# Anna University Model Question Paper BE V sem IT DIGITAL SIGNAL PROCESSING

ANNA UNIVERSITY :: CHENNAI – 600 025

**MODEL QUESTION PAPER**

**V SEMESTER**

**B.TECH. INFORMATION TECHNOLOGY**

# IF351 – DIGITAL SIGNAL PROCESSING

Time : 3 Hours Max. Marks : 100

Answer all Questions

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**PART – A (10 X 2 = 20 MARKS) **

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- State and prove the convolution property of Z transform.
- Check the system is linear or not y(n) = x(n)+ay(n-1)
- Write equations for finding DFT and IDFT using Z transform.
- Draw the radix 2 butterfly structure for DIF
- Draw the implementation for the generalized for IIR filter using direct form II.
- Explain how the addition and multiplication of (H1, H2) impulse responses implemented in filter design
- Write equations for Hanning and Blackman window.
- Why frequency prewarping procedure is adopted in the design of IIR filter?
- Write two advantages of musical sound processing and briefly explain.
- Explain the effects due to upsampling.

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**PART – B (5 x 16 = 80 Marks)**

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11.i) The impulse response of a linear TI system is h(n) = {1, 0, 1, -1}. Find the response of the system to the input signal x(n) = {1, 0, 2, 1}.

ii) Check whether the system y(n) = x(n) – x(n-1) is LTI and stable.

12.a) Develop and draw the 8 point radix-2 DIT FFT algorithm for DFT computation.

**(OR)**

12.b) Compute the DFT of the following sequence

x(n) = 0 0£ n £ 2

= 1 3£ n £ 6

= 0 n=7

Plot magnitude and phase spectra

13.a) Design a LPF with following specifications. Use Hamming window and at least 8 points.

/2 |

– /2 |

0 |

1 |

H(e |

**(OR)**

13.b)i) Obtain H(z) from H(s) when T = 1 sec.

ii) Design a digital BPF using w1 & w2 as cutoff frequencies

14.a)i) Perform the following using Floating Point arithmetic.

1.5 x 1.75 and 1.5 x 1.75

ii) Realize the following H(z) given by

using cascade and Parallel form with Direct form-I.

**(OR)**

14.b)i) What is meant by quantization error? Explain briefly.

ii) Realize the following filter using cascade technique with DF-I and DF-II.

15.a) Briefly explain

- Interpolator
- Decimator
- Effects due to sampling rate conversion

**(OR)**

15.b)i) Write a note on Musical sound processing

ii) Explain how the data compression is achieved in speech signal and discuss a technique to check the quality.

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