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Digital Signal Processing (ECE-2203) (Makeup)

This document is an examination paper for the Fourth Semester B.Tech (E & C) Degree in Digital Signal Processing, held in June 2019. It includes various questions related to signal processing concepts such as impulse response determination, DFT, FFT algorithms, filter design, and PSD estimation. The exam consists of multiple sections requiring theoretical explanations and practical calculations.

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ADVIKA SINGH
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13 views1 page

Digital Signal Processing (ECE-2203) (Makeup)

This document is an examination paper for the Fourth Semester B.Tech (E & C) Degree in Digital Signal Processing, held in June 2019. It includes various questions related to signal processing concepts such as impulse response determination, DFT, FFT algorithms, filter design, and PSD estimation. The exam consists of multiple sections requiring theoretical explanations and practical calculations.

Uploaded by

ADVIKA SINGH
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as PDF, TXT or read online on Scribd
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Reg. No.

FOURTH SEMESTER B.TECH. (E & C) DEGREE END SEMESTER EXAMINATION


JUNE 2019
SUBJECT: DIGITAL SIGNAL PROCESSING (ECE - 2203)

TIME: 3 HOURS MAX. MARKS: 50


Instructions to candidates
 Answer ALL questions.
 Missing data may be suitably assumed.

1A. An input x(n)= [1, -3.5, 1.5] is given to a stable system. The output y(n)=[3, -4]. Determine
the impulse response of the system using Z transform.
1B. i) Given two signals a(n) and b(n) with corresponding 4-point DFTs A(k)={6, 0, 2, 0} and
B(k)={6,0,-4,0}. If c(n) is the circularly convoluted output of a(n) and b(n), determine c(n).
ii) Explain overlap and save method for filtering long data sequences.
(5+5)
2A. Deduce DIF FFT algorithm and illustrate it with signal flow diagram.
2B. A length 9 linear phase digital FIR filter is required with pass band extending from 300Hz to
600Hz at a sampling frequency of 4.8kHz. The filter should have an attenuation of at least 50
dB in the stop band. Determine the filter coefficients using suitable window function. Also
write down the system function H(z).
(5+5)
3A. Explain Type-1 Chebyshev low pass IIR filter design to determine the order and poles of the
analog filter.
3B. Obtain the transfer function of first order analog prototype low pass filter. Convert this into
LPF and HPF with 3 dB frequency Ωc = 2 radians/sec. Also obtain the system function of the
above LPF using impulse invariant transformation technique at a sampling frequency of 1Hz.
(5+5)
4A. Obtain the system function for frequency sampling realization of FIR filter with impulse
response h(n). Sketch the realization structures for both even and odd length filters.
4B. Determine the lattice and ladder parameters for the following IIR filter. Sketch the lattice-
ladder structure.
1 − 0.8𝑧 −1 + 0.15𝑧 −2
(
𝐻 𝑧 =)
1 + 0.12𝑧 −1 − 0.72𝑧 −2
(5+5)
5A. Describe the general cascade realization for IIR filters. Sketch the direct form-II realization of
standard second order section.
5B. Highlight the estimation of PSD using autocorrelation function. Briefly explain one such
method of PSD estimation.
5C. Bring out the advantages of parametric method of PSD estimation. Mention different methods
of signal modelling used. (3+4+3)

ECE –2203 Page 1 of 1

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