Dear students,
Please check out the difference in syllabus ......
IIT Syllabus
Title : Digital Signal Processing
Course No : EE2004
Credits : 4
Prerequisite :
Syllabus :
- Review of
Signals and Systems: Discrete time complex exponentials and other basic
signals—scaling of the independent axis and differences from its
continuous-time counterpart—system properties (linearity, time-invariance,
memory, causality, BIBO stability)—LTI systems described by linear
constant coefficient difference equations (LCCDE)—autocorrelation.
- Discrete-Time
Fourier Transform (DTFT): Complex exponentials as eigensignals of LTI
systems—DTFT definition—inversion formula—properties—relationship to
continuous-time Fourier series (CTFS).
- Z-Transform:
Generalized complex exponentials as eigensignals of LTI
systems—z-transform definition—region of convergence (RoC)—properties of
RoC—properties of the z-transform—inverse z-transform methods (partial
fraction expansion, power series method, contour integral
approach)—pole-zero plots—time-domain responses of simple pole-zero
plots—RoC implications of causality and stability.
- Frequency
Domain Analysis of LTI Systems: Frequency response of systems with
rational transfer function—definitions of magnitude and phase
response—geometric method of frequency response evaluation from pole-zero
plot—frequency response of single complex zero/pole—frequency response of
simple configurations (second order resonator, notch filter, averaging
filter, comb filter, allpass systems)—phase response—definition of
principal phase—zero-phase response—group delay—phase response of single
complex zero/pole—extension to higher order systems—effect of a unit
circle zero on the phase response—zero-phase response representation of
systems with rational transfer function—minimum phase and allpass
systems—constant group delay and its consequences—generalized linear
phase—conditions that have to be met for a filter to have generalized
linear phase—four types of linear phase FIR filters—on the zero locations
of a linear phase FIR filter—constrained zeros at z = 1 and at z = -1 and
their implications on choice of filters Type I through Type IV when
designing filters—frequency response expressions for Type I through Type
IV filters.
- Sampling:
Impulse train sampling—relationship between impulse trained sampled
continuous-time signal spectrum and the DTFT of its discrete-time
counterpart—scaling of the frequency axis—relationship between true
frequency and digital frequency—reconstruction through sinc
interpolation—aliasing—effect of sampling at a discontinuous
point—relationship between analog and digital sinc—effects of
oversampling—discrete-time processing of continuous-time
signals—non-integer delay—up-sampling and down-sampling—introduction to
sample-rate alteration.
Discrete Fourier Transform (DFT): Definition of the DFT and inverse
DFT—relationship to discrete-time Fourier series—matrix representation—DFT
as the samples of the DTFT and the implied periodicity of the time-domain
signal—recovering the DTFT from the DFT—circular shift of signal and the
“index mod N” concept—properties of the DFT—circular convolution and its
relationship with linear convolution—sectioned convolution methods:
overlap add and overlap save—effect of zero padding—introduction to the
estimation of frequencies of sinusoids—windowing and spectral
leakage—introduction to the Fast Fourier Transform (FFT)
algorithm—decimation-in-time and decimation-in-frequency algorithms.
Text Books :
- Discrete-Time Signal Processing by Alan V. Oppenheim and Ronald W. Schafer,
3rd edition, 2010, Prentice Hall, Upper Saddle River, NJ.
- Digital Signal Processing by John G. Proakis and Dimitris K.
Manolakis, 4th edition, 2007, Prentice Hall, Upper Saddle River, NJ.
Digital Signal Processing by Sanjit Mitra, 4th edition, 2011,
McGraw-Hill, New York, NY.
Anna university Syllabus
EC6502 PRINCIPLES OF DIGITAL SIGNAL PROCESSING SYLLABUS REGULATION 2013
UNIT I DISCRETE FOURIER TRANSFORM
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 IIR FILTER DESIGN
Structures of IIR – Analog filter design – Discrete time IIR filter from analog filter – IIR filter design by Impulse Invariance, Bilinear transformation, Approximation of derivatives – (LPF, HPF, BPF, BRF) filter design using frequency translation.
UNIT III FIR FILTER DESIGN
Structures of FIR – Linear phase FIR filter – Fourier Series - Filter design using windowing techniques (Rectangular Window, Hamming Window, Hanning Window), Frequency sampling techniques – Finite word length effects in digital Filters: Errors, Limit Cycle, Noise Power Spectrum.
UNIT IV FINITE WORDLENGTH EFFECTS
Fixed point and floating point number representations – ADC –Quantization- Truncation and Rounding errors - Quantization noise – coefficient quantization error – Product quantization error - Overflow error – Roundoff noise power - limit cycle oscillations due to product round off and overflow errors – Principle of scaling.
UNIT V DSP APPLICATIONS
Multirate signal processing: Decimation, Interpolation, Sampling rate conversion by a rational factor – Adaptive Filters: Introduction, Applications of adaptive filtering to equalization.