BITT POLYTECHNIC, RANCHI

DEPARTMENT OF ELECTRONICS &

COMMUNICATION ENGINEERING
Question Set-2 Communication Systems
Date:-31/03/2020

A. Objectives Questions:

Choose the correct answer:

1. In Pulse code modulation system,
2. In modulation carrier is :
3. FM transmitting and receiving equipment as compared to AM equipment is:
4. Which one of the following is analog?
5. Which one of the following is a digital modulation technique?
6. SSB can be generated by:
7. Under ordinary circumstances, impulse noise can be reduced in:
8. PAM stands for:

B. Short Answer Types Questions:

1. What is modulation? Explain the benefits of modulation.

2. With the help of neat waveforms explain natural PAM sampling and flat top
sampling.
3. Draw the block diagram of communication system and explain the function of each
block.

C. Long Answer Types Questions:

1) Define noise. Explain detailed classification of noise. What are the sources of noise?
2) Explain the term bandwidth and information capacity.
3) Compare the following digital modulation technique on the basis of bandwidth
Requirement and S/N ratio: 1) FSK and 2) PSK
4) What do you mean by bit rate and baud rate?
5) What are the advantages of Single Sideband transmission?
6) Draw the block diagram of FM receiver and explain each block.
7) Write short notes on any two of the following:

a) PWM b) PLL demodulator c) DPSK d) Modulation of DSB-SC

Solutions:
A. Objectives Questions:
1) Large bandwidth is required.
2) Voltage for which frequency, phase or amplitude is varied.
3) Costly
4) PWM
5) all of these ( PSK, DM, PCM)
6) all of these ( filter method, phase cancellation method, good attenuation
characteristics)
7) FM only
8) Pulse Amplitude Modulation

B. Short Answer Types Questions

1) Modulation and Demodulation

The transmission of information and reception of meaningful information can

be successfully achieved with the help of two processes. The two processes

are

a) Modulation b) Demodulation

Modulation : Modulation is the process of combining the low-frequency

Audio waves with a very-high frequency radio waves. The low-frequency wave is

called Modulating Wave. The very-high frequency radio wave which carries

the low frequency audio wave information is called a Carrier Wave. The resultant

wave obtained is called Modulated Carrier.

Demodulation : Demodulation involves recovering the low-frequency audio wave

From the Modulated Carrier Wave. This process is performed at the receiving end.

It is the reverse process of modulation.

Carrier Wave : Carrier wave is a high frequency radio wave produced using

radio-frequency oscillators. The radio frequency ranges from 3kHz to 300GHz.

In radio transmission, Carrier waves in the radio frequency range from hundreds

of kHz to few MHz are preferred.

Need and benefits for Modulation

Audio frequency signals are low-frequency signals.There are disadvantages

in transmitting the unmodulated low frequency signals during communication. They

are
 1) Low frequency signals cannot propagate over long distances. They are
short range signals

2) If there are many transmissions of low frequency signals directly

modulation they interfere and the information at the receiving end will not
be clear.

3) The antenna length required for the transmission of audio-frequency

signals is around 75 m which is practically very large.

Thus a low frequency signal cannot be transmitted effectively and efficiently

without modulation. Hence radio frequency carrier waves are modulated by

low frequency signals and are transmitted to reach longer distances. Even

the antenna size required for the transmission of the radio frequency waves

is of reasonable size.

1) Natural PAM sampling and flat top sampling.

NATURAL SAMPLING:

Natural sampling is performed by multiplying w(t) by a train of pulses:

W_s(t) = w(t) s(t) $$
where ,Natural sampling takes a slice of the waveform and the top of the slice

preserves the shape of the waveform.

PAM using natural sampling is as shown below:

 FLAT TOP SAMPLING:
In flat top sampling, the top of the samples remains constant and equal to the
instantaneous value of the modulating signal at the start of the sampling.
Thus the amplitude of the pulse after sampling is kept constant and the top of
the sampled pulse do not follow the contour of the modulating signal unlike
Natural sampling.
The duration of each sample is τ and the sampling rate is fs=1/Ts
Therefore, Ts=1/fs
Sample and hold circuit is used for the generation of the sampled signal to
attain flat top sampling.
Shannon's Sampling theorem:

 Sampling theorem states that in any pulse modulation system if the sampling rate of the
samples exceeds twice the maximum signal frequency, then this ensures the
reconstruction of the original signal in the receiver with minimum distortion.

 Sampling theorem can be expressed as given below: fs≥2fm Where, fs is the sampling
frequency and fm is the maximum modulating signal frequency

 Sampling is a process of translating continuous analog signal into discrete analog signal,
where the sampled signal is the discrete time representation of the original analog signal.

Aliasing:
 Aliasing is an effect that causes different signals to become indistinguishable from each
other during sampling.
 Signal loss may occur due to aliasing effect.
Now, we may state the sampling theorem for strictly band limited signals of finite energy into
two equivalent parts:
 A band limited signal of finite energy, which only has frequency components less
than W Hertz, is completely described by specifying the values of the signal at instants of
time separated by 1/2W seconds
 A band limited signal of finite energy which only has frequency components less
than W Hertz , may be completely recovered from a knowledge of its sample taken at the
rate of 2W samples per second.
The sampling rate of 2W samples per second for a signal bandwidth of W Hertz is called the
Nyquist rate, its reciprocal 1/2W measured in seconds is called the Nyquist interval.

Fig: a) spectrum of a strictly band limited signal g(t)

b) spectrum of a sampled version of g(t) for $T_s$ = $\frac{1}{2W}$
NOTE : The concept of under sampling and over sampling is explained below.

 When sampling frequency $f_r =2W$ then this type of sampling is called correct
sampling and here there is no aliasing effect seen in this mechanism i.e when $f_s =
2W$.

 When $f_s \lt 2W$ then it is under sampling and there will be aliasing effect induced
here.
  When $f_s \gt 2W$ then it is over sampling and there will no aliasing effect.

The effect of aliasing can be reduced by :

 Pre-alias filter must be used to limit band of frequency of the required signal fm Hz.
 Sampling frequency fs must be selected such that sampling frequency is greater than
twice the maximum modulating signal frequency.

2) The block diagram of communication system and explain the function of each block.

The elements of basic communication system are as follows

 Information or input signal

 Input Transducer

 Transmitter

 Communication channel or medium

 Noise

 Receiver

 Output Transducer
 Fig.1 Block diagram of Communication system
1. Information or input signal
 The communication systems have been developed for communicating useful information
from one place to other
 The information can be in the form of sound signal like speech or music or it can be in
the form of pictures.
2. Input Transducer
 The information in the form of sound, picture or data signals cannot the transmitted as it
is.
 First it has to be converted into a suitable electrical signal.
 The input transducers commonly used in the communication systems are microphones,
TV etc.
3. Transmitter
 The function of the transmitter block is to convert the electrical equivalent of the
information to a suitable form
 It increases the power level of the signal. The power level should be increased in order to
cover a large range. The transmitter consists of the electronics circuits such as amplifier,
mixer, oscillator, and power amplifier.
4. Communication channel or medium
 The communication channel is the medium used for the transmission of electronic signals
from one place to the another.
 The communication medium can be conducting wires, cables, optical fibres or free space.
Depending upon the type of the communication medium, two types of the
communication system will exist
a. Wire communication or line communication
b. Wireless communication or radio communication
5. Noise

 Noise is an unwanted electrical signal which gets added to the transmitted signal when it
is travelling towards receiver.

 Due to noise, the quality of the transmitted information will degrade. One added the noise
cannot be separated out from the information

 Hence noise is ab big problem in the communication systems.

6. Receiver
 The reception is exactly the opposite process of transmission. The received signal is
amplified and demodulated and converted in a suitable form
 The receiver consists of the electronic circuits like mixer, oscillator, detector and
amplifier.
7. Output Transducer
 It consists of the electrical signal at the output of the receiver back to the original form
i.e. sound or TV pictures.
 The typical example of the output transducers are loud speakers, picture tubes etc.

C. Long Answer Types Questions:

1) Noise, its classifications & types

Types Of Noise In Communication

In electrical terms, noise is defined as the unwanted form of energy which tends to interface with
the proper reception and the reproduction of transmitted signals. Electronic Devices unwanted
random addition to the signal are considered as Noise. Their are various types of Noise presents.
The Acoustic Noise is observed when the signals are converted into sound which is generally
know as snow in TV or video Images. In signalling processing or computing noise can be
considered random unwanted data without meaning that is, data that is not being used to transmit
signal, but is simply produced as an unwanted by-product of other activities. “signal to noise
ratio” is sometimes used to refer to the ratio of useful to irrelevant information in an exchange.
Here in this post, Description of Noise of with Classification are explained in detail. If you are
interested you can find out more details on various topics from Note Section.
“In common use, the word noise means any unwanted sound”. Unwanted signal are called
noise.
Noise
Classification of Noise:
There are several way to classify Noise, but conveniently Noise is classified as
1)ExternalNoise

2) Internal Noise
External Noise:
External noise is defined as the type of Noise which is general externally due
to communication system. External Noise are analysed qualitatively. Now, External Noise may
be classified as
a) Atmospheric Noise : Atmospheric Noise is also known as static noise which is the natural
source of disturbance caused by lightning, discharge in thunderstorm and the natural disturbances
occurring in the nature.
b) Industrial Noise : Sources of Industrial noise are auto-mobiles, aircraft, ignition of electric
motors and switching gear. The main cause of Industrial noise is High voltage wires. These
noises is generally produced by the discharge present in the operations.
c) Extraterrestrial Noise : Extraterrestrial Noise exist on the basis of their originating source.
They are subdivided into
i) Solar Noise
ii) Cosmic Noise
Internal Noise:
Internal Noise are the type of Noise which are generated internally or within the Communication
System or in the receiver. They may be treated qualitatively and can also be reduced or
minimized by the proper designing of the system. Internal Noises are classified as
1) Shot Noise : These Noise are generally arises in the active devices due to the random
behaviour of Charge particles or carries. In case of electron tube, shot Noise is produces due to
the random emission of electron form cathodes.
2) Partition Noise : When a circuit is to divide in between two or more paths then the noise
generated is known as Partition noise. The reason for the generation is random fluctuation in the
division.
3) Low- Frequency Noise : They are also known as FLICKER NOISE. These type of noise are
generally observed at a frequency range below few kHz. Power spectral density of these noise
increases with the decrease in frequency. That why the name is given Low- Frequency Noise.
4) High- Frequency Noise : These noises are also known TRANSIT- TIME Noise. They are
observed in the semi-conductor devices when the transit time of a charge carrier while crossing a
junction is compared with the time period of that signal.
5) Thermal Noise : Thermal Noise are random and often referred as White Noise or Johnson
Noise. Thermal noise are generally observed in the resistor or the sensitive resistive components
of a complex impedance due to the random and rapid movement of molecules or atoms or
electrons.

2) The term bandwidth and information capacity.

Information Capacity, Bits, and Bit Rate I α B x t, where I= information capacity (bits per
second) B = bandwidth (hertz) t = transmission time (seconds), it can be seen that information
capacity is a linear function of bandwidth and transmission time and is directly proportional to
both. If either the bandwidth or the transmission time changes, a directly proportional change
occurs in the information capacity. The higher the signal-to-noise ratio, the better the
performance and the higher the information capacity. Mathematically stated, the Shannon
limit_for information capacity ,where I = information capacity (bps) B = bandwidth (hertz) N S
= signal-to-noise power ratio (unitless) For a standard telephone circuit with a signal-to-noise
power ratio of 1000 (30 dB) and a bandwidth of 2.7 kHz, the Shannon limit for information
capacity is I = (3.32)(2700) log10 (1 + 1000) = 26.9 kbps

Shannon's formula is often misunderstood. The results of the preceding example indicate that
26.9 kbps can be propagated through a 2.7-kHz communications channel. This may be true, but it
cannot be done with a binary system. To achieve an information transmission rate of 26.9 kbps
through a 2.7-kHz channel, each symbol transmitted must contain more than one bit.

3) The following digital modulation technique on the basis of bandwidth

Requirement and S/N ratio:

1) FSK and 2) PSK

if the information signal is digital and the amplitude (lV of the carrier is varied proportional to
the information signal, a digitally modulated signal called amplitude shift keying (ASK) is
produced. If the frequency (f) is varied proportional to the information signal, frequency shift
keying (FSK) is produced, and if the phase of the carrier (0) is varied proportional to the
information signal, phase shift keying (PSK) is produced. If both the amplitude and the phase are
varied proportional to the information signal, quadrature amplitude modulation (QAM) results.
ASK, FSK, PSK, and QAM are all forms of digital modulation:

FSK Bit Rate, Baud, and Bandwidth , it can be seen that the time of one bit (tb) is the same as
the time the FSK output is a mark of space frequency (ts). Thus, the bit time equals the time of
an FSK signaling element, and the bit rate equals the baud. The baud for binary FSK can also be
determined by substituting N = 1 : baud = fb / 1 = fb

The minimum bandwidth for FSK is given as

B = |(fs – fb) – (fm – fb)| = |(fs– fm)| + 2fb

and since |(fs– fm)| equals 2∆f, the minimum bandwidth can be approximated as

B = 2(∆f + fb) (2.15) 14

where B= minimum Nyquist bandwidth (hertz)

∆f= frequency deviation |(fm– fs)| (hertz) fb = input bit rate (bps)

Bandwidth considerations of BPSK. In a BPSK modulator. the carrier input signal is multiplied
by the binary data. If + 1 V is assigned to a logic 1 and -1 V is assigned to a logic 0, the input
carrier (sin ωc t) is multiplied by either a + or - 1 . The output signal is either + 1 sin ωc t or -1
sin ωc t the first represents a signal that is in phase with the reference oscillator, the latter a signal
that is 180° out of phase with the reference oscillator. 27 Each time the input logic condition
changes, the output phase changes. Mathematically, the output of a BPSK modulator is
proportional to

BPSK output = [sin (2πfa t)] x [sin (2πfc t)] (2.20)

where fa = maximum fundamental frequency of binary input (hertz)

fc = reference carrier frequency (hertz) Solving for the trig identity for the product of two sine
functions,

0.5cos[2π(fc – fa)t] – 0.5cos[2π(fc + fa)t]

Thus, the minimum double-sided

Nyquist bandwidth (B) is fc + fa fc + fa -(fc + fa) or -fc + fa 2fa

and because fa = fb / 2, where fb = input bit rate,

where B is the minimum double-sided Nyquist bandwidth.

the output phase-versus-time relationship for a BPSK waveform. Logic 1 input produces an
analog output signal with a 0° phase angle, and a logic 0 input produces an analog output signal
with a 180° phase angle.

b) What do you mean by bit rate and baud rate?

Solution:

Baud and Minimum Bandwidth Baud refers to the rate of change of a signal on the transmission
medium after encoding and modulation have occurred. Hence, baud is a unit of transmission rate,
modulation rate, or symbol rate and, therefore, the terms symbols per second and baud are often
used interchangeably. Mathematically, baud is the reciprocal of the time of one output signaling
element, and a signaling element may represent several information bits. Baud is expressed as
baud = s t 1 (2.7) where baud = symbol rate (baud per second) ts = time of one signaling element
(seconds) 6 The minimum theoretical bandwidth necessary to propagate a signal is called the
minimum Nyquist bandwidth or sometimes the minimum Nyquist frequency. Thus, fb = 2B,
where fb is the bit rate in bps and B is the ideal Nyquist bandwidth. The relationship between
bandwidth and bit rate also applies to the opposite situation. For a given bandwidth (B), the
highest theoretical bit rate is 2B. For example, a standard telephone circuit has a bandwidth of
approximately 2700 Hz, which has the capacity to propagate 5400 bps through it. However, if
more than two levels are used for signaling (higher-than-binary encoding), more than one bit
may be transmitted at a time, and it is possible to propagate a bit rate that exceeds 2B. Using
multilevel signaling, the Nyquist formulation for channel capacity is fb = B log2 M (2.8) where
fb = channel capacity (bps) B = minimum Nyquist bandwidth (hertz) M = number of discrete
signal or voltage level can be rearranged to solve for the minimum bandwidth necessary to pass
M-ary digitally modulated carriers B = M f b 2 log (2.9) 7 If N is substituted for log2 M, reduces
to B = N f b (2.10) where N is the number of bits encoded into each signaling element. In
addition, since baud is the encoded rate of change, it also equals the bit rate divided by the
number of bits encoded into one signaling element. Thus, Baud = N f b (2.11) By comparing
Equation 2.10 with Equation 2.11 the baud and the ideal minimum Nyquist bandwidth have the
same value and are equal to the bit rate divided by the number of bits encoded.

4)

The advantages of single-sideband modulation mainly include the following.

1. Less bandwidth requirement as SSB requires a BW of fm. This will allow more number
of signals to be transmitted in the same frequency range.
 2. Lots of power saving. This is due to the transmission of only one sideband component. At
100% modulation, the percent power saving is 83.33%
3. Reduced interference of noise. This is due to the reduced bandwidth. As the bandwidth
increases, the amount of noise added to the signal will increase.
4. It is used in the applications where the power saving and low bandwidth requirements are
important. The application areas are land and air mobile communication, telemetry,
military communications, navigation and amateur radio. Many of these applications are
point to point communication applications.

6) The block diagram of FM receiver and explain each block.

Solution:

Most of these blocks are discussed individually, and in more detail, on other pages.

See filters, mixers, frequency changers, am modulation and amplifiers.

The f.m. band covers 88-108 MHz.

There are signals from many radio transmitters in this band inducing signal voltages in the aerial.

The rf amplifier selects and amplifies the desired station from the many.

It is adjustable so that the selection frequency can be altered.

This is called TUNING.

In cheaper receivers the tuning is fixed and the tuning filter is wide enough to pass all signals in
the f.m. band.

The selected frequency is applied to the mixer.

The output of an oscillator is also applied to the mixer.

The mixer and oscillator form a FREQUENCY CHANGER circuit.

The output from the mixer is the intermediate frequency (i.f.)

The i.f. is a fixed frequency of 10.7 MHz.

No matter what the frequency of the selected radio station is, the i.f. is always 10.7 MHz.

The i.f. signal is fed into the i.f. amplifier.

The advantage of the i.f. amplifier is that its frequency and bandwidth are fixed, no matter what
the frequency of the incoming signal is.

This makes the design and operation of the amplifier much simpler.

The amplified i.f. signal is fed to the demodulator.

This circuit recovers the audio signal and discards the r.f. carrier.

Some of the audio is fed back to the oscillator as an AUTOMATIC FREQUENCY CONTROL
voltage.
This ensures that the oscillator frequency is stable in spite of temperature changes.

The audio signal voltage is increased in amplitude by a voltage amplifier.

The power level is increased sufficiently to drive the loudspeaker by the power amplifier.

7) Write short notes on any two of the following:

a) PWM

Solution:

Pulse width modulation (PWM), or pulse-duration modulation (PDM), is a method of

reducing the average power delivered by an electrical signal, by effectively chopping it up into
discrete parts. The average value of voltage (and current) fed to the load is controlled by turning
the switch between supply and load on and off at a fast rate. The longer the switch is on
compared to the off periods, the higher the total power supplied to the load. Along with
MPPT maximum power point tracking, it is one of the primary methods of reducing the output of
solar panels to that which can be utilized by a battery. PWM is particularly suited for running
inertial loads such as motors, which are not as easily affected by this discrete switching, because
they have inertia to react slow. The PWM switching frequency has to be high enough not to
affect the load, which is to say that the resultant waveform perceived by the load must be as
smooth as possible.
The rate (or frequency) at which the power supply must switch can vary greatly depending on
load and application. For example, switching has to be done several times a minute in an electric
stove; 120 Hz in a lamp dimmer; between a few kilohertz (kHz) and tens of kHz for a motor
drive; and well into the tens or hundreds of kHz in audio amplifiers and computer power
supplies. 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. PWM
has also been used in certain communication systems where its duty cycle has been used to
convey information over a communications channel.

b) PLL demodulator

Solution:

A phase-locked loop or phase lock loop (PLL) is a control system that generates an
output signal whose phase is related to the phase of an input signal. There are several different
types; the simplest is an electronic circuit consisting of a variable frequency oscillator and
a phase detector in a feedback loop. The oscillator generates a periodic signal, and the phase
detector compares the phase of that signal with the phase of the input periodic signal, adjusting
the oscillator to keep the phases matched.
Keeping the input and output phase in lock step also implies keeping the input and output
frequencies the same. Consequently, in addition to synchronizing signals, a phase-locked loop
can track an input frequency, or it can generate a frequency that is a multiple of the input
frequency. These properties are used for computer clock synchronization, demodulation,
and frequency synthesis.
Phase-locked loops are widely employed in radio, telecommunications, computers and other
electronic applications. They can be used to demodulate a signal, recover a signal from a noisy
communication channel, generate a stable frequency at multiples of an input frequency
(frequency synthesis), or distribute precisely timed clock pulses in digital logic circuits such
as microprocessors. Since a single integrated circuit can provide a complete phase-locked-loop
building block, the technique is widely used in modern electronic devices, with output
frequencies from a fraction of a hertz up to many gigahertz.

c) DPSK

Solution:

The DPSK stands for “Differential phase-shift keying”. It is one type of phase modulation used
to transmit data by altering the carrier wave’s phase. In this, the modulated signal’s phase is
moved e to the element of an earlier signal. The phase of the signal tracks the low or high state of
the earlier element. This kind of phase-shift keying doesn’t require a synchronous carrier on the
demodulator.

The binary bits input series can be changed so that the next bit depends upon the earlier bit. So,
the earlier received bits in the receiver are utilized for detecting the current bit.

The above-shown figure is the DPSK waveform. From the above waveform, once the data-bit is
‘0’, the signal’s phase will not be inverted as well as continued. Once the data-bit is a ‘1’, then
the signal’s phase will be inverted.
d) Modulation of DSB-SC

Solution:

Double sideband suppressed carrier modulation

At the beginning of the explanation of amplitude modulation, we explained the AM radio system,
but the term for amplitude in the theoretical expression was complex. If the amplitude of the
carrier wave is simply changed and mathematised, it is as follows. This modulation method is
called DSB-SC (double sideband suppressed carrier modulation). As the name suggests, there’s
no wave carrier in the modulated wave.

With DSB-SC, the amplitude of carrier wave Ac is shifted proportionally to the modulating
signal m(t).

When the modulating signal m(t) is a single sine wave, it’s as follows.
If the initial phase ΦC of the carrier wave is 0, and Kdsb-sc*Am is modulation factor m, the
result is as follows.

The first equation shows a spectrum where only the modulation signal frequency Fm is separate
on either side of the carrier frequency Fc. However note that with DSB-SC, there’s no carrier
wave. For this reason, DSB-SC is considered to have good electrical efficiency. However,
synchronous detection is necessary on the receiving end, which involves a system with advanced
technology and the associated costs.

◆Single sideband suppressed carrier modulation (SSB-SC)

The modulation method where only a single band of double sideband suppressed carrier
modulation is transmitted is known simply as SSB. It offers even better electrical efficiency and
frequency band efficiency than DSB. In addition, with digital modulation, modulation is
performed at relatively low frequency, and this method is also used to up-convert the signal to a
radio frequency. SSB-SC modulation can be achieved with various systems, but in terms of
hardware, USB or LSB can be used as a filter. As with DSB-SC, the modulating signal spectrum
is shifted directly to the carrier frequency band without loss.