Basic Simulation Lab Manual

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DEPT.OF ECE www.jntuworld.com BASIC SIMULATION LAB MANUAL
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CONTENTS
S.No Experiment Name Page No.
LIST OF EXPERIMENTS:
1. Basic operations on matrices. 2. Generation on various signals and Sequences (periodic and aperiodic), such as unit impulse, unit step, square, sawtooth, triangular, sinusoidal, ramp, sinc. 3. Operations on signals and sequences such as addition, multiplication, scaling, shifting, folding, computation of energy and average power. 4. Finding the even and odd parts of signal/sequence and real and imaginary part of signal. 5. Convolution between signals and sequences. 6. Auto correlation and cross correlation between signals and sequences. 7. Verification of linearity and time invariance properties of a given continuous /discrete system. 8. Computation of unit sample, unit step and sinusoidal response of the given LTI system and verifying its physical Realizability and stability properties. 9. Gibbs phenomenon. 10. Finding the Fourier transform of a given signal and plotting its magnitude and phase spectrum. 11. Waveform synthesis using Laplace Transform. 12. Locating the zeros and poles and plotting the pole zero maps in s-plane and z-plane for the given transfer function. 13. Generation of Gaussian Noise(real and complex), computation of its mean, M.S. Value and its skew, kurtosis, and PSD, probability distribution function. 14. Sampling theorem verification. 15. Removal of noise by auto correlation/cross correlation. 16. Extraction of periodic signal masked by noise using correlation. 17. Verification of Weiner-Khinchine relations. 18. Checking a random process for stationarity in wide sense. 1 6 18 22 26 29 33 40 43 48 53 54 57 66 70 73
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EXP.NO: 1
BASIC OPERATIONS ON MATRICES

Aim: To generate matrix and perform basic operation on matrices Using MATLAB Software. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software MATLAB on Matrices MATLAB treats all variables as matrices. For our purposes a matrix can be thought of as an array, in fact, that is how it is stored. Vectors are special forms of matrices and contain only one row OR one column. Scalars are matrices with only one row AND one column.A matrix with only one row AND one column is a scalar. A scalar can be reated in MATLAB as follows: a_value=23 a_value = A matrix with only one row is called a row vector. A row vector can be created in MATLAB as follows : rowvec = [12 , 14 , 63] rowvec = 12 14 63 A matrix with only one column is called a column vector. A column vector can be created in MATLAB as follows: colvec = [13 ; 45 ; -2] colvec = 13 45 -2 A matrix can be created in MATLAB as follows: matrix = [1 , 2 , 3 ; 4 , 5 ,6 ; 7 , 8 , 9] matrix = 123 456 789 Extracting a Sub-Matrix A portion of a matrix can be extracted and stored in a smaller matrix by specifying the names of both matrices and the rows and columns to extract. The syntax is: sub_matrix = matrix ( r1 : r2 , c1 : c2 ) ; Where r1 and r2 specify the beginning and ending rows and c1 and c2 specify the beginning and ending columns to be extracted to make the new matrix.
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A column vector can beextracted from a matrix. As an example we create a matrix below: matrix=[1,2,3;4,5,6;7,8,9] matrix = 123 456 789 Here we extract column 2 of the matrix and make a column vector: col_two=matrix( : , 2) col_two = 258 A row vector can be extracted from a matrix. As an example we create a matrix below: matrix=[1,2,3;4,5,6;7,8,9] matrix = 123 456 789 Here we extract row 2 of the matrix and make a row vector. Note that the 2:2 specifies the second row and the 1:3 specifies which columns of the row. rowvec=matrix(2 : 2 , 1 :3) rowvec =4 5 6 a=3; b=[1, 2, 3;4, 5, 6] b= 123 456 c= b+a % Add a to each element of b c= 456 789 Scalar - Matrix Subtraction a=3; b=[1, 2, 3;4, 5, 6] b= 123 456 c = b - a %Subtract a from each element of b c= -2 -1 0 123 Scalar - Matrix Multiplication a=3; b=[1, 2, 3; 4, 5, 6] b= 123 456
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c = a * b % Multiply each element of b by a c= 369 12 15 18 Scalar - Matrix Division a=3; b=[1, 2, 3; 4, 5, 6] b= 123 456 c = b / a % Divide each element of b by a c= 0.3333 0.6667 1.0000 1.3333 1.6667 2.0000 a = [1 2 3 4 6 4 3 4 5] a= 1 2 3 4 6 4 3 4 5
b=a+2 b= 3 4 5 6 8 6 5 6 7
A = [1 2 0; 2 5 -1; 4 10 -1] A= 1 2 2 0
5 -1
4 10 -1 B = A' B= 1 2 2 4
5 10
0 -1 -1 C=A*B C=
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5 12 24 12 30 59 24 59 117 Instead of doing a matrix multiply, we can multiply the corresponding elements of two matrices or vectors using the .* operator. C = A .* B C= 1 4 0
4 25 -10 0 -10 1
Let's find the inverse of a matrix X = inv(A) X= 5 2 -2 1 1
-2 -1 0 -2
and then illustrate the fact that a matrix times its inverse is the identity matrix. I = inv(A) * A I= 1 0 0 0 1 0 0 0 1
to obtain eigenvalues eig(A) ans = 3.7321
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0.2679 1.0000 as the singular value decomposition. svd(A) ans =
12.3171 0.5149 0.1577
CONCLUSION: Inthis experiment basic operations on matrices Using MATLAB

have been demonstrated.
3.perform following operations on any two matrices A+B A-B A*B A.*B A/B A./B A\B A.\B A^B,A.^B,A',A.
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EXP.NO: 2
GENERATION ON VARIOUS SIGNALS AND SEQUENCES (PERIODIC AND APERIODIC), SUCH AS UNIT IMPULSE, UNIT STEP, SQUARE, SAWTOOTH, TRIANGULAR, SINUSOIDAL, RAMP, SINC.
Aim: To generate different types of signals Using MATLAB Software. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software THEORY : UNIT IMPULSE: a) Continous signal: And
(t ) =
0 t 0 t = 0
(t )dt = 1
Also called unit impulse function. The value of delta function can also be defined in the sense of generalized function (t): Test Function b) Unit Sample sequence: (n)={ 1, i.e
0, n=0 n0
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Matlab program: %unit impulse generation clc close all n1=-3; n2=4; n0=0; n=[n1:n2]; x=[(n-n0)==0] stem(n,x)
2)Unit Step Function u(t):
u (t ) (t )dt = (t )dt
0
1 u (t ) = 0
t > 0 t < 0
b)Unit Step Sequence u(n): )=
1, 0,
n0 n< 0
% unit step generation n1=-4; n2=5; n0=0; [y,n]=stepseq(n0,n1,n2); stem(n,y); xlabel('n')
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ylabel('amplitude'); title('unit step');
Square waves: Like sine waves, square waves are described in terms of period, frequency and amplitude:
Peak amplitude, Vp , and peak-to-peak amplitude, Vpp , are measured as you might expect. However, the rms amplitude, Vrms , is greater than that of a sine wave. Remember that the rms amplitude is the DC voltage which will deliver the same power as the signal. If a square wave supply is connected across a lamp, the current flows first one way and then the other. The current switches direction but its magnitude remains the same. In other words, the square wave delivers its maximum power throughout the cycle so that Vrms is equal to Vp . (If this is confusing, don't worry, the rms amplitude of a square wave is not something you need to think about very often.) Although a square wave may change very rapidly from its minimum to maximum voltage, this change cannot be instaneous. The rise time of the signal is defined as the time taken for the voltage to change from 10% to 90% of its maximum value. Rise times are usually very short, with durations measured in nanoseconds (1 ns = 10-9 s), or microseconds (1 s = 10-6 s), as indicated in the graph
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% square wave wave generator fs = 1000; t = 0:1/fs:1.5; x1 = sawtooth(2*pi*50*t); x2 = square(2*pi*50*t); subplot(2,2,1),plot(t,x1), axis([0 0.2 -1.2 1.2]) xlabel('Time (sec)');ylabel('Amplitude'); title('Sawtooth Periodic Wave') subplot(2,2,2),plot(t,x2), axis([0 0.2 -1.2 1.2]) xlabel('Time (sec)');ylabel('Amplitude'); title('Square Periodic Wave'); subplot(2,2,3),stem(t,x2), axis([0 0.1 -1.2 1.2]) xlabel('Time (sec)');ylabel('Amplitude');
SAW TOOTH: The sawtooth wave (or saw wave) is a kind of non-sinusoidal waveform. It is named a sawtooth based on its resemblance to the teeth on the blade of a saw. The convention is that a sawtooth wave ramps upward and then sharply drops. However, there are also sawtooth waves in which the wave ramps downward and then sharply rises. The latter type of sawtooth wave is called a 'reverse sawtooth wave' or 'inverse sawtooth wave'. As audio signals, the two orientations of sawtooth wave sound identical. The piecewise linear function based on the floor function of time t, is an example of a sawtooth wave with period 1.
A more general form, in the range 1 to 1, and with period a, is This sawtooth function has the same phase as the sine function. A sawtooth wave's sound is harsh and clear and its spectrum contains both even and odd harmonics of the
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fundamental frequency. Because it contains all the integer harmonics, it is one of the best waveforms to use for synthesizing musical sounds, particularly bowed string instruments like violins and cellos, using subtractive synthesis. Applications The sawtooth and square waves are the most common starting points used to create sounds with subtractive analog and virtual analog music synthesizers. The sawtooth wave is the form of the vertical and horizontal deflection signals used to generate a raster on CRT-based television or monitor screens. Oscilloscopes also use a sawtooth wave for their horizontal deflection, though they typically use electrostatic deflection. On the wave's "ramp", the magnetic field produced by the deflection yoke drags the electron beam across the face of the CRT, creating a scan line. On the wave's "cliff", the magnetic field suddenly collapses, causing the electron beam to return to its resting position as quickly as possible. The voltage applied to the deflection yoke is adjusted by various means (transformers, capacitors, center-tapped windings) so that the half-way voltage on the sawtooth's cliff is at the zero mark, meaning that a negative voltage will cause deflection in one direction, and a positive voltage deflection in the other; thus, a center-mounted deflection yoke can use the whole screen area to depict a trace. Frequency is 15.734 kHz on NTSC, 15.625 kHz for PAL and SECAM)
% sawtooth wave generator fs = 10000; t = 0:1/fs:1.5; x = sawtooth(2*pi*50*t); subplot(1,2,1); plot(t,x), axis([0 0.2 -1 1]); xlabel('t'),ylabel('x(t)') title('sawtooth signal'); N=2; fs = 500;n = 0:1/fs:2; x = sawtooth(2*pi*50*n); subplot(1,2,2); stem(n,x), axis([0 0.2 -1 1]); xlabel('n'),ylabel('x(n)') title('sawtooth sequence');
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Triangle wave
A triangle wave is a non-sinusoidal waveform named for its triangular shape.A bandlimited triangle wave pictured in the time domain (top) and frequency domain (bottom). The fundamental is at 220 Hz (A2).Like a square wave, the triangle wave contains only odd harmonics. However, the higher harmonics roll off much faster than in a square wave (proportional to the inverse square of the harmonic number as opposed to just the inverse).It is possible to approximate a triangle wave with additive synthesis by adding odd harmonics of the fundamental, multiplying every (4n1)th harmonic by 1 (or changing its phase by ), and rolling off the harmonics by the inverse square of their relative frequency to the fundamental.This infinite Fourier series converges to the triangle wave:
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To generate a trianguular pulse A=2; t = 0:0.0005:1; x=A*sawtooth(2*pi*5*t,0.25); %5 Hertz wave with duty cycle 25% plot(t,x); grid axis([0 1 -3 3]);
%%To generate a trianguular pulse fs = 10000;t = -1:1/fs:1; x1 = tripuls(t,20e-3); x2 = rectpuls(t,20e-3); subplot(211),plot(t,x1), axis([-0.1 0.1 -0.2 1.2]) xlabel('Time (sec)');ylabel('Amplitude'); title('Triangular Aperiodic Pulse') subplot(212),plot(t,x2), axis([-0.1 0.1 -0.2 1.2]) xlabel('Time (sec)');ylabel('Amplitude'); title('Rectangular Aperiodic Pulse') set(gcf,'Color',[1 1 1]),
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%%To generate a rectangular pulse t=-5:0.01:5; pulse = rectpuls(t,2); %pulse of width 2 time units plot(t,pulse) axis([-5 5 -1 2]); grid
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Sinusoidal Signal Generation The sine wave or sinusoid is a mathematical function that describes a smooth repetitive oscillation. It occurs often in pure mathematics, as well as physics, signal processing, electrical engineering and many other fields. Its most basic form as a function of time (t) is: where:

A, the amplitude, is the peak deviation of the function from its center position. , the angular frequency, specifies how many oscillations occur in a unit time interval, in radians per second , the phase, specifies where in its cycle the oscillation begins at t = 0.
A sampled sinusoid may be written as:
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where f is the signal frequency, fs is the sampling frequency, is the phase and A is the amplitude of the signal. The program and its output is shown below: Note that there are 64 samples with sampling frequency of 8000Hz or sampling time of 0.125 mS (i.e. 1/8000). Hence the record length of the signal is 64x0.125=8mS. There are exactly 8 cycles of sinewave, indicating that the period of one cycle is 1mS which means that the signal frequency is 1KHz.
% sinusoidal signal N=64; % Define Number of samples n=0:N-1; % Define vector n=0,1,2,3,...62,63 f=1000; % Define the frequency fs=8000; % Define the sampling frequency x=sin(2*pi*(f/fs)*n); % Generate x(t) plot(n,x); % Plot x(t) vs. t title('Sinewave [f=1KHz, fs=8KHz]'); xlabel('Sample Number'); ylabel('Amplitude');
% RAMP clc close all n=input('enter the length of ramp'); t=0:n;
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plot(t); xlabel('t'); ylabel('amplitude'); title ('ramp')
SINC FUNCTION: The sinc function computes the mathematical sinc function for an input vector or matrix x. Viewed as a function of time, or space, the sinc function is the inverse Fourier transform of the rectangular pulse in frequency centered at zero of width 2 and height 1. The following equation defines the sinc function:
The sinc function has a value of 1 whenx is equal to zero, and a value of
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% sinc x = linspace(-5,5); y = sinc(x); subplot(1,2,1);plot(x,y) xlabel(time); ylabel(amplitude); title(sinc function); subplot(1,2,2);stem(x,y); xlabel(time); ylabel(amplitude); title(sinc function);
CONCLUSION: In this experiment various signals have been generated Using

MATLAB Exersize questions:generate following signals using MATLAB
1.x(t)=e-t
2.x(t)= t 2 / 2 3.generate rectangular pulse function
4.generate signum sunction sinc(t)=
1 t>0
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0 -1
t=0 t<0
5. generate complex exponential signal x(t)= e st for different values of and
EXP.NO: 3
OPERATIONS ON SIGNALS AND SEQUENCES SUCH AS ADDITION, MULTIPLICATION, SCALING, SHIFTING, FOLDING, COMPUTATION OF ENERGY AND AVERAGE POWER
Aim: To perform arithmetic operations different types of signals Using MATLAB Software. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software THEORY :
Basic Operation on Signals:
% program for time shifting of a signal clc clear all close all t = [0:.01:pi]; % independent (time) variable A = 8; % amplitude f1 = 2; s1 = A*sin(2*pi*f1*t); subplot(3,1,1) plot(t,s1) xlabel('t'); ylabel('amplitude'); title('input signal') subplot(3,1,2) plot(t+10,s1) xlabel('t'); ylabel('amplitude'); title('right shifted signal') subplot(3,1,3)
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plot(t-10,s1) xlabel('t'); ylabel('amplitude'); title('left shifted signal')
% program for time shifting of a sequence clc clear all close all n=0:1:4; h=[1,1,1,-1,2]; subplot(3,1,1)
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stem(n,h); xlabel('n'); ylabel('h(n)'); title('impulse sequence'); subplot(3,1,2) stem(n+2,h); xlabel('n') ; ylabel('h(n)'); title('folded sequence'); subplot(3,1,3) stem(n-2,h) xlabel('n') ; ylabel('h(n)'); title('shifted sequence'); subplot(3,1,3)
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Signal Addition and Substraction : Addition: any two signals can be added to form a third signal, z (t) = x (t) + y (t)
Multiplication : Multiplication of two signals can be obtained by multiplying their values at every instants . z (t) = x (t) y (t)
%Program for addition and multiplication of signals t = [0:.01:1]; % independent (time) variable A = 8; % amplitude f1 = 2; s1 = A*sin(2*pi*f1*t); % create a 2 Hz sine wave f2 = 6; s2 = A*sin(2*pi*f2*t); % create a 4 Hz sine wave figure subplot(4,1,1) plot(t, s1) title('1 Hz sine wave') ylabel('Amplitude') %plot the 4 Hz sine wave in the middle panel
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subplot(4,1,2) ;plot(t, s2) title('2 Hz sine wave') ylabel('Amplitude') %plot the summed sine waves in the bottom panel subplot(4,1,3) ;plot(t, s1+s2) title('Summed sine waves') ylabel('Amplitude') xlabel('Time (s)') xmult=s1.*s2; subplot(4,1,4);plot(xmult); title('multiplication'); ylabel('Amplitude') xlabel('Time (s)')
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%Program for addition and multiplication & addition of sequences n1=1:1:9 s1 = [1 2 3 4 5 6 2 3 1]; subplot(4,1,1) stem(n, s1) xlabel('n1')
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title('input sequence') ylabel('Amplitude') n2=-2:1:6; s2=[1 1 1 0 2 3 1 1 0 ] subplot(4,1,2) ;stem(n2, s2) title('secong sequence') ; ylabel('Amplitude') xlabel('n2') s3=s1+s2; subplot(4,1,3) ;stem(n2,s3) title('Summed squence') ylabel('Amplitude') xlabel('n2') s4=s1.*s2; subplot(4,1,4);stem(n2,s4); title('multiplication'); ylabel('Amplitude') xlabel('n2')
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Time reversal: Time reversal of a signal x(t) can be obtained by folding the signal about t=0. Y(t)=y(-t) % Folding for signal Clc Close all Clear all t = [0:.01:1]; % independent (time) variable A = 8; % amplitude f1 = 2; % create a 2 Hz sine wave lasting 1 sec x = A*sin(2*pi*f1*t); lx=length(x); nx=0:lx-1; xf=fliplr(x); nf=-fliplr(nx); subplot(2,1,1); stem(nx,x); xlabel('nx'); ylabel('amplitude)'); title('original signal'); subplot(2,1,2); stem(nf,xf); xlabel('nf'); ylabel('amplitude'); title('folded signal');
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%signal folding clc; clear all t=0:0.1:10; x=0.5*t; lx=length(x); nx=0:lx-1; xf=fliplr(x); nf=-fliplr(nx); subplot(2,1,1); stem(nx,x); xlabel('nx'); ylabel('x(nx)'); title('original signal'); subplot(2,1,2); stem(nf,xf); xlabel('nf'); ylabel('xf(nf)');
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title('folded signal');
Signal Amplification/Attuation : Y(n)=ax(n) if a < 1 attnuation a >1 amplification
% Amplitude scalling t = [0:.01:1]; A = 8; f1 = 2; % independent (time) variable % amplitude % create a 2 Hz sine wave lasting 1 sec
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s1 = A*sin(2*pi*f1*t); subplot(3,1,1) plot(s1); xlabel('t'); ylabel('amplitude'); title('input signal'); s2=2*s1; subplot(3,1,2) plot(s2); xlabel('t'); ylabel('amplitude'); title('amplified input signal'); s3=s1/2; subplot(3,1,3) plot(s3); xlabel('t'); ylabel('amplitude'); title('atteniatedinput signal');
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% Amplitude scalling for sequences n=0:1:6 s1 = [1 2 3 3 1 1 1] subplot(3,1,1) stem(n,s1); xlabel('n'); ylabel('amplitude'); title('input signal'); s2=4*s1; subplot(3,1,2) stem(n,s2); xlabel('t'); ylabel('amplitude'); title('amplified input signal'); s3=s1/4; subplot(3,1,3) stem(n,s3);
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xlabel('t'); ylabel('amplitude'); title('atteniatedinput signal');
Time scalling: The Time scalling of a signal x(t) can be accomplished by replacing t by at where t is a scalling factor. Y(t)=x(at) : a= arbotorary constant If a<1 (a=1/2) i e x(t/2)---- the signal y(t) is expanded by 2 If a>1 (a=2) i e x(2 t )---- the signal y(t) is compressed 2
% Time scalling t = [0:.05:pi]; % independent (time) variable
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A = 8; % amplitude f1 = 2; % create a 2 Hz sine wave lasting 1 sec s1 = A*sin(2*pi*f1*t); subplot(3,1,1) plot(s1); xlabel('t'); ylabel('amplitude'); title ('sine signal') s2=A*sin(2*2*pi*f1*t); % scalling by a=4 subplot(3,1,2) plot(s2); xlabel('t'); ylabel('amplitude'); title ('scaled sine signal with a=4') s3=A*sin(.5*pi*f1*t); % scalling by a=1/4 subplot(3,1,3); plot(s3); xlabel('t'); ylabel('amplitude'); title ('scaled sine signal with a=1/4')
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CONCLUSION: Inthis experiment the various oprations on signals have been performedUsing MATLAB have been demonstrated
Excersize questions: Sketch the following questions using MATLAB 1. 2. 3. 4. 5. 6. 7. x(t)= u(-t+1) x(t)=3r(t-1) x(t)=U(n+2-u(n-3) x(n)=x1(n)+x2(n)where x1(n)={1,3,2,1},x2(n)={1,-2,3,2} x(t)=r(t)-2r(t-1)+r(t-2) x(n)=2(n+2)-2(n-4), -5 n 5. X(n)={1,2,3,4,5,6,7,6,5,4,2,1} determine and plot the following sequence a. x1(n)=2x(n-5-3x(n+4)) b. x2(n)=x(3-n)+x(n)x(n-2)
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EXP.NO: 4
FINDING THE EVEN AND ODD PARTS OF SIGNAL/SEQUENCE AND REAL AND IMAGINARY PART OF SIGNAL
Aim: program for finding even and odd parts of signals Using MATLAB Software. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software THEORY :
Even and Odd Signal

One of characteristics of signal is symmetry that may be useful for signal analysis. Even signals are symmetric around vertical axis, and Odd signals are symmetric about origin. Even Signal: A signal is referred to as an even if it is identical to its time-reversed counterparts; x(t) = x(-t). Odd Signal: A signal is odd if x(t) = -x(-t). An odd signal must be 0 at t=0, in other words, odd signal passes the origin. Using the definition of even and odd signal, any signal may be decomposed into a sum of its even part, xe(t), and its odd part, xo(t), as follows:
It is an important fact because it is relative concept of Fourier series. In Fourier series, a periodic signal can be broken into a sum of sine and cosine signals. Notice that sine function is odd signal and cosine function is even signal.
close all; clear all; t=0:.005:4*pi; x=sin(t)+cos(t); % x(t)=sint(t)+cos(t) subplot(2,2,1) plot(t,x) xlabel('t'); ylabel('amplitude') title('input signal') y=sin(-t)+cos(-t) % y=x(-t)
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subplot(2,2,2) plot(t,y) xlabel('t'); ylabel('amplitude') title('input signal with t=-t') z=x+y subplot(2,2,3) plot(t,z/2) xlabel('t'); ylabel('amplitude') title('even part of the signal') p=x-y subplot(2,2,4) plot(t,p/2) xlabel('t'); ylabel('amplitude') title('odd part of the signal')
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%even and odd signals program:

t=-4:1:4; h=[ 2 1 1 2 0 1 2 2 3 ]; subplot(3,2,1) stem(t,h); xlabel('time'); ylabel('amplitude'); title('signal'); n=9; for i=1:9 x1(i)=h(n); n=n-1; end subplot(3,2,2) stem(t,x1); xlabel('time'); ylabel('amplitude'); title('folded signal'); z=h+x1 subplot(3,2,3); stem(t,z); xlabel('time'); ylabel('amplitude'); title('sum of two signal'); subplot(3,2,4); stem(t,z/2); xlabel('time'); ylabel('amplitude'); title('even signal'); a=h-x1; subplot(3,2,5); stem(t,a); xlabel('time'); ylabel('amplitude'); title('difference of two signal'); subplot(3,2,6); stem(t,a/2); xlabel('time'); ylabel('amplitude'); title('odd signal');
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ENERGY AND POWER SIGNA L:A signal can be categorized into energy signal
or power signal: energy signal has a finite energy, 0 < E < . And Power=0 Energy signals have values only in the limited time duration. Example: a signal having only one square pulse is energy signal. A signal that decays exponentially has finite energy Energy of Contineous Signals
Energy of Discrete Signals
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power signal has a finite energy, 0 < P < . And Energy = Example: sine Power of Contonous Signals
Power of Discrete Signals
% energy clc; close all; clear all; x=[1,2,3]; n=3 e=0; for i=1:n; e=e+(x(i).*x(i)); end % energy clc; close all; clear all; N=2 x=ones(1,N) for i=1:N y(i)=(1/3)^i.*x(i); end n=N; e=0; for i=1:n; e=e+(y(i).*y(i)); end % power clc; close all; clear all; N=2 x=ones(1,N) for i=1:N
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y(i)=(1/3)^i.*x(i); end n=N; e=0; for i=1:n; e=e+(y(i).*y(i)); end p=e/(2*N+1); % power N=input('type a value for N'); t=-N:0.0001:N; x=cos(2*pi*50*t).^2; disp('the calculated power p of the signal is'); P=sum(abs(x).^2)/length(x) plot(t,x); axis([0 0.1 0 1]); disp('the theoretical power of the signal is'); P_theory=3/8
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EXP.NO: 5 LINEAR CONVOLUTION Aim: To find the out put with linear convolution operation Using MATLAB Software. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software Theory:
If x(n)=h(n) [ impulse ) then output y(n) is known as impulse response of the system. x(n)=(n) y(n)=T[x(n)]=h(n) similarly (n-k)= h(n-k) x(n) cab represented as weighted sum of impulses such as
y(n)=T[x(n)] = (n-k)= h(n-k) --- Linear Convolution equation Linear Convolution involves the following operations. 1. Folding 2. Multiplication 3. Addition 4. Shifting These operations can be represented by a Mathematical Expression as follows:
x[ ]= Input signal Samples
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h[ ]= Impulse response co-efficient. y[ ]= Convolution output. n = No. of Input samples h = No. of Impulse response co-efficient. Example : X(n)={1 2 -1 0 1}, h(n)={ 1,2,3,-1}
Program: clc; close all; clear all; x=input('enter input sequence'); h=input('enter impulse response'); y=conv(x,h); subplot(3,1,1); stem(x); xlabel('n');ylabel('x(n)'); title('input signal') subplot(3,1,2); stem(h); xlabel('n');ylabel('h(n)'); title('impulse response') subplot(3,1,3); stem(y); xlabel('n');ylabel('y(n)'); title('linear convolution') disp('The resultant signal is'); disp(y)
linear convolution output: enter input sequence[1 4 3 2] enter impulse response[1 0 2 1] The resultant signal is 1 4 5 11 10 7 2
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CONCLUSION: In this experiment convolution of various signals have been performed Using MATLAB
Applications:Convolution is used to obtain the response of an LTI system to an arbitrary input signal.It is used to find the filter response and finds application in speech processing and radar signal processing. Excersize questions: perform convolution between the following signals 1. X(n)=[1 -1 4 ], h(n) = [ -1 2 -3 1]
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2. perform convolution between the. Two periodic sequences x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2
EXP.NO: 6
6. AUTO CORRELATION AND CROSS CORRELATION BETWEEN SIGNALS AND SEQUENCES.

Aim: To compute auto correlation and cross correlation between signals and
sequences
EQUIPMENTS: PC with windows (95/98/XP/NT/2000).
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MATLAB Software Theory: Correlations of sequences: It is a measure of the degree to which two sequences are similar. Given two realvalued sequences x(n) and y(n) of finite energy, Convolution involves the following operations. 1. Shifting 2. Multiplication 3. Addition These operations can be represented by a Mathematical Expression as follows: Crosscorrelation
rx , y (l ) =
n =
x (n) y ( n l )
The index l is called the shift or lag parameter Autocorrelation

rx , x (l ) =
n =
x( n) x( n l )
The special case: y(n)=x(n)
% Cross Correlation clc; close all; clear all; x=input('enter input sequence'); h=input('enter the impulse suquence'); subplot(3,1,1); stem(x); xlabel('n'); ylabel('x(n)'); title('input signal'); subplot(3,1,2); stem(h); xlabel('n'); ylabel('h(n)'); title('impulse signal'); y=xcorr(x,h);
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subplot(3,1,3); stem(y); xlabel('n'); ylabel('y(n)'); disp('the resultant signal is'); disp(y); title('correlation signal');
% auto correlation clc; close all; clear all; x = [1,2,3,4,5]; y = [4,1,5,2,6]; subplot(3,1,1); stem(x); xlabel('n'); ylabel('x(n)'); title('input signal');
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subplot(3,1,2); stem(y); xlabel('n'); ylabel('y(n)'); title('input signal'); z=xcorr(x,x); subplot(3,1,3); stem(z); xlabel('n'); ylabel('z(n)'); title('resultant signal signal');
performed Using MATLAB
CONCLUSION: In this experiment correlation of various signals have been
Applications:it is used to measure the degree to which the two signals are similar and it is also used for radar detection by estimating the time delay.it is also used in Digital communication, defence applications and sound navigation Excersize questions: perform convolution between the following signals 1. X(n)=[1 -1 4 ], h(n) = [ -1 2 -3 1] 2. perform convolution between the. Two periodic sequences
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x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2
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EXP.NO: 7
VERIFICATION OF LINEARITY AND TIME INVARIANCE PROPERTIES OF A GIVEN CONTINUOUS /DISCRETE SYSTEM.
Aim: To compute linearity and time invariance properties of a given
continuous /discrete system

EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software THEORY:
LINEARITY PROPERTY : Any system is said to be linear if it satisfies the superposition principal superposition principal state that Response to a weigted sumn of input signal equal to the corresponding weighted sum of the outputs of the system to each of the individual input signals
X(n)-----------input signal Y(n) --------- output signal Y(n)=T[x(n)] Y1(n)=T[X1(n)] : Y2(n)=T[X2(n)]
x3=[a X1(n)] +b [X2(n) ] Y3(n)= T [x3(n)] = T [a X1(n)] +b [X2(n) ] = a Y1(n)+ b [X2(n) ]= Z 3(n) Let a [Y1(n)]+ b [X2(n) ] =Z 3(n) If Y3(n )- Z 3(n)=0 then the system is stable other wise it is nit stable
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Program 1: clc; clear all; close all; n=0:40; a=2; b=1; x1=cos(2*pi*0.1*n); x2=cos(2*pi*0.4*n); x=a*x1+b*x2; y=n.*x; y1=n.*x1; y2=n.*x2; yt=a*y1+b*y2; d=y-yt; d=round(d) if d disp('Given system is not satisfy linearity property'); else disp('Given system is satisfy linearity property'); end subplot(3,1,1), stem(n,y); grid subplot(3,1,2), stem(n,yt); grid subplot(3,1,3), stem(n,d); grid
Program2:
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clc; clear all; close all; n=0:40; a=2; b=-3; x1=cos(2*pi*0.1*n); x2=cos(2*pi*0.4*n); x=a*x1+b*x2; y=x.^2; y1=x1.^2; y2=x2.^2; yt=a*y1+b*y2; d=y-yt; d=round(d); if d disp('Given system is not satisfy linearity property'); else disp('Given system is satisfy linearity property'); end subplot(3,1,1), stem(n,y); grid subplot(3,1,2), stem(n,yt); grid subplot(3,1,3), stem(n,d); grid
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Program clc; close all; clear all; x=input('enter the sequence'); N=length(x); n=0:1:N-1; y=xcorr(x,x); subplot(3,1,1); stem(n,x); xlabel(' n----->');ylabel('Amplitude--->'); title('input seq'); subplot(3,1,2); N=length(y); n=0:1:N-1; stem(n,y); xlabel('n---->');ylabel('Amplitude----.'); title('autocorr seq for input'); disp('autocorr seq for input'); disp(y)
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p=fft(y,N); subplot(3,1,3); stem(n,p); xlabel('K----->');ylabel('Amplitude--->'); title('psd of input'); disp('the psd fun:'); disp(p)
LINEAR TIME INVARIENT SYSTEMS(LTI): A system is called time invariant if its input output characteristics do not change with time
X(t)---input : Y(t) ---output X(t-T) -----delay input by T seconds : Y(t-T) ------ Delayed output by T seconds
Program1: clc; close all; clear all; n=0:40; D=10; x=3*cos(2*pi*0.1*n)-2*cos(2*pi*0.4*n); xd=[zeros(1,D) x]; y=n.*xd(n+D); n1=n+D; yd=n1.*x; d=y-yd; if d disp('Given system is not satisfy time shifting property'); else disp('Given system is satisfy time shifting property'); end subplot(3,1,1),stem(y),grid; subplot(3,1,2),stem(yd),grid; subplot(3,1,3),stem(d),grid;
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Program2: clc; close all; clear all; n=0:40; D=10; x=3*cos(2*pi*0.1*n)-2*cos(2*pi*0.4*n); xd=[zeros(1,D) x]; x1=xd(n+D); y=exp(x1); n1=n+D; yd=exp(xd(n1)); d=y-yd; if d disp('Given system is not satisfy time shifting property'); else disp('Given system is satisfy time shifting property'); end
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subplot(3,1,1),stem(y),grid; subplot(3,1,2),stem(yd),grid; subplot(3,1,3),stem(d),grid;
CONCLUSION: In this experiment Linearity and Time invariance property of given

system has bees verified performed Using MATLAB Applications:it is used to measure the degree to which the two signals are similar and it is also used for radar detection by estimating the time delay.it is also used in Digital communication defence applications and sound navigation Excersize questions: perform convolution between the following signals 1. X(n)=[1 -1 4 ], h(n) = [ -1 2 -3 1] 2. perform convolution between the. Two periodic sequences x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2
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EXP.NO:8
COMPUTATION OF UNIT SAMPLE, UNIT STEP AND SINUSOIDAL RESPONSE OF THE GIVEN LTI SYSTEM AND VERIFYING ITS PHYSICAL REALIZABILITY AND STABILITY PROPERTIES.
Aim: To Unit Step And Sinusoidal Response Of The Given LTI System And Verifying Its Physical Realizability And Stability Properties. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software A discrete time system performs an operation on an input signal based on predefined criteria to produce a modified output signal. The input signal x(n) is the system excitation, and y(n) is the system response. The transform operation is shown as,
If the input to the system is unit impulse i.e. x(n) = (n) then the output of the system is known as impulse response denoted by h(n) where, h(n) = T[(n)] we know that any arbitrary sequence x(n) can be represented as a weighted sum of discrete impulses. Now the system response is given by,
For linear system (1) reduces to
%given difference equation y(n)-y(n-1)+.9y(n-2)=x(n);
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%calculate and plot the impulse response and step response b=[1]; a=[1,-1,.9]; x=impseq(0,-20,120); n = [-20:120]; h=filter(b,a,x); subplot(3,1,1);stem(n,h); title('impulse response'); xlabel('n');ylabel('h(n)'); =stepseq(0,-20,120); s=filter(b,a,x); s=filter(b,a,x); subplot(3,1,2); stem(n,s); title('step response'); xlabel('n');ylabel('s(n)') t=0:0.1:2*pi; x1=sin(t); %impseq(0,-20,120); n = [-20:120]; h=filter(b,a,x1); subplot(3,1,3);stem(h); title('sin response'); xlabel('n');ylabel('h(n)'); figure; zplane(b,a);
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CONCLUSION: In this experiment computation of unit sample, unit step and sinusoidal response of the given lti system and verifying its physical realizability and stability properties Using MATLAB
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EXP.NO: 9
GIBBS PHENOMENON
Aim: To verify the Gibbs Phenomenon. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software the Gibbs phenomenon, the Fourier series of a piecewise continuously differentiable periodic function behaves at a jump discontinuity.the n the approximated function shows amounts of ripples at the points of discontinuity. This is known as the Gibbs Phenomina . partial sum of the Fourier series has large oscillations near the jump, which might increase the maximum of the partial sum above that of the function itself. The overshoot does not die out as the frequency increases, but approaches a finite limit The Gibbs phenomenon involves both the fact that Fourier sums overshoot at a jump discontinuity, and that this overshoot does not die out as the frequency increases Gibbs Phenomina Program : t=0:0.1:(pi*8); y=sin(t); subplot(5,1,1); plot(t,y); xlabel('k'); ylabel('amplitude'); title('gibbs phenomenon'); h=2; %k=3; for k=3:2:9 y=y+sin(k*t)/k; subplot(5,1,h); plot(t,y); xlabel('k'); ylabel('amplitude'); h=h+1; end
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CONCLUSION: In this experiment Gibbs phenomenon have been demonstrated

Using MATLAB
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EXP.NO: 10.
FINDING THE FOURIER TRANSFORM OF A GIVEN SIGNAL AND PLOTTING ITS MAGNITUDE AND PHASE SPECTRUM
Aim: to find the fourier transform of a given signal and plotting its
magnitude and phase spectrum

EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software
Fourier Transform TheoremsL:

the Fourier transform as follows. Suppose that is a function which is zero outside of some interval [L/2, L/2]. Then for any T L we may expand in a Fourier series on the interval [T/2,T/2], where the "amount" of the wave e2inx/T in the Fourier series of is given by By definition Fourier Transform of signal f(t) is defined as
Iverse Fourier Transform of signal F(w) is defined as
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PROGRAM:
Aim: To compute N-point FFT EQUIPMENTS:

PC with windows (95/98/XP/NT/2000). MATLAB Software
Theory:
DFT of a sequence X[K] =
N 1
K = 0
x [n ]e
j 2 Kn N
Where N= Length of sequence. K= Frequency Coefficient. n = Samples in time domain. FFT : -Fast Fourier transformer . There are Two methods. 1.Decimation in time (DIT FFT). 2. Decimation in Frequency (DIF FFT).
Program: clc; close all; clear all; x=input('enter the sequence'); N=length(x); n=0:1:N-1; y=fft(x,N) subplot(2,1,1); stem(n,x); title('input sequence');
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xlabel('time index n----->'); ylabel('amplitude x[n]----> '); subplot(2,1,2); stem(n,y); title('output sequence'); xlabel(' Frequency index K---->'); ylabel('amplitude X[k]------>');
FFT magnitude and Phase plot: clc close all x=[1,1,1,1,zeros(1,4)]; N=8; X=fft(x,N);
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magX=abs(X),phase=angle(X)*180/pi; subplot(2,1,1) plot(magX); grid xlabel('k') ylabel('X(K)') subplot(2,1,2) plot(phase); grid xlabel('k') ylabel('degrees')
Applications: The no of multiplications in DFT = N2.
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The no of Additions in DFT = N(N-1). For FFT. The no of multiplication = N/2 log 2N. The no of additions = N log2 N.
CONCLUSION: In this experiment the fourier transform of a given signal and plotting its magnitude and phase spectrum have been demonstrated using matlab
Exp:11 LAPLECE TRNASFORMS Aim: To perform waveform synthesis using Laplece Trnasforms of a given signal
Bilateral Laplace transform :

When one says "the Laplace transform" without qualification, the unilateral or one-sided transform is normally intended. The Laplace transform can be alternatively defined as the bilateral Laplace transform or two-sided Laplace transform by extending the limits of integration to be the entire real axis. If that is done the common unilateral transform simply becomes a special case of the bilateral transform where the definition of the function being transformed is multiplied by the Heaviside step function. The bilateral Laplace transform is defined as follows:
Inverse Laplace transform The inverse Laplace transform is given by the following complex integral, which is known by various names (the Bromwich integral, the Fourier-Mellin integral, and Mellin's inverse formula):
Program for Laplace Transform:

f=t syms f t; f=t; laplace(f)
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Program for nverse Laplace Transform

f(s)=24/s(s+8) invese LT syms F s F=24/(s*(s+8)); ilaplace(F) y(s)=24/s(s+8) invese LT poles and zeros
Signal synthese using Laplace Tnasform: clear all clc t=0:1:5 s=(t); subplot(2,3,1) plot(t,s); u=ones(1,6) subplot(2,3,2) plot(t,u); f1=t.*u; subplot(2,3,3) plot(f1); s2=-2*(t-1); subplot(2,3,4); plot(s2); u1=[0 1 1 1 1 1]; f2=-2*(t-1).*u1; subplot(2,3,5); plot(f2); u2=[0 0 1 1 1 1]; f3=(t-2).*u2; subplot(2,3,6); plot(f3); f=f1+f2+f3; figure; plot(t,f); % n=exp(-t); % n=uint8(n); % f=uint8(f); % R = int(f,n,0,6) laplace(f);
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CONCLUSION: In this experiment the Triangular signal synthesised using

Laplece Trnasforms using MATLAB
Applications of laplace transforms: 1. Derive the circuit (differential) equations in the time domain, then transform these ODEs to the s-domain; 2. Transform the circuit to the s-domain, then derive the circuit equations in the sdomain (using the concept of "impedance").
The main idea behind the Laplace Transformation is that we can solve an equation (or system of equations) containing differential and integral terms by transforming the equation in "t-space" to one in "s-space". This makes the problem much easier to solve
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EXP.NO: 12
LOCATING THE ZEROS AND POLES AND PLOTTING THE POLE ZERO MAPS IN S-PLANE AND Z-PLANE FOR THE GIVEN TRANSFER FUNCTION.
Aim: To locating the zeros and poles and plotting the pole zero maps in s-
plane and z-plane for the given transfer function

Z-transforms
the Z-transform converts a discrete time-domain signal, which is a sequence of real or complex numbers, into a complex frequency-domain representation.The Z-transform, like many other integral transforms, can be defined as either a one-sided or two-sided transform.
Bilateral Z-transform
The bilateral or two-sided Z-transform of a discrete-time signal x[n] is the function X(z) defined as
Unilateral Z-transform
Alternatively, in cases where x[n] is defined only for n 0, the single-sided or unilateral Z-transform is defined as
In signal processing, this definition is used when the signal is causal.
The roots of the equation P(z) = 0 correspond to the 'zeros' of X(z) The roots of the equation Q(z) = 0 correspond to the 'poles' of X(z)
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The ROC of the Z-transform depends on the convergence of the
clc; close all clear all; %b= input('enter the numarator cofficients') %a= input('enter the dinomi cofficients') b=[1 2 3 4] a=[1 2 1 1 ] zplane(b,a);
Applications :Z-Transform is used to find the system responses
CONCLUSION: In this experiment the zeros and poles and plotting the pole zero maps in s-plane and z-plane for the given transfer function
using MATLAB
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EXP.NO: 13
13. Gaussian noise

%Estimation of Gaussian density and Distribution Functions %% Closing and Clearing all clc; clear all; close all; %% Defining the range for the Random variable dx=0.01; %delta x x=-3:dx:3; [m,n]=size(x); %% Defining the parameters of the pdf mu_x=0; % mu_x=input('Enter the value of mean'); sig_x=0.1; % sig_x=input('Enter the value of varience'); %% Computing the probability density function px1=[]; a=1/(sqrt(2*pi)*sig_x); for j=1:n px1(j)=a*exp([-((x(j)-mu_x)/sig_x)^2]/2); end %% Computing the cumulative distribution function cum_Px(1)=0; for j=2:n cum_Px(j)=cum_Px(j-1)+dx*px1(j); end %% Plotting the results figure(1) plot(x,px1);grid axis([-3 3 0 1]); title(['Gaussian pdf for mu_x=0 and sigma_x=', num2str(sig_x)]); xlabel('--> x') ylabel('--> pdf') figure(2) plot(x,cum_Px);grid axis([-3 3 0 1]); title(['Gaussian Probability Distribution Function for mu_x=0 and sigma_x=', num2str(sig_x)]); title('\ite^{\omega\tau} = cos(\omega\tau) + isin(\omega\tau)') xlabel('--> x') ylabel('--> PDF')
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EXP.NO: 14
14. Sampling theorem verification

Aim: To detect the edge for single observed image using sobel edge detection and canny edge detection. EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software Sampling Theorem: \A bandlimited signal can be reconstructed exactly if it is sampled at a rate atleast twice the maximum frequency component in it." Figure 1 shows a signal g(t) that is bandlimited.
Figure 1: Spectrum of bandlimited signal g(t) The maximum frequency component of g(t) is fm. To recover the signal g(t) exactly from its samples it has to be sampled at a rate fs 2fm. The minimum required sampling rate fs = 2fm is called ' Nyquist rate Proof: Let g(t) be a bandlimited signal whose bandwidth is fm (wm = 2fm).
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Figure 2: (a) Original signal g(t) (b) Spectrum G(w) (t) is the sampling signal with fs = 1/T > 2fm.
Figure 3: (a) sampling signal (t) ) (b) Spectrum (w) Let gs(t) be the sampled signal. Its Fourier Transform Gs(w) isgiven by
Figure 4: (a) sampled signal gs(t) (b) Spectrum Gs(w)
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To recover the original signal G(w): 1. Filter with a Gate function, H2wm(w) of width 2wm Scale it by T.
Figure 5: Recovery of signal by filtering with a fiter of width 2wm Aliasing { Aliasing is a phenomenon where the high frequency components of the sampled signal interfere with each other because of inadequate sampling ws < 2wm.
Figure 6: Aliasing due to inadequate sampling
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Aliasing leads to distortion in recovered signal. This is the reason why sampling frequency should be atleast twice thebandwidth of the signal. Oversampling { In practice signal are oversampled, where fs is signi_cantly higher than Nyquist rate to avoid aliasing.
Figure 7: Oversampled signal-avoids aliasing t=-10:.01:10; T=4; fm=1/T; x=cos(2*pi*fm*t); subplot(2,2,1); plot(t,x); xlabel('time');ylabel('x(t)') title('continous time signal') grid; n1=-4:1:4 fs1=1.6*fm; fs2=2*fm; fs3=8*fm; x1=cos(2*pi*fm/fs1*n1); subplot(2,2,2); stem(n1,x1); xlabel('time');ylabel('x(n)') title('discrete time signal with fs<2fm') hold on subplot(2,2,2); plot(n1,x1) grid; n2=-5:1:5; x2=cos(2*pi*fm/fs2*n2); subplot(2,2,3); stem(n2,x2); xlabel('time');ylabel('x(n)') title('discrete time signal with fs=2fm') hold on subplot(2,2,3); plot(n2,x2) grid;
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n3=-20:1:20; x3=cos(2*pi*fm/fs3*n3); subplot(2,2,4); stem(n3,x3); xlabel('time');ylabel('x(n)') title('discrete time signal with fs>2fm') hold on subplot(2,2,4); plot(n3,x3) grid;
CONCLUSION: In this experiment the sampling theorem have been verified

using MATLAB
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EXP.No:15
REMOVAL OF NOISE BY AUTO CORRELATION/CROSS CORRELATION

Aim: removal of noise by auto correlation/cross correlation
EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software Detection of a periodic signal masked by random noise is of greate importance .The noise signal encountered in practice is a signal with random amplitude variations. A signal is uncorrelated with any periodic signal. If s(t) is a periodic signal and n(t) is a noise signal then T/2 Lim 1/T S(t)n(t-T) dt=0 T-- -T/2 for all T
Qsn(T)= cross correlation function of s(t) and n(t) Then Qsn(T)=0 Detection of noise by Auto-Correlation: S(t)=Periodic Signal (Transmitted) , mixed with a noise signal n(t). Then f(t) is received signal is [s(t ) + n(t) ] Let Qff(T) =Auto Correlation Function of f(t) Qss(t) = Auto Correlation Function of S(t) Qnn(T) = Auto Correlation Function of n(t) T/2 Qff(T)= Lim 1/T f(t)f(t-T) dt T-- -T/2 T/2 = Lim 1/T [s(t)+n(t)] [s(t-T)+n(t-T)] dt T-- -T/2 =Qss(T)+Qnn(T)+Qsn(T)+Qns(T) The periodic signal s(t) and noise signal n(t) are uncorrelated Qsn(t)=Qns(t)=0 ; Then Qff(t)=Qss(t)+Qnn(t)
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The Auto correlation function of a periodic signal is periodic of the same frequency and the Auto correlation function of a non periodic signal is tends to zero for large value of T since s(t) is a periodic signal and n(t) is non periodic signal so Qss(T) is a periodic where as aQnn(T) becomes small for large values of T Therefore for sufficiently large values of T Qff(T) is equal to Qss(T). Detection by Cross Correlation: f(t)=s(t)+n(t) c(t)=Locally generated signal with same frequencyas that of S(t) Qfc (t) T/2 = Lim 1/T [s(t)+n(t)] [ c(t-T)] dt T-- -T/2
= Qsc(T)+Qnc(T) C(t) is periodic function and uncorrelated with the random noise signal n(t). Hence Qnc(T0=0) Therefore Qfc(T)=Qsc(T)
a)auto correlation clear all clc t=0:0.1:pi*4; s=sin(t); k=2; subplot(6,1,1) plot(s); title('signal s'); xlabel('t'); ylabel('amplitude'); n = randn([1 126]); f=s+n; subplot(6,1,2) plot(f); title('signal f=s+n'); xlabel('t'); ylabel('amplitude'); as=xcorr(s,s); subplot(6,1,3) plot(as); title('auto correlation of s'); xlabel('t'); ylabel('amplitude'); an=xcorr(n,n);
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subplot(6,1,4) plot(an); title('auto correlation of n'); xlabel('t'); ylabel('amplitude'); cff=xcorr(f,f); subplot(6,1,5) plot(cff); title('auto correlation of f'); xlabel('t'); ylabel('amplitude'); hh=as+an; subplot(6,1,6) plot(hh); title('addition of as+an'); xlabel('t'); ylabel('amplitude');
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B)CROSS CORRELATION : clear all clc t=0:0.1:pi*4; s=sin(t); k=2; %sk=sin(t+k); subplot(7,1,1) plot(s);
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title('signal s');xlabel('t');ylabel('amplitude'); c=cos(t); subplot(7,1,2) plot(c); title('signal c');xlabel('t');ylabel('amplitude'); n = randn([1 126]); f=s+n; subplot(7,1,3) plot(f); title('signal f=s+n');xlabel('t');ylabel('amplitude'); asc=xcorr(s,c); subplot(7,1,4) plot(asc); title(' correlation of s and c');xlabel('t');ylabel('amplitude'); anc=xcorr(n,c); subplot(7,1,5) plot(anc); title(' correlation of n and c');xlabel('t');ylabel('amplitude'); cfc=xcorr(f,c); subplot(7,1,6) plot(cfc); title(' correlation of f and c');xlabel('t');ylabel('amplitude'); hh=asc+anc; subplot(7,1,7) plot(hh); title('addition of sc+nc');xlabel('t');ylabel('amplitude');
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Applications:detection of radar and sonal signals ,detection of cyclical component in brain analysis meterology etc. CONCLUSION: in this experiment the removal of noise by auto correlation/cross correlation have been verified using MATLAB
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EXP.No:16
EXTRACTION OF PERIODIC SIGNAL MASKED BY NOISE USING CORRELATION

clear all; close all; clc; n=256; k1=0:n-1; x=cos(32*pi*k1/n)+sin(48*pi*k1/n); plot(k1,x) %Module to find period of input signl k=2; xm=zeros(k,1); ym=zeros(k,1); hold on for i=1:k [xm(i) ym(i)]=ginput(1); plot(xm(i), ym(i),'r*'); end period=abs(xm(2)-xm(1)); rounded_p=round(period); m=rounded_p % Adding noise and plotting noisy signal
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y=x+randn(1,n); figure plot(k1,y) % To generate impulse train with the period as that of input signal d=zeros(1,n); for i=1:n if (rem(i-1,m)==0) d(i)=1; end end %Correlating noisy signal and impulse train cir=cxcorr1(y,d);
%plotting the original and reconstructed signal m1=0:n/4; figure plot(m1,x(m1+1),'r',m1,m*cir(m1+1));
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Application
The theorem is useful for analyzing linear time-invariant systems, LTI systems, when the inputs and outputs are not square integrable, so their Fourier transforms do not exist. A corollary is that the Fourier transform of the autocorrelation function of the output of an LTI system is equal to the product of the Fourier transform of the autocorrelation function of the input of the system times the squared magnitude of the Fourier transform of the system impulse response. This works even when the Fourier transforms of the input and output signals do not exist because these signals are not square integrable, so the system inputs and outputs cannot be directly related by the Fourier transform of the impulse response. Since the Fourier transform of the autocorrelation function of a signal is the power spectrum of the signal, this corollary is equivalent to saying that the power spectrum of the output is equal to the power spectrum of the input times the power transfer function. This corollary is used in the parametric method for power spectrum estimation.
CONCLUSION: In this experiment the Weiner-Khinchine Relation have been verified using MATLAB
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EXP.No:17
VERIFICATION OF WIENERKHINCHIN RELATION AIM: verification of wienerkhinchin relation

EQUIPMENTS: PC with windows (95/98/XP/NT/2000). MATLAB Software The WienerKhinchin theorem (also known as the WienerKhintchine theorem and sometimes as the WienerKhinchinEinstein theorem or the KhinchinKolmogorov theorem) states that the power spectral density of a wide-sense-stationary random process is the Fourier transform of the corresponding autocorrelation function.[1][2][3] Continuous case:
Where
is the autocorrelation function defined in terms of statistical expectation, and where is the power spectral density of the function . Note that the autocorrelation function is defined in terms of the expected value of a product, and that the Fourier transform of does not exist in general, because stationary random functions are not square integrable. The asterisk denotes complex conjugate, and can be omitted if the random process is realvalued. Discrete case:
Where
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and where is the power spectral density of the function with discrete values . Being a sampled and discrete-time sequence, the spectral density is periodic in the frequency domain. PROGRAM: clc clear all; t=0:0.1:2*pi; x=sin(2*t); subplot(3,2,1); plot(x); au=xcorr(x,x); subplot(3,2,2); plot(au); v=fft(au); subplot(3,2,3); plot(abs(v)); fw=fft(x); subplot(3,2,4); plot(fw); fw2=(abs(fw)).^2; subplot(3,2,5); plot(fw2);
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EXP18. CHECKING A RANDOM PROCESS FOR STATIONARITY IN WIDE SENSE. AIM:Checking a random process for stationarity in wide sense.
Theory:
a stationary process (or strict(ly) stationary process or strong(ly) stationary process) is a stochastic process whose joint probability distribution does not change when shifted in time or space. As a result, parameters such as the mean and variance, if they exist, also do not change over time or position..
Definition
Formally, let Xt be a stochastic process and let represent the cumulative distribution function of the joint distribution of Xt at times t1..tk. Then, Xt is said to be stationary if, for all k, for all , and for all t1..tk
Weak or wide-sense stationarity

A weaker form of stationarity commonly employed in signal processing is known as weak-sense stationarity, wide-sense stationarity (WSS) or covariance stationarity. WSS random processes only require that 1st and 2nd moments do not vary with respect to time. Any strictly stationary process which has a mean and a covariance is also WSS. So, a continuous-time random process x(t) which is WSS has the following restrictions on its mean function
and autocorrelation function
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The first property implies that the mean function mx(t) must be constant. The second property implies that the correlation function depends only on the difference between t1 and t2 and only needs to be indexed by one variable rather than two variables. Thus, instead of writing,
we usually abbreviate the notation and write
This also implies that the autocovariance depends only on
= t1 t2, since
When processing WSS random signals with linear, time-invariant (LTI) filters, it is helpful to think of the correlation function as a linear operator. Since it is a circulant operator (depends only on the difference between the two arguments), its eigenfunctions are the Fourier complex exponentials. Additionally, since the eigenfunctions of LTI operators are also complex exponentials, LTI processing of WSS random signals is highly tractableall computations can be performed in the frequency domain. Thus, the WSS assumption is widely employed in signal processing algorithms.
Applicatons: Stationarity is used as a tool in time series analysis, where the raw data are often transformed to become stationary, for example, economic data are often seasonal and/or dependent on the price level. Processes are described as trend stationary if they are a linear combination of a stationary process and one or more processes exhibiting a trend. Transforming these data to leave a stationary data set for analysis is referred to as de-trending
Stationary and Non Stationary Random Process:
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A random X(t) is stationary if its statistical properties are unchanged by a time shift in the time origin.When the auto-Correlation function Rx(t,t+T) of the random X(t) varies with time difference T and the mean value of the random variable X(t1) is independent of the choice of t1,then X(t) is said to be stationary in the wide-sense or wide-sense stationary . So a continous- Time random process X(t) which is WSS has the following properties 1) E[X(t)]=X(t)= X(t+T) 2) The Autocorrelation function is written as a function of T that is 3) RX(t,t+T)=Rx(T) If the statistical properties like mean value or moments depends on time then the random process is said to be non-stationary. When dealing wih two random process X(t) and Y(t), we say that they are jointly wide-sense stationary if each pocess is stationary in the wide-sense. Rxy(t,t+T)=E[X(t)Y(t+T)]=Rxy(T).
MATLAB PROGRAM: clear all clc y = randn([1 40]) my=round(mean(y)); z=randn([1 40]) mz=round(mean(z)); vy=round(var(y)); vz=round(var(z)); t = sym('t','real'); h0=3; x=y.*sin(h0*t)+z.*cos(h0*t); mx=round(mean(x)); k=2; xk=y.*sin(h0*(t+k))+z.*cos(h0*(t+k)); x1=sin(h0*t)*sin(h0*(t+k)); x2=cos(h0*t)*cos(h0*(t+k)); c=vy*x1+vz*x1; %if we solve "c=2*sin(3*t)*sin(3*t+6)" we get c=2cos(6) %which is a costant does not depent on variable 't' % so it is wide sence stationary
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1. Define Signal 2. Define determistic and Random Signal 3. Define Delta Function 4. What is Signal Modeling 5. Define Periodic and a periodic Signal 6. Define Symetric and Anti-Symmetric Signals 7. Define Continuous and Discrete Time Signals 8. What are the Different types of representation of discrete time signals 9. What are the Different types of Operation performed on signals 10. What is System 11. What is Causal Signal 12. What are the Different types of Systems 13. What is Linear System 14. What is Time Invariant System 15. What is Static and Dynamic System 16. What is Even Signal 17. What is Odd Signal 18. Define the Properties of Impulse Signal 19. What is Causality Condition of the Signal 20. What is Condition for System Stability 21. Define Convolution 22. Define Properties of Convolution 23. What is the Sufficient condition for the existence of F.T 24. Define the F.T of a signal 25. State Paesevals energy theorem for a periodic signal 26. Define sampling Theorem 27. What is Aliasing Effect 28. what is Under sampling 29. What is Over sampling 30. Define Correlation 31. Define Auto-Correlation 32. Define Cross-Correlation 33. Define Convolution 34. Define Properties of Convolution 35. What is the Difference Between Convolution& Correlation 36. What are Dirchlet Condition 37. Define Fourier Series 38. What is Half Wave Symmetry 39. What are the properties of Continuous-Time Fourier Series 40. Define Laplace-Transform
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41. What is the Condition for Convergence of the L.T 42. What is the Region of Convergence(ROC) 43. State the Shifting property of L.T 44. State convolution Property of L.T 45. Define Transfer Function 46. Define Pole-Zeros of the Transfer Function 47. What is the Relationship between L.T & F.T &Z.T 48. Fined the Z.T of a Impulse and step 49. What are the Different Methods of evaluating inverse z-T 50. Explain Time-Shifting property of a Z.T 51. what are the ROC properties of a Z.T 52. Define Initial Value Theorem of a Z.T 53. Define Final Value Theorem of a Z.T 54. Define Sampling Theorem 55. Define Nyquist Rate 56. Define Energy of a Signal 57. Define Power of a signal 58. Define Gibbs Phenomena 59. Define the condition for distortionless transmission through the system 60. What is signal band width 61. What is system band width 62. What is Paley-Winer criterion? 63. Derive relationship between rise time and band width 64.
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Basic Simulation Lab Manual

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Basic Simulation Lab Manual

Uploaded by

www.jntuworld.

BASIC OPERATIONS ON MATRICES

Let's find the inverse of a matrix X = inv(A) X= 5 2 -2 1 1

to obtain eigenvalues eig(A) ans = 3.7321

0.2679 1.0000 as the singular value decomposition. svd(A) ans =

12.3171 0.5149 0.1577

CONCLUSION: Inthis experiment basic operations on matrices Using MATLAB

2)Unit Step Function u(t):

b)Unit Step Sequence u(n): )=

% unit step generation n1=-4; n2=5; n0=0; [y,n]=stepseq(n0,n1,n2); stem(n,y); xlabel('n')

ylabel('amplitude'); title('unit step');

A sampled sinusoid may be written as:

% RAMP clc close all n=input('enter the length of ramp'); t=0:n;

plot(t); xlabel('t'); ylabel('amplitude'); title ('ramp')

CONCLUSION: In this experiment various signals have been generated Using

4.generate signum sunction sinc(t)=

5. generate complex exponential signal x(t)= e st for different values of and

Basic Operation on Signals:

plot(t-10,s1) xlabel('t'); ylabel('amplitude'); title('left shifted signal')

Signal Amplification/Attuation : Y(n)=ax(n) if a < 1 attnuation a >1 amplification

xlabel('t'); ylabel('amplitude'); title('atteniatedinput signal');

% Time scalling t = [0:.05:pi]; % independent (time) variable

Even and Odd Signal

%even and odd signals program:

Energy of Discrete Signals

Power of Discrete Signals

x[ ]= Input signal Samples

6. AUTO CORRELATION AND CROSS CORRELATION BETWEEN SIGNALS AND SEQUENCES.

The index l is called the shift or lag parameter Autocorrelation

The special case: y(n)=x(n)

performed Using MATLAB

CONCLUSION: In this experiment correlation of various signals have been

x1(t)=e-3t{u(t)-u(t-2)} , x2(t)= e -3t for 0 t 2

continuous /discrete system

X(n)-----------input signal Y(n) --------- output signal Y(n)=T[x(n)] Y1(n)=T[X1(n)] : Y2(n)=T[X2(n)]

subplot(3,1,1),stem(y),grid; subplot(3,1,2),stem(yd),grid; subplot(3,1,3),stem(d),grid;

CONCLUSION: In this experiment Linearity and Time invariance property of given

For linear system (1) reduces to

%given difference equation y(n)-y(n-1)+.9y(n-2)=x(n);

CONCLUSION: In this experiment Gibbs phenomenon have been demonstrated

magnitude and phase spectrum

Fourier Transform TheoremsL:

Iverse Fourier Transform of signal F(w) is defined as

Aim: To compute N-point FFT EQUIPMENTS:

Applications: The no of multiplications in DFT = N2.

Bilateral Laplace transform :

Program for Laplace Transform:

Program for nverse Laplace Transform

CONCLUSION: In this experiment the Triangular signal synthesised using

plane and z-plane for the given transfer function

In signal processing, this definition is used when the signal is causal.

The ROC of the Z-transform depends on the convergence of the

Applications :Z-Transform is used to find the system responses

13. Gaussian noise

14. Sampling theorem verification

Figure 4: (a) sampled signal gs(t) (b) Spectrum Gs(w)

Figure 6: Aliasing due to inadequate sampling

CONCLUSION: In this experiment the sampling theorem have been verified

REMOVAL OF NOISE BY AUTO CORRELATION/CROSS CORRELATION

EXTRACTION OF PERIODIC SIGNAL MASKED BY NOISE USING CORRELATION

%plotting the original and reconstructed signal m1=0:n/4; figure plot(m1,x(m1+1),'r',m1,m*cir(m1+1));

VERIFICATION OF WIENERKHINCHIN RELATION AIM: verification of wienerkhinchin relation

Weak or wide-sense stationarity

and autocorrelation function

we usually abbreviate the notation and write

This also implies that the autocovariance depends only on

Stationary and Non Stationary Random Process:

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