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IC304 Discrete Time Signal Processing

This document outlines a course on discrete time signal processing. The course aims to teach design techniques for digital filters, the effects of finite word length in signal processing, and non-parametric power spectrum estimation methods. The syllabus covers topics like sampling theory, the discrete Fourier transform, digital filter design, and finite word length effects. Students will learn about time and frequency domain sampling, computing the DFT, digital filter structures, and designing digital filters. The course is divided into six modules over one semester, with internal exams and an end semester exam assessing students' understanding.

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AKSHAY KRISHNA K R
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279 views2 pages

IC304 Discrete Time Signal Processing

This document outlines a course on discrete time signal processing. The course aims to teach design techniques for digital filters, the effects of finite word length in signal processing, and non-parametric power spectrum estimation methods. The syllabus covers topics like sampling theory, the discrete Fourier transform, digital filter design, and finite word length effects. Students will learn about time and frequency domain sampling, computing the DFT, digital filter structures, and designing digital filters. The course is divided into six modules over one semester, with internal exams and an end semester exam assessing students' understanding.

Uploaded by

AKSHAY KRISHNA K R
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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Course code Course Name L-T-P- Year of

Credits Introduction
IC304 DISCRETE TIME SIGNAL PROCESSING 3-0-0-3 2016
Prerequisite: Nil
Course Objectives
 To learn the design techniques for digital filters
 To learn the effects of finite word length in signal processing
 To study the non-parametric methods of power spectrum estimations
Syllabus
Sampling theorem and aliasing – Up sampling and down sampling – Frequency domain
sampling – Discrete Fourier Transform – Circular convolution – Computation of DFT – Digital
filters – Finite word length effects – IIR and FIR filter design
Expected Outcome
After the completion of the course, students should be able to
 Explain time domain and frequency domain sampling
 Compute DFT for a given discrete signal
 Explain finite word length effects in digital filters
 Design digital filters
Text Books:
1. Oppenhiem and Schafer, Discrete Time Signal Processing, PHI
2. Sanjit K Mitra, Digital Signal Processing, McGraw Hill Education (India) Private
Limited.

References
1. Proakis and Manolakis, Digital Signal Processing Principles, Algorithms and
Applications. Pearson.
2. Li Tan, Digital Signal Processing Fundamentals and Applications, Academic Press.
Course Plan
Module Contents Hours Sem.
Exam
Marks
I Sampling Theorem - Definition and proof of sampling 6 15%
theorem for band-limited signals.
Effect of Under sampling: Aliasing
Sampling of Discrete Time Signals – Impulse train sampling –
Discrete time Decimation and
Interpolation (Up sampling and down sampling)
II Frequency domain sampling and reconstruction of Discrete 6 15%
time signals -DFT- properties of
DFT
Circular convolution - Dentition, Linear convolution using
DFT – Linear Filtering method
based on DFT
FIRST INTERNAL EXAM
III Frequency analysis of signals using DFT 7 15%
Computation of DFT - Direct computation of DFT – DFT as
linear Transformation – Fast
Fourier Transform (Radix – 2 FFT only) – Decimation in time
and decimation in Frequency
algorithms
IV Digital Filter Structures 8 15%
Z-Transform representation of FIR and IIR filters, FIR filter
structures : - Direct form,
Cascade form and Lattice structures IIR filter structure: Direct form
Transpose form,
Cascade form and Parallel form structures
Finite word length effects
Quantization noise, Round off effects, Limit cycle oscillations
SECOND INTERNAL EXAM
V Digital Filter Specifications 8 20%
Design of IIR digital filters from analog filters – Butterworth
and Chebyshev filters – Low pass, High pass, Band pass and
band stop Filters – scaling the
digital transfer function – impulse invariant method, bilinear
transformation method – spectral transformation of IIR filters.
VI Linear phase characteristics – basic concepts of group delay- 7 20%
Design of FIR filters:
Estimation of filter order - FIR filter design - windowed
Fourier series method
Impulse response of ideal filters - Gibbs Phenomenon – Fixed
window functions –
Rectangular, Hamming, Blackman – Adjustable window
function – Kaiser window.
END SEMESTER EXAM

QUESTION PAPER PATTERN:

Maximum Marks: 100 Exam Duration: 3 Hours

Part A
Answer any two out of three questions uniformly covering Modules 1 and 2. Each question
carries 15 marks and can have not more than four sub divisions. (15 x 2 = 30 marks)
Part B
Answer any two out of three questions uniformly covering Modules 3 and 4. Each question
carries 15 marks and can have not more than four sub divisions. (15 x 2 = 30 marks)
Part C
Answer any two out of three questions uniformly covering Modules 5 and 6. Each question
carries 20 marks and can have not more than four sub divisions. (20 x 2 = 40 marks)

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