D 14pcom refrence manu 2015-16

ANALOG COMMUNICATION LAB (ETL-502)
REFERENCE JOURNAL
ACADEMIC YEAR: - 2015- 2016
SUBJECT: - ANALOG COMMUNICATION
BRANCH: - Electronics and Telecommunication Engineering
SEMESTER: - V
DEPARTMENT OF ELECTRONICS AND TELECOMMUNICATION
Vivekanand Education Society’s Institute of Technology
HAMC, Collector Colony, Chembur-71
Faculty In charge: Mr. Chintan S. Jethva ,Darshana Suryavanshi, Mr.Mahesh Warang
Subject: Analog Communication
LAB MANUAL (V-SEM EXTC) Page 1

Branch: Electronics and Telecommunication Engineering (Semester V) D14A, D-14B, D14C
Academic Year: 2015-16 Semester:-
ODD
LAB PLAN
Sr.
No
Name of Experiment Mapped
CO
Mapped
PO
Week
1 To perform AM modulation & Demodulation 1,2,3,4 a,c,f,g 2
2 To perform FM modulation & Demodulation 1,2,3,4 a,c,f,g 3
3 To perform Sampling Techniques. 1,2,3,4 a,b,c,g 4
4 To perform PWM modulation & Demodulation 2,3,4 a,c,f 5
5 To perform AM modulation (COMMSIM7) 2,3,4 a,c,d,f 6
6 To perform FM modulation & Demodulation
(COMMSIM7)
2,3,4 a,c,d,f 6
7 To perform PCM modulation & Demodulation 1,2,3,4 a,b,c,g 7
8 To perform DM modulation & Demodulation 1,2,3,4 a,c,f,g 8
9 To perform TDM 2,4 a,c,f 9
10 To perform FDM 2,4 a,c,f 10
Evaluation of Experiment done based on following grading system
Factor Contributing Percent
Observations 50%
Diagrams 15%
Explanation 10%
Punctuality 15%
Conclusion 10%
Extra evaluation techniques
• Mock Viva
Vivekanand Education Society’s Institute of Technology
Department of Electronics & Telecommunication Engineering
Faculty In charge: Subject: Analog Communication

Branch : Electronics and Telecommunication Engineering (Semester V) D14
Academic Year: 2015-16
VISION
• Towards developing a center of excellence in the field of Electronics and
Telecommunication and to nurture students to become technocrats with a humane outlook.
MISSION
• To empower students to meet the growing challenges of industry.
• To promote a cutting-edge research to benefit the society locally and globally.
• To develop young engineers with human and social intellectual qualities required for
practices responsible engineers.
Program Educational Objectives (PEO)
I To prepare students to aptly apply their acquired knowledge of engineering
fundamentals and core concepts in Electronics and Telecommunications.
II To contribute to the needs of society in solving real life technical challenges using
Electronics and Telecommunication engineering principle tools and practices.
III To enable students to be successful technocrats with effective communication skills
and be socially conscious with strong ethical and balanced outlook.
IV To create and provide a conducive environment suitable for lifelong learning,
successful entrepreneurship, multidisciplinary engineering challenges and to tackle
the contemporary issues.
Program Outcomes (PO)
a. Engineering Knowledge: An ability to apply fundamental concepts of Maths, science &
engineering to solve Electronics and Telecommunication problems.
b. Problem analysis & design: An ability to identify analysis /synthesis interprets data
to design /develop solutions for complex engineering problem in the field of
Electronics and Telecommunication.
c. Professional engineer practice: An ability to apply the acquired engineering skills
professionally & ethically & understand the impact of engineering solution in social and
environmental contexts.
d. Modern tool usage: Create, select, and apply appropriate techniques, resources, and
modern engineering and IT tools, including prediction and modeling to complex
engineering activities, with an understanding of the limitation.
e. Communication: communicate effectively on complex engineering activities with the
engineering community and with the society at large such as, being able to comprehend

and write effective reports and design documentation, make effective presentations,
and give and receive clear instructions.
f. Individual and team work: Function effectively as an individual, and as a member or
leader in diverse teams, and in multidisciplinary settings.
g. Life –long learning: Recognize the need for, and have the preparation and ability to
engage in independent and life -long learning in the broadest context of technological
change.
h. Industry Orientation: Demonstrate knowledge and understanding of the economic
principles, management and telecom regulation.
Course Objective:
1. The fundamentals of basic communication system.
2. Various modulation and demodulation techniques used in analog communication, noise
handling and Multiplexing.
3. The working principles of transmitters and receivers used in analog communication
systems.
Course Outcome:
Students will be able to:
1. The different modulation and demodulation techniques used in analog
communication.
2. Identify and solve basic communication problems, analyze transmitter and
receivers.
3. Detect the errors that occur due to noise during transmission.
4. Compare and contrast advantages and limitations of analog communication systems.

EXPERIMENT NO.1
AMPLITUDE MODULATION AND DEMODULATION
Aim: To study amplitude full carrier modulation and demodulation.
Apparatus: Trainer kit (AM), dual trace CRO function generator and patch cords
Circuit diagram:-

THEORY: when a low frequency audio signal controls the amplitude of carrier signal by
keeping frequency and phase constant .we get amplitude modulation. Carrier is high
frequency signal and low frequency audio signal is called modulating signal. Hundreds
of the carrier cycles during one cycle of modulating signal. A physical layout of an
emitter type AM modulator is as shown in diagram. Carrier signal Vm is the input at
CE amplifier. The circuit amplifies carrier signal by a factor A .So the gain is Av.
The modulating signal is a part of biasing .It produces the low frequency variation in
emitter current .Thus in turn produces Variation in Rc and finally changes the
amplitude output wave. The final output peaks very sinusoidally with modulating signal.
Input frequency should be choosen in such a way that the carrier frequency fx should
be at least 100 times greater than modulating frequency fy. This is Because the
capacitor should look like low impedance to carrier signal and high impedance to

modulating signal so that modulating signal can be easily coupled to the output The
amount of AM modulation can be measured as the percent modulation .Percent
modulation gives us the depth in modulation it depends on modulation Index (m) of AM
signal.
Percent modulation (%m) = Vmax – Vmin/ Vmax + Vmin*100
PROCEDURE:
• Connect the kit to main supply and switch it 'ON'
• Select sine wave at amplitude and 1KHz frequency for modulating wave input .
• Switch 'ON' the carrier kit and feed carrier input signal to input at C1 as shown in
panel
• Feed modulating input at modulating input terminals.
• Observe the output(amplitude modulated ) on CRO (select time base of CRO in
millisecond range
• Observe the output by varying amplitude of modulating input
• Calculate modulation index (m)
Observe the amplitude modulated wave at different modulating index for each
modulating signal amplitude.
Observation table:
Sr
no.
Vmax
(V)
Vmin
(V)
M=vma-
vmin/vmax+vmin
Vm=vmax-
vmin/2
Vc=vmax+vmin/2

Part B : DEMODULATION AM FC FULL CARRIER
CIRCUIT DIAGRAM:
THEORY:
At the transmitter, modulation is done in which audio is placed in the carrier to take audio
at higher distance .Thus to get audio from the modulated signal. The reverse procedure
of detecting the low frequency audio from the modulated wave is called as
"Demodulation The AM Demodulation kit demodulates the modulated wave of 30%
modulation .Basically it’s a peak detector .Ideally the peak of the input signal are to
be Detected so that the output is upper envelope .For this reason the circuit is called
envelope. During each carrier cycle diode turns 'ON' and charges the capacitor to
the peak voltage of the particular carrier cycle between peak particular cycle

discharges.
If we make the RC time much greater than the period of the carrier we get only a slight
discharge between cycle .This removes most of the outputs then look Like upper envelope
with a smaller ripple.
PROCEDURE:
• Select AM modulated waveform with appropriate amplitude.
• Give the input to trainer kit
• Observe of AM demodulator.
• The output will be the audio demodulated signal
OBSERVATION TABLE:-
Signal Amplitude Frequency
Carrier
Modulating i/p
Output
CONCLUSION:
Standard output:

EXPERIMENT NO.2
FREQUENCY MODULATION AND DEMODULATION
Aim: To study frequency modulation and demodulation (Using PLL).
Apparatus: Trainer kit, CRO, Jumper, FM modulator, Fm demodulator.
CIRCUIT DIAGRAM FOR FM MOD AND DEMODULATION:-

THEORY:
In angle modulation, information signal may be used to vary carrier Frequency giving
rise to frequency modulation or it may be used to vary angle of phase load or log. Giving
rise to phase modulation since both are parameters of carrier angle , which is a
function of time ,general term angle modulation covers both. Angle modulation results
whenever phase angle (ø) of a Sinusoidal wave is varied with respect to time .An angle
modulated wave Is expressed mathematically as
M(t)=v cos[wt+ ø(t)]

Where
M (t) ->angle modulated wave
V c->peak carrier amplitude
W c->carrier radian frequency
Ø(t)->instantaneous phase deviation
DIRECT FREQUENCY MODULATION (FM):
Varying frequency of constant amplitude carrier directly proportional to amplitude of
modulating signal at a rate equal to frequency of modulating signal.
FREQUENCY DEVIATION:
Frequency deviation is the change in frequency that occurs in carrier when it is acted
upon by a modulating signal frequency deviation is typically given as a peak frequency
Shift in hertz (Hz).the peak to peak frequency deviation (2Af) is sometimes called
carrier swing. For an fm, deviation sensitivity is often given in hertz per volt .therefore
peak frequency deviation Is simply the product of the peak modulating signal voltage
and expressed as
F=k, Vm(Hz)
Expression for modulation index in fm can be written as
M = F
Fm
M(t)=Vc cos Wc t +Af/fm sin[(Wmt)]
M(t)=Vc cos Wc t +m sin[(Wmt)]
With Fm, however both the modulation index &frequency deviation are directly
Proportional to amplitude of amplitude of modulating signal and inversely
proportional o the frequency. Here we are using IC 8038 for generation of fm
wave IC 8038 wave form generator is monolithic integrated circuit capable of
producing high accuracy sine square triangular saw tooth & pulse waveform with
minimum Of External components. Frequency can be selected externally from
0.00 Hz to 300Hz Using either resistors of capacitors, frequency modulation and
sweeping can be Accomplished with an external voltage.

PROCEDURE:-
• Take the trainer kit insert ,its main cord in the (230v)supply ,Switch it on and
see the power Led glows.
• Select the carrier frequency by selecting the c2 or c3.
• Connect CRO at the FM o/p terminal observe the carrier hence note the
carrier frequency of FM modulator (8038).
• Select 100Hz 1 volt peak to peak sine wave from sine wave generator give this
sine wave at the modulating input terminal of FM modulator.
• Again connect CRO at FM output terminal and observe the frequency
modulated wave form by varying the sine wave i/p from the function
generator.
• With the help of jumper connect FM output to the FM output (Input Of 565) of
the FM demodulator. Vary the frequency of carrier by changing c2 or c3.
->PART B
FM DEMODULATION:
FM demodulators are frequency dependent circuits designed to produce an o/p voltage
that is proportional to the instantaneous frequency as Its input. The overall transfer
function for an FM demodulator is nonlinear but when operated over its linear
range.
Kd=v(volts)/f(Hz)
Where Kd equals transfer function o/p from an fm demodulator is expressed as
Vout(t)=demodulated o/p signal.
Kd=demodulator transfer function
Delta f=difference between i/p frequency & center frequency of demodulator.
TYPES OF FM DEMODULATORS:-
(1)Slope detector
(2)Balanced slope detector
(3) Foster seely discriminator

(4)Radio detector
(5)Fm demodulators using PLL
A PLL frequency demodulator is probably simplest &easiest compensates for changes in
carrier frequency due to instability in transmit oscillator. A PLL is a closed loop feedback
(circuit) control system in which either frequency or phase of feedback signal is the
parameter of interest rather than magnitude of signals voltage or current .The basic block
diagram for phase locked circuit is as shown below.
PLL consists of four primary blocks.
(1)Phase comparator or phase detector
(2)Low pass filter
(3)Low gain operational amplifier
(4)VCO
The four circuits are modularized and placed on an integrated circuit With each circuit
provide external i/p& o/p pins ,allowing users to Interconnect the circuits are needed and
to set the break frequency Of low pass filter, the gain of the amplifier ,and the frequency
of the VCO
i/p
(b)FREQUENCY DEMODULATION
(1) Take a trainer kit, insert its mains cord in mains supply, Switch it on and see the power
Led glows. Select carrier frequency by selecting c2 or c3.
(2) With the help of a jumper connect FM output (8038) To FM i/p (565) of fm
demodulator.
Phase comparator Low pass filter Amplifier
Voltage controlled
Oscillator
Feedback loop

(3) Connect CRO at demodulator o/p and observe FM Demodulated wave form on one
channel
While on other Channel. Observe the sine wave of signal generator.
(4) Adjust the ten turn potentiometer of FM demod kit to get same modulating
frequency.
The o/p of demodulated signal will be low in amplitude.
(5) Connect demodulated signal to the built in amplifier and adjust the gain to get both
modulated
I/p and demodulated o/p same in amplitude.
(6) Vary the frequency of carrier by changing c2 or c3
OBSERVATION TABLE:-
MODULATION
AMPLITUDE FREQUENCY
CARRIER
MODULATING
SIGNAL
CONCLUSION:-
DEMODULATION
AMPLITUDE FREQUENCY
MODULATING
SIGNAL
DEMODULATING
SIGNAL

OUTPUT WAVEFORM
EXPERIMENT NO.3

ANALOG SIGNAL SAMPLING & RECONSTRUCTION KIT.
OBJECTIVE:
To study different types of signal samplings and its reconstruction.
1) Natural Sampling.
2) Sample and Hold.
3) Flat top sampling.
EQUIPMENTS:
DCL –01 Kit.
Connecting Chords.
Power supply.
THEORY:
The kit is used to study Analog Signal Sampling and its Reconstruction. It basically
consists of functional blocks, namely Function Generator, Sampling Control Logic, Clock
section, Sampling Circuitry and Filter Section.
FUNCTION GENERATOR:
This Block generates two sine wave signals of 1 KHz and 2 KHz frequency. This
sine wave generation is done by feeding 16 KHz and 32 KHz clock to the shift register.
The serial to parallel shift register with the resistive ladder network at the output
generates 1 KHz and 2 KHz sine waves respectively by the serial shift operation. The
R-C active filter suppresses the ripple and smoothens the sine wave. The unity gain
amplifier buffer takes care of the impedance matching between sine wave generation and
sampling circuit. SAMPLING CONTROL LOGIC:
This unit generates two main signals used in the study of Sampling Theorem, namely the
analog signals (5V pp, frequency 1 KHz and 2 KHz) and sampling signal of frequency 2 KHz,
4 KHz, 8 KHz, 16 KHz, 32 KHz, and 64 KHz. The 6.4 MHz Crystal Oscillator generates the

6.4 MHz clock. The decade counter divides the frequency by 10 and the ripple counter
generates the basic sampling frequencies from 2 KHz to 64 KHz and the other control
frequencies.
From among the various available sampling frequencies, required sampling frequency is
selected by using the Frequency selectable switch. The selected sampling frequency is
indicated by means of corresponding LED.
CLOCK SECTION:
This section facilitates the user to have his choice of external or internal clock feeding
to the sampling section by using a switch (SW4).
SAMPLING CIRCUITRY:
The unit has three parts namely, Natural Sampling Circuit, Flat top Sampling
Circuit, and Sample and Hold Circuit.
The Natural sampling section takes sine wave as analog input and samples the analog input
at the rate equal to the sampling signal.
For sample and hold circuit, the output is taken across a capacitor, which holds the level
of the samples until the next sample arrives. For flat top sampling clock used is Inverted
to that of sample and hold circuit. Output of flat top sampling circuit is pulses with flat
top and top corresponds to the level of analog signal at the instant of rising edge of the
clock signal.
FILTER SECTION:
Two types of Filters are provided on board, viz., 2nd Order and 4th Order Low
Pass Butterworth Filter

PROCEDURE:
• Connect power supply in proper polarity to the kit DCL-01 & switch it on.
• Keep all the switch faults (except switch 1) in OFF position.
• Connect the 1 KHz, 5Vpp Sine wave signal, generated onboard, to the
BUF IN post of the BUFFER.
•Connect the sampling frequency clock in the internal mode INT CLK using
switch (SW4).
•Using clock selector switch (S1) select 8 KHz sampling frequency. Using
switch SW2
•Select 50% duty cycle
• Connect BUF OUT post of the BUFFER to the IN post of the Flat
-Top Sampling block by means of the Connecting chords provided.
•Connect the OUT post of the Flat Top Sampling block to the input IN1 post
of the 2nd Order Low Pass Butterworth Filter.
DUTY CYCLE SWITCH POSITIONS

OBSERVATION:
1 KHz Analog Input waveform.
Sampling frequency waveform.
Flat Top signal and its corresponding reconstructed output of 2nd order Low

Pass Butterworth Filter.
CONCLUSION:
Comparing the reconstructed output of 2nd order Low Pass Butterworth Filter for all
the three types of sampling, it is observed that the output of the Sample and Hold is
the best as compared to the output of Natural Sampling and the output of the Flat
Top Sampling.
PULSE WIDTH MODULATION

AIM: - To Study Pulse Width Modulation & Demodulation
APPARATUS REQUIRED: - Trainer kit, CRO, probes , patch cords etc.
THEORY: - In PWM, we have fixed amplitude and each pulse width is made
proportional to the amplitude of the signal at that instance. The carrier wave is in the
form of rectangular pulses. The width of these pulses is directly proportional to the
modulating signal strength. PWM can be done in three ways, first the center may be
fixed and both edges of the pulses are moved to compress and expand the width.
Second the lead edge can be held fixed and rail edge is modulated. Third, the trail
edge is kept fixed and lead edge is modulated.
PWM modulator is basically a monostable multivibrator with modulating voltage is
applied at the control input. In the diagram shown the fixed width pulses are
added with the ramp signal generated by the integrator in an adder circuit. This
signal is compared with a reference signal. Output of the comparator circuit giving
the pulses whose width is varying in accordance with the modulating signal.
Pulse Width modulation is used to transmit analog signal information like speech, data,
etc. Here signals are sampled at regular intervals and signal information is sent only at
time sampling time delay along with any synchronizing pulse original wave is obtained
from information regarding sample. So they are taken frequently enough.

THEORY & CIRCUIT DETAILS PULSE WIDTH MODULATION &
DEMODULATION
The PAM Modulation and Demodulation System consist of following sections.
1. Modulating Audio Signal Generator section
2. Sampling Pulse Generator section
3. Pulse Width Modulator section - Comparator section
4. DC voltage Source
5. Pulse Width Demodulator section - Comparator section & Low pass filter section
6. Power supply section.
(1) Modulating Audio Signal Generator section: -
1C 8038 - waveform generator - is used generate sine wave signal. 10K Pot is used to
vary its frequency. The frequency range is 300 Hz to 3.4 KHz. Two 100K presets
are adjusted for proper peaks of sine wave signal. IK preset is used to adjust
duty cycle. The sine wave output signal available at pin 2 of 1C 8038 is given to
1C 356 through Amplitude pot for amplification. The amplified sine wave signals
from pin 6 of 1C 356 are then available at "SINE" terminal. 22k Pot is used to
vary the amplitude of Sine wave signal. The output amplitude varies from 0 to
5Vpp.
(2) Sampling Pulse Generator section:-
To generate PWM signal comparator circuit is used. The sampling saw tooth waveform
is required to provide sampling signal to comparator circuit. This section is based on
voltage controlled oscillator (VCO) 1C CD4046B. The pulse frequency is determined by
varying control voltage of VCO at pin 9 of this 1C. This control voltage is varied by
frequency pot (10K Pot). The pulse shape of this signal is changed to saw tooth
waveform by using mono-stable 1C LF356. The output is available at pin 6. The pulse
frequency can be varied from 2 KHz to 32 KHz.
(3) Pulse Width Modulator section:-
To generate PWM signal comparator circuit is used. Here 1C LM 311 is used as
comparator. 10K Pot is used to vary the amplitude of its output. The modulating input
signal is given to one input of comparator. The sampling saw tooth waveform is given to

second input of comparator. The output signal of comparator produces PWM signal,
which is taken out through attenuator Pot.
(4) DC voltage Source:-
To see the effect of DC voltages on PWM modulated signal -8V to +8V DC voltage is
required. These voltages are obtained from +15 V and -15V DC of Power supply by
potential divider made of 100K presets and variable POT P 401 (100K pot). By varying
this Pot -8V to + 8V dc are available at VARDC socket.
(5) Pulse Width Demodulator section:-
This section is based on comparator low pas filter
(a) Comparator:-
The PAM signal is given to one input of comparator. The 1-volt DC reference signal is
given to second input of comparator. This 1 volt reference signal is generated by 1C
741 by setting 4K7 preset. Then output of comparator is pulse width demodulated
signal.
(b) Low pass filter:
This pulse width demodulated signal is then passed through Low pass filter made of
three 741 1C. The Lo pass filter passes only low frequencies up to 3.4 KHz and
reduces all other frequencies. Thus this removes high frequency quantization noise of
PWM signal. By removing high frequency we recover original modulating signal.
(6) Power supply section:-
The regulated power supply is used for different supply voltages.
Following output D.C. Voltages are required to operate PWM Modulation demodulation
system.
+15V, 250mA, -15V, 250mA, + 5V, 250mA
Three terminal regulators are used for different output voltages i.e. 1C 7805 for
+ 5V, 1C 7815 for +15V, K 7915 for-15V,
These ICs are supplied different dc input voltages by two Half-wave rectifiers
consisting of D1-D4 and D5- D8 and Cl, C2, C3, C4. The capacitors at each input
and each output are for filtering purpose. SW 301 is main AC ON/OFF Switch.

EXPERIMENT PROCEDURE;-
1.) Connect signal input terminal of modulator section to Sine O/P terminals of
Audio frequency generator.
2.) Connect CRO at Sine wave output signal of Audio frequency generator. Set this
Audio signal frequency to 2 KHz by Freq. Pot and amplitude to 1 Vpp.
____Waveform (Tl)
3.) Keep Frequency pot of Sampling pulse generator in mid position.
4.) Keep Frequency pot of Sampling pulse generator in mid position.
5.) Observe sampling clock and Saw tooth signals. ____Waveform
(T2)______Waveform (T3)
6.) Connect CRO Channel 1 at PWM O/P signal.
7.) Observe PWM signal ______Waveform (T4)
8.) Connect CRO Channel 2 at demodulated output of demodulator section.
9.) Observe recovered Sine wave signal. _____Waveform (T5)
10.) Connect CRO Channel 1 Sine wave input to modulator section.
11.) Now vary amplitude of sine wave modulating signal and observe its effect on
PWM output as well as on recovered signal.
12.) Vary frequency of sine wave modulating signal and observe its effect on PWM
output as well as on recovered signal.
13.) Vary Pulse frequency of sampling pulse and see the effect on PWM output as
well as on recovered signal.
14.) Vary attenuator pot and see its effect on recovered signal.
15.) Now Connect variable DC signal at input terminal from VARDC source. Vary
the DC volts from -8 to +8V by control POT and see its effect on the
modulator output.

16.) To verify Nyquist's Sampling Theorem keep modulating sine wave
frequency to 2 KHz and amplitude 2Vpp Keep Sampling pulse width pot in mid
position. Now reduce sampling frequency slowly from 32 KHz to 2 KHz by
observing original signal and recovered demodulated signal. Measure the
sampling frequency for which original signal & recovered demodulated signal are
nearly same i.e. error is less. It will be more than 4 KHz which proves
Nyquist's Sampling Theorem.
Conclusion;
1. The error in recovered signal:
Increases with increase in signal amplitude, increases with increase in signal
frequency, decreases with increase in sampling pulse frequency.
2. The attenuation of amplitude of PWM signal has no effect no recovery of
modulating signal as the information is transmitted in pulse width of carrier
signal.
Output wave forms:-

EXPERIMENT NO.4
AM USING COMMSIM SOFTWARE
OBJECTIVE: Simulation of AM using COMMSIM
APPARATUS:-Computer loaded with Commsim 7 software , Printer.etc
THEORY:
In Commsim, you build system models in the form of block diagrams. Blocks are your
basic design component. Each block represents a specific mathematical function. The
function can be as simple as a sin function or as complex as a 15th order transfer
function.
Commsim offers over 90 blocks for linear, nonlinear, continuous, discrete-time, time
varying, and hybrid system design. Blocks are categorized under the Blocks menu as
follows: Animation, Annotation, Arithmetic, Boolean, DDE, Integration, Linear

Systems, Mat Lab Interface, Matrix Operations, Nonlinear, Optimization, Random
Generator, Signal Consumer, Signal Producer, Time Delay, Transcendental.
In addition, Commsim supplies five special-purpose blocks: embed, expression, user
Function.If your design requirements extend beyond the blocks supplied by Commsim,
you can create custom blocks in C, C++, FORTRAN, or Pascal
By wiring blocks together, Commsim is able to pass signals among blocks during a
simulation. Signals are simply data. Input signals (xn) represent data entering blocks;
output signals (yn) represent data exiting blocks.
Commsim offers two types of wires: Flex Wires, Vector wires
A flex Wire is a thin wire that allows a single signal to pass through it. A vector wire,
on the other hand, is a thick wire that contains multiple flex Wire. Typically, you use
vector wires when performing vector or matrix operations, or to reduce wiring clutter
at top-level diagram design.
You can manually bundle and unbundle flex Wires using the scalar To Vector and vector
To Scalar blocks.
You can alternatively use variable blocks to pass signals. A variable lets you name and
transmit a signal throughout a block diagram without using wires. Typically, you use a
variable block for system-wide variables or signals that would be laborious or visually
messy to represent as wires.
You attach flex Wires and vector wires to blocks through their connector tabs. Once
you have attached a wire to a block, Commsim maintains the connection even as you
move the block around the screen.
When you wire blocks, the following rules are in effect:
·Wires can only be drawn between an input and output connector tab pair. The
triangular shape of the connector tab lets you easily distinguish inputs from outputs.
·Input connector tabs can only have one wire attached to them; output connector tabs
can have any number of wires attached to them.
·If you draw multiple wires between two blocks, Commsim automatically skews them.
All blocks that operate on signals have connector tabs. Input connector tabs enable
signals to enter a block; output connector tabs enable signals to exit a block. The
triangular shape of the connector tab lets you easily see the direction in which the
signals travel.

Some blocks have symbols on their connector tabs that indicate how the block acts on
the data or the type of data the block is expecting
Setting up the simulation range involves choosing the start and end of the simulation,
specifying the step size of the integration algorithm, indicating whether Commsim runs
in real-time mode, and indicating whether Commsim automatically restarts the
simulation either with or without the last known system states.
Commsim can simulate linear, nonlinear, continuous, and discrete systems. Commsim can
also simulate systems containing both continuous and discrete transfer functions, as
well as systems containing multi-rate sampling for discrete transfer functions.
When you initiate a simulation, Commsim first evaluates Signal Producer blocks, like
constants and ramps, and then sends the data to intermediate blocks that have both
inputs and outputs, like gains and summing Junctions. Lastly, it sends data to Signal
Consumer blocks that have only inputs, such as plots and meters.
Commsim simulates a system according to:
·Simulation parameters set in the dialog box for the Simulate > Simulation Properties
command
·Initial conditions for the system set in the applicable blocks
If the status bar is turned on, Commsim displays current settings for the simulation
range, step size, elapsed simulation time, integration algorithm, and implicit solver.
In Commsim, discrete and continuous time blocks can be used together in a model.
Such systems are called hybrid systems. In hybrid systems, the outputs of the
discrete blocks are held constant between successive sample times, and updated at
times that correspond to the specified discrete sample time. The outputs of
continuous blocks are updated at every time step. Similarly, the inputs to the discrete
blocks are updated at times that correspond to the discrete time interval while the
inputs to continuous blocks are updated at every time step.
The plot block displays data in a two-dimensional time domain plot. You can customize
the plot and control how data is in the following ways:
·Choose between XY or frequency domain

·Select logarithmic scaling, fixed axis bounds, or a time axis scale
·Display signal traces as individual data points, line segments, or stepped line segments
·Overlay signal traces with geometric markers
·Specify the number of data points to plot
·Use crosshairs and grid lines to determine data point coordinates
·Overlap plots
AM MODULATION :
AM is the process of changing the amplitude of a high frequency carrier signal in
proportion with the instantaneous value of modulating signal .
Carrier is high frequency sinusoidal signal. Am can be measured as percent modulation.
Percent modulation gives us the depth in modulation. It depends on modulation index
(m) of AM signal.
Percent modulation = (vm/vc)*100
= vmax-vmin *100
Vmax+vmin
• When Vm <Vc , then modulation is called Undermodulation.
• When Vm=Vc , then modulation is called 100%rmodulation.
• When Vm >Vc , then modulation is called Overmodulation.
Procedure:
1. Open commsim 7 software.
2. Click on the ‘Comm’ from menubar and select the sine wave signal from the signal
generator block source.
3. In the same ‘comm’ option select AM block from the modulator complex.
4. Set modulating and Carrier Frequency amplitude.
5. Now add ‘Complex to real part converter’ block after AM block and connect all
the blocks.
6. Add a plot window to plot output.
7. Adjust simulation properties time scale and starting frequency.
8. Give names to X and Y axis and set the values.
9. Simulate and observe the waveform plot.

10. For multiple plots in one window, change the settings to add the desired no. of
pots.
11. Then change the amplitude of the carrier and observe the waveform for
different modulation index.

• Block Diagram of Amplitude Modulation

AMPLITUDE MODULATION SIMULATION WAVEFORM

• OUTPUT WAVEFORMS:
m=1
m<1

m>1
CONCLUSION:
EXPERIMENT NO 5
FM USING COMMSIM SOFTWARE
OBJECTIVE: Simulation of FM using commsim
APPARATUS: Computer loaded with Commsim 7 software , Printer.etc
THEORY:
In Commsim, you build system models in the form of block diagrams. Blocks are your
basic design component. Each block represents a specific mathematical function. The
function can be as simple as a sin function or as complex as a 15th order transfer
function.
Commsim offers over 90 blocks for linear, nonlinear, continuous, discrete-time, time
varying, and hybrid system design. Blocks are categorized under the Blocks menu as
follows: Animation, Annotation, Arithmetic, Boolean, DDE, Integration, Linear
Systems, Mat Lab Interface, Matrix Operations, Nonlinear, Optimization, Random
Generator, Signal Consumer, Signal Producer, Time Delay, Transcendental.
In addition, Commsim supplies five special-purpose blocks: embed, expression, user
Function.If your design requirements extend beyond the blocks supplied by Commsim,
you can create custom blocks in C, C++, FORTRAN, or Pascal
By wiring blocks together, Commsim is able to pass signals among blocks during a
simulation. Signals are simply data. Input signals (xn) represent data entering blocks;
output signals (yn) represent data exiting blocks.
Commsim offers two types of wires:Flex Wires, Vector wires

A flex Wire is a thin wire that allows a single signal to pass through it. A vector wire,
on the other hand, is a thick wire that contains multiple flex Wire. Typically, you use
vector wires when performing vector or matrix operations, or to reduce wiring clutter
at top-level diagram design.
You can manually bundle and unbundle flex Wires using the scalar To Vector and vector
To Scalar blocks.
You can alternatively use variable blocks to pass signals. A variable lets you name and
transmit a signal throughout a block diagram without using wires. Typically, you use a
variable block for system-wide variables or signals that would be laborious or visually
messy to represent as wires.
You attach flex Wires and vector wires to blocks through their connector tabs. Once
you have attached a wire to a block, Commsim maintains the connection even as you
move the block around the screen.
When you wire blocks, the following rules are in effect:
·Wires can only be drawn between an input and output connector tab pair. The
triangular shape of the connector tab lets you easily distinguish inputs from outputs.
·Input connector tabs can only have one wire attached to them; output connector tabs
can have any number of wires attached to them.
·If you draw multiple wires between two blocks, Commsim automatically skews them.
All blocks that operate on signals have connector tabs. Input connector tabs enable
signals to enter a block; output connector tabs enable signals to exit a block. The
triangular shape of the connector tab lets you easily see the direction in which the
signals travel.
Some blocks have symbols on their connector tabs that indicate how the block acts on
the data or the type of data the block is expecting.
Setting up the simulation range involves choosing the start and end of the simulation,
specifying the step size of the integration algorithm, indicating whether Commsim runs
in real-time mode, and indicating whether Commsim automatically restarts the
simulation either with or without the last known system states.

Commsim can simulate linear, nonlinear, continuous, and discrete systems. Commsim can
also simulate systems containing both continuous and discrete transfer functions, as
well as systems containing multi-rate sampling for discrete transfer functions.
When you initiate a simulation, Commsim first evaluates Signal Producer blocks, like
constants and ramps, and then sends the data to intermediate blocks that have both
inputs and outputs, like gains and summing Junctions. Lastly, it sends data to Signal
Consumer blocks that have only inputs, such as plots and meters.
Commsim simulates a system according to:
·Simulation parameters set in the dialog box for the Simulate > Simulation Properties
command
·Initial conditions for the system set in the applicable blocks
If the status bar is turned on, Commsim displays current settings for the simulation
range, step size, elapsed simulation time, integration algorithm, and implicit solver.
In Commsim, discrete and continuous time blocks can be used together in a model.
Such systems are called hybrid systems. In hybrid systems, the outputs of the
discrete blocks are held constant between successive sample times, and updated at
times that correspond to the specified discrete sample time. The outputs of
continuous blocks are updated at every time step. Similarly, the inputs to the discrete
blocks are updated at times that correspond to the discrete time interval while the
inputs to continuous blocks are updated at every time step.
The plot block displays data in a two-dimensional time domain plot. You can customize
the plot and control how data is in the following ways:
·Choose between XY or frequency domain
·Select logarithmic scaling, fixed axis bounds, or a time axis scale
·Display signal traces as individual data points, line segments, or stepped line segments
·Overlay signal traces with geometric markers
·Specify the number of data points to plot
·Use crosshairs and grid lines to determine data point coordinates
·Overlap plots
FM MODULATION:

The FM is the process of varying the frequency of high frequency carrier in
accordance with the amplitude of modulating signal.
FREQUENCT DEVIATION:
It is the change in frequency that occurs in carrier when it is acted upon by
modulating signal. It is nothing but peak frequency shift in Hz.
Modulation Index = Δf / Fm
Here modulation index is directly proportional to frequency deviation Δf and inversely
proportional to the modulating frequency.
PROCEDURE:
1. Open commsim 7 software.
2. Click on the ‘Comm’ from menu bar and select the sine wave signal from the
signal generator block source.
3. In the same ‘comm’ option select FM block from the modulator complex.
4. Set modulating and Carrier Frequency amplitude .
5. Now add ‘Complex to real part converter’ block after FM block and connect all
the blocks.
6. Add a plot window to plot output.
7. Adjust simulation properties time scale and starting frequency.
8. Give names to X and Y axis and set the values.
9. Simulate and observe the waveform plot.
10. For multiple plots in one window, change the settings to add the desired no. of
pots.
11. Then change the frequency deviation of the carrier and observe the waveform
for different modulation index .

Block Diagram Of Frequency Modulation And Demodulation

Simulation Properties

OUTPUT:
Modulating Signal
Frequency Modulated Signal
Demodulated Signal
CONCLUSION:

EXPERIMENT NO 7
STUDY OF PULSE CODE MODULATION AND DEMODULATION
AIM:
To study the operation of PCM Transmitter
EOUlPMENTS REQUIRED:
1) Pulse code modulation and demodulation kit
2) 4 mm Patch cords
3) Sine wave generator
4) CRO or DSO

THEORY:
The major form of DPM is pulse code modulation (PCM). Its primary advantage is much
better noise and interference immunity. In PCM the modulating signal is sampled, the
sample amplitude i s converted into a binary code and the binary code is transmitted in
groups as a train of pulses. A major difference is that in PCM, the sampled amplitude must
be transmitted as a binary number out of a limited range of binary numbers. To
accomplish this, each s a m p l e must first be converted t o the nearest standard
amplitude, called the quantum. This process of sample conversion i s called quantizing t h e
signal. A model for quantizing a modulating waveform i s shown in Figure 1 Here, we use
eight quantization l e v e l s , ranging from 0 to 1 where each level represents one volt.
Table 1 shows the binary number and the 3-bit pulse code represented by each of the
quantization levels. In figure 1 (A), we see that many of the sampling points are not
at a quantum level In those instances, the sample amplitude are represented by the nearest
quantum level. For example, the amplitude of sampling pulse 1 is represented by quantum
level 4. Another example is sampling pulse 6. Where the amplitude is about 0.4 V, which is
represented by quantum level 0. The error that ,is the difference between sampling point
amplitude and quantum levels is a distortion called quantization noise because the
errors are random. Stated another way, t he differences b e t w e e n a n y
quantum level and the amplitude of the signal at any instant is unpredictable.
Quantization n o i s e can be reduced by increasing the number of Quantization
levels.

However, a disadvantage of increasing the number of levels is an increased
transmission bandwidth requirement. Therefore, a compromise must be made
between an acceptable transmission bandwidth and acceptable quantization noise.
For example, the noise generated by a 3-bit code may be acceptable for the
narrow bandwidth used for voice transmission but not for the much greater
bandwidth required for television transmission.
After Quant izati on and before transmission as a PCM signal, each sample i s
coded as a binary number. Coding the quantized waveform of Figure 1 (A) is shown
in Fig 1(B) the eight quantum levels used are represented by 3-bit binary words,
as shown in Table. 1.
After the quantized waveform is coded, each sequential sample is
transmitted as a pulse code, shown as the 3-bit PCM pulse train in Figure 1 (C).
In practical PCM systems, the 3-bit word for this quantizing model is seldom used because of
the quantization noise introduced. Instead, 8-bit words are more common because they
provide 256 quantum levels, and therefore allow much better reproduction of the modulating
signal with very little quantization noise. Also, in practical PCM systems, synchronizing pulses
are transmitted with the pulse train to ensure that the receiver decodes the information
pulses correctly.
PROCEDURE:
1) Connect the s i g n a l f r o m the s i n e wave generator to the input ofPCM
modulator. Adjust the level potentiometer & verify that output of clamping
circuit is a DC sine wave.
2) Now vary the input amplitude f r o m minimum to maximum such that all LED of A
to D converter are ON. This indicates that analog input is in 8 bit data form.
3) Note the value of each data output for different amplitude of sine wave.
4) Now observe the PCM data at the final output of PCM.
5) Connect the Trainer kit of PCM Tx & PCM Rx to main supply and switch them ON.
6) Short the common terminal of both the kit & also short the following
PCM Tx PCM Rx
CLOCK PULSE CLOCK PULSE (62.5 Khz)
Monostable MV Latch Enable Pulse
7) Select signal of 200 Hz/ 2 volts AC peak to peak from sine wave generator and
connect it to analog input terminals of the PCM kit.

8) Connect PCM output to PCM input of receiver and observe the received analog
output. Both the input and output should be same except some delay because of filter.
Repeat the experiment with frequency of 100 Hz.
9) Refer to the theory for circuit analysis and draw the waveform.
CONCLUSION:-
WAVEFORM:-

• Input waveform (Blue trace colour)
• Clamper output waveform (pink trace colour)
• Input waveform (Blue trace colour)

• Clamper output waveform (pink trace colour)
• PCM Output waveform (yellow colour)
• Master clock waveform (Blue trace colour)-
• Clock waveform (pink trace colour)-
• RD waveform (yellow trace colour)-
• PL waveform (Green trace colour)-

• DAC (Sample &Hold ) waveform (Blue trace colour)-
• Filter output waveform (pink trace colour)-
• Input waveform (yellow trace colour)-

DELTA MODULATION AND DEMODULATION
OBJECTIVE:
Study of Delta Modulation and Demodulation.
EQUIPMENTS:
• DCL –07 kit.
• Connecting chords.
• Power supply.
• CRO/DSO
CIRCUIT DIAGRAM:-

THEORY:
DELTA MODULATION: Delta modulation is the differential pulse code Modulation
scheme in which the difference signal is encoded into just a single bit. In digital
modulation system, the analog signal is sampled and digitally coded. This code
represents the sampled amplitude of the analog signal. The digital signal is sent to the
receiver through any channel in serial form. At the receiver
the digital signal is decoded and filtered to get reconstructed analog signal.
Sufficient number of samples is required to allow the analog signal to be
reconstructed accurately. Delta modulation is a process of converting analog signal
into one bit code, means only one bit is sent per sample. This bit indicates whether
the signal is larger or smaller than the previous samples. The advantage of DM is that
the modulator and demodulator circuits are much simpler thanThose used in
traditional PCM. Delta modulation is an encoding process where the logic levels of the
transmitted pulses indicate whether the decoded output should rise or fall at each
pulse. This is a true digital encoding process as compare to PAM, PWM and PPM. If
signal amplitude has increased in DM then modulated output is a logic level 1. If the
signal amplitude has decreased the modulator output is logic level 0. Thus the output
from the modulator is a series of zeroes and ones to indicate rise and fall of the
waveform from the previous value. The block diagram (Fig. 1.1) of Delta Modulation
illustrates the components at the transmitter end. It consists of Digital Sampler and
anIntegrator at the feedback path of Digital sampler. Let assume that the base band
signal a (t) and its quantized approximation I(t) are applied as inputs to the
comparator. A comparator as its name Suggests simply makes a comparison between
inputs. The comparator has one fixed output c (t) when a (t) is greater than i (t) and
the different output when a (t) is less than i(t) the comparator output is then latched
in to a D-flip/flop which is clocked by the selected transmitterclock. Thus the output

of the D-flip/flop is latched 1 or 0 synchronous with the clock edge. This binary data
stream is transmitted to the receiver and is also fed to the input of integrator. The
integrator output is then connected to the negative terminal of voltage comparator,
thus completing the modulator circuit.
DELTA DEMODULATOR: The Delta Demodulator consists of a D-flip/flop, followed
by an integrator and a 2nd and 4th order low pass Butterworth filter. The Delta
Demodulator receives the data stream from D-flip/flop of Delta Modulator.
It latches this data at every rising edge of receiver clock. This data stream is then
fed to integrator; its output tries to follow the analog signal in ramp fashion and
hence is a good approximation of the signal itself. The integrator output contains
sharp edges, which is smoothened out by the 2nd order, and 4th order low pass
Butterworth filter whose cut-off frequency is just above the audio band.
The practical use of Delta Modulation is limited due to following drawbacks:
1) NOISE: A noise is defined, as any unwanted random waveform accompanying
the information signal. When the signal is received at the receiver irrespective of any
channel it is always accompanied by noise.
2) DISTORTION: Distortion means that the receiver output is not the true copy
of the analog input signal at the transmitter. In Delta modulation, when the analog
signal is greater than the integrator output the integrator ramps up to meet the
analog signal. The ramping rate of integrator is constant. Therefore if the rate of
change of analog input is faster than the ramping rate, the modulator is unable to
catch up with the input signal. This causes a large disparity between the information
signal and its quantized approximation. This error phenomenon is known as Slope over
loading and causes the loss of rapidly changing Information. The slope overloading

waveform is as shown in the figure. The problem of slope overload can be solved by
increasing the ramping rate of the integrator. But as it can be seen from the figure
the effect of the large step size is to add large sharp edges at the integrator output
and hence it adds to noise.
3) Another problem of Delta Modulation is that it is unable to pass DC
Information. This is not a serious limitation of the speech communication.
PROCEDURE:
 Refer to Block Diagram & Carry out the following connections.
 Connect the power supply with the proper polarity to the Kit DCL-07 and switch
it ON.
 Keep the switch faults in OFF position
 Select sine wave input 250Hz of 0V through pot P1 and connects post 250Hz to
post IN
Of input buffer.
 Connect output of buffer post OUT to Digital Sampler input post IN1.
 Then select clock rate of 8 KHz by pressing switch S1 selected clock indicated
by LED glow.
 Keep Switch S2 in Delta position.
 Connect output of Digital Sampler post OUT to input post IN of Integrator.
 Connect output of Integrator 1 post OUT to input post IN2 of Digital Sampler.
 Observe the Delta modulated output at output of Digital Sampler post OUT and
compare it with the clock rate selected. It is half the frequency of clock rate
selected.
 Observe the integrator output test point. It can be observe that as the clock

rate is increased amplitude of triangular waveform decreases. This is called minimum
step size.
 Increase the amplitude of 250Hz sine wave up to 0.5V. Signal approximating
250Hz is available at the integrator output. This signal is obtained by integrating the
digital output resulting from Delta modulation.
 Go on increasing the amplitude of selected signal through the respective pot
from 0 to 2V.
It can be observed that the digital high makes the integrator output to go upward and
digital low makes the integrator output to go downwards. Observe that the
integrator output follow the input signal.
 Increase the amplitude of 250Hz sine wave through pot P1 further high and
observe that the integrator output cannot follow the input signal. State the reason.
Select the clock rate of 32 KHz using switch S1.
 Repeat the above mention procedures with different signal sources and
selecting the different clock rates and observe the response of Delta Modulator.
 Connect Delta modulated output post OUT of Digital Sampler to the input of
Delta Demodulator section post IN of Demodulator
 Connect output of Demodulator post OUT to the input of Integrator 3post IN
 Keep Switch S4 in high position.
 Connect output of Integrator 3 post OUT to the input of output buffer post IN
 Connect output of output buffer post OUT to the input of 2nd order filter post
IN.
 Connect output of 2nd order filter post OUT to the input of 4torder filter
post IN.

OBSERVATION:
Sampling clock.
Input Signal.
Integrator 1 output at feedback loop for Delta modulator.
Digital sampler Output.

Integrator 3 output.
Filter Outputs

TIME DIVISION MULTIPLEXING AND DEMULTIPLEXING

STUDY OF TDM PULSE AMPLITUDE MODULATION / DEMODULATION.

OBJECTIVE:
To study Time Division Multiplexing and Demultiplexing, using Pulse Amplitude
Modulation and Demodulation and to reconstruct the signals at the Receiver, using
Filters. The Transmitter Clock and the Channel Identification Information is linked
Directly to the Receiver.
EQUIPMENTS REQUIRED:
• DCL-02 KIT
• Connecting Chords.
• Power supply.
• CRO/DSO
THEORY:
1. The Onboard Function Generator,
2. The Transmitter,
3. The Receiver with the associated synchronization circuitry.
ONBOARD FUNCTION GENERATOR:
This basically provides four Amplitude Variable each (0 - 5 V) synchronized sine
waves, each 250Hz, 500Hz, 1KHz, and 2Khz and an amplitude variable DC level (0-5V).
TRANSMITTER:
The Transmitter Section consists of four Analog Input signals from the Function
generator fed to the four channels of the Multiplexer where the signals fed are Time
Division Multiplexed after undergoing the sampling. The sampling process makes the
signals Pulse Amplitude Modulated. The frequencies for sampling are given from the
decoder.
RECEIVER:
The Receiver Section consists of a Demultiplexer that demultiplexes the four Time
Division Multiplexed signals, which it receives from the transmitter. This
Demultiplexed signals are then fed to the reconstruction circuit, which is the filter
section. The receiver timing logic is very similar to the transmitter timing logic. The

demultiplexer based on the control signals C0, C1, C2, C3 assigns the information to
the corresponding channels. The success of the demultiplexer operation is fully
dependent on how exactly, RXCH0, RXCH1, RXCH2, RXCH3 Signals match with the
TXCH0, TXCH1, TXCH2, TXCH3 signals. Thus, to ensure the proper demultiplexing,
two dividers are reset by the RXCH0 signal, which corresponds with the TXCH0. The
demultiplexed signals are then given to the corresponding reconstruction units. The
signal reconstruction unit is a 4th order Active Low Pass Butterworth Filter provided
for each receiver channel. They filter out the sampling frequency and their
harmonics from the demultiplexed signal and recover the base band by an integrate
action. The cut-off frequency of the 4th Order Low Pass Butterworth Filter is
3.4KHz
PROCEDURE:
 Refer to Block Diagram & Carry out the following connections and switch
settings.
 Connect power supply in proper polarity to the kit DCL-02 & switch it on.
 Keep all the switch faults in off position
 Connect 250Hz, sine wave signal from the Function Generator to the
multiplexer inputs channel CH0, by means of the connecting chords
provided.
 Connect 500Hz, sine wave signal from the Function Generator to the
provided.
 Connect 1 KHz, sine wave signal from the Function Generator to the
provided.
 Connect 2 KHz, sine wave signal from the Function Generator to the
provided
 Set the amplitude of the input sine wave as desired.
 Connect the multiplexer output TXD of the transmitter section to the
demultiplexer input RXD of the receiver section.

Connect the sampling clock TX CLK of the transmitter section to the
corresponding RX CLK of the receiver section respectively
 Connect the Channel Identification Clock TXSYNC of the transmitter
section to the corresponding RX SYNC of the receiver section
respectively.
 Connect the output of the receiver section CH0 to the IN0 of the filter
section.
section.
section.
section.
 Observe the reconstructed output of filters at out 0.
Observe the reconstructed output of filters at out 3.
OBSERVATION:- Transmitter
Sr no Channel Amplitude Time Frequency
1 CH0
2 CH1
3 CH2
4 CH3
RECIEVER
Sr no Channel Amplitude Time Frequency
1 CH0
2 CH1
3 CH2
4 CH3

CONCLUSION:
In this experiment, the transmitter clock and the channel identification
clock (Sync) are directly linked to the receiver section. Hence transmitter and
receiver are synchronized and proper reconstruction of the signal is achieved.
• CHO input waveform (Blue trace colour)-
• CH1 input waveform (pink trace colour)-
• CH2 input waveform (yellow trace colour)-
• CH3input waveform (Green trace colour)-

• TX clk/RX CLK waveform (Blue trace colour)
• TX Sync/RX Sync waveform (pink trace colour)
Time division multiplexing output waveform

DEMULTIPLEXER OUTPUT
• CHO demultiplexer o/p waveform (Blue trace colour)-
• CH1 demultiplexer o/p waveform (pink trace colour)
• CH2 demultiplexer o/p waveform (yellow trace colour)
• CH3 demultiplexer o/p waveform (Green trace colour)
• OUT 0 waveform (Blue trace colour)-
• OUT 1 waveform (pink trace colour)-
• OUT 2 waveform (yellow trace colour)-
• OUT 3 waveform (Green trace colour)-

EXPERIMENT NO:- 10
FREQUENCY DIVISION MULTIPLEXING AND DEMULTIPLEXING
AIM; To study Frequency division multiplexing and Demultiplexing
APPARATUS REQUIRED: - FDMD trainer kit , CRO , probes patch cords etc.
CIRCUIT DIAGRAM:-
THEORY :
In many communication systems, a single large frequency band is assigned to the
system & is shared among a group of users example A Microwave transmission line
connection to sites over a long distance each site has a no. of sources generating
independent stream s that are transmitted simultaneously over the Microwave line ex. AM
& FM Radio Bands which are divided among many channel or stations the stations are
selected with the radio dial by tuning variable frequency filter.
FDM means that the total bandwidth to the system is
divided into a series of non overlapping frequency sub band that are arranged to each
communication system & user points each transmitter modulate s its source into a signal
that lies in a different frequency sideband
The block dig. For FDM would mainly consist of the following blocks: as shown in the figure
the signals are then transmitted across a common channel.

At the receiving end of the system ,BPF are used to
pass the appropriate signal to the desired user and to block all unwanted signal to ensure
that the transmitted signal do not spray outside their assigned sub bands , it is also common
to place appropriate pass band filters at the o/p stage of each transmitter it is also
appropriate to design an FDM system so that the BW allocated to each sub band is slightly
larger than the BW needed by each source .This extra b/w called a guard band allows
system to use less expensive filters .
The main advantage of FDM over TDM is that it is not sensitive to propagation delays. It
therefore require less complex channel equalization techniques .On the other hand , FDM
needs large no. of BPF which are expensive & complicated to construct and design . TDM
uses less simplex and complex design circuits.
Another disadvantage of FDM is that in many practical communication circuits , the
power amplifier in the transmitter has non linear characteristics. Non linear amplification
leads to creation of out- of- band spectral component that may interfere with other fdm
channel . Thus it is necessary to use more complex linear amplifier in fdm system.
FDM is used in commercial FM receiver radio. The frequency band 88-108Mhz is
divided into 200khz sub-bands so there can be up to 100 different radio station with each
station identified by the centre frequency within the channel.
Multiplexing is the transmission of information from more than one source on the same
media. In frequency^Siivision multiplexing (FDM), many information channels are
transmitted simultaneously, with each channel occupying a different frequency band .If
each information channel originally occupied the same frequency range, the frequencies
must be translated to different areas of the frequency spectrum before they are
combined. To achieve frequency separation, each channel amplitude-modulates a different
carrier
frequency. If a carrier is amplitude-modulated with a single frequency, the resultant
waveform is mathematically described as
Asinwct+mA/2cos(wc-wm)t-Ma/2cos (wc+wm)t
(a) (b) (c)
where A = peak carrier amplitude
m = modulation coefficient
fc = carrier frequency

fm = modulating frequency
wc=2Пfc
wm=2Пfm
Expression (a) is the original carrier frequency, (b) the lower side or difference frequency,
and (c) the upper side or sum frequency.
Frequency-division multiplexing.
{If a Carrier is amplitude - modulated by a band of frequencies, an upper and a lower
sideband are produced. The upper sideband (USB) is made up of the sum of the carrier
frequency and the individual frequencies present in the modulating signals: the lower
sideband (LSB) is the made up of the difference between the carrier frequency and the
individual frequencies present in the modulating signal
In amplitude modulation the carrier contains no intelligence: therefore, it is suppressed
through some form of balanced modulator ("Ring Modulator").
Since the upper and lower sideband contain identical information, the transmission of only a
single sideband is necessary to convey to the information. With FDM, a single sideband is
transmitted without the carrier. This signal is described as single-sideband suppressed
carrier (SSBSC)A An A-type (analog) channel bank performs frequency division multiplexing
of twelve-voice band channel. Each voice band channel can carry either voice information or
digital information from a modem. Each channel amplitude - modulates a different carrier
frequency. The lower sideband of each modulation process is extract and combines with the
lower sidebands from the eleven other channels to form a group (Figure 8-3). A group has a
bandwidth of 48 KHz (12 X 4 KHz) and occupies the frequency baud from 60 to 108 KH/..
Although each Voice channel is allocated a frequency range of 0 to 4 KHz, Signal
Information is normally limited to a 300-to 3000 Hz pass and. Consequently, a group has a
natural guard band of 1.3 KHz (Fig 8- 4) Between adjacent channel signals.
If further multiplexing is desire, five groups may be similarly combining to produce a super
group (SG). The bandwidth of an SG, which results from combining.
t^ = 300 Hz to 3 kHz fc - 60 kHz
60 kHz
57 to 59.7 kHz
The LSB of each mod. Process is , 240 kHz and extended from 312-552 kHz.
Super groups may be combine to form a master group . Master groups is made up of 10
super groups & contain information from 600 voice band channel

DETAILS FREQUENCY DIVISION MULTIPLEXING/DEMULTIPLEXING
[The Frequency Division Multiplexing/Demultiplexing System consists of following sections.
1. Modulating Audio Signal Generator - 1 |
2. Modulating Audio Signal Generator - 2
3. Sub Carrier Generator )
4. Double Balanced Amplitude Modulator -1
5. Double Balanced Amplitude Modulator -2
6. Band Pass Filter- 12-16 KHz j
7. Band Pass Filter - 28-32 KHz
8. Summing Amplifier
9. R.F. Carrier Oscillator
10. Main Final Amplitude Modulator I
11. Main Balanced Demodulator (Product Detector)
14. Channel-1 Amplitude Demodulator -1
15. Channel-1 Amplitude Demodulator -2
16. Low pass Filter -1
17. Low pass Filter -2
18. Power supply.
19.
1. Modulating Audio Signal Generator -1: -
1C 8038 waveform generator 1C is used generate sine wave signal. 10K Pot is used
to vary its frequency. The frequency range is 300 Hz to 3.4 KHz. Two 100K
Presets are adjusted for proper peaks of sine wave signal. IK reset is used to
adjust duty cycle. The sine wave output signal is available at pin 2 of 8038 and it
is then amplified by 1C 356. The amplified output is available output terminals 2?
K Pot is used to vary the amplitude of Sine rave signal. The output amplitude vary
from 0 to lO Vpp. 10K Pot is used to vary the frequency of output signal
2. Modulating Audio Signal Generator -2 :-
|This section is similar to above section 1
SUB CARRIER GENERATOR SECTION;

Two Sub Carrier signals of Frequencies 16 KHz and 32 KHz are required for
modulation of two modulating Audio signals. Here 1C 74HC04 is used to generate
1.28 MHz high frequency stable TTL signal. Then this signal I divided by lO by 1C
4017 to get 128 KHz signal. Then it is divided by 8 end 4 top get 16KHz and
32KHz rf carrier
4 Double Balanced Amplitude Modulator - 1:
IC 1496 is used as Double Side Band Suppressed Carrier amplitude modulator.
The modulating audio signal-1 is connected at pin 1 through buffer transistor Ql.
This 1C has two inputs as it works as it balanced modulator. The second input can
be connected at pin 4 through buffer transistor Q2. The RF Sub carrier signal is
connected at pin 8 through coupling capacitor from Sub carrier generator section.
The modulated outputs are available at pin 12 and 6 of this 1C, which are then
balanced amplified by Q3, Q4, Q5 and Q6. The final balanced modulated o/p
signal M-l is available at output terminal. Bal-A preset is used to balance carrier
signal while Bal-Preset is used to balance input audio signal. IK presets is used to
adjust output zero DC level.
The output M-l is DSB-SC output. It contains side bands at frequencies '16 + 4
KHz i e 12-16 KHz (Lower Side "and) and 16-20 KHz (Upper Side Band), because
input modulating frequency is maximum 4 KHz only and carrier is 16 KHz.
5. Double Balanced Amplitude Modulator - 2: -
this section is similar to above section 4. Here final balanced modulated output
signal M-2 is available at put terminal. It contains side band: at frequencies 32 +
4 KHz i.e. 28-32 KHz (Lower Side Band) and 32-36 z (Upper Side Band), because
input-modulating frequency is maximum 4 KHz only and Carrier is 32 KHz.
6 Band Pass Filter - 12-16 KHz section:-
The Band pass active filter is made by two Op-amps - 1C 353. It passes signal
between 12 to 16 KHz frequencies : DSB-SC output M-l is applied as a input to
this filter. As M-l signal has side bands 12-16KHz & 1620KHz I filter has band

pass of 12-16 KHz, Upper side band 16-20 KHz will be blocked by this filter and
Lower Side id 12-16 KHz will be available at the output II.
7 .Band Pass Filter - 28-32 KHz section: -
'he Band pass active filter is made by two Op-amps - 1C 353. It passes signal
between 28 to 32 KHz frequencies i DSB-SC output M-2 is applied as a input to
this filter. As M-2 signal has side bands 28-32 KHz & 32-36Khz 1 filter has band
pass of 28-32 KHz, Upper side band 32-36KHz will be blocked by this filter and
Lower Side id 28-32 KHz will be available at the output 12.
8. Summing Amplifier: -
this section is Op-amps adder. It adds two signals II and 12 of Band Pass filter
the added output is available at pin no 6 of IC356
9.R.F. Carrier Oscillator section :-
transistor Ql (BC107B) is used generate RF sine wave signal. Pot PI (15) is used to
van- its frequency. The frequency range is 200KHz to 1 MHz. Transistor Q2
(BC177), Q3 (BC107),
Q4 (BC177) & Q5 (BC107) are used to amplified the RF oscillation signal of Ql.
22pf trimmer capacitor and IK preset are adjusted for proper peaks of sine wave
signal. The amplified sine wave output signal is available at emitter’s junction of
Q4 and Q5. Pot P2 is used to vary the amplitude of Sine wave signal. The output
amplitude vary from 0 to lOVpp. Here 455 KHz frequency is set for final
modulation.
10) Main Final Amplitude Modulator: -
This section is similar to modulator -1 and modulator -2. Here one input is 455
KHz from main RF carrier Oscillator and other modulating input is final added
output (Two band pass signals- 12-16 & 28-32 KHz). The output this modulator is
Frequency Division Multiplexed DSB-SC signal. It is complex signals having many
side band d harmonics. This output is then given to "SIGMA" Frequency Division
Demultiplexing Trainer Model 3M128B for DemuItiplexing to recover original
modulating input signals.

11.Main Balanced Demodulator (Product Detector) : -
This section is similar to Final Amplitude Modulator used in-multiplexing section.
Here one input is 455 KHz, which is from multiplexing section, and other
modulating input is FDM DSB-SC signal, which also comes from multiplexing
section. The output of this product detector is mix signal of two modulated band
pass signals of frequency bands (12-16 & 28-32 KHz).
12) Band Pass Filter - 12-16 KHz section:-
between 12 to 16 KHz frequencies the output is modulated signal between band
12-16 KHz.
31 Band Pass Filter - 28-32 KHz section: -
between 28 to 32 KHz frequencies the output is modulated signal between band
28-32 KHz. and band pass active filter is made by two Op-amps - 1C 353. It
passes signal between 28 to 32 KHz frequencies -SC output M-2 is applied as a
input to this filter. As M-2 signal has side bands 28-32 KHz & 32-36Khs filter
has band pass of 28-32 KHz, Upper side band 32-36KHz will be blocked by this
filter and Lower Sid< 32 KHz will be available at the output 12.
14.Channel-1 Amplitude Demodulator - 1: -
This section is similar to used in multiplexing board. One input is from 28-32 KHz
filter section. The other i/p signal is 32 KHz synchronized carrier signal
Low generated by PLL section. The output of demodulator is mixed signal of input
audio-1 and audio-2
15. Channel-2 Amplitude Demodulator - 2: -
This section is similar to used in multiplexing board. One input is from 12-16 KHz
filter section. The other signal is 16 KHz synchronized carrier signal generated
by PLL section. The output of demodulator is mixer of input audio-1 and audio-2.

Pass Filter -1:-
Low pass filter is made by Op-amp - 1C 353. It accepts output from Amplitude
Demodulator-1. It passes signal below 4 KHz. The output is recovered Audio
signal-1
.
Low Pass Filter -2:-
Low pass filter is made by Op-amp - 1C 353. It accepts output from Amplitude
Demodulator-2. It passes below 4KHz. The output is recovered Audio signal-2.
19. Power supply section: -
The regulated power supply is used for different supply voltages, following output
D.C. Voltages are required to operate system.
+15, 250mA, -15v, 250mA , +5v. 250mA
three terminal regulators are used for different output voltages i.e. 1C 7805 for
+ 5V, 1C 7815 for +15V, 1C for-15V, these ICs are supplied different dc input
voltages by two Half-wave rectifiers consisting of D1-D4 and D5-D8 and c1, C2,
C3, C4. The capacitors at each input and each output are for filtering purpose.
SW1 is main AC ON/FF Switch.
PROCEDURE FOR FDMD

1. Connect 1 KHz sine wave &2 KHz sine wave signals to balanced modulators
2. Connect CRO channel-1 at 1Khz sine wave audio signal & adjust its amplitude to 1Vpp
3. Connect CRO channel-1 at 2Khz sine wave audio signal & adjust its amplitude to
1Vpp
4. Connect CRO channel-1 at 16Khz sub career in sub career generator &observe it
5. Connect CRO channel-1 at 32Khz sub career in sub career generator &observe it
6. Connect CRO channel-1 at 1Khz sine wave audio signal &Connect CRO channel-2 at
balanced modulator-1 section. Trigger CRO by channel-1. the DSB-SC amplitude
modulated wave will be observed & also observed BPF o/p-1
7. Connect CRO channel-1 at 2Khz sine wave audio signal &Connect CRO channel-2 at
balanced modulator-2 section. Trigger CRO by channel-1. the DSB-SC amplitude
modulated wave will be observed & also observed BPF o/p-2
8. Connect CRO channel-1 at 1Khz sine wave o/p Trigger CRO by channel connect CRO
Channel 2 at added o/p of summing amplifier & observe o/p
9. Connect CRO channel 2 at main rf career of rf oscillator section. Trigger CRO by
channel 2. set o/p frequency of rf oscillator to 455 KHz & amplitude to 10Vpp
10. Connect CRO channel-1 at 1Khz sinewave-1 o/p Trigger CRO by channel-1 connect
CRO Channel 2 at modulated o/p of final modulator section & observe the FDM
DSB-SC wave
11. Connect CRO channel-1 at sinewave-1 o/p Trigger CRO by channel-1 connect CRO
Channel 2 at o/p of main demodulator section in demultiplexing board & observe
o/p1Khz s
12. Connect CRO channel-1 at inewave-1 o/p Trigger CRO by channel-1 connect CRO
Channel 2 at o/p of BPF 28-32Khz in demultiplexing board & observe o/p
Channel 2 at o/p of BPF 16-32Khz in demultiplexing board & observe o/p
Channel 2 at o/p channel-1 demodulator section in demultiplexing board
Channel 2 at o/p channel-2 demodulator section in demultiplexing board &observe
o/p
Channel 2 at o/p LPF-1 in demultiplexing board &observe recover o/p

Channel 2 at o/p LPF-2 in demultiplexing board &observe recovered .o/p
OBSERVATION TABLE:-
SRNO SIGNAL AMPLITUDE FREQ.
1. MOD SIG. -1 (AUDIO-1)
2. MOD SIG. -1 (AUDIO-2)
3. SUBCARRIER1
4. SUBCARRIER2
5. MOD CHANNEL-1
6. MOD CHANNEL-2
7. FDM O/P
8. RF CARRIER
9. DEMOD O/P
10. CH1 DEMOD
11. CH2 DEMOD
12. LPF-1
13. LPF-2

CONCLUSION:-
Thus FDM was studied two modulating signals were modulated by two sub carrier and then
demodulated and recovered.

D 14pcom refrence manu 2015-16

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