CS2403 Digital Signal Processing

SELVAM COLLEGE OF TECHNOLOGY,
NAMAKKAL-637003,TAMIL NADU
Collected By: Mr.R.Logarasu AP /ECE
IT6502-DIGITAL SIGNAL PROCESSING
www.selvamtech.edu.in
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Reg.No:
Question Paper Code:21321

B.E./B.Tech. DEGREE EXAMINATION, MAY/JUNE 2013
Seventh Semester
Computer Science and Engineering
CS 2403/CS 73 - DIGITAL SIGNAL PROCESSING
(Common to Fifth Semester- Information Technology)
(Regulation 2008)
Time: Three hours
Maximum: 100Marks
Answer ALL questions
PART A- (10 x 2=20 marks)
1. Define energy signals and network signals.

2. What is the correlation? What are its types?
3. What is meant by radix-2 FFT?
4. Give transform pair equation of DFT.
5. What are the characteristics of Chebyshev filter?
6. Write the transformation equation to convert low pass filter into band stop filter.
7. Write the equation for Blackman window.
8. What is zero input limit cycle oscillation?
9. What is decimation?
10. List various special audio effects that can be implemented digitally.
Page 12 of 54


PART B (5 16 = 80)
11. (a) (i).Consider the analog signal

x a (t ) = 3 cos 2000t + 5 sin 6000t + 10 cos12000t
(1).What is the Nyquist rate for this signal?
(2).Assume now that we sample this signal using sampling rate Fa = 5000 samples / Sec .What
is the discrete time signal obtained after sampling?
(3).What is the analog signal y a (t ) that we can reconstruct from the samples if we use ideal
interpolated.
(8)
(ii).Derive the equation for convolution sum and summarize the steps involved in computing
convolution.
(8)
Or
(b)
(i).Determine the Z-transform and ROC of the signal x(n) = n u (n 1) .
(6)
(ii).Check whether the discrete time system y (n) = cos[x(n)] is

(1).Static or dynamic
(2).Linear or nonlinear
(3).Time invariant or time varying
(4).Casual or non-casual
(5).Stable or unstable.
(10)
12. (a) (i).Find eight point DFT of the following sequence using direct method:
(1) {1, 1, 1, 1, 1, 1, 0, 0
(10)
(ii).State any six properties of DFT.
(6)
Or
(b)
(i).Compute eight point DFT of the following sequences using radix-2 decimation in time FFT
algorithm x(n) = {1, 1, 1, 1, 1, 1, 1, 1 }
(ii).Discuss the use of FFT in linear filtering.
(10)
(6)
Page 13 of 54

13. (a) (i).Obtain the direct form I, direct form II, cascade and parallel form realization for the system
y (n) = 0.1 y (n 1) + 0.2 y (n 2) + 3 x(n) + 3.6 x(n 1) + 0.6 x(n 2)
(ii).For the analog transfer function H ( s ) =
(8)
2
.Determine H (z ) using impulse
( s + 1)( s + 2)
invariance method. Assume T = 1sec
(8)
Or
(b)
A low pass filter meeting the following specification is required:

Passband
0-500Hz
Stopband
2-4KHz
Passband ripple
3dB
Stopband attenuation -
20dB
Determine following:
(i).Pass and stopband edge frequencies for a suitable analog prototype low pass filter.
(ii).Order N of the prototype low pass filter.
(iii).Coefficients and hence the transfer function of the discrete time filter using the bilinear ztransform.
Assume Butterworth characteristics of the filter.
14. (a) (i).Given a three stage lattice filter with coefficients K 1 =
(16)
1
1
1
, K 2 = , K 3 = , determine the
4
4
3
FIR filter coefficients for the direct form structure.
(8)
(ii).Determine the coefficients of a linear phase FIR filter of length M = 15 has a symmetric
unit sample response and a frequency response that satisfies the conditions
2k 1
H
=
15 0
k = 0, 1, 2, 3
k = 4, 5, 6, 7
(8)
Or
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(b) Design an ideal high pass filter with a frequency response
( )
H d e j
1
=
0
for
for
Find the values of h(n) for N = 11 using hamming window. Find H (z ) and compute magnitude
response.
(16)
15 (a) (i) Explain the method for converting the sampling rate by a factor I/D with block diagram and
equations.
(8)
(ii) Discuss sub band coding process in detail.
(8)
Or
(b) (i) .With block diagram explain adaptive filtering based adaptive channel equalization.
(8)
(ii) .What is image enhancement? Explain various image enhancement techniques.
(8)
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Reg.No:
Question Paper Code:20239

B.E./B.Tech. DEGREE EXAMINATION, MAY/JUNE 2012
Seventh Semester
CS 2403/CS 73 - DIGITAL SIGNAL PROCESSING
(Common to Fifth Semester Information Technology)
(Regulation 2008)
Time: Three hours
Maximum: 100Marks

1. Determine even and odd components of the signal, x(n) = exp j n + j , where j = 1 .
2
4
2. Compute autocorrelation of the signal, x(n) = (0.5)n u (n) .
3. Write the formula for Discrete Cosine Transform(DCT) pair.
4. What is FFT?
5. What are the limitations of impulse invariant technique of designing filter?
6. Give the low pass transfer function H a ( s ) =
1
.Find the high pass transfer function having a cuts =1
off frequency 10 Rad / Sec .

7. Compare the digital signal processing systems with fixed point and floating point representation.
8. List the few applications, where in linear phase is preferred.
9. Draw the basic building Blocks of Adaptive filters.
10.Let x(n) = { 1.5, 1, 0.5, 0.2, 1.5, 7.5
} Compute (a) x n
3
(b) x(4n)
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PART B (5 16 = 80)
11. (a) (i).State sampling theorem and explain aliasing graphically.
(8)
(ii).Find the circular convolution of x(n) h(n) .Given that

x(n) = {1,
0 n 99
(0.5)n ,
h( n) =
0,
9 n 49
50 n 99
(4)
(iii).Find the Z-transform of the given sequence x(n) = 8(n 5) + e n u (n 2) + u (n) .
(4)
Or
(b)
(i).Find the Linear Convolution of x(n) h(n) through circular convolution. Assume the
suitable length .M
(0.5) n ,
x ( n) =
0,
(0.8)n ,
h( n) =
0,
0n9
Otherwise
(8)
0 n 19
Otherwise
(ii).List all the properties of Analog and Digital frequencies.
(4)
(iii).Find the Z-transform of Auto correlation of signal.
(4)
12. (a) (i).Let X ( K ) = DFT {x(n)} with n: K = 0,1,......., ( N 1) .Determine the relationship between
X (K ) and the following DFTs.
(1) DFT {Re x(n)}

(2) DFT {x(n 1)}
(9)
(ii).State and prove any two properties of DFT.
(7)
Or
(b)
(i).Compute the DFT of the following sequences.

(1) x(n) = [1, 0, 1, 0]
(2) x(n) = Cos (0.25n), n = 1,1,2,...7
(ii).Write short notes on filtering methods using DFT.
(8)
(8)
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NAMAKKAL-637003,TAMIL
NADU
13. (a) (i).An impulse response, h(t ) = exp(0.5t )u (t ) of certain LTI system. Find the T.F. H (Z )
using impulse invariant technique. Assume T = 2 Sec .
(8)
(ii).Compare Analog filters with Digital filters.
(4)
(iii).Differentiate between Bilinear transformation with frequency translations of filter transfer

function.
(4)
Or
(b)
(i).Draw the Ideal Gain Vs frequency characteristics of HPF and BPF and also how the
above filters(practically) specified.
(8)
(ii).Write short notes on frequency translation in both Analog and Digital domain.
14. (a) The Hamming window is given by (n) = 0.54 046 cos
2n
, 0 n m 1 .Compute the
m 1
first 10 coefficients using above window functions having the magnitude response.
(b)
(8)
(16)
(i) Draw three different structure of H (z ).

H ( z ) = (1 + 0.5 z 1 )(1 + 0.75 z 1 ) 3
(8)
(ii) Draw three different structure of H (z ). with coefficients represented by (1) 4-bit word length (2) 6-
bit word length at frequency

H ( z ). =
=3
1
z 0.752352
(8)
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15 (a) (i) Explain the salient features of image enhancement techniques.
(10)
(ii) Write a brief note on speech compression techniques.
(6)
Or
(b) (i) Determine the output y(n) for each of the given input signal x(n)
(8)
(1) x(n) = 8(n 4)
(2) x(n) = exp( j 0.2n )u (n)
(ii) Explain the sampling by rational factor by taking an example of your own. Also state its
uses in DSP systems.
(8)
Page 22 of 54


Reg. No. :
Question Paper Code :
55310
B.E./B.Tech. DEGREE EXAMINATION, NOVEMBER/DECEMBER 2011.
Seventh Semester
CS 2403 DIGITAL SIGNAL PROCESSING
(Common to Fifth Semester Information Technology)
(Regulation 2008)
Time : Three hours
Maximum : 100 marks

Answer ALL questions.
PART A (10 2 = 20 marks)
1.
Define and express the transfer function of Nth order LTI system.
2.
Compare linear convolution and circular convolution.
3.
What is the relation between DFT and Z-Transform?
4.
What is phase factor or twiddle factor?
5.
Sketch the various tolerance limits to approximate an ideal lowpass and

highpass filter.
6.
What is the importance of poles in filter design?
7.
What is frequency warping?
8.
What is Butterworth approximation?
9.
List various voice compression and coding techniques.
10.
What are the various enhancement techniques in image processing?
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PART B (5 16 = 80 marks)
11.
(a)
(i)
(ii)
Find
the
inverse
Z2 +Z
X (z ) =
, ROC : Z > 3 ,
(Z 1 )(Z 3 )
using
(1)
Residue method and
(2)
Convolution method.
Z-Transform
of
(8)
State and prove circular convolution.
(8)
Or
(b)
12.
(a)
LTI
system
is
described
by
the
y (n ) = a y (n 1 ) + b x (n ) . Find the impulse
function and phase function. Solve b, if
magnitude and phase response for a = 0.6.
(i)
H (w ) = 1 . Sketch the
(16)
Evaluate the 8-point for the following sequences using DIT-FFT

algorithm
(8)
1 for
x (n ) =
0
(ii)
difference
equation
response, magnitude
3 n 3
otherwise
Calculate the percentage of saving in calculations in a 1024point radix

-2 FFT, when compared to direct DFT. (8)
Or
13.
(b)
Determine the response of LTI system when the input sequence

x (n ) = { 1, 1, 2 , 1, 1 } by radix 2 DIT FFT. The impulse response of the
system is h (n ) = { 1, 1, 1, 1} .
(16)
(a)
The specification of the desired lowpass filter is

1
2
H (w ) 1 .0 ; 0 w 0 . 2
H (w ) 0 . 08 ; 0 .4 w
Design a Butterworth digital filter using bilinear transformation. (16)

Or
(b)
The specification of the desired low pass filter is

0 . 9 H (w ) 1 .0
H (w ) 0 .24 ;
0 w 0 . 25
0 .5 w
Design a Chebyshev
transformation.
digital
filter
using
impulse
invariant
(16)
55310
Page 24 of 54

14.
(a)
(i)
Design a single tier notch filter to reject frequencies in the range

1 to 2 rad/sec using rectangular window with N = 7 .
(8)
(ii)
Compare Hamming window and Kaiser window.
(8)
Or
(b)
15.
(a)
(i)
Explain the characteristics of a limit cycle oscillation with

respect to the system described by the equation
y (n ) = 0 . 95 y (n 1 ) + x (n ) . Determine the dead band of the filter.(8)
(ii)
Explain Gibbs phenomenon (or Gibbs oscillation).
(8)
Explain the methods of speech analysis and synthesis in detail.
(16)
Or
(b)
Explain how image enhancement restoration and coding can be done

using signal processing.
(16)
55310
Page 25 of 54


Reg. No. :
Question Paper Code :
11273
B.E./B.Tech. DEGREE EXAMINATION, APRIL/MAY 2011

Fifth Semester
Information Technology
(Regulation 2008)
Time : Three hours
Maximum : 100 marks

PART A (10 2 = 20 marks)
1.
State sampling theorem.
2.
Distinguish between power and energy signal with an example.
3.
State and prove Parseval's theorem.
4.
Compute the DFT of the four point sequence x (n ) = [0 ,1,2 ,3 ] .
5.
What is meant by warping?
6.
What are the limitations of impulse invariance method?
7.
List out the conditions for the FIR filter to be linear phase.
8.
What is meant by limit cycle oscillations?
9.
Let x (n ) = [1,2 ,3 ,4 ,5 ,6 ] . Sketch x (n 2 ) and x (3n ) .
10.
Give any two image enhancement methods.
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PART B (5 16 = 80 marks)
11.
(a)
(i)
Suppose
a LTI system with input x (n ) and output y (n ) is
characterized by its unit sample response h (n ) = (0 .8 )n u (n ) . Find

the response y (n ) of such a system to the input signal
x ( n ) = u (n ) .
(8)
(ii) A causal system is represented by the following difference
equation
y (n ) +
1
1
y (n 1 ) = x (n ) + x (n 1 )
4
2
Compute the system function H (z ) and find the unit sample

response of the system in analytical form.
(8)
Or
(b)
(i)
Compute the normalized

x (n ) = a n u (n ),0 < a < 1 .
autocorrelation
(ii)
Determine the impulse response for the cascade of two LTI

system having impulse responses h1 (n ) = (1 / 2 )n u (n ) and
h 2 (n ) = (1 / 4 )n u (n ) .
12.
(a)
of
the
signal
(8)
(8)
By means of the DFT and IDFT, determine the response at the FIR
filter with the impulse response h (n ) = [1,2 ,3 ] and the input sequence
x (n ) = [1,2 ,2 ,1 ] .
Or
13.
(b)
Compute the DFT of the following sequence x (n ) using the

decimation in time FFT algorithm x (n ) = [1,1,1,1,1,1,1,1 ] .
(a)
(i)
Find the H (z ) corresponding to the impulse invariance design

using a sample rate of 1/T samples/sec for an analog filter H (s )
A
specified as follows : H ( s ) =
.
(6)
S +
(ii)
Design a digital low pass filter using the bilinear transform to

satisfy the following characteristics (1) Monotonic stop band
and pass band (2 ) 3 dB cutoff frequency of 0 .5 rad (3 )
magnitude down at least 15 dB at 0 .75 rad .
(10)
Or
11273
Page 27 of 54

14.
(b)
Design an IIR filter using impulse invariance technique for the given
1
H a (s ) = 2
. Assume T = 1 sec. Realize this filter using direct
s + 17 s + 12
(16)
form I and direct form II.
(a)
Design and obtain the coefficients of a 15 tap linear phase FIR low
pass filter using Hamming window to meet the given frequency
response (16)
1 for
H d (w ) =
0 for
| w|
6 .
| w|
6
Or
(b)
(i)
Determine the coefficients of a linear phase FIR filter of length

M = 15 which has a symmetric unit sample response and a
frequency response that satisfies the conditions
1
2 k
Hr
= 0 .4
15
0
(ii)
15.
(a)
k = 0 ,1,2 ,3
k = 4 .
k = 5 ,6 ,7
(8)
The output of A/D converter is applied to digital filter with the

0 .5 z
system function H ( z ) =
. Find the output noise power
z 0 .5
from the digital filter when the input signal is quantized to have
8 bits.
(8)
Derive and explain the frequency domain characteristics of the

Decimator by the factor M and interpolator by the factor L.
(16)
Or
(b)
With neat diagram explain any two applications of adaptive filter

(16)
using LMS algorithm.
11273
Page 28 of 54


Reg. No. :
Question Paper Code : 53110

B.E./B.Tech. DEGREE EXAMINATION, NOVEMBER/DECEMBER 2010
Fifth Semester
Information Technology
(Regulation 2008)
Time : Three hours
Maximum : 100 Marks

PART A (10 2 = 20 Marks)
1.
Calculate the minimum sampling frequency required for x(t ) = 0.5 sin 50 t
+ 0.25 sin 25 t , so as to avoid aliasing.
2.
State any two properties of Auto correlation function.
3.
Write down DFT pair of equations.
4.
Calculate % saving in computing through radix 2, DFT algorithm of DFT

coefficients. Assume N = 512.
5.
What are the limitations of Impulse invariant method of designing digital

filters?
6.
Draw the ideal gain Vs frequency characteristics of :
7.
(a)
HPF and
(b)
BPF.
Compare FIR filters and FIR filters with regard to :

(a)
Stability and
(b)
Complexity
Page 29 of 54

8.
Represent decimal number 0.69 in fixed point representation of length

N = 6.
9.
Prove that up sampling by a factor M is time varying system.
10.
State a few applications of adaptive filter.

PART B (5 16 = 80 Marks)
11.
(a)
(i)
Find the convolution x (n) * h(n ) , where

x (n) = a n u(n )
h(n ) = n u(n )
(ii)
Find the Z-transform of the following sequences :

x (n) = (0.5) n u(n ) + u(n 1)
x ( n ) = ( n 5) .
Or
(b)
12.
(a)
(i)
State and explain sampling theorems.
(ii)
Find the Z-transform auto correlation function.
(i)
Explain, how linear convolution of two finite sequences are obtained

via DFT.
(ii)
Compute the DFT of the following sequences :

(1)
x = [1, 0, 1, 0]
(2)
x = [ j, 0, j,1] when j = 1 .
Or
13.
(b)
Draw the flow chart for N = 8 using tadix-2, DIF algorithm for finding
DFT coefficients.
(a)
Design digital low pass filter using Bilinear transformation, Given that
Ha( s ) =
1
.
( s + 1)( s + 1.732 s + 1)
Assume sampling frequency of 100 rad/sec.

Or
53110
Page 30 of 54

(b)
Design FIR filter using impulse invariance technique. Given that

Ha( s ) =
1
( s + 5s + 6)
and implement the resulting digital filter by adder, multipliers and

delays Assume sampling period T = 1 sec.
14.
(a)
Design the first 15 coefficients of FIR filters of magnitude specification is

given below :
H ( e jw ) = 1, / w / < / 2
= 0, otherwise.
Or
(b)
Draw THREE different FIR structures for the H(z) given below:
H ( z ) = (1 + 5z 1 + 6z 2 )(1 + z 1 ) .
15.
(a)
(i)
A signal x (n) = {6,1, 5, 7, 2,1}

Find :
(ii)
(1)
x (n / 2)
(2)
x (2n ) .
Explain any one application using multirate processing of signals.

Or
(b)
Write short notes on the following :

(i)
Adaptive filter
(ii)
Image Enhancement.
53110
Page 31 of 54

Reg.No:
B.E./B.Tech. DEGREE EXAMINATION, APRIL/MAY 2010

Fourth Semester
IT1252 - DIGITAL SIGNAL PROCESSING
(Common to Seventh Semester- Computer Science and Engineering)
(Regulation 2004)
Time: Three hours
Maximum: 100 Marks

1. Consider the analog signal x a (t ) = 3 cos 2000t + 5 sin 6000t + 10 cos12000t . What is the Nyquist
rate for this signal?
2. Determine the system described by the following input and output equation is linear or nonlinear
y (n) = Ax(n) + B
3. What is impulse computation?

4. How many complex multiplication and addition are required in DFT and FFT?
5. Indicate the location in the z plane to which the ve points on the j axis in the s plane go to due to
the bilinear transformation.
6. Convert the analog filter with system function H (s ) into digital IIR filter by
H (s) =
1
( s + 0.2)( s + 0.6)
7. Write down the equation of Kaiser Window for 0 n M 1 .

8. What are the possible phase angles if the impulse response is anti-symmetric and legth of the filter
is odd?
9. What is limit cycle oscillation?
10.By:
Define
over flow error.
Collected
Mr.R.Logarasu
AP /ECE
Page 32 of 54


PART B (5 16 = 80)
11. (a) (i).
Compute and plot the convolution x(n) h(n) for the following signal:
x(n) = [1, 1, 1, 1 ]
h(n) = [ 6, 5, 4, 3, 2, 1
(ii).
(6)
Determine the response y (n) , n 0 of the system described by the second order
differential equation y (n) 4 y (n 1) + 4 y (n 2) = x(n) x(n 1) when the input is
x(n) = (1) n u (n) and initial conditions are y (1) = y (2) = 0
(10)
Or
(b)
(i).
Determine the Z-transform and ROC of the signal:
x(n) = 3(2 n ) 4(3 n ) u (n) .

(ii).
(6)
A linear time invariant system is characterized by the system function

3 4 z 1
specify the ROC of H (z ) and determine for the following
1 3.5 z 1 + 1.5 z 1
conditions:
H ( z) =
(1).The system is sable

(2).The system is causal.
(3).The system is anti-causal.
12. (a) (i).
Obtain the FFT of the sequence using decimation in frequency
x(n) = {8, 7, 6, 5, 4, 3, 2, 1
(ii).
(10)
(12)
How can you compute IDFT using FFT(4)

Or
(b)
Multiplication of the DFTs of two sequences is equivalent to the circular convolution of

the two sequences in the time domain. Prove this property by the following sequences:
x1 (n) = [2, 1, 2, 1 ]
13. (a)
x 2 (n) = [1, 2, 3, 14]
Design a digital low pass Butterworth filter using Bilinear transformation method to
meet the following specifications:
Pass band ripple 1.25 dB ,pass band edge = 200 H z ,stop band attenuation 15 dB ,Stop
Collected By: Mr.R.Logarasu

AP /ECE
band edge
frequency = 2 kH z .
= 300 H z ,sampling
Page 33 of 54


Or
(b).
(i).
Find the transfer function of low pass analog Chebyshev filter to meet the following
requirements:
Pass band ripple 0.1 dB ,pass band edge 1 Rad / Sec ,stop band attenuation is almost
40 dB for 2 Rad / Sec (12)
(ii).
14. (a) (i).
Compare FIR and IIR filter.(4)
Design a low pass FIR with 11 coefficients for the following specifications:
Pass band frequency edge is 0.25kH z and sampling frequency is 1 kH z .(Use
Rectangular window)(12)
(ii).
Realize the above designed filter by direct form structures.(4)

Or
(b)
Determine the coefficients of a linear-phase FIR filter length M = 15 which has a

symmetric unit sample response and a frequency response that satisfy the condition:
1,
2k
Hr
= 0.4,
15
0
15 (a)
(b).
k = 1, 2, 3
k=4
k = 5, 6, 7
For the all polo second order IIR digital filter described by difference equation
y (n) = a1 y (n 1) a 2 (n 2) + x(n) . Determine the dead band region governing the
limit cycle
Or
Derive the expressions for the variance at the output of a digital filter characterized by
h(n) Where the input is quantization noise.
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SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

IT6502-DIGITAL SIGNAL PROCESSING

Question Paper Code:21321

1. Define energy signals and network signals.

Collected By: Mr.R.Logarasu AP /ECE

SELVAM COLLEGE OF TECHNOLOGY,

IT6502-DIGITAL SIGNAL PROCESSING

11. (a) (i).Consider the analog signal

(i).Determine the Z-transform and ROC of the signal x(n) = n u (n 1) .

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

(ii).State any six properties of DFT.

Collected By: Mr.R.Logarasu AP /ECE

SELVAM COLLEGE OF TECHNOLOGY,

IT6502-DIGITAL SIGNAL PROCESSING

(ii).For the analog transfer function H ( s ) =

invariance method. Assume T = 1sec

A low pass filter meeting the following specification is required:

FIR filter coefficients for the direct form structure.

Collected By: Mr.R.Logarasu AP /ECE

SELVAM COLLEGE OF TECHNOLOGY,

IT6502-DIGITAL SIGNAL PROCESSING

(b) Design an ideal high pass filter with a frequency response

(ii) Discuss sub band coding process in detail.

(ii) .What is image enhancement? Explain various image enhancement techniques.

Collected By: Mr.R.Logarasu AP /ECE

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING

SELVAM COLLEGE OF TECHNOLOGY,

Collected By: Mr.R.Logarasu AP /ECE

IT6502-DIGITAL SIGNAL PROCESSING