DIGITAL SIGNAL PROCESSING

LABORATORY MANUAL
III – B. Tech., II-Semester ECE

DEPARTMENT OF ELECTRONICS AND COMMUNICATION ENGINEERING

UNIVERSITY COLLEGE OF ENGINEERING
ADIKAVI NANNAYA UNIVERSITY
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

DIGITAL SIGNAL PROCESSING LAB

(R13) III – B. Tech., ECE II- Semester

Index

S. No. Name of the

Experiment

1. Study of architecture of DSP chips – TMS 320C 5X/6X Instructions.

2. Verification of Linear convolution.

3. Verification of Circular convolution.

Design of FIR filter (LP/HP) using windowing technique
4. a) Using rectangular window b) Using triangular window c) Using
Kaiser window
5. Implementation of IIR filter (LP/HP) on DSP Processors

6. Implementation of N-point FFT algorithm.

7. MATLAB program to generate sum of sinusoidal signals.

8. MATLAB program to find IIR frequency response of analog LP/HP filters.

9. Computation of Power Density Spectrum of a Sequence.

10 Computation of the FFT of given 1-D signal.

.
Frequency responses of anti-imaging and anti-aliasing filters.
11
.

Department of Electronics & Communication 2

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

GENERAL INSTRUCTIONS:

1. The experiments have been designed to be performed with in the 3-

hour laboratory time.
2. To successfully complete the experiment in one lab, turn, come
prepared to the laboratory.
3. Read the experiment in advance.
4. List and collect the components for the experiment.
5. Be sure that the specifications and values of the components are
as per design.
6. Follow the experimental steps judiciously.
7. Record stepwise observations using proper test instruments.
8. Get the observation signed by the instructor.
9. Always take safety precautions while performing experiments.

GUIDANCE FOR THE LABORATORY REPORT:

1. Format of the report
Exp. No: Date:
Expt. Title:
Objective:
List of instruments and components:
Theory in brief
Procedure, Observations, Graph if
any Result
2. Write the experimental observations and measurements stepwise.
3. Plot the graph neatly. Always label the axes and indicate units too.
Wherever frequency response is to be drawn, use the semi-log
graph paper.
4. Compare the results with theoretical values with remarks/comments.
5. Wherever necessary, sketch the circuit diagram neatly and label the
components.

Department of Electronics & Communication 3

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

WORKING PROCEDURE WITH MATLAB:

1)Double click on Matlab icon. -> Then Matlab will be opened

2)To write the Matlab Program Goto file menu-> New -> Script(Mfile) -> In the
opened Script file write the Matlab code and save the file with an extension of .m
Ex: “linear.m”
3) To execute Matlab Program Select the all lines in matlab
program(ctrl+A) of mfile and press “F9” to execute the matlab code
4) Entering the inputs in command window
 If the command window is displaying the message like “enter the input
sequence” then enter the sequence with square brackets and each sample
values is spaced with single space
Ex: Enter input sequence [1 2 3 4]
 If it is asking a value input write the value without
brackets Ex: “enter length of sequence 4”
 After entering inputs It displays the Output Graphs.

PROCEDURE TO WORK ON CODE COMPOSER STUDIO

PROCEDURE FOR EXECUTING NON REAL TIME
PROGRAMS
(EX: LINEAR & CIRCULAR CONVOLUTION, FFT, PSD)

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

Use StartProgramsTexas InstrumentsCode Composer StudioCode Composer

Studio CDSK6713 Tools DSK6713 Diagnostic Utilities

 Select StartSelect DSK6713 Diagnostic Utility Icon from Desktop

 Select Start Option
 Utility Program will test the board
 After testing Diagnostic Status, you will get PASS

To create the New Project

Project  New (File Name. pjt , Eg: Vectors.pjt)

To Create a Source file

File  New  Type the code (Save & give file name, Eg: sum.c).

To Add Source files to Project

Project  Add files to Project  c/ccs studio3.1/my projects/your project name/
sum.c(select the file type as c/c++

source files) To Add rts.lib file &

hello.cmd:

Project  Add files to Project rts6700.lib

(Path:c/ccs studio3.1/cg tools/c6000/lib/
rts6700.lib) Note: Select Object & Library
in(*.o,*.l) in Type of files Project  Add files to
Project hello.cmd
CMD file – Which is common for all non-real time programs.
(Path: c/ccs studio3.1\tutorial\dsk 6713 \hello1\
hello.cmd) Note: Select Linker Command file(*.cmd) in Type of files

Compile:
To Compile: Project  Compile project
To Build: Project  build project,
To Rebuild: Project  rebuild,

Department of Electronics & Communication 4

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
Which will create the final .out executable file. (Eg. Vectors.out).

Department of Electronics & Communication 5

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Procedure to Load and Run program:

Load the program to DSK: File  Load program  Vectors.
out To Execute project: Debug  Run.
1. Execution should halt at break point.
2. Now press F10. See the changes happening in the watch window.
3. Similarly go to view & select CPU registers to view the changes happening in CPU
registers.
Configure the graphical window as shown below
INPUT
x[n] = {1, 2, 3, 4,0,0,0,0}
h[k] = {1, 2, 3, 4,0,0,0,0}
OUTPUT:

b) PROCEDURE FOR EXECUTING REAL TIME PROGRAMS (EX:IIR

FILTERS,FIR FILTERS DESIGNING)

CONNECTING DSP PROCESSOR TO PC

 Connect the dsp processor to the pc using usb cable connector.
 Check the DSK6713 diagnostics (IF you get the “pass”then click on ok).
 Click on ccs studio3.1 desktop icon. Then the window will be opened.
 Go to debug click on connect (then target device will be connected to pc)
TO CREATE PROJECT
 Project new given project name and select the family’TMS320C67XX’Then click ok
 File new source file write deown the ‘c’program and save it with.’c’
exetention in current project file
 File new dsp/bios.config file select dsk67xx click on dsk6713 and save it in
current project.
 Project add files to project add source file
 Project add files to project add library file by following the given path
 c/ccs studio3.1/cgtools/c6000/dsk6713/DSK6713.bs/file.
 Project add files to the project. Add the configuration file.
 Now files are generated and included in generated files. in that open the 3rd file,
and copy the header file and paste it in source file. Copy the include files named
as”dsk6713.h” and “dsk6713_aic23.h” paste it in current project folder.
 Now compile project. (project compile)
 Project build.
 Project rebuild all.
 File load program projectname.pjt debug “project name .out” file click on
open debug click on run
 Now apply the input sine wave to line in of dsk6713 kit.
 Observe the output at line out of dsk6713 by using CRO.
Department of Electronics & Communication 6
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT-1: STUDY OF ARCHITECTURE OF DSP CHIPS-TMS 320C

5X/6X

AIM: To study the Architecture of DSP chips-TMS 320c 5x/6xinstructions.

INTRODUCTION TO DSP PROCESSORS

A signal can be defined as a function that conveys information, generally about

the state or behavior of a physical system. There are two basic types of signals viz.
Analog (continuous time signals which are defined along a continuum of times) and
Digital (discrete-time). Remarkably, under reasonable constraints; a continuous time
signal can be adequately represented by samples, obtaining discrete time signals. Thus
digital signal processing is an ideal choice for anyone who needs the performance
advantage of digital manipulation along with today’s analog reality. Hence a processor
which is designed to perform the special operations (digital manipulations) on the digital
signal within very less time can be called as a Digital signal processor. The difference
between a DSP processor, conventional microprocessor and a microcontroller are listed
below.

Microprocessor or General Purpose Processor such as Intel xx86 or Motorola

680xx family Contains - only CPU
-No RAM
-No ROM
-No I/O ports
-No Timer

Microcontroller such as 8051

family Contains - CPU
- RAM
- ROM
-I/O ports
- Timer &
- Interrupt circuitry
Some Micro Controllers also contain A/D, D/A and Flash Memory

DSP Processors such as Texas instruments and

Analog Devices Contains - CPU

- RAM
-ROM
- I/O ports
- Timer

Optimized for – fast arithmetic

- Extended precision
- Dual operand fetch
- Zero overhead loop
- Circular buffering

The basic features of a DSP Processor are

Feature Use
Fast-Multiply accumulate Most DSP algorithms, including filtering, transforms, etc. are multiplication- intensive
Multiple – access memory Many data-intensive DSP operations require reading a program instruction and multiple
data items
Specialized addressing Efficient handling of data arrays and first-in, first-out buffers in memory
modes
Specialized program Efficient control of loops for many iterative DSP algorithms. Fast interrupt handling for
control frequent I/O
On-chip peripherals and I/O On-chip peripherals like A/D converters allow for small low cost system designs. Similarly

Department of Electronics & Communication 7

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
I/O

Department of Electronics & Communication 8

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

ARCHITECTURE OF 6713 DSP PROCESSOR

This chapter provides an overview of the architectural structure of the TMS320C67xx

DSP, which comprises the central processing unit (CPU), memory, and on-chip
peripherals. The C67xE DSPs use an advanced modified Harvard architecture that
maximizes processing power with eight buses. Separate program and data spaces allow
simultaneous access to program instructions and data, providing a high degree of
parallelism. For example, three reads and one write can be performed in a single cycle.
Instructions with parallel store and application-specific instructions fully utilize this
architecture. In addition, data can be transferred between data and program spaces.
Such Parallelism supports a powerful set of arithmetic, logic, and bit-manipulation
operations that can all be performed in a single machine cycle. Also, the C67xx DSP
includes the control mechanisms to manage interrupts, repeated operations, and
function calling.

Bus Structure

The C67xx DSP architecture is built around eight major 16-bit buses (four program/data
buses and four address buses):
_ The program bus (PB) carries the instruction code and immediate operands from program
memory.
_ Three data buses (CB, DB, and EB) interconnect to various elements, such as the CPU, data
address generation logic, program address generation logic, on-chip peripherals, and data
memory.
_ The CB and DB carry the operands that are read from data memory.
_ The EB carries the data to be written to memory.
_ Four address buses (PAB, CAB, DAB, and EAB) carry the addresses needed for
instruction execution.
Department of Electronics & Communication 9
Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

The C67xx DSP can generate up to two data-memory addresses per cycle using the two
auxiliary register arithmetic units (ARAU0 and ARAU1). The PB can carry data operands
stored in program space (for instance, a coefficient table) to the multiplier and adder for
multiply/accumulate operations or to a destination in data space for data move
instructions (MVPD and READA). This capability, in conjunction with the feature of dual-
operand read, supports the execution of single- cycle, 3-operand instructions such as the
FIRS instruction. The C67xx DSP also has an on-chip bidirectional bus for accessing on-
chip peripherals. This bus is connected to DB and EB through the bus exchanger in the
CPU interface. Accesses that use this bus can require two or more cycles for reads and
writes, depending on the peripheral’s structure.
Central Processing Unit (CPU)
The CPU is common to all C67xE devices. The C67x CPU contains:

_ 40-bit arithmetic logic unit (ALU)

_ Two 40-bit accumulators
_ Barrel shifter
_ 17 × 17-bit multiplier
_ 40-bit adder
_ Compare, select, and store unit (CSSU)
_ Data address generation unit
_ Program address generation unit

Arithmetic Logic Unit (ALU)

The C67x DSP performs 2s-complement arithmetic with a 40-bit arithmetic logic unit
(ALU) and two 40-bit accumulators (accumulators A and B). The ALU can also perform
Boolean operations. The ALU uses these inputs:
_ 16-bit immediate value
_ 16-bit word from data memory
_ 16-bit value in the temporary register, T
_ Two 16-bit words from data memory
_ 32-bit word from data memory
_ 40-bit word from either accumulator
The ALU can also function as two 16-bit ALUs and perform two 16-bit operations simultaneously.

Fig 2 – 2 ALU UNIT

Department of Electronics & Communication 10

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
Accumulators
Accumulators A and B store the output from the ALU or the multiplier/adder block. They
can also provide a second input to the ALU; accumulator A can be an input to the
multiplier/adder. Each accumulator is divided into three parts:
_ Guard bits (bits 39–32)
_ High-order word (bits 31–16)
_ Low-order word (bits 15–0)
Instructions are provided for storing the guard bits, for storing the high- and the low-
order accumulator words in data memory, and for transferring 32-bit accumulator words
in or out of data memory. Also, either of the accumulators can be used as temporary
storage for the other.

Barrel Shifter

The C67x DSP barrel shifter has a 40-bit input connected to the accumulators or to data
memory (using CB or DB), and a 40-bit output connected to the ALU or to data memory
(using EB). The barrel shifter can produce a left shift of 0 to 31 bits and a right shift of 0
to 16 bits on the input data. The shift requirements are defined in the shift count field of
the instruction, the shift count field (ASM) of status register ST1, or in temporary register
T (when it is designated as a shift count register). The barrel shifter and the exponent
encoder normalize the values in an accumulator in a single cycle. The LSBs of the output
are filled with 0s, and the MSBs can be either zero filled or sign extended, depending on
the state of the sign-extension mode bit (SXM) in ST1. Additional shift capabilities enable
the processor to perform numerical scaling, bit extraction, extended arithmetic, and
overflow prevention operations.

Multiplier/Adder Unit

The multiplier/adder unit performs 17 _ 17-bit 2s-complement multiplication with a 40-bit

addition in a single instruction cycle. The multiplier/adder block consists of several
elements: a multiplier, an adder, signed/unsigned input control logic, fractional control
logic, a zero detector, a rounder (2s complement), overflow/saturation logic, and a 16-bit
temporary storage register (T). The multiplier has two inputs: one input is selected from
T, a data-memory operand, or accumulator A; the other is selected from program
memory, data memory, accumulator A, or an immediate value. The fast, on- chip
multiplier allows the C54x DSP to perform operations efficiently such as convolution,
correlation, and filtering. In addition, the multiplier and ALU together execute
multiply/accumulate (MAC) computations and ALU operations in parallel in a single
instruction cycle. This function is used in determining the Euclidian distance and in
implementing symmetrical and LMS filters, which are required for complex DSP
algorithms. See section 4.5, Multiplier/Adder Unit, on page 4-19, for more details about
the multiplier/adder unit.

Department of Electronics & Communication 11

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Fig 2 – 3 MULTIPLIER/ADDER UNIT

Department of Electronics & Communication 12

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

These are the some of the important parts of the processor and you are instructed to go
through the detailed architecture once which helps you in developing the optimized code
for the required application.

RESULT: The architecture of DSP chips-TMS 320c 5x/6x is studied successfully.

VIVA QUESTIONS

1. Define signal and signal processing

2. Differentiate digital and analog signals?
3. How the DSP processor will differ from conventional processors?
4. Expand the abbreviation TMS320C 5X/6X
5. What kind of processor is DSP processor?
6. What are the main building blocks of DSP processor?
7. What is the main function of MAC unit?
8. Explain VLIW architecture?
9. What is meant by circular buffer?
10. What is meant by emulator and JTAG?

Department of Electronics & Communication 13

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT-2:
VERIFICATION OF LINEAR CONVOLUTION USING MATLAB AND CC STUDIO

AIM: To verify Linear Convolution using MATLAB AND CC STUDIO.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
DSK6713 DSP Starter kit CCS Studio 3-1
USB Cable OS: Windows XP
Power Adapter

THEORY:

Convolution is a formal mathematical operation, just as multiplication, addition, and

integration. Addition takes two numbers and produces a third number, while convolution
takes two signals and produces a third signal. Convolution is used in the mathematics of
many fields, such as probability and statistics. In linear systems, convolution is used to
describe the relationship between three signals of interest: the input signal, the impulse
response, and the output signal.

In this equation, x1(k), x2(n-k) and y(n) represent the input to and output from the system
at time n. Here we could see that one of the inputs is shifted in time by a value every
time it is multiplied with the other input signal. Linear Convolution is quite often used as a
method of implementing filters of various types.

Linear Convolution Using

MATLAB: - Program:
clc;
clear all;
close all;
disp('linear convolution program');
x=input('enter i/p x(n):');
m=length(x);
h=input('enter i/p h(n):');
n=length(h);
x=[x,zeros(1,n)];
subplot(2,2,1), stem(x);
title('i/p sequence x(n)is:');
xlabel('---->n');
ylabel('---->x(n)');grid;
h=[h,zeros(1,m)];
subplot(2,2,2), stem(h);
title('i/p sequence h(n)is:');
xlabel('---->n');
ylabel('---->h(n)');grid;
disp('convolution of x(n) & h(n) is y(n):');
y=zeros(1,m+n-1);
for i=1:m+n-1
y(i)=0;
for j=1:m+n-1
if(j<i+1)
y(i)=y(i)+x(j)*h(i-j+1);
end

Department of Electronics & Communication 14

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

end
end
y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is :');
xlabel('---->n');
ylabel('---->y(n)');grid;

Output :

Linear Convolution Using

CCSTUDIO: - Procedure to create
new Project:
1.To create project, go to Project and Select New.

Department of Electronics & Communication 15

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

2.Give project name and click on finish

(Note: Location must be c:\CCStudio_v3.1\MyProjects)

3.Click on File New Source File, to write the Source Code

Department of Electronics & Communication 16

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Mathematical Formula:
The linear convolution of two continuous time signals x(t) and h(t) is defined by

For discrete time signals x(n) and h(n), is defined by

Where x(n) is the input signal and h(n) is the impulse response of the

system. In linear convolution length of output sequence is,

Length (y(n)) = length(x(n)) + length(h(n)) – 1
Program:
#include<stdio.h>
main()
{ int m=4; /*Lenght of i/p samples sequence*/
int n=4; /*Lenght of impulse response Co-
efficients */ int i=0,j;
int x[10]={1,2,3,4,0,0,0,0}; /*Input Signal Samples*/
int h[10]={1,2,3,4,0,0,0,0}; /*Impulse Response Co-efficients*/
/*At the end of input sequences pad 'M' and 'N' no. of zero's*/
int *y;
y=(int
*)0x0000100;
for(i=0;i<m+n-
1;i++)
{ y[i]
=0;
for(j=0;j<=i;j
++) y[i]
+=x[j]*h[i-j];
}
for(i=0;i<m+n-
1;i++) printf("%d\
n",y[i]);
}

Output:
1, 4, 10, 20, 25, 24, 16.
4.Enter the source code and save the file with “.C” extension.

Department of Electronics & Communication 17

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Department of Electronics & Communication 18

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

5.Right click on source, Select add files to project and Choose “.C “ file Saved before.

6.Right Click on libraries and select add files to Project

and choose C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib
and click open.

7.a) Go to Project to Compile .

b) Go to Project to Build.
c) Go to Project to Rebuild All.

Department of Electronics & Communication 19

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

8.Go to file and load program and load “.out” file into the board..

9.Go to Debug and click on run to run the program.

Department of Electronics & Communication 20

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

10. Observe the output in output window.

Department of Electronics & Communication 21

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
11. To see the Graph go to View and select time/frequency in the Graph, And
give the correct Start address provided in the program, Display data can be taken as
per user.

12. Green line is to choose the point, Value at the point can be seen
(Highlighted by circle at the left corner).

RESULT: Hence Linear Convolution is verified successfully using MATLAB and CC Studio.

VIVA QUESTIONS
1. Explain the significance of convolution.
2. Define linear convolution.
3. Why linear convolution is called as a periodic convolution?
4. Why zero padding is used in linear convolution?
5. What are the four steps to find linear convolution?
6. What is the length of the resultant sequence in linear convolution?
7. How linear convolution will be used in calculation of LTI system response?
8. List few applications of linear convolution in LTI system design.
9. Give the properties of linear convolution.
10. How the linear convolution will be used to calculate the DFT of a signal?

Department of Electronics & Communication 22

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-3:
VERIFICATION OF CIRCULAR CONVOLUTION USING MATLAB AND CC STUDIO

AIM: To verify Circular Convolution using MATLAB AND CC STUDIO.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
DSK6713 DSP Starter kit CCS Studio 3-1
USB Cable OS: Windows XP
Power Adapter

THEORY
Circular convolution is another way of finding the convolution sum of two input signals. It
resembles the linear convolution, except that the sample values of one of the input
signals is folded and right shifted before the convolution sum is found. Also note that
circular convolution could also be found by taking the DFT of the two input signals and
finding the product of the two frequency domain signals. The Inverse DFT of the product
would give the output of the signal in the time domain which is the circular convolution
output. The two input signals could have been of varying sample lengths. But we take the
DFT of higher point, which ever signals levels to. For example, If one of the signal is of
length 256 and the other spans 51 samples, then we could only take 256 point DFT. So
the output of IDFT would be containing 256 samples instead of 306 samples, which
follows N1+N2 – 1 where N1 & N2 are the lengths 256 and 51 respectively of the two
inputs. Thus the output which should have been 306 samples long is fitted into 256
samples. The256 points end up being a distorted version of the correct signal. This
process is called circular convolution.

Circular Convolution using

MATLAB:- Program:
clc;
clear all;
close all;
disp('circular convolution program');
x=input('enter i/p x(n):');
m=length(x);
h=input('enter i/p sequence h(n)');
n=length(h);
subplot(2,2,1),
stem(x);
title('i/p sequence x(n)is:');
xlabel('---->n');
ylabel('---->x(n)');
grid;
subplot(2,2,2),
stem(h);
title('i/p sequence h(n)is:');
xlabel('---->n');
ylabel('---->h(n)');
grid;
disp('circular convolution of x(n) & h(n) is y(n):');
if(m-n~=0)
if(m>n)
h=[h,zeros(1,m-n)];
n=m;
end
x=[x,zeros(1,n-m)];
m=n;
Department of Electronics & Communication 23
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
end
y=zeros(1,n);
y(1)=0;
a(1)=h(1);
for j=2:n
a(j)=h(n-j+2);
end
%circular convolution
for i=1:n
y(1)=y(1)+x(i)*a(i);
end
for k=2:n
y(k)=0;
% circular shift
for j=2:n
x2(j)=a(j-1);
end
x2(1)=a(n);
for i=1:n
if(i<n+1)
a(i)=x2(i);
y(k)=y(k)+x(i)*a(i);
end
end
end
y
subplot(2,2,[3,4]),stem(y);
title('convolution of x(n) & h(n) is:');
xlabel('---->n');
ylabel('---->y(n)');
grid;

Department of Electronics & Communication 24

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT :

Circular Convolution using

CCStudio :- Procedure to create

new Project:
1.To create project, go to Project and Select New.

2.Give project name and click on finish.

Department of Electronics & Communication 25

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

( Note: Location must be c:\CCStudio_v3.1\MyProjects ).

3.Click on File New Source File, to write the Source Code.

Circular Convolution:
Let x1(n) and x2(n) are finite duration sequences both of length N with DFT’s
X1(k) and X2(k). Convolution of two given sequences x1(n) and x2(n) is given by
the equation,

x3(n) = IDFT[X3(k)]

X3(k) = X1(k) X2(k)

N-1
x3(n) = ∑ x1(m) x2((n-
m))N m=0

Program:

#include<stdio.h>
int
m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],
a[30]; void main()
{
int *y;
y=(int *)0x0000100;
printf(" enter the length of the first sequence\n");
Department of Electronics & Communication 26
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

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++)
printf("%d ",y[i]);
}

Output:

enter the length of the first

sequence 4
enter the length of the second
sequence 4
enter the first
sequence 4 3 2 1
enter the second

Department of Electronics & Communication 27

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
sequence 1 1 1 1

Department of Electronics & Communication 28

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

the circular
convolution is 10 10 10
10

4.Enter the source code and save the file with “.C” extension.

5.Right click on source, Select add files to project .. and Choose “.C “ file Saved before.

6.Right Click on libraries and select add files to Project..

and choose C:\CCStudio_v3.1\C6000\cgtools\lib\rts6700.lib
and click open.

Department of Electronics & Communication 29

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

7.a) Go to Project to Compile .

b) Go to Project to Build.
c) Go to Project to Rebuild All.

Department of Electronics & Communication 30

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

8.Go to file and load program and load “.out” file into the board..

9.Go to Debug and click on run to run the program.

10. Enter the input data to calculate the circular convolution.

Department of Electronics & Communication 31

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Department of Electronics & Communication 32

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

The corresponding output will be shown on the output window as shown below

11. To see the Graph go to View and select time/frequency in the Graph, and
give the correct Start address provided in the program, Display data can be taken as
per user.

Department of Electronics & Communication 33

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

12. Green line is to choose the point, Value at the point can be seen
(Highlighted by circle at the left corner).

RESULT: Hence Circular Convolution is verified successfully using MATLAB and CC Studio.

VIVA QUESTIONS
1. Give mathematical definition of circular convolution
2. Why circular convolution is called as periodic convolution?
3. Difference between linear convolution and circular convolution
4. Explain the circular shift
5. How circular convolution is used to calculate the Z-transform of a signal?
6. List few Applications of circular convolution
7. What are the different methods used to calculate circular convolution?
8. Explain properties of circular convolution?
9. Explain modulo N operation
10. What is the importance of circular convolution to realization of digital
systems or digital filters?

Department of Electronics & Communication 34

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-4:
DESIGN OF FIR FILTER (LP/HP) USING WINDOWING TECHNIQUE
AIM: To design FIR filters (LP/HP) by using following windowing techniques on MATLAB
and DSK6713 KIT:
1)Rectangular 2) Triangular 3) Kaiser

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
DSK6713 DSP Starter kit CCS Studio 3-1
USB Cable OS: Windows XP
Power
Adapter Jack
Cables
CRO, Probes, Function generator

THEORY:
A Finite Impulse Response (FIR) filter is a discrete linear time-invariant system whose
output is based on the weighted summation of a finite number of past inputs. An FIR
transversal filter structure can be obtained directly from the equation for discrete-
time convolution.
N 1

y(n)   x(k)h(n  0  n  N 1
k)
k 0
In this equation, x(k) and y(n) represent the input to and output from the filter at time
n. h(n-k) is the transversal filter coefficients at time n. These coefficients are
generated by using FDS (Filter Design Software or Digital filter design package).

FIR – filter is a finite impulse response filter. Order of the filter should be specified.
Infinite response is truncated to get finite impulse response. Placing a window of
finite length does this. Types of windows available are Rectangular, Bartlett,
Hamming, Hanning, Blackmann window etc., This FIR filter is an all zero filter.

%fir filt design window techniques %

clc;
clear all;
close all;
rp=input('enter passband ripple');
rs=input('enter the stopband ripple');
fp=input('enter passband freq');
fs=input('enter stopband freq');
f=input('enter sampling freq ');
wp=2*fp/f;
ws=2*fs/f;
num=-20*log10(sqrt(rp*rs))-13;
dem=14.6*(fs-fp)/f;
n=ceil(num/dem);
n1=n+1;
if(rem(n,2)~=0)
n1=n;
n=n-1;
end
c=input('enter your choice of window function 1. rectangular 2. triangular
3.kaiser: \n ');
if(c==1)
y=rectwin(n1);
disp('Rectangular window filter response');

Department of Electronics & Communication 35

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
end
if (c==2)
y=triang(n1);
disp('Triangular window filter response');
end
if(c==3)
y=kaiser(n1);
disp('kaiser window filter response');
end
%LPF
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,1);plot(o/pi,m);
title('LPF');
ylabel('Gain in dB-->');
xlabel('(a) Normalized frequency-->');
%HPF
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,2);plot(o/pi,m);
title('HPF');
ylabel('Gain in dB-->');
xlabel('(b) Normalized frequency-->');
%BPF
wn=[wp ws];
b=fir1(n,wn,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,3);plot(o/pi,m);
title('BPF');
ylabel('Gain in dB-->');
xlabel('(c) Normalized frequency-->');
%BSF
b=fir1(n,wn,'stop',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(2,2,4);plot(o/pi,m);
title('BSF');
ylabel('Gain in dB-->');
xlabel('(d) Normalized frequency-->');

Department of Electronics & Communication 36

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT:

Department of Electronics & Communication 37

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

C PROGRAM FOR KAISER WINDOW LPF:

#include "filtercfg.h"
#include "dsk6713.h"
#include "dsk6713_aic23.h"
#include "stdio.h"
//float filter_coeff [] = {-0.000019,-0.000170,-0.000609,-0.001451,-0.002593,

// -0.003511, -0.003150,0.000000,0.007551,0.020655,
// 0.039383,0.062306,0.086494,0.108031,0.122944,
// 0.128279,0.122944,0.108031,0.086494,0.062306,
// 0.039383,0.020655,0.007551,0.000000, -0.003150,
// -0.003511, -0.002593, -0.001451, -0.000609, -0.000710,
// -0.000019};// kaiser low pass fir filter pass band range 0-500Hz

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

//float filter_coeff[]={-0.000035,-0.000234,-0.000454,0.000000,0.001933,
// 0.004838,0.005671,-0.000000,-0.013596,-0.028462,
// -0.029370,0.000000,0.064504,0.148863,0.221349,
// 0.249983,0.221349,0.148863,0.064504,0.000000,
// -0.029370,-0.028462,-0.013596,-0.000000,0.005671,
// 0.004838,0.001933,0.000000,-0.000454,-0.000234,
// -0.000035};// kaiser low pass fir filter pass band range 0-1000Hz

float filter_coeff[]={-0.000046,-0.000166,0.000246,0.001414,0.001046,
-0.003421,-0.007410,0.000000,0.017764,0.020126,
-0.015895,-0.060710,-0.034909,0.105263,0.289209,
0.374978,0.289209,0.105263,-0.034909,-0.060710,
-0.015895,0.020126,0.017764,0.000000,-0.007410,
-0.003421,0.001046,0.001414,0.000246,-0.000166,
-0.000046};//Kaiser low pass fir filter pass band range 0-1500Hz

//static short int in_buffer[100];

Department of Electronics & Communication 38
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle
hCodec; Uint32 l_input,
r_input,l_output, r_output;
DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

1);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

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

while(!DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);
}

signed int FIR_FILTER(float * h, signed int x)

{
int i=0;
signed long output=0;
static short int in_buffer[100];
in_buffer[0] = x;

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1];

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

KAISER WINDOW HPF:

#include
"filtercfg.h"
#include
"dsk6713.h"

Department of Electronics & Communication 39

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

#include
"dsk6713_aic23.h"
#include "stdio.h"

//float filter_coeff[]={0.000050,0.000223,0.000520,0.000831,0.000845,
// -0.000000,-0.002478,-0.007437,-0.015556,-0.027071,
// -0.041538,-0.057742,-0.073805,-0.087505,-0.096739,
// 0.899998,-0.096739,-0.087505,-0.073805,-0.057742,
// -0.041538,-0.027071,-0.015556,-0.007437,-0.002478,
// -
0.000000,0.000845,0.000831,0.000520,0.000223,
// 0.000050};//FIR High pass Kaiser filter pass band range 400Hz-
3.5KHz

float filter_coeff[]={0.000000,-0.000138,-0.000611,-0.001345,-0.001607,
-0.000000,0.004714,0.012033,0.018287,0.016731,
0.000000,-0.035687,-0.086763,-0.141588,-0.184011,
0.800005,-0.184011,-0.141588,-0.086763,-0.035687,
0.000000,0.016731,0.018287,0.012033,0.004714,
-0.000000,-0.001607,-0.001345,-0.000611,-0.000138,
0.000000};//FIR High pass Kaiser filter pass band range
800Hz-3.5KHz

//float filter_coeff[]={-0.000050,-0.000138,0.000198,0.001345,0.002212,-0.000000,
// -0.006489,-0.012033,-
0.005942,0.016731,0.041539,0.035687,
// -0.028191,-0.141589,-0.253270,0.700008,-0.253270,-
0.141589,
// -0.028191,0.035687,0.041539,0.016731,-0.005942,-
0.012033,
// -0.006489,-0.000000,0.002212,0.001345,0.000198,-0.000138,
// -0.000050};//FIR High pass Kaiser filter pass
band range 1200Hz-3.5KHz

//static short int in_buffer[100];

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

1);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

Department of Electronics & Communication 40

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
l_output=(Int16)FIR_FILTER(&filter_coeff ,l_input);
r_output=l_output;

while(!DSK6713_AIC23_write(hCodec,

l_output)); while(!

DSK6713_AIC23_write(hCodec, r_output));

Department of Electronics & Communication 41

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

DSK6713_AIC23_closeCodec(hCodec);
}

signed int FIR_FILTER(float * h, signed int x)

{
int i=0;
signed long output=0;
static short int in_buffer[100];
in_buffer[0] = x;

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1];

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

RECTANGULAR LPF:

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//float filter_coeff[]={-0.008982,-0.017782,-0.025020,-0.029339,-0.029569,
// -0.024895,-0.014970,0.000000,0.019247,0.041491,
// 0.065053,0.088016,0.108421,0.124473,0.134729,
// 0.138255,0.134729,0.124473,0.108421,0.088016,
// 0.065053,0.041491,0.019247,0.000000,-0.014970,
// -0.024895,-0.029569,-0.029339,-0.025020,-0.017782,
// -0.008982};//FIR Low pass Rectangular Filter
pass band range 0-500Hz

//float filter_coeff[]={-0.015752,-0.023869,-0.018176,0.000000,0.021481,
// 0.033416,0.026254,-0.000000,-0.033755,-0.055693,
// -0.047257,0.000000,0.078762,0.167080,0.236286,
// 0.262448,0.236286,0.167080,0.078762,0.000000,
// -0.047257,-0.055693,-0.033755,-0.000000,0.026254,
// 0.033416,0.021481,0.000000,-0.018176,-0.023869,
// -0.015752};//FIR Low pass Rectangular Filter
pass band range 0-1000Hz

float filter_coeff[]={-0.020203,-0.016567,0.009656,0.027335,0.011411,
-0.023194,-0.033672,0.000000,0.043293,0.038657,
-0.025105,-0.082004,-0.041842,0.115971,0.303048,
0.386435,0.303048,0.115971,-0.041842,-0.082004,
-0.025105,0.038657,0.043293,0.000000,-0.033672,
-0.023194,0.011411,0.027335,0.009656,-0.016567,
-0.020203};//FIR Low pass Rectangular Filter pass band range 0-1500Hz

//static short int in_buffer[100];

Department of Electronics & Communication 42

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};
void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

1);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

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

l_output=(Int16)FIR_FILTER(&filter_coeff ,l_input);
r_output=l_output; while(!

DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);
}

signed int FIR_FILTER(float * h, signed int x)

{
int i=0;
signed long output=0;
static short int in_buffer[100];
in_buffer[0] = x;

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1];

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

RECTANGULAR HPF:

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//float filter_coeff[]={0.021665,0.022076,0.020224,0.015918,0.009129,
// -0.000000,-0.011158,-0.023877,-0.037558,-0.051511,

Department of Electronics & Communication 43

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

// -0.064994,-0.077266,-0.087636,-0.095507,-0.100422,
// 0.918834,-0.100422,-0.095507,-0.087636,-0.077266,
// -0.064994,-0.051511,-0.037558,-0.023877,-0.011158,
// -
0.000000,0.009129,0.015918,0.020224,0.
022076,
// 0.021665};//FIR High pass Rectangular filter pass band range 400Hz-
3.5KHz

float filter_coeff[]={0.000000,-0.013457,-0.023448,-0.025402,-0.017127,
-0.000000,0.020933,0.038103,0.043547,0.031399,
0.000000,-0.047098,-0.101609,-0.152414,-0.188394,
0.805541,-0.188394,-0.152414,-0.101609,-0.047098,
0.000000,0.031399,0.043547,0.038103,0.020933,
-0.000000,-0.017127,-0.025402,-0.023448,-0.013457,
0.000000};//FIR High pass Rectangular filter pass band
range 800Hz-3.5KHz

//float filter_coeff[]={-0.020798,-0.013098,0.007416,0.024725,0.022944,
// -0.000000,-0.028043,-0.037087,-
0.013772,0.030562,
// 0.062393,0.045842,-0.032134,-0.148349,-
0.252386,
// 0.686050,-0.252386,-0.148349,-
0.032134,0.045842,
// 0.062393,0.030562,-0.013772,-0.037087,-
0.028043,
// -0.000000,0.022944,0.024725,0.007416,-0.013098,
// -0.020798};//FIR High pass Rectangular filter pass band range 1200Hz-3.5KHz

//static short int in_buffer[100];

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

1);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

l_output=(Int16)FIR_FILTER(&filter_coeff ,l_in

Department of Electronics & Communication 44

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
put); r_output=l_output;

while(!DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);

Department of Electronics & Communication 45

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

signed int FIR_FILTER(float * h, signed int x)

{
int i=0;
signed long output=0;
static short int in_buffer[100];
in_buffer[0] = x;

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1];

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

TRANGULAR LPF:

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//float filter_coeff[]={0.000000,-0.001185,-0.003336,-0.005868,-0.007885,
// -0.008298,-0.005988,0.000000,0.010265,0.024895,
// 0.043368,0.064545,0.086737,0.107877,0.125747,
// 0.138255,0.125747,0.107877,0.086737,0.064545,
// 0.043368,0.024895,0.010265,0.000000,-0.005988,
// -0.008298,-0.007885,-0.005868,-0.003336,-0.001185,
// 0.000000};//FIR Low pass Triangular Filter pass band range 0-500Hz

//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};//FIR Low pass Triangular Filter pass band range 0-
1000Hz

float filter_coeff[]={0.000000,-0.001104,0.001287,0.005467,0.003043,
-0.007731,-0.013469,0.000000,0.023089,0.023194,
-0.016737,-0.060136,-0.033474,0.100508,0.282844,
0.386435,0.282844,0.100508,-0.033474,-0.060136,
-0.016737,0.023194,0.023089,0.000000,-0.013469,
-0.007731,0.003043,0.005467,0.001287,-0.001104,
0.000000};//FIR Low pass Triangular Filter pass band range 0-1500Hz

//static short int in_buffer[100];

Department of Electronics & Communication 46

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

Department of Electronics & Communication 47

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

1);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

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

l_output=(Int16)FIR_FILTER(&filter_coeff ,l_input);
r_output=l_output; while(!

DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);
}

signed int FIR_FILTER(float * h, signed int x)

{
int i=0;
signed long output=0;
static short int in_buffer[100];
in_buffer[0] = x;

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1];

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

TRANGULAR HPF

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//float filter_coeff[]={0.000000,0.001445,0.002648,0.003127,0.002391,
// -0.000000,-0.004383,-0.010943,-0.019672,-0.030353,
// -0.042554,-0.055647,-0.068853,-0.081290,-0.092048,
// 0.902380,-0.092048,-0.081290,-0.068853,-0.055647,
// -0.042554,-0.030353,-0.019672,-0.010943,-0.004383,

Department of Electronics & Communication 48

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

// -0.000000,0.002391,0.003127,0.002648,0.001445,
// 0.000000};//FIR High pass Triangular filter pass
band range 400Hz-3.5KHz

//float filter_coeff[]={0.000000,-0.000897,-0.003126,-0.005080,-0.004567,
// -0.000000,0.008373,0.017782,0.023225,0.018839,
// 0.000000,-0.034539,-0.081287,-0.132092,-
0.175834,
// 0.805541,-0.175834,-0.132092,-0.081287,-
0.034539,
// 0.000000,0.018839,0.023225,0.017782,0.0083
73,
// -0.000000,-0.004567,-0.005080,-0.003126,-
0.000897,
// 0.000000};//FIR High pass Triangular filter pass band
range
800Hz-3.5KHz

float filter_coeff[]={0.000000,-0.000901,0.001021,0.005105,0.006317,
-0.000000,-0.011581,-0.017868,-0.007583,0.018931,
0.042944,0.034707,-0.026541,-0.132736,-0.243196,
0.708287,-0.243196,-0.132736,-0.026541,0.034707,
0.042944,0.018931,-0.007583,-0.017868,-0.011581,
-0.000000,0.006317,0.005105,0.001021,-0.000901,
0.000000};//FIR High pass Triangular filter pass band
range
1200Hz-3.5KHz

//static short int in_buffer[100];

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

1);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

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

while(!DSK6713_AIC23_write(hCodec, l_output));

Department of Electronics & Communication 49

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);
}

Department of Electronics & Communication 50

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

signed int FIR_FILTER(float * h, signed int x)

{
int i=0;
signed long output=0;
static short int in_buffer[100];
in_buffer[0] = x;

for(i=30;i>0;i--)
in_buffer[i] = in_buffer[i-1];

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

DSP STARTER KIT DSK6713

SNAP SHOTS:
1. APPLIED INPUT TO FILTER

Fig : applying 2V p-p as input for the filter

Department of Electronics & Communication 51

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUT PUT OF L.P.FILTER AT CUTOFF FREQUENCY

L.P.FILTER OUT PUT WILL BE ZERO AFTER CUTOFF FREQUENCY

EXPERMENTAL SETUP OF FINAL IMPLIMENTATION OF LPF

Department of Electronics & Communication 52

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

RESULT: The design of FIR filters (LP/HP) by using various windowing techniques is
done successfully using MATLAB and DSK6713 KIT.

VIVA QUESTIONS
1. What is a filter?
2. Differentiate analog filter and digital filter.
3. Define FIR filter.
4. What are the differences between recursive and non-recursive systems?
5. List a few Applications of FIR filters.
6. Explain advantages of FIR filters over IIR filters.
7. Explain limitations of FIR filters.
8. What is the different method to design FIR filters?
9. Explain different window functions.
10. Differentiate rectangular, triangular and Kaiser windows.

Department of Electronics & Communication 53

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-5:
IIR FILTER (LP/HP) IMPLEMENTATION ON DSP PROCESSORS

AIM: To implement the IIR FILTERS (LPF/HPF) on DSK 6713 DSP starter kit

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
DSK6713 DSP Starter kit CCS Studio 3-1
USB Cable OS: Windows XP
Power Adapter
CRO, Probes, Function Generator

THEORY:

The IIR filter can realize both the poles and zeroes of a system because it has a rational
transfer function, described by polynomials in z in both the numerator and the
denominator

m and n are order of the two polynomials b (nb + 1) and a (na + 1) are the filter
coefficients. These filter coefficients are generated using FDS (Filter Design software
or Digital Filter design package. IIR filters can be expanded as infinite impulse response
filters. In designing IIR filters, cutoff frequencies of the filters should be mentioned. The
order of the filter can be estimated using Butterworth polynomial. That’s why the filters
are named as Butterworth filters. Filter coefficients can be found and the response can
be plotted.

C PROGRAM IIR_BUTERWORTH_LP FILTER

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//const signed int filter_coeff[] =

{2366,2366,2366,32767,-18179,13046};//IIR_BUTTERWORTH_LP FILTER pass band
range 0-2.5kHZ
//const signed int filter_coeff[] =
{312,312,312,32767,-27943,24367};//IIR_BUTTERWORTH_LP FILTER pass band
range 0-800Hz
const signed int filter_coeff[] =
{15241,15241,15241,32761,10161,7877};//IIR_BUTERWORTH_LP FILTER pass band
range 0-8kHz

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle
hCodec; Uint32 l_input,
r_input,l_output, r_output;
DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

Department of Electronics & Communication 54

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
&config); DSK6713_AIC23_setFreq(hCodec,

3);

Department of Electronics & Communication 55

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

l_output=IIR_FILTER(&filter_coeff ,l_input);
r_output=l_output; while(!

DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);
}
signed int IIR_FILTER(const signed int * h, signed int x1)
{
static signed int x[6] = {0,0,0,0,0,0};

static signed int y[6] =

{0,0,0,0,0,0}; int temp=0;

temp = (short int)x1;

x[0] = (signed int)

temp; temp =

( (int)h[0] * x[0]);

temp += ( (int)h[1] * x[1]);

temp += ( (int)h[1] * x[1]);
temp += ( (int)h[2] * x[2]);

temp -= ( (int)h[4] * y[1]);

temp -= ( (int)h[4] * y[1]);
temp -= ( (int)h[5] * y[2]);

temp >>=15;

if ( temp > 32767 )

{
temp = 32767;
}
else if ( temp < -32767)
{
temp = -32767;
}
y[0] = temp;

y[2] = y[1];
y[1] = y[0];

x[2] = x[1];

Department of Electronics & Communication 56

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
x[1] = x[0];

Department of Electronics & Communication 57

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

return (temp<<2);
}

IIR BUTTERWORTH HP FILTER

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//const signed int filter_coeff[] = {20539,-20539,20539,32767,-

18173,13406};//IIR_BUTTERWORTH_HP FILTER pass band range 2.5kHz-11KHz
//const signed int filter_coeff[] =
{15241,-15241,15241,32761,-10161,7877};//IIR_BUTTERWORTH_HP FILTER pass band
range 4kHz-11KHz
const signed int filter_coeff[] = {
7215,-7215,7215,32767,5039,6171};//IIR_BUTTERWORTH_HP FILTER pass band
range 7kHz-11Khz

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

3);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

l_output=IIR_FILTER(&filter_coeff ,l_input);
r_output=l_output; while(!

DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

DSK6713_AIC23_closeCodec(hCodec);
}

signed int IIR_FILTER(const signed int * h, signed int x1)

{
Department of Electronics & Communication 58
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
static signed int x[6] = {0,0,0,0,0,0};

static signed int y[6] = {0,0,0,0,0,0};

Department of Electronics & Communication 59

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

int temp=0;

temp = (short int)x1;

x[0] = (signed int)

temp; temp =

( (int)h[0] * x[0]);

temp += ( (int)h[1] * x[1]);

temp += ( (int)h[1] * x[1]);
temp += ( (int)h[2] * x[2]);

temp -= ( (int)h[4] * y[1]);

temp -= ( (int)h[4] * y[1]);
temp -= ( (int)h[5] *

y[2]); temp >>=15;

if ( temp > 32767 )

{
temp = 32767;
}
else if ( temp < -32767)
{
temp = -32767;
}
y[0] = temp;

y[2] = y[1];
y[1] = y[0];

x[2] = x[1];
x[1] = x[0];

return (temp<<2);
}

IIR CHEBYSHEV LP FILTER

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//const signed int filter_coeff[] = {1455,1455,1455,32767,-23410,21735};//IIR_CHEB_LP

FILTER pass band range 0-2.5kHz
//const signed int filter_coeff[] = {168,168,168,32767,-30225,28637};//IIR_CHEB_LP
FILTER pass band range 0-800Hz
const signed int filter_coeff[] = {11617,11617,11617,32767,8683,15506};//IIR_CHEB_LP
FILTER pass band range 0-8kHz

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

Department of Electronics & Communication 60

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

3);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

l_output=IIR_FILTER(&filter_coeff ,l_input);
r_output=l_output; while(!

DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}
DSK6713_AIC23_closeCodec(hCodec);
}

signed int IIR_FILTER(const signed int * h, signed int x1)

{
static signed int x[6] = {0,0,0,0,0,0};

static signed int y[6] =

{0,0,0,0,0,0}; int temp=0;

temp = (short int)x1;

x[0] = (signed int)

temp; temp =

( (int)h[0] * x[0]);

temp += ( (int)h[1] * x[1]);

temp += ( (int)h[1] * x[1]);
temp += ( (int)h[2] * x[2]);

temp -= ( (int)h[4] * y[1]);

temp -= ( (int)h[4] * y[1]);
temp -= ( (int)h[5] * y[2]);

temp >>=15;

if ( temp > 32767 )

{

Department of Electronics & Communication 61

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
temp = 32767;
}
else if ( temp < -32767)

Department of Electronics & Communication 62

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

{
temp = -32767;
}
y[0] = temp;

y[2] = y[1];
y[1] = y[0];

x[2] = x[1];
x[1] = x[0];

return (temp<<2);
}
IIR CHEBYSHEV HP FILTER

#include "filtercfg.h"
#include "dsk6713.h"
#include
"dsk6713_aic23.h"
#include "stdio.h"

//const signed int filter_coeff[] =

{12730,-12730,12730,32767,-18324,21137};//IIR_CHEB_HP FILTER pass band range
2.5kHz-11KHz
//const signed int filter_coeff[] = {9268,-9268,9268,32767,-7395,18367};//IIR_CHEB_HP
FILTER pass band range 4kHz-11KHz
const signed int filter_coeff[] = {
3842,-3842,3842,32767,12360,19289};//IIR_CHEB_HP FILTER pass band range 7kz-
11KHz

DSK6713_AIC23_Config
config={0x0017,0x0017,0x00d8,0x00d8,0x0011,0x0000,0x0000,0x0043,0x0081,0x0001};

void main()
{
DSK6713_AIC23_CodecHandle hCodec;
Uint32 l_input, r_input,l_output,
r_output; DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0,

&config); DSK6713_AIC23_setFreq(hCodec,

3);

while(1)
{
while(!DSK6713_AIC23_read(hCodec, &l_input));

while(!DSK6713_AIC23_read(hCodec,

&r_input));

l_output=IIR_FILTER(&filter_coeff ,l_inp
ut); r_output=l_output;

while(!DSK6713_AIC23_write(hCodec, l_output));

while(!DSK6713_AIC23_write(hCodec, r_output));
}

Department of Electronics & Communication 63

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
DSK6713_AIC23_closeCodec(hCodec);
}

Department of Electronics & Communication 64

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

signed int IIR_FILTER(const signed int * h, signed int x1)

{
static signed int x[6] = {0,0,0,0,0,0};

static signed int y[6] =

{0,0,0,0,0,0}; int temp=0;

temp = (short int)x1;

x[0] = (signed int)

temp; temp =

( (int)h[0] * x[0]);

temp += ( (int)h[1] * x[1]);

temp += ( (int)h[1] * x[1]);
temp += ( (int)h[2] * x[2]);

temp -= ( (int)h[4] * y[1]);

temp -= ( (int)h[4] * y[1]);
temp -= ( (int)h[5] *

y[2]); temp >>=15;

if ( temp > 32767 )

{
temp = 32767;
}
else if ( temp < -32767)
{
temp = -32767;
}
y[0] = temp;

y[2] = y[1];
y[1] = y[0];

x[2] = x[1];
x[1] = x[0];
return (temp<<2);
}

Department of Electronics & Communication 65

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

DSP STARTER KIT DSK6713

OUTPUT SNAP SHOTS

1. APPLIED INPUT TO FILTER

Fig : applying 2V p-p as input for the filter

OUT PUT OF L.P.FILTER AT CUTOFF FREQUENCY

Department of Electronics & Communication 66

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
L.P.FILTER OUT PUT WILL BE ZERO AFTER CUTOFF FREQUENCY

EXPERMENTAL SETUP OF FINAL IMPLIMENTATION OF LPF

RESULT: Hence implementation of IIR FILTERS (LPF/HPF) on DSK 6713 DSP starter
kit is done successfully.

VIVA QUESTIONS
1. List some advantages of digital filters over analog filters.
2. Write some differences between FIR and IIR filters.
3. What are the different methods to design IIR filters?
4. Why IIR filters are not reliable?
5. What are different applications of IIR filters?
6. What are advantages of IIR filters?
7. What are disadvantages of IIR filters?
8. Differentiate Butterworth and Chebyshev approximations.
9. What is meant by impulse response?
10. What is the importance of impulse response to calculate the o/p response of the filter?

Department of Electronics & Communication 67

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-6:
VERIFICATION OF N-POINT FAST FOURIER TRANSFORM ALGORITHM

AIM: To verify N-point Fast Fourier Transform Algorithm.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB

THEORY:
The Fast Fourier Transform is useful to map the time-domain sequence into a
continuous function of a frequency variable. The FFT of a sequence {x(n)} of length N is
given by a complex-valued sequence X(k).

The above equation is the mathematical representation of the DFT. As the number of
computations involved in transforming an N point time domain signal into its
corresponding frequency domain signal was found to be N 2 complex multiplications, an
alternative algorithm involving lesser number of computations is opted.

PROGRAM:

clc;
clear all;
close all;
tic;
x=input('enter the sequence');
n=input('enter the length of fft');
%compute fft
disp('fourier transformed signal');
X=fft(x,n)
subplot(1,2,1);stem(x);
title('i/p signal');
xlabel('n --->');
ylabel('x(n) -->');grid;
subplot(1,2,2);stem(X);
title('fft of i/p x(n) is:');
xlabel('Real axis --->');
ylabel('Imaginary axis -->');grid;

Department of Electronics & Communication 68

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT:

RESULT: Hence N-point Fast Fourier Transform Algorithm is verified successfully.

VIVA QUESTIONS
1. Define transform. What is the need for transform?
2. Differentiate Fourier transform and discrete Fourier transform.
3. Differentiate DFT and DTFT.
4. What are the advantages of FFT over DFT?
5. Differentiate DITFFT and DIFFFT algorithms.
6. What is meant by radix?
7. What is meant by twiddle factor and give its properties?
8. How FFT is useful to represent a signal?
9. Compare FFT and DFT with respect to number of calculation required?
10. How the original signal is reconstructed from the FFT of a signal?

Department of Electronics & Communication 69

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-7:
GENERATION OF SUM OF SINUSOIDAL SIGNALS USING MATLAB

AIM: to generate Sum of Sinusoidal Signals using MATLAB.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
OS: Windows XP

THEORY:
When two sinusoids are added together the result depends upon their amplitude, frequency and phase. The
effects are easiest to observe when only one of these is varied between the two sinusoids being added.

In the simplest case, when two sinusoids with the same frequency and phase but with different amplitudes are
added together the result is a sinusoid who's amplitude is the sum of the originals and who's frequency and
phase remain unchanged.

When the two sinusoids have different frequencies the result is more complicated. The new signal is no longer a
sinusoid since it doesn't follow the simple up and down pattern. Instead we see the higher frequency sinusoid as
a `ripple' superimposed on the lower frequency sinusoid (see figure). The frequency of the resulting signal will
be the lower of the two original frequencies. In the figure we can see that the resulting signal goes through two
and a half cycles (that is it repeats itself this many times) just like the second sinusoid. The amplitude of the
resulting signal is the sum of the originals and the phase is unchanged.

Department of Electronics & Communication 70

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Adding together sinusoids with different phases can have surprising results. If two sinusoids are `in phase' then
their peaks and troughs coincide and the result is as observed earlier. If the sinusoids are `out of phase' or in
other words if they differ in phase by π radians then their peaks and troughs oppose each other and they will
cancel each other out. The phase of the resulting sinusoid is the sum of the phases of the constituents.

Adding up more than two sinusoids can produce complex looking waveforms. When a number of different
frequencies are combined the frequency of the result will in general be that of the lowest frequency component.
This is often called the fundamental frequency of the signal.

PROGRAM:
clc;
clear all;
close all;
tic;
%giving linear spaces
t=0:.01:pi;
% t=linspace(0,pi,20);
%generation of sine signals
y1=sin(t);
y2=sin(3*t)/3;
y3=sin(5*t)/5;
y4=sin(7*t)/7;
y5=sin(9*t)/9;
y = sin(t) + sin(3*t)/3 + sin(5*t)/5 + sin(7*t)/7 + sin(9*t)/9;
plot(t,y,t,y1,t,y2,t,y3,t,y4,t,y5);
legend('y','y1','y2','y3','y4','y5');
title('generation of sum of sinusoidal signals');grid;
ylabel('---> Amplitude');
xlabel('---> t');
toc;

Department of Electronics & Communication 71

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT:

RESULT: Hence Sum of Sinusoidal Signals using MATLAB is generated successfully.

VIVA QUESTIONS
1. Define Gibb’s phenomena.
2. What is meant by ringing effect?
3. Why do we need to represent a signal in frequency domain?
4. Why Fourier Series converges only for periodic signals?
5. How the ringing effect can be rectified?
6. What are the different forms of Fourier series?
7. What are the limitations of Fourier series?
8. Write few applications of Fourier series.
9. Explain the MATLAB functions ‘tic’ and ‘toc’.
10. Explain the MATLAB function ‘legend’.

Department of Electronics & Communication 72

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-8:
VERIFICATION THE FREQUENCY RESPONSE OF ANALOG LP/HP FILTERS USING
MATLAB

AIM: To verify the frequency response of analog LP/HP filters using MATLAB.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
OS: Windows XP

THEORY:
Analog Low pass filter & High pass filter are obtained by using Butterworth or Chebyshev
filter with coefficients are given. The frequency – magnitude plot gives the frequency
response of the filter.

PROGRAM:

clc;
clear all;
close all;
warning off;
disp('enter the IIR filter design specifications');
rp=input('enter the passband ripple');
rs=input('enter the stopband ripple');
wp=input('enter the passband freq');
ws=input('enter the stopband freq');
fs=input('enter the sampling freq');
w1=2*wp/fs;w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs,'s');
c=input('enter choice of filter 1. LPF 2. HPF \n ');
if(c==1)
disp('Frequency response of IIR LPF is:');
[b,a]=butter(n,wn,'low','s');
end
if(c==2)
disp('Frequency response of IIR HPF is:');
[b,a]=butter(n,wn,'high','s');
end
w=0:.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure,subplot(2,1,1);plot(om/pi,m);
title('magnitude response of IIR filter is:');
xlabel('(a) Normalized freq. -->');
ylabel('Gain in dB-->');
subplot(2,1,2);plot(om/pi,an);
title('phase response of IIR filter is:');
xlabel('(b) Normalized freq. -->');
ylabel('Phase in radians-->');

Department of Electronics & Communication 73

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT:

Department of Electronics & Communication 74

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

RESULT: Hence the frequency response of analog LP/HP filters using MATLAB is
verified successfully.

VIVA QUESTIONS
1. What are the filter specifications required to design the analog filters?
2. What is meant by frequency response of filter?
3. What is meant by magnitude response?
4. What is meant by phase response?
5. Differentiate ideal filter and practical filter responses.
6. What are the different types of analog filter approximations?
7. Define order of the filter and explain important role it plays in designing of a filter.
8. Explain advantages and disadvantages of Butterworth filter
9. Explain advantages and disadvantages of Chebyshev filter
10. Why Chebyshev is better than Butterworth filter?

Department of Electronics & Communication 75

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-9:
COMPUTATION OF POWER DENSITY SPECTRUM OF A SEQUENCE USING MATLAB

AIM: to compute Power Density Spectrum of a sequence using MATLAB.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
OS: Windows XP
THEORY:
By the definition of energy spectral density require that the Fourier transforms of
the signals exist, that is, that the signals are integrable/summable or
square-integrable/square-summable. (Note: The integral definition of the Fourier
transform is only well-defined when the function is integrable. It is not sufficient for a
function to be simply square-integrable. In this case one would need to use the
Plancherel theorem.) An often more useful alternative is the power spectral density
(PSD), which describes how the power of a signal or time series is distributed with
frequency. Here power can be the actual physical power, or more often, for convenience
with abstract signals, can be defined as the squared value of the signal, that is, as the
actual power dissipated in a load if the signal were a voltage applied across it. This
instantaneous power (the mean or expected value of which is the average power) is then
given by
P(t) = s(t)2 for a signal s(t).

PROGRAM:

t = 0:0.001:0.6;
x = sin(2*pi*50*t)+sin(2*pi*120*t);
y = x + 2*randn(size(t));
figure, plot(1000*t(1:50),y(1:50)) ;
title('Signal Corrupted with Zero-Mean Random Noise');
xlabel('time (milliseconds)');
Y = fft(y,512);
%The power spectral density, a measurement of the energy at various
frequencies, is:
Pyy = Y.* conj(Y) / 512;
f = 1000*(0:256)/512;
figure, plot(f,Pyy(1:257));
title('Frequency content of y');
xlabel('frequency (Hz)');

Department of Electronics & Communication 76

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT:

RESULT: Hence the Power Density Spectrum of a given sequence using MATLAB is
computed successfully.

VIVA QUESTIONS
1. Define power signal.
2. Define energy signal.
3. Define power spectral density of a signal.
4. How the energy of a signal can be calculated?
5. Explain difference between energy spectral density and power spectral density.
6. Explain the PSD plot.
7. What is the importance of PSD?
8. What are the applications of PSD?
9. Explain MATLAB function randn(size(n)).
10. What is the need to represent the signal in frequency domain?

Department of Electronics & Communication 77

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT NO-10:
COMUTATION OF FFT OF 1-D SIGNAL USING MATLAB

AIM: To find the FFT of given 1-D signal and plot.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
DSK6713 DSP Starter kit CCS Studio 3-1
USB Cable, Power Adapter OS: Windows XP

THEORY:
A fast Fourier transform (FFT) is an efficient algorithm to compute the discrete
Fourier transform (DFT) and it’s inverse. There are many distinct FFT algorithms involving
a wide range of mathematics, from simple complex-number arithmetic to group theory
and number theory; this article gives an overview of the available techniques and some
of their general properties, while the specific algorithms are described in subsidiary
articles linked below.
A DFT decomposes a sequence of values into components of different
frequencies. This operation is useful in many fields (see discrete Fourier transform for
properties and applications of the transform) but computing it directly from the definition
is often too slow to be practical. An FFT is a way to compute the same result more
quickly: computing a DFT of N points in the naive way, using the definition, takes (N2)
arithmetical operations, while an FFT can compute the same result in only
(N log N) operations. The difference in speed can be substantial, especially for long data
sets where N may be in the thousands or million in practice, the computation time can be
reduced by several orders of magnitude in such cases, and the improvement is roughly
proportional to N / log(N).

C Program:
#include<stdio.h>
#include<math.h>
#define N 32
#define pI
3.14159
typedef struct
{
float real,imag;
}
complex;
float
iobuffer[N];
float y[N];
main()
{
int i;
complex w[N];
complex x[N];
complex
temp1,temp2;
int
j,k,upper_leg,lower_leg,leg_diff,index,ste
p; for(i=0;i<N;i++)
{
iobuffer[i]=sin((2*pI*2*i)/32.0);
}
for(i=0;i<N;i++)
{
x[i].real=iobuffer[i];
Department of Electronics & Communication 78
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
x[i].imag=0.0;
}
for(i=0;i<N;i++)

Department of Electronics & Communication 79

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

{
w[i].real=cos((2*pI*i)/(N*2.0));
w[i].imag=sin((2*pI*i)/(N*2.0));
}
leg_diff=N/2;
step=2;
for(i=0;i<5;i
++)
{
index=0;
for(j=0;j<leg_diff;j
++)
{
for(upper_leg=j;upper_leg<N;upper_leg+=(2*leg_diff))
{
lower_leg=upper_leg+leg_diff; temp1.real=(x[upper_leg]).real+
(x[lower_leg]).real; temp1.imag=(x[upper_leg]).imag+
(x[lower_leg]).imag; temp2.real=(x[upper_leg]).real-
(x[lower_leg]).real; temp2.imag=(x[upper_leg]).imag-
(x[lower_leg]).imag; (x[lower_leg]).real=temp2.real*(w[index]).real-
temp2.imag*(w[index]).imag;
(x[lower_leg]).imag=temp2.real*(w[index]).imag+temp2.imag*(w[index
]).real; (x[upper_leg]).real=temp1.real;
(x[upper_leg]).imag=temp1.imag;
}
index+=step;
}
leg_diff=(leg_diff)/
2; step=step*2;
}
j=
0;
for(i=1;i<(N-1);i++)
{
k=N/2;
while(k<=
j)
{
j=j-k;
k=k/2
;
}
j=j+
k;
if(i<j
)
{
temp1.real=(x[j]).real;
temp1.imag=(x[j]).imag;
(x[j]).real=(x[i]).real;
(x[j]).imag=(x[i]).imag;
(x[i]).real=temp1.real;
(x[i]).imag=temp1.imag;
}
}
for(i=0;i<N;i++)
{
y[i]=sqrt((x[i].real*x[i].real)+(x[i].imag*x[i].imag));
}
Department of Electronics & Communication 80
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
for(i=0;i<N;i++)
{
printf("%f\t",y[i]);
}

Department of Electronics & Communication 81

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

return(0);
}

OUTPUT:
Graph Of Input Signal: sinusoidal Signal

OUTPUT GRAPH: FFT of Sinusoidal Signal

RESULT: Hence the FFT of given 1-D signal is computed and plotted successfully.

VIVA QUESTIONS
1. Define signal, Give Examples for 1-D, 2-D, 3-D signals.
2. Explain mathematical formula for calculation of FFT.
3. Explain mathematical formula for calculation of IFFT.
4. How to calculate FT for 1-D signal?
5. Define DFT for 1-D signal.
6. What is meant by magnitude plot, phase plot, power spectrum?
7. Explain the importance of FFT.
8. Explain the applications of FFT.
9. What are separable transforms?
10. Explain Modulation property of Fourier Transform.

Department of Electronics & Communication 82

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

EXPERIMENT-11:
FREQUENCY RESPONSE OF ANTI IMAGING AND ANTI ALIASING FILTERS

AIM: To observe the frequency responses of anti imaging and anti aliasing filters.

EQUIPMENT REQUIRED:

Hardware required Software Required

PC MATLAB
OS: Windows XP

THEORY:
The FIR Interpolator object up samples an input by the integer up sampling factor,
L, followed by an FIR anti-imaging filter. The filter coefficients are scaled by the
interpolation factor. A poly phase interpolation structure implements the filter. The
resulting discrete-time signal has a sampling rate L times the original sampling rate. The
demo versions illustrate two possible decimator design solutions. The floating-point
version model uses a cascade of three poly phase FIR decimators. This approach reduces
computation and memory requirements as compared to a single decimator by using
lower-order filters. Each decimator stage reduces the sampling rate by a factor of four.
The fixed-point version uses a five-section CIC decimator to reduce the sampling rate by
the same factor of 64. While not as flexible as a FIR decimator, the CIC decimator has the
advantage of not requiring any multiply operations. It is implemented using only
additions, subtractions, and delays. Therefore, it is a good choice for a hardware
implementation where computational resources are limited.

PROGRAM FOR ANTI IMAGING FILTER

clear all;
n=0:1:1023;
x=1/4*sinc((1/4)*(n-512)).^2
i=1:1024;
y=[zeros(1,2048)];
y(2*i)=x;
f=-2:1/512:2;
h1=freqz(x,1,2*pi*f);
h2=freqz(y,1,2*pi*f);
subplot(3,1,1);
plot(f,abs(h1));
xlabel('frequency');
ylabel('magnitude');
title('frequecny response of input sequence');
subplot(3,1,2);
plot(f,abs(h2));
xlabel('frequency');
ylabel('magnitude');
title('frequency reponse of up sampled input sequence ');
p=fir1(127,.3);
xf=filter(p,1,y);
h4=freqz(xf,1,2*pi*f);
subplot(3,1,3);
plot(f,abs(h4));
title('frequency response of output of an anti imanaging filter');
xlabel('frequency');
ylabel('magnitude');

Department of Electronics & Communication 83

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

OUTPUT:

PROGRAM FOR ANTI ALIASING FILTER:

clear all;
F=input('enter the highest normalized frequency
component'); D=input('emter the dicimation factor');
n=0:1:1024;
xd=(F/2*sinc(F/2)*(n-
512)).^2; f=-2:1/512:2;
h1=freqz(xd,1,pi*f);
subplot(3,1,1);
plot(f,abs(h1));
xlabel('frequency
');
ylabel('magnitud
e');
title('frequency response of input
sequence'); if(F*D<=1)
xd1=F/2*sinc(F/2*(n-
512)*D).^2;
h2=freqz(xd1,1,pi*f*D);
subplot(3,1,3)
plot(f,abs(h2));
axis([-2 2 0 1]);
h2=freqz(xd1,1,pi*f*D);
subplot(3,1,2);
plot(f*D,abs(h2));
axis([-2*D 2*D 0
1]); else
p=fir1(127,1/D);
xf=filter(p,1,xd);
h4=freqz(xf,1,pi*f);
subplot(3,1,3);
plot(f,abs(h4));
title('FREQUENCY RESPONSE OF OUTPUT OF ANTI ALIASING FILTER');
xlabel('frequency');
ylabel('magnitude');
Department of Electronics & Communication 84
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
i=1:1:1024/D;
xr=xf(i*D);
h5=freqz(xr,1,pi*f*D);
subplot(3,1,2);
plot(f*D,abs(h5));
title('frequency response of output of
downsampler'); xlabel('frequency');
ylabel('magnitud
e'); end;

enter the highest normalized frequency

component 0.25 enter the decimation factor 5

OUTPUT:

RESULT: Hence the frequency responses of anti-imaging and anti-aliasing filters

are observed successfully.

VIVA QUESTIONS
1. Explain about multi rate digital signal processing.
2. List the Applications of multi rate digital signal processing.
3. Define interpolation.
4. Define decimation.
5. Define aliasing.
6. What is meant anti-aliasing?
7. What is the effect of anti-imaging filter?
8. Define sampling rate.
9. What is the use of sampling rate convertors?
10. Explain advantages of anti-aliasing filters.

Department of Electronics & Communication 85

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

PROCEDURE FOR HARDWARE EXECUTION

I. Consignment list
1.A main board of DSP6713 -1No
2.Power Supply 5V, 3A. - 1No
3.USB Programmer-1No
4.USB Cable -1No
4.Head Phone with MIC -1No
5.Audio Jack -3No [1+2]
6.Sample and Syllabus Programs CD -1No
7.DSP Lab Manual -1No

II. Introduction of main board

1.USB2.0 CY7C68013-56PVC, compatible with USB2.0 and USB1.1, including 8051
2.DSP TMS320C6713 TQFP-208 Package Device with, 4 layers board
3.SDRAM MT48LC4M16A2 1meg*16 *4 bank micron
4.FLASH AM29LV800B 8Mbit1Mbyte of AMD
5.RESET chip specialize for reset with button for manually reset
6.POWER supply externally, special 5V, 3.3V, 1.6V chip for steady voltage with
remaining for other devices.
7.EEPROM 24LC64 for download of USB firmware
8.CPLD XC95144XL
9.AIC TLV320AIC23B sampling with 8-96KHZ, 4 channels

III. The setup of XDS510 USB emulator in CCS V3.1

1.Install CCS V3.1 software according to the Custom Install default Custom Install_could
changes the installation directory. Following is an example of install the software under C
disk root directory.

2.Install USB Emulator choose to install directory from CCS v3.1 that is if CCSv3.1 is
installed in C: / CCStudio_v3.1 directory, then install the USB emulator driver in this
directory.

3.Replace TIXDS510_Connection.xml file [From \CCStudio_v3.1\drivers\TargetDB\

connection with the TIXDS510_Connection.xml in xds510_setup directory.]
Department of Electronics & Communication 86
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

Procedure to Setup Emulator:

Start->Program->Texas Instrument-> Code Composer Studio 3.1 ->Setup Code

Composer Studio v3.1, the window of Code Composer Studio Setup will show up.

1.Open the Setup CCStudio v3.1

2.Select Create Board (Marked in Circle, can witness in the below Figure)

Department of Electronics & Communication 87

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

3.Right Click on the TI XDS510 emulator and select add to system. Enter,
(After selecting that options a Connection Properties window will be opened as
shown in step 4).

4.Provide Connection Name as ChipMax_6713 and click on Next.

Department of Electronics & Communication 88

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

5.Choose the option falling edge is JTAG Standard in TMS/TDO Output Timing.

6.Right Click on TMS320C6710 and Select Add to System…Enter.

7.a) Provide Processor Name as TMS320C6713_0

b) Select GEL File, Click on browse icon and select DSP621x_671x.gel.
c) Select N/A in Master/Slave.

Department of Electronics & Communication 89

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

d) Click on Ok.

8.Click on Save and quit (highlighted by Circle).

9.Click on Yes to start Code Composer Studio.

Department of Electronics & Communication 90

Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

10. Go to Debug and select the option connect by holding Reset button on the Board.

11. Now Target is connected

Department of Electronics & Communication 91

Engineering
 UCE, AKNU DIGITAL SIGNAL PROCESSING LAB

FFT

MAT LAB SOURCE

clc;
clear all;
close all;
x=input('Enter the sequence : ')
N=length(x)
xK=fft(x,N)
xn=ifft(xK)
n=0:N-1;
subplot (2,2,1);
stem(n,x);
xlabel('n---->');
ylabel('amplitude');
title('input sequence');
subplot (2,2,2);
stem(n,abs(xK));
xlabel('n---->');
ylabel('magnitude');
title('magnitude response');
subplot (2,2,3);
stem(n,angle(xK));
xlabel('n---->');
ylabel('phase');
title('Phase responce');
subplot (2,2,4);
stem(n,xn);
xlabel('n---->');
ylabel('amplitude');

GENERATION OF SINUSOIDAL SIGNAL THROUGH FILTERING A

IM: To generate a sinusoidal signal through filtering. S
Department of Electronics & Communication 92
Engineering
UCE, AKNU DIGITAL SIGNAL PROCESSING LAB
oftware: MATLAB
Theory: One application of an LTI discrete-time system is to pass certain frequency
components in an input sequence without any distortion and block other frequency
components. Such systems are called digital filters. The key to the filtering process
is the inverse Discrete Fourier transform, which expresses an arbitrary input
sequence as a linear weighed sum of an infinite number of exponential sequences,
or equivalently, as a linear weighted sum of sinusoidal sequences. As a result, by
appropriately choosing the values of magnitude function of the LTI digital filter at
frequencies corresponding to the frequencies of the sinusoidal components of the
input, some of these sinusoidal sequences can be selectively heavily attenuated or
filtered with respect to the others.
SINUSOIDSL SINAL
close all;
clear all;
clc;
b=[1];
a=[1,-1,0.9];
n=[-20:120];
t=0:0.1:2*pi;
x=sin(t);
s=filter(b,a,x);
stem(t,s);
title('sinusoidal response');
xlabel('n');
ylabel('s(n)');

VIVA QUESTIONS:
1. Define sequence and signal?
2. Differentiate periodic and non-periodic signals?
3. Define period of the signal?
4. Define LTI system. 5. What is filtering?

Department of Electronics & Communication 93

Engineering