RAJALAKSHMI INSTITUTE OF TECHNOLOGY

CHENNAI 602107

DEPARTMENT
Of
ELECTRONICS AND COMMUNICATION ENGINEERING

EC 8561-COMMUNICATION SYSTEMS LABORATORY

INSTRUCTOR MANUAL
Regulation 2017
Academic Year: 2019-2020

Prepared By
Mrs.K.JAYAMANI, AP(SS), ECE
Mr.L.SARAVANAN, AP/ECE,
Ms.R.ARUNA JAYASREE, AP/ECE,

EC8561 COMMUNICATION SYSTEMS LABORATORY LTPC

0032
OBJECTIVES:
The student should be made to:
To visualize the effects of sampling and TDM.
To Implement AM & FM modulation and demodulation.
To implement PCM & DM.
To implement FSK, PSK and DPSK schemes.
To implement Equalization algorithms.
To implement Error control coding schemes.
LIST OF EXPERIMENTS:
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
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)
TOTAL: 45 PERIODS
OUTCOMES:
At the end of the course, the student should be able to:
CO1: Understand the basic concepts of analog modulation and to evaluate the performance of
various communication systems

CO2: Simulate end-to-end Communication Link.

CO3: Demonstrate their knowledge in base band signaling schemes through implementation of
FSK, PSK and DPSK.

CO4: Apply various channel coding schemes & demonstrate their capabilities towards the
improvement of the noise performance of communication system.
CO5: Simulate & validate the various functional modules of a communication system

EXPERIMENT MAPPING WITH COURSE OUTCOMES

List of Experiments as per Syllabus:

S. No Name Of the Experiment COURSE OUTCOME

CO1 CO2 CO3 CO4 CO5
Analog Communication Systems
1 Signal Sampling and Reconstruction * *
2 Time Division Multiplexing * *
3 AM Modulation and Demodulation * *
4 FM Modulation and Demodulation * *
Digital Communication Systems
5 Pulse Code Modulation and * *
Demodulation
6 Delta Modulation and Demodulation * *
Simulation
7 FSK,PSK and DPSK Schemes * * * *
8 Observation of signal constellations * * * *
of BPSK,QPSK and QAM
9 Error Control Coding Schemes * * * *
10 Equalization-Zero Forcing & LMS
Algorithms
List of Experiments beyond Lab course syllabus:

S.No Name Of the Experiment COURSE OUTCOME

CO1 CO2 CO3 CO4 CO5
Analog Communication Systems
2 AM Modulation using IC2206 *
 BRIDGING THE CURRICULUM GAP
Course outcomes CO1-CO4 is satisfied by Anna university syllabus. To bridge the gap
between Anna University and IIT , experiments like ASK, PWM and PAM are implemented
using hardware and Linear delta modulation and OFDM spectrum simulated using Matlab
and Experiments like FM radio receiver, ASK, FSK and BPSK using Software defined Radio
are included from NIT.
- Course Instructors

INDEX
 S.NO LIST OF EXPERIMENTS PAGE
NO.
1. AM Modulation and Demodulation

2. FM Modulation and Demodulation

3. Sampling and Reconstruction

4. Time Division Multiplexing

Pulse Code Modulation
5.
Delta and Adaptive Delta Modulation
6.

8. Simulation of Digital Modulation Techniques-

ASK,FSK,PSK,QPSK,DPSK

9. Signal Constellation of BPSK, QPSK & QAM

10. Error Control Coding using MATLAB

11. Simulation of Equalization Techniques

12. Content Beyond Syllabus

CIRCUIT DIAGRAM

AMPLITUDE MODULATION
DEMODULATION

EXPT.NO.1 AM MODULATION AND DEMODULATION

AIM:

To construct amplitude modulator and demodulator circuit and plot the waveforms.

COMPONENTS REQUIRED:
 THEORY :

Modulation can be defined as the process by which the characteristics of carrier wave
are varied in accordance with the modulating wave (signal). Modulation is performed in a

S.NO. NAME OF THE EQUIPMENT / RANGE QUANTITY

COMPONENT
1 Transistor BC 107 1
2 Diode 1N4001 1
3 Capacitors 0.1µF, 0.01µF 2,1
4 Resistors 100K,22K,500Ω,20 2, 1each
0K,10Ω
5 Decade Inductance Box 10 mH 1
6 Function Generators 1 MHz 2
7 CRO 20MHz 1
8 Bread board - 1
9 Regulated Power supply 0-30V 1

transmitter by a circuit called a modulator.

Need for modulation is as follows:
 Avoid mixing of signals
 Reduction in antenna height
 long distance communication
 Multiplexing
 Improve the quality of reception
 Ease of radiation
MODEL GRAPH:

Amplitude Modulation is the process of changing the amplitude of a relatively highfr

equency carrier signal in proportion with the instantaneous value of the modulating signal.
 The output waveform contains all the frequencies that make up the AM signal and is used totr

ansport the information through the system. Therefore the shape of the modulated wave is

called the AM envelope. With no modulating signal the output waveform is simply the carrier

signal. Coefficient of modulation is a term used to describe the amount of amplitude change

present in an AM waveform. There are three degrees of modulation available based on value

of modulation index.

1) Under modulation : m<1, Em < Ec

2) Critical modulation: m-1, Em = Ec

3) Over modulation: m>1, Em > Ec

Demodulation is the reverse process of modulation and converts the modulated carrier
back to the original information. Demodulation is performed in a carrier by a circuit called a
demodulator.

ADVANTAGES:

1) Relatively inexpensive
2) Low quality form of modulation

DISADVANTAGES:

1) Low efficiency
2) Small operating range

APPLICATION:

1) Commercial broadcasting of both audio and video signals

2) Two way mobile radio communication such as citizen band (CB) radio.

PROCEDURE:
1. Rig up the circuit as per the circuit diagram.
2. Set the carrier signal using function generator and measure the amplitude and time
period.
3. Set the modulating signal and measure the amplitude and time period.
4. Vary the amplitude around the carrier voltage.
TABULATION:

Time period
Signals Amplitude (V) Frequency (KHz)
(ms)

Modulating signal

Carrier signal

INPUT SIGNAL:

MODULATED SIGNAL:

DETECTED SIGNAL:
 Emax Emin Type of
m = (Emax – Emin)/ (Emax + Emin) %
(V) (V) modulation

5.
Not e do
wn the maximum (Emax) and minimum (Emin) voltages from the CRO.
6. Calculate the modulation index using the formula.
7. Apply the AM signal to the detector circuit.
8. O bserve the ampl
itude demod Amplitude (V) Time period (ms) Frequency (KHz) ulated output o
n the CRO.
9. C ompare the de
modulated si gnal with the o
riginal modu lating signal (B
oth must
be same in all parameters). Plot the observed waveforms.
CONCLUSION:
Thus the characteristics of AM Transmitter and Receiver are studied and the
waveforms are observed and plotted.

Questions
1. Define Amplitude Modulation.
AM is a process in which the amplitude of the carrier wave is varied in accordance with
some characteristics of the modulating signal.
2. What is the need for Modulation?
a) Difficult in transmitting signals at low frequencies.
b) To minimize signal loss.
 c) To reduce antenna length.
3. What are the applications of AM?
Amplitude modulation is utilized in many services such as television, standard
broadcasting, aids to navigation, telemeter, radar, facsimile etc.
4. What are the different types of AM?
Single Side Band, Double Side Band and Vestigial Side Band Modulation are the
different types of AM.
5. What are the disadvantages of AM?
Low efficiency, small operating range, noisy reception.
 BLOCK DIAGRAM

FM MODULATOR AND DEMODULATOR

EXPT.NO:2 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.

NAME OF THE EQUIPMENT /

S.NO. RANGE QUANTITY
COMPONENT

1 FM Transmitter and receiver kit 1

2 CRO 20 MHz 1

COMPONENTS 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 DEVIATION f and MODULATION INDEX m f :

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

fmax  fmin
f 
2

We define modulation index m f the ratio between f and the modulating frequency f.
m f  f
f
MODEL GRAPH:
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.

ADVANTAGES:
1. Noise reduction
2. Improved system fidelity
3. Efficient use of power

DISADVANTAGE:
1. Requires a wider bandwidth
2. Utilizing more complex circuit in both transmitters and receivers.
APPLICATION:
1. Television sound transmission
2. Two way mobile radio
3. Cellular Radio
4. Microwave
5. Satellite Communication System

TABULATION:
 Signals Amplitude (V) Time period (ms) Frequency(KHz)

Modulating signal

Carrier signal

Tmin= fmax=
Modulated signal
Tmax= fmin=

Demodulated signal

PROCEDURE:
 i) Connect the power supply with proper polarity to the kit. While connecting this
ensures that the Power supply is OFF.
ii) Switch on the power supply and carry out the following presetting as shown in
circuit Diagram.
iii) In the FM modulator set the level about 2Vpp and frequency knob to the
minimum and switch on 1500 KHz.
iv) Observe the Fm modulated waveform from the RF/FM output of the FM
modulator measure frequency deviation and modulation index of FM.
v) For demodulation switch on the demodulator and carry out the following
demodulation connection as shown in circuit diagram.
vi) Observe the demodulated waveform and plot the graph.

CONCLUSION:
Thus the modulation characteristics of FM modulator and demodulator are observed
and plotted.

Questions:
1. What is Frequency Modulation?
Frequency modulation (FM) is a technique in which the frequency of the carrier wave is
varied in accordance with the amplitude of the message signal.
3. What is the frequency band for FM radio?
The frequency band for FM radio is about 88 to 108 MHz.
4. What is the bandwidth of FM signal?
Bandwidth of a FM signal may be predicted using:
BW = 2 ( + 1 ) fm; where  is the modulation index and fm is the maximum modulating
frequency used.
5.What are the disadvantages of FM?
Compared to AM, the FM signal has a larger bandwidth
6. What are the disadvantages of FM?
High efficiency and better immunity to noise.
 WORKSHEET
BLOCK DIAGRAM

CIRCUIT DIAGRAM: (USING MOSFET)

 Signal Sampling and Reconstruction
Experiment: 03 Date:

OBJECTIVE

To study the process of sampling and to reconstruct the signals at the receiver using
filters.

REQUIREMENTS

S.No Requirements Name &Range Quantity

1 Equipments Sampling and Reconstruction kit

DSO(0-25MHZ), Each one

Power Supply
2 Accessories Patch cords As Required

THEORY
Sampling is a process by which an analog signal is converted in to corresponding square of
samples that spread uniformly in time. PAM is where the analog signal after sampling takes on the
form of a sequence of pulses Ts seconds apart. Each pulse carries information about the analog
signal’s amplitude at the time the pulse was generated. There are two variants of PAM. The first,
currently the dominant approach, called Natural Sampling is where the shape of each pulse is
affected by the changing input during the pulse. The second is called Uniform Sampling (mainly used
in the older laboratory grade samplers) where all of the pulses have the same shape, but different
amplitudes. Both types of sampling are practical. One, Natural Sampling, is simpler in the frequency
domain while the other, Uniform Sampling, is simpler in the time domain. In the limiting case, as the
pulse width approaches zero while holding the pulse area constant, both systems yield the same
CONCLUSION. This limiting case is called Ideal (or Impulse) Sampling where each pulse is now the

“Impulse” or “Dirac Delta” function,   t  n * Ts  .

The second type of sampling, Uniform Sampling, CONCLUSIONs in a train of pulses in which
each pulse has the same shape. Each pulse has a “strength” which is set by the value of the input at
the sampling “instant”. This method also goes by the name “Flat Top Sampling” when the
transmitted pulses are rectangular. One way to implement uniform sampling is to precede a natural
sampler with a “Sample and Hold” circuit.

TABULAR COLUMN
MODULATING SIGNAL:

Amplitude (V) Time period (ms) Frequency (KHz)

SAMPLED SIGNAL:

Time period (ms) Total

Sampling
Amplitude Duty No. of (for each sample) Time Frequency
frequency
(V) Cycle (%) Samples period (KHz)
(KHZ) Ton Toff
(ms)

RECONSTRUCTED SIGNAL:

Duty cycle c Amplitude (V) Time period (ms) Frequency (KHz) alculation:

D = Ton / (Ton + T off) = ---------- %

ADVANTAGES:

It can store retrieve and transmit signals without any loss
With higher sampling rate they can relax low pass filter design requirements for
ADC and DAC
PROCEDURE:
1. Give the connections as per the block diagram.
2. Apply the modulating signal and measure its amplitude and time period.
3. Set the sampling frequency to 80 KHz and note down the amplitude and time
period of the sampled signal.
4. Give the sampled signal to the reconstruction circuit and observe the reconstructed
signal.
5. Note down the amplitude and time period of the reconstructed signal.
6. Repeat the same procedure for different sampling frequencies.
7. Plot the above waveforms in the graph.

CONCLUSION:
Thus the given signal is sampled with different sampling frequencies and the
waveforms are plotted.
Questions:
1. What is aliasing effect?
A. The original analog waveform can be recovered from the PAM type samples simply by
low pass filtering them If fs <fnyquist (2fm) then overlapping of adjacent spectrum
replicates occurs. This is known as aliasing .Due to under- sampling (for f s<2fm) exact
analog waveform cannot be recovered,
2. What is the function of Op-amps in this circuit and what is the effect of frequency of
sampling signal?
A. Op-amps acts as voltage followers, if the f s<2fm , then distorted waveform is
Observed, so to recover the exact signal the sampling signal frequency should be
maintained greater than or equal to the 2fm.
3. What are the different types of sampling?
A. Instantaneous sampling, Natural sampling and Flat top sampling.
4. State Sampling Theorem.
A. The sampling theorem for a band limited signal of finite energy can be stated as,” A
Band limited signal of finite energy, which has no frequency component higher than W
Hz is completely described by specifying the values of the signal at instants of time

MODEL GRAPH
BLOCK DIAGRAM
 TIME DIVISION MULTIPLEXING

Experiment: 4 Date:

OBJECTIVE
To perform four channel Time Division multiplexing and De multiplexing.

REQUIREMENTS

S.No Requirements Name &Range Quantity

1 Equipments TDM Kit

DSO(0-25MHZ), Each one

Power Supply
2 Accessories Patch cords As Required

THEORY
Time-division multiplexing (TDM) is a type of digital (or rarely analog) multiplexing in which
two or more bit streams or signals are transferred apparently simultaneously as sub-channels in one
communication channel, but are physically taking turns on the channel. The time domain is divided
into several recurrent timeslots of fixed length, one for each sub-channel. A sample byte or data
block of sub-channel 1 is transmitted during timeslot 1, sub-channel 2 during timeslot 2, etc. One
TDM frame consists of one timeslot per sub-channel plus a synchronization channel and sometimes
error correction channel before the synchronization.
The sampling theorem provides the basis for transmitting the information contained into a
band limited message signal on a sequence of samples taken uniformly at a rate slightly higher that
then nyquist rate. An important feature of it is conversion of time i.e., Td of message signal engages
the communication channel for only a fraction of the sampling interval a periodic basis and in this
way same of the interval between adjacent samples is closed for use by other independent message
sources without mutual interference from other.
TABULAR COLUMN

1. TRANSMITTED SIGNALS: 3. RECEIVED SIGNALS:

Time
Amplitude Frequency
Channel period
(V) (KHz)
(ms)
2. SAMPLED(TDM) SIGNAL

Time period (ms)

Amplitude No.of Total Time Frequency
Channel (for each sample)
(V) Samples period(ms) (KHz)
Ton Toff

PROCEDURE
1. Connections are given as per block diagram.
2. The transmitted and received clocks are synchronized by interconnecting the two sides.
3. Now a minimum of two input messages of different frequencies on transmitter side are
multiplexed.
4. On receives side, signals are made available at different channel

CONCLUSION:

Thus the Time division multiplexing and demultiplexing waveforms are obtained.
Questions
1. What is multiplexing?
 A. It is a process in which a single transmission channel is shared by a number of
base band signals.
2. What is TDM?
A. In TDM, different time intervals rather than frequencies are allotted to different
signals. During these intervals these signals are sampled and transmitted. Thus, this
system transmits information intermittently rather than continuously.
3. What are the advantages of TDM?
A. a)Low cost equipment b) Ease of installation and maintenance c) Low and constant
delay d) unsurpassed voice quality and e) standards based.
4. What are the applications of TDM?
A. In telecommunications and signal processing applications

BLOCK DIAGRAM:

MODEL DIAGRAM
 PULSE CODE MODULATION

Experiment: 05 Date:

OBJECTIVE
To generate a PCM signal using PCM modulator and detect the message signal from PCM
signal by using PCM demodulator.

REQUIREMENTS

S.No Requirements Name &Range Quantity

1 Equipments Pulse code transmitter and receiver

DSO(0-25MHZ), Each one

Power Supply
2 Accessories Patch cords As Required

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 Amplitude (V) Time period (ms) Frequency (KHz) 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 through 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.

TABULATION:

TRANSMITTED SIGNAL:

SAMPLED SIGNAL:
 No. of Time period (ms) Total Time Frequency
samples (for each sample) Period (KHz)
Channel Amplitude(V)
(ms)
Ton Toff

RECEIVED SIGNAL:

PCM OUTPUT:

Amplitude (V) Time period (ms) Frequency (KHz)

DC Voltage Encoded values

(V) D6 D5 D4 D3 D2 D1 D0
-5
-4
-3
-2
-1
0
1
2
3
4
5
ADVANTAGES:
1. Secrecy
2. Noise resistant and hence free from channel interference
DISADVANTAGES:
1. Requires more bandwidth

APPLICATION:
1. Compact DISC for storage
2. Military Applications.

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 in the channel can be seen in 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.

VIVA QUESTIONS:

1. What is the expression for transmission bandwidth in a PCM system?

2. What is the expression for quantization noise /error in PCM system?

3. What are the applications of PCM?

4. What are the advantages of the PCM?

5. What are the disadvantages of PCM?

CONCLUSION

Thus, the pulse code modulation was studied and the output waveforms were observed and
plotted.
BLOCK DIAGRAM

MODEL GRAPH:
 DELTA MODULATION AND DEMODULATION

Experiment: 06 Date:

OBJECTIVE
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.
REQUIREMENTS

S.No Requirements Name &Range Quantity

1 Equipments Delta modulation and demodulation

kit
Each one
DSO(0-25MHZ),
Power Supply
2 Accessories Patch cords As Required

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
TABULAR COLUMN

AMPLITUDE (V) TIME PERIOD (ms) FREQUENCY

(HZ)

Input Signal

Integrator 1 output

Sampler output

Integrator 3 output

Filter output

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.
5. Calculate the sampling frequency required for no slope overload distortion. Compare the
calculated and measured values of the sampling frequency.

VIVA QUESTIONS:

1. What are the advantages of Delta modulator?

2. What are the disadvantages of delta modulator?

3. How to overcome slope overload distortion?

4. How to overcome Granular or ideal noise?

5. What are the differences between PCM & DM?

6. Define about slope over load distortion?

7. What is the other name of Granular noise?

8. What is meant by staircase approximation?

9. What are the disadvantages of Delta modulator?

10. Write the equation for error at present sample?

CONCLUSION
 Thus the delta modulations are demodulation was constructed and desired signal has been
modulated and demodulated.

Amplitude shift keying [ASK]

 SHIFT KEYING TECHNIQUES
Experiment: 07 Date:
OBJECTIVE
To study the various shift keying techniques
i) Amplitude shift keying [ASK]
ii) Frequency shift keying [FSK]
iii) Phase shift keying [PSK]

S.No Requirements Name &Range Quantity

1 Equipments Transmitter (Modulator) kit

Receiver (demodulator) kit.
Each one
DSO(0-25MHZ),
Power Supply
2 Accessories Patch cords As Required

THEORY
The modulation process making the transmission possible involves switching (keying) the
amplitude, phase frequency of a sinusoidal carrier in same fashion as incoming data. Thus, these are
the basic signaling schemes and respectively called ASK, FSK and PSK.
Amplitude-shift keying (ASK) is a form of modulation that represents digital data as variations
in the amplitude of a carrier wave. The amplitude of an analog carrier signal varies in accordance
with the bit stream (modulating signal), keeping frequency and phase constant. The level of
amplitude can be used to represent binary logic 0s and 1s. We can think of a carrier signal as an ON
or OFF switch. In the modulated signal, logic 0 is represented by the absence of a carrier, thus giving
OFF/ON keying operation and hence the name given.
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. The time domain of an FSK modulated carrier is illustrated in the figures to the right.
Phase-shift keying (PSK) is a digital modulation scheme that conveys data by changing, or modulating,
the phase of a reference signal (the carrier wave). Any digital modulation scheme uses
a finite number of distinct signals to represent digital data.
ASK PROGRAM:
clear;
clc;
close;
f =2;
t = 0:1/152:1;
x = sin (2* %pi *f *t);
I = input ("Enter the digital binary data");
Xask = [];
for n = 1: length (I )
if (( I ( n ) ==1)&( n ==1) )
Xask = [x , Xask ];
elseif (( I ( n ) ==0) &( n ==1) )
Xask = [ zeros (1 , length ( x ) ) , Xask ];
elseif (( I ( n ) ==1)&( n ~=1) )
Xask = [ Xask , x ];
elseif (( I ( n ) ==0)&( n ~=1) )
Xask = [ Xask , zeros (1 , length ( x )) ];
end
end
figure
plot(t, x)
xtitle("Analog Carrier Signal for Digital Modulation");
xgrid
figure
plot (Xask)
xtitle ("Amplitude Shift Keying");
xgrid
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. This requires the receiver to be able to compare the phase of the
received signal to a reference signal — such a system is termed coherent (and referred to as CPSK).
FSK WAVEFORM:
SOFTWARE IMPLEMENTATION

FSK PROGRAM:
clear;
clc;
close;
f =2;
t = 0:1/152:1;
x = sin (2* %pi *f *t);
I = input ("Enter the digital binary data");
Xfsk = [];
x1 =sin(2* %pi * f *t);
x2 =sin(2* %pi *(2* f) * t);
for n = 1: length (I)
if ( I ( n ) ==1)
Xfsk = [Xfsk, x2];
elseif(I(n)~=1)
Xfsk = [Xfsk, x1];
end
end
figure
plot(t, x)
xtitle("Analog Carrier Signal for Digital Modulation");
xgrid
figure
plot (Xfsk)
xtitle ("Frequency Shift Keying");
xgrid
VIVA QUESTIONS-ASK
1. What is the difference between PSK&ASK?
2. What is the band width requirement of a ASK?
3. Explain the operation of ASK detection?
4. What are the advantages of APSK?
5. What is meant by ASK?

VIVA QUESTIONS-FSK
1. Define Binary FSK signal?
2. What is meant by carrier swing?
3. Define Frequency deviation of FSK signal?
4. What are the advantages of this FSK signal?
5. Give the differences between FSK & FM?

VIVA QUESTIONS-BPSK
1. What is the bandwidth requirement of BPSK?
2. What is the expression for error probability of BPSK reception?
3. What are the draw backs of BPSK?
4. Draw the Power spectral density of BPSK?
5. What are the major differences between DPSK&BPSK?
6. What are the advantages of BPSK over a PSK signal?
PSK PROGRAM:
clear;
clc;
close;
f =2;
t = 0:1/152:1;
x = sin (2* %pi *f *t);
I = input ("Enter the digital binary data");
Xpsk = [];
x1 = sin (2* %pi * f *t ) ;
x2 = -sin (2* %pi * f* t ) ;
for n = 1: length (I )
if ( I ( n ) ==1)
Xpsk = [ Xpsk , x1 ];
elseif ( I ( n ) ~=1)
Xpsk = [ Xpsk , x2 ];
end
end
figure
plot(t, x)
xtitle("Analog Carrier Signal for Digital Modulation");
xgrid
figure
plot (Xpsk)
xtitle ("Phase Shift Keying");
xgrid
CONCLUSION

Thus, the various techniques for shift keying such as ASK, FSK, PSK are constructed and output
waveforms were plotted.

OBSERVATION OF SIGNAL CONSTELLATIONS

Experiment: 09 Date:

OBJECTIVE
To observe the signal constellations of
i) Binary Phase shift keying [BPSK]
ii) Quadrature Phase shift keying [QPSK]
iii) Quadrature Amplitude Shift Keying [QAM]

REQUIREMENTS

i) PC
ii) MATLAB 2014

THEORY
A constellation diagram is a representation of a signal modulated by a digital modulation scheme
such as quadrature amplitude modulation or phase-shift keying. It displays the signal as a two-
dimensional X-Y plane scatter diagram in the complex plane at symbol sampling instants. In a more
abstract sense, it represents the possible symbols that may be selected by a given modulation
scheme as points in the complex plane. Measured constellation diagrams can be used to recognize
the type of interference and distortion in a signal.

Fig 1: A constellation diagram for Gray encoded 8-PSK.

By representing a transmitted symbol as a complex number and modulating a cosine and sine carrier
signal with the real and imaginary parts (respectively), the symbol can be sent with two carriers on
the same frequency. They are often referred to as quadrature carriers. A coherent detector is able to
independently demodulate these carriers. This principle of using two independently modulated
carriers is the foundation of quadrature modulation. In pure phase modulation, the phase of the
modulating symbol is the phase of the carrier itself and this is the best representation of the
modulated signal.
As the symbols are represented as complex numbers, they can be visualized as points on the complex
plane. The real and imaginary axes are often called the in phase, or I-axis, and the quadrature, or Q-
axis, respectively. Plotting several symbols in a scatter diagram produces the constellation diagram.
The points on a constellation diagram are called constellation points. They are a set of modulation
symbols which comprise the modulation alphabet.
Also a diagram of the ideal positions, signal space diagram, in a modulation scheme can be called a
constellation diagram. In this sense the constellation is not a scatter diagram but a representation of
the scheme itself. The example shown here is for 8-PSK, which has also been given a Gray coded bit
assignment.
Interpretation

Fig 2:A constellation diagram for rectangular 16-QAM.

Upon reception of the signal, the demodulator examines the received symbol, which may have been
corrupted by the channel or the receiver (e.g. additive white Gaussian noise, distortion, phase
noise or interference). It selects, as its estimate of what was actually transmitted, that point on the
constellation diagram which is closest (in a Euclidean distance sense) to that of the received symbol.
Thus it will demodulate incorrectly if the corruption has caused the received symbol to move closer
to another constellation point than the one transmitted. This is maximum likelihood detection. The
constellation diagram allows a straightforward visualization of this process — imagines the received
symbol as an arbitrary point in the I-Q plane and then decides that the transmitted symbol is
whichever constellation point is closest to it. For the purpose of analyzing received signal quality,
some types of corruption are very evident in the constellation diagram. For example:
 Gaussian noise shows as fuzzy constellation points
 Non-coherent single frequency interference shows as circular constellation points
 Phase noise shows as rotationally spreading constellation points
 Attenuation causes the corner points to move towards the center
A constellation diagram visualizes phenomena similar to those an eye pattern does for one-
dimensional signals. The eye pattern can be used to see timing jitter in one dimension of modulation.
BPSK

Fig3:Signal Constellation of BPSK

QPSK

M File Script
 Fig4:Signal Constellation of QPSK

QAM:

Fig4:Signal Constellation of QAM

VIVA QUESTIONS

1. Identify which of the following signal constellations given below are required to construct a 16 –
level QAM signal constellation.

a) 2 octal phase shift keying systems with different radius

b) 2 QPSK systems with different energies
c) 3 QPSK systems with different energies
d) 1 octal phase shift keying and 2 QPSK systems with different energies.

2. ASK, PSK, FSK, and QAM are examples of ________ conversion.

3.A constellation diagram shows us the __________ of a signal element, particularly when we are
using two carriers (one in-phase and one quadrature).

4. Quadrature amplitude modulation (QAM) is a combinati on of ___________.

5.The constellation diagram of QPSK has ______ dots.

Conclusion:

The signal constellations are observed by using scatter plots. Depends on the selected
modulation scheme, the number of signal points varied in XY plane.
 ERROR CONTROL CODING
Experiment: 11 Date:

OBJECTIVE
a. To generate parity check matrix & generator matrix for a (7,4) Hamming code.
b. To generate parity check matrix given generator polynomial g(x) = 1+x+x3.
c. To determine the code vectors.
d. To perform syndrome decoding

REQUIREMENTS

1. PC with Win XP
2. Sci lab 6.2 or above

THEORY
In information theory and coding theory with applications in computer
science and telecommunication, error detection and correction or error control are techniques that
enable reliable delivery of digital data over unreliable communication channels. Many
communication channels are subject to channel noise, and thus errors may be introduced during
transmission from the source to a receiver. Error detection techniques allow detecting such errors,
while error correction enables reconstruction of the original data. Error detection is the detection of
errors caused by noise or other impairments during transmission from the transmitter to the
receiver. Summer is another name for error detection Error correction is the detection of errors and
reconstruction of the original, error-free data.
Linear codes are used in forward error correction and are applied in methods for transmitting
symbols (e.g., bits) on communications channel so that, if errors occur in the communication, some
errors can be corrected or detected by the recipient of a message block. The codeword in a linear
block code are blocks of symbols which are encoded using more symbols than the original value to
be sent. A linear code of length n transmits blocks containing n symbols. For example, the [7, 4,
3] Hamming code is a linear binary code which represents 4-bit messages using 7-bit codewords. Two
distinct codewords differ in at least three bits. As a consequence, up to two errors per codeword can
be detected while a single error can be corrected. This code contains 24=16 codewords.
Implementing one such error correction codes is done as Mat lab simulation below.
The generated matrix ‘K’ by ‘n’ is given by G= [P:Ik] where
‘P’ is the coefficient matrix ‘K’ by ‘n-k’
‘IK’ is the ‘K’ by ‘K’ identity matrix
Codeword X= Mg where
‘Mk’ is ‘1’ by ‘k’ message vector
‘H’ is denoted by ‘n-k’ by ‘n’ matrix
PROGRAM:
ALGORITHM

1. Generate the message sequence of length ‘k

2. Generate the coefficient matrix “p” of order ‘k’ by ‘n-k’
3. Generate identify matrix and multiply with ‘P’ matrix to from generated matrix.
4. Linear block code is calculated [#] * [G]
5. Obtain the parity check matrix [P]
6. Add the noise to code word
[r] = [c] + [P]
7. Construct the syndrome table
8. Decode the desired word message from code word.

VIVA QUESTIONS

1. What is Error Control Coding?

2. What are linear block codes?
3. What are cyclic codes?
4. What is Parity check matrix?
5. What do you mean by generator polynomial?

CONCLUSION

Thus, the program to implement (7,4) linear block code using MATLA 6.5 was executed and
output was verified successfully.
PROGRAM
clc;clear all;close all;
M=4;
msg=randint(1500,1,M);
modmsg=pskmod(msg,M);
sigconst=pskmod([0:M-1],M);
trainlen=500;
chan=[.986;.845;.237;.123+.31i];
filtmsg=filter(chan,1,modmsg);
eqobj =lineareq(8,lms(0.01),sigconst,1);
[symbolest,yd]=equalize(eqobj,filtmsg,modmsg(1:trainlen));
h=scatterplot(filtmsg,1,trainlen,'bx');hold on;
scatterplot(symbolest,1,trainlen,'r.',h);
scatterplot(sigconst,1,0,'k*',h);
legend('fitered signal','equalized signal','ideal signal constellation');
hold off;
demodmsg_noeq=pskdemod(filtmsg,M);
demodmsg =pskdemod(yd,M);
[nnoeq,rnoeq]=symerr(demodmsg_noeq(trainlen+1:end),msg(trainlen+1:end));
[neq,req] = symerr(demodmsg(trainlen+1:end),msg(trainlen+1:end));
disp('symbol error rate with equalizer:');
disp(req);
disp('symbol error rate without equalizer:');
disp(rnoeq)
SIMULATED OUTPUT:

Symbol error rate with equalizer: 0

Symbol error rate without equalizer: 0.3310

 SIMULATION OF EQUALIZATION TECHNIQUES

Experiment: 12 Date:
1. Simulation of Zero Forcing Equalizer.
AIM:
To simulate the Zero Forcing Equalizer using MATLAB.
SOFTWARE USED: MATLAB
THEORY:
Equalizer can be employed to mitigate the ISI for a smooth recovery of transmitted
symbols and to improve the receiver performance Zero forcing (or) linear equalizer
which processes the incoming signal with a linear filter. It is classified into two
(a) Symbol spaced equalizer
(b) Fractionally spaced equalizer
Symbol spaced equalizer:
A symbol spaced linear equalizer consist of a tapped delay line that stores samples from the input
signal. Here the sample rates of both input & output signals are equal to 1/T.
Fractionally spaced equalizer: A Fractionally spaced linear equalizer is similar to a
symbol spaced equalizer,but the former receives K input samples before it produces one
output sample & updates the weights, where K is an integer. Here the output sample
rates is 1/T,while that of input sample is K/T.

2. Equalization using LMS Algorithm

AIM:
To simulate Least Mean Square (LMS) algorithm to adaptively adjust the coefficients of
an FIR filter.

SOFTWARE USED:
MATLAB

THEORY:
The LMS recursive algorithm used for adjusting the filter coefficients adaptively so as to
minimize the sum of squared error is described below.
Let x[n] be the input sequence and y[n] be the output sequence of an FIR filter. Then,the
output is given by the expression

Y[n]=∑ h[k]x[n-k], n=0,1,……M

Where h[n] is the adjustable coefficients of FIR filter.
Let the desired sequence be d[n].Then, the error sequence e[n] is given by
 e[n] = d[n] – y[n] , n=0,1,……M
The LMS algorithm starts with any arbitrary choice of h[k],say h0[k].For example, we
may begin with h0[k]=0,0 ≤ k ≤ N-1.After that each new sample x[n] enters the adaptive filter
,we compute the corresponding output, say y[n], form the error signal e[n]=d[n]-y[n],and update
PROGRAM:
clc;clear all;close all;
N=input('enter the system order,N=');
M=input('enter the number of iterations,M=');
if((N>=2)&&(M>=2))
x=rand(M,1);
b=fir1(N-1,0.5);
n=0.1*randn(M,1);
d=filter(b,1,x)+n;
h=zeros(N,1);
Px=(1/length(x))*sum(x.^2);
mu=1/(5*N*Px);
for n=N:M
u=x(n:-1:n-N+1);
y(n)=h'*u;
e(n)=d(n)-y(n);
h=h+mu*u*e(n);
end
hold on;plot(d,'g');
plot(y(),'r');
semilogy((abs(e())),'m');
title('system output');
xlabel('number of iterations');
ylabel('true and estimated output');
legend('desired','output','error');
hold off;
figure,plot(b','k+');
hold on,plot(h,'r*');
legend('actual weights','estimated weights');
hold off;
title('comparison of actual weights and estimated weights');
else('system order and number of iterations should be greater than 1');
end
SIMULATED OUTPUT:

Enter the system order,N=5

Enter the number of iterations,M=200
the filter coefficients according to the equation hn[k] = hn-1[k] +µ.e[n].x[n-k], 0 ≤ k ≤ N-
1,n=0,1…..where µ is called step size parameter, x[n-k] is the sample of input signal located
at the kth tap of the filter at time n and e[n]x[n-k] is an pproximation(estimate) of the negative
of the gradient for the kth filter coefficients.
The step size parameter µ controls the rate of convergence. Large value of µ leads to
rapid convergence and smaller value leads to slower convergence. If µ is made too large,the
algorithm becomes unstable.
In order to ensure convergence and good tracking capabilities in slowly varying channels,
the step size parameters is given by µ=1/5NPx where N is the length of the adaptive FIR filter
and Px is the average power in the input signal which is approximated by

SIMULATED OUTPUT:

Enter the system order,N=5

Enter the number of iterations,M=200
SIMULATED OUTPUT:

comparison of actual weights and estimated weights

 EC6512 COMMUNICATION SYSTEMS LABORATORY

CONCLUSION:

Thus the Zero Forcing Equalizer Least Mean Square (LMS) algorithm is simulated in MATLAB .

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 EC6512 COMMUNICATION SYSTEMS LABORATORY

CIRCUIT DIAGRAM:

DEMODULATOR

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 EC6512 COMMUNICATION SYSTEMS LABORATORY

CONTENT BEYOND SYLLABUS

EXPT.NO.1 AM MODULATION AND DEMODULATION USING IC2206

AIM:

To construct amplitude modulator and demodulator circuit and plot the waveforms.

COMPONENTS REQUIRED:

NAME OF THE EQUIPMENT /

S.NO. RANGE QUANTITY
COMPONENT
1
IC2206 1

2 Resistors 47K,1K,10K, 3,1,1,1

220Ω
3. Capacitors 0.01µF,0.1µF 1,2

THEORY:
MODULATOR:
An amplitude modulated signal is composed of both low frequency and high frequencycomp
onents. The amplitude of the high frequency (Carrier) of the signal is controlled by the lowfrequency (
modulating) signal. The envelope of the signal is created by the low frequency signal.If the modulating
signal is sinusoidal, then the envelope of the modulated radio frequency (RF)signal will also be sinu
soidal. The circuit for generating an AM modulated waveform must
produce the product the of the carrier and the modulating signal.

Semester 05 Department of ECE Rajalakshmi Institute of Technology

 EC6512 COMMUNICATION SYSTEMS LABORATORY

TABULATION:
MODULATING SIGNAL

Time period
Signals Amplitude (V) Frequency (KHz)
(ms)

Modulating signal

Carrier signal

MODULATED SIGNAL

Emax (V) Emin (V) m = (Emax – Emin) / (Emax + Emin) %

DETECTED SIGNAL

Amplitude
Frequency (KHz) Time period (ms)
(V)

Semester 05 Department of ECE Rajalakshmi Institute of Technology

 EC6512 COMMUNICATION SYSTEMS LABORATORY

DEMODULATOR:
A single diode can be used to detect the AM signal and is called PN diode detector orenvel
ope detector. The diode acts as a rectifier in removing half the envelope resulting in thebase band s
ignal with a Dc offset. The offset is removed with a series capacitor, producing theoutput.

Envelope detectors are not perfect. All diodes are nonlinear, and will distort the envelopewhen
it is near the zero voltage level. This effect can be minimized by using a diode with a lowforward volt
age drop and a strong signal(several 100mV) at the detector.

PROCEDURE:
1. Rig up the circuit as per the circuit diagram.
2. Set the carrier signal to 8V, 10 KHz using function generator and measure the
amplitude and time period.

3. Set the modulating signal 4V,1 KHz and measure the amplitude and time period.

4. Vary the amplitude around the carrier voltage.

5. Note down the maximum (Emax) and minimum (Emin) voltages from the CRO.

6. Calculate the modulation index using the formula.

7. Apply the AM signal to the detector circuit.

8. Observe the amplitude demodulated output on the CRO.

9. Compare the demodulated signal with the original modulating signal (Both must
be same in all parameters). Plot the observed waveforms.

Semester 05 Department of ECE Rajalakshmi Institute of Technology

 EC6512 COMMUNICATION SYSTEMS LABORATORY

MODEL GRAPH:

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 EC6512 COMMUNICATION SYSTEMS LABORATORY

CONCULSION:

Semester 05 Department of ECE Rajalakshmi Institute of Technology

 EC6512 COMMUNICATION SYSTEMS LABORATORY

Thus the characteristics of AM Transmitter and Receiver are studied and the waveforms
are observed and plotted.

Semester 05 Department of ECE Rajalakshmi Institute of Technology

 EC6512 COMMUNICATION SYSTEMS LABORATORY

Semester 05 Department of ECE Rajalakshmi Institute of Technology

 EC6512 COMMUNICATION SYSTEMS LABORATORY

Semester 05 Department of ECE Rajalakshmi Institute of Technology