EC6502 Principles of Digital Signal Processing

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EC6502 PRINCIPLES OF DIGITAL SIGNAL PROCESSING                   L T P C    3 1 0 4 

OBJECTIVES: 
 To learn discrete Fourier transform and its properties 
 To know the characteristics of IIR and FIR filters learn the design of infinite and finite impulse response filters for filtering undesired signals 
 To understand Finite word length effects 
 To study the concept of Multirate and adaptive filters

 UNIT I DISCRETE FOURIER TRANSFORM                                                      9 
Discrete Signals and Systems- A Review – Introduction to DFT – Properties of DFT – Circular Convolution - Filtering methods based on DFT – FFT Algorithms –Decimation in time Algorithms, Decimation in frequency Algorithms – Use of FFT in Linear Filtering. 

UNIT II IIR FILTER DESIGN                                                                                9 
Structures of IIR – Analog filter design – Discrete time IIR filter from analog filter – IIR filter design by Impulse Invariance, Bilinear transformation, Approximation of derivatives – (LPF, HPF, BPF, BRF) filter design using frequency translation. 

UNIT III FIR FILTER DESIGN                                                                             9 
Structures of FIR – Linear phase FIR filter – Fourier Series - Filter design using windowing techniques (Rectangular Window, Hamming Window, Hanning Window), Frequency sampling techniques – Finite word length effects in digital Filters: Errors, Limit Cycle, Noise Power Spectrum.

UNIT IV FINITE WORDLENGTH EFFECTS                                                     9 
Fixed point and floating point number representations – ADC –Quantization- Truncation and Rounding errors - Quantization noise – coefficient quantization error – Product quantization error - Overflow error – Roundoff noise power - limit cycle oscillations due to product round off and overflow errors – Principle of scaling 

UNIT V DSP APPLICATIONS                                                                             9 
Multirate signal processing: Decimation, Interpolation, Sampling rate conversion by a rational factor – Adaptive Filters: Introduction, Applications of adaptive filtering to equalization. 

 TOTAL (L:45+T:15):                                                                                        60 PERIODS 

OUTCOMES: 
Upon completion of the course, students will be able to 
 apply DFT for the analysis of digital signals & systems 
 design IIR and FIR filters 
 characterize finite Word length effect on filters 
 design the Multirate Filters 
 apply Adaptive Filters to equalization 

TEXT BOOK: 
1. John G. Proakis & Dimitris G.Manolakis, “Digital Signal Processing – Principles, Algorithms & Applications”, Fourth Edition, Pearson Education / Prentice Hall, 2007. 

REFERENCES: 
1. Emmanuel C..Ifeachor, & Barrie.W.Jervis, “Digital Signal Processing”, Second Edition, Pearson Education / Prentice Hall, 2002. 
2. Sanjit K. Mitra, “Digital Signal Processing – A Computer Based Approach”, Tata Mc Graw Hill, 2007. 
3. A.V.Oppenheim, R.W. Schafer and J.R. Buck, “Discrete-Time Signal Processing”, 8th Indian Reprint, Pearson, 2004. 4. Andreas Antoniou, “Digital Signal Processing”, Tata Mc Graw Hill, 2006. 



ANNAUNIVERSITY QUESTIONS 

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IMPORTANT QUESTIONS  

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NOTES FOR 5 UNITS  

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IMPORTANT QUESTIONS	ANNA UNIVERSITY QUESTIONS	Subject Related PDF	Subject Related PPT
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