{{{credits}}}
| L | T | P | C |
| 3 | 0 | 0 | 3 |
- To review discrete-time signals and systems
- To introduce DFT and its properties for discrete-time signal analysis
- To teach the design of IIR and FIR filters for filtering the undesired signals
- To introduce the DSP for engineering applications.
{{{unit}}}
| UNIT I | REVIEW OF SIGNALS, SYSTEMS AND SIGNAL PROCESSING | 9 |
Basic elements of DSP system, Discrete-Time (DT) signals – Classification of DT signals, Concepts of frequency in Analog and Digital Signals, Need for sampling and sampling theorem, Discrete – Time systems – Types of DT system, Analysis of discrete time LTI systems, Linear convolution, Z transform and its applications for LTI system analysis.
{{{unit}}}
| UNIT II | FREQUENCY ANALYSIS OF DISCRETE-TIME SIGNALS | 9 |
Introduction to Discrete Fourier Transform (DFT) – Properties of DFT – Circular convolution, Linear filtering based on DFT, Filtering of long data sequences using Overlap – save method and Overlap – add method, FFT Algorithms – Decimation-in-time (DIT) FFT Algorithm, Decimation-in-frequency (DIF) FFT Algorithm.
{{{unit}}}
| UNIT III | DESIGN OF IIR FILTER | 9 |
Design of analog IIR filter – Butterworth and Chebyshev IIR filters, Design of discrete time IIR filter from analog filter – IIR filter design by Impulse Invariance technique (IIT) and Bilinear transformation technique (BLT), Structures of IIR filter – Direct form-I, Direct form-II, Cascade and Parallel form, Finite word length effects of IIR filter – limit cycle oscillations, dead band, signal scaling.
{{{unit}}}
| UNIT IV | DESIGN OF FIR FILTER | 9 |
Linear phase FIR filter, FIR Filter design using Rectangular, Hamming and Hanning windowing techniques, Structures of FIR filter – Transversal, linear-phase and poly-phase structures, Finite word length effects of FIR filter.
{{{unit}}}
| UNIT V | APPLICATIONS OF DIGITAL SIGNAL PROCESSING | 9 |
Multi-rate signal processing - Decimation, Interpolation, Multi-stage design of Decimation and Interpolation filters, Sub-band coding of speech signals, Adaptive filtering – Noise cancellation, Speech enhancement.
\hfill Total Periods: 45
After the completion of this course, students will be able to:
- Understand signals, systems and basics of digital signal processing (K2)
- Understand and apply DFT and FFT for the analysis of DT signals & systems (K3)
- Design and realize IIR filters using impulse invariant and bilinear transformation techniques (K3)
- Design and realize FIR filters using various window techniques (K3)
- Understand the need of DSP for engineering applications (K2).
- John G Proakis, Dimitris G Manolakis, “Digital Signal Processing – Principles, Algorithms and Applications”, 4th Edition, Pearson Education, Prentice Hall, 2007.
- Sanjit K Mitra, “Digital Signal Processing – A Computer Based Approach”, 3rd Edition, Tata McGraw Hill, 2007.
- S Salivahanan, “Digital Signal Processing”, Third Edition, McGraw Hill Education (India) Private Limited, 2015.
- V Oppenheim, R W Schafer, J R Buck, “Discrete-Time Signal Processing”, 8th Indian Reprint, Pearson 2004.
- Emmanuel C Ifeachor, Barrie W Jervis, “Digital Signal Processing”, 2nd Edition, Pearson Education, Prentice Hall 2002.
- Andreas Antoniou, “Digital Signal Processing”, Tata McGraw Hill,
- P Ramesh Babu, “Digital Signal Processing”, 4th Edition, Scitech Publications (India) Pvt Ltd, 2011.