Lab Manual DSP Lab 5 Sem

Dr H.S.
Prashantha, Dept of ECE, PESIT

1

DSP LABORATORY MANUAL

Prepared by
Dr H.S.Prashantha
Associate Professor
Department of Electronics Communication Engineering
PES Institute of Technology
Bangalore

Dr H.S.Prashantha, Dept of ECE, PESIT

2

DSP LABORATORY MANUAL

CONTENTS
DSP using Matlab
1. Verification of sampling theorem.
2. Impulse response of a given system.
3. Linear convolution of two given sequences.
4. Circular convolution of two given sequences.
5. Autocorrelation of a given sequence and verification of its properties.
6. Cross correlation of a given sequence and verification of its properties
7. Solving a given difference equation.
8. Computation of N point DFT of a given sequence and to plot magnitude and phase
Spectrum.
9. Linear convolution of two given sequences using DFT and IDFT.
10. Circular convolution of two given sequences using DFT and IDFT
11. Design and implementation of FIR filter to meet given specification.
12. Design and implementation of IIR filter to meet given specification.

DSP Experiments using DSP processor
3. Computation of N point DFT of a given sequence.
4. Realization of an FIR filter (any type) to meet given specification. The input can be a signal
from function generator/speech signal.
5. Audio specification such as to plot time and frequency display of microphone plus a cosine
using DSP. Read a wav file and match with their respective spectrograms.
6. Noise removal: Add noise above 3KHz and then remove; Interference suppression using 400
Hz ton.
7. Impulse response of a given system of a given system of first and second order.


3

Cycle I

DSP using Matlab
1. Verification of sampling theorem.
2. Impulse response of a given system.
5. Autocorrelation of a given sequence and verification of its properties.
6. Cross correlation of a given sequence and verification of its properties
7. Solving a given difference equation.

10. Impulse response of a given system of a given system of first and second order.

Cycle II

DSP using Matlab
1. Computation of N point DFT of a given sequence and to plot magnitude and phase
Spectrum.
2. Linear convolution of two given sequences using DFT and IDFT.
3. Circular convolution of two given sequences using DFT and IDFT
4. Design and implementation of FIR filter to meet given specification.
5. Design and implementation of IIR filter to meet given specification.

1. Computation of N point DFT of a given sequence.
2. Realization of an FIR filter (any type) to meet given specification. The input can be a signal
from function generator/speech signal.
3. Audio specification such as to plot time and frequency display of microphone plus a cosine
using DSP. Read a wav file and match with their respective spectrograms.
4. Noise removal: Add noise above 3KHz and then remove; Interference suppression using 400
Hz ton.


4

Cycle 1

DSP USING MATLAB

MATLAB is a high-level language and interactive environment that enables you to perform
computationally intensive tasks faster than with traditional programming languages such as C,
C++ etc. The key features of MATLAB are as follows
High-level language for technical computing
Development environment for managing code, files, and data
Interactive tools for iterative exploration, design, and problem solving
Mathematical functions for linear algebra, statistics, Fourier analysis, filtering,
optimization, and numerical integration
2-D and 3-D graphics functions for visualizing data
Tools for building custom graphical user interfaces

MATLAB (MATrix LABoratory) is a special purpose computer program optimized to perform
engineering and scientific calculations.
The MATLAB program implements the MATLAB language, and provides a very extensive
library of predefined functions to make technical programming tasks easier and more efficient.
The extremely wide variety of functions makes it much easier to solve technical problems in
MATLAB than in other languages such as FORTRAN or C.
The advantages of using MATLAB for technical programming are:
Ease of use
Platform independence
Predefined functions
Device-independent plotting
Graphical User Interface
MATLAB Compiler
MATLAB is a huge program, with an incredibly rich variety of functions. There are more
than 1000 functions in the basic MATLAB.
The disadvantages of MATLAB are:
It is interpreted language, and therefore can execute more slowly than compiled
languages.
Cost. (Student version of MATLAB is available for low cost with almost identical
feature) full copy of MATLAB is 5 TO 10 times more expensive than a conventional C
or FORTRAN compiler.
The functional unit of data in any MATLAB program is the array. An array is a
collection of data values organized into rows and columns.
When MATLAB executes, it can display several types of windows that accept commands
or display information. The three most important types of windows are command
windows, where commands may be entered: figure windows, which display plots and
graphs: and edit/debug windows which permit a user to create and modify MATLAB
programs.
The major tools within or accessible from the MATLAB desktop are:
The command window: a user can enter interactive commands at the command
prompt (>>) in the command window and they will be executed on the spot.
Instead of typing commands directly in the command window, a series of
commands may be placed in to a single file and the entire file may be executed by

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typing its name in the command window. Such files are called script files. Script
files are also called M-Files because they have file extension of . M.
The command history window: the command history window displays a list of the
commands that a user has entered in the command window.
The launch pad: it is a special tool that collects references to the documentation,
demos, and related tools for MATLAB itself and for each toolkit that you own.
The edit/debug window: used to create new M-Files, or to modify existing ones.
An edit/debug window is created automatically when you create a new M-File or
open an existing one. The edit/debug window is essentially a programming text
editor, with the MATLAB languages features highlighted in different colors.
Comments in an M-File appear in green, variables and numbers appear in black,
character strings appear in red, and language keywords appear in blue.
Figure windows: used to display MATLAB graphics. A figure can be two or
three-dimensional plot of data, an image or a GUI.
The MATLAB workspace: it is a collection of all the variables and arrays that can
be used by MATLAB when a particular command, M-File, or function is
executing. All commands executed in the command window share a common
workspace, so they can all share variables. (Clear deletes all variables from the
current workspace).
The workspace browser: it provides a graphic display of the same information as
whos command, and the display is dynamically updated whenever the contents of
workspace change. The workspace browser also allows user to change the
contents of any variables in the workspace. (A list of the variables and arrays in
the current workspace can be generated with the whos command)

Getting help: by selecting the? Help icon bar from the desktop tool bar, or by typing
helpdesk or helpwin in the command window. The other way of obtaining help is by
using lookfor command. Help command searches for an exact function name match,
whereas the lookfor command searches the quick summary information in each function
for a match. Lookfor is slower than help but it improves the chances of getting useful
information.
For new users of MATLAB, a few demonstrations may help to give you a feel for its
capabilities. To run MATLABs built in demonstrations, type demo in the command
window, or select demos from the launch pad.
The contents of the command window can be cleared at any time using the clc command,
and the contents of the current figure window can be cleared at any time using the clf
command. The variables in the workspace can be cleared with the clear command. It is a
good idea to issue the clear command at the start of each new independent calculation.
Another important command is the abort command. If an m-file appears to be running for
too long, it might contain an infinite loop, and it will never terminate. In this case the user
can regain the control by typing control-c (^c) in the command window. When MATLAB
detects a ^c, it interrupts the running program and returns a command prompt.
After diary command is typed, a copy of all input and most output typed in the command
window is echoed in the diary file. The command diary off suspends input in to the diary
file, and the command diary on resumes input again.

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MATLAB includes a special command (which) to help you to find out just which version
of a file is being executed. and where it is located. The format of this command is which
functionname, where functionname is the name of the function that you are trying to
locate.
The path related function include:
* addpath Add directory to MATLAB search path
* path Display MATLAB search path
* path2rc Add current directory to MATLAB search path
* rmpath Remove directory from MATLAB search path

The functional unit of data in any MATLAB program is the array. An array is a
collection of data values organized into rows and columns, and known by a single name.
MATLAB variable is a region of memory containing an array, which is known by a user-
specified name. MATLAB variable names must begin with a letter, followed by any
combination of letters, numbers, and the underscore( _ ) character. Only the first 31
characters are significant; if more than 31 are used, the remaining characters will be
ignored. If two variables are declared with names that only differ in the 32
nd
character,
MATLAB will treat them as same variable.
Spaces cannot be used in MATLAB variable names, underscore letters can be substituted
to create meaningful names.
It is important to include a data dictionary in the header of any program that you write. A
data dictionary lists the definition of each variable used in a program. The definition
should include both a description of the contents of the item and the units in which it is
measured.
MATLAB language is case-sensitive. It is customary to use lower-case letters for
ordinary variable names.
The most common types of MATLAB variables are double and char.
MATLAB is weakly typed language. Variables are not declared in a program before it is
used.
MATLAB variables are created automatically when they are initialized. There are three
common ways to initialize variables in MATLAB:
Assign data to the variable in an assignment system.
Input data into the variable from the keyboard.
Read data from a file.
The semicolon at the end of each assignment statement suppresses the automatic echoing
of values that normally occurs whenever an expression is evaluated in an assignment
statement.

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How to invoke MATLAB?
*) Double Click on the MATLAB icon on the desktop .
*) You will find a Command window where in which you can type the commands and see the
output. For example if you type PWD in the command window, it will print current working
directory.
*)If you want to create a directory type mkdir mydir in the command window, It will create a
directory called pes.
*) If you want delete a directory type rmdir mydir in the command window.
How to open a file in MATLAB?
*) Go to File New M-File and click
Then type the program in the file and save the file with an extension of .m. While giving file
name we should make sure that given file name should not be a command. It is better to the file
name as myconvlution .

How to run a MATLAB file?
*) Go to Debugrun and click

1. Program to Verify Sampling Theorem

% program for verification of sampling theorem

clc; % clears the command window
clear all; % clears the variables declared

t=0:.001:0.1;
f1=input ('Enter the input freequency1 = ');
f2=input ('Enter the input freequency2 = ');

y=cos(2*pi*f1*t)+cos(2*pi*f2*t);
f3=max(f1,f2);

% under sampling
fs=f3; %fs = sampling freequency
ts=1/fs;
tx=0:ts:0.1;
m=max(size(tx));
ys=cos(2*f1*pi*tx)+cos(2*pi*f2*tx);
figure(1);
subplot(3,1,1);
plot(t,y);
grid on;
title('The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t)');
xlabel('Time in seconds');
ylabel('Amplitude in volts');

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subplot(3,1,2);
stem(tx,ys);
grid on;
title('The sinusoidal signal sampled at fs Hz');
subplot(3,1,3);
plot(tx,ys);
grid on;
title('The recovered sinusidal sampled at fs Hz');

% Right sampling
fs=2*f3; %fs = sampling freequency
ts=1/fs;
tx=0:ts:0.1;
m=max(size(tx));
ys=cos(2*pi*f1*tx)+cos(2*pi*f2*tx);
figure(2);
subplot(3,1,1);
plot(t,y);
grid on;
subplot(3,1,2);
stem(tx,ys);
grid on;
subplot(3,1,3);
plot(tx,ys);
grid on;

% over sampling
fs=3*f3; %fs = sampling freequency
ts=1/fs;
tx=0:ts:0.1;
m=max(size(tx));
ys=cos(2*pi*f1*tx)+cos(2*pi*f2*tx);
figure(3);

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subplot(3,1,1);
plot(t,y);
grid on;
subplot(3,1,2);
stem(tx,ys);
grid on;
subplot(3,1,3);
plot(tx,ys);
grid on;

OUTPUT
Under sampling

Right sampling
0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t)
Time in seconds
A
m
p
l
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t
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i
n

v
o
l
t
s
0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
0
1
2
The sinusoidal signal sampled at fs Hz
Time in seconds
A
m
p
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t
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d
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i
n

v
o
l
t
s
0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
0
1
2
The recovered sinusidal sampled at fs Hz
Time in seconds
A
m
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v
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10

Over sampling

0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
Time in seconds
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v
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0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
Time in seconds
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i
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v
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0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
Time in seconds
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v
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0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
Time in seconds
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v
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0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
Time in seconds
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v
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0 0.01 0.02 0.03 0.04 0.05 0.06 0.07 0.08 0.09 0.1
-2
0
2
Time in seconds
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2. Program to find the Impulse response of the system

%To find the Impulse response of a given System
%a
0
*y(n)-a
1
*y(n-1)+a
2
*y(n-2)=b
0
*x(n)+b1*x(n-1)+b
2
*x(n-2)
clc;
clear all;
b=input('Enter the coefficiants of x(n) in the order x(n),x(n-1)...in the Matrix form = ');
a=input('Enter the coefficiants of y(n) in the order y(n),y(n-1)...in the Matrix form = ');
n=[-5:20];
x=[(n==0)];
h=filter(b,a,x)
figure(1);
subplot(2,1,1);
stem(n,x);
xlabel('n');
ylabel('amplitude');
title('impulse sequence');
subplot(2,1,2);
stem(n,h);
xlabel('n');
title('impulse response');

OUTPUT
Enter the coefficiants of x(n) in the order x(n),x(n-1)...in the Matrix form = [0.2066 0.4131
0.2066]
Enter the coefficiants of y(n) in the order y(n),y(n-1)...in the Matrix form = [1 -0.3695 0.1968]

-5 0 5 10 15 20
0
0.5
1
n
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impulse sequence
-5 0 5 10 15 20
-0.2
0
0.2
0.4
0.6
n
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impulse response

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Amplitude= [0 0 0 0 0 0.2066 0.4894 0.3468 0.0318 -0.0565 -0.0271 0.0011
0.0057 0.0019 -0.0004 -0.0005 -0.0001 0.0001 0.0000 0.0000 -0.0000 -0.0000
0.0000 0.0000 0.0000 -0.0000]

3. Program to perform linear Convolution of two given sequences

%This program gives the convolution of two sequences
clc;
clear all;
close all;
x1=input('enter the first sequence'); %input of first sequence
x2=input('enter the second sequence'); %input of second sequence
n1=length(x1);
n2=length(x2);
N=n1+n2-1;

x1=[x1 zeros(1,N-n1)];
x2=[x2 zeros(1,N-n2)];
m=[0:1:N-1];

for n=0:N-1
y(n+1)=sum(x2(mod(n-m,N)+1).*x1);
end;
disp('linear convoluted sequence');
y

disp('first sequence');
x1
disp('second sequence');
x2
disp('convolved sequence');
y
subplot(3,1,1);
stem(x1);
title('first sequence');
xlabel('signal');
ylabel('time');

subplot(3,1,2);
stem(x2);
title('second sequence');
xlabel('signal');
ylabel('time');

subplot(3,1,3);
stem(y);

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title('convolved sequence');
xlabel('signal');
ylabel('time');

Input:
x1 =
1 2 3 4 0 0 0
second sequence
x2 =
1 2 3 4 0 0 0
convolved sequence
y =
1 4 10 20 25 24 16

4. Program to perform Circular Convolution of two given sequences

%This program calculates the circular convolution
clc;
clear all;
x=input('Input first sequence');
h=input('Input second sequence');
N1=length(x);
N2=length(h);
N=max(N1,N2);
1 2 3 4 5 6 7
0
2
4
first sequence
s
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g
n
a
l
time
1 2 3 4 5 6 7
0
2
4
second sequence
s
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n
a
l
time
1 2 3 4 5 6 7
0
20
40
convolved sequence
s
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n
a
l
time

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x=[x zeros(1,N-N1)];
h=[h zeros(1,N-N2)];

for m=1:N
y(m)=0;
for n=1:N
i=m-n+1;
if(i<=0)
i=N+i;
end
y(m)=y(m)+x(n)*h(i);
end
end

x
h
y
n=0:N-1;
subplot(3,1,1);
stem(x);
ylabel('signal');
xlabel('time');

subplot(3,1,2);
stem(h);
ylabel('signal');
xlabel('time');

subplot(3,1,3);
stem(y);
ylabel('signal');
xlabel('time');

OUTPUT

Input first sequence [1 2 3 4]
Input second sequence [1 2 3 4]

first sequence

15

x =

1 2 3 4

second sequence

h =

1 2 3 4

convolved sequence

y =

26 28 26 20

5. Autocorrelation of a given sequence
x=input('enter the sequence');
y=xcorr(x,x);
figure;
subplot(2,1,1);
stem(x);
1 1.5 2 2.5 3 3.5 4
0
2
4
first sequence
s
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g
n
a
l
time
1 1.5 2 2.5 3 3.5 4
0
2
4
second sequence
s
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n
a
l
time
1 1.5 2 2.5 3 3.5 4
0
20
40
convolved sequence
s
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g
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a
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time

16

xlabel('n');
subplot(2,1,2);
stem(fliplr(y));
y
xlabel('n');
disp('the resultant signal is ');
fliplr(y);

Input:
Enter the sequence[1 2 3 4]
The resultant signal is

y = [ 4 11 20 30 20 11 4]

6. Cross correlation of a given sequence
clc;
close all;
clear all;
x=input('enter the sequence');
h=input('enter the second sequence');
y=xcorr(x,h);
figure;
subplot(3,1,1);
1 1.5 2 2.5 3 3.5 4
0
1
2
3
4
a
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n
1 2 3 4 5 6 7
0
10
20
30
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n

17

stem(x);
xlabel('n');
subplot(3,1,2);
stem(h);
xlabel('n');
subplot(3,1,3);
stem(fliplr(y));
y
xlabel('n');
disp('the resultant signal is ');
fliplr(y);

Input:
enter the sequence[1 2 3 4]
enter the second sequence[4 3 2 1]

y =

1.0000 4.0000 10.0000 20.0000 25.0000 24.0000 16.0000

The resultant signal is

1 1.5 2 2.5 3 3.5 4
0
2
4
a
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n
1 1.5 2 2.5 3 3.5 4
0
2
4
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1 2 3 4 5 6 7
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10
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7. Program to Solve difference equation

%program for solution of difference equation
% y(n) -0.5 y(n-1) = x(n) with x(n)=u(n)-u(n-4)
b=input('enter the coefficients of x');
a=input('enter the coefficients of y');
n=[-5:50];
x=[(n>=0)]-[(n>4)];
figure(1);
subplot(2,1,1);
stem(n,x);
title('input sequence x(n)');
xlabel('n');
ylabel('x');
subplot(2,1,2);
y=filter(b,a,x);
stem(n,y);
title('output sequence y(n)');
xlabel('n');
ylabel('y');

% y(n) -0.25 y(n-1) -0.125 y(n-2)= x(n) with x(n)=u(n)-u(n-10) and initial condition y(-1)=1,
y(-2)=-2

-10 0 10 20 30 40 50
0
0.5
1
input sequence x(n)
n
x
-10 0 10 20 30 40 50
0
0.5
1
1.5
2
output sequence y(n)
n
y

19

b=input('enter the coefficients of x');
a=input('enter the coefficients of y');
c=input('enter the initial conditions');
n=[-5:50];
ic=filtic(b,a,c);

x=[(n>=0)]-[(n>10)];
y=filter(b,a,x,ic);

subplot(2,1,1);
stem(n,x);
title('input sequence x(n)');
xlabel('n');
ylabel('x');
subplot(2,1,2);

stem(n,y);
title('output sequence y(n)');
xlabel('n');
ylabel('y');

Output:

-10 0 10 20 30 40 50
0
0.5
1
input sequence x(n)
n
x
-10 0 10 20 30 40 50
0
0.5
1
1.5
2
output sequence y(n)
n
y

20

8. Program to Calculate N point DFT and N point IDFT of the given sequence

%to compute dft of the sequence and plot magnitude and phase spectra
clc;
clear all;
close all;
% input sequence
xn=input('enter the input sequence');
N=input('enter the number of points');

xk=dft(xn,N);

%magnitude of fft
magxk=abs(xk);

%phase of fft
phasexk=angle(xk);

k=0:N-1;

subplot(2,1,1);
stem(k,magxk);
title('fft sequence');
xlabel('frequency');
ylabel('magnitude');

subplot(2,1,2);
stem(k,phasexk);
title('phase of fft sequence');
xlabel('frequency');
ylabel('phase');

function xk=dft(xn,N)
L=length(xn);
if(N<L)
error('N must be >=L');
end
x1=[xn zeros(1,N-L)]

for k=0:1:N-1;
for n=0:1:N-1;
p=exp(-i*2*pi*n*k/N)
x2(k+1,n+1)=p;
end
end
xk=x1*x2;

OUTPUT:
X=[1 1 1 1 1 1 1 1]
N=32

21

%to compute idft of the sequence
clc;
clear all;
close all;
% input sequence
Xk=input('enter the input sequence');
N=length(Xk)

xn=idft(Xk,N);

k=0:N-1;

stem(k,xn);
title('ifft sequence');
xlabel('time');
disp(xn)

function xn=idft(Xk,N)

for k=0:1:N-1;
for n=0:1:N-1;
p=exp(i*2*pi*n*k/N)
x2(k+1,n+1)=p;
end
end
0 5 10 15 20 25 30 35
0
2
4
6
8
fft sequence
frequency
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0 5 10 15 20 25 30 35
-4
-2
0
2
4
phase of fft sequence
frequency
p
h
a
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e

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xn=(Xk*x2)./N;

9. Linear convolution of two sequences using DFT and IDFT
clc;
clear all;
n1=length(x);
n2=length(h);
N=n1+n2-1;

x1=[x zeros(1,N-n1)];
x2=[h zeros(1,N-n2)];
a=fft(x,N);
b=fft(h,N);
c=a.*b;
d=ifft(c,N);

x
h
d
n=0:N-1;
subplot(3,1,1);
stem(x);
ylabel('signal');
xlabel('time');

subplot(3,1,2);
stem(h);
ylabel('signal');
xlabel('time');

subplot(3,1,3);
stem(d);
ylabel('signal');
xlabel('time');
Input:
x = 1 1 1 1
second sequence
h = 1 1 1 1
convolved sequence

23

d = 1.0000 2.0000 3.0000 4.0000 3.0000 2.0000 1.0000

10. Circular convolution of two sequences using DFT and IDFT
clc;
clear all;
n1=length(x);
n2=length(h);
N=max(n1,n2);

if N>n1
x1=[x zeros(1,N-n1)];
end
if N>n2
x2=[h zeros(1,N-n2)];
end
a=fft(x,N);
b=fft(h,N);
c=a.*b;
d=ifft(c,N);

x
h
d
n=0:N-1;
1 1.5 2 2.5 3 3.5 4
0
0.5
1
first sequence
s
i
g
n
a
l
time
1 1.5 2 2.5 3 3.5 4
0
0.5
1
second sequence
s
i
g
n
a
l
time
1 2 3 4 5 6 7
0
2
4
convolved sequence
s
i
g
n
a
l
time

24

subplot(3,1,1);
stem(x);
ylabel('signal');
xlabel('time');

subplot(3,1,2);
stem(h);
ylabel('signal');
xlabel('time');

subplot(3,1,3);
stem(d);
ylabel('signal');
xlabel('time');

Result:
Input first sequence[1 2 3]
Input second sequence[1 2 3 4]
first sequence
x = 1 2 3
second sequence
h = 1 2 3 4
convolved sequence
d = 18 16 10 16


25

11. Design and implementation of FIR filters to meet given specifications
clear all;
close all;
clc;
%cutoff frequency of low pass response at 0.25 pi and a minimum stopband
%attenuation of 39 dB. transition band is 0.2 pi
wc=0.25*pi;
tw=0.2*pi;

%normalize the cutoff frequency and transition width
fcn=wc/pi;
twn=tw/pi;

%the required transition width is 39 dB. Choose Hanning window
%order of the filter
disp('order of the filter');
N=ceil(4/twn)
if(rem(N,2)==0);
N=N+1;
end;

1 1.2 1.4 1.6 1.8 2 2.2 2.4 2.6 2.8 3
0
2
4
first sequence
s
i
g
n
a
l
time
1 1.5 2 2.5 3 3.5 4
0
2
4
second sequence
s
i
g
n
a
l
time
1 1.5 2 2.5 3 3.5 4
0
10
20
convolved sequence
s
i
g
n
a
l
time

26

y=hanning(N)
disp('fir filter coefficients');

b=fir1(N-1,fcn,y)

disp('frequency response');
[H,f]=freqz(b,1,256)
H_mag=20*log10(abs(H));
plot(f/pi,H_mag);
xlabel('frequency in rad/sec');
ylabel('gain in dB');

output:

%passband edge frequency of 2 kHz, a minimum stopband attenuation of 50 dB
%transition band is 0.5 kHzat a sampling frequency of 8 kHz.

fs=8000;
%normalizing frequency
fn=fs/2;
%passband and stopband frequencies
fp=2000;
tw=500;
fs=fp+tw;
%normalize the cutoff frequency and transition width
fcn=(fp+fs)/(2*fn);
twn=tw/fs;
%the required transition width is 50 dB. Choose Hamming window
%order of the filter
disp('order of the filter');
N=ceil(4/twn)
if(rem(N,2)==0);
N=N+1;
end;
y=hamming(N)
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-140
-120
-100
-80
-60
-40
-20
0
20
frequency in rad/sec
g
a
i
n

i
n

d
B

27

disp('fir filter coefficients');
b=fir1(N-1,fcn,y)
disp('frequency response');
[H,f]=freqz(b,1,256)
H_mag=20*log10(abs(H));
plot(f/pi,H_mag);
xlabel('frequency in rad/sec');
ylabel('gain in dB');

Output:

12. Programs to Design IIR filters

% Program to Design the Digital IIR Butterworth Low pass filter using bilinear
transformation

clc;
clear all;

pb=input('Enter the pass band edge frequency ');
sb=input('Enter the stop band edge frequency ');
pbr=input('Enter the pass band attenuation ');
sbr=input('Enter the stop band attenuation ');

fs= input('Enter the sampling frequency ');

pbrad=pb*2*pi;
sbrad=sb*2*pi;

[n,wn]=buttord(pbrad,sbrad,pbr,sbr,'s');
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
-120
-100
-80
-60
-40
-20
0
20
frequency in rad/sec
g
a
i
n

i
n

d
B

28

[b,a]=butter(n,wn,'s');

[z,p,k]=tf2zp(b,a);
[zd,pd,kd]=bilinear(z,p,k,fs);
[num,den]=zp2tf(zd,pd,kd);

figure(1);
freqz(num,den,512,fs);
title('Butter worth freequency response');

[n,wn]=cheb1ord(pbrad,sbrad,pbr,sbr,'s');
[b,a]=cheby1(n,pbr,wn,'s');

[z,p,k]=tf2zp(b,a);

figure(2);
title('Chebychev1 Freequency Response');

[b,a]=cheby2(n,sbr,wn,'s');

[z,p,k]=tf2zp(b,a);

figure(3);
title('Chebychev 2 Freequency Response');

OUTPUT
Enter the pass band edge frequency 1500
Enter the stop band edge frequency 2000
Enter the pass band attenuation 0.5
Enter the stop band attenuation 30
Enter the sampling frequency 8000


29

% Program to Disign the Digital IIR Butterworth Low pass filter using impulse invariance
method

clc;
clear all;

pb=input('Enter the pass band edge frequency ');
sb=input('Enter the stop band edge frequency ');

pbr=input('Enter the pass band attenuation in dB ');
0 500 1000 1500 2000 2500 3000 3500 4000
-2000
-1000
0
Frequency (Hz)
P
h
a
s
e

(
d
e
g
r
e
e
s
)
0 500 1000 1500 2000 2500 3000 3500 4000
-500
0
500
Frequency (Hz)
M
a
g
n
i
t
u
d
e

(
d
B
)
Butter worth freequency response
0 500 1000 1500 2000 2500 3000 3500 4000
-1000
-500
0
Frequency (Hz)
P
h
a
s
e

(
d
e
g
r
e
e
s
)
0 500 1000 1500 2000 2500 3000 3500 4000
-500
0
500
Frequency (Hz)
M
a
g
n
i
t
u
d
e

(
d
B
)
Chebychev1 Freequency Response
0 500 1000 1500 2000 2500 3000 3500 4000
-500
0
500
Frequency (Hz)
P
h
a
s
e

(
d
e
g
r
e
e
s
)
0 500 1000 1500 2000 2500 3000 3500 4000
-100
-50
0
Frequency (Hz)
M
a
g
n
i
t
u
d
e

(
d
B
)
Chebychev 2 Freequency Response

30

sbr=input('Enter the stop band attenuation in dB ');

fs= input('Enetr the sampling frequency ');

pbrad=pb*2*pi;
sbrad=sb*2*pi;

[n,wn]=buttord(pbrad,sbrad,pbr,sbr,'s');
[b,a]=butter(n,wn,'s');
[num,den]=impinvar(b,a,fs);

figure(1);
title('Butter worth frequency response');

[b,a]=cheby1(n,pbr,wn,'s');

figure(2);
title('Chebychev1 Frequency Response');

[b,a]=cheby2(n,sbr,wn,'s');

figure(3);
title('Chebychev 2 Frequency Response');


31

DSP KIT TMS306713:
Troubleshooting DSK Connectivity

If Code Composer Studio IDE fails to configure your port correctly, perform the following steps:

Test the USB port by running DSK Port test from the start menu

Use StartProgramsTexas InstrumentsCode Composer StudioCode Composer Studio
C6713 DSK ToolsC6713 DSK Diagnostic Utilities
The below Screen will appear
Select StartSelect 6713 DSK Diagnostic Utility Icon from Desktop
The Screen Look like as below
Select Start Option
Utility Program will test the board
After testing Diagnostic Status you will get PASS

If the board still fails to detect


32

Go to CMOS setup Enable the USB Port Option
(The required Device drivers will load along with CCS Installation)

INTRODUCTION TO CODE COMPOSER STUDIO

Code Composer is the DSP industry's first fully integrated development environment (IDE) with
DSP-specific functionality. With a familiar environment liked MS-based C++TM, Code
Composer lets you edit, build, debug, profile and manage projects from a single unified
environment. Other unique features include graphical signal analysis, injection/extraction of data
signals via file I/O, multi-processor debugging, automated testing and customization via a C-
interpretive scripting language and much more.

CODE COMPOSER FEATURES INCLUDE:

IDE
Debug IDE
Advanced watch windows
Integrated editor
File I/O, Probe Points, and graphical algorithm scope probes
Advanced graphical signal analysis
Interactive profiling
Automated testing and customization via scripting
Visual project management system
Compile in the background while editing and debugging
Multi-processor debugging
Help on the target DSP
Procedure to work on Code Composer Studio

1. To create a New Project
Project New (SUM.pjt)


33

2. To Create a Source file
File New

type the code (Save & give a name to file, Eg: sum.c).

3. To Add Source files to Project
Project Add files to Project sum.c


34

4. To Add rts6700.lib file & hello.cmd:

Project Add files to Project rts6700.lib
Path: c:\CCStudio\c6000\cgtools\lib\rts6700.lib

35

Note: Select Object & Library in(*.o,*.l) in Type of files

Project Add files to Project hello.cmd
Path: c:\ti\tutorial\dsk6713\hello1\hello.cmd
Note: Select Linker Command file(*.cmd) in Type of files

5. To Compile:
Project Compile File

6. To build or Link:
Project build,
Which will create the final executable (.out) file.(Eg. sum.out).

7. Procedure to Load and Run program:
Load program to DSK:
File Load program sum. out

8. To execute project:
Debug Run.

TMS320C6713 is a floating point DSP processor.Basically it is a cisc processor.It has
following features
*) It can execute eight 32-bit instructions/cycle
*) Advanced Very Long Instruction Word DSP core.
*) It has eight independent functional units.In which there are 2 ALUs of fixed point, 4 ALUs
of Floating and fixed point and 2 multipliers of floating and fixed point.
*) Two multi channel audio ports
*) 32 bit External memory Interface.
DSP Starter kit consists of TMS320C6713 ,CPLD,Flash Memory,SDRAM,Codec.

Code Composer Studio

36

Code Composer is the Integrated Development Environment(IDE) with DSP specific
functionality.Basically code composer studio is a cross compiler.That means the programs
written in Higher level languages like Care converted into processor specific languages.

Procedure to work with code composer studio
Open the code composer studio,make sure DSP Starter Kit(DSK) is turned on.
Create a new project as follows
Go to Project->new and click on it , give project_name ,Select target as TMS320C67xx.
Then save it in C:\CCStudio\MyProjects
Then go Project -> Add Files to Project and click on it , then add the run time support library
C:\CCStudio\c6000\cgtools\lib\rts6700.lib
Then go Project-> Add Files to Project and click on it , then add the linker command file
C:\CCStudio\tutorial\dsk6713\hello1\hello.cmd
Then compile the program using Project->compile->click
Then go to Project -> Rebuild All.
Then go to File->Load program -> project_name.out into the DSP Starter kit.
How to view graphical output

To view the graphical output go to View Graph Time/Frequency

1 C PROGRAM TO IMPLEMENT LINEAR CONVOLUTION

/* prg to implement linear convolution */
#include<stdio.h>

#define LENGHT1 6 /*Lenght of i/p samples sequence*/
#define LENGHT2 4 /*Lenght of impulse response Co-efficients */

int x[2*LENGHT1-1]={1,2,3,4,5,6,0,0,0,0,0}; /*Input Signal Samples*/
int h[2*LENGHT1-1]={1,2,3,4,0,0,0,0,0,0,0}; /*Impulse Response Co-efficients*/

int y[LENGHT1+LENGHT2-1];

main()
{
int i=0,j;

for(i=0;i<(LENGHT1+LENGHT2-1);i++)
{
y[i]=0;
for(j=0;j<=i;j++)

y[i]+=x[j]*h[i-j];

}
for(i=0;i<(LENGHT1+LENGHT2-1);i++)
printf("%d\n",y[i]);

}


37

PROCEDURE:

Open Code Composer Studio, make sure the DSP kit is turned on.

Start a new project using Project-new pull down menu, save it in a
separate directory(c:\ti\myprojects) with name lconv.pjt.

Add the source files conv.c
to the project using Projectadd files to project pull down menu.

Add the linker command file hello.cmd .
(Path: c:\ti\tutorial\dsk6713\hello1\hello.cmd)

Add the run time support library file rts6700.lib
(Path: c:\ti\c6000\cgtools\lib\rts6700.lib)

Compile the program using the Project-compile pull down menu or by
clicking the shortcut icon on the left side of program window.

Build the program using the Project-Build pull down menu or by

Load the program(lconv.out) in program memory of DSP chip using the
File-load program pull down menu.

To View output graphically
Select view graph time and frequency.
Configure the graphical window as shown below


38


39

2 program to Implement Circular Convolution
#include<stdio.h>
int m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],a[30];
void main()
{
printf(" enter the length of the first sequence\n");
scanf("%d",&m);
printf(" enter the length of the second sequence\n");
scanf("%d",&n);
printf(" enter the first sequence\n");
for(i=0;i<m;i++)
scanf("%d",&x[i]);
printf(" enter the second sequence\n");
for(j=0;j<n;j++)
scanf("%d",&h[j]);
if(m-n!=0) /*If length of both sequences are not equal*/
{
if(m>n) /* Pad the smaller sequence with zero*/
{
for(i=n;i<m;i++)
h[i]=0;
n=m;
}
for(i=m;i<n;i++)
x[i]=0;
m=n;
}
y[0]=0;
a[0]=h[0];
for(j=1;j<n;j++) /*folding h(n) to h(-n)*/
a[j]=h[n-j];
/*Circular convolution*/
for(i=0;i<n;i++)
y[0]+=x[i]*a[i];
for(k=1;k<n;k++)
{
y[k]=0;
/*circular shift*/
for(j=1;j<n;j++)
x2[j]=a[j-1];
x2[0]=a[n-1];
for(i=0;i<n;i++)
{
a[i]=x2[i];
y[k]+=x[i]*x2[i];
}
}
/*displaying the result*/
printf(" the circular convolution is\n");
for(i=0;i<n;i++)

40

printf("%d \t",y[i]);

}

PROCEDURE:

Open Code Composer Studio, make sure the DSP kit is turned on.

Start a new project using Project-new pull down menu, save it in a
separate directory(c:\ti\myprojects) with name cir conv.pjt.

Add the source files Circular Convolution.C
to the project using Projectadd files to project pull down menu.

Add the linker command file hello.cmd .
(Path: c:\ti\tutorial\dsk6713\hello1\hello.cmd)

Add the run time support library file rts6700.lib
(Path: c:\ti\c6000\cgtools\lib\rts6700.lib)

Compile the program using the Project-compile pull down menu or by

Build the program using the Project-Build pull down menu or by

Load the program(lconv.out) in program memory of DSP chip using the
File-load program pull down menu.

DFT USING DSP KIT

#include <stdio.h>
#include <math.h>
float x[4]={1,3,2,5}; //input only real sequence

main()
{
float y[16]; //for 8 point DFT to store real & imaginary
float w;
int n,k,k1,N=8,xlen=4;
for(k=0;k<2*N;k=k+2)
{
y[k]=0;
y[k+1]=0; //initialize real & imag parts
k1=k/2; //actual k index
for(n=0;n<xlen;n++)

41

{
w=-2*3.14*k1*n/N; //careful about minus sign
y[k]=y[k]+x[n]*cos(w);
y[k+1]=y[k+1]+x[n]*sin(w);
}
printf("%f+j%f \n",y[k],y[k+1]);
}
}//end of main

Procedure for Real time Programs :

1. Connect CRO to the Socket Provided for LINE OUT.

2. Connect a Signal Generator to the LINE IN Socket.

3. Switch on the Signal Generator with a sine wave of frequency 500 Hz. and Vp-p=1.5v

4. Now Switch on the DSK and Bring Up Code Composer Studio on the PC.

5. Create a new project with name codec.pjt.

6. From the File Menu new DSP/BIOS Configuration select
dsk6713.cdb and save it as xyz.cdb


42

7. Add xyz.cdb to the current project.

8. Add the given codec.c file to the current project which has the main function and
calls all the other necessary routines.

9. Add the library file dsk6713bsl.lib to the current project

Path C:\CCStudio\C6000\dsk6713\lib\dsk6713bsl.lib

10. Copy files dsk6713.h and dsk6713_aic23.h from
C:\CCStudio\C6000\dsk6713\include and paste it in current project.

11. Build, Load and Run the program.

12. You can notice the input signal of 500 Hz. appearing on the CRO verifying the codec
configuration.

13. You can also pass an audio input and hear the output signal through the speakers.

14. You can also vary the sampling frequency using the DSK6713_AIC23_setFreq
Function in the codec.c file and repeat the above steps.

43

5.0 Conclusion:
The codec TLV320AIC23 successfully configured using the board support library
and verified.

codec.c

#include "xyzcfg.h"

#include "dsk6713.h"
#include "dsk6713_aic23.h"

/* Codec configuration settings */
DSK6713_AIC23_Config config = { \
0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Left line input channel volume */ \
0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume */\
0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */ \
0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */ \
0x0011, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */ \
0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */ \
0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */ \
0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \
0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */ \
0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \
};

/* main() - Main code routine, initializes BSL and generates tone */

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
int l_input, r_input,l_output, r_output;

/* Initialize the board support library, must be called first */
DSK6713_init();

/* Start the codec */
hCodec = DSK6713_AIC23_openCodec(0, &config);

/*set codec sampling frequency*/
DSK6713_AIC23_setFreq(hCodec, 3);

while(1)
{
/* Read a sample to the left channel */
while (!DSK6713_AIC23_read(hCodec, &l_input));

/* Read a sample to the right channel */

44

while (!DSK6713_AIC23_read(hCodec, &r_input));

/* Send a sample to the left channel */
while (!DSK6713_AIC23_write(hCodec, l_input));

/* Send a sample to the right channel */
while (!DSK6713_AIC23_write(hCodec, l_input));
}

/* Close the codec */
DSK6713_AIC23_closeCodec(hCodec);
}

USING MATLAB TO DETERMINE FILTER COEFFICIENTS :
Using FIR1 Function on Matlab
B = FIR1(N,Wn) designs an N'th order lowpass FIR digital filter and returns the filter
coefficients in length N+1 vector B.
The cut-off frequency Wn must be between 0 < Wn < 1.0, with 1.0 corresponding to half the
sample rate. The filter B is real and has linear phase, i.e., even symmetric coefficients obeying
B(k) = B(N+2-k), k = 1,2,...,N+1.
If Wn is a two-element vector, Wn = [W1 W2], FIR1 returns an order N bandpass filter with
passband W1 < W < W2.
B = FIR1(N,Wn,'high') designs a highpass filter.
B = FIR1(N,Wn,'stop') is a bandstop filter if Wn = [W1 W2].
4. IMPLEMENTATION OF AN FIR FILTER : ALGORITHM TO IMPLEMENT :
We need to realize an advance FIR filter by implementing its difference equation as per the
specifications. A direct form I implementation approach is taken. (The filter coefficients are
taken as a
i
as generated by the Matlab program.)

C PROGRAM TO IMPLEMENT FIR FILTER:

fir.c


45

#include "filtercfg.h"

#include "dsk6713.h"
#include "dsk6713_aic23.h"

float filter_Coeff[] ={0.000000,-0.001591,-0.002423,0.000000,0.005728,
0.011139,0.010502,-0.000000,-0.018003,-0.033416,-0.031505,0.000000,
0.063010,0.144802,0.220534,0.262448,0.220534,0.144802,0.063010,0.000000,
-0.031505,-0.033416,-0.018003,-0.000000,0.010502,0.011139,0.005728,
0.000000,-0.002423,-0.001591,0.000000 };

static short in_buffer[100];

DSK6713_AIC23_Config config = {\
0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Leftline input channel volume */\
0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume*/\
0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */\
0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */\
0x0011, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */\
0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */\
0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */\
0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */\
0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */\
0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \
};

/*
* main() - Main code routine, initializes BSL and generates tone
*/

void main()
{
DSK6713_AIC23_CodecHandle hCodec;

Uint32 l_input, r_input,l_output, r_output;

/* Initialize the board support library, must be called first */
DSK6713_init();

/* Start the codec */
hCodec = DSK6713_AIC23_openCodec(0, &config);

DSK6713_AIC23_setFreq(hCodec, 1);

while(1)
{ /* Read a sample to the left channel */
while (!DSK6713_AIC23_read(hCodec, &l_input));

/* Read a sample to the right channel */
while (!DSK6713_AIC23_read(hCodec, &r_input));

l_output=(Int16)FIR_FILTER(&filter_Coeff ,l_input);
r_output=l_output;

/* Send a sample to the left channel */
while (!DSK6713_AIC23_write(hCodec, l_output));


46

/* Send a sample to the right channel */
while (!DSK6713_AIC23_write(hCodec, r_output));
}

/* Close the codec */
DSK6713_AIC23_closeCodec(hCodec);
}

signed int FIR_FILTER(float * h, signed int x)
{
int i=0;
signed long output=0;

in_buffer[0] = x; /* new input at buffer[0] */

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1]; /* shuffle the buffer */

for(i=0;i<32;i++)
output = output + h[i] * in_buffer[i];

return(output);

}

PROCEDURE :

Switch on the DSP board.
Open the Code Composer Studio.
Create a new project
Project New (File Name. pjt , Eg: FIR.pjt)
Initialize on board codec.

Note: Kindly refer the Topic Configuration of 6713 Codec using BSL

Add the given above C source file to the current project (remove codec.c source file from the
project if you have already added).
Connect the speaker jack to the input of the CRO.
Build the program.
Load the generated object file(*.out) on to Target board.
Run the program
Observe the waveform that appears on the CRO screen.
Vary the frequency on function generator to see the response of filter.

Lab Manual DSP Lab 5 Sem

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Lab Manual DSP Lab 5 Sem

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Dr H.S.

Prashantha, Dept of ECE, PESIT

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