Solve any one question from Q1 and Q2

1 (a)
Explain Advantages of Digital Signal Processing over Analog Signal
Processing.

5 M

1 (b)
Explain the concept of orthogonality.

2 M

1 (c)
Check whether the functions given are orthogonal or not over an time interval [0, 1] f(t)=1; x(t)=√3 (1-2t).

3 M

2 (a)
Compute 4-point DFT of sequence x(n)={1231} using DIT-FFT radix x-2 algorithm. What is Computational Complexity of Radix x-2 FFT algorithm?

6 M

2 (b)
Compute circular convolution of x

_{1}(n)={1,2,3,4} & x_{2}(n)={2,1,2,1}.
4 M

Solve any one question from Q3 and Q4

3 (a)
Plot the magnitude and phase spectrum of the sampled data sequence = {2, 0, 0, 1} which was obtained using a sampling frequency of 20 Hz.

5 M

3 (b)
Explain linear filtering effect for long duration sequences.

5 M

4 (a)
State any four properties of Z-Transform.

4 M

4 (b)
Compute Z-Transform and ROC of the following sequence [ x(n) = \left [ \dfrac {-1}{3} \right ]^n u[n] - \left [ \dfrac {1}{3} \right ]^n u[-n -1] ]

6 M

Solve any one question from Q5 and Q6

5 (a)
Design a Butterworth filter using impulse invariant method transformation
to satisfy the following specifications. [ \begin {align*} &0.707 \le |H(e^{jw})le 1 & for 0\le w \le 0.2 \pi \ &|H(e^{jw})le 0.2 & &for 0.6 \pi \le w \le \pi \end{align*} ]

8 M

5 (b)
Explain impulse invariant method for S-plane to Z-plane mapping. Explain its limitations.

8 M

6 (a)
What is frequency warping effect? How the mapping is done in bilinear
transformation method?

7 M

6 (b)
Draw the direct form-I and II structures for the following systems.

i) 3y(n)-2y(n-1)+y(n-2)=4x(n)-3x(n-1)+2x(n-2)

ii) y(n)=0.5 [x(n)+x(n-1)].

i) 3y(n)-2y(n-1)+y(n-2)=4x(n)-3x(n-1)+2x(n-2)

ii) y(n)=0.5 [x(n)+x(n-1)].

9 M

Solve any one question from Q7 and Q8

7 (a)
Explain the characteristics of the FIR filters.

8 M

7 (b)
Determine the impulse response h(n) of a filter having desired frequency
response. [ H_d (e^{jw\omega}) = \left{egin{matrix}
e^{-j(N-1)omega} & for \ 0\le | \omega \le \frac {pi}{2} \ 0
& for \ \frac {pi}{2} \le |omega|le \pi
\end{matrix} ight. ] N = 7, use windowing technique approach. Use hamming window.

8 M

8 (a)
Explain Gibbs phenomenon. Compare between windows available.

8 M

8 (b)
Determine the impulse response h (n) of a filter having desired frequency response, [ H_d (e^{jw}) = \left{egin{matrix}
e^{-j (N-1)w} & for \ 0 \le |w| \le \frac {pi}{2}\0
& for \ \frac {pi}{2} \le |w| \le \pi
\end{matrix} ight. ] N = 7, Use frequency sampling approach.

8 M

Solve any one question from Q9 and Q10

9 (a)
What is sampling rate conversion? What is multirate DSP? Why it is
Required?

6 M

9 (b)
A signal x (n), at a sampling frequency of 2.048 KHz is to be decimated
by a factor of 32 to yield a signal at sampling frequency 64Hz. The signal
band of interest extends from 0-30 Hz. The anti-aliasing filter should
satisfy the following specifications:

Pass Band deviation : 0.01dB

Stop Band deviation : 80dB

Pass Band : 0 - 30Hz

Stop Band : 32 - 64Hz

The signal components in the range from 30 to 32 Hz should be protected from aliasing. Design a suitable one-stage decimator.

Pass Band deviation : 0.01dB

Stop Band deviation : 80dB

Pass Band : 0 - 30Hz

Stop Band : 32 - 64Hz

The signal components in the range from 30 to 32 Hz should be protected from aliasing. Design a suitable one-stage decimator.

8 M

9 (c)
How the DSP processors are selected? (any four points)

4 M

10 (a)
State four important features of DSP processors.

4 M

10 (b)
Draw the architecture of typical DSP processor TMS320C67XX and
explain it in short.

6 M

10 (c)
5 Design a two stage decimator that down samples an audio signal by a
factor of 30, satisfying the following constraints:

Input sampling frequency : 240 Khz

Highest frequency of interest : 3.4 Khz

Passband ripple : 0.05

Stopband attenuation : 0.01

Input sampling frequency : 240 Khz

Highest frequency of interest : 3.4 Khz

Passband ripple : 0.05

Stopband attenuation : 0.01

8 M

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