VELS UNIVERSITY SCHOOL OF ENGINEERING (Common to ECE third year A,B,&C sections and third year EEE) SUB

CODE: 09CBEC52/09CBEE52 SUB NAME: DIGITAL SIGNAL PROCESSING SEM: V UNIT-I-DISCRETE FOURIER TRANSFORM Part-A
1. List any two properties of Fourier Transform. 2. Draw the basic butterfly diagram for the computation in the radix 2 decimation-in-frequency FFT algorithm. 3. Distinguish between Discrete time Fourier transform and Discrete Fourier transform. 4. What are the differences and similarities between DIF and DIT algorithms? 5. What is FFT? 6. How many multiplications and additions are required to compute N point DFT using radix 2 FFT? 7. What are the applications of FFT algorithm? 8. Give the formula for DFT and IDFT. 9. State the property for convolution theorem. 10. Compute the number of multiplications needed in the FFT computation of DFT of a 32 point sequence. 11. State Parsevals relation for DFT. 12. Describe twiddle factor with equation. .

PART B 1. i) Compute the 4 point DFT of the following sequence using DIF- FFT algorithm x(n)={1,1,1,1 } ii) State and explain any two properties of DFT. (10) (6)

2. For the given sequence x(n)={0,1,2,3} Find X(k) using DIT-FFT algorithm. (16)

3. Determine the 4 point DFT of the signal x(n)={1,1,0,0}

(16)

4. (i) Compute 4 point DFT using DIF FFT radix 2 algorithm x(n)=[0.5,0.5,0.5,0.5} (ii) What are the differences and similarities between DIT and DIF algorithms. 5. Draw the butterfly diagram and compute the DIT-FFT of the following sequence Using decimation in time algorithm x(n)={1,1,1,1,1,1,1,1} (16) (8) (8)

UNIT-II: INFINITE IMPULSE RESPONSE DIGITAL FILTERS Part-A 1. What is meant by prewarping in the design of digital filters? 2. What is known as Gibbs phenomenon? 3. What is Frequency warping? 4. State any two properties of Butterworth filter. 5. State any two properties of Chebyshev filter.

6. State the advantages and disadvantages of bilinear transformation. 7. Explain the warping effect. 8. State the relationship between analog and digital frequency in impulse invariant transformation 9. Compare the properties of Butterworth and Chebyshev filters. 10. What are the limitations of impulse invariant mapping? 11. Give the relationship between Analog and Digital frequency in Bilinear transformation. 12. Give the equation for the order of N and cutoff frequency c of the Butterworth filter.

Part-B 1. For the analog transfer function H(s)= method. Assume T=1 s. determine the H(z) using Impulse invariance (16)

2. Using bilinear transformation, design a high pass filter, monotonic in pass band with cutoff frequency of 1000 Hz and down 10dB at 350 Hz. The sampling frequency is 5000 Hz . (16) 3.(i) Show that a stable analog filter is mapped to a stable digital filter using bilinear transform. (8) (ii) An analog filter has a transfer function H(s) = 10 (s 2+7s+10) Design a digital filter equivalent to this using impulse invariant method for T=0.2 sec. 4. (i) Design a digital Butterworth filter that satisfies the following constraints 0.707 H( |H ( ) | 0.2, )| 1, for for 0 3 (8)

with T=1 s using bilinear transformation method.

(16)

5. Explain in detail the steps involved in the design of IIR filter using bilinear transformation. (16) 6. Obtain the direct form I ,cascade realization for the system y(n)= -0.1y(n-1)+0.2y(n-2)+3x(n)+3.6x(n-1)+0.6x(n-2). (16)

UNIT-III: FINITE IMPULSE RESPONSE DIGITAL FILTERS

Part-A 1. What are the advantages of FIR filters? 2. Why is FIR filter always stable? 3. State the characteristic of FIR filter designed using window functions. 4. State the Kaiser Window function. 5. Under what condition a FIR filter will exhibit linear phase response? 6. What are the properties of FIR filter? 7. State the condition satisfied by linear phase FIR filter. 8. Compare FIR and IIR filters. 9. Define Hanning Window functions. 10. Define Blackmann Window functions. 11. What are the design techniques of designing FIR filters? 12. Discuss the stability of FIR filters. Part-B 1. Design an ideal Hilbert transformer having frequency response H( ) = j; for 0

=-j; for 0 . Using (i) Rectangular window for N=11. (16)

2. (i) Find the frequency response of linear phase FIR filter with symmetric impulse response and N-odd. (8) (ii) Design a FIR filter approximating the ideal frequency response: H d( )= for . (8) -

= 0 for

Determine the filter coefficient of N=13. 3. Design the FIR filter with H d ( ) ={

{0 Using a Hanning window with N=7.

< (16) (16)

4. Obtain a direct form realization for a linear phase FIR system for N-odd and N-even. 5. Design a FIR low pass filter having following specifications: H d( ) = 1 for

= 0 for otherwise And given that N=7 using (i) (ii) (iii) Hanning window Hamming window Blackman window. (16)

6. The desired response of a low pass filter is

H d(

<

={0 Determine H(

< ) for M=7 using a Hamming window. (16)

UNIT IV Part A

FINITE WORD LENGTH EFFECTS

1. Define sampling rate conversion. 2. What is the need for signal scaling? 3. Define product quantization error? 4. What is meant by (Zero input) limit cycle oscillations? 5. What is meant by overflow oscillations? 6. Define Dead band. 7. Compare fixed point and floating point arithmetic. 8. What are the three types of quantization errors due to finite word length registers in digital filters? 9. What are the Advantages of floating point arithmetic? 10. What is the range of quantization error when truncation is performed, while representing positive numbers? 11. List the different types of quantization effects on DSP system design. 12. Distinguish between rounding and truncation. Part B

1.

i)Realize

the

first

order

transfer

function

H(z)=

and

draw (8)

its

quantization model. find the steady state noise power due to product round off

(ii) Explain in detail about the zero-input limit cycle oscillations due to finite word length of registers. (8) 2 .Discuss about product quantization, input quantization and co-efficient quantization in DSP system design. (16) 3. i) Explain in detail about binary floating point representation of numbers. ii) Derive the equation for quantization noise power. (8) (8) errors

4. Find the steady state variance of the noise in the output due to quantization of input for the first order filter. Y(n)=ay(n-1) + x(n) 5. Consider a second order IIR filter with H(z)= Find the effect on quantization on pole location of the system function in direct and in cascade form. (8+8) (16)

UNIT V-MULTIRATE SIGNAL PROCESSING PART A 1. What are the two techniques of sampling rate conversion? 2. What are the necessities of multistage approach? 3. What is oversampling? 4. Give any two applications of sub band coding. 5. Define sub band coding.

6. State the expression for interpolation. 7. 8. State the principle of interpolation by factor I. What are the applications of sampling rate conversion?

9. What are the disadvantages of D/A conversion and resampling at required rate? 10. What are the advantages of multi-rate processing? PART B 1. (i) Explain oversampling A/D and D/A conversion with the block diagram. (ii) Write the necessity of multistage approaches for sampling rate conversion. 2. Explain sampling rate conversion by a rational factor I/D. 3. Explain Multirate digital signal processing. 4. What is sub band coding? Discuss about its applications of sub band coding? (10) (6) (16) (16) (2+14)