PARISUTHAM INSTITUTE OF TECHNOLOGY &

SCIENCE

Department Of ECE

III Year / V Semester

EC 6512 COMMUNICATION SYSTEM LABORATORY

LAB MANUAL

SYLLABUS
EC6512 COMMUNICATION SYSTEMLABORATORY

LIST OF EXPERIMENTS:
CYCLE: 1
1. Signal Sampling and reconstruction
2. Time Division Multiplexing
3. AM Modulator and Demodulator
4. FM Modulator and Demodulator
5. Pulse Code Modulation and Demodulation
6. Delta Modulation and Demodulation
CYCLE: 2
7. Observation (simulation) of signal constellations of BPSK, QPSK and QAM
8. Line coding schemes
9. FSK, PSK and DPSK schemes (Simulation)
10. Error control coding schemes - Linear Block Codes (Simulation)
11. Communication link simulation
12. Equalization Zero Forcing & LMS algorithms(simulation)

Exp-No: 1, 2
Date:

SIGNAL SAMPLING AND RECONSTRUCTION

AIM:
To study the process of sampling and time division multiplexing of four signals using
pulse amplitude modulation and De-modulation and to reconstruct the signals at the receiver
using filters.
APPARATUS REQUIRED:
1. Sampling and TDM Communication trainer kit:
2. Multi Output Power Supply.
3. Patch cords.
4. CRO (60MHz)
THEORY
The Sample and Hold circuit uses two buffers to keep a voltage level stored in a
capacitor. Sample will charge the capacitor to the present signal level, while the input buffer
ensures the signal won't be changed by the charging process. From there, the output buffer will
make sure that the voltage level across the storage cap won't decrease over time. Sclear will short
out the storage cap, discharging it and setting the output to 0V.In actual practice, the switches
used are various forms of transistor switch, which provides cleaner switching and also allows
another circuit to control the sample and clearing operations. Excellent Sample and Hold circuits
like the LF398 are available on a single chip for cheap and easy use.
Sample and Hold circuits are used internally in Analog to Digital conversion. We might
also use them to hold a given signal value from any particular sensor on a robot, for analysis and
later use.
In TDM, by interleaving samples of several source waveforms in time, it is possible to
transmit enough information to a receiver, via only one channel to recover all message
waveforms.
The conceptual implementation of the time multiplexing of N similar messages f n(t)
where n= 1,2,3,..N is illustrated in fig 1. the time allocated to one sample of one message is
called time slot. The time intervals over which all message signals are sampled atleast once is

called a Frame. The portion of the time slot not used by the system may be allocated to other
functions like signaling, monitoring, synchronization, etc.
The four channels CH0, CH1, CH2, and CH3 are multiplexed on a single line TXD with the aid
of a electronic switch CD 4016. The CD 4016 latches one of the four inputs I0-I3 depending on
the control inputs C0, C1, C2, C3 which are generated by a 2: 4 line decoder.
The decoder, depending on the A0 and A1, which start from 00 to 11, generates 0000 to
0011 on the output lines Y0, Y1, Y2 and Y3. On receiving the control signals, the CD4016
latches the first information signal I0 on the first count 0000. In the next clock, the control inputs
change their state to 0001 and the input II is latched to the output on the same line. Similarly, all
the information signals are multiplexed without any interference on the line
PROCEDURE:
The sample and hold circuit is assembled with the desired components. The input signal
is given to the circuit from the function generator. The amplitude of the input signal should not
exceed 10 volts. The frequency of the input signal is set to 600 Hz. The frequency of the sample
signal is set to 5600 Hz. The next sample available is zero order holding device, integrate the
signal between consequence sampling inputs.

MODEL GRAPH FOR SAMPLING

MODEL GRAPH FOR TDM

RESULT
Thus the sampling process was studied and the different types of signals are multiplexed using
TDM Technique.

Exp-No:3
Date:

AM MODULATION AND DEMODULATION

AIM
To transmit a modulating signal after amplitude modulation using AM transmitter and
receive the signal back after demodulating using AM receiver.
APPARATUS REQUIRED:
1.
2.
3.
4.

AM transmitter trainer kit

AM receiver trainer kit
CRO
Patch cards

THEORY:
AMPLITUDE MODULATION:
Amplitude Modulation is a process by which amplitude of the carrier signal is
varied in accordance with the instantaneous value of the modulating signal, but frequency and
phase of carrier wave remains constant.
The modulating and carrier signal are given by
Vm(t) = Vm sinmt
VC(t) = VC sinCt
The modulation index is given by, ma = Vm / VC.
Vm = Vmax Vmin and VC = Vmax + Vmin
The amplitude of the modulated signal is given by,
VAM(t) = VC (1+ma sinmt) sinCt
Where
Vm = maximum amplitude of modulating signal
VC = maximum amplitude of carrier signal
Vmax = maximum variation of AM signal
Vmin = minimum variation of AM signal

PROCEDURE:

1.
2.
3.
4.

The circuit wiring is done as shown in diagram

A modulating signal input given to the Amplitude modulator
Now increase the amplitude of the modulating signal to the required level.
The amplitude and the time duration of the modulating signal are observed using
CRO.
5. Finally the amplitude modulated output is observed from the output of amplitude
modulator stage and the amplitude and time duration of the AM wave are noted
down.
6. Calculate the modulation index by using the formula and verify them. The final
demodulated signal is viewed using an CRO at the output of audio power amplifier
stage. Also the amplitude and time duration of the demodulated wave are noted down.

TABULATION:
Waveform
Message

Amplitude (V)

Time Period (msec)

Frequency

Carrier
modulated
Demodulated

MODEL GRAPH
Message signal
Vm

time
Vc

Carrier signal

time

AM signal
Vmc

time

RESULT
Thus the AM signal was transmitted using AM trainer kit and the AM signal detected
using AM detector kit.

Exp-No: 4
Date:

FREQUENCY MODULATION AND DEMODULATION

AIM
To transmit a modulating signal after frequency modulation using FM transmitter and
receive the signal back after demodulating using FM receiver.
APPARATUS REQUIRED:
1.
2.
3.
4.

FM transmitter trainer kit

FM receiver trainer kit
CRO
Patch cards

THEORY:
Frequency modulation (FM) is a form of modulation that represents information as
variations in the instantaneous frequency of a carrier wave. (Contrast this with amplitude
modulation, in which the amplitude of the carrier is varied while its frequency remains constant.)
In analog applications, the carrier frequency is varied in direct proportion to changes in the
amplitude of an input signal. Shifting the carrier frequency among a set of discrete values can
represent digital data, a technique known as frequency-shift keying. FM is commonly used at
VHF radio frequencies for high-fidelity broadcasts of music and speech (see FM broadcasting).
Normal (analog) TV sound is also broadcast using FM. A narrowband form is used for voice
communications in commercial and amateur radio settings. The type of FM used in broadcast is
generally called wide-FM, or W-FM. In two-way radio, narrowband narrow-fm (N-FM) is used
to conserve bandwidth. In addition, it is used to send signals into space.
FM is also used at intermediate frequencies by most analog VCR
systems, including VHS, to record the luminance (black and white) portion of
the video signal. FM is the only feasible method of recording video to and
retrieving video from magnetic tape without extreme distortion, as video
signals have a very large range of frequency components from a few hertz
to several megahertz, too wide for equalizers to work with due to electronic
noise below -60 dB. FM also keeps the tape at saturation level, and therefore
acts as a form of noise reduction, and a simple limiter can mask variations in
the playback output, and the FM capture effect removes print-through and
pre-echo. A continuous pilot-tone, if added to the signal as was done on
V2000 and many Hi-band formats can keep mechanical jitter under control
and assist time base correction.

PROCEDURE:

1.
2.
3.
4.

The circuit wiring is done as shown in diagram

A modulating signal input given to the Frequency modulator
Now increase the modulated signal to the required level.
The amplitude and the time duration of the modulating signal are observed using
CRO.
5. Finally the frequency modulated output is observed from the output of frequency
modulator stage and the amplitude and time duration of the FM wave are noted down.

MODEL GRAPH

TABULATION:
Waveform

Amplitude (V)

Time Period (msec)

Frequency

Message
Carrier
modulated
Demodulated
RESULT
Thus the FM signal was transmitted using FM trainer kit and the FM signal detected
using FM detector kit.

Exp-No: 5
Date:

PULSE CODE MODULATION

AIM
To generate a PCM signal using PCM modulator and detect the message signal from
PCM signal by using PCM demodulator.
APPARATUS REQUIRED
PCM kit, CRO and connecting probes
THEORY
Pulse code modulation is a process of converting an analog signal into digital. The voice
or any data input is first sampled using a sampler (which is a simple switch) and then quantized.
Quantization is the process of converting a given signal amplitude to an equivalent binary
number with fixed number of bits. This quantization can be either midtread or mid-raise and it
can be uniform or non-uniform based on the requirements. For example in speech signals, the
higher amplitudes will be less frequent than the low amplitudes. So higher amplitudes are given
less step size than the lower amplitudes and thus quantization is performed non-uniformly. After
quantization the signal is digital and the bits are passed through a parallel to serial converter and
then launched into the channel serially.
At the demodulator the received bits are first converted into parallel frames and each
frame is de-quantized to an equivalent analog value. This analog value is thus equivalent to a
sampler output. This is the demodulated signal.
In the kit this is implemented differently. The analog signal is passed trough a ADC
(Analog to Digital Converter) and then the digital codeword is passed through a parallel to serial
converter block. This is modulated PCM. This is taken by the Serial to Parallel converter and
then through a DAC to get the demodulated signal. The clock is given to all these blocks for
synchronization. The input signal can be either DC or AC according to the kit. The waveforms
can be observed on a CRO for DC without problem.
AC also can be observed but with poor resolution.

PROCEDURE
1. Power on the PCM kit.
2. Measure the frequency of sampling clock.
3. Apply the DC voltage as modulating signal.
4. Connect the DC input to the ADC and measure the voltage.
5. Connect the clock to the timing and control circuit.
6. Note the binary work from LED display. The serial data through the channel can be
observed in the CRO.
7. Also observe the binary word at the receiver end.
8. Now apply the AC modulating signal at the input.
9. Observe the waveform at the output of DAC.
10. Note the amplitude of the input voltage and the codeword. Also note the value of the
output voltage. Show the codeword graphically for a DC input.

MODEL GRAPH:

TABULAR COLUMN

S.No

Name of the signal

Modulating Signal

Carrier Signal

Modulated Signal

Demodulated
Signal

RESULT

Amplitude in V

Time period in Sec

Frequency in Hz

Thus the PCM signal was generated using PCM modulator and the message signal was
detected from PCM signal by using PCM demodulator.

Exp-No: 6
Date:

DELTA MODULATION
AIM
To transmit an analog message signal in its digital form and again reconstruct back the
original analog message signal at receiver by using Delta modulator.
APPARATUS REQUIRED
DM kit, CRO and connecting probes
THEORY
Delta modulation is the DPCM technique of converting an analog message signal to a
digital sequence. The difference signal between two successive samples is encoded into a single
bit code. The block and kit diagrams show the circuitry details of the modulation technique. A
present sample of the analog signal m(t) is compared with a previous sample and the difference
output is level shifted, i.e. a positive level (corresponding to bit 1) is given if difference is
positive and negative level (corresponding to bit 0) if it is negative. The comparison of samples
is accomplished by converting the digital to analog form and then comparing with the present
sample. This is done using an Up counter and DAC as shown in block diagram. The delta
modulated signal is given to up counter and then a DAC and the analog input is given to OPAMP
and a LPF to obtain the demodulated output.
PROCEDURE
1. Switch on the kit. Connect the clock signal and the modulating input signal to the
modulator block. Observe the modulated signal in the CRO.
2. Connect the DM output to the demodulator circuit. Observe the demodulator output on
the CRO.
3. Also observe the DAC output on the CRO.
4. Change the amplitude of the modulating signal and observe the DAC output.
Notice the slope overload distortion. Keep the tuning knob so that the distortion is
gone. Note this value of the amplitude. This is the minimum required value of the
amplitude to overcome slope overload distortion.
1. Calculate the sampling frequency required for no slope overload distortion. Compare
the calculated and measured values of the sampling frequency.

MODEL GRAPH

TABULAR COLUMN
S.No
Name of the signal
1

Modulating Signal

Carrier Signal

Modulated Signal

Demodulated

Amplitude in V

Time period in Sec

Frequency in Hz

Signal
RESULT
Thus the analog message signal in its digital form was transmitted and again the original
analog message signal was reconstructed at receiver by using Delta modulator and Demodulator.

Exp-No: 7
Date:

OBSERVATION OF SIGNAL CONSTELLATIONS OF BPSK, QPSK AND

QAM USING MATLAB
AIM:
To write a program in MATLAB for design of BPSK, QPSK and QAM.

PROGRAM:
QPSK
clc
clear all;
close all;
N=20;
X=randint(1,N);
L=100;
l=(N/2*L*0.01)-0.01
i=1;
for t=0:0.01:1
I(i)=cos(2*pi*t);
i=i+1;
end
i=1;
for t=0:0.01:1
Q(i)=sin(2*pi*t);
i=i+1;
end
for i=1:N/2
if X((i-1)*2+1)==1
for j=((i-1)*L+1):(i*L)
y(j)=1;
QMI(j)=y(j)*I(j);
end
else
for j=((i-1)*L+1):(i*L)
y(j)=-1;
QMI(j)=y(j)*I(j);
end

end
k=((i-1)*2)+2;
if X(k)==1
for j=((i-1)*L+1):(i*L)
y(j)=1;
QMQ(j)=y(j)*Q(j);
end
else
for j=((i-1)*L+1):(i*L)
y(j)=-1;
QMQ(j)=y(j)*Q(j);
end
end
end
for i=1:(N/2*L)
QP(i)=QMI(i)+QMQ(i);
end
for i=1:(N/2*L)
re1(i)=QP(i)*I(i);
reQ(i)=QP(i)*Q(i);
end
k=1;
for i=1:N/2
rI=0;
rQ=0;
for j=((i-1)*L+1):(i*L)
rI=rI+re(j);
rQ=rQ+reQ(j);
end
if rI>=0
real(i)=1;
else
real(i)=0;
end
if rQ>=0
imag(i)=1;
else
imag(i)=0;
end
det(k)=real(i);
det(k+1)=imag(i);

k=k+2;
end

RESULT:
Thus the FSK, PSK and DPSK was designed using MATLAB.

Exp-No: 8
Date:

LINE CODING
AIM :
To study different line coding techniques.
APPARATUS REQUIRED:
1. Communication trainer kit
2. Multi Output Power Supply.
3. Patch cords.
4. DSO/CRO
THEORY:
We need to represent PCM binary digits by electrical pulses in order to transmit them
through a base band channel.
The most commonly used PCM popular data formats are being realized here.
Line coding refers to the process of representing the bit stream (1s and 0s) in the form of
voltage or current variations optimally tuned for the specific properties of the physical channel
being used. The selection of a proper line code can help in so many ways: One possibility is to
aid in clock recovery at the receiver. A clock signal is recovered by observing transitions in the
received bit sequence, and if enough transitions exist, a good recovery of the clock is guaranteed,
and the signal is said to be self-clocking.
Another advantage is to get rid of DC shifts. The DC component in a line code is called the
bias or the DC coefficient. Unfortunately, most long-distance communication channels cannot
transport a DC component. This is why most line codes try to eliminate the DC component
before being transmitted on the channel.Such codes are called DC balanced, zero-DC, zero-bias,
or DC equalized.Some common types of line encoding in common-use nowadays are unipolar,
polar, bipolar, Manchester, MLT-3 and Duobinary encoding. These codes are explained here:
1. Unipolar (Unipolar NRZ and Unipolar RZ):
Unipolar is the simplest line coding scheme possible. It has the advantage of being compatible
with TTL logic. Unipolar coding uses a positive rectangular pulse p(t) to represent binary 1, and
the absence of a pulse (i.e., zero voltage) to represent a binary 0. Two possibilities for the pulse
p(t) exist3: Non-Return-to-Zero (NRZ) rectangular pulse and Return-to-Zero (RZ) rectangular
pulse. The difference between Unipolar NRZ and Unipolar RZ codes is that the rectangular pulse
in NRZ stays at a positive value (e.g., +5V) for the full duration of the logic 1 bit, while the pule
in RZ drops from +5V to 0V in the middle of the bit time.

A drawback of unipolar (RZ and NRZ) is that its average value is not zero, which means
it creates a significant DC-component at the receiver (see the impulse at zero frequency in the
corresponding power spectral density (PSD) of this line code

UNIPOLAR NRZ CODE

The disadvantage of unipolar RZ compared to unipolar NRZ is that each rectangular

pulse in RZ is only half the length of NRZ pulse. This means that unipolar RZ requires twice the
bandwidth of the NRZ code.
Polar (Polar NRZ and Polar RZ):
In Polar NRZ line coding binary 1s are represented by a pulse p(t) and binary 0s are
represented by the negative of this pulse -p(t) (e.g., -5V). Polar (NRZ and RZ) signals .Using the
assumption that in a regular bit stream a logic 0 is just as likely as a logic 1,polar signals
(whether RZ or NRZ) have the advantage that the resulting Dccomponent is very close to zero.

The rms value of polar signals is bigger than unipolar signals, which means that polar
signals have more power than unipolar signals, and hence have better SNR at the receiver.
Actually, polar NRZ signals have more power compared to polar RZ signals. The drawback of
polar NRZ, however, is that it lacks clock information especially when a long sequence of 0s or
1s is transmitted. Non-Return-to-Zero, Inverted (NRZI): NRZI is a variant of Polar NRZ. In
NRZI there are two possible pulses, p(t) and p(t). A transition from one pulse to the other
happens if the bit being transmitted is a logic 1, and no transition happens if the bit being
transmitted is a logic 0.

This is the code used on compact discs (CD), USB ports, and on fiber-based Fast Ethernet
at 100-Mbit/s .

MANCHESTER ENCODING:
In Manchester code each bit of data is signified by at least one transition. Manchester
encoding is therefore considered to be self-clocking, which means that accurate clock recovery
from a data stream is possible. In addition, the DC component of the encoded signal is zero.
Although transitions allow the signal to be self-clocking, it carries significant overhead as there
is a need for essentially twice the bandwidth of a simple NRZ or NRZI encoding

POWER SPECTRA OF LINE CODES:

Unipolar most of signal power is centered around origin and there is waste of power due to
DC component that is present.
Polar format most of signal power is centered around origin and they are simple to
implement.
Bipolar format does not have DC component and does not demand more bandwidth, but
power requirement is double than other formats.
Manchester format does not have DC component but provides proper clocking.

PROCEDURE
1. Connect the PRBS (test point P5) to various line coding formats. Obtain the coded output
as per the requirement.
2. Connect coded signal test point to corresponding decoding test point as inputs.
3. Set the SW1 as per the requirement.
4. Set the potentiometer P1 in minimum position.
5. Switch ON the power supply. Press the switch SW2 once.
6. Display the encoded signal on one channel of CRO and decoded signal on second channel
of CRO.

MODEL GRAPH:

TABULAR COLUMN
S.No

Name of the signal

Modulating Signal

Carrier Signal

Modulated Signal

Demodulated

Amplitude in V

Time period in Sec

Signal
RESULT
Thus the different line coding techniques was studied .

Frequency in Hz

Exp-No: 9
Date:

FSK, PSK and DPSK schemes USING MATLAB

AIM:
To write a program in MATLAB for design of FSK,PSK and DPSK.

PROGRAM:
FSK
clc
clear all
close all
N=10;
x=randint(1,N);
k=1;
for t=0.01:0.01:10
c1(k)=sin(2*pi*t);
c2(k)=sin(4*pi*t);
k=k+1;
end
for j=1:1:N;
if x(j)==0
for i=(j-1)*100+1:1:j*100
y(i)=0;
tr(i)=c2(i);
end
end
if x(j)==1
for i=(j-1)*100+1:1:j*100
y(i)=1;
tr(i)=c1(i);
end
end
end
for i=1:1:1000
re(i)=tr(i)*c1(i)*c2(i);
end
for j=1:1:N

d=0;
for i=(j-1)*100+1:1:j*100
d=d+re(i);
end
if d>0.5
det(j)=1;
else
det(j)=0;
end
end
for j=1:1:N
if det(j)==0
for i=(j-1)*100+1:1:j*100
det(i)=0;
end
end
if x(j)==1
for i=(j-1)*100+1:1:j*100
det(i)=1;
end
end
end
subplot(6,1,1);
plot(y);
title('message signal');
subplot(6,1,2);
plot(c1);
title('Carrier Signal-1');
subplot(6,1,3);
plot(c2);
title('Carrier Signal-2');
subplot(6,1,4);
plot(tr);
title('Transmitted Signal');
subplot(6,1,5);
plot(re);
title('Received Signal');
subplot(6,1,6);
plot(det);
title('Detected Signal');

PSK
clc
clear all;
close all;
N=10;%No.of Data
x=randint(1,N);
k=1;
for t=0.01:0.01:10
c(k)=2*sin(2*pi*t);
k=k+1;
end
for j=1:1:N
if x(j)==0
for i=(((j-1)*100)+1):1:(j*100)
y(i)=0;
tr(i)=-c(i);
end
else
for i=(((j-1)*100)+1):1:(j*100)
y(i)=1;
tr(i)=c(i);
end
end
end
for i=1:1:1000
re(i)=tr(i)*c(i);
end
for j=1:1:N
d=0;
for i=(((j-1)*100)+1):1:(j*100)
d=d+re(i)
end
if d>=0
det(j)=1;
else
det(j)=0;
end
end
for j=1:1:N
if det(j)==0

for i=(((j-1)*100)+1):1:(j*100)
det(i)=0;
end
end
if x(j)==1
for i=(((j-1)*100)+1):1:(j*100)
det(i)=1;
end
end
end
subplot(5,1,1);
plot(y);
title('Message Signal');
subplot(5,1,2);
plot(c);
title('Carrier Signal');
subplot(5,1,3);
plot(tr);
title('Transmitted Signal');
subplot(5,1,4);
plot(re);
title('Received Signal');
subplot(5,1,5);
plot(det);
title('Detected Signal');
RESULT:
Thus the FSK, PSK and DPSK was designed using MATLAB.

Exp-No:10

Date:

ERROR CONTROL CODING USING MATLAB

AIM:
To write a program in MATLAB for error control coding techniques.
ALGORITHM:
1.Get the input binary sequcence.
2.Calculate the reundancy bits for the corrosponding code.
3.Transmit the signal that contains message bits+redundancy bits added at the end.
4.Calculate the redundancy bits once again for the received bits.
5.If the redundancy bits=0 then no error in the transmission otherwise some error in
the transmission.
PROGRAM:
clc;
clear all;
close all;
k=input('Number of message bits');
n=input('Number of coded bits');
P=[1 1 1;0 1 1;1 0 1;1 1 0]
G=[eye(k) P]
for i=1:2^k
str=dec2base(i-1,2,4);
for j=1:k
m(i,j)=str(j);
end
end
for i=1:(2^k)

for r=1:n
o=0;
for j=1:k
o=o+(m(i,j)*G(j,r));
end
c(i,r)=mod(o,2);
end
end
e=zeros(n,n)
for i=1:n
e(i,i)=1;
end
% Syndrome Table
H=[P' eye(n-k)];
H1=H';
for i=1:n
for r=1:n-k
o=0;
for j=1:n
o=o+(e(i,j)*H1(j,r));
end
er(i,r)=mod(o,2);
end
end
for i=1:n

rec1=c(2^k,i)+e(1,i);
rec(1,i)=mod(rec1,2);
end
for i=1:1
for r=1:n-k
o=0;
for j=1:n
o=o+(rec(i,j)*H1(j,r));
end
sy(i,r)=mod(o,2);
end
end
i=1;
j=1;
while sy(1,j)==er(i,j)&&sy(1,j+1)==er(i,j+1)&&sy(1,j+2)==er(i,j+2)
rec_er=e(i,:);
i=i+1;
end
rec_er
%Error Corrected Message
for i=1:n
Det=rec(1,i)+rec_er(1,i);
det_rec(1,i)=mod(Det,2);
end
det_rec

RESULT:
Thus the error control coding techniques are executed using MATLAB programs.