SIGNALS AND DSP (BEE502)

SIGNALS AND DSP

Course Code BEE502 
CIE Marks 50
Teaching Hours/Week (L:T:P: S) 3:0:2:0 
SEE Marks 50
Total Hours of Pedagogy 40 hours Theory + 12 Lab slots
Total Marks 100
Credits 04 
Exam Hours 03

MODULE-1

Signals, systems and signal processing, classification of signals, Basic Operations on Signals, Basic

Elementary Signals, properties of systems. concept of frequency in continuous and Discrete time

signals, sampling of analog signals, the sampling theorem , quantization of continuous amplitude

and sinusoidal signals , coding of quantized samples, digital to analog conversion,

Time-domain representations for LTI systems: Convolution, impulse response representation,

Convolution Sum and Convolution Integral, properties of impulse response representation, solution

of difference equations.

MODULE-2

Discrete Fourier Transforms (DFT):

Introduction to DFT, definition of DFT and its inverse, matrix relation to find DFT and IDFT

,Properties of DFT, linearity, circular time shift, circular frequency shift, circular folding, symmetry

of : real valued sequences, real even and odd sequences, DFT of complex conjugate sequence,

multiplication of two DFTs- the circular convolution, Parseval’s theorem, circular correlation,

Digital linear filtering using DFT. Signal segmentation , overlap-save and overlap-add method.

MODULE-3

Fast-Fourier-Transform (FFT) algorithms: Direct computation of DFT, need for efficient computation of the

DFT (FFT algorithms)., speed improvement factor, Radix-2 FFT algorithm for the computation of DFT and

IDFT–decimation-in-time and Decimation-in-frequency algorithms , calculation of DFT when N is not a power of 2.

MODULE-4

IIR filter design: Classification of analog filters, generation of Butterworth polynomials, frequency

transformations. Design of Butterworth filters, low pass, high pass, band pass and band stop filters,

Generation of Chebyshev polynomials, design of Chebyshev filters, design of Butterworth and

Chebyshev filters using bilinear transformation and Impulse invariance method, representation of IIR

filters using direct form one and two, series form and parallel form.

MODULE 5

FIR filter design:

Introduction to FIR filters, symmetriv and antisymmetric FIR filters, design of linear phase FIR

filters using - Rectangular, Bartlett, Hamming, Hanning and Blackman windows, design of FIR

differentiators and Hilbert transformers, FIR filter design using frequency sampling Technique.

Representation of FIR filters using direct form and lattice structure.

Experiments

1 Verification of Sampling Theorem in time and frequency domains

2 Generation of different signals in both continuous and discrete time domains

3 To perform basic operations on given sequences- Signal folding, evaluation of even and odd

signals

4 Evaluation of impulse response of a system.

5. Solution of a difference equation.

6. Evaluation of linear convolution and circular convolution of given sequences

7 Computation of N- point DFT and IDFT of a given sequence by use of (a) Defining equation; (b)

FFT method

8 Evaluation of circular convolution of two sequences using DFT and IDFT approach.

9 Design and implementation of IIR filters to meet given specification (Low pass, high pass, band

pass and band reject filters).

10 Design and implementation of FIR filters to meet given specification (Low pass, high pass, band

pass and band reject filters) using different window functions.

11 Design and implementation of FIR filters to meet given specification (Low pass, high pass, band

pass and band reject filters) using frequency sampling technique.

12 Realization of IIR and FIR filters.

13 Following experiments to be done using DSP Kit:

a)Obtain the linear convolution of two sequences

b)Compare circular convolution of two sequences

c)To find N –point DFT of given sequence

d)To find impulse response of first and second order system

e)Generation of sine wave and standard test signals

Text Books/Reference Books:

1.Introduction to Digital Signal Processing, Jhonny R. Jhonson, Pearson 1 st Edition, 2016.

2.Digital Signal Processing – Principles, Algorithms, and Applications,Jhon G. Proakis Dimitris G.

Manolakis, Pearson, 4 th Edition, 2007.

3. Digital Signal Processing, A.NagoorKani, McGraw Hill, 2nd Edition, 2012.

4. Digital Signal Processing, Shaila D. Apte,Wiley, 2nd Edition, 2009.

5. Digital Signal Processing, Ashok Amberdar, Cengage, 1st Edition, 2007.

6. Digital Signal Processing, Tarun Kumar Rawat, Oxford, 1st Edition, 2015.