EEC 552 COMMUNICATION LAB-I

1. To study DSB/ SSB amplitude modulation & determine its

modulation factor & power in side bands.

2. To study amplitude demodulation by linear diode detector

3. To study frequency modulation and determine its modulation

factor

4. To study PLL 565 as frequency demodulator.

5. To study sampling and reconstruction of Pulse Amplitude

modulation system.

6. To study the Sensitivity, Selectivity, and Fidelity

characteristics of super heterodyne receiver.

7. To study Pulse Amplitude Modulation

a. using switching method
b. by sample and hold circuit

8. To demodulate the obtained PAM signal by 2nd order LPF.

9. To study Pulse Width Modulation and Pulse Position

Modulation.

10. To plot the radiation pattern of a Dipole, Yagi-uda and

calculate its beam width.
11. To plot the radiation pattern of Horn, Parabolic & helical
antenna. Also calculate beam width & element
current.
12. Design and implement an FM radio receiver in 88-108 MHz.

5
 OBJECT-To study sampling and reconstruction of Pulse
Amplitude modulation system.

Equipment Required
1. ST2101 with power supply cord
2. Oscilloscope with connecting probe
3. Connecting cords
Theory :-
The signals we use in the real world, such as our voice, are called "analog" signals.
to "digital" form. While an analog signal is continuous in both time and amplitude,
a signal is measured at certain intervals in time. Each measurement is referred to as
a sample. A precondition of the sampling theorem is that the signal to be band
limited. However, in practice, no time-limited signal can be band limited. Since
signals of interest are almost always time-limited (e.g., at most spanning the
lifetime of the sampling device in question), it follows that they are not band
limited. However, by designing a sampler with an appropriate guard band, it is
possible to obtain output that is as accurate as necessary. Aliasing is the presence of
unwanted components in the reconstructed signal. These components were not
present when the original signal was sampled. In addition, some of the frequencies
in the original signal may be lost in the reconstructed signal. Aliasing occurs
because signal frequencies can overlap if the sampling frequency is too low. As a
result, the higher frequency components roll into the reconstructed signal and cause
distortion of the signal Frequencies "fold" around half the sampling
frequency.
 Signal Sampling Connection Diagram :-

Signal Reconstruction Connection Diagram:

Principle of sampling

Principle of sampling
Principle of sampling:-
that can be viewed as a continuous function of time,
Consider an analogue signal x(t) hat can be viewed as a continuous function of
time, as shown in figure1. We can represent this signal as a discrete time signal
by using values of x(t) at intervals of nTs to form x(nTs) as shown in figure 1.
We are
"grabbing" points from the function x(t) at regular intervals of time, Ts, called the
sampling period

Basic Sampling Process

Sampling of signal at sampling interval (period) T

 t=nT
wheres n is
d(
oft)
d(
at t)
regular
interval of time.
 7. Connect
Figure 2 depictsthe
theSample output
sampling of atosignal
Inputatofregular
Fourthinterval
Order low pass Filters & observe
(period)t=nT
reconstructed
where output
n is an integer. onsampling
The (TP46) with helpis of
signal oscilloscope.
a regular Theofdisplay
sequence narrowshows
pulsesthe
d
reconstructed original 1 KHz sine wave Sa
( t ) of amplitude 1.Figure 3 shows the sampled output of narrow pulses d ( t ) at
m ple
regular interval of time.
8.By successive presses of sampling Frequency Selector switch, change the d
(Sampling frequency is 1/10th of the frequency indicated by the illuminated
LED). Observe how SAMPLE output changes in each cases and how the lower
sampling frequencies introduce distortion into the filter’s output waveform. This
is due to the fact that the filter does not attenuate the unwanted frequency
component significantly. Use of higher order filter would improve the output
waveform

9. So far, we have used sampling frequencies greater than twice the maximum
input frequency.

Result:- As the input sampling frequency is smaller than the applied input signal
then the output is distorted means the original signal can not be reconstructed.

Output of narrow pulses d(t)

The time distance T is called sampling interval or sampling period,

fs =1/ Ts is called as sampling frequency (Hz or samples/sec), also called sampling
rate.

Procedure:

1. Connect the power cord to the trainer. Keep the power switch in ‘Off’ position.
2. Connect 1 KHz Sine wave to signal Input.
3. Switch ‘On’ the trainer's power supply & Oscilloscope.
4. Connect BNC connector to the CRO and to the trainer’s output port.
5. Select 320 KHz (Sampling frequency is 1/10th of the frequency indicated by the
illuminated LED) sampling rate with the help of sampling frequency selector
switch.
6. Observe 1 KHz sine wave (TP12) and Sample Output (TP37) on Oscilloscope.
The display shows 1 KHz Sine wave being sampled at 32 KHz, so there are 32
Equipment Required
1. ST2101 with power supply cord
2. Oscilloscope with connecting probe
3. Connecting cords

7
OBJECT:- To study Pulse Amplitude Modulation
a. using switching method
b. by sample and hold circuit
Sample and Hold Waveform

Sample & Hold circuit:-

In electronics, a sample and hold circuit is used to interface real-world signals, by
changing analogue signals to a subsequent system. The purpose of this circuit is to
Now,
hold from figure 17 the areasteady
under for
the curve (which
time iswhile
equivalent to the signal
the analogue value a short the converter or other
power) is greater and so the filter output amplitude and quality of reproduced signal
following system performs some operation that takes a little time.
is
improved.
Sampling mode:
InInmost circuits,
this mode, thea switch
capacitor is the
is in usedclosed
to store the analogue
position and thevoltage andcharges
capacitor an electronic
to the
switch or gate is used to alternately
instantaneous input voltage. connect and disconnect the capacitor from the

Hold mode:
In this mode,
analogue input.the
Theswitch
rate atiswhich
in the this
openswitch
position. The capacitor
is operated is now disconnected
is the sampling rate of the
from
system. the input. As there is no path for the capacitor to discharge, it will hold the
Involtage on and
a sample it just before
hold opening
circuit the switch.
the switch opens The
for acapacitor
very shortwill hold this
duration. Thevoltage
sampletill
the hold
and next circuit
sampling instant.for a short duration charge into a capacitor.
integrates
The 'hold' facility can be provided by a capacitor, when the switch connects the
capacitor to PAM output it charges to the instantaneous value.
 A buffered sample and hold circuit consists of unit gain buffer preceding and
succeeding the charging capacitor. The high input impedance of the preceding
buffer
prevents the loading of the message source and also ensures that the capacitor
charges
by a constantrate irrespective of the source impedance see figure

Procedure:
1. Disconnect the Sample Output from Fourth Order low pass filter input.
Connect
Sample & Hold output to Fourth Order low pass Filter's Input. Set the Duty
Cycle Selector switch to position 5. (figure 2.1)
2. Observe the waveform at Sample & Hold output (TP39) on oscilloscope. Vary
the sampling frequency selector from 32 KHz to 2 KHz to illustrate how each
sample is held at the sample/hold output. Also observe the filter output at TP46.
(figure 2.2)
3. Vary the position of Duty Cycle Selector switch from 0 to 9 and note that in
contrast to step 7, the filter's output amplitude is now independent of the
sampling duty cycle and is equal to the amplitude of the original input signal.
This is an important result - with Sample and Hold Output, the proportion of
sampling time to holding time has no effect on reconstructed waveform
provided that Nyquist criteria has been followed. If sample/hold feature is
utilized in digital communication system many channels can be multiplexed
with maximum amplitude of reconstructed signal.

RESULT:-

For transmitting the signal if a sample and hold amplifier is used just before the
transmission channel, the signal will be less suffered from distortion as
compared to when only sample amplifier is used.

3
 OBJECT:- To study frequency modulation and determine its
modulation factor.

APPARATUS:

S.No Name of the Equipment Qty.

1. Frequency Modulation trainer kit. 1

2. C.R.O (30MHz) 1

3. Connecting cords & probes.

THEORY:
The frequency modulation process generates a large number of side frequencies.
Theoretically the sidebands are infinitely wide with the power levels becoming lower
and lower as we move away from the carrier frequency. The bandwidth of 250 KHz
was chosen as a convenient value to ensure a low value of distortion in the received
signal whilst allowing many stations to be accommodated in the VHF broadcast band
Communication signals, which do not require the high quality associated with
broadcast stations can, adopt a narrower bandwidth to enable more transmissions
within their allotted frequency band. Marine communications for ship to ship
communications, for example, use a bandwidth of only 25 KHz but this is only for
speech and the quality is not important.
These bandwidth figure bear no easy relationship with the frequency of the
information signal or with the frequency deviation or, it seems anything else. FM is
unlike AM in this respect.

FM Transmitter
The audio oscillator supplies the information signal and could, if we wish, can be
replaced by a microphone and AF amplifier to provide speech and music instead of
the sine wave signals that we are using with FM KIT.
The FM modulator is used to combine the carrier wave and the information signal
much in the same way as in the AM transmitter. The only difference in this case is
that the generation of the carrier wave and the modulation process is carried out in the
same block. It is not necessary to have the two processes in same block, but in our
case, it is. The output amplifier increases the power in the signal before it is applied to
the antenna for transmission just as it did in the corresponding block in the FM
transmitter.
The only real difference between the AM and FM transmitters are the modulations, so
we are only going to consider this part of the transmitter.
We are going to investigate two types of modulator; they are called the varactor
modulator and the reactance modulator.

Procedure :-

1.Make the connections as shown in figure.

2. Turn on power to the KIT.
4. Turn the audio oscillator block's amplitude potentiometer to its fully clockwise
(Maximum) positions, and examines the block's output (TP1) on an
Oscilloscope.
This is the audio frequency sine wave, which will be used as our modulating
signal. Note that the sine wave's frequency can be adjusted from about 300 Hz
to approximately 3.4 KHz by adjusting the audio oscillator's frequency
potentiometer Note also that the amplitude of this audio modulating signal can
be reduced to zero, by turning the audio oscillator's amplitude potentiometer to
its fully counter clockwise position.
5. Connect the output socket of the audio oscillator block to the audio input socket
of the modulator circuit’s block, as shown in figure 19.
6. Put the reactance /varactor switch in the reactance position. This switches the
output of the reactance modulator through to the input of the mixer/amplifier
block~ and also switches off the varactor modulator block to avoid interference
between the two modulators.
7. The output signal from the reactance modulator block appears at TP13, before
being buffered and amplified by the mixer/amplifier block. Although the output
from the reactance modulator block can be monitored directly at TP13, any
capacitive loading affect this point (e.g. due to an Oscilloscope probe) may
slightly affect the modulator's output frequency.
In order to avoid this problem we will monitor the buffered FM output signal
from the mixer/amplifier block at TP34.
8. Put the reactance modulator's potentiometer in its midway position (arrow
pointing towards top of PCB) then examine TP34.
9. The amplitude of the FM carrier (at TP34) is adjustable by means of the
mixer/amplifier block's amplitude potentiometer, from zero to its present level.
Try turning this potentiometer slowly anticlockwise, and note that the amplitude
of the FM signal can be reduced to zero.
Return the amplitude potentiometer to its fully clockwise position.
10. The frequency of the FM carrier signal (at TP34) should be approximately 455
KHz at the moment This carrier frequency can be varied from 453 KHz to 460
KHz (approximately) by adjusting the carrier frequency potentiometer in the
reactance modulator block.
Turn this potentiometer over its range of adjustment and note that the frequency
of the monitored signal can be seen to vary slightly. Note also that the carrier
frequency is maximum when the potentiometer is in fully clockwise position.
11. Try varying the amplitude & frequency potentiometer in audio oscillators block,
and also sees the effect of varying the carrier frequency potentiometer in the
mixer/amplifiers block.
12. Monitor the audio input (at TP6) and the FM output (at TP34) triggering the
Oscilloscope on the audio input signal. Turn the audio oscillator's amplitude
potentiometer throughout its range of adjustment and note that the amplitude of
the FM output signal does not change. This is because the audio information is
contained entirely in the signal's frequency, and not in its amplitude.
13. The complete circuit diagram for the reactance modulator is given at the end of
operating manual. If you wish, follow this circuit diagram and examine the test
points in the reactance modulator block, to make sure that you fully understand
how the circuit is working.
14. By using the optional audio input module, the human voice can be used as the
audio modulating signal, instead of using ST2203’s audio oscillator block.
If you have an audio input module, connect the module's output to the audio
input socket in the modulator circuit’s block
The input signal to the audio input module may be taken from an external
microphone (supplied with the module), or from a cassette recorder, by
choosing the appropriate switch setting on the modules.

RESULT:-

4
OBJECT:-To study PLL 565 as frequency demodulator

APPARATUS:

S.No Name of the Equipment Qty.

1. Frequency Modulation trainer kit. 1

2. C.R.O (30MHz) 1

3. Connecting cords & probes.

THEORY:
This is another demodulator that employs a phase comparator circuit. It is a very good
demodulator and has the advantage that it is available, as a self-contained integrated
circuit so there is no set up required. You plug it in and in works. For these reasons, it
is often used in commercial broadcast receivers. It has very low levels of distortion
and is almost immune from external noise signals and provides very low levels of
distortion. Altogether a very nice circuit.

The overall action of the circuit may, at first, seem rather pointless. As we can see in
Figure 31, there is a Voltage-Controlled Oscillator (VCO). The DC output voltage
from the output of the low pass filters controls the frequency of this oscillator. Now
this DC voltage keeps the oscillator running at the same frequency as the original
input signal and 90° out of phase. And if we did, then why not just add a phase
shifting circuit at the input to give the 90phase shift? The answer can be seen by
imagining what happens when the input frequency changes - as it would with a FM
signal. If the input frequency increases and decreases, the VCO frequency is made to
follow it. To do this, the input control voltage must increase and decrease. These
change of DC voltage level that forms the demodulated signal. The AM signal then
passes through a signal buffer to prevent any loading effects from disturbing the VCO
and then through an audio amplifier if necessary. The frequency response is highly
linear as shown in figure 32.

Procedure :-
2. Make the connections shown in figure 35.
3. Turn on power to the ST2203 module.
4. Now monitor the audio input signal to the varactor modulator block (at TP14)
together with the output from the phase-locked loop detector block (at TP60),
triggering the Oscilloscope in TP14. The signal at TP68 should contain three
components.
· A positive DC offset voltage.
· A sine wave at the same frequency as the audio signal at TP14.
· A high - frequency ripple component.

5. The low pass filter/amplifier block strongly attenuates the high-frequency ripple
component at the detector's output and also blocks the DC offset voltage.
Consequently the signal at the output of the low- pass filter/amplifier block (at
TP73) should be very closely resemble the original audio making signal, if not
then slowly adjust the frequency adjust potentiometer of PLL block.
6. Adjust the audio oscillator block's amplitude and frequency potentiometer and
compare the original audio signal with the final demodulated signal.
7. We can investigate the effect of noise on the system by following the procedure
given in earlier experiments. The only change will be that we will use phase
locked loop detector instead of quadrature or detuned resonant circuit.

RESULT:-

12
OBJECT:-Design and implement an FM radio receiver in 88-108
MHz.