DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
DIGITAL SIGNAL PROCESSING LAB
FILE (LCECE325-G)
ECE Vth SEMESTER
SUBMITTED BY: ……………………….. SUBMITTED TO:………………………………….
ROLL NO:……………………………… ………………………………..
Department Of Electronics & Communication Engg
SAT KABIR INSTITUTE OF TECH & MGMT
LADRAWAN,BAHADURGARH
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
INDEX
S.NO NAME OF EXPERIMENT SIGNATURE
1. To present basic signals(unit step, unit impulse, ramp, exponent, sine and
cosine)
2. To develop a program for discrete convolution.
3. To develop a program for discrete correlation
4. To design Infinite Impulse Response (low pass filter) filter with cut of
frequency of 4000 HZ .
5. To Design Finite Impulse Response.(FIR filter)
6. To develop program for interpolation and decimation of sequences.
7. Implementation of FFT of given sequence
8. To find DFT / IDFT of given DT signal
9. To find the z-transform & inverse z-transform of a expression
10. To develop program for Circular Convolution
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
EXPERIMENT NO. 1
AIM:-To present basic signals(unit step, unit impulse, ramp, exponent, sine and cosine)
PROGRAM:-
** for unit step, unit impulse, ramp and exponent
clc;
clear all; close
all;
n=[0:0.5:20];
x=input('enter value of x');
y=exp(x*n); subplot(2,2,1);
stem(n,y);
xlabel ('x');
ylabel ('exponent');
title('expo1');
%clc;
%clear all;
%close all;
n=[0:0.5:20];
a=input('enter value of a');
y=(a).^n;
subplot(2,2,2);
stem(n,y);
xlabel('a');
ylabel('exponent');
title('expo2');
%clc;
%clear all;
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%close all;
n=[0:0.5:20];
subplot(2,2,3);
stem(n,n);
xlabel('x');
ylabel('y');
title('unit ramp');
%clc;
%clearall;
%close all;
t=[-3:1:2];
n1=[zeros(1,3),ones(1,1),zeros(1,2)];
subplot(2,2,4);
stem(t,n1);
xlabel('x');
ylabel('y'); title('unit
impulse');
**for sine and cosine
clc;
clear all;
close all;
t=[0:0.1:pi];
y=cos(2*pi*t);
subplot(2,2,1);
stem(t,y);
xlabel('t');
ylabel('coz');
title('cos(2pit)');
%clc;
%clear all;
%close all;
t=[0:0.1:pi];
y=sin(2*pi*t);
subplot(2,2,2);
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stem(t,y);
xlabel('t');
ylabel('sin');
title('sin(2pit)');
OUTPUT:-
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EXPERIMENT NO. 2
AIM:-To develop a program for discrete convolution.
PROGRAM:-
clc, clear
all;
closeall;
x=[1 2 35];
h=[2 3 56];
y=conv(x,h);
subplot(3,1,1);
stem(x); xlabel('x');
ylabel('amplitude');
title('x sequence');
subplot(3,1,2);
stem(h); xlabel('h');
ylabel('amplitude');
title('h sequence');
subplot(3,1,3);
stem(y); xlabel('y');
ylabel('amplitude');
title('ysequence');
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OUTPUT:-
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
EXPERIMENT NO. 3
AIM:-To develop a program for discrete correlation
PROGRAM:-
clc, clear
all;
closeall;
x=[1 2 35];
h=[2 3 56];
y=xcorr(x,h);
subplot(3,1,1);
stem(x); xlabel('x');
ylabel('amplitude');
title('x sequence');
subplot(3,1,2);
stem(h); xlabel('h');
ylabel('amplitude');
title('h sequence');
subplot(3,1,3);
stem(y); xlabel('y');
ylabel('amplitude');
title('ysequence');
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
OUTPUT:-
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
EXPERIMENT NO. 4
AIM :To design Infinite Impulse Response (low pass filter) filter with cut of frequency of 4000hz .
PROGRAM :
#include"dsk6713.h" //this file is added to initialize the DSK6713
#include"dsk6713_aic23.h"
Uint32 fs = DSK6713_AIC23_FREQ_8KHZ; // set the sampling frequency, Different sampling frequencies
supported by AIC23 codec are 8, 16, 24, 32,
44.1, 48, and
96 kHz.
// FILTER COFFICIENTS IS CALCULATED BY MATLAB
float fc []={
2.338110787e-019,6.936318823e-005,-0.0003181171778,0.0008399875951,
- 0.001779771759,0.003340889933,-0.005792469252,0.00948059652,-
0.01485389285,
0.02252536267,-0.03342207149, 0.04916161299, -0.07310581207,
0.1139752045,
- 0.2039434612, 0.6338219047, 0.6338219047, -0.2039434612,0.1139752045,
- 0.07310581207, 0.04916161299, -0.03342207149, 0.02252536267, -
0.01485389285,
0.00948059652, -0.005792469252, 0.003340889933,-
0.001779771759,0.0008399875951,
- 0.0003181171778, 6.936318823e-005,2.338110787e-019
};
static short in_buffer[18] ;
void main( )
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{
comm_intr(); // ISR function is called, using the givencommand
while(1); // program execution halts and it starts listening for the
interrupt which occur at everysampling
period Ts.
}
interrupt voidc_int11() // ISR call, At each Interrupt, program execution goes to
the interrupt service routine
{
Uint32indata; //variable declaration
inti=0;
signedint output=0;
indata= input_sample(); //newest input @ top ofbuffer
in_buffer[0]=indata; //new input at buffer[0]
for(i=17;i>=0;i--)
in_buffer[i] =in_buffer[i-1]; //shuffle thebuffer
for(i=0;i<18;i++)
output = output + fc[i] * in_buffer[i];
output_sample(output); //output filter,the value in the buffer yn indexedby
the variable loop is written on to the codec.
}
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
OUTPUT :
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
EXPERIMENT NO. 5
AIM :To Design Finite Impulse Response.
FIR filter:
lowpass 1500 Hz, High pass,2200hz, Bandpass 1750 Hz, Band stops 790 Hz.
PROGRAM :
#include"DSK6713_AIC23.h" //this file is added to initialize the DSK6713
#include"lowp1500.cof" // coefficient of low-pass filter file calculatedfrom
MATLAB
#include"highp2200.cof" // coefficient of high-pass filter file calculated from
MATLAB
#include"bpass1750.cof" // coefficient of band-pass filter file calculatedfrom
MATLAB
#include"bstop790.cof" // coefficient of band-stop filter file calculated from
MATLAB
Uint32fs=DSK6713_AIC23_FREQ_8KHZ; // set the sampling frequency, Differentsampling
Frequencies supported by AIC23 codec are 8, 16,
24, 32, 44.1, 48, and 96 kHz.
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shortFIR_number=0; //filter number
intyn = 0; //variable declaration
shortdly[N]; //declare delay buffer of n values
shorth[4][N]; //co-efficients of 4 different filters
interruptvoidc_int11() // ISR call, At each Interrupt, program execution goes to
the interrupt serviceroutine
shorti; //variable declaration
dly[0]= input_sample(); //newest input @ top of buffer
yn = 0; //initialize filteroutput
for (i = 0; i<N;i++) //for loop takes in the value of i from 0 to N yn
+=(h[FIR_number][i]*dly[i]); //y(n) +=h(LP#,i)*x(n-i)
for (i = N-1; i>0;i--) //starting @ bottom ofbuffer
dly[i] = dly[i-1]; //update delays with datamove
output_sample(yn>>15); //output filter,the value in the buffer yn indexedby
the variable loop is written on to the codec.
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return; // program execution goes back to while(1) and then
again starts listening for next interrupt and this
process goes on
}
void main()
shorti; //variabledeclaration
for (i=0;i<N;i++) //for loop which takes in the value of i from 0 toN=4
and switches to corresponding filter co-efficients
dly[i] = 0; //init buffer
h[0][i] = hlp[i]; //start addr of lp1500 coeff
h[1][i] = hhp[i]; //start addr of hp2200 coeff
h[2][i] = hbp[i]; //start addr of bp1750 coeff
h[3][i] = hbs[i]; //start addr of bs790 coeff
comm_intr(); // ISR function is called, using thegiven
command
while(1); //program execution halts and it starts listening for
the interrupt which occur at every sampling
period Ts.
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
OUTPUT :
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DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
EXPERIMENT NO. 6
AIM :program to verify the decimation and interpolation of given sequence.
PROGRAM for Decimation:
PROGRAM:
Clc;
Clear all;
Close all;
D=input('enter the downsampling factor');
L=input('enter the length of the input signal');
f1=input('enter the frequency of first sinusodal');
f2=input('enter the frequency of second sinusodal');
n=0:L-1;
x=sin(2*pi*f1*n)+sin(2*pi*f2*n);
y=decimate(x,D,'fir');
subplot(2,1,1);
stem(n,x(1:L));
title('input sequence');
xlabel('time(n)');
ylabel('amplitude');
subplot(2,1,2)
m=0:(L/D)-1;
stem(m,y(1:L/D));
title('Decimated sequence');
xlabel('time(n)');
ylabel('amplitude'DIGITAL SIGNAL PROCESSING LAB (LCECE325-G)
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INPUT :
enter the downsampling factor = 5
enter the length of the input signal = 100
enter the frequency of first sinusoidal = 0.01
enter the frequency of second sinusoidal = 0.0
OUTPUT :
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PROGRAM for Interpolation:
Clc;
Clear all;
Close all;
L=input('enter the upsampling factor');
N=input('enter the length of the input signal'); % Length should be greater than 8
f1=input('enter the frequency of first sinusodal');
f2=input('enter the frequency of second sinusodal');
n=0:N-1;
x=sin(2*pi*f1*n)+sin(2*pi*f2*n);
y=interp(x,L);
subplot(2,1,1)
stem(n,x(1:N))
title('input sequence');
xlabel('time(n)');
ylabel('amplitude');
subplot(2,1,2)
m=0:N*L-1;
stem(m,y(1:N*L))
title('output sequence ');
xlabel('time(n)');
ylabel('amplitude');
INPUT:
enter the upsampling factor = 5
enter the length of the input signal = 9
enter the frequency of first sinusoidal = 0.1
enter the frequency of second sinusoidal = 0.3
Output Waveforms:
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EXPERIMENT NO. 7
AIM :Implementation of FFT of a given sequence.
PROGRAM:
%MATLAB program to plot the frequency response (magnitude and phase response)of
agiven difference equation.
clc;
clear all;
b=input('Enter the numerator coefficients:');
a=input('Enter the denominator coefficients:');
[h,w]=freqz(b,a);
subplot(2,1,1);
plot(w/pi,abs(h));
grid;
xlabel('Normalised Frequency');
ylabel('Magnitude in dB');
title('Magnitude Response');
subplot(2,1,2);
plot(w/pi,angle(h));
grid;
xlabel('Normalised Frequency');
ylabel('phase in radians');
title('Phase Response');
Output:
Enter the numerator coefficients: [1]
Enter the denominator coefficients: [1 -1/6 -1/6]
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Output Waveforms:
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EXPERIMENT NO. 8
AIM :To find DFT / IDFT of given DT signal.
PROGRAM:
clc;
close all;
clear all;
xn=input('Enter the sequence x(n)'); %Get the sequence from user
ln=length(xn); %find the length of the sequence
xk=zeros(1,ln); %initialize an array of same size as that of input sequence
ixk=zeros(1,ln); %initialize an array of same size as that of input sequence
%DFT of the sequence
%-----------------------------------------------------
for k=0:ln-1
for n=0:ln-1
xk(k+1)=xk(k+1)+(xn(n+1)*exp((i)*2*pi*k*n/ln));
end
end
%------------------------------------------------------
%Plotting input sequence
%-----------------------------------------------------
t=0:ln-1;
subplot(221);
stem(t,xn);
ylabel ('Amplitude');
xlabel ('Time Index');
title('Input Sequence');
%---------------------------------------------------------------
magnitude=abs(xk); % Find the magnitudes of individual DFT points
% plot the magnitude response
%------------------------------------------------------------
t=0:ln-1;
subplot(222);
stem(t,magnitude);
ylabel ('Amplitude');
xlabel ('K');
title('Magnitude Response');
%------------------------------------------------------------
phase=angle(xk); % Find the phases of individual DFT points % plot the magnitude
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sequence
%------------------------------------------------------------
t=0:ln-1;
subplot(223);
stem(t,phase);
ylabel ('Phase');
xlabel ('K');
title ('Phase Response');
%------------------------------------------------------------
%IDFT of the sequence
%------------------------------------------------------------
for n=0:ln-1
for k=0:ln-1
ixk(n+1)=ixk(n+1)+(xk(k+1)*exp(i*2*pi*k*n/ln));
end
end
ixk=ixk./ln;
%------------------------------------------------------------
%code block to plot the input sequence
%------------------------------------------------------------
t=0:ln-1;
subplot(224);
stem(t,ixk);
ylabel ('Amplitude');
xlabel ('Time Index');
title ('IDFT sequence');
%------------------------------------------------------
Output:
Xn=[1 2 3 4 5]
Xk = 15,-2.50+3.44i,-2.50+0.81i,-2.49-0.81i,-2.49-3.44i
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Output Waveforms:
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EXPERIMENT NO. 9
AIM :TO FIND THE Z-TRANSFORM & INVERSE Z-TRANSFORM OF A
EXPRESSION
Program:
%X(n) = [1/16n ]u(n)
syms z n
a=ztrans(1/16^n)
% X(Z) = 3*Z / (Z+1)
syms Z n
y=iztrans(3*Z/(Z+1))
Result:
a = 16*z/(16*z-1)
y= 3*(-1)^n
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EXPERIMENT NO. 10
AIM: To develop program for Circular Convolution
ALGORITHM:
Step 1: Start
Step 2: Read the first sequence
Step 3: Read the second sequence
Step 4: Find the length of the first sequence
Step 5: Find the length of the second sequence
Step 6: Perform circular convolution MatLab for both the sequences using inbuilt function
Step 7: Plot the axis graph for sequence
Step 8: Display the output sequence
Step 9: Stop
clc;
close all;
clear all;
x1=input('Enter the first sequence :
');
x2=input('Enter the second sequence
: ');
N1=length(x1);
N2=length(x2);
N=max(N1,N2);
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if(N2>N1)
x4=[x1,zeros(1,N-N1)];
x5=x2;
elseif(N2==N1)
x4=x1;
x5=x2;
else
x4=x1;
x5=[x2,zeros(1,N-N2)];
end
x3=zeros(1,N);
for m=0:N-1
x3(m+1)=0;
for n=0:N-1
j=mod(m-n,N);
x3(m+1)=x3(m+1)+x4(n+1).*x5(j+1);
end
end
subplot(4,1,1)
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stem(x1);
title('First Input Sequence');
xlabel('Samples');
ylabel('Amplitude');
subplot(4,1,2)
stem(x2);
title('Second Input Sequence');
xlabel('Samples');
ylabel('Amplitude');
subplot(4,1,3)
stem(x3);
title('Circular Convolution Using
Modulo Operator');
xlabel('Samples');
ylabel('Amplitude');
%In built function
y=cconv(x1,x2,N);
subplot(4,1,4)
stem(y);
title('Circular Convolution using
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Inbuilt Function');
xlabel('Samples');
ylabel('Amplitude');
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