Sunday 21 October 2018

New Interpolation Decimation Program_reg

Dear students,

New decimation  & Interpolation program without 'sinwave' function , Verify this program...... Most expected question in University practical examination.....


Interpolation:
clear all;
close all;
clc;
N=12; % sequence length
n=0:N-1;
% X=sin(2*pi*n/20)+sin(2*pi*n/15) %input sequence
X=[1,-1,1,-1, 2,-2,2,-2,3,-3,3,-3]; %input sequence
I=3; %up sampling factor
X1=zeros(1,I*N); 
j=1:I:I*N;
X1(j)=X;%up sampler output sequence
length(X1)
subplot(2,1,1)
stem(X)
subplot(2,1,2)
stem(X1)

Decimation:
clear all;

close all;
clc;

x=[1,-1,1,-1, 2,-2,2,-2,3,-3,3,-3];
n=length(x);
D=3;
y=x(1:D:n);
subplot(2,1,1)
stem(x);
subplot(2,1,2)
stem(y);
stem(y);



Regards
Rajesh.K
AP/ECE,
SSMIET

Tuesday 9 October 2018

question bank for 4,


Dear Students ,

Find the question bank for 4,5 units below......


Question Bank (UNIT-4,5)
Sub: EC6502-PDSP                                                                                                  SEM:V
PART -A
1.      What are the different types of fixed point representation with example?
2.      What is meant by fixed point arithmetic? Write an example.
3.      Represent 15.75 using fixed point and floating point arithmetic.
4.      What are the methods used to prevent overflow?
5.      Compare truncation with rounding errors.
6.      What is meant by finite word Length effects in digital system?
7.      List out the factors degrading the performance of a finite word length digital filter
8.      Compare the fixed point and floating point number representations?
9.      Why Rounding is preferred  over truncation in realizing the digital filter
10. State the need for scaling in filter implementation
11. Define Adaptive filter?
12. Write the input output relationship for a decimator
13. State the applications of adaptive filtering?
14.  Write the expression  for the output y(n) as a function of input x(n) for the given Multirate system shown below

15. What is the need for Anti imaging filter after Up sampling signal?
16. How is sampling rate converted by a factor I/D?
17. . What is echo cancellation?
18. What is Quadrature Mirror Filter (QMF)?
19. What is sub-band coding?
20. Compare efficient transversal structure with direct form structure

PART –B

1. Explain the characteristics of Limit cycle oscillation with respect to system described by difference equation: y(n)=0.95y(n-1)+x(n); x(n)=0 and y(-1)=13                                                         
2. Explain coefficient quantization in IIR filter
3. Draw the product quantization noise model of second order IIR system.
4. Find the output round-off noise power for the system having transfer function H(z) = 1/(1-0.5Z-1)( 1-0.4Z-1) which is realized in direct and in cascade forms. Assume word length is 4 bits.
5. How does sampling rate increase by an integer factor I? Derive the input-output relationship in both time and frequency domains.
6. Briefly explain the following: (a) coefficient quantization error (b) Product quantization error (c) Truncation and Rounding
7. Design Two stage decimator for the following specifications Sampling  rate of the i/p signal=20kHz, M=100, pass band=0  to 40 Hz,  Transition band=40 to 50 Hz, PB ripple=0.01, SB ripple=0.002                                                  
8. Explain Sub Band coding in detail
9. Explain in detail about applications of adaptive filter for echo cancellation and equalization                                                                                                          
10. Explain Poly phase structure of the decimator and interpolator                                                                       

Thursday 14 June 2018

IIT -- Annauniversity DSP syllabus comparison

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 :



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.

Wednesday 13 June 2018

EC6502-Principles of Digital Signal Processing Question Bank -Reg

Dear Students,

welcome you all to this Blogspot . Please take print out of the question bank for Fifth semester subject EC6502-Principles of Digital signal Processing. The link is given below. Have a nice day.
All the best.


https://drive.google.com/file/d/1XroEVX7Qy3zk4EIARGHXH5flUeG_eKhX/view?usp=sharing

Thursday 9 July 2015

about "maram seyya virumbu"


see this photos of MSV Projects :








Monday 29 June 2015

Dear Students,


After you gone through the basics of MatLab ,here you have 'Solving equations and curve fitting' in MatLab Try this