COMMUNICATION SYSTEM LAB MANUAL

(ECC305)

DEPARTMENT OF ELECTRONICS ENGINEERING

INDIAN INSTITUTE OF TECHNOLOGY (ISM),
DHANBAD
JHARKHAND-826004
Do’s:

1. Be PUNCTUAL for your laboratory session.

2. Be attentive in the lab.
3. Keep silence in the lab.
4. All bags must be left at the indicated place.
5. Follow the instructions of the lab in-charges and lab supervisor.
6. Switch off the power supply after finishing the experiment.
7. Handle the systems and interfacing kits with care.
8. Maintain the lab records on a regular basis.

Don’ts:

1. Do not use internet or open any other programs other than multisim software in your laptops.
2. Do not make noise in the lab.
3. Do not use mobile phones in the lab.
4. Do not switch on the power supply until you finish all the connections.
5. Do not remove connecting wires or probes when the power is ON.
6. Do not mishandle the kits and instruments.
7. Do not pluck the ICs or other components from the trainer kits.
Contents:

Sl.No Experiment
1. i. To design a second order unity gain active low pass filter for cut off frequency
1KHz.
ii. To design a second order Butterworth active low pass filter for cut off frequency
1KHz.
2. To design a second order active bandpass filter for two different center frequencies.
3. To generate the white noise and limit the frequency range of the noise.
4. To study Amplitude modulation
i. Measure modulation index.
ii. Study under, over, and 100% modulation
5. To study Amplitude Demodulation.
i. To demodulate AM signal and observe the trapezoidal figure on CRO.
6. To perform pulse amplitude modulation and demodulation.
7. Study of pulse width Modulation and Demodulation.
8. Study of Pulse Position Modulation (PPM) and Demodulation.
9. To perform analog multiplexing and de-multiplexing.
10. To generate the Different Modulated wave for message signal frequency 100Hz
and carrier frequency 2kHz using Matlab.
i. Amplitude modulated wave
ii. Frequency modulated wave
iii. Phase modulated wave
11. Sampling and Reconstruction of Low pass signal using matlab.
 Experiment- 1

1. Aim:
i. To design a second order unity gain active low pass filter for cut off frequency 1KHz.
ii. To design a second order Butterworth active low pass filter for cut off frequency 1KHz.

2. Learning Outcome:
After the realization of active low pass filters, one should be able to:
i. Design second order low pass unity gain and Butterworth active filter according to given
specifications.
ii. Draw the frequency response of the low pass unity gain and Butterworth active filters.
iii. Discuss the difference between the unity gain and the Butterworth active filters.

3. Theory:
An electric filter is often a frequency selective circuit that passes a specified band of frequencies
and blocks or attenuates signals of frequencies outside this band. A low pass filter passes low
frequency signals but attenuate signal with frequency higher than the cutoff frequency. A first
order low-pass filter can be converted into second order type simply by using additional RC circuit.
Second order filters are important because higher order filters can be designed using them.

Filter analysis:
The op-amp is connected as non-inverting amplifier and hence,
𝑅
𝑣𝑜 = (1 + 𝑅𝐹) 𝑣𝐵 = 𝐴𝑜 𝑣𝐵 ……(i)
1

𝑅
Where, 𝐴𝑜 = (1 + 𝑅𝐹 ) ……(ii)
1

𝑣𝐵 is the voltage at the non-inverting terminal.

The transfer function of the second order low pass filter is given as:
𝑣𝑜 𝐴𝑜
= (𝑅3 𝐶3 +𝑅2 𝐶3 +𝑅2 𝐶2 −𝐴𝑜 𝑅2 𝐶2 ) 1 …..(iii)
𝑣𝑖𝑛 2
𝑠 + +
𝑅2 𝑅3 𝐶2 𝐶3 𝑅2 𝑅3 𝐶3 𝐶3

Where 𝐴𝑜 is the gain of the filter in pass band.

The standard form of transfer function of any 2nd order system,
𝑣𝑜 (𝑠) 𝐴
= 𝑠2 +2𝜁𝜔 2 …..(iv)
𝑣𝑖𝑛 (𝑠) 𝑛 𝑠+𝜔𝑛

Where, 𝐴= overall gain, 𝜁= damping of system, 𝜔𝑛 = natural frequency of oscillations

Comparing (iii) & (iv), we get,
1
𝑓𝑐 = 2𝜋√𝑅 ….(required cut off frequency) ………(v)
2 𝑅3 𝐶3 𝐶3

4. Apparatus & Components Required:

Software Simulation Hardware Implementation
Multisim Software in Laptop  Resistors
 Op-amp IC 741  Capacitors
 Resistors  Op-amp IC 741
 Capacitor  Bread board
 Function Generator  Function generator
 Oscilloscope  MSO
 Power supply

5. Experimental Procedure:
i. Choose the cut off frequency 𝑓𝑐 .
ii. The design can be simplified by selecting 𝑅2 = 𝑅3 = 𝑅 𝑎𝑛𝑑 𝐶2 = 𝐶3 = 𝐶 . Choose the
value of 𝐶 less than 1µF.
iii. Calculate the value of 𝑅 from the cut frequency formula.
1
𝑓𝑐 =
2𝜋√𝑅2 𝑅3 𝐶3 𝐶3
iv. For 2nd order Butterworth Active LPF, When 𝑅2 = 𝑅3 = 𝑅 𝑎𝑛𝑑 𝐶2 = 𝐶3 = 𝐶 the
voltage gain of the second order low pass filter becomes 𝐴𝑜 = 1.586.
v. Design the circuits as shown, take the readings as per the observation table from function
generator and oscilloscope.
vi. Plot the frequency response curve and find 3dB below cut off frequency and find the %
Error in the cut off frequency for the two filters.
6. Observations & Calculations.
i. Unity gain 2nd order active low pass filter

Fig 1.1 Circuit Diagram of unity gain active low-pass filter

a. Input voltage= 1 Vp-p (Peak to Peak Voltage)

b. Observation Table
S. No. Frequency (KHz) Output Voltage (Vp-p) Gain = 20 log(Vout /Vin)

c. Cut off frequency of unity gain filter =

d. Calculate the percentage error found in the cut off frequency.
𝑓𝑡ℎ𝑒𝑜𝑟𝑒𝑡𝑖𝑐𝑎𝑙 − 𝑓𝑝𝑟𝑎𝑐𝑡𝑖𝑐𝑎𝑙
%𝐸𝑟𝑟𝑜𝑟 = × 100 %
𝑓𝑡ℎ𝑒𝑜𝑟𝑒𝑡𝑖𝑐𝑎𝑙

ii. Butterworth second order low pass active low pass filter:

a. Gain required : 1.58 =1+RF/R1

b. From the standard Resistance values, 27k and 47k we get expected ratio.
c. Input voltage =1 Vp-p
 Fig 1.2 Circuit Diagram of active low-pass filter

d. Obsevation Table.
S. No. Frequency(KHz) Output Voltage(Vp-p) Gain = 20log(Vout /Vin)

e. Cut off frequency of Butterworth filter =

f. Calculate the percentage error found in the cut off frequency.
ftheoretical − fpractical
%Error = × 100 %
ftheoretical

7. Questions:
i. Design a low pass filter with cut off frequency 1KHz and a pass band gain of 2.
ii. Using frequency scaling technique, how can you change the cut off frequency of above
question from 1KHz to 1.6KHz?
iii. What will be the value of new resistor if C=0.01µF ?
iv. What is the voltage gain magnitude equation for a 2nd order Butterworth Active LPF?
v. What will be the roll off rate for an nth –order low pass filter?
 Experiment- 2

1. Aim: To design a second order active bandpass filter for two different center frequencies

2. Learning Outcome:
After the realization of band pass filters, one should be able to design of a band pass filter with
desired pass band frequencies.

3. Theory
A band pass filter has a pass band between two cut off frequencies 𝑓𝐻 and 𝑓𝐿 such that
𝑓𝐻 > 𝑓𝐿 . Any signal beyond this pass band is attenuated. There are 2 types of band pass filter
wide band pass and narrow band pass filter. The wide band pass filter circuit is formed by
simply cascading a high pass and a low pass section. A narrow band pass filter uses a single
op-amp and consists of two feedback paths so also known as multiple feedback filter. A
narrow band pass filter is designed for specific values of 𝑓𝑜 and Q. To simple the design
calculations, 𝐶1 = 𝐶1 = 𝐶.
𝑓𝑜 = center frequency
𝐴𝑜 = gain at 𝑓𝑜
BW=band width
Q=Quality factor=𝑓𝑜 /BW
An inverting band passes circuit shown in the figure. The expression for the nature of the
resistance is as:
𝑄
𝑅1 =
𝐴𝐶𝑤0
𝑄
𝑅2 =
(2𝑄 2 − 𝐴)𝐶𝑤0
𝑄
𝑅3 =
𝜋𝑓𝑜 𝐶

The response of narrow band circuit is similar to that obtained with a simple series parallel resonant
circuit having a moderate Q. For this situation, the upper and lower 3dbB frequency 𝑓𝐻 & 𝑓𝐿
respectively are quite close so that so the center frequency 𝑓𝑜 = √𝑓𝐻 𝑓𝐿 and bandwidth (𝑓𝐻 − 𝑓𝐿 )
is 𝑓𝑜 /Q.

4. Apparatus & Components Required:

Software Simulation Hardware Implementation
Multisim Software in Laptop  Resistors
 Op-amp IC 741  Capacitors
 Resistors  Op-amp IC 741
 Capacitors  Bread board
 Function Generator  Function generator
 Oscilloscope  MSO
 Power supply

Narrow Band Pass Filter

Fig. 2.1 Circuit Diagram of active band-pass filter

5. Experimental Procedure:
i. Choose the center frequency 𝑓𝑜 = 10 𝐾𝐻𝑧.
 ii. Calculate the values of R1, R2 and R3 as per given equations for A=10 and Q=10.
iii. The design can be simplified by selecting 𝐶2 = 𝐶3 = 𝐶 . Choose the value of 𝐶 less than
1µF.
iv. Design the circuit in and simulate the result.
v. Note the output voltage from oscilloscope by changing the input frequency and keeping
input voltage 1Vp-p.
vi. Plot the frequency response curve and find the bandwidth and % Error in the center
gain n frequency curve
frequency.
vii. Repeat the above steps for center frequency 20KHz.

6. Observations and Calculation

i. For 𝑓𝑜 = 10kHz
𝐴𝑜 =10, 𝐶1 =𝐶2 =1nf, Q=10
𝑅1 =……
𝑅2 =……
𝑅3 =……
Input Voltage= 1Vp-p
S. No. Frequency(KHz) Output Voltage(Vp-p) Gain=20log(𝑉𝑜 /𝑉𝑖𝑛 )

a. Observed center frequency =

b. Low cut off frequency =
c. Maximum Gain =
d. Bandwidth =
𝑓𝑜(𝑡ℎ𝑒𝑜𝑟𝑒𝑡𝑖𝑐𝑎𝑙) −𝑓𝑜(𝑝𝑟𝑎𝑐𝑡𝑖𝑐𝑎𝑙)
e. %𝐸𝑟𝑟𝑜𝑟 = × 100 %
𝑓𝑜(𝑡ℎ𝑒𝑜𝑟𝑒𝑡𝑖𝑐𝑎𝑙)

ii. For 𝑓𝑜 = 20kHz

𝐴𝑜 =10, 𝐶1 =𝐶2 =1nf, Q=10
 𝑅1 =……
𝑅2 =……
𝑅3 =……

Input Voltage= 1Vp-p

S. No. Frequency(KHz) Output Voltage(Vp-p) Gain=20log(𝑉𝑜 /𝑉𝑖𝑛 )

a. Observed center frequency =

b. Low cut off frequency =
c. Maximum Gain =
d. Bandwidth =
fo(theoretical) −fo(practical)
e. %Error = × 100 %
fo(theoretical)

7. Questions:
i. A band pass filter has a bandwidth of 250Hz and center frequency of 866Hz. Find the
quality factor of the filter?
ii. What is the center frequency of wide band-pass filter?
iii. Which filter attenuates any frequency outside the pass band?
iv. What is the difference between wide band and narrow band pass filters?
v. What is quality factor Q? What is the range of Q for narrow band pass filter?
vi. What will be the roll over rate of the resultant filter when a second order high pass filter
and second order low pass sections are cascaded?
 Experiment No: 3

1. Aim: To generate the white noise and limit the frequency range of the noise.

2. Learning Outcome:
After the completing this experiment we can learn how to generate a band limited white noise
using zener diode, amplifier and filters for security of information of the communication signals.

3. Theory:
White noise is a random signal with a constant power spectral density. The term is used, with this
or similar meanings, in many scientific and technical disciplines, including physics, acoustic
engineering, telecommunications, statistical forecasting, and many more. White noise refers to a
statistical model for signals and signal sources, rather than to any specific signal. The samples of
a white noise signal may be sequential in time, or arranged along one or more spatial dimensions.
In digital image processing, the pixels of a white noise image are typically arranged in a
rectangular grid, and are assumed to be independent random variables with uniform probability
distribution over some interval. The concept can be defined also for signals spread over more
complicated domain. An infinite-bandwidth white noise signal is a purely theoretical
construction. The bandwidth of white noise is limited in practice by the mechanism of noise
generation, by the transmission medium and by finite observation capabilities. Thus, a random
signal is considered "white noise" if it is observed to have a flat spectrum over the range of
frequencies that is relevant to the context.
Narrow Band:
All most communication system of often deal with band pass filter of signals. This filter’s
bandwidths is just large enough to pass the modulator components of the receiving signal without
distortion. This wide band is shaped into band limited noise. If the band width of this band limited
noise is relatively small compared to the carrier frequency, then it is called narrowband band
noise.The spectral component of narrow band noise is around some mid-band frequency +fc. where
fc is the carriers frequency. The sample to function n(t) appear as somewhat similar to sine wave
of frequency fc which modulate slowly in both amplitude and phase.The power spectral density of
the narrowband noise can be derive and used to analyses the performance of liner system.
 𝑛(𝑡) = 𝑛𝐼 (𝑡)𝑐𝑜𝑠2𝑛𝑓𝑐 𝑡 − 𝑛𝑞 (𝑡)𝑠𝑖𝑛2𝑛𝑓𝑐 𝑡
Where 𝑓𝑐 carrier frequency within the band
𝑛𝐼 (𝑡) in- phase component of n(t), 𝑛𝑞 (𝑡) quadrature component of n(t)

4. Apparatus and Component required:

Software Simulation Hardware Implementation

Multisim Software in Laptop  Resistors
 Resistor (15 kΩ, 100 kΩ ,  Capacitors
33 kΩ , 470 kΩ ),  Op-amp IC 741
 Capacitors,  Zener Diode
 Zener Diode(12V),  Bread board
 IC741  Function generator
 Mixed signal oscilloscope,  MSO
 Function generator,  Power supply

5. Experiment Step/ Numerical Step / Simulated Circuit :

Fig. 3.1. Band limited White Noise Generation Circuit diagram

6. Results and observation:
i. Observe the generated noise from zener diode
ii. Observe the noises after the first and second amplifiers.
iii. Observe the noise after the active band pass filter and note down the bandwidth of the
noise.
iv. Comments on the time period of the noise over the bandwidth of the band pass filter

7. Questions:
i. What is the role of noise in applications of communication?
ii. Why it is called as white noise?
iii. Write the basic properties of a noise?
 Experiment- 4

1. Aim:
To study Amplitude modulation
i. Measure modulation index.
ii. Study under, over, and 100% modulation

2. Learning Outcome:
After the completing this experiment we can understand
i. How the amplitude of a higher frequency signal (carrier signal) is modulated by a lower
frequency signal (modulating signal) for AM transmitter
ii. And the modulation index in terms of modulation.

3. Theory:
Amplitude Modulation:
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 (modulated) 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:
i. The carrier frequency fc must be much greater then the highest frequency components
fm of the message signal m (t) i.e. fc >> fm
ii. The modulation index must be less than unity. if the modulation index is greater than
unity, the carrier wave becomes over modulated.

Amplitude modulation index:

Emax – Emin
m = Emax + Emin

Emax = maximum peak to peak value of modulated signal.

Emin = minimum peak to peak value of modulated signal
the modulation index must not allowed to exceed unity; else the –ve peak of the modulating
waveform is clipped. This clipping not only causes distortion but also Interference. Depending
on the value of modulating index there are three types of modulation
i. Under modulation (m<1),
ii. Perfect modulation (m=1),
iii. Overmodulation(m>1).

Over modulation occurs when the magnitude of the peak negative voltage of modulating wave
exceeds the peak carrier voltage. To ensure that peak value of modulating signal should not exceed
the peak value of carrier signal.

4. Apparatus and Component required:

Simulated Implementation Hardware Implementation
Multisim Software on Laptop  Resistor
 Resistor  Capacitor,
 Capacitor,  OPAMP IC741,
 OPAMP IC741,  OA 79 germanium
 OA 79 germanium diode (In Multisim use 1BH62 or  MSO,
virtual diode  function generator,
 MSO,  probes,
 function generator,  connecting wires,
 probes,  power supply.
 connecting wires,
 power supply.

5. Experiment Step/ Numerical Step / Simulated Circuit :

Amplitude Modulating Simulated Circuit in Multisim

Fig.4.1 Circuit diagram of Amplitude modulation

6. Results and observation:
i. Observe the waveform after adder circuit.
ii. Observe the waveform after zener diode.
iii. Observe the waveform after band pass filter and the modulating index from that waveforms
for under modulation, perfect modulation and over modulation.
iv. Compare simulating modulating indexes with theoretical modulation index.

7. Questions:
i. What is importance of modulation in communication?
ii. What are the advantages and disadvantages of Amplitude Modulation over the other types
analog modulations?
iii. Suppose there are two sinusoidal signals with frequency 30 kHz and 1 kHz respectively.
Among of them which signal should be carrier signal and why?
iv. A sinusoidal carrier has amplitude of 10 V. It is amplitude modulated by a sinusoidal
voltage of amplitude 3 V. Determine the modulation index and transmission efficiency.
 Experiment- 5

1. Aim: To study Amplitude Demodulation.

2. Learning Outcome:
After the competing this experiment we can learn how the original signal (modulating or base
band signal) can be recovered from amplitude modulated signal. This principle can be used in
AM receiver.

3. Theory:
Amplitude Demodulation
The process of detection provides a means of recovering the modulating signal from modulated
signal. Demodulation is the reverse process of modulation. The detector circuit is employed to
separate the carrier wave and eliminate the side bands. There mainly two types detectors are
used, Envelope detector and square law detectors. Since the envelope of an AM wave has the
same shape as the message, independent of the carrier frequency and phase, demodulation can
be accomplished by extracting envelope. This circuit is essentially a rectifier circuit followed
by a capacitor across the output terminal.

Envelope Detector:
On the positive half cycle of the input signal, the capacitor C charges up to the peak voltage of
the input signal, as the input falls below this voltage, the diode is cutoff because Vc is greater
than V1.The capacitor discharges through R during the negative cycle. When V1>Vc diode
conducts .The capacitor charges up to the new value of this cycle. The capacitor discharges
slowly during cutoff and hence Vc charges negligibly. Thus during each +ve cycle C charges
up to the new peak value and holds on to it till the next +ve cycle. The time constant RC is
adjusted so that exponential decay of Vc follows the envelope
4. Apparatus and Component required:
Simulated Implementation Hardware Implementation
Multisim Software on Laptop  Resistor
 Resistor  Capacitor,
 Capacitor,  OPAMP IC741,
 OPAMP IC741,  OA 79 germanium
 OA 79 germanium diode (In  MSO,
Multisim use 1BH62 or virtual  function generator,
diode  probes,
 MSO,  connecting wires,
 function generator,  power supply.
 probes,
 connecting wires,
 power supply.

5. Experiment Step/ Numerical Step / Simulated Circuit :

AM Modulated
Signal

Fig. 5.1 Amplitude Demodulating Simulated Circuit in Multisim

6. Results and observation:
i. Observe the waveforms after the zener diode and active filters.
ii. Find the frequency of the output signal and comment on it.

7. Questions:
i. Why it is called envelope detector?
ii. What are advantages and disadvantages of Envelope detector over the Square law
detector?
iii. What do you mean by the diagonal clipping and negative peak clipping in envelope
detectors?
 Experiment No. 6

1. Aim:
To perform pulse amplitude modulation and demodulation.

2. Learning Outcome:
After completing this experiment we will be able to understand how pulse amplitude modulation
and demodulation can be accomplished.

3. Theory:
Pulse Amplitude Modulation:
Pulse amplitude modulation is a scheme, which alters the amplitude of regularly spaced
rectangular pulses in accordance with the instantaneous values of a continuous message signal.
Then amplitude of the modulated pulses represents the amplitude of the intelligence. A train of
very short pulses of constant amplitude and fast repetition rate is chosen, the amplitude of these
pulse is made to vary in accordance with that of a slower modulating signal the result is that of
multiplying the train by the modulating signal the envelope of the pulse height corresponds to the
modulating wave. The PAM wave contains upper and lower side band frequencies besides the
modulating and pulse signals.
Pulse Amplitude Demodulation:
The demodulated PAM waves, the signal is passed through a low pass filter having a cut off
frequencies equal to the highest frequency in the modulating signal. At the output of the filter is
availability of modulating signal along with DC component. PAM has same signal to noise ratio
as AM and so it is not employed in practical circuit.
 3. Apparatus and Component required:

Simulated Implementation Hardware Implementation

Multisim Software on Laptop  Resistor
 Resistor  Capacitor,
 Capacitor,  OPAMP IC741,
 OPAMP IC741,  Transistor (BC 547/548)
 Transistor (BC 547/548)  MSO, Breadboard
 MSO,  function generator,
 function generator,  probes and connecting
 probes wires,
 connecting wires,  power supply.
 power supply.

5. Experimental Procedure:
Modulation:
i. Make the circuit as shown in Fig.6.1.
ii. Set the pulse generator output to be ( 0 to 4.7 Vpp) at 10 kHz
iii. Set the voltage (0 to 2 Vpp) (sine wave) at the input of function generator with
frequency of 1 kHz.
iv. Vary the amplitude of the modulating signal and observe the changes that take place in the
modulating output
v. Observe the output waveform on the dual channel oscilloscope.
vi. Tabulate the reading.
Demodulation:
i. Connect the circuit as shown in Fig.6.2.
ii. Given the pulse modulated output signal to the input of demodulator circuit.
iii. Observe the amplitude of demodulated signal (output) on dual channel oscilloscope and
verify with that of input modulating signal.
iv. Draw the demodulated waveform
 (a)

(b)
Fig.6.1 Pulse Amplitude Modulation (a) Circuit, (b) Input and Output Waveform

Fig.6.2 Circuit diagram for Pulse Amplitude Demodulation

5. Observation:
Amplitude of 10 kHz Pulse 0-4.7V
S. No. Input DC Voltage Output Pulse Amplitude

Offset Voltage for sinusoidal signal = …….V

7. Questions:
i. Why PAM is not preferred in digital transmission?
ii. What is the process of sampling an analog signal at a high rate?
 Experiment: 7

1. Aim: Study of pulse width Modulation and Demodulation.

2. Learning Outcome: After completing this experiment we will be able to understand how
pulse width modulation and demodulation can be accomplished.

3. Theory:
Pulse width Modulation (PWM):
In PWM system, the width of the pulse is varied in accordance with the instantaneous level of
modulating signal. PWM is powerful technique for controlling analog circuits with a digital
output. PWM is a way of digitally encoding analog signal levels. Through the use of high
resolution counters, the duty cycle of a square wave is modulated to encode a specific analog
signal level. The PWM signal is digital after modulation because at any given instant of time
the signal is either ON or OFF. The term duty cycle describes the proportion of ON time and
OFF time. The pulses of low duty cycle correspond to low power and high duty cycle
corresponds to high power.
The main advantage of PWM is that power loss in the switching device is very low. It is used
for controlling power to inertial electrical devices, made practical by modern electronic power
switches. When a switch is off there is practically no current and when it is on, power is
transferred to the load, there is a voltage of voltage and current is thus in both case close to
zero. PWM also work with digital control, which become of their on/off nature can easily set
the duty the duty cycle.
The circuit diagram of pulse width modulation is described by IC 555 timer.
The 555 timer IC is an integrated circuit (chip) used in a variety of timer, pulse generation,
and oscillator applications. The 555 can be used to provide time delays, as an oscillator, and
as a flip-flop element. Derivatives provide up to four timing circuits in one package. By
applying a voltage to the control voltage input one can alter the timing characteristics of the
device. In most applications, the control voltage input is not used. It is usual to connect a 10 nf
capacitor between pin 5 and 0 V to prevent interference. The control voltage input can be used
to build an astable multivibrator with a frequency-modulated output.
 4. Apparatus and Component required:

Simulated Implementation Hardware Implementation

Multisim Software on Laptop  Resistor
 Resistor  Capacitor,
 Capacitor,  IC 555 timer,
 IC 555 timer,  Transistor (BC 547/548)
 Transistor (BC 547/548)  MSO, Breadboard
 MSO,  function generator,
 function generator,  probes,
 probes,  connecting wires,
 connecting wires,  power supply.
 power supply.

5. Experimental Procedure:
i. Connect the circuit as per circuit diagram shown in Fig.7.1.
ii. Apply a trigger signal (Pulse wave) of frequency 10 KHz with amplitude of 5 Volt (p-
p).
iii. Observe the sampled output signal at the pin3.
iv. Apply the input ac modulating signal with frequency of 1 kHz at the pin 5 and vary
the amplitude.
v. Note that as the control voltage is varied output pulse width is also varied.
vi. Observe that the pulse width increases during positive slope condition & decreases
under negative slope condition. Pulse width will be maximum at the +ve peak and
minimum at the –ve peak of sinusoidal waveform. Record the observations.
vii. Feed PWM waveform to the circuit of Fig.7.2 and observe the resulting demodulated
waveform.
 (a)

(b)
Fig.7 .1. Pulse width Modulation (a) IC 555 timer circuit, (b) input and output waveform

In PWM, how long a rectangular pulse stays “on” within a constant period is determined by the
value of the information signal. The “on-off” behavior changes the average power of the signal.
Duty cycle is the ratio of the duration of the “on” event to the total period of the rectangular pulse
train, and is given by
𝑂𝑛𝑡𝑖𝑚𝑒
𝐷𝑢𝑡𝑦 𝐶𝑦𝑐𝑙𝑒 = × 100%
𝑃𝑒𝑟𝑖𝑜𝑑
Demodulation Circuit:

Fig.7.2. Pulse width demodulation circuit

6. Observation:

S. No. Control Voltage Output Pulse Width( m-sec)

(Vpp)

On the bases of observations we can observed that the duty cycle of the pulse width modulation
signal varied according to modulating signal. Thus the information carried by the modulating
signal was transferred and demodulated according to the variation in the width of the pulse train
(changes in duty cycle)

7. Questions:
i. Where does PWM technology find its applicability?
ii. What is PWM?
 Experiment- 8

1. Aim: Study of Pulse Position Modulation (PPM) and Demodulation.

2. Learning Outcome:
After the completion of the experiment, one should be able to:
i. Understand the Pulse position modulation and demodulation.
ii. Design the circuit and generate the Pulse position modulated and demodulated signals.

3. Theory
In pulse position modulation, the amplitude and width of the pulse are kept constant, while the
position of a reference pulse is changed according to the instantaneous sampled value of the
modulating circuit. Hence, for proper communication transmitter and receiver must be in
synchronization. The graphical representation of the generation of PPM is shown in fig.1.
Moreover, as the amplitude and width of the pulse are constant, the transmitter handles constant
power output which is a definite advantage over PWM and PAM.

Pulse position modulator: It consist of a PWM generator followed by the monostable

multivibrator. Since in PPM , output remain high for fixed duration from trailing edges of the
PWM signal, the trailing edge of the PWM signal is used as a trigger input for the monostable
multivibrator.

Monostable multivibrator (DM54/54121): It is a monostable multivibrator IC having both

positive and negative edge triggering with complementary outputs. An internal 2kΩ resistor is
provided for design convenience. A single external capacitor is used. Input (A) is active low
trigger transition input and input (B) is an active high Schmitt trigger input that allows jitter free
triggering input.
 Fig. 8.1 Waveforms for PPM generation

Fig 8.2 Schematic diagram and truth table of IC 54121

 4. Apparatus & Components Required:

Software Simulation Hardware Implementation

Multisim Software in Laptop  Resistors
 Op-amp IC 741 IC 555, IC  Capacitors
54121, transistor (BC  Op-amp IC 7 IC 555, IC
548), IC 741 54121, transistor (BC
 Resistors 548), IC 741
 Capacitors  Bread board
 Function Generator  Function generator
 Oscilloscope  MSO
 Power supply

4. Experimental Procedure:
i. The block diagram to generate PPM signal is shown in Fig.8.3.
ii. To generate a PPM signal, first of all a PWM signal is needed, which can be generated
using the Fig.8.4.
iii. Now, this PWM signal then passed through the monoshot multivibrator IC 54121 at pin
no.4 shown in Fig.8.5
iv. The output can be taken from pin no. 6 of IC 54121, which is a PPM signal.
v. Lastly the demodulation of PPM signal can be done using Fig. 8.6.

Modulation Circuit:

PWM Mono shot Pulse Position

Modulation Multivibrator IC Modulated
Signal

Fig.8.3. Block Diagram of PPM

 Fig.8.4. PWM Modulation.

.
Fig.8.5. Circuit for Pulse position modulation
 Fig.8.6. Circuit for Pulse position demodulation

6. Observations
The PPM output and the demodulated output can be observed on the MSO and same need to be
plotted in the lab notebook.

7. Questions:
i. What are the advantages of PPM over PAM and PWM?
ii. What are the Disadvantages of PPM?
iii. What is the significance of synchronization in PPM
iv. Comment on robustness of PPM over noise?
 Experiment- 9

1. Aim: To perform analog multiplexing and de-multiplexing.

4. Learning Outcome:
After the completion the experiment, one should be able to:
i. Understand the significance of multiplexing.
ii. Perform 8×1 multiplexing.
iii. Perform 1×8 de-multiplexing.

3. Theory:
It has been observed that most of the individual data-communicating devices typically require
modest data rate. But, communication media usually have much higher bandwidth. As a
consequence, two communicating stations do not utilize the full capacity of a data link. Moreover,
when many nodes compete to access the network, some efficient techniques for utilizing the data
link are very essential. When the bandwidth of a medium is greater than individual signals to be
transmitted through the channel, a medium can be shared by more than one channel of signals. The
process of making the most effective use of the available channel capacity is called Multiplexing.
For efficiency, the channel capacity can be shared among a number of communicating stations.
Most common use of multiplexing is in long-haul communication using coaxial cable, microwave
and optical fiber. Fig.1 depicts the functioning of multiplexing functions in general. The
multiplexer is connected to the de-multiplexer by a single data link. The multiplexer combines
(multiplexes) data from these ‘n’ input lines and transmits them through the high capacity data
link, which is being de-multiplexed at the other end and is delivered to the appropriate output lines.
Thus, Multiplexing can also be defined as a technique that allows simultaneous transmission of
multiple signals across a single data link.
 Fig.9.1. Multiplexing and De-multiplexing
sampling
In Time-division multiplexing all signals operate with same frequency at different times.
This is a base band transmission system, where an electronic commutator sequentially samples all
data source and combines them to form a composite base band signal, which travels through the
media and is being de-multiplexed into appropriate independent message signals by the
corresponding commutator at the receiving end. The incoming data from each source are briefly
buffered. Each buffer is typically one bit or one character in length. The buffers are scanned
sequentially to form a composite data stream. The scan operation is sufficiently rapid so that each
buffer is emptied before more data can arrive. Composite data rate must be at least equal to the
sum of the individual data rates. The composite signal can be transmitted directly or through a
modem. The multiplexing operation is shown in Fig. 9.2.

Fig. 9.2. Multiplexing Operation for Synchronous TDM

The time slots are transmitted irrespective of whether the sources have any data to send or not.
Hence, for the sake of simplicity of implementation, channel capacity is wasted. Although fixed
assignment is used, TDM devices can handle sources of different data rates. This is done by
assigning fewer slots per cycle to the slower input devices than the faster devices. Both
multiplexing and de-multiplexing operation for synchronous TDM are shown in Fig.9. 3.

Fig. 9.3. Multiplexing and De-multiplexing Operation for Synchronous TDM

4. Apparatus& Components Required:

Software Simulation Hardware Implementation

Multisim Software in Laptop  Resistors
 IC7493, IC4051/MPC508  Capacitors
 Capacitor  IC7493, IC4051/MPC508
 Function Generator  Bread board
 Oscilloscope  Function generator
 MSO
 Power supply

5. Experimental Procedure:

1) Multiplexing:
i. Different input signals are provided at the input lines of analog multiplexer IC
(IC4051).
ii. To select any of the given input signals a control signal is given in the form of a 3-bit
combination of the three control lines varying from 000 to 111.
iii. Each corresponds to the one of the input signal being transferred to output lines as per
the control signal.
iv. The different signals for the input line of multiplexer IC are generated through potential
divider consist of ten resistors in series.
v. The output of the multiplexer IC4051 is connected to the MSO along with the control
lines connected through 3-bit binary counter (IC7493) which has been provided with
TTL clock frequency.
vi. The output of at the MSO is observed as a stair case.

2) De-multiplexing:
i. The multiplexed output then connected to the analog demultiplexer (IC4051)
 ii. The control lines are the same as the three bit counter.
iii. We get 8-output signals which are same as being multiplexed earlier as can be viewed
at MSO.
iv. Connection diagram has been given in Fig. 9.4.

Fig.9.4. Circuit Diagram

6. Observations:-
Observation of the corresponding waveforms can be done on the MSO and same will be produced
on the lab notebook.

7. Questions:
i. What are the differences in frequency division multiplexing and time division
multiplexing?
ii. State benefits of multiplexing in communication?
iii. What is the significance of synchronization in TDM?
iv. What are the applications of multiplexing?
 Experiment- 10

1. Aim:
To generate the Different Modulated wave for message signal frequency 100Hz and
carrier frequency 2 KHz using Matlab.
i. Amplitude modulated wave
ii. Frequency modulated wave
iii. Phase modulated wave

2. Learning Outcome:
i. After the completing of the experiment one should be able to understand:
How amplitude, frequency and phase of a high frequency signal (carrier signal) is
modulated with lower frequency signal (modulating signal).
ii. Comparison between the performances of different analog modulation
3. Theory:
Amplitude Modulation:
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 (modulated) 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.
m(t)= Am cos(𝜔𝑚 𝑡) (message signal)
c(t) = Ac cos(𝜔𝑐 𝑡) (carrier signal)
The amplitude modulated signal is given by
𝑠(𝑡) = 𝐴𝑐 [1 + 𝑘𝑎 𝑚(𝑡)] cos(𝜔𝑐 𝑡)
𝑘𝑎 is amplitude sensitivity. The carrier frequency fc must be much greater then the highest
frequency components fm of the message signal m(t) i.e. fc>>fm:
 Fig. 10.1 Waveform of Amplitude modulation

Frequency Modulation
Frequency modulation is that from of angle modulation in which the instantaneous
frequency f (t) is varied linearly with the message signal m(t):
𝑓(𝑡) = 𝑓𝑐 + 𝑘𝑓 𝑚(𝑡)
The term 𝑓𝑐 represents the frequency of the unmodulated carrier and the constant
𝑘𝑓 represents the frequency sensitivity of the modulator.
The frequency modulated signal s(t) is thus described in the time domain by
𝑡
𝑠(𝑡) = 𝐴𝑐 cos[2π𝑓𝑐 t + 𝑘𝑓 ∫0 𝑚(𝑡) ]

Fig. 10.2 Waveform of frequency modulation

Phase Modulation
It is a form of angle modulation in which the angle 𝜑𝑖 (𝑡) is varied linearly with the
message signal m(t),
𝜑𝑖 (𝑡) = 2π𝑓𝑐 t + φ(t)
Where φ(t) = 𝑘𝑝 m(t) and not a constant
The term 2π𝑓𝑐 t represents the angle of the carrier , the constant 𝑘𝑝 represents the phase
sensitivity of the modulator.
The phase modulated signal s(t) is thus described in the time domain by
𝑠(𝑡) = 𝐴𝑐 cos[2π𝑓𝑐 t + 𝑘𝑝 m(t) ]

Fig. 10.3 Waveform of phase modulation

4. Apparatus and Component required:
Laptop installed with Matlab Software.

5. Experimental procedure / Programming

Flow chart for Matlab program :

 START

Define input variables i.e

message signal (fm)
carrier signal (fc)

Define mathematical
Expression for Amplitude,
frequency and phase
modulated signals

Plot generated AM, FM

and PM waveform

END

6. Observation:
Observation of performance of different analog modulation techniques and comparison
between them.
7. Questions:
i. What is modulation?
ii. What is the major advantage of Amplitude Modulation over frequency modulations?
iii. What is angle modulation? What are different types of angle modulation?
iv. Why Frequency translation is required?
 Experiment- 11

1. Aim: Sampling and Reconstruction of Low pass signal using matlab.

2. Learning Outcome:
After the completing of the experiment one should be able to :
Observe and understand the concept of the periodically sampling a continuous signal at sample
rates and reconstructing the original signal using matlab.

3. Theory:
Sampling and Reconstruction:
In order to store, transmit or process analog signals using digital hardware, we must first
convert them into discrete-time signals by sampling.
The processed discrete-time signal is usually converted back to analog form by interpolation,
resulting in a reconstructed analog signal 𝑥(𝑡).
The sampler reads the values of the analog signal 𝑥(𝑡) at equally spaced sampling instants.
The time interval 𝑇𝑠 between adjacent samples is known as the sampling period (or
sampling interval).
The sampling rate, measured in samples per second, is 𝑓𝑠 = 1/𝑇𝑠.
𝑋[𝑛] = 𝑥(𝑛𝑇𝑠) , 𝑛 = ⋯ , −1, 0, 1, 2, …. Also it possible to reconstruct x(t) from its
samples:
𝑥(𝑡) = 𝑥[𝑡𝐹𝑠].
 Fig.11.1 Sampling and reconstruction process
Sampling Theorem
The uniform sampling theorem states that a bandlimited signal having no spectral components
above 𝑓𝑚 hertz can be determined uniquely by values sampled at uniform intervals of:
1
𝑇𝑠 =≤
2𝑓𝑚
The upper limit on 𝑇𝑠 can be expressed in terms of sampling rate, denoted 𝑓𝑠 = 1/𝑇𝑠.
The restriction, stated in term of the sampling rate, is known as the Nyquist criterion. The statement
is: Nyquist rate
𝑓𝑠 ≤ 2𝑓𝑚

The sampling rate 𝑓𝑠 =2𝑓𝑚 is also called.The allow Nyquist criterion is a theoretically sufficient
condition to allow an analog signal to be reconstructed completely from a set of a uniformly spaced
discrete-time samples.

Signal Reconstruction:

The process of reconstructing a continuous time signal x(t) from its samples is known
as interpolation. In the sampling theorem signal x(t) is band limited to some frequency in Hz
which can be reconstructed from its samples. This reconstruction is accomplished by passing the
sampled signal through an ideal low pass filter of the same bandwidth as x(t) .
The reconstruction process consists of replacing each sample by a sinc function, centered at the
time of the sample and scaled by the sample value x(nT) times 2fc/ fs and adding all the functions
so created.

Suppose the signal is sampled at exactly Nyquist rate fs= 2fm,

fs
Then fm= 2 = fs- fm and
fm = 1/2 = 1 − fm .

The requirement fm < fc < 1 − fm cannot be met, in this case we must allow fc = fm , which
means that
fc = fm = fs/2.

This will work till the signals spectrum does not have an impulse at fm . (If there is an impulse
at, it will be aliased in the sampling process). In this limiting case, the interpolation is described
by the simpler expression
+∞
𝑡 − 𝑛𝑇
𝑥(𝑡) = ∑ 𝑥(𝑛𝑇) 𝑠𝑖𝑛𝑐( )
𝑇
𝑛=−∞
4. Apparatus and Component required: Laptop, Matlab Software

5. Experimental procedure / programming :

START

Define input sinusoidal signal

x(t) which has to be sampled

Define sampling frequency

according to Nyquist criterion

Plot generated sampled

waveform

Reconstruct the original

signal by interpolation

Plot reconstructed Waveform

END
6. Observation:
.
Observing how the original analog signal is converted to a digital signal using sampling theorem and
again how the original signal is reconstructed.

7. Questions:
i. What is sampling theorem and Nyquist criteria?
ii. What is aliasing effect?
iii. What happened to its reconstructed signal when the original analog signal is sampled
below the Nyquist rate?