Department Of Computer Science and

Engineering
University Institute of Technology
Rajiv Gandhi Proudyogiki Vishwavidyalaya Bhopal

LAB MANUAL
Subject Code : CS-405

Subject Name : Analog & Digital Communication

Semester : IV

Session : 2023-24
 List of Experiments
1. Study of sampling process and signal reconstruction and
aliasing.
2. Study of PAM PPM and PDM.
3. Study of PCM transmitter and receiver.
4. Time division multiplexing (TDM) and De multiplexing.
5. Study of ASK PSK and FSK transmitter and receiver.
6. Study of AM modulation and Demodulation techniques
(Transmitter and Receiver) Calculate of parameters
7. Study of FM modulation and demodulation (Transmitter and
Receiver) & Calculation of parameters.
8. To construct and verify pre emphasis and de-emphasis and plot
the wave forms.
9. Study of super heterodyne receiver and characteristics of ratio
radio receiver.
10. To construct frequency multiplier circuit and to observe the
waveform.
11. Study of AVC and AFC.
 Experiment 1
Aim
Study of Sampling Process and Signal Reconstruction and Aliasing.
Apparatus 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 process these signals for digital communication, we
need to convert analog signals to "digital" form. While an analog signal is
continuous in both time and amplitude, a digital signal is discrete in both
time and amplitude. To convert continuous time signal to discrete time
signal, a process is used called as sampling. The value of the signal is
measured at certain intervals in time. Each measurement is referred to
as a sample.
Principle of sampling:- Consider an analogue signal x(t) that can be
viewed as a continuous function of time, as shown in figure. 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 . We are “grabbing"
points from the function x(t) at regular intervals of time, Ts, called the
sampling period.

Aliasing: 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. This type of signal distortion
is called aliasing. We only sample the signal at intervals. We don't know
what happened between the samples. A crude example is to consider a
'glitch' that happened to fall between adjacent samples. Since we don't
measure it, we have no way of knowing the glitch was there at all.

In a less obvious case, we might have signal components that are

varying rapidly in between samples. Again, we could not track these
rapid inter-sample variations. We must sample fast enough to see the
most rapid changes in the signal. Sometimes we may have some a prior
knowledge of the signal, or be able to make some assumptions about
how the signal behaves in between samples.
Circuit Diagram:-
Signal Reconstruction:-

Procedure:-
A. Set up for Sampling and reconstruction of signal. Initial set up of
trainer: Duty cycle selectors switch position: Position 5. Sampling
selector switch: Internal position.
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 samples for every cycle of the sine
wave.
7. Connect the Sample output to Input of Fourth Order low pass Filter
& observe reconstructed output on (TP46) with help of
oscilloscope. The display shows the reconstructed original 1 KHz
sine wave.
 8. By successive presses of sampling Frequency Selector switch,
change the sampling frequency to 2KHz, 4KHz, 8KHz, 16KHz and
back to 32KHz (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.

Conclusion:-As the sampling frequency increases the output of sample

port has more number of samples of applied input signal.

B. Setup of Nyquist criteria and aliasing:-Initial set up of trainer: Duty

cycle selector switch position: Position 5 Sampling selector switch:
Internal position.
1. Keep the power switch in ‘Off’ position.
2. Connect 2 volts peak, 2 KHz sine wave from 600 ohms output of
the Function Generator to the signal Input of the trainer.
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. Connect the sample output to fourth order low pass filter & observe
the output (TP46) on oscilloscope. Observe the two waveforms
(applied input signal & filter output) which are similar but the
second waveform (filter output) is lagging in phase. This is as
expected from filters phase/ frequency response.
7. Decrease the sampling rate from 32 KHz to 2 KHz. Observe the
distorted waveform at filter's output (TP46). This is due to the fact
that we under-sampled the input waveform overlooking the Nyquist
criteria and thus the output was distorted even though the signal
lies below the cut-off frequency (3.4 KHz) of the filter. This explains
the phenomena of Aliasing.

Conclusion:-As the input sampling frequency is smaller than the applied

input signal then the output is distorted means the original signal cannot
be reconstructed.
 Experiment 2
Aim

Study of PAM, PPM and PDM.

(a)Study of Pulse Amplitude Modulation using Natural & Flat top Sampling.

Apparatus Required:

ST2110 with power supply cord.

Oscilloscope with connecting probe

Connecting cords.

Theory

Most digital modulation systems are based on pulse modulation. It involves

variation of a pulse parameter in accordance with the instantaneous value of the
information signal. This parameter can be amplitude, width, repetitive frequency
etc. Depending upon the nature of parameter varied, various modulation systems
are used. Pulse amplitude modulation, pulse width modulation, pulse code
modulation are few modulation systems cropping up from the pulse modulation
technique. In pulse amplitude modulation (PAM) the amplitude of the pulses are
varied in accordance with the modulating signal. In true sense, pulse amplitude
modulation is analog in nature but it forms the basis of most digital
communication and modulation systems. The pulse modulation systems require
analog information to be sampled at predetermined intervals of time. Sampling is
a process of taking the instantaneous value of the analog information at a
predetermined time interval. A sampled signal consists of a train of pulses, where
each pulse corresponds to the amplitude of the signal at the corresponding
sampling time. The signal sent to line is modulated in amplitude and hence the
name Pulse Amplitude Modulation (PAM).

Natural sampling:In the analogue-to-digital conversion process an analogue

waveform is sampled to form a series of pulses whose amplitude is the amplitude
of the sampled waveform atb the time the sample was taken. In natural sampling
the pulse amplitude takes the shape of the analogue waveform for the period of
the sampling pulse as shown in figure.

Flat Top sampling:-After an analogue waveform is sampled in the analogue-to-

digital conversion process, the continuous analogue waveform is converted into
a series of pulses whose amplitude is equal to the amplitude of the analogue signal
at the start of the sampling process. Since the sampled pulses have uniform
amplitude, the process is called flat top sampling as shown in figure

Circuit-Diagram
Signal Reconstruction:-

Procedure:-

Connect the circuit as shown in Figure.

Output of sine wave to modulation signal input in PAM block keeping the switch
in 1 KHz position.

8 KHz pulse output to pulse input.

Switch ‘On’ the power supply & oscilloscope.

Observe the outputs at TP(3 & 5) these are natural & flat top outputs respectively.

Observe the difference between the two outputs.

Vary the amplitude potentiometer and frequency change over switch & observe
the effect on the two outputs.

Vary the frequency of pulse, by connecting the pulse input to the 4 frequencies
available i.e. 8, 16, 32, 64 kHz in Pulse output block.

Switch ‘On’ fault No. 1, 2, 3, 4 one by one & observe their effect on Pulse
Amplitude Modulation output and try to locate them.

Switch ‘Off’ the power supply.

Related O/P Waveforms:-

(b) Study of PPM using DC Input, Sine wave Input.

Apparatus Required:

ST2110 with power supply cord.

Oscilloscope with connecting probe

Connecting cords.

Theory

The Amplitude and width of the pulses is kept constant in this system, while the
position of each pulse, in relation to the position of a recurrent reference pulse is
varied by each instantaneous sampled value of the modulating wave. As
mentioned in connection with pulse width modulation, pulse-position
modulations has the advantage of requiring constant transmitter power output,
but the disadvantages of depending on transmitter receiver is synchronization.

(a)analog-signal(b)pulse-amplitude-modulation(c)pulse-width-
modulation(d)pulse-position-modulation

There may be a sequence of signal sample amplitudes of (say) 0.9, 0.5, 0 and -
0.4V. These can be represented by pulse widths of 1.9, 1.5, 1.0 and 0.6μs
respectively. The width corresponding to zero amplitude was chosen in this
system to be 1.0μs, and it has been assumed that signal amplitude at this point
will vary between the limits of + 1 V (width = 2μs) and -1 V (width = 0μs). Zero
amplitude is thus the average signal level, and the average pulse width of 1μs has
been made to correspond to it. In this context, a negative pulse width is not
possible. It would make the pulse end before it began, as it were, and thus throw
out the timing in the receiver. If theb pulses in a practical system have a
recurrence rate of 8000 pulses per second, the time between the commencements
of adjoining pulses is 106 /8000 =125μs. This is adequate not only to
accommodate the varying widths but also to permit time-division multiplexing.
Pulse width modulation has the disadvantage, when compared with pulse position
modulation, which will be treated next, that its pulses are of varying width and
therefore, of varying power content. This means that the transmitter must be
powerful enough to handle the maximum-width pulses, although the average
power transmitted is perhaps only half of the peak power. On the other hand, puls
width modulation still works if synchronization between transmitter and receiver
fails, whereas pulse-position modulation does not, as will be seen.

For DC Input:

circuit-diagram:

Procedure:

Connect the circuit as shown in Figure and also described below for clarity.

a. Connect the DC output to input of PPM block.

Switch ‘On’ the power supply & oscilloscope.

Observe the output of PPM block at TP7.

Vary the DC output while observing the output of PPM block.

Switch ‘On’ the switched faults No. 1, 2, & 6 one by one & observe their effects
PPM input and try to locate them.

Switch ‘Off’ the power supply.

Connection Diagram:-

connection-diagram
related-wave-form

For Sine wave Inputs:

Circuit Diagram:
Procedure:-

Connect the circuit as shown in Figure and also described below for clarity. a.
Input of pulse position modulation blocks to sine wave output of FG block.

Switch ‘On’ the power supply & oscilloscope.

Keep the oscilloscope at 0.5mS / div, time base speed and in X-5 mode, and
observe the pulse position modulated waveform at the pulse position modulation
block output.

Vary the amplitude of sine wave and observe the pulse position modulation, keep
the amplitude preset in center. Here you can best observe the pulse modulation.

Switch ‘On’ fault No. 1, 2, & 6 one by one & observe their effects in pulse
position modulation output and try to locate them.

Switch ‘Off’ the power supply.

(c) Study of PWM using different Sampling Frequency.

Apparatus Required:

ST2110 with power supply cord.

Oscilloscope with connecting probe

Connecting cords.

Theory

In pulse width modulation of pulse amplitude modulation is also often called

PDM (pulse duration modulation) and less often, PLM (pulse length modulation).
In this system, as shown in Figure, we have fixed amplitude and starting time of
each pulse, but the width of each pulse is made proportional to the amplitude of
the signal at that instant.

(a)analog-signal(b)pulse-amplitude-modulation(c)pulse-width-
modulation(d)pulse-position-modulation

there may be a sequence of signal sample amplitudes of (say) 0.9, 0.5, 0 and -
0.4V. These can be represented by pulse widths of 1.9, 1.5, 1.0 and 0.6μs
respectively. The width corresponding to zero amplitude was chosen in this
system to be 1.0μs, and it has been assumed that signal amplitude at this point
will vary between the limits of + 1 V (width = 2μs) and -1 V (width = 0μs). Zero
amplitude is thus the average signal level, and the average pulse width of 1μs has
been made to cores to it. In this context, a negative pulse width is not possible. It
would make the pulse end before it began, as it were, and thus throw out the
timing in the receiver. If the pulses in a practical system have a recurrence rate of
8000 pulses per second, the time between the commencements of adjoining
pulses is 106 /8000 = 125μs. This is adequate not only to accommodate the
varying widths but also to permit time-division multiplexing. Pulse width
modulation has the disadvantage, when compared with pulse position
modulation, which will be treated next, that its pulses are of varying width and
therefore, of varying power content. This means that the transmitter must be
powerful enough to handle the maximum-width pulses, although the average
power transmitted is perhaps only half of the peak power. On the other hand,
pulse width modulation still works if synchronization between transmitter and
receiver fails, whereas pulse-position modulation does not, as will be seen.

Connection Diagram:-

Procedure:-

Connect the circuit as shown in Figure and also described below for clarity.

a. 1 KHz sine wave output of function generator block to modulation input of

PWM block

b. 64 KHz square wave output to pulse input of PWM block.

Switch ‘On’ the power supply & oscilloscope.

Observe the output of PWM block.

Vary the amplitude of sine wave and see its effect on pulse output.

Vary the sine wave frequency by switching the frequency selector switch to 2
KHz.

Also, change the frequency of the pulse by connecting the pulse input to different
pulse frequencies viz. 8 KHz, 16 KHz, 32 KHz and see the variations in the PWM
output.

Switch ‘On’ fault No. 1, 2, & 5 one by one & observes their effect on PWM
output and tries to locate them.

Switch ‘Off’ the power supply.

 Experiment 4
Aim
Study of PCM Transmitter and Receiver.
(a)Study of Pulse Amplitude Modulation using Natural & Flat top
Sampling.
Apparatus Required:
1. ST2103 trainer with power supply cord
2. Oscilloscope with connecting probe
3. Connecting cords.
Theory
The ST2103 & 2104, TDM PCM transmitter & receiver trainer
demonstrates the basic scheme used to transmit an information signal
using coding technique. It covers very basic concepts like role of sample
Amplifier, Analog to digital conversion Pseudo random synch code
generator, Digital to analog conversion, Pseudo random synch code
detector of sampling pulse while transmitting a signal. It also
demonstrates signal recovery using low pass filters of different orders.
Steps in Pulse Code Modulation:-
Sampling:-
The signals which are required to be transmitted as information is
known as information signal and in the case of voice communication
this will be a continuously changing signal containing speech
information. The aim of the kit is to transmit the signals in digital form
and is to reproduce this information signal in analog form at the
receiving end of the communication system with the help of sampling
and reconstruction trainer. In the exercises to follow, you will simulate
audio signal by a 1 KHz test signal provided On-board. The repetitive,
non-changing waveform does not contain information. Provided the
frequency of the test-signal lies within the frequency range which an
information signal will occupy, a test signal of this type can be extremely
helpful in system analysis and testing The voice signals are limited to
the range 300 Hz to 3.4 KHz, a 1 KHz frequency fits conveniently in this
range and can be used to demonstrate and test many techniques used
in communication system. Theory of sampling: The signals we use in
the real world, such as our voice, are called "analog" signals. To process
these signals for digital communication, we need to convert analog
signals to "digital" form. While an analog signal is continuous in both
time and amplitude, a digital signal is discrete in both time and
amplitude. To convert continuous time signal to discrete time signal, a
process is used called as sampling. The value of the signal is measured
at certain intervals in time. Each measurement is referred to as a
sample.
Principle of Sampling:-
Consider an analogue signal x(t) that can be viewed as a continuous
function of time, as shown in figure. 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. We are "grabbing" points from the function
x(t) at regular intervals of time, Ts, called the sampling period.

Figure depicts the sampling of a signal at regular interval (period) t= nTs

where n is an integer. The sampling signal is a regular sequence of
narrow pulses δ (t) of an amplitude. Figure shows the sampled output of
narrow pulses δ (t) at regular interval of time.
Circuit Diagram:-

Procedure:-
Initial set up for trainer ST2103:
Mode Switch Position: FAST position.
Function generator setting:
DC l & DC 2 amplitude controls: fully clockwise direction.
1 KHz & 2 KHz signal levels: 10 V peak -peak.
Pseudo random sync code generator switch: OFF position.
Error check code selector switches A & B: A = 0 & B =0 Position ('Off'
Mode).
All switched faults: OFF position.
1. Make the following connections as shown in figure.
2. I. DC 1 To CH 0
3. II. DC 2 To CH 1
4. Turn ‘On’ the power supply and oscilloscope. Adjust the DC1
amplitude control such that the voltage measured at TP10 (CH 0)
with the help of DMM / oscilloscope is + 3 Volts. Adjust the DC 2
amplitude control so that the voltage at TP12 (CH 1) is 2 V.
5. The LED outputs of A/D Converter & shift register are a
combination of the two input voltages. Also since the trainer is
working in fast mode, it is impossible to detect the code.
6. As stated earlier, the two channels are sampled at different time.
Approximately, after 10 seconds, when the system has settled
down to slow mode, observe the LEDs of A/D converter Block.
Notice that a particular combination of LEDs is lit in the A/D
converter Block for approximately 7 seconds. These LEDs
represent the latched output from the A/D Converter for every
sample of CH 0 & CH 1 Channels. Note the output of the A/D
Converter, Note: You may find the A/D Converter's output may not
be identical every time you switch the circuit from fast to slow
mode for the same DC Control setting. This is due to the slight
change in voltage at Sample / Hold circuit at the time of switching.
However the change in code will only be 1 Bit.
7. The parallel data from the A/D Converter is then loaded in the shift
register which converts in serial output. Connect the oscilloscope
at following points :
8. Oscilloscope channel 1 to TX. Clock output (TP3)
9. Oscilloscope channel 2 to S/L test point (TP9)
10. External trigger to TX. to output (TP4) You may have to
adjust the oscilloscope trigger levels to obtain a stable display.
11. Observe the interdependence of S/L, TX clock output and
the shift register outputs as shown by their respective LEDs.
Record the waveforms. The timing diagram for the process is
shown in figure.
Conclusion:
As the controlling signals are properly synchronized the output of the
two input waveforms are also synchronized.
 Experiment 5
Aim
Study Time Division Multiplexing (TDM) and De multiplexing.
Apparatus Required:
1. ST2102 trainer with power supply cord
2. Oscilloscope with connecting probe
3. Connecting cords.
Theory
Multiplexing: A sampled signal consists of a train of pulses, where each
pulse corresponds to the amplitude of the signal at the corresponding
sampling time. The signal sent to line is modulated in amplitude and
hence the name Pulse Amplitude Modulation (PAM). Multiplexing is the
process of combining signals from different information sources so that
they can be transmitted over a common channel. Multiplexing is
advantageous in cases where it is impracticable and uneconomical to
provide separate links for the different information sources. The price
that has to be paid to acquire this advantage is in the form of increased
system complexity and bandwidth. The two most commonly used
methods of multiplexing are:
1. Frequency division multiplexing (FDM)
2. Time division multiplexing (TDM)
Time Division Multiplexing:-
Time division multiplexing is the process of combining the samples from
different information signals, in time domain so that they can be
transmitted over the common channel. The fact utilized in TDM
technique is that there are large intervals between the message
samples. The samples from the other sources can be placed within
these time intervals. Thus every sample is separated from other in time
domain. The time division multiplexing system can be simulated by two
rotating switches, one at transmitter and the other at receiver. (See
figure) The two wipers rotate and establish electrical contact with one
channel at a time.
Each signal is sampled over one sampling interval and transmitted one
after the other along a common channel. Thus part of message 1 is
transmitted first followed by part of message 2, message 3 and then
again message 1 so on. The switches connect the transmitter and the
receiver to each of the channels in turn for a specific interval of time. In
effect each channel is sampled and the sample is transmitted when the
switches are in the channel 1 position, channel 1 forms a PAM channel
with an LPF for reconstruction, and so on for channels 2 and 3. The
result is that the amplitudes samples from each channel share the line
sequentially, becoming interleaved to form a complex PAM wave, as
shown above. A major problem in any TDM system is the
synchronization of the transmitter and receiver timing circuits. The
transmitter and receiver must switch at the same time and frequency.
Also SW1 must be in the channel 1 position when SW2 is in the channel
1 position, so that the switches must be synchronized in position also. In
a system that uses analogue modulation (PAM) the time slots are
separated by guard slots to prevent crosstalk between channels.
Circuit Diagram:-

Procedure:
Initial Setup of Trainer.
Function Generator pot direction.
Anti clock wise Duty cycle Position.
Delay control.
Anti clockwise Comparator Threshold level.
Anti clockwise.
1. Connect the power cord to the trainer. Keep the power switch in
‘Off’ position.
2. Switch ‘On’ the trainer's power supply & Oscilloscope.
3. Connect BNC connector to the CRO and to the trainer’s output
port.
4. Observe the clock signal (TP5) provided on Transmitter Timing
Logic block on CRO. It will be a train of pulses having clock
frequency of 64 KHz.
5. Display the clock signal (TP5) along with channel 0(TP6). Observe
the relation between two signals.
6. Vary the Duty Cycle Selector switch and observe the variation in
both signals (TP5 & TP6).
7. Observe the waveforms at TP7, 8, 9 & 10. Sketch the relative time
graph between the waveforms observed at TP 5, 7, 8, 9 & 10.
8. With the same links, observe the waveform at Transmitter CH0
output (TP6) on channel 1 of the oscilloscope.
9. Observe the waveforms at TP7, 8, 9, & 10 on the other channel.
10. Plot the wave forms in time domain with reference to the
Transmitter CH0 Signal.
Conclusion:
1. The ‘On’ and ‘Off” time of Sync signal is changed as we vary the
Duty Cycle Selector Switch.
2. The switching time is varying as we change duty cycle, so
accordingly we can change sampling period of input signals.
 Experiment 4
Aim
Study of ASK PSK and FSK transmitter and receiver.
(a) Study of ASK (Amplitude Shift Keying) Modulation and
Demodulation.
Apparatus Required:
1. ASK kit (Model No. ME 749-I)
2. Connecting probes
3. CRO
Theory
The binary ASK was one of the earliest forms of digital modulation used
in wireless telegraphy. this simplest form of the digital modulation is no
longer used widely in digital communication. Nevertheless it serves as a
useful model which helps in understanding certain concepts. In an ASK
system binary symbol 1 is represented by transmitting a sinusoidal
carrier wave of fixed amplitude Ac and fixed frequency Fc for the bit
duration Tb seconds whereas binary symbol is represented by switching
of the carrier for Tb seconds. This signal can be generated simply by
tuning the carrier of sinusoidal oscillator on end of for the prescribed
period by the modulating pulse trend. For this reason, the scheme is also
known as on-off scheme.
Circuit Diagram

Procedure:-
For Ask Modulation:
1. Connect the output of carrier wave generator (carrier output) to
carrier input of ASK modulator & connect the modulating output to
modulating input through patchcords.
2. Connect the CRO across output of ASK modulator.
3. Switch on the CRO as well as instrument.
 4. Observe the output wave shape on CRO.
5. Change the amplitude & frequency of modulating signal & observe
corresponding effect on ASK modulated wave. set the output
shape (Ask modulated) through offset potentiometer provided on
the front panel.
For Ask Demodulation:-
1. Connect the output of ASK modulator to the input of demodulator
through patchcord.
2. Connect CRO probe across output of demodulator.
3. Observe the demodulated (square wave) shape on CRO. We will
observe that the demodulated output is of same frequency as the
modulated signal with little distortion.
Precautions:
1. Check the cont unity of the connecting probes.
2. Handle the CRO properly.
(b) Study of PSK (Phase Shift Keying) Modulation and Demodulation.
Apparatus Required:
1. PSK Kit (Model No. ME 750-I),
2. Connecting probes
3. CRO
Theory
Digital communication became important with the expansion of the use
of the computers and data processing and have continue to develop in
to a major industry providing the interconnection of computer
peripherals and transmission of data between distance sides. PSK is a
relatively new system in which carrier may be phase shifted by 90° for a
mark and by minus 90° for a space. PSK has a number of similarities to
FSK in many aspects, as in FSK, frequency of the carrier is shifted
according to modulating square wave.
Circuit Diagram
Procedure:-
For PSK Modulation:
1. Connect the carrier Output of the carrier wave generator (IC 8038)
to carrier input of PSK modulator through patch cords. Also
connect any data output from the data outputs of the data
generator to the data input of the PSK modulator.
2. Connect the channel 1 of CRO across output of PSK. And channel
2 across data input on a dual trace oscilloscope.
3. Switch on the instrument using ON/OFF toggle switch provided on
the front panel.
4. Observe the output wave shape on CRO.
5. Change the data inputs and observe the PSK output on CRO.
For PSK Demodulation:
1. Connect the output of PSK to the demodulator input through patch
cord.
2. Connect the carrier to the carrier input of the PSK demodulator.
3. Connect the channel 1 of CRO across output of demodulator.
4. Observe the demodulated output on CRO (it will be same as data
input applied in the modulated in the modulator section).
Precautions:
1. Check the cont unity of the connecting probes.
2. Handle the CRO properly.
(c) Study of Frequency Shift Keying.
Apparatus Required:
Data generator, FSK modulation kit, CRO and connecting leads.
Theory
FSK is one of the basic modulation techniques for the transmission of
digital data .If the frequency of the sinusoidal carrier is switched
depending upon the input digital signal , then it is known as frequency
shift keying. As the amplitude remains constant in FSK, so the effect of
non-linear ties, noise interference is minimum on digital detection. So
FSK is preferred over ASK. Frequency shift keying consists of shifting of
frequency of carrier from a mask frequency to a space frequency
according to the base band digital signal Frequency shift keying is
identical to modulating an FM carrier with a binary digital signal In an
FSK system, two sinusoidal carrier waves of the same amplitude Ac but
different frequencies fc1 and fc2 are used to represent binary symbols
1and 0 respectively. It can be easily verified that binary FSK waveform is
a superposition of two binary ASK waveforms, one with a frequency fc1
and other with a frequency fc2. No discrete components appear in the
signal spectrum of FSK signal. The main advantage of FSK lies in its
easy hardware implementation.
Generation of FSK signal:-
The PSK signal can be generated by applying the incoming binary data
to a frequency modulator. To the other input a sinusoidal carrier wave of
constant amplitude Ac and frequency fc is applied. As the modulating
voltages changes from one level to another, the frequency modulator
output changes its frequency in the corresponding fashion.
Detection of FSK signal:-
FSK can be demodulated by using coherent and non-coherent detector.
The detector based on coherent detection requires phase and timing
synchronization. Non coherent detection can be done by using envelop
detector.

Result:FSK output is obtained on CRO.

Precautions:
1. Do not use open ended wires for connecting to 230 V power
supply.
2. Before connecting the power supply plug in to socket, ensure
power supply should be switched off
3. Ensure all connections should be tight before switching on the
power supply.
4. Take the reading carefully.
 Experiment 6
Aim
Study of AM modulation and Demodulation techniques (Transmitter and
Receiver) Calculate of parameters.
(a) To Generate Amplitude Modulated Wave and Determine the
Percentage modulation.
Apparatus Required:
1. Amplitude Modulation and Demodulation Trainer
2. Function Generator
3. Oscilloscope
4. Connecting Wires
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
Procedure:-
1. Switch on the trainer and check the O/P of carrier generator on
oscilloscope.
2. Connect 1 KHz 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 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.
% of Modulation (ma)= Vmax-Vmin/Vmax+Vmin x 100
or Modulation factor (ma) = Vmax-Vmin/Vmax+Vmin
Precautions:
1. Connect the circuit as shown in the circuit diagram.
2. Apply the required voltages wherever needed.
 Do not apply stress on the components.
Result:Amplitude modulated wave generated and studied.
(b)To Demodulate the Amplitude modulated wave using envelope
detector.
Apparatus Required:
1. Amplitude Modulation and Demodulation Trainer
2. Function Generator
3. Oscilloscope
4. Connecting Wires
Theory
Demodulation is the act of extracting the original information-bearing
signal from a modulated carrier wave. A demodulator is an electronic
circuit (or computer program in a software-defined radio) that is used to
recover the information content from the modulated carrier wave. There
are many types of modulation so there are many types of demodulators.
The signal output from a demodulator may represent sound (an analog
audio signal), images (an analog video signal) or binary data (a digital
signal).
These terms are traditionally used in connection with radio receivers, but
many other systems use many kinds of demodulators. For example in a
modem, which is a contraction of the terms modulator/demodulator. a
demodulator is used to extract a serial digital data stream from a carrier
signal which is used to carry it through a telephone line, coaxial cable, or
optical fiber
There are two methods used to demodulate AM signals.
The envelope detector is a very simple method of demodulation. It
consists of a rectifier (anything that will pass current in one direction
only) or other non-linear that enhances one half of the received signal
over the other, and a low-pass filter. The rectifier may be in the form of a
single diode, or may be more complex. Many natural substances exhibit
this rectification behavior, which is why it was the earliest modulation
and demodulation technique used in radio. The filter is usually a RC low-
pass type, but the filter function can sometimes be achieved by relying
on the limited frequency response of the circuitry following the rectifier.
The crystal set exploits the simplicity of AM modulation to produce a
receiver with very few parts, using the crystal as the rectifier, and the
limited frequency response of the headphones as the filter.
The product detector multiplies the incoming signal by the signal of a
local oscillator with the same frequency and phase as the carrier of the
incoming signal. After filtering, the original audio signal will result. This
method will decode both AM and SSB, although if the phase cannot be
determined a more complex setup is required.
An AM signal can be rectified without requiring a coherent demodulator.
For example, the signal can be passed through an envelope detector (a
diode rectifier and a low-pass filter). The output will follow the same
curve as the input baseband signal.
Procedure:-
1. Switch on the trainer and check the O/P of carrier generator on
oscilloscope.
2. Connect 1 KHz 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=0.1μF
9. Connect the circuit diagram as shown in Fig.
10. Feed the AM wave to the demodulator circuit and observe
the output
11. Note down frequency and amplitude of the demodulated
output waveform.
12. Draw the demodulated wave form.

Precautions:
1. Connect the circuit as shown in the circuit diagram.
2. Apply the required voltages wherever needed.
3. Do not apply stress on the components.
Result:
Original baseband signal recovered from Amplitude modulated wave.
 Experiment 7
Aim
Study of FM modulation and demodulation (Transmitter and Receiver) &
Calculation of parameters.
(a) To generate frequency modulated signal and determine the
modulation index and Bandwidth.
Apparatus Required:
1. Frequency Modulation and Demodulation Trainer
2. Function Generator
3. Oscilloscope
4. 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 FM signal is expressed as where
Ac is amplitude of the carrier signal; f c is the carrier frequency β is the
modulation index of the FM wave

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.

Precautions:
1. Connect the circuit properly.
2. Apply the required voltages wherever needed.
3. Do not apply stress on the components.
Result:Frequency of the Carrier signal varied according to the baseband
signal and waveform observed.
(b) To Demodulate a Frequency Modulated signal using FM detector.
Apparatus Required:
1. Amplitude Modulation and Demodulation Trainer
2. Function Generator
3. Oscilloscope
4. Connecting Wires
Theory
Demodulation is the act of extracting the original information-bearing
signal from a modulated carrier wave. A demodulator is an electronic
circuit (or computer program in a software-defined radio) that is used to
recover the information content from the modulated carrier wave.
There are many types of modulation so there are many types of
demodulators. The signal output from a demodulator may represent
sound (an analog audio signal), images (an analog video signal) or
binary data (a digital signal).
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 FM signal is expressed as where
Ac is amplitude of the carrier signal; f c is the carrier frequency β is the
modulation index of the FM wave.
In FM demodulation process we extract message signal from the FM
modulated signal
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

Precautions:
1. Connect the circuit properly.
2. Apply the required voltages wherever needed.
3. Do not apply stress on the components.
Result:Original baseband signal recovered from Frequency modulated
wave.
 Experiment 8
Aim
To Construct and Verify Pre-emphasis and De-emphasis and Plot the
Waveforms.
(a) To generate frequency modulated signal and determine the
modulation index and Bandwidth.
Apparatus Required:
1. Resistors (10 K-2, 47K, 75K, 1K)
2. Capacitors (22μF, 0.1μ-2,)
3. Transistor BC107
4. Function generators
5. CRO
6. Connecting Wires
7. RPS (15V)
8. Connecting Wires
Theory
The noise has a effect on the higher modulating frequencies than on the
lower ones. Thus, if the higher frequencies were artificially boosted at
the transmitter and correspondingly cut at the receiver, an improvement
in noise immunity could be expected, thereby increasing the SNR ratio.
This boosting of the higher modulating frequencies at the transmitter is
known as pre-emphasis and the compensation at the receiver is called
de-emphasis
Circuit Diagram
Procedure:-
1. Connect the circuit as per circuit diagram as shown in Fig.
2. Apply the sinusoidal signal of amplitude 20mV as input signal to
pre emphasis circuit.
3. Then by increasing the input signal frequency from 500Hz to 20
KHz, observe the output voltage (VO) and calculate gain 20 log
(vo/vi).
4. Plot the graph between gain Vs frequency.
5. Repeat above steps 2 to 4 for de-emphasis circuit (shown in
Fig.2). by applying the sinusoidal signal of 5V as input signal./li>

Model Graph

Result:
Thus the pre-emphasis and de-emphasis characteristics are studied
 Experiment 9
Aim
Study of Super-heterodyne Receiver and Characteristics of Radio
Receiver.
Theory
In super heterodyne radio receivers, the incoming radio signals arc
intercepted by the antenna arid converted into the corresponding
currents and voltages. In the receiver, the incoming signal frequency is
mixed with a locally generated frequency. The output of the mixer
consists of the sum and difference of the two frequencies. The mixing of
the two frequencies is termed heterodyning. Out of the two resultant
components of the mixer, the sum component is rejected and the
difference component is selected. The value of the difference frequency
component varies with the incoming frequencies, if the frequency of the
local oscillator is kept constant. It is possible to keep the frequency of
the difference components constant by varying the frequency of the
local oscillator according to the incoming signal frequency. In this case,
the process is called Super heterodyne and the receiver is known as a
super heterodyne radio receiver.

Antenna: As with all radio stations, the antenna will pick up the
electromagnetic radio waves from the atmosphere and convert these
into very small electrical currents.
Tuned R.F. Amplifier.: This block amplifies the very small currents
created in the antenna, to improve the sensitivity of the radio receiver, in
the same way that it was used in the TRF radio.
Local Oscillator: A new addition to the superheteradio. This is a sine
wave generator which is mechanically linked to the tuning capacitor.
This ensures that it always produces a frequency at a fixed amount
above the resonant frequency of the tuned amplifier. This is typically in
the range 450 kHz to 480 kHz.
Mixer: Again a new addition to the super heterodyne radio, but is the
critical addition, as it combines the received modulated radio frequency
carrier (fc) from the R.F.Amplifier, and the Local Oscillator (fo). The
output of the mixer produces at its output four different frequency
signals containing the following frequencies, fc, fo-fc, fo+fc, fo. Three of
these frequencies fc, fo-fc, fo+fc are amplitude modulated signals each
containing all the information about the original audio signal. The only
one that does not contain the original signal is fo, the local oscillator
frequency which is a pure sine wave. The most important of these is fo-
fc because irrespective of the carrier frequency that is tuned in, this
frequency will always be the same, since the output of the local
oscillator tracks the carrier frequency tuned in. This modulated
frequency is called the intermediate frequency (I.F.) and contains the
audio signal from the original radio station no matter what station is
tuned in.
IF Filter: The I.F. Filter is a fixed range band pass filter with very high
selectivity, specifically designed to pass only the intermediate
frequency. It is this which gives the super heterodyne receiver it’s big
advantage because no matter what radio station is tuned in, it will be
transferred by the mixer to the intermediate frequency and this highly
specialised band pass filter will be able to select this single frequency
from the four produced by the mixer every time with perfect rejection of
the others.
IF Amplifier: This stage provides extra amplification for the signal after
the IF filter, and again is carefully designed to provide maximum gain at
the IF frequency. The combined effect of the IF filters and IF amplifier
gives the super heterodyne receiver its excellent selectivity. Commercial
radio receivers may have several pairs of IF filters and amplifiers all
tuned to an identical IF Frequency. This is often referred to as the IF
Strip.
Detector / demodulator: As in the simple radio receiver, the
detector/demodulator block contains the diode and RF filter to produce
the non-zero signal, and remove the remaining RF carrier.
AF Amplifier: The recovered audio signal is now amplified so that it can
provide a meaning full signal to the loudspeaker.
Loudspeaker: Converts the amplified audio signal into sound.
Result:Study of Super-heterodyne Receiver Completed
 Experiment 10
Aim: To construct a frequency multiplier circuit and observe the
waveform.

Apparatus Required:

1. Breadboard
2. Resistors (various values)
3. Capacitors (various values)
4. Operational amplifier (Op-amp)
5. Function generator
6. Oscilloscope
7. Connecting wires

Theory: A frequency multiplier circuit is designed to take an input signal

of a certain frequency and generate an output signal with a frequency
that is a multiple of the input frequency. This is commonly achieved
using frequency multiplication techniques employing operational
amplifiers and various passive components like resistors and capacitors.
Procedure:

1. Set up the breadboard and connect the operational amplifier (Op-

amp) as per the datasheet or manufacturer's instructions.
2. Connect the function generator to the input of the circuit.
3. Connect the output of the circuit to an oscilloscope.
4. Adjust the function generator to provide the desired input
frequency.
5. Observe the output waveform on the oscilloscope.
6. Adjust component values if necessary to achieve the desired
frequency multiplication.
7. Record observations and measurements.

Conclusion:

The frequency multiplier circuit successfully generated an output

waveform with a frequency that is a multiple of the input frequency. The
waveform observed on the oscilloscope confirmed the functionality of
the circuit. Further adjustments and optimizations can be made to
improve the performance of the circuit for specific applications in analog
and digital communication systems.
 Experiment 11

Aim
To study the operation of AGC and AFC.
Theory
Automatic Gain Control (AGC): To control the gains of the amplifiers of
the system, AGC is employed. The AGC voltage is used to keep the
volume of the receiver constant to the level set by the listener. The
output of the second IF amplifier is also given to the AGC detector.
The AGC detector produces a dc voltage, called the AGC bias voltage,
which is proportion to the carrier strength of the received signal. The
signal is also amplified by the AG amplifier before being detected to
generate the AGC bias voltage. The AGC bias voltage generated is
applied to the RF amplifier and first IF amplifier, to control their gains.

In communication receivers, the simple AGC technique discussed in AM

receivers is not employed. This is because the simple AGC also reduces
the gain of the amplifiers for the weak signals. In communication
receivers, improved AGC techniques are used so that the weak signals
are satisfactorily processed. The two techniques or AGC that are
employed in communication receivers are:
 Delayed ACC
 Auxiliary AGC
Delayed AGC: In delayed AGC, the AGC remains inoperative below a
predetermined input carrier voltage. If predetermined level, it is
considered a weak signal. The received signal strength is below the
predetermined level, it is considered a weak signal. The AGC bias
voltage is applied to the RF and IF amplifiers only if the level or input
carrier voltage goes above this predetermined level. In other words, the
AGC is delayed in applying the bias voltage to the amplifiers the bias
voltage to the amplifier for a certain predetermined level.
A typical circuit diagram of a delayed AGC is illustrated in Figure (b). In
delayed AGC, the output of the last IF amplifier, which is the second IF
amplifier, is taken through a coupling capacitor, Cc. This is applied to a
diode, D1. The cathode of diode D1 is provided with a positive dc
voltage, Vdc, through a variable resistor, R. This sets the predetermined
level up to which the AGC is not to be applied. If the received signal is
weak, the anode voltage of the diode is less than the cathode voltage,
and is reverse-biased. This results in the diode not conducting. The
received signal thus goes to R1, R2, and C1 networks, as shown in Figure
(b). This is an AC signal and passes through C1. Thus, no dc voltage is
available for AGC and the amplifiers operate at their usual gains.
When a strong signal is received, the anode voltage goes above the
cathode voltage of the diode, and the diode starts conducting. The
signal, passed to capacitor C1 in this condition, gets negative peaks and
filters them. This results in a constant negative voltage, which is used as
a delayed AGC and applied, to the RF and IF amplifiers. The variable
resistor, R can adjust the level of delay AGC. This control is given at the
panel of the receiver so that the operator can adjust the delayed AGC
level according to the signal conditions. If a weak signal is received, then
the operator can adjust it so that no AGC is applied.
AGC Characteristics Curves: A comparison between simple AGC and
delayed AGC is shown in Figure (c). A curve is also drawn for an ideal
AGC.
These curves are also compared with the generated when AGC is not
applied. An ideal AGC provides a constant output signal level for all input
carrier signals, after a particular input level, marked as A in Figure (c). Up
to point A, the output level linearly increases with the input carrier signal.
A simple AGC continuously increases with an increase in the input
carrier signal. Thus, there is no control on the output signal level for a
simple AGC. On the other hand, the characteristic curve for delayed AGC
shows that it is very close to the ideal AGC.
Auxiliary AGC:In communication receivers, an auxiliary AGC is provided
in addition to the delayed AGC. The auxiliary AGC becomes operative for
very strong signals. The auxiliary AGC circuit includes only one diode,
which is connected between the collector of the first IF amplifier, Q3,
and the collector of the converter, Q4, as shown in Figure (d).
The anode of the diode is connected to the collector of the IF amplifier
and the cathode is connected to the collector of the convertor. When the
signal is not strong the diode is reverse biased and does not affect the
normal operation of the circuit. When a very strong signal is received, the
diode becomes forward biased and starts conducting. The diode
resistances lowers and it loads the first IFT and capacitor C3, as shown
in Figure (d). This circuit is connected to the base of first 1F amplifier,
and its gain reduces due to the loading of IFT1 and C3. This accordingly
reduces the output signal level. The auxiliary AGC provides another
means to reduce the gain of the IF amplifier in the presence of strong
signals.
Automatic Frequency Control AFC: circuits are used in situations where
you must accurately control the frequency of an oscillator by some
external signal. Basically, this type circuit does two things: It senses the
difference between the actual oscillator frequency and the frequency
that is desired and produces a control voltage proportional to the
difference; it also uses the control voltage to change the oscillator to the
desired frequency. AFC circuits are used to control the frequency of
sinusoidal oscillators and no sinusoidal oscillators. Only sinusoidal AFC
circuits will be covered here. AFC circuits are used in radio receivers, fm
transmitters, and frequency synthesizers to maintain frequency stability.
Figure (e) is a block diagram illustrating AFC operation in a receiver.
Let’s run through the applicable parts of this block diagram.

Figure (f) shows another widely used type of AFC and its circuitry. This
type is commonly referred to as a BALANCED-PHASE DETECTOR or
PHASE-DISCRIMINATOR. This circuit uses fixed capacitors and the
varying conductance of the diodes to achieve a variable reactance. As
seen in the block diagram, an AFC circuit requires two sections, a
frequency detector and a variable reactance. Our detector output is a dc
control voltage proportional to the amount of frequency change. This dc
voltage is applied directly to the oscillator. The phase inverter input
signals are discriminated IF outputs fed to the two diodes 180 degrees
out of phase.

A reference voltage is also applied to both diodes. The diodes are biased
to conduct only during the peak portions of the input signals. Any
change in oscillator frequency will alter the phase relationship between
the saw tooth reference voltage and the incoming signals. If this
happens, one diode will conduct more than the other and produce a
control signal. This system remains unbalanced at all times because any
change in frequency is instantaneously corrected. The network between
the diodes and oscillator is essentially a low-pass filter. This filter
prevents discriminator pulses from reaching the oscillator.