III.B.TECH-I-SEM - AC LAB HHJ
III.B.TECH-I-SEM - AC LAB HHJ
ANALOG COMMUNICATION
LAB
LABORATORY MANUAL
CONTENTS
S.NO NAME OF THE EXPERIMENT
EXPERIMENT NO-1
AMPLITUDE MODULATION &DEMODULATION
AIM: To study the function of Amplitude Modulation & Demodulation (under modulation,
perfect modulation & over modulation) and also to calculate the modulation index.
APPARATUS :
1. Amplitude Modulation & De modulation trainer kit.
2. C.R.O (20MHz)
THEORY:
Modulation is defined as the process of changing the characteristics (Amplitude, Frequency or
Phase) of the carrier signal (high frequency signal) in accordance with the intensity of the
message signal (modulating signal).
Amplitude modulation is defined as a system of modulation in which the amplitude of the carrier
is varied in accordance with amplitude of the message signal (modulating signal).
The message signal is given by the expression.
Em(t) =Em cosWmt
Where Wm is -----> Angular frequency
Em --------Amplitude
Carrier voltage Ec(t)= Ec cosWct
E(t)=Ec + KaEm cosWmt
KaEm cosWmt -----change in carrier amplitude
Ka----- constant
The amplitude modulated voltage is given by
E=E(t) cosWct
From above two equations
E= ( Ec+KaEm cosWmt) cosWct.
E= (1+KaEm/Ec cosWmt) Ec cosWct
E= Ec(1+Ma cosWmt)cosWct
Where Ma-----depth of modulation/ modulation index/modulation factor
Ma=KaEm/Ec
100* Ma gives the percentage of modulation
BLOCK DIAGRAM:
Modulation
Demodulation
CIRCUIT DIAGRAM:
AM DEMODULATION
PROGRAM:
% program for AM modulation and demodulation
close all
clear all
clc
fs=8000;
fm=20;
fc=500;
Am=1;
Ac=1;
t=[0:0.1*fs]/fs;
m=Am*cos(2*pi*fm*t);
c=Ac*cos(2*pi*fc*t);
ka=0.5;
u=ka*Am;
s1=Ac*(1+u*cos(2*pi*fm*t)).*cos(2*pi*fc*t);
subplot(4,3,1:3);
plot(t,m);
title('Modulating or Message signal(fm=20Hz)');
subplot(4,3,4:6);
plot(t,c);
title('Carrier signal(fc=500Hz)');
subplot(4,3,7);
plot(t,s1);
title('Under Modulated signal(ka.Am=0.5)');
Am=2;
ka=0.5;
u=ka*Am;
s2=Ac*(1+u*cos(2*pi*fm*t)).*cos(2*pi*fc*t);
subplot(4,3,8);
plot(t,s2);
title('Exact Modulated signal(ka.Am=1)');
Am=5;
ka=0.5;
u=ka*Am;
s3=Ac*(1+u*cos(2*pi*fm*t)).*cos(2*pi*fc*t);
subplot(4,3,9);
plot(t,s3);
title('Over Modulated signal(ka.Am=2.5)');
r1= s1.*c;
[b a] = butter(1,0.01);
mr1= filter(b,a,r1);
subplot(4,3,10);
plot(t,mr1);
title(' deModulated signal for(ka.Am=0.5)');
r2= s2.*c;
[b a] = butter(1,0.01);
mr2= filter(b,a,r2);
subplot(4,3,11);
plot(t,mr2);
title(' deModulated signal for(ka.Am=1)');
r3= s3.*c;
[b a] = butter(1,0.01);
mr3= filter(b,a,r3);
subplot(4,3,12);
plot(t,mr3);
title(' deModulated signal for(ka.Am=2.5)');
PROCEDURE:-
1. Connect the AC Adapter to the mains and the other side to the Experimental Trainer. Switch
„ON‟ the power.
2. Observe the carrier and modulating waveforms and note their frequencies. (Carrier frequency
is around 100 KHz and amplitude is variable from 0 -8Vp-p, modulating signal is 1KHz).
5. Connect Carrier I/P to ground and apply a 2V peak to peak AF Signal input to (modulating
I/P) and adjust P1 in anti-clock wise position to get minimum A.C output.
6. Connect modulating I/P to ground and apply a 3V peak to peak carrier signal to carrier I/P and
adjust P2 in clock wise direction to get minimum A.C ouyput..
7. Connect modulating input &carrier input to ground and adjust P3 for zero D.C output.
8. Make modulating i/p 2 Vpp and carrier i/p 3 Vpp peak to peak and adjust potentiometer P4 for
maximum output.
9. Calculate maximum and minimum points on the modulated envelope on a CRO and calculate
the depth of modulation.
10. Observe that by varying the modulating voltage, the depth of modulation varies.
11. During demodulation connect this AM output to the input of the demodulator.
12. By adjusting the RC time constant (i.e., cut off frequency) of the filter circuit we get
minimum distorted output.
13. Observe that this demodulated output is amplified has some phase delay because of RC
components.
EXPECTED WAVEFORMS:-
OBSERVATIONS:
Modulation
RESULT:
QUESTIONS
1. Define AM and draw its spectrum?
2. Draw the phase representation of an amplitude modulated wave?
3. Give the significance of modulation index?
4. What are the different degrees of modulation?
5. What are the limitations of square law modulator?
6. Compare linear and nonlinear modulators?
7. Compare base modulation and emitter modulation?
EXPERIMENT NO-2
DSB-SC MODULATOR & DETECTOR
AIM: To study the working of the Balanced Modulator and demodulator.
APPARATUS:
1. Balanced modulator trainer kit
2. C.R.O (20MHz)
3. Connecting cards and probes
4. Function generator (1MHz)
THEORY:
Balanced modulator circuit is used to generate only the two side bands DSB-SC. The balanced
modulation system is a system is a system of adding message to carrier wave frequency there by
only the side bands are produced. It consists of two AM modulators arranged in a balanced
configuration. The AM modulator is assumed to be identical. The carrier input to the two
modulators is same.
If we eliminate or suppress the carrier then the system becomes suppressed carrier
DSB-SC. In this we need reinsert the carrier is complicated and costly. Hence the suppressed
carrier DSB system may be used in point to point communication system.
Generation of suppressed carrier amplitude modulated volt balanced modulator may be of the
following types.
1. Using transistors or FET.
2. Using Diodes
BLOCK DIAGRAM: Modulation
Demodulation
CIRCUIT DIAGRAM:
DSB-SC MODULATOR
PROGRAM:
% program for dsbsc modulation and demodulation
close all
clear all
clc
t =0:0.000001:.001;
Vm= 1;
Vc= 1;
fm = 2000;
fc= 50000;
m_t = Vm*sin(2*pi*fm*t);
subplot(4,1,1);
plot(t,m_t);
c_t = Vc*sin(2*pi*fc*t);
subplot(4,1,2);
plot(t,c_t);
subplot(4,1,3);
s_t = m_t.*c_t;
hold on;
plot(t,s_t);
plot(t,m_t,'r:');
plot(t,-m_t,'r:');
hold off;
r = s_t.*c_t;
[b a] = butter(1,0.01);
mr= filter(b,a,r);
subplot(4,1,4);
plot(t,mr);
PROCEDURE:-
1. Connect the circuit as per the given circuit diagram.
3. Apply a 100KHz, 0.1 peak sinusoidal to the carrier input and a 5KHz, 0.1 peak sinusoidal to
the modulation input.
4. Measure the output signal frequency and amplitude by connecting the output to CRO.
EXPECTED WAVEFORMS:-
OBSERVATIONS:
RESULT:
QUESTIONS
1. What are the two ways of generating DSB_SC?
2. What are the applications of balanced modulator?
3. What are the advantages of suppressing the carrier?
4. What are the advantages of balanced modulator?
5. What are the advantages of Ring modulator?
6. Write the expression for the output voltage of a balanced modulator?
7. Explain the working of balanced modulator and Ring Modulator using diodes
EXPERIMENT.NO-3
SSB-SC MODULATOR & DETECTOR
(PHASE SHIFT METHOD)
AIM:- To generate SSB using phase method and detection of SSB signal using Synchronous
detector.
APPARATUS:-
1. SSB trainer kit
2. C.R.O (20MHz)
3. Patch cards
4. CRO probes
THEORY:
AM and DSBSC modulation are wasteful of band width because they both require a
transmission bandwidth which is equal to twice the message bandwidth In SSB only one side
band and the carrier is used. The other side band is suppressed at the transmitter, but no
information is lost. Thus the communication channel needs to provide the same band width,
when only one side band is transmitted. So the modulation system is referred to as SSB system.
The base band signal may not be recovered from a SSB signal by the Use of a diode
modulator. The bae band signal can be recovered if the spectral component of the output i.e
either the LSB or USB is multiplied by the carrier signal.
Consider the modulating signal
M(t)=Am cos Wmt
C(t)=Ac cosWct
M(t)c(t)= AcAm cosWmt cosWct
The above signal when passed through a filter, only one of the above component is obtained
which lays the SSB signal.
BLOCK DIAGRAM: -
SSB MODULATION
CIRCUIT DIAGRAM:
PROGRAM:-
% program for ssb modulation and demodulation
close all
clear all
clc
fs=8000;
fm=20;
fc=50;
Am=1;
Ac=1;
t=[0:0.1*fs]/fs;
subplot(5,1,1);
m1=Am*cos(2*pi*fm*t);
plot(t,m1);
title('Message Signal');
m2=Am*sin(2*pi*fm*t);
subplot(5,1,2)
c1=Ac*cos(2*pi*fc*t);
plot(t,c1)
title('Carrier Signal');
c2=Ac*sin(2*pi*fc*t);
subplot(5,1,3)
% Susb=0.5* Am*cos(2*pi*fm*t).* Ac*cos(2*pi*fc*t) -- 0.5* Am*sin(2*pi*fm*t).*
Ac*sin(2*pi*fc*t);
Susb=0.5*m1.*c1-0.5*m2.*c2;
plot(t,Susb);
title('SSB-SC Signal with USB');
subplot(5,1,4);
Slsb=0.5*m1.*c1+0.5*m2.*c2;
plot(t,Slsb);
title('SSB-SC Signal with LSB');
r = Susb.*c1;
subplot(5,1,5);
[b a] = butter(1,0.0001);
mr= filter(b,a,r);
plot(t,mr);
title('demodulated output');
PROCEDURE:-
SSB MODULATION
1. Connect the Adaptor to the mains and the other side to the Experimental Trainer Switch „ON‟
the power.
2. (a) Connect carrier fc 900 to Ain of Balanced Modulator –A and adjust its amplitude to
0.1Vpp.
(b). Connect modulating signal fm 00 5Vpp to Bin of the Balanced Modulator-A.
3. Observe the DSB-A output on CRO.
4. Connect fc 00 at 0.1 Vpp at Cin of Balanced Modulator B. Connect fm 900 at 5 Vpp at Din of
Balanced Modulator B.
5. Connect the DSB-A output and DSB-B output to the summing amplifier. Observe the output
(SSB output) on the spectrum analyzer. This gives single side band (upper) only while the lower
side band is cancelled in the summing Amplifier.
SSB DEMODULATION
1. Connect the carrier fc 00 and SSB output to the synchronous detector.
2. Connect the demodulator output on the oscilloscope which is the recovered modulating signal.
OBSERVATIONS:
RESULT:
QUESTIONS
1. What are the different methods to generate SSB-SC signal?
2. What is the advantage of SSB-SC over DSB-SC?
3. Explain Phase Shift method for SSB generation.
4. Why SSB is not used for broadcasting?
5. Give the circuit for synchronous detector?
6. What are the uses of synchronous or coherent detector?
7. Give the block diagram of synchronous detector?
8. Why the name synchronous detector?
EXPERMENT NO-4
FREQUENCY MODULATION AND DEMODULATION
AIM: To study the process of frequency modulation and demodulation and calculate the depth
of modulation by varying the modulating voltage.
APPARATUS :
1. FM modulation and demodulation kit
3. CRO probes
4. Patch cards.
THEORY:
The modulation system in which the modulator output is of constant amplitude, in which the
signal information is super imposed on the carrier through variations of the carrier frequency.
The frequency modulation is a non-linear modulation process. Each spectral component of the
base band signal gives rise to one or two spectral components in the modulated signal. These
components are separated from the carrier by a frequency difference equal to the frequency of
base band component. Most importantly the nature of the modulators is such that the spectral
components which produce decently on the carrier frquency and the base band frequencies.The
spetral components in the modulated wave form depend on the amplitude.
The modulation index for FM is defined as
Mf= max frequency deviation/ modulating frequency.
BLOCK DIAGRAM:
Modulation
Demodulation
CIRCUIT DIAGRAM:
FM MODULATION FM DEMODULATOR
PROGRAM:-
% program for fm modulation and demodulation
close all
clear all
clc
%fm=35HZ,fc=500HZ,Am=1V,Ac=1V,B=10
fs=10000;
Ac=1;
Am=1;
fm=35;
fc=500;
B=10;
t=(0:.1*fs)/fs;
wc=2*pi*fc;
wm=2*pi*fm;
m_t=Am*cos(wm*t);
subplot(5,1,1);
plot(t,m_t);
title('Modulating or Message signal(fm=35Hz)');
c_t=Ac*cos(wc*t);
subplot(5,1,2);
plot(t,c_t);
title('Carrier signal(fm=500Hz)');
s_t=Ac*cos((wc*t)+B*sin(wm*t));
subplot(5,1,3);
plot(t,s_t);
title('Modulated signal');
d=demod(s_t,fc,fs,'fm');
subplot(5,1,4);
plot(t,d);
title('demodulated signal');
PROCEDURE:
1. Switch on the experimental board.
2. Observe the FM modulator output without any modulator input which is the carrier signal and
note down its frequency and amplitude.
3. Connect modulating signal to FM modulator input and observe modulating signal and FM
output on two channels of the CRO simultaneously.
4. Adjust the amplitude of the modulating signal until we get less distorted FM output.
5. Apply the FM output to FM demodulator and adjust the potentiometer in demodulation until
we get demodulated output.
OBSERVATIONS:
Modulation
EXPECTED WAVEFORMS:-
FM modulated output:
RESULT:
QUESTIONS
1. Define FM & PM.
EXPERIMENT NO-5
STUDY OF SPECTRUM ANALYZER AND ANALYSIS OF AM AND FM
SIGNALS
AIM: To verify the spectrum of AM and FM signals using spectrum analyzer.
APPARATUS / SOFTWARE REQUIRED:
1. Pc with windows(95/98/XP/NT/2000)
PROGRAM:
%program of spectrum analyzer and analysis of am and fm signals
close all
clear all
clc
Fs = 100; %sampling frq
t = [0:2*Fs+1]'/Fs;
Fc = 10; % Carrier frequency
x = sin(2*pi*2*t); % message signal
Ac=1;
% compute spectra of am
xam=ammod(x,Fc,Fs,0,Ac);
zam = fft(xam);
zam = abs(zam(1:length(zam)/2+1));
frqam = [0:length(zam)-1]*Fs/length(zam)/2;
% compute spectra of dsbsc
ydouble = ammod(x,Fc,Fs, 3.14,0);
zdouble = fft(ydouble);
zdouble = abs(zdouble(1:length(zdouble)/2+1));
frqdouble = [0:length(zdouble)-1]*Fs/length(zdouble)/2;
% compute spectra of ssb
ysingle = ssbmod(x,Fc,Fs,0,'upper');
RESULT:
EXPERIMENT.NO-6
PRE-EMPHASIS & DE-EMPHASIS
AIM: To study the frequency response of Pre-Emphasis and De-Emphasis circuits.
APPARATUS:
1. Pre-emphasis & De-emphasis trainer kits.
2. C.R.O (20 MHz )
3. Function generator (1MHz).
4. Patch cards and Probes.
THEORY:
Frequency modulation is much immune to noise than amplitude modulation and significantly
more immune than phase modulation. A single noise frequency will affect the output of the
receiver only if it falls with in its pass band.
The noise has a greater effect on the higher modulating frequencies than on lower ones. Thus, if
the higher frequencies were artificially boosted at the transmitter and correspondingly cut at the
receiver, improvement in noise immunity could be expected. This booting of the higher
frequencies, in accordance with a pre-arranged curve, is termed pre-emphasis, and the
compensation at the receiver is called de-emphasis.
If the two modulating signals have the same initial amplitude, and one of them is pre-emphasized
to (say) twice this amplitude, whereas the other is unaffected (being at a much lower frequency)
then the receiver will naturally have to de-emphasize the first signal by a factor of 2, to ensure
that both signals have the same amplitude in the output of the receiver. Before demodulation, I.e.
while susceptible to noise interference the emphasized signal had twice the deviation it would
have had without pre-emphasis, and was thus more immune to noise. Alternatively, it is seen that
when this signal is de-emphasized any noise sideband voltages are de-emphasized with it, and
therefore have a correspondingly lower amplitude than they would have had without emphasis
again their effect on the output is reduced. Apart from that, it would be difficult to introduce pre-
emphasis and de-emphasis in existing AM services since extensive modifications would be
needed, particularly in view of the huge numbers is receivers in use.
CIRCUIT DIAGRAM:
PRE EMPHASIS
DE EMPHASIS
PROGRAM:-
% program for Pre-Emphasis and De-Emphasis
close all
clear all
clc
num_samples = 2^13;
fs=5000;
Ts=1/fs;
fm1=20;
fm2=30;
fc=200;
t=(0:num_samples-1)*Ts;
f=(-num_samples/2:num_samples/2-1)*fs/num_samples;
mt=sin(2*pi*fm1*t);
Mf=fftshift(abs(fft(mt)));
f_cutoff_pe=15;
Wn_pe=f_cutoff_pe/(fs/2);
[b_pe,a_pe]=butter(1,Wn_pe);
[H_pe,W]=freqz(a_pe,b_pe);
a_de=b_pe;
b_de=a_pe;
[H_de,W]=freqz(a_de,b_de);
mt_pe=filter(a_pe,b_pe,mt);
Mf_pe=fftshift(abs(fft(mt_pe)));
figure(1)
subplot(211);plot(t,mt)
axis([0 .6 min(mt)-1 max(mt)+1])
grid on;title('Modulating Signal (Time Domain)')
subplot(212);plot(f,Mf)
grid on;axis([-50 50 0 max(Mf)+100])
PROCEDURE:
I-PRE-EMPHASIS
1. Connect the circuit as per the circuit diagram
3. By varying the input frequency with fixed amplitude, note down the output amplitude (Vo)
with respect to the input frequency.
EXPECTED WAVEFORMS
TABLE:-
Pre-emphasis
De-emphasis
RESULT :
QUESTIONS
1. What is the need for pre-emphasis?
2. Explain the operation of pre-emphasis circuit?
3. Pre emphasis operation is similar to high pass filter explain how?
4. De emphasis operation is similar to low pass filter justify?
5. What is de-emphasis?
6. Draw the frequency response of a pre-emphasis circuit?
7. Draw the frequency response of a de-emphasis circuit?
8. Give the formula for the cutoff frequency of the pre-emphasis circuit?
9. What is the significance of the 3db down frequency
EXPERIMENT NO-7
TIME DIVISION MULTIPLEXING & DEMULTIPLEXING
AIM:
1. Study of 4 Channel Analog Multiplexing and De multiplexing Techniques.
2. Study of the effect of sampling frequency variation on the output.
3. Study of input signal amplitude on the output.
APPARATUS:
1. TIME DIVISION MULTIPLEXING & DEMULTIPLEXING Trainer.
2. C.R.O (30 MHz)
3. Patch cards.
THEORY:-
The TDM is used for transmitting several analog message signals over a communication channel
by dividing the time frame into slots, one slot for each message signal. The four input signals, all
band limited by the input filters are sequentially sampled, the output of which is a PAM
waveform containing samples of the input signals periodically interlaced in time. The samples
from adjacent input message channels are separated by Ts/M, where M is the number of input
channels. A set of M pulses consisting of one sample from each of the input M-input channels is
called a frame.
At the receiver the samples from individual channels are separated by carefully synchronizing
and are critical part TDM. The samples from each channel are filtered to reproduce the original
message signal. There are two levels of synchronization. Frame synchronization is necessary to
establish when each group of samples begin and word synchronization is necessary to properly
separate the samples within each frame.
Besides the space diversity & frequency diversity there is a method of sending multiple analog
signals on a channel using “TIME DIVISION MULTIPLEXING & DEMULTIPLEXING”
Technique.
CIRCUIT DESCRIPTION:-
Function Generator Circuit:-
A 4.096 MHz clock is used to derive the modulating signal, which is generated by an oscillator
circuit comprising a 4.096MHz crystal and three 74HC04(U2) inverter gates. this 4.096 MHz
clock is then divided down in frequency by a factor of 4096, by binary counter 74HC4040(U3),
to produce 50% duty cycle, 64 KHz square wave on pin no.1 of U4, and 32KHz square wave on
pin no.4. 32 KHz square wave is given to pin no.2 of IC NE555(U7) which act as a monostable
multivibrator. Potentiometer P5 is used to adjust the pulse width. 64KHz square wave is fed to
the four bit binary counter on pin no.1 to produce 4KHz square wave at pin no.6. this goes to pin
no.13. this signal clocks to the second half of the counter to produce square wave at following
frequencies.
Counter output Frequency
2QD 250Hz
2QC 500Hz
2QB 1 KHz
2QA 2 KHz
Each of these square wave outputs is then fed to its own low pass filter circuits TL072 (U8, U9).
Which generates corresponding sine wave outputs? The amplitude of this sine wave can be
varied by potentiometers P1, P2, P3, P4 respectively. These sine wave outputs are available at
TP1, TP2, TP3, and TP4 respectively and have amplitudes up to 10V max.
Transmitter Block:-
Each modulating signal is applied to IC TLO74(U6)(pin nos. 3,5,10,12 respectively). This IC
buffers the applied signal and is fed to pin nos.3,14,11,6 respectively of IC DG211(U5). The
pulse input(32KHz(clock)) is applied to pin nos. 1,16,9,8 of U5. Corresponding PAM outputs are
available at test points Tp5,Tp6,Tp7,Tp8 respectively. These each PAM outputs are applied to IC
4052(U1). Which will act as multiplexer.
4 Channel Demultiplexer:-
The multiplexed PAM signal is given to the 4 channel Demultiplexer input at pin 3(TP12). The
A& B timing wave forms selects the channel and accordingly connects the same to the output.
This at the PAM signal of each channel are separated these separated demultiplexed outputs are
monitored at test points 13,14,15,16 respectively.
Low Pass Filter:-
Each separated PAM outputs are being connected to corresponding channel‟s butter worth Low
Pass Filter. This is 4th order filter having roll of rate of 24db/octave(40db/decade) and cut-off
frequency of 250Hz,500Hz,1KHz,2KHz respectively. The output of these filters goes to
CIRCUIT DIAGRAM:
PROGRAM:-
%program for time division multiplexing and demultiplexing
clc;
close all;
clear all;
% Signal generation
x=0:.5:4*pi; % siganal taken upto 4pi
sig1=8*sin(x); % generate 1st sinusoidal signal
l=length(sig1);
sig2=8*triang(l); % Generate 2nd traingular Sigal
% Display of Both Signal
subplot(2,2,1);
plot(sig1);
title('Sinusoidal Signal');ylabel('Amplitude--->');xlabel('Time--->'); subplot(2,2,2);
plot(sig2);
title('Triangular Signal');ylabel('Amplitude--->');xlabel('Time--->');
% Display of Both Sampled Signal
subplot(2,2,3);
stem(sig1);
title('Sampled Sinusoidal Signal');
ylabel('Amplitude--->');xlabel('Time--->');
subplot(2,2,4);
stem(sig2);
title('Sampled Triangular Signal');
ylabel('Amplitude--->');xlabel('Time--->');
l1=length(sig1);
l2=length(sig2);
for i=1:l1
sig(1,i)=sig1(i); % Making Both row vector to a matrix
sig(2,i)=sig2(i);
end
PROCEDURE:-
Multiplexing:-
1. Connect the circuit as shown in diagram.
4. Monitor the outputs at test points 5,6,7,8. these are natural sampling PAM outputs.
5. Observe the outputs varying the duty cycle pot(P5). The PAM outputs will varying with 10%
to 50% duty cycle.
6. Try varying the amplitude of modulating signal corresponding each channel by using
amplitude pots P1,P2,P3,P4. Observe the effect on all outputs.
7. Observe the TDM output at pin no.13 (at TP9) OF 4052. all the multiplexer channel are
observed during the full period of the clock(1/32 KHz).
3. Observe by varying the duty cycle pot P5 and see the effect on the outputs.
4. Observe the low pass filter outputs for each channel at test points 17,18,19,20 and at sockets
channels CH1,CH2,CH3,CH4. These signals are true replica of the inputs. These signals have
lower amplitude.
EXPECTED WAVEFORMS
RESULT:
QUESTIONS
1. Draw the TDM signal with 2 signals being multiplexed over the channel?
2. Define guard time & frame time?
EXPERIMENT NO-8
FREQUENCY DIVISION MULTIPLEXING
& DE MULTIPLEXING
AIM: To study the frequency division Multiplexing and De multiplexing Techniques.
APPARATUS/SOFTWARE REQUIRED:
1. Pc with windows(95/98/XP/NT/2000)
2. Matlab Software
PROGRAM:
%program for frequency division multiplexing and demultiplexing
close all
clear all
clc
Fs = 100; % sampling freq
t = [0:2*Fs+1]'/Fs;
x1 = sin(2*pi*2*t); % signal 1 signal
z1 = fft(x1);
z1=abs(z1);
x2 = sin(2*pi*10*t); % signal 2 signal
z2 = fft(x2);
z2=abs(z2);
figure;
subplot(4,1,1); plot(x1);
title('signal 1');xlabel('time');ylabel('amplitude');
subplot(4,1,2); plot(x2);
title('signal 2');xlabel('time');ylabel('amplitude');
subplot(4,1,3); plot(z1);
title('Spectrum of signal 1');xlabel('freqency');ylabel('magnitude');
subplot(4,1,4); plot(z2);
title('Spectrum of signal 2');xlabel('freqency');ylabel('magnitude');
% freqency multiplexing
z=z1+z2;
figure;
plot(z);
title('frequency multiplexed signals');
figure;
% freqency demultiplexing
f1=[ones(10,1); zeros(182,1);ones(10,1)];%applying filter for signal 1
dz1=z.*f1;
d1 = ifft(dz1);
subplot(2,1,1)
plot(t*100,d1);
f2=[zeros(10,1); ones(182,1);zeros(10,1)];% applying filter for signal 2
dz2=z.*f2;
d2 = ifft(dz2);
title('recovered signal 1');xlabel('time');ylabel('amplitude');
subplot(2,1,2)
plot(t*100,d2);
title('recovered signal 2');xlabel('time');ylabel('amplitude');
EXPECTED WAVEFORMS:
RESULT:
EXPERIMENT NO-9
VERIFICATION OF SAMPLING THEOREM
AIM:
1. To study the sampling theorem and its reconstruction.
2. To study the effect of amplitude and frequency variation of modulating signal on the output.
3. To study the effect of variation of sampling frequency on the demodulated output.
APPARATUS:
1. Sampling and reconstruction Trainer.
2. C.R.O(30Mhz)
3. Patch cards.
THEORY:
Pulse Modulation is used to transmit analog information. In this system continuous wave forms
are sampled at regular intervals. Information regarding the signal is transmitted only at the
sampling times together with synchronizing signals.
At the receiving end, the original waveforms may be reconstituted from the information
regarding the samples.
Sampling Theorem Statement:
A band limited signal of finite energy which has no frequency components higher than fm Hz, is
completely described by specifying the values of the signal at instants of time separated by ½ fm
seconds.
The sampling theorem states that, if the sampling rate in any pulse modulation system exceeds
twice the maximum signal frequency, the original signal can be reconstructed in the receiver
with minimum distortion.
Fs > 2fm is called Nyquist rate.
Where fs – sampling frequency
Fm – Modulation signal frequency.
If we reduce the sampling frequency fs less than fm, the side bands and the information signal
will overlap and we cannot recover the information signal simply by low pass filter. This
phenomenon is called fold over distortion or aliasing.There are two methods of sampling. (1)
Natural sampling (2) Flat top sampling.
Sample & Hold circuit holds the sample value until the next sample is taken.
Sample & Hold technique is used to maintain reasonable pulse energy.The duty cycle of a signal
is defined as the ratio of Pulse duration to the Pulse repetition period. The duty cycle of 50% is
desirous taking the efficiency into account.
Circuit Description:-
Pulse and Modulating Signal Generator:-
A 4.096 MHz clock is used to derive the modulating signal, which is generated by an oscillator
circuit comprising a 4.096MHz crystal and three 74HC04(U9) inverter gates. This 4.096MHz
clock is then divided down in frequency by a factor of 4096, by binary counter 74HC4040(U10),
to produce 50% duty cycle, 1KHz square wave on pin no.1 of U10, and 2KHz square wave on
pin no.15. the frequency is selectable by means of SW1. this input of fourth order low pass filter
U11(TL072) is used to produce sine wave from the square wave. The amplitude of this sine
wave can be varied.
The square wave which is generated by the oscillator is buffered by inverter 74HC04(U9), to
produce 32KHz square wave at pin no. 4 of the 74HC4040. This pulse is given to the monostable
multi(U4) to obtain the 16KHz and 32KHz square wave at the output which are selected by the
frequency pot.
Sampling Circuit:-
The IC DG211(U3) is used as analog switch which is used in pulse amplitude modulation in this
circuit. The modulation signal & pulse signal are given as the input to TL074(U2), 7400(U1)
IC‟s respectively. These IC output are fed to the inputs of the DG211.
The sampled output is available at the pin no.2 of DG211 and it is buffered by using TL074(U2)
and then output is available at TP5.
Similarly the sample & hold output and the flat top output are available at pin no15 & 10 of
DG211 respectively. These are buffered by TL074(U2) and then output is available at TP6 &
TP7 respectively.
Reconstruction Circuit:-
The demodulation section comprises of a fourth order low pass filter and an AC amplifier. The
TL074 (U5) is used as a low pass filter and AC amplifier. The output of the modulator is given
as the input to the low pass filter.
The low pass filter output is obviously less and it is fed to the AC amplifier which comprises of a
single op amp and whose output is amplified.
CIRCUIT DIAGRAM:
PROGRAM:-
%program for verification of sampling theorem
close all;
clear all
clc
t=-10:.01:10;
T=4;
fm=1/T;
x=cos(2*pi*fm*t); % input signal
subplot(2,2,1);
plot(t,x);
xlabel('time');ylabel('x(t)');title('continous time signal');
grid;
n1=-4:1:4;
fs1=1.6*fm;
fs2=2*fm;
fs3=8*fm;
%discrete time signal with fs<2fm
x1=cos(2*pi*fm/fs1*n1);
subplot(2,2,2);
stem(n1,x1);
xlabel('time');ylabel('x(n)');
title('discrete time signal with fs<2fm');
hold on
subplot(2,2,2);
plot(n1,x1)
grid;
%discrete time signal with fs=2fm
n2=-5:1:5;
x2=cos(2*pi*fm/fs2*n2);
subplot(2,2,3);
stem(n2,x2);
xlabel('time');ylabel('x(n)');
title('discrete time signal with fs=2fm');
hold on
subplot(2,2,3);
plot(n2,x2)
%discrete time signal with fs>2fm
grid;
n3=-20:1:20;
x3=cos(2*pi*fm/fs3*n3);
subplot(2,2,4);
stem(n3,x3);
xlabel('time');ylabel('x(n)');
title('discrete time signal with fs>2fm');
hold on
subplot(2,2,4);
plot(n3,x3)
grid;
PROCEDURE:
Sampling:-
1. Connect the circuit as shown in diagram
a. The output of the modulating signal generator TP1 is connected to modulating signal input
TP4 of the sampling circuit keeping the frequency switch in 1KHz position, and amplitude knob
to max position.
b. The output of pulse generator TP2 is connected to sampling pulse input TP3 of the sampling
circuit keeping the frequency switch in 16KHz position.(Adjust the duty cycle pot to mid
position i.e.50%).
2. Switch ON the power supply.
3. Observe the outputs of sampling, sampling and hold, flat top output at TP7, TP8 and TP9
respectively. By varying the amplitude pot also observe the effect on outputs.
4. By varying Duty cycle pot observe the effect on sampling outputs (Duty cycle is varying from
10-15%).
5. Vary the switch position in the pulse generator circuit to 32 KHz and now observe the outputs
at TP7, TP8 and TP9.By varying the amplitude pot also observe the effect on outputs.
6. Now, vary the switch position in modulating signal generator to 2 KHz and repeat all the
above steps 3&4.
b. The output of pulse generator TP2 is connected to sampling pulse input TP3 of the sampling
circuit keeping the frequency switch in 16KHz position.(Adjust the duty cycle pot to mid
position i.e.50%).
c. Connect the sample output from TP7 to the input of low pass filter TP10.
d. Output of low pass filter from TP11 to input of AC amplifier TP12, keep the gain pot in AC
amplifier to max position.
2. Switch ON the power supply.
3. Observe the output of AC amplifier at TP13. The output will be the replica of the input. By
varying the gain pot observe the demodulating signal amplification.
4. Similarly connect the sample and hold output and flat top output to TP10 and observe
reconstructed the signal.
5. Vary the switch position in the sampling frequency circuit to 32KHz and now repeat the steps
3&4.
6. Vary the switch position in the modulating signal generator to 2KHz and repeat all the above
steps 3 to 5.
EXPECTED WAVEFORMS:
RESULT:
QUESTIONS
1. What are the types of sampling?
2. State sampling theorem?
3. What happens when fs < 2 fm?
4. How will be the reconstructed signal when fs >= 2fm?
5. Explain the operation of sampling circuit?
6. Explain the operation of re-construction circuit?
7. Who formalized the sampling theorem?
8. What are the applications of the above theorem?
9. Is the sampling theorem basis for the modern digital communications?
10. Is the voice signal sampling of 8000 Hz, follows sampling theorem in Land line
Telephone Exchange
EXPERIMENT NO-10
PULSE AMPLITUDE MODULATION
AIM:- 1.To study the Pulse amplitude modulation & demodulation Techniques.
2. To study the effect of amplitude and frequency variation of modulating signal on the output.
APPARATUS:-
1. Pulse amplitude modulation & demodulation Trainer.
3. Patch cards.
THEORY:-
Pulse modulation is used to transmit analog information. In this system continuous wave forms
are sampled at regular intervals. Information regarding the signal is transmitted only at the
sampling times together with syncing signals.
At the receiving end, the original waveforms may be reconstituted from the information
regarding the samples.
The pulse amplitude modulation is the simplest form of the pulse modulation. PAM is a pulse
modulation system is which the signal is sampled at regular intervals, and each sample is made
proportional to the amplitude of the signal at the instant of sampling. The pulses are then sent by
either wire or cables are used to modulated carrier.
The two types of PAM are i) Double polarity PAM, and ii) the single polarity PAM, in which a
fixed dc level is added to the signal to ensure that the pulses are always positive. Instantaneous
PAM sampling occurs if the pulses used in the modulator are infinitely short.
Natural PAM sampling occurs when finite-width pulses are used in the modulator, but the tops of
the pulses are forced to follow the modulating waveform.
Flat-topped sampling is a system quite often used because of the ease of generating the
modulated wave.
PAM signals are very rarely used for transmission purposes directly. The reason for this lies in
the fact that the modulating information is contained in the amplitude.
factor of the pulses, which can be easily distorted during transmission by noise, crosstalk, other
forms of distortion. They are used frequently as an intermediate step in other pulse-modulating
methods, especially where time-division multiplexing is used.
Circuit description:-
Pulse and Modulation Signal Generator:-
A 4.096 MHz clock is used to derive the modulating signal, which is generated by an oscillator
circuit comprising a 4.096MHz crystal and three 74HC04(U9) inverter gates. This 4.096MHz
clock is then divided down in frequency by a factor of 4096, by binary counter 74HC4040(U10),
to produce 50% duty cycle, 1 KHz square wave on pin no.1 of U10, and 2KHz square wave on
pin no.15. the frequency is selectable by means of SW1. this goes to input of fourth order low
pass filter U11(TL072) is used to produce sine wave from the square wave. The amplitude of this
sine wave can be varied.
The square wave which is generated by the oscillator is buffered by inverter 74HC04(U9), to
produce 32KHz square wave at pin no.4 of the 74HC4040(U10). This pulse is given to the
monostable multi to obtain the 16 KHz and 32 KHz square wave at the output which are selected
by the frequency pot.
Modulation:-
The ICDG211 (U3) is used as a pulse amplitude modulation in this circuit. The modulation
signal & pulse signals are given to TL074 (U2) & 7400(U1) IC‟s respectively. These outputs are
fed to the inputs the D4211 (U3).
The sampled output is available at the pin no 2 of DG211 and it is buffered by using TL074 (U2)
and then output is available at TP5.
Similarly the sample & hold output and the flat top output are available at pin no.15 &10 of
DG211 respectively. These are buffered by TL074 (U2) and then output is available at TP6&TP7
respectively.
Demodulation:-
The demodulation section comprises of fourth order low pass filter and an AC amplifier. The
TL074(U5) is used as a low pass filter and AC amplifier. The output of the modulator is given as
the input to the low pass filter.
The low pass filter output is obviously less and it is fed to the AC amplifier which comprises of a
single op amp and whose output is amplified.
CIRCUIT DIAGRAM:
PROGRAM:-
% pulse amplitude modulation
close all
clear all
clc
t = 0 : 1/1e3 : 1; % 1 kHz sample freq for 1 sec
d = 0 : 1/5 : 1;
x = sin(2*pi/4*2*t); %message signal
figure;
subplot(3,1,1)
plot(x);
title('message');
xlabel('time');ylabel('amplitude');
PROCEDURE:
Double Polarity:-
Modulation:-
1. Connect the circuit as shown in diagram 1.
a. The output of the modulating signal generator is connected to the modulating signal input TP2
keeping the frequency switch in 1KHz position, and amplitude knob to max position
b. 16KHz pulse output to pulse input TP1.(Keep the frequency in minimum position in pulse
generator block).
2. Switch ON the power supply.
3. Monitor the outputs at TP5, TP6& TP7. And observe the outputs also by varying amplitude
pot (Which is in modulation signal generator block).
4. Now vary the frequency selection which position in modulating signal generator block to 2
KHz, amplitude pot to max position.
5. Observe the output at TP5, TP6& TP7 and observe the outputs also by varying amplitude pot
(Which is in modulation signal generator block).
6. Repeat all the above steps for the pulse frequency 32KHz ( By varying the frequency pot in
the pulse generator block).
a. The output of the modulating signal generator is connected to the modulating signal input TP2
keeping the frequency switch in 1KHz position, and amplitude knob to max position
b. 16KHz pulse output to pulse input TP1 .
9. Switch ON the power supply.
10. Repeat above step 3 to 6 and observe the outputs.
11. Vary DC output pot until you get single polarity PAM at TP5,TP6,TP7.
12. Switch OFF the power supply.
Demodulation:-
1. Connect the circuit as shown in diagram 3.
a. The output of the modulating signal generator is connected to the modulating signal input TP2
keeping the frequency switch in 1KHz position, and amplitude knob to max position
b. 16KHz pulse output to pulse input TP1.
c. Sample output, sample and hold output and flat top outputs
Respectively to the input of low pass filter(TP9) and LPF
EXPECTED WAVEFORMS
RESULT:
QUESTIONS
1. TDM is possible for sampled signals. What kind of multiplexing can be used in
continuous modulation systems?
2. What is the minimum rate at which a speech signal can be sampled for the purpose of PAM?
3. What is cross talk in the context of time division multiplexing?
4. Which is better, natural sampling or flat topped sampling and why?
5. Why a dc offset has been added to the modulating signal in this board? Was it essential for the
working of the modulator? Explain?
6. If the emitter follower in the modulator section saturates for some level of input signal, then
what effect it will have on the output?
7. Derive the mathematical expression for frequency spectrum of PAM signal.
EXPERIMENT NO-11
PULSE WIDTH MODULATION & DEMODULATION
AIM:
1. To study the Pulse Width Modulation (PWM) and Demodulation Techniques.
2. To study the effect of Amplitude and Frequency of Modulating Signal on PWM output.
APPARATUS:
1. PWM trainer kit
2. C.R.O(30MHz)
3. Patch Cards.
THEORY:-
Pulse modulation is used to transmit analog information. In this system continuous wave forms
are sampled at regular intervals. Information regarding the signal is transmitted only at the
sampling times together with synchronizing signals.
At the receiving end, the original waveforms may be reconstituted from the information
regarding the samples.
The pulse Width Modulation of the PTM is also called as the Pulse Duration Modulation (PDM)
& less often Pulse length Modulation (PLM).
In pulse Width Modulation method, we have fixed and starting time of each pulse, but the width
of each pulse is made proportional to the amplitude of the signal at that instant.
This method converts amplitude varying message signal into a square wave with constant
amplitude and frequency, but which changes duty cycle to correspond to the strength of the
message signal.
Pulse-Width modulation has the disadvantage, that its pulses are of varying width and therefore
of varying power content. This means that the transmitter must be powerful enough to handle the
maximum-width pulses. But PWM still works if synchronization between transmitter and
receiver fails, whereas pulse-position modulation does not.
Pulse-Width modulation may be generated by applying trigger pulses to control the starting time
of pulses from a mono stable multivibrator, and feeding in the signal to be sampled to control the
duration of these pulses.
When the PWM signals arrive at its destination, the recovery circuit used to decode the original
signal is a sample integrator (LPF).
CIRCUIT DESCRIPTION:-
Pulse & Modulating Signal Generator:-
A 4.096MHz clock is used to derive the modulating signal, which is generated by an oscillator
circuit comprising a 4.096MHz crystal and three 74HC04(U9) inverter gates. This 4.096MHz
clock is then divided down in frequency by a factor of 4096, by binary counter 74HC4040(U2),
to produce 50% duty cycle, 1KHz square wave on pin no.1 of U4, and 2KHz square wave on pin
no.15. the frequency is selectable by means of SW1. This goes to input of fourth order low pass
filter U3 is used to produce sine wave from the square wave. The amplitude of this sine wave can
be varied.
The square wave which is generated by the oscillator is buffered by inverter 74HC04, to produce
32KHz square wave at pin no.4 of the 74HC4040(U2). This pulse is given to the monostable
multi to obtain the 16KHz and 32KHz square wave at the output which are selected by the
frequency pot.
Modulation:-
The PWM circuit uses the 555 IC(U1) in monostable mode. The Modulating signal input is
applied to pin no.5 of 555IC, and there Pulse input is applied to pin no.2.
The output of PWM is taken at the pin no.3 of 555IC i.e., TP3.
Demodulation:-
The demodulation section comprises of a fourth order low pass filter and an AC amplifier. The
TL074(U5) is used as a low pass filter and an AC amplifier. The output of the modulator is given
as the input to the low pass filter.
The low pass filter output is obviously less and it is feed to the AC amplifier which comprises of
a single op amp and whose output is amplified.
CIRCUIT DIAGRAM:
CLOCK GENERATOR
PROGRAM:-
% pulse width modulation & demodulation
close all
clear all
clc
fc=1000;
fs=10000;
f1=200;
t=0:1/fs:((2/f1)-(1/fs));
x1=0.4*cos(2*pi*f1*t)+0.5;
%modulation
y1=modulate(x1,fc,fs,'pwm');
subplot(311);
plot(x1);
axis([0 50 0 1]);
title('original signal taken mesage,f1=500,fs=10000')
subplot(312);
plot(y1);
axis([0 500 -0.2 1.2]);
title('PWM')
%demodulation
x1_recov=demod(y1,fc,fs,'pwm');
subplot(313);
plot(x1_recov);
title('time domain recovered, single tone,f1=200')
axis([0 50 0 1]);
PROCEDURE:
Modulation:-
1. Connect the circuit as shown in the diagram 1.
a. The output of the modulating signal generator is connected to the modulating signal input TP2
keeping the frequency switch in 1KHz position, and amplitude knob to max position
b. 16KHz pulse output (by varying the frequency pot (put it min position) in pulse generator
block) from pulse generator to pulse input(TP1).
2. Switch ON the power supply.
3. Observe the output of pulse width modulation block at TP3.(By varying the amplitude pot).
4. Vary the modulating signal generator frequency by switching the frequency selector switch to
2 KHz.
5. Now, again observe the PWM output at TP3.(By varying the amplitude pot).
6. Repeat the above steps (3 to 5) for the pulse frequency of 32KHz(by varying the frequency
pot(put it in max position) in pulse generator block).
7. Switch OFF the power supply.
Demodulation:-
EXPECTED WAVEFORMS
RESULT:
QUESTIONS
1. An audio signal consists of frequencies in the range of 100Hz to 5.5KHz.What is the
minimum frequency at which it should be sampled in order to transmit it through pulse
modulation?
2. Draw a TDM signal which is handling three different signals using PWM?
EXPERIMENT NO-12
PULSE POSITION MODULATION AND DEMODULATION
AIM:
1. To study the generation Pulse Position Modulation (PPM) and Demodulation.
2. To study the effect of Amplitude and the frequency of modulating signal on its output and
observe the wave forms.
APPARATUS:
1. Pulse Position Modulation (PPM) and demodulation Trainer.
2. C.R.O(30MHz)
3. Patch cards.
THEORY:-
Pulse Modulation is used to transmit analog information in this system continuous wave forms
are sampled at regular intervals. Information regarding the signal is transmitted only at the
sampling times together with synchronizing signals.
At the receiving end, the original waveforms may be reconstituted from the information
regarding the samples. Pulse modulation may be subdivided in to two types analog and digital. In
analog the indication of sample amplitude is the nearest variable. In digital the information is a
code.
The pulse position modulation is one of the methods of the pulse time modulation.PPM is
generated by changing the position of a fixed time slot.
The amplitude& width of the pulses is kept constant, while the position of each pulse, in relation
to the position of the recurrent reference pulse is valid by each instances sampled value of the
modulating wave. Pulse position modulation into the category of analog communication. Pulse-
Position modulation has the advantage of requiring constant transmitter power output, but the
disadvantage of depending on transmitter receiver synchronization.
Pulse-position modulation may be obtained very simply from PWM. However, in
PWM the locations of the leading edges are fixed, whereas those of the trailing edges are not.
Their position depends on pulse width, which is determined by the signal amplitude at that
DEPT.OF ECE Page 73
JMTK ACLAB
instant. Thus, it may be said that the trailing edges of PWM pulses are, in fact, position-
modulated. This has positive-going narrow pulses corresponding to leading edges and negative-
going pulses corresponding to trailing edges. If the position corresponding to the trailing edge of
an un modulated pulse is counted as zero displacement, then the other trailing edges will arrive
earlier or later. They will therefore have a time displacement other than zero; this time
displacement is proportional to the instantaneous value of the signal voltage. The differentiated
pulses corresponding to the leading edges are removed with a diode clipper or rectifier, and the
remaining pulses, is position-modulated.
Circuit Description:-
Modulating Signal Generator:-
A 4.096 MHz clock is used to derive the modulating signal, which is generated by an oscillator
circuit comparing a 4.096MHz crystal and three 74HC04(U9) inverter gates. This 4.096 MHz
clock is then divided down in frequency by a factor of 4096, by binary counter 74HC4040(U4),
to produce 50% duty cycle, 1 KHz square wave on pin no.1 of U4, and 2 KHz square wave on
pin no.15. The frequency is selectable by means of SW1. This goes to input of fourth order low
pass filter U3 (TL072) is used to produce sine wave from the square wave. The amplitude of this
sine wave can be varied.
Modulation:-
The circuit uses the IC 555(U1) a Mono stable Multivibrator to perform the pulse position
Modulation action.
The Modulating signal is given to Pin No. 5 at Pin No.2 the pulse is 32 KHz which is connected
internally.
The PWM is available at TP2; this PWM output is differentiated by using differentiated circuit.
This differentiated output is available at TP8. This differentiated output is fed to the 555 IC (U2)
(Mono stable Mode) Pin No.2. The PPM output is available at TP3.
CIRCUIT DIAGRAM:
PROGRAM:-
% pulse position modulation
close all
clear all
clc
fc=100;
fs=1000;
f1=80;
t=0:1/fs:((2/f1)-(1/fs));
x1=0.4*cos(2*pi*f1*t)+0.5;
%modulation
y1=modulate(x1,fc,fs,'ppm');
subplot(311);
plot(x1);
axis([0 15 0 1]);
title('original signal taken mesage,f1=80,fs=1000')
subplot(312);
plot(y1);
axis([0 250 -0.2 1.2]);
title('PPM')
%demodulation
x1_recov=demod(y1,fc,fs,'ppm');
subplot(313);
plot(x1_recov);
title('time domain recovered, single tone,f1=80')
axis([0 15 0 1]);
PROCEDURE:
Modulation:
1. Connect the circuit as shown in diagram 1.
a. Connect the modulating signal generator output to modulating signal input (TP1) in PPM
block.
b. Keep the switch in 1 KHz position and amplitude pot in max position.
2. Switch ON the power supply
3. Observe the PWM output at TP2, and the differentiated output signal at TP8.
5. Try varying the amplitude and frequency of sine wave by varying amplitude pot.
6. Repeat Step 5 for frequency of 2 KHz and observe the PPM output.
Demodulation:-
8. Connect the circuit as shown in diagram2.
a. Connect the modulating signal generator output to modulating signal input (TP1) in PPM
block.
b. Keep the switch in 1 KHz position and amplitude pot in max position.
11. Thus the recovered signal is true replica of the input signal
12. a. As the output of LPF has less amplitude, connect the output of LPF to the input of an AC
amplifier (TP5 to TP6).
b. Observe the demodulated out put on the oscilloscope at TP7 and also observe the amplitude of
demodulated signal by varying gain pot. This is amplitude demodulated output.
13. Repeat the steps (7 to 9) for the modulating signal for frequency 2 KHz.
EXPECTED WAVEFORMS:
RESULT:
QUESTIONS:
1. What is the advantage of PPM over PWM?
2. Is the synchronization is must between Tx and Rx
EXPERIMENT.NO-13
FREQUENCY SYNTHESIZER
AIM: To study the operation of frequency synthesizer using PLL
APPARATUS :
1. Frequency synthesizer trainer
4. Patch chords
THEORY:
PLL stands for „Phase locked loop‟ and it is basically a closed loop frequency control system,
whose functioning is based on phase sensitive detection of phase difference between the input
and output signals of controller oscillator.
Before the input is applied the PLL is in free running state. Once the input frequency is applied
the VCO frequency starts change and phase locked loop is said to be in captured mode. The
VCO frequency continues to change until it equals the input frequency and PLL is then in the
phase locked state. When phase locked the loop tracks any change in the input frequency through
its repetitive action.
Frequency Synthesizer:
The frequency divider is inserted between the VCO and the phase comparator. Since the output
of the divider is locked to the input frequency fin, VCO is running at multiple of the input
frequency. The desired amount of multiplication can be obtained by selecting a proper divide by
N network. Where N is an integer. For example fout = 5 fin a divide by N=10, 2 network is
needed as shown in block diagram. This function performed by a 4 bit binary counter 7490
configured as a divide by 10, 2 circuit. In this circuit transistor Q1 used as a driver stage to
increase the driving capacity of LM565 as shown in fig.b.
To verify the operation of the circuit, we must determine the input frequency range and then
adjust the free running frequency Fout of VCO by means of R1 (between 10th and 8th pin) and
CI (9th pin), so that the output frequency of the 7490 driver is midway within the predetermined
input frequency range. The output of the VCO now should 5Fin.
Free running frequency(f0):
Where there is no input signal applied, it is in free running mode.
F0 = 0.3 / (RtCt) where Rt is the timing resistor
Ct is the timing capacitor.
Lock range of PLL(fL)
FL = + 8f0/Vcc where f0 is the free running frequency
=2VCC
Capture range (fC)
CIRCUIT DIAGRAM:
PROGRAM:-
% program for frequency synthesizer
close all;
clear all;
clc
fs = 10000;
t = 0:1/fs:1.5;
f=50;
x1 = square(2*pi*f*t);
subplot(3,1,1)
plot(t,x1); axis([0 0.2 -1.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude');
title('Square wave input with freq=50HZ');
t = 0:1/fs:1.5;
x2 = square(2*pi*2*f*t);
subplot(3,1,2)
plot(t,x2); axis([0 0.2 -1.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude');
title('frequency multiplication by a factor of 2');
x3 = square(2*pi*f/2*t);
subplot(3,1,3)
plot(t,x3); axis([0 0.2 -1.2 1.2])
xlabel('Time (sec)');ylabel('Amplitude');
title('frequency division by a factor of 2');
PROCEDURE:
1. Switch on the trainer and verify the output of the regulated power supply i.e. + 5V. These
supplies are internally connected to the circuit so no extra connections are required.
2. Observe output of the square wave generator using oscilloscope and measure the range with
the help of frequency counter, frequency range should be around 1 KHz to 10 KHz.
3. Calculate the free running frequency range of the circuit (VCO output between 4th pin and
ground). For different values of timing resistor R1 (to measure Rt switch off the trainer and
measure Rt value using digital multimeter between given test points) . and record the frequency
values in tabular 1. Fout = 0.3 /(RtCt) where Rt is the timing resistor and Ct is the timing
capacitor =0.01 μf.
4. Connect 4th pin of LM 565 (Fout) to the driver stage and 5th pin (Phase comparator)
connected to 11th pin of 7490. Output can be taken at the 11th pin of the 7490. It should be
divided by the 10, 2 times of the fout.
EXPECTED WAVEFORMS:
Input waveforms
RESULT:
QUESTIONS:
1. What are the applications of PLL?
2. What is PLL?
4. What is a VCO?
EXPERIMENT NO-14
AGC CHARADTERISTICS
AIM: To study the operation of AGC in communication system.
APPARATUS:
1. Trainer Kit
THEORY:
A Simple AGC is a system by means of which the overall gain of a radio receiver is varied
automatically with the changing strength of the received signal, to keep the output substantially
constant. A dc bias voltage, derived from the detector. The devices used in those stages are ones
who‟s trans-conductance and hence gain depends on the applied bias voltage or current. It may
be noted in passing that, for correct AGC operation, this relationship between applied bias and
trans-conductance need not to be strictly linear, as long as trans-conductance drops significantly
with increased bias. All modern receivers are furnished with AGC, which enables tuning to
stations of varying signal strengths without appreciable change in the size of the output signal
thus AGC “irons out” input signal amplitude variations, and the gain control dose not have to be
re adjusted every time the receiver is tuned from one station to another, except when the change
in signal strengths is enormous. In addition, AGC helps to smooth out the rapid fading which
may occur with long-distance short-wave reception and prevents the overloading of last IF
amplifier which might otherwise have occurred.
CIRCUIT DESCRIPTION:
RF Generator:
Colpitts oscillator using FET is used here to generate RF signal of 455 KHz frequency to use as
carrier signal in this experiment. Adjustments for amplitude and frequency are provided on panel
for easy operation
AF generator:
Low frequency signal of approximately 1 KHz is generated using op-amp based wine bridge
oscillator; required application and adjustable attenuation are provided Regulated power
supply:
This consist of bridge rectifiers, capacitor filters and thee terminal regulators to provide required
dc voltages in the circuit i.e. +12v, -12v, +6v @150mA each
AM MODULATOR:
Modulator section illustrates the circuit of modulating amplifier employing transistor (BC 107)
as an active device in common emitter amplifier mode. R1 and R2 establish quiescent forward
bias for the transistor. The modulating signal is fed at the emitter section causes the bias to
increase or decrease in accordance with the modulating signal. R4 is emitter resistance and C3 is
by pass capacitor for carrier. Thus the carrier signal applied at the base gets amplified more when
the amplitude of the modulating signal is at its maximum and less when the signal by the
modulating signal output is amplitude- modulated signal. C2 couples the modulated signal to
output of the modulator.
Detector and AGC Stage:
This circuit incorporates two-stage amplifier, diode detector and AGC circuit.
1st IF amplifier
Q2 (BF 495c) acts as 1st if amplifier. The base of Q2 is connected through R5 (68k0 to the
detector output .R6 (100E) and C4 (47n) is decoupling filter for +B line. The base potential
depends on R4 (220k) base biasing resistor and detector current supplied by R5. The detector
current is proportional to the signal strength received. This is called A.G.C C6 (4.7/16) is used as
base bias and AGC decoupling capacitor C18 (2n7). This is given to the base of Q3 (BF 495D).
2nd IF AMPLIFIER
Q3 (BF 195C) acts as 2nd IF amplifier. The base bias for Q3 is provided by R7 (180k), C7 (47)
is used to keep the end 4of L8 (IFT2) at ground potential for if signal. The collector of Q3 is
connected to the L9 (IFT3). L9 contains 200pf capacitor inside across the primary. The output of
Q3 is available across the secondary of L9, the primary of which tuned by the internal 200pf
capacitor. R8 (220e), C8 (47n) consists the decoupling circuit for the collector supply of Q3. The
output of Q3 is coupled to detector diode D1 (OA 79).
Detector
Modulated IF signal from the secondary of L9 (IFT3) is fed to the detector diode D1 rectifies the
modulated if signal & if component of modulated signal is filtered by c8
(22n),r9(680e0&c14(22n).r9 is the detector load resistor. The detected signal (AF signal) is
given to the volume control P2 (10k Log) tough maximum audio output-limiting resister r21
(10k). It is also given to AGC circuit made of R5 (68k) and C6 (a.7/16).
AGC:
The Sound received from the LS will depend on the strength of the signals received at the
antenna. The strength of the received signals can vary widely due to fading. This will cause
variations in sound which can be annoying. Moreover, the Strength of signals can also be too
large in close vicinity of MW transmitters causing overloading of 2nd IF amplifier.
Automatic gain control (AGC) is used to minimize the variations in sound with changes in
Signals strength & to prevent overloading. The operation of AGC depends on the fact that The
gain obtained from any transistor depends on its collector current & becomes less When the
collector current is reduced to cut off (or increased to saturation For AGC, DC voltage obtained
from the detection of IF signals is applied to the 1st amplifier transistor base in such a way that
an increase in this voltages reduces the gain of the transistor. The result is that when the strength
of the incoming signal increases, the DC voltage also increases and this tends to reduce the gain
of the amplifier thus not permitting the output to change much. Here R5 (68k) C6 (4.7/16)
performs this function. C6 (4.7/16) is the AGC decoupling capacitor to by pass any AF signals
and keep the bias steady.
CIRCUIT DIAGRAM:
PROGRAM:
% program for AGC
close all
clear all
clc
Fs = 100e3; %sampling freq
plot(z);
title('am demodulation ');xlabel('time');ylabel('amplitude');
PROCEDURE:
1. As the circuit is already wired you just have to trace the circuit according to the Circuit
diagram given above Fig1.1.
2. Connect the trainer to the mains and switch on the power supply.
3. Measures the output voltages of the regulated power supply circuit i.e. +12v and -
12v,+6@150ma.
4. Observe outputs of RF and AF signal generator using CRO, note that RF voltage is
approximately 50mVpp of 455KHz frequency and AF voltage is 5Vpp of 1KHZ frequency.
5. Now vary the amplitude of AF signal and observe the AM wave at output, note the Percentage
of modulation for different value of AF signal.
%Modulation= (B-A)/ (B+A) X 100
6. Now adjust the modulation index to 30% by varying the amplitudes of RF & AF Signals
simultaneously.
7. Connect AM output to the input of AGC and also to the CRO channel-1.
8. Connect AGC link to the feedback network through OA79 diode
9. Now connect CRO channel-2 at output. The detected audio signal of1 KHz will be observed.
10. Calculate the voltage gain by measuring the amplitude of output signal (Vo) waveform, using
formula A = Vo/Vi.
11. Now vary input level of 455 KHz IF signal and observe detected 1 KHz audio Signal with
and without AGC link. The output will be distorted when AGC link removed i.e. there is no
AGC action.
12. This explains AGC effect in Radio circuit.
EXPECTED WAVEFORMS:
AF Modulated RF Output:
RESULT:
EXPERIMENT NO-15
PLL AS FM DEMODULATOR
AIM: .
To study phase lock loop and its capture range, lock range and free running VCO
APPARATUS:
THEORY:
PLL has emerged as one of the fundamental building block in electronic technology. It is used
for the frequency multiplication, FM stereo detector , FM demodulator , frequency shift keying
decoders, local oscillator in TV and FM tuner. It consists of a phase detector, a LPF and a
voltage controlled oscillator (VCO) connected together in the form of a feedback system. The
VCO is a sinusoidal generator whose frequency is determined by a voltage applied to it from an
external source. In effect, any frequency modulator may serve as a VCO. The phase detector or
comparator compares the input frequency, fin , with feedback frequency , f out , ( output
frequency). The output of the phase detector is proportional to the phase difference between f in ,
and f out , . The output voltage of the phase detector is a DC voltage and therefore m is often
refers to as error voltage . The output of the phase detector is then applied to the LPF , which
removes the high frequency noise and produces a DC lend. The DC level, in term is the input to
the VCO. The output frequency of the VCO is directly proportional to the input DC level. The
VCO frequency is compared with the input frequencies and adjusted until it is equal to the input
frequency. In short, PLL keeps its output frequency constant at the input frequency. Thus, the
PLL goes through 3 states. 1. Free running state. 2. Capture range / mode 3. Phase lock state.
Before input is applied, the PLL is in the free running state. Once the input frequency is applied,
the VCO frequency starts to change and the PLL is said to be the capture range/mode. The VCO
frequency cantinues to change (output frequency ) until it equals the input frequency and the
PLL is then in the phase locked state. When phase is locked, the loop tracks any change in the
input frequency through its repetitive action. Lock Range or Tracking Range: It is the range of
frequencies in the vicinity of f O over which the VCO, once locked to the input signal, will
remain locked
Capture Range : (f C) : Is the range of frequencies in the vicinity of „f O‟ over which the loop
will acquire lock with an input signal initially starting out of lock .
CIRCUIT DIAGRAMS:
circuit diagram
DEPT.OF ECE Page 93
JMTK ACLAB
PROCEDURE:
7. Connect pin 2 to oscillator or function generator through a 1µf capacitor, adjust the amplitude
aroung 2Vpp.
11. By varying the frequency in different steps observe that of one frequency the wave form will
be phase locked.
12. Change R-C components to shift VCO center frequency and see how lock range of the input
varies.
EXPECTED WAVEFORMS:-
FM modulated input:
RESULT: