Course
UndergraduateSemester
ElectivesSubject Code
AV464Subject Title
Advanced DSP and Adaptive FilterSyllabus
Discrete Random Process: Expectation, Variance and Co‐variance, Uniform, Gaussian and Exponentially distributed noise, Hillbert space and inner product for discrete signals, Energy of discrete signals, Parseval’s theorem, Wiener Khintchine relation, power spectral density, Sum decomposition theorem, Spectral factorization theorem.
Spectrum Estimation: periodogram, Non – parametric methods of spectral estimation Correlation method, WELCH method – AR, MA, ARMA models. Tule – Walker method. Linear Estimation and Prediction: ML estimate –Efficiency of estimator, Cramer Rao bound ‐ LMS criterion. Wiener filter – Recursive estimator – Kalman estimator – Linear prediction, Analysis and synthesis filters, Levinson resursion, Lattice realization. Adaptive filters: FIR adaptive filter – Newton’s Steepest descent algorithm – Widrow Hoff LMS adaptation algorithms – Adaptive noise cancellation, Adaptive equalizer, Adaptive echo cancellors.
Text Books
Same as Reference
References
- M. Hays: Statistical Digital Signal Processing and Modelling, John Willey and Sons, 1996.
- Simon Haykin: Adaptive Filter Theory, Prentice Hall, 1996
- "Adaptive Filters :Theory and Applications", by B. Farhang ‐ Boroujeny, John Wiley and Sons, 1999.
- John G Proakis and Manolakis, “ Digital Signal Processing Principles, Algorithms and Applications”, Pearson, Fourth Edition, 2007.
- Sophocles J. Orfanidis, Optimum Signal Processing, An Introduction, McGraw Hill,1990
Course Outcomes (COs):
CO1: Understand the use of random process in signal analysis
CO2: Analyse and apply spectral estimation techniques
CO3: Evaluate the different estimation and prediction techniques
CO4: Apply the different adaptive filters for different signal processing applications