MADANAPALLE INSTITUTE OF TECHNOLOGY & SCIENCE

ANGALLU, MADANAPALLE 517325

(UGC Autonomous)

DEPARTMENT OF
ELECTRONICS & COMMUNICATION ENGINEERING

COMMUNICATIONS SYSTEMS LAB MANUAL

 INDEX
Exp No EXPERIMENT NAME Page.No
Hardware
I Syllabus
II Instructions to the Candidates
III Introduction to Analog Communication
Amplitude Modulation and Demodulation
1 a) Generation of AM with Carrier.
b) Demodulation of AM with Carrier.
Frequency modulation
2 a) Generation of Frequency Modulated Signals.
b) Demodulation of Frequency Modulated Signal.
a) Pulse Amplitude Modulation (PAM)
3
b) Time Division Multiplexing / De Multiplexing(TDM).
Sampling and Reconstruction
a) Generation of sampling Signal and its properties.
4
b) Sampling and Reconstruction.
c) Study of under sampling and aliasing.
5 Pre-emphasis and de-emphasis.
MATLAB
DSB-SC Modulation and Demodulation
a) Generation of DSB-SC Modulated Signal.
6 b) Demodulation of DSB-SC Signal.
c) Demodulation of DSB-SC Signal: Effect of LO Phase errors.
d) Demodulation of DSB-SC Signal: Effect of LO Frequency errors.
7 Generation and Demodulation of SSB-SC Signals.
Angle Modulation and Demodulation
8 a) Generation of Narrowband Phase Modulation (NBPM).
b) Characterization of VCO Module.
Study of Analog Filters Using Matlab
a) Simple RC Filters.
9
b) Higher order Filters.
c) Butterworth and Chebyshev Filters.
10 Study of Analog Filters Using RLC components.
Signal and Noise Experiments.
a) Generation of Signals.
11 b) Generation of Noise.
c) Studies on Signal Plus Noise.
d) Filtering of Noise.
Study of Baseband Detection performance in the Presence of Noise Using
Matlab
12
a) Generation of Unipolar, Bipolar and Noise Samples at the Receiver.
b) BER Performance of Unipolar system as a function of Eb/No.

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 c) BER Performance of Bipolar system as a function of Eb/No.

B.Tech. III Year I Semester

14ECE206 COMMUNICATION SYSTEMS PRACTICALS

L T P C

0 0 3 2
Course Prerequisite: 14ECE103 & 14ECE201
Course Description:
These courses provide simulation of various filters using MATLAB and implementation of Analog and
digital communication models.

Course Objectives:
1. To learn the basics of Communication systems.
2. To learn about both analog and digital systems of communication.

LIST OF EXPERIMENTS

1. Study of Analog Filters Using Matlab

d) Simple RC Filters.
e) Higher order Filters.
f) Butterworth and Chebyshev Filters.
2. Study of Analog Filters Using RLC components.
3. Signal and Noise Experiments.
e) Generation of Signals.
f) Generation of Noise.
g) Studies on Signal Plus Noise.
h) Filtering of Noise.
4. Amplitude Modulation and Demodulation
c) Generation of AM with Carrier.
d) Demodulation of AM with Carrier.
5. DSB-SC Modulation and Demodulation
e) Generation of DSB-SC Modulated Signal.
f) Demodulation of DSB-SC Signal.
g) Demodulation of DSB-SC Signal: Effect of LO Phase errors.
h) Demodulation of DSB-SC Signal: Effect of LO Frequency errors.
6. Generation and Demodulation of SSB-SC Signals.
7. Angle Modulation and Demodulation
c) Generation of Narrowband Phase Modulation (NBPM).
d) Characterization of VCO Module.
8. Frequency modulation
c) Generation of Frequency Modulated Signals.
d) Demodulation of Frequency Modulated Signal.
9. Sampling and Reconstruction
d) Generation of sampling Signal and its properties.

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 e) Sampling and Reconstruction.
f) Study of under sampling and aliasing.

10. a) Pulse Amplitude Modulation (PAM)

e) Time Division Multiplexing / De Multiplexing(TDM).
11. Study of Baseband Detection performance in the Presence of Noise Using Matlab
d) Generation of Unipolar, Bipolar and Noise Samples at the Receiver.
e) BER Performance of Unipolar system as a function of Eb/No.
f) BER Performance of Bipolar system as a function of Eb/No.
12. Pre-emphasis and de-emphasis.

Course Outcomes:
Upon successful completion of the course, students will be able to

1. Analyze the basic system of communication.

2. Analyze the analog and digital means of communication systems.

Mode of Evaluation: Continuous Internal Evaluation, Practical Examination.

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Instructions to the Candidates
Students are required to attend all labs.
Students should work in a group of three in hardware laboratories and individually in
computer laboratories.
While coming to the lab the student should bring the observation book, record, HB
pencil, scale, eraser & sharpener.
Before coming to the lab, the student should go through the procedure of the lab
experiments.
The student should utilize the prescribed time allotted for lab session properly to
perform the experiment & to note down the readings.
The student should complete the circuit design, calculations (if necessary), and
graph within the allotted time and should take the signature from faculty incharge of
the laboratory.
If the experiment is not completed in the prescribed time, the pending work has to
be done in the session(s) allotted for repetition only .
You will be expected to submit the completed record book according to the
deadlines set up by your instructor.
For practical subjects there shall be a continuous evaluation during the semester for
40 sessional marks and 60 end examination marks. Of the 40 marks for internal, 30
marks shall be awarded for day-to-day work and 10 marks to be awarded by
conducting an internal laboratory test.
The end examination shall be conducted by the teacher concerned & another member of
the staff of the same department.
Student should come with thorough preparation for the experiment to be conducted.
Student should take prior permission from the concerned faculty before availing the leave.
Student should come with proper dress code and to be present on time in the laboratory.
Student will not be permitted to attend the laboratory unless they bring the practical record fully
completed in all respects pertaining to the experiment conducted in the previous class.
Student will not be permitted to attend the laboratory unless they bring the observation book
fully completed in all respects pertaining to the experiment to be conducted in present class.
Experiment should be started conducting only after the staff-in-charge has checked the circuit
diagram.
Wherever graphs to be drawn, A-4 size graphs only should be used and the same should be
firmly attached in the practical record.
Student should obtain the signature of the staff-in-charge in the observation book after
completing each experiment.
Theory related to each experiment should be written in the practical record before procedure in
your own words with appropriate references.
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Introduction to Analog Communication
Communication is the transfer of information from one place to another. Radio communication
uses electrical energy to transmit information. The transmitted information is the intelligence
signal or message signal. Message signals are in the Audio Frequency (AF) range of low
frequencies from about 20 Hz to 20 kHz. The Radio Frequency (RF)is the carrier signal. Carrier
signals have high frequencies that range from 10 kHz up to about 1000 GHz. A radio transmitter
sends the low frequency message signal at the higher carrier signal frequency by combining the
message signal with the carrier signal. Modulation is the process of changing a characteristic of
the carrier signal with the message signal. In the transmitter, the message signal modulates the
carrier signal. The modulated carrier signal is sent to the receiver where demodulation of the
carrier occurs to recover the message signal.

IMPORTANT TERMS

Electromagnetic waves - the radiant energy produced by oscillation of an electric charge.

Message signal - any signal that contains information; it is also called the intelligence signal.

Audio Frequency (AF)-frequencies that a person can hear. AF signals range from about 20
Hz to 20 kHz.

Radio Frequency (RF) - the transmission frequency of electromagnetic (radio) signals. RF

frequencies are from about 300 kHz to the 1,000,000 kHz range.

Carrier signal - a single, high-frequency signal that can be modulated by a message signal
and transmitted.

Modulation - the process of combining the message signal with the carrier signal that causes
the message signal to vary a characteristic of the carrier signal.

Demodulation-the process of recovering or detecting the message signal from the modulated
carrier frequency.

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 Amplitude Modulation (AM) - the process of combining the message signal with the carrier
signal and the two sidebands: the lower sideband and the upper sideband.

Frequency Modulation (FM) - the process of combining the message signal with the carrier
signal that causes the message signal to vary the frequency of the carrier signal.

Phase Modulation (PM)-the process of combining the message signal with the carrier signal
that causes the message signal to vary the phase of the carrier signal.

Angle modulation - the process of combining the message signal with the carrier signal that
causes the message signal to vary the frequency and/or phase of the carrier signal.

Balanced modulator - an amplitude modulator that can be adjusted to control the amount of
modulation.

Double-Sideband (DSB)-an amplitude modulated signal in which the carrier is suppressed,

leaving only the two sidebands: the lower sideband and the upper sideband.

Mixer- an electronic circuit that combines two frequencies.

Phase detector-an electronic circuit whose output varies with the phase dierential of the
two input signals.

Envelopes- the waveform of the amplitude variations of an amplitude modulated signal.

Sidebands - the frequency bands on each side of the carrier frequency that are formed during
modulation; the sideband frequencies contain the intelligence of the message signal.

AM - an amplitude modulated signal that contains the carrier signal and the two sidebands: the
lower sideband and the upper sideband.

Bandwidth - the frequency range, in hertz (Hz), between the upper and lower frequency limits.
Harmonics - signals with frequencies that are an integral multiple of the fundamental frequency

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Expirement:

Amplitude Modulation & Demodulation

Amplitude Modulation and Demodulation
e) Generation of AM with Carrier.
f) Demodulation of AM with Carrier.

Aim: 1. To generate amplitude modulated wave and determine the percentage modulation.
2.To Demodulate the modulated wave using envelope detector.
Apparatus Required:
1. Amplitude Modulation and Demodulation Trainer
2. Function Generator
3. Oscilloscope
4. BNC Probes & Connecting Wires

Circuit Diagram For modulation:

Fig 1: CIRCUIT FOR AMPLITUDE MODULATION

Circuit Diagram For Demodulation:

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 Fig 2: CIRCUIT FOR AM DEMODULATOR
Carrier Generator Theory

Fig 3: Carrier Generator Theory

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.
Amplitude modulation is defined as the process in which is the amplitude of the carrier wave
is varied about a means values linearly with the base band signal. The envelope of the
modulating wave has the same shape as the base band signal provided the following two
requirements are satisfied
(1). the carrier frequency fc must be much greater then the highest frequency components
fm of the message signal m (t)
i.e. fc >> fm
(2) The modulation index must be less than unity. if the modulation index is greater than unity,
the carrier wave becomes over modulated

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Procedure:
Modulation
1.Switch on the trainer and check the O/P of carrier generator on oscilloscope.
2. Connect 1KHz with 2 Volts A.F signal at AF I/P to the modulator circuit.
3. Connect the carrier signal at carrier I/P of modulator circuit.
4. Observe the modulator output signal at AM O/P Spring by making necessary changes in
A.F. signal.
5. Vary the modulating frequency and amplitude and observe the effects on the modulated
waveform.
6. The depth of modulation can be varied using the variable knob (potentiometer) provided at
A.F. input.
7. The percentage of modulation or modulation factor can be calculated using the following
formulas.

8. Find the value of R from fm=1/(2*Pi*R*C) , C=0.1F

9. Connect the circuit diagram as shown in Fig.2.

Demodulation :
1. Connect the output received above to the input of demodulating circuit of the kit.
2. Connect the required terminals internally on the kit marked with dotted lines.
3. See the wave for demodulated signal on DSO by connecting the DSO probes on output of
demodulated circuit.
4. Feed the AM wave to the demodulator circuit and observe the output
5. Note down frequency and amplitude of the demodulated output waveform.
6. Draw the demodulated wave form., m=1
Tabular Column:

Demodulator
Message signal Carrier signal Modulator O/P m
S. O/P
No Vm Vc (Vp- Fm Fmin Vmax Vmin Fo V0
Fm(Hz)
(Vp-p) p) (Hz) (Hz) (V) (V) (Hz) (V)

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Model wave forms
Message signal

Demodulated signals

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PRECAUTIONS:

1. Switch off the experimental kit during making connections.

2. Set the proper amplitude and frequency of the modulating signal to get a reasonable AM
waveform.
3. Use the DSO carefully.

Observation:
Plot the graphs of input and output waveforms as observed on a DSO

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RESULT: Amplitude Modulation & Demodulation is studied.

TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:

Q1: Define Amplitude Modulation.

Q2: What is Modulation Index or Factor?

Q3: What is the range of commercial AM broadcast bands?

Q4: Which kind of modulation is used in picture signal in Television broadcast?

Q5: What are the different degree of modulation?

Q6: What are the limitations of square law modulator?

Q7: Compare linear and nonlinear modulators?
Q8: Compare base modulation and emitter modulation?

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Q9: Explain how AM wave is detected?
Q10 Define detection process?
Q11 What are the different types of distortions that occur in an envelop detector? How can they
be eliminated?

Expirement:

Frequency Modulation & Demodulation

Frequency modulation
a) Generation of Frequency Modulated Signals.
b) Demodulation of Frequency Modulated Signal.

Aim: 1. To generate frequency modulated signal and determine the modulation index and
bandwidth for various values of amplitude and frequency of modulating signal.

2. To demodulate a Frequency Modulated signal using FM detector.

Apparatus required:

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 1. Amplitude Modulation and Demodulation Trainer Kit
2. Function Generator
3. DSO
4. BNC Probes & Connecting Wires
Theory:
The process, in which the frequency of the carrier is varied in accordance with the instantaneous
amplitude of the modulating signal, is called Frequency Modulation. The phase of the carrier wave is
also kept constant. The instantaneous frequency of the resulting frequency modulated signal equals,

i = c + K f(t)

The term Kf(t) represents the frequency sensitivity of the modulator. The instantaneous frequency of
FM signal varies with time. The maximum change in instantaneous frequency from the average
frequency i.e. , is known as frequency deviation ().

= Kf Em

A disadvantage of the AM, DSBSC and other form of amplitude-modulation communication systems is
that they are susceptible to picking up electrical noise in the transmission medium (the channel). This is
because noise changes the amplitude of the transmitted signal and the demodulators of these systems
are affected by amplitude variations.

As its name implies, frequency modulation (FM) uses a messages amplitude to vary the
frequency of a carrier instead of its amplitude. This means that the FM demodulator is designed to look
for changes in frequency instead. As such, it is less affected by amplitude variations and so FM is less
susceptible to noise. This makes FM a better communications system in this regard.

There are several methods of generating FM signals but they all basically involve an oscillator
with an electrically adjustable frequency. The oscillator uses an input voltage to affect the frequency of
its output. Typically, when the input is 0V, the oscillator outputs a signal at its rest frequency (also
commonly called the free-running or centre frequency). If the applied voltage varies above or below 0V,
the oscillators output frequency deviates above and below the rest frequency. The amount of deviation
is affected by the amplitude of the input voltage. That is, the bigger the input voltage, the greater the
deviation. Figure 1 below shows a simple message signal and an unmodulated carrier. It also shows the
result of frequency modulating the carrier with the message

There are a few things to notice about the FM signal. First, its envelopes are flat recall that FM
doesnt vary the carriers amplitude. Second, its period (and hence its frequency) changes when the
amplitude of the message changes. Third, as the message alternates above and below 0V, the signals
frequency goes above and below the carriers frequency. (Note: Its equally possible to design an FM

P a g e | 15
modulator to cause the frequency to change in the opposite direction to the change in the messages
polarity.)

In frequency modulation, the amplitude of the carrier wave is kept constant but its frequency is varied in
accordance with the amplitude of the audio frequency signal.

FM is widely used for broadcasting music and speech, two-way radio systems, magnetic tape-recording
systems and some video-transmission systems. In radio systems, frequency modulation with sufficient
bandwidth provides an advantage in cancelling naturally occurring noise.

Fig 1: Circuit Diagram for Frequency Modulator

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 Fig 2 : Circuit for Frequency Demodulator

PROCEDURE:
1. Switch on the FM experimental board.
2. Connect Oscilloscope to the FM O/P and observe that carrier frequency at that point without
any A.F. input.
3. Connect around 7KHz sine wave (A.F. signal) to the input of the frequency modulator (At AF
input).
4. Now observe the frequency modulation output on the 1st channel of on CRO and adjust the
amplitude of the AF signal to get clear frequency modulated wave form.
5. Vary the modulating frequency (A.F Signal) and amplitude and observe the effects on the
modulated waveform.
6. Connect the FM o/p to the FM i/p of De-modulator
7. Vary the potentiometer provided in the demodulator section.
8. Observe the output at demodulation o/p on second channel of CRO.
9. Draw the demodulated wave form

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Observations:
AF Frequency : Carrier Frequency:
Message Carrier Demodulator
Modulator O/P
S. signal signal O/P
Tmax Tmin Fmax Fmin
No V0
Vm (Vp-p) Vc (Vp-p) V(p-p) Fo (Hz)
(usec) (usec) (khz) (khz) (V)

Model wave forms

PRECAUTIONS:
1. Switch off the experimental kit during making connections.
2. Set the proper amplitude and frequency of the modulating signal to get a reasonable FM
waveform.
3. Use the CRO carefully.

Observation:
Plot the graphs of input and output waveforms as observed on a DSO

RESULT: Frequency Modulation & Demodulation has been studied.

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TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:
Q1: Define Frequency modulation.

Q2: What is Frequency deviation?

Q3: What is carrier swing?

Q4: What is modulation index or factor in FM?

Q5: What is the range of band allotted for commercial FM broadcast?

Q6: What is main advantage of FM over AM?

Q7: What is main disadvantage of FM?

Q8: What method is used to compensate low frequency boost?

Q9: At what stage Pre-emphasis and de-emphasis circuits are used?

Q10: What is the bandwidth of commercial FM broadcast?

Q11: State Carson rule.

Q12: Why Armstrong method of FM is superior to reactance modulator?

Q13: Differentiate between narrow band FM and wideband FM.

Q14: Distinguish between Direct and Indirect FM?

Q15: Differentiate FM and AM?

P a g e | 19
Expirement:

Pulse Amplitude and Modulation

Aim: To generate the Pulse Amplitude modulated signal and demodulated signals.

Apparatus required:
1.Pulse amplitude modulation trainer.
2. Signal generator
3. DSO
4. BNC probes, connecting wires.
Theory:
PAM is the simplest form of the data modulation. The amplitude of uniformly spaced
pulses is varied in proportion to the corresponding sample values of a continuous message
m(t).
A PAM waveform consists of a sequence of list-topped pulses. The amplitude of each
pulse corresponds to the value of the message signal x(t) at the leading edge of the pulse.
The pulse amplitude modulation is the process in which the amplitude of regularity
spaced rectangular pulses vary with the instantaneous sample values of a continuous message
signal in a one-one fashion.
Pulse Modulation is used to transmit analog information. In this system continuous wave forms
are sampled at regular intervals. Information regarding the signal is transmitted only at the sampling
times together with syncing signals.
At the receiving end, the original waveforms may be reconstituted from the information regarding the
samples.
The Pulse Amplitude Modulation is the simplest form of the pulse modulation. PAM is a pulse
modulation system in which the signal is sampled at regular intervals, and each sample is made
proportional to the amplitude of the signal at the instant of sampling. The pulses are then sent by either
wire or cable are used to module division multiplexing is used.

PAM is of two types :

1. Double polarity PAM This is the PAM wave which consists of both positive and
negative pulses.
2. Single polarity PAM This consists of PAM wave of only either negative or positive
pulses. In this the fixed dc level added to the signal to ensure single polarity signal.

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Pulse Amplitude modulated signal and demodulated signals Kit

Procedure:

1 . Switch on pulse Amplitude modulation and demodulation trainer.

2 . In clock generator section connect pin 6 of 555IC to the 33pfcapacitor terminal.

P a g e | 21
 3 . Check the clock generator (RF)output signal.
4 . Connect RF output of clock generator to the RF input of modulator section.
5 . Connect a 1KHz; 2vp-p of sine wave from function generator to the AF input of
modulator section.
6 . Short the 10F terminal and 10k terminal of modulator.
7 . Connect 10k terminal to pin 1 of IC 4016.
8 . Connect the CRO to modulated output of modulator section.
9 . Adjust the 1k potentiometer to vary the amplitude of the modulated signal.
10. Adjust the AF signal frequency from 1KHZ-10KHZ to get stable output waveform.
While increases the AF signal frequency decreases the output signal pulses.
11 During demodulation, connect the modulated output to the PAM input of
Demodulator section.
12 . Connect channel 1 of CRO to modulating signal and channel-2 to demodulated
output. Observe the two waveforms that they are 1800out of phase, since the transistor
detector operates in CE configuration.

Sample Readings:

RF frequency = AF frequency =

S. AF input voltage RF Voltage PAM output voltage Demodulated Output

No Vp-p (volts) Vp-p (volts)
Vmax (volts) Vmin (volts) Vp-p (volts) Fo (Hz)

Expected Waveforms:

P a g e | 22
PRECAUTIONS:
1. Switch off the experimental kit during making connections.
2. Adjust the frequency of pulse trains carefully to get reasonable PAM waveforms.
3. Use the CRO carefully.

RESULT: Pulse Amplitude Modulation & Demodulation is studied.

TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:

Q1: Define Pulse Amplitude Modulation (PAM).

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Q2: What is the disadvantage of PAM?

Q3: Where do you find the application of PAM?

Q4: State sampling theorem.

Q6: Is PAM & Demodulation is sensitive to Noise ?

Q7: TDM is possible for sampled signals. What kind of multiplexing can be used in continuous
modulation systems?

Q8: What is the minimum rate at which a speech signal can be sampled for the purpose of
PAM?
Q9: What is cross talk in the context of time division multiplexing?

Experiment
TIME DIVISION MULTIPLEXING

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 Aim: To verify the operation of Time Division Multiplexing & De-Multiplexing

Apparatus:

1. Time Division Multiplexing and De-Multiplexing Trainer Kit

2. DSO
3. BNC Probes & Connecting Wires
Theory:

The Sampling Theorem provides the basis for transmitting the information contained in a
band limited message signal m(t)as a sequence of samples of m(t) taken uniformly at a rate that
is usually slighter higher than the Nyquist rate. An important feature of the sampling process is
a conservation of time. That is, the transmission the message samples engages the
communication channel s for only a fraction of the sampling interval on aperiodic basis, and in
this way some of the time interval between adjacent samples is cleared for use by other
independent message sources on a time shared basis. We there by obtain a time division
multiplexing(TDM) system, which enables the joint utilization of a common communication
channel by a plurality of independent message sources without mutual interference among
them.

The TDM system is highly sensitive to dispersion in the common channel, that is, to
variations of amplitude with frequency or lack of proportionality of phase with frequency.
accordingly, accurate equalization of both magnitude and phase response of a channel is
necessary to ensure a satisfactory operation of the system. Unlike FDM, TDM is immune to
nonlinearities in the channel as a source of cross talk. There as on for this is, the different
message signals are not simultaneously applied to the channel.

The primary advantage of TDM is that several channels of information can be transmitted
simultaneously over a single cable.

In the circuit diagram the555 timer is used as a clock generator. This timer is a
highly stable device for generating accurate time delays. In this circuit this timer generates
clock signal, which is of 100 KHz frequency(approximately). This clock signal is connected to
the74163IC, itis synchronous presettable binary counter. It divides the clock signal frequency
into three parts and those are used as selection lines for multiplexer and de-multiplexer.
Inbuilt signal generator is provided with sine, square and triangle outputs with variable

P a g e | 25
 frequency. These three signals can be used as inputs to the multiplexer. IC 4051 is an 8 to 1
Analog Multiplexer. It selects one-of-eight signal

sources as a result of unique three bit binary code at the select inputs. Again IC 4051 is wired
as one to eight de-multiplexers. Demux input receives the data source and transmits the data
signals on different channels.

Time Division Multiplexing and De-Multiplexing Trainer Kit

Input Signals

Multiplexed output:

P a g e | 26
Demultiplexed Output:

Observations:

S. No Signals Amplitude(vp-p) Time period (ms)

i 1 Square wave
n 2 Sinusoidal wave
p
3 Triangular wave
u
t 4 Clock signal

s
m

Square wave
Multiple

signals

i
xed

g Sinusoidal wave
n Triangular wave
a
l
Square wave
Multiple

s
xing
De-

Sinusoidal wave
Triangular wave

Procedure:
1. Generate different signals such as Sine , Square & Triangular signals
P a g e | 27
 2. Connect these signals to the respective input channels of MUX
3. Turn on the power supply of the trainer.
4. Observe the TX. output along-with CH.0 input for reference with the aid of oscilloscope.
5. The transmitter circuit samples all channels at different time intervals.
6. The Time Division Multiplexed samples appear at the TX. output.
7. Plot the Multiplexed Output
8. Connect the Multiplexed output to the de multiplexer
9. Using DSO connect to the respective channels to study respective outputs
10. Plot the outputs

Note : Vary the amplitude of the input sine-waves by varying the potentiometers in the
function generator block. This will help identifying which sample belongs to which input
channel.

PRECAUTIONS:
1. Switch off the experimental kit during making connections.
2. Use the CRO carefully.

RESULT:
Time Division Multiplexing and De-multiplexing has been studied.

TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:

P a g e | 28
Q1: Difference between TDM and TDMA.

Q2: What is aliasing?

Q3: What are the different multiplexing techniques used in digital communication?

P a g e | 29
Experiment

Sampling and Reconstruction

Sampling and Reconstruction
Generation of sampling Signal and its properties.
Sampling and Reconstruction.
Study of under sampling and aliasing.

Aim:
To verify the Sampling Theorem and observe the Aliasing effect

Apparatus Used:-
1. Sampling and Reconstruction Trainer kit,
2. Function generator
3. A 20 MHz, dual channel Oscilloscope.
4. BNC Probes & Connecting Wires
5. Patch cords
Theory

NYQUIST CRITERION (Sampling Theorem):-

The Nyquist criterion states that a continuous signal band limited to fm Hz, can be completely
represented by and reconstructed from, the samples taken at a rate greater than or equal to 2 f m
samples per second. This minimum frequency is called as "Nyquist Rate". Thus, for the faithful
reconstruction of the information signal from its samples, it is necessary that the sampling rate, fs
must be greater than 2f m.
Preliminary discussion So far, the experiments in this manual have concentrated on
communications systems that transmit analog signals. However, digital transmission is fast
replacing analog in commercial communications applications. There are several reasons for this
including the ability of digital signals and systems to resist interference caused by electrical
noise.
Many digital transmission systems have been devised and several are considered in later
experiments. Whichever one is used, where the information to be transmitted (called the message
) is an analog signal (like speech and music), it must be converted to digital first. This involves
sampling which requires that the analog signals voltage be measured at regular intervals.
Figure 1a below shows a pure sinewave for the message. Beneath the message is the digital
sampling signal used to tell the sampling circuit when to measure the message. Beneath that is
the result of naturally sampling the message at the rate set by the sampling signal. This type of
sampling is natural because, during the time that the analog signal is measured, any change in
its voltage is measured too. For some digital systems, a changing sample is unacceptable. Figure

P a g e | 30
1b shows an alternative system where the samples size is fixed at the instant that the signal
measured. This is known as a sample-and-hold scheme (and is also referred to as pulse amplitude
modulation ).

Regardless of the sampling method used, by definition it captures only pieces of the message. So,
how can the sampled signal be used to recover the whole message? This question can be
answered by considering the mathematical model that defines the sampled signal:
Sampled message = the sampling signal the message
As you can see, sampling is actually the multiplication of the message with the sampling signal.
And, as the sampling signal is a digital signal which is actually made up of a DC voltage and
many sinewaves (the fundamental and its harmonics) the equation can be rewritten as:
Sampled message = (DC + fundamental + harmonics) message
When the message is a simple sinewave (like in Figure 1) the equations solution (which
necessarily involves some trigonometry that is not shown here) tells us that the sampled signal
consists of:
A sinewave at the same frequency as the message
A pair of sinewaves that are the sum and difference of the fundamental and message frequencies
Many other pairs of sinewaves that are the sum and difference of the sampling signals
harmonics and the message
This ends up being a lot of sinewaves but one of them has the same frequency as the message.
So, to recover the message, all that need be done is to pass the sampled signal through a lowpass
filter. As its name implies, this type of filter lets lower frequency signals through but rejects
higher frequency signals.
Aliasing;- If the information signal is sampled at a rate lower than that stated by Nyquist
criterion, than there is an overlap between the information signal and the side bands of the
harmonics. Thus the lower and the higher frequency components get mixed and cause unwanted
signals to appear at the demodulator output. This phenomenon is termed as Aliasing or Fold-over

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Distortion. The various reasons for aliasing and the ways for its prevention may be summarized
as under:-
A) Aliasing due to under-Sampling:- If the signal is sampled at a rate lower than 2 f m
, then it causes aliasing, as illustrated in the following figure, where a sinusoidal signal of freque
ncy f m is being sampled at a rate f s <2 f m, and the dots represent the sample points.
The LPF at demodulator effectively joins the sample causing an unwanted frequency component
to appear at the output. This unwanted component has a frequency = (fs fm)
B) Aliasing due to wide band signal: The system is designed to take samples at a frequency sl
ightly greater than that stated ny Nyquist rate. If higher frequencies are ever present in the infor
mation signal, or it is affected by H. F. noise, then the aliasing will occur. To prevent the aliasin
g, Antialiasing filters are usually installed prior to sampling. In telephone networks, the speech
signals are bandlimited by filters before sampling to avoid the effect of aliasing.
C) Aliasing due to noise: If very small duty cycle is used in sampleandhold circuit, aliasing
may occur if the signal has been affected by the noise. High frequency noise generally mix with
the High frequency component of the signal. and hence causes undesirable frequency componen
ts to appear at the output. This type of aliasing may, therefore, be prevented by slightly increasin
g the duty cycle of the sampling pulses.
D) Aliasing due to Filter Rolloff : Aliasing may also occur, if appropriate filter response is not
chosen and the frequencies above the nominal cutoff frequency of the filter, have significant a
mplitudes at the filter's output. This is called Aliasing due to Filter Rolloff.
Observations:

Voltage (Vp-p) Time Frequency

Message signal
Clock Signal
Sampled Signal
Reconstructed
signal
Model wave forms

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Procedure

1. With the help of Function generator generate 1 KHz sinusoidal signal to SIGNAL INPUT on the
Sampling Circuit Kit
2. Now, turn the ON/OFF switch of the kit to ON.
3. Observe the information signal (SIGNAL INPUT of kit) on one channel and the Sample output
on the other channel of the DSO.
4. Adjust the scopes Time base control to view two or so cycles of the Master Signals modules
1kHz SINE output.
5. Activate the scopes Channel B input by pressing the Channel B Display controls ON/OFF
button to observe the sampled message out of the Dual Analog Switch module as well as the
message.
Tip: To see the two waveforms clearly, you may need to adjust the scope so that the two signals
are not over layed.
6. Draw the two waveforms to scale in the space provided on the next page leaving room to draw a
third waveform.
Tip: Draw the message signal in the upper third of the graph and the sampled signal in the middle
third.

PRECAUTIONS:
1. Switch off the experimental kit during making connections.
2. Use the DSO carefully.
3. Vary the amplitude of the input sine-waves by varying the potentiometers in the function
generator block. This will help identifying & compare the output. plot sample note down the time
period & pulse width of each samples & no of samples in a cycle

Result:
The analog signals are sampled and reconstructed and the results are plotted on the graph. Thus verified
sampling theorem.

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TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:
1. State the sampling theorem.
2. What is Nyquist rate?
3. What is meant by Aliasing?
4. What are the effects of Aliasing?
5. How to avoid Aliasing effect?
6. What happens when fs < 2 fm ?
7. How will be the reconstructed signal when fs >= 2fm?
8. Explain the operation of sampling circuit?
9. Explain the operation of re-construction circuit?
10. Who formalised the sampling theorem ?
11. What are the applications of the above theorem?
12. Is the sampling theorem basis for the modern digital communications?
13. What are the various Sampling techniques?
14. Explain various sampling circuits?
15. Why you need a hold circuit?

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Experiment
PRE EMPHASIS AND DE-EMPHASIS CIRCUITS

AIM: In this lab, you will learn how the characteristics of Pre-emphasis and De-emphasis differ
from each other.

HARDWARE REQUIRED
1. Function Generator,
2. Resistors,
3. Capacitor,
4. DSO & Probes
5. Connecting Wires

THEORY

Pre-emphasis

The circuits are the transmitting side of the frequency modulator. It is used to increase the
gain of the higher frequency component as the input signal frequency increased, the impendence
of the collector voltage increase. If the signal frequency is lesser then the impendence decrease
which increase the collector current and hence decrease the voltage.

Pre-emphasis refers to boosting the relative amplitudes of the modulating voltage for higher
audio frequencies from 2 to approximately 15 KHz.

De-emphasis means attenuating those frequencies by the amount by which they are boosted.
However pre-emphasis is done at the transmitter and the de-emphasis is done in the receiver.
The purpose is to improve the signal-to-noise ratio for FM reception. A time constant of 75s is
specified in the RC or L/Z network for pre-emphasis and deemphasis.

Pre-emphasis circuit

At the transmitter, the modulating signal is passed through a simple network which amplifies the
high frequency, components more than the low-frequency components. The simplest form of
such a circuit is a simple high pass filter of the type shown in fig (a). Specification dictate a time
constant of 75 microseconds (s) where t = RC. Any combination of resistor and capacitor
(or resistor and inductor) giving this time constant will be satisfactory. Such a circuit has a cutoff
frequency fco of 2122 Hz. This means that frequencies higher than 2122 Hz will he linearly

P a g e | 35
enhanced. The output amplitude increases with frequency at a rate of 6 dB per octave.
The pre-emphasis curve is shown in Fig (b). This pre-emphasis circuit increases the energy
content of the higherfrequency signals so that they will tend to become stronger than the high
frequency noise components. This improves the signal to noise ratio and increases intelligibility
and fidelity.

The pre-emphasis circuit also has an upper break frequency fu where the signal enhancement
flattens out. See Fig (b). This upper break frequency is computed with the expression.
fu = R1 +(R2/2R1R1C)

It is usually set at some very high value beyond the audio range. An fu of greater than 30KHz is
typical.
De-emphasis

To return the frequency response to its normal level, a de-emphasis circuit is used at the receiver.
This is a simple low-pass filter with a constant of 75 s. See figure (c). It features a cutoff of
2122 Hz and causes signals above this frequency to be attenuated at the rate of 6bB per octave.
The response curve is shown in Fig (d). As a result, the preemphasis at the transmitter is exactly
offset by the de-emphasis circuit in the receiver, providing a normal frequency response. The
combined effect of pre-emphasis and deemphasis is to increase the high-frequency components
during transmission so that they will be stronger and not masked by noise.

P a g e | 36
 The circuit is placed at the receiving side. De emphasis is the inverse process of
preemphasis, used to attenuate the high frequency signal that is boosted at the transmitter section.
The deemphasis network at the receiver section restores the original amplitude Vs frequency
characteristics of the information signal, after the demodulation process. The pre-emphasis and
deemphasis produces a more uniform SNR throughout the modulating signal frequency
spectrum. It acts as allow pass filter. The boosting gain for higher frequency signal in the
transmitting side is done by the pre-emphasis circuit is filtered to the same value by the low pass
filter.

The cut off frequency is given by the formula

fc = 1/(2p RC) (4-1) Where R = 2 p fc L

DESIGN FORMULA

fc = 1/(2 p RC) (assume =R = 10 KO, C = 0.01f)

R = 2 pfcL; L=1/(2pfc)

PRE-EMPHASIS:

Vi =
S.No Frequency(Hz) O/p voltage(V0) V0/Vi Gain in dB 20log(V0 /V i )

DE-EMPHASIS:

Vi =
S.No Frequency(Hz) O/p voltage(V0) V0/Vi Gain in dB 20log(V0 /V i )

P a g e | 37
Model plot

PROCEDURE

1. The circuit connection are made as shown in the circuit diagram for the pre-emphasis and de-
emphasis circuits
2. A input signal with 100Hz 20KHz with Constant input voltage 2 Vp-p is given to the circuit
3. For a constant value of input voltage the values of the frequency is varied and the output is
noted on the DSO
4. A graph is plotted between gain and frequency
5. The cut frequencies are practical values of the values of cut off frequency are found, compared
and verified

PREPARATION (PRE-LAB)
Do the complete revision of Pre-emphasis and De-emphasis theory.

RESULTS

The characteristics of pre emphasis and de emphasis circuits were studied and a graph was
drawn between gain (in db) and frequency.

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TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:
1. Which range of frequencies are more prone to noise interference?
2. How to reduce the noise during transmission in FM ?
3. Which technique is used at the receiver side to reconstruct the original signal?
4. What should be the time constant for the de emphasis circuit?
5. Why pre-emphasis is done after modulation?
6. What is the need for pre-emphasis?
7. Explain the operation of pre-emphasis circuit.
8. Pre emphasis operation is similar to high pass filter with gain in pass band explain
how?
9. De emphasis operation is similar to low pass filter with attenuation in stop band Justify?
10. What is de-emphasis?
11.Draw the frequency response of a pre-emphasis circuit
12. Draw the frequency response of a de-emphasis circuit.
13. Give the formula for the cutoff frequency of the pre-emphasis circuit.
14. What is the significance of the 3dB bandwidth.

P a g e | 39
SOFTWARE

MATLAB

P a g e | 40
Expirement:

DSB-SC Modulation and Demodulation

DSB-SC Modulation and Demodulation
a) Generation of DSB-SC Modulated Signal.
b) Demodulation of DSB-SC Signal.
c) Demodulation of DSB-SC Signal: Effect of LO Phase errors.
d) Demodulation of DSB-SC Signal: Effect of LO Frequency errors.

Aim
To Design Double Side Band Suppressed Carrier modulation. To implement a demodulator to obtain the
message signal.

Software:
MATLAB 7.10.0 ( R 2010a)

Theory
DSB-SC is basically an amplitude modulation wave without the carrier, therefore reducing power
waste, giving it a 50% efficiency. This is an increase compared to normal AM transmission, (DSB) has a
maximum efficiency of 33.33%, since 2/3 of the power is in the carrier which carries no intelligence, and
each sideband carries the same information. Single Side Band (SSB) Suppressed Carrier is 100% efficient

DSB-SC is a kind of amplitude modulation in which the carrier frequency component is absent. It is
generated by multiplying the carrier and modulating signals. If ec is the carrier and emis the message
signal, where

ec = Ec sin 2fct (6.1)

em = Em sin 2fmt (6.2)

Multiplication is done using AD633 (See A.5) multiplier IC. Applying em to X and ec to Y with Z grounded,

(6.3)

(6.4)

(6.5)

This wave contains both the sidebands at fc fm and fc + fm, but not the wave at carrier frequency1. Hence
the name double sideband suppressed carrier modulation(DSB-SC).
1sin A. sin B = cos (AB)2cos (A+B)

P a g e | 41
 The following figure 1 shows1 the DSB-SC signal in blue and the original message is shown in red. (It is
an indicative graph, not to scale as per the experimental set-up.)

Figure 1: DSB-SC signal in blue, original message shown in red.

Multiplying the DSB-SC with the carrier once again will result in the following output.

(6.6)

(6.7)
Thus the signal consists of various frequencies of which, the smallest is the message frequency. It can be
extracted by filtering using a low pass filter. Since the amplitude of the message frequency is very small,
It may be amplified using a simple non-inverting amplifier using an opamp.

Design
To the X input of the IC, feed the message sinusoid of amplitude Em = 2.5 V (ie., peak to peak amplitude
of 5 V) and frequency fm = 1 kHz.
To the Y input of the IC, feed the carrier sinusoid of amplitude Ec = 2.5 V and frequency fc = 100 kHz.

P a g e | 42
Ground the Z input of the IC.

Figure 1.2: Message and carrier signals

The output signal will have a waveform as given by,

(6.8)

(6.9)

(6.10)
This is the DSB-SC waveform.

Demodulation is by multiplying the DSB-SC signal once again with the carrier. This can be implemented
by connecting another AD633 IC in cascade with the first one.

The multiplication will result in the following output, as per the theory already explained.

Figure 1.3: AM(DSB-FC) and Demodulation stage-1 signals

(6.11)

This waveform is shown in Figure 1.3, which is the stage -1 in demodulation. The next step is to obtain
the message signal. This is done by lowpass filtering the above signal at a cut-off frequency of 1.5 kHz.

P a g e | 43
 To design an RC lowpass filter of cut-off frequency 1.5 kHz,

(6.12)

Choose C1 = 0.01 F R1 = 10 k
A non-inverting amplifier may be used to amplify this signal. Using a feedback resistor of Rf = 100
and an input resistance of Ri = 10 k will result in a gain of .

PROGRAM:

P a g e | 44
Procedure
1. Open the MATLABsoftware by double clicking its icon.
2. MATLABlogo will appear and after few moments Command Prompt will appear.
3. Go to the File Menu and select a New Mfile. (File NewMfile) or in the left hand corner a blank w
hite paper icon will be there. Click it once.
4. A blank Mfile will appear with a title untitled.
5. Now start typing your program. After completing, save the Mfile with appropriate name.
To execute the program Press F5 or go to Debug Menu and select Run.
6. After execution output will appear in the Command window. If there is an error then with an alarm,
type of error will appear in red color.
7. Rectify the error if any and go to Debug Menu and select Run.

Result
The AM-DSBSC wave is generated for the given message and carrier signals and the message signal is
recovered from the modulated waveform & were plotted.

P a g e | 45
TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:

1.Define AM?
2.Define DSB-SC system?
3.What is the working of Balanced Modulator?
4.Define Modulator and Demodulator?
5.Define coherent detector?
6.What are the different detectors used in
a) Critical modulation b) Under modulation c) Over modulation
7. What are the advantages and disadvantages of DSB-SC modulation?
8. What is a squelch circuit?
9. What are the different types of fading and solution?
11. What is the difference between DSB SC and SSB SC?
12. What are the applications of DSBSC?
13. Write the methods of DSBSC generation.
14. What is the BW for single tone modulating signal with frequency ?
15. What is the percentage of power saving for DSBSC when compared with AM having 100%
depth of modulation?

P a g e | 46
Expirement:

SSB-SC Modulation and Demodulation

Aim
To Design Single Side Band Suppressed Carrier modulation. To implement a demodulator to obtain the
message signal.

Software:
MATLAB 7.10.0 ( R 2010a)

Theory:
Single sideband suppressed carrier modulation (SSB-SC)
There are two approaches to eliminating one of the sidebands, one is the filter method and the
other is the phasing method. The process of selective filtering of the upper or lower sideband is
difficult due to the stringent filters required, especially if there's signal content close to DC. This
experiment will discuss the alternative, the phasing method, which uses a Hilbert Transformer to
implement SSB Modulation.
Figure 1 below shows a simple message signal and an unmodulated carrier. It also shows the
result of modulating the carrier with the message using SSB-SC. If you look closely, you'll
notice that the modulated carrier is not the same frequency as either the message or the carrier.

Fig.1 Simple message, unmodulated carrier & SSB Signal.

How to Generate SSB-SC

For the generation of SSB-SC, DSB-SC is used, where one of the sidebands of the modulated
signal is filtered out. Since filters are available only with finite edge steepness, SSB-SC can only
be implemented for the signal having a lower cut-off frequency not equal to zero. This is the case
with speech signals, where the frequency range spans from 0.3 kHz < f < 3.4 kHz.

Many different filter methods can be used for the suppression of the unwanted sideband. If a
Nyquist-filter is used instead of a filter with very good edge steepness, the modulation method is
called vestigial (residual) sideband AM. It enables the transfer of signals with an only slightly
higher bandwidth than as for example with SSB-SC. The advantage is that it can also carry DC
P a g e | 47
voltage signals like TV video signal. SSB- modulation can be performed also by a so-called
Hilbert-Transformer. This procedure is of interest, if SSB- modulation is implemented by digital
signal processing.

Fig.2 Block diagram showing phasing type of SSB modulator.

Ideal Hilbert Transform

The discrete Hilbert Transform is a process by which a signal's negative frequencies are phase-
advanced by 90 degrees and the positive frequencies are phase-delayed by 90 degrees. Shifting
the results of the Hilbert Transform (+j) and adding it to the original signal creates a complex
signal as we'll see below.

If mi[n] is the Hilbert Transform of mr[n], then:

is a complex signal known as the Analytic Signal. The diagram below shows the generation of an
analytic signal by means of the ideal Hilbert Transform.

P a g e | 48
Single - Sideband Modulation

The SSB modulated signal, f[n] can be written as

where mc[n] is the analytic signal defined as

Expanding that equation and taking the real part we get

which results in a single sideband, upper sideband (SSBU). Similarly, we can define the SSB
lower sideband (SSBL) by

The SSBU equation above suggests a more efficient way of implementing SSB. Rather than
performing the complex multiplication of mc[n] with exp(j*2*pi*fo*n/fs)and then throwing
away the imaginary part, we can compute only the quantities we need by implementing SSBU as
shown below.

Single-sideband modulation (SSB) or Single-sideband suppressed-carrier (SSB-SC) is a

refinement of amplitude modulation that more efficiently uses electrical power and bandwidth.
Amplitude modulation produces a modulated output signal that has twice the bandwidth of the
original baseband signal. Single-sideband modulation avoids this bandwidth doubling, and the
power wasted on a carrier, at the cost of somewhat increased device complexity and more
difficult tuning at the receiver.

The modulation method where only a single band of double sideband suppressed carrier
modulation is transmitted is known simply as SSB. It offers even better electrical efficiency and

P a g e | 49
frequency band efficiency than DSB. In addition, with digital modulation, modulation is
performed at relatively low frequency, and this method is also used to up-convert the signal to a
radio frequency. SSB-SC modulation can be achieved with various systems, but in terms of
hardware, USB or LSB can be used as a filter. As with DSB-SC, the modulating signal spectrum
is shifted directly to the carrier frequency band without loss.

The baseband or modulating signal can be recovered from the SSB-SC signal by using the
synchronous detection. Coherent Demodulation of SSB signals SSB (t) is multiplied with
cos(c t) and passed through low pass lter to get back the original signal.
A true SSB demodulator must have the ability to select sidebands. All the methods of SSB
generation so far discussed have their counterparts as demodulators. In this experiment you will
be examining the phasing-type demodulator, shown in Fig.1.

P a g e | 50
Program

P a g e | 51
Procedure
1. Open the MATLAB software by double clicking its icon.
2. MATLABlogo will appear and after few moments Command Prompt will appear.
3. Go to the File Menu and select a New Mfile. (New Script) or in the left hand corner a blank
white paper icon will be there. Click it once.
4. A blank Mfile will appear with a title untitled.
5. Now start typing your program. After completing, save the Mfile with appropriate name.
To execute the program Press F5 or go to Debug Menu and select Run.
6. After execution output will appear in the Command window. If there is an error then with an alarm,
type of error will appear in red color.
7. Rectify the error if any and go to Debug Menu and select Run.
8. Give input data in command window is any
9. Save the output plots
10. Record all the output observations of the command window if any

Result
The SSB-SC wave is generated for the given message and carrier signals and the message signal is
recovered from the modulated waveform & were plotted.

TYPICAL VIVA-VOCE QUESTIONS FOR REFERENCE:

Q.1 Can SSB Reception possible with the help of Envelope Detector?
Q.2 SSB can be viewed as a hybrid form of _____________?
Q.3 Why SSB is not used for video broad casting.
Q.4 Write one advantage of SSB modulation Process?
Q.5 Write one disadvantage of SSB modulation Process?
Q.6 The DSB-SC signal occupies _________ the space necessary than required for holding
the information?
Q.7 An input signal of 1.8Mhz is mixed with a local oscillator of 5Mhz. A filter selects the
difference signal. What is the output?

P a g e | 52
P a g e | 53