JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD

JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD B.Tech III DIGITAL SIGNAL PROCESSING Previous Question Paper (October - 2020 - R15)

B.Tech | III | III-I | R15 | October - 2020

Answer PaperPaper Code: 125DWMaximum Marks: 75Time: 2 hours

Answer any five questions. All questions carry equal marks.

Main

115 marks
a

Determine the impulse response and the unit step response of the systems described by the difference equation:

y(n) = 0.7y(n-1) - 0.1y(n-2) + 2x(n) - x(n-2)
8 marks

Answer

To find the impulse response h(n), we take the Z-transform of the difference equation y(n) - 0.7y(n-1) + 0.1y(n-2) = 2x(n) - x(n-2). Assuming zero initial conditions, we get Y(z)(1 - 0.7z^-1 + 0.1z^-2) = X(z)(2 - z^-2). The transfer function is H(z) = Y(z)/X(z) = (2 - z^-2) / (1 - 0.7z^-1 + 0.1z^-2). Factoring the denominator, we have (1 - 0.2z^-1)(1 - 0.5z^-1). Using partial fraction expansion: H(z) = A/(1 - 0.2z^-1) + B/(1 - 0.5z^-1). Solving for constants, we find A = -10 and B = 12. Thus, the impulse response is h(n) = -10(0.2)^n u(n) + 12(0.5)^n u(n).

The unit step response s(n) is the response to x(n) = u(n), which is the convolution of h(n) with u(n), or the inverse Z-transform of S(z) = H(z) / (1 - z^-1). Performing partial fraction expansion on S(z) = (2 - z^-2) / [(1 - 0.2z^-1)(1 - 0.5z^-1)(1 - z^-1)], we obtain coefficients for terms (1-z^-1)^-1, (1-0.2z^-1)^-1, and (1-0.5z^-1)^-1, resulting in s(n) = [2.5 - 2.5(0.2)^n + 1(0.5)^n] u(n).

b

Determine the canonic form realization of the LTI system:

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

Answer

The given difference equation is y(n) = x(n) - x(n-1) + 2x(n-2) - 3x(n-4). The transfer function is H(z) = 1 - z^-1 + 2z^-2 + 0z^-3 - 3z^-4. In the canonic form (Direct Form II), the realization uses a single delay chain to represent the system. Since the denominator coefficients are all zero except for the leading coefficient (a0 = 1), the feedback path is absent.

  1. The input x(n) enters the delay line consisting of four unit delays (z^-1).
  2. The output y(n) is formed by the weighted sum of the input and the outputs of the delay elements: y(n) = 1*x(n) - 1*x(n-1) + 2*x(n-2) + 0*x(n-3) - 3*x(n-4).
  3. The structure consists of a feedforward path where the input x(n) is passed through four delay elements in series. Multipliers with coefficients 1, -1, 2, 0, and -3 are applied to the input and the taps after each delay, respectively, and summed to produce y(n).
215 marks
a

Consider a system with impulse response:

h(n) = (1/2)^n ; 0 ≤ n ≤ 4; 0 ; elsewhere

Determine the input x(n) for 0 ≤ n ≤ 8 that will generate the output sequence x(n) = {1, 2, 2.5, 3, 3, 3, 2, 1, 0, ...}

8 marks

Answer

Sign in to view the complete answer paper.

b

The step response of an LTI system is s(n) = (1/3)^(n-2) u(n+2). Find the system function H(z) and sketch the pole-zero plot.

7 marks

Answer

Sign in to view the complete answer paper.

3

For the sequences x1(n) = cos(2πn/N) and x2(n) = sin(2πn/N) for 0 ≤ n ≤ N-1. Determine the N-point circular convolution of x1(n) and x2(n).

15 marks

Answer

Sign in to view the complete answer paper.

4

Compute eight-point DFT using DIT and DIF algorithm for the sequence:

x(n) = n ; 0 ≤ n ≤ 7; 0 ; elsewhere
15 marks

Answer

Sign in to view the complete answer paper.

5

Determine the system function H(z) of the lowest-order Chebyshev digital filter that meets the following specifications:

a) 1-dB ripple in the passband 0 < |ω| < 0.3π

b) At least 60 dB attenuation in the stopband 0.35π < |ω| < π. Use the bilinear transformation.

15 marks

Answer

Sign in to view the complete answer paper.

6

Design a Chebyshev lowpass filter with the specifications αp = 1dB ripple in the pass band 0 ≤ ω ≤ 0.2π, αs = 15dB in the stop band 0.3π ≤ ω ≤ π. Using Impulse Invariance and Bilinear transformation.

15 marks

Answer

Sign in to view the complete answer paper.

7

Explain the type I and type II frequency sampling method of designing an FIR digital filter.

15 marks

Answer

Sign in to view the complete answer paper.

815 marks
a

Consider a signal x(n) with Fourier transform X(ω) = 0 for ωm < ω ≤ π. Show that the signal x(n) can be recovered from its samples x(mD) if the sampling frequency ωs = 2π/D ≥ 2ωm (f = 1/D ≥ 2fm).

8 marks

Answer

Sign in to view the complete answer paper.

b

Explain about the round-off noise in digital filters.

7 marks

Answer

Sign in to view the complete answer paper.

About this paper

This page contains the solved question paper with answers of DIGITAL SIGNAL PROCESSING from JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY HYDERABAD for B.Tech ETM III III-I, October - 2020, R15.

For the question paper without answers, switch to the Question Paper tab.

Other papers in DIGITAL SIGNAL PROCESSING

Loading Question Papers

Fetching data from database...