MCKV Institute of Engineering

MODEL QUESTIONS
DIGITAL SIGNAL PROCESSING

Discrete time Signals

Theory
1. Define the followings:
i. Signal and system ii. Unit sample sequence iii. Unit step signal iv. Deterministic and non-
deterministic signal v. Periodic and aperiodic signal vi. Even and odd signal vii. Energy and power
signal viii. Static system and dynamic system ix. Causal and non-causal system x. Time invariant and
time varying system
2. How can you differentiate continuous time signal, discrete time signal and digital signal?
3. Write advantages and disadvantages of digital signal processing. Write its applications.
4. Show that a discrete signal is periodic if its frequency is a rational number.
5. State the “sampling theorem”.

Problems
1. Determine the power and energy of unit step sequence. [ans. ½, infinity].

2. Determine the energy of the following sequence

⎧⎛ 1 ⎞n
⎪⎜ ⎟ for n ≤ 0
x[n] = ⎨⎝ 2 ⎠ [ans: 2]
⎪
⎩0 for n < 0

3. Determine whether the following sequences are periodic or not. If periodic, determine the fundamental
period.
i. x[n]=sin(6πn/7) [Hints: f=3/7,ratio of two integer, periodic with period N=7 samples]
ii. x[n]=sin(n/8) [Hints: f=1/16π, not the ratio of two integer, aperiodic ]

4. Test if the following systems are time invariant or time variant

i. y[n]=x[n]-x[n-1] ii. y[n]=nx[n] iii. y[n]=x[-n] iv. y[n]=x[n]cos ω0n
[ans: i. Invariant ii. Variant iii. Variant iv. Variant ]

5. Test if the following systems are linear or non-linear

i. y[n]=nx[n] ii. y[n]=x[n2] iii. y[n]=x2[n] iv. y[n]=Ax[n]+B v. y[n]=ex[n]
[ans: i. Linear ii. Linear iii. Non- Linear iv. Non- Linear v. Non- Linear ]

6. Test if the following systems are causal or non-causal

n
i. y[n]=x[n]-x[n-1] ii. y[n] = ∑ x ( k ) iii. y[n]=ax[n] iv. y[n]=x[n]+3x[n+4] v. y[n]=x[n2]
k =−∞
∞
vi. y[n]=x[2n] vii. y[n]=x[-n] viii. y[n] = ∑ x (k )
k =−∞
[ans: i. causal ii. causal iii. causal iv. Non-causal v. Non-causal vi. Non-causal vii. Non-causal
viii. Non-causal]
7. Test if the following systems are stable or not
n +1
i. y[n]=x[n]cos x[n] ii. y[n] = ∑ x ( k ) iii. y[n]=x[n]+ e α
x[n]
k =−∞
[ans: i. Unstable ii. Unstable]

8. A signal x(t)=3cos200πt+2cos500πt is uniformly sampled at a rate 150samples/second. Determine the

frequency information carried by the sampled version of x(t). (Proakis)

Discrete-time LTI Systems

Theory
1. What do you mean by LTI system? Write the necessary condition for LTI system.
2. Define convolution. Write the application of it.
3. State and prove the condition of causality.
4. State and prove the condition of BIBO stability.

Problems

1. Determine the convolution sum of two sequences

{
x[n]={3,2,1,2}; h [ n ] = 1, 2,1, 2
↑
} (Ramesh Babu)

2. Determine the impulse response for the cascade of two LTI system having impulse responses
n
⎛1⎞
h1 [ n ] = ⎜ ⎟ u [ n ]
⎝2⎠
n
⎛1⎞
and h2 [ n ] = ⎜ ⎟ u [ n ] (Proakis)
⎝4⎠

3. Determine the unit step response of the LTI system with impulse response
h [ n] = a n u [ n] (Proakis)

4. A DTLTI system with impulse response h [ n ] = {1,1,1} is excited by a sequence x[n]={4,3,2,1}.

Determine the output.

z-transform

Theory
1. Define z-transform. What do you mean by ROC? Explain it for right handed sequence and left hand
sequence.
2. Point out the properties of ROC of z-transform.
3. State and prove the convolution property of z-transform.
4. What are the different methods to find inverse z-transform?

Problems
(All problems are from “Ramesh Babu”)

1. Find the z transform and ROC of the following signals:

i. x [ n ] = a n u [ n ] ii. x [ n ] = −b n u [ −n − 1] iii. x [ n ] = sin nθ u [ n ] iv. x [ n ] = r n cos nθ u [ n ]
2. Find the inverse z transform using long division method
z + 0.2
i. X ( z ) = z >1
( z + 0.5)( z − 1)
z
ii. X ( z ) = z <3
( )( z − 4 )
z − 3

3. Find the inverse z transform using partial fraction expansion method

X (z) =
(
z z2 − 4z + 5 )
( z − 3)( z − 2 )( z − 1)
For i. 2 < z < 3
ii. z > 3
iii. z < 1

4. Find the inverse z transform using residue method

z +1
X (z) = z >1
( z + 0.2 )( z − 1)

Realization of Digital Filter

Theory

1. What are the different type of structures for IIR systems realizations?
2. Write computational load and storage requirements of Directform I and II.
3. Why cascade form realization is popular?
4. When cascade form realization is preferred in FIR filter? (complex zeros with absolute magnitude less
than one)

Problems

1. Obtain direct form I and II and cascade form realization of a system described by
3 1 1
y [ n ] − y [ n − 1] + y [ n − 2] = x [ n ] + x [ n −1]
4 8 2

2. Determine a cascade realization of the system characterized by the transfer function

2 ( z + 2)
H (z) =
z ( z − 0.1)( z + 0.5 )( z + 0.4 )

3. Obtain the parallel realization of the following IIR digital filter

H ( z) =
(
3 2z 2 + 5z + 4 )
( 2 z + 1)( z + 2 )

IIR Filter

Theory
1. What are the advantages of digital filter over analog filter?
2. What do you mean by IIR and FIR filter? Write there merits and demerits.
3. Compare the IIR filter with the FIR filter.
4. What is the impulse invariant technique? Obtain the mapping formula for the impulse invariant
transformation. Discuss the stability of this technique. Write the demerits of this method.
5. What is bilinear transformation? Prove that in bilinear transformation the mapping from s plane to z
2 1 − z −1
plane is s = . Comment on the stability of this method. How does the bilinear transformation
T 1 + z −1
method differ from the impulse invariance method?
6. What is warping effect and what is its consequence? How can you avoid this effect?
7. What is meant by order of a filter?

Problems
(All problems are from “Ramesh Babu”)
2
1. For the analog transfer function H ( s ) = determine H(z) using impulse invariance
( s + 1)( s + 2 )
method. Assume T=1 sec.

2. Repeat prob. 1 for sampling frequency 2Hz.

10
3. For the analog transfer function H ( s ) = determine H(z) using impulse invariance method.
s + 7 s + 10
2

Assume T= 0.2sec.

4. Given the specification α p = 1dB , α s = 30dB , Ω p = 200rad/sec and Ω s = 600rad/sec . Determine

the order of the filter.

5. Using bilinear transformation design a digital bandpass Butterworth filter with the following
specifications:
Sampling frequency =8 KHz
α p = 2dB in the pass-band 800Hz ≤ f ≤ 100Hz
α s = 20dB in the stop-band 0 ≤ f ≤ 400Hz and 2000Hz ≤ f ≤ ∞

6. Design a digital Butterworth filter satisfying the constraints

( ) for 0 ≤ ω ≤ π2
0.707 ≤ H e jω ≤ 1

H ( e jω ) ≤ 0.2 for
3π
≤ω ≤π
4
with T=1sec using (a) the bilinear transformation (b) Impulse invariance.

FIR Filter

Theory
1. What are the conditions for the impulse response of FIR filter to satisfy for constant group delay and
constant phase delay?
2. What do you mean by the term “window” in designing FIR filter?
3. Describe in detail any one type of window method of designing a FIR filter.
4. What are the desirable properties of a “window”?
Problems
1. Design an ideal high pass filter with a frequency response
( )
H d e jω = 1 for π 4 ≤ ω ≤ π
= 0 for ω ≤ π 4
Find the values of h[n] for N=11. Use rectangular window and Hamming window. Plot the magnitude
response.
(Ramesh Babu)

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

( )
H d e jω = e − j 2ω for −π 4 ≤ ω ≤ π 4
= 0 for π 4 < ω < π
Find the values of h[n] for following window function
⎧1 0 ≤ n ≤ 4
w [ n] = ⎨
⎩0, otherwise
Determine the frequency response of the designed filter.
(Salivahanan)

DFT and FFT

Theory
(See Questions and Answers of “Ramesh Babu” )

1. Define discrete Fourier transform (DFT)?

2. What is Twiddle factor? What are the properties of Twiddle factor?
3. Differentiate discrete time Fourier transform (DTFT) and DFT.
4. State and prove the time shifting property of DFT.
5. Explain the relationship between DFT and z transform.
6. Discuss various problems of pitfalls in using DFT.
7. Explain Parseval’s theorem for discrete time sequences.
8. State the difference between (i) overlap-save method and (ii) overlap-add method.
9. What do you understand by periodic convolution? Define circular convolution.
10. Distinguish between linear and circular convolution.
11. What is zero padding? What are its uses?
12. What is fast Fourier transform (FFT)? What is butterfly structure?
13. What is meant by radix-2 FFT?
14. Differentiate between DIT and DIF algorithm.
15. How many stages are there for 16 point FFT?
16. How many multiplication terms are required for doing 64 point DFT by expressional method and FFT
method? Or Show that when algorithm is used to find out DFT, the number of complex addition and
multiplication is reduced.

Problems
(All problems are from “Ramesh Babu”)

1. Find the DFT of a sequence x[n]={1,1,0,0} and DFT of Y(k)={1,0,1,0}.

2. Compute the DFT of a sequence (-1)n for N=4.

3. Find the output y[n] of a filter whose impulse response is h[n]={1,1,1} and input signal x[n]={3,-1,0,
1,3,2,0,1,2,1} using (i) overlap-save method (ii) overlap-add method.
4. Find the circular convolution of two finite duration sequences x1[n]={1,-1,-2,3,-1}; x2[n]={1,2,3}.

5. Find the DFT of a sequence x[n]={1,2,3, 4,4,3,2,1} using DIT algorithm.

6. Find the DFT of a sequence x[n]={0,1,2,3} using DIT algorithm.

Short notes.

1. Circular Convolution
2. Gibb’s phenomenon
3. Design of FIR filter using window method
4. overlap-add and overlap-save method
5. BIBO stability
6. Mapping of S-plane to z-plane
7. DIF algorithm
8. Architecture of Digital Signal Processor