Exp.No.

:
Date:
SAMPLING AND RECONSTRUCTION OF ANALOG SIGNALS
AIM:
To study the signal sampling and reconstruction of analog signals.

APPARATUS REQUIRED:
1. Sampling and Reconstruction Kit
2. Patch Cords
3. Probes
4. DSO

THEORY:
A band limited signal of finite energy has no frequency components higher than ‘W’
hertz is completely described by specified the values of the signal of instants of time separated
by 1/2W seconds, where ‘W’ is the higher frequency content. The zero order hold circuit is used
for practical reconstruction. It simply holds the value x(n) for ‘T’ seconds. Here ‘T’ is the
sampling period; The output of zero order hold is stair case signal. The reconstructed signal is
the succession of sine pulses weighted by x(nTs) these pulses are interpolated with the help of a
LPF. It is also called reconstruction filter or interpolation filter Natural sampling is chopper
sampling because the waveform of the sampled signal appears to be chopped off from the
original signal waveform. The top of the samples remains constant and equal to instantaneous
value of x(t) at start of sampling fs = 1/Ts

PROCEDURE:
1. Connect the main plug in to the main board. Keep the power switch in OFF position.
2. Put the duty cycle selector switch in position 50%
3. Link 25 Hz sine wave output to analog input.
4. Turn on the trainer.
5. Turning on the trainer select 250 Hz sampling rate by default.
6. Display 25Hz sine wave and sampled output on t oscilloscope. This display shows
25Hz sine wave being sampled at 200 Hz there are 10 samples for every cycle of the
sine wave.
FUNCTIONAL BLOCK DIAGRAM:
7. Link the sample output to the fourth order low pass filter display sample output and
output of the filter in the oscilloscope. The display shows the reconstructed original 21
Hz sine wave.
8. We had used sampling frequency greater than twice the maximum input frequency.
9. Remove the line from 25KHz sine wave output to the modulating input.
10. By successive process of frequency selector switch change the sampling frequency 32
KHz, 16KHz, 8 KHz,4 KHz,2 KHz,1 KHz,50 Hz and back to 250 Hz
11. Observe how sample output changes in each cases and how the lower sampling
frequencies introduce distortion in to the filter output waveform. This is due to the fact
that the filter does not attenuate the unwanted next frequency component significantly
use of higher order filter would improve the output waveform.
12. So far we have used sampling frequencies greater than twice the maximum input
frequency. To set the nyquist criteria set sampling rate 4 Hz 50% duty cycle.
13. Remove the link 25 Hz sine wave output to the modulating input.
14. Connect the link from 250 Hz or 500 Hz sine wave output to the modulating input and
link the sampled output to fourth order LPF. Display sample output and output of the
filter on the oscilloscope. The display shows the reconstruction signal 250 Hz or 500Hz
sine wave.
15. Now decrease the sampling rate to 32 KHz and then to 500 Hz. Observe the distorted
fact that we under sampled the input waveform overlooking the nyquist criteria and
thus the output was distorted even though the signal below the cutoff frequency of the
filter. This is also describes the phenomenon of aliasing.
TABULATION:

Amplitude Time period Frequency

Parameters
(V) (ms) (Hz)

Modulation signal

Sampled output

Sampled & hold output

Flat top output

Demodulated signal
RESULT:
Thus the signal sampling and reconstruction techniques were performed and graph
plotted.
 Exp.No.:
Date:
TIME DIVISION MULTIPLEXING (TDM)

AIM
To Perform the time division multiplexing using PAM Modulation and Demodulation
using the trainer kit.

APPARATUS REQUIRED

1.TDM Trainer.

2.CRO.

3.Patch Chords.

4.CRO probe

TIME DIVISION MULTIPLEXING

It is the process of taking the samples from different information signals, in time domain so
that they can be transmitted over the same channel. The main fact in the TDM technique is that
there are large intervals between the message samples. The samples from the other sources are
placed with in these time intervals. Thus every sample is separated from other in time domain.
Here, each signal is sampled over one sampling interval and transmitted one after the other along a
common channel. But the receiving end has to follow some constraints. i. It must receive and show
the signal as the transmitted.
ii. It must start at the same time as the transmitting end and establish electrical contact with the
same channel of the input channel.
When the two conditions are met then the receiver end is said to be in synchronization with the
transmitter end. If the 1st condition is not met then the samples different sources would get mixed
out the receiver end and if the 2nd condition is not met then the information from source '1' will be
received by same other channel which is not intending to accept the information from that
particular channel.
TABULATION
PROCEDURE
1. Switch ON the power supply to the board.
2. Make initial settings on VCT- 02 as follows.
a) Set all sine wave voltages to 2V,
b) Make the wiring connections as in wiring diagram which is provided at
the end of this experiment.
3. Display the multiplexed signal at test point T14 on channel 1 and 250Hz sinewave at
test point T2 on channel 2 of oscilloscope, note down waveforms.
4. Display the 500Hz sinewave at test point T3 on channel 2 in place of
250Hz, identify sampled version of this sinewave in TDM signal and note
down.
5. Similarly observe 1KHz and 2KHz waveforms at test point T4 and T5
respectively on oscilloscope and note down.
6. Display the TDM waveform (test point T14) on channel 1 and channel
synchronization signal (test point T13) on channel 2 of oscilloscope and note down
waveforms.
7. Display 250Hz sinewave at test point T2 on channel 1 and output sinewave at
test point T16 on channel 2 of oscilloscope and note down waveforms.
8. Similarly, observe input and output 500Hz, 1KHz and 2KHz sine waves on
oscilloscope and note down.

RESULT

Thus the Perform time division multiplexing using PAM Modulation and Demodulation
using the trainer kit and understand the concept using graph.
FUNCTIONAL BLOCK DIAGRAM:
Exp. No.:
Date:
AMPLITUDE MODULATION AND DEMODULATON
AIM:
To perform the amplitude modulation and demodulation using AM Kit.

APPARATUS REQUIRED:
1. Amplitude modulation kit
2. DSO
3. Probes
4. Patch cords

MODULATION THEORY:
Modulation is defined as the process by which some characteristics of a carrier signal is
varied in accordance with a modulating signal. The base band signal is referred to as the modulating
signal and the output of the modulation process is called as the modulation signal.
The carrier frequency fc must be much greater than the highest frequency components fm of
the message signal m(t) i.e. fc >>fm
The modulation index must be less than unity. if the modulation index is greater than
unity, the carrier wave becomes over modulated.

DEMODULATION THEORY:
The process of detection provides a means of recovering the modulating Signal from
modulating signal. Demodulation is the reverse process of modulation. The detector circuit is
employed to separate the carrier wave and eliminate the side bands. Since the envelope of an AM
wave has the same shape as the message, independent of the carrier frequency and phase,
demodulation can be accomplished by extracting envelope.
The depth of modulation at the detector output greater than unity and circuit impedance is
less than circuit load (Rl>Zm) results in clipping of negative peaks of modulating signal. It is called
“negative clipping “.
MODEL GRAPH:

TABULATION:

Amplitude Time period Frequency

Parameters
Volts sec Hz

Message Signal

Carrier Signal

Modulated Signal

Demodulated Signal
PROCEDURE:

A. Amplitude Modulation

1. Connect the mains cord of the trainer unit to AC 220V, 50 Hz supply.

2.Switch ON the trainer kit. The neon lamp will glow indicating that the unit is ready
for operation.
3.Observe the waveforms of modulating signal and carrier signal in an Oscilloscope.

4.Using patch cords, connect the modulating signal and the carrier signal to
‘AM MODULATION’.

5.Observe the amplitude modulated output waveform across sockets marked

‘AM OUTPUT’.

B. AM Demodulation

1.Set the amplitude of modulating and carrier signal in Amplitude Modulation.

2. Using patch cords, connect the ‘AM OUTPUT’ from the AM Modulation to
the sockets marked ‘AM INPUT‘ in the AM Demodulation.
3.Connect the detector output to filter input using patch cords.

4.Connect the filter output to amplifier input.

5.Connect the amplifier output to inverting amplifier input.

6.Observe the demodulated output waveform across sockets marked

‘DEMOD OUTPUT’.

RESULT:

Thus the amplitude modulation and demodulation operation has been performed.
FUNCTIONAL BLOCK DIAGRAM:
Exp. No.:

Date:
FREQUENCY MODULATION AND DEMODULATION
AIM:

To perform the frequency modulation and demodulation using FM kit.

APPARATUS REQUIRED:

1. Frequency modulation kit

2. DSO
3. Probes
4. Patch cords

THEORY:

Frequency modulation is a process of changing the frequency of a carrier wave in

accordance with the slowly varying base band signal. The main advantage of this modulation is
that it can provide better discrimination against noise.

PROCEDURE:

1. Connect the mains cord of the trainer unit to AC 220V, 50 Hz supply.

2. Switch ON the trainer kit. The neon lamp will glow indicating that the unit is ready
for operation.
3. Observe the Modulating Signal in an Oscilloscope.
4. Observe the FM Source in the Oscilloscope.
5. Using patch cords, connect the FM Source to sockets marked ‘FM INPUT’ in
FM Detector Circuit.
6.Observe the Frequency Demodulated Output Signal across sockets marked
‘DEMOD OUTPUT’.
MODEL GRAPH:

TABULATION:

Amplitude Time period Frequency

Parameters
Volts Sec Hz

Message signal

Carrier signal

Modulated signal

Demodulated signal
RESULT:

Thus the frequency modulation and demodulation has been performed and also the
modulation index was found.
FUNCTIONAL BLOCK DIAGRAM:
Exp. No.:

Date:

PULSE CODE MODULATION AND DEMODULATION

AIM:

To perform Pulse code Modulation and demodulation and to plot the waveform for
binary data at different frequencies

APPARATUS REQUIRED:

1. PCM kit
2. DSO
3. Probe
4. Patch cord

THEORY:

PCM is a method of converting an analog in to digita signals . Information in a analog

form cannot be processes by digital computers so its necessary to convert them in to digital PCM
is term which was formed during the development of digital audio transmission standards.
Digital data can be transported robustly over long distances unlike the analog data and can be
interleaved with other digital data sos various combinations of transmission channels can be
used.

PROCEDURE:

1.Connect the mains cord of the trainer unit to AC 220V, 50 Hz supply.

2.Switch ON the trainer kit. The neon lamp will glow indicating that the unit is ready
for operation.
3.Observe the Modulating Signal in an Oscilloscope.
4.Observe the FM Source in the Oscilloscope.
5.Using patch cords, connect sinewave signal source to the sample & hold circuits.
6.Connect the clock signal to the respective stages.
7. Observe the PCM output signal across the sockets marked “PCM OUTPUT”
8. Pulse mode modulated can also be observed by variable DC supply.

DEMODULATION:

1. Using patch cord connect the output from pulse code modulation to the sockets .
2. Observe the PCK demodulated output signal across the sockets marked “DEMOD
OUTPUT”
BLOCK DIAGRAM:

Sources of
Low –
Continuous- Sampler Quantizer Encoder
Time pass filter
message
PCM signal applied to channel input

(a) Transmitter
Disorted PCM Regenarated PCM
signal produced Regenarative Regenarative signal produced
at channel output ------------ at channel output
repeater repeater

(b) Transmission path

Final
Regenaration
Channel Decoder Reconstruction Destination
circuit
output filter

(b) Receiver

TABULATION:

Amplitude Time period Frquency

Parameters
Volts Sec Hz

Message signal

Modulated signal

Demodulated signal
RESULT:

Thus the Pulse Code Modulation and Demodulation was performed and output the
verified.
FUNCTIONAL BLOCK DIAGRAM:
Exp. No.:

Date:
DELTA MODULATION AND DEMODULATION
AIM:

To perform the Delta Modulation and Demodulation using hardware kit.

APPARATUS REQUIRED:

1. DM kit
2. DSO
3. Probe
4. Patch cord

THEORY:

Delta Modulation is a form of pulse modulation where a sample value is represented as a

single bit. This is almost similar to differential PCM, as the transmitted bit is only one per
sample just to indicate whether the present sample is larger or smaller than the previous one. The
encoding, decoding and quantizing process become extremely simple but this system cannot
handle rapidly varying samples. This increases the quantizing noise.

PROCEDURE:

A. Delta Modulation:
1.Connect the mains cord of the trainer unit to AC 220V, 50 Hz supply.

2.Switch ON the trainer kit. The neon lamp will glow indicating that the unit is ready
for operation.

3.Observe the waveforms of Modulating Signal Generator and Clock Signal

Generator in an Oscilloscope.

4.Using patch cords, connect the modulating signal to the sockets marked ‘MOD
SIGNAL’
in the Delta Modulation.

5.Using patch cords, connect the clock signal to the sockets marked ‘CLK’ in the
Signal Reconstructed.
6.Connect the ‘DELTA MOD OUTPUT’ in Delta Modulator to the sockets marked
BLOCK DIAGRAM OF DM MODULATOR AND DEMODULATOR:

Quantizer
e(n) + e(n) e(n)= ±1 To channel
∑
_

g(n)

Accumulator
Encoder

Low Pass
e(n) Accumulator Filter Output
Decoder

TABULATION:

Amplitude Time period Frequency

Parameters
Volts Sec Hz

Message signal

Demodulated signal

MODEL GRAPH:
 ‘DELTA MOD INPUT’ in the Signal Reconstructed.
7.Using patch cords, connect the Delta Reconstructed Output marked (#) from Signal
reconstructed to the Delta Modulator marked (#).

B. Delta Demodulation:
1.Using patch cords, connect the ‘DELTA RECONSTRUCTED OUTPUT’ from the
Signal.

2.Reconstructor to the sockets marked ‘DELTA RECONSTRUCTED INPUT‘ in the

Delta Demodulation.

3.Observe the demodulated output waveform across sockets marked ‘DEMOD

OUTPUT’.

RESULT:

Thus the Delta modulation and demodulation were performed and graph plotted.
MATLAB CODING:

BPSK:
clc;
clear all;
bits=1000000;
data=randint (1,bits)>0.5;
ebno=0:10;
BER=zeros(1,length(ebno));
for i=1:length(ebno)
%---Transmitter---------
%mapping of bits into symbols
symb=2.*data-1;
%----Filter
psf=ones(1,1);
M=length(psf);
% inserting zeros between the bits
% w.r.t number of coefficients of
 % PSF to pass the bit stream from the
PSF z=zeros(M-1,bits);
upsamp=[symb;z];
upsamp2=reshape(upsamp,1,(M)*bits);
%Passing the symbols from PSF
tx_symb=conv(upsamp2,psf);
%--------CHANNEL-----------
%Random noise generation and addition to the signal
ebnos=10.^(ebno(i)/10);
n_var=1/sqrt(2.*ebnos);
rx_symb=tx_symb+n_var*randn(1,length(tx_symb));
%xxxxxxxxxxxxxxxxxxxxxxxxxx
%-------RECEIVER-----------
rx_match=conv(rx_symb,psf);
rx=rx_match(M:M:length(rx_match));
rx=rx(1:1:bits);
recv_bits=(sign(rx)+1)./2;
%xxxxxxxxxxxxxxxxxxxxxxxxxxx
%---SIMULATED BIT ERROR RATE----
errors=find(xor(recv_bits,data));
errors=size(errors,2);
BER(i)=errors/bits;
%xxxxxxxxxxxxxxxxxxxxxxxxxxx
end
fs=1;
n_pt=2^9;
tx_spec=fft(tx_symb,n_pt);
f= -fs/2:fs/n_pt:fs/2-fs/n_pt;
figure
plot(f,abs(fftshift(tx_spec)));
title('Signal Spectrum for Signal with Rectangular Pulse
Shaping for BPSK');
xlabel('Frequency [Hz]');
ylabel('x(F)');
figure
semilogy(ebno,BER,'b.-');
hold on
thr=0.5*erfc(sqrt(10.^(ebno/10)));
semilogy(ebno,thr,'rx-');

xlabel('Eb/No (dB)')
ylabel('Bit Error rate')
title('Simulated Vs Theoritical Bit Error Rate for BPSK')
legend('simulation','theory')
grid on;
Exp. No.:

Date:

OBSERVATION (SIMULATION) OF SIGNAL CONSTELLATIONS OF BPSK, QPSK

AND QAM

AIM:

To simulate BPSK QPSK and QAM using MAT Lab .

APPARATUS REQUIRED:

1. Personal Computer
2. Matlab software R2014a

PROCEDURE:

1. Open Matlab version R2014a

2. Open new file and enter the program and save it.
3. Add the path to the location of the file in the system.
4. Compile the program and check for any error and debug it.
5. Note down the output.
QPSK:
clc
clear all
bits=1000000;
data=randint(1,bits)>0.5;
%---debugging---
%data=[1 1 1]
%xxxxxxxxxx
ebno=0:10;
BER=zeros(1,length(ebno));
%---Transmitter---------
%Gray mapping of bits into symbols
col=length(data)/2;
I=zeros(1,col);
Q=I;

I=data(1:2:bits-1);
Q=data(2:2:bits);

I= -2.*I+1;
Q= -2.*Q+1;

symb=I+j.*Q;
%----Filter
psf=ones(1,1);
%----
M=length(psf);
for i=1:length(ebno)
% inserting zeros between the bits
% w.r.t number of coefficients of
% PSF to pass the bit stream from the PSF
z=zeros(M-1,bits/2);

upsamp=[symb;z];
upsamp2=reshape(upsamp,1,(M)*bits/2);
%Passing the symbols from PSF

%tx_symb=conv(real(upsamp2),psf)+j*conv(imag(upsamp2),psf);
tx_symb=conv(upsamp2,psf);
%--------CHANNEL-----------
%Random noise generation and addition to the signal
npsd=10.^(ebno(i)/10);
n_var=1/sqrt(2.*npsd);
rx_symb=tx_symb+(n_var*randn(1,length(tx_symb))
+j*n_var*randn(1,length(tx_symb)) );
%xxxxxxxxxxxxxxxxxxxxxxxxxx
 %-------RECEIVER-----------
rx_match=conv(rx_symb,psf);
rx=rx_match(M:M:length(rx_match));
rx=rx(1:1:bits/2);
recv_bits=zeros(1,bits);
%demapping
k=1;
for ii=1:bits/2
recv_bits(k)= -( sign( real( rx(ii))) -1)/2;
recv_bits(k+1)=-( sign( imag( rx(ii)))-1)/2;
k=k+2;
end
%sign( real( rx ) )
%sign( imag( rx ) )
%data
%tx_symb
%rx_symb

%recv_bits
%xxxxxxxxxxxxxxxxxxxxxxxxxxx
%---SIMULATED BIT ERROR RATE----

errors=find(xor(recv_bits,data));
errors=size(errors,2);
BER(i)=errors/bits;
%xxxxxxxxxxxxxxxxxxxxxxxxxxx
end
fs=1;
n_pt=2^9;
tx_spec=fft(tx_symb,n_pt);
f= -fs/2:fs/n_pt:fs/2-fs/n_pt;
figure
plot(f,abs(fftshift(tx_spec)));
title('Signal Spectrum for Signal with Rectangular
Pulse Shaping for QPSK');
xlabel('Frequency [Hz]');
ylabel('x(F)');
figure
semilogy(ebno,BER,'b.-');
hold on
thr=0.5*erfc(sqrt(10.^(ebno/10)));
semilogy(ebno,thr,'rx-');
xlabel('Eb/No (dB)')
ylabel('Bit Error rate')
title('Simulated Vs Theoritical Bit Error Rate for
QPSK')
legend('Simulation','Theory'; grid on;
QAM:
clc
clear all
bits=3000000;
data=randint(1,bits)>0.5;
%---debugging---
%data=[1 1 1]
%xxxxxxxxxx
ebno=0:10;
BER=zeros(1,length(ebno));
thr=BER;
%---Transmitter---------
%Gray mapping of bits into symbols

col=length(data)/3;
I=zeros(1,col);
Q=I;
k=1;
for i=1:3:length(data)
if(data(i:i+2)==[0 0 0])
I(k)=1;
Q(k)=1;
k=k+1;
elseif(data(i:i+2)==[0 0 1])
I(k)=3;
Q(k)=1;
k=k+1;
elseif(data(i:i+2)==[0 1 0])
I(k)=-1;
Q(k)=1;
k=k+1;
elseif(data(i:i+2)==[0 1 1])
I(k)=-3;
Q(k)=1;
k=k+1;
elseif(data(i:i+2)==[1 0 0])
I(k)=1;
Q(k)=-1;
k=k+1;
elseif(data(i:i+2)==[1 0 1])

I(k)=3;
Q(k)=-1;
k=k+1;
elseif(data(i:i+2)==[1 1 0])
I(k)=-1;
Q(k)=-1;
 k=k+1;
elseif(data(i:i+2)==[1 1 1])

I(k)=-3;
Q(k)=-1;
k=k+1;
end
end
symb=I+j*Q;
%real(symb)
%imag(symb)
%----Filter
psf=ones(1,1);
Es=sum(psf.^2);
eb=Es/3;
eb=2;
%----
M=length(psf);
for i=1:length(ebno)
% inserting zeros between the bits
% w.r.t number of coefficients of
% PSF to pass the bit stream from the PSF
z=zeros(M-1,bits/3);
upsamp=[symb;z];
upsamp2=reshape(upsamp,1,(M)*bits/3);
%Passing the symbols from PSF

%tx_symb=conv(real(upsamp2),psf)+j*conv(imag(upsamp2),psf);
tx_symb=conv(upsamp2,psf);
%--------CHANNEL-----------
%Random noise generation and addition to the signal
ebno2=10.^(ebno(i)/10);
%no=eb/ebno2;
%n_var=sqrt(no/2);
n_var=sqrt(eb/(2*ebno2));
rx_symb=tx_symb+(n_var*randn(1,length(tx_symb))
+j*n_var*randn(1,length(tx_symb)) );
%xxxxxxxxxxxxxxxxxxxxxxxxxx
%------- RECEIVER-----------
rx_match=conv(rx_symb,psf);
rx=rx_match(M:M:length(rx_match));
rx=rx(1:1:bits/3);
recv_bits=zeros(1,bits);
%demapping
k=1;
for n=1:bits/3
I=real(rx(n));

Q=imag(rx(n));
if (I > 0) && (I < 2) && (Q > 0)
recv_bits(k:k+2)=[00 0];
elseif (I > 0) && (I < 2) && (Q < 0)
recv_bits(k:k+2)=[10 0];
elseif (I > 2) && (Q >0)
 recv_bits(k:k+2)=[00 1];
elseif (I > 2) && (Q < 0)
recv_bits(k:k+2)=[10 1];
elseif (I < 0) && (I > -2) && (Q > 0)
recv_bits(k:k+2)=[01 0];
elseif (I < 0) && (I > -2) && (Q < 0)
recv_bits(k:k+2)=[11 0];
elseif (I < -2) && (Q > 0)
recv_bits(k:k+2)=[01 1];
elseif (I < -2) && (Q < 0)
recv_bits(k:k+2)=[11 1];
end
k=k+3;
end
tx_symb;
rx_symb;
data;
recv_bits;
%xxxxxxxxxxxxxxxxxxxxxxxxxxx
%---SIMULATED BIT ERROR RATE----
errors=find(xor(recv_bits,data));
errors=size(errors,2);
BER(i)=errors/bits;
ebno_lin=(10^(ebno(i)/10))
thr(i)=(5/12)*erfc(sqrt(ebno_lin/2));
%xxxxxxxxxxxxxxxxxxxxxxxxxxx
end
fs=1;
n_pt=2^9;
tx_spec=fft(tx_symb,n_pt);
f= -fs/2:fs/n_pt:fs/2-fs/n_pt;
figure
plot(f,abs(fftshift(tx_spec)));
title('Signal Spectrum for Signal with Rectangular
Pulse Shaping for 8QAM');
xlabel('Frequency [Hz]');
ylabel('x(F)');
figure;
semilogy(ebno,BER,'b.-');
hold on
%ebno2=(10.^(ebno/10));
 %thr=(5/12).*erfc(sqrt((10.^(ebno/10))./2));
semilogy(ebno,thr,'rx-'); xlabel('Eb/No
(dB)')
ylabel('Bit Error rate')
title('Simulated Vs Theoritical Bit Error Rate for 8-QAM')
legend('Simulation','Theory'); grid on;

SIMULATED VS THEORTICAL BIT ERROR RATE FOR BPSK:

RECTANGULAR PULSE SHAPING FOR BPSK:

SIMULATED VS THEORITICAL BIT ERROR RATE FOR QPSK:

– 636 703.
SIGNAL SPECTRUM FOR SIGNAL WITH RECTANGULAR PULSE SHAPING FOR
QPSK:

SIGNAL SPECTRUM FOR SIGNAL WITH RECTANGULAR PULSE SHAPING FOR

8 QAM:
SIMULATED VS THEORITICAL BIT ERROR RATE FOR 8- QAM:

RESULT:

Thus the Signal Constellation of BPSK, QPSK and QAM were plotted.
Exp. No.:

Date:

LINE CODING AND DECODING TECHNIQUES

AIM:

1. To study the different line coding techniques with the communication trainer kit
.

APPARATUS REQUIRED:

2. Communication trainer kit.

3. Patch cords.
4. DSO/CRO

THEORY:

Line coding refers to the process of representing the bit stream (1’s and 0’s) 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.
Some common types of line encoding in common-use nowadays are unipolar, polar,
bipolar, Manchester and Duobinary encoding. These codes are explained here:

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 plus 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.
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
Ø Unipolar most of signal power is centered around origin and there is waste of power
due to DC component that is present.
Ø Polar format most of signal power is centered around origin and they are
simple to implement.
Ø Bipolar format does not have DC component and does not demand more bandwidth,
but power requirement is double than other formats.
Ø 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 and decoded signal on the cro
FUNCTIONAL BLOCK DIAGRAM:
TABULAR COLUMN:

Name of the Amplitude Time period Frequency

S.no
Signal Volt Sec Hz

MODEL GRAPH:

RESULT:

Thus the line coding and decoding techniques was studied

Exp. No.:

Date:
ASK, FSK AND PSK SIMULATION USING MATLAB

AIM:

To simulate ASK, FSK and PSK using matlab.

APPARATUS REQUIRED:

1. Personal Computer
2. Matlab software R2014a

PROCEDURE:

1. Open Matlab version R2014a.

2. Open new file and enter the program and save it.
3. Add the path to the location of the file in the system.
4. Compile the program and check for any error and debug it.
5. Note down the output.

MATLAB CODING:

ASK, FSK & PSK:

%matlab code for digital modulation (ask, fsk and psk)
pi=3.14;
f=5;
f2=10;
phi=pi;

x=[1 0 1 1 0];
nx=size(x,2);

i=1;
while i<nx+1
t = i:0.001:i+1;
if x(i)==1
ask=sin(2*pi*f*t);
fsk=sin(2*pi*f*t);
psk=sin(2*pi*f*t);
else
ask=0;
fsk=sin(2*pi*f2*t);
psk=sin(2*pi*f*t+phi);
end

subplot(3,1,1);
plot(t,ask);
xlabel('time')
ylabel('amplitude')
title('amplitude shift keying')
holdon;
gridon;
axis([1 10 -2 2]);

subplot(3,1,2);
plot(t,fsk);
xlabel('time')
ylabel('amplitude')
title('frequency shift keying')
holdon;
gridon;
axis([1 10 -2 2]);

subplot(3,1,3);
plot(t,psk);
xlabel('time')
ylabel('amplitude')
title('Phase shift keying')
holdon;
gridon;
axis([1 10 -2 2]);

i=i+1;
end
MODEL GRAPH:

RESULT:

The simulation of ASK FSK and PSK has been done using MATLAB and the outputs
were recorded.
Exp. No.:

Date:
ERROR CONTROL CODING
AIM:

To study error linear block code error control coding technique using MATLAB.

APPARATUS REQUIRED:

1. Personal Computer
2. Matlab software R2014a

THEORY:

In coding theory, a linear code is an error-correcting code for which any linear
combination of codewords is also a codeword. Linear codes are traditionally partitioned into
block codes and convolutional codes, although turbo codes can be seen as a hybrid of these two
types. Linear codes allow for more efficient encoding and decoding algorithms than other codes.
Linear codes are used in forward error correction and are applied in methods for transmitting
symbols (e.g., bits) on a communications channel so that, if errors occur in the communication,
some errors can be corrected or detected by the recipient of a message block.

PROCEDURE:

1. Open Matlab version R2014a

2. Open new file and enter the program and save it.
3. Add the path to the location of the file in the system.
4. Compile the program and check for any error and debug it.
5. Note down the output.
MATLAB CODING:
clc;clearall;
%g=input('Enter The Generator Matrix: ');%row value
separate by semicolon
disp('The Generator Matrix is : ');
g= [1 1 0 1 0 0 0 ;0 1 1 0 1 0 0;1 1 1 0 0 1 0;1 0 1 0 0 0
1];
disp(g);
disp ('The Order of Linear Block Code for given Generator
Matrix is:');
[n,k] = size(transpose(g));
disp('The Code Word Length is : ');disp(n);
disp('The Parity Bit Length is : ');disp(k);
for i = 1:2^k
for j = k:-1:1
if rem(i-1,2^(-j+k+1))>=2^(-j+k)
m(i,j)=1;
else
m(i,j)=0;
end
end
end
disp('The Possible Message Bits are : ');
disp(' c0 c1 c2 c3');
disp(m);
disp('The Possible Codewords are :')
disp(' b0 b1 b2 c0 c1 c2 c3 Hamming
weight')
c = rem(m*g,2);
d_min = sum((c(1:2^k,:))');
d_min2=d_min';
s= [ c d_min2];
disp(s);
disp('The Minimum Hamming Weight for the given Block
Code is= ');
d_min1 = min(sum((c(2:2^k,:))'));
disp(d_min1);
% DECode
p = [g(:,1:n-k)];
h = [eye(n-k),transpose(p)];
disp('The H Matrix is ');
disp(h);
ht = transpose(h);
disp('The H Transpose Matrix is ');
disp(ht);
r=[0 0 1 1 1 0 1];
e=rem(r*ht,2);
disp('Syndrome of a Given Codeword
is :'); disp(e);
for i = 1:1:size(ht)
if(ht(i,1:3)==e)
r(i) = 1-r(i);
break;
end
end
disp('The Error is in bit:');
disp(i);
disp('The Corrected Codeword is :');disp(r);
OUTPUT:
The Generator Matrix is :
1 1 0 1 0 0 0
0 1 1 0 1 0 0
1 1 1 0 0 1 0
1 0 1 0 0 0 1

The Order of Linear Block Code for given Generator Matrix is:
The Code Word Length is : 7
The Parity Bit Length is : 4
The Possible Message Bits are :
c0 c1 c2 c3
0 0 0 0
0 0 0 1
0 0 1 0
0 0 1 1
0 1 0 0
0 1 0 1
0 1 1 0
0 1 1 1
1 0 0 0
1 0 0 1
1 0 1 0
1 0 1 1
1 1 0 0
1 1 0 1
1 1 1 0
1 1 1 1

The Possible Codewordsare :

b0 b1 b2 c0 c1 c2 c3 Hamming weight
0 0 0 0 0 0 0 0
1 0 1 0 0 0 1 3
1 1 1 0 0 1 0 4
0 1 0 0 0 1 1 3
0 1 1 0 1 0 0 3
1 1 0 0 1 0 1 4
1 0 0 0 1 1 0 3
0 0 1 0 1 1 1 4
1 1 0 1 0 0 0 3
0 1 1 1 0 0 1 4
 0 0 1 1 0 1 0 3
1 0 0 1 0 1 1 4
1 0 1 1 1 0 0 4
0 0 0 1 1 0 1 3
0 1 0 1 1 1 0 4
1 1 1 1 1 1 1 7
The Minimum Hamming Weight for the given Block Code is= 3
The H Matrix is
1 0 0 1 0 1 1
0 1 0 1 1 1 0
0 0 1 0 1 1 1

The H Transpose Matrix is

1 0 0
0 1 0
0 0 1
1 1 0
0 1 1
1 1 1
1 0 1

Syndrome of a Given Codewordis :

0 0 1

The Error is in bit: 3

The Corrected Codewordis :
0 0 0 1 1 0 1

RESULT:

Thus the program for error control coding was done using MATLAB and the output
verified.
Exp. No.:

Date:
COMMUNICATION LINK
AIM:

To simulate the communication link using MATLAB simulation tool.

APPARATUS REQUIRED:

1. Personal Computer
2. Matlab software R2014a

PROCEDURE:

1. Open Matlab version R2014a.

2. And click the start button.
3. Now select the simulink library browser.
4. Open new file for a new project and fix the tools from the simulink library browser,
the connect the all tools and save it.
5. Now press run button for to simulate communication link.
6. Note down the output.

SIMULINK DIAGRAM OF COMMUNICATION LINK:

OUTPUT:

RESULT:

Thus the simulation of the communication link was done using MATLAB and the output
verified.
Exp. No.:

Date:
Equalization – Zero Forcing & LMS algorithms (simulation)
AIM:

To simulate the zero forcing and LMS algorithms equalizer using MATLAB simulation
tool.

APPARATUS REQUIRED:

1. Personal Computer
2. Matlab software R2014a

PROCEDURE:

1. Open Matlab version R2014a.

2. And click the start button.
3. Now select the simulink library browser.
4. Open new file for a new project and fix the tools from the simulink library browser,
the connect the all tools and save it.
5. Now press run button for to simulate communication link.
6. Note down the output.

MATLAB CODE FOR ZERO FORCING EQUALIZER:

%Project: Zero Forcing Equalizer
%Discription: Zero Forcing Equalizer is a type of
linear %equalizers used to
%combat ISI(inter symbol interference). This codes is a
%demostration of a
%simple implemenation of Zero Forcing Equalizer using
MatLab tools.

function Xh = ZF(h,r)
%r --- signal at the receiver
% h--- impulse response of the channel
%Computing inverse impulse response
gD=tf(h,1); %taking impulse response and transforming
%it to S domain
f=1/gD; % taking inverse of a transfer %function
[num,den]=tfdata(f,'v'); % extracting numerator and
%denominator %coefficients
 %Zero forcing
Xh=filter(num,den,r); % filtering

Xh=Xh(2:end); %this done for techniqal

reasons End
SIMULINK DIAGRAM OF LMS LINEAR EQUALIZER:

OUTPUT:

RESULT:

Thus the simulations of zero forcing and LMS algorithms equalizer has been done using
MATLAB and the output verified.