VINAYAKA MISSIONS UNIVERSITY

AARUPADAI VEEDU INSTITUTE OF TECHNOLOGY, CHENNAI

&
V.M.K.V.ENGINEERING COLLEGE, SALEM

QUESTION BANK

DEGREE / BRANCH : B.E.-MECHATRONICS

SUBJECT : DIGITAL SIGNAL PROCESSING
SUBJECT CODE :
YEAR & SEMESTER : III / V
REGULATION : CBCS R 2015

UNIT–I
DISCRETE FOURIER TRANSFORMS & FAST FOURIER TRANSFORMS:
PART-A

1. What is Digital Signal Processing?

2. Mention the applications of FFT algorithm.
3. What is Radix-2 FFT?
4. Define Correlation.
5. State Time shifting property of DFT.
6. Perform 4-Point DFT for the sequence x(n)= {0,1,2,3}.
7. Distinguish between DIT and DIF algorithm.
8. How many stages are there for 8 point DFT?
9. List the advantages of FFT algorithm.
10. Draw the basic flow graph of DIT radix-2 FFT.
11. Give any two applications of DFT.
12. List out any four properties of DFT.
13. Arrange the 8-point sequence x(n)={1,2,3,4,-1,-2,-3,-4} in bit reversed order.
14. What is the phase factors involved in the first stage of computation in the 8 point DIF
radix-2 FFT?
15. Why FFT is needed?
16. How many multiplications and additions are involved in radix-2 FFT.
17. What is circular convolution?
18. Differentiate between DTFT and DFT.
19. What is phase factor or twiddle factor?
20. Why circular convolution is important in DSP?

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 PART-B
1. Compute the 4-Point DFT of causal three sample sequence given by

2. i) Find the circular convolution of the two sequences

and
using concentric circle method.
ii) Find the circular convolution of the two sequences
and
using matrices method.
3. Find the circular convolution of the two sequences x1(n) = (2, 1, 2, 1) and
x2 (n) =(1,2,3,4) using DFT and IDFT method .
4. An 8 point sequence is given by x(n)=(2,1,1,2,1,1,1,1) compute 8 point DFT of x(n) by
radix-2 DIT-FFT
5. An 8 point sequence is given by x(n)=(2,1,1,2,1,1,1,1) compute 8 point DFT of x(n) by
radix-2 DIF-FFT
6. An 8 point sequence is given by x(n)=(2,2,0,2,1,1,0,1) compute 8 point DFT of x(n) by
radix-4 DIT-FFT.
7. 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 .
8. Find the linear and circular convolution of the sequences x(n) & h(n),x(n)={1,0.5} &
h(n)={0.5,1}.
9. An 8 point sequence is given by x(n)=(1,2,3,4,4,3,2,1) compute 8 point DFT of x(n) by
radix-2 DIF-FFT.
10. In an LTI system the input x(n)={1,1,1} and the impulse response
h(n)={-1,-1}.determine the response of LTI system by radix-2 DIT FFT.

UNIT-II
IIR FILTER DESIGN
PART-A

1. Mention the advantages of Digital filters.

2. What is the use of impulse invariance method?
3. List out any two advantages of IIR filters.
4. Mention some realization methods available to realize IIR filter.
5. What do you mean by bilinear transformation?
6. Write down the relation between analog and digital filter using impulse invariance
method?
7. Give any two properties of Butterworth filter and chebyshev filter.
8. Distinguish between Chebyshev filter and Butterworth filter?
9. Mention two transformations to digitize an analog filter.
10. Give the bilinear transform equation between S plane and Z plane.
11. What is Warping effect?
11. List the disadvantages of digital filters.
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 12. What is the main objective of impulse invariant method?
13. Compare the digital and analog filter.
14. What is impulse invariant transformation?
15. Write the impulse invariant transform used to transform complex conjugate poles.
16. How bilinear transformation is performed?
17. Write the transfer function of normalized Butterworth low pass filter.
18. Classify the filters based on frequency response.
19. What is chebyshev approximation?
20. Sketch the magnitude response of type-1 chebyshev filters.

PART-B

1. Convert the analog filter with transfer function H(s) into digital filter using bilinear
transformation.
2s s3
H(s) = ( s  1)( s  s  1)
2
i) H(s) = s  0.2s  1 ii)
2

2. i) Convert the analog filter with system transfer function

(S+0.1)

H(s) = -------------

(S+0.1)2+9

into a digital IIR filter by means of the impulse invariance technique.

ii) Write the design procedure for lowpass digital Butterworth IIR filter.

3. Convert the analog filter with system transfer function

2 0.0325
ii) H(s) = ( s  0.25)  (0.0325)
2 2
i) H(s) = ( s  1)( s  2)

into a digital filter by means of the impulse invariance technique.

If i) T= 1 sec and ii) T= 0.1 sec.

4. i)Obtain the Direct form–I realizations of the LTI system governed by the equation

y(n)= 0.5y(n-1)-0.25 y(n-2)+x(n)+3x(n-1)

ii) Determine the direct form II realizations for the following system

y(n)= - 0.1y(n-1)+0.72y(n-2)+0.7x(n)-0.252x(n-2)

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 5. Design a Butterworth IIR filter with the following specifications

0.8 ≤ |H (ω)| ≤ 1.0 ; 0 ≤ ω ≤ 0.2

|H (ω)| ≤ 0.2 ; 0.32 ≤ ω ≤ 

Using impulse invariant method.

6. Discuss and draw various IIR realization structures of Direct form-II forms for the
difference equation given by,

y(n)= -3/8 Y(n-1) + 3/32 y(n-2) + 1/64 y(n-3) + x(n) + 3 x(n-1) + 2x(n-2).

7. Convert the analog filter with system transfer function

1 5
i) H(s) = ( s  0.1)  16
2
ii) H(s) = s  0.1

into a digital filter by means of the approximation of derivatives technique.

If i) T= 1 sec and ii) T= 0.2 sec.

8. i) Write the design procedure for low pass digital Chebyshev IIR filter.

ii) Contrast the Butterworth & Chebyshev type-1 filters.

9. Design a Chebyshev IIR filter with the following specifications

0.9 ≤ |H (ω)| ≤ 1.0 ; 0 ≤ ω ≤ 0.25

|H (ω)| ≤ 0.24 ; 0.5 ≤ ω ≤ 

Using impulse invariant method.

2
10. i. Apply bilinear transformation to H(s) = ( s  1)( s  2) with T=1sec and find H (Z).

ii. A digital filter with a 3dB bandwidth of 0.25π is to be designed from the analog filter
whose system response is

c
H(S) = S  c

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 UNIT-III
FIR FILTER DESIGN
PART-A
1. Obtain the digital transfer function for the given analog transfer function

by using impulse invariant technique with T=1 sec.

2. State the characteristics of linear phase FIR filters.
3. What are the types of windowing technique?
4. Write the frequency response of linear phase LTI system with constant phase delay and
constant group delay.
5. What are the conditions to be satisfied for constant phase delay in linear phase FIR
filters?
6. List the features of FIR filter design using rectangular window.
7. What are the advantages and disadvantages of FIR filter?
8. Contrast about FIR and IIR filters.
9. Write down the magnitude and phase response of Anti symmetric FIR filter.
10. Mention the necessary and sufficient condition for linear phase characteristics in FIR
filter.
11. Mention the properties of FIR filter?
12. What is Gibbs phenomenon?
13. Write the characteristic equation of Hamming window.
14. How phase distortion and delay distortion are introduced?
15. Write the steps involved in FIR filter design.
16. Compare rectangular window and Hanning window.
17. What is the principle of designing FIR filter using frequency sampling method?
18. List out the disadvantages of Fourier series method.
19. What is window? Why is it necessary?
20. List the features of Hanning window spectrum.

PART-B
1. Design a FIR low pass filter with cutoff frequency 1 kHZ and sampling rate of 4 kHZ
with 11 samples using fourier series method.
2. Design a low pass filter using rectangular window by taking 9 samples of w (n) and with
a cutoff frequency of 1.2 radians/sec.
3. Determine the coefficients of a linear phase FIR filter of length M=15 has a symmetric
unit sample response and a frequency response that satisfies the condition.

H (2k/15) = {1 ; for k=0, 1, 2, 3

0 ; for k=4, 5, 6, 7}
4. A low pass filter is to be designed with the following desired frequency response

,-/4    /4
j  j 2
H( e )={ e
0 , /4     }
determine the filter coefficients h d (n) if the window function is defined as
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 w(n)= {1, 0  n  4
0, otherwise

5. Design a band stop filter to reject frequencies in the 1 to 2 rad/sec using rectangular
window, with N=7.
6. Determine the coefficients of a linear phase FIR filter of length N=15 which has a
symmetric unit sample response that satisfies the conditions
H (2k/15) = {1 ; for k=0, 1, 2, 3
0.4 ; for k=4
0 ; for k=5, 6, 7}
7. Design a linear phase FIR Highpass filter using hamming window, with cutoff frequency
wc = 0.8π rad/sample and N= 7.
8. A filter is to be designed with the following desired frequency response
H( e )= {0, -/4    /4
j

, /4     }
 j 2
e
determine the filter coefficients h d (n) if the window function is defined as
w(n)= {1, 0  n  4
0, otherwise
9. Design a linear phase FIR bandpass filter to pass frequencies in the range 0.4π to 0.65π
rad/sample by taking 7 samples of hanning window sequence.
10. Design a FIR low pass filter with cutoff frequency 2 kHZ and sampling rate of 5 kHZ
with 9 samples using fourier series method.

UNIT-IV
FINITE WORD LENGTH EFFECTS
PART-A
1. Differentiate Rounding and Truncation.
2. List out the different quantization methods.
3. What are the two kinds of limit cycle behavior in DSP?
4. Discriminate fixed and floating point numbers.
5. What is meant by limit cycle oscillations?
6. How can we avoid over flow error?
7. Draw the quantization noise model for a first order system.
8. What is product quantization error?
9. Give the expression for signal to quantization noise ratio and calculate the improvement
with an increase of 2 bits to the existing bit.
10. What do you understand by A/D conversion noise?
11. What is meant by finite word length effects in digital filters?
12. List some of the finite word length effects in digital filters.
13. Give the advantages of floating point arithmetic.
14. Sketch the noise probability density functions for rounding.
15. How the digital filter is affected by quantization of filter coefficients?
16. What is zero input limit cycle?
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 17. What is dead band?
18. Mention why rounding is preferred for quantizing the product.?
19. What is meant by quantization step size?
20. Why rounding is preferred to truncation in realizing digital filter?

PART-B
1. Discuss the operation of analytical model of sample and hold circuit with necessary
diagrams.
2. Describe about fixed point and binary floating point number representation with an
example.
3. a) i) Give a brief note on various number representations used in DSP.
ii) Explain briefly about the analytical model of sample and hold operations.
4. Briefly explain finite word length effect in digital filters>

5. For the digital network shown in figure find H(z) and scale factor S0 to avoid overflow in
register A1.

x(n) So A1 w(n) 0.245 y(n)

0.245
0.509

w(n-1)

6. For the recursive filter shown in figure the input x(n) has a peak value of 10V,
represented by 6 bits. Compute the variance of output due to A/D conversion process?

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 x(n) y1(n)

Z-1
0.93

7. Write detailed notes on

i) Channel vocoder ii) Homomorphic vocoder

8. Explain the characteristics of a limit cycle oscillation with respect to the system described
by the difference equation y(n)=0.95y(n-1)+x(n).Determine the dead band of the filter.

9. The output of an A/D converter is applied to a digital filter with the system function
0.5 z 0.6 z
H (Z )  H (Z ) 
a) z  0.5 b) z  0.6
Find the output noise power from the digital filter, when the input signal is quantized to
have eight bits.

10. Give short notes on the following:

(i)Overflow error (ii) Truncation error (iii) Coefficient Quantization Error.

UNIT-V
PART-A

1. What are the special features of digital signal processors?

2. List out the types of digital signal processors.
3. How is fast data access achieved in digital signal processors?
4. Compare Von Neumann and Harvard architectures.
5. What are the advantages of special addressing modes in DSPs?
6. How fast computation is achieved in DSP?
7. List out the special addressing modes of TMS320C5x Processing.
8. What are the on-chip peripherals of TMS320C5x processor?
9. What is pipelining in DSPs?
10. Write brief note on MAC unit.
11. Mention the internal buses of TMS320C5x processor.
12. What is meant by guard bits?
13. State the components of data address generation unit of TMS320C5x processor.
14. What is memory-mapped register addressing?
15. Give examples of accumulator addressing.
16. What the operations performed by auxiliary register arithmetic unit?
17. What accumulator addressing?
18. Mention the addressing modes of TMS320C5x processor.
19. How DMA is useful in the DSP processor?
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 20. What is the advantage of using multiprocessor DSP?

PART-B
1. Explain Von Neumann and Harvard architecture with simple sketch.
2. Discuss about the fast data access and fast computation requirements of digital signal
processors.
3. Explain in detail the pipelining of instruction execution.
4. Elucidate simplified architecture of TMS320C5x processor.
5. Write short notes on data and program address generation units of TMS320C5x
processor.
6. Explain any four addressing modes of TMS320C5x processors with examples.
7. Discuss the numerical fidelity and fast execution control requirements of digital signal
processors.
8. Explain the various on-chip peripherals of TMS320C5x processor.
9. Write the salient features of TMS320C5x family of digital signal processors.
10. Write detailed notes on various functional units of CPU of TMS320C5x processors.

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