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ADVANCED SIGNAL PROCESSING

L T P C
3 0 2 4

Aims and Objectives: This course examines the fundamentals of detection and
estimation for signal processing. It will help the students to implement new
algorithms for signal processing applications in the frequency and time domains.

Pre-requisites: Basic knowledge in Signals and Systems, Digital Signal Processing.

Course Outcome: Students will learn to design a filter by properly estimating the
spectrum for the signal when it is deterministic or random in nature.

REVIEW OF DISCRETE TIME SYSTEM: (5 Hours)


Sampling and Reconstruction, Discrete time signals and systems, Convolution and
correlation, Z- transform and its properties, Importance of ROC and its significance in
signals and system analysis, Recursive – Non-recursive filters.

FREQUENCY DOMAIN ANALYSIS: (6 Hours)


Discrete Time Fourier Transform, Discrete and Fast Fourier Transform

DESIGN OF DIGITAL FILTERS: (14 Hours)


Frequency transformations, Design of digital IIR filter – Impulse Invariance and
Bilinear Transformation.
Design of FIR filter using window functions, Importance of Linear Phase
characterization.

SPECTRUM ESTIMATION (6 Hours)


Periodogram Estimator. MA, AR and ARMA models. Parameter estimation using
Yule-Walker method

LINEAR ESTIMATION AND PREDICTION (10 Hours)


Maximum likelihood criterion, Efficiency of estimator, review of least mean squared
error criterion – Wiener filter – Discrete Wiener Hopf equations.

DIGITAL SIGNAL PROCESSORS (4 Hours)


General-purpose digital signal processors - Fixed point and floating point DSP. Tools
and aids for firmware development, typical implementation of DSP algorithms.

Text Books:

1. Monson H. Hayes, “Statistical Digital Signal Processing and Modeling”,


John Wiley & Sons, Inc, Newyork, 2002
2. John G. Proakis & Dimitris G. Manolakis, “Digital Signal Processing”,
Prentice Hall of India, New Delhi, 1995.
3. Emmanuel C.Ifeachor, “ Digital Signal Processing A Practical Approach” 2nd
edition, Pearson Education, 2001.

Proceedings of the 21st Academic Council of VIT [30.11.2010] 529


References:

1. Dimitris G. Manolakis, Vinay K. Ingle & Stephen M. Kogon, “Statistical and


Adaptive Signal Processing”, Artech House, Inc., 2005
2. S.K.Mitra, “Digital Signal Processing”, 3rd edition, TMH, 2006
3. Andreas Antoniou, “Digital Signal Processing”, TMH, 2006
4. Dr. D. Ganesh Rao, “Digital Signal Processing”, Technical Publishers, 2005

SIGNAL PROCESSING LABORATORY


Laboratory Outcome: Students will learn to design the Filter in MATLAB and
implement in TEXAS Instruments DSK6713 kit for the signal when it is deterministic
or random in nature.
List of Experiments:

Processor Details- Architecture Information


1. Processor Basic functions - ALU, MAC, shifter (ASM)( 8 bit and 16 bit)
2. Sum of Products – Using ASM & C-coding-clock cycle verification
3. Linear and circular convolution -Using C coding
4. Correlation studies (Auto and Cross correlation) - Using C coding
5. FFT Implementation using DIT (using C Coding)
6. Design of FIR filter to smoothen the sharp transition of ECG signal
7. Design of Three Band Equalizer with IIR & FIR filters in real time using
TEXAS Instruments DSK6713
8. BASS & TREBLE CONTROL OF MUSIC USING FIR FILTER (USING SIMULINK &
CCS along with TEXAS Instruments DSK6713 in Real Time)
9. Periodogram based Power Spectrum Estimation
10. Weiner Filter Design through Weiner-Hopf Equation.

Proceedings of the 21st Academic Council of VIT [30.11.2010] 530

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