H.S.

Prashantha, Dept of TE, PESIT

DSP LABORATORY MANUAL Semester : V Sub Code: 06ECL57

Prepared by H.S.Prashantha Assistant Professor Department of Telecommunication PES Institute of Technology Bangalore

H.S.Prashantha, Dept of TE, PESIT DSP LABORATORY MANUAL Semester : V CONTENTS DSP using Matlab 1. Verification of sampling theorem. 2. Impulse response of a given system. 3. Linear convolution of two given sequences. 4. Circular convolution of two given sequences. 5. Autocorrelation of a given sequence and verification of its properties. 6. Cross correlation of a given sequence and verification of its properties 7. Solving a given difference equation. 8. Computation of N point DFT of a given sequence and to plot magnitude and phase Spectrum. 9. Linear convolution of two given sequences using DFT and IDFT. 10. Circular convolution of two given sequences using DFT and IDFT 11. Design and implementation of FIR filter to meet given specification. 12. Design and implementation of IIR filter to meet given specification. DSP Experiments using DSP processor 1. Linear convolution of two given sequences. 2. Circular convolution of two given sequences. 3. Computation of N point DFT of a given sequence. 4. Realization of an FIR filter (any type) to meet given specification. The input can be a signal from function generator/speech signal. 5. Audio specification such as to plot time and frequency display of microphone plus a cosine using DSP. Read a wav file and match with their respective spectrograms. 6. Noise removal: Add noise above 3KHz and then remove; Interference suppression using 400 Hz ton. 7. Impulse response of a given system of a given system of first and second order. Sub Code: 06ECL57

H.S.Prashantha, Dept of TE, PESIT Cycle I DSP using Matlab 1. Verification of sampling theorem. 2. Impulse response of a given system. 3. Linear convolution of two given sequences. 4. Circular convolution of two given sequences. 5. Autocorrelation of a given sequence and verification of its properties. 6. Cross correlation of a given sequence and verification of its properties 7. Solving a given difference equation. DSP Experiments using DSP processor 8. Linear convolution of two given sequences. 9. Circular convolution of two given sequences. 10. Impulse response of a given system of a given system of first and second order. Cycle II DSP using Matlab 1. Computation of N point DFT of a given sequence and to plot magnitude and phase Spectrum. 2. Linear convolution of two given sequences using DFT and IDFT. 3. Circular convolution of two given sequences using DFT and IDFT 4. Design and implementation of FIR filter to meet given specification. 5. Design and implementation of IIR filter to meet given specification. DSP Experiments using DSP processor 1. Computation of N point DFT of a given sequence. 2. Realization of an FIR filter (any type) to meet given specification. The input can be a signal from function generator/speech signal. 3. Audio specification such as to plot time and frequency display of microphone plus a cosine using DSP. Read a wav file and match with their respective spectrograms. 4. Noise removal: Add noise above 3KHz and then remove; Interference suppression using 400 Hz ton.

H.S.Prashantha, Dept of TE, PESIT

Cycle 1
DSP USING MATLAB MATLAB is a high-level language and interactive environment that enables you to perform computationally intensive tasks faster than with traditional programming languages such as C, C++ etc. The key features of MATLAB are as follows High-level language for technical computing Development environment for managing code, files, and data Interactive tools for iterative exploration, design, and problem solving Mathematical functions for linear algebra, statistics, Fourier analysis, filtering, optimization, and numerical integration 2-D and 3-D graphics functions for visualizing data Tools for building custom graphical user interfaces
MATLAB (MATrix LABoratory) is a special purpose computer program optimized to perform engineering and scientific calculations. The MATLAB program implements the MATLAB language, and provides a very extensive library of predefined functions to make technical programming tasks easier and more efficient. The extremely wide variety of functions makes it much easier to solve technical problems in MATLAB than in other languages such as FORTRAN or C. The advantages of using MATLAB for technical programming are: Ease of use Platform independence Predefined functions Device-independent plotting Graphical User Interface MATLAB Compiler

MATLAB is a huge program, with an incredibly rich variety of functions. There are more than 1000 functions in the basic MATLAB.
The disadvantages of MATLAB are: It is interpreted language, and therefore can execute more slowly than compiled languages. Cost. (Student version of MATLAB is available for low cost with almost identical feature) full copy of MATLAB is 5 TO 10 times more expensive than a conventional C or FORTRAN compiler.

The functional unit of data in any MATLAB program is the array. An array is a collection of data values organized into rows and columns. When MATLAB executes, it can display several types of windows that accept commands or display information. The three most important types of windows are command windows, where commands may be entered: figure windows, which display plots and graphs: and edit/debug windows which permit a user to create and modify MATLAB programs. The major tools within or accessible from the MATLAB desktop are: The command window: a user can enter interactive commands at the command prompt (>>) in the command window and they will be executed on the spot. 4

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

Getting help: by selecting the? Help icon bar from the desktop tool bar, or by typing helpdesk or helpwin in the command window. The other way of obtaining help is by using lookfor command. Help command searches for an exact function name match, whereas the lookfor command searches the quick summary information in each function for a match. Lookfor is slower than help but it improves the chances of getting useful information. For new users of MATLAB, a few demonstrations may help to give you a feel for its capabilities. To run MATLABs built in demonstrations, type demo in the command window, or select demos from the launch pad. The contents of the command window can be cleared at any time using the clc command, and the contents of the current figure window can be cleared at any time using the clf command. The variables in the workspace can be cleared with the clear command. It is a good idea to issue the clear command at the start of each new independent calculation. Another important command is the abort command. If an m-file appears to be running for too long, it might contain an infinite loop, and it will never terminate. In this case the user can regain the control by typing control-c (^c) in the command window. When MATLAB detects a ^c, it interrupts the running program and returns a command prompt. 5

H.S.Prashantha, Dept of TE, PESIT After diary command is typed, a copy of all input and most output typed in the command window is echoed in the diary file. The command diary off suspends input in to the diary file, and the command diary on resumes input again. MATLAB includes a special command (which) to help you to find out just which version of a file is being executed. and where it is located. The format of this command is which functionname, where functionname is the name of the function that you are trying to locate. The path related function include: * addpath Add directory to MATLAB search path * path * path2rc * rmpath Display MATLAB search path Add current directory to MATLAB search path Remove directory from MATLAB search path

The functional unit of data in any MATLAB program is the array. An array is a collection of data values organized into rows and columns, and known by a single name. MATLAB variable is a region of memory containing an array, which is known by a userspecified name. MATLAB variable names must begin with a letter, followed by any combination of letters, numbers, and the underscore( _ ) character. Only the first 31 characters are significant; if more than 31 are used, the remaining characters will be ignored. If two variables are declared with names that only differ in the 32 nd character, MATLAB will treat them as same variable. Spaces cannot be used in MATLAB variable names, underscore letters can be substituted to create meaningful names. It is important to include a data dictionary in the header of any program that you write. A data dictionary lists the definition of each variable used in a program. The definition should include both a description of the contents of the item and the units in which it is measured. MATLAB language is case-sensitive. It is customary to use lower-case letters for ordinary variable names. The most common types of MATLAB variables are double and char. MATLAB is weakly typed language. Variables are not declared in a program before it is used. MATLAB variables are created automatically when they are initialized. There are three common ways to initialize variables in MATLAB: Assign data to the variable in an assignment system. Input data into the variable from the keyboard. Read data from a file. 6

H.S.Prashantha, Dept of TE, PESIT The semicolon at the end of each assignment statement suppresses the automatic echoing of values that normally occurs whenever an expression is evaluated in an assignment statement.

How to invoke MATLAB? *) Double Click on the MATLAB icon on the desktop . *) You will find a Command window where in which you can type the commands and see the output. For example if you type PWD in the command window, it will print current working directory. *)If you want to create a directory type mkdir mydir in the command window, It will create a directory called pes. *) If you want delete a directory type rmdir mydir in the command window. How to open a file in MATLAB? *) Go to File New M-File and click Then type the program in the file and save the file with an extension of .m. While giving file name we should make sure that given file name should not be a command. It is better to the file name as myconvlution . How to run a MATLAB file? *) Go to Debugrun and click 1. Program to Verify Sampling Theorem % program for verification of sampling theorem clc; clear all; t=0:.001:0.1; f1=input ('Enter the input freequency1 = '); f2=input ('Enter the input freequency2 = '); y=cos(2*pi*f1*t)+cos(2*pi*f2*t); f3=max(f1,f2); % under sampling fs=f3; ts=1/fs; tx=0:ts:0.1; m=max(size(tx)); ys=cos(2*f1*pi*tx)+cos(2*pi*f2*tx); figure(1); subplot(3,1,1); plot(t,y); grid on; %fs = sampling freequency % clears the command window % clears the variables declared

H.S.Prashantha, Dept of TE, PESIT title('The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t)'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); subplot(3,1,2); stem(tx,ys); grid on; title('The sinusoidal signal sampled at fs Hz'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); subplot(3,1,3); plot(tx,ys); grid on; title('The recovered sinusidal sampled at fs Hz'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); % Right sampling fs=2*f3; %fs = sampling freequency ts=1/fs; tx=0:ts:0.1; m=max(size(tx)); ys=cos(2*pi*f1*tx)+cos(2*pi*f2*tx); figure(2); subplot(3,1,1); plot(t,y); grid on; title('The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t)'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); subplot(3,1,2); stem(tx,ys); grid on; title('The sinusoidal signal sampled at fs Hz'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); subplot(3,1,3); plot(tx,ys); grid on; title('The recovered sinusidal sampled at fs Hz'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); % over sampling fs=3*f3; ts=1/fs; tx=0:ts:0.1; %fs = sampling freequency

H.S.Prashantha, Dept of TE, PESIT m=max(size(tx)); ys=cos(2*pi*f1*tx)+cos(2*pi*f2*tx); figure(3); subplot(3,1,1); plot(t,y); grid on; title('The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t)'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); subplot(3,1,2); stem(tx,ys); grid on; title('The sinusoidal signal sampled at fs Hz'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); subplot(3,1,3); plot(tx,ys); grid on; title('The recovered sinusidal sampled at fs Hz'); xlabel('Time in seconds'); ylabel('Amplitude in volts'); OUTPUT Under sampling
Amplitude in volts The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t) 2 0 -2

0.01

0.02

Amplitude in volts

0.04 0.05 0.06 0.07 0.08 Time in seconds The sinusoidal signal sampled at fs Hz

0.03

0.09

0.1

2 1 0 0 0.01 0.02 0.04 0.05 0.06 0.07 0.08 Time in seconds The recovered sinusidal sampled at fs Hz 0.03 0.09 0.1

Amplitude in volts

2 1 0

0.01

0.02

0.03

0.04 0.05 0.06 Time in seconds

0.07

0.08

0.09

0.1

Right sampling 9

H.S.Prashantha, Dept of TE, PESIT

The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t) 2 0 -2

Amplitude in volts

0.01

0.02

Amplitude in volts

0.04 0.05 0.06 0.07 0.08 Time in seconds The sinusoidal signal sampled at fs Hz

0.03

0.09

0.1

2 0 -2 0 0.01 0.02 0.04 0.05 0.06 0.07 0.08 Time in seconds The recovered sinusidal sampled at fs Hz 0.03 0.09 0.1

Amplitude in volts

2 0 -2

0.01

0.02

0.03

0.04 0.05 0.06 Time in seconds

0.07

0.08

0.09

0.1

Over sampling
Amplitude in volts The sinusoidal signal cos(2*pi*f1*t)+cos(2*pi*f2*t) 2 0 -2

0.01

0.02

Amplitude in volts

0.04 0.05 0.06 0.07 0.08 Time in seconds The sinusoidal signal sampled at fs Hz

0.03

0.09

0.1

2 0 -2 0 0.01 0.02 0.04 0.05 0.06 0.07 0.08 Time in seconds The recovered sinusidal sampled at fs Hz 0.03 0.09 0.1

Amplitude in volts

2 0 -2

0.01

0.02

0.03

0.04 0.05 0.06 Time in seconds

0.07

0.08

0.09

0.1

10

H.S.Prashantha, Dept of TE, PESIT 2. Program to find the Impulse response of the system %To find the Impulse response of a given System %a0*y(n)-a1*y(n-1)+a2*y(n-2)=b0*x(n)+b1*x(n-1)+b2*x(n-2) clc; clear all; b=input('Enter the coefficiants of x(n) in the order x(n),x(n-1)...in the Matrix form = '); a=input('Enter the coefficiants of y(n) in the order y(n),y(n-1)...in the Matrix form = '); n=[-5:20]; x=[(n==0)]; h=filter(b,a,x) figure(1); subplot(2,1,1); stem(n,x); xlabel('n'); ylabel('amplitude'); title('impulse sequence'); subplot(2,1,2); stem(n,h); xlabel('n'); ylabel('amplitude'); title('impulse response'); OUTPUT Enter the coefficiants of x(n) in the order x(n),x(n-1)...in the Matrix form = [0.2066 0.4131 0.2066] Enter the coefficiants of y(n) in the order y(n),y(n-1)...in the Matrix form = [1 -0.3695 0.1968]
impulse sequence 1

amplitude

0.5

0 -5

10 n impulse response

15

20

0.6 0.4 amplitude 0.2 0 -0.2 -5

5 n

10

15

20

11

H.S.Prashantha, Dept of TE, PESIT Amplitude= [0 0 0 0 0 0.2066 0.4894 0.3468 0.0318 -0.0565 -0.0271 0.0011 0.0057 0.0019 -0.0004 -0.0005 -0.0001 0.0001 0.0000 0.0000 -0.0000 -0.0000 0.0000 0.0000 0.0000 -0.0000] 3. Program to perform linear Convolution of two given sequences %This program gives the convolution of two sequences clc; clear all; close all; x1=input('enter the first sequence'); %input of first sequence x2=input('enter the second sequence'); %input of second sequence n1=length(x1); n2=length(x2); N=n1+n2-1; x1=[x1 zeros(1,N-n1)]; x2=[x2 zeros(1,N-n2)]; m=[0:1:N-1]; for n=0:N-1 y(n+1)=sum(x2(mod(n-m,N)+1).*x1); end; disp('linear convoluted sequence'); y disp('first sequence'); x1 disp('second sequence'); x2 disp('convolved sequence'); y subplot(3,1,1); stem(x1); title('first sequence'); xlabel('signal'); ylabel('time'); subplot(3,1,2); stem(x2); title('second sequence'); xlabel('signal'); ylabel('time'); subplot(3,1,3); stem(y); 12

H.S.Prashantha, Dept of TE, PESIT title('convolved sequence'); xlabel('signal'); ylabel('time'); Input: x1 = 1 2 3 4 0 0 0 second sequence x2 = 1 2 3 4 0 0 0 convolved sequence y= 1 4 10 20 25 24 16
first sequence 4 signal 2 0

4 time second sequence

4 signal 2 0 1 2 3 4 time convolved sequence 5 6 7

40 signal 20 0

4 time

4. Program to perform Circular Convolution of two given sequences %This program calculates the circular convolution clc; clear all; x=input('Input first sequence'); h=input('Input second sequence'); N1=length(x); N2=length(h); N=max(N1,N2); 13

H.S.Prashantha, Dept of TE, PESIT x=[x zeros(1,N-N1)]; h=[h zeros(1,N-N2)]; for m=1:N y(m)=0; for n=1:N i=m-n+1; if(i<=0) i=N+i; end y(m)=y(m)+x(n)*h(i); end end disp('first sequence'); x disp('second sequence'); h disp('convolved sequence'); y n=0:N-1; subplot(3,1,1); stem(x); title('first sequence'); ylabel('signal'); xlabel('time'); subplot(3,1,2); stem(h); title('second sequence'); ylabel('signal'); xlabel('time'); subplot(3,1,3); stem(y); title('convolved sequence'); ylabel('signal'); xlabel('time'); OUTPUT Input first sequence Input second sequence first sequence 14 [1 2 3 4] [1 2 3 4]

H.S.Prashantha, Dept of TE, PESIT x= 1 2 3 4

second sequence h= 1 2 3 4

convolved sequence y= 26 28
4 signal 2 0

26

20
first sequence

1.5

2.5 time second sequence

3.5

4 signal 2 0 1 1.5 2 2.5 time convolved sequence 3 3.5 4

40 signal 20 0

1.5

2.5 time

3.5

5. Autocorrelation of a given sequence x=input('enter the sequence'); y=xcorr(x,x); figure; subplot(2,1,1); stem(x); ylabel('amplitude'); 15

H.S.Prashantha, Dept of TE, PESIT xlabel('n'); subplot(2,1,2); stem(fliplr(y)); y ylabel('amplitude'); xlabel('n'); disp('the resultant signal is '); fliplr(y); Input: Enter the sequence[1 2 3 4] The resultant signal is y= [4 11 20 30 20 11 4]

4 3 amplitude 2 1 0

1.5

2.5 n

3.5

30

amplitude

20

10

4 n

6. Cross correlation of a given sequence clc; close all; clear all; x=input('enter the sequence'); h=input('enter the second sequence'); y=xcorr(x,h); figure; subplot(3,1,1); 16

H.S.Prashantha, Dept of TE, PESIT stem(x); ylabel('amplitude'); xlabel('n'); subplot(3,1,2); stem(h); ylabel('amplitude'); xlabel('n'); subplot(3,1,3); stem(fliplr(y)); y ylabel('amplitude'); xlabel('n'); disp('the resultant signal is '); fliplr(y); Input: enter the sequence[1 2 3 4] enter the second sequence[4 3 2 1] y= 1.0000 4.0000 10.0000 20.0000 25.0000 24.0000 16.0000

The resultant signal is

4 amplitude 2 0

1.5

2.5 n

3.5

4 amplitude 2 0

1.5

2.5 n

3.5

30 amplitude 20 10 0 1 2 3 4 n 5 6 7

17

H.S.Prashantha, Dept of TE, PESIT 7. Program to Solve difference equation %program for solution of difference equation % y(n) -0.5 y(n-1) = x(n) with x(n)=u(n)-u(n-4) b=input('enter the coefficients of x'); a=input('enter the coefficients of y'); n=[-5:50]; x=[(n>=0)]-[(n>4)]; figure(1); subplot(2,1,1); stem(n,x); title('input sequence x(n)'); xlabel('n'); ylabel('x'); subplot(2,1,2); y=filter(b,a,x); stem(n,y); title('output sequence y(n)'); xlabel('n'); ylabel('y');
input sequence x(n) 1

0.5

0 -10

10

20 30 n output sequence y(n)

40

50

2 1.5 y 1 0.5 0 -10

10

20 n

30

40

50

% y(n) -0.25 y(n-1) -0.125 y(n-2)= x(n) with x(n)=u(n)-u(n-10) and initial condition y(-1)=1, y(-2)=-2 18

H.S.Prashantha, Dept of TE, PESIT b=input('enter the coefficients of x'); a=input('enter the coefficients of y'); c=input('enter the initial conditions'); n=[-5:50]; ic=filtic(b,a,c); x=[(n>=0)]-[(n>10)]; y=filter(b,a,x,ic); subplot(2,1,1); stem(n,x); title('input sequence x(n)'); xlabel('n'); ylabel('x'); subplot(2,1,2); stem(n,y); title('output sequence y(n)'); xlabel('n'); ylabel('y'); Output:
input sequence x(n) 1

0.5

0 -10

10

20 30 n output sequence y(n)

40

50

2 1.5 y 1 0.5 0 -10

10

20 n

30

40

50

19

H.S.Prashantha, Dept of TE, PESIT 8. Program to Calculate N point DFT and N point IDFT of the given sequence
%to compute dft of the sequence and plot magnitude and phase spectra clc; clear all; close all; % input sequence xn=input('enter the input sequence'); N=input('enter the number of points'); xk=dft(xn,N); %magnitude of fft magxk=abs(xk); %phase of fft phasexk=angle(xk); k=0:N-1; subplot(2,1,1); stem(k,magxk); title('fft sequence'); xlabel('frequency'); ylabel('magnitude'); subplot(2,1,2); stem(k,phasexk); title('phase of fft sequence'); xlabel('frequency'); ylabel('phase'); function xk=dft(xn,N) L=length(xn); if(N<L) error('N must be >=L'); end x1=[xn zeros(1,N-L)] for k=0:1:N-1; for n=0:1:N-1; p=exp(-i*2*pi*n*k/N) x2(k+1,n+1)=p; end end xk=x1*x2; OUTPUT: X=[1 1 1 1 1 1 1 1] N=32

20

H.S.Prashantha, Dept of TE, PESIT

fft sequence 8 6 magnitude 4 2 0

10

15 20 frequency phase of fft sequence

25

30

35

4 2 phase 0 -2 -4

10

15 20 frequency

25

30

35

%to compute idft of the sequence clc; clear all; close all; % input sequence Xk=input('enter the input sequence'); N=length(Xk) xn=idft(Xk,N); k=0:N-1; stem(k,xn); title('ifft sequence'); xlabel('time'); ylabel('amplitude'); disp(xn) function xn=idft(Xk,N) for k=0:1:N-1; for n=0:1:N-1; p=exp(i*2*pi*n*k/N) x2(k+1,n+1)=p; end end

21

H.S.Prashantha, Dept of TE, PESIT

xn=(Xk*x2)./N;

9. Linear convolution of two sequences using DFT and IDFT clc; clear all; x=input('Input first sequence'); h=input('Input second sequence'); n1=length(x); n2=length(h); N=n1+n2-1; x1=[x zeros(1,N-n1)]; x2=[h zeros(1,N-n2)]; a=fft(x,N); b=fft(h,N); c=a.*b; d=ifft(c,N); disp('first sequence'); x disp('second sequence'); h disp('convolved sequence'); d n=0:N-1; subplot(3,1,1); stem(x); title('first sequence'); ylabel('signal'); xlabel('time'); subplot(3,1,2); stem(h); title('second sequence'); ylabel('signal'); xlabel('time'); subplot(3,1,3); stem(d); title('convolved sequence'); ylabel('signal'); xlabel('time'); Input: x= 1 1 1 1 second sequence h= 1 1 1 1 convolved sequence 22

H.S.Prashantha, Dept of TE, PESIT d= 1.0000

1 signal 0.5 0

2.0000

3.0000

4.0000

3.0000

2.0000

1.0000

first sequence

1.5

2.5 time second sequence

3.5

1 signal 0.5 0 1 1.5 2 2.5 time convolved sequence 3 3.5 4

4 signal 2 0

4 time

10. Circular convolution of two sequences using DFT and IDFT clc; clear all; x=input('Input first sequence'); h=input('Input second sequence'); n1=length(x); n2=length(h); N=max(n1,n2); if N>n1 x1=[x zeros(1,N-n1)]; end if N>n2 x2=[h zeros(1,N-n2)]; end a=fft(x,N); b=fft(h,N); c=a.*b; d=ifft(c,N); disp('first sequence'); x disp('second sequence'); h disp('convolved sequence'); d n=0:N-1; 23

H.S.Prashantha, Dept of TE, PESIT subplot(3,1,1); stem(x); title('first sequence'); ylabel('signal'); xlabel('time'); subplot(3,1,2); stem(h); title('second sequence'); ylabel('signal'); xlabel('time'); subplot(3,1,3); stem(d); title('convolved sequence'); ylabel('signal'); xlabel('time'); Result: Input first sequence[1 2 3] Input second sequence[1 2 3 4] first sequence x= 1 2 3 second sequence h= 1 2 3 4 convolved sequence d = 18 16 10 16

24

H.S.Prashantha, Dept of TE, PESIT

first sequence 4 signal 2 0

1.2

1.4

1.6

1.8

2 2.2 time second sequence

2.4

2.6

2.8

4 signal 2 0 1 1.5 2 2.5 time convolved sequence 3 3.5 4

20 signal 10 0

1.5

2.5 time

3.5

11. Design and implementation of FIR filters to meet given specifications clear all; close all; clc; %cutoff frequency of low pass response at 0.25 pi and a minimum stopband %attenuation of 39 dB. transition band is 0.2 pi wc=0.25*pi; tw=0.2*pi; %normalize the cutoff frequency and transition width fcn=wc/pi; twn=tw/pi; %the required transition width is 39 dB. Choose Hanning window %order of the filter disp('order of the filter'); N=ceil(4/twn) if(rem(N,2)==0); N=N+1; end; 25

H.S.Prashantha, Dept of TE, PESIT y=hanning(N) disp('fir filter coefficients'); b=fir1(N-1,fcn,y) disp('frequency response'); [H,f]=freqz(b,1,256) H_mag=20*log10(abs(H)); plot(f/pi,H_mag); xlabel('frequency in rad/sec'); ylabel('gain in dB'); output:
20 0 -20 -40 gain in dB -60 -80 -100 -120 -140

0.1

0.2

0.3

0.4 0.5 0.6 frequency in rad/sec

0.7

0.8

0.9

%passband edge frequency of 2 kHz, a minimum stopband attenuation of 50 dB %transition band is 0.5 kHzat a sampling frequency of 8 kHz. fs=8000; %normalizing frequency fn=fs/2; %passband and stopband frequencies fp=2000; tw=500; fs=fp+tw; %normalize the cutoff frequency and transition width fcn=(fp+fs)/(2*fn); twn=tw/fs; %the required transition width is 50 dB. Choose Hamming window %order of the filter disp('order of the filter'); N=ceil(4/twn) if(rem(N,2)==0); N=N+1; end; y=hamming(N) 26

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

-20

gain in dB

-40

-60

-80

-100

-120

0.1

0.2

0.3

0.4 0.5 0.6 frequency in rad/sec

0.7

0.8

0.9

12. Programs to Design IIR filters % Program to Design the Digital IIR Butterworth Low pass filter using bilinear transformation clc; clear all; pb=input('Enter the pass band edge frequency '); sb=input('Enter the stop band edge frequency '); pbr=input('Enter the pass band attenuation '); sbr=input('Enter the stop band attenuation '); fs= input('Enter the sampling frequency '); pbrad=pb*2*pi; sbrad=sb*2*pi; [n,wn]=buttord(pbrad,sbrad,pbr,sbr,'s'); 27

H.S.Prashantha, Dept of TE, PESIT [b,a]=butter(n,wn,'s'); [z,p,k]=tf2zp(b,a); [zd,pd,kd]=bilinear(z,p,k,fs); [num,den]=zp2tf(zd,pd,kd); figure(1); freqz(num,den,512,fs); title('Butter worth freequency response'); [n,wn]=cheb1ord(pbrad,sbrad,pbr,sbr,'s'); [b,a]=cheby1(n,pbr,wn,'s'); [z,p,k]=tf2zp(b,a); [zd,pd,kd]=bilinear(z,p,k,fs); [num,den]=zp2tf(zd,pd,kd); figure(2); freqz(num,den,512,fs); title('Chebychev1 Freequency Response'); [n,wn]=cheb2ord(pbrad,sbrad,pbr,sbr,'s'); [b,a]=cheby2(n,sbr,wn,'s'); [z,p,k]=tf2zp(b,a); [zd,pd,kd]=bilinear(z,p,k,fs); [num,den]=zp2tf(zd,pd,kd); figure(3); freqz(num,den,512,fs); title('Chebychev 2 Freequency Response'); OUTPUT Enter the pass band edge frequency 1500 Enter the stop band edge frequency 2000 Enter the pass band attenuation 0.5 Enter the stop band attenuation 30 Enter the sampling frequency 8000

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H.S.Prashantha, Dept of TE, PESIT

Magnitude (dB) Butter w orth freequency response 500 0 -500 0 500 1000 1500 2000 2500 Frequency (H z) 3000 3500 4000

Phase (degrees)

0 -1000 -2000 0 500 1000 1500 2000 2500 Frequency (H z) 3000 3500 4000

Magnitude (dB)

C e ych v F e u n R sp n h b e 1 re q e cy e o se 50 0 0 -5 0 0

50 0

10 00

Phase (degrees)

10 50 20 00 20 50 F q e cy (H re u n z)

30 00

30 50

40 00

0 -5 0 0 -1 0 00

50 0

10 00

10 50 20 00 20 50 F q e cy (H re u n z)

30 00

30 50

40 00

Chebychev 2 Freequency Response Magnitude (dB) 0 -50 -100

500

1000

1500 2000 2500 Frequency (Hz)

3000

3500

4000

Phase (degrees)

500 0 -500

500

1000

1500 2000 2500 Frequency (Hz)

3000

3500

4000

% Program to Disign the Digital IIR Butterworth Low pass filter using impulse invariance method clc; clear all; pb=input('Enter the pass band edge frequency '); sb=input('Enter the stop band edge frequency '); pbr=input('Enter the pass band attenuation in dB '); 29

H.S.Prashantha, Dept of TE, PESIT sbr=input('Enter the stop band attenuation in dB '); fs= input('Enetr the sampling frequency '); pbrad=pb*2*pi; sbrad=sb*2*pi; [n,wn]=buttord(pbrad,sbrad,pbr,sbr,'s'); [b,a]=butter(n,wn,'s'); [num,den]=impinvar(b,a,fs); figure(1); freqz(num,den,512,fs); title('Butter worth frequency response'); [n,wn]=cheb1ord(pbrad,sbrad,pbr,sbr,'s'); [b,a]=cheby1(n,pbr,wn,'s'); [num,den]=impinvar(b,a,fs); figure(2); freqz(num,den,512,fs); title('Chebychev1 Frequency Response'); [n,wn]=cheb2ord(pbrad,sbrad,pbr,sbr,'s'); [b,a]=cheby2(n,sbr,wn,'s'); [num,den]=impinvar(b,a,fs); figure(3); freqz(num,den,512,fs); title('Chebychev 2 Frequency Response');

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DSP KIT TMS306713: Troubleshooting DSK Connectivity If Code Composer Studio IDE fails to configure your port correctly, perform the following steps: Test the USB port by running DSK Port test from the start menu

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

If the board still fails to detect 31

H.S.Prashantha, Dept of TE, PESIT Go to CMOS setup Enable the USB Port Option (The required Device drivers will load along with CCS Installation) INTRODUCTION TO CODE COMPOSER STUDIO Code Composer is the DSP industry's first fully integrated development environment (IDE) with DSP-specific functionality. With a familiar environment liked MS-based C++TM, Code Composer lets you edit, build, debug, profile and manage projects from a single unified environment. Other unique features include graphical signal analysis, injection/extraction of data signals via file I/O, multi-processor debugging, automated testing and customization via a Cinterpretive scripting language and much more. CODE COMPOSER FEATURES INCLUDE: IDE Debug IDE Advanced watch windows Integrated editor File I/O, Probe Points, and graphical algorithm scope probes Advanced graphical signal analysis Interactive profiling Automated testing and customization via scripting Visual project management system Compile in the background while editing and debugging Multi-processor debugging Help on the target DSP

Procedure to work on Code Composer Studio

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

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H.S.Prashantha, Dept of TE, PESIT 2. To Create a Source file File New

type the code (Save & give a name to file, Eg: sum.c). 3. To Add Source files to Project Project Add files to Project sum.c

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H.S.Prashantha, Dept of TE, PESIT

4. To Add rts6700.lib file & hello.cmd: Project Add files to Project rts6700.lib Path: c:\CCStudio\c6000\cgtools\lib\rts6700.lib 34

H.S.Prashantha, Dept of TE, PESIT Note: Select Object & Library in(*.o,*.l) in Type of files Project Add files to Project hello.cmd Path: c:\ti\tutorial\dsk6713\hello1\hello.cmd Note: Select Linker Command file(*.cmd) in Type of files

5. To Compile: Project Compile File 6. To build or Link: Project build, Which will create the final executable (.out) file.(Eg. sum.out). 7. Procedure to Load and Run program: Load program to DSK: File Load program sum. out 8. To execute project: Debug Run. TMS320C6713 is a floating point DSP processor.Basically it is a cisc processor.It has following features *) It can execute eight 32-bit instructions/cycle *) Advanced Very Long Instruction Word DSP core. *) It has eight independent functional units.In which there are 2 ALUs of fixed point, 4 ALUs of Floating and fixed point and 2 multipliers of floating and fixed point. *) Two multi channel audio ports *) 32 bit External memory Interface. DSP Starter kit consists of TMS320C6713 ,CPLD,Flash Memory,SDRAM,Codec. Code Composer Studio 35

H.S.Prashantha, Dept of TE, PESIT

Code Composer is the Integrated Development Environment(IDE) with DSP specific functionality.Basically code composer studio is a cross compiler.That means the programs written in Higher level languages like Care converted into processor specific languages. Procedure to work with code composer studio Open the code composer studio,make sure DSP Starter Kit(DSK) is turned on. Create a new project as follows Go to Project->new and click on it , give project_name ,Select target as TMS320C67xx. Then save it in C:\CCStudio\MyProjects Then go Project -> Add Files to Project and click on it , then add the run time support library C:\CCStudio\c6000\cgtools\lib\rts6700.lib Then go Project-> Add Files to Project and click on it , then add the linker command file C:\CCStudio\tutorial\dsk6713\hello1\hello.cmd Then compile the program using Project->compile->click Then go to Project -> Rebuild All. Then go to File->Load program -> project_name.out into the DSP Starter kit. How to view graphical output To view the graphical output go to View Graph Time/Frequency 1 C PROGRAM TO IMPLEMENT LINEAR CONVOLUTION

/* prg to implement linear convolution */ #include<stdio.h> #define LENGHT1 6 #define LENGHT2 4 /*Lenght of i/p samples sequence*/ /*Lenght of impulse response Co-efficients */ /*Input Signal Samples*/ /*Impulse Response Co-efficients*/

int x[2*LENGHT1-1]={1,2,3,4,5,6,0,0,0,0,0}; int h[2*LENGHT1-1]={1,2,3,4,0,0,0,0,0,0,0}; int y[LENGHT1+LENGHT2-1]; main() { int i=0,j; for(i=0;i<(LENGHT1+LENGHT2-1);i++) { y[i]=0; for(j=0;j<=i;j++) y[i]+=x[j]*h[i-j]; } for(i=0;i<(LENGHT1+LENGHT2-1);i++) printf("%d\n",y[i]); }

36

H.S.Prashantha, Dept of TE, PESIT PROCEDURE: Open Code Composer Studio, make sure the DSP kit is turned on. Start a new project using Project-new pull down menu, save it in a separate directory(c:\ti\myprojects) with name lconv.pjt. Add the source files conv.c to the project using Projectadd files to project pull down menu. Add the linker command file hello.cmd . (Path: c:\ti\tutorial\dsk6713\hello1\hello.cmd) Add the run time support library file rts6700.lib (Path: c:\ti\c6000\cgtools\lib\rts6700.lib) Compile the program using the Project-compile pull down menu or by clicking the shortcut icon on the left side of program window. Build the program using the Project-Build pull down menu or by clicking the shortcut icon on the left side of program window. Load the program(lconv.out) in program memory of DSP chip using the File-load program pull down menu. To View output graphically Select view graph time and frequency. Configure the graphical window as shown below

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H.S.Prashantha, Dept of TE, PESIT

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H.S.Prashantha, Dept of TE, PESIT

program to Implement Circular Convolution

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

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H.S.Prashantha, Dept of TE, PESIT

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

PROCEDURE: Open Code Composer Studio, make sure the DSP kit is turned on. Start a new project using Project-new pull down menu, save it in a separate directory(c:\ti\myprojects) with name cir conv.pjt. Add the source files Circular Convolution.C to the project using Projectadd files to project pull down menu. Add the linker command file hello.cmd . (Path: c:\ti\tutorial\dsk6713\hello1\hello.cmd) Add the run time support library file rts6700.lib (Path: c:\ti\c6000\cgtools\lib\rts6700.lib) Compile the program using the Project-compile pull down menu or by clicking the shortcut icon on the left side of program window. Build the program using the Project-Build pull down menu or by clicking the shortcut icon on the left side of program window. Load the program(lconv.out) in program memory of DSP chip using the File-load program pull down menu. DFT USING DSP KIT #include <stdio.h> #include <math.h> float x[4]={1,3,2,5}; //input only real sequence main() { float y[16]; //for 8 point DFT to store real & imaginary float w; int n,k,k1,N=8,xlen=4; for(k=0;k<2*N;k=k+2) { y[k]=0; y[k+1]=0; //initialize real & imag parts k1=k/2; //actual k index 40

H.S.Prashantha, Dept of TE, PESIT for(n=0;n<xlen;n++) { w=-2*3.14*k1*n/N; //careful about minus sign y[k]=y[k]+x[n]*cos(w); y[k+1]=y[k+1]+x[n]*sin(w); } printf("%f+j%f \n",y[k],y[k+1]); } }//end of main Procedure for Real time Programs : 1. Connect CRO to the Socket Provided for LINE OUT. 2. Connect a Signal Generator to the LINE IN Socket. 3. Switch on the Signal Generator with a sine wave of frequency 500 Hz. and Vp-p=1.5v 4. Now Switch on the DSK and Bring Up Code Composer Studio on the PC. 5. Create a new project with name codec.pjt. 6. From the File Menu new DSP/BIOS Configuration select dsk6713.cdb and save it as xyz.cdb

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H.S.Prashantha, Dept of TE, PESIT

7. Add xyz.cdb to the current project. 8. Add the given codec.c file to the current project which has the main function and calls all the other necessary routines. 9. Add the library file dsk6713bsl.lib to the current project Path C:\CCStudio\C6000\dsk6713\lib\dsk6713bsl.lib 10. Copy dsk6713.h and dsk6713_aic23.h C:\CCStudio\C6000\dsk6713\include and paste it in current project. files from

11. Build, Load and Run the program. 12. You can notice the input signal of 500 Hz. appearing on the CRO verifying the codec configuration. 13. You can also pass an audio input and hear the output signal through the speakers. 14. You can also vary the sampling frequency using the DSK6713_AIC23_setFreq Function in the codec.c file and repeat the above steps. 42

H.S.Prashantha, Dept of TE, PESIT

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

codec.c #include "xyzcfg.h" #include "dsk6713.h" #include "dsk6713_aic23.h" /* Codec configuration settings */ DSK6713_AIC23_Config config = { \ 0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Left line input channel volume */ \ 0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume */\ 0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */ \ 0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */ \ 0x0011, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */ \ 0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */ \ 0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */ \ 0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */ \ 0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */ \ 0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \ }; /* main() - Main code routine, initializes BSL and generates tone */ void main() { DSK6713_AIC23_CodecHandle hCodec; int l_input, r_input,l_output, r_output; /* Initialize the board support library, must be called first */ DSK6713_init(); /* Start the codec */ hCodec = DSK6713_AIC23_openCodec(0, &config); /*set codec sampling frequency*/ DSK6713_AIC23_setFreq(hCodec, 3); while(1) { /* Read a sample to the left channel */ while (!DSK6713_AIC23_read(hCodec, &l_input)); /* Read a sample to the right channel */

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H.S.Prashantha, Dept of TE, PESIT

while (!DSK6713_AIC23_read(hCodec, &r_input)); /* Send a sample to the left channel */ while (!DSK6713_AIC23_write(hCodec, l_input)); /* Send a sample to the right channel */ while (!DSK6713_AIC23_write(hCodec, l_input)); } /* Close the codec */ DSK6713_AIC23_closeCodec(hCodec); }

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

fir.c

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H.S.Prashantha, Dept of TE, PESIT

#include "filtercfg.h" #include "dsk6713.h" #include "dsk6713_aic23.h" float filter_Coeff[] ={0.000000,-0.001591,-0.002423,0.000000,0.005728, 0.011139,0.010502,-0.000000,-0.018003,-0.033416,-0.031505,0.000000, 0.063010,0.144802,0.220534,0.262448,0.220534,0.144802,0.063010,0.000000, -0.031505,-0.033416,-0.018003,-0.000000,0.010502,0.011139,0.005728, 0.000000,-0.002423,-0.001591,0.000000 }; static short in_buffer[100]; DSK6713_AIC23_Config config = {\ 0x0017, /* 0 DSK6713_AIC23_LEFTINVOL Leftline input channel volume */\ 0x0017, /* 1 DSK6713_AIC23_RIGHTINVOL Right line input channel volume*/\ 0x00d8, /* 2 DSK6713_AIC23_LEFTHPVOL Left channel headphone volume */\ 0x00d8, /* 3 DSK6713_AIC23_RIGHTHPVOL Right channel headphone volume */\ 0x0011, /* 4 DSK6713_AIC23_ANAPATH Analog audio path control */\ 0x0000, /* 5 DSK6713_AIC23_DIGPATH Digital audio path control */\ 0x0000, /* 6 DSK6713_AIC23_POWERDOWN Power down control */\ 0x0043, /* 7 DSK6713_AIC23_DIGIF Digital audio interface format */\ 0x0081, /* 8 DSK6713_AIC23_SAMPLERATE Sample rate control */\ 0x0001 /* 9 DSK6713_AIC23_DIGACT Digital interface activation */ \ }; /* * main() - Main code routine, initializes BSL and generates tone */ void main() { DSK6713_AIC23_CodecHandle hCodec; Uint32 l_input, r_input,l_output, r_output; /* Initialize the board support library, must be called first */ DSK6713_init(); /* Start the codec */ hCodec = DSK6713_AIC23_openCodec(0, &config); DSK6713_AIC23_setFreq(hCodec, 1); while(1) { /* Read a sample to the left channel */ while (!DSK6713_AIC23_read(hCodec, &l_input)); /* Read a sample to the right channel */ while (!DSK6713_AIC23_read(hCodec, &r_input)); l_output=(Int16)FIR_FILTER(&filter_Coeff ,l_input); r_output=l_output; /* Send a sample to the left channel */ while (!DSK6713_AIC23_write(hCodec, l_output));

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/* Send a sample to the right channel */ while (!DSK6713_AIC23_write(hCodec, r_output)); } /* Close the codec */ DSK6713_AIC23_closeCodec(hCodec); } signed int FIR_FILTER(float * h, signed int x) { int i=0; signed long output=0; in_buffer[0] = x; /* new input at buffer[0] */ */

for(i=30;i>0;i--) in_buffer[i] = in_buffer[i-1]; /* shuffle the buffer for(i=0;i<32;i++) output = output + h[i] * in_buffer[i]; return(output); }

PROCEDURE : Switch on the DSP board. Open the Code Composer Studio. Create a new project Project New (File Name. pjt , Eg: FIR.pjt) Initialize on board codec. Note: Kindly refer the Topic Configuration of 6713 Codec using BSL

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

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