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DSP Papers

The document outlines examination papers for Digital Signal Processing courses at BMS College of Engineering, detailing various units covering topics such as DFT, convolution, filter design, and FFT algorithms. Each unit includes multiple questions requiring calculations, derivations, and explanations related to digital signal processing concepts. The exams are structured to assess students' understanding and application of these concepts in practical scenarios.

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0% found this document useful (0 votes)
32 views17 pages

DSP Papers

The document outlines examination papers for Digital Signal Processing courses at BMS College of Engineering, detailing various units covering topics such as DFT, convolution, filter design, and FFT algorithms. Each unit includes multiple questions requiring calculations, derivations, and explanations related to digital signal processing concepts. The exams are structured to assess students' understanding and application of these concepts in practical scenarios.

Uploaded by

rohit adhikary
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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U.S.N.

BMS College of Engineering, Bangalore-560019


(Autonomous Institute, Affiliated to VTU, Belgaum)

July / August 2017 Supplementary Semester Examinations


Course: Digital Signal Processing Duration: 3 hrs
Course Code: 16EC5DCDSP Max Marks: 100

Date: 25.07.2017
Instructions: Answer any FIVE question
Chebyshev Filter Table may be used

UNIT 1
1 a Compute the 3 point and 6 point DFTs of the sequence 08
𝑥[𝑛] = [2, 1, 2] and compare the results.
b The DFT of a real sequence is 06
𝑋[𝑘] = {1, 𝐴, −1, 𝐵, −7, −𝑗 2, 𝐶, −1 + 𝑗}. What is the energy of the signal
c Given the sequences 𝑥[𝑛] = {1, −2, 3, 0, −1, 1}for 0 ≤ n ≤ 6, with 06
a 6 point DFT X(k), evaluate the following without computing the DFT. (i) X(0), (ii)
X(3), (iii) ∑5𝑘=0 𝑋(𝑘)and (iv) ∑5𝑘=0|𝑋(𝑘)|2

UNIT 2
2 a Given 𝑥(𝑛) = [1, −1, 2, −2, 3, −3, 4, −4]; ℎ(𝑛) = [−1, 1], perform convolution 08
by using overlap and add method. Use N = 4 circular convolution.
b Explain the relationship between DFT and Z –transform 06
c Perform linear convolution of the two sequences x(n) = {1,2} and 06
y(n) = {3, 4}via circular convolution.

UNIT 3
3 a A 8 point sequence is given by x(n) = [ 2, 2, 2, 2, 1, 1, 1, 1]. 10
Compute 8 point DFT by radix – 2 DIT -FFT algorithm. Draw the
butterfly diagram and clearly indicate the output samples for every
stage of computation
b Derive the DIF FFT algorithm to compute N-point DFT and show all the 10
intermediate values in the signal flow graph.

UNIT 4
4 a 3(𝑧 2 +5𝑧+4) 08
Given 𝐻(𝑧) = (2𝑧+1)(𝑧+2)
for an IIR system, obtain the parallel form of realization.
b Determine the system function H(Z) of the lowest order Chebyshev IIR filter with 12
the following specifications:
i. 3 dB ripple in passband 0 ≤ ω ≤ 0.2 π
ii. 25 dB attenuation in stopband 0.45 π ≤ ω ≤ π.
Use Bilinear transformation. [ Filter Table may be used]
OR

5 a Obtain direct form –I for the digital filter H(s) given by 04


8𝑧 3 − 4𝑧 2 + 11𝑧 − 2
𝐻(𝑠) =
(𝑧 − 1)(𝑧 3 − 𝑧 + 1)
𝑠+1
b Transform the analog filter 𝐻𝑎 (𝑠) = 𝑠2 +5𝑠+6 into a digital filter H(z) using impulse 06
invariant transformation. Take T = 0.1 sec 6
c Design an IIR low-pass Butterworth digital filter using Bilinear transformation, to 10
satisfy the following analog specifications:
(i) Pass band = 0 – 400 Hz (ii) Pass band ripple = 2 dB
(iii) Stop band = 2.1 – 4 kHz. (iv) Stop band attenuation = 20dB
(iv)Sampling frequency = 10kHz.
Use bilinear transformation.

UNIT 5
6 a Draw the linear phase realization of the digital filter 04
1 1 1 1
𝐻(𝑧) = 1 + 3 𝑧 −1 + 4 𝑧 −2 + 4 𝑧 −3 + 3 𝑧 −4 + 𝑧 −5
b Design a low-pass filter using rectangular window, with a cutoff frequency of 1.2 10
rad/ sec and N = 9.
c Explain the Gibbs phenomenon. 06

OR

7 a Determine the impulse response h(n) of a filter having desired frequency response, 12

  j ( N 1) 
H d (e jw )  e 2 for 0  w 
 2

  0 for   
2

N=7. Use frequency sampling approach


b Obtain the Direct form and linear phase realization for 08

3 1 17 2 3 3
H (Z )  1  Z  Z  Z  Z 4
4 8 4

*******
U.S.N.

BMS College of Engineering, Bangalore-560019


(Autonomous Institute, Affiliated to VTU, Belgaum)

December 2016 Semester End Main Examinations


Course: Digital Signal Processing Duration: 3 hrs
Course Code: 16EC5DCDSP Max Marks: 100

Date: 15.12.2016
Instructions: Answer any FIVE question
Chebyshev Filter Table may be used
UNIT 1
1 a Find the N-point DFT of the sequence 05
1 for 0  n  N  1

x(n)  0 otherwise for N  10

Also sketch the DFT of the sequence


b State and prove time reversal property and convolution property of DFT 08
c State and verify parseval’s theorem for the sequence x (n) whose DFT is given as 07
X(K)={6,0,-2,0}
UNIT 2
2 a Obtain the response of a system given x(n)=[1,2,3,4] and h(n)=[2, 1] using 08
i. Linear Convolution
ii. Circular Convolution
b Determine the Output y(n) of a filter whose impulse response is h(n)={1, 1, 1} and 08
input signal x(n)={3,-1,0,1,3,2,0,1,2,1} using
i) Overlap Save Method
ii) Overlap Add Method
c Derive the relationship between DFT and Z-transform 04
UNIT 3
3 a Find the DFT of the following sequence using DIT FFT algorithm and draw the 10
flow graph indicating the intermediate values of the graph

x(n)= {1, -1, -1, -1, 1, 1, 1, -1}


b Evaluate the speed improvement factor in calculating 64-point DFT of a sequence 04
using direct computation and FFT algorithm
c Find the IDFT of the sequence X(K)={3, 5-j8, -1, 5+j8) using decimation in 06
frequency fast Fourier transform
UNIT 4
4 a Design an analog low pass Chebyshev type-1 filter that has -3dB cutoff frequency of 10
2 radians/ sec and stop band attenuation of 25dB or greater for all radian
frequencies greater than 5 radians / sec.(Filter table may be used)
b 2 05
For the analog transfer function H ( s)  . Design of IIR filter using
( s  1)(s  2)
impulse invariance method. Assume T=1 sec.
c Realize the system with difference equation 05
3 1 1
y(n)  y(n  1)  y(n  2)  x(n)  x(n  1) in cascade form
4 8 3
OR
5 a Design a digital low pass Butterworth filter using Bilinear Transformation for the 10
following specifications,
 p  0.2 , s  0.6 , K p  2dB K s  14dB
Assume T=2 seconds
b Obtain the cascade and parallel realization for the following function: 10

(1  Z 1 )
H (Z ) 
(1  0.5Z 1 )(1  0.25Z 1 )(1  0.125Z 1 )
UNIT 5
6 a A low pass FIR filter is to be designed with the following desired frequency 10
response


H d ( w)  e  j 2 w for w 
4

0 for  w 
4

Determine FIR filter coefficients using windowing method. The length of filter N=5.
Window to be used is Hanning Window.
b Draw the direct form and linear phase structure for an FIR Filter characterized by 06
1 1 1
h(n)   (n)   (n  1)   (n  2)   (n  3)   (n  4)
2 4 2
c Differentiate between IIR and FIR Filter 04
OR
7 a With necessary mathematical analysis explain the frequency sampling technique of 06
FIR filter design
b A lowpass FIR filter has the desired frequency response, 10


H d ( w)  e  j 3 w for 0  w 
2

 0 for  w
2
Determine the impulse response of the filter by frequency sampling methods. The
length of filter N=7
c Discuss the necessary steps involved in design of FIR filter using Kaiser Window 04
Technique
*******
U.S.N.

BMS College of Engineering, Bangalore-560019


(Autonomous Institute, Affiliated to VTU, Belgaum)

January 2017 Semester End Make Up Examinations


Course: Digital Signal Processing Duration: 3 hrs
Course Code: 16EC5DCDSP Max Marks: 100

Date: 12.01.2017
Instructions: Answer any FIVE question
Chebyshev Filter Table may be used

UNIT 1
1 a Let xp(n) be a periodic sequence with fundamental period N. Let X1(K) denote the 07
N-point DFT one period of xp(n) and X3(K) be the 3N-point DFT of three periods of
xp(n). Build the relationship between X1(K) and X3(K) for 0≤ k≤ N-1

b Consider the DFT Pair 05

x(n)  X ( K )  {4,  j 2, 0, j 2}
with N  4. Solve the DFT of x(  n N
)
c Let x(n) be the sequence x(n)=2δ(n)+δ(n-1)+δ(n-3) 08
Find y(n)=x(n)*x(n) ie. 5 point circular convolution x(n) with x(n) itself.

UNIT 2
2 a Determine the Output y(n) of a filter whose impulse response is h(n)={1, 1, 1} and 10
input signal x(n)={3,-1,0,1,3,2,0,1,2,1} using
i) Overlap Save Method
ii)Overlap Add Method
b Given x1(n)={1, -1, 1} and x2(n) ={ 2, 2, 2} Find the linear convolution using 05
Circular convolution
c Establish the relation between DFT and Z-transform ? 05
UNIT 3
3 a Derive the complete decimation in frequency FFT algorithm for a 8-point sequence. 10
Draw the neat signal flow graph mentioning all intermediate outputs.
b Compare computational complexity of DFT and DIT-/ DIF-FFT 04
c In calculating a DFT of a 4 point sequence, x(n) was noted down as x(n)= [j, 1, -j,1] 06
and the DFT was noted down as X(K)= { 1+j, 3+2j, 2+3j, 3}.
Later it was noted found that only x(0), x(1) and X(1), X(3) were correct values.
Illustrate the correct values of X (K) ?
UNIT 4
4 a Design an analog Butterworth lowpass filter that has a gain of -2dB at 20 rad / sec 07
and attenuation in excess of 10 dB beyond 30 rad/sec.
(Do not use filter tables)

b 2 07
For the analog transfer function H ( s)  . Design of IIR filter using
( s  1)(s  2)
impulse invariance method. Assume T=1 sec.
c Realize the system with difference equation 06
3 1 1
y(n)  y(n  1)  y(n  2)  x(n)  x(n  1) in cascade form
4 8 3
OR
5 a Design and realize a digital LPF using BLT to meet the following specifications. 10

a) Monotonic passband and stopband


b) -3dB cutoff at 0.5П rad/sec
c) -15dB attenuation at 0.75П rad/ sec
( Do not use filter table)

b Design a lowpass 1.4531 rad / sec bandwidth chebyshev filter with the following 10
specifications
i) Acceptable passband ripple of 1 dB
ii) Cutoff random frequency of 1.4531 rad/sec
i) Stopband attenuation of 20 dB or greater beyond 6.1536 rad/ -15dB
attenuation at 0.75П rad/ sec
(Filter table may be used)

UNIT 5
6 a Design a lowpass digital filter to be used in an A/D-H(z)-D/A structure that will 10
have a -3dB cutoff at 30П rad/sec and an attenuation of 50 dB at 45П rad/sec. The
filter is required to have a linear phase and the system will use a sampling rate of
100 samples / sec.

b Realize the following system function in Direct form 05

3 17 3
H ( Z )  1  Z 1  Z 2  Z 3  Z 4
4 8 4

c Explain the Gibbs phenomenon 05


OR
7 a Determine the impulse response h(n) of a filter having desired frequency response, 12



H d (e jw )  e  j ( N 1) / 2 for 0 w 
2

0  w 
for
2
N=7 use frequency sampling approach

b Obtain the linear phase realization for the following equation 08


3 17 3
H ( z )  1  Z 1  Z 2  Z 3  Z 4
4 8 4

*******
U.S.N.

BMS College of Engineering, Bengaluru-560019


Autonomous Institute Affiliated to VTU

December 2017 Semester End Main Examinations

Course: Digital Signal Processing Duration: 3 hrs


Course Code: 16EC5DCDSP Max Marks: 100
Date: 14.12.2017
Instructions: Answer five full questions choosing one from each unit

UNIT 1
1 a Compute the N-point DFT of x(n)=a for
n
0≤n≤N-1. 4
b For x(n)={1,-2,3,-4,5,-6} without computing its DFT, find following. 12
i) X(0) ii)∑𝑘=0 𝑋 𝐾 iii)X(3) iv)∑𝑘=0 |𝑋 𝐾 |2.
c Find 4-point DFT of x(n)=cos[nπ/4] using matrix method or using fundamental 4
definition.
OR
2 a An analog signal is sampled at 10 KHz and the DFT of 512 samples is computed. 6
Determine the frequency spacing between spectral samples of DFT. Determine the
frequency spacing of 1024-point DFT is calculated.
b Determine 8-point DFT of the signal x(n)={1,1,1,1,1,1,0,0}. 14
Also sketch its magnitude and phase.
UNIT 2
3 a Given x(n)={1,2,3,4} and h(n)={1,2,2}. Compute 8
i) Circular convolution
ii) Linear convolution
iii)Linear convolution using circular convolution
b Determine the Output y(n) of a filter whose impulse response is h(n)={1, 1, 1} and 12
input signal x(n)={3,-1,0,1,3,2,0,1,2,1} using
i) Overlap Save Method
ii) Overlap Add Method
UNIT 3
4 a What is the speed improvement factor in calculating 64-point DFT of a sequence 6
using direct computation and FFT algorithm? Also mention the number of registers
required.
b Write a note on geortzel algorithm. 4
c Compute IDFT of {0, 2-j4.8284, 0, 2+j0.8284, 0, 2-j0.8284, 0, 2+j0.8284} using 10
DIF-FFT algorithm.
OR
5 a Obtain the digital filter equivalent of the analog filter shown in fig below, where 12
R=1 ohm, C=1F

using (i)IIT (ii)BLT.


Assuming fs=8fc, where Fc is cut off frequency of the filter.
b The transfer function of a discrete causal system is given as 8
H (Z)=1-Z-1/1-0.2Z-1-0.15Z-2
i) Find the difference equation
ii) draw cascade &parallel realization
UNIT 4
6 a Design a digital IIR LP Butterworth filter that has 2dB pass band attenuation at a 10
frequency of 300𝜋 rad/sec & at least 60dB stop band attenuation at 4500𝜋
rad/sec.Use BLT.
b Design a lowpass 1.4531 rad / sec bandwidth chebyshev filter with the following 10
specifications
i) Acceptable passband ripple of 1 dB
ii) Cutoff random frequency of 1.4531 rad/sec
iii) Stopband attenuation of 20 dB or greater beyond 6.1536 rad/ Sec
( Filter table may be used)
UNIT 5
7 a 12
Design an Ideal BP FIR filter with frequency response H(ejw)= 1, for,
𝜋 𝜋
≤𝜔≤
use rectangular window with N=11.
b Bring out the difference between FIR & IIR filter 4
c Explain Gibb’s phenomenon in design of digital filter 4
*******
U.S.N.

BMS College of Engineering, Bengaluru-560019


Autonomous Institute Affiliated to VTU

January 2018 Semester End Make Up Examinations

Course: Digital Signal Processing Duration: 3 hrs


Course Code: 16EC5DCDSP Max Marks: 100
Date: 09.01.2018

Instructions: Chebyshev filter table to be supplied

UNIT 1
1 a a) Find N point DFT of 10
i) x(n) = Cos2(2πkon/N)
ii) x(n) = (-1)n for N=8
b 4 Point DFT of x(n) is X(k)= [10,-2+2j,-2,-2-2j]. Find DFT of y(n) = (-1)n x(n). 5
c Given the sequences x1(n) = [1,2,3,4] and x2(n) = [2,1,2,1] find y(n) such that 5
Y(k) = X1(k)X2(k)
OR
2 a Find N point DFT of 8
i) x(n) = sin(2πkon/N)
ii) x(n) = cosh(an)

b If x(n)= [ 1,2,3,4,5,6], find the sequence y(n) with 6-point DFT Y(K)=W2kX(K) 6
c If x(n) is a real valued sequence with N point DFT X(K), prove that 6
i) X(0) is real.
ii) X(N/2) is real.
UNIT 2
3 a Obtain the response of a system given, x(n) = [1,2,3,4] and h(n) = [2,1] using 8
i) Linear Convolution
ii) Circular Convolution
b Find the output of a system whose impulse response h(n) = [1,1,1] for an input 12
x(n) = [ 3,-1,0,1,3,2,0,1,2,1] using
i) overlap add method
ii) overlap save method ,with 5 point circular convolution
UNIT 3
4 a Obtain the 8 point DFT of a sequence x(n) = [ 0,1,2,3,4,5,6,7] using radix 2 DIT- 8
FFT algorithm
b Obtain the 8 point IDFT of X(K) = [4,1-j2.414,0,1-j0.414,0,1+j0.414,0,1+j2.414] 8
using radix 2 DIF-FFT algorithm for computing IDFT.
c Given x(n)=[1,0,1,0] find X(2) using the Goertzel algorithm . 4
OR
5 a Design a analog Butter worth low pass filter to meet the following specifications 10
passband gain = 0.89, passband edge frequency =30 Hz, attenuation = 0.20, stop
band edge frequency = 75 Hz.
b Obtain the series and parallel form realization for a digital filter described by the 10
system function H(z) = (8z3-4z2+11z-2) /(z-1/4)(z2-z+1/2)
UNIT 4
6 a Design a analog chebyshev low pass filter to meet the following specifications 8
passband ripple=-0.5db, passband edge frequency =1 rad/sec, attenuation = -30db
stop band edge frequency = 3 rad/sec.
b Design an IIR digital low pass filter which is monotonic in PB and SB to meet the 12
following specifications -1db cutoff at 100π rad/sec, stop band attenuation of
35 db at1000π rad/sec and a sampling rate of 2000 samples/sec use bilinear
transformation technique
UNIT 5
7 a Design a band pass FIR filter using Hamming window given lower cut off frequency 10
of 1 rad/sec, higher cut off frequency of 2 rad/sec and window length = 7. Obtain its
magnitude response.
b Design a linear phase FIR filter with 7 taps and cut off frequency of0.3π rad using 10
the frequency sampling method
*******
U.S.N.

B.M.S. College of Engineering, Bengaluru-560019


Autonomous Institute Affiliated to VTU

July / August 2019 Supplementary Examinations


Programme: B.E. Semester : V
Branch : ELECTRONICS AND COMMUNICATION ENGG Duration: 3 hrs.
Course Code: 16EC5DCDSP Max Marks: 100
Course: Digital Signal Processing Date: 27.07.2019

Instructions: 1. Answer any FIVE full questions, choosing one full question from each unit.
2. Missing data, if any may suitably assumed.
3. Normalised filter table can be used.

UNIT - I
Revealing of identification, appeal to evaluator will be treated as
Important Note: Completing your answers, compulsorily draw diagonal cross lines on the remaining

1 a) 1.a) The DFT of a sequence x(n) is given by X(K) ={10, -2+j2 , -2, -2-j2}. 6
Determine i) x1(n) = x((−n)) 4 ii) x2(n) = x((n − 2)) 4
b) Obtain the relations for DFT X(k) using symmetry property for a complex 8
valued sequence x(n); 0≤n≤N-1. Considering x(n) being even, odd, real
imaginary
c) Obtain the Parseval’s relations for energy for the N length sequence x(n) 6
with its DFT X(k). Compute the energy |𝑋(𝑘)|2 for N=6 length sequence
x(n)=[1, 3, -2, 1, -3, 4],
OR
2 a) If x(n) is N point sequence with DFT X(K). if x1(n) = x((n-1))8, and given 8
first five points of 8 point DFT of x(n) is
{16, (2√2+j2√2), 0, (2√2-j2√2), 0}, find X1(K) .
b) A complex sequence z(n)=x(n)+jy(n) with DFT Z(k). Find X(k), Y(k). 8
Given Z(k)={2+j2, 1.41+j3.41, 0, -0.58+j1.41}.
c) An analog signal x(t) =2 sin(2000πt) is sampled at fs=8000Hz. Frequency 4
resolution required to compute N point DFT is 8Hz. Determine frequency in
Hz that corresponds to bin X(250) and X(500).

UNIT - II
3 a) A long sequence x(n) is filtered through a filter with impulse response h(n) 8
to yield output y(n). If x(n)=[3,-1,0,1,3,2,0,1,2,1] and h(n)= [1,1,1]
Implement the 6-point circular convolution by overlap save method
.Indicate number of DFTs required for computation
b) Express the relation between DFT and Z transform 6
c) Find the response of the linear filter(linear convolution) with impulse 6
response h(n)= [1 2 3] when excited with input x(n)=[1 2 2 1] using circular
convolution.
blank pages.
malpractice.

UNIT - III
4 a) Compute the 8-point DFT of the sequence 𝑥(𝑛) = (1)𝑛 , 0 ≤ n ≤ 7 using the 10
DIT-FFT algorithm. Show all the intermediate results. Indicate
computational complexity of this algorithm and direct computation
b) Find the output y(n)of the system with impulse response h(n) =[1 0 -1 0 ] 10
excited by input x(n)=[1 2 3 4] using DIF FFT algorithm, with N=4.

UNIT - IV
5 a) A IIR digital LPF has following specifications. Compute the order for 10
i) Butterworth approximation ii) Chebyshev type 1 approximation. Use
bilinear transformation given with sampling frequency 100kHz.
PB ripple ≤ 1.5dB PB edge frequency 0.1π rad/sec
SB attenuation ≥ 50dB SB edge frequency 0.5π rad/sec
iii) Why is pre-warping required in bilinear transformation
b) Realize the digital IIR filter using DFI and DFII and transpose structures 10

0.72𝑧 −2 + 1.42𝑧 −1 + 0.72


𝐻(𝑧) =
0.51𝑧 −2 + 1.342𝑧 −1 + 1
OR
6 a) Design a digital Butterworth LPF with transfer function H(z) using bilinear 12
transformation with T=1sec.Specifications are given below.
PB Ripple 7dB, SB attenuation of 16dB
PB edge frequency Ω𝑝 = 0.2𝜋, SB edge frequency Ω𝑠 = 0.3𝜋,
b) Obtain parallel form structure and cascade structure of digital IIR filter 8
1
1 − 2 𝑧 −1
𝐻(𝑧) =
1 1
(1 − 3 𝑧 −1 )(1 − 4 𝑧 −1 )

UNIT - V
7 a) Design an symmetric causal LP FIR filter with Hanning window . assume 10
𝑒 𝑗3𝑤 ; −𝜔𝑐 ≤ 𝜔 ≤ 𝜔𝑐
M=7, 𝜔𝑐 =1rad/sec, given 𝐻𝑑 (𝑒 𝑗𝑤 ) = {
0 ; 𝜔𝑐 ≤ 𝜔 ≤ 𝜋

b) Obtain the DF I structure of FIR filter system given by 5


1 1 1 1 1
𝐻(𝑧) = 1 + 𝑧 −1 + 𝑧 −2 + 𝑧 −3 + 𝑧 −4 + 𝑧 −5 + 𝑧 −6
2 3 6 3 2
c) What is Gibbs phenomenon? How is this reduced in FIR filter? 5

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