Generation OF Signals: Digital Signal Processing Lab

Digital Signal processing Lab
Ex.No:
Date :
GENERATION OF SIGNALS
AIM:
To write a MATLAB for the following
(i) Plot the unit impulse function.
(ii) Plot the unit step function.
(iii)Plot the unit ramp and exponential function.
(iv)Plot the sine, cosine, saw tooth and square wave function.
ESSENTIAL REQUIRED:
Hardware: PC
Software: MATLAB 7.4
ALGORITHM:
IMPLUSE FUNCTION:
Step1: Get the time interval.
Step2: Give the respect impulse function using zeros and ones function.
Step3: Subplot and stem the sequence.
Step4: Give x label, y label and title.
Step5: Stop the program and execute it.
Department of ECE Page 1

STEP FUNCTION:
Step1: Give the number of sequences.
Step2: Get the time interval.
Step3: Give the matrix using one’s function.
Step4: Subplot and stem the sequence.
Step5: Give the x label, y label and title.
Step6: Stop the program.
RAMP FUNCTION:
Step1: Give the length of the sequence.
Step2: Give the time interval.
Step3: Subplot and stem the sequence
SINUSOIDAL FUNCTION:
Step1: Enter the length of the sinusoidal sequence.
Step2: Enter the number of cycle.
Step3: Get the interval and obtain sinusoidal waveform using sine function.
Step4: Subplot and stem the waveform.

COSINE FUNCTION
Step1: Enter the length of cosine sequence.
Step2: Enter the number of cycles.
Step3: Get the interval and obtain cosine function.
Step4: Subplot and stem the waveform.
EXPONENTIAL FUNCTION:
Step1: Start the exponential function.
Step2: Plot the exponential sequence in discrete domain.
SAWTOOTH FUNCTION:
Step1: Set the time domain from 0 to 5.
Step2: Generate the saw tooth wave and store it.
Step3: Subplot and stem the waveform for the given time domain.

SQUARE FUNCTION:
Step1: Set the time domain from 0 to 5.
Step2: Generate the square wave and store it.
Step3: subplot and stem the waveform for the given time domain.
Step4: Given x label, y label and title.
FLOWCHART:
IMPLUSE FUNCTION:
START
Get the time interval
Compute the impulse function using zero

and ones
Plot output sequence and give

title, x label and y label
STOP

STEP FUNCTION:
START
Get the number of sequences and

the time interval
Compute the step function using ones function
Plot the output sequence and give

STOP
RAMP FUNCTION:
START
Get the length of the sequence

and time interval
Compute the ramp function using zero and ones

STOP

SINUSOIDAL FUNCTION:
START
Enter the length of the sinusoidal

sequence and number of cycle
Get the interval and obtain sine waveform using

sine function

x label, y label and title
STOP
COSINE FUNCTION
START
Enter the length of sinusoidal

sequences and number of cycle
Get the time interval and obtain cosine using cosine

function
Plot the output sequence and

give x label, y label and title
STOP

EXPONENTIAL FUNCTION:
START
Set the exponential
Plot the output sequence x label,y label and tittle
STOP
SAWTOOTH FUNCTION:
START
Set the time domain from 0 to

5
Generate the saw tooth wave and store it
Plot the output sequence x

label,y label and tittle
STOP

‘SQUARE FUNCTION:
START
Set the time domain form

0 to 5
Generate the sequence wave form and store it
Plot the output sequence x label,

y label and title
STOP
PROGRAM:
% TO PLOT THE UNIT IMPULSE FUNCTION
s1= [zeros(1,5),1,zeros(1,5)];
n1=-5:5;
subplot (2,2,1);
stem (n1,s1);
title ('unit impulse function');
x label ('time');
y label('amplitude');

% TO PLOT THE UNIT STEP FUNCTION
s2=[zeros(1,5),ones(1,6)];
n2=-5:5;
subplot(2,2,2);
stem(n2,s2);
title('unit step function');
xlabel('time');
ylabel('amplitude');
%TO PLOT THE UNIT RAMP FUNCTION
n3=input('enter the limit= ');
n4=0:n3;
subplot(2,2,3);
stem(n4,n4);
title('unit ramp function');
xlabel('time');
%TO PLOT THE EXPONENTIAL FUNCTION
n5=0:0.1:1;
subplot(2,2,4);
stem(n5,exp(n5));
title(' exponential function ');
xlabel('time');

%TO PLOT THE SAWTOOTH WAVEFORM
t1=0:0.1:1;
y1=sawtooth(2*pi*t1);
subplot(1,2,1);
stem(t1,y1);
title('sawtooth function ');
xlabel('time ');
% TO PLOT THE SQUARE WAVEFORM
t2=0:0.1:1;
y2=square(2*pi*t2);
subplot(1,2,2);
stem(t2,y2);
title('square function ');
xlabel('time');
%TO PLOT THE COSINE WAVEFORM
clear all;
close all;
vm=input('enter the amplitude');
f=input('enter frequency');

t1=0:0.1:2;
y=vm*cos(2*pi*f*t1);
stem(t1,y);
title('cos wave');
xlabel('time');
%TO PLOT THE SINE WAVEFORM
clc;
clear all;
close all;
vm=input('enter the amplitude');
f=input('enter frequency');
t1=0:0.1:2;
y=vm*sin(2*pi*f*t1);
stem(t1,y);
title('sine wave');
xlabel('time');

OUTPUT:
enter the limit= 3
unit impulse function unit step function

1 1
amplitude
amplitude
0.5 0.5
0 0
-5 0 5 -5 0 5
time time
unit ramp function exponential function
3 3
2 2
amplitude
amplitude
1 1
0 0
0 1 2 3 0 0.5 1
time time
sawtooth function square function

0.8 1
0.6 0.8
0.6
0.4
0.4
0.2
0.2
0
amplitude
amplitude
0
-0.2
-0.2
-0.4
-0.4
-0.6
-0.6
-0.8 -0.8
-1 -1
0 0.2 0.4 0.6 0.8 1 0 0.2 0.4 0.6 0.8 1
time time

Enter the amplitude: 3
Enter frequency: 1
cos wave
3
1
amplitude
-1
-2
-3
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
time
Enter the amplitude: 3
Enter frequency: 1
sine wave
3
1
amplitude
-1
-2
-3
0 0.2 0.4 0.6 0.8 1 1.2 1.4 1.6 1.8 2
time
RESULT:
Thus the MATLAB program to generate the various sequences and waves were executed
and the results are obtained.

Ex.No:
Date :
LINEAR CONVOLUTION AND CIRCULAR CONVOLUTION
AIM:
To write a MATLAB program for
1. Computing linear convolution using FFT.
2. Computing circular convolution using FFT.
ESSENTIALS REQUIRED:
HARDWARE: Zenith PC or compatible
SOFTWARE: MATLAB version 7.4.
ALGORITHM:
LINEAR CONVOLUTION USING FFT:
Step1: start the program
Step2: first enter the sequence x (n).
Step3: place the input sequence x (n).
Step4: compute the FFT for the input sequence x (n).
Step5: enter the impulse response h (n).
Step6: plot the impulse response h (n).
Step7: compute the impulse response h (n).

Step8: multiply x (n) and h (n).
Step9: compute IFFT for y (n).
Step10: display the convolued sequence.
CIRCULAR CONVOLUTION USING FFT:
Step1: First enter the input sequence x(n).
Step2: Plot the input sequence x(n).
Step3: Compute the FFT for the input sequence x(n).
Step4: Enter the impulse response h(n).
Step5: Plot the impulse response h(n).
Step6: Compute FFT for the impulse response h(n).
Step7: Multiply x(n) and h(n).
Step8: Compute IFFT for y(n).
Step9: Display the convolued sequence.

FLOWCHART:
LINEAR CONVOLUTION USING FFT:
START
Enter the sequence x(n)
Plot the input sequence x(n)
Compute the FFT for x(n)
Enter the impulse response h(n)
Plot the impulse response h(n)
Compute FFT for h(n)
Y1=x1*h1
Compute IFFT for y1
Plot the convolued sequence y(n)
STOP

CIRCULAR CONVOLUTION USING FFT:
START
Plot the input sequence x(n)
Compute FFT for x(n)
Enter the impulse response h(n)
Plot the impulse response h(n)
Compute FFT for h(n)
Y1=x1*h1
Compute IFFT for y1
Plot the convolued sequence y(n)
STOP

% LINEAR CONVOLUTION USING FFT
clc;
x=input('enter the sequence x(n)');
len1=length(x);
n1=0:1:len1-1;
subplot(3,1,1);
stem(n1,x);
xlabel('time period');
title('input sequence x(n)');
h=input('enter the sequence h(n)');
len2=length(h);
n2=0:1:len2-1;
subplot(3,1,2);
stem(n2,h);
n3=0:1:len-1;
x1=fft(x,len);
h1=fft(h,len);
y=x1.*h1;
title('impulse response h(n)');
len=len1+len2-1;
y=ifft(y,len);

subplot(3,1,3);
disp('convoluted sequence');
disp(y);
stem(n3,y);
OUTPUT:
enter the sequence x(n):[1,-1,-1,-1,1,1,1,-1]
enter the sequence h(n):[4,3,2,1,0,0,0,0]
convolued sequence
Columns 1 through 9
4.0000 -1.0000 -5.0000 -8.0000 -2.0000 4.0000 8.0000 2.0000 0
Columns 10 through 15
-1.0000 -1.0000 0 0.0000 -0.0000 -0.0000

input sequence x(n)

1
amplitude
0
-1
1 2 3 4 5 6 7 8
time index
impulse response h(n)
4
amplitude
0
1 2 3 4 5 6 7 8
time index
convolued sequence y(n)
10
amplitude
-10
0 2 4 6 8 10 12 14
time index
%CIRCULAR CONVOLUTION:
clc;
xn=input('enter the input sequence');
len1=length(xn);
n1=0:1:len1-1;
subplot(3,1,1);
stem(xn);
hn=input('enter the impulse response');
len2=length(hn);
subplot(3,1,2);
stem(hn);
len=len1+len2-1;

N=len;
x=[xn zeros(1,N-len1)];
n1=length(x);
h=[hn zeros(1,N-len2)];
n2=length(h);
x1=fft(x,len);
h1=fft(h1,len);
y1=x.*h1;
y=ifft(y,len);
subplot(3,1,1);
disp('convoluted sequence');
disp(y);
n=0:1:len-1;
subplot(3,2,1);
stem(n,y);
title('convoluted sequence');

Output:
enter the input sequence[1 1 1 -1]
enter the impulse response[-1 -1 1 0]
convoluted sequence
Columns 1 through 5
-0.2857 -0.6435 - 0.2854i 0.4453 - 0.1669i -0.1589 - 0.2632i -0.1589 + 0.2632i
Columns 6 through 7
0.4453 + 0.1669i -0.6435 + 0.2854i
convoluted sequence
0.5
amplitude
-0.5
-1
0 2 4 6
time period
1
-1
1 1.5 2 2.5 3 3.5 4
convolued sequence y(n)
2
amplitude
-2
0 1 2 3 4 5 6
time index
RESULT:
Thus the MATLAB program to perform (i) linear convolution using FFT
(ii) circular convolution using FFT has been written and the output sequence is obtained.

Ex.No:
Date :
SAMPLING AND EFFECT OF ALIASING
AIM:
To write a MATLAB code to explain sampling a continuous time signal also obtains a graph
to obtain aliasing in time domain and frequency domain.
ESSENTIAL REQUIRED:
Hardware: Zenith Pc or Compatible.
Software: MATLAB version 5.1
ALGORITHM:
Step 1: Initialize the continuous time variable to the range between 0 and 1.
Step 2: Obtain the frequency of ct signal as input.
Step 3: Compute the ct signals.
Step 4: Display and obtain the sampling frequency > the nyquist rate of ct signal.
Step 5: Compute the sampling period and allow the time variable of the sampled signal to range
between 0 and 1 in the range of sampling period.
Step 6: Compute the sampled signal and display it if fc < nyquist rate then go to step 8.
Step 7: Obtain the sampling and frequency less than nyquist rate as input and repeat 5&6.
Step 8: Compute the dft of sampled signal for both cases and display it.

FLOW CHART:
START
INITIALISE CT VARIABLE
0:0.001:1
Display
x=cos(2*pi*f*t)
Input fs>=2fm
ts=1/(fs)
x(n)=cos(2*pi*f*t)
Input fs<2fm
tsln=1/(fsln)
x(n)=cos(2*pi*f*tln)
display
DFT signal and xln area display
STOP

%PROGRAM FOR SAMPLING IN TIME DOMAIN AND ALIASING IN FREQUENCY

DOMAIN
t=0:0.001:1;
f=input('Enter the frequency in cycles per sec');
x=cos(2*pi*f*t);
subplot(3,1,1);
plot(t,x);
title('continuous time signal');
xlabel('time index t');
ylabel('signal x(t)');
%SAMPLING IN TIME DOMAIN AT NYQUIST RATE
fs=input('Enter the nyquist rate fs');
ts=1/fs;
t=0:ts:1;
xn=cos(2*pi*f*t);
subplot(3,1,2);
stem(t,xn);
title('Analog signal at nyquist rate');
ylabel('discrete time signal x(n)');

%SAMPLING IN TIME DOMAIN LESS THAN NYQUIST RATE
fsln=input('Enter the sampling frequency less than nyquist');
tsln=1/fsln;
tln=0:tsln:1;
xln=cos(2*pi*f*tln);
subplot(3,1,3);
stem(tln,xln);
title('Analog signal samples less than nyquist rate');
figure;
%ALIASING IN FREQUENCY DOMAIN AT NYQUIST RATE
xnf=fft(xn);
subplot(2,2,1);
plot(abs(xnf));
title('frequency domain representation of signal aliased at nyquist');
xlabel('index n');

%ALIASING IS FREQUENCY DOMAIN LESS THAN NYQUIST RATE
xlnf=fft(xln);
subplot(2,1,2);
plot(abs(xlnf));
title('frequency domain of signal aliased at a frequency less than nyquist rate');
xlabel('index n');
OUTPUT:
Enter the frequency in cycles per sec 5
continuous time signal

1
signal x(t)
0
continuous time signal
1
signal x(t)
-1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
0 time index t
time index t
discrete time signal x(n)
Analog signal at nyquist rate

1
Enter the nyquist rate fs 50
-1
0 0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
time index t
-1
0 0.1 0.2 0.3 0.4 0.5
Analog signal0.6at nyquist
time index t
0.7 0.8
rate 0.9 1
1 Analog signal samples less than nyquist rate

1
amplitude
00
-1
-1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
0 0.1 0.2 0.3 time0.4
index t 0.5 0.6 0.7 0.8 0.9 1
time index t
Analog signal samples less than nyquist rate
1
amplitude

0
-1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
time index t

Analog signal at nyquist rate
1
0
Enter the sampling

-1 frequency less than nyquist 20
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
time index t
Analog signal samples less than nyquist rate
1
amplitude
-1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
time index t
frequency domain representation of signal aliased at nyquist

30
20
amplitude
10
0
0 20 40 60
index n
frequency domain of signal aliased at a frequency less than nyquist rate
10
8
amplitude
0
0 5 10 15 20 25
index n
RESULT:
Thus the MATLAB program to explain the concept of sampling a continuous time signal
was written and graph to explain aliasing in time domain and in frequency domain was written.

Ex.No :
Date :
DESIGN OF FIR FILTERS
AIM:
To design and plot the FIR filter using hamming, hanning and Blackman window
techniques.
ESSENTIAL REQUIRED:
Hardware:PC or compatible
Software:MATLAB version 7.4
ALGORITHM:
HAMMING WINDOW:
Step 1: For the given length N, calculate alpha= (N-1)/2 and assign eps=0.001 and
wc=0.5*pi.
Step2: Calculate delayed sample response (hd) using sine function.
Step3: Calculate the rectangular window coefficients as resonant frequency (wr).
Step4: Calculate unit sample response (hn) as delayed sample tresponse(hd) multiplied by resonant
frequency wr.
Step5: Calculate the frequency of unit sample response and plot it.
Step6: Calculate the hamming window coefficients (wn) using hamming function.
Step7: Calculate hn=hd*wn.
Step8: Calculate the frequency and plot it.
HANNING WINDOW

Step1: For the given length N. calculate alpha= (N-1)/2 and assign eps=0.001 and cut off
frequency wc=0.3*pi.
Step2: Calculate delayed sample response (hd) using sine function.
Step3: Claculate the rectangular window coefficient (wr).
Step4: Calculate the sample response h (n) as hd.*wr’.
Step5: Calculate the frequency of h (n) and plot.
Step6: Calculate the hanning window coefficient (wh) using hanning functions.
Step7: Calculate hn=hd.*wh’.
Step8: Calculate the frequency and plot it.
BLACKMANN WINDOW:
Step1:For the given length N,calculatealpha=(N-1)/2 and assign eps=0.001 and
wc=0.7*pi.
Step2:Calculate delayed sample response using sine function.
Step3:Calculate the rectangular window coefficient as wr.
Step4:Calculate unit sample response h(n) as hd.*wr’.
Step5:Calculate the frequency of h(n) and plot.
Step6:Calculate the blackmann window coefficient (wn) using blackmann function.
Step7:Calculate hn=hd.*wp’.
Step8:Calcul;ate the frequency the frequency and plot it.

FLOWCHART:
HAMMING WINDOW:
START
Get the sequence length
Calcutate α=N-1/2 and wc=0.5*pi
Calculate the filter coefficient as hd

rect. window coefficient as wr
Calculate the FIR filter coefficient hn =hd*wr’
Find the frequency domain response of h(n)
Plot the frequency domain response
Compute hamming window coefficient wh and corresponding filter

coefficient
Calculate frequency domain representation of
h(n)
H=freq(h,1,w) and plot it

Plot the frequency domain with label
STOP

HANNING WINDOW:
START

rect. window coefficient as wr
Compute hanning window coefficient wh and corresponding filter

coefficient
h(n)

STOP

BLACKMANN WINDOW:
START

rect. window coefficient as wn
Compute blackmann window coefficient wh and corresponding filter

coefficient
h(n)

STOP

PROGRAM:
%TO DESIGN BLACKMAN WINDOW
clear all;
wc=0.7*pi;
N=9;
alpha=(N-1)/2;
eps=0.001;
n=0:1:N-1;
hd=sin(wc*(n-alpha+eps))./(pi*(n-alpha+eps));
wr=boxcar(N);
hn=hd.*wr';
w=0:0.1:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h));
hold on;
wh= blackman(N);
hn=hd.*wh';
w=0:0.1:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h),'-*');
grid;
xlabel('normalised frequency');
title('blackman window');
hold off;

OUTPUT:
%TO DESIGN HAMMING WINDOW
clear all;
wc=0.5*pi;
N=9;
alpha=(N-1)/2;
eps=0.0001;
n=0:1:N-1;
wr=boxcar(N);
hn=hd.*wr';
w=0:0.1:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h));
hold on;
wh= hamming(N);

hn=hd.*wh';
w=0:0.1:pi;
h=freqz(hn,1,w);
grid;
ylabel('magnitude');
title('hamming window');
hold off;
OUTPUT:

%TO DESIGN HANNING WINDOW
clear all;
wc=0.3*pi;
N=9;
alpha=(N-1)/2;
eps=0.001;
n=0:1:N-1;
wr=boxcar(N);
hn=hd.*wr';
w=0:0.1:pi;
h=freqz(hn,1,w);
plot(w/pi,abs(h));
hold on;
wh= hanning(N);
hn=hd.*wh';
w=0:0.1:pi;
h=freqz(hn,1,w);
grid;
ylabel('magnitude');
title('hanning window');
hold off;

OUTPUT:
RESULT:
Thus the FIR filter is designed and plotted using hamming, hanning and blackmann window
techniques.

Ex.No :
Date :
DESIGN OF IIR FILTERS USING BUTTERWORTH TECHNIQUE
AIM:
To write a MATlab program to design a Butterworth low pass, high pass, band pass and band
reject filter.
Hardware: pc or compatible
Software: MATlab version 7.4
ALGORITHM:
Butterworth low pass filter:
Step1: Start the program.
Step2: Assign alphap=4, alphas=30, fp=400, fs=800, f=2000.
Step3: Perform omp=2*fp/f & oms=2*fs/f.
Step4: Calculate cutoff frequency and order of the filter using buttord function.
Step5: Butter function is used to calculate system function.
Step6: Assign w=0:0.01: pi.
Step7: Calculate z transform using freqz function.
Step8: Perform m=abs (h) & an=angle (h).

Butterworth high pass filter:
Step2: Assign alphap=0.4, alphas=30, fp=400, fs=800, f=2000.
Step3: Perform omp=2*fp/f & oms=2*fs/f.
Step8: Perform m=20*log(abs (h)) & an=angle (h).
Butterworth band pass filter:
Step2: Assign alphap=2, alphas=20,wp=[0.2*pi,0.4*pi],ws=[0.1*pi,0.5*pi].

Step8: Plot phase in radians in y axis and normalized frequency in x axis.
Step9:Stop the program.

Butterworth bandstop filter:
Step2: Assign alphap=2, alphas=20,wp=[0.2*pi,0.4*pi],ws=[0.1*pi,0.5*pi].

Step8: Plot phase in radians in y axis and normalized frequency in x axis.
Step9:Stop the program.

FLOWCHART:
Butterworth low pass filter:
Start
Assign alphap=4, alphas=30,

fp=400, fs=800, f=2000.
Calculate omp=2*fp/f & oms=2*fs/f.
Calculate cutoff frequency and order of the filter using

buttord function
Assign w=0:0.01:pi
Calculate z transform using freqz function.
And Perform m=abs (h) & an=angle (h).
Display butterworth lowpass

filter response
Stop

%TO DESIGN A BUTTERWORTH LOWPASS FILTER
clear all;
alphap=4;
alphas=30;
fp=400;
fs=800;
f=2000;
omp=2*fp/f;
oms=2*fs/f;
%to find the cutoff frequency and order of the filter
[n,wn]=buttord(omp,oms,alphap,alphas)
%system function of the filter
[b,a]=butter(n,wn);
w=0:0.01:pi;
[h,om]=freqz(b,a,w,'whole');
m=abs(h);
an=angle(h);
subplot(2,1,1);
plot(om/pi,20*log(m));
grid;
ylabel('gain in dB');
subplot(2,1,2);
plot(om/pi,an);
grid;
ylabel('phase in radians');

OUTPUT:

Butterworth high pass filter:
Start
Assign alphap=0.4, alphas=20,

fp=400, fs=800, f=2000.
Calculate omp=2*fp/f & oms=2*fs/f.

buttord function
Assign w=0:0.01: pi
And Perform m=20*log(abs (h)) & an=angle (h).
Display butterworth highpass

filter response
Stop

%TO DESIGN A BUTTERWORTH HIGHPASS FILTER
clear all;
alphap=0.4;
alphas=30;
fp=400;
fs=800;
f=2000;
omp=2*fp/f;
oms=2*fs/f;
[n,wn]=buttord(omp,oms,alphap,alphas)
[b,a]=butter(n,wn,'high');
w=0:0.01:pi;
[h,om]=freqz(b,a,w);
m=20*log(abs(h));
an=angle(h);
subplot(2,1,1);
plot(om/pi,m);
grid;
subplot(2,1,2);
plot(om/pi,an);
grid;

OUTPUT:

Butterworth band pass filter:
Start
Assign alphap=2, alphas=20.
Calculate wp=[0.2*pi,0.4*pi],ws=[0.1*pi,0.5*pi].

buttord function
Assign w=0:0.01:pi
Display butterworth
bandpass filter response
Stop

%TO DESIGN A BUTTERWORTH BANDPASS FILTER
clear all;
alphap=2;
alphas=20;
wp=[0.2*pi,0.4*pi];
ws=[0.1*pi,0.5*pi];
[n,wn]=buttord(wp/pi,ws/pi,alphap,alphas)
[b,a]=butter(n,wn);
w=0:0.01:pi;
[h,ph]=freqz(b,a,w);
m=20*log(abs(h));
an=angle(h);
subplot(2,1,1);
plot(ph/pi,m);
grid;
subplot(2,1,2);
plot(ph/pi,an);
grid;

OUTPUT:

Butterworth band stop filter:
Start
Assign alphap=2, alphas=20.
Calculate wp=[0.2*pi,0.4*pi],ws=[0.1*pi,0.5*pi].

buttord function
Assign w=0:0.01:pi
Display butterworth bandstop

filter response
Stop

%TO DESIGN A BUTTERWORTH BANDREJECT FILTER
clear all;
alphap=2;
alphas=20;
wp=[0.2*pi,0.4*pi];
ws=[0.1*pi,0.5*pi];
[n,wn]=buttord(wp/pi,ws/pi,alphap,alphas)
[b,a]=butter(n,wn,'stop');
w=0:0.01:pi;
m=20*log(abs(h));
an=angle(h);
subplot(2,1,1);
plot(ph/pi,m);
grid;
subplot(2,1,2);
plot(ph/pi,an);
grid;

OUTPUT:
RESULT:
Thus the MATLAB program to design a Butterworth low pass, high pass, band pass and
band reject filter is performed and graph is drawn.

Ex.No :
Date :
DESIGN OF IIR FILTER USING CHEBYCHEV TECHNIQUE
AIM:
To write a MATLAB program to design a chebychev IIR filter low pass, band pass and band
reject filter.
ESSENTIAL REQUIRED:
Hardware: PC or Compatible
Software: MATLAB version 7.4.
ALGORITHM:
CHEBYSHEV-LOW PASS FILTER:
Step1: Start the program
Step2: Assign alphap=1, alphas=15
Step3: Perform the function wp=0.2*pi,ws=0.3*pi
Step4: Calculate the cutoff frequency and the order of the filter.
Step5: Assign the value w=0:0.01:pi
Step6: Calculate the z-transform using the frequency function.
Step7: Perform m=20*log(abs(h)) and an=angle(h)
Step8: Display the chebyshev low pass filter function

CHEBYSHEV-BAND PASS FILTER:
Step3: Perform the function wp=0.2*pi, 0.4*pi ws=0.1*pi, 0.5*pi
Step4: Calculate the cutoff frequency and the order of the function
Step8: Display the chebyshev band pass filter function
CHEBYSHEV-BAND REJECT FILTER:
Step3: Perform the function wp=0.2*pi, 0.4*pi ws=0.1*pi, 0.5*pi
Step4: Calculate the cutoff frequency and the order of the function.
Step7: Perform m=20*log (abs (h)) and an=angle (h)
Step8: Display the chebyshev band pass filter function

CHEBYSHEV-HIGH PASS FILTER:
Step3: Perform the function wp=0.2*pi,ws=0.3*pi
Step4: Calculate the cutoff frequency and the order of the filter.
Step8: Display the chebyshev low pass filter function

FLOW CHART:
LOW PASS AND HIGH PASS FILTER
Start
Assign alphap=1, alphas=15
Calculate wp=0.2*pi& ws=0.3*pi.
Calculate cutoff frequency and order of the filter.
Assign w=0:0.01:pi
Perform m=20*log(abs(h)) and an=angle(h)
Display chebychev low pass

&high pass filter response
Stop

BAND PASS AND BAND REJECT FILTERS:
Start
Assign alphap=2, alphas=20
Calculate wp=[0.2*pi,0.4*pi]&
ws=[0.1*pi,0.5*pi]
Calculate cutoff frequency and order of the filter.
Assign w=0:0.01:pi
Perform m=20*log10(abs(h)) for band pass and

m=20*log(abs(h)) for band reject &an=angle(h)
Display chebychev band pass and

band reject filter response
Stop

PROGRAM:
%TO DESIGN A CHEBYCHEV LOW PASS FILTER
clear all;
alphap=1;
alphas=15;
wp=0.2*pi;
ws=0.3*pi;
%to find cuttoff frequency and order of filter
[n,wn]=cheb1ord(wp/pi,ws/pi,alphap,alphas);
[b,a]=cheby1(n,alphap,wn);
w=0:0.01:pi;
m=20*log(abs(h));
an=angle(h);
subplot(2,1,1);
plot(ph/pi,m);
grid;
ylabel('Gain in dB');
xlabel('Normalised frequency');
subplot(2,1,2);
plot(ph/pi,an);
grid;
title('low pass filter');

ylabel('Phase in radians');
%TO DESIGN A CHEBYCHEV HIGH PASS FILTER
clear all;
alphap=1;
alphas=15;
wp=0.2*pi;
ws=0.1*pi;
[b,a]=cheby1(n,alphas,wn,'high');
w=0:0.01:pi;
m=abs(h);
an=angle(h);
subplot(2,1,1);
plot(ph/pi,m);
grid;
title('high pass filter');
subplot(2,1,2);
plot(ph/pi,an);

grid;
title('high pass filter');
%TO DESIGN A CHEBYCHEV BAND PASS FILTER
clear all;
alphap=2;
alphas=20;
wp=[0.2*pi,.4*pi];
ws=[0.1*pi,.5*pi];
[n,wn]=buttord(wp/pi,ws/pi,alphap,alphas);
[b,a]=cheby1(n,alphap,wn);
w=0:0.01:pi;
m=20*log10(abs(h));
an=angle(h);
subplot(2,1,1);
plot(ph/pi,m);
grid;
title('BAND PASS FILTER');

subplot(2,1,2);
plot(ph/pi,an);
grid;
title('band pass filter');
%TO DESIGN A CHEBYCHEV BAND REJECT FILTER
clear all;
alphap=2;
alphas=20;
wp=[0.2*pi,.4*pi];
ws=[0.1*pi,.5*pi];
[b,a]=cheby2(n,alphas,wn,'stop');
w=0:0.01:pi;
m=20*log(abs(h));
an=angle(h);
subplot(2,1,1);
plot(ph/pi,m);
grid;

title('Band reject filter');
subplot(2,1,2);
plot(ph/pi,an);
grid;
title('band reject filter');
xlabel('Normalised frequency'
OUTPUT:
LOW PASS FILTER
low pass filter

0
-200
Gain in dB
-400
-600
-800
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Normalised frequency
low pass filter
4
Phase in radians
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

BAND PASS FILTER
BAND PASS FILTER

0
-100
Gain in dB
-200
-300
-400
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
band pass filter
4
Phase in radians
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
BAND REJECT FILTER
Band reject filter

0
-50
Gain in dB
-100
-150
-200
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
band reject filter
4
Phase in radians
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

HIGH PASS FILTER
high pass filter

1
Gain in dB
0.5
0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
high pass filter
4
Phase in radians
-2
-4
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
RESULT:
Thus the MATLAB program to design and plot the low pass, high pass, band pass, band
reject using chebyshev filter technique and the output is verified.

Ex.No :
Date :
CALCULATION OF FFT OF A SIGNAL
AIM:
To write the MATLAB program for the following given sequence

x(n)=[0,1,2,3,4,5,6,7] .compute and plot the 8 point DFT and IDFT of the sequence .
Hardware: pc or compatible
Software: MATLAB version7.4
ALGORITHM:
DISCRETE FOURIER TRANSFORM
Step2: First enter the sequence x (n).
Step3: Plot the input sequence x (n).
Step4: Compute DFT of the sequence using FFT algorithm.
Step5: Display the DFT sequences.
Step6: Plot the real part.
Step7: Plot the imaginary part.
Step8: Use abs command to obtain magnitude plot.
Step9: Use angle command to obtain the phase plot and plot it.

INVERSE DISCRETE FOURIER TRANSFORM
Step2: First enter the sequence X (k).
Step3: Plot the input sequence X (k).
Step4: Compute IDFT of the sequence using IFFT command.
Step5: Display the IDFT sequence.
Step6: Plot the real part.
Step7: Plot the imaginary part.
Step8: Use abs command to obtain magnitude plot.
Step9: Use angle command to obtain the phase plot and plot it.

FLOW CHART:
DISCRETE FOURIER TRANSFORM
Start
Plot the sequence x(n)
Compute DFT using FFT algorithm
Display the DFT sequence
Plot the real part
Plot the imaginary part
Plot the magnitude using
abs command
Plot the phase using angle

command
Stop

INVERSE DISCRETE FOURIER TRANSFORM
Start
Plot the sequence x(n)
Compute IDFT using IFFT command
Display the IDFT sequence
Plot the real part
Plot the imaginary part
Plot the magnitude using
abs command
Plot the phase using angle

command
Stop

PROGRAM:
%COMPUTATION OF DFT
x=input ('Enter the sequence');
n=length(x);
y=fft(x,n);
subplot (2,2,1);
stem (1:n,real(y));
title ('Real part of x(k):');
xlabel('Index k');
ylabel('real of x(k)');
subplot (2,2,2);
stem (1:n,imag(y));
title ('Imaginary part of x(k):');
xlabel('index of k');
ylabel('imag of x(k)');
m=abs(y);
subplot (2,2,3);
stem (m);
title ('magnitude of x(k):');
xlabel('index of f');
ylabel('magnitude of x(k)');
disp(real(y));
disp('imag of x(k)');
disp(imag(y));

disp('Magnitude of x(k):');
disp(m);
p=angle(y);
subplot (2,2,4);
stem (p);
title ('phase of x(k)');
ylabel('phase of x(k)');
disp('phase of x(k):');
disp(p);
INPUT:
Enter the sequence[1 2 3 4 5]
15.0000 -2.5000 -2.5000 -2.5000 -2.5000
imag of x(k)
0 3.4410 0.8123 -0.8123 -3.4410
Magnitude of x(k):
15.0000 4.2533 2.6287 2.6287 4.2533
phase of x(k):
0 2.1991 2.8274 -2.8274 -2.1991

OUTPUT:
Real part of x(k): Imaginary part of x(k):

15 4
10 2
imag of x(k)
real of x(k)
5 0
0 -2
-5 -4
1 2 3 4 5 1 2 3 4 5
Index k index of k
magnitude of x(k): phase of x(k)
15 4
magnitude of x(k)
2
phase of x(k)
10
0
5
-2
0 -4
1 2 3 4 5 1 2 3 4 5
index of f index of k
%COMPUTATION OF IDFT
clc;
clear all;
close all;
x=input('enter the sequence:');
n=length(x);
y=ifft(x,n);
subplot(2,2,1);
stem(1:n,real(y));
title('real part of x(k)');
subplot(2,2,2);

stem(1:n,imag(y));
title('imaginary part of x(k)');
ylabel('imaginary of x(k)');
m=abs(y);
subplot(2,2,3);
stem(m);
title('magnitude of x(k)');
xlabel('index of f');
disp('real of x(k)');
disp(real(y));
disp('imaginary of x(k)');
disp(imag(y));
disp('magnitude of x(k)');
disp(m);
p=angle(y);
subplot(2,2,4);
stem(p);
title('phase of x(k)');
disp('phase of x(k)');
disp(p);

INPUT:
enter the sequence:[2 4 6 8]
real of x(k)
5 -1 -1 -1
imaginary of x(k)
0 -1 0 1
magnitude of x(k)
5.0000 1.4142 1.0000 1.4142
phase of x(k)
0 -2.3562 3.1416 2.3562
OUTPUT:
real part of x(k) imaginary part of x(k)

6 1
imaginary of x(k)
4 0.5
real of x(k)
2 0
0 -0.5
-2 -1
1 2 3 4 1 2 3 4
index of k index of k
magnitude of x(k) phase of x(k)
6 4
2
real of x(k)
real of x(k)
4
0
2
-2
0 -4
1 2 3 4 1 2 3 4
index of f index of k
RESULT:
Thus the computation of N-point DFT and IDFT was done and the output was obtained.

Ex.No:
Date :
WAVEFORM GENERATION
AIM:
 To generate triangular waveform of DAC channel.

 To generate the sine waveform at DAC port.
 To generate the square waveform at DAC port.
DSP trainer kit: TMS 320C50
Universal debugger for C50
PROGRAM:
;square waveform generation.
.mmregs
.text
start:
ldp #100h
lacc #0fffh ;change this value for amplitude.
loop: sacl 0
rpt #0ffh ;change this value for frequency.
out 0,04 ;address for dac.
cmpl
b loop
.end

;triangle waveform generation
AMPLITUDE .SET 4
FREQ .SET 350
TEMP .SET 0
.mmregs
.text
START:
LDP #100H
splk #0,TEMP
CONT1:
lar AR2,#FREQ
CONT:
out TEMP,4
LACC TEMP
ADD #AMPLITUDE
SACL TEMP
MAR *,AR2
BANZ CONT,*-
LAR AR2,#FREQ
CONTx:
OUT TEMP,4
LACC TEMP

SUB #AMPLITUDE
SACL TEMP
MAR *,AR2
BANZ CONTx
B CONT1
.end
Sine Wave Generation. With these default values sine generated is 100 Hz.
INCFREQ .set 10 ;minimum value 1 - change for increasing frequency
;every count will increase frequency in steps of 100hz
DECFREQ .set 0 ;minimu value 0 - change for decreasing frequency
;every count will decrease frequency by half
LENGTH .set 360
AMPLITUDE .set 5
delay .set 7 ;change to increase/decrease frequency in
;different steps;
TEMP .set 0
TEMP1 .set 1
.mmregs
.text
START:
LDP #100H

SPLK #TABLE,TEMP ;load start address of table
lar AR2,#( (LENGTH/INCFREQ)+(LENGTH*DECFREQ) )-1
lar ar3,#DECFREQ ;repeat counter for reducing frequency
CONT:
CALL DELAY
LACC TEMP ;load address of sine value
TBLR TEMP1 ;read sine data to TEMP1
LT TEMP1
MPY #AMPLITUDE
PAC
SACL TEMP1
mar *,ar3
banz repeat,*-
lar ar3,#DECFREQ ;repeat counter for reducing frequency
LACC TEMP
ADD #INCFREQ ;increase table value
repeat:
SACL TEMP ;store it
OUT TEMP1,4 ;send sine data to DAC
MAR *,AR2
BANZ CONT,*-
b START

DELAY:
lar ar7,#delay
mar *,ar7
back: banz back,*-
ret
TABLE
.word 100
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.word 2
.word 1
.word 1
.word 1
.word 0
.word 0
.word 0
.word 0
.word 0
.word 0
.word 0
.word 0

.word 0
.word 0
.word 0
.word 0
.word 0
.word 0
.word 0
.word 0
.word 0
.word 1
.word 1
.word 1
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.end
SAWTOOTH WAVEFORM GENERRATION
.MMREGS
.TEXT
START:
LDP #120H
LACC #0H ;change lower amplitude
SACL 0
LOOP: LACC 0
OUT 0,04H
ADD #05h ;change frequency
SACL 0
SUB #0FFFh ;change upper amplitude
BCND LOOP,LEQ
B START
.END
Result:
Thus the waveforms are generated using DSP processor.

Ex.No:
Date :
IMPLEMENTATION OF CONVOLUTION
AIM:
To calculate the convolution using DSP processor for the sequence.
x (n)=[1 3 1 3]
h (n)=[0 1 2 1]
PROGRAM:
.mmregs
.text
START:
LDP #02H
LAR AR1,#8100H ; x(n) datas
lar ar0,#08200H ;h(n) datas
LAR AR3,#8300H ;y(n) starting
LAR AR4,#0007 ;N1+N2-1

;to fold the h(n) values
lar ar0,#8203H ; data mem 8200 to program mem c100(tblw)
lacc #0c100h
mar *,ar0
rpt #3
tblw *- ;to move 8203- 8200 to c100- c103
;padding of zerros for x(n) values
lar ar6,#8104h
mar *,ar6
lacc #0h
rpt #3h
sacl *+
;convalution operation starts
LOP: MAR *,AR1
LACC *+
SACL 050H ;starting of the scope of multiplication
LAR AR2,#0153H ; end of the array, to be multiplied with h(n) {150+N1-1}
MAR *,AR2

ZAP
RPT #03H ;N1-1 times so that N1 times
MACD 0C100H,*-
APAC ;to accmulate the final product sample
MAR *,AR3
SACL *+
MAR *,AR4
BANZ LOP,*-
H: B H
RESULT:
Thus the linear convolution using DSP processor was calculated and the output is verified.

Ex.No:
Date :
IMPLEMENTATION OF FIR FILTER
AIM:
To implement a band pass and band reject FIR filter using Blackmann window.
APPARATUS REQUIRED:
PROGRAM:
BAND PASS:
* Approximation type: Window design - Blackmann Window
* Filter type: bandpass filter
* Filter Order: 52
* lower Cutoff frequency in KHz = 3.000000Hz
* upper Cutoff frequency in KHz = 5.000000Hz
.mmregs
.text
B START
CTABLE:
.word 024AH

.word 010FH
.word 0FH
.word 0FFECH
.word 0C6H
.word 0220H
.word 0312H
.word 02D3H
.word 012FH
.word 0FEBDH
.word 0FC97H
.word 0FBCBH
.word 0FCB0H
.word 0FE9EH
.word 029H
.word 0FFDCH
.word 0FD11H
.word 0F884H
.word 0F436H
.word 0F2A0H
.word 0F58AH
.word 0FD12H
.word 075FH
.word 01135H
.word 01732H
.word 01732H
.word 01135H
.word 075FH
.word 0FD12H
.word 0F58AH
.word 0F2A0H
.word 0F436H
.word 0F884H
.word 0FD11H
.word 0FFDCH
.word 029H
.word 0FE9EH
.word 0FCB0H
.word 0FBCBH
.word 0FC97H
.word 0FEBDH
.word 012FH
.word 02D3H
.word 0312H
.word 0220H
.word 0C6H

.word 0FFECH
.word 0FH
.word 010FH
.word 024AH
*
* Move the Filter coefficients
* from program memory to data memory
START:
MAR *,AR0
LAR AR0,#0200H
RPT #33H
BLKP CTABLE,*+
SETC CNF
* Input data and perform convolution
ISR: LDP #0AH
LACC #0
SACL 0
OUT 0,05 ;pulse to find sampling frequency
IN 0,06H
LAR AR7,#0 ;change value to modify sampling freq.

MAR *,AR7
BACK: BANZ BACK,*-
IN 0,4
NOP
NOP
NOP
NOP
MAR *,AR1
LAR AR1,#0300H
LACC 0
AND #0FFFH
SUB #800H
SACL *
LAR AR1,#333H
MPY #0
ZAC
RPT #33H
MACD 0FF00H,*-
APAC
LAR AR1,#0300H

SACH * ;give as sach *,1 incase of overflow
LACC *
ADD #800H
SACL *
OUT *,4
LACC #0FFH
SACL 0
OUT 0,05
NOP
B ISR
.end
BAND REJECT:
* Approximation type: Window design - Blackmann Window
* Filter type: bandreject filter
* Filter Order: 52
* lower Cutoff frequency in KHz = .000000Hz
* upper Cutoff frequency in KHz = .000000Hz
.mmregs
.text
B START

CTABLE:
.word 0FEB9H
.word 14EH
.word 0FDA1H
.word 155H
.word 0FE1BH
.word 282H
.word 0FEAFH
.word 2ACH
.word 0FD35H
.word 8DH
.word 0F9D9H
.word 0FE07H
.word 0F7CCH
.word 0FEE2H
.word 0FA2FH
.word 4BAH
.word 1AH
.word 25CH
.word 420H
.word 1008H
.word 89H
.word 0D61H
.word 0F3F2H
.word 0AF9H
.word 0DB7EH
.word 045DFH
.word 045DFH
.word 0DB7EH
.word 0AF9H
.word 0F3F2H
.word 0D61H
.word 89H
.word 1008H
.word 420H
.word 25CH
.word 1AH
.word 4BAH
.word 0FA2FH
.word 0FEE2H
.word 0F7CCH
.word 0FE07H
.word 0F9D9H
.word 8DH
.word 0FD35H
.word 2ACH

.word 0FEAFH
.word 282H
.word 0FE1BH
.word 155H
.word 0FDA1H
.word 14EH
.word 0FEB9H
* Move the Filter coefficients
* from program memory to data memory
START:
MAR *,AR0
LAR AR0,#0200H
RPT #33H
BLKP CTABLE,*+
SETC CNF
* Input data and perform convolution
ISR: LDP #0AH

LACC #0

SACL 0
OUT 0,05 ;pulse to find sampling frequency
IN 0,06H
LAR AR7,#0 ;change value to modify sampling freq.
MAR *,AR7
BACK: BANZ BACK,*-
IN 0,4
NOP
NOP
NOP
NOP
MAR *,AR1
LAR AR1,#0300H
LACC 0
AND #0FFFH
SUB #800H
SACL *
LAR AR1,#333H

MPY #0
ZAC
RPT #33H
MACD 0FF00H,*-
APAC
LAR AR1,#0300H
SACH * ;give as sach *,1 incase of overflow
LACC *
ADD #800H
SACL *
OUT *,4
LACC #0FFH
SACL 0
OUT 0,05
NOP
B ISR
.end
Result:
Thus a band pass and band reject FIR filter using Blackmann window is implemented.

Generation OF Signals: Digital Signal Processing Lab

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Generation OF Signals: Digital Signal Processing Lab

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Digital Signal processing Lab

To write a MATLAB for the following

(i) Plot the unit impulse function.

(ii) Plot the unit step function.

(iii)Plot the unit ramp and exponential function.

Software: MATLAB 7.4

Step1: Get the time interval.

Step3: Subplot and stem the sequence.

Step4: Give x label, y label and title.

Step5: Stop the program and execute it.

Department of ECE Page 1

Step1: Give the number of sequences.

Step2: Get the time interval.

Step3: Give the matrix using one’s function.

Step4: Subplot and stem the sequence.

Step5: Give the x label, y label and title.

Step6: Stop the program.

Step1: Give the length of the sequence.

Step2: Give the time interval.

Step3: Subplot and stem the sequence

Step4: Give x label, y label and title.

Step5: Stop the program.

Step1: Enter the length of the sinusoidal sequence.

Step2: Enter the number of cycle.

Step4: Subplot and stem the waveform.

Step5: Give the x label, y label and title.

Step6: Stop the program.

Department of ECE Page 2

Step1: Enter the length of cosine sequence.

Step2: Enter the number of cycles.

Step3: Get the interval and obtain cosine function.

Step4: Subplot and stem the waveform.

Step5: Give x label, y label and title.

Step6: Stop the program.

Step1: Start the exponential function.

Step2: Plot the exponential sequence in discrete domain.

Step3: Give the x label, y label and title.

Step4: Stop the program.

Step1: Set the time domain from 0 to 5.

Step2: Generate the saw tooth wave and store it.

Step4: Give x label, y label and title.

Step5: Stop the program.

Department of ECE Page 3

Step1: Set the time domain from 0 to 5.

Step2: Generate the square wave and store it.

Step4: Given x label, y label and title.

Step5: Stop the program.

Get the time interval

Compute the impulse function using zero

Plot output sequence and give

Department of ECE Page 4

Get the number of sequences and

Compute the step function using ones function

Plot the output sequence and give

Get the length of the sequence

Compute the ramp function using zero and ones

Plot the output sequence and give

Department of ECE Page 5

Enter the length of the sinusoidal

Get the interval and obtain sine waveform using

Plot the output sequence and give

Enter the length of sinusoidal

Get the time interval and obtain cosine using cosine