OIes a butter

asbutterfly line diagram.

oWn.
utterfly, hence it is 8-point DIT-FFT using radix-2 method requires 3 stage
computation i.e., N = 8 23
s the 8-point FFT calculation, the
8ie, N = 8 length of the sequence In DIT-FFT algorithm, the input sequence is arranged
in bit reversal order and the output obtained will be in normal
But, Ne order.

The required twiddle factors can be calculated as,

2 tk

Here N 8,
w e
For k 0, w*=e = 1

e'i
Fork- 1, - -

Ifk =0 ~ W=e
com-/i
If k 1 = w =T) 0.707-j0.707)

For k 2, w? =
=e?
= cos

sin-4
=0.707-j (0.707) 3
Fork-3, w= =
e
k=2 W-
co:2-/sin
=-0.707-0.707)
Twiddle factors,

1 , w 0.707-(O.707)
k=3 wi=
wJw-0.707-(0.707)

c-/sin Bit Reversal Order Normal Order

x(0) 2 x(0)=2
-0.707-j0.707)
x(4)= 1 x(1)=2
In the figure, the input of the DIT-FFT is bit reversal
x(2) =2 x(2)=2
der and the outputisin normal order.
2,2, x(3) =2
x{n)={2,
An 8-point sequence is glven byDFT
x(6) =1
of k(n) by
2, 1, 1, 1,,1}. Compute 8-point x(1)= 2 x(4)= 1
radix-2 DIT-FFT.
x(5)= 1
Q5(a))
Ans (Model Paper-1, Q5 May-17, (R13),
x(6) = 1
(3)2
The given sequence 1s,
x(7)- 1
x(7)1
x(n) =
{2, 2, 2, 2, 1, 1, 1, 1}
Table
 Flow DIaTAL
graph for N-point DET using 1T
Flgure
The DFT of the given sequence x(») using 8-point radix-2 DIT-FFT is,
xk(12,1-/2.414), 0,1-40414),0, 1+/(0.414), 0, 1+/J2.414)!
 P R O C E S S I N G
INTU-HYDER
RABAD
W N T U - H Y D E

2.54 SIGNAL
(3).
DIGIT
DIGITAL
as
shown
in figure
Output normal
Using DIT-FFT, the output of system i.e., IFFT OT computed

order
Input bit reversal 4oN
order
Y(0) 12- 12

12=y(1)N
12
Y)-0

20 y(2)N
Y0)=-4 W

12-y (3)N
Y3)-4 W w
Figure (3): DFT Sequence Y(k)
DFT of r*(k)= [4, 12, 20,
12]
Substituting the above values in equation (2), we get,
in) -4,12, 20,12*
= [1,3,5, 3]

Thus, the output of system is,

n) =[1, 3, 5, 3]
Q52. Compare DIT-FFT and DIF-FFT algorithms in calculating DFT of a sequence.
Ans:
The comparison between DIT and DIF-FFT algorithms is mentioned below.
DIT-FFT DIF-FFT
1.In this algorithm, time domain sequence is 1. In this algorithm, frequency domain sequence is decimat-|
decimated. ed.
2. Number of input samples decimated is, 2. Number ofinput samples decimated is,
N 2 N 2.
3. Number of stages required for computation is 3. Number of stages required for computation is, i log,N=

,i=logN
Input sequence is in normal order and output 4. Input sequence is in bit reversal order and output sequence
sequence is in bit reversal order. is in nofmal order.
5. Complex additions required, N log,N 5. Complex additions required, Nlog,N
N
6.
Complex multiplicationsrequired,logN 6.
Complex multiplications requiredlog N
7. Flow graph for two stage DIT-FFT algorithm is 7. Flow
shown in figure.
graph for two stage DIF-FFT algorithm is shown in
figure.
-A+BWN A
A+B

B - A - BW
B -(A - B)WN
WN WN
Figure
Figure
 PART-A
SHORT QUESTIONS WITH SOLUTIONS
01. Define liIR filter and write its features.
Anst

Inhnite impulse Response (IIR) filter is a type of digital filter, which is obtained by considering all infinite samples of
impulse response.

Features

1. Physically realizable IIR filters have no linear phase.

2. The lIR filter specifications possess desired characteristics
for magnitude response,
Q2. What are the properties of Butterworth Low pass filter?
Ans: (Model Paper-1, Q1(e) | April-18, (R13), Q1(e))
The characteristics of Butterworth low pass filter are mentioned below,
The magnitude response of this filter depends on 2, ie., as '2' increases, it response decreases monotonically,
2. The transition band is high.
3. The poles lie on a circle in the s-plane.
4. The order of the filter is given as,

100.1-1
N2
V 100.1-1

5. The magnitude response consists of flat pass-band and flat stop band.

6. It has non-linear phase response.

7, It is lightly damped with damping coefficient of a = 1.414

8. t has poor roll-off rate.

Q3.
23. Write the expressions for order of butterworth and chebyshev filters.

Ans
The order of Butterworth filter is given as,
/100la
lo8 100.Iap-1
N2
log
 The order of Chebyshev filter is given as,

cosh'
100.1, 11
N2 V 1001ap-L
cosh

Where.
- Analog filter stop band frequency
2 - Analog filter pass band frequency
a,-Stop band attenuation
a - Passband attenuation.
Q4. What are the advantages of Butterworth flter?
(Model Paper4, Q1()| May-16, (R13), 11)
Anss
The advantages of Butterworth filter are,
1. It has more transition band than the Chebyshev filter.
less number of parameters to be calculated.
2. The design of Butterworth filter takes
3. t doesn't have ripples in stop band unlike Chebyshev filter.

4. It contains less ringing in step response.

filter.
5. The phase response of Butterworth filter is more linear than Chebyshev
is better than Chebyshev filter.
6. For unit step impulse response, both phase delay and group of Butterworth filter
05. Prove that physically realizable and stable lIR filters cannot have linear phase.
(Model Paper-il, Q1(0) May-17, (R13), a1)
Ans:
The filter has to satisfy the following conditions to have a linear phase.

i) m)=hHN-1 - )
Where,
N-Length ofthe filter.
For each and every pole which lie in side the unit circle ofzplane, the filter should have mior image of the pole outside
the unit circle.

But, satisfying these two conditions leads to unstability in the fiter. Thus, physically realizable and stable IIR flters can
not have a linear phase.
. What is step invariance method of lIR filter design?

Step invariance method is one of the techniques used to design an Infinite Impulse Response (UR) filter from an equivalent
log filter. In this method, linear frequency response of analog flter iS converted into its equivalent digital format by mapping
filter.
step response to the step response of digital
Discuss the stabilty of the impulse invariant mapping technlque.
(Model Paper-, Q1(0) April-18, (R13), Q1(D)
The transfer function of an IIR digital filter is expressed as,

k=0
H)=
1-24*
e fnction is stable, only when the poles of the filter lie inside the unit circle of a z-plane.
 PART-B3
WITH SOLUTIONS
QUESTIONS
ESSAY CHEBYSHEV
BUTTERWORTH AND
APPROXIMATIONS
ANALOG FILTER
a using Butterworth approximation technique,
D I S c u s s in detail the procedure of designina an
analog filter

Ans:
mentioned below.
technique is
Butterworth approximation
n i n g procedure of an analog filter using
1. Determine the order of the filter N* from the given specifications.
2. Round off the order to highest integer in sequence.
3. Evaluate the transfer function H(s) for 2 = I rad/sec for the value of .

4 Determine the cut-off frequency

'2,'.
by replacings by s/2. in H(s).
y,derermine the transfer function H(5) for the obtained value ofcut-offfrequency 2,
5.
c design procedure is understood by considering an example as a low pass Butterworth filter having a 3-dB cut-off

frequency of 1.5 kHz and an attenuation of 40 dB at 3 kHz.

i.e., Passband or
cutofffrequency, f=S= 1.5 kHz
Passband attenuation, o, = 3 dB

Stopband frequency., =3 kHz

Stopband attenuation, a, = 40 dB

Thus, the corresponding ana>og frequencies are obtained as,

T , d and T, = o
As,
o,2,and o,-2,
( 2 1 ) (3 kHz) = 6n kHz = 6000r rad sec-

( 2 7 ) (1.5 kHz) =3000m rad sec-

Considering T=1, we have,
, 6000n rad secl and
2, 3000n rad sec
Then, the expression for order ofa lowpass Butterworth filter is given by,
Nog(/s)

Where,
V100las1= /10*1 99:.995
E=/10.1ap -1 =/10.3-1 =0.998
99.995 )
N2
log 0.9988
6000 T
iog3000
N26.65
.
gtal Filtersi
The root% o1 the
normalized (i.e., O» 1) 3.5
Buttervworth filter are,
Where, 4,+
since, N7,
for k-1, 2,. N
km1,2,3, 4, 5, 6,7
The different values of
, for k- I to 7 are,

1
Then, the corresponding roots are obtained as,

cosjsin 4-02+097
S e l 4 .

cos cos jsin -0.62 +j0.78

s j6)=co --09+/043
s e=cos(7) +jsin(r)=-1
- 0)-con)si --09-04
s= en) = cos =-0.62-j0.78
107
s,
ell0un)=cos
jsin --0.22-0.97.

function is obtained as,

and the denominator of the transfer
= -5)-s6-s) 6-s)s)-s)-s)

(s + 1) ( +0.9 +j0.43) (s +0.62+j0.78) (6 +0.22+j0.97)

+ 0.22-j0.97) ( +0.62-j0.78) (s +0.9 -j043)
=
0 in the above expression as,
function is obtained by substituting s
The numerator of the transfer
4+0.78)(0.93 +0.43*).1
(0.223 +0.97) (0.622
0.998
is written as,
for transfer function
Hence, the expression +j0.78)(+0.22+J0.97)
-j0.43)6+1)+0.9 +j0.43)(6+0.62
H)Ts+0.22-0.97)g+0.62- J0.78)(s +0.9
of the required filter is obtained as,
transfer function
actual
Substituting s
= (s/3000r) the (30001
+ 2700 j1290n)(s + 2700TT + j1290)
j23407) (s
-

+ 1860% +

j291 0T) (s
660% + j29101)s
* 1860T- J2s40T)(G
n010T)(s + 660% +
H) (s + 3000n)(s
+ 660-
j2910)(s

ENGINEERING
STUDENT SIA GROUP
IIlnM MONE 10URNAL FOR
 DIGITAL SIGNAL PROCESSING |JNTU-HYDERABAn BADI
3.6 approximation techni
filter using Chebyshev ique.
Discuss in detail the procedure
of designing an analog Model Paper-4, Q6(a)
Ans: mentioned below.
technique is
Chebyshev approximation
ne procedure of an analog filter, using
design
.
Determine the order of the filter N from the given specifications.
2. Round off the order to highest integer in sequence.
below,
Determine a and b values, by using the formulae mentioned

Where,
a-Minor axis of the ellipse
b- Major axis of
the ellipse.
=E+VE+T
E=1001=1
-Passband frequency
a-Maximum allowable attenuation in the passband.
4. Compute the poles of Chebyshev filter by using the respective formula. In Chebyshev the poles lie on an ellipse.

a cos o+jbsine, k=1, 2,.., N

Where,-+2 k=1,2,..N
5. Obtain the denominator polynomial D(s)
by using the poles,
Where,
NG)
6.
HO)D(
The transfer function numerator depends on 'N value.
) If the value of Nis odd, then the
numerator of transfer function is obtained by replacing 's° by 0' value in D(s)
(ii) Ifthe value of "N is even, then
.

replace 's' by '0" value in D) and divide the output

A this obtained value is the
of D($) by V1+E.
same as the numerator. The
design procedure can be understood better by
example or Chebyshev filter having a 3dB cut off frequency of IkHz and an considering an
attenuation of 16dB at 2kHz.
ie., Passband or cut off frequency,
f=f=1kHz
Passband attenuation, a,= 3dB
Stopband frequency,f =2kHz
Stop band attenuation, a40dB
Thus, the corresponding analog frequencies are obtained as,
T=o,and Tao
As, ,=2n, o,=21,
(27) (1000 Hz)=2000m rad/sec
o(2n) (2000 Hz)= 4000n rad/sec
Considering T= I we have,
=4000 rad/sec and
2000r rad/sec
T-3 (IR Digital Filters)
NIT
Then, the expression for order of 3.
chebyshev filter is given by,
cos
os "h- 100.-1
V 10 cosh/100%-1
N2 0.lap-1 cosh 101 = 1.91
cos h- cosh(400T

N=2
Here, the value ofNis even, the
oscillartory curve starts from eE7
e=(10-1)s=(103-1)=1
H =el+ V1+e =2.414
The value of a and b are,

- 200m2.414}(2.414)
2 910
b -2000241424142197
The poles are given by.
sacoso,+jb sino where, k=1,2
(2k-1)
+ 2 N T , k =1,2and N=2
- 135

+ -225°

sino,=(-643.46T +j 154r)
s acos +jb
1554
+jb sin0, =643.46T-j
s a cos
D) (6*643.467j]54T)6 t 643.467
= +j1554a)
The denominator of HG)
=

+643.46 7R+ (1554H-

43.46m+(1554
T
(1414.38 *n*
(1414.38
The numerator of H()
=
Ns)= T1+E

function is,
Hence, the transfer (1414.38
N
HG)p(s)+1287ns +(1682 r
a n analog Butterworth filter
and give ts pole locations.
of an
aracteristics of
characteristics
Chebyshev filters.
digltal Ghebyshev
14. Discuss
Discuss magnitude locations
magnitude cnaions for the algital
pole
discuss aboutthe
 FILTERS FROM
3.2 DESIGN OF IIR DIGITAL
IMPULSE
ANALOG FILTERS, STEP AND
INVARIANT TECHNIQUES, BILINEAR
TRANSFORMATION METHOD, SPECTRAL
TRANSFORMATIONS

Q21. What is an lIR digital filter?

Ans:

If all the infinite samples of impulse response are

considered for désigning a filter, then the filter obtained is known
as Infinite Impulse Response (R) filter. IIR filters are recursive
type i.e., present output depends on present, past inputs and past
outputs. The differential equation ofthe IIR system is given by,

n)24.»n -k)+2B,x(n -)
k=1 k = 0

Applying Z-transform, we get,

Ye-Ao+2ax kl k=0

re-4 -
k=0
X

Y(z)
H)X
2B2*
H(2)=k0
1-A k=1
Where, H(z)= Transfer fiunat
 ages and disadvantages of lfhn) is the unit s a n p . filter
digital iiters over analog filters? sample response) ofdigital
then the impulse response (unit
is given by
dranages
es ooff Digital Filters Over Analog Filters hn)=hnT)
tinlike analog filters, variation in component values with =r".ul), a>0
i.e., if h ()
fime and temperature do not effect the performance of
Then, hn)- un), a>0
system digital filters.
in
shown in
represented as
Digital filters have high precision than analog filters. This response is graphically

These filters are less expensive compared to analog figure below:

filters. h,()
Digital filters are programmable whereas, analog filters
are not programmable.

These filters can handle low frequency signals easily

whereas, it is difficult for analog filters to process low
frequency signals.

odcadVantages of Digital Filters Over and Analog Filters

Digital filters consume high power when compared
to

analog filters.
Figure
constraints
complex to design than analog the various realizability
These filters are more 024. What are
function of an lIR digital
filters. Imposed on transfer
for ilter?
3. Digital require longer time duration
filters
filters.
development when compared to analog Anss
filters cannot process very
Unlike analog filters, digital of the IIR system is given by,
4. The differential equation
highfrequencysignals. in time
023. Describe digital IR filter
characterization
in-Ava-b+axn-
k =0
domain.
Ans of
is an infinite duration k=0
A digital filter, whose response of
. H) =
IIR filter. The response
unit samples is referred to as digital
as,
1-4
digital IR filter can be expressed
Transfer function of IIR system.
Where, H()
=

hn)=0 for n <0

#0 forn >0 constraints imposed on transfer
The various realizability
unit sample response and stability i.e.,
Thus, a digital IIR
filter produces
described by the function of an IIR digital filter are causality
filter should be causal and
the transfer function of IR digital
are
These filters
the interval of 0 to
oo.
conditions H(z) should satisfy the
a
dúference equation, obtainaboveo
stable. Thus, to obtain above

n-b+b,am-k) following requirements,

n)=-24n-i)+
=0
1. Thesystemfunction ofIR digitalfilter must bearational
and input should be real
are the output function ofz and it's coefficients
and x{n)
Where, y(n)
the unit circle of
tspectively. using recursive All the poles of H(2) should lie with in
filters can be
easily analysed
Digital IIR IR flter depends on the z-plane.
of the digital
tysten and hence, the output and outputs as shown in equation
ens 3. The number of poles of H{) should be greater than or
inputs the analog
sent inputs, past obtained from
equal to number of zeros of H().
are generally

e digital lIR fiters

R Alter.
E 100RNAL FOR
ENOINEERING STUDENTS -SIA GROUP
 9 n e d Tom

analog filters.
May-16, (R13), Q6(b)
(or)
Discuss IR filter design using bilinear
transformation.
Refer Only BilinearTransformarionMethod
Ans
Impulse Invariant Method

For Answer Refer Unit-IlI, Q26

Bilinear Transformation Method: Bilinear transformation
method is used to design highpass and bandstop filters. To avoid
aliasingof frequency components, this technique transforms
All the
thej2-axis into the unit circle in the z-plane only
once.

into the lef half

poles of stable analog filter are transformed
filter results from the
of the s-plane. Therefore, a stable digital
transformation of a stable analog filter.
function is,
Consider the analog filter whose transfer
(1)
H) 3+A
Y(s) B

sY)+AY)= = BX)

inverse Laplace transform,

we get,
Applying

dy()+Ay()= Bx(?) (22)

dt
SIA GROUP
 DIGITAL SIGNALI AD
By integrating equation (2), we get,
d+A
(7-1)T y)dt =B r()dt
(n-1)T (-1)T

bOl-)r tA y()dt = B
x()dt
(-1)T (-1

ban-vtn -1)]7]+A odt-Bx0d

x(nd
-1)T» -1)T

By using trapezoidal rule the above expression can be written as,

nT)-{n-1)r]+42DMaT)+n-1)T]= xtT) +x{(n - 17]

For discrete time
system, the above equation can be written as,
r)-yln- 1)] + n) +{n-1)=2n)+x(n-1)]
Applying Z-transform, we get,
Y)-r'Y) 4(r)+rY)- x)+X
+

X[1 +r]

BT- B

(3)
H)=

From equations (1) and (3), we get,

using Bilinear Transformation

Design Steps of IIR Filter
IIR filter using bilinear transformation.
The following are the design steps
frequencies(s2), from the given specification,
Determine the pre warping analog

Where,
a-tan2
transfer function H(s) of the' analog filter.
Determine the
filter is to be chosen.
of the digital
The sampling rate
's' in H()
Substitute the values

Where,

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