Ac Lab Record Demo Exp

Analog Communications Lab Record
DEPARTMENT
OF
ELECTRONICS & COMMUNICATION ENGINEERING
ANALOG COMMUNICATION LAB

RECORD
SR ENGINEERING COLLEGE
Ananthasagar, Hasanparthy
Warangal - 506371
Dept. of ECE SR Engineering college

Analog Communication Lab

Program: B.Tech Lecture: -
Year/Sem: III/I Tutorial: -
CIE: 30 Practical: 3/Week
SEE: 70 Credits: 2
COURSE OUTCOMES:
At the end of the course the students will develop ability to
1. Measure modulation index of AM and FM and simulate using MATLAB.
2. Generate the modulated and demodulated waves of SSB-SC and DSB-SC and simulate
using MATLAB.
3. Perform the FM demodulation using PLL.
4. Analyze the operation of TDM and FDM and simulate using MATLAB.
5. Analyze various pulse modulation schemes and simulate using MATLAB.
LIST OF EXPERIMENTS:
Note: Minimum 10 experiments should be conducted.
All these experiments are to be simulated first either using Commsim, MATLAB, SCILAB,
OCTAVE or any other simulation package and then to be realized in hardware.
1. Amplitude modulation and demodulation
2. DSB-SC Modulator & Detector
3. SSB-SC Modulator & Detector (Phase Shift Method)
4. Frequency modulation and demodulation
5. Study of spectrum analyzer and analysis of AM and FM signals
6. Pre- emphasis & De-emphasis
7. Time Division Multiplexing & De Multiplexing
8. Frequency Division Multiplexing & De multiplexing
9. Verification of Sampling Theorem
10. Pulse Amplitude Modulation & De Modulation
11. Pulse width Modulation & De modulation
12. Pulse position Modulation and De modulation
13. Frequency Synthesizer
14. AGC Characteristics
15. PLL as FM Demodulator

Instructions to the students
Do’s
1. The students should come to the lab in time.
2. They should maintain discipline and decorum during the lab hours.
3. They should wear their ID Cards.
4. They are required to come prepared for the experiments to be conducted.
5. They should bring the observation book and submit the lab record immediately for
evaluation.
6. Take the components as per the requirements of the experiment.
7. See that the equipments/ components are returned after completion of experiment to
the concerned person.
8. Switch off the power supply as soon on the experiment is completed.
Dont’s
1. Do not damage the IC’s/ Components.
2. Do not switch on the supply with out permission of the lab supervisor.
3. Do not talk during the lab session.
4. Do not walk around the lab.
5. Do not displace the apparatus after completion of experiment.
6. Mobile phones are not permitted into the lab.
7. Do not attempt to repair the apparatus or apply excessive voltage to the equipments.
8. Do not carry eatables inside the laboratory.

INTRODUCTION TO ANALOG COMMUNICATION
Communication is the transfer of information from one place to another. Radio

communication uses electrical energy to transmit information.
The transmitted information is the intelligence signal or message signal. Message
signals are in the Audio Frequency (AF) range of low frequencies from about 20 Hz to 20
kHz.
The Radio Frequency (RF) is the carrier signal. Carrier signals have high frequencies
that range from 3 kHz up to about 300 GHz. A radio transmitter sends the low frequency
message signal at the higher carrier signal frequency by combining the message signal with the
carrier signal.
Modulation is the process of changing a characteristic of the carrier signal with the message
signal. In the transmitter, the message signal modulates the carrier signal. The modulated
carrier signal is sent to the receiver where demodulation of the carrier occurs to recover the
message signal.
Three basic blocks in any communication system are: 1) transmitter 2) Channel and 3)
Receiver
Fig. Basic communication system
The transmitter puts the information from the source (meant for the receiver) onto the
channel. The channel is the medium connecting the transmitter and the receiver and the
transmitted information travels on this channel until it reaches the destination. Channels can be
of two types: i) wired channels or ii) wireless channels. Examples of the first type include:
twisted pair telephone channels, coaxial cables, fiber optic cable etc. Under the wireless
category, we have the following examples: earth’s atmosphere (enabling the propagation of
ground wave and sky wave), satellite channel, sea water etc.
The main disadvantage of wired channels is that they require a man-made medium to
be present between the transmitter and the receiver. Though wired channels have been put to
extensive use, wireless channels are equally (if not more) important and have found a large
number of applications.

In order to make use of the wireless channels, the information is to be converted into a
suitable form, say electromagnetic waves. This is accomplished with the help of a transmitting
antenna. The antenna at the receiver (called the receiving antenna) converts the received
electromagnetic energy to an electrical signal which is processed by the receiver.
For efficient radiation, the size of the antenna should be λ 10 or more (preferably
around λ 4 ), where λ is the wavelength of the signal to be radiated. Take the case of audio,
which has spectral components almost from DC upto 20 kHz. Assume that we are designing
the antenna for the mid frequency; that is,10kHz. Then the length of the antenna that is
required, even for the λ 10 situation
c 3 × 108 3
is, 10⋅f = 10 × 104 = 3 × 10 meters, c being the velocity of light.
Even an antenna of the size of 3 km, will not be able to take care of the entire spectrum of the
signal because for the frequency components around 1 kHz, the length of the antenna would be
λ 100 . Hence, what is required from the point of view of efficient radiation is the conversion
of the baseband signal into a narrowband, band-pass signal. Modulation process helps us to
accomplish this; besides, modulation gives rise to some other features which can be exploited
for the objective of efficient communication. We describe below the advantages of modulation.
Table illustrates the different radio-frequency bands that are in use today, and their practical
applications.

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

• Message signal - any signal that contains information; it is also called the intelligence
signal.
• Audio Frequency (AF) - frequencies that a person can hear. AF signals range from about
20 Hz to 20 kHz.
• Radio Frequency (RF) - the transmission frequency of electromagnetic (radio) signals.
RF frequencies are from about 3 kHz to the 300 GHz range.
• Carrier signal - a single, high-frequency signal that can be modulated by a message signal
and transmitted.
• Amplitude Modulation (AM) – is the process of changing amplitude of the carrier signal
in accordance with the instantaneous values of message signal by keeping frequency and phase
of the carrier signal as constant.
• Frequency Modulation (FM) - is the process of changing frequency of the carrier signal
in accordance with the instantaneous values of message signal by keeping amplitude and phase
of the carrier signal as constant.
• Phase Modulation (PM) is the process of changing phase of the carrier signal in
accordance with the instantaneous values of message signal by keeping amplitude and
frequency of the carrier signal as constant.
• Double-Sideband (DSB) - an amplitude modulated signal in which the carrier is
suppressed, leaving only the two sidebands: the lower sideband and the upper sideband.
• Mixer- an electronic circuit that combines two frequencies.
• Phase detector - an electronic circuit whose output varies with the phase differential of the
two input signals.
• Envelopes- the waveform of the amplitude variations of an amplitude modulated signal.
• Sidebands - the frequency bands on each side of the carrier frequency that are formed
during modulation; the sideband frequencies contain the intelligence of the message signal.
• Bandwidth - the frequency range, in hertz (Hz), between the upper and lower frequency
limits.
• Harmonics - signals with frequencies that are an integral multiple of the fundamental
frequency.

Concept of Carrier in Communication Systems

What kinds of electrical signals flow through communication systems? To answer this
question, we need to recognize that communication systems are essentially transporters that
carry information across the world in the form of telephone calls, e-mails, texts, Internet, radio,
and television, which are nowadays a part of our daily life. These transport systems can be
compared to trains or planes that move people from place to place; just as a modern
commercial plane is a strong and reliable transport, similarly there is need for a strong, stable
signal to carry the information. The latter signal is called as the carrier, and is just a simple
sinusoid given by the following equation:
x(t) = A cos(ωct + θ)
where
A is carrier amplitude, V
ω c is the carrier frequency radians/s
θ is the carrier phase (rad)
The carrier frequency is fc (Hz) = ωc/2π, and varies with application, from kHz to GHz, as
shown in Table
In addition to frequency, important properties of the carrier signal include its power and
bandwidth, which controls data speed, including upload and download speeds.
TABLE
Communication Systems and Frequency Bands
Application Carrier Frequency
AM radio Long wave (LF): 153–279 kHz
Medium wave (MF) 531–1611 kHz
Short wave (HF) 2.3–26.1 MHz
FM radio 76.0–108.0 MHz
Satellite 1–75 GHz
Line of sight (LOS) microwave links 1–25 GHz
Cellular 800 MHz–2.5 GHz
Introduction to Simulink
Programming software that is utilized in communication systems modeling and applications
can be categorized into the following two categories:
• Simulation software: These software, such as MATLAB ® and Simulink®, are utilized to
model communication systems, and hence are very valuable tools to design practical
systems. While MATLAB requires programs to be written, Simulink is a graphical tool,
which has built-in system blocks.
Simulink
• Basics
After logging into MATLAB, you will receive the prompt >>. In order to open up Simulink,
type
• in the following:

>> Simulink
•
Alternately, click on the Simulink icon in the MATLAB Command window, or in newer
editions of the software, use the command Open New Model.
General Simulink Operations

Two windows will open up: the model window and the library window. The model
window is the space utilized for creating your simulation model.
In order to create the model of the system, components will have to be selected and dragged
from the library using the computer mouse, and inserted into the model window. If you browse
the library window, the fol-lowing sections will be seen. Each section can be accessed by
clicking on it.
•\ Sources—This section consists of different signal sources such as sinusoidal, triangular,
pulse, random, or files containing audio or video signals.
•\ Sinks—This section consists of measuring instruments such as scopes and displays.
•\ Linear—This section consists of linear components that perform operations like
summing, integration, and product.
•\ Nonlinear—Nonlinear operations.
•\ Connections—Multiplexers, demultiplexers.
•\ Toolboxes—These specify different areas of Simulink, for example: •\
Communications
•\ DSP
•\ Neural nets
•\ Simscape (power systems)
•\ Control systems
Editing, Running, and Saving Simulink Files

The complete system is created in the model window by utilizing compo-nents from the
various available libraries. Once a complete model is created, save the model into a file. Click
on simulation and select run, or in newer mod-els, click on the green arrow. The simulation
will run, and the output data can be read by clicking on the appropriate sinks, such as
oscilloscopes or displays. Save the output plots also into files. The model and output files can
be printed out from the files.
Demo Files
Try out the demo files, both in the main library window, and in the Toolboxes window. There
are several illustrative demonstration files in the areas of sig-nal processing, image processing,
and communications.
Digital Storage Oscilloscopes

A digital storage oscilloscope (often abbreviated DSO) is an oscilloscope which stores

and analyses the signal digitally rather than using analog techniques. It is now the most
common type of oscilloscope in use because of the advanced trigger, storage, display and
measurement features which it typically provides.
The input analogue signal is sampled and then converted into a digital record of the amplitude
of the signal at each sample time. The sampling frequency should be not less than the Nyquist
rate to avoid aliasing. These digital values are then turned back into an analogue signal for
display on a cathode ray tube (CRT), or transformed as needed for the various possible types of
output—liquid crystal display, chart recorder, plotter or network interface.
Digital storage oscilloscope costs vary widely; bench-top self-contained instruments (complete
with displays) start at US$300 or even less, with high-performance models selling for tens of
thousands of dollars. Small, pocket-size models, limited in function, may retail for as little as
US$50.
Comparison with analog storage
The principal advantage over analog storage is that the stored traces are as bright, as sharply
defined, and written as quickly as non-stored traces. Traces can be stored indefinitely or
written out to some external data storage device and reloaded. This allows, for example,
comparison of an acquired trace from a system under test with a standard trace acquired from a
known-good system. Many models can display the waveform prior to the trigger signal.
Digital oscilloscopes usually analyze waveforms and provide numerical values as well as
visual displays. These values typically include averages, maxima and minima, root mean
square (RMS) and frequencies. They may be used to capture transient signals when operated in
a single sweep mode, without the brightness and writing speed limitations of an analog storage
oscilloscope.
The displayed trace can be manipulated after acquisition; a portion of the display can be
magnified to make fine detail more visible, or a long trace can be examined in a single display
to identify areas of interest. Many instruments allow a stored trace to be annotated by the user.
Many digital oscilloscopes use flat panel displays similar to those made in high volumes for
computers and television displays.

1. AMPLITUDE MODULATION AND DEMODULATION

Date:
Grade:
Objective:
1. To understand the operation of amplitude modulation and demodulation and to calculate

depth of modulation for various modulating voltages.
2. To simulate Amplitude modulation and demodulation.
Apparatus Required:
□ AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit
□FM Trainer kit □Sampling trainer kit □ TDM Trainer kit
□FDM Trainer kit □ PAM Trainer kit □PWM Trainer kit
□PPM Trainer kit □AGC Trainer kit □Frequency Synthesizer
Trainer kit □PLL as FM Demod Trainer kit □Bread board
□Spectrum Analyzer □Function Generator □Digital Multi-meters
□ CRO with probes □Patch cards □RPS
□Micro phone □Speakers
THEORY:
In order that a steady radio signal or "radio carrier" can carry information it must be
changed or modulated in one way so that the information can be conveyed from one place to
another. There are a number of ways in which a carrier can be modulated to carry a signal -
often an audio signal and the most obvious way is to vary its amplitude.
Amplitude Modulation has been in use since the very earliest days of radio technology. The
first recorded instance of its use was in 1901 when a signal was transmitted by a Canadian
engineer named Reginald Fessenden. To achieve this, he used a continuous spark transmission
and placed a carbon microphone in the antenna lead. The sound waves impacting on the
microphone varied its resistance and in turn this varied the intensity of the transmission.
Although very crude, signals were audible over a distance of a few hundred metres. The
quality of the audio was not good particularly as a result of the continuous rasping sound
caused by the spark used for the transmission.
Later, continuous sine wave signals could be generated and the audio quality was greatly
improved. As a result, amplitude modulation, AM became the standard for voice transmissions.

It remains in use today in many forms of communication; for example it is used in portable two
way radios, VHF aircraft radio, Citizen's Band Radio, and in computer modems (in the form of
QAM). "AM" is often used to refer to medium wave AM radio broadcasting.
Amplitude Modulation is defined as a process in which the amplitude of the carrier
wave c(t) is varied linearly with the instantaneous amplitude of the message signal m(t).The
standard form of an amplitude modulated (AM) wave is defined by
Sam(t) = Ac [1 + Ka m(t)] Cos(2пfct)
Sidebands along with the carrier can be represented as follows
Where Ka is constant called the amplitude sensitivity of the modulator. The

demodulation circuit is used to recover the message signal from the incoming AM wave at the
receiver. An envelope detector is a simple and yet highly effective device that is well suited for
the demodulation of AM wave, for which the percentage modulation is less than 100%.Ideally,
an envelope detector produces an output signal that follows the envelop of the input signal
wave form exactly; hence, the name. Some version of this circuit is used in almost all
commercial AM radio receivers.
One disadvantage of all amplitude modulation techniques (not only standard AM) is that the
receiver amplifies and detects noise and electromagnetic interference in equal proportion to the
signal. Increasing the received signal to noise ratio, say, by a factor of 10 (a 10 decibel
improvement), thus would require increasing the transmitter power by a factor of 10. This is in
contrast to frequency modulation (FM) and digital radio where the effect of such noise
following demodulation is strongly reduced so long as the received signal is well above the
threshold for reception. For this reason AM broadcast is not favored for music and high fidelity
broadcasting, but rather for voice communications and broadcasts (sports, news, talk radio
etc.).
To ensure that an amplitude modulated signal does not create spurious emissions outside the
normal bandwidth it is necessary to ensure that the signal does not become over -modulated -
this is a conditions that occurs when the modulation exceeds 100%. At this point the carrier
breaks up and intermodulation distortion occurs leading to large levels of unwanted noise
spreading out either side of the carrier and beyond the normal bandwidth. This can cause
interference to other users.
If over-modulation occurs, the carrier becomes phase inverted and this leads to sidebands
spreading out either side of the carrier. The different types of amplitude modulation are

a. double sideband modulation (DSB-AM)

b. double sideband suppressed carrier modulation (DSB SC-AM)
c. single sideband carrier modulation(SSB AM)
d. single sideband suppressed carrier modulation (SSB SC-AM)
e. vestigial sideband modulation (VSB AM)
Applications
Amplitude modulation is used in a variety of applications. Even though it is not as widely used
as it was in previous years in its basic format it can nevertheless still be found.
 Broadcast transmissions: AM is still widely used for broadcasting on the long,

medium and short wave bands. It is simple to demodulate and this means that radio
receivers capable of demodulating amplitude modulation are cheap and simple to
manufacture. Nevertheless many people are moving to high quality forms of
transmission like frequency modulation, FM or digital transmissions.
 Air band radio: VHF transmissions for many airborne applications still use AM. . It is
used for ground to air radio communications as well as two way radio links for ground
staff as well.
 Single sideband: Amplitude modulation in the form of single sideband is still used for
HF radio links. Using a lower bandwidth and providing more effective use of the
transmitted power this form of modulation is still used for many point to point HF
links.
 Quadrature amplitude modulation: AM is widely used for the transmission of data in
everything from short range wireless links such as Wi-Fi to cellular
telecommunications and much more. Effectively it is formed by having two carriers 90°
out of phase.
These form some of the main uses of amplitude modulation. However in its basic form, this
form of modulation is being sued less as a result of its inefficient use of both spectrum and
power.
Amplitude modulation advantages & disadvantages
Like any other system of modulation, amplitude modulation has several advantages
and disadvantages. These mean that it is used in particular circumstances where its advantages
can be used to good effect.

ADVANTAGES DISADVANTAGES
 It is simple to implement  An amplitude modulation signal
 It can be demodulated using is not efficient in terms of its
a circuit consisting of very power usage
few components  It is not efficient in terms of its
use of bandwidth, requiring a
bandwidth equal to twice that of
 AM receivers are very cheap the highest audio frequency
as no specialised components
are needed.
 An amplitude modulation signal
is prone to high levels of noise
because most noise is
amplitude based and obviously
AM detectors are sensitive to it.
In view of its characteristics advantages and disadvantages, amplitude modulation is being used less
frequently. However it is still in widespread use for broadcasting on the long, medium and short wave bands
as well as for a number of mobile or portable communications systems including some aircraft
communications.
Power in AM:
Pt=Pc(1+m2/2)
Pt =total power in AM
Pc=power in carrier
m=modulation index
Block Diagram

Circuit Diagram:
Demodulator
Procedure:
1. Switch ON the trainer board.
2. Observe the output of RF and AF signal generator using CRO, note that RF voltage is
300mV PP of 1MHz frequency and AF voltage is 10V PP of 2KHz frequency.
Modulation:
3. Now connect RF and AF signals to the respective inputs of modulation.
4. Initially set both the signals at zero level.
5. Connect one of the input of oscilloscope to modulator output and other input to AF
signal.
6. Adjust the RF signal amplitude with the help of potentiometer. So that output of the
modulator is 300mV PP by keeping AF signal at zero level.
7. Now vary the amplitude of AF signal and observe the amplitude modulated wave at
output, note the percentage of modulation for different values of AF signals.
% modulation = ((Vmax - Vmin) / (Vmax + Vmin) )x 100

8. Observe 100% amplitude modulation and over modulation by varying amplitude of the
AF signal.
Demodulation:
9. Now connect the modulated output to the demodulator input.

10. Observe demodulated signal at output of demodulator at approximately 50%

modulation using oscilloscope.
11. Compare demodulated signal with the original AF signal.
12. And plot the all graphs at 50% modulation
Calculations & Observations
Vmax  Vmin
Modulation Index 
V V
max min
Modulating Signal Generator
Amplitude = Time Period = Frequency =
Carrier Signal Generator:
Demodulated Output:
Tabular Form:
Modulating Modulation index
V V
signal Vmax Vmin m  max min
a V
mi
V
amplitude (V) max n

Model Wave Forms
Precautions
1. Avoid loose connections.

2. Avoid parallax error while taking observations.
Result:

S. No. Marks Split up Maximum marks Marks obtained
1 Pre-lab work 05
2 Post-lab work 05
3 Procedure & Result Analysis 05
4 Record 03
5 Viva 02
Total 20
Pre lab work :
1. Define Amplitude modulation.

2. What is mean by demodulation?
3. Define short wave, Long wave and Medium wave radios.
4. Transmission Bandwidth of AM wave?
5. Define Range of AM in Medium wave band?
6. Standard channel Bandwidth AM station in medium wave band?
7. Define broadcasting.
8. Given an example for Simplex, Half duplex and full duplex communications.
9. Why AM covers long distance?
10. What is trapezoidal waveform?
11. Why we need modulation?
12. When does a carrier is said to be over, under modulated in Amplitude modulation?
13. What is single tone and multi tone modulation?
Post Lab Work:
1. A 5O0 W carrier is modulated to a depth of 60 percent Calculate the total power in

modulated wave.
2. An AM transmitter radiates 10 KW of power without modulation with 0 65 depth of
modulation what is the radiated power?
3. A carrier is amplitude modulated to a depth of 75 percent. Calculate the total Power in
the modulated wave, if the carrier power is 40 watt.
4. A transmitter radiates 9 kW without modulation and 10.125 kW after modulation.
Determine depth of modulation.
2. DSB – SC MODULATOR AND DETECTOR

Date:
Grade:
Objective:

1. To observe the generation and detection processes of DSB-SC signal using Balanced
Modulator and observe the outputs.
2.
2.To simulate DSB-SC modulation and demodulation
Apparatus Required

Theory
Double-sideband suppressed-carrier transmission (DSB-SC) is transmission in which

frequencies produced by amplitude modulation (AM) are symmetrically spaced above and
below the carrier frequency and the carrier level is reduced to the lowest practical level, ideally
being completely suppressed.
DSB-SC is basically an amplitude modulation wave without the carrier, therefore reducing
power waste, giving it a 50% efficiency. This is an increase compared to normal AM
transmission (DSB), which has a maximum efficiency of 33.333%, since 2/3 of the power is in
the carrier which carries no intelligence, and each sideband carries the same information.
Single Side Band (SSB) Suppressed Carrier is 100% efficient.
In the DSB-SC modulation, unlike in AM, the wave carrier is not transmitted; thus, much of
the power is distributed between the sidebands, which implies an increase of the cover in DSB-
SC, compared to AM, for the same power used.
DSB-SC transmission is a special case of double-sideband reduced carrier transmission. It is

used for radio data systems.
Generation: DSB-SC is generated by a mixer. This consists of a message signal multiplied by

a carrier signal. The mathematical representation of this process is shown below, where the
product-to-sum trigonometric identity is used.

Demodulation is done by multiplying the DSB-SC signal with the carrier signal just like the
modulation process. This resultant signal is then passed through a low pass filter to produce a
scaled version of original message signal. DSB-SC can be demodulated by a simple envelope
detector, like AM, if the modulation index is less than unity. Full depth modulation requires
carrier re-insertion.
The equation above shows that by multiplying the modulated signal by the carrier
signal, the result is a scaled version of the original message signal plus a second term. Since,
this second term is much higher in frequency than the original message. Once this signal
passes through a low pass filter, the higher frequency component is removed, leaving just the
original message.
. With synchronous detection as shown in this applet, the modulated wave (receiving wave) is
multiplied with the carrier frequency which is exactly the same frequency and phase as the
transmission carrier wave. 　 For demodulation, the demodulation oscillator's frequency and

phase must be exactly the same as modulation oscillator's, otherwise, distortion and/or
attenuation will occur.
To see this effect, take the following conditions:
 Message signal to be transmitted:
 Modulation (carrier) signal:

 Demodulation signal (with small frequency and phase deviations from the modulation
signal):
Two types of generating methods:

1.Balanced modulator.
2.Ring modulator.
Block Diagram:
Circuit Diagram:

Procedure:
1. Observe the output of the RF generator using CRO. Output should be a sine wave of
100 KHz and 300 mV amplitude.
2. Observe the output of the AF generator using CRO. Output should be a sine wave of 5
KHz and 100mV amplitude.
3. connect the both RF and AF input of the balanced modulator and connect to the output
of the RF generator and AF generater.

4. Connect CRO CH1 input to input of the balanced modulator and CH2 input to the
output of the balanced modulator.
5. Observe the balanced modulator output by slowly increasing the amplitude of AF
signal
6. Compare the balanced modulator output with wave form DSB-SC.
7. Connect DSB-SC signal input of the synchronous detector to the output of the balanced
modulator and RF input to the output of the RF generator.
8. Observe the synchronous detector output using CRO and compare with the original AF
signal.
Observations
Message Signal:
Carrier Signal:
Demodulated Signal:
Amplitude = Time Period = Frequency
Model Wave Forms

Precautions
1. Avoid loose connections

Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work:
1. Draw the magnitude response or amplitude spectrum of DSBSC signal?

2. Amount of power saving in DSBSC signal is------------

3. Coherent detection means?
4. Give the practical applications of balanced modulator?
5. Differentiate synchronous and non synchronous detection techniques in analog
modulators?
6. If the circuit is operating in balanced state, the modulation index value is ----------
7. Transmission Bandwidth of DSB SC?
8. Prove that DSB−SC signal can be generated from two AM modulator, using
mathematics describe signal at each point?
Post Lab work:
1. How DSBSC is more efficient than AM in terms power saving, explain?

2. The phase shift at zero crossings in DSBSC wave is----------
3. Draw the wave form of DSBSC wave and AM wave, and differentiate those two
waveforms?
4. Why the circuit is called balanced modulator?
5. List out the applications of DSB SC.
3. SSB – SC MODULATOR AND DETECTOR
Objective

1.To understand the Process of single side band signal generation using phase shift method and
to demodulate the same using synchronous detector.
2.To simulate Single Sideband (SSB) modulation.
Apparatus Required

Theory
An SSB signal is produced by passing the DSB signal through a highly selective band
pass filter. This filter selects either the upper or the lower sideband. Hence transmission
bandwidth can be cut by half if one sideband is entirely suppressed. This leads to single side
band modulation (SSB). In SSB modulation bandwidth saving is accompanied by a
considerable increase in equipment complexity.
Single Sideband Suppressed Carrier (SSB-SC) modulation was the basis for all long
distance telephone communications up until the last decade. It was called "L carrier." It
consisted of groups of telephone conversations modulated on upper and/or lower sidebands of
contiguous suppressed carriers. The groupings and sideband orientations (USB, LSB)
supported hundreds and thousands of individual telephone conversations.
Due to the nature of-SSB, in order to properly recover the fidelity of the original audio,
a pilot carrier was distributed to all locations (from a single very stable frequency source), such
that, the phase relationship of the demodulated (product detection) audio to the original
modulated audio was maintained.
Also, SSB was used by the U.S. Air force's Strategic Air Command (SAC) to insure
reliable communications between their nuclear bombers and NORAD. In fact, before satellite
communications SSB-was the only reliable form of communications with the bombers.
The main reason-SSB-is superior to-AM,-and most other forms of modulation are:

(1) Since the carrier is not transmitted in SSB, there is a reduction by 50% of the transmitted
power. In AM out of 100% modulation: 67% of the power is comprised of the carrier; with the
remaining 33% power in both sidebands.
(2) Because in SSB, only one sideband is transmitted, there is a further reduction by 50% in
transmitted power.
(3) Finally, because only one sideband is received, the receiver's needed bandwidth is reduced
by one half--thus effectively reducing the required power by the transmitter another 50%
Generation of SSB-SC:
1.Filter method
2.Phase shift method
Coherent Demodulation of SSB signals :ΦSSB(t) is multiplied with cos(ωct) and passed through
low pass ﬁlter to get back the orignal signal
Advantages
Single sideband modulation is often compared to AM, of which it is a derivative. It has several
advantages for two way radio communication that more than outweigh the additional
complexity required in the SSB receiver and SSB transmitter required for its reception and
transmission.

1. As the carrier is not transmitted, this enables a 50% reduction in transmitter power level
for the same level of information carrying signal. [NB. for an AM transmission using
100% modulation, half of the power is used in the carrier and a total of half the power
in the two sideband – each sideband has a quarter of the power.]
2. As only one sideband is transmitted there is a further reduction in transmitter power.
3. As only one sideband is transmitted the receiver bandwidth can be reduced by half.
This improves the signal to noise ratio by a factor of two, i.e. 3 dB, because the
narrower bandwidth used will allow through less noise and interference.
The summary of this is that SSB modulation offers a far more effective solution for two way
radio communication because it provides a significant improvement in efficiency. SSB assures
the optimum exploitation of transmitted power and transmission bandwidth among the CW
modulation schemes.
Disadvantages
Complex transmitter and receiver conﬁgurations are required and Hard to demodulate.
Amplitude modulation typically produces a modulated output signal that has twice the
bandwidth of the modulating signal, with a significant power component at the center carrier
frequency. Single-sideband modulation improves this, at the cost of extra complexity.
To produce an SSB signal, a filter removes one of the sidebands. Most often, the carrier is
reduced (suppressed) or removed entirely. What remains still contains the entire information
content of the AM signal, using substantially less bandwidth and power, but cannot now be
demodulated by a simple envelope detector.
Practical implementations
A Collins KWM-1, an early Amateur Radio transceiver that featured SSB voice capability
Bandpass filtering
One method of producing an SSB signal is to remove one of the sidebands via filtering, leaving
only either the upper sideband (USB), the sideband with the higher frequency, or less
commonly the lower sideband (LSB), the sideband with the lower frequency. Most often, the
carrier is reduced or removed entirely (suppressed), being referred to in full as single sideband
suppressed carrier (SSBSC). Assuming both sidebands are symmetric, which is the case for a
normal AM signal, no information is lost in the process. Since the final RF amplification is
now concentrated in a single sideband, the effective power output is greater than in normal AM

(the carrier and redundant sideband account for well over half of the power output of an AM
transmitter). Though SSB uses substantially less bandwidth and power, it cannot be
demodulated by a simple envelope detector like standard AM.
Hartley modulator
An alternate method of generation known as a Hartley modulator, named after R. V. L.

Hartley, uses phasing to suppress the unwanted sideband. To generate an SSB signal with this
method, two versions of the original signal are generated, mutually 90° out of phase for any
single frequency within the operating bandwidth. Each one of these signals then modulates
carrier waves (of one frequency) that are also 90° out of phase with each other. By either
adding or subtracting the resulting signals, a lower or upper sideband signal results. A benefit
of this approach is to allow an analytical expression for SSB signals, which can be used to
understand effects such as synchronous detection of SSB.
Shifting the baseband signal 90° out of phase cannot be done simply by delaying it, as it
contains a large range of frequencies. In analog circuits, a wideband 90-degree phase-
difference network[7] is used. The method was popular in the days of vacuum tube radios, but
later gained a bad reputation due to poorly adjusted commercial implementations. Modulation
using this method is again gaining popularity in the homebrew and DSP fields. This method,
utilizing the Hilbert transform to phase shift the baseband audio, can be done at low cost with
digital circuitry.
Weaver modulator
Another variation, the Weaver modulator,[8] uses only lowpass filters and quadrature mixers,
and is a favored method in digital implementations.
In Weaver's method, the band of interest is first translated to be centered at zero, conceptually
by modulating a complex exponential with frequency in the middle of the voiceband, but
implemented by a quadrature pair of sine and cosine modulators at that frequency (e.g. 2 kHz).
This complex signal or pair of real signals is then lowpass filtered to remove the undesired
sideband that is not centered at zero. Then, the single-sideband complex signal centered at zero
is up-converted to a real signal, by another pair of quadrature mixers, to the desired center
frequency.
BlockDiagram:

Fig: Modulator
Fig:Demodulator
Experimental Procedure:
1. Study the circuit operation of SSB system thoroughly.

2. Observe the output of the RF Generator, using CRO. There are two outputs from the RF
Generator, one is direct output and other is 90° phase shift with the direct output. The output
frequency is 100KHZ and the amplitude is ≥ 0.2 Vpp (potentiometers are provided to vary the
output amplitude
3. Observe the output of the AF Generator using CRO.
4. There are two outputs from the AF generator, one is direct output. And the other is 90° phase
shift with the direct output. A switch is provided to select the required frequency (2K, 4K, or
6kHz). AGC potentiometer is provided to adjust the gain of the Oscillator (or to set the output
to good shape). And the amplitude is ≡ 10 Vpp (potentiometers are provided to vary the output
amplitude.
5. Measure and record the RF signal frequency using frequency counter.
6. Set the amplitude of the RF signals to 0.2 Vpp and connect 0° phase shift signal to one
balanced modulator and 90° phase shift signal to another balanced modulator as shown in
fig1.1
7. Select the required frequency (6KHz) of the AF generator with the help of switch and adjust the
AGC potentiometer until the output amplitude is 10 Vpp (when amplitude controls are in
maximum condition).
8. Measure and record the AF signal frequency using frequency counter.

9. Set the AF signal amplitude to 8 Vpp using amplitude control and connect to the balanced
modulators as shown in figure
10. Observe the outputs of both the balanced modulators simultaneously using dual trace
oscilloscope, and adjust the balance control until you get the output waveforms (DSB-SC) as
shown in below figure 1.3
11. To get SSB lower side band signal, connect balanced modulator outputs (DSB-SC signal) to
subtractor as shown in fig1.1
12. Measure and record the SSB signal frequency-using counter.
13. Calculate theoretical frequency of SSB (LSB) and compare it with the practical value
LSB = RF frequency – AF frequency
EX: If RF frequency is 100KHZ and AF frequency is 6KHZ Then LSB = 100khz – 6khz = 94khz
14. To get SSB upper side band signal, connect the output of the balanced modulator to the summer
circuit as shown in figure.
15. Measure and record the SSB upper side band frequency-using counter.
16. Calculate theoretical value of the SSB-USB frequency and compare and compare it with
practical value
USB = RF frequency + AF frequency.
EX: If RF frequency is100 kHz and AF frequency is 6 kHz then USB = 100kHz + 6 kHz = 106 kHz
17. Connect SSB signal from the summer or subtractor to the SSB signal input of the synchronous
detector and RF signals (0°) to the RF input of the synchronous detector as shown in fig 1.2
18. Observe the detector output using CRO and compare it with the modulating signal (AF
signal)as shown in fig 1.3
19. Observe the SSB signal for the different frequencies of the modulating (AF) signal.
Calculations & Observations

a. Theoretical frequency of LSB = RF – AF
b. Theoretical frequency of USB = RF + AF
RF Generator (Carrier Waveform):
AF Generator (Message Waveform):
Subtractor Output (fc - fm) =
Adder Output (fc + fm) =

LSB Output
USB Output
Demodulated Output:
Model Graphs
DSB-SC Signal

SSB – SC SIGNAL
Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20

Pre lab work:

1. SSB-SC is suitable for speech signals and not for video signals. Why?
2. What is the power saving from AM to DSB-SC and AM to SSB-SC ?
3. Give the methods of generating SSB-SC. And mention some applications of SSB-SC.
4. Compare AM with DSB-SC and SSB-SC.
5. Explain the practical difficulties of implementing SSB-SC.
6. Define Hilbert transform?
7. Compare SSB-SC and VSB.
8. What the percentage efficiency of SSB-SC.
Post Lab Work:
1. Evaluate the Hilbert Transform of

i. x(t) = sin t/ t* cos 200πt
ii. x(t) = sin t/t*sin 200πt.
2. A music signal with a bandwidth of 20 kHz is transmitted from an AM radio
station, with a carrier power of 50 W and transmission frequency of 650 kHz. If the
average music signal power is 3 W, determine the following:
(a) Power and bandwidth of the modulated signal, if conventional AM
modulation is used
(b) Power and bandwidth of the modulated signal, if SSB modulation is
used
3. What is the percentage of power saved in SSB when compared with AM system?
4. FREQUENCY MODULATION AND DEMODULATION

Objective:
1.To observe the process of frequency modulation and demodulation and to calculate the
modulation index (or depth of modulation).
2.To simulate Frequency modulation and demodulation.
Apparatus Required

□PPM Trainer kit □AGC Trainer kit □Frequency
Synthesizer Trainer kit □PLL as FM Demod Trainer kit □Bread board
Theory
The process, in which the frequency of the carrier is varied in accordance with the
instantaneous amplitude of the modulating signal, is called “Frequency Modulation”. The
FM signal is expressed as
Where Ac is amplitude of the carrier signal, fc is the carrier frequency β is the modulation
Index of the FM wave
Power in the FM/PM signal is relatively easier to calculate; since it is a cosine wave
with constant amplitude A, the average power is given by
The variation in frequency of the FM signal is called the maximum frequency deviation,
Δf.
∆f  fc  fa (or) ∆f  fb  fc

where fb  1 And fa  1
T T
min max
There are two ways to generate a frequency modulation signal, direct FM and indirect FM.
With frequency modulation, the information is incorporated in the frequency, so a
nonlinear power amplifier can be used which ensures good electrical efficiency.
・ DirectFM
With direct FM, the modulating signal is input in a VCO (voltage control oscillator) which
generates a frequency shift proportionally to the added voltage. The signal input of the
VCO has a variable capacitance diode, and when a modulating signal voltage is applied
here, the capacitance of the variable capacitance diode changes and the oscillation
frequency(carrierfrequency)changes
 The direct methods cannot be used for the broadcast applications. Thus the
alternative method i.e. indirect method called as the Armstrong method of FM
generation is used.
 In this method the FM is obtained through phase modulation. A crystal oscillator

can be used hence the frequency stability is very high.
Demodulation
The frequency discrimination method or PLL detection method can be used to
demodulate the frequency modulation signal. Demodulation in this applet uses envelope
detection after differentiation of the modulated wave. With frequency modulation,
information is carried by the frequency of the carrier wave, so it’s said to be resistant to
noise in the amplitude direction. FM detectors detect frequencies and convert them into
voltages, and since they respond to amplitude fluctuations, these cause errors. The
received signal is affected by fading and outside noise which affects the amplitude
direction and appears as distortion or noise in the detector output signal. Therefore a
limiter circuit is built in before the detector in order to eliminate amplitude fluctuations
There are various reasons why FM is used, but the main reasons are as follows.
1, It has good tone quality
2, It’s resistant to noise

3, The hardware is cheap

4, It has a good track record with reliable communication
5, You can set an appropriate modulation factor to improve signal to noise
6, It’s resistant to fading
7, The technology is well established
8, Component parts are readily available
Applications
Magnetic tape storage
FM is also used at intermediate frequencies by analog VCR systems (including

VHS) to record the luminance (black and white) portions of the video signal. Commonly,
the chrominance component is recorded as a conventional AM signal, using the higher-
frequency FM signal as bias. FM is the only feasible method of recording the luminance
("black and white") component of video to (and retrieving video from) magnetic tape
without distortion; video signals have a large range of frequency components – from a few
hertz to several megahertz, too wide for equalizers to work with due to electronic noise
below −60 dB. FM also keeps the tape at saturation level, acting as a form of noise
reduction; a limiter can mask variations in playback output, and the FM capture effect
removes print-through and pre-echo. A continuous pilot-tone, if added to the signal – as
was done on V2000 and many Hi-band formats – can keep mechanical jitter under control
and assist time base correction.
These FM systems are unusual, in that they have a ratio of carrier to maximum modulation
frequency of less than two; contrast this with FM audio broadcasting, where the ratio is
around 10,000. Consider, for example, a 6-MHz carrier modulated at a 3.5-MHz rate; by
Bessel analysis, the first sidebands are on 9.5 and 2.5 MHz and the second sidebands are
on 13 MHz and −1 MHz. The result is a reversed-phase sideband on +1 MHz; on
demodulation, this results in unwanted output at 6−1 = 5 MHz. The system must be
designed so that this unwanted output is reduced to an acceptable level.

Sound
FM is also used at audio frequencies to synthesize sound. This technique, known as

FM synthesis, was popularized by early digital synthesizers and became a standard feature
in several generations of personal computer sound cards.
Radio
Edwin Howard Armstrong (1890–1954) was an American electrical engineer who

invented wideband frequency modulation (FM) radio. He patented the regenerative circuit
in 1914, the superheterodyne receiver in 1918 and the super-regenerative circuit in 1922.
Armstrong presented his paper, "A Method of Reducing Disturbances in Radio Signaling
by a System of Frequency Modulation", (which first described FM radio) before the New
York section of the Institute of Radio Engineers on November 6, 1935. The paper was
published in 1936
As the name implies, wideband FM (WFM) requires a wider signal bandwidth than
amplitude modulation by an equivalent modulating signal; this also makes the signal more
robust against noise and interference. Frequency modulation is also more robust against
signal-amplitude-fading phenomena. As a result, FM was chosen as the modulation
standard for high frequency, high fidelity radio transmission, hence the term "FM radio"
(although for many years the BBC called it "VHF radio" because commercial FM
broadcasting uses part of the VHF band—the FM broadcast band). FM receivers employ a
special detector for FM signals and exhibit a phenomenon known as the capture effect, in
which the tuner "captures" the stronger of two stations on the same frequency while
rejecting the other (compare this with a similar situation on an AM receiver, where both
stations can be heard simultaneously). However, frequency drift or a lack of selectivity
may cause one station to be overtaken by another on an adjacent channel. Frequency drift
was a problem in early (or inexpensive) receivers; inadequate selectivity may affect any
tuner.
An FM signal can also be used to carry a stereo signal; this is done with multiplexing and
demultiplexing before and after the FM process. The FM modulation and demodulation
process is identical in stereo and monaural processes. A high-efficiency radio-frequency

switching amplifier can be used to transmit FM signals (and other constant-amplitude

signals). For a given signal strength (measured at the receiver antenna), switching
amplifiers use less battery power and typically cost less than a linear amplifier. This gives
FM another advantage over other modulation methods requiring linear amplifiers, such as
AM and QAM.
FM is commonly used at VHF radio frequencies for high-fidelity broadcasts of music and
speech. Analog TV sound is also broadcast using FM. Narrowband FM is used for voice
communications in commercial and amateur radio settings. In broadcast services, where
audio fidelity is important, wideband FM is generally used. In two-way radio, narrowband
FM (NBFM) is used to conserve bandwidth for land mobile, marine mobile and other
radio services.
There are reports that on October 5, 1924, Professor Mikhail A. Bonch-Bruevich, during a
scientific and technical conversation in the Nizhny Novgorod Radio Laboratory, reported
about his new method of telephony, based on a change in the period of oscillations.
Demonstration of frequency modulation was carried out on the laboratory model.
Block Diagram:

Circuit diagrams:
Procedure:
1. Connections are made as per the circuit diagram.
2. Observe the output of AF generator using CRO. Note that AF voltage is approximately
20V PP of 500Hz frequency.

Modulator:
3. Observe the carrier signal at the modulator output. The carrier signal is approximately 5V
PP of 100 KHz frequency.
4. Connect AF signal to the AF input of the modulator.
5. Observe the frequency modulated wave at the modulator output by varying deviation
potentiometer, which varies the amplitude of the incoming AF signal.
6. Set deviation potentiometer in minimum position and calculated the output signal
frequency of modulator which indicates RF frequency i.e fc.
7. Now set the deviation potentiometer to middle position and observe the output signal.
Calculate the maximum frequency i.e, S = fb-fc.
8. Now find the modulation index M – S/Fm.
Fm = Maximum frequency deviated / Modulating frequency.
9. Now set the AF signal to 5 KHz frequency.
10. Connect one of the input of oscilloscope to the modulator output and another input to AF
generator.
11. Observe trace, the frequency modulated wave and AF signal simultaneously.
Demodulator:
12. Connect the modulator output to the demodulator input.
13. Observe trace the demodulated signal at the output of modulator and compare it with the
original AF signal.
Calculations
a) Modulation index
S
mF  where f m is the modulating signal frequency
fm
b) Frequency deviation, S
S  fc  f a (or) S  fb  fc
Where fb  1 And fa  1
T T
min max
Modulating Signal Generator:

Carrier Signal Generator:
Demodulated Output:
Observations
Modulating Modulation
Modulating Signal Frequency index
signal Tmin Tmax
amplitude deviation (S) S
frequency(fm) f
mf  m
(V)

Waveforms:

Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20

Pre lab work:

1. Define Frequency modulation? How it is different from phase modulation?
2. Give the Carson’s rule in FM?
3. Differentiate Narrow band FM with Wide band FM?
4. How FM wave is different from PM wave?
5. In commercial FM broadcasting, the audio frequency range handled is only up-to
--------.
6. FM Broadcasting range?
7. Standard IF value of FM receiver?
8. How FM wave is generated from PM?
9. Compare AM and FM.
10. Define modulation index β, frequency deviation?
11. Draw the amplitude spectrum of FM wave?
12. Write equation of FM wave, explain each parameter in it?
13. State advantages and disadvantages of FM?
Post Lab work:

1. FM is more robust to noise compared to AM, why?
2. List the Applications of FM.
3. The transmission band width required for commercial FM broadcasting is
-----------

5. STUDY OF SPECTRUM ANALYZER AND ANALYSIS OF

AM AND FM SIGNALS
Objective:
1.To observe the spectrum of AM and FM signals and obtain the power levels in dBm of
fundamental frequency components by using spectrum Analyzer.
2.To simulate the spectrum of AM and FM signals .
Apparatus Required

Theory:
A spectrum analyzer provides a calibrated graphical display on its CRT with frequency
on the horizontal axis and amplitude on the vertical axis. Displayed as vertical lines
against these coordinates are sinusoidal components of which the input signal in
composed. The height represents the absolute magnitude, and horizontal location
represents the frequency. This instrument provide a display of the frequency spectrum over
a given frequency band.

The common way of observing electrical signals is to view them in the time domain
using an oscilloscope. The time domain is used to recover relative timing and phase
information which is needed to characterize the electric circuit behavior. However not all
circuits can be characterized from just time domain information. Circuit such as
amplifiers, oscillators, mixers, modulators, detectors and filters are best characterized by
their frequency response information. This frequency information is best obtained by
viewing electrical signals in the frequency domain. To display the frequency domain
requires a device that can discriminate between frequencies while measuring the power
level at each. One instrument which displays the signals in frequency domains is the
spectrum analyzer. It graphically displays Voltage or power as a function of frequency on a
CRT (Cathode Ray Tube) In the time domain all frequency components of a signal are
seen summed together. In the frequency domain, complex signals (i.e., signals composed
of more than on frequency) are separated into their frequency components and the power
level at each frequency is displayed in the frequency domain as a graphical representation.
The frequency domain contains information not found in the time domain and therefore
the spectrum analyzer has certain advantages compared to oscilloscope.
Spectrum analyzer types are distinguished by the methods used to obtain the spectrum of a
signal. There are swept-tuned and Fast Fourier Transform (FFT) based spectrum
analyzers:
 A swept-tuned analyzer uses a superheterodyne receiver to down-convert a portion

of the input signal spectrum to the center frequency of a narrow band-pass filter,
whose instantaneous output power is recorded or displayed as a function of time.
By sweeping the receiver's center-frequency (using a voltage-controlled oscillator)
through a range of frequencies, the output is also a function of frequency. But
while the sweep centers on any particular frequency, it may be missing short-
duration events at other frequencies.
 An FFT analyzer computes a time-sequence of periodograms. FFT refers to a
particular mathematical algorithm used in the process. This is commonly used in
conjunction with a receiver and analog-to-digital converter. As above, the receiver
reduces the center-frequency of a portion of the input signal spectrum, but the
portion is not swept. The objective of the receiver is to reduce the sampling rate
that the analyzer must contend with. With a sufficiently low sample-rate, FFT
analyzers can process all the samples (100% duty-cycle), and are therefore able to
avoid missing short-duration events.
Typical functionality
Center frequency and span
In a typical spectrum analyzer there are options to set the start, stop, and center frequency.
The frequency halfway between the stop and start frequencies on a spectrum analyzer
display is known as the center frequency. This is the frequency that is in the middle of the
display’s frequency axis. Span specifies the range between the start and stop frequencies.
These two parameters allow for adjustment of the display within the frequency range of
the instrument to enhance visibility of the spectrum measured.
Resolution bandwidth
The resolution bandwidth filter or RBW filter is the bandpass filter in the IF path. It's the
bandwidth of the RF chain before the detector (power measurement device).[7] It
determines the RF noise floor and how close two signals can be and still be resolved by
the analyzer into two separate peaks.[7] Adjusting the bandwidth of this filter allows for the
discrimination of signals with closely spaced frequency components, while also changing
the measured noise floor. Decreasing the bandwidth of an RBW filter decreases the
measured noise floor and vice versa. This is due to higher RBW filters passing more
frequency components through to the envelope detector than lower bandwidth RBW
filters, therefore a higher RBW causes a higher measured noise floor.
Video bandwidth
The video bandwidth filter or VBW filter is the low-pass filter directly after the envelope
detector. It's the bandwidth of the signal chain after the detector. Averaging or peak
detection then refers to how the digital storage portion of the device records samples—it
takes several samples per time step and stores only one sample, either the average of the
samples or the highest one.[7] The video bandwidth determines the capability to
discriminate between two different power levels.[7] This is because a narrower VBW will
remove noise in the detector output.[7] This filter is used to “smooth” the display by

removing noise from the envelope. Similar to the RBW, the VBW affects the sweep time
of the display if the VBW is less than the RBW. If VBW is less than RBW, this relation for
sweep time is useful:
Here tsweep is the sweep time, k is a dimensionless proportionality constant, f2 − f1 is the

frequency range of the sweep, RBW is the resolution bandwidth, and VBW is the video
bandwidth.[8]
Detector
With the advent of digitally based displays, some modern spectrum analyzers use analog-
to-digital converters to sample spectrum amplitude after the VBW filter. Since displays
have a discrete number of points, the frequency span measured is also digitised. Detectors
are used in an attempt to adequately map the correct signal power to the appropriate
frequency point on the display. There are in general three types of detectors: sample, peak,
and average
 Sample detection – sample detection simply uses the midpoint of a given interval
as the display point value. While this method does represent random noise well, it
does not always capture all sinusoidal signals.
 Peak detection – peak detection uses the maximum measured point within a given
interval as the display point value. This insures that the maximum sinusoid is
measured within the interval; however, smaller sinusoids within the interval may
not be measured. Also, peak detection does not give a good representation of
random noise.
 Average detection – average detection uses all of the data points within the
interval to consider the display point value. This is done by power (rms) averaging,
voltage averaging, or log-power averaging.
Displayed average noise level
The Displayed Average Noise Level (DANL) is just what it says it is—the average noise
level displayed on the analyzer. This can either be with a specific resolution bandwidth
(e.g. −120 dBm @1 kHz RBW), or normalized to 1 Hz (usually in dBm/Hz) e.g. −170

dBm(Hz).This is also called the sensitivity of the spectrum analyzer. If a signal level equal
to the average noise level is fed there will be a 3 dB display. To increase the sensitivity of
the spectrum analyzer a preamplifier with lower noise figure may be connected at the
input of the spectrum analyzer.
APPLICATIONS:
Spectrum analyzers are widely used to measure the frequency response, noise and
distortion characteristics of all kinds of radio-frequency (RF) circuitry, by comparing the
input and output spectra.For example in RF mixers, spectrum analyzer is used to find the
levels of third order inter-modulation products and conversion loss. In RF oscillators,
spectrum analyzer is used to find the levels of different harmonics.
In telecommunications, spectrum analyzers are used to determine occupied

bandwidth and track interference sources. For example, cell planners use this equipment to
determine interference sources in the GSM frequency bands and UMTS frequency bands.
In EMC testing, a spectrum analyzer is used for basic precompliance testing;

however, it can not be used for full testing and certification. Instead, an EMI receiver is
used.

A spectrum analyzer is used to determine whether a wireless transmitter is working

according to defined standards for purity of emissions. Output signals at frequencies other
than the intended communications frequency appear as vertical lines (pips) on the display.
A spectrum analyzer is also used to determine, by direct observation, the bandwidth of a
digital or analog signal.
A spectrum analyzer interface is a device that connects to a wireless receiver or a

personal computer to allow visual detection and analysis of electromagnetic signals over a
defined band of frequencies. This is called panoramic reception and it is used to determine
the frequencies of sources of interference to wireless networking equipment, such as Wi-Fi
and wireless routers.
Spectrum analyzers can also be used to assess RF shielding. RF shielding is of

particular importance for the siting of a magnetic resonance imaging machine since stray
RF fields would result in artifacts in an MR image.[10]
A typical application is to measure the distortion of a nominally sinewave signal; a

very-low-distortion sinewave is used as the input to equipment under test, and a spectrum
analyser can examine the output, which will have added distortion products, and determine
the percentage distortion at each harmonic of the fundamental
Block diagram:
Procedure:
1. AM signal is given to the spectrum analyzer.
2. Adjust the zero marker to carrier frequency and measure spectrum of AM.

3. For different values of fc and fm, observe the spectrum of AM.

4. Now remove AM signal and give FM signal to the spectrum analyzer.
5. Adjust the zero marker to carrier frequency and observe spectrum of FM.
6. Plot the spectrums of FM and AM.
Observation Table:
Model Graphs:

Precautions
Results:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work:

1. Distinguish between CRO and Spectrum analyzer?
2. What are the functions of span/div control and reference level control?

3. Difference between time domain signal and frequency domain signal.

4.What are the types of spectrum analyzer.
5.List different detectors used.
6.What is video bandwidth.
7.what is meant by Displayed average noise level.
8.How to calculate center frequency.
Post Lab work:

1.List different applications of spectrum analyzer.
2.What is Resolution bandwidth.
3. Draw the phasor representation of an amplitude modulated wave.
4. Draw the phasor representation of frequency modulated wave.
6. PRE-EMPHASIS AND DE-EMPHASIS
Objective:
1. To observe the Process of Pre-emphasis and De-emphasis. To calculate the gain of Pre-
emphasis and de-emphasis and draw the corresponding frequency response curves.
2. To simulate the frequency response of Pre-Emphasis and De-Emphasis.
Apparatus Required

□FDM Trainerkit □ PAM Trainer kit □PWM Trainer kit
Theory:
Pre-emphasis:
Frequencies contain in human speech mostly occupy the region from 100 to 10,000
Hz, but most of the power is contained in the region of 500 Hz for men and 800 Hz for
women. Common voice characteristics emit low frequencies higher in amplitude than
higher frequencies. The problem is that in FM system the noise output of the receiver
increases linearly with the frequency, which means that the signal to noise ratio becomes
poorer as the modulating frequency increases.
Also, noise can make radio reception less readable and unpleasant. This noise is
greatest in frequencies above 3KHz.The high frequency noise causes interference to the
already weak high frequency voice. To reduce the effect of this noise and ensure an even
power spread of audio frequencies, Pre emphasis is used at the Transmitter side.
A pre-emphasis network in the transmitter accentuates the audio frequencies above
3 KHz, so providing the higher average deviation across the voice spectrum, thus
improving the signal to noise ratio.
The pre-emphasis is obtained by using the simple audio filter; even simple RC
filter will do the job. The pre-emphasis circuit produces higher output at higher
frequencies because the capacitive reactance is decreased as the frequency increases.

De-emphasis:
The problem in FM broadcasting is that noise and hiss tend to be more noticeable,
especially when receiving the weaker stations. To reduce this effect, the treble response of
the audio signal is artificially boosted prior to transmission. This is known as pre-emphasis
At the receiver side a corresponding filter or “de-emphasis” circuit is required to

reduce the treble response to correct level. Since most noise and hiss tend to be at the
higher frequencies, the de-emphasis removes a lot of this. Pre-emphasis and de-emphasis
thus allow an improved signal to noise ratio to be achieved while maintaining the
frequency response of the original audio signal. The de-emphasis stage is used after the
detector stage.
Fig : Frequency response of Pre-emphasis & De-emphasis circuits

Magnitude response:
x(f)=(1/sqrt(1+(f1/f)^2));
y(f)=(1/sqrt(1+(f/f1)^2));
The FM broadcasting industry uses pre-emphasis and de-emphasis techniques to

improve their signal-to-noise ratios. It's been correctly pointed out that audio frequencies
below the breakpoint are transmitted flat, and audio frequencies above the breakpoint are
transmitted pre-emphasized. (There have been other such "curves" used to tailor response,
such as the RIAA curve in phonograph records, and the NAB curve in tape recording.)
But that isn't the original reason pre-emphasis and de-emphasis were used in
narrow-band radio. The early transmitters were PM (phase modulated), not FM, so they
naturally had a 6 dB/octave pre-emphasis. PM became the standard modulation method.
When FM transmitters came along, their audio had to be intentionally pre-emphasized to
maintain compatibility with the PM transmitters already in service. In very early narrow-
band literature, you won't even find the terms "pre-emphasis" and "de-emphasis".
Engineers simply "rolled off" the audio in the receiver with a single pole filter because
they had to defeat the PM transmitter's characteristic "roll-up". The pre-emph and de-emph
terms came from the broadcast people. (I wish the narrow-band radio industry had better
terms for these characteristics. Unlike the broadcasters with their middle-of-the-band
breakpoint, in narrow-band radio the breakpoints are outside the voice bandwidth.) So, de-
emphasis has little to do with signal-to-noise radio and everything to do with making the
response correct. If FM had always been used, there never would have been pre-emph or
de-emph in narrow-band radio
Circuit Diagram:

Procedure:
Pre-emphasis
1. First observe the output of the AF generator using CRO. It is sine wave of 10V and
frequency range from 200 Hz to 100 KHz.
2. Connect AF signal to the one of the pre-emphasis network.
3. Output of the pre-emphasis observed on CRO.
4. Adjust the AF signal to required amplitude level.
5. By varying AF signal frequency in steps note down the corresponding input and output
voltages. (Keep AF input signal amplitude constant for all frequencies)
6. Plot a graph between frequency Vs output voltage
7. From graph note the frequency at which the output voltage is 70.7% of the input voltage
and compare it with the theoretical frequency.
8. Repeat the steps from 5 to 10 for different pre-emphasis network.
Theoretical frequency is f = 1 / (2ΠRC)

De-emphasis:
1. Connect AF signal to the one of the de-emphasis networks

2. Output of the de-emphasis is observed on CRO.

3. Adjust AF signal to required amplitude level.
4. By varying the AF signal frequency in steps note down the corresponding input and output
voltage.
5. Plot the graph between frequency Vs voltage.
6. From graph between at which the output voltage is 70.7% of the input voltage and
compare it with theoretical frequency.
7. Repeat the steps from 5 to 10 for different de-emphasis network.
Theoretical frequency is f = 1 / (2ΠRC)
Graphs:
Pre-emphasis De-emphasis
Observation Table:
For T=75us Input voltage:10V
Frequency O/P voltage Gain in db= Frequency O/P voltage Gain in db=
(Hz) (Vpp) 20log(Vo/Vin) (Hz) (Vpp) 20log(Vo/Vin)
200 200
400 400

600 600
800 800
1k 1k
2k 2k
4k 4k
8k 8k
10k 10k
12k 12k
14k 14k
16k 16k
18k 18k
20k 20k
50k 50k
100k 100k
200k 200k
For T=50us
Frequency O/P voltage Gain in db= Frequency O/P voltage Gain in db=
(Hz) (Vpp) 20log(Vo/Vin) (Hz) (Vpp) 20log(Vo/Vin)
200 200
400 400
600 600
800 800
1k 1k
2k 2k
4k 4k
8k 8k
10k 10k
12k 12k
14k 14k
16k 16k
18k 18k++
20k 20k
50k 50k
100k 100k
200k 200k
Precautions
Result:

1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work :

1. Define pre-emphasis and De-emphasis processes in FM.
2. Why Pre-emphasis is used at Transmitter of FM and de-emphasis at FM receiver?
3. Draw the pre-emphasis circuit and explain its working in detail?
4. Draw de-emphasis circuit and explain its working in detail?
5. Define 3dB frequencies?
6. What is the necessity of boosting up high frequencies in frequency modulation
communication system?
7. Draw the frequency response characteristics of pre-emphasis and de-emphasis
explain each one in detail?

8. Design a circuit to boost the baseband signal amplitude in the FM Transmitter for
the cut off frequency fc=2KHz.
Post Lab Work:

1. Pre-emphasis circuit operation is similar to -----------
2. De-emphasis circuit operation is similar to -----------
3. Calculate the cut-off frequencies of pre-emphasis and de-emphasis circuits
practically.
7. TIME DIVISION MULTIPLEXING & DEMULTIPLEXING
Objective:
1. To study Time Division Multiplexing and De-multiplexing, using Pulse Amplitude
Modulation and Demodulation and to reconstruct the signals at the Receiver, using Filters.
2. To simulate Time division multiplexing (TDM) and de-multiplexing.
Apparatus Required


Theory:
Time-division multiplexing (TDM) is a method of transmitting and receiving

independent signals over a common signal path by means of synchronized switches at
each end of the transmission line so that each signal appears on the line only a fraction of
time in an alternating pattern. It is used when the data rate of the transmission medium
exceeds that of signal to be transmitted. This form of signal multiplexing was developed in
telecommunications for telegraphy systems in the late 19th century, but found its most
common application in digital telephony in the second half of the 20th century
Time-division multiplexing is used primarily for digital signals, but may be applied in
analog multiplexing in which two or more signals or bit streams are transferred appearing
simultaneously as sub-channels in one communication channel, but are physically taking
turns on the channel. The time domain is divided into several recurrent time slots of fixed
length, one for each sub-channel. A sample byte or data block of sub-channel 1 is
transmitted during time slot 1, sub-channel 2 during time slot 2, etc. One TDM frame
consists of one time slot per sub-channel plus a synchronization channel and sometimes
error correction channel before the synchronization. After the last sub-channel, error
correction, and synchronization, the cycle starts all over again with a new frame, starting
with the second sample, byte or data block from sub-channel 1, etc
TDM can be further extended into the time-division multiple access (TDMA) scheme,
where several stations connected to the same physical medium, for example sharing the
same frequency channel, can communicate. Application examples include:

 The GSM telephone system

 The Tactical Data Links Link 16 and Link 22
Types of TDM
1. Synchronous TDM
2. Asynchronous TDM
In synchronous TDM, each device is given same time slot to transmit the data over the
link, irrespective of the fact that the device has any data to transmit or not. Hence the name
Synchronous TDM. Synchronous TDM requires that the total speed of various input lines
should not exceed the capacity of path
Asynchronous TDM
1. It is also known as statistical time division multiplexing.
2. Asynchronous TDM is called so because is this type of multiplexing, time slots are not
fixed i.e. the slots are flexible.
3. Here, the total speed of input lines can be greater than the capacity of the path.

4. In synchronous TDM, if we have n input lines then there are n slots in one frame. But in
asynchronous it is not so.
Advantages of TDM :
1. Full available channel bandwidth can be utilized for each channel.

2.lntermodulation distortion is absent.
3. TDM circuitry is not very complex.
4. The problem of crosstalk is not severe.
Disadvantages of TDM :
1.Synchronization is essential for proper operation.

2. Due to slow narrowband fading, all the TDM channels may get wiped out.
Applications:
The T1 system for wireline telephone networks

The T1 system is used for wireline long-distance service in North America and is
an excellent example of TDM. Speech from a telephone conversation is sampled once
every 125 msec and each sample is converted into eight bits of digital data (see Chapter 8
for more details). Using this technique, a transmission speed of 64,000 bits/sec is required
to transmit the speech. A T1 line is essentially a channel capable of transmitting at a speed
of 1.544 Mbit/sec. This is a much higher transmission speed than a single telephone
conversation needs, so TDM is used to allow a single T1 line to carry 24 different speech
signals between, say, two different telephone substations (called central offices) within a
city. As shown in Figure . the 1.544 Mbit/sec bit stream is divided into 193-bit frames. The
duration of each frame is
corresponding to the period between samples of the speech. Each frame is divided into 24
slots, which are each eight bits wide (corresponding to the number of bits needed to
digitize a speech sample). One additional bit at the end of the frame is used for signaling.

The eight bits of data corresponding to a sample of the speech are placed into one of the
24 slots in the frame.
For longer distances (say, between two large cities) higher-capacity channels are used and
multiple T1 lines are time division multiplexed onto the new channels. A T3 channel for
example, has a transmission speed of 44.736 Mbit/sec and uses TDM to carry 28 T1 lines
(a total of 672 different speech signals) plus signaling. For more information on this
hierarchical multiplexing system.
Fig:Time division multiplexing on a T1 line.
TDM with sources having different data rates

Consider the case of three streams with bit rates of 8 kbit/sec,16 kbit/sec, and 24
kbit/sec, respectively. We want to combine these streams into a single high-speed stream
using TDM. The high-speed stream in this case must have a transmission rate of 48
kbit/sec, which is the sum of the bit rates of the three sources. To determine the number of
time slots to be assigned to each source in the multiplexing process. we must reduce the
ratio of the rates, 8:16:24, to the lowest possible form, which in this case is 1:2:3. The sum
of the reduced ratio is 6, which will then represent the minimum length of the repetitive
cycle of slot assignments in the multiplexing process. The solution is now readily
obtained: In each cycle of six time slots we assign one slot to Source A (8 kbit/sec), two
slots to Source B (16 kbit/sec), and three slots to Source: C (24 kbit/sec). Figure 7-4
illustrates this assignment, using “a” to indicate data from Source A, “b” to indicate data
from Source B, and “c” to indicate data from Source C.
Figure—Multiplexing input lines with different transmission speeds.
A more complex TDM system
 Consider a system with four low-bit-rate sources of 10 kbit/sec, 15 kbit/sec, 20

kbit/sec, and 30 kbit/sec. Determine the slot assignments when the data streams are
combined using TDM.
Solution
 The rate ratio 10:15:20:30 reduces to 2:3:4:6. The length of the cycle is therefore 2
+ 3 + 4 + 6 = 15 slots. Within each cycle of 15 slots, we assign two slots to the 10
kbit/sec source, three slots to the 15 kbit/sec source, four slots to the 20 kbit/sec
source, and six slots to the 30 kbit/sec source.
 So far we have considered a form of TDM that is based on fixed slot assignments
to each of the low-bit-rate data streams. In other words, each stream has predefined
slot positions in the combined stream, and the receiver must be aware which slots
belong to which input stream. Both transmission ends, the transmitter and the
receiver, must be perfectly synchronized to the slot period. For this reason, the
technique is usually called synchronous TDM.
 There is another important version of TDM, usually referred to as statistical TDM.

Statistical TDM is useful for applications in which the low-bit-rate streams have
speeds that vary in time. For example, a low-bit-rate stream to a single terminal in
a computer network may fluctuate between 2 kbit/sec and 50 kbit/sec during an
active connection session (we've all seen variable speeds during Internet
connections, for instance). If we assign the stream enough slots for its peak rate
(that is, for 50 kbit/sec), then we will be wasting slots when the rate drops well
below the peak value. This waste can be especially significant if the system has
many variable-speed low-bit-rate streams.

 Statistical TDM works by calculating the average transmission rates of the streams
to be combined, and then uses a high-speed multiplexing link with a transmission
rate that is equal to (or slightly greater than) the statistical average of the combined
streams. Since the transmission rates from each source are variable, we no longer
assign a fixed number of time slots to each data stream. Rather, we dynamically
assign the appropriate number of slots to accommodate the current transmission
rates from each stream. Because the combined rate of all the streams will also
fluctuate in time between two extreme values, we need to buffer the output of the
low-bit-rate streams when the combined rate exceeds the transmission rate of the
high-speed link.
 With statistical TDM, we are no longer relying on synchronized time slots with
fixed assignments for each input stream, as we did with synchronous TDM. So
how does the demultiplexer in statistical TDM know which of the received bits
belongs to which data stream? Prior to transmission, we divide each stream of bits
coming from a source into fixed-size blocks. We then add a small group of bits
called a header to each block, with the header containing the addresses of the
source and intended user for that block. The block and the header are then
transmitted together across the channel. Combined, the block and header are called
a packet.
 Actually, the header may contain other information besides the source and user
addresses, such as extra bits for error control or additional bits for link control
(used, for example, to indicate the position of a particular block in a sequence of
blocks coming from the same user, or to indicate priority level for a particular
message). Extra bits can also be added to the beginning and end of a block for
synchronization; a particular pattern of bits, called a start flag, can be used in the
header to mark the start of a block, and another particular pattern of bits, called an
end flag, can be used to conclude the block. Each block transmitted across the
channel thus contains a group of information bits that the user wants, plus
additional bits needed by the system to ensure proper transmission. These
additional bits, while necessary to system operation, reduce the effective
transmission rate on the channel. Figures present the statistical TDM technique and
the structure of a typical packet.

 Figure 7-5—Statistical TDM.
Block Diagram:
Procedure for Modulation:

1. Connect the four channel input 250 KHz, 500 KHz, 1 KHz, 2 KHz to the input of
transmitter CH0, CH1, CH2 and CH3 respectively.
2. Connect the TX clock transmitter clock to RX clock (Receiver clock).
3. Connect the TX CH0 (Transmitter Sync) to RX CH0 (Receiver Sync)
4. Connect the TXD (Transmitter Data) to RXD(receiver Data).
5. Observe the multiplexed Data at TDX
Procedure for Demodulation:
6. Connect the TDX output the RDX Receiver.

7. Observe the Demultiplexed signals at the receiver across the output of fourth order low
pass filter at CH0, CH0, CH1, CH2, and CH3 respectively.
Calculations:
Transmitter side:
Message signal (CH0)
Receiver side:
De-multiplexed signal (OUT0)

Expected Graph:

Wave forms:

Precautions
1.Avoid loose connections.
2.Avoid parallax error while taking observations.
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work

1. 1 What is meant by multiplexing?
2. Mention the types of multiplexing?

3. What is the need for multiplexing?

4. Distinguish between the two basic multiplexing techniques?
5. Why sync pulse is required in TDM?
6. What are the functions of commutator switch?
7. Give the advantages of multiplexing.
8. State the applications of TDM.
9. Two signals g1(t) and g2(t) are to be transmitted over a common channel by means
of time division multiplexing. The highest freq of g1(t) is 1 KHz and that g2(t) is
1.5 KHz. What is the minimum value of the permissible sampling rate? Justify
your answer.
Post Lab Work:

1. Twenty four voice signals are sampled uniformly and then time division multiplexed.
The sampling operation uses flat top samples with 1μs duration. The multiplexing
operation includes provision for synchronization by adding an extra pulse of sufficient
amplitude and also 1 μs duration. The highest frequency component of each voice signal is
3.4KHz.
a. Assuming a sampling rate of 8 KHz, Find the spacing between successive
pulses of the multiplexed signal.
b. Repeat your calculation assuming the use of nyquist rate sampling.
2.Three signals m1,m2 and m3 are to be multiplexed.m1 and m2 have a 5 KHz bandwidth
and m3 has 10KHz bandwidth. Design a commutator switching system so that each signal
is sampled at its Nyquist rate.
8. Frequency Division Multiplexing & De-multiplexing
Objective:

1. To Study the Frequency Division multiplexing/ De-multiplexing with Sinusoidal waves.

2. To simulate Frequency Division multiplexing/ De-multiplexing with Sinusoidal waves.
Apparatus Required

Theory:
Frequency-division multiplexing (FDM) is a scheme in which numerous signals

are combined for transmission on a single communications line or channel. Each signal is
assigned a different frequency (sub-channel) within the main channel.
Suppose a long-distance cable is available with a bandwidth allotment of three
megahertz (3 MHz). This is 3,000 kHz, so in theory, it is possible to place 1,000 signals,
each 3 kHz wide, into the long-distance channel. The circuit that does this is known as a
multiplexer. It accepts the input from each individual end user, and generates a signal on a
different frequency for each of the inputs. This results in a high-bandwidth, complex
signal containing data from all the end users. At the other end of the long-distance cable,
the individual signals are separated out by means of a circuit called a de-multiplexer, and
routed to the proper end users. A two-way communications circuit requires a
multiplexer/de-multiplexer at each end of the long-distance, high-bandwidth cable.

Fig. Principle of FDM & De-multiplexing
The most natural example of frequency-division multiplexing is radio and television

broadcasting, in which multiple radio signals at different frequencies pass through the air
at the same time. Another example is cable television, in which many television channels
are carried simultaneously on a single cable. FDM is also used by telephone systems to
transmit multiple telephone calls through high capacity trunklines, communications
satellites to transmit multiple channels of data on uplink and downlink radio beams, and
broadband DSL modems to transmit large amounts of computer data through twisted pair
telephone lines, among many other uses.
An analogous technique called wavelength division multiplexing is used in fiber-optic

communication, in which multiple channels of data are transmitted over a single optical
fiber using different wavelengths (frequencies) of light.
FDM can also be used to combine signals before final modulation onto a carrier wave.
In this case the carrier signals are referred to as subcarriers: an example is stereo FM
transmission, where a 38 kHz subcarrier is used to separate the left-right difference signal
from the central left-right sum channel, prior to the frequency modulation of the composite
signal. An analog NTSC television channel is divided into subcarrier frequencies for
video, color, and audio. DSL uses different frequencies for voice and for upstream and
downstream data transmission on the same conductors, which is also an example of
frequency duplex.
Where frequency-division multiplexing is used as to allow multiple users to share a

physical communications channel, it is called frequency-division multiple access

One of the examples of FDM system is as shown in below figure
Advantages of FDM:
1. A large number of signals (channels) can be transmitted simultaneously.
2. FDM does not need synchronization between its transmitter and receiver for proper
operation.
3. Demodulation of FDM is easy.
4. Due to slow narrow band fading only a single channel gets affected.
Disadvantages:
The communication channel must have a very large bandwidth.
2. Intermodulation distortion takes place.
3. Large number of modulators and filters are required.
4. FDM suffers from the problem of crosstalk.
5. All the FDM channels get affected due to wideband fading.
Applications of FDM
1. FDM is used for FM & AM radio broadcasting. Each AM and FM radio station uses a
different carrier frequency. In AM broadcasting, these frequencies use a special band from

530 to 1700 KHz. All these signals/frequencies are multiplexed and are transmitted in air.
A receiver receives all these signals but tunes only one which is required. Similarly FM
broadcasting uses a bandwidth of 88 to 108 MHz
2. FDM is used in television broadcasting.
3. First generation cellular telephone also uses FDM.
Other examples
FDM can also be used to combine signals before final modulation onto a carrier
wave. In this case the carrier signals are referred to as subcarriers: an example is stereo
FM transmission, where a 38 kHz subcarrier is used to separate the left-right difference
signal from the central left-right sum channel, prior to the frequency modulation of the
composite signal. An analog NTSC television channel is divided into subcarrier
frequencies for video, color, and audio. DSL uses different frequencies for voice and for
upstream and downstream data transmission on the same conductors, which is also an
example of frequency duplex.
Where frequency-division multiplexing is used as to allow multiple users to share a

physical communications channel, it is called frequency-division multiple access (FDMA).
[1]
FDMA is the traditional way of separating radio signals from different transmitters.
In the 1860s and 70s, several inventors attempted FDM under the names of
acoustic telegraphy and harmonic telegraphy. Practical FDM was only achieved in the
electronic age. Meanwhile, their efforts led to an elementary understanding of
electroacoustic technology, resulting in the invention of the telephone.

Blok Diagram:
Procedure:
1. Set the modulating frequency of ch 1 with the help of potentiometer to 2 KHz
and ch 2 to 4 KHz.
2. Observe the carrier frequency 100 KHz and 200 KHz on the oscilloscope.
3. Connect the ch 1 output to left input of modulator ch 1.
4. Repeat step 3 for ch 2 also.
5. Connect carrier generator outputs (100 KHz and 200KHZ) to CH 1 and CH 2
respectively.
6. Observe the modulator output on oscilloscope.
7. Connect the modulator output of ch 1 and ch 2 to adder circuit.
8. Connect the adder output to demodulator inputs in both the sections.
9. Connect the respective carrier frequency to demodulator second input.
10. Connect the output of demodulator of ch 1 and ch 2 to LPF 1 and LPF 2.
11. Observe the output of low pass filter on the scope and compare it with the modulating
signal.
Calculations:
Message signal (AF1)
Message signal (AF2)


Carrier signal (RF1)
Carrier signal (RF2)
DSB-SC 1
DSB-SC 2
De-multiplexed signal 1
De-multiplexed signal 2

WAVE FORMS:

Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work :

1. What is FDM?
2. In what situation multiplexing is used?
3. What is the difference between Frequency Division Multiplexing and Time
Division Multiplexing?
4. What are the advantages and disadvantages of FDM.
Post Lab Work:

1. What are the applications of FDM.
2. Why guard bands are used in FDM?
3. Will multiplexing create additional harmonics in the system?

9. VERIFICATION OF SAMPLING THEOREM
Objective:
1. To study different types of signal sampling techniques.
a) Sample and Hold b) Flat top sampling.
2.To study the effect of different sampling frequencies on the reconstructed signal.
3.To study the effect of 2nd Order and 4th Order Low Pass Butterworth Filters on the
reconstruction of the signal.
4. To verify the Sampling theorem using simulation.
Apparatus Required

Theory
The sampling theorem is a fundamental bridge between continuous-time signals
(often called "analog signals") and discrete-time signals (often called "digital signals"). It
establishes a sufficient condition for a sample rate that permits a discrete sequence of
samples to capture all the information from a continuous-time signal of finite bandwidth.
The sampling theorem introduces the concept of a sample rate that is sufficient for
perfect fidelity for the class of functions that are band-limited to a given bandwidth, such
that no actual information is lost in the sampling process. It expresses the sufficient sample
rate in terms of the bandwidth for the class of functions. The theorem also leads to a
formula for perfectly reconstructing the original continuous-time function from the
samples.

The symbol T = 1/fs is customarily used to represent the interval between samples and is
called the sample period or sampling interval.
Statement: A continuous time signal can be represented in its samples and can be
recovered back when sampling frequency fs is greater than or equal to the twice the highest
frequency component of message signal. i. e.
fs ≥ 2fm.
Aliasing Effect
When the bandlimit is too high (or there is no bandlimit), the reconstruction exhibits
imperfections known as aliasing The overlapped region in case of under sampling represents
aliasing effect, which can be removed by
 considering fs >2fm
 By using anti aliasing filters.

Application to multivariable signals and images
The sampling theorem is usually formulated for functions of a single variable.

Consequently, the theorem is directly applicable to time-dependent signals and is normally
formulated in that context. However, the sampling theorem can be extended in a
straightforward way to functions of arbitrarily many variables. Grayscale images, for
example, are often represented as two-dimensional arrays (or matrices) of real numbers
representing the relative intensities of pixels (picture elements) located at the intersections
of row and column sample locations. As a result, images require two independent
variables, or indices, to specify each pixel uniquely—one for the row, and one for the
column.
Color images typically consist of a composite of three separate grayscale images, one to
represent each of the three primary colors—red, green, and blue, or RGB for short. Other
colorspaces using 3-vectors for colors include HSV, CIELAB, XYZ, etc. Some
colorspaces such as cyan, magenta, yellow, and black (CMYK) may represent color by
four dimensions. All of these are treated as vector-valued functions over a two-
dimensional sampled domain.
Similar to one-dimensional discrete-time signals, images can also suffer from aliasing if
the sampling resolution, or pixel density, is inadequate. For example, a digital photograph
of a striped shirt with high frequencies (in other words, the distance between the stripes is
small), can cause aliasing of the shirt when it is sampled by the camera's image sensor.
The aliasing appears as a moiré pattern. The "solution" to higher sampling in the spatial
domain for this case would be to move closer to the shirt, use a higher resolution sensor, or
to optically blur the image before acquiring it with the sensor.
Another example is shown to the right in the brick patterns. The top image shows the
effects when the sampling theorem's condition is not satisfied. When software rescales an
image , in effect, runs the image through a low-pass filter first and then downsamples the
image to result in a smaller image that does not exhibit the moiré pattern. The top image is
what happens when the image is downsampled without low-pass filtering: aliasing results.

The application of the sampling theorem to images should be made with care. For
example, the sampling process in any standard image sensor (CCD or CMOS camera) is
relatively far from the ideal sampling which would measure the image intensity at a single
point. Instead these devices have a relatively large sensor area at each sample point in
order to obtain sufficient amount of light. In other words, any detector has a finite-width
point spread function. The analog optical image intensity function which is sampled by the
sensor device is not in general bandlimited, and the non-ideal sampling is itself a useful
type of low-pass filter, though not always sufficient to remove enough high frequencies to
sufficiently reduce aliasing. When the area of the sampling spot (the size of the pixel
sensor) is not large enough to provide sufficient spatial anti-aliasing, a separate anti-
aliasing filter (optical low-pass filter) is typically included in a camera system to further
blur the optical image. Despite images having these problems in relation to the sampling
theorem, the theorem can be used to describe the basics of down and up sampling of
images.
Audio sampling
Digital audio uses pulse-code modulation and digital signals for sound reproduction. This
includes analog-to-digital conversion (ADC), digital-to-analog conversion (DAC), storage,
and transmission. In effect, the system commonly referred to as digital is in fact a discrete-
time, discrete-level analog of a previous electrical analog. While modern systems can be
quite subtle in their methods, the primary usefulness of a digital system is the ability to
store, retrieve and transmit signals without any loss of quality
Video sampling:
In digital video, the temporal sampling rate is defined the frame rate – or rather the field
rate – rather than the notional pixel clock. The image sampling frequency is the repetition
rate of the sensor integration period. Since the integration period may be significantly
shorter than the time between repetitions, the sampling frequency can be different from the
inverse of the sample time
3D sampling:
The process of volume rendering samples a 3D grid of voxels to produce 3D renderings of

sliced (tomographic) data. The 3D grid is assumed to represent a continuous region of 3D
space. Volume rendering is common in medical imaging, X-ray computed tomography

(CT/CAT), Magnetic resonance imaging (MRI), Positron Emission Tomography (PET) are
some examples. It is also used for Seismic tomography and other applications.
Block Diagram:
Procedure:
1. Connect the 2KHz, 5U P-P signal generated onboard to the ANALOG INPUT, by
means of the patch chords provided.
2. Connect the sampling frequency of 4KHz in INTERNAL mode, by means of the a
shorting pins provided.
3. Observe the output of sampling amplifier and output of sample hold amplifier.
4. Connect the sample hold output to the input of record order and fourth order low
pass filer to be the reconstructed signals.
5. Vary the switch positions of DIP switch to observe the duty cycle effect on the
sampling frequency.
6. Duty cycle table is shown below.
7. Vary the duty cycle of the sampling frequency signal from 10% to 90% in steps of
10% each.
8. Observe the effect of duty cycle on sampling amplifier at sample output.

9. Observe the reconstructed output at the second order and fourth order low pass
filter for different laiplin frequency of 32KHz, 16KHz, 8 KHz, 4 KHz, 2 KHz.
Calculations:
Sampling pulse
Analog signal
Flat-top sampled signal
Sample and hold signal
2nd order reconstructed output
4th order reconstructed output

Expected Graph:

Flat-top sampled signal
Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20

Pre lab work :

1. Define sampling theorem? What is the need for sampling?
2. Define Nyquist rate and Nyquist interval in sampling theorem?
3. What are different types of sampling techniques?
4. What was the effect on sampled signal if fs < 2 fm?
5. What is aliasing effect in sampling? How to avoid it?
6. Why do we use pre-filtering in sampling?
7. What are the types of filters used in reconstruction?
8. What is the need for converting a continuous signal into a discrete signal.
9. As the number of samples increases, the reconstruction of original signal becomes
--------
10. How sampling is different from PAM?
11. What do you mean by reconstruction of sampling theorem?
12. Draw the amplitude spectrum of sampled signal if fs < 2 fm,fs =2 f,fs > 2 fm.
13. What is the difference between discrete and a digital signal?
Post Lab work:
1. Differentiate second order, fourth order and sixth order low pass filters in
reconstruction process.
2. Explain about zero-order hold circuit.
3. How to convert an analog signal into a digital signal?

10. PULSE AMPLITUDE MODULATION & DEMODULATION
Objective:
1. To study the operation of Pulse Amplitude modulation & Demodulation and plot the waveforms.
2.To simulate Pulse amplitude modulation and demodulation
Apparatus Required

Theory:
Pulse amplitude modulation (PAM) is the transmission of data by varying the
amplitudes (voltage or power levels) of the individual pulses in a regularly timed sequence
of electrical or electromagnetic pulses. The number of possible pulse amplitudes can be
infinite (in the case of analog PAM), but it is usually some power of two so that the
resulting output signal can be digital . For example, in 4-level PAM there are 2^2 possible
discrete pulse amplitudes; in 8-level PAM there are 2^3 possible discrete pulse amplitudes;
and in 16-level PAM there are 2^4 possible discrete pulse amplitudes.
In some PAM systems, the amplitude of each pulse is directly proportional to the
instantaneous modulating-signal amplitude at the time the pulse occurs. In other PAM
systems, the amplitude of each pulse is inversely proportional to the instantaneous
modulating-signal amplitude at the time the pulse occurs. In still other systems, the

intensity of each pulse depends on some characteristic of the modulating signal other than
its strength, such as its instantaneous frequency or phase .
There are two types of pulse amplitude modulation:
1. Single polarity PAM: In this a suitable fixed DC bias is added to the signal to
ensure that all the pulses are positive.
2. Double polarity PAM: In this the pulses are both positive and negative.
Pulse-amplitude modulation is widely used in modulating signal transmission of digital

data, with non-baseband applications having been largely replaced by pulse-code
modulation, and, more recently, by pulse-position modulation.
In particular, all telephone modems faster than 300 bit/s use quadrature amplitude
modulation (QAM). (QAM uses a two-dimensional constellation).
The number of possible pulse amplitudes in analog PAM is theoretically infinite. Digital
PAM reduces the number of pulse amplitudes to some power of two. For example, in 4-
level PAM there are possible discrete pulse amplitudes; in 8-level PAM there are
possible discrete pulse amplitudes; and in 16-level PAM there are possible discrete pulse
amplitudes.
In PAM the amplitude of the message or modulating signal is mapped to a series of pulses
with two possible variant
1) Flat Top PAM:- The amplitude of each pulse is directly proportional to instantaneous
modulating signal amplitude at the time of pulse occurrence and then keeps the
amplitude of the pulse for the rest of the half cycle.
2) 2) Natural PAM:- The amplitude of each pulse is directly proportional to the
instantaneous modulating signal amplitude at the time of pulse occurrence and then
follows the amplitude of the modulating signal for the rest of the half cycle.
Advantages of PAM:
← PAM allows data to be transmitted more effectively, efficiently and quickly

using conventional copper wires in greater volume.
← The frequency modulations available are infinite; hence PAM formulas can
be developed continually to allow increased data throughput over existing
networks.
← PAM is also the simplest form of modulation.
Applications:

The PAM modulation technique is widely used in high speed digital communications like
telephone modems. They are used to drive LED lights more efficiently than using PWM
method. Unlike the PPM the transmitter and receiver synchronization is not required for
the PAM.
They are also used in Ethernet. For example, 100BASE-T2 – operating at 100Mb/s –
Ethernet medium is using 5 level PAM modulations running at 25 mega pulses/sec over
four wires. Later developments include the 100BASE-T medium which raised the bar to 4
wire pairs, running each at 125 mega pulses/sec in order to achieve 1000 Mbps data
transfer rates, but still with the same PAM5 for each pair.
More recently, PAM12 and PAM8 have gained consideration in the newly proposed IEEE
802.3an standard for 10GBase-T — ten gigabyte Ethernet over copper wire.
Block Diagram:
Circuit Diagrams:
PAM Modulator Circuit Diagram:

PAM De-modulator Circuit Diagram:
1. As the circuitry is already wired you just have to trace the circuit according to the
circuit diagram given above.
2. Connect trainer to mains and switch on the power.
3. Observe the output of AF generator and pulse generator using CRO and note that
AF signal is approximately 3V P-P of 400Hz frequency and pulse generator output
is pulse train of 10V P-P with frequency between 1 KHz and 6 KHz.
Modulator:
4. Connect pulse output and AF output to the respective inputs of modulator circuit.
5. Connect one of the inputs of oscilloscope to the modulator output and another to
AF signal.
6. Initially set the amplitude of the AF generator to minimum level and sampling
frequency to 1 KHz (by adjusting the preset provided in pulse generator block).
Note down the output of modulator, by varying amplitude of modulating signal

observe the modulator output so that you can notice the amplitude of the sampling
pulses is varying in accordance with the modulating signal.
Demodulator:
7. Connect PAM wave input to demodulator input and set sampling pulse frequency
to maximum (6 KHz).
8. Observe demodulated signals at output of demodulator; compare it with the
original AF signal.
(Note: Only shape, amplitude will be attenuated).
9. You can observe the amplified signal by applying demodulated signal to amplifier.
10. Find the detected signal is same as the AF signal applied. Thus no information is
lost in the process of modulation.
Calculations:
Pulse Train
Modulating signal
PAM signal
Demodulated signal

Waveforms:
Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03

5 Viva 02
Total 20
Pre lab work

1. What is PAM?
2. Write the transmission bandwidth of PAM signal?
3. What is the purpose of Equalizer in PAM demodulator?
4. Draw Flat‐ Top and natural sampling of PAM With respect to input signal.
5. Mention the applications of PAM signal.
6. Why PAM is not preferable in digital transmission?
7. Mention the advantages and disadvantages of PAM signal.
8. What is meant by aperture effect?
Post Lab Work:

1. Consider An analog signal x(t)=30cos(2000πt)+5sin(6000πt)+10cos(12000πt).Find
The Nyquist Rate and Nyquist Interval of this signal.
2. For The analog signal x(t)=3cos(100πt),the signal is sampled at the rate of
fs=75Hz,what is the discrete time signal obtained after sampling?

11&12 .PWM & PPM DEMONSTRATOR

Objective:
1.To study the operation of PWM & PPM and observe the waveforms
2. To simulate Pulse width modulation (PWM) and demodulation.
3. To simulate Pulse Position modulation (PPM) and demodulation
Apparatus Required :

Theory:
Pulse Time Modulation is also known as Pulse Width Modulation or Pulse Length
Modulation. In PWM, the samples of the message signal are used to vary the duration of
the individual pulses. Width may be varied by varying the time of occurrence of leading
edge, the trailing edge or both edges of the pulse in accordance with modulating wave. It
is also called Pulse Duration Modulation.
Pulse-width modulation (PWM), or pulse-duration modulation (PDM), is a

modulation technique used to encode a message into a pulsing signal. Although this
modulation technique can be used to encode information for transmission, its main use is
to allow the control of the power supplied to electrical devices, especially to inertial loads
such as motors. In addition, PWM is one of the two principal algorithms used in
photovoltaic solar battery chargers,[1] the other being maximum power point tracking.

The term duty cycle describes the proportion of 'on' time to the regular interval or 'period'
of time; a low duty cycle corresponds to low power, because the power is off for most of
the time. Duty cycle is expressed in percent, 100% being fully on.
The main advantage of PWM is that power loss in the switching devices is very low.
When a switch is off there is practically no current, and when it is on and power is being
transferred to the load, there is almost no voltage drop across the switch. Power loss, being
the product of voltage and current, is thus in both cases close to zero. PWM also works
well with digital controls, which, because of their on/off nature, can easily set the needed
duty cycle.
Applications:
PWM has also been used in certain communication systems where its duty cycle has been
used to convey information over a communications channel. PWM is used to control
servomechanisms.
Telecommunications
In telecommunications, PWM is a form of signal modulation where the widths of the

pulses correspond to specific data values encoded at one end and decoded at the other.
Pulses of various lengths (the information itself) will be sent at regular intervals (the
carrier frequency of the modulation
Power delivery
PWM can be used to control the amount of power delivered to a load without incurring the
losses that would result from linear power delivery by resistive means. Drawbacks to this
technique are that the power drawn by the load is not constant but rather discontinuous
(see Buck converter), and energy delivered to the load is not continuous either.

Voltage regulation
PWM is also used in efficient voltage regulators. By switching voltage to the load with the
appropriate duty cycle, the output will approximate a voltage at the desired level. The
switching noise is usually filtered with an inductor and a capacitor.
One method measures the output voltage. When it is lower than the desired voltage, it
turns on the switch. When the output voltage is above the desired voltage, it turns off the
switch.
Audio effects and amplification
PWM is sometimes used in sound (music) synthesis, in particular subtractive synthesis, as

it gives a sound effect similar to chorus or slightly detuned oscillators played together. (In
fact, PWM is equivalent to the difference of two saw tooth waves with one of them
inverted. The ratio between the high and low level is typically modulated with a low
frequency oscillator
Electrical
SPWM (Sine–triangle pulse width modulation) signals are used in micro-inverter design
(used in solar and wind power applications). These switching signals are fed to the FETs
that are used in the device. The device's efficiency depends on the harmonic content of the
PWM signal.
. In Pulse Position Modulation, both the pulse amplitude and pulse duration are held
constant but the position of the pulse is varied in proportional to the sampled values of the
message signal. Pulse time modulation is a class of signaling techniques that encodes the
sample values of an analog signal on to the time axis of a digital signal and it is analogous
to angle modulation techniques. The two main types of PTM are PWM and PPM. In PPM
the analog sample value determines the position of a narrow pulse relative to the clocking
time. In PPM rise time of pulse decides the channel bandwidth. It has low noise
interference.

APPLICATIONS:
For RF communications:
Narrowband RF (radio frequency) channels with low power and long wavelengths
(i.e., low frequency) are affected primarily by flat fading, and PPM is better suited than M-
FSK to be used in these scenarios. One common application with these channel
characteristics, first used in the early 1960s with top-end HF (as low as 27 MHz)
frequencies into the low-end VHF band frequencies (30 MHz to 75 MHz for RC use
depending on location), is the radio control of model aircraft, boats and cars, originally
known as "digital proportional" radio control. PPM is employed in these systems, with the
position of each pulse representing the angular position of an analogue control on the
transmitter, or possible states of a binary switch. The number of pulses per frame gives the
number of controllable channels available. The advantage of using PPM for this type of
application is that the electronics required to decode the signal are extremely simple,
which leads to small, light-weight receiver/decoder units. (Model aircraft require parts that
are as lightweight as possible).Servos made for model radio control include some of the
electronics required to convert the pulse to the motor position – the receiver is required to
first extract the information from the received radio signal through its intermediate
frequency section, then de-multiplex the separate channels from the serial stream, and feed
the control pulses to each servo.
PPM encoding for radio control
A complete PPM frame is about 22.5 ms (can vary between manufacturer), and
signal low state is always 0.3 ms. It begins with a start frame (high state for more than
2 ms). Each channel (up to 8) is encoded by the time of the high state (PPM high state +
0.3 × (PPM low state) = servo PWM pulse width).
More sophisticated radio control systems are now often based on pulse-code modulation,
which is more complex but offers greater flexibility and reliability. The advent of 2.4 GHz
band FHSS radio-control systems in the early 21st century changed this still further.
Pulse-position modulation is also used for communication to the ISO/IEC 15693

contactless smart card, as well as the HF implementation of the Electronic Product Code
(EPC) Class 1 protocol for RFID tags.
Advantages:
 has the advantage over pulse amplitude modulation (PAM) in that it has a higher
noise immunity
 requiring constant transmitter power since the pulses are of constant amplitude and
duration
 signal and noise separation is very easy.
Disadvantages:
 depending on transmitter-receiver synchronization
 highly sensitive to multipath way interference
Block diagram:

Circuit Diagrams:
PWM Modulator:
PWM – PPM (Monostable):
PPM – PWM (JK Flip-Flop):

PWM-Demodulator:
Observation of PWM and PPM with DC input voltage:
1. Study circuit operation thoroughly.
2. Switch on the trainer and measure the output voltages of the regulated power
supply i.e. +5V and –5V.
3. Observe the output of the AF generator using CRO, note that the output is 5V pp
400 Hz frequency.
4. Observe the output of the control signal generator i.e. ramp and reference pulse
using CRO.
5. Connect ramp signal to the ramp input of the PWM modulator and dc source
output to AF input.
6. Connect one DMM to the dc source output and CH 1 input of the scope to the
PWM modulator output.
7. Measures the output pulse width at different input voltages starting from zero and
note down the readings. (By this we can observe the output pulse width is varying
in accordance with the input voltage as per theory of PWM, the amplitude and
position are fixed only width is varying)
8. Now connect output of the PWM modulator to mono-stable multivibrator input

and CH 2 input of the oscilloscope to the mono-stable output i.e. PPM output (Set
scope in dual mode and trigger source in CH 1).
9. Observe PWM and PPM waveforms for different values of the input voltage
starting from zero (By this we can notice the output of mono-stable is PPM i.e. the
pulse width is fixed and amplitude is constant only position is varying).
Observation of PWM and PPM with AC input signal:

10. Now connect AF signal instead of dc voltage to the modulator and observe output
waveform (condition: scope is in dual mode, CH 1 is connected to AF signal and
CH 2 is connected to PWM output, trigger source in CH 1,if you are using storage
oscilloscope after setting AF input voltage observe output in stop mode). Similarly
PPM waveform.
PWM Demodulation:
11. Remove connection from mono-stable input and connect it to PWM demodulator
input.
12. Connect CH 1 to input AF signal and CH 2 to demodulator output and observe the
output, compare it with original AF signal
PPM demodulation:
13. Connect PPM and reference pulse signals to respective inputs of PPM – PWM
converter circuit and output of the same circuit to PWM demodulator. (Scope
should be set in dual mode, CH 1 is connected to input AF Signal, CH 2 to
demodulator output and trigger source to CH 1). Observe the output signal and
compare it with input AF signal.
Note: The main problem in this experiment will be in triggering the oscilloscope to
Observe the waveforms, especially PPM
Calculations:
Control signal
Modulating signal


PWM signal
PPM signal
Demodulated signal

Waveforms:

Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work :

1. What is mean by pulse time modulation?

2. What are the applications of PWM?
3. What are the advantages and disadvantages of PWM?
4. What are the methods to generate PWM?
5. What are the applications of PPM?
6. What are the advantages and disadvantages of PPM?
7. What is the relation between PWM and PPM?
8. What are the analog analogies of PAM, PPM & PWM?
9. Mention different types of Pulse Width Modulation techniques.
Post Lab Work:

1. Why Schmitt trigger circuit is used in PWM signal reception.
2. How to generate PPM from PWM?
3. Why is PWM used rarely in any sort of communication or broadcasting?
13. FREQENCY SYNTHESIZER

Objective:

1. To study the operation of Frequency Synthesizer.

2. To simulate Frequency synthesis.
Apparatus Required

Theory:
The frequency divider is inserted between the VCO and the phase comparator of
PLL. Since the output of the divider is locked to the input frequency fin, the VCO is
actually running at a multiple of the input frequency. The desired amount of multiplication
can be obtained by selecting a proper divide– by – N network ,where N is an integer. To
obtain the output frequency fOUT=5fIN, a divide – by – N = 5 network is needed. One must
determine the input frequency range and then adjust the free running fOUT of the VCO by
means of R1 (20kΩpot) and C1 (10µF) so that the output frequency of the divider is
midway within the predetermined input frequency range. The output of the VCO now
should be 5fIN. The output of the VCO now should be adjusted from 1.5 KHz to 15 KHz
by varying potentiometer R1 .this means that the input frequency fin range has to be with
in 300Hz to 3KHz. In addition, the input wave form may be applied to inputs pin2 or pin3.
Input – output waveforms forms for fOUT= 5fIN. A small capacitor typically 1000pf is
connected between pin7 and pin8 to eliminate possible oscillations. Also, capacitor
C2should be large enough to stabilize the VCO frequency.
A frequency synthesizer is an electronic circuit for generating any of a range of

frequencies from a single fixed time base or oscillator. They are found in many modern
devices, including radio receivers, televisions, mobile telephones, radiotelephones, walkie-

talkies, CB radios, cable television converter boxes satellite receivers, and GPS systems. A
frequency synthesizer uses the techniques of frequency multiplication, frequency division,
direct digital synthesis, and frequency mixing to generate new frequencies which have the
same stability and accuracy as the master oscillator.
PLL frequency synthesizers are widely used in all forms of radio communications
equipment today.
These frequency synthesizers are found in a variety of items from cellular phones to all
forms of wireless products and domestic radios and televisions to professional radio
frequency equipment like signal generators and spectrum analyzers as well as professional
radio equipment and much more.
PLL frequency synthesizers offer very many advantages over the use of other forms of
oscillator.
Frequency synthesizers not only offer high levels of stability and accuracy (determined by
the reference which is normally a crystal oscillator); they are also easy to control from
digital circuitry such as microprocessors. This enables facilities such as keypad frequency
entry, channel memories and more to be implemented - all of which are expected as basic
functionality in today's equipment.
In view of all their advantages, PLL frequency synthesizers are usually the preferred form
of radio frequency oscillator for most applications. Accordingly synthesizers are included
in many radio chip-sets from cellular phones to radio and televisions.
Frequency synthesizer types / categories
There are several different types of categories of synthesizer. Each of them obviously has
its own advantages and disadvantages. There are often choices that can be made about
which type to choose
 Direct: The direct forms of frequency synthesizer, are as the name suggests
implemented by creating a waveform directly without any form of frequency
transforming element. Direct techniques including forms of oscillator and mixer
are used.

Direct Analogue Frequency Synthesis: This form of frequency synthesizer was

sometimes called a mix-filter-divide architecture. The direct analogue frequency
synthesizer gained this name because it accurately defines one of the more popular
architectures for this form of synthesis. The direct analogue frequency synthesizer had
several drawbacks: it required a considerable amount of critical circuitry which today does
not lend itself to integration; the successive mix processes introduced significant numbers
of spurious signals; the spurious signals required considerable levels of filtering, again
adding to the cost. As a result, this type of frequency synthesis was only used as a last
resort before the widespread availability of RF ICs and the possibility of utilizing other
forms of frequency synthesis.
Direct Digital Frequency Synthesis: Direct digital synthesizers, DDS are widely used
now. They create the signal by having a stored version of the waveform required, and then
advancing the phase in fixed increments. The phase advance increments determine the
signal frequency that is generated.
Indirect: Indirect frequency synthesis is based around phase locked loop technology.
Here the output signal is generated indirectly. In other words the final signal is generated
by an oscillator that is controlled by other signals. In this way the signals used in creating
the output are indirectly replicated by the output oscillator, thereby giving the name to this
technique.
Indirect Analogue Frequency Synthesis: Indirect analogue frequency synthesis uses

phase locked loop technology with a mixer placed between the voltage controlled
oscillator and phase detector. This enables and offset frequency to be introduced into the
loop.
Indirect Digital Frequency Synthesis: The indirect digital frequency synthesis

techniques introduce a digital divide into the phase locked loop between the voltage
controlled oscillator and the phase detector. The VCO runs at a frequency equal to the
phase comparison frequency times the division ratio. By altering he division ratio, it is
possible to alter the frequency of the output signal. Typically the comparison frequency is
equal to the channel spacing required. This could be 100 of 50 kHz for an FM tuner, 25 or

12.5 kHz for professional mobile communications systems, etc. It could be much smaller
for general radio applications.
Today's frequency synthesizers may use a variety of techniques and technologies.

Typically the direct analogue approach is not used these days, but the other three may be
used in various applications, the choices being dependent upon the requirements of the
application.
Block Diagram of Frequency synthesizer:
Circuit Diagram:
Wave Forms:

1. Switch on the trainer and verify the output of the regulated power supply i.e. +5v.
These supplies are internally connected to the circuit so no extra connections are
required.
2. Observe output of the square wave generator-using oscilloscope and measure the
range with the help of frequency counter; frequency range should be around 1 KHz
to 10 KHz.
3. Calculate the free running frequency range of the circuit (VCO output between 4 th
pin and ground). For different values of timing resistor Rt (to measure Rt switch
off the trainer and measure Rt value using digital multimeter between given test
points). And record the frequency values in tabular 1.
Fout= 0.3/RtCt where Rt is the timing resistor and Ct is timing capacitor=0.01uf.
Fin KHz Fout=Nfin in KHz Divide by 10, 2
4. Connect 4th pin of LM 565(Fout) to the driver stage and 5th pin (phase comparator)
connected to 11th pin of 7490. Output can be taken at the 11 th pin of the 7490. It
should be divided by 10, 2 times of the fout.

Precautions
RESULT:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work
1. What is frequency synthesizer?

2. How does a frequency synthesizer work?
3. What is frequency divider?
4. What is frequency multiplier?

5. What is 74LS90?
6. What is 74LS93?
7. Mention different types of Frequency synthesizers.
8. How to achieve fout = 2 fin ?
9. What is mean by harmonic frequency?
Post Lab Work:
1. What is the effect of C1 on the output frequency?

2. List out the applications of frequency synthesizer.
3. What is the difference between heterodyning and synthesizer?

14. AGC CHARACTERISTICS
Objective: To observe the AGC characteristics to check the output is constant for the
variations of input voltage.
Apparatus Required

Theory:
Automatic gain control (AGC), also called automatic voltage gain, is a closed-
loop feedback regulating circuit, the objective of which is to provide a controlled signal
amplitude at its output, despite variation of the amplitude in the input signal. The average
or peak output signal level is used to dynamically adjust the input-to-output gain to a
suitable value, enabling the circuit to work satisfactorily with a greater range of input
signal levels. It is used in most radio receivers to equalise the average volume (loudness)
of different radio stations due to differences in received signal strength, as well as
variations in a single station's radio signal due to fading. Without AGC the sound emitted
from an AM radio receiver would vary to an extreme extent from a weak to a strong
signal; the AGC effectively reduces the volume if the signal is strong and raises it when it
is weaker

The signal to be gain controlled (the detector output in a radio) goes to a diode &
capacitor, which produce a peak-following DC voltage. This is fed to the RF gain blocks
to alter their bias, thus altering their gain. Traditionally all the gain-controlled stages came
before the signal detection, but it is also possible to improve gain control by adding a gain-
controlled stage after signal detection.
Types of AGC:
1.Simple AGC
2.Delayed AGC
3.Forward AGc
Applications:
1.AM radio receivers
AGC is afrom linearity in AM radio receivers.[3] Without AGC, an AM radio would
have a linear relationship between the signal amplitude and the sound waveform – the
sound amplitude, which correlates with loudness, is proportional to the radio signal
amplitude, because the information content of the signal is carried by the changes of
amplitude of the carrier wave. If the circuit were not fairly linear, the modulated signal
could not be recovered with reasonable fidelity. However, the strength of the signal
received will vary widely, depending on the power and distance of the transmitter, and
signal path attenuation. The AGC circuit keeps the receiver's output level from fluctuating
too much by detecting the overall strength of the signal and automatically adjusting the
gain of the receiver to maintain the output level within an acceptable range. For a very
weak signal, the AGC operates the receiver at maximum gain; as the signal increases, the
AGC reduces the gain.
2.Radar
A related application of AGC is in radar systems, as a method of overcoming
unwanted clutter echoes. This method relies on the fact that clutter returns far outnumber
echoes from targets of interest. The receiver's gain is automatically adjusted to maintain a
constant level of overall visible clutter. While this does not help detect targets masked by
stronger surrounding clutter, it does help to distinguish strong target sources. In the past,
radar AGC was electronically controlled and affected the gain of the entire radar receiver.
As radars evolved, AGC became computer-software controlled, and affected the gain with
greater granularity, in specific detection cells. Many radar countermeasures use a radar's

AGC to fool it, by effectively "drowning out" the real signal with the spoof, as the AGC
will regard the weaker, true signal as clutter relative to the strong spoof.
3. Telephone recording
Devices to record both sides of a telephone conversation must record both the
relatively large signal from the local user and the much smaller signal from the remote
user at comparable loudnesses. Some telephone recording devices incorporate automatic
gain control to produce acceptable-quality recordings.
Block Diagram:
Procedure:
1. As the circuit is already wired you just have to trace the circuit according to the circuit
diagram given above.
2. Connect the trainer to the mains and switch on the power supply.
3. Measures the output voltages of the regulated power supply circuit i.e. +12v and -12v,
+6@150mA.
4. Observe outputs of RF and AF signal generator using CRO, note that RF voltage is approximately
50mVpp of 455 KHz frequency and AF voltage is 5Vpp of1 KHz frequency.
5. Now vary the amplitude of AF signal and observe the AM wave at output, note the percentage of
modulation for different values of AF signal.% Modulation= (Emax -Emin) /
(Emax+Emin) × 100
6. Now adjust the modulation index to 30% by varying the amplitudes of RF & AF signals
simultaneously.
7. Connect AM output to the input of AGC and also to the CRO channel -1
8. Connect AGC link to the feedback network through OA79 diode
9. Now connect CRO channel - 2 at output. The detected audio signal of 1 KHz will be observed.

10. Calculate the voltage gain by measuring the amplitude of output signal (Vo) waveform, using Formula
A =Vo/V i.
11. Now vary input level of 455 KHz IF signal and observe detected 1 KHz audio signal
with and Without AGC link. The output will be distorted when AGC link removed i.e.
there is no AGC action.
12. This explains AGC effect in Radio circuit
Observation Table:
SIGNAL TYPE FREQUENCY AMPLITUDE
Modulating signal
Carrier signal
Modulated signal
De modulated Signal
(without AGC)
Demodulated Signal
(with AGC)

Waveforms:
AM modulated wave

Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20
Pre lab work

1.Classify AGC?
2Explain various applications of AGC?

3. What are the functions of IF Amplifier?
4. What are the functions of RF Amplifier?
5. What is the need for AGC?
6. What are the drawbacks of AGC and solution?
Post Lab Work:

1. What is the necessity of AGC.
2.Explain about forward AGC.
3. Explain about delayed AGC.
4..Explain about Simple AGC.

15. PLL AS FM DEMODULATOR

Objective:
1.To Study of the operation of Phase Locked Loop as FM Demodulator.
2. To simulate Phase Locked Loop as FM Demodulator.
Equipment Required:
Apparatus Required

Theory:
PLL has emerged as one of the fundamental building block in electronic technology. it
is used for the frequency multiplication, fm stereo detector , fm demodulator , frequency
shift keying decoders, local oscillator in tv and fm tuner. it consists of a phase detector, a
lpf and a voltage controlled oscillator (vco) connected together in the form of a feedback
system. the vco is a sinusoidal generator whose frequency is determined by a voltage
applied to it from an external source. in effect, any frequency modulator may serve as a
vco.
Applications:
Phase-locked loops are widely used for synchronization purposes; in space

communications for coherent demodulation and threshold extension, bit synchronization,
and symbol synchronization. Phase-locked loops can also be used to demodulate
frequency-modulated signals. In radio transmitters, a PLL is used to synthesize new

frequencies which are a multiple of a reference frequency, with the same stability as the
reference frequency.
Other applications include:
 Demodulation of both FM and AM signals

 Recovery of small signals that otherwise would be lost in noise (lock-in amplifier
to track the reference frequency)
 Recovery of clock timing information from a data stream such as from a disk drive
 Clock multipliers in microprocessors that allow internal processor elements to run
faster than external connections, while maintaining precise timing relationships
 DTMF decoders, modems, and other tone decoders, for remote control and
telecommunications
 DSP of video signals; Phase-locked loops are also used to synchronize phase and
frequency to the input analog video signal so it can be sampled and digitally
processed
 Atomic force microscopy in tapping mode, to detect changes of the cantilever
resonance frequency due to tip–surface interactions
 DC motor drive
Clock recovery
Some data streams, especially high-speed serial data streams (such as the raw stream of
data from the magnetic head of a disk drive), are sent without an accompanying clock. The
receiver generates a clock from an approximate frequency reference, and then phase-aligns
to the transitions in the data stream with a PLL. This process is referred to as clock
recovery. In order for this scheme to work, the data stream must have a transition
frequently enough to correct any drift in the PLL's oscillator. Typically, some sort of line
code, such as 8b/10b encoding, is used to put a hard upper bound on the maximum time
between transitions.
Deskewing
If a clock is sent in parallel with data, that clock can be used to sample the data. Because
the clock must be received and amplified before it can drive the flip-flops which sample

the data, there will be a finite, and process-, temperature-, and voltage-dependent delay
between the detected clock edge and the received data window. This delay limits the
frequency at which data can be sent. One way of eliminating this delay is to include a
deskew PLL on the receive side, so that the clock at each data flip-flop is phase-matched
to the received clock. In that type of application, a special form of a PLL called a delay-
locked loop (DLL) is frequently used.
Clock generation
Many electronic systems include processors of various sorts that operate at hundreds of
megahertz. Typically, the clocks supplied to these processors come from clock generator
PLLs, which multiply a lower-frequency reference clock (usually 50 or 100 MHz) up to
the operating frequency of the processor. The multiplication factor can be quite large in
cases where the operating frequency is multiple gigahertz and the reference crystal is just
tens or hundreds of megahertz.
Spread spectrum
All electronic systems emit some unwanted radio frequency energy. Various regulatory
agencies (such as the FCC in the United States) put limits on the emitted energy and any
interference caused by it. The emitted noise generally appears at sharp spectral peaks
(usually at the operating frequency of the device, and a few harmonics). A system designer
can use a spread-spectrum PLL to reduce interference with high-Q receivers by spreading
the energy over a larger portion of the spectrum. For example, by changing the operating
frequency up and down by a small amount (about 1%), a device running at hundreds of
megahertz can spread its interference evenly over a few megahertz of spectrum, which
drastically reduces the amount of noise seen on broadcast FM radio channels, which have
a bandwidth of several tens of kilohertz.
Clock distribution

Typically, the reference clock enters the chip and drives a phase locked loop (PLL), which
then drives the system's clock distribution. The clock distribution is usually balanced so
that the clock arrives at every endpoint simultaneously. One of those endpoints is the
PLL's feedback input. The function of the PLL is to compare the distributed clock to the
incoming reference clock, and vary the phase and frequency of its output until the
reference and feedback clocks are phase and frequency matched.
PLLs are ubiquitous—they tune clocks in systems several feet across, as well as clocks in
small portions of individual chips. Sometimes the reference clock may not actually be a
pure clock at all, but rather a data stream with enough transitions that the PLL is able to
recover a regular clock from that stream. Sometimes the reference clock is the same
frequency as the clock driven through the clock distribution, other times the distributed
clock may be some rational multiple of the reference.
Jitter and noise reduction
One desirable property of all PLLs is that the reference and feedback clock edges be
brought into very close alignment. The average difference in time between the phases of
the two signals when the PLL has achieved lock is called the static phase offset (also
called the steady-state phase error). The variance between these phases is called
tracking jitter. Ideally, the static phase offset should be zero, and the tracking jitter
should be as low as possible.
Phase noise is another type of jitter observed in PLLs, and is caused by the oscillator itself
and by elements used in the oscillator's frequency control circuit. Some technologies are
known to perform better than others in this regard. The best digital PLLs are constructed
with emitter-coupled logic (ECL) elements, at the expense of high power consumption. To
keep phase noise low in PLL circuits, it is best to avoid saturating logic families such as
transistor-transistor logic (TTL) or CMOS.
Another desirable property of all PLLs is that the phase and frequency of the generated
clock be unaffected by rapid changes in the voltages of the power and ground supply lines,
as well as the voltage of the substrate on which the PLL circuits are fabricated. This is
called substrate and supply noise rejection. The higher the noise rejection, the better.

To further improve the phase noise of the output, an injection locked oscillator can be
employed following the VCO in the PLL.
Frequency synthesis
In digital wireless communication systems (GSM, CDMA etc.), PLLs are used to provide
the local oscillator up-conversion during transmission and down-conversion during
reception. In most cellular handsets this function has been largely integrated into a single
integrated circuit to reduce the cost and size of the handset. However, due to the high
performance required of base station terminals, the transmission and reception circuits are
built with discrete components to achieve the levels of performance required. GSM local
oscillator modules are typically built with a frequency synthesizer integrated circuit and
discrete resonator VCOs.
Block diagram:

Procedure:
Connect +12V and -12V DC power supplies at their indicated positions on AB25 board
from external source or ST2612 Analog Lab.
1. Connect a 2mm patch cord between VCO output and phase comparator input (2).
2. Set VCO free-running frequency to 50 KHz by varying the VCO free running
frequency pot and observing the same signal at test point TP1 on oscilloscope.
3. Apply a frequency modulated output signal from Function Generator (ST4063) with
carrier frequency 50 KHz amplitude 2Vpp and modulating signal amplitude 2Vpp
frequency between 1 KHz to 3.3 KHz. Observe the modulating signal on CH1 of
oscilloscope.
4. Connect output of loop filter (AM output) to the input of LPF on AB25 board using a
2mm patch cord.
5. Observe the demodulated output between Vout (LPF & Amplifier output) and ground. If
demodulated signal is not properly obtained then vary the VCO frequency adjust pot to
obtain a pure modulating signal at output.
6. For obtaining output signal amplitude exactly equal to the modulating signal, vary the
amplifier gain adjustment pot.
Precautions
Result:
1 Pre-lab work 05
2 Post-lab work 05
4 Record 03
5 Viva 02
Total 20

Pre lab work:

What are the different modules of PLL?
2. What are the applications of PLL?
3. What is a PLL?
4. What are the three stages through which PLL operates?
5. Draw the pin diagram of IC 566.
6. What is a VCO?
7. Define the lock range of a PLL.
8. Define the capture range of PLL.
9.What is the need to generate corrective control voltage?
Post Lab Work:

1. Give the expression for free running frequency f0 of a PLL.
2. What is meant by free running frequency of a PLL?
3. Give the formulae for the lock range and capture range of the PLL.
4. What is the effect of R1 and C1 values and Vcc on output signal?

Ac Lab Record Demo Exp

Caricato da

Informazioni sul documento

Titolo originale

Copyright

Formati disponibili

Condividi questo documento

Condividi o incorpora il documento

Opzioni di condivisione

Hai trovato utile questo documento?

Questo contenuto è inappropriato?

Copyright:

Formati disponibili

Ac Lab Record Demo Exp

Caricato da

Copyright:

Formati disponibili

Analog Communications Lab Record

ANALOG COMMUNICATION LAB

Dept. of ECE SR Engineering college

Analog Communication Lab

Note: Minimum 10 experiments should be conducted.

Dept. of ECE SR Engineering college

Instructions to the students

Dept. of ECE SR Engineering college

INTRODUCTION TO ANALOG COMMUNICATION

Communication is the transfer of information from one place to another. Radio

Fig. Basic communication system

Dept. of ECE SR Engineering college

Dept. of ECE SR Engineering college

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

Dept. of ECE SR Engineering college

Concept of Carrier in Communication Systems

Dept. of ECE SR Engineering college

General Simulink Operations

Editing, Running, and Saving Simulink Files

Digital Storage Oscilloscopes

Dept. of ECE SR Engineering college

A digital storage oscilloscope (often abbreviated DSO) is an oscilloscope which stores

Comparison with analog storage

Dept. of ECE SR Engineering college

1. AMPLITUDE MODULATION AND DEMODULATION

1. To understand the operation of amplitude modulation and demodulation and to calculate

Dept. of ECE SR Engineering college

Where Ka is constant called the amplitude sensitivity of the modulator. The

Dept. of ECE SR Engineering college

a. double sideband modulation (DSB-AM)

 Broadcast transmissions: AM is still widely used for broadcasting on the long,

Amplitude modulation advantages & disadvantages

Dept. of ECE SR Engineering college

Dept. of ECE SR Engineering college

% modulation = ((Vmax - Vmin) / (Vmax + Vmin) )x 100

Dept. of ECE SR Engineering college

10. Observe demodulated signal at output of demodulator at approximately 50%

Calculations & Observations

Modulating Signal Generator

Amplitude = Time Period = Frequency =

Carrier Signal Generator:

Amplitude = Time Period = Frequency =

Amplitude = Time Period = Frequency =

Dept. of ECE SR Engineering college

Model Wave Forms

1. Avoid loose connections.

Dept. of ECE SR Engineering college

S. No. Marks Split up Maximum marks Marks obtained

Pre lab work :

1. Define Amplitude modulation.

Post Lab Work:

1. A 5O0 W carrier is modulated to a depth of 60 percent Calculate the total power in

2. DSB – SC MODULATOR AND DETECTOR

Dept. of ECE SR Engineering college

□ AM Trainer kit □DSB-SC TRAINER KIT □SSB-SC Trainer kit

Double-sideband suppressed-carrier transmission (DSB-SC) is transmission in which

DSB-SC transmission is a special case of double-sideband reduced carrier transmission. It is

Generation: DSB-SC is generated by a mixer. This consists of a message signal multiplied by

Dept. of ECE SR Engineering college

Dept. of ECE SR Engineering college

To see this effect, take the following conditions:

 Message signal to be transmitted:

 Modulation (carrier) signal: