1

CONNECTION DIAGRAM

NATURAL SAMPLING

SAMPLE AND HOLD

FLAT – TOP SAMPLING

2
 1. SIGNAL SAMPLING AND RECONSTRUCTION

AIM:
To sample and reconstruct the given signal using natural sampling, sample - hold and
flat top sampling techniques using DCL-01Kit.
APPARATUS REQUIRED:

S.No. Name of the Equipment/ Component Range Quantity

1. DCL-01 trainer kit - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required

THEORY
Sampling is the process of splitting the given analog signal into different samples of
equal amplitudes with respect to time. There are two types of sampling namely natural
sampling, flat top sampling. Sampling should follow strictly the Nyquist Criterion i.e. the
sampling frequency should be twice higher than that of the highest frequency signal.
fs ≥ 2fm
Where,
fs= Minimum Nyquist Sampling rate (Hz), fm= Maximum analog input frequency (Hz).
In natural sampling, the top of the sampled pulse follows the shape of the original
signal. Since the natural sampling increases the system complexity, the flat top sampling is
mostly preferred in practical case. The flat top sampling is achieved with the help of sample
and hold circuit. Sample and Hold circuits are used internally in analog to digital conversion.
Here, the sampled signal obtained at each sampling instant is hold until the next sampling
instant. The sampling process must follow the sampling theorem for proper signal
reconstruction. In other words, the sampling frequency must be equal to twice that of the
highest frequency component present in the original 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.

3
TABULATION
Amplitude Time Period
Parameters
in Volts in Seconds
Sampling input
Ton =
Clock pulse
Toff=
Sampled output Ton =
Toff=

MODEL GRAPH

4
PROCEDURE:

1. Give the connections as per the connection diagram.

2. Put the duty cycle selector switch in position 50%.
3. Connect the modulating signal of 1 KHz frequency to BUFIN and measure its
amplitude and time period.
4. Connect the sampling frequency clock in the internal mode INT CLK using the
sampling signal selector switch SW4.
5. Set the sampling frequency to 8 KHz and note down the amplitude and time period of
the sampling signal.
6. Observe the natural sampling output at OUT terminal and note down the amplitude
and time period of the sampled signal.
7. Give the sampled signal to the 2nd order low pass filter circuit and observe the
reconstructed signal.
8. Note down the amplitude and time period of the reconstructed signal.
9. Repeat the same procedure for flat top sampling, Sample and hold and note down the
readings.
10. Plot the readings in the graph.

RESULT:

5
 CONNECTION DIAGRAM

MODEL GRAPH

6
 2. TIME DIVISION MULTIPLEXING

AIM:
To perform four channel Time Division multiplexing and De multiplexing using DCL –
02 trainer kit.
APPARATUS REQUIRED:
S.No. Name of the Equipment/ Component Range Quantity
1. DCL-02 trainer kit - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required
THEORY:
Time Division Multiplexing (TDM) is a technique of transmitting different source signals
on the same channel at different time slots. That is several information can be transmitted over
a single channel by sending samples from different information sources at different moments.
TDM is widely used in digital communication systems to increase the efficiency of the
transmitting medium. TDM can be achieved by electronically switching the samples such that
they interleave sequentially at correct instant in time without mutual interference.A major
problem in any TDM system is the synchronization of the transmitter and receiver timing
circuits. The transmitter and receiver must switch at the same time and frequency.TDM based
on analog modulation, the time slots are separated by guard slots to prevent crosstalk between
the channels.
In PAM, PPM the pulse is present for a short duration and for most of the time between
the two pulses no signal is present. This free space between the pulses can be occupied by
pulses from other channels. Thus, time division multiplexing makes maximum utilization of
the transmission channel. Each channel to be transmitted is passed through the low pass filter.
The outputs of the low pass filters are connected to the rotating sampling switch (or)
commutator.
It takes the sample from each channel per revolution and rotates at the rate of f s. Thus
The sampling frequency becomes fs the single signal composed due to multiplexing of input
channels. These channels signals are then passed through low pass reconstruction filters. If
the highest signal frequency present in all the channels is fm, then by sampling theorem, the
sampling frequency fs must be such that fs≥2fm. Therefore, the time space between
successive samples from any one input will be Ts=1/fs, and Ts≤ 1/2fm.

7
TABULATION:

Amplitude Time Period

Parameters
in Volts in Seconds
Message signal

Carrier signal

Modulated signal

8
PROCEDURE:

1. Give the connections as per the connection diagram.

2. Connect the power supply with proper polarity to the kit DCL-02 and switch it ON.
3. Connect 250 Hz, 500 Hz, 1 KHz, 2 KHz sine wave signal from the function generator
to the multiplexer input channel CH0, CH1, CH2 and CH3 by means of connecting
cords.
4. Connect the multiplexer OUT TXD to the transmitter section to the de-multiplexer
input RXD of the receiver section.
5. Connect the output of the receiver section CH0, CH1, CH2 and CH3 to the IN0, IN1,
IN2 and IN3 of the filter section.
6. Connect the sampling clock TXCLK and the channel identification clock SYNC of the
receiver section respectively.
7. Set the amplitude of the input sine wave as desired.
8. Take the observations and draw the required graphs.

RESULT:

9
AM TRANSMITTER (MODULATOR)

10
 3. AM MODULATOR AND DEMODULATOR
AIM

To transmit a modulating signal after amplitude modulation using AM transmitter and

receive the signal back after demodulating using AM receiver.
APPARATUS REQUIRED:

S.No. Name of the Equipment/ Component Range Quantity

1. AM Transmitter kit - ACL 01 - 1
2. AM Receiver kit - ACL 01 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required
THEORY:
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.
The modulating, carrier and modulated signals are given by
Vm(t) =Vmsinωmt ; VC(t) = VCsinωCt ; VAM(t) = VC (1+ma sinωmt)
sinωCt 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,
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
DEMODULATION THEORY:
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 “.

11
TABULATION:

Amplitude Time Period Frequency

Parameter
(V) in seconds in Hz
Message signal

Carrier signal

Modulated signal

Demodulated signal

Calculation of modulation index:

Practical calculation

Theoretical calculation

12
PROCEDURE:

1. The circuit wiring is done as shown in diagram

2. A modulating signal input given to the Amplitude modulator
3. Now increase the amplitude of the modulating signal to the required level.
4. 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
the 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

7. The final demodulated signal is viewed using CRO at the output of audio
power amplifier stage. Also the amplitude and time duration of the
demodulated wave are noted down.

13
Model Graph :

14
RESULT:

15
FM MODULATOR

FM DEMODULATOR

16
 4. FM MODULATION AND DEMODULATION

AIM
To plot the modulation characteristics of FM modulator and demodulator and also to
Observe and measure frequency deviation and modulation index of FM.
APPARATUS REQUIRED:
S.No. Name of the Equipment/ Component Range Quantity
1. FM Transmitter kit - ACL 03 - 1
2. FM Receiver kit - ACL 04 - 1
3. CRO 30 MHz 1
4. Power supply 5V, ±12V 1
5. Patch chords - Required

THEORY:
Frequency modulation is a type of modulation in which the frequency of the high
frequency (carrier) is varied in accordance with the instantaneous value of the modulating
signal.
FREQUENCY MODULATION GENERATION:
The circuits used to generate a frequency modulation must vary the frequency of a
high frequency signal (carrier) as function of the amplitude of a low frequency signal
(modulating signal). In practice there are two main methods used to generate FM.
DIRECT METHOD
An oscilloscope is used in which the reactance of one of the elements of the resonant
circuit depends on the modulating voltage. The most common device with variable reactance
is the Varactor or Varicap, which is a particular diode which capacity varies as function of the
reverse bias voltage. The frequency of the carrier is established with AFC circuits (Automated
frequency control) or PLL (Phase locked loop).
INDIRECT METHOD:
The FM is obtained in this case by a phase modulation, after the modulating signal has
been integrated. In this phase modulator the carrier can be generated by a quartz oscillator, and
so its frequency stabilization is easier. In the circuit used for the exercise, the frequency
modulation is generated by a Hartley oscillator, which frequency is determined by a fixed
inductance and by capacity (variable) supplied by varicap diodes.

17
TABULATION

Time Period
Parameter Amplitude (V) Frequency in Hz
in seconds

Message signal

Carrier signal
Tmin = Fmin =
Modulated signal
Tmax = Fmax =

Demodulated signal

MODEL GRAPH

18
FREQUENCY Deviation f and MODULATION INDEX fm:

The frequency deviation f represents the maximum shift between the modulated signal frequency,
over and under the frequency of the carrier.

We define modulation index m f the ratio between f and the modulating frequency f.

PROCEDURE:
1. Connect the power supply with proper polarity to the kit. While connecting this
ensures that the Power supply is OFF.
2. Switch on the power supply and carry out the following presetting as shown
incircuit Diagram.
3. In the FM modulator set the level about 2Vpp and frequency knob to theminimum
and switch on 1500 KHz.

4. Observe the FM modulated waveform from the RF/FM output of the FMmodulator

measure frequency deviation and modulation index of FM.

5. For demodulation switch on the demodulator and carry out the followingdemodulation
connection as shown in circuit diagram.
6. Observe the demodulated waveform and plot the graph.

Result :

19
CONNECTION DIAGRAM:

20
 5. PULSE CODE MODULATION AND DEMODULATION

AIM
To generate a PCM signal using PCM modulator and detect the message signal from
PCM signal by using PCM demodulator.
APPARATUS REQUIRED:

S.No. Name of the Equipment/ Component Range Quantity

1. PCM Trainer kit - DCL 03, DCL 04 - 1
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required

THEORY:
Pulse code modulation is known as digital pulse modulation technique. It is the process
in which the message signal is sampled and the amplitude of each sample is rounded off to the
nearest one of the finite set of allowable values. It consists of three main parts transmitter,
transmitter path and receiver. The essential operation in the transmitter of a PCM system are
sampling, quantizing and encoding. The band pass filter limits the frequency of the analog
input signal. The sample and hold circuit periodically samples the analog input signal and
converts those to a multi-level PAM signal. The ADC converts PAM samples to parallel PCM
codes which are converted to serial binary data in parallel to serial converter and then
outputted on the transmission line as serial digital pulse. The transmission line repeaters are
placed at prescribed distance to regenerate the digital pulse.
In the receiver serial to parallel converter converts serial pulse received from the
transmission line to parallel PCM codes. The DAC converts the parallel PCM codes to multi-
level PAM signals. The hold circuit is basically a Low Pass Filter that converts the PAM
signal back to its original analog form.
ADVANTAGES:
1. Secrecy
2. Noise resistant and hence free from channel interference
DISADVANTAGE:
1. Requires more bandwidth
APPLICATION:
1. Compact DISC for storage2. Military Applications.

21
TABULATION

Parameters Amplitude in V Time period in Sec

TON TOFF
Input Signal
Modulating Signal
Demodulating Signal

MODEL GRAPH :

22
PROCEDURE:
1. Refer the block diagram and carry out the following connections and switch settings.
2. Connect the power supply with power polarity to the kit and switch it ‘ON’.
3. Put the switch sw1 to ‘FAST’ mode.
4. Select 500Hz and 1 KHz sine wave signals generated on board.
5. Connect the signals to ‘CH0’ and ‘CH1’ of sample and hold circuit.
6. The output of sample and hold circuit ‘OUT0’ and ‘OUT1’ are given to the multiplexer, then to the
pulse code modulation logic.
7. The output ‘TXDATA’ is connected to ‘RXDATA’ of pulse code demodulation logic.
8. The output ‘DAC OUT’ is given to ‘IN’ of ‘DEMUX’.
9. Then the output of ‘DEMUX’ –‘OUT0’ and ‘OUT1’ are given to ‘IN0’ and ‘IN1’ of filter.
10. Observe the pulse code demodulated output at ‘OUT0’ and ‘OUT1’.

RESULT
.

23
CONNECTION DIAGRAM

DELTA MODULATION AND DEMODULATION

24
 6. DELTA MODULATION AND DEMODULATION

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:
S.No Name of the Equipment/ Component Range Quantity
Delta modulation/ demodulation
1. - 1
trainer kit, DCL 07
2. CRO 30 MHz 1
3. Power supply 5V, ±12V 1
4. Patch chords - Required
THEORY:
Delta modulation uses a single bit PCM code to achieve digital transmission of analog
signal. With conventional PCM, each code is a binary representative of both the sign and
magnitude of a particular sample. The algorithm of delta modulation is simple if the current
sample is smaller than the previous sample a logic0 is transmitted. If the current sample is
larger than the previous sample logic 1 is transmitted.
ADVANTAGES:
1. Simple system/circuitry - Cheap
2. Single bit encoding allows us to increase the sampling rate or to transmit more
information at some sampling rate for the given system BW.
DISADVANTAGES:
1. Noise and distortion
2. .Major drawback is that it is unable to pass DC information.
APPLICATIONS:
Digital voice storage, Voice transmission, Radio communication devices such TV remotes.
Adaptive delta modulation is delta modulation system where the step size of DAC is
automatically varied, depending on the amplitude characteristics of the analog input signal. A
common algorithm for an adaptive delta modulator is when three consecutive 1s or 0soccurs
the step size of the DAC is increased or decreased by a factor of 1.5
APPLICATION:
Audio communication system

25
MODEL GRAPH

TABULATION

Time Period
Parameter Amplitude (V)
in seconds
Modulating Signal

Carrier Signal

Modulated Signal

Demodulated Signal

26
PROCEDURE:
1. Refer to the block diagram and carry out the following connections
2. Connect the power supply with proper polarity to the kit DCL-07 and switch it ON.
3. Select the sine wave input 250Hz or through port P1 and connect port 250Hz to port
IN of input buffer.
4. Connect the output of buffer ‘OUT’ to digital samples input port ‘IN1’.
5. Then select clock rate of 8 KHz by pressing switch S1. Selected clock is indicated by
LED glow.
6. Keep switch S2 in (∆) deltaposition.
7. Connect output of a digital sampler port ‘OUT’ to input port ‘IN’ of integrator.
8. Connect output of integrator 1 port ‘OUT’ to input port ‘IN2’ of digital sampler.
9. The digital sampler ‘OUT’ is given to the input of the output buffer.
10. The output of the buffer is given to the second order input of the fourth order
Butterworth filter.

RESULT:

27
 CONNECTION DIAGRAM
NON-RETURN TO ZERO-SPACE (NRZ-S)

NON-RETURN TO ZERO-MARK (NRZ-M)

NON-RETURN TO ZERO-LEVEL (NRZ-L)

28
 7. LINE CODING SCHEMES

AIM

To study the various encoding and decoding techniques and observe the output waveforms
of NRZ-S, NRZ-M, NRZ-L, BIPHASE -L, BIPHASE –M, BIPHASE –S, URZ,AMI using
trainer kit DCL-05 and DCL-06.
APPARATUS REQUIRED:

S.No Name of the Equipment/ Component Range Quantity

1. Line encoding trainer kit DCL-05 - 1

2. Line decoding trainer kit DCL-06 - 1

3. CRO 30 MHz 1

4. Power supply 5V, ±12V 1

5. Patch chords - Required

THEORY:
NON-RETURN TO ZERO signal are the easiest formats that can be generated. These
signals do not return to zero with the clock. The frequency component associated with these
signals are half that of the clock frequency. The following data formats come under this
category. Non-return to zero encoding is commonly used in slow speed communications
interfaces for both synchronous and asynchronous transmission. Using NRZ, logic 1 bit is sent
as a high value and logic 0 bit is sent as a low value.
a) NON-RETURN TO ZERO-LEVEL (NRZ-L)
This is the most extensively used waveform in digital logics. All “ones” are
represented by “high” and all “zeros” by“low”. The data format is directly available at the
output of all digital data generation logics and hence very easy to generate. Here all the
transitions take place at the rising edge of the clock.
b) NON-RETURN TO ZERO-MARK (NRZ-M)
These waveforms are extensively used in tape recording. All „ones‟ are marked by
change inlevels and all “zeros” by no transitions, and the transitions take place at the rising
edge of the clock.
c) UNIPOLAR AND BIPOLAR
Unipolar signals are those signals, which have transition between 0 to +VCC. Bipolar
signals are those signals, which have transition between +VCC to –VCC.

29
 CONNECTION DIAGRAM
BIPHASE – LINE
CODING

BIPHASE MARK CODING

BIPHASE SPACE CODING

30
d) NON-RETURN TO ZERO-SPACE (NRZ-S)
This type of waveform is marked by change in levels for „zeros‟ and no transition for
“ones” and the transitions take place at the rising edge of the clock. This format is also used
in magnetic tape recording.
e) BIPHASE – LINE CODING (BIPHASE -L):
With the Biphase – L one is represented by a half bit wide pulse positioned during
thefirst half of the bit interval and a zero is represented by a half bit wide pulse positioned
during the second half of the bit interval.
f) BIPHASE MARK CODING(BIPHASE-M):
With the Biphase-M, a transition occurs at the beginning of every bit interval. A “one”
isrepresented by a second transition, half bit later, whereas a zero has no second transition.
g) BIPHASE SPACE CODING(BIPHASE-S):
With a Biphase-S, a transition occurs at the beginning of every bit interval. A “zero”
ismarked by a second transition, one half bit later; „one‟ has no second transition.
h) RETURN TO ZERO SIGNALS:
These signals are called “Return to Zero signals” since they return to “zero” with
theclock. In this category, only one data format, i.e, the unipolar return to zero (URZ); With
the URZ a “one” is represented by a half bit wide pulse and a “zero” is represented by the
absence of pulse.
i) MULTILEVEL SIGNALS:
Multilevel signals use three or more levels of voltages to represent the binary digits,
“one” and “zero” – instead of normal “highs” and “lows” Return to zero – alternative mark
inversion (RZ - AMI) is the most commonly used multilevel signal. This coding scheme is
most often used in telemetry systems. In this scheme, “one” are represented by equal
amplitude ofalternative pulses, which alternate between a +5 and -5. These alternating pulses
return to 0volt, after every half bit interval. The “Zeros” are marked by absence of pulses.

31
UNIPOLAR RETURN TO ZERO

ALTERNATIVE MARK INVERSION

32
PROCEDURE:

1. Connect power supply in proper polarity to the kits DCL-05 and DCL-06 and switch it on.
2. Connect CLOCK and DATA generated on DCL-05 to CODING CLOCK IN and
DATA INPUT respectively by means of the patch-chords provided.
3. Connect the coded data NRZ-L on DCL-05 to the corresponding DATA INPUT
NRZ-L, of the decoding logic on DCL-06.
4. Keep the switch SW2 for NRZ-L to ON position for decoding logic as shown in the
block diagram.
5. Observe the coded and decoded signal on the oscilloscope.
6. Connect the coded data NRZ-M on DCL-05 to the corresponding DATA INPUT NRZ-M,
of the decoding logic on DCL-06.
7. Keep the switch SW2 for NRZ –M to ON position for decoding logic as shown in the
block diagram.
8. Observe the coded and decoded signal on the oscilloscope.
9. Connect the code data NRZ-S on DCL-05 to the corresponding DATA INPUT
NRZ-S, of the decoding logic on DCL-06.
10. Keep the switch SW2 for NRZ-S to ON position for decoding logic as shown in the
block diagram.
11. Observe the coded and decoded signal on the oscilloscope.
12. Use RESET switch for clear data observation if necessary.
13. Unipolar to Bipolar/Bipolar to Unipolar:
a. connect NRZ-L signal from DCL-05 to the input post IN Unipolar to Bipolar
and Observe the Bipolar output at the post OUT.
b. Then connect bipolar output signal to the input post IN of Bipolar to
Unipolar and Observe Unipolar out at post OUT.

33
MODEL GRAPH:

34
RESULT:

35
Program
%ASK Modulation
clc;
clear
all;
close
all;
%GENERATE CARRIER SIGNAL
Tb=1; fc=10;
t=0:Tb/100:1;
c=sqrt(2/Tb)*sin(2*pi*fc*t);
%generate message
signal N=8;
m=rand(1,);
t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2]
if m(i)>0.5
m(i)=1;
m_s=ones(1,length(t);
else
m(i)=0;
m_s=zeros(1,length(t;
end
message(i,:)=m_s;
%product of carrier and
message ask_sig(i,:)=c.*m_s;
t1=t1+(Tb+.01);
t2=t2+(Tb+.01);
%plot the message and ASK signal
subplot(5,1,2);axis([0 N -2
2]);plot(t,message(i,:),'r');
title('message signal');xlabel('t--->');ylabel('m(t)');grid
on hold on
subplot(5,1,4);plot(t,ask_sig(i,:));
title('ASK signal');xlabel('t--->');ylabel('s(t)');grid

36
 8.SIMULATION OF ASK, FSK AND BPSK GENERATION SCHEME

Aim:
To generate amplitude shift keyed (ASK) signal using MATLAB.
Generation of ASK
Amplitude shift keying - ASK - is a modulation process, which imparts to a sinusoid two
or more discrete amplitude levels. These are related to the number of levels adopted by the digital
message. For a binary message sequence there are two levels, one of which is typically zero. The
data rate is a sub-multiple of the carrier frequency. Thus the modulated waveform consists of
bursts of a sinusoid. One of the disadvantages of ASK, compared with FSK and PSK, for example,
is that it has not got a constant envelope. This makes its processing (eg, power amplification)
more difficult, since linearity becomes an important factor. However, it does make for ease of
demodulation with an envelope detector.
modulation
1. Generate carrier signal.
2. Start FOR loop
3. Generate binary data, message signal(on-off form)
4. Generate ASK modulated signal.
5. Plot message signal and ASK modulated signal.
6. End FOR loop.
7. Plot the binary data and carrier.

Generation of FSK
Frequency-shift keying (FSK) is a frequency modulation scheme in which digital
information is transmitted through discrete frequency changes of a carrier wave. The simplest FSK
is binary FSK (BFSK). BFSK uses a pair of discrete frequencies to transmit binary (0s and 1s)
information. With this scheme, the "1" is called the mark frequency and the "0" is called the space
frequency.
In binary FSK system, symbol 1 & 0 are distinguished from each other by transmitting one of the
two sinusoidal waves that differ in frequency by a fixed amount.

37
 end
hold
off
%Plot the carrier signal and input binary
data subplot(5,1,3);plot(t,c);
title('carrier signal');xlabel('t--->');ylabel('c(t)');grid
on subplot(5,1,1);stem(m);
title('binary data bits');xlabel('n--->');ylabel('b(n)');grid on

MODEL GRAPH :

38
 Si (t) = √2E/Tb cos 2πf1t 0≤ t
≤Tb 0 elsewhere
Where i=1, 2 &Eb=Transmitted energy/bit
Transmitted freq= ƒi = (nc+i)/Tb, and n = constant (integer), Tb = bit
interval Symbol 1 is represented by S1 (t)
Symbol 0 is represented by S0 (t)

Generation of PSK signal

PSK is a digital modulation scheme that conveys data by changing, or modulating, the
phase of a reference signal (the carrier wave). PSK uses a finite number of phases, each assigned a
unique pattern of binary digits. Usually, each phase encodes an equal number of bits. Each pattern
of bits forms the symbol that is represented by the particular phase. The demodulator, which is
designed specifically for the symbol-set used by the modulator, determines the phase of the
received signal and maps it back to the symbol it represents, thus recovering the original data.

In a coherent binary PSK system, the pair of signal S1(t) and S2 (t) used to represent
binary symbols 1 & 0 are defined by
S1 (t) = √2Eb/ Tb Cos 2πfct
S2 (t) =√2Eb/Tb (2πfct+π) = - √ 2Eb/Tb Cos 2πfct where 0 ≤ t< Tb
and Eb = Transmitted signed energy for bit
The carrier frequency fc =n/Tb for some fixed integer n.

39
Program
% FSK Modulation
clc;
clear
all;
close
all;
%GENERATE CARRIER
SIGNAL Tb=1; fc1=2;fc2=5;
t=0:(Tb/100):Tb;
c1=sqrt(2/Tb)*sin(2*pi*fc1*t);
c2=sqrt(2/Tb)*sin(2*pi*fc2*t);
%generate message
signal N=8;
m=rand(1,);
t1=0;t2=Tb
for i=1:N
t=[t1:(Tb/100):t2]
if m(i)>0.5
m(i)=1;
m_s=ones(1,length(t));
invm_s=zeros(1,length(t));
else
m(i)=0;
m_s=zeros(1,length(t));
invm_s=ones(1,length(t));
end
message(i,:)=m_s;
%Multiplier
fsk_sig1(i,:)=c1.*m_s;
fsk_sig2(i,:)=c2.*invm;
fsk=fsk_sig1+fsk_sig2;
%plotting the message signal and the modulated
signal subplot(3,2,2);axis([0 N -2
2]);plot(t,message(i,:),'r'

40
41
MODEL GRAPH :

Program

% BPSK modulation
clc;
clear
all;
close
all;
%GENERATE CARRIER SIGNAL
Tb=1;
t=0:Tb/100:Tb;
fc=2;
c=sqrt(2/Tb)*sin(2*pi*fc*;
%generate message
signal N=8;
m=rand(1,N
);
t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2]
if m(i)>0.5
m(i)=1;
m_s=ones(l
ength(t));

42
43
 else
m(i)=0;
m_s=-1*ones(1,length(t));
end
message(i,:)=m_s;
%product of carrier and message
signal bpsk_sig(i,:)=c.*m_s;
%Plot the message and BPSK modulated signal
subplot(5,1,2);axis([0 N -2
2]);plot(t,message(i,:),'r');
title('message signal(POLAR form)');xlabel('t--->');ylabel('m(t)');
grid on; hold on;
subplot(5,1,4);plot(t,bpsk_sig(i,:));
title('BPSK
signal');xlabel('t--->');ylabel('s(t)');
grid on; hold on;

MODEL GRAPH

44
RESULT :

45
PROGRAM FOR DPSK GENERATION SCHEME:
clc;
clear
all;
rng default
M = 6; % Alphabet size
dataIn = randi([0 M-1],1011,1); % Random
message txSig = dpskmod(dataIn,M); % Modulate
rxSig =
txSig*exp(2i*pi*rand());
dataOut =
dpskdemod(rxSig,M); errs =
symerr(dataIn,dataOut)
errs =
symerr(dataIn(2:end),dataIn(2:end))
figure
subplot(2,2,1)
plot(dataIn)
title('DATA')
subplot(2,2,2)
plot(txSig)
title('DPSK
SIGNAL')
subplot(2,2,3)
plot(rxSig)
title('Received
DPSK')
subplot(2,2,4)
plot(dataOut)
title('DATA RECEIVED')

46
 9.SIMULATION OF DPSK, QPSK AND QAM GENERATION SCHEMES

AIM

To simulate DPSK, QPSK and QAM Generation Schemes using Matlab.

SOFTWARE REQUIRED:

MATLAB

PROCEDURE:

1. Click on the Matlab icon on desktop.

2. Matlab windows open.

3. Click on the 'file' menu on the menu bar.

4. Click on NEW file on file menu.

5. An editor window open, start typing command.

6. Now finish the command.

7. And run the program.

47
SIMULATION WAVEFORM OF DPSK

PROGRAM FOR QPSK GENERATION SCHEME:

QPSK
clc; clear all;
t=0:0.0001:0.2
5;
m=square(2*pi*10*t);
c1=sin(2*pi*60*t);
c2=sin(2*pi*60*t+180
); for i=1:2500
if(mod(i,1000))<500
s(i)=c1(i);
else
s(i)=-c2(i
); end
end
subplot(4,1;
plot(t,m,'k','linewidth',5);
title('polor representation of message 1 0 1 0 1
0'); xlabel('time'); ylabel('amplitude')

48
49
 subplot(4,1,2);
plot(c1);
title('frequency 1');
xlabel('time');
ylabel('amplitude');
subplot(4,1,3); plot(c2);
title('frequency 2');
xlabel('time'); ylabel('amplitude');
subplot(4,1,4); plot(s);
title('quadrature phase shift
keying'); xlabel('time');
ylabel('amplitude');

SIMULATION WAVEFORM OF OPSK

50
RESULT :

51
 PROGRAM BPSK
clc;
clear
all;
close
all;
M=2;
k=log2(M);
n=3*1e5;
nsamp=8;
X=randint(n,1
);
xsym =
bi2de(reshape(X,k,length(X)/k).','left-msb');
Y_psk= modulate(modem.pskmod(M),xsym);
Ytx_psk = Y_psk;
EbNo=30;
SNR=EbNo+10*log10(k)-10*log10(nsamp);
Ynoisy_psk = awgn(Ytx_psk,SNR,'measured');
Yrx_psk = Ynoisy_psk;
h1=scatterplot(Yrx_psk(1:nsamp*5e3),nsamp,0,'r.
'); hold on;
scatterplot(Yrx_psk(1:5e3),1,0,'k*',h1);
title('constellation diagram BPSK');
legend('Received signal' ,'signal
constellation'); axis([-5 5 -5 5]);
hold off;

Program for QPSK & QAM:

clc;
clear
all;
close
all;
M=16;
k=log2(M);
n=3*1e5;
nsamp=8;
X=randint(n,1
);
xsym =
bi2de(reshape(X,k,length(X)/k).','left-msb');
Y_qam= modulate(modem.qammod(M),xsym);
Y_qpsk= modulate(modem.pskmod(M),xsym);
Ytx_qam = Y_qam;
Ytx_qpsk =
Y_qpsk;
EbNo=30;
SNR=EbNo+10*log10(k)-10*log10(nsamp);
Ynoisy_qam =
awgn(Ytx_qam,SNR,'measured'); Ynoisy_qpsk

52
 10. SIMULATION OF SIGNAL CONSTELLATIONS OF BPSK, QPSK AND QAM

AIM:
To plot the constellation diagram of digital modulation system BPSK, QPSK & QAM
using MATLAB.

APPARATUS REQUIRED:
1. PC
2. MATLAB SOFTWARE

THEORY:
A constellation diagram is a representation of a signal modulated by an arbitrary
digital modulation scheme. It displays the signal as a two dimensional scatter diagram in the
complex plane at symbol sampling instants. It can also be viewed as the possible symbols that
may be selected by a given modulation scheme as points in the complex plane.

53
 = awgn(Ytx_qpsk,SNR,'measured'); Yrx_qam = Ynoisy_qam;
Yrx_qpsk = Ynoisy_qpsk;
h1=scatterplot(Yrx_qam(1:nsamp*5e3),nsamp,0,'r.'
); hold on;
scatterplot(Yrx_qam(1:5e3),1,0,'k*',h1);
title('constellation diagram 16 QAM');
legend('Received signal' ,'signal constellation');
axis([-5 5 -5 5]);
hold off;
h2=scatterplot(Yrx_qpsk(1:nsamp*5e3),nsamp,0,'
r.'); hold on;
scatterplot(Yrx_qpsk(1:5e3),1,0,'k*',h2);
title('constellation diagram 16 PSK');
legend('Received signal' ,'signal
constellation'); axis([-5 5 -5 5]);
hold off;
title('constellation diagram 16 PSK');
legend('Received signal' ,'signal
constellation'); axis([-5 5 -5 5]);
holdoff;

BPSK CONSTELLATION :

54
55
QPSK CONSTELLATION:

QAM CONSTELLATION:

56
RESULT:

57
ASK Demodulation

t1=0;t2=Tb

for i=1:N

t=[t1:Tb/100:t2]

%correlator

x=sum(c.*ask_sig(i,:));

%decision device

if x>0

demod(i)=1;

else

demod(i)=0;

end

t1=t1+(Tb+.01);

t2=t2+(Tb+.01);

end

%plot demodulated binary data bits

subplot(5,1,5);stem(demod);

title('ASK demodulated signal'); xlabel('n--->');ylabel('b(n)');grid on

FSK Demodulation

t1=0;t2=Tb

for i=1:N

t=[t1:(Tb/100):t2]

%correlator

x1=sum(c1.*fsk_sig1(i,:));

x2=sum(c2.*fsk_sig2(i,:));

x=x1-x2;

%decision device

if x>0

58
 11. SIMULATION OF ASK, FSK AND BPSK DETECTION SCHEMES

AIM:
To demodulate Binary phase shift keyed (BPSK) signal using MATLAB

APPARATUS REQUIRED:
MATLAB

Demodulation

ASK signal has a well defined envelope. Thus it is amenable to demodulation by an envelope
detector. Some sort of decision-making circuitry is necessary for detecting the message. The signal is
recovered by using a correlator and decision making circuitry is used to recover the binary sequence.

ASK demodulation

Start FOR loop

Perform correlation of ASK signal with carrier to get decision variable

Make decision to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’

Plot the demodulated binary data.

FSK demodulation

Start FOR loop

Perform correlation of FSK modulated signal with carrier 1 and carrier 2 to get two decision
variables x1 and x2.

Make decisionon x = x1-x2 to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’.

Plot the demodulated binary data.

BPSK demodulation

Start FOR loop

Perform correlation of PSK signal with carrier to get decision variable

Make decision to get demodulated binary data. If x>0, choose ‘1’ else choose ‘0’

Plot the demodulated binary data.

59
 demod(i)=1;
else
demod(i)=0;
end
t1=t1+(Tb+.01);
t2=t2+(Tb+.01);
end
%Plotting the demodulated data bits
subplot(3,2,6);stem(demod);
title(' demodulated data');xlabel('n---->');ylabel('b(n)'); grid on;

% PSK Demodulation

t1=0;t2=Tb
for i=1:N
t=[t1:.01:t2]
%correlator
x=sum(c.*bpsk_sig(i,:));
%decision device
if x>0
demod(i)=1;
else
demod(i)=0;
end
t1=t1+1.01;
t2=t2+1.01;
end
%plot the demodulated data bits
subplot(5,1,5);stem(demod);
title('demodulated data');xlabel('n--->');ylabel('b(n)');

SIMULATION WAVEFORM ASK

60
61
SIMULATION WAVEFORM BPSK

SIMULATION WAVEFORM FSK

62
RESULT :

63
MATLAB CODE:

% Input Generator Matrix

g=input('Enter The Generator Matrix: ')
disp ('G = ')
disp ('The Order of Linear block Code for given Generator Matrix is:')
[n,k] = size(transpose(g))
fori = 1:2^k
forj = k:-1:1
ifrem(i-1,2^(-j+k+1))>=2^(-j+k)
u(i,j)=1;
else
u(i,j)=0;
end
end
end
disp('The Possible Codewords are :')
c = rem(u*g,2)
disp('The Minimum Hamming Distance dmin for given Block Code is= ')
d_min = min(sum((c(2:2^k,:))'))

% Code Word

r= input('Enter the Received Code Word:')

P= [g(:,n-k+2:n)];
h= [transpose(p),eye(n-k)]; disp('HammimgCode')

ht = transpose(h)

disp('Syndrome of a Given Codeword is :')

s= rem(r*ht,2)

fori = 1:1:size(ht)
if(ht(i,1:3)==s)

r(i) = 1-r(i);
break;

end
end

disp('The Error is in bit:')

disp(i);
disp('The Corrected Codeword is :')
disp(r);

64
12. SIMULATION OF LINEAR BLOCK AND CYCLIC ERROR CONTROL CODING SCHEME

AIM:

To simulate and study the error control coding scheme of linear block code using MATLAB

ALGORITHM:

STEP 1: Give the generator matrix

STEP 2: Find the order of the linear block code for the given generator matrix

STEP 3: Obtain the possible code words

STEP 4: Find the minimum hamming distance

STEP 5: Give the received code word

STEP 6: Calculate the syndrome vector and compare it with transpose of

hamming matrix.

STEP 7: Find the error bit position and display the corrected code word

65
PROGRAM:
Generation of parity check matrix and generator matrix for a (7, 4) Hamming code.
[h,g,n,k] = hammgen(3);
Generation of parity check matrix for the generator polynomial g(x) = 1+x+x3.
h1 = hammgen(3,[1011]);
Computation of code vectors for a cyclic code
clc;
close all;
n=7;
k=4;
msg=[1 0 0 1; 1 0 1 0; 1 0 1 1];
code = encode(msg,n,k,'cyclic');
msg
code
Syndrome decoding
clc;
close all;
q=3;
n=2^q-1;
k=n-q;
parmat = hammgen(q); % produce parity-check matrix
trt = syndtable(parmat); % produce decoding table recd
= [1 0 1 1 1 1 0 ] %received vector
syndrome = rem(recd * parmat',2);
syndrome_de = bi2de(syndrome, 'left-msb'); %convert to decimal
disp(['Syndrome = ',num2str(syndrome_de),.....
(decimal), ',num2str(syndrome),' (binary) ']); corrvect
= trt(1+syndrome_de, :);%correction vector
correctedcode= rem(corrvect+recd,2);
parmat corrvect
correctedcode

66
67
 OUTPUT

Enter the Generator Matrix: [1 0 0 0 1 0 1; 0 1 0 0 1 1 1; 0 0 1 0 1 1 0; 0 0 0 1 0 1 1] g =

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

G=
The Order of Linear block Code for given Generator Matrix is:
=7 k =4

The Possible Code words are:

c=

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

1 0 0 0 1 01
1 0 0 1 1 10
1 0 1 0 0 11
1 0 1 1 0 00
1 1 0 0 0 10
1 1 0 1 0 01
1 1 1 0 1 00
1 1 1 1 1 11

68
69
The Minimum Hamming Distance dmin for given Block Code is=

d_min = 3

Enter the Received Code Word:[1 0 0 0 1 0 0]

r=
1 0 0 0 1 0 0
Hamming Code
ht =

1 01
1 11
1 10
1 1
0 0
1 0
0 1
Syndrome of a Given Code word is :
s=
0 0 1
The Error is in bit:
7
The Corrected Code word is:
1 0 0 0 1 0 1

70
71
 OUTPUT

COMPUTATION OF CODE VECTORS FOR A CYCLIC CODE

Msg=

1001

1010

1011

Code =

1101001

0111010

0001011

SYNDROME DECODING

Recd=

1011110

Syndrome=7(decimal), 1 1 1(binary)

format=

1001011

0101110

0010111

Correct=

0000010

Corrected code=

1011100

72
RESULT:

73
CONVOLUTIONAL CODING

74
 13. SIMULATION OF CONVOLUTIONAL CODING SCHEME

AIM
To simulate convolutional coding scheme using MATLAB Simulink tool.

APPARATUS REQUIRED:
PC with MATLAB Software

ALGORITHM:

Simulate the link by following these steps:

Generate binary data.
Encode the data with a rate 2/3 convolutional code.
Modulate the encoded data.
Pass the signal through an AWGN channel.
Demodulate the received signal.
Decode the demodulated signal by using a Viterbi decoder.
Collect the error statistics
.
THEORY:
Generating Random Data

The Bernoulli Binary Generator block produces the information source for this
simulation. The block generates a frame of three random bits at each sample time. The Samples
per frame parameter determines the number of rows of the output frame.
Convolutional Encoding with Puncturing

The Convolutional Encoder block encodes the data from the Bernoulli Binary Generator.
This example uses the same code as described in Soft-Decision Decoding.
The puncture pattern is specified by the Puncture vector parameter in the mask. The
puncture vector is a binary column vector. A 1 indicates that the bit in the corresponding
position of the input vector is sent to the output vector, while a 0 indicates that the bit is
removed.

For example, to create a rate 3/4 code from the rate 1/2, constraint length 7
convolutional code, the optimal puncture vector is [1 1 0 1 1 0].' (where the .' after the vector
indicates the transpose). Bits in positions 1, 2, 4, and 5 are transmitted, while bits in positions 3
and 6 are removed. Now, for every 3 bits of input, the punctured code generates 4 bits of output
(as opposed to the 6 bits produced before puncturing). This makes the rate 3/4.In this example,
the output from the Bernoulli Binary Generator is a column vector of length 3. Because the rate
1/2 Convolutional Encoder doubles the length of each vector, the length of the puncture vector
must divide 6.

75
BIT ERROR RATE (BER)

76
 Transmitting Data

The AWGN Channel block simulates transmission over a noisy channel. The
parameters for the block are set in the mask as follows:
The Mode parameter for this block is set to Signal to noise ratio (Es/No).
The Es/No parameter is set to 2 dB. This value typically is changed from one simulation run to
next.
The preceding modulation block generates unit power signals so the Input signal power is set to
Watt.
The Symbol period is set to 0.75 seconds because the code has rate 3/4.
Demodulating
In this simulation, the Viterbi Decoder block is set to accept unquantized inputs. As a
result, the simulation passes the channel output through a Simulink® Complex to Real-Image
block that extracts the real part of the complex samples.
Viterbi Decoding of Punctured Codes
The Viterbi Decoder block is configured to decode the same rate 1/2 code specified in
the Convolutional Encoder block.
In this example, the decision type is set to Unquantized. For codes without puncturing,
you would normally set the Traceback depth for this code to a value close to 40. However,
for decoding punctured codes, a higher value is required to give the decoder enough data to
resolve the ambiguities introduced by the punctures.Since the punctured bits are not
transmitted, there is no information to indicate their values. As a result they are ignored in the
decoding process.
The Puncture vector parameter indicates the locations of the punctures or the bits to
ignore in the decoding process. Each 1 in the puncture vector indicates a transmitted bit while
each 0 indicates a puncture or the bit to ignore in the input to the decoder.In general, the two
Puncture vector parameters in the Convolutional Encoder and Viterbi Decoder must be the
same.
Calculating the Error Rate
The Error Rate Calculation block compares the decoded bits to the original source bits.
The output of the Error Rate Calculation block is a three-element vector containing the
calculated bit error rate (BER), the number of errors observed, and the number of bits
processed.
In the mask for this block, the Receive delay parameter is set to 96, because the
Traceback depth value of 96 in the Viterbi Decoder block creates a delay of 96. If there were
other blocks in the model that created delays, the Receive delay would equal the sum of all
the delays.
BER simulations typically run until a minimum number of errors have occurred, or
until the simulation processes a maximum number of bits. The Error Rate Calculation block
uses its Stop simulation mode to set these limits and to control the duration of the simulation.

RESULT:

77
AM MODULE

OUTPUT (AM)

78
 14. COMMUNICATION LINK SIMULATION

AIM

To generate modulation and demodulation of AM, ASK, DM and PAM using Simulink.

APPARATUS REQUIRED:

PC with MATLAB Software with Simulink tool

PROCEDURE:

1. Start simulink section

2. Select file New Model in the simulink library to construct a new model
Go to simulink library select appropriate module and add to model
Connect all the inserted models
Set the simulation parameters
Run the simulation and observe and save all the plots and values.

79
ASK MODULE

OUTPUT

80
DM MODULE

SIMULATION OUTPUT
PAM MODULE

SIMULATION OUTPUT

 
RESULT: