Unit I Signals and system

PART A
1.

Define Signal.
A Signal is defined as any physical quantity that varies with time, space or any other
independent variables.

2.

Define a system.
A System is a physical device (i.e., hardware) or algorithm (i.e., software) that performs an
operation on the signal.

3. What are the steps involved in digital signal processing?

Converting the analog signal to digital signal, this is performed by A/D converter

Processing Digital signal by digital system.

Converting the digital signal to analog signal, this is performed by D/A converter.
4.

5.

6.

Give some applications of DSP?

Speech processing Speech compression & decompression for voice storage system

Communication Elimination of noise by filtering and echo cancellation.

Bio-Medical Spectrum analysis of ECG,EEG etc.
Write the classifications of DT Signals.

Continuous & Discrete signals

Energy & Power signals

Periodic & Aperiodic signals

Even & Odd signals

Deterministic & Random signals
What is an Energy and Power signal?
MAY 2012
Energy signal: A finite energy signal is periodic sequence, which has a finite energy but zero
average power.
Power signal: An Infinite energy signal with finite average power is called a power signal.

7. What is Discrete Time Systems?

The function of discrete time systems is to process a given input sequence to generate output
sequence. In practical discrete time systems, all signals are digital signals, and operations on such
signals also lead to digital signals. Such discrete time systems are called digital filter.
8.

Write the Various classifications of Discrete-Time systems.

Linear & Non linear system

Causal & Non Causal system

Time variant & Time invariant system

Stable & Un stable system

Static & Dynamic system

FIR & IIR system

9.

Define Linear system

A system is said to be linear system if it satisfies Super position principle. Let us consider
x1(n) & x2(n) be the two input sequences & y1(n) & y2(n) are the responses respectively

  +
  = 
  +

 

10. Define Static & Dynamic systems

When the output of the system depends only upon the present input sample, then it is called
static system, otherwise if the system depends past values of input then it is called dynamic system
11. Define causal system.
When the output of the system depends only upon the present and past input sample, then it is
called causal system, otherwise if the system depends on future values of input then it is called noncausal system
12. What are FIR and IIR systems?
The impulse response of a system consist of infinite number of samples are called IIR system
& the impulse response of a system consist of finite number of samples are called FIR system.
13. What are the basic elements used to construct the block diagram of discrete time system?
The basic elements used to construct the block diagram of discrete time Systems are Adder,
Constant multiplier &Unit delay element.
14. What is ROC in Z-Transform?
The values of z for which z transform converges is called region of convergence (ROC).
The z-transform has an infinite power series; hence it is necessary to mention the ROC along with ztransform.
15. List the properties of Z-Transform.

Linearity property

Time Shifting property

Frequency shift or Frequency translation

Time reversal property

Scaling property

Differentiation property

Convolution property

Parsevals theorem

Initial value theorem

Final value theorem
16. What are the different methods of evaluating inverse z-transform?

Partial fraction expansion

Long division method

Contour integration (Residue method)
17. Define sampling theorem.
MAY2011/NOV2013
A continuous time signal can be represented in its samples and recovered back if the
sampling frequency Fs > 2f. Here Fs is the sampling frequency and f is the maximum frequency
present in the signal.
18. What are the properties of convolution?
NOV2013
1. Commutative property x(n) * h(n) = h(n) * x(n)
2. Associative property [x(n) * h1(n)]*h2(n) = x(n)*[h1(n) * h2(n)]
3. Distributive property x(n) *[ h1(n)+h2(n)] = [x(n)*h1(n)]+[x(n) * h2(n)]

19. Why the result of circular and linear convolution is not same?
Circular convolution contains same number of samples as that of x (n) and h (n), while in linear
convolution, number of samples in the result (N) are,
N=L+M-1 Where L= Number of samples in x (n) M=Number of samples in h (n)
20. What is meant by aliasing? How it can be avoided?
Nov2012
When the sampling frequency is less than twice of the highest frequency content of the signal,
then the aliasing is frequency domain takes place. In aliasing, the high frequencies of the signal mix
with lower frequencies and create distortion in frequency spectrum.

Part B
1. State and Explain sampling theorem.

NOV 2010/NOV 2013

 It is a process of converting a continuous time signal to discrete time Signal. The continuous time
signal is sampled at regular interval.
 The sampling interval is defined as time interval between two successive samples. It is also called
as sampling time.

Where,

 =

  Sampling Rate

  Sampling Time



Sampling Theorem states that a band limited signal with highest frequency component (fm)
can be determined from its samples. If the sampling frequency is greater than or equal to twice the
maximum frequency of the signal,
 2

If  < 2 , Aliasing problem will occurs.

What is Aliasing effect?

Aliasing is a problem due to interference of information between two band of frequencies.

To avoid Aliasing effect, using sampling theorem, sampling continuous time signal at high
rate.
Proof of Sampling Theorem:
Let x(t) be the continuous time signal(band limited) to fm. the sampling function samples the
signal regularly at the rate of fs samples per seconds.
x(t) is sampled by  
  function.

Impulse function is given as,

  =   
 



After sampling, we get

  =   
 
 



Fourier transform of equation 1 & 2, we get

 =     



 =     



Equation 4 can be written as

  =   0   +     



  =   +     



Fourier Transform of x(t):

If we take Fourier transform of continuous time x(t), we get
Xf = $ xt e()* dt

,

For discrete put t=nTs

  =  
 e()-./



fs = 2f

  =   +     



Rearranging,

1
 =      


 =
If frequency lies between f to +f
Substitute equation 7 in 9

1
 
 



1
 567
7 8 ,
 = 1   2 3 4





  +

Reconstruction of x(t):
x(t) can be represented from 10 by putting fs = 2f.
1
 567
7 8
 =
1  2 34
2
2



Taking Inverse Fourier Transform(IFT),

7

)1
n
xt = $ ;  x < > e() ? e()* df
2f
2f

-

7

n 1
xt =  x < > $ e()<*)> df
2f 2f

-

7

Interchanging order of summation and integration,

n sin 2ft n
xt =  x < > @
C
2f
2ft n

-

Since,
sin DE =

FG HE
HE

sin 2ft n
= sin C2ft n
2ft n

n
xt =  x < > sin C2ft n ,
2f

-

< n <

This is known as interpolation formula, by expanding 14, we get

x(t)=x(0) sin C2ft + x < ) > sin C2ft 1 +


2. List various properties of z-transform.

1.) Linearity property
2.) Time shifting Property
3.) Time reversal Property
4.) Scaling Property
5.) Differentiation Property
6.) Convolution Property
7.) Parsevals Theorem
8.) Initial Value Theorem
9.) Final Value Theorem
1.) Linearity Property:
Let  ,    Discrete sequence
M

   O
M

Then,

   O
M

Proof:

   +      O +   O

By definition of Z transform,
Xz =  xnz 



z   +    =     +   z 



=    z - +    z 





=   



z -

z   +    =  X +  X 

2.) Time Shifting Property:

Let  ,    Discrete sequence
M

 O
M

 Q OR S

+    z 



Proof:

zxn =  xnz 



zxn k =  xn kz 

Put, n k = m
n=m+k
When  = , V =
 = , V =
V WXG4F XYV Y +

zxn k =  xmz [\]



=  xmz [ z ]

]

zxn k = z

3.) Time Reversal Property:

Let  ,    Discrete sequence

Xz

 O
M

 O  

Proof:

zxn =  xnz 



Put, n = m
m = n
When  = , V =
 = , V =
V WXG4F XYV Y +

zxn =  xmz [



=  xmz  [


zxn = Xz  

4.) Scaling Property:

Let  ,    Discrete sequence
M

 O
M
R
   < >


Proof:

za xn =  a- xnz 



=  xna z

z =  xn < >

a

_

za xn = X < >

-

5.) Differentiation Property:

Let  ,    Discrete sequence
M

 O
M

Proof:

 R ab R
a

zxn = Xz =  xnz 



c
c
Xz =
;  xnz - ?
cR
cR
= 





c
xnz - 
cR

=  xnnz -


=  xnnz - z 


c
1
Xz =  nxnz cR
z



c
z Xz =  nxnz cR



6.) Convolution Property:

Let  ,    Discrete sequence
M

   O
M

   O

Then,

R    =  O  O

Proof:

    =   Q   Q

]

R    =  1   Q   Q8 z 

 ]

Changing order of summation,

R    =   Q    Q z - z ] z \] 

S

-

=   Qz ]    Q z -]

S

-

= X z 1    Q z -] 8

Put, n k = m
n=m+k
When  = , V =
 = , V =
V WXG4F XYV Y +

-

R    = X z 1   V z [ 8

[,

R    = X z X  z

7.) Parsevals Theorem:

Let  ,    Discrete sequence
M

   O
M

Then,

   O
,

     =

,

1
1
f  R  2 3 R  cR
2He
R

8.) Initial Value Theorem:

Let  ,    Discrete sequence
4,

 O

0 = lim  O

b,

Proof:

zxn = Xz =  xnz ,

Xz = x0z j + x1z  + x2z  +

Apply the limit R

lim Xz = lim x0z j + lim x1z  + lim x2z  +
b,

9.) Final Value Theorem:

Then,

b,

If  O

b,

lim Xz = 0

b,

b,

lim  = lim 1 R  O

,

|b|

3. Check whether the following systems are static or dynamic, linear or non-linear, time variant
or invariant, causal or non-causal, stable or unstable.
Nov/Dec 2013
i.
y(n) = Cos[x(n)]
ii.
y(n) = x(-n+2)
iii. y(n) = x(2n)
iv.
y(n) = x(n). coslj (n)
Solution:
A) static: depends on present input state, dynamic:- depends on past and future input state
B) causal:- depends on present and past inputs, non-causal:- depends on future input
C) Linear:- satisfies the super position principle

  +
  = 
  +

 
Otherwise non-linear
D) Time invariant: system do not change with time
m, Q = m Q

Time variant: system change with time.

m, Q m Q

E) Stable: it produces a bounded output sequence for every bounded input sequence
Unstable: it produces a unbounded(infinite) output sequence for every bounded input sequence
i.
ii.
iii.
iv.

y(n) = Cos[x(n)] static, non linear, time invariant, causal, stable

y(n) = x(-n+2)
y(n) = x(2n)
y(n) = x(n). cosop (n)

4. Describe the classification of system.

i.
Linear & Non linear system
ii.
Causal & Non Causal system
iii. Time variant & Time invariant system
iv.
Stable & Un stable system
v.
Static & Dynamic system
vi.
FIR & IIR system
i.
ii.
iii.

static: depends on present input state, dynamic:- depends on past and future input state
causal:- depends on present and past inputs, non-causal:- depends on future input
Linear:- satisfies the super position principle

  +
  = 
  +

 
Otherwise non-linear

iv.

Time invariant: system do not change with time

m, Q = m Q
Time variant: system change with time.

v.

m, Q m Q
Stable: it produces a bounded output sequence for every bounded input sequence
Unstable: it produces a unbounded(infinite) output sequence for every bounded input
sequence

vi.

FIR:- The system is of finite duration, IIR:- The system is of infinite duration.

5. Describe the different types of digital signal representation.

There are different types of representations for signal
a) Graphical representation
b) Functional representation
c) Tabular representation
d) Sequence representation

MAY 2013

6. Perform the convolution of the signals

7. Find the inverse Z-transform of st = uwux, ROC: |Z|>3, using

uv \u

a) Residue Method
b) Convolution Method
Solution:
a) Residue Method:

NOV 2011
yz{|}~z =

Where m order of pole at Z=a

w
}w
w u  stuw
 w! u }u

~ yz{|}~z = w + vx ~

b) Convolution Method:
 = w  v 
,

 =  w . v  
,

 = w + vx ~

8. A causal system is represented y the following difference equation

MAY 2011/NOV 2012
w
w
 +  w =  +  w

Find the system transfer function H(z), unit sample response, magnitude Response and phase
function of the system.
Solution:I)

II)

Transfer function:

1
1 + O 
2
R =
1
1 + O 
4

1
1 + 2 4 5
4 5 =
1
1 + 4 4 5

Magnitude response

|| = l l

|| =
III)

Phase response:
l = tan

l = tan 2

1.25 + YFl
1.06 + 0.5YFl

GVGXm Y l

4 Y l

0.25 sin l
0.5 sin l
3 tan 2
3
1 + 0.25 cos l
1 + 0.5 cos l

9. A) Determine the Z transform and ROC of the signal.

 = ~ w
Solution: -

From the definition

,

R =  O 
,


R =    O 
,

ROC: |Z|<|

R =

1
1 O 

The ROC is now the interior of a circle having radius | |

B) Check whether the discrete time system y(n)=cos[x(n)] is

i.)
Static or dynamic
ii.)
Linear or non Linear
iii.)
Time variant or Time invariant
iv.)
Causal or non causal
v.)
Stable or unstable
Solution:
i.
ii.
iii.
iv.
v.

static: depends on present input state

Non Linear:- since doesnt satisfies the super position principle

  +
  
  +

 
Time invariant: system do not change with time
m, Q = m Q
causal:- depends on present and past inputs
Stable: it produces a bounded output sequence for every bounded input sequence

UNIT II Frequency Transformations

PART A
1. What is DFT?
NOV2013
It is a finite duration discrete frequency sequence, which is obtained by sampling one period of
Fourier transform. Sampling is done at N equally spaced points over the period extending from l =
0Y 2H.
2.

Define N point DFT.

The DFT of discrete sequence x(n) is denoted by X(K). It is given by, Here k=0,1,2N-1 Since
this summation is taken for N points, it is called as N-point DFT.
3. What is DFT of unit impulse (n)?
The DFT of unit impulse (n) is unity.
4. List the properties of DFT.

Periodicity

Linearity

Symmetry property

Circular convolution of two sequences

Time reversal of sequence

Circular Time shift of sequence

Circular Frequency shift of sequence

Circular correlation of two sequences

Multiplication of two sequences

Parsevals theorem.
5. State Linearity property of DFT.
DFT of linear combination of two or more signals is equal to the sum of linear combination of
DFT of individual signal.
6. What is circular time shift of sequence?
Shifting the sequence in time domain by 1 samples is equivalent to multiplying the sequence in
frequency domain by WNkl
7. What is the disadvantage of direct computation of DFT?
For the computation of N-point DFT, N2 complex multiplications and N[N-1] Complex additions
are required. If the value of N is large than the number of computations will go into lakhs. This proves
inefficiency of direct DFT computation.
8. What is the way to reduce number of arithmetic operations during DFT computation?
Number of arithmetic operations involved in the computation of DFT is greatly reduced by using
different FFT algorithms as follows.
1. Radix-2 Decimation in Time (DIT) algorithm.
2. Radix-2 Decimation in Frequency (DIF) algorithm.
3. Radix-4 FFT algorithm.
9. What is the computational complexity using FFT algorithm?

Complex multiplications = N/2 log2N

Complex additions = N log2N

10. How linear filtering is done using FFT?

Correlation is the basic process of doing linear filtering using FFT. The correlation is nothing but
the convolution with one of the sequence, folded. Thus, by folding the sequence h (n), we can compute
the linear filtering using FFT.
11. What is zero padding? What are its uses?
Let the sequence x (n) has a length L. If we want to find the N point DFT (N>L) of the sequence x
(n). This is known as zero padding. The uses of padding a sequence with zeros are
(i) We can get better display of the frequency spectrum.
(ii) With zero padding, the DFT can be used in linear filtering.
12. Why FFT is needed?
The direct evaluation of the DFT using the formula requires N2 complex multiplications and N
(N-1) complex additions. Thus for reasonably large values of N (inorder of 1000) direct evaluation of the
DFT requires an inordinate amount of computation. By using FFT algorithms the number of
computations can be reduced. For example, for an N-point DFT, The number of complex multiplications
required using FFT is N/2log2N. If N=16, the number of complex multiplications required for direct
evaluation of DFT is 256, whereas using DFT only 32 multiplications are required.
13. What is the speed of improvement factor in calculating 64-point DFT of a sequence using direct
computation and computation and FFT algorithms?
The number of complex multiplications required using direct computation is N2=642=4096. The
number of complex multiplications required using FFT is N/2 log2N = 64/2log264=192. Speed
improvement factor = 4096/192=21.33
14. What is FFT?
MAY2012
The fast Fourier transforms (FFT) is an algorithm used to compute the DFT. It makes use of the
Symmetry and periodically properties of twiddles factor WKN to effectively reduce the DFT computation
time. It is based on the fundamental principle of decomposing the computation of the DFT of a sequence
of length N into successively smaller discrete Fourier transforms. The FFT algorithm provides speedincrease factors, when compared with direct computation of the DFT, of approximately 64 and 205 for
256-point and 1024-point transforms, respectively.
15. How many multiplications and additions are required to compute N-point DFT using redix-2
FFT?
The number of multiplications and additions required to compute N-point DFT using redix-2 FFT
are N log2N and N/2 log2N respectively.
16. What is meant by radix-2 FFT?
The FFT algorithm is most efficient in calculating N-point DFT. If the number of output points N
can be expressed as a power of 2, that is, N=2M, where M is an integer, Then this algorithm is known as
radix-s FFT algorithm.
17. What is a decimation-in-time algorithm?
Decimation-in-time algorithm is used to calculate the DFT of a N-point Sequence. The idea is to
break the N-point sequence into two sequences, the DFTs of which can be combined to give the DFT of
the original N-point sequence. Initially the N-point sequence is divided into two N/2-point sequences
xe(n) and x0(n), which have the even and odd members of x(n) respectively. The N/2 point DFTs of these

two sequences are evaluated and combined to give the N point DFT. Similarly the N/2 point DFTs can be
expressed as a combination of N/4 point DFTs. This process is continued till we left with 2-point DFT.
This algorithm is called Decimation-in-time because the sequence x(n) is often splitted into smaller sub
sequences.
18. What are the differences and similarities between DIF and DIT algorithms?
Differences:
1. For DIT, the input is bit reversal while the output is in natural order, whereas for DIF, the
input is in natural order while the output is bit reversed.
2. The DIF butterfly is slightly different from the DIT butterfly, the difference being that the
complex multiplication takes place after the add-subtract operation in DIF.
Similarities: Both algorithms require same number of operations to compute the DFT. Both
algorithms can be done in place and both need to perform bit reversal at some place during the
computation.
19. What are the applications of FFT algorithms?
1. Linear filtering
2. Correlation
3. Spectrum analysis

NOV2012

20. What is a decimation-in-frequency algorithm?

In this the output sequence X (K) is divided into two N/2 point sequences and each N/2 point
sequences are in turn divided into two N/4 point sequences.
21. Distinguish between DFT and DTFT.
DFT
DTFT

Obtained by performing

Sampling is performed
sampling operation in both
only in time domain.

Continuous function of
the time and frequency
domains.

Discrete frequency spectrum

22. Distinguish between Fourier series and Fourier transform.

Fourier Series

Gives the harmonic content of
a periodic time function.

Discrete frequency spectrum

Fourier transform

Gives the frequency information
for an aperiodic signal.

Continuous frequency spectrum

23. How linear filtering is done using FFT?

Nov 2011
Correlation is the basic process of doing linear filtering using FFT. The correlation is nothing
but the convolution with one of the sequence, folded. Thus, by folding the sequence h (n), we can
compute the linear filtering using FFT.

Part B
1. A)Find eight point DFT of the following sequence using direct method:
{1,1,1,1,1,1,0,0}
Solution:


Q =  4 56S/ ;

MAY 2013

Q = 0,1, 1

j

For the given sequence N=8, by substituting k and N values in above equation, we get
X(0)=
X(1)=
X(2)=
X(3)=
X(4)=
X(5)=
X(6)=
X(7)=

6
-0.707- j1.707
1- j
0.707+j0.293
0
0.707- j0.293
1+ j
- 0.707+j1.707

B) State any six properties of DFT (6 marks)

1. Periodicity
2. Linearity
3. Symmetry Property
4. Circular Convolution of two sequences.
5. Time reversal of sequence.
6. Parsevals Theorem.

MAY 2013

2. A) Compute eight point DFT of the following sequence using radix 2 Decimation in time
FFT algorithm.
MAY 2011(16 mark)/MAY 2013(8 mark)
X(n)={1,-1,-1,-1,1,1,1,-1}
Solution:Butterfly diagram should be drawn
(2 marks)
The output of each stage are given below
(each stage 2 marks)
Input of I stage
Output of I stage
Output of II stage
Output of III stage
1

2j

-1.414+j3.414

-1

2-2j

-2

-2j

1.414 - j0.586

-1

-2

-2

-2

1.414+j0.586

-1

-2

2+2j

-1

-2

-1.414 j3.414

B) Discuss the use of FFT in linear filtering

MAY 2013

The overlap-save and overlap-add methods are used for filtering a long data sequence with an
FIR filter based on the use of DFT.
FFT algorithm can be used for computing DFT and IDFT.
(2 marks)
Comparison of Overlap-save and overlap-add method in filtering.
(4 marks)
3. Compute the FFT of the sequence  = v + w p w, where N=8 using DIT
algorithm.
NOV 2012
Solution: - N=8
 = v + w p
x(0)=1
x(1)=2
x(2)=5
x(3)=10
x(4)=17
x(5)=26
x(6)=37
x(7)=50
Butterfly diagram should be drawn
The output of each stage are given below
Input of I stage
Output of I stage
Output of II stage
Output of III stage
1
18
60
148
17
-16
-16+32j
-4.688+13.248j
5
42
-24
-24+32j
37
-32
-16-32j
-27.312+13.248j
2
28
88
-28
26
-24
-24+40j
-27.312-13.248j
10
60
-32
-24-32j
50
-40
-24-40j
-4.688-77.248j
4. Find DFT for {1,1,2,0,1,2,0,1} using FFT DIT butterfly algorithm.
Butterfly diagram should be drawn
The output of each stage are given below
Input of I stage
Output of I stage
1
2
1
0
2
2
0
2
1
3
2
-1
0
1
1
-1

Output of II stage
4
-2j
0
2j
4
-1+j
2
-1-j

NOV 2013

Output of III stage

8
-0.586j
-2j
3.414j
0
-3.414j
2j
0.586j

5. Discuss the properties of DFT.

1. Periodicity
2. Linearity
3. Symmetry Property
4. Circular Convolution of two sequences
5. Time Reversal of sequence
6. Circular time shift of sequence
7. Circular frequency shift of sequence
8. Circular Correlation of two sequences
9. Multiplication of two sequences
10. Parsevals Theorem

NOV 2013

1. Periodicity:
 If discrete time signal is periodic, then its DFT is also periodic.
 If x(n) is a discrete signal of length N, then
i.
x(n+N)=x(n)
ii.
X(k+N)=x(k)
Where x Input signal (time domain)
X DFT[x(n)] (frequency domain)
2. Linearity:
It satisfies superposition principle


 Q
YG


Similarly,

   Q
YG


   Q
YG
By Linearity Property,


  +
   Q +
 Q
YG
Where a,b constants
3. Symmetry Property:
If signal or sequence repeats its waveform in negative direction after N/2 number of
samples, then it is called symmetric signal or sequence.
From periodicity property,
If x(n)=x(N-n)
Then X(k)=X(N-k)
For Symmetric signal,
If

 Q
YG

Then 

 Q
YG



4. Circular Convolution of two sequences:

Circular Convolution is defined as



 V =   .  V 
j

It is denoted by
  =   

5. Time reversal of a sequence:

If

 Q
YG

Where m=0,1,2,..N-1

[ DGXXDYWYGY]


    Q +  Q

YG

Then  =    Q = Q

YG


6. Circular Time Shift of a Sequence:

 Q
YG

If

Then
 V = 4 


4 

YG
7. Circular Frequency Shift:
i.e.  V

If

Then

 Q
YG
4

Q

Q

 Q V
YG

8. Circular Correlation of two sequences:

Let

   Q
YG


   Q
YG

V
Q
R 
= 
 Q
Then circular correlation is given by


  V =      V
j

Circular correlation property is

  V =  Q Q
(i.e)


     V  Q Q

YG
j

9. Multiplication of two sequences:

Let

   Q
YG


Then

   Q
YG


1
  
 Q Q
YG

10. Parsevals Theorem:

For complex valued sequences   c  

[ DGXXDYWYGY]

   Q
YG


Then

   Q
YG





1
     =   Q  Q

j

Sj

if   =   = 

then  Q =  Q = Q
and
.   = ||
Q.  Q = |Q|

is the general form of Parsevals theorem





j

Sj

1
|| = |Q|

6. Draw the flow chart for N=8 using radix 2 DIF algorithm for finding DFT coefficient. (Nov
2010)
Decimation-in-frequency (DIF) is another important radix 2 FFT algorithm. In case of
Decimation-in-Time, the input data sequence x(n) are decimated but in case of Decimation-infrequency, the data coefficients X(k) are decimated. In this algorithm, we first divide the DFT

formula into two summations. The first term contains first

last

points.


Xk =  xnW-] ,
j

- points and the second term contains the

k = 0,1,2, , N 1






Xk =  xnW-] +  xnW-]

j




Xk = 

j

xnW-]









<-\ >]
+ x 2n + 3 W 
2
j




]
Xk =  xnW-] + W x 2n + 3 W-]
2

j

W = 1]

Since

j










Xk =  xnW-] + 1] x 2n + 3 W-]

2
j

j

Xk = @xn + 1] x 2n + 3C W-]

2

w

j

Now decimate X9K) into odd and even-indexed samples.

For even decimation,




X2k = @xn + x 2n + 3C W-] ,
2
j

k = 0,1,2, , 2 13
2





X2k = @xn + x 2n + 3C W-] ,

2

j

k = 0,1,2, , 2 13
2

v

For odd decimation,




-]\
X2k + 1 = @xn x 2n + 3C W
,
2
j




k = 0,1,2, , 2 13
2

X2k + 1 = @xn x 2n + 3C W- W-] ,

2

j

k = 0,1,2, , 2 13
2

x

The equation (2) and (3) can be redefined as




k = 0,1,2, , 2 13
2

X2k =  g n W-] ,



j

g n =  +  2 + 3
2

Where




X2k + 1 =  g  n W-] ,

j

Where



k = 0,1,2, , 2 13
2



g  n = @xn x 2n + 3C W2

Decimation in frequency stands for splitting the sequences in terms of frequency. That means we
have split output sequences into smaller subsequences. This decimation is done as follows.
First stage of decimation:
first stage of decimation as shown in fig. below.

Second stage of decimation: In the first stage of decimation we have used 4-point DFT. We
can further decimate the sequence by using 2 point DFT. The second stage of decimation is shown in
fig below.

Third stage of decimation : In the second stage of decimation we have used 2- point DFT. So
further decimation is not possible. Now we will use a butterfly structure to obtain 2-point DFT. Thus
the total flow graph of 8 point DIF-FFT is shown below.

7. Develop a Radix-2, 8-point DIT FFT algorithm

The principle of decimation-in-time (DIT) can be obtained by considering the special case of
N, an integer power of 2, i.e.

If r=2, radix-2

= X

= 2

(1)
(2)

In radix-2,N is an even integer. Let us consider computing X(k) by separating x(n) into two
N/2 point subsequences consisting of the even-numbered points in x(n) and the odd-numbered
points in x(n).


Xk =  xnW-] , k = 0,1,2, , N 1

j

x

Separate x(n) into odd-numbered points and even numbered points.

Xk =  xnW-] +  xnW-] , k = 0,1,2, , N 1
--

-

For even-numbered points, replace n=2r

For odd-numbered points, replace n=2X + 1.










j

j

Xk =  x2rW] +  x2r + 1W

\]

S =  S

6
5< >
4

6
5

4

 S = S







Xk = x2rS + W] x2r + 1S

Where,

Q = Q + W] Hk,

j

j

k = 0,1,2, , N 1






Q = x2rS
j




Q = x2r + 1S

Each term [G(k) and H(k)] in equation (5) is a

j

- point DFT of 2X and 2X + 1 respectively.

G(k) and H(k) are periodic, with period  . Therefore,

2Q + 3 = Q
2

2Q + 3 = Q
2

<S\ >


= S

Hence, equation (5) becomes,

Q = Q + W] Hk,

k = 0,1,2, , 2 13
2

S\
 2Q + 3 = 2Q + 3 + W  H 2Q + 3
2
2
2

 2Q + 3 Q W] Hk,
2

Now k ranges between 0 to < 1>

k = 0,1,2, , 2 13
2

Fig 1: Decimation in time of a length N-DFT into two lengths





DFT followed by a combined stage

Fig 2: Radix-2 Decimation-in-Time FFT algorithm for a length-8 signal

8. Draw a 8 point radius 2 FFT DIT flow graphs and obtain DFT of the following sequence
x(n)={0,1,-1,0,0,2,-2,0}

UNIT III - IIR FILTER DESIGN

PART A
1. What is bilinear transformation?
Nov2013
The bilinear transformation is conformal mapping that transforms the s-plane to z-plane. In
this mapping the imaginary axis of s-plane is mapped into the unit circle in z-plane, The left half of splane is mapped into interior of unit circle in z-plane and the right half of s-plane is mapped into
exterior of unit circle in z-plane
2. What are the characteristics of Chebyshev filter?
May 2013

The magnitude response of the chebyshev filter exhibits ripples either in pass band or in
stop band according to type

The poles of the filter lies on an ellipse
3. What is impulse invariant transformation?
May2012
The transformation of analog filter to digital filter without modifying the impulse response of
the filter is called impulse invariant transformation.
4. What is the importance of poles in filter design?
Nov 2011
The stability of a filter is related to the location of the poles. For a stable analog filter the
poles should lie on the left half of s-plane. For a stable digital filter the poles should lie inside the unit
circle in the z-plane
5.

Mention the properties of Butterworth filter?

Nov 2013

All pole design.

The poles lie on a circle in s-plane.

The magnitude response is maximally flat at the origin and monotonically decreasing function
of . The normalized magnitude response has a value of 1 / 2 at the cutoff frequency c.

Only few parameters have to be calculated to determine the transfer function.

6. What are the properties of bilinear transformation?

May 2011

The mapping for the bilinear transformation is a one-to-one mapping that is for every point Z,
there is exactly one corresponding point S, and vice-versa.

The j -axis maps on to the unit circle |z|=1,the left half of the s-plane maps to the interior of
the unit circle |z|=1 and the half of the s-plane maps on to the exterior of the unit circle |z|=1.
7. What are the various methods to design IIR filters?

Approximation of derivatives

Impulse invariance

Bilinear transformation.
8. Write the transformation equation to convert low pass filter into band stop filter? MAY 2013
2  1 Q
O 
O +
1+Q
1+Q
O 
1 Q 
2 
O
O +1
1+Q
1+Q

Where,

l + l
2
=
l l
cos 2
cos

Q = tan @

l
l + l
C tan
2
2

9. Distinguish between FIR and IIR filters.

FIR filter
IIR filter

These filters can be easily designed to
These filters do not have linear phase.
have perfectly linear phase.

FIR filters can be realized recursively
IIR filters can be realized recursively.
and non-recursively.

Greater flexibility to control the shape of
Less flexibility,usually limited to kind of
their magnitude response.
filters.

Errors due to roundoff noise are less
The roundoff noise in IIR filters are
severe in FIR filters, mainly because
more.
feedback is not used.
10. What is prewarping?
Prewarping is the method of introducing nonlinearly in frequency relationship to compensate
warping effect.
11. What are the various methods to design IIR filters?

Approximation of derivatives

Impulse invariance

Bilinear transformation.
12. What is meant by warping effect?
MAY 2011
The relation between the analog and digital frequencies in bilinear transformation is given by
2
l
= tan


2
For smaller values of l there exist linear relationship between l and . But for large values
of l the relationship is non-linear. The non-linearity introduces distortion in the frequency axis. This
is known as wrapping effect.
PART B
\j.
1. For given analog filter system function H(s)=
into digital IIR filter by means of bilinear z
\j. \

transformation. Digital filter is to have resonant frequency o = v

Nov2012/13
Ans:

From above eqn f=4

Value of o = v

Substitute s=

 M
 M\

2 A chebyshev low pass filter has the following specifications:

Nov2013/2012
(a) Order of the filter = 3
(b) Ripple in pass-band = 1 db
(c) Cut off frequency = 100 Hz
(d) Sampling frequency = 1 kHz.
Determine H(z) of the corresponding IIR digital filter using bilinear transformation technique
Ans
Oder of the filter = N= 3

Step 1:Calculation of required design specification of digital filter

Given =1 db
For normalize filter F sample = 1 Khz
Calculate sk=acosQ+jbsinQ,
where Q= +

S

Calculate =-1+(1+-2)

Calculate a = p [

b = p [

] = 0.297 and

] = 1.043

.Calculate Poles
ES =  +
6

S\6


, Q = 0,1,2

k=0, Ej =  + =
6

k=1, E =  +
6

k=2, E =  +
6

S k=a cosk+jbsink
S 0=a cos0+jbsin0,

S 1=a cos1+jbsin1,

S 2=a cos2+jbsin2,

=H

Calculate system function

3. Convert the analog filter transfer function to digital filter using Impulse Invariance methodH(S) =
{\v
{\w{\x

Ans
Using Partial Fraction

Apply Impulse invariance transformation

R = F/ F =

 b

4. Design a digital chebyshev filter for the following specifications using bilinear transformation

MAY 2012
Given:

5. Design a digital chebeshev filter for the following specifications using Impulse in variance
transformation

Ans:

6.Design a digital lowpass butterworth filter where transfer function is given by

p. z  w
z  p. x

Ans:

w\v

= p.

=3.179

w\v

p p. v

p.

= p. x wp = p. v ws = p.

=1.02

Using Impulse invariant transform w= T ,T=1 Sec

p = p. v

s= p. v

N= log(( /)/log1(s/p) =1.03 =2

Ha(s) =

 \  \

It is a LPF s = s/ c
c = p /

1/N

=0.622 rad/sec

s = s/ 0.622
 =
Using partial fraction A = 0.439 j B = -0.439 j
 =

p. x
+ p. + p. x

p. x
p. x
p. x
=

v + p. + p. x
 p. x p. x  p. x + p. x

Using impulse invariance method

{|

wz|

uw 

H(Z)= 0.2401 Z-1 /1- 1.165Z-1+0.414 Z -2

7. Design a digital butterworth loepass filter using bilinear transformation with passband and stop
band frequencies 800 rad/sec and 1800 rad/sec.The passband and stop band attenuation are -3dB and 10 dB respectively.
Ans :
p =-3dB s =-3dB wp =800 rad/sec ws =1800 rad/sec
Using Bilinear transformation



= , p = 3.2397 rad/sec s =30.12 rad/sec

N= log(( /)/log1(s/p) =1.0.491 =1

= 10j. 1 = 0.997

= 10j. 1 = 3
H(s) = 1/(S+1)
H(s) = 1/ (s/3.24)+1

S.S

S= S/c

c = p /

1/N

=3.24

Using Bilinear transformation Substitute s by

H(Z) =3.24 (1+Z-1)/2-2Z-1 +3.24(1+Z-1)

 M
 M\

UNIT IV - FIR FILTER DESIGN

PART A
1. What is frequency warping?
Nov 2011
Because of the non-linear mapping: the amplitude response of digital IIR filter is expand at
lower frequencies and compressed at higher frequencies in comparison to the analog filter.
2. What is the frequency response of Butterworth filter?
Butterworth filter has monotonically reducing frequency response

Nov 2011

3. What is Gibbs phenomenon (or Gibbs Oscillation)?

Nov 2013/2012
In FIR filter design by Fourier series method the infinite duration impulse response is
truncated to finite duration impulse response. The abrupt truncation of impulse response introduces
oscillations in the pass band and stop band. This effect is known as Gibbs phenomenon.
4. What is the condition for linear phase of a digital filter?
May 2012
The necessary and sufficient condition for linear phase characteristic in FIR filter is, the
impulse response h(n) of the system should have the symmetry property i.e.,
 =  1 , where N is the duration of the sequence.
5. What is meant by limit cycle oscillation?
May 2008, May 2011, Nov 2012
In recursive system when the input is zero or same non-zero constant value the non linearities
due to finite precision arithmetic operation may cause periodic oscillation in the output. Thus the
oscillation is called as Limit cycle.
6. What are the different types of arithmetic in digital systems?
Nov2011
There are three types of arithmetic used in digital systems. They are fixed point arithmetic,
floating point, block floating point arithmetic.
7. What is meant by fixed point number?
May2013
In fixed point number the position of a binary point is fixed. The bit to the right represent the
fractional part and those to the left is integer part.
8. What is zero input limit cycle oscillation?
May2013
When a stable IIR filter is excited by a finite input sequence, the output will ideally decay to
zero. But due to non linearities in the finite precision arithmetic operation cause periodic oscillation
to occur in the output
9. Write the equation for blackman window.
2H
4H
 = 0.42 + 0.5 cos 2
3 + 0.08 cos 2
3,
1
1

May2013
1
1
2
3  2
3
2
2

10. State the condition for a digital filter to be causal and stable.
A digital filter is causal if its impulse response h(n) = 0 for n<0 A digital filter is stable if its
impulse response is absolutely summable,
11. What are the properties of FIR filter?
i.
FIR filter is always stable.
ii.
A realizable filter can always be obtained.
iii. FIR filter has a linear phase response.

12. How phase distortion and delay distortions are introduced?

The phase distortion is introduced when the phase characteristics of a filter is not linear within the
desired frequency band. The delay distortion is introduced when the delay is not constant within the
desired frequency range.
13. Write the steps involved in FIR filter design.

Choose the desired (ideal) frequency response Hd(w).

Take inverse fourier transform of Hd(w) to get hd(n).

Convert the infinite duration hd(n) to finite duration h(n).

Take Z-transform of h(n) to get the transfer function H(z) of the FIR filter.
14. What are the advantages of FIR filters?

Linear phase FIR filter can be easily designed.

Efficient realization of FIR filter exist as both recursive and nonrecursive structures.

FIR filters realized nonrecursively are always stable.

The roundoff noise can be made small in nonrecursive realization of FIR filters.
15. What are the disadvantages of FIR filters?

The duration of impulse response should be large to realize sharp cutoff filters.

The non-integral delay can lead to problems in some signal processing applications.

PART B
1. Prove that an FIR filter has Linear phase if the unit sample response satisfies the condition
h(n)=h(N-1-n).Also discuss the symmetric and anti symmetric cases of FIR filter when N is
even.
NOV 2013
Case 1:- Symmetric Impulse Response with Even Length:
The frequency response of h(n) is
4



 =  4 5
j

If filter length is even,






4 5 =  4 5 +  4 5




j

< >


4 5 =  4 5 +  N 1 n4 5-

j

j

We know that h(n)=h(N-1-n)




j

j

4 5 =  4 5 +  n4 5-


 




5<
>
5<
>
5<
> 
5



4 = 4
 4
+  4
 j

j




 


1
5<
>
5

4 = 4
2   cos @2
3 C
2
 j




 



1
5<
>
5


4 = 4
 2 2 3 cos @2 3 lC
2
2
j




 


1
5<
>
5

4 = 4

 cos @2 3 lC
2
j




Where
= 2 < >

4 5 = 4 5<


>
 4 5

= 4 5 4 5

1
4 5 = 
 cos 2 3 l
2


+1
El = 2
3l
2

Case 2:- Anti Symmetric Impulse Response with Even Length:

The frequency response of h(n) is



4 5  =  4 5
j

If filter length is even,






4 5 =  4 5 +  4 5

j

< >


j

j

4 5 =  4 5 +  N 1 n4 5-

We know that h(n)= h(N-1-n)

j

j

4 5 =  4 5  n4 5-




 




5<
>
5<
>
5<
> 
5



4 = 4
 4
 4
 j

j



 


6

1
5<
> 5 
5
 4   2 2 3 sin @2 3 lC
4 = 4
2
2





 


6
1
5<
> 5 
5


4 = 4
4
 c cos @2 3 lC
2





Where c= 2 < >

4 5 = 4 5<


6
>
 4 5  4 5

= 4 5 4 5

1
4 5 = 
 cos 2 3 l
2


El =

H
1
2
3l
2
2

2. Prove that an FIR filter has Linear phase if the unit sample response satisfies the condition
h(n)=h(N-1-n).Also discuss the symmetric and anti symmetric cases of FIR filter when N is odd.
Case 1:- Symmetric Impulse Response with Odd Length
The frequency response of h(n) is



4 5  =  4 5
j

If filter length is odd,

4

=  4
j

5

1 5<>

+2
34
+
2



<

\
>


4 5

We know that h(n)=h(N-1-n)




4 5 =  4 5 + 2
j

1 5<>

34
+  4 5-
2
j


 





1
5<
>
5<
>
5<
> 
5




4 = 4
 4
+2
3 +  4
2
 j

j




Let <

>  = ,

 



1
5<
>
5<
>
5


4 = 4
2  4
+2
3
2 
 j








 


1
1
5<
>
5

4 = 4
 2. 2
3 cos l + 2
3
2
2 
 




 


1
1
5<
>
5

4 = 4
 2. 2
3  cos l + 2
3
2
2 
 


4 5 = 4 5<

Where 0 = <

> c  = 2. 2<





>


> 3




4 5 = 4 5<




  cos l

j


>
 4 5



= 4 5 4 5

4 5 =   cos l
j

1
El = 2
3l
2

Case 2:- Anti Symmetric Impulse Response with Odd Length:

For this type of sequence,
2
The frequency response of h(n) is
4

1
3=0
2


 =  4 5
j

If filter length is odd,

4

=  4
j

We know that h(n)= h(N-1-n)

5

1 5<>

+2
34
+
2

j

j



<

\
>


4 5

4 5 =  4 5  4 5-




 




5<
>
5<
>
5<
> 
5




4 = 4
 4
 4
 j

j




 


1
5<
> 
5
 e  2 sin @2
4 = 4
3 C
2
 j





 


6
1
5<
> 5 
5
 4   2. 2
4 = 4
3  sin l 
2
 




Where c = 2. 2<

> 3




4 5 = 4 5<


6
>
 4 5




  cos l

j

4 5 = 4

5<


>


4 5  4 5 = 4 5 4 5



4 5 =   sin l
El =



H
1
2
3l
2
2

3. Design an ideal band pass filter with a frequency response.

3H
| |
Hd 4el  = 1 for 4 l 4 
0 Y4XGF4

NOV 2011/MAY 2012

Find the values of h(n) for N=7. Find the realizable filter transfer function and magnitude
function of H4el 
Soln. Step 1. Draw the ideal desired frequency response of bandpass filter.

Form the desired frequency response, we can find that the given response is symmetric N odd
Step 2. To find

Step 3. To find h(n).

For symmetry response

Step 4. To find filter transfer function,

Step 5. To find the realizable filter transfer function

Therefore, the filter co-efficients of the causal filters are,

Step 6. To find the magnitude response of H4 5

4. Design an ideal high pass filter with a frequency response

|l| H
4
5

H 4 = 
H
0 YX |l| <
4
1 for

Find the value of h(n) for N=11 using

a. Hamming window
b. Hanning window

Sol. (a) Hamming Window

Step 1. Draw the desired frequency response of ideal highpass filter.

Step 2. To find hd(n)

We know that,

MAY 2013/MAY 2011

Step3. To find the Hamming window sequence.

Step 4. To find the filter co-efficient hd(n)

Step 5. To find the filter co-efficients using Hamming window sequence.

Step 6. To find the transfer function of the filter.

Step 7. To find transfer function of the realizable filter

The filter co-efficients of causal filters are,

(b) Hanning Window

Step 1. The filter co-efficient can be obtained from part (a), step (2) and step (5)

Step 2. To find the Hanning window sequence

The Hanning window sequence is given by

Step 3. To find the filter co-efficients using Hanning window.

The filter co-efficients using Hanning window are

Step 4. To find the transfer function of the filter.

The transfer function of the filter is given by,

Step 5. To find the transfer function of realizable filter.

5. Design an ideal low pass filter with a frequency response.

1 for l
2
2 
Hd 4el  =
H
0 YX l H
2

NOV 2011/NOV 2013

Find the values of h(n) for N=1. Find the realizable filter transfer function and magnitude
function of H4el 

Sol. Step 1. Draw the desired frequency response:

From the frequency response, we can find that the given response is a symmetrical
N odd response.
Step 2. To find
In general,

Step 3. To find h(n):

For symmetric response,

Step 4. To find the filter transfer function.

6. Design an ideal low pass filter with a frequency response.

1 for 0 |l|
6
Hd 4el  =
H
0 YX |l| H
6

Use 10 tap filter and obtain the impulse response of the desired filter

Ans. The filter co-efficients are given by :

NOV 2013/MAY 2012

7. Explain Finite word length effects in FIR filters

Parameter quantization in digital filters
The common method of quantization is Truncation and Rounding.
Truncation: - Truncation is the process of discarding all bits less significant than least significant
bits that is retained.
Rounding: - Rounding of a number of b bits is choosing the rounded results as the b bit closet to the
original number unrounded.
In the realization of FIR and IIR filters hardware or in software on a general purpose
computer, the accuracy with which filter coefficients can be specified is limited by word length of the
computer. Since the coefficients used in implementing a given filter are not exact, the poles and zeros
of system function will be different from desired poles and zeros. Consequently, we obtain a filter
having a frequency response that is different from the frequency response of the filter with
unquantized coefficients. Also it sometimes affects stability of filter.
Round off noise in multiplication
As already explained when a signal is sampled or a calculation in the computer is performed,
the results must be placed in a register or memory location of fixed bit length. Rounding the value to
the required size introduces an error in the sampling or calculation equal to the value of the lost bits,
creating a nonlinear effect. Round-off error is a characteristic of computer hardware.
Sampling/Digitization Error
There is another, different, kind of error that is a characteristic of the program or algorithm
used, independent of the hardware on which the program is executed. Many numerical algorithms
compute discrete approximations to some desired continuous quantity. For example, an integral
is evaluated numerically by computing a function at a discrete set of points, rather than at every
point. Or, a function may be evaluated by summing a finite number of leading terms in its infinite
series, rather than all infinity terms. In cases like this, there is an adjustable parameter, e.g., the
number of points or of terms, such that the true answer is obtained only when that parameter goes
to infinity. Any practical calculation is done with a finite, but sufficiently large, choice of that
parameter. The difference between the true answer and the answer obtained in a practical calculation
is called the truncation error. Truncation error would persist even on a hypothetical, perfect
computer that had an infinitely accurate representation and no round off error.

 Overflow in addition
Overflow in addition of two or more binary numbers occurs when the sum exceeds the word
size available in the digital implementation of the system.
Limit cycles
Since quantization inherent in the finite precision arithmetic operations render the system
nonlinear, in recursive system these nonlinearities often cause periodic oscillation to occur in the
output, even when input sequence is zero or some nonzero value. Such an oscillation in recursive
systems are called limit cycles.
As explained in above paragraphs finite word length affects LTI system in many ways. We
have concentrated on effects due to coefficient quantization on filter response and in that also on IIR
filters. Later we have given brief overview of effects of coefficient quantization in FIR system for the
sack of completeness.
8. Explain about limit Cycle oscillations
When a stable IIR digital filter is excited by finite input sequence that is constant the output will
ideally zero. However non linearity's due to finite precision arithmetic operation often cause periodic
oscillation to occur in the output. Such oscillations in recursive systems are called zero input limit
cycles Consider a first order IIR filter the difference equation)
y(n) =x(n) +a y(n-1)
Let us assume a=1/2 and the data register length is 3 bits plus a sign bit. If the input is

n
0
1
2
3
4
5

x(n)
0.875
0
0
0
0
0

0.875 YX  = 0 
x(n) =
0 Y4XGF4
y(n-1)
0.0
7/8
1/2
1/4
1/8
1/8

ay(n-1)
0.0
7/16
1/4
1/8
1/16
1/16

Q[ay(n-1]
0.000
0.110
0.010
0.001
0.001
0.001

Y(n)
7/8
1/2
1/4
1/8
1/8
1/8

Note that beyond n=4 the value of ay(n-1) is 1/16 and in binary 0.000100 an 0.000100 which
whenrounded gives 1.001 and 0.001 exhibiting oscillatory.
Dead ban d will b calculated by using y(n-1)

w 
v
v

w||

UNIT V APPLICATIONS
PART A
1. Define multi rate digital signal processing?
NOV 2012
Digital signal processing system handles processing at multiple sampling rates and then it is
called multi-rate signal processing.
2. What are the two techniques of sampling rate conversion?
i. D/A conversion and resampling at required rate.
ii. Sampling rate conversion in digital domain (multi-rate processing)

NOV2012

3. What are the applications of sampling rate conversion?

i.
Narrow band filters.
ii.
Quadrature mirror filters.
iii. Digital filter banks.

MAY 2013

4. What is meant by decimation?

MAY 2013
Decimation by a factor D, means to reduce the sampling rate by a factor D. It is also called
down sampling.
5. What is meant by interpolation? Write the interpolation equation?
MAY 2012
Interpolation by a factor I, means to increase the sampling rate by a factor I. It is also called as
up sampling by I.

MAY2012
6. Define speech compression and decompression.

Speech analysis by a vocoder becomes the compression and synthesis by a vocoder becomes
decompression.

Vocoder extracts the spectral envelope of speech and information regarding voicing and pitch.

This data is coded and transmitted. The synthesizer generates speech from the received data.
7.

Write the principle of adaptive filters.

NOV 2013
The coefficients of the filter are changed automatically according to the changes in input
signal. This means the filtering characteristics of the adaptive filter are changed or adapted according
to the changes in input signal.

8. How the image enhancement is achieved using DSP?

i.
Local neighbourhood operations as in convolution.
ii.
Transform operations as in DFT.
iii. Mapping operations as in pseudo coloring and gray level mapping.

NOV 2013

9. List the different methods of image enhancement.

i.
Contrast and edge enhancement.
ii.
Pseudo coloring.
iii. Noise filtering.
iv.
Sharpening.
v.
Magnifying.

MAY 2011/NOV 2011

10. What are the applications of image enhancement?

NOV 2011

i.
ii.

iii.

Feature extraction in an image.

Image analysis.
Visual information display.

11. What is adaptive equalization?

NOV 2012
Adaptive equalization is the technique used to reliably transmit data through a communication
channel. Ideally, if the channel is ideal (without and channel distortion and additive noise), we can
demodulate the signal perfectly at the output without causing any error.
12. State a few applications of adaptive filter.

NOV 2013/NOV 2010

Noise cancellation
Signal prediction
Adaptive feedback cancellation
Echo cancellation

13. List various special audio effects that can be implemented digitally.

Echo effect

Reverberation

Chorus effect

Phasing effect

Flanging

MAY 2013

PART B
1. Discuss sub band coding process in detail.
MAY 2013
Digitized speech signals may be transmitted over a limited bandwidth channel or it can be
stored. Reducing the size of the signal before transmission or storage is known as speech
compression. The signal compression can be achieved by sub band coding. This method is making
use of the non uniform distribution of signal energy in the frequency component.
Transmitter:

The signal is spit into many narrow band signals which occupy continuous frequency
bands using analysis filter bank.

Down sampling these signals gives sub band signals.

Then compress it using encoders and the compressed signal is multiplexed and transmtted.
Receiver:

The received signal is demultiplexed, decoded, up sampled and then passed through a
synthesis filter.

The output of synthesis filter bank are combined to get the original uncompressed signal.

Block diagram of analysis and synthesis section of sub band coding

2. With lock diagram explain adaptive filtering based adaptive channel equalization. MAY 2013
 Filters with adjustable coefficients are called adaptive filters
 In digital communication system the adaptive equalizer is used to compensate for the distortion
caused by the transmission medium.
 Block diagram with explanation of each block.

3. A.) Explain how the speech compression is achieved.

NOV 2013
The human speech in its pristine form is an acoustic signal. For the purpose of communication
and storage, it is necessary to convert it into an electrical signal. This is accomplished with the help of
certain instruments called transducer.
This electrical representation of speech has certain properties:
1. It is one- dimensional signal, with time as its independent variable.
2. It is random in nature
3. It is non stationary, that is, all the characters of the signal changes with time.
With the advent of digital computing mechanism, it was propounded to exploit the powers of
the same for processing of speech signals. This required a digital representation of speech. To achieve
this, the analog signal is sampled at some frequency and then quantized at discrete levels. Thus
parameters of digital speech are
1. Sampling rate
2. Bits per second
3. Number of channels
Compression is the process of converting an input speech data stream into another data stream
that has a smaller size. Compression is possible only because data is normally represented in the
computer in a format that is longer than necessary that is, the input data has some amount of
redundancy associated with it. The main objective of compression system is to eliminate the
redundancy.
Some application of speech compression:
B.) Discuss about multirate signal processing.
NOV 2013
1. Decimation: Decimation is a process of reducing the sampling rate by a factor M.
Prove the Decimator (Down- sampler)
Let x(n) be a sequence which has been sampled at rate 1(unity), i.e. x(n) is
obtained by sampling a continuous time sequence x(t) at Nyquist rate
X(n)=x(t)|t=n
The decimation (or down sampling) operator M converts the input sequence
x(n) into a new sequence y(n), having the rate 1/M.
y(n)=x(Mn)
1. Interpolation: Interpolation is a process of increasing the sampling rate by a factor L.
Prove the Interpolation (up- sampler)
The Interpolation pads L-1 new samples between successive values of the signal. The
interpolation process increases the sampling rate from I to IFs. since interpolation process
increases the sampling rate, it is symbolically represented by a up arrow (L)

m =  <  >
4. How the image enhancement is achieved using DSP?
i) Local neighborhood operations as in convolution.
ii) Transform operations as in DFT.
iii) Mapping operations as in pseudo coloring and gray level mapping.
Different methods of image enhancement.
i) Contrast and edge enhancement.
ii) Pseudo coloring.

NOV2013/2012

iii) Noise filtering.

iv) Sharpening.
v) Magnifying.
Applications of image enhancement
i) Feature extraction in an image.
ii) Image analysis.
iii) Visual information display.
5. Explain Adaptive noise cancellation with a neat diagram.
NOV 2012

Linear Filtering will be optimal only if it is designed with some knowledge about the input
data.

If this information is not known, then adaptive filters are used.

The adjustable parameters in the filter are assigned with values based on the estimated
statistical nature of the signals.

Filters are adaptable to the changing environment.

Adaptive filtering finds its application in adaptive noise cancelling, line enhancing, frequency
tracking, channel equalizations, etc.

The noise cancellers are used to eliminate intense background noise. This configuration is
applied in mobile phones and radio communications, because in some situations these devices are
used in high-noise environments. Figure 6 shows an adaptive noise cancellation system.
The canceller employs a directional microphone to measure and estimate the instantaneous
amplitude of ambient noise r(n), and another microphone is used to take the speech signal which is
contaminated with noise d(n) + r(n). The ambient noise is processed by the adaptive filter to make it
equal to the noise contaminating the speech signal, and then is subtracted to cancel out the noise in
the desired signal. In order to be effectively the ambient noise must be highly correlated with the
noise components in the speech signal, if there is no access to the instantaneous value of the
contaminating signal, the noise cannot be cancelled out, but it can be reduced using the statistics of
the signal and the noise process.
6. Explain any one application using multirate processing of signals.
Some application of Multirate signal processing are
1. Sampling rate conversion
2. Design of phase shifters
3. Interfacing of digital systems with different sampling rate.

NOV 2010

4. Improved digital-to-analog conversion (DAC) and analog-to-digital conversion(ADC)

5. Frequency division multiplexing (FDM) channel modulation and processing
6. Sub band coding of speech and images.
Sub band coding of speech and images.
Digitized speech signals may be transmitted over a limited bandwidth channel or it can be
stored. Reducing the size of the signal before transmission or storage is known as speech
compression. The signal compression can be achieved by sub band coding. This method is making
use of the non uniform distribution of signal energy in the frequency component.
Transmitter:

The signal is spit into many narrow band signals which occupy continuous frequency
bands using analysis filter bank.

Down sampling these signals gives sub band signals.

Then compress it using encoders and the compressed signal is multiplexed and transmtted.
Receiver:

The received signal is demultiplexed, decoded, up sampled and then passed through a
synthesis filter.

The output of synthesis filter bank are combined to get the original uncompressed signal.

Block diagram of analysis and synthesis section of sub band coding

7. Derive and explain the frequency domain characteristics of the decimator by the factor M and
interpolator by the factor L.
MAY 2011/MAY 2013
1. Decimation: Decimation is a process of reducing the sampling rate by a factor M.
Prove the Decimator (Down- sampler):
Let x(n) be a sequence which has been sampled at rate 1(unity), i.e. x(n) is
obtained by sampling a continuous time sequence x(t) at Nyquist rate
X(n)=x(t)|t=n
The decimation (or down sampling) operator M converts the input sequence
x(n) into a new sequence y(n), having the rate 1/M.
y(n)=x(Mn)

Spectral analysis of Decimator:

BLOCK DIAGRAM OF DECIMATOR AND DERIVATION
2. Interpolation: Interpolation is a process of increasing the sampling rate by a factor I.
Prove the Interpolation (up- sampler)
The Interpolation pads L-1 new samples between successive values of the
signal. The interpolation process increases the sampling rate from I to IFs. since interpolation
process increases the sampling rate, it is symbolically represented by a up arrow (L)

m =  <  >

Spectral analysis of Interpolator:

BLOCK DIAGRAM OF INTERPOLATOR AND DERIVATION
8. Explain the methods of speech analysis and synthesis in details.
NOV 2011
Vocoders (Voice Coder) were originally designed to reduce the bandwidth requirements of
transmission of normal voice signal. The vocoder implements analysis and synthesis
s
sections, which
is nothing but application of multirate signal processing. In the analysis section, natural speech is
analyzed, typically by a bank of filters as shown in figure. The output of each filter is coded by one of
a variety of different methods, and this coded information is transmitted across the channel.
Block diagram of analysis and synthesis of subband encoded speech signal