Wednesday, October 28, 2009
DIGITAL SIGNAL PROCESSING |
PART-A (10 x 2 = 20 Marks) |
1. Differentiate between analog and digital signal. Why Digital signal processing is widely |
used than analog signal processing. |
2. State Shannon’s Sampling theorem.3. Determine the Z-transform of (1/2)n[ u[n]-u[n- |
8]] and indicate its ROC. |
4. Compare FIR and IIR filter. |
5. What are the advantages of linear phase characteristics? Which systems exhibit linear |
phase? |
6. Show that the system described by the difference equation is an all pass system3 y(n) – |
y(n-1) = -x(n) + 3x(n-1) |
7. Mention few application areas where speech coding is required. |
8. Explain the circular addressing mode of DSP processor |
9. Find the DFT of the signal x(n)= {1,3,5,7}. |
10. Distinguish between recursive and non-recursive realizations of filters. |
PART-B (5x16 = 80 Marks) |
11.i) Show that Z-Transform of x*(n) is X*(z*). (4) |
ii) Consider a linear shift-invariant discrete system with input x(n) and output y(n) for |
whichy(n-2) – 2.5 y(n-1) +y(n) =x(n)By considering the pole-zero pattern associated with |
the difference equation, determine the three possible choices for the unit-sample |
response of the system. Comment on the stability of the system in each case. (12) |
12.a)i) Find the DFT of the sequence {1,1,1,1,2,2,2,2} using radix-2 Decimation-in-Time |
FFT. Sketch the magnitude and phase plot. (12) |
ii) What is the need for FFT? (4) |
(OR) |
12.b)i) Find the DFT of the sequence {1,1,1,1,2,2,2,2} using radix-2 Decimation-in- |
Frequency FFT. (12) |
ii) Write about over lap save method. (4) |
13.a) The specification of the desired low pass filter are:Amin = 22 dB and Amax = 3 dB ?p |
= 0.2? and ?s = 0.4?Design a Butterworth digital filter using Bilinear Transformation. (Amin |
and Amax are attenuation) (16) |
(OR) |
13.b) Design and also realize a high pass FIR filter with a cutoff frequency of 1.3 rad/sec |
and N=9. (16) |
14.a)i) Perform the linear convolution of (1/4)n u(n) and (1/2)n u(n). (6) |
ii) Is it possible to perform linear convolution through circular convolution. If so |
how?(2)iii) Find the Discrete Fourier Series of the following periodic sequence. (8)(OR) |
14.b)i) Explain about the Frequency Transformation that will be adopted in IIR filter |
design. (4) |
ii) The specification of the desired low pass digital filter areAmin = 12.4 dB and Amax = |
0.915 dB ?p = 0.25? and ?s = -0.5?Design a Chebyshev digital filter using impulse invariant |
transformation. (Amin and Amax are attenuation). (12) |
15.ai) Highlight the special blocks of the Digital Signal Processor Architecture over the |
regular Micro-Controller based Architectures. (16) |
(OR) |
15.b)i) Explain how bit-reversal is achieved in the Texas based DSP Processor. (8) |
ii) Show that FFT can be evaluated with lesser machine cycles using DSP processor |
compared to any of Micro-controller. (8) |
http://www.ziddu.com/download/7201561/dsp3.pdf.html