Guru Nanak Institutions Technical Campus (Autonomous)

ECE Department
DSP (EC0642) Question Bank (Previous Papers)

PART A
Unit I

1) What are the applications of digital signal processing? (2 marks) (March 2017)
2) Explain region of convergence and its properties. (3 marks) ( March 2017)
3) Define decimation and interpolation. (2 marks) (November 2016)
4) State linear and static system with example. (3 marks) (November 2016)
5) Write four advantages of digital signal processing over analog signal
Processing? (2 marks) (May 2016)
6) Show that the frequency response of a discrete system is a periodic function of frequency.
(3 marks) (May 2016)
7) Show that δ ( n )=u ( n ) −u ( n−1 ) . (2marks)(April 2018)
8) Find the z-transform of f ( n )=n2 u(n) (3marks)(April 2018)

Unit II

1) What is zero padding? What are its uses? (2 marks) (March 2017)
2) State and prove time shifting property of DFT. (3 marks) ( March 2017)
Π
3) If x(n)= cos n , find spectra of the signal? (2 marks) (November 2016)
3
4) How many multiplications and additions are required to compute N-point DFT using
radix – 2 FFT? (3 marks) (November 2016)
5) Give the relation between DTFT and Z- Transform. (2 marks) (May 2016)
6) Distinguish between linear convolution and circular convolution. (3 marks) (May 2016)
7) State and prove the any three properties of DFT.(2marks)(April2018)
8) What is the basic operation of DIF algorithm? (3marks)(April2018)

Unit III

1) Give any two properties of Butterworth low pass filter. (2 marks) (March 2017)
2) State the properties of IIR filter. (2 marks) (November 2016)
3) State the methods used to prevent overflow. (3 marks) (November 2016)
4) What are the advantages of Butterworth filter? (2 marks) (May 2016)
 5) What are the advantages and disadvantages of Chebyshev filter. (3 marks) (May 2016)
6) What are the properties of Butterworth Low pass filter? (2marks)(April2018)
7) Discuss the stability of the impulse invariant mapping technique. (3marks)(April2018)

Unit IV

1) What is Gibbs phenomenon? (2 marks) (March 2017)

2) Explain the procedure for designing FIR filters using windows. (3 marks) ( March 2017)
3) What is warping effect? What is its effect on magnitude and phase response? (3 marks)
(March 2017)
4) Give the equation for Hamming window and Blackman window. (2 marks)
(November 2016)
5) What is dead band of a filter? (3 marks) (November 2016)
6) Define impulse response. (2 marks) (May 2016)
7) Define sampling and Nyquist rate. (3 marks) (May 2016)
8) Explain the effects of truncating an infinite Fourier series into a finite series. (2marks)
(April2018)
9) What is the condition for the impulse response of FIR filter to satisfy for constant group
and phase delay and for constant group delay? (3marks)(April2018)

Unit V

1) What is over flow oscillations? (2 marks) (March 2017)

2) What is the need for anti-aliasing filter prior to down sampling? (3 marks) ( March 2017)
3) Give the steps in multistage sampling rate converter design. (2 marks) (November 2016)
4) Write any four applications of Multi – rate signal processing. (3 marks) (November 2016)
5) Define Decimation. (2 marks) (May 2016)
6) What is the need for Multi rate Digital Signal Processing? (3 marks) (May 2016)
7) What is the need for MultiMate Digital Signal Processing? (2marks)(April2018)
10) What do you mean by quantization step size? (3marks)(April2018)
 PART B

Unit I

1) Determine the impulse response h(n) for the system described by the second order
difference equation y (n) – 4 y (n-1) + 4 y (n-2) = x (n-1) ( March 2017)
2) Find the magnitude and phase response for the system characterized by the difference
equation y (n) = x (n) + x (n-1) + x (n-2) ( March 2017)
3) Check the following filter for time invariant, causal and linear
i) y (n) = (n-1) (n+1)
ii) y(n) = x (n-2) ( March 2017)
4) Draw the structures of cascade and parallel realizations of
(1−z −1)
H(z) =
( 1 −2
) 1 −1 ( March 2017)
1− z (1− z )
2 8

5) If y(n) = 12x(n-1) + 11x(n-2), Find whether given system is time variant or not?.
( ( November 2016)
6) Find the frequency response of 1st order system y(n) = x(n) + ay(n-1).
(November 2016)

7) y(n)-3y(n-1) – 4y(n-2) = 0 determine zero-input response of the system; Given y(-2) =0

and y(-1) =5 ( November 2016)
8) Obtain the cascade and parallel form realization for the system
y(n) = −0.1 y(n −1) + 0.2 y(n − 2) + 3x(n) + 3.6x(n −1) + 0.6 x(n − 2). (November 2016)
9) Test the following systems for linearity, Time invariance, causality and stability
y(n) = sin(2npif/F)x(n)
(May 2016)
10) A digital system is characterized by the following difference equation:
Y(n) = x(n)+ay(n-1). Assuming that the system is relaxed initially, determined its
impulse response. (May 2016)
11) By taking an example compute DFT by using overlap save method. (May 2016)

12) A causal system is represented by the following difference equation:

 y(n)+(1/4)y(n-1) = x(n)+(1/2)x(n-1). Find the system transfer function H(z), unit sample
response, magnitude and phase function of the system. Determine direct form I and II
for the second order filter given by:
Y(n)=2bcoswoy(n-1)-b2y(n-2)+x(n)-bcoswoy(n-1) (May 2015)

13) Determine the range of values of ‘n’ for which an LTI system represented below is stable,
h[n] = an for n≥0
= 0 otherwise (June 2014)

14) Write whether an LTI system with an impulse response represented below is stable or
not? Justify.
h[n] = 3n u[-n-1] (June 2014)

15) Write whether an LTI system with an impulse response represented below is causal or
not? Justify.
h[n] =u[n+2]-u[n-2] (June 2014)
16) Write a short note on Time and Frequency domain input-output relationships of an LTI
system. (Dec 2013)
17) An LTI system is characterized by its impulse response h(n) = (½)n u(n). Determine the
spectrum and energy density spectrum of the output signal when the system is excited by
the signal x(n) = (¼)n u(n). (Dec 2013)
18) a) An LTI system is characterized by an impulse response

()
n
3
h (n)= u (n)
4
Find the step response of the system. Also, evaluate the output of the system at n=±5.
(April 2018)
19) Consider adiscrete time system characterized by the following input-output relationship
( n ) =x ( n−2 )−2 x ( n−17 ) . Determine whether the system is memory less, time-invariant,
linear, causal and stable. (April 2018)

20) Given the difference equation y ( n ) +b2 y ( n−2 ) =0 for n≥0 and |b|<1. With initial
conditions y(-1)=0 and y(-2)=-1, show that
nπ
y ( n )=bn+2 cos ⁡( ) (April 2018)
2
21) Find the z-transform of the sequence f(n) defined below: (April 2018)
 {
n
3 n<0
1 n
f ( n )= 3 ) n=0,2,4 …
(

() 1 n
2
n=1,3,5 …

Unit II
1) Determine the 8 point DFT of the sequence

X(n) = {1;0 ; −4 ≤n ≤ 4
ot h erwise
( March 2017)

2) Compare overlap-save method and overlap-add method. ( March 2017)

3) Compute 4-point DFT of a sequence x (n) = { 0,1,2,3} using DIT algorithm.
( March 2017)
4) Find the IDFT of the sequence using DIF algorithm X (k) = {10, -2-j2, -2, -2+j2}
( March 2017)
5) Explain all properties of DTFT. (Nov 2016)
6) Find the DFT of a sequence x(n)= {1,1,0,0} and find the IDFT of Y(k)={1,0,1,0}
(Nov 2016)
7) Explain Radix – 2. Decimation in time FFT algorithm with example. (Nov 2016)
8) Compute the circular convolution of the sequences

X1(n) = {1,2,0,1} and

X2(n) = {2,2,1,1} using DFT approach. (May 2016)

9) What is FFT? Calculate the number of multiplications needed in the calculation of DFT
using FFT algorithm with 32 point sequence. (May 2016)
10) Prove the following properties
x*(n)→X*((-K))NRN(K)
x*((-n))NRN(n) →Xep(k) = ½[X(K)N+X*(-K)N]RN(K) (May 2016)
11) Compute FFT for the sequence
x(n) = {1,0,1,1,0,1,1,1} (May 2016)
12) State and prove the following properties of DFT:
(a) Linearity
(b) Frequency shifting (May 2015)
13) Explain the following
(a) Overlap Add Method
 (b) Overlap Save Method (May 2015)
14) Develop the DIF FFT algorithm for N=8. Using the resulting signal flow graph compute
the 8-point DFT of the sequence x(n) = sin (pi/2 n), 0≤n≤7. (May 2015)
15) If x2(n) = x1(-n) without performing FFT find X2(k) using X1(k). (May 2015)
16) Perform Linear convolution of the x1[n] and x2[n] sequences using Over-lap add method
x1[n] = {3,-1,0,1,2,3,0,1,1,2} and x2[n] = {1, 1, 1} (June 2014)
17) Obtain the relationship between DTFT and DFS. (June 2014)
18) Compute the DFT of the sequence x[n] = {1,2,3,4,3,4,1,2} using DIT FFT algorithm.
Write the signal involved. (June 2014)
19) Compute IDFT of X(k) = {4,-1,0,-1} using FFT techniques. (June 2014)
20) Discuss the procedure of computing linear convolution using Over-lap add method.
(Dec 2013)
21) Perform Linear convolution of the two sequences x(n) = {1, 2, 3, -1, -2, -3, 4, 5, 6} and
h(n) = {2, 1, -1} using Over-lap add method. (Dec 2013)
22) Give the general procedure of computing FFT using DIF algorithm and develop the basic
butterfly structure and necessary equations. (Dec 2013)
23) How the computational speed of FFT algorithm has been improved over DFT.
(Dec 2013)
24) Find the IDFT of the sequence X(k) given below
X(K)={1,0,0,j,0,-j,0,0} (April 2018)

25) Obtain the 10 point DFT of the sequence x ( n )=δ ( n ) +2 δ ( n−5 ) (April 2018)
Find the IDFT of the sequence X(k)={20,-5.828-j2.414,0,-0.712-j0.414,0,-
0.172+j0.414,0,-5.828+j2.414} using DIT-FFT algorithm. (April 2018)

26) Using FFT and IFFT, determine the output of system if input x(n)={2,2,4} and impulse
response h(n)={1,1} (April 2018)

Unit III

1) What are the steps to design an analog Chebyshev low pass filter. ( March 2017)
2) Apply bilinear transformation to ( March 2017)
 with T=1 Sec and find H (z).
3) Consider an analog filter with transfer function

Is this a Butterworth or Chebyshev filter? Obtain the transfer function

of an IIR digital filter using impulse invariant transformation. Assume T = 1 Sec.

(March 2017)

4) Design a Butterworth high pass filter satisfying:

fp = 0.32Hz ; αp = 0.5 dB

fs = 0.16Hz ; αs = 30 dB ; F = 1 Hz (Nov 2016)

5) Discuss the steps in design of IIR filter using bilinear transformation for any one type of
filter. (Nov 2016)
6) Discuss in detail about spectral transformations. (May 2016)
7) Explain how IIR digital filters are designed from analog filters. (May 2016)
8) Compare the impulsive invariance and bilinear transformation methods.
(May 2016)
9) Find the order and poles of a low pass Butterworth filter that has a -3dB bandwidth of
400 Hz and an attenuation of 20dB at 1 KHz. (May 2016)
10) Design a digital IIR low pass Butterworth filter that has a 2dB pass band attenuation at a
frequency of 300pi rad/sec and at least 60dB stop band attenuation at 4500pi rad/sec. Use
backward reference transformation. (May 2015)

11)Determine the order and the poles of a type-I low pass Chebyshev filter that satisfies the
following constrains
0.8 ≤ |H(w)| ≤ 1; 0 ≤ W ≤ 0.2pi;

And |H(w)| ≤0.2; 0.6pi≤W ≤ pi. (May 2015)

11) For the analog filter with transfer function H(s) = 2(s+1)(s+3), determine H(z) using
bilinear transformation technique. Use T =0.1sec. (June 2014)
12) Explain impulse invariant method of IIR filter design. (June 2014)
13) Explain briefly the frequency response of LTI systems. (Dec 2013)
14) Discuss direct form-I & II IIR realization structures in detail with necessary flow graphs.
(Dec 2013)
15) List out the merits and demerits of Butterworth and Chebyshev filter
 approximation techniques. (Dec 2013)
16) Prove that the relationship between analog (Ω) and digital (ω) frequency in bilinear
transformation is given by Ω = (2/T) Tan(ω/2) (Dec 2013)
17) Design a digital low pass filter using Chebyshev filter that meets the following
specifications: Pass band magnitude characteristics that is constant to within 1 dB for
recurrences below ω=0.2 π and stop band attenuation of at least 15dB for frequencies
between ω=0.3 π and π .Use bilinear transformation. (April 2018)

18) An analog filter has the following system function. Convert this filter into a digital filter
by using the impulse invariant technique: (April 2018)
1
H ( s )=
(s+ 0.1)2 +9
19) Using a bilinear transformation, design a Butterworth filter which satisfy the following
conditions: (April 2018)
jω
0.8 ≤∨H ( e ) ≤1 0 ≤ ω ≤0.2 π
¿ H (e jω )≤ 0.2 0.6 π ≤ ω ≤ π
20) Determine H(z) using impulse invariance method for the following system function:
(April 2018)

1
H ( s )= 2
( s +0.5 ) (s +0.5 s+2)

Unit IV

1) Explain the type -1 FIR filter design procedure using frequency sampling method.
( March 2017)
2) List the features of Blackman window spectrum. ( March
2017)
3) Explain the design procedure of linear phase FIR filter using Fourier series method.
( March 2017)
2 2 −1
4) Realize the system function z +1+ z by linear phase FIR structure. (Nov 2016)
3 3
5) Explain the designing of FIR filters using windows. (Nov 2016)
6) Draw and explain frequency response of FIR digital filter. (May 2016)
7) List the designing steps of FIR filters using fourier method. (May 2016)
 8) Design a high pass filter using hamming window with a cut-off frequency of 1.2
radians/second and N=9. (May 2016)
9) Design a low pass digital FIR filter using Kaiser Window satisfying the specifications
given below.

Pass band cut-off frequency = 100 Hz.

Stop band cut-off frequency = 200 Hz
Pass band ripple = 0.1dB
Stop band attenuation =20 dB
Sampling frequency = 1000 Hz. (May 2016)

10) Determine the transfer function H(z) of an FIR filter to implement h(n) =
δ ( n ) +2 δ ( n−1 )+ δ (n−2) using frequency sampling techniques. (May 2015)
11) Give the comparison between FIR and IIR filters with examples. (May 2015)
12) Compare FIR and IIR filters. (June 2014)
13) Design a high pass FIR filter whose cut-off frequency is 1.4 rad/sec and N=5 using
Hamming window. (June 2014)
14) Compare various windowing techniques with respect to sidelobes and beamwidth.
(Dec 2013)
15) Design an FIR Digital High pass filter using Hanning window whose cutoff freq is 1.2
rad/s and length of window N=9. (Dec 2013)
16) The desired frequency response of a low pass filter is given (April 2018)

{
−3 π
− jω 3π
e 4
≤ ω≤
jω 4
H d (e )=
3π
¿0 ≤ ω≤ π
4

Find H (e jω ) for M=7 using a rectangular window.

17) Explain the type II frequency sampling method of designing an FIR digital filter.
(April 2018)

18) Design a band pass filter which approximates the ideal filter with cutoff frequencies at

0.2rad/sec and 0.3 rad/sec. The filter order is M=7. Use the Hanning window function.

(April 2018)
 18) Design an ideal band pass filter with a frequency response.
(April 2018)

{
π
3π
jω 1 4 ≤∨ω∨≤
H d (e )= 4
¿ 0 ot h erwise

Unit V

1) What are the effects of finite word length in digital filters? ( March 2017)
2) Explain limit cycles in recursive structures. ( March 2017)
3) Explain interpolation process with an example. ( March
2017)
4) Explain with block diagrams how can sampling rate be converted by a rational factor
M/L both in time domain and frequency domain. ( March 2017)
5) With respect to finite word length effects in digital filters, explain over flow limit cycle
oscillation with example. (Nov 2016)
6) Explain the applications of Multi rate signal processing. (Nov 2016)
7) What is meant by signal scaling? Explain. (Nov 2016)
8) What are the dead band effects? Discuss. (May 2016)
9) What is mean by sampling rate conversion? Explain. (May 2016)
10) What are limit cycles and discuss various types of limit cycles in brief. (May 2016)
11) Discuss the process of performing sampling rate conversion by an rational
factor I/D. (May 2016)
12) Explain the characteristics of a limit cycle oscillation with respect to the system described
by the equation Y(n)=0.85y(n-2)+0.72y(n-1)+x(n)
Determine the dead band of the filter x(n) = ¾ δ (n). (May 2015)
13) Explain about multi rate signal processing and gives its examples. (May 2015)
14) Consider a single stage interpolator with the following specifications:

Original sampling rate = 1KHz

Interpolation factor L = 2
Frequency of interest = 0 – 150 Hz
Passband ripple = 0.02 dB
Stopband attenuation = 45 db

(i) Draw the block diagram for the interpolator.

 ii) Determine the window type filter length and cutoff frequency, if the window
method is used for the anti imaging FIR filter design. (May 2015)
15) Explain the process of decimation using relevant expression and block diagram.
(June 2014)
16) Explain the implementation of polyphase filter structure for interpolator.
(June 2014)
17) Write notes on:
(a) Limit cycles
(b) Over flow oscillations
© Dead band effects. (June 2014)
18) Discuss the process of Interpolation by a factor I with a neat block diagram (Dec 2013)
19) Explain the process of Interfacing of digital systems with different sampling rates with a
neat block diagram. (Dec 2013)
20) Discuss quantization errors occuring in the computation of DFT. (Dec 2013)
21) What are limit cycles and explain the types of it and also give their remedies. (Dec 2013)
22) Explain the interpolation process for an integer factor I with an example.
(April 2018)
23) Given the limit cycle behavior y ( n )=0.7 y ( n−2 ) +0.52 y ( n−1 ) + x (n). Find the
dead band of the above two systems. (April 2018)

24) The signal x(n) is defined by (April 2018)

g ( n )=
{ An
0 ot h erwise
n>0

i) Obtain the decimated signal with a factor of 3

ii) Obtain the interpolated signal with a factor of 3

1
25) Given the system y ( n )= y ( n−1 )+ x ¿) (April 2018)
2

1 n
i) Calculate the system response to the input x(n)=( ) u(n) assuming infinite precision
4
arithmetic.
ii) Calculate the response y(n), 0 ≤ n ≤ 5 to the same input assuming finite precision with five
bits, one sign bit plus four fractional bits. The quantization is performed by truncation. Discuss
the results.