BM17601 DIGITAL SIGNAL PROCESSING TECHNIQUES L T P C

4 0 0 4
OBJECTIVES:
1. To learn discrete Fourier transform and its properties
2. To know the characteristics of IIR and FIR filters
3. To learn the design of infinite and finite impulse response filters for filtering undesired signals
4. To understand Finite word length effects
5. To study the concept of Multirate and adaptive filters

UNIT I DISCRETE FOURIER TRANSFORM 12

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 FINITE IMPULSE RESPONSE (FIR) FILTER 12

Introduction to FIR filter - phase delay and group delay – linear phase transfer function. Design of FIR filter
using Fourier method, Rectangular window, Hanning window, Hamming window, Kaiser window. Design
using frequency sampling technique. Structure realization of FIR system – direct form, cascade form, linear
phase FIRsystem.

UNIT III INFINITE IMPULSE RESPONSE (IIR) FILTER 12

Introduction to IIR filter - Impulse-invariant transformation technique – Bilinear transformation technique –
frequency transformation in digital domain - design of Butterworth filter and Chebyshev filter (type-1)
(restricted to 3rd order). Structure realization of IIR system – lattice structure and lattice-ladder structure.

UNIT IV ANALOG TO DIGITAL CONVERSION AND QUANTISATION ERROR 12

Sampling of analog signals- Sampling theorem- ADC - Quantization of continuous amplitude signals,
Sinusoidal signal- Fixed point and floating point number representations –Quantization- Truncation and
Rounding errors - Quantization noise – coefficient quantization error – Product quantization error - Overflow
error – Round off noise power - limit cycle oscillations due to product round off and overflow errors –
Principle of scaling

UNIT V BASICS OF RANDOM SIGNAL PROCESSING 12

(ONLY QUALITATIVE ANALYSIS)
Introduction to probability function, joint probability, conditional probability – estimation parameters – joint
distribution function, probability density function, ensemble average – mean squared value, variance, standard
deviation, moments, correlation, covariance, orthogonality, auto-covariance, auto-correlation, cross-
covariance and cross-correlation – stationarity – ergodic – white noise –Decimator (down sampling) –
frequency-domain analysis of decimator – interpolation (up sampling) – frequency-domain analysis of
interpolator
TOTAL: 60 PERIODS
OUTCOMES:
At the end of the course, the student should be able to:
1. Apply DFT for the analysis of digital signals & systems
2. Design and implement IIR filters.
3. Design and implement FIR Filters.
4. Characterize finite Word length effect on filters
5. Analyse random signals and apply it for real time processing

TEXT BOOKS:
1. John G. Proakis and Dimitris G. Manolakis, “Digital Signal Processing, Algorithms and
Applications”, Pearson education, New Delhi, 4th Edition, 2007.
2. E. C. Ifeachor and B.W. Jervis, “Digital Signal processing – A Practical Approach”,Pearson
education, New Delhi, 4th Edition, 2004.

REFERENCES:
1. Sanjit K. Mitra, “Digital Signal Processing – A computer Based Approach”,TMH, New Delhi, 1998
2. Andreas Antoniou, Digital filter Analysis and Design”, Prentice Hall India
3. R. Rabiner and B. Gold, “Theory and Application of Digital Signal processing”, PHI.