VELS UNIVERSITY

School of Engineering
Department of Electronics and Communication Engineering

Title of the Paper: Digital Signal Processing Sem: III BE Part time
Subject code:

QUESTION BANK

UNIT I : DISCRETE TIME SIGNALS AND SYSTEMS

PART A – 2 Marks

1. What are the advantages of DSP?

2. Define aliasing?
3. Find 4-point DFT of the sequence x(n) = {1,1,-1,-1}
4. How many stages are required in the case of a 64 point radix 2 DIT-FFT algorithm?
5. What are the advantages of FFT over DFT?
6. What is meant by bit reversal?
7. What is zero padding? What is the purpose of it?
8. How many multiplications and additions are required to compute N-point DFT using Radix 2
FFT?
9. State Sampling theorem?
10. What is meant by Gibbs Phenomenon?

PART B – 16 Marks

1. What is sampling. Explain the different components of an analog to digital convertor?

2. Elaborate on Discrete Fourier Transform. List and prove its properties.
3. Determine the Fourier Transform of the discrete time signal,
𝑥(𝑛) = 𝑎−𝑛 𝑢(𝑛), |𝑎| < 1
4. Find X(k) using DIT-FFT algorithm of the sequence x(n)={1,1,0,0,1,0,1,1}
5. Explain Radix 2 DIF-FFT algorithm. Compare it with DIT-FFT algorithm?
6. With appropriate diagrams describe (i) overlap-save method and (ii) overlap-add method?

UNIT II : INFINITE IMPULSE RESPONSE FILTER DESIGN

PART A – 2 Marks

1. What are IIR filters? List the different methods to design IIR filters.
2. List the steps in the design of digital filter from analog filters?
1
3. Convert the analog transfer function 𝐻(𝑠) = 𝑠+𝑎 into digital by impulse invariant method.
 4. What is warping effect?
5. Give any two properties of Chebyshev filters.
6. Why do we go for analog approximation to design digital filters?
7. What are the advantages of cascade realization?
8. What are the disadvantages of direct realization?
9. What are the conditions needed for a digital filter to be stable?
1
10. Find 𝐻(𝑍) for the IIR filter whose 𝐻(𝑠) = (𝑠+6) with T=0.1 sec.

PART B – 16 Marks

1. Explain Butterworth filter approximation in detail?

2. Determine 𝐻(𝑍) using impulse invariant technique for the analog system function,
1
𝐻(𝑆) = (𝑠+0.5)(𝑠2
+0.5𝑠+2)
3. Discuss the steps in the design of IIR filter using Bilinear Transformation technique for any one
type of filter.
2
4. Convert the given analog filter with transfer function 𝐻(𝑠) = (𝑠+1)(𝑠+2) into a digital IIR filter
using bilinear transformation.
5. Convert the following pole-zero IIR filter into a lattice ladder structure
[1 + 2𝑧 −1 + 2𝑧 −2 + 𝑧 −3 ]
𝐻(𝑧) =
13 5 1
1 + ( ) 𝑧 −1 + ( ) 𝑧 −2 + ( ) 𝑧 −3
24 8 3
6. Obtain the parallel form realization for the equation
3 3 1
𝑦(𝑛) = − 𝑦(𝑛 − 1) + 𝑦(𝑛 − 2) + 𝑦(𝑛 − 3) + 𝑥(𝑛) + 3𝑥(𝑛 − 1) + 2𝑥(𝑛 − 2)
8 22 64

UNIT III : FINITE IMPULSE RESPONSE FILTER DESIGN

PART A – 2 Marks

1. State the properties of FIR filter?

2. Differentiate symmetric and asymmetric FIR filters?
3. Write down the equation of Hamming window.
4. Why is window function used in FIR filter design?
5. What are the desirable characteristics of a window?
6. Draw a causal FIR filter structure for length m=5.
2 2
7. Realize the following causal linear phase FIR system function, 𝐻(𝑍) = 3 + 𝑧 −1 + 3 𝑧 −2
8. The length of an FIR filter is 9. If the filter has linear phase show that
𝑚−1

∑ ℎ(𝑛) sin(𝜔𝑐 − 𝜔𝑛 ) = 0
𝑛=0
9. What is the need of equiripple approximation?
 10. What is frequency sampling method?

PART B – 16 Marks

1. A low pass filter has the desired response as given below,

𝜋
𝑒 −𝑗3𝜔 , 0 ≤ 𝜔 ≤
𝐻𝑑 (𝑒 𝑗𝜔 ) = { 2
𝜋
0, ≤ 𝜔 ≤ 𝜋
2
Determine the filter coefficients 𝐻(𝑛) if 𝑀 = 7 using Type I frequency sampling techniques.
2. Derive the frequency response of a linear phase FIR filter?
3. Determine the frequency response of an IIR filter defined by
𝑦(𝑛) = 0.25𝑥(𝑛) + 𝑥(𝑛 − 1) + 0.25𝑥(𝑛 − 2). Calculate the phase delay and group delay.
4. Discuss the design procedure for FIR filter by frequency sampling method?
5. What is a linear phase filter? What are the conditions to be satisfied by the impulse response of
an FIR system in order to have linear phase.
6. Explain the principle and procedure for designing FIR filter using rectangular window.

UNIT IV : FINITE WORD LENGTH EFFECTS

PART A – 2 Marks

1. What are the two types of quantization employed in a digital system?

2. What is fixed point arithmetic? Give examples.
3. Define zero input limit cycle oscillations?
4. What is meant by input quantization error?
5. Define truncation error?
6. State the methods used to prevent overflow?
7. Explain briefly quantization noise.
8. List the types of limit cycle oscillation?
9. Define scaling?
10. What is noise transfer function?

PART B – 16 Marks

1. Explain briefly about

a. Product quantization error
b. Limit cycle oscillations
2. Derive the equations for
a. Rounding and Truncation error
b. Quantization Noise power
3. Explain the various formats of the fixed point representation of binary numbers.
4. Explain the quantization process and the errors introduced due to quantization?
 5. Explain how signal scaling is used to prevent overflow limit cycle in digital filter implementation
with examples.
6. Explain how the reduction of product round-off error is achieved in digital filters.

UNIT V : SPECTRUM ESTIMATION AND MULTIRATE SIGNAL PROCESSING

PART A – 2 Marks

1. What is the need of periodogram estimation?

2. What are the applications of multi-rate signal processing?
3. State the basic operations in multi-rate signal processing?
4. Define decimator and interpolator?
5. Differentiate Bartlett and Welch methods.
6. What are the various parametric methods of spectrum estimation?
7. What is interpolation?
8. Differentiate upsampling and downsampling?
9. What are QMF filters?
10. What is meant by sub band coding?

PART B – 16 Marks

1. Discuss the non-parametric methods of periodogram estimation in detail.

2. Explain in detail about the parametric methods of periodgram estimation.
3. Discuss the principles of multirate digital signal processing.
4. Explain in detail about decimation and interpolation.
5. Discuss QMF filters in detail.
6. Explain sub band coding of speech signals.