I.

Basics and Amplitude Modulation

Electromagnetic Spectrum and Communication Applications

The electromagnetic spectrum is the range of frequencies (the spectrum) of electromagnetic
radiation and their respective wavelengths and energies. The electromagnetic spectrum covers
electromagnetic waves with frequencies ranging from below one hertz to above 1025 hertz,
corresponding to wavelengths from thousands of kilometres down to a fraction of the size of
an atomic nucleus. This frequency range is divided into separate bands, and the electromagnetic
waves within each frequency band are called by different names; beginning at the low
frequency (long wavelength) end of the spectrum these are: radio
waves, microwaves, infrared, visible light, ultraviolet, X-rays, and gamma rays at the high-
frequency (short wavelength) end.
Table 1.1 Frequency spectrum

Elements of Communication Systems

The basic components of a communication system are information source, input transducer,
transmitter, communication channel, receiver, output transducer, and destination.

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 Fig 1.1 Basic Elements of Communication System

Information Source
The information generated by the source may be in the form of sound (human speech), picture
(image source), words (plain text in some particular language such as English, French, German
etc.) The message is the part of a communication which involves sending information from
source to destination. Information is a meaningful data that the receiver consumes.
Input Transducer
A transducer is a device which converts one form of energy or signal into another form of
energy or signal. The transducer is present at the input side and output side of the
communication system. The transducer that is present at the input side of the communication
system is called input transducer. Generally, the input transducer converts the non-electrical
signal (sound signal or light signal) into an electrical signal. The best example of an input
transducer is the microphone which is placed between the information source and the
transmitter section. A microphone is a device which converts your voice signals (sound signals)
into electrical signals.

Transmitter

The transmitter is a device which converts the signal produced by the source into a form that
is suitable for transmission over a given channel or medium. Transmitters use a technique
called modulation to convert the electrical signal into a form that is suitable for transmission
over a given channel or medium. Modulation is the main function of a transmitter.

Communication Channel

The communication channel is a wired or wireless medium through which the signal
(information) travels from source (transmitter) to destination (receiver).

Noise

Noise is an unwanted signal that enters the communication system via the communication
channel and interferes with the transmitted signal. The noise signal (unwanted signal) degrades
the transmitted signal (signal containing information).

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Receiver

The receiver is a device that receives the signal (electrical signal) from the channel and converts
the signal (electrical signal) back to its original form (light and sound) which is understandable
by humans at the destination.

Output Transducer

The transducer that is present at the output side of the communication system is called output
transducer. Generally, the output transducer converts the electrical signal into a non-electrical
signal (sound signal, light signal, or both sound and light signal).

Need for Modulation

The primary purpose of modulation in a communication system is to generate a modulated
signal which is well suited to the characteristics of transmission medium. The need for
modulation is listed as follows:

 To reduce the antenna height

 To overcome hardware system limitations
 To reduce the interference, noise & distortions made when we transmit the signals with
nearly same frequency in the audio frequency range (20-20k) Hz.
 To multiplex the more number of signals
 To the assignment of channel frequency
 To narrow banding the signal
 To reduce the complexity of the transmission system
 To increase the bandwidth of the signal

Amplitude Modulation
The amplitude of the carrier signal varies in accordance with the instantaneous amplitude of
the modulating signal.” Which means, the amplitude of the carrier signal containing no
information varies as per the amplitude of the signal containing information, at each instant.
This can be well explained by the following figures.

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 Fig 1.2 Amplitude Modulation

The first figure shows the modulating wave, which is the message signal. The next one is the
carrier wave, which is a high frequency signal and contains no information. While, the last
one is the resultant modulated wave.
It can be observed that the positive and negative peaks of the carrier wave, are interconnected
with an imaginary line. This line helps recreating the exact shape of the modulating signal.
This imaginary line on the carrier wave is called as Envelope. It is the same as that of the
message signal.
Mathematical Expressions
Following are the mathematical expressions for these waves.
Time-domain Representation of the Waves
Let the modulating signal be,

m(t)=Am cos(2πfmt)
and the carrier signal be,
c(t)=Ac cos(2πfct)
Where,
Am and Ac are the amplitude of the modulating signal and the carrier signal respectively.
fm and fc are the frequency of the modulating signal and the carrier signal respectively.
Then, the equation of Amplitude Modulated wave will be
s(t)=[Ac+Amcos(2πfmt)]cos(2πfct)

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Modulation Index of AM

A carrier wave, after being modulated, if the modulated level is calculated, then such an
attempt is called as Modulation Index or Modulation Depth. It states the level of modulation
that a carrier wave undergoes.
Rearrange the Equation 1 as below.
s(t)=Ac[1+(AmAc)cos(2πfmt)]cos(2πfct)
s(t)=Ac[1+μcos(2πfmt)]cos(2πfct) (Equation 2)
Where, μ is Modulation index and it is equal to the ratio of Am and Ac. Mathematically, we
can write it as
μ=Am/Ac (Equation 3)
Hence, we can calculate the value of modulation index by using the above formula, when the
amplitudes of the message and carrier signals are known.
The modulation index or modulation depth is often denoted in percentage called as
Percentage of Modulation. We will get the percentage of modulation, just by multiplying the
modulation index value with 100.
For a perfect modulation, the value of modulation index should be 1, which implies the
percentage of modulation should be 100%.
For instance, if this value is less than 1, i.e., the modulation index is 0.5, then the modulated
output would look like the following figure. It is called as Under-modulation. Such a wave is
called as an under-modulated wave.

Fig 1.3 Under-Modulated Wave

If the value of the modulation index is greater than 1, i.e., 1.5 or so, then the wave will be
an over-modulated wave. It would look like the following figure.

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 Fig 1.4 Over-Modulated Wave

As the value of the modulation index increases, the carrier experiences a 180o phase reversal,
which causes additional sidebands and hence, the wave gets distorted. Such an over-
modulated wave causes interference, which cannot be eliminated.

Frequency Spectrum

AM Power Distribution in AM_DSB_FC

Consider the following equation of amplitude modulated wave.

Power of AM wave is equal to the sum of powers of carrier, upper sideband, and lower
sideband frequency components.

We know that the standard formula for power of cos signal is

Where,
vrms is the rms value of cos signal.
vm is the peak value of cos signal.
First, let us find the powers of the carrier, the upper and lower sideband one by one.
Carrier power

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Upper sideband power

Similarly, we will get the lower sideband power same as that of the upper side band power.

Now, let us add these three powers in order to get the power of AM wave.

We can use the above formula to calculate the power of AM wave, when the carrier power
and the modulation index are known.
If the modulation index μ=1μ=1 then the power of AM wave is equal to 1.5 times the carrier
power. So, the power required for transmitting an AM wave is 1.5 times the carrier power for
a perfect modulation.

Generation of AM_DSB_SC using FET Balance Modulator

A balanced modulator is a device that modifies a signal, usually in the form of an amplitude
modulated (AM) radio signal. It takes the original signal that has both sidebands and a carrier
signal, and then modulates it so that only the sideband signals come through the output
modulator. This creates a balanced signal, as there is less noise because the carrier signal has
been removed.

Fig 1.5 FET Balance Modulator

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The balanced modulator can also be built using FETs. Figure shows the circuit diagram of
balanced modulator using FETs. There are three transformers T1,T2 and T3. The carrier signal
is applied to the center taps of the input transformer T1 and the output transformer T3 through
the Transformer T1. The modulating signal is applied to the input transformer T1.The carrier
signal is applied to the primary of transformer T2. This signal is further applied to two gates
of FETs in phase through the secondary of T2. The modulating voltage appearing 180 degree
out of phase at the gates, since these are the opposite ends of the center tapped transformer.
Consider that there is no modulating signal is applied. Then FET currents due to carrier signal
are equal in amplitude but opposite in the directions. These opposite and equal currents are
the primary of the output transformer cancel each other. Hence, no output is produced at the
secondary of T3. Thus the carrier is suppressed.
When modulating signal is applied, the current id1 and id2 flow in the primary of T3 due to
carrier signal as well as the modulating signal. The FET currents due to carrier are equal and
opposite and cancel each other. Seems modulating signal is applied 180 degree out of phase
at the gates, the FET currents due to modulating signal for equal but not opposite, hence do
not cancel each other. Thus DSB output is produced by FET balanced modulator.

Generation of AM_SSB using Phase Shift Method

Fig 1.6 Generation of AM_SSB using Phase Shift Method

This block diagram consists of two product modulators, two −900 phase shifters, one local
oscillator and one summer block. The product modulator produces an output, which is the
product of two inputs. The −900 phase shifter produces an output, which has a phase lag
of −900 with respect to the input.
The local oscillator is used to generate the carrier signal. Summer block produces an output,
which is either the sum of two inputs or the difference of two inputs based on the polarity of
inputs.

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The modulating signal Am cos(2πfmt) and the carrier signal Ac cos(2πfct) are directly applied
as inputs to the upper product modulator. So, the upper product modulator produces an output,
which is the product of these two inputs.
The output of upper product modulator is

The modulating signal Am cos(2πfmt) and the carrier signal Ac cos(2πfct) are phase shifted
by 900 before applying as inputs to the lower product modulator. So, the lower product
modulator produces an output, which is the product of these two inputs.
The output of lower product modulator is

Add s1(t) and s2(t) in order to get the SSBSC modulated wave s(t)s(t) having a lower sideband.

Subtract s2(t) from s1(t) in order to get the SSBSC modulated wave s(t) having a upper
sideband.

Hence, by properly choosing the polarities of inputs at summer block, we will get SSBSC
wave having a upper sideband or a lower sideband.

Generation of AM_SSB using Filter Method

This is the filter method of SSB suppression for the transmission. Fig

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 Fig 1.7 SSB Single Side Band Transmission Filter Method

A crystal controlled master oscillator produces a stable carrier frequency fc (say 100 KHz).
This carrier frequency is then fed to the balanced modulator through a buffer amplifier which
isolates these two stages.The audio signal from the modulating amplifier modulates the carrier
in the balanced modulator. Audio frequency range is 300 to 2800 Hz. The carrier is also
suppressed in this stage but allows only to pass the both side bands. (USB & LSB).A band
pass filter (BPF) allows only a single band either USB or LSB to pass through it. This side
band is then heterodyned in the balanced mixer stage with 12 MHz frequency produced by
crystal oscillator or synthesizer depends upon the requirements of our transmission. So in
mixer stage, the frequency of the crystal oscillator or synthesizer is added to SSB signal. The
output frequency thus being raised to the value desired for transmission. Then this band is
amplified in driver and power amplifier stages and then fed to the aerial for the transmission.

AM VSB
In case of SSB modulation, when a sideband is passed through the filters, the band pass filter
may not work perfectly in practice. As a result of which, some of the information may get lost.
Hence to avoid this loss, a technique is chosen, which is a compromise between DSB-
SC and SSB, called as Vestigial Sideband (VSB) technique. The word vestige which means
“a part” from which the name is derived. Vestigial Sideband Modulation or VSB
Modulation is the process where a part of the signal called as vestige is modulated, along with
one sideband. A VSB signal can be plotted as shown in the following figure.

Fig 1.8 VSB Modulation

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Along with the upper sideband, a part of the lower sideband is also being transmitted in this
technique. A guard band of very small width is laid on either side of VSB in order to avoid
the interferences. VSB modulation is mostly used in television transmissions.
Transmission Bandwidth
The transmission bandwidth of VSB modulated wave is represented as −
B=( f_{m}+ f_{v}) Hz
Where,
fm = Message bandwidth
fv = Width of the vestigial sideband

Advantages
Following are the advantages of VSB −
 Highly efficient.
 Reduction in bandwidth.
 Filter design is easy as high accuracy is not needed.
 The transmission of low frequency components is possible, without difficulty.
 Possesses good phase characteristics.

Disadvantages
Following are the disadvantages of VSB −
 Bandwidth when compared to SSB is greater.
 Demodulation is complex.
 VSB Modulation − Application

The most prominent and standard application of VSB is for the transmission of television
signals. Also, this is the most convenient and efficient technique when bandwidth usage is
considered.

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Comparison of AM Schemes

Table 1.2 Comparison of AM Schemes

Envelope Detector
The AM diode detector is an envelope detector – it provides an output of the envelope of the
signal. As such the diode detector or demodulator is able to provide an output proportional to
the amplitude of the envelope of the amplitude modulated signal.
The signal diode detector consists of two main elements to the circuit:
Diode / rectifier: The diode in the detector serves to that enhances one half of the received
signal over the other. In many instances Schottky diodes are used for this form of detector,
because signal levels may be low, and Schottky diodes have a much lower turn on voltage
(typically around 0.2 V) than standard silicon diodes (typically around 0.7 or 0.7 V).
Low pass filter: The low pass filter is required to remove the high frequency elements that
remain within the signal after detection / demodulation. The filter usually consists of a very
simple RC network but in some cases It can be provided simply by relying on the limited
frequency response of the circuitry following the rectifier. As the capacitor in the circuit stores
the voltage, the output voltage reflects the peak of the waveform. Sometimes these circuits are
used as peak detectors.

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When selecting the value of the capacitor used int he circuit, it should be large enough to hold
the peak of the RF waveform, but not so large that it attenuates any modulation on the signal,
i.e. it should act as a filter for the RF carrier and not the audio modulation.

Fig 1.9 Circuit of an envelope detector as used in an AM radio receiver

The circuit typically has a relatively high source impedance. When linking the circuit to a following
stage of the circuit, care should be taken not to land the detector too much otherwise the operation
will be impaired.

Normally a resistor is placed across the capacitor - this may either be the load of the next stage, a
volume control, or resistor in the circuit. This level of this should be determined by calculating the
time constant of the capacitor and the load. This should be between the RF signal and audio
modulation so that the RF is satisfactorily removed, but the audio modulation is left untouched.

It is worth noting in this circuit that the secondary of the transformer provides a DC return to ground.
Sometimes when the AM signal detector is used using a capacitor connection tot he previous
stage, then a resistor or choke (inductor) to ground must be used at the input so that a DC return
path is provided. If not the circuit will not operate correctly.

Fig 1.10 Capacitor coupled envelope signal detector showing resistor providing DC
return path.

The value of the resistor on the input providing the DC return path is normally critical, but it can
help provide the require match without absorbing too much signal.

AM diode detection process

In rectifying the RF signal, the AM diode detector provides an output equivalent to the envelope of
one half of the signal, i.e. it is an envelope detector.

In view of the operation of the diode detector, it may sometimes be referred to as an envelope
detector.

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The incoming amplitude modulated RF signal consists of a waveform of both positive and negative
going voltages as shown. Any audio transducer would not respond to this.

Fig 1.11 AM diode envelope detection process.

The diode envelope detector rectifies the waveform leaving only the positive or negative half of the
waveform.

The high frequency element of this is then filtered out, typically using a capacitor which forms the
low pass filter and effectively ‘fills in’ the high frequency elements, leaving a waveform to which a
transducer like a pair of earphones or a loudspeaker could respond to and convert into sound
waves.

Significance of RC time Constant in envelope Detector

By keeping the time constant, RC large, the capacitor discharging a small that is negligible hence
spikes can be reduced.But the large values of RC create another problem called diagonal clipping.
Hence we cannot increase it beyond the certain limit.

Choice of time constant RC

It is desired to keep the time constant RC very high as compared to time of carrier dev in order to
minimize spikes fluctuation in detected envelope. On the other hand, if it is kept too high the
discharge curve becomes approximately horizontal. In that case, negative pics of detected
envelope baby completely or partially missing. Therefore the recovered baseband signal is
distorted at negative peaks. This type of distortion is called diagonal clipping.

Diagonal clipping maybe caused

 When the time constant of detector is not selected properly

 Increase in depth of modulation index make the envelope slope steeper than the discharge
path of capacitor

 To avoid diagonal clipping proper value of time constant may be obtained as follows

 During the non-conducting period of diode, the voltage across RC combination at an instant
―t‖ is given by

𝑉c(𝑡) = 𝑉0𝑒−𝑡/𝑅

 The slope of capacitor discharge or rate of fall may be obtained by differentiating Vc(t) with
respect to time t

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  If the distortion is avoided, the decrease in capacitor voltage must follows envelope. We
know the envelope of the modulated voltage signal which is given by VAM (t)= Vc[1+ ma
sinωmt] sinωct

VAM=envelope voltage VAM= Vc[1+ ma sinωmt]

The slope of the envelope is given by

Where K=proportionality constant

 The diagonal clipping is avoided, rate of fall for slope of capacitor is algebraically greater
than or equal to slope of envelope.

Question Bank

Part A

1. Define modulation index of an AM signal

2. Draw the circuit diagram of an envelope detector
3. Draw the spectrum of AM.
4. Draw the spectrum of DSB-SC.
5. Define the transmission efficiency of AM signal.
6. Draw the phasor diagram of AM signal.
7. Advantages of SSB.
8. Disadvantages of DSB-SC.
9. Draw the spectrum of SSB
10. Distinguish between low level and high level modulator.

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Part B

1. Explain the generation of AM signals using square law modulator.

2. Explain the detection of AM signals using envelope detector.
3. Explain about Balanced modulator to generate DSB-SC signal.
4. Explain the power distribution of DSB_FC AM wave.

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 II. Angle Modulation

Frequency Modulation

In amplitude modulation, the amplitude of the carrier signal varies. Whereas, in Frequency
Modulation (FM), the frequency of the carrier signal varies in accordance with the
instantaneous amplitude of the modulating signal.
Hence, in frequency modulation, the amplitude and the phase of the carrier signal remains
constant. This can be better understood by observing the following figures.

Fig 2.1 Frequency Modulation

The frequency of the modulated wave increases, when the amplitude of the modulating or
message signal increases. Similarly, the frequency of the modulated wave decreases, when the
amplitude of the modulating signal decreases. Note that, the frequency of the modulated wave
remains constant and it is equal to the frequency of the carrier signal, when the amplitude of
the modulating signal is zero.

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Mathematical Representation

The equation for instantaneous frequency fi in FM modulation is

Where,
fc is the carrier frequency
kt is the frequency sensitivity
m(t) is the message signal
We know the relationship between angular frequency ωi and angle θi(t) as

This is the equation of FM wave.

If the modulating signal is m(t)=Am cos(2πfmt), then the equation of FM wave will be

Modulation Index of FM

The ratio of frequency deviation to the modulating frequency is knwn as the modulation
index of FM.

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Frequency Deviation

The difference between FM modulated frequency (instantaneous frequency) and normal carrier
frequency is termed as Frequency Deviation. It is denoted by Δf, which is equal to the product
of kf and Am.

Deviation Ratio

Accordingly the FM deviation ratio can be defined as: the ratio of the maximum carrier
frequency deviation to the highest audio modulating frequency.

m=Max frequency deviation/Max modulation frequency

Carson’s Rule for Bandwidth of FM

This rule states that the bandwidth of an FM system is double the sum of the maximum
frequency deviation and the highest modulating frequency fm. Thus, if B is the bandwidth of
the system; then according to Carson's rule:

B=2( fd + fm)

Comparison of AM and FM

Table 2.1 Comparison of AM and FM

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Narrow Band and Wideband FM Comparison

Narrowband FM

Following are the features of Narrowband FM.

 This frequency modulation has a small bandwidth when compared to wideband FM.
 The modulation index ββ is small, i.e., less than 1.
 Its spectrum consists of the carrier, the upper sideband and the lower sideband.
 This is used in mobile communications such as police wireless, ambulances, taxicabs,
etc.

Wideband FM

Following are the features of Wideband FM.

 This frequency modulation has infinite bandwidth.

 The modulation index ββ is large, i.e., higher than 1.
 Its spectrum consists of a carrier and infinite number of sidebands, which are located
around it.
 This is used in entertainment, broadcasting applications such as FM radio, TV, etc.

Generation of FM Using Varactor Diode Modulator (Direct Method)

 The varactor diode FM modulator has been shown below in figure .

Fig 2.2 Varactor Diode Modulator

 A varactor diode is a semiconductor diode whose junction capacitance varies linearly

with the applied bias and the varactor diode must be reverse biased.

Working Operation

 The varactor diode is reverse biased by the negative dc source –Vb.

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  The modulating AF voltage appears in series with the negative supply voltage. Hence,
the voltage applied across the varactor diode varies in proportion with the modulating
voltage.
 This will vary the junction capacitance of the varactor diode.
 The varactor diode appears in parallel with the oscillator tuned circuit.
 Hence the oscillator frequency will change with change in varactor dioide capacitance
and FM wave is produced.
 The RFC will connect the dc and modulating signal to the varactor diode but it offers a
very high impedance at high oscillator frequency. Therefore, the oscillator circuit is
isolated from the dc bias and modulating signal.

Indirect Method of FM Generation

In the direct methods of generation of FM, LC oscillators are to be used. The crystal oscillator
cannot be used. The LC oscillators are not stable enough for the communication or broadcast
purpose. Thus, the direct methods cannot be used for the broadcast applications.

The alternative method is to use the indirect method called as the Armstrong method of FM
generation.

In this method, the FM is obtained through phase modulation. A crystal oscillator can be used
hence the frequency stability is very high and this method is widely used in practice.

Fig 2.3 Armstrong Method for FM Generation

The Armstrong method uses the phase modulator to generate a frequency modulated wave.

Working Principle

The working operation of this system can be divided into two parts as follows:

Part I: Generate a narrow band FM wave using a phase modulator.

Part II: Use the frequency multipliers and mixer to obtain the required values of
frequency deviation, carrier and modulation index.

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Part I: Generate a narrow band FM using Phase Modulator

As discussed carrier, we can generate FM using a phase modulator.

The modulating signal x(t) is passed through an integrator before applying it to the phase
modulator as shown in figure 1.

Let the narrow band FM wave produced at the output of the phase modulator be represented
by s1(t) i.e.,

where Vc1 is the amplitude and f1 is the frequency of the carrier produced by the crystal
oscillator.

The phase angle Φ1(t) of s1(t) is related to x(t) as follows:

where k1 represents the frequency sensitivity of the modulator.

If Φ1(t) is very small then,

Hence, the approximate expression for s1(t) can be obtained as follows:

After approximation, we get,

Substituting,

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This expression represents a narrow band FM. Thus, at the output of the phase modulator, we
obtain a narrow band FM wave.

Part II: Implementation of the Phase Modulator

Figure.2.4 shows the block diagram of phase modulator circuit.

Fig 2.4 Phase Modulator Circuit

Working Principle

The crystal oscillator produces a stable unmodulated carrier which is applied to the 90° phase
shifter as well as the combining network through a buffer.

The 90° phase shifter produces a 90° phase shifted carrier. It is applied to the balanced
modulator along with the modulating signal.

Thus, the carrier used for modulation is 90° shifted with respect to the original carrier.

At the output of the product modulator, we get DSB SC signal i.e., AM signal without carrier.

This signal consists of only two sidebands with their resultant in phase with the 90° shifted
carrier.

The two sidebands and the original carrier without any phase shift are applied to a combining
network (∑). At the output of the combining network, we get the resultant of vector addition
of the carrier and two sidebands as shown in figure 2.5.

Fig 2.5 Phasor explaining the generation of PM

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Now, as the modulation index is increased, the amplitude of sidebands will also increase.
Hence, the amplitude of their resultant increases. This will increase the angle Φ made by the
resultant with unmodulated carrier.

The angle Φ decreases with reduction in modulation index as shown in figure2.6.

Fig 2.6 Effect of modulation index on frequency

Thus, the resultant at the output of the combining network is phase modulated. Hence, the block
diagram operates as a phase modulator.

Part III: Use of Frequency Multipliers Mixer and Amplifier

The FM signal produced at the output of phase modulator has a low carrier frequency and low
modulation index. They are increased to an adequately high value with the help of frequency
multipliers and mixer.

Phase Modulation

In frequency modulation, the frequency of the carrier varies. Whereas, in Phase Modulation
(PM), the phase of the carrier signal varies in accordance with the instantaneous amplitude of
the modulating signal.

So, in phase modulation, the amplitude and the frequency of the carrier signal remains constant.
This can be better understood by observing the following figures.

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 Fig 2.7 Phase Modulation

The phase of the modulated wave has got infinite points, where the phase shift in a wave can
take place. The instantaneous amplitude of the modulating signal changes the phase of the
carrier signal. When the amplitude is positive, the phase changes in one direction and if the
amplitude is negative, the phase changes in the opposite direction.

Mathematical Representation

The equation for instantaneous phase ϕi in phase modulation is

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Phase modulation is used in mobile communication systems, while frequency modulation is
used mainly for FM broadcasting.

Principle of Slope Detection

According to the principle of the slope detector, the received FM signal is applied to an LC
circuit whose output is an amplitude and frequency-modulated signal. This signal is then passed
to an AM detector, which uses a detector diode, D, as shown in Figure (a) to recover the
modulating signal, Vo.
The circuit diagram of a slope detector is shown in Figure 2.7. This circuit is also known as a
single-tuned slope detector.

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 Fig 2.7 FM Slope Detector Circuit

The transformer, T, shown in Figure (a), passes the received signal to the diode D. The
secondary winding of the transformer T used as the inductor, and a capacitor. CT is connected
in parallel to constitute an LC resonating circuit. The secondary winding is tuned to a frequency
slightly less than the resonating frequency of the LC resonating circuit the resonating frequency
of the resonating circuit. fo is greater than the central frequency of the input signal fc.
Symbolically. fo > fc as shown in below Figure 2.8.

Fig 2.8 Slope Detector Waveform

If the maximum frequency deviation in the input FM signal is ±df. the operating frequency
range of the voltage versus frequency curve of Figure (b) will be (fc -fd) to (fc + fd), as clearly
shown in Figure (b). This range covers the linear region of the curve. The frequency variation
is converted into the corresponding voltage variation, and the voltage available at the anode of
the diode D carries both the amplitude variation and the frequency variation in direct proportion
to the modulating signal. This is marked as Vo in Figure 2.7.
The detector diode D rectifies the secondary voltage VD, which is marked as r(t) in Figure 2.7.
The rectified voltage is used to charge the capacitor C up to the peak values.
The capacitor discharges through the resistance R to develop the modulating, voltage Vo. This
modulating voltage is the voltage em, as marked in Figure 2.7. As a result, the slope-detector
circuit demodulates the received signal and recovers the original modulating signal.

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Balance Slope Detector

A balanced slope detector is an improved version of the slope detector. The drawback of
harmonic distortion is removed in this detector by using two slope detectors instead of one as
in a single-tuned slope detector.

Fig 2.9 FM Balanced Slope Detector Circuit

Circuit Description
The circuit diagram, shown in Figure 2.9, has two slope detectors marked slope detector 1 and
slope detector 2. Both the slope detractors are called balanced because they have identical
components as follows:
Slope detector 1: It consists of a detector diode D1, filter capacitor C1, load resistor R1, and
variable capacitor CT1. The variable capacitor CT1 is called the tuning capacitor because it is
adjusted to tune the upper winding of the secondary Winding of the input transformer T.
Slope detector 2: It consists of a detector diode D2, that is identical to D1. It also has filter
capacitor C2, load resistor R2, and variable capacitor CT2. The tuning capacitor CT2 tunes the
lower winding of the secondary winding of the input transformer T.
The two slope detectors are balanced because C1 = C2, R1 = R2, and D1 is identical to D2.
The upper and lower windings of the secondary windings of the center-tap transformer T are
also identical.
The primary winding of the input transformer T is tuned to the central frequency of the carrier
signal fc by using the tuning. capacitor CT. The secondary windings are tuned to different
frequencies so that the circuit is staggered tuned. The outputs of the detector diodes D1 and
D2, are filtered by C1, R1 and C2, R2, respectively. The voltage V1. which is developed across
R1, and voltage V2, which is developed across R2 are added together between the points E and
F to get the final output voltage Vo, which is the modulating voltage em.
Circuit Operation
The operation of the circuit can be explained by considering the two slope detectors separately.
Slope Detectorn 1
The resonating frequency of slope detector 1 is set to f1, by adjusting CT1 so that it is greater
than fc by an amount of ∆f. As a result, slope detector 1 is tuned to fc, given as:

28
f1 = fc + ∆f
The signal coupled to the upper winding of the secondary winding of the transformer T has a
central frequency fc. If the maximum frequency deviation of the incoming FM signal, r(t),
is ±fd, the operating range of slope detector 1 is between fc and (fc + fd), shown in figure (b).
Which illustrates the frequency response curve of both slope detectors.
When the incoming signal frequency deviation between fc and (fc + fd), diode D1 is forward
biased because voltage VD1 increases according to the frequency response curve of slope
detector 1. The diode rectifies this amplitude and frequency modulated signal.
Capacitors C1 and resistor R1 then filter the rectified voltage. The voltage so developed across
R1 is the positive half of the modulating signal. This can explained by the frequency-deviation
curve applied at fc on the frequency-axis of Figure (b). The positive half of the frequency-
deviation curve lies in the response curve of slope detector 1. Therefore, the frequency
deviations that are greater than are converted into corresponding amplitude by slope detector
1.
Slope Detector 2
Slope detector 2 is tuned to f2, by adjusting the tuning capacitor C2 such that:
f2 = fc - ∆f
The input FM signal whose frequency deviation lies between fc and (fc — df) is converted into
corresponding amplitude variations by slope detector 2 because this part of the frequency-
deviation curve lies toward the frequency-response curve of slope detector 2. The voltage VD2,
developed across the lower winding of the secondary winding of the transformer T is
amplitude-and frequency-modulated, which corresponds to the frequency deviation from fc to
(fc — df). This voltage, VD2 is rectified by diode D2, because it is forward-biased and the
capacitor C2, filters this rectified voltage. The filtered voltage is developed across R2 and, as
a result, the output of the balanced slope detector is the negative half of the modulating signal
corresponding to the frequency deviations lower than fc.
Combined Response of Slope Detectors 1 and 2
The output of the balanced slope detector is the combined output of the individual slope
detectors. Slope detector 1 provides the positive half of the modulating signal across the Output
terminals E and F. When D1 is forward-biased, the diode D, is also slightly forward-biased
because the voltage developed across the lower winding is very small. This is because slope
detector 2 is tuned to (fc — ∆f), while the incoming signal lies between fc and (fc + ∆f)
The conduction of diode D2, results in a small current that flows though R2 from F to D. The
direction of the current due to D1 is from E to D through R1
Therefore, when a positive voltage is developed across R1 a very small negative voltage also
develops across R2. The sum of these two voltages appears between output terminals E and F.
Thus, the final output is sligtly reduced by a negative voltage developed across R2, because: -
Vo = V1 - V2

29
A similar action takes place when the incoming FM signal contains the frequency deviations
between fc and (fc - df). The diode D2 is forward-biased because appreciable positive voltage
VD2 appears across D2. This is because slope detector 2 is tuned towards (fc - df). This
frequency-deviation range lies in the negative half of the modulating signal. As a result, the
voltage developed across R2 after filtering the rectified output of D2 is the negative half of the
modulating signal.
During the conduction of D2, a small voltage also appears across the upper winding of the
secondary winding of transformer T. This voltage is small because the upper winding is tuned
to (fc — ∆f), while the incoming voltage lies in the frequency range fc to (fc — ∆f). Due to
this voltage, diode D2, also supplies a positive voltage across R1. This positive voltage slightly
reduces the negative voltage appearing across R2, when summed up between the output
terminals E and F. The net voltage is still a negative half cycle of the modulating signal.
The positive and negative halves of he modulating signal are available across the output
terminals of the balanced slope detector marked as shown in Figure (b). The overall response
of both slope detectors is shown in Figure (b), and the operating range at the curve is marked
between the points K and L. This operating range of the overall response curve is a straight
line, and therefore, the operation is linear. This removes the; nonlinear behavior of a single-
tuned slope detector, and the higher harmonics are not generated in a balanced slope detector.
The overall response curve takes the shape of the letter S, as shown in the Figure 2.10. This is
called S-response of the FM detectors.

Fig 2.10 FM Balanced Slope Detector Frequency Response Curve

Foster Seely Discriminator

The Foster Seeley detector or as it is sometimes described the Foster Seeley discriminator is
quite similar to the ratio detector at a first look. It has an RF transformer and a pair of diodes,
but there is no third winding - instead a choke is used.

30
 Fig 2.11 Foster Seeley Discriminator Circuit

In many respects the Foster Seeley FM demodulator resembles the circuit of a full wave bridge
rectifier - the format that uses a centre tapped transformer, but additional components are added
to give it a frequency sensitive aspect.
The basic operation of the circuit can be explained by looking at the instances when the
instantaneous input equals the carrier frequency, the two halves of the tuned transformer circuit
produce the same rectified voltage and the output is zero. If the frequency of the input changes,
the balance between the two halves of the transformer secondary changes, and the result is a
voltage proportional to the frequency deviation of the carrier.
Looking in more detail at the circuit, the Foster-Seeley circuit operates using a phase difference
between signals. To obtain the different phased signals a connection is made to the primary
side of the transformer using a capacitor, and this is taken to the centre tap of the transformer.
This gives a signal that is 90° out of phase.
When an un-modulated carrier is applied at the centre frequency, both diodes conduct, to
produce equal and opposite voltages across their respective load resistors. These voltages
cancel each one another out at the output so that no voltage is present. As the carrier moves off
to one side of the centre frequency the balance condition is destroyed, and one diode conducts
more than the other. This results in the voltage across one of the resistors being larger than the
other, and a resulting voltage at the output corresponding to the modulation on the incoming
signal.
The choke is required in the circuit to ensure that no RF signals appear at the output. The
capacitors C1 and C2 provide a similar filtering function.
Both the ratio detector and Foster-Seeley detectors are expensive to manufacture. Any wound
components like the RF transformers are expensive to manufacture when compared with
integrated circuits produced in vast numbers. As a result the Foster Seeley discriminator as well
as the ratio detector circuits are rarely used in modern radio receivers as FM demodulators.

31
Foster Seeley circuit for frequency control
Prior to the introduction of very stable local oscillators within superhet radios - the universal
format for radios receiving FM, local oscillators had a tendency to drift. Drift was a major
factor in domestic radio receivers, although it was present in all radios.
When receiving FM signals the drift meant that the incoming FM signal might drift away from
being at the centre of the FM detector slope onto the non-linear portions. This meant that the
signal would become distorted.
To overcome this, radio receivers would incorporate a facility known as automatic frequency
control was implemented. Using this, the DC offset from the FM demodulator is used to tune
the receiver local oscillator to bring it back on frequency.

Fig 2.12 FM Demodulator Curve

A DC offset is produced when the centre frequency of the carrier is not on the centre of the
demodulator curve. By filtering off the audio, only a DC component remains. Typically a long
time constant RC combination is used to achieve this. The time constant of this RC network
can be quite long as the drift of the oscillator occurs gradually over a period of seconds, and it
must also be longer than that of the lowest frequency of the audio.

Fig 2.13 AFC Circuit

The filtered voltage is applied to a varactor diode within the local oscillator such that it causes
the local oscillator to remain on tune for the FM signal being received. In this way the receiver

32
can operate so that the signal being received is demodulated within the linear region of the FM
demodulator.

Essentially the effect of the AFC circuitry is to create a form of negative feedback loop that
seeks to keep the centre of the FM signal at the centre of the FM demodulation S curve. It is
essentially a frequency locked loop.

Most radios used for FM reception that have free running local oscillators incorporate an
automatic frequency control, AFC circuit. It uses only a few components and it provides for a
significant improvement in the performance of the receiver, enabling the FM signal to be
demodulated with minimum distortion despite the drift of the local oscillator signal.

Prior to the widespread introduction of frequency synthesizers, AFC was not always used in
radios such as walkie talkies and handhelds radios aimed at for two way radio communications
applications as they tended to use crystal controlled oscillators and these did not drift to any
major degree. Hence there was less requirement for an AFC.

Ratio Detector

In the Foster-Seeley discriminator, changes in the magnitude of the input signal will give rise
to amplitude changes in the resulting output voltage. This makes prior limiting necessary. It is
possible to modify the discriminator circuit to provide limiting, so that the amplitude limiter
may be dispensed with. A circuit so modified is called a Ratio Detector Circuit. As we now,
the sum Vao + Vbo remains constant, although the difference varies because of changes in
input frequency. This assumption is not completely true. Deviation from this ideal does not
result in undue distortion in the Ratio Detector Circuit, although some distortion is undoubtedly
introduced. It follows that any variations in the magnitude of this sum voltage can be considered
spurious here. Their suppression will lead to a discriminator which is unaffected by the
amplitude of the incoming signal. It will therefore not react to noise amplitude or spurious
amplitude modulation.

Fig 2.14 Ratio Detector Circuit

33
Operation:

With diode D2 reversed, o is now positive with respect to b’, so that Va′b′ is now a sum voltage,
rather than the difference it was in the discriminator. It is now possible to connect a large
capacitor between a’ and b’ to keep this sum voltage constant. Once C5 has been connected, it
is obvious that Va′b′ is no longer the output voltage; thus the output voltage is now taken
between o and o′. It is now necessary to ground one of these two points, and o happens to be
the more convenient, as will be seen when dealing with practical Ratio Detector Circuit.
Bearing in mind that in practice R5 = R6, Vo is calculated as follows:

Equation shows that the ratio detector output voltage is equal to half the difference between
the output voltages from the individual diodes. Thus (as in the phase discriminator) the output
voltage is proportional to the difference between the individual output voltages. The Ratio
Detector Circuit therefore behaves identically to the discriminator for input frequency changes.
The S curve of Figure 6-40 applies equally to both circuits.
Amplitude limiting by the ratio detector:
It is thus established that the ratio detector behaves in the same way as the phase discriminator
when input frequency varies (but input voltage remains constant). The next step is to explain
how the Ratio Detector Circuit reacts to amplitude changes. If the input voltage V12 is constant
and has been so for some time, C5 has been able to charge up to the potential existing between
a’ and b’. Since this is a dc voltage if V12 is constant, there will be no current either flowing
in to charge the capacitor or flowing out to discharge it. In other words, the input impedance
of C5 is infinite. The total load impedance for the two diodes is therefore the sum of R3 and
R4, since these are in practice much smaller than R5 and R6. If V12 tries to increase, C5 will
tend to oppose any rise in Vo. The way in which it does this is not, however, merely to have a
fairly long time constant, although this is certainly part of the operation. As soon as the input
voltage tries to rise, extra diode current flows, but this excess current flows into the capacitor
C5, charging it. The voltage Va′b′ remains constant at first because it is not possible for the
voltage across a capacitor to change instantaneously. The situation now is that the current in
the diodes load has risen, but the voltage across the load has not changed. The conclusion is
that the load impedance has decreased. The secondary of the ratio detector transformer is more
heavily damped, the Q falls, and so does the gain of the amplifier driving the Ratio Detector
Circuit. This neatly counteracts the initial rise in input voltage. Should the input voltage fall,
the diode current will fall, but the load voltage will not, at first, because of the presence of the
capacitor. The effect is that of an increased diode load impedance; the diode current has fallen,
but the load voltage has remained constant. Accordingly, damping is reduced, and the gain of
the driving amplifier rises, this time counteracting an initial fall in the input voltage. The ratio
detector provides what is known as diode variable damping. We have here a system of varying
the gain of an amplifier by changing the damping of its tuned circuit. This maintains a constant
output voltage despite changes in the amplitude of the input.

34
Question Bank

Part A

1. What do you mean by narrowband and wideband FM?

2. Give the frequency spectrum of narrowband FM?
3. Why Armstrong method is superior to reactance modulator.
4. Define frequency deviation in FM?
5. State Carson’s rule of FM bandwidth?
6. Differentiate between narrow band and wideband FM.?
7. What are the advantages of FM.?
8. Define PM.
9. What is meant by indirect FM generation?

10. Define modulation index of FM and PM.

Part B

1. Explain the indirect method of generation of FM wave and any one method of
demodulating an FM wave.
2. Derive the expression for the frequency modulated signal. Explain what is meant
by narrow band FM and wide band FM?
3. Explain the FM detection using balanced slope detector.
4. Explain the Foster Seeley discriminator.

35
 III. Pulse Modulation and Multiplexing

Pulse Modulation
Pulse modulation is a technique in which the signal is transmitted with the information by
pulses. This is divided into Analog Pulse Modulation and Digital Pulse Modulation.
Analog pulse modulation is classified as
Pulse Amplitude Modulation (PAM)
Pulse Width Modulation (PWM)
Pulse Position Modulation (PPM)

Digital modulation is classified as

Pulse Code Modulation
Delta Modulation
Pulse Amplitude Modulation
Pulse amplitude modulation is a technique in which the amplitude of each pulse is controlled
by the instantaneous amplitude of the modulation signal. It is a modulation system in which
the signal is sampled at regular intervals and each sample is made proportional to the amplitude
of the signal at the instant of sampling. This technique transmits the data by encoding in the
amplitude of a series of signal pulses.

Fig 3.1 Pulse Amplitude Modulation Signal

There are two types of sampling techniques for transmitting a signal using PAM. They are:
Flat Top PAM

36
Natural PAM
Flat Top PAM
The amplitude of each pulse is directly proportional to modulating signal amplitude at the time
of pulse occurrence. The amplitude of the signal cannot be changed with respect to the analog
signal to be sampled. The tops of the amplitude remain flat.

Fig 3.2 Flat Top Pulse Amplitude Modulation

Natural PAM
The amplitude of each pulse is directly proportional to modulating signal amplitude at the time
of pulse occurrence. Then follows the amplitude of the pulse for the rest of the half-cycle.

Fig 3.3 Natural Pulse Amplitude Modulation

Pulse Width Modulation (PWM)

In PWM, the width of the modulated pulses varies in proportion with the amplitude of
modulating signal. The waveforms of PWM is shown in fig.

37
 Fig 3.4 PWM Signal

Generation of PWM Signal

The block diagram of a PWM signal generator is shown in fig.2 below. This circuit can also
be used for the generation of PPM signal.

Fig 3.5 PWM and PPM Generator

A sawtooth generator generates a sawtooth signal of frequency fs, and this sawtooth signal in
this case is used as a sampling signal.It is applied to the inverting terminal of a comparator.The
modulating signal x (t) is applied to the non-inverting terminal of the same comparator.The
comparator output will remain high as long as the instantaneous amplitude of x (t) is higher
than that of the ramp signal.This gives rise to a PWM signal at the comparator output as shown
in fig.

Fig 3.6 Fig Waveforms

38
Here, it may be noted that the leading edges of the PWM waveform coincide with the falling
edges of the ramp signal. Thus, the leading edges of PWM signal are always generated at fixed
time instants.
However, the occurance of its trailing edges will be dependent on the instantaneous amplitude
of x(t). Therefore, this PWM signal is said to be trail edge modulated PWM.

Detection of PWM Signal

The circuit for the detection of PWM signal is shown in fig.

Fig 3.7 PWM Detection Circuit

The working operation of the circuit may be explained as under:
The PWM signal received at the input of the detection circuit is contaminated with noise. This
signal is applied to pulse generator circuit which regenerates the PWM signal.
Thus, some of the noise is removed and the pulses are squared up.
The regenerated pulses are applied to a reference pulse generator. It produces a train of constant
amplitude, constant width pulses.
These pulses are synchronized to the leading edges of the regenerated PWM pulses but delayed
by a fixed interval.
The regenerated PWM pulses are also applied to a ramp generator. At the output of it, we get
a constant slope ramp for the duration of the pulse. The height of the ramp is thus proportional
to the width of the PWM pulses.
At the end of the pulse, a sample and hold amplifier retains the final ramp voltage until it is
reset at the end of the pulse.
The constant amplitude pulses at the output of reference pulse gtenerator are then added to the
ramp signal.
The output of the adder is then clipped off at a thereshold level to generate a PAM signal at
the output of the clipper.
A low pass filter is used to recover the original modulating signal back from the PAM signal.
The waveforms for this circuit have been shown in fig.

39
 Fig 3.9 Waveforms for PWM detection circuit
Advantages of PWM

 Less effect of noise i.e., very good noise immunity.

 Synchronization between the transmitter and receiver is not essential (Which is
essential in PPM).
 It is possible to reconstruct the PWM signal from a noise, contaminated PWM, as
discussed in the detection circuit. Thus, it is possible to separate out signal from noise (which
is not possible in PAM).

Disadvantages of PWM
 Due to the variable pulse width, the pulses have variable power contents. Hence, the
transmission must be powerful enough to handle the maximum width, pulse, though the
average power transmitted can be as low as 50% of this maximum power.
 In order to avoid any waveform distortion, the bandwidth required for the PWM
communication is large as compared to bandwidth of PAM.

Multiplexing is used in the cases where the signals of lower bandwidth and the transmitting

40
media is having higher bandwidth. In this case, the possibility of sending a number of signals is
more. In this the signals are combined into one and are sent over a link which has greater
bandwidth of media than the communicating nodes.

Time Division Multiplexing (TDM)

This happens when data transmission rate of media is greater than that of the source, and each
signal is allotted a definite amount of time. These slots are so small that all transmissions appear
to be parallel. In frequency division multiplexing all the signals operate at the same time with
different frequencies, but in time division multiplexing all the signals operate with same
frequency at different times.

Fig 3.10 Fig Time Division Multiplexing

Synchronous TDM
The time slots are pre-assigned and fixed. This slot is even given if the source is not ready with
data at this time. In this case the slot is transmitted empty. It is used for multiplexing digitized
voice stream.

Fig 3.11 Synchronous Time Division Multiplexing

41
Asynchronous (or statistical) TDM
The slots are allocated dynamically depending on the speed of source or their
ready state. It dynamically allocates the time slots according to different input
channel’s needs, thus saving the channel capacity.

Fig 3.12 Asynchronous Time Division Multiplexing

Frequency Division Multiplexing (FDM)

In this a number of signals are transmitted at the same time, and each source transfers its signals
in the allotted frequency range. There is a suitable frequency gap between the 2 adjacent signals
to avoid over-lapping. Since the signals are transmitted in allotted time so this decreases the
probability of collision. The frequency spectrum is divided into several logical channels, in which
every user feels that they possess a particular bandwidth. A number of signals are sent
simultaneously on the same time allocating separate frequency band or channel to each signal. It
is used in radio and TV transmission. Therefore, to avoid interference between two successive
channels Guard bands are used.

Fig 3.13 Frequency division Multiplexing

42
Pulse Code Modulation

A signal is Pulse Code modulated to convert its analog information into a binary sequence, i.e.,
1s and 0s. The output of a Pulse Code Modulation (PCM) will resemble a binary sequence.
The following figure shows an example of PCM output with respect to instantaneous values
of a given sine wave.

Fig 3.14 Output of Pulse Code Modulation

Instead of a pulse train, PCM produces a series of numbers or digits, and hence this process is
called as digital. Each one of these digits, though in binary code, represent the approximate
amplitude of the signal sample at that instant. In Pulse Code Modulation, the message signal
is represented by a sequence of coded pulses. This message signal is achieved by representing
the signal in discrete form in both time and amplitude.

Basic Elements of PCM

The transmitter section of a Pulse Code Modulator circuit consists of Sampling, Quantizing
and Encoding, which are performed in the analog-to-digital converter section. The low pass
filter prior to sampling prevents aliasing of the message signal.

Fig 3.15 Block Diagram of PCM

43
The basic operations in the receiver section are regeneration of impaired signals, decoding, and
reconstruction of the quantized pulse train. The following figure is the block diagram of PCM
which represents the basic elements of both the transmitter and the receiver sections.
Elements of PCM

Low Pass Filter (LPF)

This filter eliminates the high frequency components present in the input analog signal which
is greater than the highest frequency of the message signal, to avoid aliasing of the message
signal.

Sampler

This is the circuit which uses the technique that helps to collect the sample data at instantaneous
values of the message signal, so as to reconstruct the original signal. The sampling rate must
be greater than twice the highest frequency component W of the messagesignal, in accordance
with the sampling theorem.

Quantizer

Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer, reduces the redundant bits and compresses the value.

Encoder

The digitization of analog signal is done by the encoder. It designates each quantized level bya
binary code. The sampling done here is the sample-and-hold process. These three sections will
act as an analog to the digital converter. Encoding minimizes the bandwidth used.

Regenerative Repeater

The output of the channel has one regenerative repeater circuit to compensate the signal loss
and reconstruct the signal. It also increases the strength of the signal.

Decoder

The decoder circuit decodes the pulse coded waveform to reproduce the original signal. This
circuit acts as the demodulator.
Reconstruction Filter

44
After the digital-to-analog conversion is done by the regenerative circuit and the decoder, a
low pass filter is employed, called as the reconstruction filter to get back the original signal.
Hence, the Pulse Code Modulator circuit digitizes the analog signal given, codes it, and
samples it. It then transmits in an analog form. This whole process is repeated in a reverse
pattern to obtain the original signal.
Quantization

The digitization of analog signals involves the rounding off of the values which are
approximately equal to the analog values. The method of sampling chooses a few points on the
analog signal and then these points are joined to round off the value to a near stabilized value.
Such a process is called as Quantization.

Quantizing an Analog Signal

The analog-to-digital converters perform this type of function to create a series of digital values
out of the given analog signal. The following figure represents an analog signal. This signal to
get converted into digital, has to undergo sampling and quantizing.

Fig 3.16 Analog Signal

The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels. Quantization is representing the sampled values of the amplitude by a finite
set of levels, which means converting a continuous-amplitude sample into a discrete-
time signal. The following figure shows how an analog signal gets quantized. The blue line
represents analog signal while the brown one represents the quantized signal.

45
 Fig 3.17 Quantizing an Analog Signal

Quantization

Both sampling and quantization result in the loss of information. The quality of a Quantizer
output depends upon the number of quantization levels used. The discrete amplitudes of the
quantized output are called as representation levels or reconstruction levels. The spacing
between the two adjacent representation levels is called a quantum or step-size. The below
figure shows the resultant quantized signal which is the digital form for the given analog signal.

Fig 3.18 Stair Case Signal

Resultant Quantized Signal: This is also called as Stair-case waveform, in accordance with its
shape.

Types of Quantization

There are two types of Quantization - Uniform Quantization and Non-uniform Quantization.

The type of quantization in which the quantization levels are uniformly spaced is termed as a
Uniform Quantization. The type of quantization in which the quantization levels are unequal
and mostly the relation between them is logarithmic, is termed as a Non-uniform Quantization.
There are two types of uniform quantization. They are Mid-Rise type and Mid-Tread type. The
following figures represent the two types of uniform quantization.

46
 Fig 3.19 Mid-rise TreadQuantization Types
Figure 3.19 shows the mid-rise type and the mid-tread type of uniform
quantization.

The Mid-Rise type is so called because the origin lies in the middle of a raising part of the
stair-case like graph. The quantization levels in this type are even in number. The Mid-tread
type is so called because the origin lies in the middle of a tread of the stair-case like graph. The
quantization levels in this type are odd in number. Both the mid-rise and mid-tread type of
uniform quantizers are symmetric about the origin.

Quantization Error

For any system, during its functioning, there is always a difference in the values of its input
and output. The processing of the system results in an error, which is the difference of those
values. The difference between an input value and its quantized value is called a Quantization
Error. A Quantizer is a logarithmic function that performs Quantization rounding off the value.
An analog-to-digital converter (ADC) works as a quantizer.

Quantization Noise

It is a type of quantization error, which usually occurs in analog audio signal, while quantizingit
to digital. For example, in music, the signals keep changing continuously, where a regularityis
not found in errors. Such errors create a wideband noise called as Quantization Noise.

Companding in PCM

The word Companding is a combination of Compressing and Expanding, which means that it
does both. This is a non-linear technique used in PCM which compresses the data at the
transmitter and expands the same data at the receiver. The effects of noise and crosstalk are
reduced by using this technique.

47
There are two types of Companding techniques. They are

A-law Companding Technique

 Uniform quantization is achieved at A = 1, where the characteristic

curve is linear andno compression is done.

 A-law has mid-rise at the origin. Hence, it contains a non-zero value.

 A-law companding is used for PCM telephone systems.

µ-law Companding Technique

 Uniform quantization is achieved at µ = 0, where the characteristic

curve is linear andno compression is done.

 µ-law has mid-tread at the origin. Hence, it contains a zero value.

 µ-law companding is used for speech and music signals.

 µ-law is used in North America and Japan.

For the samples that are highly correlated, when encoded by PCM technique, leave redundant
information behind. To process this redundant information and to have a better output, it is a wise
decision to take a predicted sampled value, assumed from its previous output and summarize them

withthe quantized values. Such a process is called as Differential PCM DPCM technique.

DPCM Transmitter
The DPCM Transmitter consists of Quantizer and Predictor with two summer circuits.
Following is the block diagram of DPCM transmitter.

Fig 3.20 DPCM Transmitter

48
 The signals at each point are named as −
 x(nTs) is the sampled input
 x^(nTs) is the predicted sample
 e(nTs) is the difference of sampled input and
predicted output, often called asprediction error
 v(nTs) is the quantized output
 u(nTs) is the predictor input which is actually the
summer output of the predictor outputand the
quantizer output
The predictor produces the assumed samples from the previous outputs of the transmitter
circuit. The input to this predictor is the quantized versions of the input signal x(nTs).

Quantizer Output is represented as

v(nTs) = Q[e(nTs)]

= e(nTs)+q(nTs)

Where q (nTs) is the quantization error

Predictor input is the sum of quantizer output and predictor output,

u(nTs)=x^(nTs)+v(nTs)

u(nTs)=x^(nTs)+e(nTs)+q(nTs)u(nTs) = x(nTs)+q(nTs)

The same predictor circuit is used in the decoder to reconstruct the original input.

DPCM Receiver

The block diagram of DPCM Receiver consists of a decoder, a predictor, and a

summer circuit.Following is the diagram of DPCM Receiver.

49
 Fig 3.21 DPCM Receiver

The notation of the signals is the same as the previous ones. In the absence of noise, the
encoded receiver input will be the same as the encoded transmitter output. As mentioned
before, the predictor assumes a value, based on the previous outputs. The input given to the
decoder is processed and that output is summed up with the output of the predictor, to obtaina
better output.

The sampling rate of a signal should be higher than the Nyquist rate, to achieve better sampling.
If this sampling interval in Differential PCM is reduced considerably, the sample to-sample
amplitude difference is very small, as if the difference is 1-bit quantization, then the step-size
will be very small i.e., Δ delta.
Delta Modulation

The type of modulation, where the sampling rate is much higher and in which the step size
after quantization is of a smaller value Δ, such a modulation is termed as delta modulation.

Features of Delta Modulation

Following are some of the features of delta modulation.

An over-sampled input is taken to make full use of the signal correlation.

The quantization design is simple.

The input sequence is much higher than the Nyquist rate.

The quality is moderate.

The design of the modulator and the demodulator is simple.

50
The stair-case approximation of output waveform.

The step-size is very small, i.e., Δ delta.

The bit rate can be decided by the user.

This involves simpler implementation.

Delta Modulation is a simplified form of DPCM technique, also viewed as 1-bit DPCM
scheme. As the sampling interval is reduced, the signal correlation will be higher.

Delta Modulator

The Delta Modulator comprises of a 1-bit quantizer and a delay circuit along with two summer
circuits. Following is the block diagram of a delta modulator.

Fig 3.22 Delta Modulator

The predictor circuit in DPCM is replaced by a simple delay circuit in DM.From the above

diagram, we have the notations as −

 x(nTs) = over sampled input

 ep(nTs) = summer output and quantizer input
 eq(nTs) = quantizer output = v(nTs)
 x^(nTs) = output of delay circuit
 u(nTs) = input of delay circuit

Using these notations, now we shall try to figure out the process of delta modulation.

51
ep(nTs)=x(nTs)−x^(nTs) ---------------------------- equation 1
=x(nTs)−u([n−1]Ts)
=x(nTs)−[x^[[n−1]Ts]+v[[n−1]Ts]] ------------------------------------- equation 2

Further,

v(nTs)=eq(nTs)=S.sig.[ep(nTs)] --------------------------- equation 3

u(nTs)=x^(nTs)+eq(nTs)Where,

 x^(nTs) = the previous value of the delay circuit

 eq(nTs) = quantizer output = v(nTs)Hence,

u(nTs)=u([n−1] Ts)+v(nTs) ---------------------------- equation 4

Which means,

The present input of the delay unit

=The previous output of the delay unit + the present quantizer outputAssuming zero

condition of Accumulation,

u(nTs) =S∑ sig[ep(jTs)]

j=1 n
Accumulated version of DM output = ∑ v(jTs)-------------------------------- equation 5
j=1

Now, note that

x^(nTs) = u([n−1] Ts)

n−1

= ∑ v(jTs) equation 6

j=1

Delay unit output is an Accumulator output lagging by one sample. From equations 5 & 6, we
get a possible structure for the demodulator. A Stair-case approximated waveform will be the
output of the delta modulator with the step-size as delta (Δ). The output quality of the waveform
is moderate.

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Delta Demodulator

The delta demodulator comprises of a low pass filter, a summer, and a delay circuit. The
predictor circuit is eliminated here and hence no assumed input is given to the demodulator.

Following is the diagram for delta demodulator.

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 Fig 3.23 Delta Demodulator

From the above diagram, we have the notations as −

 v^(nTs) is the input sample

 u^(nTs) is the summer output
 x¯(nTs) is the delayed output

A binary sequence will be given as an input to the demodulator. The stair-case approximated
output is given to the LPF. Low pass filter is used for many reasons, but the prominent reason
is noise elimination for out-of-band signals. The step-size error that may occur at the
transmitter is called granular noise, which is eliminated here. If there is no noise present, then
the modulator output equals the demodulator input.

Advantages of DM Over DPCM

1-bit quantizer

Very easy design of the modulator and the demodulator however, there exists some

noise in DM.

Slope Over load distortion (when Δ is small)

Granular noise (when Δ is large)

Question Bank

Part A

1. Define Pulse Code Modulation?

2. What is line coding?
3. What are the disadvantages of PCM?
4. What are the disadvantages of delta modulation?
5. What is Guard band?
6. Define Aliasing.
7. Define Nyquist rate.

54
 8. What is quantization?
9. Mention the types of digital modulation techniques.

Part B

1. Explain the detection of PWM signal.

2. Explain Frequency division multiplexing and Time division multiplexing
3. Explain Pulse code modulation.
4. Explain the generation of Binary and Quadrature phase shift keying.

55
 IV. Transmitters And Receivers

AM transmitter - High level modulator

Below figure's show the block diagram of high-level and low-level transmitters. The basic
difference between the two transmitters is the power amplification of the carrier and
modulating signals.

Figure (a) shows the block diagram of high-level AM transmitter.

Fig 4.1 Block Diagram of High Level AM Transmitter

Figure (a) is drawn for audio transmission. In high-level transmission, the powers of the carrier
and modulating signals are amplified before applying them to the modulator stage, as shown
in figure (a). In low-level modulation, the powers of the two input signals of the
modulator stage are not amplified. The required transmitting power is obtained from the last
stage of the transmitter, the class C power amplifier.

The various sections of the figure (a) are:

 Carrier oscillator
 Buffer amplifier
 Frequency multiplier
 Power amplifier
 Audio chain
 Modulated class C power amplifier

Carrier oscillator

The carrier oscillator generates the carrier signal, which lies in the RF range. The frequency of
the carrier is always very high. Because it is very difficult to generate high frequencies with
good frequency stability, the carrier oscillator generates a sub multiple with the required carrier
frequency. This sub multiple frequency is multiplied by the frequency multiplier stage to get
the required carrier frequency. Further, a crystal oscillator can be used in this stage to generate
a low frequency carrier with the best frequency stability. The frequency multiplier stage then
increases the frequency of the carrier to its required value.

56
Buffer Amplifier

The purpose of the buffer amplifier is two fold. It first matches the output impedance of the
carrier oscillator with the input impedance of the frequency multiplier, the next stage of the
carrier oscillator. It then isolates the carrier oscillator and frequency multiplier.

This is required so that the multiplier does not draw a large current from the carrier oscillator.
If this occurs, the frequency of the carrier oscillator will not remain stable.

Frequency Multiplier

The sub-multiple frequency of the carrier signal, generated by the carrier oscillator , is now
applied to the frequency multiplier through the buffer amplifier. This stage is also known as
harmonic generator. The frequency multiplier generates higher harmonics of carrier oscillator
frequency. The frequency multiplier is a tuned circuit that can be tuned to the requisite carrier
frequency that is to be transmitted.

Power Amplifier

The power of the carrier signal is then amplified in the power amplifier stage. This is the
basic requirement of a high-level transmitter. A class C power amplifier gives high power
current pulses of the carrier signal at its output.

Audio Chain

The audio signal to be transmitted is obtained from the microphone, as shown in figure (a). The
audio driver amplifier amplifies the voltage of this signal. This amplification is necessary to
drive the audio power amplifier. Next, a class A or a class B power amplifier amplifies the
power of the audio signal.

AM transmitter – Low level modulator

This is the output stage of the transmitter. The modulating audio signal and the carrier signal,
after power amplification, are applied to this modulating stage. The modulation takes place at
this stage. The class C amplifier also amplifies the power of the AM signal to the reacquired
transmitting power. This signal is finally passed to the antenna, which radiates the signal into
space of transmission.

Fig 4.2 Block Diagram of Low Level AM Transmitter

The low-level AM transmitter shown in the figure (b) is similar to a high-level transmitter,
except that the powers of the carrier and audio signals are not amplified. These two signals are
directly applied to the modulated class C power amplifier.

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Modulation takes place at the stage, and the power of the modulated signal is amplified to the
required transmitting power level. The transmitting antenna then transmits the signal.

Coupling of Output Stage and Antenna

The output stage of the modulated class C power amplifier feeds the signal to the transmitting
antenna. To transfer maximum power from the output stage to the antenna it is necessary that
the impedance of the two sections match. For this , a matching network is required. The
matching between the two should be perfect at all transmitting frequencies. As the matching is
required at different frequencies, inductors and capacitors offering different impedance at
different frequencies are used in the matching networks.

The matching network must be constructed using these passive components. This is shown in
below Figure Fig 4.3.

Fig 4.3 Double Pi Matching Network

The matching network used for coupling the output stage of the transmitter and the antenna is
called double π-network. This network is shown in figure (c). It consists of two inductors ,
L1 and L2 and two capacitors, C1 and C2. The values of these components are chosen such that
the input impedance of the network between 1 and 1'. Shown in figure (c) is matched with the
output impedance of the output stage of the transmitter. Further, the output impedance of the
network is matched with the impedance of the antenna.

The double π matching network also filters unwanted frequency components appearing at the
output of the last stage of the transmitter. The output of the modulated class C power amplifier
may contain higher harmonics, such as second and third harmonics, that are highly undesirable.
The frequency response of the matching network is set such that these unwanted higher
harmonics are totally suppressed, and only the desired signal is coupled to the antenna.

preemphasis concept

At the transmitter, the modulating signal is passed through a simple network which amplifies
the high frequency, components more than the low-frequency components. The simplest form
of such a circuit is a simple high pass filter of the type shown in fig (a). Specification dictate a
time constant of 75 microseconds (µs) where t = RC. Any combination of resistor and capacitor
(or resistor and inductor) giving this time constant will be satisfactory. Such a circuit has a
cutoff frequency fco of 2122 Hz. This means that frequencies higher than 2122 Hz will he
linearly enhanced. The output amplitude increases with frequency at a rate of 6 dB per octave.
The pre-emphasis curve is shown in Fig (b). This pre-emphasis circuit increases the energy

58
content of the higher-frequency signals so that they will tend to become stronger than the high
frequency noise components. This improves the signal to noise ratio and increases intelligibility
and fidelity.

Fig 4.4 Pre-emphasis Circuit & Curve

The pre-emphasis circuit also has an upper break frequency fu where the signal enhancement
flattens out.
See Fig (b). This upper break frequency is computed with the expression.
fu = R1 +(R2/2πR1R1C)
It is usually set at some very high value beyond the audio range. An fu of greater than 30KHz
is typical.

FM stereo broad cast transmitter

Fig 4.5 Stereo Broadcast Transmiiter

59
AM Receivers-super heterodyne receiver

Fig 4.6 Superheterodyne AM Receiver

AGC

An Automatic Gain Control (AGC) circuit is a circuit that is designed to maintain a constant
output signal level after amplification, despite variations in signals at the input of the amplifier
or system. This is achieved by providing more amplification to weak signals and less
amplification to strong signals thus maintaining a constant signal amplitude level at the output.

60
 Fig 4.7 Automatic Gain Control

Deemphasis concept

Fig 4.8 De-emphasis Circuit Curve

To return the frequency response to its normal level, a de-emphasis circuit is used at the
receiver. This is a simple low-pass filter with a constant of 75 πs. See figure (c). It features a
cutoff of 2122 Hz and causes signals above this frequency to be attenuated at the rate of 6bB
per octave. The response curve is shown in Fig (d). As a result, the pre-emphasis at the
transmitter is exactly offset by the de-emphasis circuit in the receiver, providing a normal
frequency response. The combined effect of pre-emphasis and de-emphasis is to increase the
high-frequency components during transmission so that they will be stronger and not masked
by noise.

Fig 4.9 Combined Frequency Response

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AFC.
In radio equipment, Automatic Frequency Control (AFC), also called Automatic Fine
Tuning (AFT), is a method or circuit to automatically keep a resonant circuit tuned to
the frequency of an incoming radio signal. It is primarily used in radio receivers to keep the
receiver tuned to the frequency of the desired station.

Fig 4.10 Automatic Frequency Control

Question Bank

Part A

1. Define super heterodyne principle.

2. Define AGC.
3. Define AFC.
4. Mention the use of Pre-emphasis circuit?
5. Mention the use of De-emphasis circuit?

62
 Part B

1. Explain Super hetrodyne AM receiver.

2. Explain the low level and high level AM transmitters.
3. Compose short notes on

Broad Band Communication System

Facsimile system

Fig 5.1 Block Diagram of Fax-Machine

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64
Satellite communication system

In general terms, a satellite is a smaller object that revolves around a larger object in space.
For example, moon is a natural satellite of earth.
We know that Communication refers to the exchange (sharing) of information between two
or more entities, through any medium or channel. In other words, it is nothing but sending,
receiving and processing of information.
If the communication takes place between any two earth stations through a satellite, then it is
called as satellite communication. In this communication, electromagnetic waves are used as
carrier signals. These signals carry the information such as voice, audio, video or any other
data between ground and space and vice-versa.
Soviet Union had launched the world's first artificial satellite named, Sputnik 1 in 1957.
Nearly after 18 years, India also launched the artificial satellite named, Aryabhata in 1975.
Need of Satellite Communication
The following two kinds of propagation are used earlier for communication up to some
distance.
 Ground wave propagation − Ground wave propagation is suitable for frequencies up
to 30MHz. This method of communication makes use of the troposphere conditions of
the earth.
 Sky wave propagation − The suitable bandwidth for this type of communication is
broadly between 30–40 MHz and it makes use of the ionosphere properties of the earth.
The maximum hop or the station distance is limited to 1500KM only in both ground wave
propagation and sky wave propagation. Satellite communication overcomes this limitation. In
this method, satellites provide communication for long distances, which is well beyond the
line of sight.
Since the satellites locate at certain height above earth, the communication takes place
between any two earth stations easily via satellite. So, it overcomes the limitation of
communication between two earth stations due to earth’s curvature.
How a Satellite Works
A satellite is a body that moves around another body in a particular path. A communication
satellite is nothing but a microwave repeater station in space. It is helpful in
telecommunications, radio and television along with internet applications.
A repeater is a circuit, which increases the strength of the received signal and then transmits
it. But, this repeater works as a transponder. That means, it changes the frequency band of the
transmitted signal from the received one.
The frequency with which, the signal is sent into the space is called as Uplink frequency.
Similarly, the frequency with which, the signal is sent by the transponder is called as Downlink
frequency. The following figure illustrates this concept clearly.

65
 Fig 5.5 Satellite Communication

The transmission of signal from first earth station to satellite through a channel is called
as uplink. Similarly, the transmission of signal from satellite to second earth station through a
channel is called as downlink.
Uplink frequency is the frequency at which, the first earth station is communicating with
satellite. The satellite transponder converts this signal into another frequency and sends it
down to the second earth station. This frequency is called as Downlink frequency. In similar
way, second earth station can also communicate with the first one.
The process of satellite communication begins at an earth station. Here, an installation is
designed to transmit and receive signals from a satellite in an orbit around the earth. Earth
stations send the information to satellites in the form of high powered, high frequency (GHz
range) signals.
The satellites receive and retransmit the signals back to earth where they are received by other
earth stations in the coverage area of the satellite. Satellite's footprint is the area which receives
a signal of useful strength from the satellite.

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Pros and Cons of Satellite Communication
In this section, let us have a look at the advantages and disadvantages of satellite
communication.
Following are the advantages of using satellite communication:
 Area of coverage is more than that of terrestrial systems
 Each and every corner of the earth can be covered
 Transmission cost is independent of coverage area
 More bandwidth and broadcasting possibilites
Following are the disadvantages of using satellite communication −
 Launching of satellites into orbits is a costly process.
 Propagation delay of satellite systems is more than that of conventional terrestrial
systems.
 Difficult to provide repairing activities if any problem occurs in a satellite system.
 Free space loss is more
 There can be congestion of frequencies.
Applications of Satellite Communication
Satellite communication plays a vital role in our daily life. Following are the applications of
satellite communication −
 Radio broadcasting and voice communications
 TV broadcasting such as Direct To Home (DTH)
 Internet applications such as providing Internet connection for data transfer, GPS
applications, Internet surfing, etc.
 Military applications and navigations
 Remote sensing applications
 Weather condition monitoring & Forecasting

Electronic Mail
Electronic Mail (e-mail) is one of most widely used services of Internet. This service allows
an Internet user to send a message in formatted manner (mail) to the other Internet user in any
part of world. Message in mail not only contain text, but it also contains images, audio and
videos data. The person who is sending mail is called sender and person who receives mail is
called recipient. It is just like postal mail service.
Components of E-Mail System

The basic components of an email system are User Agent (UA), Message Transfer Agent
(MTA), Mail Box, and Spool file. These are explained as following below.

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User Agent (UA)

The UA is normally a program which is used to send and receive mail. Sometimes, it is called
as mail reader. It accepts variety of commands for composing, receiving and replying to
messages as well as for manipulation of the mailboxes.

Message Transfer Agent (MTA)

MTA is actually responsible for transfer of mail from one system to another. To send a mail,
a system must have client MTA and system MTA. It transfer mail to mailboxes of recipients
if they are connected in the same machine. It delivers mail to peer MTA if destination mailbox
is in another machine. The delivery from one MTA to another MTA is done by Simple Mail
Transfer Protocol.

Fig 5.6 Electronic Mail

Mailbox

It is a file on local hard drive to collect mails. Delivered mails are present in this file. The user
can read it delete it according to his/her requirement. To use e-mail system each user must
have a mailbox. Access to mailbox is only to owner of mailbox.

Spool file

This file contains mails that are to be sent. User agent appends outgoing mails in this file using
SMTP. MTA extracts pending mail from spool file for their delivery. E-mail allows one name,
an alias, to represent several different e-mail addresses. It is known as mailing list, Whenever
user have to sent a message, system checks recipients’s name against alias database. If mailing

68
list is present for defined alias, separate messages, one for each entry in the list, must be
prepared and handed to MTA. If for defined alias, there is no such mailing list is present, name
itself becomes naming address and a single message is delivered to mail transfer entity.
Services provided by E-mail system:
Composition
The composition refer to process that creates messages and answers. For composition any
kind of text editor can be used.
Transfer
Transfer means sending procedure of mail i.e. from the sender to recipient.
Reporting
Reporting refers to confirmation for delivery of mail. It help user to check whether their mail
is delivered, lost or rejected.
Displaying
It refers to present mail in form that is understand by the user.
Disposition
This step concern with recipient that what will recipient do after receiving mail i.e save mail,
delete before reading or delete after reading.
Power Line Carrier Communication

The figure 1 shows a basic PLCC network used in power substations. The Power line carrier
Communication (PLCC) uses the existing power infrastructure for the transmission of data
from sending to receiving end. It works in full duplex mode. PLCC system consists of three
parts:
The terminal assemblies include the receivers transmitters and protective relays.
The coupling equipment is the combination of line tuner, coupling capacitor and the wave or
line trap.
The 50/60 Hz power transmission line serves as path for relaying data in the PLCC bandwidth.

Coupling Capacitor
It forms the physical coupling link between transmission line and the terminal assemblies for
the relaying of carrier signals. Its function is to provide high impedance to power frequency
and low impedance to carrier signal frequencies. They are usually made up of paper or liquid
dielectric system for high voltage application. The ratings of coupling capacitors range from
0.004-0.01µF at 34 kV to 0.0023-0.005µF at 765kV (source: IEEE).
Drain Coil
As shown in the figure 1 the purpose of drain coil is to provide high impedance for carrier
frequency and low impedance for power frequency.
Line Tuner

69
It is connected in series with the coupling capacitor to form a resonant circuit or carrier signal
frequency high pass filter or band pass filter. Its function is to match the impedance of the
PLC terminal with the power line in order to impress the carrier frequency over the power
line. In addition it also provides isolation from power frequency and transient overvoltage
protection.
Line Trap or Wave Trap
It is a parallel L-C tank filter or band-stop filter connected in series with the transmission
line. It presents high impedance to carrier signal frequencies and very low impedance to the
power frequency. It consists of
1. Main coil

An inductor that is connected directly to the high voltage power line carries power
frequency.
2. Tuning device

It may be a capacitor or a combination of capacitor, inductor and resistor, connected

across the main coil in order to tune the line trap to the desired blocking frequency.
3. Protective device

It is usually a gap type surge arrester used to protect the line trap from damage due to
transient over-voltages.
The line trap or wave trap prevents unwanted loss of carrier signal power and also prevents
carrier signal transmission to adjacent power lines. Line traps or wave traps are available for
narrow-band and wide-band carrier frequency blocking applications.

Power Line Channel Characteristics

 Characteristic Impedance

The characteristics impedance of transmission line is given by :

Where, L is the inductance per unit length in Henry(H).

C is the capacitance per unit length in Farad(F).
It varies in the range of 300-800 Ω for power line communication.
 Attenuation

It is measured in decibels(db). Attenuation losses can be due to the impedance

mismatching, resistive losses, coupling losses and various other losses that occur in
the line trap, line tuner, power line etc.

 Noise

The signal-to-noise ratio(SNR) must be high at the receiving end, other wise the
carrier frequency shows erratic patterns at the receiving end. The noise level limit the
attenuation that PLCC channels can tolerate.
 Bandwidth

70
 The wider bandwidth means faster the channel, but it also leads to the accentuation of
noise. For relaying purpose, AM channel bandwidth is around 1000Hz to 1500Hz and
for FSK bandwidth it is 500Hz to 600Hz (source: IEEE).

Applications of PLCC in Power Systems

 Protective Relaying

For the purpose of carrier aided protection, PLCC channels use modulation schemes
namely the Amplitude modulation(AM) for blocking schemes and Frequency Shift
keying(FSK) for unblocking, permissive and direct-trip schemes.

 Telemetry

It is used to monitor electrical quantities like voltage, current, power etc. at remote
locations. The analog data is converted in binary which is used to shift the FSK
frequency HIGH and LOW and then transmitted over narrow band SSB channel.
 Telephony

Voice messages are sent over SSB narrow band mode with bandwidth ~3khz.

 Home Automation and Home Networking

It is classified as low voltage power line communication. Using low voltage electrical
network at home to control appliances by sending or receiving data through power
line. It is used as narrow-band PLCC for home automation and metering purposes,
and broadband PLCC for internet.

71