VNRVJIET

Name of the Laboratory: Name of the Experiment: _____

Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

ELECTRONICS AND COMMUNICATION

IV B.Tech. I Semester

Signal Processing and

Communication Applications
Laboratory

Submitted By:
BADRI ROHITH
18071A0465

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

INDEX

1. Removal of various types of Noise from signals acquired by sensors.

2. Removal of noise from Image data for further Image processing.
3. Design of FIR adaptive filter as an Equalization filter
4. Develop OFDM standard complaint waveforms using FFT and other blocks
5. Digital modulation and demodulation in AWGN and other channel setting.
6. Evaluate different performance aspects required for lossy and lossless Compression methods.
7. Perform Automatic gain and frequency control for an Audio signal.
8. Audio compression algorithms (Using Sub band coding and Linear predictive code)

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

1. Removal of various types of Noise from signals acquired by sensors.

Aim: To verify removal of noise by Moving Mean / Average.

Software used: MATLAB, Python.

Theory: Types of noise: Wind, Electrical Noise, Interference, Ground Loops, Rumble,
Mechanical Noises, Equipment Self Noise, Background Noise. One of the problems is having
sharp disturbances or noise while having a smooth conversation. Moving mean method is used to
reduce or smoothen the sharp disturbances in a audio file or a conversation. In this method, the
sliding window technique is used where the window will be moving, and we calculate the mean
or average of the elements present in that window.

Python Program:

# Importing Libraries
import numpy as np
import matplotlib.pyplot as plt

# Signal
Signal = np.random.uniform(10,40,50)

# Moving Average Filter

WindowSize = 3
FilteredSignal = np.convolve(Signal, np.ones(WindowSize), 'valid') / WindowSize

# Plots
plt.plot(Signal, label='Signal', color='cyan')
plt.plot(FilteredSignal,label='Filtered Signal', color='crimson')
plt.legend()
plt.xlabel('Time')
plt.ylabel('Magnitude')
plt.title('Denoising Sensor Data using Moving Average Filter')
plt.show()

MATLAB Program:
clear
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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

close all
clc

%To load your own Audio file

%[y,fs] = audioread('D.wav');
% "handel" is a sample audio file present in Matlab
load handel
y = y(1:256);

%sound(y)%function to hear the audio file

subplot(1,2,1)
plot(y)
title('Audio File With Noise');
ysmm = movmean(y, 3, 'endpoints', 'discard' );
subplot(2,2,1)
plot(ysmm)
title('Audio File After Filtering');

Output Graphs:

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

2. Removal of noise from Image data for further Image processing.

Aim: To remove the noise (salt pepper noise) from Image data using averaging filter and median
filters.

Software used: MATLAB.

Python Program:

# Importing Libraries
from skimage import io, data, color, filters
from skimage.morphology import disk
import skimage
import matplotlib.pyplot as plt

# Image
Image = color.rgb2gray(data.rocket())

plt.figure('Original Image')
io.imshow(Image)

# Salt and Pepper Noise

ImageSP = skimage.util.random_noise(Image, 's&p')
ImageSPF = filters.median(ImageSP)

plt.figure('Salt and Pepper Noise')

plt.subplot(1,2,1)
io.imshow(ImageSP)
plt.title('Noisy Image')
plt.subplot(1,2,2)
io.imshow(ImageSPF)
plt.title('Filtered Image')

# Speckle Noise
ImageS = skimage.util.random_noise(Image, 'speckle')
ImageSF = filters.rank.mean(ImageS, disk(2))

plt.figure('Speckle Noise')

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

plt.subplot(1,2,1)
io.imshow(ImageS)
plt.title('Noisy Image')
plt.subplot(1,2,2)
io.imshow(ImageSF)
plt.title('Filtered Image')

# Gaussian Noise
ImageG = skimage.util.random_noise(Image, 'gaussian')
ImageGF = filters.gaussian(ImageG)

plt.figure('Gaussian Noise')
plt.subplot(1,2,1)
io.imshow(ImageG)
plt.title('Noisy Image')
plt.subplot(1,2,2)
io.imshow(ImageGF)
plt.title('Filtered Image')
plt.show()

MATLAB program:

Filter the noise image with an averaging filter:

a=imread('eight.tif');
imshow(a);
j=imnoise(a,'salt & pepper',0.02);
figure,imshow(j);
k=filter2(fspecial('average',3),j)/255;
figure,imshow(k);

Original noised image Filtered Image

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

Median filters to filter the noisy image:

i=imread('pout.tif');
imshow(i);
j=imnoise(i,'salt & pepper',0.02);
figure,imshow(j);
l=medfilt2(j,[3 3]);
figure,imshow(l);

Original noise image

Medfilter add

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

3. Design of FIR adaptive filter as an Equalization filter

Aim: To design an FIR adaptive filter as an equalization filter
Software Used: Python
Python Program:
# Importing Libraries
import audio_dspy as adsp
import numpy as np
import matplotlib.pyplot as plt

fs = 44100 # sample rate

worN = np.logspace(1, 3.3, num=1000, base=20) # frequencies to plot

# Equalizer Design
eq = adsp.EQ(fs)
eq.add_LPF(10000, 0.707)
eq.add_lowshelf(200, 1.4, 2)
eq.add_notch(880, 0.707)

# Equalizer Magnitude Response

eq.plot_eq_curve(worN=worN)
plt.title('Adaptive Equalization Filter')
plt.show()

Output Graph:

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

4. Develop OFDM standard complaint waveforms using FFT and other

blocks
Aim: Develop OFDM standard complaint waveforms using FFT and other blocks.
Software used: MATLAB
Theory:
In telecommunications, orthogonal frequency-division multiplexing (OFDM) is a type of digital
transmission and a method of encoding digital data on multiple carrier frequencies. OFDM has
developed into a popular scheme for wideband digital communication, used in applications such
as digital television and audio broadcasting, In OFDM, multiple closely
spaced orthogonal subcarrier signals with overlapping spectra are transmitted to carry data in
parallel. Demodulation is based on fast Fourier transform algorithms. Each subcarrier (signal) is
modulated with a conventional modulation scheme (such as quadrature amplitude
modulation or phase-shift keying) at a low symbol rate. This maintains total data rates like
conventional single-carrier modulation schemes in the same bandwidth.
Program:
fsMHz = 20; % sampling frequency
fcMHz = 1.5625; % signal frequency
N = 128; % fft size

% generating the time domain signal

x1T = exp(j*2*pi*fcMHz*[0:N-1]/fsMHz);
x1F = fft(x1T,N); % 128 pt FFT
figure;
plot([-N/2:N/2-1]*fsMHz/N,fftshift(abs(x1F))) ; % sub-carriers from [-128:127]
xlabel('frequency, MHz')
ylabel('amplitude')
title('frequency response of complex sinusoidal signal');

% generating the frequency domain signal with subcarrier indices [-N/2:-1 dc 1:N/2-1]
x2F = [zeros(1,N/2) 0 zeros(1,9) 1 zeros(1,N/2-10-1)]; % valid frequency on 10th subcarrier, rest
all zeros
x2T = N*ifft(fftshift(x2F)); % time domain signal using ifft()

% comparing the signals

diff = x2T – x1T;

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

err = diff*diff’/length(diff)

% impulse response of a sinc() filter

hT = sinc([-20:20]/3); % consider sample at 30MHz sampling;
hF = fft(hT,1024); % 128 pt FFT
figure;
plot([-512:511]*30/1024,fftshift(abs(hF))) ;
xlabel('frequency, MHz')
ylabel('amplitude')
title('frequency response of sinc filter');

Output:

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

5. Digital modulation and demodulation in AWGN and other channel

setting.
Aim: To plot the waveforms for the QPSK signal subjected to AWGN using MATLAB.

Software used: MATLAB

Theory: Quadrature Phase Shift Keying (QPSK) is a digital modulation technique. Quadrature
Phase Shift Keying (QPSK) is a form of Phase Shift Keying in which two bits are modulated at
once, selecting one of four possible carrier phase shifts (0, Π/2, Π, and 3Π/2). QPSK is
performed by changing the phase of the In-phase (I) carrier from 0° to 180° and the Quadrature-
phase (Q) carrier between 90° and 270°. Additive white Gaussian noise (AWGN) is a channel
model in which the only impairment to communication is a linear addition of wideband or white
noise with a constant spectral density (expressed as watts per hertz of bandwidth) and a Gaussian
distribution of amplitude.

Program:
Signal = randi([0 3],1000,1);
BER =[];
n = 20;
for i = 1:n
Errors = signal_awgn(Signal,i);
BER = [BER Errors/1000]; end
semilogy(BER);
function Errors = signal_awgn(Signal,SNR)
TxSignal = pskmod(Signal,4,pi/4);
RxSignal = awgn(TxSignal,SNR);
Signal2 = pskdemod(RxSignal,4,pi/4);
Errors = symerr(Signal,Signal2);
end

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

Output Graph:

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VNRVJIET
Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

6. Evaluate different performance aspects required for lossy and lossless

compression methods.

Aim: Evaluate different performance aspects required for Lossy and Lossless Image
Compression
Software used: MATLAB.
Theory: Digital files such as image files are often "compressed" to reduce their size and/or to
change various attributes, such as file type, dimensions, resolution, bit depth. Compression
reduces the size of a file, often without appreciable loss of information. It can be either lossless
or lossy.
Lossless compression restores and rebuilds file data in its original form after the file is
decompressed. The file can be decompressed to its original quality without any loss of data. This
compression method is also known as reversible compression.

In lossy compression, the data in a file is removed and not restored to its original form after
decompression. Specifically, data is permanently removed, which is why this method is also
known as irreversible compression.

Program:
clc
clear all
close all
rgb = imread('peppers.png');
imshow(rgb)
imwrite(rgb,'mypeppers.tif')
imwrite(rgb, 'test_image.study_013', 'tif')
bytes_in_memory_peppers = numel(rgb)
imwrite(rgb,'mypeppers.png')
info = imfinfo('mypeppers.png');
bytes_on_disk_peppers_png = info.FileSize
compression_ratio = bytes_in_memory_peppers / bytes_on_disk_peppers_png
peppers2 = imread('mypeppers.png');
isequal(rgb, peppers2)
url = 'https://blogs.mathworks.com/images/steve/2012/plot_screen_shot.png';
rgb_plot = imread(url);

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

bytes_in_memory_plot = numel(rgb_plot)
info = imfinfo(url);
bytes_on_disk_plot = info.FileSize
compression_ratio = bytes_in_memory_plot / bytes_on_disk_plot
imwrite(rgb,'peppers.jpg');
info = imfinfo('peppers.jpg');
bytes_on_disk_peppers_jpg = info.FileSize
compression_ratio_peppers_jpg = bytes_in_memory_peppers / ...
bytes_on_disk_peppers_jpg
imshow('peppers.jpg')
subplot(1,2,1)
imshow('peppers.png')
%limits = [232 276 215 248];
%axis(limits)
title('Original')
subplot(1,2,2)
imshow('peppers.jpg')
%axis(limits)
title('JPEG compressed')
imwrite(rgb_plot,'plot_screen_shot.jpg')
rgb_plot_compressed = imread('plot_screen_shot.jpg');
subplot(2,2,1)
imshow(rgb_plot)
title('Original image')
limits = [80 105 210 230];
axis(limits);
subplot(2,2,2)
imshow(rgb_plot_compressed)
title('JPEG compressed image')
axis(limits);

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

OUTPUT:
bytes_in_memory_peppers = 589824
bytes_on_disk_peppers_png = 287589
compression_ratio = 2.0509
ans = logical
bytes_in_memory_plot = 336000
bytes_on_disk_plot = 3284
compression_ratio = 102.3143
bytes_on_disk_peppers_jpg = 23509
compression_ratio_peppers_jpg = 25.0893

Original image jpeg compressed image

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

7. Perform Automatic gain and frequency control for an Audio signal.

Aim: To Perform Automatic gain and frequency control for an audio signal.
Software used: MATLAB.
Theory:
Automatic Gain Control: It is a closed loop feedback regulating circuit in an amplifier or chain of
amplifiers. It is a system that controls the increase in amplitude of an electrical signal from the
original input to the amplified output automatically.
Gain: Ratio of amplitude of output signal from an amplifier circuit to amplitude of input signal.
AGC circuits are basically designed for radio receiver circuits which receive highly varying
signal strength according to climatic conditions. They are used in audio amplifier circuits, audio
ICs, signal analysers etc. They apply high gain whenever the signals are week and as the signal
strength decreases, they automatically decrease their gain.
Program:
clear;
clc;
%% time instances
dt=0.001
t=0:dt:20;
%% input signal
A=10; f=0.25;
xin=A*sin(2*pi*f.*t);

%% AGC parameters
% loop filter gain
L=round(10*f*(1/dt));

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

% Target power
powerTarget=0.2;

%% AGC process
yout=zeros(1,length(xin));
gainAGC=ones(1,length(xin));
for n=1:length(t)

% amplify current input sample

yout(n)=gainAGC(n)*xin(n);
% adjust agc gain, with feedback branch
gainAGC(n+1)=gainAGC(n)*(1 - (1/L)*(yout(n)^2-powerTarget) );
end

%% Plot input and output

figure;
subplot(2,1,1);
plot(t, xin, '-b'); hold on;
plot(t, yout, '-r'); zoom on; grid on;
xlabel('t in sec'); ylabel('amplitude');
legend('xin', 'yout'); title('AGC input and output');
subplot(2,1,2);
plot(t, gainAGC(1:end-1)); zoom on; grid on;
title('gain of AGC');

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

Output Graph:

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

8. Audio compression algorithms (Using Sub band coding and Linear

predictive code)
Aim: Audio compression algorithms using sub band coding.

Software used: MATLAB

Theory:
In signal processing, sub-band coding (SBC) is any form of transform coding that breaks a signal
into several different frequency bands, typically by using a fast Fourier transform, and encodes
each one independently. This decomposition is often the first step in data compression for audio
and video signals.
The basic idea of SBC is to enable a data reduction by discarding information about frequencies
which are masked. The result differs from the original signal, but if the discarded information is
chosen carefully, the difference will not be noticeable, or more importantly, objectionable.
First, a digital filter bank divides the input signal spectrum into some number of subbands. The
psychoacoustic model looks at the energy in each of these subbands, as well as in the original
signal, and computes masking thresholds using psychoacoustic information. Each of the subband
samples is quantized and encoded to keep the quantization noise below the dynamically
computed masking threshold. The final step is to format all these quantized samples into groups
of data called frames, to facilitate eventual playback by a decoder.
Decoding is much easier than encoding since no psychoacoustic model is involved. The frames
are unpacked, subband samples are decoded, and a frequency-time mapping reconstructs an
output audio signal.

Program:

close all;
clear all;
num=36000;
[x,fs] = audioread('C:\Users\vnrvjiet\Downloads\eee.wav');
x=x(:,1)';

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

lnx=length(x);
L = 2;
len = 25;
wc = 1/L; %cut-off frequency is pi/2.
freq=-pi:2*pi/(lnx-1):pi;% the frequency vector
lp = fir1(len-1, wc,'low');
hp = fir1(len-1, wc,'high');
yl=conv(x,lp);
yh=conv(x,hp);

%Time domain plots of signal and filters

figure(1);
subplot(311);
plot(x);axis([0 lnx min(x) max(x)]);ylabel('speech');
title('Speech and filters in time domain');
subplot(312);
stem(lp);axis([0 length(lp) (min(lp)+0.1) (max(lp)+0.1)]);
ylabel('lp');
subplot(313);
stem(hp);axis([0 length(hp) min(hp)+0.1 max(hp)+0.1]);
ylabel('hp');
pause

%plotting filter response of filters and the two speech bands(lower and upper) in freq domian
figure(2);
X=fftshift(fft(x,lnx));
Lp=fftshift(fft(lp,lnx));
Hp=fftshift(fft(hp,lnx));
YL=fftshift(fft(yl,lnx));
Yh=fftshift(fft(yh,lnx));
subplot(321), plot(freq/pi, abs(X));ylabel('|X|');axis([0 pi/pi min(abs(X))
max(abs(X))]);title('Freq domain representation of speech and the two bands');
subplot(323), plot(freq/pi, abs(Lp),'g');ylabel('|Lp|');axis([0 pi/pi min(abs(Lp)) max(abs(Lp))]);
subplot(324), plot(freq/pi, abs(Hp), 'g');ylabel('|Hp|');axis([0 pi/pi min(abs(Hp)) max(abs(Hp))]);
subplot(325), plot(freq/pi, abs(YL), 'y');ylabel('|YL|');axis([0 pi/pi min(abs(YL))
max(abs(YL))]);legend('Low bandafter filtering');

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

subplot(326), plot(freq/pi, abs(Yh), 'y');ylabel('|Yh|');axis([0 pi/pi min(abs(Yh))

max(abs(Yh))]);legend('High band after filtering');
pause

ydl =yl(1:2:length(yl));
ydh=yh(1:2:length(yh));
s0=conv(ydl,lp);
s1=conv(ydl,hp);
s2=conv(ydh,lp);
s3=conv(ydh,hp);

% now finally decimating to get the four bands

b0 =s0(1:2:length(s0));
b1=s1(1:2:length(s1));
b2 =s2(1:2:length(s2));
b3=s3(1:2:length(s3));

%freq plots of decimated signals(four bands)

figure(3);
title('Four bands in freq domain');
subplot(411);
plot(freq/pi,abs(fftshift(fft(b0,lnx))));ylabel('|B0|');axis([0 pi/pi min(abs(fft(b0)))
max(abs(fft(b0)))]);title('Four bands in freq domain');
subplot(412);
plot(freq/pi,abs(fftshift(fft(b1,lnx))));ylabel('|B1|');axis([0 pi/pi min(abs(fft(b0)))
max(abs(fft(b1)))]);
subplot(413);
plot(freq/pi,abs(fftshift(fft(b2,lnx))));ylabel('|B2|');axis([0 pi/pi min(abs(fft(b2)))
max(abs(fft(b2)))]);
subplot(414);
plot(freq/pi,abs(fftshift(fft(b3,lnx))));ylabel('|B3|');axis([0 pi/pi min(abs(fft(b3)))
max(abs(fft(b3)))]);
pause;

% now synthesizing
L=2;

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

N1=length(b0);
Ss0=zeros(1,L*N1);
Ss1=zeros(1,L*N1);
Ss2=zeros(1,L*N1);
Ss3=zeros(1,L*N1);
Ss0(L:L:end)=b0;
Ss1(L:L:end)=b1;
Ss2(L:L:end)=b2;
Ss3(L:L:end)=b3;

%Passing through reconstruction filters

% making a low pass filter with cutoff at 1/L and gain L
reconst_fil=L*fir1(len-1,1/L);

% finding the freq response of the filter

sb0=conv(reconst_fil,Ss0);
sb1=conv(reconst_fil,Ss1);
sb2=conv(reconst_fil,Ss2);
sb3=conv(reconst_fil,Ss3);
Slow=sb0-sb1;
Shigh=sb2-sb3;
subl=zeros(1,length(Slow)*2);
subh=zeros(1,length(Shigh)*2);
subl(L:L:end)=Slow;
subh(L:L:end)=Shigh;
subll=conv(reconst_fil,subl);
subhh=conv(reconst_fil,subh);
sub=subll-subhh;

%Freq plots of final two bands and their merging into a single band
figure(4);
subplot(311);
plot(freq/pi,abs(fftshift(fft(subll,lnx))));ylabel('|low band|');axis([0 pi/pi min(abs(fft(subll)))
max(abs(fft(subll)))]);title('Final two bands in synthesis');
subplot(312);

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

plot(freq/pi,abs(fftshift(fft(subhh,lnx))));ylabel('|High band|');axis([0 pi/pi min(abs(fft(subhh)))

max(abs(fft(subhh)))]);
subplot(313);
plot(freq/pi,abs(fftshift(fft(sub,lnx))));ylabel('|Band|');axis([0 pi/pi min(abs(fft(sub)))
max(abs(fft(sub)))]);
pause;

%Comparison
figure(5);
subplot(211),
plot(freq/pi, abs(X));ylabel('|X|');axis([0 pi/pi min(abs(X)) max(abs(X))]);title('Comparison');
legend('original band');
subplot(212);
plot(freq/pi,abs(fftshift(fft(sub,lnx))),'r');ylabel('|Band|');axis([0 pi/pi min(abs(fft(sub)))
max(abs(fft(sub)))]);
legend('Synthesized Band');

OUTPUT:

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

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Name of the Laboratory: Name of the Experiment: _____
Signal Processing and Experiment No.________

Communication Applications Laboratory Date _______________

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