DSP LAB RECORD

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

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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 rerouted 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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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 Project  Add files to Project (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]; SCT COLLEGE OF ENGINEERING Page 13

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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 Enter the length of 2nd sequence Enter the 1st sequence Enter the 2nd sequence The circular convolution is 3 1 22 4 4 2 2 1 3 24 4 4 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 Project  Add files to Project (path: C:\CCstudio\tutorial\dsk6713\hello\hello.cmd) 6. Add the run time support library file rts6700.lib Project  Add files to Project (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++) SCT COLLEGE OF ENGINEERING Page 15

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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. 2. 3. 4. 5. Connect Speaker to the LINE OUT socket. Connect the line out from the PC to the LINE IN socket. Now switch ON the DSK and bring up Code Composer Studio on PC Create a new project with name codec.pjt 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" SCT COLLEGE OF ENGINEERING Page 19

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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.

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Experiment 4

SINE WAVE USING DSK6713

AIM
To generate a real time sine wave using TMS320C6713 DSK

PROCEDURE
1. 2. 3. 4. Connect Speaker to the LINE OUT socket. Now switch ON the DSK and bring up Code Composer Studio on PC Create a new project with name sinewave.pjt From File menu  New  DSP/BIOS Configuration  Select dsk6713.cdb and save it as “xyz.cdb” 5. Add xyz.cdb to the current project 6. Create a new source file and save it as sinewave.c 7. Add the source file sinewave.c to the project 8. Add the library file “dsk6713bsl.lib” to the project ( Path: C:\CCStudio\C6000\dsk6713\lib\dsk6713bsl.lib) 9. Copy files “dsk6713.h” and “dsk6713_aic23.h” to the Project folder 10. Build (F7) and load the program to the DSP Chip ( File  Load Program) 11. Run the program (F5) 12. Give an audio output from the PC and notice the output in the speaker

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() { SCT COLLEGE OF ENGINEERING Page 21

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int i=0; DSK6713_AIC23_CodecHandle hCodec; 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)); } } DSK6713_AIC23_closeCodec(hCodec); }

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

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Experiment 5

ADVANCED DISCRETE TIME FILTER ( FIR )

AIM
To generate filter coefficients using MATLAB and to implement Audio filter using DSK6713

PROCEDURE
1. 2. 3. 4. 5. Connect Speaker to the LINE OUT socket. Connect the line out from the PC to the LINE IN socket. Now switch ON the DSK and bring up Code Composer Studio on PC Create a new project with name firfilter.pjt 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 firfilter.c 8. Add the source file firfilter.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

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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, 0.5652971907268, -0.3229392723481, -0.05042696667111, -0.3229392723481, -0.05042696667111, 0.09663870571612,

0.04761022772871, -0.04556930460411, -0.04315598597689, 0.02059285910535, 0.03739961263014, -0.005410850195565, -0.03076334540056 }; SCT COLLEGE OF ENGINEERING Page 25

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short x_buff[FIR_ORD+1]; DSK6713_AIC23_Config config = DSK6713_AIC23_DEFAULTCONFIG; void main() { DSK6713_AIC23_CodecHandle hCodec; Uint32 l_input,r_input,l_output,r_output; DSK6713_init(); hCodec = DSK6713_AIC23_openCodec(0,&config); DSK6713_AIC23_setFreq(hCodec,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)); } DSK6713_AIC23_closeCodec(hCodec); } 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 SCT COLLEGE OF ENGINEERING Page 26

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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
clc; close all; clear all; 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
clc; close all; clear all; 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
clc; clear all; close all; 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); SCT COLLEGE OF ENGINEERING Page 33

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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)'); xlabel(' Normalised Frequency'); %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)'); xlabel(' Normalised Frequency'); %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); SCT COLLEGE OF ENGINEERING Page 34

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title('4. BSF'); ylabel('Gain(dB)'); xlabel(' Normalised Frequency');

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
clc; clear all; close all; rp= input('Enter the passband ripple ');

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

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency 200 Enter the stopband frequency 400 Enter the sampling frequency 1000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

HPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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,'high'); 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 HPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency 1200 Enter the stopband frequency 1000 Enter the sampling frequency 4000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

BPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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 BPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .4

Enter the stop band ripple 50 Enter the passband frequency [800 1200] Enter the stopband frequency [500 1500] Enter the sampling frequency 4000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

BSF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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,'stop'); 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 BSF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .4

Enter the stop band ripple 50 Enter the passband frequency [800 1500] Enter the stopband frequency [1000 1200] Enter the sampling frequency 4000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

Experiment 10

CHEBYSHEV TYPE 1 FILTERS

AIM
To design and simulate Chebyshev type 1 filters using MATLAB

PROGRAM LPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb1ord(w1,w2,rp,rs); [b,a]=cheby1(n,rp,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('Chebyshev Type 1 LPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2);

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency 1000 Enter the stopband frequency 1500 Enter the sampling frequency 4000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

HPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb1ord(w1,w2,rp,rs); [b,a]=cheby1(n,rp,wn,'high'); 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('Chebyshev Type 1 HPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency 1500 Enter the stopband frequency 1000 Enter the sampling frequency 5000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

BSF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb1ord(w1,w2,rp,rs); [b,a]=cheby1(n,rp,wn,'stop'); 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('Chebyshev Type 1 BSF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency [1000 1200] Enter the stopband frequency [800 1500] Enter the sampling frequency 4500

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

Experiment 11

CHEBYSHEV TYPE 2 FILTERS
AIM
To design and simulate Chebyshev type 2 filters using MATLAB

PROGRAM LPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb2ord(w1,w2,rp,rs); [b,a]=cheby2(n,rp,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('Chebyshev Type 2 LPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); SCT COLLEGE OF ENGINEERING Page 50

DSP LAB RECORD

subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency 1000 Enter the stopband frequency 1500 Enter the sampling frequency 4000

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

HPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb2ord(w1,w2,rp,rs); [b,a]=cheby2(n,rp,wn,'high'); 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('Chebyshev Type 2 HPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency 1000 Enter the stopband frequency 800 Enter the sampling frequency 4500

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

BPF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb2ord(w1,w2,rp,rs); [b,a]=cheby2(n,rp,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('Chebyshev Type 2 BPF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency [800 1500] Enter the stopband frequency [1000 1200] Enter the sampling frequency 4500

SCT COLLEGE OF ENGINEERING

Page 55

DSP LAB RECORD

BSF
clc; clear all; close all; rp= input('Enter the passband ripple ');

rs= input('Enter the stop band ripple '); 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]=cheb2ord(w1,w2,rp,rs); [b,a]=cheby2(n,rp,wn,'stop'); 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('Chebyshev Type 2 BSF'); ylabel('Gain(dB)'); xlabel('Normalised frequency'); subplot(2,1,2); plot(om/pi,an); ylabel('Phase(radians)'); xlabel('Normalised frequency');

SCT COLLEGE OF ENGINEERING

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DSP LAB RECORD

RESULT
Enter the passband ripple .5

Enter the stop band ripple 50 Enter the passband frequency [800 1500] Enter the stopband frequency [1000 1200] Enter the sampling frequency 4500

SCT COLLEGE OF ENGINEERING

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