DSP Lab Maual

LABORATORY MANUAL
DIGITAL SIGNAL PROCESSING LAB

III B.Tech. ECE - II Sem (2014-15)
Department of Electronics & Communication Engineering
BALAJI INSTITUTE OF ENGINEERING & SCIENCES

Laknepally, Narsampet, Warangal
Dept. of Electronics & Communication Engineering
LIST OF EXPERIMENTS
Academic Year: 2014 15 (II Sem)
Name of the Lab : DIGITAL SIGNAL PROCESSING LAB

Class
: III B.Tech., ECE
Regulation
: R 09
The programs shall be implemented in software (Using MATLAB / Lab

view / C programming/ Equivalent) and hardware (Using TI / Analog
devices / Motorola / Equivalent DSP processors).
Note: - Minimum of 12 experiments has to be conducted.
1. Determination of Power Spectrum of a given signal(s).
2. To find frequency response of a given system given in (Transfer
Function/ Differential equation form).
3. To find DFT / IDFT of given DT signal
4. Implementation of FFT of given sequence
5. Implementation of LP IIR filter for a given sequence
6. Implementation of HP IIR filter for a given sequence
7. Implementation of LP FIR filter for a given sequence
8. Implementation of HP FIR filter for a given sequence
9. Generation of Sinusoidal signal through filtering
10.
Implementation of Decimation Process
11.
Implementation of Interpolation Process
12.
Impulse response of first order and second order systems.
13.
Noise removal: Add noise above 3 KHz and then remove,
interference suppression using 400 Hz tone.
14.
Audio application such as to plot a time and frequency
display of microphone plus a cosine using DSP. Read a .wav file
and match with their respective spectrograms.
15.
Generation of DTMF signals
16.
Generation of Sinusoidal waveform / signal based on
recursive difference equations
17.
Implementation of I/D sampling rate converters
Experiment No. 1
PROGRAM FOR DETERMINATION OF POWER SPECTRUM

OF A GIVEN SIGNAL
Aim: To determine the power spectrum of a given signal.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab 7.5 & CCStudio3.1
MAT Lab program for Power spectrum of a signal

t = 0:0.001:0.6;
x = sin(2*pi*50*t)+sin(2*pi*120*t);
y = x + 2*randn(size(t));
figure,plot(1000*t(1:50),y(1:50))
title('Signal Corrupted with Zero-Mean Random Noise')
xlabel('time (milliseconds)');
Y = fft(y,512);
%The power spectral density, a measurement of the energy at various
frequencies, is:
Pyy = Y.* conj(Y) / 512;
f = 1000*(0:256)/512;
figure,plot(f,Pyy(1:257))
title('Frequency content of y');
xlabel('frequency (Hz)');
Signal Corrupted with Zero-Mean Random Noise
Frequency content of y
Result: The power spectrum of a given signal is determined
Experiment No. 2
PROGRAM TO FIND THE FREQUENCY RESPONSE

OF A GIVEN SYSTEM
Aim: To find the frequency response of a given system
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program for frequency response of a given system

num=input('type the numerator vector');
den=input('type the denominator vector');
N=input('number of frequency points');
w=0:pi/N:pi;
H=freqz(num,den,w);
figure;
subplot(2,1,1);plot(w/pi,real(H));
xlabel('\omega /pi');ylabel('Amplitude')
title('Real part');
subplot(2,1,2);
plot(w/pi,imag(H));
xlabel('\omega /pi');ylabel('Amplitude')
title('Imaginary part');
figure;
subplot(2,1,1);
plot(w/pi,abs(H));
xlabel('\omega /pi');ylabel('Magnitude');
title('Magnitude spectrum');
subplot(2,1,2);
plot(w/pi,angle(H));
xlabel('\omega /pi');ylabel('Phase(radians)');
title('Phase Spectrum');
INPUT & OUTPUT
type the numerator vector
type the denominator vector
number of frequency points
Graph for Real Part
Graph for Imaginary Part
Magnitude Spectrum
Phase Spectrum
Result: Frequency response of a given system is plotted.

Experiment No. 3
PROGRAM TO FIND THE DISCRETE FOURIER TRANSFORM OF

A GIVEN SEQUENCE
Aim: To find the DFT of a given sequence.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program for DFT of a given sequence

close all;
clear all;
N=input('Howmany point DFT do you want');
x2=input('Enter the sequence=');
n2=length(x2);
c=zeros(N);
x2=[x2 zeros(1,N-n2)];
for k=1:N
for n=1:N
w=exp(-2*pi*i*(k-1)*(n-1)/N);
%prev. step=>evaluating w-matrix
x(n)=w;
end
c(k,:)=w;
end
r=[c]*[x2']
%plotting magnitude and angle
subplot(2,1,1);
stem(abs(r));
title('DFT absolute value:');

subplot(2,1,2);
stem(angle(r));
title('DFT angle:');
INPUT & OUTPUT:

Enter the Sequence
Output Sequence
:
Magnitude Plot
Phase Plot
RESULT: DFT of a given sequence is determined.

Experiment No. 4
PROGRAM TO IMPLEMENT FAST FOURIER TRANSFORM

OF A GIVEN SEQUENCE
Aim: To implement FFT of a given sequence.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program for FFT of a given sequence

clc;
clear all;
close all;
tic;
x=input('Enter the sequence:');
n=input('Enter the length of FFT:');
%compute fft
disp('Fourier transformed signal');
X=fft(x,n)
subplot(1,2,1);stem(x);
title('i/p signal');
xlabel('n --->');
ylabel('x(n) -->');grid;
subplot(1,2,2);stem(real(X));
title('fft of i/p x(n) is:');
xlabel('Real axis --->');
ylabel('Imaginary axis -->');grid;
INPUT & OUTPUT
Enter the sequence:
Enter the length of FFT:
Fourier transformed signal
Input Sequence Graph
Output Sequence Graph
RESULT: FFT of a given sequence is determined.

Experiment No. 5
PROGRAM TO IMPLEMENT LOW PASS IIR FILTER

FOR A GIVEN SEQUENCE
Aim: To implement low pass IIR filter for a given sequence.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to implement low pass IIR filter for a given sequence
clc;
clear all;
close all;
disp('Enter the IIR filter design specifications:');
rp=input('Enter the passband ripple:');
rs=input('Enter the stopband 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,'s');
[b,a]=butter(n,wn,'low','s');
w=0:0.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure,subplot(2,1,1);plot(om/pi,m);
title('Magnitude response of IIR filter is:');
xlabel('(a) Normalized freq. -->');

ylabel('Gain in dB-->');
subplot(2,1,2);plot(om/pi,an);
title('Phase response of IIR filter is:');
xlabel('(b)Normalized frequency -->');
ylabel('Phase in radians-->');
INPUT & OUTPUT:

Enter the IIR filter design specifications:
Enter the passband ripple:
Enter the stopband ripple:
Enter the passband frequency:
Enter the stopband frequency:
Enter the sampling frequency:
Magnitude Response of Low Pass IIR Filter
Phase Response of Low Pass IIR Filter
RESULT: Frequency response of the Low pass IIR filter is plotted.
Experiment No. 6
PROGRAM TO IMPLEMENT HIGH PASS IIR FILTER

Aim: To implement high pass IIR filter for a given sequence.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to implement high pass IIR filter for a given sequence
clc;
clear all;
close all;
disp('Enter the IIR filter design specifications:');
rp=input('Enter the passband 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,'s');
[b,a]=butter(n,wn,'high','s');
w=0:0.01:pi;
[h,om]=freqs(b,a,w);
m=20*log10(abs(h));
an=angle(h);
figure,subplot(2,1,1);plot(om/pi,m);
title('Magnitude response of IIR filter is:');
xlabel('(a) Normalized freq. -->');

subplot(2,1,2);plot(om/pi,an);
title('Phase response of IIR filter is:');
xlabel('(b)Normalized frequency -->');
ylabel('Phase in radians-->');
INPUT & OUTPUT

Enter the IIR filter design specifications:
Enter the passband ripple:
Enter the passband frequency:
Enter the stopband frequency:
Enter the sampling frequency:
Magnitude Response of High Pass IIR Filter
Phase Response of High Pass IIR Filter
RESULT: Frequency response of the high pass IIR filter is plotted.
Experiment No. 7
PROGRAM TO IMPLEMENT LOW PASS FIR FILTER

Aim: To implement low pass FIR filter for a given sequence.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to implement low pass FIR filter for a given sequence
clc;
clear all;
close all;
rp=input('Enter passband ripple:');
fp=input('Enter passband frequency:');
fs=input('Enter stopband frequency:');
f=input('Enter 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
c=input('Enter your choice of window function 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);
disp('Kaiser window filter response');
end
%Low Pass filter
b=fir1(n,wp,y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(1,1,1);plot(o/pi,m);
title('LPF');
xlabel('(a)Normalized frequency-->');
INPUT & OUTPUTS
Enter passband ripple:
Enter passband frequency:
Enter stopband frequency:
Enter sampling frequency:
Enter your choice of window function 1. Rectangular 2. Triangular 3.Kaiser:
Rectangular window filter response
INPUT & OUTPUTS

Triangular window filter response
INPUT & OUTPUTS

Kaiser window filter response
RESULT:
The frequency responses of Low pass FIR filter using Rectangular, Triangular &
Kaiser windows are plotted.
Experiment No. 8
PROGRAM TO IMPLEMENT HIGH PASS FIR FILTER

Aim: To implement high pass FIR filter for a given sequence.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to implement high pass FIR filter for a given sequence
clc;
clear all;
close all;
rp=input('Enter passband ripple:');
fp=input('Enter passband frequency:');
fs=input('Enter stopband frequency:');
f=input('Enter 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
c=input('Enter your choice of window function 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);
disp('Kaiser window filter response');
end
%High pass filer
b=fir1(n,wp,'high',y);
[h,o]=freqz(b,1,256);
m=20*log10(abs(h));
subplot(1,1,1);plot(o/pi,m);
title('HPF');
xlabel('(b)Normalized frequency-->');
INPUT & OUTPUTS
Rectangular Window Filter Response
INPUT & OUTPUTS

Triangular Window Filter Response
INPUT & OUTPUTS

Kaiser Window Filter Response
RESULT:
The frequency responses of High pass FIR filter using Rectangular, Triangular &
Kaiser windows are plotted.
Experiment No. 9
PROGRAM TO GENERATE A SINUSOIDAL SIGNAL THROUGH

FILTERING
Aim: To generate a sinusoidal signal through filtering.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to generate a sinusoidal signal through filtering.

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');
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');
OUTPUT:
RESULT:
The Sinusoidal signal is filtered from a modulated signal using butter worth
lowpass filter.
Experiment No. 10,11
PROGRAM TO IMPLEMENT DECIMATION & INTERPOLATION

PROCESSES
Aim: To implement decimation & interpolation on a given signal.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to implement decimation & interpolation on a given signal

clc;
close all;
clear all;
t=0:0.001:0.5;
F=100;
X=2*sin(2*pi*F*t);
Xp=X(1:100);
de=decimate(Xp,2);
tp=2*t(1:50);
interp=interp(de,2);
subplot(3,1,1);
stem(t(1:100),X(1:100));
grid;
xlabel('t');
ylabel('X(t)');
title('Plot of input sequence');
subplot(3,1,2);
stem(tp,de);
grid;
xlabel('t');
ylabel('Xd(t)');
title('decimated output sequence');
subplot(3,1,3);
stem(t(1:100),interp);
grid;
xlabel('t');
ylabel('xi(t)');
title('Interpolated output sequence');
OUTPUT:
Plot of Input Sequence
Plot of Decimated Sequence
Plot of Interpolated Sequence
RESULT: Given signal is decimated & interpolated.
Experiment No. 12
PROGRAM TO PLOT THE IMPULSE RESPONSE OF

SECOND ORDER SYSTEM
Aim: To plot the impulse response of second order system .
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program to plot the impulse response of second order system.
clc;
clear all;
close all;
a = [-0.5572 -0.7814;0.7814 0];
b = [1 -1;0 2];
c = [1.9691 6.4493];
sys = ss(a,b,c,0);
impulse(sys);
INPUT & OUTPUTS
RESULT: The impulse response of second order system is plotted.
Experiment No. 13
PROGRAM FOR NOISE REMOVAL

Aim: To add noise above 3kHz and then remove interference suppression using 400 Hz
tone.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
C Program
#include "NCcfg.h"
#include "dsk6713.h"
#include "dsk6713_aic23.h"
#define beta 1E-13
#define N 30
//rate of convergence
//adaptive FIR filter length-vary this parameter & observe
float delay[N];
float w[N];
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,T;
/* Initialize the board support library, must be called first */
DSK6713_init();
hCodec = DSK6713_AIC23_openCodec(0, &config);
/* Start the codec */
DSK6713_AIC23_setFreq(hCodec, 1);
for (T = 0; T < 30; T++)

{
w[T] = 0;
delay[T] = 0;
//Initialize the adaptive FIR coeffs=0

//init buffer for weights
//init buffer for delay samples
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=(short int)adaptive_filter(l_input,r_input);
r_output=l_output;
while (!DSK6713_AIC23_write(hCodec, l_output)); /* Send o/p to the left channel
*/
while (!DSK6713_AIC23_write(hCodec, r_output)); /* Send o/p to the right
channel*/
}
DSK6713_AIC23_closeCodec(hCodec); /* Close the codec */
}
signed int adaptive_filter(int l_input,int r_input)
//ISR
{
short i,output;
float yn=0, E=0, dplusn=0,desired,noise;
desired = l_input;
noise = r_input;
dplusn = (short)(desired + noise);
delay[0] = noise;
for (i = 0; i < N; i++)
//noise as input to adapt FIR
//to calculate out of adapt FIR
yn += (w[i] * delay[i]);
E = (desired + noise) - yn;
for (i = N-1; i >= 0; i--)
//desired+noise
//output of adaptive filter

//"error" signal=(d+n)-yn
//to update weights and delays
{
w[i] = w[i] + beta*E*delay[i]; //update weights
delay[i] = delay[i-1];
//update delay samples
//output=((short)E); //error signal as overall output

output=((short)dplusn);//output (desired+noise)
//overall output result
return(output);
}
Result:
Experiment No. 14
PROGRAM FOR AUDIO APPLICATION

Aim: To plot a time & frequency display of micro phone plus a cosine using DSP read a
wav file and match with their respective spectrograms.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
Spectrogram_rtdx_mtl.c Time-Frequency analysis of signals Using RTDX-MATLAB

#include "dsk6713_aic23.h"
fs=DSK6713_AIC23_FREQ_8KHZ;
#include <rtdx.h>
#include <math.h>
#include "hamming.cof"
#define PTS 256
#define PI 3.14159265358979
typedef struct {float real,imag;} COMPLEX;
void FFT(COMPLEX *Y, int n);
float iobuffer[PTS],iobuffer1[PTS],a[PTS];
float x1[PTS];
short i;
int j, k,l, curr_block = 0;
short buffercount = 0;
short flag = 0;
COMPLEX w[PTS];
COMPLEX samples[PTS];
RTDX_CreateOutputChannel(ochan);
main()
{
for (i = 0 ; i<PTS ; i++)
{
w[i].real = cos(2*PI*i/512.0);
w[i].imag =-sin(2*PI*i/512.0);
}
comm_intr();
while(!RTDX_isOutputEnabled(&ochan))
puts("\n\n Waiting . . . ");
for(l=0;l<256;l++)
a[l]=cos(2*3.14*1500*l/8000);
for(k=0;k<5000;k++)
{
while (flag == 0) ;
flag = 0;
for (i = 0 ; i < PTS ; i++)
{ iobuffer1[i]=iobuffer[i]+a[i];
samples[i].real=h[i]*iobuffer1[i];
iobuffer1[i] = x1[i];
}
for (i = 0 ; i < PTS ; i++)
samples[i].imag = 0.0;
FFT(samples,PTS);
for (i = 0 ; i < PTS ; i++)

{
x1[i] = (samples[i].real*samples[i].real
+ samples[i].imag*samples[i].imag)/16;
}
RTDX_write(&ochan, x1, sizeof(x1)/2); //send 128 samples to PC
}
}
interrupt void c_int11()
{
output_sample((short)(iobuffer[buffercount]));
iobuffer[buffercount++]=(short)(input_sample());
if (buffercount >= PTS)
{
buffercount = 0;
flag = 1;
}
}
FFT.c C callable FFT function in C

#define PTS 256
//# of points for FFT
typedef struct {float real,imag;} COMPLEX;

extern COMPLEX w[PTS];
//twiddle constants stored in w
void FFT(COMPLEX *Y, int N)
//input sample array, # of points
{
COMPLEX temp1,temp2;
int i,j,k;
//temporary storage variables
//loop counter variables
int upper_leg, lower_leg;

int leg_diff;
//index of upper/lower butterfly leg
//difference between upper/lower leg
int num_stages = 0;
int index, step;
//number of FFT stages (iterations)

//index/step through twiddle constant
i = 1;
//log(base2) of N points= # of stages
do
{
num_stages +=1;
i = i*2;
}while (i!=N);
leg_diff = N/2;
//difference between upper&lower legs
step = 512/N;
//step between values in twiddle.h
for (i = 0;i < num_stages; i++) //for N-point FFT

{
index = 0;
for (j = 0; j < leg_diff; j++)
{
for (upper_leg = j; upper_leg < N; upper_leg += (2*leg_diff))

{
lower_leg = upper_leg+leg_diff;
temp1.real = (Y[upper_leg]).real + (Y[lower_leg]).real;
temp1.imag = (Y[upper_leg]).imag + (Y[lower_leg]).imag;
temp2.real = (Y[upper_leg]).real - (Y[lower_leg]).real;
temp2.imag = (Y[upper_leg]).imag - (Y[lower_leg]).imag;
(Y[lower_leg]).real = temp2.real*(w[index]).real
-temp2.imag*(w[index]).imag;
(Y[lower_leg]).imag = temp2.real*(w[index]).imag
+temp2.imag*(w[index]).real;
(Y[upper_leg]).real = temp1.real;
(Y[upper_leg]).imag = temp1.imag;
}
index += step;
}
leg_diff = leg_diff/2;
step *= 2;
}
j = 0;
for (i = 1; i < (N-1); i++)
//bit reversal for resequencing data
{
k = N/2;
while (k <= j)
{
j = j - k;
k = k/2;
}
j = j + k;
if (i<j)
{
temp1.real = (Y[j]).real;
temp1.imag = (Y[j]).imag;
(Y[j]).real = (Y[i]).real;
(Y[j]).imag = (Y[i]).imag;
(Y[i]).real = temp1.real;
(Y[i]).imag = temp1.imag;
}
}
return;
}
Spectrogram_RTDX.m For spectrogram plot using RTDX with MATLAB
clc;
ccsboardinfo
%board info
cc=ccsdsp('boardnum',0);
%set up CCS object
reset(cc);
%reset board
visible(cc,1);
%for CCS window
enable(cc.rtdx);
%enable RTDX
if ~isenabled(cc.rtdx);
error('RTDX is not enabled')
end
cc.rtdx.set('timeout',50);
%set 50sec timeout for RTDX
open(cc,'spectrogram1.pjt');
%open CCS project
load(cc,'./debug/spectrogram1.out');
%load executable file
run(cc);
%run program
configure(cc.rtdx,2048,1);
open(cc.rtdx,'ochan','r');
%configure one RTDX channel

%open output channel
pause(3)
%wait for RTDX channel to open
enable(cc.rtdx,'ochan');
%enable channel from DSK
isenabled(cc.rtdx,'ochan');
M = 256;
%window size
N = round(M/2);
B = 128;
%No. of blocks (128)
fs = 8000;
%sampling rate
t=(1:B)*(M/fs);
%spectrogram axes generation
f=((0:(M-1)/2)/(M-1))*fs;
set(gcf,'DoubleBuffer','on');
y = ones(N,B);
column = 1;
set(gca,'NextPlot','add');
axes_handle = get(gcf,'CurrentAxes');
set(get(axes_handle,'XLabel'),'String','Time (s)');
set(get(axes_handle,'YLabel'),'String','Frequency (Hz)');
set(get(axes_handle,'Title'),'String','\fontname{times}\bf Real-Time Spectrogram');
set(gca,'XLim', [0 4.096]);
set(gca,'YLim', [0 4000]);
set(gca,'XLimMode','manual');
set(gca,'YLimMode','manual');
for i = 1:32768
w=readmsg(cc.rtdx,'ochan','single');
%read FFT data from DSK
w=double(w(1:N));
y(:, column) = w';
imagesc(t,f,dB(y));
%plot spectrogram
column = mod(column, B) + 1;
end
halt(cc);
%halt processor
close(cc.rtdx,'ochan');
%close channel
clear cc
%clear object
Experiment No. 15
PROGRAM FOR GENERATION OF DTMF SIGNALS

Aim: To generate dual tone multiple frequency signals.
Software Required:
Operating System
Windows Xp
Constructor
Simulator
Software
MAT Lab program for generation of dual tone multiple frequency signals
fs = 4000;
dt = 1/fs;
frow = [697 770 852 941];
fcol = [1209 1336 1477];
ftone = [350 440];
fbusy = [480 620];
fring = [440 480];
Tkey = 0.25;
Ttone = 2;
Tring = [2 4];
Tbusy = [0.5 0.5];
task = 'go';
while strcmp(task,'end') == 0
task = input('task: ', 's');
switch task
case '1'
row = 1; col = 1;
t = 0:dt:Tkey;
x = cos(2*pi*frow(row)*t) + cos(2*pi*fcol(col)*t);
soundsc(x, fs);
case '2'
row = 1; col = 2;
t = 0:dt:Tkey;
soundsc(x, fs);
case '3'
row = 1; col = 3;
t = 0:dt:Tkey;
soundsc(x, fs);
case '4'
row = 2; col = 1;
t = 0:dt:Tkey;
soundsc(x, fs);
case '5'
row = 2; col = 2;
t = 0:dt:Tkey;
soundsc(x, fs);
case '6'
row = 2; col = 3;
t = 0:dt:Tkey;
soundsc(x, fs);
case '7'
row = 3; col = 1;
t = 0:dt:Tkey;
soundsc(x, fs);
case '8'
row = 3; col = 2;
t = 0:dt:Tkey;
soundsc(x, fs);
case '9'
row = 3; col = 3;
t = 0:dt:Tkey;
soundsc(x, fs);
case '*'
row = 4; col = 1;
t = 0:dt:Tkey;
soundsc(x, fs);
case '0'
row = 4; col = 2;
t = 0:dt:Tkey;
soundsc(x, fs);
case '#'
row = 4; col = 3;
t = 0:dt:Tkey;
soundsc(x, fs);
case 'tone'
t = 0:dt:Ttone;
x = cos(2*pi*ftone(1)*t) + cos(2*pi*ftone(2)*t);
soundsc(x, fs);
case 'ring'
ton = 0:dt:Tring(1);
toff = 0:dt:Tring(2);
xon = cos(2*pi*fring(1)*ton) + cos(2*pi*fring(2)*ton);
xoff = zeros(size(toff));
x = [xon xoff];
for k=1:2
sound(x, fs);
pause(Tring(1)+Tring(2));
end
case 'busy'
ton = 0:dt:Tbusy(1);
toff = 0:dt:Tbusy(2);
xon = cos(2*pi*fbusy(1)*ton) + cos(2*pi*fbusy(2)*ton);
xoff = zeros(size(toff));
x = [xon xoff];
for k=1:4
sound(x, fs);
pause(Tbusy(1)+Tbusy(2));
end
case 'end'
fprintf('\nthanks for playing! \n');
otherwise
fprintf('invalid task\n');
end
end
INPUT & OUTPUT
task: 1
task: 2
task: 3
task: 4
task: 5
task: 6
task: 7
task: 8
task: 9
task: 0
task: *
task: #
task: end
thanks for playing!

RESULT: Dual tone multiple frequency signals are generated.

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LABORATORY MANUAL

DIGITAL SIGNAL PROCESSING LAB

Department of Electronics & Communication Engineering

BALAJI INSTITUTE OF ENGINEERING & SCIENCES

Dept. of Electronics & Communication Engineering

Name of the Lab : DIGITAL SIGNAL PROCESSING LAB

: III B.Tech., ECE

The programs shall be implemented in software (Using MATLAB / Lab

PROGRAM FOR DETERMINATION OF POWER SPECTRUM

MAT Lab 7.5 & CCStudio3.1

MAT Lab program for Power spectrum of a signal

Result: The power spectrum of a given signal is determined

Dept. of Electronics & Communication Engineering

PROGRAM TO FIND THE FREQUENCY RESPONSE

MAT Lab 7.5 & CCStudio3.1

MAT Lab program for frequency response of a given system

Dept. of Electronics & Communication Engineering

Graph for Imaginary Part

Result: Frequency response of a given system is plotted.

PROGRAM TO FIND THE DISCRETE FOURIER TRANSFORM OF

MAT Lab 7.5 & CCStudio3.1

MAT Lab program for DFT of a given sequence

title('DFT absolute value:');

INPUT & OUTPUT:

RESULT: DFT of a given sequence is determined.

PROGRAM TO IMPLEMENT FAST FOURIER TRANSFORM

MAT Lab 7.5 & CCStudio3.1

MAT Lab program for FFT of a given sequence

Input Sequence Graph

Output Sequence Graph

RESULT: FFT of a given sequence is determined.

PROGRAM TO IMPLEMENT LOW PASS IIR FILTER

MAT Lab 7.5 & CCStudio3.1

xlabel('(a) Normalized freq. -->');

INPUT & OUTPUT:

Magnitude Response of Low Pass IIR Filter

Dept. of Electronics & Communication Engineering

Phase Response of Low Pass IIR Filter

RESULT: Frequency response of the Low pass IIR filter is plotted.

Dept. of Electronics & Communication Engineering

PROGRAM TO IMPLEMENT HIGH PASS IIR FILTER

MAT Lab 7.5 & CCStudio3.1

xlabel('(a) Normalized freq. -->');

INPUT & OUTPUT

Dept. of Electronics & Communication Engineering

Phase Response of High Pass IIR Filter

RESULT: Frequency response of the high pass IIR filter is plotted.

Dept. of Electronics & Communication Engineering

PROGRAM TO IMPLEMENT LOW PASS FIR FILTER

MAT Lab 7.5 & CCStudio3.1

Dept. of Electronics & Communication Engineering

Dept. of Electronics & Communication Engineering

Dept. of Electronics & Communication Engineering

INPUT & OUTPUTS

Dept. of Electronics & Communication Engineering

INPUT & OUTPUTS

PROGRAM TO IMPLEMENT HIGH PASS FIR FILTER

MAT Lab 7.5 & CCStudio3.1