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It1252 Digital Signal Processing

This document outlines the units and topics covered in the IT1252 Digital Signal Processing course. The five units cover: 1) signals and systems, sampling theory, and discrete time signals; 2) fast Fourier transforms and efficient FFT algorithms; 3) IIR filter design using impulse invariance and bilinear transformation; 4) FIR filter design using windowing techniques; and 5) finite word length effects, quantization noise, and applications of digital signal processing including speech processing. The course relies on three textbooks and references for digital signal processing principles, algorithms, and applications.

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Subramani M
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485 views1 page

It1252 Digital Signal Processing

This document outlines the units and topics covered in the IT1252 Digital Signal Processing course. The five units cover: 1) signals and systems, sampling theory, and discrete time signals; 2) fast Fourier transforms and efficient FFT algorithms; 3) IIR filter design using impulse invariance and bilinear transformation; 4) FIR filter design using windowing techniques; and 5) finite word length effects, quantization noise, and applications of digital signal processing including speech processing. The course relies on three textbooks and references for digital signal processing principles, algorithms, and applications.

Uploaded by

Subramani M
Copyright
© Attribution Non-Commercial (BY-NC)
We take content rights seriously. If you suspect this is your content, claim it here.
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IT1252 DIGITAL SIGNAL PROCESSING UNIT I SIGNALS AND SYSTEMS Basic elements of digital signal Processing Concept of frequency

in continuous time and discrete time signals Sampling theorem Discrete time signals. Discrete time systems Analysis of Linear time invariant systems Z transform Convolution and correlation. UNIT II FAST FOURIER TRANSFORMS Introduction to DFT Efficient computation of DFT Properties of DFT FFT algorithms Radix-2 and Radix-4 FFT algorithms Decimation in Time Decimation in Frequency algorithms Use of FFT algorithms in Linear Filtering and correlation. UNIT III IIR FILTER DESIGN Structure of IIR System Design of Discrete time IIR filter from continuous time filter IIR filter design by Impulse Invariance. Bilinear transformation Approximation derivatives Design of IIR filter in the Frequency domain. UNIT IV FIR FILTER DESIGN Symmetric & Antisymteric FIR filters Linear phase filter Windowing technique Rectangular, Kaiser windows Frequency sampling techniques Structure for FIR systems. UNIT V FINITE WORD LENGTH EFFECTS Quantization noise derivation for quantization noise power Fixed point and binary floating point number representation comparison over flow error truncation error co-efficient quantization error - limit cycle oscillation signal scaling analytical model of sample and hold operations Application of DSP Model of Speech Wave Form Vocoder. TEXT BOOK 1. John G Proakis and Dimtris G Manolakis, Digital Signal Processing Principles, Algorithms and Application, PHI/Pearson Education, 2000, 3rd Edition. REFERENCES 1. Alan V Oppenheim, Ronald W Schafer and John R Buck, Discrete Time Signal Processing, PHI/Pearson Education, 2000, 2nd Edition. 2. Johny R.Johnson, Introduction to Digital Signal Processing, Prentice Hall of India/Pearson Education, 2002. 3. Sanjit K.Mitra, Digital Signal Processing: A Computer Based Approach, Tata McGraw-Hill, 2001, Second Edition.

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