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Syllabus For ADSP

This document outlines an advanced digital signal processing course that covers topics like linear algebra applications in DSP, statistical properties of signals, stochastic processes, parametric signal modeling, adaptive filtering, and spectrum estimation. The course aims to teach students how to apply these advanced DSP concepts and design filters like Wiener, Kalman, and adaptive filters. It includes 30 hours of lectures organized in 8 modules, 30 hours of experiments, and projects applying DSP techniques to areas like speech processing, image processing, and biomedical signal processing.

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400 views

Syllabus For ADSP

This document outlines an advanced digital signal processing course that covers topics like linear algebra applications in DSP, statistical properties of signals, stochastic processes, parametric signal modeling, adaptive filtering, and spectrum estimation. The course aims to teach students how to apply these advanced DSP concepts and design filters like Wiener, Kalman, and adaptive filters. It includes 30 hours of lectures organized in 8 modules, 30 hours of experiments, and projects applying DSP techniques to areas like speech processing, image processing, and biomedical signal processing.

Uploaded by

anittadevadas
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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ECE XXXX Advanced Digital Signal Processing L T P J C

2 0 2 4 4
Prerequisite: -

Objectives:
 To build advanced concepts in digital signal processing applicable for processing and
analysing random process
 To expose the students to adaptive algorithms.

Expected Outcome:
Students will be able to
 Apply linear algebra for DSP computation
 Determine the statistical properties of signals
 Model the signal and stochastic process using parametric method
 To design effective lattice system for random signal processing.
 Implement effective power spectrum analyzer.
 To design advanced filters like Weiner, Kalman and adaptive filters.

Student Learning Outcomes (SLO): 1, 2, 4, 5, 6,7, 9, 12

Module 1 Introduction 5 Hours SLO: 1,2,7

Discrete-Time Signal Processing: Discrete-Time Signals, Discrete-Time Systems, Time-Domain


Descriptions of LSI Filters, The Discrete-Time Fourier Transform, and The z-Transform, Special
Classes of Filters, Filter Flow graphs, The DFT and FFT. Linear and Circular convolution.

Module 2 Discrete Time Random Processes 5 Hours SLO: 1, 4, 5


Random Variables: Definitions, Ensemble Averages, Jointly Distributed Random Variables,
Joint Moments, Independent, Uncorrelated and Orthogonal Random Variables, Linear Mean
Square Estimation, Gaussian Random Variables Parameter Estimation: Bias and Consistency.
Random Processes: Review, The auto-covariance and autocorrelation Matrices Ergodicity, White
Noise, Power Spectrum.
Filtering Random Processes, Spectral Factorization.
Special Types of Random Processes: Autoregressive Moving Average Processes,
Autoregressive Processes, Moving Average Processes, Harmonic Processes.

Module 3 Signal Modelling 4 Hours SLO: 1, 4, 5


’
Introduction, The Least Squares (Direct) Method, The Pad e Approximation, Prony's Method-
Pole-Zero Modeling, Shanks' Method
Stochastic Models: Autoregressive Moving Average Models, Autoregressive Models, Moving
Average Models

Module 4 The Levinson –Durbin Recursion 3 Hours SLO: 4, 5, 12


The Levinson-Durbin Recursion: Development of the Recursion, The Lattice Filter, Properties
Module 5 Optimal filters 4 Hours SLO: 4, 5, 12
The FIR Wiener Filter: Filtering, Linear Prediction, Noise Cancellation, Lattice Representation
for the FIR Wiener Filter.
The IIR Wiener Filter: Non-causal IIR Wiener Filter, The Causal IIR Wiener Filter, Causal
Wiener Filtering, Causal Linear Prediction, Wiener Deconvolution.

Module 6 Introduction Adaptive Filters 3 Hours SLO: 5, 6, 7


Discrete Kalman Filter, steepest descent algorithm, LMS, RLS

Module 7 Spectrum Estimation 4 Hours SLO:7, 9


Non Parametric Methods Periodogram, The Modified Periodogram, Bartlett's Method, Welch's
Method, Blackman-Tukey Approach: Periodogram Smoothing, Performance Comparisons.
Parametric Methods
Autoregressive Spectrum Estimation, Moving Average Spectrum Estimation, Autoregressive
Moving Average Spectrum Estimation.

Module 8 Contemporary Issues 2 Hours SLO: 2


Recent trends in advanced digital signal processing

Total Lectures: 30 Hours


Text Books:
1. Monson H. Hayes, Statistical digital signal processing and modeling, John Wiley & Sons,
2009

Reference Books:
1. Manolakis, Dimitris G., Vinay K. Ingle, and Stephen M. Kogon. Statistical and adaptive
signal processing: spectral estimation, signal modeling, adaptive filtering, and array
processing. Vol. 46. Norwood: Artech House, 2005
2. Mitra, Sanjit Kumar, and Yonghong Kuo. Digital signal processing: a computer-based
approach. Vol. 2. New York: McGraw-Hill, 2006
3. Moon, Todd K., and Wynn C. Stirling. Mathematical methods and algorithms for signal
processing. Vol. 1. New York: Prentice hall, 2000
4. http://freevideolectures.com/Course/3042/Advanced-Digital-Signal-Processing

Typical List of Experiments: 30 Hours SLO: 5, 12


I. Real time experiments using TMS6713 Processor
1. Interfacing a function generator with TMS 6713 Processor through codec with sampling rate
of 96 KHz and display of the signal as a graph in CC-Studio in a time window of 256 samples.

2. Interfacing a function generator with TMS 6713 Processor through codec with sampling rate
of 96 KHz and display of the magnitude spectrum of signal as a graph in CC-Studio for a time
window of 256 samples by applying FFT for the samples.

3. FIR-filtering (low/high/bandpass) of an audio input obtained through microphone interface


and output the result in the loud speaker.
4. IIR-filtering (low/high/bandpass) of an audio input obtained through microphone interface
and output the result in the loud speaker.

II. Simulation Experiments using Matlab


1. Decimation and Interpolation of Band limited speech signal and frequency domain analysis.
2. Generation Various Random Processes MA, AR, ARMA.
3. Implementation of FIR and IIR Wiener Filter for separating the desired signal corrupted by
AWGN and MSE calculation.
4. Implementation of digital Kalman filter.
5. ECHO Cancellation.
6. Power spectrum estimation parametric method.
7. Power spectrum estimation non parametric method.
8. Implementation of Adaptive filter using LMS recursive algorithm.

Typical Projects: SLO: 6, 12


1) Linear Prediction coding for speech
2) Active noise cancellation.
3) Kalman filter for target detection/tracking
4) Signal Denoising
5) Weather forecasting algorithms.
6) Image fusion
7) Speech segregation
8) Image compression
9) Water marking for medical signals and images
10) Speech assessment software.
11) Active noise control using adaptive filters
12) Removal of power line interference in bio-signals
13) Adaptive I/Q mismatch compensation.
14) Image restoration using Wiener Filter
15) Compressive sensing
16) Estimation of fetal ECG using optimal filter for heart rate monitoring
17) Blind acoustic source separation using Non linear Adaptive Techniques
18) Adaptive channel equalization
19) Empirical mode decomposition based de-noising of speech/bio-signals
20) Removal of noise from speech/bio-signals using LMS approach

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