circle. the right
half of the strip maps into the e),,1:erior of the unit circle, and the imaginary axis in
the strip maps onto the unit circle. Adjacent strips in the S-plane are thus aliased
or folded over into each other in the Z-plane. That is, the impulse invariance method
does not correspond to a simple algebraic mapping of S-plane to the Z-plane. From
the Fig 5.4, it is clear that the frequency response of an analog filter and the
equivalent digital filter obtained by impulse invariant transformation are to be

identical, the analog filter must be band limited to the range ~7t::;n s ¥. Otherwise,
the mapping produces aliasing errors (interferences).
 Let us consider a analog system with transfer function in its partial fraction
expansion, in order to show the method of mapping in impulse invariant
transformation.

N AK
(5.19)
Ha (8) L
= k=l 8 -8 K

The corresponding impulse response is given by,

N
ha (t) = L AK' eSKt u (t)
k=l

where u (t) is a continuous-time unit-step function.

The unit sample response of the digital filter is then obtained by replacing t by
nT, where T is the sampling period. That is,
N
h (n) = ha (nT) = L AK eSKnT u (n).
K=l
N
\n
L
= K=l AK (eSK T ) u (n).

The transfer function of the digital filter is given by the Z-transform of h (n) as,

N AK
(5.20)
L
H (Z) = K=l 1 - eSK T Z

In comparing Eqns (5.':'9) and (5.20) we observed that a pole at S =SK in the 8-plane
transforms to a pole at eSKT in the Z-plane and the coefficients in the partial fraction
expansions of Ha (8) and H (Z) are equal. Therefore, if the analog filter is stable, then
the resulting digital filter is also stable. However, for complex poles and multi-order
poles cases, this mapping approach should be modified.

Except for aliasing (due to absense of band limitation) the relationship between
analog and digital frequencies is linear, that is, the shape of the frequency response
is preserved. It should be noted that the impulse invariance transformation is
appropriate for band limited filters. For example, highpass or band-stop filters would
require additional band limiting (guard) filter to avoid severe aliasing distortion.

Let us illustrate the way of converting an analog filter's transfer function into
a digital transfer function through two examples.
Example 5.3: Convert the analog filter transfer function,
1
H (8) = - -

into H (Z) using impulse invariance transformation. Assume an sampling frequency

of 5 sps.

Solution: The partial fraction expansion of given H (s) yields,

1 --1-
H (8) =8 + 1 - 8 + 2

The impulse response can be obtained, by taking inverse Laplace, which results in,

h (t) =e- t - e- 2t ; t ~ O.
By replacing t by nT, we obtain,

h(nT)=e-nT-e-2nT; n~O.

By applying Z-transform, we have,

1 --1:
H (Z) = 1-e - T Z- 1 - - 2T
1 -e Z- 1

1 - e- 2TZ-l - 1 + e- T z- 1
1 - e- T Z- 1 - e- 2T z- 1 + e- 3T Z- 2

- (e-T -e-2T)Z-1
-1-(e-T+e-ZT)Z-I+e-3TZ-2

i
For T = see, the transfer function of the corresponding digital filter is given by,

H (Z) =1 - 0.148r
1.489Z- 1 +
1
0.549Z - 2

In general, there are four steps involved in the design of digital filter from its
analog counterpart through impulse invariant method. They are,
1. Use the given specifications and fmd the order N and formulate H (8).

2. Find the il1Verse Laplace transform of H (8) ~ h (t).

::J. Replace t by nT ~ h (nT).

4. Find the Z-transform of h (nT) ~ H (Z).

 Example 5.4: Convert the analog filter with system function H (8) = 8 + 0'21
(8 + 0.1) + 9
into a digital IIR filter by means of the impulse invariant method.
Solution:

The impulse response of the given analog filter can be obtained using inverse
Laplace table. (see appendix - F). Thus, we have,

h (t) = e- O.lt cos 3t; t ;::0

'3t - i3t 1
- - O.lt e1 + e. i
-e [ 2 J
= 1. e(- 0.1+ j3) t + 1. e(- 0.1 - j3) t
2 2

Then, replacing t by nT, provides,

h l. nT ) = 1. e(- 0.1 + j3) nT + 1. el- 0.1 - j3) nT . n;::O

2 2 '

Applying Z-transform leads to the required digital transfer function,

1 1.
- 2
2 + .
H (Z) =1- e(- 0.1 + j3) T Z- 1 1- e(- 0.1- .13)T Z- 1

1.
2 (1- e(- 0.1-j3i T Z-l) + 1.
2 (1- e(- 0.1+j3) T Z-l)
(1 - e(- 0.1+j3)TZ-1) (1- e(- 0.1- j3) T Z-l)

- 1 - (e- 0.1 cos 3T) Z- 1 .

- 1- (2e- 0.1 cos 3T) Z-l + e- 0.2T Z- 2
Thus we obtained the digital IIR filter transfer function in a realizable form.

5.2.3. Bilinear Transformation

A simple conformal mapping from the S-plane to the Z-plane which eliminates
the aliasing problem mentioned above and preserves the desired simple algebraic
form is the bilinear transformation defined by,

2 (1 - Z- 1)
(5.21)
S ~ T (1 + Z- 1)

The nature of this mapping is shown in Fig 5.5. As seen in this figure, the entire
j n axis in the S-plane is mapped onto the unit circle. The Eqn (5.21) can be rewritten
for Z in terms of S as,
 j?=
z =(2/T)+ 8 (5.22)
(2 T)-8
When S =j n, we find, \;f\ ~

Z (2/T) + j n
==
(2/T) -j n
where, we have, I Z I = 1. Also from Eqn (5.22), we find that for real part of S
negative, the I Z I < 1 and for real part of S positive, the I Z I > 1. This implies, the
left of the S-plane is mapped inside the unit circle and the right half of the S-plane
is mapped outside the unit circle in the Z-plane. Therefore, the bilinear
transformation yields stable digital filters from stable analog filters. Also, the bilinear
transformation avoid the problem ()f aliasing encountered with the use of impulse
invariance, because it maps the entire j Q axis in the S-plane onto the unit circle ~
the Z-plane.

S-plane
Z-plane

Fig. 5.5 The mapping of the S-plane into the

Z-plane using the bilinear transformation.
The transfer function of the digital filter H (Z) is obtained from the bilinear
transformation by making the algebraic substitution of Eqn (5.21).
That is.

H(Z)= H(S) Is=! (l_Z-I) (s,tV

T (l+Z-I)

However, there is a highly non-linear relationship between analog frequency n and

digital frequency w. Evaluation Eqn (5.21) for Z ==e?'"and S = j n giving,
 '
. 2 (l-e-j(J)) . c)}'1J.' ~"" I v
I
~ 'l-,V

which can be written as,

JQ=T (1 + e-j(J)) ~ ( t
e.-

~~2-
. 2 [ei(J)/2
J Q = T [ei(J)/2
- e-j(J)/2]
+ e-jw/2]
:::
T ~j 5;\,> CJ/~
.;l ~ '112-
or

j Q = ¥"j. tan ( ~)

The inverse relationship

I n=¥tan(
is given by,
~ n- (5~24)

OJ= 2 tan-l nT I (5.25)

( 2 )
The nonlinear relationship between Q and OJis shown in Fig 5.6 for the case T =2.
For small values of OJ,the mapping is almost linear. But for most of the frequency
scale, the mapping is highly non-linear (Frequency compression effect). This
imposes a strong restriction on bilinear transformation. It implies that the amplitude

OJ
OJ =2 tan" (nT/2)

n
the analog filter transfer function is transformed to the digital domain using Eqn
(5.23) and the resulting digital filter will meet the desired specifications.

8 ro

s
"""""''''''''''''''''.
Q = 2fT tan (ro'2)

rQ)
Qp: ;ils Q

~
IH(jQ)1

Qp Q

Fig. 5.7. Techniques for compensating the nonlinear frequency

warping of the bilinear transformation.

Although the bilinear transformation can be used to transform the stable analog
filter into the stable digital filter, the distortion in the frequency axis will manifest
itself in terms of distortion in the phase characteristic associated with the filter.
Therefore, a linear phase analog filter cannot be mapped into a linear phase digital
filter through the bilinear transformation.

Let us illustrates the procedure for bilinear transformation to design a digital

IIR filter from analog transfer function through the following examples.

Example 5.5 Design a single pole lowpass digital IIR filter with - 3 dB bandwidth
of 0.2 11,by use of bilinear transformation.

Solution: The first-order (single pole), normalized analog transfer function is given
by, (Butterworth case),
I"
H (8) = 8 + 1

The cutoff frequency given is 0.2 1trad. It is prewarped using.

 2 We
ne ="Ttan :2
Assuming T = 1 see, we h'ave
" 0.21t
2 0 ~5
~<e= tan 2"" = .0 .

Next, we do transformation from prototype to new analog lowpass filter with

ne = 0.65 by,

H(8) = H (8) 18=8/0.65

0.65
8 + 0.65

Then performing the bilinear transformation yields,

A

H (Z) = H (8) 18 = 2 11- Z- 1) where T = 1 see

T (1+Z-1)
0.65
2 (1 - Z- 1)
1 +
-
0 .6 0
(1+ Z- )

0.65 (1 + Z- 1)
2.65 - 1.35T 1
0.245 + 0.245r 1
1- 0.509 T 1

Thus, we have -the required digital filter transfer function in a suitable form for
realization.

Therefore, basically there are three steps involved in the design procedure of
IIR filter by means of bilinear transformation. They are,
1. Prewarp the given frequency specifications and find the order of the filter to
formulate Hj\ (8), the normalized lowpass filter system function.

2. Form the new analog transfer function for the prewarped specifications. That
is,

/I H (8) = H (8) 18=8/5:2

2 ~
wheren="Ttan
()
2
170 INFINITE IMPULSE RESPONSE FILTER
"
3. Apply bilinear transformation to H (8) and find H (Z). That is,
"
H (Z) = H (8) I 8 -i (l - Z-1)
T (l+Z"1

Example 5.6. Design a digital Butterworth lowpass filter using the bilinear
transformation method to satisfy the following specifications:

a) - 3.01 dB cutoff frequency at 0.5 1trad

b) at least 15 dB attenuation at 0.75 1trad

Solution:

Step 1: Prewarping 001and OJ:!using T =1 sec.

~
~
01 =-tan
2
-=2
001
tan-= 2
0.5 1t
2.
Lp.\-O"'"t-
T 2 0'

02 =2 tan 0.752 1t = 4.8282.

The order of normalized Butterworth lowpass filter can be obtained using,

N>.
Jog10 [ (103.01/10 - 1)/(1015/10 - 1) ]
- 2 log [2/4.8282]

~ 1.9412.

We choose the next larger integer as N =2 and formulate the transfer function.

1
H (8) = 82 + "28 +1

Step 2: The prewared transfer function is obtained using.

H(8) = H (8) 18= 8/n1

4
- 82 + 2 ..J28+ 4
Step 3: Applying the bilinear transformation to get H (Z) yields,
"
H (Z) = H (8) Is=1T (1+- Z-l)
(1
Z-l)
4
= 2
2 (1 - Z-l)+ 2 {2 2 (1 - Z-l) +4
[ (1 + Z- 1) ] [ (1 + Z- 1) ] .
1+2Z-1+Z-2
3.144 + 0.586 Z- 2

- 0.318 + 0.636 Z-1 + 0.318 Z- 2

- 1+ 0.186Z-z
This is the required digital filter transfen.function.

In the similar way, other types of digital IIR filters (highpass, bandpass, and
bandstop) are designed using required frequency transformation formulas. It is to

note that the factor ~ gets cancelled in the above procedure through steps 2 and 3.
That is, in,

H (8) = H (8) Is= (2/T) t~ (0)/2)

"
and
H (Z) = H (8) Is=1T (1+Z-1)
- Z-l)
(1

. That is why we assumed T = 1 when it was not given. Therefore, for simplicity.

We can avoid this factor ~ in steps 2 and 3.

Example 5.7: Design a first-order digital Butterworth highpass filter which is

equivalent to an analog filter with cutoff frequency 1 KHz at a sampling rate of
104 sps. Use Bilinear transformation.
Solution:

The first-order normalized transfer function is given by,

1
H (8) = 8 + 1

Using lowpass to highpass transformation, we have,

 HI (S)= H(S) IS=I/S

S
-S+l

Prewarped cutoff frequency is given by,

nc = tan ( 2 7t (100~) (10- 4) )

= tan ( ;~ J= 0.325
(Note: Digital frequency OJ= n T, wheren = 2 7tf).
Therefore, the prewarped transfer function is given by,
H(S ) = H 1 (S) IIS =S/O.325

=[ S :1
1 =S/O.325
S
- S + 0.325

The required digital transfer function is obtained through bilinear transformation as,

H (Z) = H(S) Is= 11 -+ Z-l

Z-l

1- Z- 1
1+Z-1
- -1-Z-1 + 0.325
1 +Z- 1

1-Z-1
- (1 - Z- 1) + 0.325 (1 + Z- 1)

1-Z-1
- 1.325 - 0.675 Z- 1

- 0.755- 0.755Z-1
- 1 - 0.509Z-1
This transfer function is in a suitable form for realization.
5.2.4. Digital - to - Digital Transformation
In Section 5.3.1, we have seen analog-to-analog transformation to obtain the
non-normalized lowpass, highpass, bandpass and bandstop filters from the normalized
lowpass prototype filter transfer function. Similarly, a set of digital transformation
can be obtained that take a digital prototype lowpass filter and turn it into a digital
highpass, bandpass, bandstop or another digital iowpass filter. These transformations
are given below.

Lowpass to Lowpass

Z-l Z-l-a
1- aZ-1

where a = sin [(OOp- OOn)/2]

sin [(OOp+ OOn)/2]

OOp- passband edge frequency of prototype.

OOn- passband edge frequency of transformed (new)

Lowpass to Highpass

Z-1-7- Z-I+a
l+ar1

where a=- cos [(00n + OOp)/2]

cos [(OOn- OOp)/2]

Lowpass to Bandpass

Z- 2 - 2.a K Z- 1+-K - 1
Z-1 -7 I~+l
')
'K+l
K
-= !.Z- 2- ~ K Z- 1+ 1
K+ 1 K+ 1

where a = cos [(002+ Wl)/2]

cos [(Wz -
(01)/2]

K= cot [(0)<2 Wl)/2] - tan ( wp/2 )

Note: wI and Wz are the lower and upper passband edge frequencies of transformed
(bandpass) filter.
Lowpass to Bandstop

Z-2_~Z-1+1-K
Z-1-. l+K l+K
1-
l+K
K z- 2- ~z-
l+K
1+ 1

h
were a- cos [(~ + (01)/2]
cos [(002- (01)/2]

K= tan [(002- (01)/2] tan (~ )

Therefore, there are generally two approaches to obtain the desired digital filter
using digitizing procedure for the given specification. The steps involved in these
approaches are given below in block diagramatic forms.

Perform Desired
1. I
I
Design a
prototype
I analog-to-analog
Digitize the
resultant analog digital
I Lowpass filter I transformation filter filter.
~ of order N. I

2. i Design a
prototype
i Lowpass filter
of order N.
- Digitize to
obtain digital
prototype
- Perform
digital-to-digital
transformation. -
Desired
digital
filter.
Lovlpass filter.

The frequency band transformation can be performed either in the analog

domain or in the digital domain. But depending on the types of filter we must choose
the appropriate domain for transformation. AI3we have seen that there are severe
aliasing problem with designing highpass and bandstop filters through the impulse
invariant method, it is better to perform mapping from an analog lowpass filter into
a digital lowpass filters and then to perform the frequency transformation in digital
domain to avoid aliasing problem. In the case of bilinear transformation, where there
is no aliasing problem, it does not matter whether frequency transformation is
performed in the analog or in the digital domain. In fact, in this case only, the above
two approaches result in identical digital filters.

5.2.5. Comparision of FIR and IIR Filters

AI3we have seen, there are number of design techniques to design both FIR
and IIR filters. Each one has its own merits and demerit,;. Therefore, comparision
of these filters in strict sense to give a precision answer as what is best, is a difficult
task. However, a general comparision between FIR and IIR filters can be carried out -,
in the following manner.
 1. Linear phase FIR filter design is easy. But IIR filters are not linear phase
filters.
2. Finite word length effects, like coefficients inaccuracy error, round off noise,
etc are severe with IIR filters, but is not so severe with FIR filters because
of non-recursive realizations.

3. There are closed form formulas to design IIR filters. But the computations
involved with FIR filters are iterative and lengthy.
4. The magnitude response of IIR filters is better (sharp cutoff) than that of
FIR filters of the same order. That is, FIR filters requires more coefficients
for sharp cutoff which needs more processing time and storage.
5. Further, the inefficient direct computations involved with FIR filters can be
significantly improved using fast convolution techniques which employ the
FFT.
6. There are no zero-input limit cycle oscillations with non-recursive FIR filters,
since these structures have no feed back.
In economical point of view, the terms hardware complexity, chip area, and
computational speed are directly related to the order of the filter required to meet
a given specifications. If we put aside phase consideration, it is generally true that
a given magnitude specification can be met most efficiently with an IIR filter.

Review Questions:
1. Give the magnitude response of lowpass Butterworth filter and show the effect
of the order N on this response.
2. Give any two properties of chebyshev filters.
3. Distinguish between the frequency response of chebyshev type I and II filters.
4. Mention the most general form of the Z-transform of llR filter.
5. Why do we go for analog approximations to design a digital filter?
6. What do you mean by backward difference in the mapping of differentials?
7. Why impulse invariance method is not preferred in the design of highpass
and bandstop IIR filters?
8. Using bilinear transformation, What is the image of S = ~ 11/2in the Z-plane?
9. Why do we need prewarping in the design procedure using bilinear
transformation?
10. Mention the disadvantages of bilinear transformation technique.