New User Manual5416

TMS320C5416 DSK
USER MANUAL
Cranes Software International Limited

(TI-Division)
#5, Airport Road, Domlur Layout, Bangalore 560 071.
Phone: 91-80-25352636/37/25353038/4532/25354496 Fax : 25356299
tisupport@cranessoftware.com
Cranes Software Int. Ltd.
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Copyright 2008 Cranes Software International Limited. All rights reserved.

Cranes Software believes the information in this publication is accurate as of its publication date.
The information is subject to change without notice. THE INFORMATION IN THIS
PUBLICATION IS PROVIDED "AS IS." CRANES SOFTWARE MAKES NO
REPRESENTATIONS OR WARRANTIES OF ANY KIND WITH RESPECT TO THE
INFORMATION IN THIS PUBLICATION, AND SPECIFICALLY DISCLAIMS IMPLIED
WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
Use, copying, and distribution of any part in this publication requires an applicable
Software license.
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TABLE OF CONTENTS
1. INSTALLATION PROCEDURE. 4
2. DSK FEAUTURES. 8
3. INTRODUCTON TO CODE COMPOSER STUDIO13
4. PROCEDURE TO WORK ON CCS....14
5. EXPERIMENTS USING DSK
a.
SINE WAVE GENERATION
b.
SOLUTION OF DIFFERENCE EQUATIONS
c.
TO VERIFY LINEAR CONVOLUTION
d.
TO VERIFY CIRCULAR CONVOLUTION
e.
PROCEDURE TO WORK IN REALTIME
f.
TO DESIGN FIR FILTER (LOW PASS/HIGH PASS) USING WINDOWING

TECHNIQUE.
a.
USING RECTANGULAR WINDOW
b.
USING TRIANGULAR WINDOW
c.
USING KAISER WINDOW
g.
TO DESIGN IIR FILTER (LP/HP).
h.
NOISE CANCELLATION USING ADAPTIVE FILTERS.
i.
TO FIND THE FFT OF GIVEN 1-D SIGNAL AND PLOT
j.
TO COMPUTE POWER DENSITY SPECTRUM OF A SEQUENCE
k.
MINI PROJECT
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INSTALLATION PROCEDURE
SYSTEM REQUIREMENTS
Minimum
233MHz or Higher PentiumCompatible CPU

600MB of free hard disk space
128MB of RAM or more
SVGA (800 x 600 ) display
Internet Explorer (4.0 or later) or
Netscape Navigator (4.7 or later)
Local CD-ROM drive
Recommended
500MHz or Higher Pentium

Compatible CPU
128MB RAM
16bit Color
Supported Operating Systems
Windows NT 4.0 Service Pack 4 or higher

Windows 2000 Service Pack 1
Windows Me
Windows XP
DSK HARDWARE INSTALLATION
Shut down and power off the PC

Connect the supplied USB port cable to the board
Connect the other end of the cable to the USB port of PC
Note: If you plan to install a Microphone, speaker, or
Signal generator/CRO these must be plugged in properly
before you connect power to the DSK
Plug the power cable into the board

Plug the other end of the power cable into a power outlet
The user LEDs should flash several times to indicate board is operational
When you connect your DSK through USB for the first time on a Windows loaded PC the
new hardware found wizard will come up. So, Install the drivers (The CCS CD contains
the require drivers for C5416 DSK).
Install the CCS software for C5416 DSK.
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DSK SOFTWARE INSTALLATION

You must install the hardware before you install the software on your system.
The requirements for the operating platform are;
Insert the installation CD into the CD-ROM drive

An install screen appears like below; if not, go to the windows Explorer
And run setup.exe
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Choose the option to install Code Composer Studio
If you already have C5000 CC Studio IDE installed on your PC,

do not install DSK software. CC Studio IDE full tools supports
the DSK platform
Respond to the dialog boxes as the installation program runs.

The Installation program automatically configures CC Studio IDE for operation with your
DSK and creates a 5416 DSK CCStudio_v3.1 IDE DSK icon on your desktop.
5416 DSK CCStudio v3.1.lnk
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DIAGNOSTIC:
Test the USB port by running DSK Port test from the start menu
Use StartProgramsTexas InstrumentsCode Composer StudioCode Composer Studio

C5416 DSK ToolsC5416 DSK Diagnostic Utilities
5416 DSK Diagnostics Utility v3.1.lnk
(Available on desktop)
Select StartSelect 5416 DSK Diagnostic Utility
The Screen Look like as below. The below Screen will appear
Select Start Option
Utility Program will test the board
After testing Diagnostic Status you will get PASS
If the board still fails to detect
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TROUBLESHOOTING:1.
If installing on Windows XP and your PC is connected to the internet through a firewall the
USB install may take up to 15 minutes if you let it complete normally. The work-around for
this issue is to simply disconnect your network cable during the USB hardware install.
2. Make sure all of the Configuration Switches (SW3) are set in the off position. This configures
the DSK for the factory default settings of little endian processor mode booting out of the onboard Flash memory.
3. Some of the Help Files are links to Adobe Acrobat PDF files. If you intend to access these files
you must have Adobe Acrobat installed on your system.
4. If you want to verify a successful USB driver install, open your device manager by right
clicking on the My Computer icon on your desktop and selecting Properties --> HW -->
Device Manager. You should see a new class SD USB Based Debug Tools and one
Spectrum Digital TMS320C5416 DSK installed.
5. The BUSY LED above the USB connector comes on when power is applied to the DSK. Do not
launch Code Composer until the LED is off.
6. Go to CMOS setup Enable the USB Port Option
(The required Device drivers will load along with CCS Installation)
7. Message: Failed: USB device is NOT enumerated or plugged in.
Problem: Windows cannot find the DSK. Check power and USB Cable
8. Messages: Waiting for USB Enumeration.
Status: This is a status message which indicates that Code Composer Studio is waiting for
communication with the on-board JTAG emulator to be established. This should take less than 10
seconds. Do not hit the close button unless it is clear that something is wrong (the progress bar
runs forever).
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TMS320C5416 DSK BOARD FEATURES

Package Contents
The 5416 DSP Starter Kit (DSK) is a low-cost platform, which lets enables customers to evaluate
and develop applications for the TI C54X DSP family.
The primary features of the DSK are:
160 MHz TMS320VC5416 DSP
PCM3002 Stereo Codec
Four Position User DIP Switch and Four User LEDs
On-board Flash and SRA
DSK Board Features
Features
TMS320VC5416 DSP
CPLD
External SRAM
External Flash
PCM3002 Codec
4 User LEDs
4 User DIP Switches
4 Jumpers
Daughter card Expansion Interface
HPI Expansion Interface
Embedded JTAG Emulator
Details
160MHz, fixed point, 128Kwords internal RAM
Programmable "glue" logic
64Kwords, 16-bit interface
256Kwords, 16-bit interface
Stereo, 6KHz 48KHz sample rate, 16 or 20 bit
samples, mic-in, line-in, line-out and speaker jacks
Writable through CPLD
Readable through CPLD
Selects power-on configuration and boot modes
Allows user to enhance functionality with add-on
daughter cards
Allows high speed communication with another DSP
Provides high speed JTAG debug through widely
accepted USB host interface
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Key Features of the C5416 DSK

C5416 DSK Tutorial: Quick overview of the DSK and its contents, getting started with
Code Composer.
Hardware: Detailed descriptions of the C5416 DSK hardware components.
Board Support Library: Functions that provide an easy interface to DSK hardware.
Examples: Example code that runs on the DSK.
Diagnostic Utility: Host based diagnostics.
Schematics: Schematics for the DSK.
Datasheets: Datasheets related to the DSK.
Troubleshooting/Support: Solutions for problems you might encounter.
TMS320VC5416 KEY FEATURES
128Kwords memory high-speed internal memory for maximum performance.
On-chip PLL generates processor clock rate from slower external clock reference.
Timer generates periodic timer events as a function of the processor clock. Used by
DSP/BIOS to create time slices for multitasking.
DMA Controller 6 channel direct memory access controller for high speed data
transfers without intervention from the DSP.
3 McBSPs Multichannel buffered serial ports. Each McBSP can be used for high-speed
serial data transmission with external devices or reprogrammed as general purpose I/Os.
EMIF External memory interface. The EMIF provides a 16-bit bus on which external
memories and other devices can be connected. It also includes internal wait state and bank
switching controls.
Block diagram of 5416 DSK
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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
Note:Launch the DSK help file by opening the following file using Windows Explorer.
C:\CCStudio_v3.1\docs\hlp\c5416dsk.hlp
Documents for Reference:
spru509 Code Composer Studio getting started guide.
spru598 Instruction set guide for 54xx family
spru131 CPU & Peripherals guide
pcm3002
ADC & DAC(CODEC) Data Manual.
spru172 mnemonic instruction guide
spru179 Algebraic Instruction set.
spru173 Application Guide.
Soft Copy of datasheets are available at : C:\CCStudio_v3.1\docs\pdf.
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Starting Code Composer

To start Code Composer Studio, double click the 5416 DSK CCStudio_v3.1 icon on your desktop.
5416 DSK CCStudio v3.1.lnk
Start Code Composer Studio (ignore this if CCS is already running) by double-clicking on the
C5416 DSK icon on your desktop.
Use the Debug Connect (Alt+c) menu option to open a debug connection to the DSK board
Procedure to work on Code Composer Studio

Generation of sine waveform
1. To create a New Project
Project New (Sin.pjt)
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2. To Create a Source file

File New source file.
Files of type C Source File (*.c, *.ccc)
Type the code in editor window.

Save the file in project folder. (Eg: Sin.c).
Important note: Save your source code with preferred language extension. For ASM codes save
the file as code (.asm) For C and C++ codes code (*.c,*.cpp) respectively.
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3. To Add Source files to Project

Project Add files to Project Sin.c
Files of type C Source File (*.c, *.ccc)
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4. To Add rts_EXT.lib file.

The run-time-support object libraries do not contain functions involving signals and locale
issues. They do contain the following:
ISO C/C++ standard library
C I/O library
Low-level support functions that provide I/O to the host operating system
Intrinsic arithmetic routines
System startup routine, _c_int00
Functions and macros that allow C/C++ to access specific instructionsrts5400.lib

Run-Time-Support (RTS) object libraries for use with little-endian C/C++ code
Project
Add files to Project rts_EXT.lib
Files of type Object and library files (*.o*, *.l*)
Path: C:\CCStudio_v3.1\c5400\cgtools\lib\rts_EXT.lib
5. Add Command and linking file
Linker command files allow you to put linking information in a file; this is useful when you invoke
the linker often with the same information. Linker command files are also useful because they
allow you to use the MEMORY and SECTIONS directives to customize your application.
Project
Add files to Project hello.cmd
Files of type Linker command file (*.cmd, *.lcf)
Path: C:\CCStudio_v3.1\tutorial\sim54xx\hello1\hello.cmd
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5. To Compile:
Project Compile File
6. To build or Link:
Project build,
Which will create the final executable (.out) file.(Eg. Sin. out).
7. Procedure to Load and Run program:
Load program to DSK:
File Load program Sin. Out

From Debug folder
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8. To execute project:
Debug Run.
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Sin.c
#include<stdio.h>
#include<math.h>
float a[100];
main()
{
int i;
for(i=0;i<99;i++)
{
a[i]= sin(2*3.14*5*i/100);
printf("%f",a[i]);
}
}
Graph Plotting
1. Goto CCS tool bar View Graph Time and frequency.
(In any graph the Start address is the important parameter to be checked.)
Refer fig.1.for the graph properties
Fig. 1
OUTPUT
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To Perform Single Step Debugging:

1. Keep the cursor on the on to the line from where u want to start single step debugging.
(eg: set a break point on to first line int i=0; of your project.)
To set break point select
icon from tool bar menu.
2. Load the Sin. out file onto the target.

3. Go to view Watch window.
4. Debug Run.
5. Execution should halt at break point.
6. Press F10 for step over. See the changes happening in the watch window.
7. Similarly go to view & select CPU registers to view the changes happening in CPU
registers.
8. Repeat steps 2 to 6.
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Difference Equation
Description:An Nth order linear constant coefficient difference equation can be represented as
N
M
ak y(n-k) =
k=0
br x(n-r)
r=0
If we assume that the system is causal a linear difference equation provides an explicit relationship
between the input and output. This can be seen by rewriting above equation.
M
y(n) =
br/a0 x(n-r) -r=0
ak/a0 y y(n-k)
k=1
C Program to Implement Difference Equation

#include <stdio.h>
#include<math.h>
#define FREQ 400
float y[3]={0,0,0};
float x[3]={0,0,0};
float z[128],m[128],n[128],p[128];
main()
{
int i=0,j;
float a[3]={0.072231,0.144462,0.072231};
float b[3]={ 1.000000,-1.109229,0.398152};
for(i=0;i<128;i++)
{
m[i]=sin(2*3.14*FREQ*i/24000);
}
for(j=0;j<128;j++)
{
x[0]=m[j];
y[0] = (a[0] *x[0]) +(a[1]* x[1] ) +(x[2]*a[2]) - (y[1]*b[1])-(y[2]*b[2]);
z[j]=y[0];
y[2]=y[1];
y[1]=y[0];
x[2]=x[1];
x[1] = x[0];
}
}
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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:\CCStudio_v3.1\myprojects) with name Diff Eq.pjt.
Add the source files DIFF EQ1.c
to the project using Project
add files to project pull down menu.
Add the linker command file hello.cmd
(Path: C:\ CCStudio_v3.1\tutorial\sim54xx\hello1\hello.cmd)
Files of type Linker command file (*.cmd, *.lcf)
Add the run time support library file rts_EXT.lib
(Path: C:\ CCStudio_v3.1\c5400\cgtools\lib\rts_EXT.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(Diff Eq.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.
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Note: To verify the Diffence Equation , Observe the output for high
frequency and low frequency by changing variable FREQ
in the program.
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LINEAR CONVOLUTION
Description:Linear Convolution Involves the following operations.
1. Folding
2. Multiplication
3. Addition
4. Shifting
These operations can be represented by a Mathematical Expression as follows:
x[ ]= Input signal Samples

h[ ]= Impulse response co-efficient.
y[ ]= Convolution output.
n = No. of Input samples
h = No. of Impulse response co-efficient.
Algorithm to implement C or Assembly program for Convolution:
Eg:
x[n] = {1, 2, 3, 4}
h[k] = {1, 2, 3, 4}
Where: n=4, k=4.
;Values of n & k should be a multiple of 4.

If n & k are not multiples of 4, pad with zeros to make
multiples of 4
r= n+k-1
; Size of output sequence.
= 4+4-1
= 7.
r=
n= 0
1
2
3
0
1
2
3
4
5
x[0]h[0] x[0]h[1] x[0]h[2] x[0]h[3]
x[1]h[0] x[1]h[1] x[1]h[2] x[1]h[3]
x[2]h[0] x[2]h[1] x[2]h[2] x[2]h[3]
x[3]h[0] x[3]h[1] x[3]h[2]
Output:
x[3]h[3]
y[r] = { 1, 4, 10, 20, 25, 24, 16}.
NOTE: At the end of input sequences pad n and k no. of zeros
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C PROGRAM TO IMPLEMENT LINEAR CONVOLUTION
conv.c:
/* prg to implement linear convolution */
#include<stdio.h>
#include<math.h>
int y[20];
main()
{
int m=6;
/*Lenght of i/p samples sequence*/
int n=6;
/*Lenght of impulse response Co-efficients */
int i=0,j;
int x[15]={1,2,3,4,5,6,0,0,0,0,0,0}; /*Input Signal Samples*/
int h[15]={1,2,3,4,5,6,0,0,0,0,0,0}; /*Impulse Response Co-efficients*/
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]);
}
PROCEDURE:
Open Code Composer Studio, make sure the DSP kit is turned on.
Use the Debug Connect menu option to open a debug connection to the DSK board
separate directory(C:\CCStudio_v3.1\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:\CCStudio_v3.1\tutorial\sim54xx\hello1\hello.cmd)
(Path: C:\CCStudio_v3.1\c5400\cgtools\lib\rts_EXT.lib)
Load the program(lconv.out) in program memory of DSP chip using the
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To View output graphically

Select view graph time and frequency.
Configure the graphical window as shown below
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CIRCULAR CONVOLUTION
Description:Steps for Cyclic Convolution
Steps for cyclic convolution are the same as the usual convolution, except all index calculations
are done "mod N" = "on the wheel"
Steps for Cyclic Convolution
Step1: Plot f[m] and h[m]
Subfigure 1.1 Subfigure 1.2

Step 2: "Spin" h[m] n times Anti Clock Wise (counter-clockwise) to get h[n-m]
(i.e. Simply rotate the sequence, h[n], clockwise by n steps)
Figure 2: Step 2
Step 3: Point wise multiply the f[m] wheel and the h[nm] wheel. Sum=y[n]
Step 4: Repeat for all 0nN1
Example 1: Convolve (n = 4)
Subfigure 3.1
Subfigure 3.2
Figure 3: Two discrete-time signals to be convolved.
h[m] =
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Figure 4
Multiply f[m] and sum to yield: y[0] =3
h[1m]
Figure 5
h[2m]
Figure 6
h[3m]
Figure 7
C Program to Implement Circular Convolution

#include<stdio.h>
#include<math.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);
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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 \t",y[i]);
}
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IN PUT:
Eg:
OUT PUT
x[4]={1, 2, 3,4}
h[4]={1, 2, 3,4}
y[4]={26, 28, 26,20}
PROCEDURE:
Open Code Composer Studio; make sure the DSP kit is turned on.
separate directory(C:\CCStudio_v3.1\myprojects) with name cir conv.pjt.
Add the source files Circular Convolution
to the project using Projectadd files to project pull down menu.
Add the linker command file hello.cmd.
(Path: C:\CCStudio_v3.1\tutorial\sim54xx\hello1\hello.cmd)
(Path: C:\CCStudio_v3.1\c5400\cgtools\lib\rts_EXT.lib)
Load the program(lconv.out) in program memory of DSP chip using the
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Frequency shift Keying (FSK)

Program for FSK.
/*Fsk.c*/
#include<stdio.h>
#include<string.h>
#include<math.h>
void freq_10( void );
float c[1000];
int k=0;
main()
{
int i, length;
char b[10];
puts("\n Enter the value for FSK in 1's and 0's\n");
gets(b);
length = strlen(b);
for(i=0;i<length;i++)
{
if (b[i]=='1')
freq_10();
else
freq_5();
}
}
void freq_10 ()
{
int j;
for(j=k;j<(k+100);j++)
{
c[j]= sin(2*3.14*10*j/500);
printf("\n%d\t%d\t\t%.3f\n",j,k,c[j]);
}
k=j;
}
void freq_5 ()
{
int j;
for(j=k;j<(k+100);j++)
{
c[j]= sin(2*3.14*5*j/500);
printf("\n%d\t %d\t\t%.3f\n",j,k,c[j]);
}
k=j;
}
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HOW TO PROCEED
Separate directory (C:\CCStudio\myprojects) with name FSK.pjt.
Add the source files FSK.c in the project using
Project add files to project pull down menu.
Add the rts file rts_EXT.lib
Load the program in program memory of DSP chip using the File-load program pull
down menu.
Run the program and observe output using graph utility.
Graphical display:
1. After run, Go to View<Graph<Time and Frequency.
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Amplitude Shift Keying (ASK)

ASK.C
#include<stdio.h>
#include<string.h>
#include<math.h>
float c[1000];
int k=0;
main()
{
int i, length;
char b[10];
puts("\n Enter the value for ASK in 1's and 0's\n");
gets(b);
length = strlen(b);
for(i=0;i<length;i++)
{
if (b[i]=='1')
freq_10();
else
freq_0();
}
}
void freq_10 ()
{
int j;
for(j=k;j<(k+100);j++)
{
c[j]= sin(2*3.14*10*j/200);
printf("\n%d\t%d\t\t%.3f\n",j,k,c[j]);
}
k=j;
}
void freq_0 ()
{
int j;
for(j=k;j<(k+100);j++)
{
c[j]= 0;
printf("\n%d\t %d\t\t%.3f\n",j,k,c[j]);
}
k=j;
}
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HOW TO PROCEED
Separate directory (C:\CCStudio\myprojects) with name FSK.pjt.
Add the source files ASK.c in the project using
Project
Add the rts file rts_EXT.lib
Load the program in program memory of DSP chip using the File-load program pull
down menu.
Run the program and observe output using graph utility.
Graphical display:
1. After run, Go to View
Graph
Time and Frequency.
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TMS320C5416DSK CODEC(PCM3002) Configuration Using Board

Support Library
Description:1.0 Unit Objective:
To configure the codec PCM3002 for a talk through program using the board support library.
2.0 Prerequisites
TMS320C5416DSP Starter Kit, PC with Code Composer Studio, CRO, Audio Source, Speakers
and Signal Generator.
3.0 Discussion on Fundamentals:
Refer BSL API Module under, help contents TMS5416 DSK.
Procedure for Real time Programs :

1. Connect a Signal Generator/audio input to the LINE IN Socket or connect a microphone to the
MICIN Socket.
2. Connect CRO/Desktop Speakers to the Socket Provided for LINE OUT or connect a
headphone to the Headphone out Socket.
3. Now Switch on the DSK and Bring Up Code Composer Studio on the PC.
4. Use the Debug Connect menu option to open a debug connection to the DSK board
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5. Create a new project with name codec.pjt.

6. Open the File Menu new DSP/BIOS Configuration select
dsk5416.cdb and save it as xyz.cdb
7. Add xyz.cdb to the current project.

Project
Add files to project xyz.cdb
8. Automatically three files are added in the Generated file folder in project pane
xyzcfg.cmd Command and linking file
xyzcfg.s62 optimized assembly code for configuration
xyzcfg_c.c Chip support initialization
9. Double click on the xyz.cdb from the project explorer and double click on the chip
support library explorer.
10. Double click on the MCBSP under the chip support library where you can see MCBSP
Configuration Manager and MCBSP Resource Manager.
11. Right click on the MCBSP Configuration Manager and select Insert mcbspCfg where
you can see mcbspCfg0 appearing under MCBSP Configuration Manager.
12. Right click on mcbspCfg0 and select properties where mcbspCfg0 properties window
35
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appears.
13. Under General property set Breakpoint Emulation to Do Not Stop.
14. Under Transmit modes property set clock polarity to Falling Edge.
15. Under Transmit Lengths property set Word Length Phase1 to 32-bits and set
Words/Frame phase1 to 2.
16. Under Receive modes property set clock polarity to Rising Edge.
17. Under Receive Multichannel property set Rx Channel Enable to All 128 Channels.
18. Under Transmit Multichannel property set Tx Channel Enable to All 128 Channels.
19. Under the Receive Lengths property set Word Length Phase1 to 32-bits and set
Words/Frame phase1 to 2.
20. Under the Sample-Rate Gen property set Generator Clock Source to BCLKR pin.
Set Frame Width to 32 and Frame period to 64.
21. Select Apply and click O.K.
22. Select McBSP2 under the MCBSP Resource Manager.
23. Right click on McBSP2 and select properties where a McBSP2 Properties Window
appears. Enable the Open handle to McBSP option and pre-initialization option. Select
msbspCfg0 under the Pre-initialize pop-up menu and change the Specify Handle
Name property to C54XX_DMA_MCBSP_hMcbsp. Select Apply and click O.K.
24. Add the generated YYYYcfg.cmd file to the current project.
25. Add the given mcbsp_io.c file to the current project which has the main function and calls
all the other necessary routines.
26. View the contents of the generated file YYYYcfg_c.c and copy the include header file
YYYYcfg.h to the mcbsp_io.c file.
27. Add the library file dsk5416f.lib from the location C:\ti\C5400\dsk5416\lib\dsk5416f.lib
to the current project
28. Select project
build options
Compiler
Advance and enable the use Far calls option.
29. Build, Load and Run the program.
30. You can notice the input signal of 500 Hz. appearing on the CRO verifying the McBSP
configuration.
31. You can also pass an audio input and hear the output signal through the speakers.
32. You can also vary the sampling frequency using the DSK5416_PCM3002_setFreq Function
in the mcbsp_io.c file and repeat the above steps.
33. Open the File Menu new Source file
34. Type the code in editor window. Save the file in project folder. (Eg: Codec.c).
Important note: Save your source code with preferred language extension. For ASM
codes save the file as code(.asm) For C and C++ codes code (*.c,*.cpp) respectively.
35. Add the saved Codec.c file to the current project which has the main function and calls all
the other necessary routines.
Project Add files to Project Codec.c
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36. Add the library file dsk5416f.lib to the current project

Path C:\CCStudio_v3.1\C5400\dsk5416\lib\dsk5416f.lib
37. Copy header files dsk5416.h and dsk5416_pcm3002.h from and paste it in current
project folder.
C:\CCStudio_v3.1\C5400\dsk5416\include.
38. Add the header file generated within xyzcfg_c.c to codec.c
Note:- Double click on xyzcfg_c.c. Copy the first line header (eg.#include xyzcfg.h) and
paste that in source file (eg.codec.c).
39. Compile the program using the Project-compile pull down menu or by
40. Build the program using the Project-Build pull down menu or by clicking the shortcut icon
on the left side of program window.
41. Load the program (Codec. Out) in program memory of DSP chip using the
42. Debug
Run
43. You can notice the input signal of 500 Hz. appearing on the CRO verifying the codec
configuration.
44. You can also pass an audio input and hear the output signal through the speakers.
5.0 Conclusion:
The codec PCM3002 successfully configured using the board support library
and verified.
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Mcbsp_io.c:
#include "YYYYcfg.h"
#include <dsk5416.h>
#include <dsk5416_pcm3002.h>
short left_input,right_input;
DSK5416_PCM3002_Config setup = {
0x1ff,
0x1ff,
0x2,
0x0
// Set-Up Reg 0 - Left channel DAC attenuation

// Set-Up Reg 1 - Right channel DAC attenuation
// Set-Up Reg 2 - Various ctl e.g. power-down modes
// Set-Up Reg 3 - Codec data format control
};
void main ()
{
DSK5416_PCM3002_CodecHandle hCodec;
// Initialize the board support library
DSK5416_init();
// Start the codec
hCodec = DSK5416_PCM3002_openCodec(0, &setup);
// Set codec frequency
DSK5416_PCM3002_setFreq(hCodec, 48000);
// Endless loop IO audio codec
while(1){
// Read 16 bits of codec data, loop to retry if data port is busy
while(!DSK5416_PCM3002_read16(hCodec, &left_input));
while(!DSK5416_PCM3002_read16(hCodec, &right_input));
// Write 16 bits to the codec, loop to retry if data port is busy
while(!DSK5416_PCM3002_write16(hCodec, left_input));
while(!DSK5416_PCM3002_write16(hCodec, right_input));
}
}
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FIR filter Designing Experiments

GENERAL CONSIDERATIONS :
In the design of frequency selective filters, the desired filter characteristics are specified
in the frequency domain in terms of the desired magnitude and phase response of the filter. In the
filter design process, we determine the coefficients of a causal FIR filter that closely approximates
the desired frequency response specifications.
IMPLEMENTATION OF DISCRETE-TIME SYSTEMS :

Discrete time Linear Time-Invariant (LTI) systems can be described completely by constant
coefficient linear difference equations. Representing a system in terms of constant coefficient
linear difference equation is its time domain characterization. In the design of a simple frequency
selective filter, we would take help of some basic implementation methods for realizations of LTI
systems described by linear constant coefficient difference equation.
UNIT OBJECTIVE :
The aim of this laboratory exercise is to design and implement Digital Kaiser low pass FIR Filter
& observe its frequency response. In this experiment we design a simple FIR filter so as to pass
required frequencies components (less than 1000 Hz) and stop or attenuate the frequency
components which are outside the required band ( greater than 1824 Hz).
BACKGROUND CONCEPTS :
A Finite impulse response (FIR) filter possesses an output response to an impulse which is of
infinite duration. The impulse response is "finite" since there is no feedback in the, filter that is if
you put in an impulse, then its output must produced for finite duration of time.
PREREQUISITES :
Concept of Discrete time signal processing.

Analog filter design concepts.
TMS320C5416 Architecture and instruction set.
EQUIPMENTS NEEDED :
Host (PC) with windows (95/98/Me/XP/NT/2000).

DSP kit with TMS 320C5416 chip and driver software.
Oscilloscope and Function generator.
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ALGORITHM TO IMPLEMENT:
We need to realize the Kaiser high pass FIR filter by implementing the difference equation y[n] =
b0x[n] + b1x[n-1])+b2x[n-2]b2x[n-25] where b0 b25 are feed forward coefficients [Assume
25thorder of filter] . These coefficients are calculated using MATLAB. A direct form I
implementation approach is taken.
Step 1 - Initialize the McBSP, the DSP board and the on board codec.
Kindly refer the Topic Configuration of 5416 McBSP using CSL
Step 2 - Initialize the discrete time system, that is, specify the initial conditions. Generally
zero initial conditions are assumed.
Step 3 - Take sampled data from codec while input is fed to DSP kit from the signal
generator. Since Codec is stereo , take average of input data read from left and right channel .
Store sampled data at a memory location.
Step 4 - Perform filter operation using above said difference equation and store filter Output
at a memory location.
Step 5 - Output the value to codec (left channel and right channel) and view the output at
Oscilloscope.
Step 6 - Go to step 3.
DESIGNING AN FIR FILTER:

Following are the steps to design linear phase FIR filters Using Windowing Method.
I.
Clearly specify the filter specifications.

Eg: Order
= 450;
Sampling Rate = 8000 samples/sec
Cut off Freq.
= 400 Hz.
II.
Compute the cut-off frequency Wc

Eg: Wc = 2*pie* fc / Fs
= 2*pie* 400/8000
= 0.1*pie
III.
Compute the desired Impulse Response h d (n) using particular Window

Eg: b_rect1=fir1(order, Wc , 'high',boxcar(31));
IV.
Convolve input sequence with truncated Impulse Response x (n)*h (n)
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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].
If Wn is a multi-element vector,
Wn = [W1 W2 W3 W4 W5 ... WN],
FIR1 returns an order N multiband filter with bands
0 < W < W1, W1 < W < W2, ..., WN < W < 1.
B = FIR1(N,Wn,'DC-1') makes the first band a passband.
B = FIR1(N,Wn,'DC-0') makes the first band a stopband.
For filters with a passband near Fs/2, e.g., highpass
and bandstop filters, N must be even.
By default FIR1 uses a Hamming window. Other available windows,
including Boxcar, Hanning, Bartlett, Blackman, Kaiser and Chebwin
can be specified with an optional trailing argument. For example,
B = FIR1(N,Wn,kaiser(N+1,4)) uses a Kaiser window with beta=4.
B = FIR1(N,Wn,'high',chebwin(N+1,R)) uses a Chebyshev window.
By default, the filter is scaled so the center of the first pass band
has magnitude exactly one after windowing. Use a trailing 'noscale'
argument to prevent this scaling, e.g. B = FIR1(N,Wn,'noscale'),
B = FIR1(N,Wn,'high','noscale'), B = FIR1(N,Wn,wind,'noscale').
Matlab Program to generate FIR Filter-Low Pass Coefficients using FIR1

% FIR Low pass filters using rectangular, triangular and kaiser windows
% sampling rate - 8000
order = 50;
cf=[500/4000,1000/4000,1500/4000];
cf--> contains set of cut-off frequencies[Wc ]
41
% cutoff frequency - 500

b_rect1=fir1(order,cf(1),boxcar(51));
b_tri1=fir1(order,cf(1),bartlett(51));
b_kai1=fir1(order,cf(1),kaiser(51,8));
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Rectangular
Triangular
Kaisar [Where 8-->Beta Co-efficient]

fid=fopen('FIR_lowpass_rectangular.txt','wt');
fprintf(fid,'\t\t\t\t\t\t%s\n','Cutoff -400Hz');
fprintf(fid,'\nfloat b_rect1[51]={');
fprintf(fid,'%f,%f,%f,%f,%f,%f,%f,%f,%f,%f,\n',b_rect1);
fseek(fid,-1,0);
fprintf(fid,'};');
fprintf(fid,'\n\n\n\n');
fseek(fid,-1,0);
fprintf(fid,'};');
fprintf(fid,'\n\n\n\n');
fseek(fid,-1,0);
fprintf(fid,'};');
fclose(fid);
winopen('FIR_highpass_rectangular.txt');
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T.1 : Matlab generated Coefficients for FIR Low Pass Kaiser filter:
Cutoff -500Hz
float b_kai1[51]={-0,0,0.0001,0.0002,0.0005,0.0008,0.0012,0.0013,0.001,-0,-0.0018,-0.0045,
-0.0076,-0.0106,-0.0125,-0.0121,0.0082,0,0.0129,0.0302,0.0506,0.0723,0.0929,0.1099,0.1211,0.125,0.1211,0.1099,0.0929,0.0723,
0.0506,0.0302,0.0129,0,-0.0082,-0.0121,-0.0125,-0.0106,-0.0076,-0.0045,-0.0018,0,0.001,0.0013,0.0012,0.0008,0.0005,0.0002,0.0001,0,-0};
Cutoff -1000Hz
IMPLEMENTATION OF AN FIR FILTER :
float b_kai2[51]={- 0,-0,-0.0001,-0.0003,-0.0004,0,0.0009,0.0019,0.0018,-0,-0.0034,-0.0064,0.0059,0,0.0096,0.0171,0.0152,-0,-0.0238,-0.0426,ALGORITHM TO IMPLEMENT :
0.0387,0,0.0711,0.1554,0.2237,0.25,0.2237,0.1554,0.0711,0,-0.0387,-0.0426,-0.0238,0,0.0152,0.0171,0.0096,0,-0.0059,-0.0064,-0.0034,-0,0.0018,0.0019,0.0009,0,-0.0004,-0.0003,We need to realize an advance FIR filter by implementing its difference equation as per the
0.0001,-0,0};
specifications. A direct form I implementation approach is taken. (The filter coefficients are taken
as ai as generated by the Matlab program.)
Cutoff -1500Hz
float b_kai3[51]={ 0,0.0001,0.0002,0.0001,-0.0003,-0.0008,-0.001,0,0.002,0.0035,0.0024,
0.0022,-0.0078,-0.0092,-0.0023,0.011,0.0214,0.0173,-0.0059,-0.0369,-0.0514,0.0247,0.0503,0.153,0.2424,0.2778,0.2424,0.153,0.0503,-0.0247,-0.0514,-0.0369,0.0059,0.0173,0.0214,0.011,-0.0023,-0.0092,-0.0078,-0.0022,0.0024,0.0035,0.002,0,-0.001,0.0008,-0.0003,0.0001,0.0002,0.0001,0};
T.2 :Matlab generated Coefficients for FIR Low Pass Rectangular filter
Cutoff -500Hz
float b_rect1[51]={-0.0046,0,0.005,0.0097,0.0133,0.0151,0.0147,0.0118,0.0068,-0,-0.0077,0.0152,-0.0214,-0.0251,-0.0253,-0.0213,0.0128,0,0.0165,0.0355,0.0557,0.0754,0.0929,0.1066,0.1154,0.1184,0.1154,0.1066,0.0929,0.0754
,0.0557,0.0355,0.0165,0,-0.0128,-0.0213,-0.0253,-0.0251,-0.0214,-0.0152,-0.0077,0,0.0068,0.0118,0.0147,0.0151,0.0133,0.0097,0.005,0,-0.0046};
Cutoff -1000Hz
float b_rect2[51]={-0.0091,-0,-0.0099,-0.0147,-0.0109,0,0.012,0.0179,0.0134,-0,-0.0152,0.0231,-0.0176,0,0.0207,0.0323,0.0254,-0,-0.0326,-0.0538,0.0456,0,0.0761,0.1614,0.2282,0.2535,0.2282,0.1614,0.0761,0,-0.0456,-0.0538,-0.0326,0,0.0254,0.0323,0.0207,0,-0.0176,-0.0231,-0.0152,-0,0.0134,0.0179,0.012,0,-0.0109,-0.0147,0.0099,-0,0.0091};
Cutoff -1500Hz
float b_rect2[51]={- 0.0021,0.0112,0.0126,0.0048,-0.0074,-0.0152,0.0125,0,0.0139,0.019,0.0103,-0.0076,-0.0224,-0.0223,-0.0049,0.0199,0.0344,0.0249,-0.0077,0.0447,-0.0582,-0.0265,0.0516,0.1524,0.237,0.27,0.237,0.1524,0.0516,-0.0265,-0.0582,-0.0447,T.3
: Matlab generated Coefficients for FIR Low Pass Triangular filter
0.0077,0.0249,0.0344,0.0199,-0.0049,-0.0223,-0.0224,-0.0076,0.0103,0.019,0.0139,0,-0.0125,0.0152,-0.0074,0.0048,0.0126,0.0112,0.0021};
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T.3 : Matlab generated Coefficients for FIR Low Pass Triangular filter
Cutoff -500Hz
float b_tri1[51]={ [0,0,0.0005,0.0013,0.0024,0.0034,0.004,0.0037,0.0024,-0,-0.0035,-0.0075,0.0116,-0.0147,-0.016,-0.0144,0.0092,0,0.0134,0.0304,0.0502,0.0713,0.092,0.1105,0.1248,0.1334,0.1248,0.1105,0.092,0.071
3,0.0502,0.0304,0.0134,0,-0.0092,-0.0144,-0.016,-0.0147,-0.0116,-0.0075,-0.0035,0,0.0024,0.0037,0.004,0.0034,0.0024,0.0013,0.0005,0,0;]};
Cutoff -1000Hz
float b_tri2[51]={ 0,-0,-0.0008,-0.0018,-0.0018,0,0.0029,0.0051,0.0044,-0,-0.0062,-0.0103,0.0086,0,0.0118,0.0197,0.0165,-0,-0.0239,-0.0416,0.0372,0,0.0682,0.1512,0.2232,0.2582,0.2232,0.1512,0.0682,0,-0.0372,-0.0416,-0.0239,0,0.0165,0.0197,0.0118,0,-0.0086,-0.0103,-0.0062,-0,0.0044,0.0051,0.0029,0,-0.0018,-0.0018,0.0008,-0,0};
Cutoff -1500Hz
float b_tri3[51]={ 0,0.0005,0.0011,0.0006,-0.0012,-0.0032,-0.0032,0,0.0047,0.0073,0.0044,0.0035,-0.0114,-0.0123,-0.0029,0.0126,0.0233,0.0179,-0.0058,-0.0359,-0.0492,0.0235,0.048,0.1483,0.2407,0.2857,0.2407,0.1483,0.048,-0.0235,-0.0492,-0.0359,0.0058,0.0179,0.0233,0.0126,-0.0029,-0.0123,-0.0114,-0.0035,0.0044,0.0073,0.0047,0,0.0032,-0.0032,-0.0012,0.0006,0.0011,0.0005,0};
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FLOWCHART FOR FIR:

Start
Initialize the DSP Board.
Take a new input in data

from the analog in of codec in
data
Initialize Counter = 0
Initialize Output = 0 , i = 0
Output += coeff[N-i]*val[i]
Shift the input value by one
No
Is the loop
Cnt = order
Poll the ready bit, when
asserted proceed.
Yes
Output += coeff[0]*data
Put the data in val array.
Write the value Output to

Analog output of the codec
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C Program to implement FIR filter:

#include "xyzcfg.h"
Int16 left_input;
Int16 left_output;
Int16 right_input;
Int16 right_output;
static short in_buffer[100];
float filter_Coeff[]={- 0,-0,-0.0001,-0.0003,-0.0004,0,0.0009,0.0019,0.0018,-0,-0.0034,-0.0064,0.0059,0,0.0096,0.0171,0.0152,-0,-0.0238,-0.0426,0.0387,0,0.0711,0.1554,0.2237,0.25,0.2237,0.1554,0.0711,0,-0.0387,-0.0426,-0.0238,0,0.0152,0.0171,0.0096,0,-0.0059,-0.0064,-0.0034,-0,0.0018,0.0019,0.0009,0,-0.0004,-0.0003,0.0001,-0,0};
/*Select particular set of co-efficients different Cut-off frequencies from above given Tables */
0x1ff,
0x1ff,
0x2,
0x0

};
void main ()
{
DSK5416_init();
// Start the codec
DSK5416_PCM3002_setFreq(hCodec,8000);
while(1)
{
left_output=FIR_FILTER(&filter_Coeff ,left_input);
right_output=left_output;
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while(!DSK5416_PCM3002_write16(hCodec, left_output));
while(!DSK5416_PCM3002_write16(hCodec, right_output));
}
}
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=51;i>0;i--)
in_buffer[i] = in_buffer[i-1]; /* shuffle the buffer */
for(i=0;i<51;i++)
output = output + h[i] * in_buffer[i];
return(output);
}
HOW TO PROCEED:
1. Connect a Signal Generator/audio input to the LINE IN Socket or connect a microphone to

the MIC IN Socket.
5. Create a new project with name Fir.pjt.
Project
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10. Type the code in editor window. Save the file in project folder. (Eg: Codec.c).
11. Add the saved Fir.c file to the current project which has the main function and calls all
project folder.
14. Add the header file generated within xyzcfg_c.c to Fir.c
Note:- Double click on xyzcfg_c.c Copy the first line header (eg.xyzcfg.h) and paste that in
source file (eg.Fir.c).
16. Build the program using the Project-Build pull down menu or by clicking the shortcut
icon on the left side of program window.
17. Load the program (Codec.out) in program memory of DSP chip using the
18. Debug
Run
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MATLAB GENERATED FREQUENCY RESPONSE

High Pass FIR filter(Fc= 800Hz).
Low Pass FIR filter (Fc=1000Hz)
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Advance Discrete Time Filter Design(IIR)

IIR filter Designing Experiments
GENERAL CONSIDERATIONS:
In the design of frequency selective filters, the desired filter characteristics are specified in the
frequency domain in terms of the desired magnitude and phase response of the filter. In the filter
design process, we determine the coefficients of a causal IIR filter that closely approximates the
desired frequency response specifications.
IMPLEMENTATION OF DISCRETE-TIME SYSTEMS:
Discrete time Linear Time-Invariant (LTI) systems can be described completely by constant
coefficient linear difference equations. Representing a system in terms of constant coefficient
linear difference equation is its time domain characterization. In the design of a simple frequency
selective filter, we would take help of some basic implementation methods for realizations of LTI
systems described by linear constant coefficient difference equation.
UNIT OBJECTIVE:
The aim of this laboratory exercise is to design and implement a Digital IIR Filter & observe its
frequency response. In this experiment we design a simple IIR filter so as to stop or attenuate
required band of frequencies components and pass the frequency components which are outside
the required band.
BACKGROUND CONCEPTS:
An Infinite impulse response (IIR) filter possesses an output response to an impulse which is of an
infinite duration. The impulse response is "infinite" since there is feedback in the filter, that is if
you put in an impulse, then its output must produced for infinite duration of time.
PREREQUISITES:
Concept of Discrete time signal processing.

Analog filter design concepts.
TMS320C5416 Architecture and instruction set.
EQUIPMENTS NEEDED:
Host (PC) with windows (95/98/Me/XP/NT/2000).

TMS320C5416 DSP Starter Kit (DSK).
Oscilloscope and Function generator.
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ALGORITHM TO IMPLEMENT:
We need to realize the Butter worth band pass IIR filter by implementing the difference equation
y[n] = b0x[n] + b1x[n-1]+b2x[n-2]-a1y[n-1]-a2y[n-2] where b0 b2, a0-a2 are feed forward and
feedback word coefficients respectively [Assume 2nd order of filter].These coefficients are
calculated using MATLAB.A direct form I implementation approach is taken.
Step 1 - Initialize the McBSP, the DSP board and the on board codec.
Kindly refer the Topic Configuration of 5416Codec using BSL
Step 2 - Initialize the discrete time system, that is, specify the initial conditions. Generally
zero initial conditions are assumed.
Step 3 - Take sampled data from codec while input is fed to DSP kit from the signal
generator. Since Codec is stereo, take average of input data read from left and right channel.
Store sampled data at a memory location.
Step 4 - Perform filter operation using above said difference equation and store filter Output
at a memory location.
Step 5 - Output the value to codec (left channel and right channel) and view the output at
Oscilloscope.
Step 6 - Go to step 3.
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FLOWCHART FOR IIR IMPLEMENTATION:
Start
Initialize the DSP Board.
Set initial conditions of discrete

time system by making x[0]-x[2]
and y[0]-y[2] equal to zeros and
a0-a2,b0-b2 with MATLAB filter
coefficients
Take a new input and store it in
x[0].
Do y[-3] = y[-2],y[-2]=y[-1]
and Y[-1] = output .
x[-3] = x[-2], x[-2]=x[-1]
output = x[0]b0+x[-1]b1+
x[-2]b2 - y[-1]a1 - y[-2]a2
x[-1]=x[0]
Poll for ready bit
Poll the ready bit, when
asserted proceed.
Write output to analog i/o.
Stop
F.1 : Flowchart for implementing IIR filter.
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MATLAB PROGRAM TO GENRATE FILTER CO-EFFICIENTS

% IIR Low pass Butterworth and Chebyshev filters
% sampling rate - 24000
order = 2;
cf=[2500/12000,8000/12000,1600/12000];
[num_bw1,den_bw1]=butter(order,cf(1));
[num_cb1,den_cb1]=cheby1(order,3,cf(1));
[num_bw2,den_bw2]=butter(order,cf(2));
[num_cb2,den_cb2]=cheby1(order,3,cf(2));
fid=fopen('IIR_LP_BW.txt','wt');
fprintf(fid,'\t\t-----------Pass band range: 0-2500Hz----------\n');
fprintf(fid,'\t\t-----------Magnitude response: Monotonic-----\n\n\');
fprintf(fid,'\n float num_bw1[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',num_bw1);
fprintf(fid,'\nfloat den_bw1[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',den_bw1);
fprintf(fid,'\n\n\n\t\t-----------Pass band range: 0-8000Hz----------\n');
fprintf(fid,'\t\t-----------Magnitude response: Monotonic-----\n\n');
fprintf(fid,'\nfloat num_bw2[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',num_bw2);
fprintf(fid,'\nfloat den_bw2[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',den_bw2);
fclose(fid);
winopen('IIR_LP_BW.txt');
fid=fopen('IIR_LP_CHEB Type1.txt','wt');
fprintf(fid,'\t\t-----------Pass band range: 2500Hz----------\n');
fprintf(fid,'\t\t-----------Magnitude response: Rippled (3dB) -----\n\n\');
fprintf(fid,'\nfloat num_cb1[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',num_cb1);
fprintf(fid,'\nfloat den_cb1[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',den_cb1);
fprintf(fid,'\n\n\n\t\t-----------Pass band range: 8000Hz----------\n');
fprintf(fid,'\t\t-----------Magnitude response: Rippled (3dB)-----\n\n');
fprintf(fid,'\nfloat num_cb2[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',num_cb2);
fprintf(fid,'\nfloat den_cb2[9]={');
fprintf(fid,'%f,%f,%f,%f,%f,\n%f,%f,%f,%f};\n',den_cb2);
fclose(fid);
winopen('IIR_LP_CHEB Type1.txt');
%%%%%%%%%%%%%%%%%%
figure(1);
[h,w]=freqz(num_bw1,den_bw1);
w=(w/max(w))*12000;
plot(w,20*log10(abs(h)),'linewidth',2)
hold on
[h,w]=freqz(num_cb1,den_cb1);
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w=(w/max(w))*12000;
plot(w,20*log10(abs(h)),'linewidth',2,'color','r')
grid on
legend('Butterworth','Chebyshev Type-1');
xlabel('Frequency in Hertz');
ylabel('Magnitude in Decibels');
title('Magnitude response of Low pass IIR filters (Fc=2500Hz)');
figure(2);
[h,w]=freqz(num_bw2,den_bw2);
w=(w/max(w))*12000;
plot(w,20*log10(abs(h)),'linewidth',2)
hold on
[h,w]=freqz(num_cb2,den_cb2);
w=(w/max(w))*12000;
plot(w,20*log10(abs(h)),'linewidth',2,'color','r')
grid on
legend('Butterworth','Chebyshev Type-1 (Ripple: 3dB)');
xlabel('Frequency in Hertz');
ylabel('Magnitude in Decibels');
title('Magnitude response in the passband');
axis([0 12000 -20 20]);
IIR_CHEB_LP FILTER CO-EFFICIENTS:

CoEffici
ents
B0
B1
B2
A0
A1
A2
Fc=2500Hz
Floating Point
Values
0.044408
0.088815
0.044408
1.000000
-1.412427
0.663336
Fixed Point
Values(Q15)
1455
1455[B1/2]
1455
32767
-23140[A1/2]
21735
Fc=800Hz
Floating Point
Values
0.005147
0.010295
0.005147
1.000000
-1.844881
0.873965
Fixed Point
Values(Q15)
168
168[B1/2]
168
32767
-30225[A1/2]
28637
Fc=8000Hz
Floating Point
Values
0.354544
0.709088
0.354544
1.000000
0.530009
0.473218
Fixed Point
Values(Q15)
11617
11617[B1/2]
11617
32767
8683[A1/2]
15506
Note: We have Multiplied Floating Point Values with 32767(215) to get Fixed Point Values.
IIR_BUTTERWORTH_LP FILTER CO-EFFICIENTS:

CoEffici
ents
B0
B1
B2
A0
A1
A2
Fc=2500Hz
Floating Point
Values
0.072231
0.144462
0.072231
1.000000
-1.109229
0.398152
Fixed Point
Values(Q15)
2366
2366[B1/2]
2366
32767
-18179[A1/2]
13046
Fc=800Hz
Floating Point
Values
0.009526
0.019052
0.009526
1.000000
-1.705552,
0.743655
Fixed Point
Values(Q15)
312
312[B1/2]
312
32767
-27943[A1/2]
24367
Fc=8000Hz
Floating Point
Values
0.465153
0.930306
0.465153
1.000000
0.620204
0.240408
Fixed Point
Values(Q15)
15241
15241[B1/2]
15241
32767
10161[A1/2]
7877
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IIR_CHEB_HP FILTER CO-EFFICIENTS:

CoEffici
ents
B0
B1
B2
A0
A1
A2
Fc=2500Hz
Floating Point
Values
0.388513
-0.777027
0.388513
1.000000
-1.118450
0.645091
Fixed Point
Values(Q15)
12730
-12730[B1/2]
12730
32767
-18324[A1/2]
21137
Fc=4000Hz
Floating Point
Values
0.282850
-0.565700
0.282850
1.000000
-0.451410
0.560534
Fixed Point
Values(Q15)
9268
-9268[B1/2]
9268
32767
-7395[A1/2]
18367
Fc=7000Hz
Floating Point
Values
0.117279
-0.234557
0.117279
1.000000
0.754476
0.588691
Fixed Point
Values(Q15)
3842
-3842[B1/2]
3842
32767
12360[A1/2]
19289
IIR_BUTTERWORTH_HP FILTER CO-EFFICIENTS:

CoEffici
ents
B0
B1
B2
A0
A1
A2
Fc=2500Hz
Floating Point
Values
0.626845
-1.253691
0.626845
1.000000
-1.109229
0.398152
Fixed Point
Values(Q15)
20539
-20539[B1/2]
20539
32767
-18173[A1/2]
13046
Fc=4000Hz
Floating Point
Values
0.465153
-0.930306
0.465153
1.000000
-0.620204
0.240408
Fixed Point
Values(Q15)
15241
-15241[B1/2]
15241
32767
-10161[A1/2]
7877
Fc=7000Hz
Floating Point
Values
0.220195
-0.440389
0.220195
1.000000
0.307566
0.188345
Fixed Point
Values(Q15)
7215
-7215[B1/2]
7215
32767
5039[A1/2}
6171
C PROGRAM TO IMPLEMENT IIR FILTER

#include "xyzcfg.h"
Int16 left_input;
Int16 left_output;
Int16 right_input;
Int16 right_output;
const signed int filter_Coeff[ ] =
{
48,48,48, 32767, -30949, 29322
//96,96,96,32767,-30918,29614 /*fc=600*/
//188,259,188,32767,-29498,27695/*fc=1000*/
//11617,11617,11617,32767,8683,15505/*fc=8000*/
//1455,1455,1455,32767,-23140,21735/*fc=2500*/
//5058,-5058,5058,32767,9995,15803/*fc=7000 -HP */
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//22586,-22586,22586,32767,-20964,15649
//20539,-20539,20539,32767,-18173,13046
//15241,-15241,15241,32767,-10161,7877
};
/*Select particular set of co-efficients different Cut-off frequencies from given Tables */
0x1ff,
0x1ff,
0x2,
0x0

};
void main ()
{
DSK5416_init();
// Start the codec
DSK5416_PCM3002_setFreq(hCodec,24000);
while(1)
{
left_output=IIR_FILTER(&filter_Coeff , left_input);
right_output=left_output;
while(!DSK5416_PCM3002_write16(hCodec, left_output));
while(!DSK5416_PCM3002_write16(hCodec, right_output));
}
}
signed int IIR_FILTER(const signed int * h, signed int x1)
{
signed int x[6] = { 0, 0, 0, 0, 0, 0 }; /* x(n), x(n-1), x(n-2). Must be static */
signed int y[6] = { 0, 0, 0, 0, 0, 0 }; /* y(n), y(n-1), y(n-2). Must be static */
long temp=0;
temp = x1;
x[0] = (signed int) temp;
temp = ( (long)h[0] * x[0]) ;
temp += ( (long)h[1] * x[1]);
temp += ( (long)h[1] * x[1]);
/* Copy input to temp */

/* Copy input to x[stages][0] */
/* B0 * x(n) */
/* B1/2 * x(n-1) */
/* B1/2 * x(n-1) */
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temp += ( (long)h[2] * x[2]);
/* B2 * x(n-2) */
temp -= ( (long)h[4] * y[1]);
/* A1/2 * y(n-1) */
temp -= ( (long)h[4] * y[1]);
/* A1/2 * y(n-1) */
temp -= ( (long)h[5] * y[2]);
/* A2 * y(n-2) */
/* Divide temp by coefficients[A0] */

temp >>= 15;
if ( temp > 32767 )
{
temp = 32767;
}
else if ( temp < -32767)
{
temp = -32767;
}
y[0] = (short int) ( temp );
/* Shuffle values along one place for next time */
y[2] = y[1]; /* y(n-2) = y(n-1) */
y[1] = y[0]; /* y(n-1) = y(n) */
x[2] = x[1]; /* x(n-2) = x(n-1) */
x[1] = x[0]; /* x(n-1) = x(n) */
/* temp is used as input next time through */
return ((short int)temp*1);
}
HOW TO PROCEED:
1. Connect a Signal Generator/audio input to the LINE IN Socket or connect a microphone to
the MIC IN Socket.
5. Create a new project with name Iir.pjt.
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Project
10. Type the code in editor window. Save the file in project folder. (Eg: Iir.c).
11. Add the saved Iir.c file to the current project which has the main function and calls all
project folder.
14. Add the header file generated within xyzcfg_c.c to Iir.c
Note:- Double click on xyzcfg_c.c Copy the first line header (eg.xyzcfg.h) and paste that in
source file (eg.Fir.c).
Clicking the shortcut icon on the left side of program window.
16. Build the program using the Project-Build pull down menu or by clicking the shortcut
icon on the left side of program window.
17. Load the program (Iir.out) in program memory of DSP chip using the
18. Debug
Run
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Fast Fourier Transforms(FFT)

The DFT Equation
Twiddle Factor
In the Definition of the DFT, there is a factor called the Twiddle Factor
where N = number of samples.
If we take an 8 bit sample sequence we can represent the twiddle factor as a vector in the unit
circle. e.g.
Note that
It is periodic. (i.e. it goes round and round the circle !!)
1. That the vectors are symmetric
2. The vectors are equally spaced around the circle.
Why the FFT?
If you look at the equation for the Discrete Fourier Transform you will see that it is quite
complicated to work out as it involves many additions and multiplications involving complex
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TI-DIVISION
numbers. Even a simple eight sample signal would require 49 complex multiplications and 56
complex additions to work out the DFT. At this level it is still manageable, however a realistic
signal could have 1024 samples which requires over 20,000,000 complex multiplications and
additions. As you can see the number of calculations required soon mounts up to unmanageable
proportions.
The Fast Fourier Transform is a simply a method of laying out the computation, which is much
faster for large values of N, where N is the number of samples in the sequence. It is an ingenious
way of achieving rather than the DFT's clumsy P^2 timing.
The idea behind the FFT is the divide and conquer approach, to break up the original N point
sample into two (N / 2) sequences. This is because a series of smaller problems is easier to solve
than one large one. The DFT requires (N-1)^2 complex multiplications and N(N-1) complex
additions as opposed to the FFT's approach of breaking it down into a series of 2 point samples
which only require 1 multiplication and 2 additions and the recombination of the points which is
minimal.
For example Seismic Data contains hundreds of thousands of samples and would take months to
evaluate the DFT. Therefore we use the FFT.
FFT Algorithm
The FFT has a fairly easy algorithm to implement, and it is shown step by step in the list below.
Thjis version of the FFT is the Decimation in Time Method
1. Pad input sequence, of N samples, with ZERO's until the number of samples is the nearest
power of two.
e.g. 500 samples are padded to 512 (2^9)
2. Bit reverse the input sequence.
e.g. 3 = 011 goes to 110 = 6
3. Compute (N / 2) two sample DFT's from the shuffled inputs.
See "Shuffled Inputs"
4. Compute (N / 4) four sample DFT's from the two sample DFT's.
5. Compute (N / 2) eight sample DFT's from the four sample DFT's.
6. Until the all the samples combine into one N-sample DFT
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Shuffled Inputs
The process of decimating the signal in the time domain has caused the INPUT samples to be reordered. For an 8 point signal the original order of the samples is
0, 1, 2, 3, 4, 5, 6, 7
But after decimation the order is
0, 4, 2, 6, 1, 5, 3, 7
At first it may look as if there is no order to this new sequence, BUT if the numbers are
represented as binary a patter soon becomes apparent.
What has happened is that the bit patterns representing the sample number has been reversed. This
new sequence is the order that the samples enter the FFT.
ALGORITHM TO IMPLEMENT FFT:
Step 1 - Select no. of points for FFT(Eg: 64)
Step 2 Generate a sine wave of frequency f (eg: 10 Hz with a sampling rate = No. of
Points of FFT(eg. 64)) using math library function.
Step 3 - Take sampled data and apply FFT algorithm.
Step 4 Use Graph option to view the Input & Output.
Step 5 - Repeat Step-1 to 4 for different no. of points & frequencies.
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C PROGRAM TO IMPLEMENT FFT:

Main.c (fft 256.c):
#include <math.h>
#define PTS 64
#define PI 3.14159265358979
//# of points for FFT
typedef struct {float real,imag;} COMPLEX;

void FFT(COMPLEX *Y, int n);
//FFT prototype
float iobuffer[PTS];
//as input and output buffer
float x1[PTS];
//intermediate buffer
short i;
//general purpose index variable
short buffercount = 0;
//number of new samples in iobuffer
short flag = 0;
//set to 1 by ISR when iobuffer full
COMPLEX w[PTS];
//twiddle constants stored in w
COMPLEX samples[PTS];
//primary working buffer
main()
{
for (i = 0 ; i<PTS ; i++)
// set up twiddle constants in w
{
w[i].real = cos(2*PI*i/(PTS*2.0)); //Re component of twiddle constants
w[i].imag =-sin(2*PI*i/(PTS*2.0)); //Im component of twiddle constants
}
for (i = 0 ; i < PTS ; i++) //swap buffers
{
iobuffer[i] = sin(2*PI*10*i/64.0);/*10- > freq,
64 -> sampling freq*/
samples[i].real=0.0;
samples[i].imag=0.0;
}
{
samples[i].real=iobuffer[i]; //buffer with new data
}
for (i = 0 ; i < PTS ; i++)
samples[i].imag = 0.0;
//imag components = 0
FFT(samples,PTS);
//call function FFT.c
for (i = 0 ; i < PTS ; i++) //compute magnitude
{
x1[i] = sqrt(samples[i].real*samples[i].real
+ samples[i].imag*samples[i].imag);
}
}
//end of main
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fft.c:
#define PTS 64
extern COMPLEX w[PTS];
void FFT(COMPLEX *Y, int N) //input sample array, # of points
{
COMPLEX temp1,temp2;
//temporary storage variables
int i,j,k;
//loop counter variables
int upper_leg, lower_leg;
//index of upper/lower butterfly leg
int leg_diff;
//difference between upper/lower leg
int num_stages = 0;
//number of FFT stages (iterations)
int index, step;
//index/step through twiddle constant
i = 1;
//log(base2) of N points= # of stages
do
{
num_stages +=1;
i = i*2;
}while (i!=N);
leg_diff = N/2;
//difference between upper&lower legs
step = (PTS*2)/N; //step between values in twiddle.h
for (i = 0;i < num_stages; i++) //for N-point FFT
{
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 = (Y[upper_leg]).real + (Y[lower_leg]).real;
temp1.imag = (Y[upper_leg]).imag + (Y[lower_leg]).imag;
temp2.real = (Y[upper_leg]).real - (Y[lower_leg]).real;
temp2.imag = (Y[upper_leg]).imag - (Y[lower_leg]).imag;
(Y[lower_leg]).real = temp2.real*(w[index]).real
-temp2.imag*(w[index]).imag;
(Y[lower_leg]).imag = temp2.real*(w[index]).imag
+temp2.imag*(w[index]).real;
(Y[upper_leg]).real = temp1.real;
(Y[upper_leg]).imag = temp1.imag;
}
index += step;
}
leg_diff = leg_diff/2;
step *= 2;
}
64
j = 0;
for (i = 1; i < (N-1); i++)
{
k = N/2;
while (k <= j)
{
j = j - k;
k = k/2;
}
j = j + k;
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//bit reversal for resequencing data
if (i<j)
{
temp1.real = (Y[j]).real;
temp1.imag = (Y[j]).imag;
(Y[j]).real = (Y[i]).real;
(Y[j]).imag = (Y[i]).imag;
(Y[i]).real = temp1.real;
(Y[i]).imag = temp1.imag;
}
}
return;
}
HOW TO PROCEED
1. Open Code Composer Studio; make sure the DSP kit is turned on.
2. Start a new project using Project-new pull down menu, save it in a
separate directory(C:\CCstudio_v3.1\myprojects) with name FFT.pjt.
3. Add the source files FFT256.c and FFT.C in the project using
Project
4. Add the linker command file hello.cmd
5. Add the rts file rts_EXT.lib
7. Project
Build
8. Load the program in program memory of DSP chip using the File-load program pull
down menu.
9. Run the program and observe output using graph utility.
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Input:
Output:
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POWER SPECTRUM
Description:The total or the average power in a signal is often not of as great an interest. We are most often
interested in the PSD or the Power Spectrum. We often want to see is how the input power has
been redistributed by the channel and in this frequency-based redistribution of power is where
most of the interesting information lies. The total area under the Power Spectrum or PSD is equal
to the total avg. power of the signal. The PSD is an even function of frequency or in other words
To compute PSD:
The value of the auto-correlation function at zero-time equals the total power in the signal. To
compute PSD We compute the auto-correlation of the signal and then take its FFT. The autocorrelation function and PSD are a Fourier transform pair. (Another estimation method called
period gram uses sampled FFT to compute the PSD.)
E.g.: For a process x(n) correlation is defined as:
R ( ) = E { x ( n ) x ( n + ) }
= lim
1
N
x(n ) x (n + )
n =1
Power Spectral Density is a Fourier transform of the auto correlation.

N 1
S ( ) = F T ( R ( )) = lim
R ( ) = F T
( S ( )) =
R ( ) e j
= N +1
1
2
S ( ) e
ALGORITHM TO IMPLEMENT PSD:
Step 1 - Select no. of points for FFT(Eg: 64)
Step 2 Generate a sine wave of frequency f (eg: 10 Hz with a sampling rate = No. of
Points of FFT (eg. 64)) using math library function.
Step 3 - Compute the Auto Correlation of Sine wave
Step4 - Take output of auto correlation, apply FFT algorithm.
Step 4 - Use Graph option to view the PSD.
Step 5 - Repeat Step-1 to 4 for different no. of points & frequencies.

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C Program to Implement PSD:

PSD.c:
/*************************************************************
* FILENAME
* Non_real_time_PSD.c
* DESCRIPTION
* Program to Compute Non real time PSD
***************************************************************
* DESCRIPTION
* Number of points for FFT (PTS)
* x --> Sine Wave Co-Efficients
* iobuffer --> Out put of Auto Correlation.
* x1 --> use in graph window to view PSD
/*===========================================================*/
#include <math.h>
#define PTS 128
#define PI 3.14159265358979

void FFT(COMPLEX *Y, int n);
//FFT prototype
float iobuffer[PTS];
//as input and output buffer
float x1[PTS],x[PTS];
//intermediate buffer
short i;
//general purpose index variable
short buffercount = 0;
//number of new samples in iobuffer
short flag = 0;
//set to 1 by ISR when iobuffer full
float y[128];
COMPLEX w[PTS];
COMPLEX samples[PTS];
//primary working buffer
main()
{
float j,sum=0.0 ;
int n,k,i,a;
for (i = 0 ; i<PTS ; i++)
// set up twiddle constants in w
{
w[i].real = cos(2*PI*i/(PTS*2.0));
/*Re component of twiddle constants*/
w[i].imag =-sin(2*PI*i/(PTS*2.0));
/*Im component of twiddle constants*/
}
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/****************Input Signal X(n) *************************/

for(i=0,j=0;i<PTS;i++)
{ x[i] = sin(2*PI*5*i/PTS);
// Signal x(Fs)=sin(2*pi*f*i/Fs);
samples[i].real=0.0;
samples[i].imag=0.0;
}
/********************Auto Correlation of X(n)=R(t) ***********/
for(n=0;n<PTS;n++)
{
sum=0;
for(k=0;k<PTS-n;k++)
{
sum=sum+(x[k]*x[n+k]); // Auto Correlation R(t)
}
iobuffer[n] = sum;
}
/********************** FFT of R(t) ***********************/
{
samples[i].real=iobuffer[i]; //buffer with new data
}
for (i = 0 ; i < PTS ; i++)
samples[i].imag = 0.0;
FFT(samples,PTS);
//imag components = 0
//call function FFT.c
/******************** PSD ********************/

for (i = 0 ; i < PTS ; i++) //compute magnitude
{
x1[i] = sqrt(samples[i].real*samples[i].real
+ samples[i].imag*samples[i].imag);
}
//end of main
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FFT.c:
#define PTS 128

extern COMPLEX w[PTS];
void FFT(COMPLEX *Y, int N) //input sample array, # of points

{
COMPLEX temp1,temp2;
//temporary storage variables
int i,j,k;
//loop counter variables
int upper_leg, lower_leg; //indexof upper/lower butterfly leg
int leg_diff;
//difference between upper/lower leg
int num_stages = 0;
//number of FFT stages (iterations)
int index, step;
//index/step through twiddle constant
i = 1;
//log(base2) of N points= # of stages
do
{
num_stages +=1;
i = i*2;
}while (i!=N);
leg_diff = N/2;
//difference between upper&lower legs
step = (PTS*2)/N;
//step between values in twiddle.h// 512
for (i = 0;i < num_stages; i++) //for N-point FFT
{
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 = (Y[upper_leg]).real + (Y[lower_leg]).real;
temp1.imag = (Y[upper_leg]).imag + (Y[lower_leg]).imag;
temp2.real = (Y[upper_leg]).real - (Y[lower_leg]).real;
temp2.imag = (Y[upper_leg]).imag - (Y[lower_leg]).imag;
(Y[lower_leg]).real = temp2.real*(w[index]).real
-temp2.imag*(w[index]).imag;
(Y[lower_leg]).imag = temp2.real*(w[index]).imag
+temp2.imag*(w[index]).real;
(Y[upper_leg]).real = temp1.real;
(Y[upper_leg]).imag = temp1.imag;
}
index += step;
}
leg_diff = leg_diff/2;
70
TI-DIVISION
step *= 2;
}
j = 0;
for (i = 1; i < (N-1); i++)
//bit reversal for resequencing data
{
k = N/2;
while (k <= j)
{
j = j - k;
k = k/2;
}
j = j + k;
if (i<j)
{
temp1.real = (Y[j]).real;
temp1.imag = (Y[j]).imag;
(Y[j]).real = (Y[i]).real;
(Y[j]).imag = (Y[i]).imag;
(Y[i]).real = temp1.real;
(Y[i]).imag = temp1.imag;
}
}
return;
}
HOW TO PROCEED
1. Open Code Composer Studio, make sure the DSP kit is turned on.
2. Start a new project using Project-new pull down menu, save it in a
separate directory(c:\ti\myprojects) with name PSD.pjt.
3. Add the source files PSD.c and FFT.c in the project using
Projectadd files to project pull down menu.
4. Add the linker command file hello.cmd .
5. Add the rts file rts_EXT.lib
7. Load the program in program memory of DSP chip using the File-load program pull
down menu.
8. Run the program and observe output using graph utility.
71
TI-DIVISION
INPUT:
OUTPUT
72
TI-DIVISION
73

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TMS320C5416 DSK

Cranes Software International Limited

Cranes Software Int. Ltd.

Copyright 2008 Cranes Software International Limited. All rights reserved.

Cranes Software Int. Ltd.

SINE WAVE GENERATION

SOLUTION OF DIFFERENCE EQUATIONS

TO VERIFY LINEAR CONVOLUTION

TO VERIFY CIRCULAR CONVOLUTION

PROCEDURE TO WORK IN REALTIME

TO DESIGN FIR FILTER (LOW PASS/HIGH PASS) USING WINDOWING

USING RECTANGULAR WINDOW

USING TRIANGULAR WINDOW

USING KAISER WINDOW

TO DESIGN IIR FILTER (LP/HP).

NOISE CANCELLATION USING ADAPTIVE FILTERS.

TO FIND THE FFT OF GIVEN 1-D SIGNAL AND PLOT

TO COMPUTE POWER DENSITY SPECTRUM OF A SEQUENCE

Cranes Software Int. Ltd.

233MHz or Higher PentiumCompatible CPU

500MHz or Higher Pentium

Supported Operating Systems

Windows NT 4.0 Service Pack 4 or higher

DSK HARDWARE INSTALLATION

Shut down and power off the PC

Plug the power cable into the board

Cranes Software Int. Ltd.

DSK SOFTWARE INSTALLATION

Insert the installation CD into the CD-ROM drive

Cranes Software Int. Ltd.

Choose the option to install Code Composer Studio

If you already have C5000 CC Studio IDE installed on your PC,

Respond to the dialog boxes as the installation program runs.

5416 DSK CCStudio v3.1.lnk

Cranes Software Int. Ltd.

Use StartProgramsTexas InstrumentsCode Composer StudioCode Composer Studio

5416 DSK Diagnostics Utility v3.1.lnk

If the board still fails to detect

Cranes Software Int. Ltd.

Cranes Software Int. Ltd.

TMS320C5416 DSK BOARD FEATURES

Cranes Software Int. Ltd.

Key Features of the C5416 DSK

Block diagram of 5416 DSK

Cranes Software Int. Ltd.

INTRODUCTION TO CODE COMPOSER STUDIO

Cranes Software Int. Ltd.

Starting Code Composer

5416 DSK CCStudio v3.1.lnk

Procedure to work on Code Composer Studio

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2. To Create a Source file

Type the code in editor window.

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3. To Add Source files to Project

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4. To Add rts_EXT.lib file.

ISO C/C++ standard library

Intrinsic arithmetic routines

System startup routine, _c_int00

Functions and macros that allow C/C++ to access specific instructionsrts5400.lib

Cranes Software Int. Ltd.