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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.
PART − B (5 × 16 = 80)
11. (a) (i).Consider the analog signal
x a (t ) = 3 cos 2000πt + 5 sin 6000πt + 10 cos12000πt
(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) = −α 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: (10)
(1) {1, 1, 1, 1, 1, 1, 0, 0 }
(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 } (10)
(ii).Discuss the use of FFT in linear filtering. (6)
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) (8)
2
(ii).For the analog transfer function H ( s ) = .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 z-
transform.
Assume Butterworth characteristics of the filter. (16)
1 1 1
14. (a) (i).Given a three stage lattice filter with coefficients K 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
 2πk  1 k = 0, 1, 2, 3
H = (8)
 15   0 k = 4, 5, 6, 7
Or
(b) Design an ideal high pass filter with a frequency response
 π
1 for ≤ ω ≤π
( )
H d e jω = 4
π
0 for ω ≤
 4
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)
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

 π π
1. Determine even and odd components of the signal, x(n) = exp j n + j  , where j = − 1 .
 4 2
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?
1
6. Give the low pass transfer function H a ( s ) = .Find the high pass transfer function having a cut-
s =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  (b) x(4n)
3
↑
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 , 9 ≤ n ≤ 49  (4)
h( n) =  
 0, 50 ≤ n ≤ 99
(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 , 0≤n≤9
x ( n) =  
0, Otherwise 
(8)
(0.8)n , 0 ≤ n ≤ 19 
h( n) =  
 0, 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 (0.25πn), n = 1,1,2,...7 (8)
(ii).Write short notes on filtering methods using DFT. (8)
(t ) = exp(−0.5t )u (t ) of certain LTI system. Find the T.F. H (Z )
13. (a) (i).An impulse response, hwww.Vidyarthiplus.com
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. (8)
2πn
14. (a) The Hamming window is given by ω (n) = 0.54 − 046 cos , 0 ≤ n ≤ m − 1 .Compute the
m −1
first 10 coefficients using above window functions having the magnitude response. (16)
(b) (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-
ω =π3
bit word length at frequency
1
H ( z ). =
z − 0.752352
(8)
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 0.2nπ )u (n)
(ii) Explain the sampling by rational factor by taking an example of your own. Also state its
uses in DSP systems. (8)
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?
PART B — (5 × 16 = 80 marks)
11. (a) (i) Find the inverse Z-Transform of

Z2 +Z
X (z ) = , ROC : Z > 3 ,
(Z − 1 )(Z − 3 )
using
(1) Residue method and
(2) Convolution method. (8)
(ii) State and prove circular convolution. (8)
Or
(b) LTI system is described by the difference equation
y (n ) = a y (n − 1 ) + b x (n ) . Find the impulse response, magnitude
function and phase function. Solve b, if H (w ) = 1 . Sketch the
magnitude and phase response for a = 0.6. (16)
12. (a) (i) Evaluate the 8-point for the following sequences using DIT-FFT
algorithm (8)
1 for −3 ≤ n ≤ 3
x (n ) = 
0 otherwise
(ii) Calculate the percentage of saving in calculations in a 1024-

point radix -2 FFT, when compared to direct DFT. (8)
Or
(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)
13. (a) The specification of the desired lowpass filter is

1
≤ H (w ) ≤ 1 .0 ; 0 ≤ w ≤ 0 . 2π
2
H (w ) ≤ 0 . 08 ; 0 .4 π ≤ w ≤ π
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 ; 0 ≤ w ≤ 0 . 25 π
H (w ) ≤ 0 .24 ; 0 .5 π ≤ w ≤ π
Design a Chebyshev digital filter using impulse invariant

transformation. (16)
2 55310
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) (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 Gibb’s phenomenon (or Gibb’s oscillation). (8)
15. (a) 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)
————————
3 55310
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.
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
1 1
y (n ) + 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 autocorrelation of the signal
x (n ) = a n u (n ),0 < a < 1 . (8)
(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 ) . (8)
12. (a) 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
(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 ] .
13. (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 +α
(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 π rad (3 )
magnitude down at least − 15 dB at 0 .75 π rad . (10)
Or
2 11273
(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
form I and direct form II. (16)
14. (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)
 π 
1 for | w| ≤
H d (w ) =  6 .
π 
0 for | w| ≤ π 
 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 k = 0 ,1,2 ,3 
2π k   
Hr  = 0 .4 k = 4 . (8)
 15  
 0 k = 5 ,6 ,7 
(ii) 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)
15. (a) 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
using LMS algorithm. (16)
——————
3 11273
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 π t
+ 0.25 sin 25 π 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 :
(a) HPF and
(b) BPF.
7. Compare FIR filters and FIR filters with regard to :
(a) Stability and
(b) Complexity
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 ) = β n u(n )
(ii) Find the Z-transform of the following sequences :
x (n) = (0.5) n u(n ) + u(n − 1)
x ( n ) = δ ( n − 5) .
Or
(b) (i) State and explain sampling theorems.
(ii) Find the Z-transform auto correlation function.
12. (a) (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
(b) Draw the flow chart for N = 8 using tadix-2, DIF algorithm for finding
DFT coefficients.
13. (a) Design digital low pass filter using Bilinear transformation, Given that
1
Ha( s ) = .
( s + 1)( s + 1.732 s + 1)
Assume sampling frequency of 100 rad/sec.
Or
2 53110
(b) Design FIR filter using impulse invariance technique. Given that
1
Ha( s ) =
( 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 / < π / 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 :
(1) x (n / 2)
(2) x (2n ) .
(ii) Explain any one application using multirate processing of signals.
Or
(b) Write short notes on the following :
(i) Adaptive filter
(ii) Image Enhancement.
—————————
3 53110
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 2000πt + 5 sin 6000πt + 10 cos12000πt . 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 jω 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
1
H (s) =
( 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. Define over flow error.
PART − B (5 × 16 = 80)
11. (a) (i). Compute and plot the convolution x(n) ∗ h(n) for the following signal: (6)
x(n) = [1, 1, 1, 1 ] h(n) = [ 6, 5, 4, 3, 2, 1 ]

↑ ↑
(ii). 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) . (6)
(ii). A linear time invariant system is characterized by the system function
3 − 4 z −1
H ( z) = specify the ROC of H (z ) and determine for the following
1 − 3.5 z −1 + 1.5 z −1
conditions:
(1).The system is sable
(2).The system is causal.
(3).The system is anti-causal. (10)
12. (a) (i). Obtain the FFT of the sequence using decimation in frequency (12)
x(n) = {8, 7, 6, 5, 4, 3, 2, 1 }
(ii). 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 ] x 2 (n) = [1, 2, 3, 14]
13. (a) 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
band edge = 300 H z ,sampling frequency = 2 kH z .
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). Compare FIR and IIR filter.(4)
14. (a) (i). 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, k = 1, 2, 3
 2πk  
Hr   = 0.4, k=4
 15  
0 k = 5, 6, 7
15 (a) 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
(b). 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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www.Vidyarthiplus.

Question Paper Code:21321

B.E./B.Tech. DEGREE EXAMINATION, MAY/JUNE 2013

Time: Three hours Maximum: 100Marks

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?

7. Write the equation for Blackman window.

8. What is zero input limit cycle oscillation?

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

x a (t ) = 3 cos 2000πt + 5 sin 6000πt + 10 cos12000πt

(1).What is the Nyquist rate for this signal?

is the discrete time signal obtained after sampling?

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

(3).Time invariant or time varying

(5).Stable or unstable. (10)

algorithm x(n) = {1, − 1, − 1, − 1, 1, 1, 1, − 1 } (10)

(ii).Discuss the use of FFT in linear filtering. (6)

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) (8)

invariance method. Assume T = 1sec (8)

(b) A low pass filter meeting the following specification is required:

Passband ripple - 3dB

Stopband attenuation - 20dB

(ii).Order N of the prototype low pass filter.

Assume Butterworth characteristics of the filter. (16)

FIR filter coefficients for the direct form structure. (8)

(ii) Discuss sub band coding process in detail. (8)

Question Paper Code:20239

B.E./B.Tech. DEGREE EXAMINATION, MAY/JUNE 2012

Time: Three hours Maximum: 100Marks

PART A- (10 x 2=20 marks)

2. Compute autocorrelation of the signal, x(n) = (0.5)n u (n) .

3. Write the formula for Discrete Cosine Transform(DCT) pair.

5. What are the limitations of impulse invariant technique of designing filter?

8. List the few applications, where in linear phase is preferred.

9. Draw the basic building Blocks of Adaptive filters.

(ii).Find the circular convolution of x(n) ∗ h(n) .Given that

(ii).List all the properties of Analog and Digital frequencies. (4)

(iii).Find the Z-transform of Auto correlation of signal. (4)

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)

(b) (i).Compute the DFT of the following sequences.

(1) x(n) = [1, 0, − 1, 0]

(2) x(n) = Cos (0.25πn), n = 1,1,2,...7 (8)

(ii).Write short notes on filtering methods using DFT. (8)

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

above filters(practically) specified. (8)

(b) (i) Draw three different structure of H (z ).

H ( z ) = (1 + 0.5 z −1 )(1 + 0.75 z −1 ) 3 (8)

(ii) Write a brief note on speech compression techniques. (6)

(1) x(n) = 8(n − 4)

(2) x(n) = exp( j 0.2nπ )u (n)

uses in DSP systems. (8)

Question Paper Code :

Computer Science and Engineering

CS 2403 — DIGITAL SIGNAL PROCESSING

(Common to Fifth Semester Information Technology)

Time : Three hours Maximum : 100 marks