EC6502-PRINCIPLES OF DIGITAL SIGNAL PROCESSING

QUESTION BANK
UNIT-1
2 marks
1.
2.
3.
4.
5.

Define DT system
How do you obtain a digital signal for DT signal?
The first five DFT values for N=8 is as follows
X(K)={28,-4+j9.656,-4+4j,-4+j1.656,-4}
Compute 4-point IDFT for X(K)={2,3+j,-4,3-j}

16 marks
1. Illustrate the construction of 8 point DFT from two 4 point DFTs. Also illustrate the reduction
of 8 point DFT to two 4 point DFTs
2. Using Decimation in time FFT algorithm compute DFT for x(n)={1,0,0,0,0,0,0,0,0}
3. Discuss the properties of DFT
4. Find the IDFT of the sequence X(K)={4,1-j2.414,0,1-j0.414,0,1+j0.414,0,1+j2.414} using
DIF algorithm
5. Find the IDFT of a sequence X(k) = {5,0,1,-j,0,1,0,1+j,0}
UNIT-2
2 marks
1. Define pass band
2. Use the backward difference for the derivative to convert analog LPF with system function
H(s)=1/S+2
3. Compare butterworth and chebyshev filter
4. What is warping effect? How to avoid it?
5. Design a digital filter using impulse invariant method for the analog filter having transfer
function H(s)=1/s+1
16 marks
1

Design a butterworth filter using bilinear transformation that satisfies the following constraint
0.7 |H(ej )| 1; 0 0.2

|H(ej )| 0.3; 0.6

( s +0.2 )2 +9

Convert the analog filter into a digital filter whose system function is H(s)=
s +0.2

use impulse invariant technique. Assume T=1 sec

( s +0.1 )2 +9

Convert the analog filter into a digital filter whose system function is H(s)
use
s+ 0.1

bilinear transformation. The digital filter should have the resonant frequency of
3

Design digital IIR LP butterworth filter to meet the following specifications

Passband gain=0.89
Passband edge frequency=30Hz
Stop band attenuation=20
Stop band edge frequency= 75Hz. Use bilinear transformation assume sampling frequency
200Hz.
Design a digital butterworth filter satisfying the constraints
0.707 |H(ej )| 1; 0 /2

|H(ej )| 0.20; 3 / 4
5

r =
4

with T=1 sec use bilinear transformation.

Describe the concept of impulse invariance method of designing IIR filter

UNIT- 3
2 marks
1. Write the frequency response of linear phase FIR filters when impulse response is antisymmetric and N is odd.
1. List the desirable window characteristics
2. List the advantages & disadvantages of FIR filter
3. Write the necessary condition for a linear phase FIR filter
4. State Gibbs phenomenon
5. State the properties of FIR filter
16 marks
j

1. Design a low pass filter whose desired frequency response is given as H d( e =

e j ; c
0 ; else

the length of the filter should be 7 and

j

2. Design a filter with Hd( e =

/ 4
{1 ;0;
/4

c = 1 rad/sec.

using Hamming and Hanning

windows
3. Determine the frequency response of FIR filter defined by y(n)=0.25 x(n) +x(n-1)+0.25 x(n2). Calculate the phase delay and group delay (8m)
4. Discuss the design procedure of FIR filter using frequency sampling method (8m)
5. Design a bandpass filter which approximates the ideal filter with cut off frequencies at 0.2
rad/sec and 0.3 rad/sec. the filter order is M=7. Use the Hanning window function.

6. A low pass filter has the desired response as given as H d( e =

/2
{ 0; ;/20

j3

UNIT- 4
2 marks
1.
2.
3.
4.
5.

What is scaling?
What is dead band of a filter?
What is product round off noise?
Define zero limit cycle oscillation
What do you understand by input quantization error?

16 marks
1.
2.
3.
4.
5.
6.

Explain about scaling? (8m)

What is meant by input quantization error? (8m)
Explain about limit cycle oscillation (8m)
Explain about product quantization error (8m)
Discuss the various common methods of quantization (8m)
Represent the following numbers in floating point format with five bits for mantissa and three
bits for exponent 7, 0.25, -7, -0.25
7. Explain the characteristics of limit cycle oscillation with respect to the system described by
the difference equation y(n)=0.95 y(n-1)+x(n),x(n)=0 and y(-1)=13. Determine the dead band
of the system.
8. The output of an ADC is applied to a digital filter with system function H(z) =0.5 z / z-0.5.
Find the output noise power from digital filter when input signal is quantized to have 8 bits.
(8m)
UNIT-5
2 marks
1.
2.
3.
4.
5.
6.
7.

State the applications of multirate signal processing

State the applications of adaptive filtering
What is meant by sampling rate conversion?
What is meant by multirate signal processing?
Define decimator and interpolator
Draw the basic building block of adaptive filter
What is sub band coding?

16 Marks
1. Implement a two stage decimator for the following specifications. Sampling rate of input
signal =20000Hz, M=100,passband=0 to 40 Hz, transition band=40 to 50 Hz, pass band
ripple=0.01, stop band ripple= 0.02
2. Explain sampling rate increase by an integer factor I and derive the input output relationship
in both time and frequency domains
3. For the multirate system shown in figure, find the relation between x(n) and y(n).

x(n)
z-

z2

4.
5.
6.
7.
8.

Discuss the need for multirate signal processing. Illustrate the decimation by a factor D.
Explain about multistage implementation of sampling rate conversion
Explain the applications of multirate DSP
Explain the application of adaptive channel equalization
Explain adaptive equalization