D.S.P/MATLAB Theory & Experiments

DSP LAB RECORD

SCT COLLEGE OF ENGINEERING Page 1

CONTENTS

1. INTRODUCTION TO TMS320C6713 DSK ........................................................... 2

2. FEATURES OF C6713 DSK................................................................................... 4

3. INTRODUCTION TO CODE COMPOSER STUDIO ............................................ 6

4. LINEAR & CIRCULAR CONVOLUTION .......................................................... 11

5. SINE WAVE AND SQUARE WAVE ................................................................... 15

6. TMS320C6713 DSK CODEC (TLV320AIC23) CONFIGURATION USING -

BOARD SUPPORT LIBRARY .................................................................................. 19

7. SINE WAVE USING DSK6713 ............................................................................ 21

8. ADVANCED DISCRETE TIME FILTER ( FIR ) ................................................. 23

9. AM FM & PWM USING MATLAB ..................................................................... 27

10. CONVOLUTION .................................................................................................. 31

11. FIR FILTERS......................................................................................................... 33

12. BUTTERWORTH IIR FILTERS .......................................................................... 36

13. CHEBYSHEV TYPE 1 FILTERS ......................................................................... 44

14. CHEBYSHEV TYPE 2 FILTERS ......................................................................... 50

DSP LAB RECORD


INTRODUCTION TO TMS320C6713 DSK

The C6713 DSK builds on Ti‟s industry-leading line of low cost, easy-to-use DSP

Starter Kit (DSK) development boards. The high performance board features the

TMS320C6713 floating point DSP, capable of performing 1350 million floating point

operations per second(MFLOPS) , the C6713 DSP makes the C6713 DSK the most

powerful DSK development board.

The DSK is USB port interfaced platform that allows efficiently develop and testing

applications for C6713. The DSK consists of a C6713-based printed circuit board that

will serve as a hardware reference design for TI‟s customer‟s product s. With the

extensive host PC and target DSP software support, including bundled TI tools, the

DSK provides ease-of -use and capabilities that are attractive to DSP engineers.

The C6713 DSK has a TMS 320C6713 DSP onboard that allows full speed

verification of the Code Composer Studio. The C6713 DSK provides

A USB interface

SDRAM and ROM

An analog interface circuit for data conversion (AIC)

An I/O port

Embedded JTAG emulation support

Connectors on the C6713 DSK provide DSP external memory interface (EMIF) and

peripheral signals that enable its functionality to be expanded with custom or third

party daughter boards.

The DSK provides a C6713 hardware reference design that can assist in the

development of C6713 based products. In addition to providing a reference for

interfacing the DSP to various types of memories and peripheral interfaces, the design

also address power, clock, JTAG and parallel peripheral interfaces.

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The C6713 DSK includes a stereo codec. The analog interface circuit (AIC) has the

following characteristics:

High performance Stereo Codec

90 dB SNR Multibit Sigma –Delta ADC (A- weighted at 48kHz)

100 dB SNR Multibit Sigma Delta DAC (A- weighted at 48kHz)

1.42 V – 3.6 V Core Digital Supply: compatible With TI C54x DSP Core

Voltages

2.7 V – 3.6 V Buffer and Analog Supply: Compatible with both TI C54x DSP

Buffer Voltages

8 kHz – 96 KHz Sampling Frequency Support

Software Control via TI McBSP –Compatible Multiprotocol Serial Port

12 C compatible and SPI – compatible Serial Port Protocols

Glueless Interface to TI McBSP's

Audio- Data Input/Output via TI McBSP- Compatible Programmable Audio Interface

12 S- compatible interface requiring only one McBSP for both ADC and DAC

Standard 12S MSB or LSB justified – Data Transfers

16/20/24/32- Bit Word Length

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FEATURES OF C6713 DSK

The 6713 DSK is a low-cost standalone development platform that enables customers

to evaluate and develop applications for the TI C67XX DSP family. The DSP also

serves as a hardware reference design for the TMS320C6713 DSP. Schematics, logic

equations and application notes are available to ease hardware development and

reduce time to market.

The DSK uses the 32bit EMIF for the SDRAM (CE0) and daughtercard expansion

interface ( CE2 and CE3). The Flash is attached to CE1 of the EMIF in 8-bit mode.

An on-board AIC23 codec allows the DSP to transmit and receive analog signals.

McBSP0 is used for the codec control interface and McBSP1 is used for data. Analog

audio I/O is done through four 3.5mm audio jacks that correspond to microphone

input, line input, line output and headphone output. The codec can select the

microphone or line input as the active input. The analog output is driven to both the

line out (fixed gain) and headphone (adjustable gain) connectors. McBSP1 can be re-

routed to the expansion connectors in software.

A programmable logic device called CPLD is used to implement glue logic that ties

the board components together. The CPLD has a register based user interface the lets

the user configure the board by reading and writing to the CPLD registers. The

registers reside at the midpoint of CE1.

The DSK include 4 LEDs and 4 DIP switches as a simple way to provide the user

with interactive feedback. Both are accessed by the reading and writing to the CPLD

register.

An included 5V external power supply is used to power is used to power the board.

On-board voltage regulators provide 1.26V DSP core voltage, 3.3V digital and 3.3V

analog voltages. A voltage supervisor monitors the internally generated voltage and

will hold the board in reset until the supplies are within operating specifications and

the reset button is released. If desired, JP1 and JP2 can be used as power test points

the core and I/O power supplies.

Code Composer communicates with the DSP through an embedded JTAG emulator

with a USB host interface. The DSK can also be used with an external emulator

through the external JTAG connector.

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DSP LAB RECORD


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 like MS- based

C++, 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 languages and much more.

CODE COMPOSER FEATURE 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

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PROCEDURE TO WORK ON CODE COMPOSER STUDIO

1. To create a new project

Project New

2. To create source file

File New Source file

Save the source file in project folder

File Save

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3. To add source file to the project

Project Add files to project <source file>

4. To add rts6700.lib & hello.cmd

Project Add files to project rts6700.lib

Path : C:\CCstudio\c6000\lib\rts6700.lib

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

Project Add files to project hello.cmd

Path: C:\CCstudio\tutorial\dsk6713\hello1\hello.cmd

Note: Select Linker command file (*.cmd) in Type of file

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5. To compile

Project Compile file ( or use icon or ctrl+F7 )

6. To build or link

Project build (or use F7 )

(which will create a .out file in project folder)

7. To load the program:

File Load Program <select the .out file in debug folder in project folder>

8. To run the program

Debug Run ( or use icon or F5)

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To perform Single Step Debugging:

1. Keep the cursor on the line from point to start single step debugging.

To set the break point click icon from tool bar menu.

2. Load the .out file onto the target

3. Go to View and select Watch window

4. Select Debug Run (F5)

5. Execution should halt at break point

6. Now press F10. See the changes happening 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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Experiment no 1

LINEAR & CIRCULAR CONVOLUTION

AIM

To implement circular and linear convolution using Code Composer Studio.

PROCEDURE

1. Open Code Composer Setup and select C6713 simulator, click save and quit

2. Start a new project using „Project New‟ pull down menu, save it in a

separate directory (D:\My projects) with file name lconv.pjt

3. Create a new source file using File New Source file menu and save it in

the project folder(linear.c)

4. Add the source file (linear.c) to the project

Project Add files to Project Select linear.c

5. Add the linker command file hello.cmd

Project Add files to Project

(path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd)

6. Add the run time support library file rts6700.lib


(path: C\CCStudio\cgtools\lib\rts6700.lib)

7. Compile the program using „project Compile‟ menu or by Ctrl+F7

8. Build the program using „project Build‟ menu or by F7

9. Load the linear.out file (from project folder lcconv\Debug) using

File Load Program

10. Run the program using „Debug Run‟ or F5

11. To view the output graphically

Select View Graph Time and Frequency

12. Repeat the steps 2 to 11 for circular convolution

PROGRAM

Linear Convolution

#include<stdio.h>

#define LENGTH1 6

#define LENGTH2 4

int x[LENGTH1+LENGTH2-1]={1,2,3,4,5,6,0,0,0};

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int h[LENGTH1+LENGTH2-1]={1,2,3,4,0,0,0,0,0};

int y[LENGTH1+LENGTH2-1];

main()

{

int i=0,j;

for(i=0;i<LENGTH1 + LENGTH2-1;i++)

{

y[i]=0;

for(j=0;j<=i;j++)

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

}

for(i=0;i<LENGTH1 + LENGTH2-1;i++)

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

}

RESULT

The convoluted sequence is obtained as

1 4 10 20 30 40 43 38 24

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Circular convolution

#include<stdio.h>

int m,n,x[30],h[30],y[30],i,j,temp[30],k,x2[30],a[30];

void main()

{

printf("Enter the length of 1st sequence\n");

scanf("%d",&m);

printf("Enter the length of 2nd sequence\n");

scanf("%d",&n);

printf("Enter the 1st sequence\n");

for(i=0;i<m;i++)

scanf("%d",&x[i]);

printf("Enter the 2nd sequence\n");

for(j=0;j<n;j++)

scanf("%d",&h[j]);

if(m-n!=0)

{

if(m>n)

{

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++)

a[j]=h[n-j];

for(i=0;i<n;i++)

y[0]+=x[i]*a[i];

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for(k=1;k<n;k++)

{

y[k]=0;

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];

}

}

printf("The circular convolution is \n");

for(i=0;i<n;i++)

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

}

RESULT

Enter the length of 1st sequence 4

Enter the length of 2nd sequence 4

Enter the 1st sequence 3 2 1 4

Enter the 2nd sequence 1 2 3 4

The circular convolution is 22 24 30 24

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Experiment 2

SINE WAVE AND SQUARE WAVE

AIM

To generate a sine wave and square wave using C6713 simulator

PROCEDURE

1. Open Code Composer Setup and select C6713 simulator, click save and quit

2. Start a new project using „Project New‟ pull down menu, save it in a

separate directory (D:\My projects) with file name sinewave.pjt

3. Create a new source file using File New Source file menu and save it in

the project folder(sinewave.c)

4. Add the source file (sinewave.c) to the project

Project Add files to Project Select sinewave.c

5. Add the linker command file hello.cmd


(path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd)

6. Add the run time support library file rts6700.lib


(path: C\CCStudio\cgtools\lib\rts6700.lib)

7. Compile the program using „project Compile‟ menu or by Ctrl+F7

8. Build the program using „project Build‟ menu or by F7

9. Load the sinewave.out file (from project folder lcconv\Debug) using

File Load Program

10. Run the program using „Debug Run‟ or F5

11. To view the output graphically

Select View Graph Time and Frequency

12. Repeat the steps 2 to 11 for square wave

PROGRAM

Sine Wave

#include <stdio.h>

#include <math.h>

float a[500];

void main()

{

int i=0;

for(i=0;i<500;i++)

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{

a[i]=sin(2*3.14*10000*i);

}

}

Square wave

#include <stdio.h>

#include <math.h>

int a[1000];

void main()

{

int i,j=0;

int b=5;

for(i=0;i<10;i++)

{ for (j=0;j<=50;j++)

{

a[(50*i)+j]=b;

}

b=b*(-1) ;

}

}

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RESULT

The sine wave and square wave has been obtained.

Sine Wave

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Square Wave:

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Experiment 3

TMS320C6713 DSK CODEC (TLV320AIC23)

CONFIGURATION USING BOARD SUPPORT LIBRARY

AIM

To configure the codec TLV320AIC23 for a talk through program using the board

support library

PREREQUISITES

TMS320C6713 DSP starter kit, PC with Code Composer Studio, CRO, Audio source,

Speakers, Signal Generator.

PROCEDURE

1. Connect Speaker to the LINE OUT socket.

2. Connect the line out from the PC to the LINE IN socket.

3. Now switch ON the DSK and bring up Code Composer Studio on PC

4. Create a new project with name codec.pjt

5. From File menu New DSP/BIOS Configuration Select dsk6713.cdb

and save it as “xyz.cdb”

6. Add xyz.cdb to the current project

7. Create a new source file and save it as codec.c

8. Add the source file codec.c to the project

9. Add the library file “dsk6713bsl.lib” to the project

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

10. Copy files “dsk6713.h” and “dsk6713_aic23.h” to the Project folder

11. Build (F7) and load the program to the DSP Chip ( File Load Program)

12. Run the program (F5)

13. Give an audio output from the PC and notice the output in the speaker

14. Vary the sampling frequency using the DSK6713_AIC23_SetFreq

PROGRAM

Codec.c

#include"xyzcfg.h"

#include"dsk6713.h"

#include"dsk6713_aic23.h"

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DSK6713_AIC23_Config config= DSK6713_AIC23_DEFAULTCONFIG;

void main()

{

DSK6713_AIC23_CodecHandle hCodec;

int l_input,r_input,l_output,r_output;

DSK6713_init();

hCodec=DSK6713_AIC23_openCodec(0,&config);

DSK6713_AIC23_setFreq= DSK6713_AIC23_FREQ_48KHZ;

while(1)

{

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

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

r_output = r_input;

l_output = l_input;

while(!DSK6713_AIC23_write(hCodec,l_output));

while(!DSK6713_AIC23_write(hCodec,r_output));

}

DSK6713_AIC23_closeCodec(hCodec);

}

RESULT

The Codec TMS320AIC23 Successfully configured using the board support library

and output is verified.

DSP LAB RECORD


Experiment 4

SINE WAVE USING DSK6713

AIM

To generate a real time sine wave using TMS320C6713 DSK

PROCEDURE



3. Create a new project with name sinewave.pjt




6. Create a new source file and save it as sinewave.c

7. Add the source file sinewave.c to the project







PROGRAM

Sinewave.c

#include "xyzcfg.h"

#include "dsk6713.h"

#include "dsk6713_aic23.h"

#include <stdio.h>

#include <math.h>

float a[500],b;

DSK6713_AIC23_Config config= DSK6713_AIC23_DEFAULTCONFIG;

void main()

{

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int i=0;


Int l_output,r_output;

DSK6713_init();

hCodec=DSK6713_AIC23_openCodec(0,&config);

DSK6713_AIC23_setFreq=DSK6713_AIC23_FREQ_48KHZ;

for(i=0;i<500;i++)

{

a[i]=sin(2*3.14*10000*i);

}

while(1)

{

for(i=0;i<500;i++)

{

b=400*a[i];

while(!DSK6713_AIC23_write(hCodec,b));

}

}


}

RESULT

The sine wave tone have been successfully generated and obtained through speaker

DSP LAB RECORD


Experiment 5

ADVANCED DISCRETE TIME FILTER ( FIR )

AIM

To generate filter coefficients using MATLAB and to implement Audio filter using

DSK6713

PROCEDURE


2. Connect the line out from the PC to the LINE IN socket.


4. Create a new project with name firfilter.pjt




7. Create a new source file and save it as firfilter.c

8. Add the source file firfilter.c to the project







DSP LAB RECORD


Procedure for Generating Filter Coefficients in MATLAB

1. Open Matlab

2. Start Toolboxes Filter Design Filter Design & Analysis Tool

3. Select the filter type and give the order and specifications for the filter and

click Design filter

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4. From the File option select Export (or use ctrl + E)

5. Select export to Coefficient File(ASCII) and save the coefficients as txt file

6. Use these coefficients to form the coefficients in the C-file.

PROGRAM

#include"xyzcfg.h"

#include"dsk6713.h"

#include"dsk6713_aic23.h"

#define FIR_ORD 20

short fir_filter(Uint32 in,short *in_buff,float *coeff);

float fir_coeff[FIR_ORD+1]={ -0.03076334540056, -0.005410850195565,

0.03739961263014, 0.02059285910535, -0.04315598597689, -0.04556930460411,

0.04761022772871, 0.09663870571612, -0.05042696667111, -0.3229392723481,

0.5652971907268, -0.3229392723481, -0.05042696667111, 0.09663870571612,

0.04761022772871, -0.04556930460411, -0.04315598597689, 0.02059285910535,

0.03739961263014, -0.005410850195565, -0.03076334540056 };

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short x_buff[FIR_ORD+1];

DSK6713_AIC23_Config config = DSK6713_AIC23_DEFAULTCONFIG;

void main()

{


Uint32 l_input,r_input,l_output,r_output;

DSK6713_init();

hCodec = DSK6713_AIC23_openCodec(0,&config);

DSK6713_AIC23_setFreq(hCodec,DSK6713_AIC23_FREQ_48KHZ);

while(1)

{

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

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

l_output=5*fir_filter(l_input,x_buff,fir_coeff);

r_output=5*fir_filter(l_input,x_buff,fir_coeff);

while(!DSK6713_AIC23_write(hCodec,l_output));

while(!DSK6713_AIC23_write(hCodec,r_output));

}


}

short fir_filter(Uint32 in, short *in_buff, float *coeff)

{

int out=0;

int i;

for(i=FIR_ORD;i>0;i--)

{

in_buff[i]=in_buff[i-1];

out=out+in_buff[i]*coeff[i];

}

in_buff[0]=in;

out=out+in_buff[0]*coeff[0];

return out;

}

RESULT

The FIR filter has been implemented and the output is verified

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Experiment 6

AM FM & PWM USING MATLAB

AIM

To implement a program in MATLAB to generate AM, FM and PWM wave.

PROGRAM

Amplitude Modulated wave

clc; clear all; close all;

t = 0:0.0001:0.015;

m = input(„Modulation Index = ‟);

fm=input(„Frequency of input wave=‟);

fc=input(„Frequency of carrier wave=‟);

x=sin(2*pi*fm*t);

y=sin(2*pi*fc*t);

e=(1+(m.*x)).*y;

plot(e);

title(„AM WAVE‟);

xlabel(„TIME INDEX‟);

ylabel(„AMPLITUDE‟);

RESULT

Modulation Index = 0.5

Frequency of input wave= 100

Frequency of carrier wave= 1000

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Frequency modulated wave

clc; close all; clear all;

a= input („enter amplitude‟);

fc=input(„enter carrier freq‟);

fm=input(„enter modulation freq‟);

m=input(„enter modulation index‟);

n=0:0.001:.2;

y=a*sin(2*pi*fc*n-(m*cos(2*pi*fm*n)));

plot(n,y);

title(„FM‟);

xlabel(„TIME‟);

ylabel(„AMPLITUDE‟);

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RESULT

enter amplitude = 1

enter carrier freq =100

enter modulation freq =10

enter modulation index =5

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Pulse Width Modulated Wave


fs=100;

t= 0:1/(5*fs):2;

x=sawtooth(2*pi*20*t);

m=0.75*sin(2*pi*t);

k=length(x);

for i=1:k

if (m(i)>=x(i))

pwm(i)=1;

else if (m(i)<x(i)

pwm(i)=0;

end;

end;

subplot(2,2,3);

plot(t,pwm,t,m);

axis([0,0.5,-2,2]);

RESULT

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Experiment 7

CONVOLUTION

AIM

To write a program in MATLAB for convolution of two sequence.

PROGRAM


a=input(„enter first sequence ‟);

b= input(„enter second sequence ‟);

c= fliplr(b);

d=length(a);

e=length(b);

f=[zeros(1,e-1),a,zeros(1,e)];

g=0;

i=d+e-1;

for h=0:(d+e-1);

j=[zeros(1,g),c,zeros(1,i)];

k=sum(f.*j);

con(g+1)=k;

g=g+1;

i= i-1;

end

m= 0:d+e-1;

stem(m,con);

title(„convoluted sequence‟);

xlabel(„time‟);

ylabel(„amplitude‟);

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RESULT

enter first sequence [1 2 3 4]

enter second sequence [4 3 2 1 ]

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Experiment 8

FIR FILTERS

AIM

To write a program in MATLAB for implementing FIR filters.

PROGRAM


rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple ');

fp= input('Enter the passband frequency ');

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

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

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

y=boxcar(n1);

%LOW PASS FILTER

b=fir1(n,wp,y);

[h,o]=freqz(b,1,256);

m=20*log10(abs(h));

subplot(2,2,1);

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plot(o/pi,m);

title('1. LPF');

ylabel('Gain(dB)');

xlabel(' Normalised Frequency');

%HIGH PASS FILTER

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('2. HPF');

ylabel('Gain(dB)');


%BAND PASS FILTER

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('3. BPF');

ylabel('Gain(dB)');


%BAND STOP FILTER

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);

DSP LAB RECORD


title('4. BSF');

ylabel('Gain(dB)');


RESULT

Enter the passband ripple .05

Enter the stopband ripple .04

Enter the passband frequency 1500

Enter the stopband frequency 2000

Enter the sampling frequency 9000

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Experiment 9

BUTTERWORTH IIR FILTERS

AIM

To design and simulate Butterworth IIR filters using MATLAB

PROGRAM

Low Pass Filter




wp= input('Enter the passband frequency ');

ws=input('Enter the stopband frequency ');

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

w1=2*wp/fs;

w2=2*ws/fs;

[n,wn]=buttord(w1,w2,rp,rs);

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

w=0:0.01:pi;

[h,om]=freqz(b,a,w);

m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('IIR LPF');

ylabel('Gain(dB)');

xlabel('Normalised frequency');

subplot(2,1,2);

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plot(om/pi,an);

ylabel('Phase(radians)');


RESULT


Enter the stop band ripple 50




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HPF







w1=2*wp/fs;

w2=2*ws/fs;


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

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('IIR HPF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






DSP LAB RECORD


BPF







w1=2*wp/fs;

w2=2*ws/fs;


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

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('IIR BPF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT



Enter the passband frequency [800 1200]

Enter the stopband frequency [500 1500]


DSP LAB RECORD


BSF







w1=2*wp/fs;

w2=2*ws/fs;


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

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('IIR BSF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






DSP LAB RECORD


Experiment 10

CHEBYSHEV TYPE 1 FILTERS

AIM

To design and simulate Chebyshev type 1 filters using MATLAB

PROGRAM

LPF







w1=2*wp/fs;

w2=2*ws/fs;

[n,wn]=cheb1ord(w1,w2,rp,rs);

[b,a]=cheby1(n,rp,wn);

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 1 LPF');

ylabel('Gain(dB)');


subplot(2,1,2);

DSP LAB RECORD


plot(om/pi,an);



RESULT






DSP LAB RECORD


HPF







w1=2*wp/fs;

w2=2*ws/fs;


[b,a]=cheby1(n,rp,wn,'high');

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 1 HPF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






DSP LAB RECORD


BSF







w1=2*wp/fs;

w2=2*ws/fs;


[b,a]=cheby1(n,rp,wn,'stop');

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 1 BSF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






DSP LAB RECORD


Experiment 11

CHEBYSHEV TYPE 2 FILTERS

AIM

To design and simulate Chebyshev type 2 filters using MATLAB

PROGRAM

LPF







w1=2*wp/fs;

w2=2*ws/fs;



w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 2 LPF');

ylabel('Gain(dB)');


DSP LAB RECORD


subplot(2,1,2);

plot(om/pi,an);



RESULT






DSP LAB RECORD


HPF







w1=2*wp/fs;

w2=2*ws/fs;


[b,a]=cheby2(n,rp,wn,'high');

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 2 HPF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






DSP LAB RECORD


BPF







w1=2*wp/fs;

w2=2*ws/fs;

[n wn]=cheb2ord(w1,w2,rp,rs);


w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 2 BPF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






DSP LAB RECORD


BSF







w1=2*wp/fs;

w2=2*ws/fs;

[n wn]=cheb2ord(w1,w2,rp,rs);

[b,a]=cheby2(n,rp,wn,'stop');

w=0:0.01:pi;


m=20*log10(abs(h));

an=angle(h);

subplot(2,1,1);

plot(om/pi,m);

title('Chebyshev Type 2 BSF');

ylabel('Gain(dB)');


subplot(2,1,2);

plot(om/pi,an);



DSP LAB RECORD


RESULT






Documents

D.S.P/MATLAB Theory & Experiments