EC 8553 – DISCRETE TIME SIGNAL PROCESSING

QUESTION BANK

UNIT I DISCRETE FOURIER TRANSFORM

PART B
1. i) Discuss any four properties of DFT (Nov 2014/ April 2018)
ii) Perform the Circular convolution of the of the given sequence X1(n) = {1,1,2,1},
X2(n)=(1,2,3,4}
2. Develop a Radix-2, 8-point DIF FFT algorithm with neat flow chart.(NOV2015)
3. Draw a 8 point radix 2 DIT-FFT flow graphs and obtain DFT of the following sequence
x(n)={0,1,-1,0,0,2,-2,0}
4. Compute 4-point DFT of causal three sample sequence given by.

5. Find 8- point DFT of sequence of using radix-2 DIT FFT algorithm

6. Compute the DFT of sequence defined by : for (a) N = even (b)N= odd.
7. a) A finite duration sequence of length L is given as

Determine N-point DFT of this sequence for

b) Find the inverse DFT of X(k) = {1, 2, 3, 4}.
8. By means of the DFT & IDFT, determine the sequence x3 (n) corresponding to the circular
convolution of the sequence x1 (n) and x2(n).

9. By means of the DFT & IDFT, determine the response with impulse response.

10. An 8-point sequence is given by x (n) = {2, 2, 2, 2, 1, 1, 1, 1). Compute 8-point DFT of
x(n) by radix -2 DIT FFT. Also sketch the magnitude and phase spectrum. (Nov 2017)
11. An 8-point sequence is given by x(n) = (2, 2, 2, 2, 1, 1, 1, 1). Compute 8 point DFT of
x(n) by radix -2 DIF-FFT.
12. Discuss Linear filtering approach for the computation of DFT.
13. Perform the linear convolution of the of the given sequence
x(n) = {1,-1,1,-1} , h(n)=(1,2,3,4}
14. Compute the DFT of the sequence whose values for one period is given by
x(n)={1,1,-2,-2} (Nov-2013).
15. Summerize the different between overlap add and over lap save method using DFT (or)
Discuss Linear filtering approach for the computation of DFT (Nov 2013/May 2014)
16. Compute the8 Point DFT of sequence using DIT-FFT x(n)= {1 for -3≤n≤3
0 otherwise ( Nov-2013)
17.Develop a 8 point DIT FFT algorithm. Draw the signal flow graph. Determine the DFT of
the following sequence x(n)= { 1, 1, 1, 1, 0, 0, 0, 0 } using the signal flow graph. Show all
the intermediate result on the signal flow graph. (May 2014)
18. Derive DIT FFT algorithm and also find the DFT of x(n)= { 1,2,3,4,4,3,2,1} Using DIT
FFT algorithm (Nov 2014/Nov 2016)
19. a) Using linear convolution, find the response of the sequence x(n)={1,2,-1,2,3,-2,-3,
-1,1,1,1,2,-1} and h(n)={1,2}. Compare the result by solving the problem using overlap
add and overlap save method. (Nov 2016)
b) Find the IDFT of X(k) = { 6,-2+2j,-2, -2-2j} (Nov 2014)
20. Find the impulse response of the causal system y(n)-y(n-1)=x(n)+x(n-1) (Nov2015)
21. (i) Compute the IDFT of X(k) = { 6,-2+2j,-2, -2-2j}
22. State and Prove if x3(k)= x1(k). x2(k), then N-1
x3(n)= ∑ x1(m) x2((n-m))N (May 2016)
m=0
23. (i) Computer the 8 point circular convolution of x1(n)= {1,1,1,1,0,0,0,0} ,
x2(n)= sin 3πn/8, o≤ n ≤7 using matrix method
(ii) State the differences between overlap-save and overlap-add method (May 2016)
24.Determine the circular convolution of the sequence x1(n)={1,2,3,1,1,2,3,1} and
x2(n)={4,3,2,2,2,2,3,4}using DFT and IDFT. (Nov 2017)
25. Compute the DFT of x(n)= {0,1,2,3} using DIT and DIF algorithm (April 2017)
26. In an LTI system the input x(n) = {1,1,2,1} and the impulse response h(n)= {1,2,3,4}
Perform the circular convolution using DFT and IDFT (April 2017)
27. Compute 4 point DFT of x(n)= { 1,2,3,4,4,3,2,1} Using radix 2 DIF FFT algorithm
(April 2018)
28. Let xp(n) be a periodic sequence with fundamental period N. Consider the following
DFTs. (Nov 2018)
N point DFT(xp(n))= X1(k)
3N point DFT(xp(n))= X3(k)
i) What is the relationship between x1(k) and x3(k)?
ii) Verify the result in part(i) using the sequence
xp(n)= {……,1,2,1,2,1,2,1,2,…..}

UNIT II IIR FILTER DESIGN

PART B
1. Determine the cascade and parallel realizations for the system described by the system
function

2.Obtain direct form-I, direct form-Il, cascade and parallel structure for the system described
 by
(May 2013)
3.What are the limitations of IIR filter design by impulse invariance method? How they are
over come? Convert the analog filter with system function H(s) = S(S+0.1) /(S+0.1)2 + 16
into digital IIR filter by means of bilinear transformation.
4.a)Given an analog transfer function as H(S) = 1 / (S+1) (S+2). Obtain H (z) using impulse
invariant method. Take T =5 Sec. (May 2016)
b)For given analog filter system function H(S) = S+0.1 / (S+0.1)2 + 16 into digital IIR filter
by means of bilinear z-transformation. Digital filter is to have resonant frequency ωr= π/2
5. An Butterworth IIR low pass filter is to be designed to meet the following specifications.
(a) Pass-band frequency = 0 to 1.2 KHz
(b) Stop-band edge = 2 KHz
(c)Pass-band attenuation 8.5dB
(d) Stop-band attenuation 15 dB
using Bilinear transformation, Obtain the desired IIR digital filter.
6. A Chebyshev low pass filter has the following specifications:
(a) Order of the filter = 3
(b) Ripple in pass-band = 1 dB
(c) Cut off frequency = 100 Hz
(d) Sampling frequency = 1 KHz.
Determine H(z) of the corresponding IIR digital filter using bilinear transformation
technique.
7. Design a chebyshev filter for the following specification using IIT (May2013)
0.8 ≤ │H(ejω)│ ≤ 1 0 ≤ ω ≤ 0.2 π
jω
│H(e )│ ≤ 0.2 0.6 π ≤ ω ≤ π
8. The specifications of the desired low pass filter is
0.8 ≤ │H(ejω)│ ≤ 1 0 ≤ ω ≤ 0.2 π
jω
│H(e )│ ≤ 0.2 0.32 π ≤ ω ≤ π
Design Butterworth digital filter using impulse invariant transformation
9. Design a low pass Butterworth digital filter with the following specification ωs = 4000,
Ωp = 3000, Ap = 3 dB, As = 20 dB, T= 0.0001 Sec. (May 2014)
10. A system is represented by a transfer function H(z) is given by
H(z)= 3 + [4z / z-(1/2)] – [z / z-(1/4)]
i) Does this H(z) represent a FIR or IIR filter?
ii) Give a difference equation realization of this system using direct form-1
iii)Draw the block diagram for the direct form 2 canonic realization and give the governing
equation for implementation (May 2014)
11. Design a Chebyshev filter for the following specification using BLT (Nov 2014)
0.707 ≤ │H(ejω)│ ≤ 1 0 ≤ ω ≤ 0.2 π
jω
│H(e )│ ≤ 0.1 0.5 π ≤ ω ≤ π
12. i) )Convert the analog filter with system functions H(s) = 2 / (s+1) (s+2) into the digital
IIR filter by means of Bilinear Transformation method.
 ii)Obtain cascade and parallel realization of
H(z)= (1+(1/4) z-1) / (1+1/2 z-1)( 1+1/2 z-1+ ¼ z-2) (Nov 2014/ April 2018)
13.An analog system has the following system function. Convert this filter into a digital filter
using backward difference for the derivative H(s)= 1/ (s+0.1)2 + 9 (Nov 2015)
14. Convert the analog filter with system function H(s) = S+0.2 /(S+0.2)2 + 9 into digital
IIR filter by means of bilinear transformation.
15. Convert the analog filter with system function H(s) = S+0.2 /(S+0.2)2 + 9 into digital
filter. Use Impulse Invariant transformation. Assume T=1sec (Nov 2015)
16. Convert the analog filter with system function H(s) = S+0.1 /(S+0.1)2 + 16 into digital

filter using BLT. Digital filter is to have resonant frequency ωr= (Nov 2015)
4
17. A digital filter with a 3dB bandwidth of 0,25  is to be designed from the analog filter

whose system function is H(s) = c Use BLT and obtain H(z).

s  c
18. Design a third order butterworth digital filter using IIT. Assume T= 1 sec. (Nov 2016)
19.Write down steps to design digital filter using bilinear transform technique and using this
design a HPF with a pass band cutoff frequency of 1000 Hz & down 10 dB at 350 Hz the
sampling frequency is 5000 Hz. (May 2016/Nov2018)
20. Enumerate the steps for IIR filter design by impulse invariance with an example.
(Nov 2017)
21.Analyse the design of discrete time IIR filter from analog filter. (Nov 2017)
22. Obtain direct form-I, direct form-Il, cascade structure for the system described by
y(n) = 0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2) (Nov 2017)
23. Convert the analog filter with system functions H(s) = 1 / (s+a) into the digital IIR filter
by means of the impulse invariance method. (Nov 2017)
24. Design a digital Butterworth filter for the following specification using BLT (April 2017)
0.707 ≤ │H(ejω)│ ≤ 1 0 ≤ ω ≤ 0.5 π
jω
│H(e )│ ≤ 0.2 0.75 π ≤ ω ≤ π
25. Determine the system function of the lowest order Chebyshev filter with the following
specifications. 3dB ripple in the passband 0 ≤ ω ≤ 0.2 π and 25dB attenuation in
stopband 0.45π ≤ ω ≤ π
26. i) Design an analog Butterworth filter for the following specification (April 2018)
0.9 ≤ │H(jΩ)│ ≤ 1 0 ≤ Ω ≤ 0.2 π
│H (jΩ)│ ≤ 0.2 0.4 π ≤ Ω ≤ π \
ii) Apply IIT method and find H(z) of H(s) = S+a /(S+a)2 + b2 (April 2018)
27. Given the specifications αp = 3dB; αs= 12dB; fp =1KHz and fs=2KHz. Determine the
order of the filter using Chebyshev approximation. Find H(s)
28. Determine the system function of the lowest order Chebyshev filter with the following
specifications. 3dB ripple in the passband 0 ≤ ω ≤ 0.2 π and 25dB attenuation in
stopband 0.45π ≤ ω ≤ π . Use Bilinear Transformation technique
 UNIT III FIR FILTER DESIGN

PART B
1. a) Obtain a cascade realization using minimum number of multiplications for the system.

b) Realize the system function.

by using direct form structure.

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

Find the values of h (n) for N=11 using Hanning window.

3. Design an ideal band pass filter with a frequency response.

Find the values of h(n) for N=11 and plot the frequency response (Nov 2016)
4. Design an ideal band reject filter with a desired frequency response

Find the value of h(n) for N = 7 and also find H(z) using Hamming window.
5. Design an ideal highpass filter with a frequency response

Find the value of h(n) for N = 11 using (a) Hamming window (b) Hanning window
6. Discuss various steps for the design of linear phase FIR filters using window method and
explain the characteristics of window function
7. 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
H(2k/15) = 0.4 k=4
0 k = 5,6,7 (Nov 2018)
8. Determine the filter coefficients h(n) obtained by sampling
Hd(ej) = e-j(N-1)/2 0     /2
0        for N=7.
(Nov 2014/April 2018)
9. Using Hanning window tecchnique ,design a LPF which approximates an ideal filter with
cutoff frequency of 1000 Hz and sampling frequency of 8KHZ. Order of filter is 7.
10.Explain the digital FIR filter design using frequency sampling method.
(May 2014/Nov 2017)
11. i) State and explain the properties of FIR filters. State their importance.
ii) Explain linear phase FIR structures and its advantages? (May 2014)
12. Design band pass filter with cut off frequencies 0.2 rad/sec and 0.3 rad/sec with M=7.
Use the Hanning window function. (Nov 2014).
13.List the advantages of FIR filters.(Nov 2015)
14. Design a linear phase FIR filter with a cut off frequency of /2 rad/sec. Take N=17 using
frequency sampling techniques. (Nov 2016)
15. Design a filter coefficients with
Hd(ej) = e-j3/4 (May 2016)
 for N=7. Use hamming window

16. Design a FIR filter with the following desired specifications using Hanning window
with N=5
Hd(ej) = e-j2  (Nov 2017)
0 /4

17.Design a FIR HPF filter with the following desired specifications using Hanning window
with N=11
Hd(ej = 1  (April 2017)
0 /4
18.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
H(2k/15) = 1 k = 0,1,2,3
0 k = 4,5,6,7 (April 2017)
19. Write the expression for the frequency response of Rectangular window and Hamming
window and explain (April 2018)
20. Design an FIR low pass filter satisfying the following specifications
αp ≤ 0.1 dB; αs ≥ 44 dB; ωp =20 rad/sec and ω s =30 rad/sec, ωsf =100 rad/sec (Nov2018)
21. Using a rectangular window technique design a lowpass filter with pass band gain of unity,
cutoff frequency of 1000Hz and working at a sampling frequency of 5KHz. The length of the
impulse response should be 7. (Nov2018)

UNIT IV FINITE WORD LENGTH EFFECTS

PART B
1.For a system described by the equation y(n) = 0.8 y(n-1) + x(n) with the range of input is
(-1, +1) and represented by 5 bits. Compute the output noise power due to input quantization.
2. A second order system is described by y(n) = 0.35 y(n-2) + 0.92 y(n-1) + x(n)
Study the effect of shift in pole locations with 4 bit coefficient representation in direct and
cascade form realization. (May 2014)
3. The transfer function of an IIR system is given by, H(z) = 1 / (1-0.48z-1) (1-0.79z-1) Find the
output round off noise power in direct form realization.(Assume that the products are rounded
to 3 bits)
4 i) Draw the quantization noise model for second order system in direct and cascade form
ii) Study the limit cycle oscillation of the system which is defined as y(n) = 0.9y(n-1) + x(n)
with zero input and y(-1) = 12. Determine the dead band (May 2014/ Nov 2015).
5. Study the limit cycle behavior of the system described by the equation y(n) = 0.95y(n-1) +
x(n) Determine the dead band of the filter (May 2016/April 2017/Nov2018)
6. For a system described by the equation, y(n) = 0.93 y(n-1) + x(n) the range of input has a
peak value of 10V, represented by 6 bits. Compute the variance of output due to A/D
conversion process.
7. i) Write short notes on overflow and zero input limit cycle oscillation
ii) Derive an expression for quantization error of input (Nov 2014)
8. Study the effects of shift in pole location of second order IIR filter with b=3 bits

9. a)Explain Briefly about various number representation in digital computer

b) Explain the finite word length effects in digital filters (Nov 2013)
10. Consider the transfer function H(Z)=H1(Z)H2(Z) where H1(Z) =1/1- a1Z-1 ;
H2(z) =1/ 1- a2Z-1 Find the o/p Round of noise power Assume a1=0.5 and a2= 0.6 and
find output round off noise power. (May 2016/April 2017)
11. Discuss the various common methods of quantization (May 2014/Nov 2014)
12. Explain how signal scaling is used to prevent overflow LCO (May 2013)
13. Determine the dead band of the filter y(n) = 0.2 y(n-1) + 0.5 y(n-2)+x(n) (May 2013)
14. Derive the signal to quantization noise ratio of A/D converter (May 2014)
0.5z
15.The output of an ADC is applied to a digital filter with system function Find the
z  0.5
output noise power from digital filter when input signal is quantized to have 8 bits(Nov 2015)
16. Study the limit cycle behavior of the system described by the equation
 y(n) = Q[7/8 y(n-1) + x(n)] Assume x(0) =3/4 & x=0 for n˃0 Choose 4 bit sign magnitude.
Determine the dead band of the filter (Nov 2016)
17. For the digital network shown in figure find H(z) and Scale factor. So to avoid over flow
in register A1 (Nov 2016)

18. Draw the quantization noise model for a second order system
H(z)= 1/ (1-2rcosϴ z-1 +r2z-2) and find the steady state output noise variance ( May 2016)
19. Explain the quantization process and errors introduced due to quantization (Nov 2017)
20. i) Study the limit cycle oscillation of the system which is defined as
y(n) = 0.95y(n-1) + x(n) with zero input and y(-1) = 13. Determine the dead band
ii) Define zero input limit cycle oscillation and explain (Nov 2017)
21. The output signal of A/D converter is passed through a first order low pass filter, with
transfer function given by H(z) = (1-a)z / (z-a) , aFind the steady state output noise
power due to quantization at the output of digital filter (April 2018)
22. Briefly explain the following
i) Coefficient quantization error
ii) Product quantization error
iii) Truncation and Rounding (April 2018)
23. The input to the system y(n) = 0.999 y(n-1) + x(n) is applied to an ADC. What is the power
produced by the quantization noise at the output of the filter is the input is quantized to a (i) 8
bits (ii) 16 bits. (Nov2018)
24. For a system described by the equation y(n) = 0.8 y(n-1) + x(n) with the range of input is
(-100, +100) and represented by 8 bits. Compute the output noise power due to A/D conversion
process. (Nov2018)