SVS Institute of Technology

JAWAHARLAL NEHRU TECHNOLOGICAL UNIVERSITY

HYDERABAD

III Year B.Tech. ECE - II Sem L T/P/D C

0 -/3/- 2

DIGITAL SIGNAL PROCESSING LAB (EC604PC)

Note:
1. The Programs shall be implemented in Software (Using MATLAB / Lab View / C
Programming/ Equivalent) and Hardware (Using TI / Analog Devices / Motorola /
Equivalent DSP processors).
2. Minimum of 12 experiments to be conducted.
List of Experiments:
1. Generation of Sinusoidal Waveform / Signal based on Recursive Difference Equations
2. Histogram of White Gaussian Noise and Uniformly Distributed Noise.
3. To find DFT / IDFT of given DT Signal
4. To find Frequency Response of a given System given in Transfer Function/Differential
equation form.
5. Obtain Fourier series coefficients by formula and using FET and compare for half sine wave.
6. Implementation of FFT of given Sequence
7. Determination of Power Spectrum of a given Signal(s).
8. Implementation of LP FIR Filter for a given Sequence/Signal.
9. Implementation of HP IIR Filter for a given Sequence/Signal
10. Generation of Narrow Band Signal through Filtering
11. Generation of DTMF Signals
12. Implementation of Decimation Process
13. Implementation of Interpolation Process
14. Implementation of I/D Sampling Rate Converters
15. Impulse Response of First order and Second Order Systems

Note: - Minimum of 12 experiments has to be conducted.

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Introduction to MAT Lab

The name MATLAB stands for matrix laboratory. It was invented in the late
1970s by Cleve Moler, then chairman of the computer science department at the University of
New Mexico. MATLAB has evolved over a period of years with input from many users. In
university environments, it is the standard instructional tool for introductory and advanced
courses in mathematics, engineering, and science. In industry, MATLAB is the tool of choice for
high-productivity research, development and

analysis.

MATLAB was first adopted by control design engineers, Little's specialty, but quickly spread to
many other domains. It is now also used in education, in particular the teaching of linear algebra
and numerical analysis, and is popular amongst scientists involved with image processing.

MATLAB is a high-performance language for technical computing. It integrates

computation, visualization, and programming in an easy-to-use environment where problems and
solutions are expressed in familiar mathematical notation. Its wide range of commands,
functions, and language constructs permit users to solve and analyze difficult computational
problems from science and engineering without programming in a general purpose language.
Typical uses include:

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1. Math and computation

2. Algorithm development
3. Modeling, simulation and prototyping
4. Data analysis, exploration and visualization
5. Scientific and engineering graphics
6. Application development, including graphical user interface building

REAL TIME APPLICATIONS

1. DATA SECURITY
2. IMAGE MANIPULATIONS
3. SIGNAL PROCESSING

MATLAB Windows:

MATLAB works with through three basic windows

Command Window : This is the main window .it is characterized by MATLAB command
prompt >> when you launch the application program MATLAB puts you in this window all
commands including those for user-written programs ,are typed in this window at the MATLAB
prompt

Graphics window: the output of all graphics commands typed in the command window are
flushed to the graphics or figure window, a separate gray window with white background color the
user can create as many windows as the system memory will allow

Edit window: This is where you write edit, create and save your own programs in files called M
files.

Input-output:

MATLAB supports interactive computation taking the input from the screen and flushing, the
output to the screen. In addition it can read input files and write output files

Data Type: the fundamental data -type in MATLAB is the array. It encompasses several distinct
data objects- integers, real numbers, matrices, character strings, structures and cells. There is no
need to declare variables as real or complex, MATLAB automatically sets the variable to be real.

Dimensioning: Dimensioning is automatic in MATLAB. No dimension statements are required for

vectors or arrays .we can find the dimensions of an existing matrix or a vector with the size and
length commands.

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Basic Instructions in Mat lab

1. T = 0: 1:10

This instruction indicates a vector T which as initial value 0 and final value 10 with an
increment of 1

Therefore T = [0 1 2 3 4 5 6 7 8 9 10]

2. F= 20: 1: 100

Therefore F = [20 21 22 23 24 ……… 100]

3. T= 0:1/pi: 1

Therefore T= [0, 0.3183, 0.6366, 0.9549]

4. zeros (1, 3)

The above instruction creates a vector of one row and three columns whose values are zero

Output= [0 0 0]

5. zeros( 2,4)

Output = 0 0 0 0

0000

6. ones (5,2)
The above instruction creates a vector of five rows and two columns

9. plot (t, x)

If x = [6 7 8 9] t = [1 2 3 4]

This instruction will display a figure window which indicates the plot of x versus t

10. stem (t,x) :- This instruction will display a figure window as shown

11. Conv

Syntax: w = conv(u,v)

Description: w = conv(u,v) convolves vectors u and v. Algebraically, convolution is the same

operation as multiplying the polynomials whose coefficients are the elements of u and v.

12.Disp

Syntax: disp(X)
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Description: disp(X) displays an array, without printing the array name. If X contains a text
string, the string is displayed.Another way to display an array on the screen is to type its name,
but this prints a leading "X=," which is not always desirable.Note that disp does not display
empty arrays.

13.xlabel

Syntax: xlabel('string')

Description: xlabel('string') labels the x-axis of the current axes.

14. ylabel

Syntax : ylabel('string')

Description: ylabel('string') labels the y-axis of the current axes.

15.Title

Syntax : title('string')

Description: title('string') outputs the string at the top and in the center of the current axes.

16.grid on

Syntax : grid on

Description: grid on adds major grid lines to the current axes.

17.FFT Discrete Fourier transform.

FFT(X) is the discrete Fourier transform (DFT) of vector X. For matrices, the FFT operation is
applied to each column. For N-D arrays, the FFT operation operates on the first non-singleton
dimension. FFT(X,N) is the N-point FFT, padded with zeros if X has less than N points and
truncated if it has more.

18. ABS Absolute value.

ABS(X) is the absolute value of the elements of X. When X is complex, ABS(X) is the complex
modulus (magnitude) of the elements of X.

19. ANGLE Phase angle.

ANGLE(H) returns the phase angles, in radians, of a matrix with complex elements.

20.INTERP Resample data at a higher rate using lowpass interpolation.

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Y = INTERP(X,L) resamples the sequence in vector X at L times the original sample rate. The
resulting resampled vector Y is L times longer, LENGTH(Y)= L*LENGTH(X).

21. DECIMATE Resample data at a lower rate after lowpass filtering.

Y = DECIMATE(X,M) resamples the sequence in vector X at 1/M times the original sample
rate. The resulting resampled vector Y is M times shorter, i.e., LENGTH(Y) =
CEIL(LENGTH(X)/M). By default, DECIMATE filters the data with an 8th order Chebyshev
Type I lowpass filter with cutoff frequency .8*(Fs/2)/R, before resampling.

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Experiment No.1
Generation of sinusoidal waveform / signal based on recursive difference equations

Aim: Generation of Sinusoidal waveform / signal based on recursive difference equations

EQUIPMENTS: PC with windows (95/98/XP/NT/2000).
MATLAB Software
THEORY: For the given difference equation, a sinusoidal signal/sequence is applied as the
input. Using the given initial conditions, the sinusoidal response of the given discrete system is
to be computed.
1. The difference equation is y(n)=x(n)+y(n-1), which is a first order system, with the initial
condition y(-1)=4.

PROGRAM:

I=input ('Enter the value of initial condition y(-1)')

n=0:0.001:1
f=input ('Enter the frequency')
x=sin (2*pi*f*n)
y=zeros (1, length(n))
for i=1:length(n)
y(i)=x(i)+I
I=y(i)
end
subplot(2,1,1)
plot(n,x)
title('Input signal x(n) applied')
xlabel('Time')
ylabel('Amplitude')
subplot(2,1,2)
plot(n,y)
title('Sinusoidal Response of the given first order system')
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xlabel('Time')
ylabel('Amplitude')
Enter the value of initial condition y(-1)
I=4
Enter the frequency
f =10

Expected Waveform:

Input signal x(n) applied

1

0.5
Amplitude
0

-0.5

-1
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Time
Sinusoidal Response y(n) of the given first order system
40
Amplitude
30

20

10

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1
Time

Result:

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Experiment No: 2
Histogram of White Gaussian Noise and Uniformly Distributed Noise.

Aim: Histogram of White Gaussian Noise and Uniformly Distributed Noise using MAT Lab.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:-

1) Open MATLAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window
Program:
clc;
clear all;
x1=randn(1,5000);
x2=randn(1,5000);
figure;
plot(x1,x2,'.')
title('scattered plot of Gaussian Distribution Random numbers');
x1=rand(1,5000);
x2=rand(1,5000);
figure;
plot(x1,x2,'.')
title('scattered plot of uniform Distribution Random numbers');
x3=rand(1,100000);
figure;
subplot(2,1,1);
hist(x3)
title('uniform Distribution ');
y=randn(1,100000);

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subplot(2,1,2);
hist(y)
title('Gaussian Distribution');
ymu=mean(y);
ymsq=sum(y.^2)/length(y);
ysigma=std(y);
yvar=var(y);
yskew=skewness(y);
ykurt=kurtosis(y);

Expected Graph:

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uniform Distribution
15000

10000

5000

0
0 0.1 0.2 0.3 0.4 0.5 0.6 0.7 0.8 0.9 1

4
x 10 Gaussian Distribution
4

0
-5 -4 -3 -2 -1 0 1 2 3 4 5

Result:.

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Experiment No: 3
To find DFT / IDFT of given DT signal.

Aim: To compute the N-point DFT of a sequence using MAT Lab.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:-

1) Open MATLAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window

Program:

%N-point DFT of a sequence.

clc;
clear all;
close all;
x=input('Enter the sequence x[n]= ');
N=input('Enter the value N point= ');
L=length(x);
x_n=[x,zeros(1,N-L)];
for i=1:N
for j=1:N
temp=-2*pi*(i-1)*(j-1)/N;
DFT_mat(i,j)=exp(complex(0,temp));
end
end
X_k=DFT_mat*x_n';
disp('N point DFT is X[k] = ');
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disp(X_k);
mag=abs(X_k);
phase=angle(X_k)*180/pi;
subplot(2,1,1);
stem(mag);
grid on;
xlabel('frequency index k');
ylabel('Magnitude of X[k]');
axis([0 N+1 -2 max(mag)+2]);
subplot(2,1,2);
stem(phase);
grid on;
xlabel('frequency index k');
ylabel('Phase of X[k]');
axis ([0 N+1 -180 180]);

Output:
Enter the sequence x[n]= [ ]
Enter the value N point=
N point DFT is X[k] =

Expected Graph:

Result:.

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Experiment No.4
To find frequency response of a given system given in (Transfer Function/ Differential
equation form).

Aim: To find the frequency response of the following difference equation.

y ( n) – 5 y(n–1) = x(n) + 4 x(n–1)
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:-
1) Open MATLAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window

Program:
clc;
clear all;
close all;
b=[1,4]; %Numerator coefficients
a=[1,-5]; %Denominator coefficients
w=-2*pi:pi/256:2*pi;
[h]=freqz (b, a, w);
subplot(2, 1,1),
plot(w,abs(h));
xlabel('Frequency \omega');
ylabel('Magnitude');
grid on;
subplot(2, 1, 2);
plot(w, angle(h));
xlabel('Frequency \omega');
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ylabel('Phase - Radians');
grid on;
Expected Waveform:

Result:

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Experiment No: 5
Implementation of FFT of given sequence.
Aim: To determine the N-point FFT of a given sequence.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:-
1) Open MATLAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window

Program:
clc;
clear all;
close all;
x=input ('enter the sequence='); %x[n] sequence
n=input ('enter the length='); %N points for FFT
xk=fft(x,n); %computes FFT
disp (xk);
subplot(2,2,1);
stem(x,'filled');
title('plot of input sequence');
subplot(2,2,2);
stem(real(xk),'filled');
title('plot of real xk');
subplot(2,2,3);
stem(abs(xk),'filled');
title('Magnitude spectrum');
subplot(2,2,4);

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stem(angle(xk),'filled');
title('Phase spectrum');

Expected Waveform:

Enter the sequence= [1 2 1 0]

Enter the length=5
4.0000 0.8090 - 2.4899i -0.3090 - 0.2245i -0.3090 + 0.2245i

0.8090 + 2.4899i

Result:

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Experiment No: 6
Determination of Power Spectrum of a given signal(s).

Aim: To obtain power spectrum of given signal using MATLAB.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:-
1) Open MAT LAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window

Program:

%Power density spectrum of a sequence.

clc;
clear all;
close all;
x=input ('enter the length of the sequence=');
y=input ('enter the sequence=');
fs=input ('enter sampling frequency=');
N=input ('enter the value of N=');
q=input ('enter the window name1=');
m=input ('enter the window name2=');
s=input ('enter the window name3=');
pxx1=psd(y, N, fs, q)
plot (pxx1,'c');
hold on;
grid on;
pxx2=psd(y, N, fs, m)
plot (pxx2,'k');
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hold on;
pxx3=psd(y, N, fs, s)
plot (pxx3,'b');
hold on;
xlabel ('-->Frequency');
ylabel ('-->Magnitude');
title ('POWER SPECTRAL DENSITY OF A SEQUENCE');
hold off;

Output:
enter the length of the sequence=5
enter the sequence=[3 4 5 6 7]
enter sampling frequency=200
enter the value of N=5
enter the window name1=triang (N)
enter the window name2=Kaiser (N, 4.5)
enter the window name3=rectwin (N)

pxx1 = 106.5789 pxx2 = 77.3378 pxx3 =125.0000

11.0127 24.6721 3.6180
0.3031 0.2229 1.3820

Result:
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Experiment No: 7
Implementation of LP FIR & HP FIR filters for a given sequence.

Aim: To Design FIR LP Filter using Rectangular /Triangular/Kaiser Windowing Technique.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
Procedure:-
1) Open MATLAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window

Algorithm:
1) Enter the pass band ripple (rp) and stop band ripple (rs).
2) Enter the pass band frequency (fp) and stop band frequency (fs).
3) Get the sampling frequency (f), beta value.
4) Calculate the analog pass band edge frequencies, w1 and w2.
w1 = 2*fp/f
w2 = 2*fs/f
5) Calculate the numerator and denominator
6) Use an If condition and ask the user to choose either Rectangular Window or Triangular
Window or Kaiser Window.
7) Use rectwin, triang, Kaiser Commands
8) Calculate the magnitude of the frequency response in decibels (dBm=20*log10(abs( h))
9) Plot the magnitude response [magnitude in neither dB Vs nor malized frequency (om/pi)]
10) Give relevant names to x and y axes and gives an appropriate ate title for the plot.
11) Plot all the responses in a single figure window. [Make use of subplot]

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Program:

%FIR Filter design window techniques.

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

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y=kaiser( n1,beta);
disp('Kaiser window filter response');
end
%LPF
b=fir1( n,wp,y) ;
[h,o]=freqz(b,1,256);
m=20*log10 (abs (h));
plot(o/pi, m);
title('LPF Rectangular window filter response');
ylabel('Gain in dB-->');
xlabel(' Normalized frequency-->');
Expected Output:
Rectangular window filter response
enter pass band ripple =0.02
enter the stop band ripple =0.04
enter passband freq =1000
enter stopband freq =2000
enter sampling freq =5000
enter beta value =
enter your choice of window function 1. rectangular 2. triangular 3.kaiser: 1
Rectangular window filter response

Result:

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Experiment No: 8
Implementation of HP FIR filters for a given sequence.

Aim: To Design FIR HP Filter using Rectangular /Triangular/Kaiser Windowing Technique.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:-
1) Open MAT LAB
2) Open new M-file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window

Algorithm:

1) Enter the pass band ripple (rp) and stop band ripple (rs).
2) Enter the pass band frequency (fp) and stop band frequency (fs).
3) Get the sampling frequency (f), beta value.
4) Calculate the analog pass band edge frequencies, w1 and w2.
w1 = 2*fp/f
w2 = 2*fs/f
5) Calculate the numerator and denominator
6) Use an If condition and ask the user to choose either Rectangular Window or Triangular
Window or Kaiser Window.
7) Use rectwin, triang, Kaiser Commands
8) Calculate the magnitude of the frequency response in decibels (dBm=20*log10(abs( h))
9) Plot the magnitude response [magnitude in neither dB Vs nor malized frequency (om/pi)]
10) Give relevant names to x and y axes and gives an appropriate ate title for the plot.
11) Plot all the responses in a single figure window. [Make use of subplot]

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

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disp('kaiser window filter response');

end
%HPF
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs( h));
plot(o/pi,m);
title('HPF Rectangular window filter response');
ylabel('Gain in dB-->');
xlabel('Normalized frequency-->');

Expected Output:
Rectangular window filter response
enter passband ripple =0.04
enter the stopband ripple =0.05
enter passband freq =2000
enter stopband freq =2500
enter sampling freq =6000
enter beta value =
enter your choice of window function 1. rectangular 2. triangular 3.kaiser: 1
Rectangular window filter response

Result:

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Experiment No: 9
Implementation of LP IIR filter for a given sequence

Aim: -To Design and generate IIR Butterworth Analog LP Filter using MATLAB
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
Procedure:
1)Open MATLAB
2)Open new M- file
3)Type the program
4)Save in current directory
5)Compile and Run the program
6)For the output see command window \ Figure window
Algorithm:-
1) Enter the pass band ripple (rp) and stop band ripple (rs).
2) Enter the pass band frequency (fp) and stop band frequency ( fs).
3) Get the sampling frequency ( f).
4) Calculate the analog pass band edge frequencies, w1 and w2.
w1 = 2*fp/f
w2 = 2*fs/f
5) Calculate the order and 3dB cutoff frequency of the analog filter. [Make use of the following
function] [n,wn]=buttord (w1,w2,rp,rs, s )
6) Des ign an nth order analog lowpass Butter worth filter using the following statement.
[b,a]=butter (n,wn, s )
7) F ind the complex frequency response of the filter by using , freqs( ) function
[h,om]=freqs(b,a,w) where, w = 0:.01:pi
This function retur ns complex frequency response vector „ h and frequency vector „ om in
radians /samples of the filter.
8) Calculate the magnitude of the fr equency response in decibels (dB m=20*log10(abs( h))
9) Plot the magnitude response [magnitude in dB Vs nor malized frequency (om/pi)]

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10) Calculate the phase response using an = angle( h)

11) Plot the phase response [phase in radians Vs nor malized frequency (om/pi) ]
12) Give relevant names to x and y axes and give an appropr iate title for the plot.
13)Plot all the responses in a single figure window.[Make use of subplot]

Program:
% IIR LP filters.
clc;
clear all;
close all;
warning off;
disp('enter the IIR LP filter design specifications');
rp=input('enter the passband ripple =');
rs=input('enter the stopband ripple =');
wp=input('enter the passband freq =');
ws=input('enter the stopband freq =');
fs=input('enter the sampling freq =');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord (w1, w2, rp, rs,'s'); % Find the order n and cut-off frequency
disp ('Frequency response of IIR HPF is :');
[b,a]=butter (n, wn,'low','s'); % Find the filter co-efficients of LPF
w=0:.01:pi;
[h,om]=freqs(b,a,w); % Plot the frequency response
m=20*log10(abs( h));
subplot(2,1,1);
plot(om/pi,m);
title('magnitude response of IIR Low Pass filter is:');
xlabel('(a)Nor malized freq. -->');
ylabel('Gain in dB-->');
an=angle(h) ;

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subplot(2,1,2);
plot(om/pi,an);
title('phase respons e of IIR Low Pass filter is:');
xlabel('(b)Normalized freq. -->');
ylabel('Phase in radians -->');

Expected Output:
enter the IIR LP filter design specifications
enter the passband ripple =0.5
enter the stopband ripple =60
enter the passband freq =400
enter the stopband freq =600
enter the sampling freq =1000
Frequency response of IIR LPF is :

Result:

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%Design of IIR Chebyshew type I Low pass filter

Program:
%Design of IIR Chebyshew type I Low pass filter .
clc;
clear all;
close all;
wp=input('Enter the passband edge Normalised frequency = ');
ws=input('Enter the stopband edge Normalised frequency = ');
Rp=input('Enter the passband attenuation level (dB) = ');
Rs=input('Enter the stopband attenuation level (dB) = ');
[N,wn]=cheb1ord(wp,ws,Rp,Rs);
[b,a]=cheby1(N,Rp,wn); %replace by following for other filters
%[b,a]=cheby1(N,Rp,wn,'high'); Highpass filter
[h,w]=freqz(b,a);
mag=20*log10(abs(h));
phase=180*angle(h)/pi;
figure(1);
plot(w,abs(h));
title('Chebyshev Lowpass Filter');
xlabel('Normalised frequency');
ylabel('Magnitude');
grid;
figure(2);
subplot(2,1,1);
plot(w,mag);
title('Magnitude response');
xlabel('Normalised frequency');
ylabel('Magnitude in dB');
grid;
subplot(2,1,2);
plot(w,phase);
title('Phase response');

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xlabel('Normalised frequency');
ylabel('Phase in degrees');
grid;

Output:

Enter the pass band edge Normalized frequency = 0.2*pi

Enter the stop band edge Normalized frequency = 0.1*pi
Enter the pass band attenuation level (dB) = 1
Enter the stop band attenuation level (dB) = 15

Figure: 1

Figure: 2

Result:

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Experiment No: 10
Implementation of HP IIR filter for a given sequence

Aim: To Design and generate IIR Butterworth Analog HP Filter using MATLAB

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:
1)Open MATLAB
2)Open new M- file
3)Type the program
4)Save in cur rent directory
5)Compile and Run the program
6)For the output see command window \ Figure window

Algorithm:
1) Enter the pass band ripple (rp) and stop band ripple (rs).
2) Enter the pass band frequency ( fp) and stop band frequency ( fs).
3) Get the sampling frequency ( f).
4) Calculate the analog pass band edge frequencies, w1 and w2.
w1 = 2*fp/f
w2 = 2*fs/f
5) Calculate the order and 3dB cutoff frequency of the analog filter. [Make use of the following
function] [n,wn]=buttord (w1, w2, rp,rs, s )
6) Des ign an nth order analog lowpass Butter worth filter using the following statement.
[b,a]=butter (n,wn, s )
7) F ind the complex frequency response of the filter by using „freqs( ) function
[h,om]=freqs(b,a,w) where, w = 0:.01:pi This function retur ns complex frequency
response vector „ h and frequency vector „ om in radians /samples of the filter.
8) Calculate the magnitude of the fr equency response in decibels (dB m=20*log10(abs( h))

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9) Plot the magnitude response [magnitude in dB Vs nor malized frequency (om/pi)]

10) Calculate the phase response using an = angle ( h)
11) Plot the phase response [phase in radians Vs nor malized frequency (om/i) ]
12)Give relevant names to x and y axes and give an appropriate ate title for the plot.
13)Plot all the responses in a single figure window.[Make use of subplot]
Program:
% IIR HP butter worth filter.
clc;
clear all;
close all;
warning off;
disp('enter the IIR HP filter design specifications');
rp=input('enter the passband ripple =');
rs=input('enter the stopband ripple =');
wp=input('enter the passband freq =');
ws=input('enter the stopband freq =');
fs=input('enter the sampling freq =');
w1=2*wp/fs;
w2=2*ws/fs;
[n,wn]=buttord(w1,w2,rp,rs,'s'); % Find the order n and cut-off frequency
disp('Frequency response of IIR HYPF is :');
[b,a]=butter(n,wn,'HIGH','s'); % Find the filter co-efficients of LPF
w=0:.01:pi;
[h,om]=freqs(b,a,w); % Plot the frequency response
m=20*log10(abs( h));
subplot(2,1,1);
plot(om/pi,m);
title('magnitude response of IIR HIGH Pass filter is:');
xlabel('(a)Normalized freq. -->');
ylabel('Gain in dB-->');
an=angle(h) ;

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subplot(2,1,2);
plot(om/pi,an);
title('phase respons e of IIR HIGH Pass filter is:');
xlabel('(b)Normalized freq. -->');
ylabel('Phase in radians -->');

Output:
enter the IIR HP filter design specifications
enter the passband ripple =0.5
enter the stopband ripple =60
enter the passband freq =400
enter the stopband freq =600
enter the sampling freq =1000
Frequency response of IIR HPF is:

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Result:
%IIR Chebyshew type I high pass filter
Program:
% IIR Chebyshew type I high pass filter
clear all;
close all;
clc;
wp=input('Enter the passband edge Normalised frequency = ');
ws=input('Enter the stopband edge Normalised frequency = ');
Rp=input('Enter the passband attenuation level (dB) = ');
Rs=input('Enter the stopband attenuation level (dB) = ');
[N,wn]=cheb1ord(wp,ws,Rp,Rs);
[b,a]=cheby1(N,Rp,wn,'high'); %High pass filter
[h,w]=freqs(b,a);
mag=20*log10(abs(h));
phase=180*angle(h)/pi;
figure(1);
plot(w,abs(h));
title('Chebyshev Lowpass Filter');
xlabel('Normalised frequency');
ylabel('Magnitude');
grid;
figure(2);
subplot(2,1,1);
plot(w,mag);
title('Magnitude response');
xlabel('Normalised frequency');
ylabel('Magnitude in dB');
grid;
subplot(2,1,2);

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plot(w,phase);
title('Phase response');
xlabel('Normalised frequency');
ylabel('Phase in degrees');
grid;
Expected Output:
Enter the passband edge Normalised frequency = 0.2*pi
Enter the stopband edge Normalised frequency = 0.1*pi
Enter the passband attenuation level (dB) = 1
Enter the stopband attenuation level (dB) = 15
>> 2
Figure: 1

Figure: 2

Result: .

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Experiment No: 11
Generation of Narrow Band Signal through Filtering

Aim: To generate a sinusoidal signal through filtering.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).

MATLAB Software

PROGRAM:
clc;
clear all;
close all;
echo on;
t0=.2; % signal duration
ts=0.001; % sampling interval
fc=250; % carrier frequency
fs=1/ts; % sampling frequency
t=[-t0/2:ts:t0/2]; % time vector
kf=100; % deviation constant
m=cos(2*pi*10*t); % the message signal
int_m(1)=0;
for i=1:length(t)-1 % integral of m
int_m(i+1)=int_m(i)+m(i)*ts;
echo off ;
end
echo on ;
u=cos(2*pi*fc*t+2*pi*kf*int_m); % modulated signal
%now lets filter the signal
[z,p] = butter(1,2*50/fs,'low');

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filter_out = filter(50,[1 50],u); %this damn filter doesn't work!

subplot(2,1,1);
plot(t,u);
hold on;
plot(t,m,'r');
subplot(2,1,2);
plot(t,filter_out);
hold on;
plot(t,m,'r');

Expected Output:

Result: .

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Experiment No: 12

Generation of DTMF Signals

Aim: To generate DTMF signals

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software
THEORY: Introduction Analog DTMF telephone signaling is based on encoding standard
telephone. Keypad digits and symbols in two audible sinusoidal signals of frequencies FL and
FH. Thus the scheme gets its name as dual tone multi frequency (DTMF).
12 13 14 16
Hz
09 36 77 33
697 1 2 3 A
770 4 5 6 B
852 7 8 9 C
941 * 0 # D
Each digit or symbol represented in figure 1 has 2 distinct high and low frequency components.
Thus each high-low frequency pair uniquely identifies the corresponding telephone keypad digit
or symbol. Each key pressed can be represented as a discrete time signal of form
dt[n] = sin[ωLn] + sin[ωLn] , 0 ≤ n ≤ N-1 (1)
Where N is defined as number of samples taken. Typically in the sampling frequency used is
8khz. Thus if the two individual frequency components of the signal can be identified then the
number dialed can be decoded.
-- In this report I have used (dual tone and digit/symbols) interchangeably but both mean the
same. Dual tone means the encoded samples of the corresponding DTMF digits/symbols.
The DTMF encoder is implemented in MATLAB function dtmfe.m. The implementation is
based on a digital oscillator, that will generate sinusoidal tones at frequencies Fo in response to an
input signal x[n] = δ[n]. Note :Implementation of DTMF Encoder

x[n] H[n] y[n] {y[n] = x[n]*H[n]

Consider a causal filter with

y(n) - 2*cos(2*pi * f *Ts)y(n-1) + y(n-2) =0*x(n) sin(f)x(n-1) + 0*x(n-2).
The impulse response of this system tells us that this indeed is a digital oscillator. The H[n] is
plotted and is sinusoidal and hence any input to this system will oscillate .

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PROGRAM:
%Generation of DTMF signals
Fs = 8000; % Sampling frequency
Ts = 1/Fs;
Numof_samples =input('enter the no of samples');
dialnumber=input('enter the dial number');
T = Ts*(0:Numof_samples-1)';
switch dial_num
case 0
F1 = 941;
F2 = 1336;
case 1
F1 = 697;
F2 = 1209;
case 2
F1 = 697;
F2 = 1336;
case 3
F1 = 697;
F2 = 1477;
case 'A'
F1 = 697;
F2 = 1633;
case 4
F1 = 770;
F2 = 1209;

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case 5
F1 = 770;
F2 = 1336;
case 6
F1 = 770;
F2 = 1477;
case 'B'
F1 = 770;
F2 = 1633;
case 7
F1 = 852;
F2 = 1209;
case 8
F1 = 852;
F2 = 1336;
case 9
F1 = 852;
F2 = 1477;
case 'C'
F1 = 852;
F2 = 1633;
case '*'
F1 = 941;
F2 = 1209;
case '#'
F1 = 941;
F2 = 1477;

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otherwise
F1 = 941;
F2 = 1633;
end;
firstsine = cos(2*pi*F1*T); % first sinusoidal signal
secondsine = cos(2*pi*F2*T); % second sinusoidal signal
d = firstsine + secondsine;
dtmfoutput = d ;
figure(1);
title('THE DTMF OUTPUT');
plot(dtmfoutput);

Expected Output:: enter the no of samples 128

enter the dial number 3

RESULT:

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Experiment No: 13
Implementation of Decimation Process

Aim: To perform Decimation process using Mat lab.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:

1) Open MAT LAB

2) Open new M- file

3) Type the program

4) Save in current directory

5) Compile and Run the program

6) For the output see command window \ Figure window

Theory:

Sampling rate conversion (SRC) is a process of converting a discrete-time signal at a given rate
to a different rate. This technique is encountered in many application areas such as:

 Digital Audio

 Communications systems

 Speech Processing

 Antenna Systems

 Radar Systems etc

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Sampling rates may be changed upward or downward. Increasing the sampling rate is called
Interpolation and decreasing the sampling rate is called decimation. Reducing the sampling rate
by a factor of M is achieved by discarding ever y M- 1 samples, or, equivalently keeping
every M th sample. Increasing the sampling rate by a factor of L ( interpolation by factor L) is
achieved by inserting L- 1 zeros into the output stream after ever y sample from the input
stream of samples. This system can perform SRC for the following cases:

• Decimation by a factor of M

• Interpolation by a factor of L

• SRC by a rational factor of L/M.

Decimator:

To reduce the sampling rate by an integer factor M, assume a new sampling period.The re-
sampled signal is The system for performing this operation, called down- sampler, is shown
below: Down-sampling generally results in aliasing. Therefore, in order to prevent aliasing, x (n)

Should be filtered prior to down-sampling with a low-pass filter that has a cutoff frequency The
cascade of a low-pass filter with a down- sampler illustrated below and is called
Decimator.

Program:

% Illustration of Decimation Process.

clc;
close all;
clear all;
M=input('enter Down-sampling factor : ');
N=input('enter number of samples :') ;
n=0:N- 1;
x=sin(2*pi*0.043*n)+ sin(2*pi*0.031*n);
y=decimate(x,M,'fir');

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subplot(2,1,1);
stem(n,x(1:N) );
title('Input Sequence') ;
xlabel('Time index n');
ylabel('Amplitude');
subplot(2,1,2);
m=0:(N/M)-1;
stem(m,y(1:N /M));
title('Output Sequence');
xlabel('Time index n');
ylabel('Amplitude') ;

Expected Output:

Enter Down-sampling factor: 4

Enter number of samples: 100

Result:
Note: Observe the Output Sequence for Different values of M.

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Experiment No: 14
Implementation of Interpolation Process.
Aim: To perform interpolation process using Mat lab.
EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:
1) Open MAT LAB
2) Open new M- file
3) Type the program
4) Save in current directory
5) Compile and Run the program
6) For the output see command window \ Figure window
Theory:
To increase the sampling rate by an integer factor L. I f xa(t) is sampled with a sampling
frequency fs = 1/Ts, then To increase the sampling rate by an integer factor L, it is necessary to
extract the samples from x(n). The samples of xi( n) for values of n that are integer multiples of
L are easily extracted from x(n) as follows:
The system per forming the operation is called up- sampler and is shown below:
After up-sampling, it is necessary to remove the frequency scaled images in xi(n), except
those that are at integer multiples of 2p. This is accomplis hed by filter ing xi(n) with a low- pass
filter that has a cutoff frequency of p /L and a gain of L. In the time domain, the low -pass filter
inter - polates between the samples at integer multiples of L as shown below and is called
interpolator.
Program:
% Illustration of Interpolation Process
clc;
close all;
clear all;
L=input('Up-sampling factor = ') ;

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N=input('enter number of samples :') ;

n=0:N- 1;
x=sin(2*pi*0.043*n)+sin(2*pi*0.031*n);
y=interp(x,L);
subplot(2,1,1);
stem(n,x(1:N) );
title('Input Sequence') ;
xlabel('Time index n');
ylabel('Amplitude');
subplot(2,1,2);
m=0:(N*L)-1;
stem(m,y(1:N*L));
title('Output Sequence');
xlabel('Time index n');
ylabel('Amplitude');
Expected Output:
Up-sampling factor = 5
Enter number of samples: 50

Result:

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Experiment No: 15
Implementation of I/D sampling rate converters.

Aim: To study sampling rate conversion by a rational for m using M ATLAB

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

Procedure:

1) Open MAT LAB

2) Open new M- file

3) Type the program

4) Save in current directory

5) Compile and Run the program

6) For the output see command window \ Figure window

Theory:

SRC by rational factor:

SRC by L/M requires performing an interpolation to a sampling rate which is divisible by

both L and M. The final output is then achieved by decimating by a factor of M. The need for a
non-Integer sampling rate conversion appears when the two systems operating at different
sampling rates have to be connected, or when there is a need to convert the sampling rate of the
recorded data into another sampling rate for further processing or reproduction. Such
applications are very common in telecommunications, digital audio, multimedia and others. An
example is transferring data from compact disc (CD) system at a rate of 44.1 kHz to a digital
audio tape at 48 kHz. This can be achieved by increasing the data rate of the CD by a factor of
48/44.1, a non- integer.

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Illustration for sampling rate converter is:

I f M> L, the resulting operation is a decimation process by a non- integer, and when M<L it is

Interpolation. I f M=1, the generalized system reduces to the simple integer interpolation and if

L=1 it reduces to integer decimation.

Program:

clc;
close all;
clear all;
L=input('Enter Up-sampling factor :');
M=input('Enter Down-sampling factor :');
N=input('Enter number of samples :');
n=0:N- 1;
x=sin(2*pi*0.43*n)+sin(2*pi*0.31*n);
y=resample(x,L,M) ;
subplot(2,1,1);
stem(n,x(1:N) );
axis([0 29 -2.2 2.2]);
title('Input Sequence') ;
xlabel('Time index n');
ylabel('Amplitude') ;
subplot(2,1,2);
m=0:(N*L/M)-1;
stem(m,y(1:N*L/M));
axis([0 (N*L/M)-1 -2.2 2.2]);
title('Output Sequence');
xlabel('Time index n');

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ylabel('Amplitude') ;

Expected Output:
Enter Up-sampling factor: 6
Enter Down-sampling factor: 3
Enter number of samples: 40

Result:

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Experiment No: 16
IMPULSE RESPONSE OF FIRST ORDER AND SECOND ORDER SYSTEM
Aim: To determine the response of first order and second order systems.

EQUIPMENTS:
PC with windows (95/98/XP/NT/2000).
MATLAB Software

PROGRAM:
%impulse response of a first order system
clear all;
b=[1];
a=[1,-.8];
w=0:0.1:2*pi;
[h]=freqz(b,a,w);
subplot(2,1,1);
plot(w/pi,abs(h));
xlabel('normalised frequency\omega/\pi'), ylabel('magnitude');
title('impulse response of first order system h(n)=(0.8)^n u(n)');
subplot(2,1,2);
plot(w/pi,angle(h));
xlabel('normalised frequency'), ylabel('phase angle in radians');

Expected Output:

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RESULT:

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