UNIT-1

AMPLITUDE AND ANGLE MODULATION

Pre-requisite: Basic Knowledge of Electronic Devices, Digital System Design, Analog and Digital
Signals & Systems

Communication can be defined as the process of exchange of information through means such as words,
actions, signs, etc., between two or more individuals.

Parts of a Communication System

Any system, which provides communication consists of the three important and basic parts as shown in the
following figure.

 Sender is the person who sends a message. It could be a transmitting station from where the signal is
transmitted.
 Channel is the medium through which the message signals travel to reach the destination.
 Receiver is the person who receives the message. It could be a receiving station where the transmitted
signal is being received.

Types of Signals

Conveying an information by some means such as gestures, sounds, actions, etc., can be termed as signaling.
Hence, a signal can be a source of energy which transmits some information. This signal helps to establish a
communication between the sender and the receiver.
An electrical impulse or an electromagnetic wave which travels a distance to convey a message, can be termed
as a signal in communication systems.
Depending on their characteristics, signals are mainly classified into two types: Analog and Digital. Analog
and Digital signals are further classified, as shown in the following figure.
Analog Signal

A continuous time varying signal, which represents a time varying quantity can be termed as an Analog
Signal. This signal keeps on varying with respect to time, according to the instantaneous values of the
quantity, which represents it.

Example

Let us consider a tap that fills a tank of 100 liters capacity in an hour (6 AM to 7 AM). The portion of filling
the tank is varied by the varying time. Which means, after 15 minutes (6:15 AM) the quarter portion of the tank
gets filled, whereas at 6:45 AM, 3/4th of the tank is filled.
If we try to plot the varying portions of water in the tank according to the varying time, it would look like the
following figure.
As the result shown in this image varies (increases) according to time, this time varying quantity can be
understood as Analog quantity. The signal which represents this condition with an inclined line in the figure, is
an Analog Signal. The communication based on analog signals and analog values is called as Analog
Communication.

Digital Signal

A signal which is discrete in nature or which is non-continuous in form can be termed as a Digital signal. This
signal has individual values, denoted separately, which are not based on the previous values, as if they are
derived at that particular instant of time.

Example

Let us consider a classroom having 20 students. If their attendance in a week is plotted, it would look like the
following figure.

In this figure, the values are stated separately. For instance, the attendance of the class on Wednesday is 20
whereas on Saturday is 15. These values can be considered individually and separately or discretely, hence
they are called as discrete values.
The binary digits which has only 1s and 0s are mostly termed as digital values. Hence, the signals which
represent 1s and 0s are also called as digital signals. The communication based on digital signals and digital
values is called as Digital Communication.

Periodic Signal

Any analog or digital signal, that repeats its pattern over a period of time, is called as a Periodic Signal. This
signal has its pattern continued repeatedly and is easy to be assumed or to be calculated.

Example
If we consider a machinery in an industry, the process that takes place one after the other is a continuous
procedure. For example, procuring and grading the raw material, processing the material in batches, packing
a load of products one after the other, etc., follows a certain procedure repeatedly.
Such a process whether considered analog or digital, can be graphically represented as follows.

Aperiodic Signal

Any analog or digital signal, that doesn’t repeat its pattern over a period of time is called as Aperiodic Signal.
This signal has its pattern continued but the pattern is not repeated. It is also not so easy to be assumed or to
be calculated.

Example

The daily routine of a person, if considered, consists of various types of work which take different time
intervals for different tasks. The time interval or the work doesn’t continuously repeat. For example, a person
will not continuously brush his teeth from morning to night, that too with the same time period.
Such a process whether considered analog or digital, can be graphically represented as follows.
In general, the signals which are used in communication systems are analog in nature, which are transmitted
in analog or converted to digital and then transmitted, depending upon the requirement.

Signal representation:
The signal representation in communication systems is of two types −

 Frequency domain representation, and

 Time domain representation.
Consider two signals with 1 kHz and 2 kHz frequencies. Both of them are represented in time and frequency
domain as shown in the following figure.
Time domain analysis, gives the signal behavior over a certain time period. In the frequency domain, the
signal is analyzed as a mathematical function with respect to the frequency.Frequency domain representation
is needed where the signal processing such as filtering, amplifying and mixing are done.For instance, if a
signal such as the following is considered, it is understood that noise is present in it.

The frequency of the original signal may be 1 kHz, but the noise of certain frequency, which corrupts this
signal is unknown. However, when the same signal is represented in the frequency domain, using a spectrum
analyzer, it is plotted as shown in the following figure.

Here, we can observe few harmonics, which represent the noise introduced into the original signal. Hence, the
signal representation helps in analyzing the signals.Frequency domain analysis helps in creating the desired
wave patterns. For example, the binary bit patterns in a computer, the Lissajous patterns in a CRO, etc. Time
domain analysis helps to understand such bit patterns.
Modulation Introduction:
For a signal to be transmitted to a distance, without the effect of any external interferences or noise addition
and without getting faded away, it has to undergo a process called as Modulation. It improves the strength of
the signal without disturbing the parameters of the original signal.

What is Modulation?

A message carrying a signal has to get transmitted over a distance and for it to establish a reliable
communication, it needs to take the help of a high frequency signal which should not affect the original
characteristics of the message signal.
The characteristics of the message signal, if changed, the message contained in it also alters. Hence, it is a
must to take care of the message signal. A high frequency signal can travel up to a longer distance, without
getting affected by external disturbances. We take the help of such high frequency signal which is called as
a carrier signal to transmit our message signal. Such a process is simply called as Modulation.
Modulation is the process of changing the parameters of the carrier signal, in accordance with the
instantaneous values of the modulating signal.
Needs for modulation: In order to carry the low frequency message signal to a longer distance,
the high frequency carrier signal is combined with it.

a) Reduction in antennaheight
b) Long distancecommunication
c) Ease ofradiation
d) Multiplexing
e) Improve the quality ofreception
f) Avoid mixing up of othersignals

Frequency modulation: Frequency Modulation is the changing frequency of the carrier signal
with respect to the instantaneous change in message signal.
 Phase modulation: Phase Modulation is defined as changing the phase of the carrier
signal with respect to the instantaneous change in message signal.

Deviation ratio:Deviation ratio is the worst case modulation index and is equal to the
maximum peak frequency deviation divided by the maximum modulating signal frequency.
Mathematically the deviation ratio is DR= f (max)/fm(max).

Amplitude modulation: Amplitude Modulation is defined as changing the amplitude of

the carrier signal with respect to the instantaneous change in message signal.

Carson’s rule: Carson‘s rule states that the bandwidth required to

transmit an angle modulated wave as twice the sum of the peak frequency deviation and the
highest modulating signal frequency. Mathematically carson‘s rule is B=2( f +fm) Hz.

Modulation index: It is defined as ratio of amplitude of the message signal to the

amplitude of the carrier signal. m=Em/Ec.

Percentage modulation: It is the percentage change in the amplitude of the output wave
when the carrier is acted on by a modulating signal. M=(Em/Ec)*100

THEORY OF AMPLITUDEMODULATION
Amplitude Modulation
Amplitude Modulation is the changing the amplitude of the carrier signal with respect to
the instantaneous change in message signal.
The amplitude modulated wave form, its envelope and itsfrequencyspectrum and
bandwidth. Fig (a) Sinosoidal modulation signal (b)High frequency carrier (c) AMsignal.
 Fig:1.2.3 : Frequency domain Representation of AM Wave

Amplitude Modulation of Power distribution:

AM Power Distribution:
This equation relates total power of AM wave to carrier power, Maximum Value of modulation index, m=1 to
avoid distortion. At this value of modulation index, Ptotal = 1.5 Pc. From the above equation we have

Example Problems:
Double Side Band AM:
In the process of Amplitude Modulation, the modulated wave consists of the carrier wave
and two sidebands. The modulated wave has the information only in the
sidebands. Sideband is nothing but a band of frequencies, containing power, which are the
lower and higher frequencies of the carrier frequency.
The transmission of a signal, which contains a carrier along with two sidebands can be
termed as Double Sideband Full Carrier system or simply DSBFC. It is plotted as shown in
the following figure.

However, such a transmission is inefficient. Because, two-thirds of the power is being

wasted in the carrier, which carries no information.
If this carrier is suppressed and the saved power is distributed to the two sidebands, then
such a process is called as Double Sideband Suppressed Carrier system or simply DSBSC.
It is plotted as shown in the following figure.
Mathematical Expressions

Let us consider the same mathematical expressions for modulating and carrier signals as we
have considered in the earlier chapters.
i.e., Modulating signal
m(t)=Amcos(2πfmt)

Carrier signal
c(t)=Accos(2πfct)

Mathematically, we can represent the equation of DSBSC wave as the product of

modulating and carrier signals.
s(t)=m(t)c(t)

⇒s(t)=AmAccos(2πfmt)cos(2πfct)

Bandwidth of DSBSC Wave

We know the formula for bandwidth (BW) is

BW=fmax−fmin

Consider the equation of DSBSC modulated wave.

s(t)=AmAccos(2πfmt)cos(2πfct)

⇒s(t)=(AmAc/2)cos[2π(fc+fm)t]+(AmAc/2)cos[2π(fc−fm)t]

The DSBSC modulated wave has only two frequencies. So, the maximum and minimum
frequencies are fc+fmand fc−fmrespectively.
i.e.,
fmax=fc+fmand fmin=fc−fm

Substitute, fmax and fmin values in the bandwidth formula.

BW=fc+fm−(fc−fm)

⇒BW=2fm

Thus, the bandwidth of DSBSC wave is same as that of AM wave and it is equal to twice the
frequency of the modulating signal.

Power Calculations of DSBSC Wave

• The total power transmitted in DSB-FC

𝑷𝑻 = 𝑷𝑪 + 𝑷𝑼𝑺𝑩 + 𝑷𝑼𝑺𝑩
 𝒎𝟐 𝑽𝒄 𝟐
𝑷𝑻 = 𝑷𝑪 𝟏+ , 𝒘𝒉𝒆𝒓𝒆 𝑷𝑪 =
𝟐 𝟐𝑹

• If the carrier is suppressed, then the total power transmitted in DSB-SC –AM is
𝟐 𝟐
𝒎𝟐 𝑽 𝒄 𝒎𝟐 𝑽 𝒄
𝑷′𝑻 = 𝑷𝑳𝑺𝑩 + 𝑷𝑼𝑺𝑩 = +
𝟖𝑹 𝟖𝑹
𝟐
𝒎 𝟐 𝑽𝒄 𝒎𝟐
= = 𝑷
𝟐 𝟐𝑹 𝟐 𝑪

𝑷𝑻 − 𝑷′𝑻
𝑷𝑫𝑺𝑩 𝑺𝑪 =
𝑷𝑻

𝒎𝟐 𝒎𝟐
𝑷𝑪 𝟏 + − 𝑷𝑪
𝟐 𝟐
=
𝒎𝟐
𝑷𝑪 𝟏 +
𝟐

𝒎𝟐 𝒎𝟐
𝑷𝑪 + 𝑷𝑪 − 𝑷𝑪 𝑷𝑪 𝟏 𝟐
𝟐 𝟐
= = = =
𝒎𝟐 𝒎𝟐 𝒎𝟐 𝟐 + 𝒎𝟐
𝑷𝑪 𝟏 + 𝑷𝑪 𝟏 + 𝟏+
𝟐 𝟐 𝟐

𝟐
The Percentage of Power saving = × 𝟏𝟎𝟎
𝟐 𝒎𝟐

• If modulation index m =1 for 100% modulation. Then, the power saving is

𝟐
𝑷𝑫𝑺𝑩 𝑺𝑪 = × 𝟏𝟎𝟎 = 𝟔𝟔. 𝟕%
𝟑

• In DSB-SC from the total power only 66.7% of power is used for transmission due
to the suppression of the carrier wave.

Single Sideband Suppressed Carrier

The DSBSC modulated signal has two sidebands. Since, the two sidebands carry the same
information, there is no need to transmit both sidebands. We can eliminate one sideband.
The process of suppressing one of the sidebands along with the carrier and transmitting a
single sideband is called as Single Sideband Suppressed Carrier system or simply SSBSC.
It is plotted as shown in the following figure.
In the above figure, the carrier and the lower sideband are suppressed. Hence, the upper
sideband is used for transmission. Similarly, we can suppress the carrier and the upper
sideband while transmitting the lower sideband.
This SSBSC system, which transmits a single sideband has high power, as the power allotted
for both the carrier and the other sideband is utilized in transmitting this Single Sideband.

Mathematical Expressions

Let us consider the same mathematical expressions for the modulating and the carrier
signals as we have considered in the earlier chapters.
i.e., Modulating signal
m(t)=Amcos(2πfmt)
Carrier signal
c(t)=Accos(2πfct)
Mathematically, we can represent the equation of SSBSC wave as
s(t)=AmAc2cos[2π(fc+fm)t] for the upper sideband
Or
s(t)=AmAc2cos[2π(fc−fm)t] for the lower sideband

Bandwidth of SSBSC Wave

We know that the DSBSC modulated wave contains two sidebands and its bandwidth is 2fm.
Since the SSBSC modulated wave contains only one sideband, its bandwidth is half of the
bandwidth of DSBSC modulated wave.
i.e., Bandwidth of SSBSC modulated wave =2fm/2=fm
Therefore, the bandwidth of SSBSC modulated wave is fm and it is equal to the frequency of
the modulating signal.

Power Calculations of SSBSC Wave

• The total power transmitted in DSB-FC

𝑃 =𝑃 +𝑃 +𝑃

𝑚 𝑉
𝑃 =𝑃 1+ , 𝑤ℎ𝑒𝑟𝑒 𝑃 =
2 2𝑅

• If the carrier is suppressed, then the total power transmitted in SSB-SC –AM is

𝑚 𝑉 𝑚 𝑉
𝑃′ = 𝑃 𝑜𝑟 𝑃 = 𝑜𝑟
8𝑅 8𝑅

𝑚 𝑉 𝑚
= = 𝑃
4 2𝑅 4

𝑃 − 𝑃′
𝑃 =
𝑃

𝑃 1+ − 𝑃
=
𝑃 1+

𝑃 +𝑃 − 𝑃 𝑃 +𝑃 𝑃 1+ 1+
= = = =
𝑃 1+ 𝑃 1+ 𝑃 1+ 1+

4+𝑚 2 4+𝑚
𝑃 = × =
4 2+𝑚 4 + 2𝑚

The Percentage of Power saving = × 100

• If modulation index m =1 for 100% modulation. Then, the power saving is

5
𝑃 = × 100 = 83.3%
6

• In SSB-SC from the total power only 83.33% of power is used for transmission due to
the suppression of the carrier wave and Sideband signal.
Advantages

 Bandwidth or spectrum space occupied is lesser than AM and DSBSC waves.

 Transmission of more number of signals is allowed.
 Power is saved.
 High power signal can be transmitted.
 Less amount of noise is present.
 Signal fading is less likely to occur.

Disadvantages

 The generation and detection of SSBSC wave is a complex process.

 The quality of the signal gets affected unless the SSB transmitter and receiver have
an excellent frequency stability.

Applications

 For power saving requirements and low bandwidth requirements.

 In land, air, and maritime mobile communications.
 In point-to-point communications.
 In radio communications.
 In television, telemetry, and radar communications.
 In military communications, such as amateur radio, etc.

VSBSC Modulation
In the previous chapters, we have discussed SSBSC modulation. SSBSC modulated signal
has only one sideband frequency. Theoretically, we can get one sideband frequency
component completely by using an ideal band pass filter. However, practically we may not
get the entire sideband frequency component. Due to this, some information gets lost.
To avoid this loss, a technique is chosen, which is a compromise between DSBSC and
SSBSC. This technique is known as Vestigial Side Band Suppressed Carrier
(VSBSC) technique. The word “vestige” means “a part” from which, the name is derived.
VSBSC Modulation is the process, where a part of the signal called as vestige is modulated
along with one sideband. The frequency spectrum of VSBSC wave is shown in the following
figure.
Along with the upper sideband, a part of the lower sideband is also being transmitted in this
technique. Similarly, we can transmit the lower sideband along with a part of the upper
sideband. 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.

Bandwidth of VSBSC Modulation

We know that the bandwidth of SSBSC modulated wave is fm. Since the VSBSC modulated
wave contains the frequency components of one side band along with the vestige of other
sideband, the bandwidth of it will be the sum of the bandwidth of SSBSC modulated wave
and vestige frequency fv.
i.e., Bandwidth of VSBSC Modulated Wave = fm+fv

Advantages

Following are the advantages of VSBSC modulation.

 Highly efficient.
 Reduction in bandwidth when compared to AM and DSBSC waves.
 Filter design is easy, since high accuracy is not needed.
 The transmission of low frequency components is possible, without any difficulty.
 Possesses good phase characteristics.

Disadvantages

Following are the disadvantages of VSBSC modulation.

 Bandwidth is more when compared to SSBSC wave.
 Demodulation is complex.
Applications

The most prominent and standard application of VSBSC is for the transmission of television
signals. Also, this is the most convenient and efficient technique when bandwidth usage is
considered.
Now, let us discuss about the modulator which generates VSBSC wave and the demodulator
which demodulates VSBSC wave one by one.

Generation of VSBSC

Generation of VSBSC wave is similar to the generation of SSBSC wave. The VSBSC
modulator is shown in the following figure.

In this method, first we will generate DSBSC wave with the help of the product modulator.
Then, apply this DSBSC wave as an input of sideband shaping filter. This filter produces an
output, which is VSBSC wave.
• The modulating signal m(t) and carrier signal Accos(2πfct) are applied as inputs to
the product modulator
• The output of the product modulator is
p(t)=Accos(2πfct)m(t)
• This filter has the input p(t) and the output is VSBSC modulated wave s(t). The
Fourier transforms of p(t) and s(t) are P(t) and S(t) respectively.
• Mathematically, we can write S(f) as
• S(t)=P(f)H(f)
• Substitute P(f) value in the above equation.
• S(f)=Ac2[M(f−fc)+M(f+fc)]H(f)
• The above equation represents the equation of VSBSC frequency spectrum.
Demodulation of VSBSC

Demodulation of VSBSC wave is similar to the demodulation of SSBSC wave. Here, the
same carrier signal (which is used for generating VSBSC wave) is used to detect the
message signal. Hence, this process of detection is called as coherent or synchronous
detection. The VSBSC demodulator is shown in the following figure.

In this process, the message signal can be extracted from VSBSC wave by multiplying it with
a carrier, which is having the same frequency and the phase of the carrier used in VSBSC
modulation. The resulting signal is then passed through a Low Pass Filter. The output of
this filter is the desired message signal.
• Let the VSBSC wave be s(t) and the carrier signal is Accos(2πfct)
• we can write the output of the product modulator as
v(t)=Accos(2πfct)s(t)
• Apply Fourier transform on both sides
V(f)=(Ac/2)[S(f−fc)+S(f+fc)]
• We know that S(f)=(Ac/2)[M(f−fc)+M(f+fc)]H(f)
• From the above equation, let us find S(f−fc) and S(f+fc)
• S(f−fc)=(Ac/2)[M(f−fc−fc)+M(f−fc+fc)]H(f−fc)
⇒S(f−fc)=(Ac/2)[M(f−2fc)+M(f)]H(f−fc)
• S(f+fc)=(Ac/2)[M(f+fc−fc)+M(f+fc+fc)]H(f+fc)
⇒S(f+fc)=Ac/2[M(f)+M(f+2fc)]H(f+fc)
• Substitute, S(f−fc) and S(f+fc) values in V(f)
• V(f)=(Ac/2)[Ac/2[M(f−2fc)+M(f)]H(f−fc)+ (Ac/2)[M(f)+M(f+2fc)]H(f+fc)]
 • ⇒V(f)=(Ac2/4)M(f)[H(f−fc)+H(f+fc)] + (Ac2/4)[M(f−2fc)H(f−fc)+M(f+2fc)H(f+fc)]
• In the above equation, the first term represents the scaled version of the desired
message signal frequency spectrum. It can be extracted by passing the above signal
through a low pass filter.
V0(f)=(Ac2/4)M(f)[H(f−fc)+H(f+fc)]

SUPERHETERODYNE RECEIVER:

A superheterodynereceiver(often shortened to superhet) uses frequency mixing to convert a

received signal to a fixed intermediate frequency (IF) which can be more conveniently
processed than the original radio carrier frequency.

Basic Superheterodyne Block Diagram and Functionality:

The basic block diagram of a basic superheterodyne receiver is shown below. This details the
most basic form of the receiver and serves to illustrate the basic blocks and their function.

The way in which the receiver works can be seen by following the signal as is passes through
the receiver.
· Front end amplifier and tuning block: Signals enter the front end circuitry from the
antenna. This circuit block performs two main functions:

Tuning: Broadband tuning is applied to the RF stage. The purpose of this is to reject the
signals on the image frequency and accept those on the wanted frequency. It must also be
able to track the local oscillator so that as the receiver is tuned, so the RF tuning remains on
the required frequency. Typically the selectivity provided at this stage is not high. Its main
purpose is to reject signals on the image frequency which is at a frequency equal to twice that
of the IF away from the wanted frequency. As the tuning within this block provides all the
rejection for the image response, it must be at a sufficiently sharp to reduce the image to an
acceptable level. However the RF tuning may also help in preventing strong off-channel
signals from entering the receiver and overloading elements of the receiver, in particular the
mixer or possibly even the RF amplifier.

Amplification: In terms of amplification, the level is carefully chosen so that it does not
overload the mixer when strong signals are present, but enables the signals to be amplified
sufficiently to ensure a good signal to noise ratio is achieved. The amplifier must also be a
low noise design. Any noise introduced in this block will be amplified later in the receiver.

· Mixer / frequency translator block: The tuned and amplified signal then enters one
port of the mixer. The local oscillator signal enters the other port. The performance of the
mixer is crucial to many elements of the overall receiver performance. It should eb as linear
as possible. If not, then spurious signals will be generated and these may appear as
'phantom' received signals.

· Local oscillator: The local oscillator may consist of a variable frequency oscillator
that can be tuned by altering the setting on a variable capacitor. Alternatively it may be a
frequency synthesizer that will enable greater levels of stability and setting accuracy.

· Intermediate frequency amplifier, IF block : Once the signals leave the mixer
they enter the IF stages. These stages contain most of the amplification in the receiver as well
as the filtering that enables signals on one frequency to be separated from those on the next.
Filters may consist simply of LC tuned transformers providing inter-stage coupling, or they
may be much higher performance ceramic or even crystal filters, dependent upon what is
required.

· Detector / demodulator stage: Once the signals have passed through the IF stages of
the superheterodyne receiver, they need to be demodulated. Different demodulators are
required for different types of transmission, and as a result some receivers may have a
variety of demodulators that can be switched in to accommodate the different types of
transmission that are to be encountered. Different demodulators used may include:

o AM diode detector: This is the most basic form of detector and this circuit block would
simple consist of a diode and possibly a small capacitor to remove any remaining RF. The
detector is cheap and its performance is adequate, requiring a sufficient voltage to overcome
the diode forward drop. It is also not particularly linear, and finally it is subject to the effects
of selective fading that can be apparent, especially on the HF bands.

Synchronous AM detector: This form of AM detector block is used in where improved

performance is needed. It mixes the incoming AM signal with another on the same frequency
as the carrier. This second signal can be developed by passing the whole signal through a
squaring amplifier. The advantages of the synchronous AM detector are that it provides a far
more linear demodulation performance and it is far less subject to the problems of selective
fading.

o SSB product detector: The SSB product detector block consists of a mixer and a local
oscillator, often termed a beat frequency oscillator, BFO or carrier insertion oscillator, CIO.
This form of detector is used for Morse code transmissions where the BFO is used to create
an audible tone in line with the on-off keying of the transmitted carrier. Without this the
carrier without modulation is difficult to detect. For SSB, the CIO re-inserts the carrier to
make the modulation comprehensible.

Audio amplifier: The output from the demodulator is the recovered audio. This is
passed into the audio stages where they are amplified and presented to the headphones or
loudspeaker.
ANGLE MODULATION:
The expression for frequency deviation, phase deviation and

modulation index in angle modulation.

Narrowband FM and Wideband FM

FM can be divided into Narrowband FM and Wideband FM based on the values of

modulation index ββ.

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.
FM Modulators:
In this chapter, let us discuss about the modulators which generate NBFM and WBFM
waves. First, let us discuss about the generation of NBFM.

Generation of NBFM

We know that the standard equation of FM wave is

s(t)=Accos(2πfct+2πkf∫m(t)dt)

⇒s(t)=Accos(2πfct) cos(2πkf ∫m(t)dt) − Acsin(2πfct)sin(2πkf∫m(t)dt)

For NBFM,
|2πkf∫m(t)dt|<<1
We know that cosθ≈1 and sinθ≈1when θ is very small.
By using the above relations, we will get the NBFM equation as
s(t)=Accos(2πfct)−Acsin(2πfct)2πkf∫m(t)dt
The block diagram of NBFM modulator is shown in the following figure.

Here, the integrator is used to integrate the modulating signal m(t)m(t). The carrier
signal Accos(2πfct) is the phase shifted by −900to get Acsin(2πfct) with the help
of −900phase shifter. The product modulator has two inputs ∫m(t)dtand Acsin(2πfct) It
produces an output, which is the product of these two inputs.

This is further multiplied with 2πkf by placing a block 2πkf in the forward path. The summer
block has two inputs, which are nothing but the two terms of NBFM equation. Positive and
negative signs are assigned for the carrier signal and the other term at the input of the
summer block. Finally, the summer block produces NBFM wave.
Generation of WBFM

The following two methods generate WBFM wave.

 Direct method
 Indirect method

Direct Method

This method is called as the Direct Method because we are generating a wide band FM
wave directly. In this method, Voltage Controlled Oscillator (VCO) is used to generate
WBFM. VCO produces an output signal, whose frequency is proportional to the input signal
voltage. This is similar to the definition of FM wave. The block diagram of the generation of
WBFM wave is shown in the following figure.

Here, the modulating signal m(t)m(t) is applied as an input of Voltage Controlled Oscillator
(VCO). VCO produces an output, which is nothing but the WBFM.
Fi α m(t)

⇒fi=fc+kfm(t)
Where,
fi is the instantaneous frequency of WBFM wave.

Indirect Method

This method is called as Indirect Method because we are generating a wide band FM wave
indirectly. This means, first we will generate NBFM wave and then with the help of
frequency multipliers we will get WBFM wave. The block diagram of generation of WBFM
wave is shown in the following figure.
This block diagram contains mainly two stages. In the first stage, the NBFM wave will be
generated using NBFM modulator. We have seen the block diagram of NBFM modulator at
the beginning of this chapter. We know that the modulation index of NBFM wave is less than
one. Hence, in order to get the required modulation index (greater than one) of FM wave,
choose the frequency multiplier value properly.
Frequency multiplier is a non-linear device, which produces an output signal whose
frequency is ‘n’ times the input signal frequency. Where, ‘n’ is the multiplication factor.
If NBFM wave whose modulation index ββ is less than 1 is applied as the input of frequency
multiplier, then the frequency multiplier produces an output signal, whose modulation index
is ‘n’ times ββ and the frequency also ‘n’ times the frequency of WBFM wave.
Sometimes, we may require multiple stages of frequency multiplier and mixers in order to
increase the frequency deviation and modulation index of FM wave.

FM Demodulators:
In this chapter, let us discuss about the demodulators which demodulate the FM wave. The
following two methods demodulate FM wave.

 Frequency discrimination method

 Phase discrimination method

Frequency Discrimination Method

We know that the equation of FM wave is

s(t)=Accos(2πfct+2πkf∫m(t)dt)
Differentiate the above equation with respect to 't'.
ds(t)dt=−Ac(2πfc+2πkfm(t))sin(2πfct+2πkf∫m(t)dt)
We can write, −sinθas sin(θ−1800)
⇒ds(t)dt=Ac(2πfc+2πkfm(t))sin(2πfct+2πkf∫m(t)dt−1800)
⇒ds(t)dt=Ac(2πfc)[1+(kfkc)m(t)]sin(2πfct+2πkf∫m(t)dt−1800)
In the above equation, the amplitude term resembles the envelope of AM wave and the angle
term resembles the angle of FM wave. Here, our requirement is the modulating
signal m(t)m(t). Hence, we can recover it from the envelope of AM wave.
The following figure shows the block diagram of FM demodulator using frequency
discrimination method.

This block diagram consists of the differentiator and the envelope detector. Differentiator is
used to convert the FM wave into a combination of AM wave and FM wave. This means, it
converts the frequency variations of FM wave into the corresponding voltage (amplitude)
variations of AM wave. We know the operation of the envelope detector. It produces the
demodulated output of AM wave, which is nothing but the modulating signal.

Phase Discrimination Method

The following figure shows the block diagram of FM demodulator using phase
discrimination method.

This block diagram consists of the multiplier, the low pass filter, and the Voltage Controlled
Oscillator (VCO). VCO produces an output signal v(t), whose frequency is proportional to
the input signal voltage d(t). Initially, when the signal d(t) is zero, adjust the VCO to
produce an output signal v(t), having a carrier frequency and −90 0 phase shift with respect
to the carrier signal.
FM wave s(t) and the VCO output v(t) are applied as inputs of the multiplier. The multiplier
produces an output, having a high frequency component and a low frequency component.
Low pass filter eliminates the high frequency component and produces only the low
frequency component as its output. This low frequency component contains only the term-
related phase difference. Hence, we get the modulating signal m(t) from this output of the
low pass filter.

Course Outcomes:
At the end of the unit the students will be able to
comparedifferentanalogmodulationschemesfortheirefficiencyandbandwidth.
 UNIT-2
INFORMATION THEORY AND NOISE

Pre-requisite: Basic Knowledge of discrete signal, continuous wave modulation and

demodulation, Information theory and noise signal

Information Theory:

Information is the source of a communication system, whether it is analog or

digital. Information theory is a mathematical approach to the study of coding of
information along with the quantification, storage, and communication of information.

Conditions of Occurrence of Events

If we consider an event, there are three conditions of occurrence.

 If the event has not occurred, there is a condition of uncertainty.
 If the event has just occurred, there is a condition of surprise.
 If the event has occurred, a time back, there is a condition of having
some information.
These three events occur at different times. The difference in these conditions help us gain
knowledge on the probabilities of the occurrence of events.
ENTROPY:
Entropy is the average amount of information contained in each message received.
Here, message stands for an event, sample or character drawn from a distribution or data
stream. Entropy thus characterizes our uncertainty about our source of information. (Entropy
is best understood as a measure of uncertainty rather than certainty as entropy is larger for
more random sources.) The source is also characterized by the probability distribution of the
samples drawn from it.
Formula for entropy:
Information strictly in terms of the probabilities of events. Therefore, let us suppose that we
have a set of probabilities (a probability distribution) P = {p1, p2, . . . ,pn}. We
define entropy of the distribution P by

. Shannon defined the entropy of the a discrete random variable X with possible values {x1,
..., xn} and probability mass function P(X) as: Here E is the expected value operator, and I is
the information content of X. I(X) is itself a random variable. One may also define the
conditional entropy of two events X and Y taking values xi and yj respectively, as
 Properties:
If X and Y are two independent experiments, then knowing the value of Y doesn't
influence our knowledge of the value of X (since the two don't influence each other by
independence):

· The entropy of two simultaneous events is no more than the sum of the entropies of each
individual event, and are equal if the two events are independent. More specifically,
ifX and Y are two random variables on the same probability space, and (X,Y) denotes their
Cartesian product, then

Channel Capacity

We have so far discussed mutual information. The maximum average mutual information, in
an instant of a signaling interval, when transmitted by a discrete memoryless channel, the
probabilities of the rate of maximum reliable transmission of data, can be understood as
the channel capacity.
It is denoted by C and is measured in bits per channel use.

Discrete Memoryless Source

A source from which the data is being emitted at successive intervals, which is independent
of previous values, can be termed as discrete memoryless source.
This source is discrete as it is not considered for a continuous time interval, but at discrete
time intervals. This source is memoryless as it is fresh at each instant of time, without
considering the previous values.
DISCRETE MEMORYLESS CHANNEL:
1.We denote (X , p(y|x), Y) a discrete channel, where X and Y are finite sets and for the collection of
probability mass functions p(y|x) (one for each x ∈ X )the following holds: ∀x, y : p(y|x) ≥ 0 and ∀x : X
y∈Y p(y|x) = 1.

2. We call (X n, p(y n|x n), Y n) the n-th extension of the discrete channel, where p(yk|xk) = p(yk|x k ,
yk−1 ). For a memoryless channel yk does not depend on the previous inputs x k−1 nor the previous
Outputs y k−1 . Therefore p(y n|x n) = Yn i=1 p(yi |xi).

In the following channel is meant to be discrete and memoryless.

Capacity of a noisy channel

Examples
Ø
All these transition probabilities from xi to yj are gathered in a transition matrix.
Ø
The (i ; j) entry of the matrix is P(Y = yj /jX = xi ), which is called forward transition
probability.
Ø
In DMC the output of the channel depends only on the input of the channel at the same
instant and not on the input before or after.
Ø
The input of a DMC is a RV (random variable) X who selects its value from a discrete
limited set X.
Ø
The cardinality of X is the number of the point in the used constellation.
Ø
In an ideal channel, the output is equal to the input.
Ø
In a non-ideal channel, the output can be different from the input with a given probability.
Transmission rate:
Ø
H(X) is the amount of information per symbol at the input of the channel.
Ø
H(Y ) is the amount of information per symbol at the output of the channel.
Ø
H(XjY ) is the amount of uncertainty remaining on X knowing Y .
Ø
The information transmission is given by:I (X; Y ) = H(X) − H(XjY ) bits/channel use
Ø
For an ideal channel X = Y , there is no uncertainty over X when we observe Y . So all the
information is transmitted for each channel use: I (X;Y ) = H(X)
Ø
If the channel is too noisy, X and Y are independent. So the uncertainty over X remains the
same knowing or not Y , i.e. no information passes through the channel: I (X; Y ) = 0.
Hard and soft decision:
Ø
Normally the size of constellation at the input and at the output are the same, i.e., jXj = jYj
Ø
In this case the receiver employs hard-decision decoding.
Ø
It means that the decoder makes a decision about the transmitted symbol.
Ø
It is possible also that jXj 6= jY j.
Ø
In this case the receiver employs a soft-decision.

Channel models and channel capacity:

1. The encoding process is a process that takes a k information bits at a time and maps
each k-bit sequence into a unique n-bit sequence. Such an n-bit sequence is called a code
word.
2. The code rate is defined as k/n.
3. If the transmitted symbols are M-ary (for example, M levels), and at the receiver the
output of the detector, which follows the demodulator, has an estimate of the transmitted data
symbol with
(a). M levels, the same as that of the transmitted symbols, then we say the detector has made
a hard decision;
(b). Q levels, Q being greater than M, then we say the detector has made a soft decision.
Channels models:
1. Binary symmetric channel (BSC):

If (a) the channel is an additive noise channel, and (b) the modulator and
demodulator/detector are included as parts of the channel. Furthermore, if the modulator
employs binary waveforms, and the detector makes hard decision, then the channel has a
discrete-time binary input sequence and a discrete-time binary output sequence.
Note that if the channel noise and other interferences cause statistically independent errors
in the transmitted binary sequence with average probability p, the channel is called a BSC.
Besides, since each output bit from the channel depends only upon the corresponding input
bit, the channel is also memoryless.
2. Discrete memoryless channels (DMC):
A channel is the same as above, but with q-ary symbols at the output of the channel encoder,
and Q-ary symbols at the output of the detector, where Q ³ q . If the channel and the
modulator are memoryless, then it can be described by a set of qQ conditional probabilities

P (Y = y i | X = x j ) º P ( y i | x j ), i = 0,1,...,Q - 1; j = 0,1,..., q -1

Such a channel is called discrete memory channel (DSC).

If the input to a DMC is a sequence of n symbols u1 , u2 ,..., un selected from the alphabet X
and the corresponding output is the sequence v1 , v 2 ,..., vn of symbols from the alphabet Y,
the joint conditional probability is

the probability transition matrix for the channel.

3. Discrete-input, continuous-output channels:

Suppose the output of the channel encoder has q-ary symbols as above, but the output of the
detector is unquantized (Q = ¥) . The conditional probability density functions

p ( y | X = x k ), k = 0,1, 2,..., q -1

AWGN is the most important channel of this type.

Y=X+G

For any given sequence X i , i = 1, 2,..., n , the corresponding output is Yi , i = 1, 2,..., n

Yi = X i + Gi , i = 1, 2,..., n

If, further, the channel is memoryless, then the joint conditional pdf of the detector‘s output is

4. Waveform channels:
If such a channel has bandwidth W with ideal frequency response C ( f ) = 1 , and if the
bandwidth-limited input signal to the channel is x ( t) , and the output signal, y ( t) of the
channel is corrupted by AWGN, then
y ( t ) = x ( t ) + n ( t)

The channel can be described by a complete set of orthonormal functions:

Since { ni } are uncorrelated and are Gaussian, therefore, statistically independent. So

Channel Capacity:
Channel model: DMC
Input alphabet: X = {x0 , x1 , x 2 ,..., xq-1}
Output alphabet: Y = {y 0 , y1 , y 2 ,..., yq-1}
Suppose x j is transmitted, yi is received, then
The mutual information (MI) provided about the event {X = x j } by the occurrence of the
event

Hence, the average mutual information (AMI) provided by the output Y about the input X is

To maximize the AMI, we examine the above equation:

(1). P ( y i) represents the jth output of the detector;

(2). P ( y i | x j ) represents the channel characteristic, on which we cannot do anything;
(3). P ( x j ) represents the probabilities of the input symbols, and we may do something or
control them. Therefore, the channel capacity is defined by

Hartley Shannon Law

In information theory, the Shannon–Hartley theorem tells the maximum rate at which
information can be transmitted over a communications channel of a specified bandwidth in
the presence of noise.

The Shannon-Hartley theorem states that the channel capacity is given by

C = B log2 (1 + S/N)
where C is the capacity in bits per second,
B is the bandwidth of the channel in Hertz, and
S/N is the signal-to-noise ratio.

Source coding theorem

In information theory, the source coding theorem informally states that
N random variables each with entropy H(X) can be compressed into more
than N H(X) bits with negligible risk of information loss, as N → ∞; but conversely, if they
are compressed into fewer than N H(X) bits it is virtually certain that information will be lost.

SHANNON-FANO CODING:
This is a basic information theoretic algorithm. A simple example will be used to illustrate
the algorithm:

Encoding for the Shannon-Fano Algorithm:

A top-down approach
1. Sort symbols according to their frequencies/probabilities, e.g., ABCDE.
2. Recursively divide into two parts, each with approx. same number of counts.
Procedure for shannonfano algorithm:
A Shannon–Fano tree is built according to a specification designed to define an effective
code table. The actual algorithm is simple:
1. For a given list of symbols, develop a corresponding list of probabilities or frequency
counts so that each symbol‘s relative frequency of occurrence is known.
2. Sort the lists of symbols according to frequency, with the most frequently occurring
symbols at the left and the least common at the right.
3. Divide the list into two parts, with the total frequency counts of the left part being as
close to the total of the right as possible.
 4. The left part of the list is assigned the binary digit 0, and the right part is assigned the
digit 1. This means that the codes for the symbols in the first part will all start with 0, and the
codes in the second part will all start with 1.
5. Recursively apply the steps 3 and 4 to each of the two halves, subdividing groups and
adding bits to the codes until each symbol has become a corresponding code leaf on the tree.

Tree diagram:

Table For Shannon Fano Coding:

HUFFMAN CODING:
 The Shannon–Fano algorithm doesn't always generate an optimal code. In 1952, David A.
Huffman gave a different algorithm that always produces an optimal tree for any given
probabilities. While the Shannon–Fano tree is created from the root to the leaves, the
Huffman
Procedure for Huffman Algorithm:
1. Create a leaf node for each symbol algorithm works from leaves to the root in the
opposite direction and add it to frequency of occurrence.
2. While there is more than one node in the queue:
· Remove the two nodes of lowest probability or frequency from the queue
· Prepend 0 and 1 respectively to any code already assigned to these nodes
· Create a new internal node with these two nodes as children and with probability
equal to the sum of the two nodes' probabilities.
· Add the new node to the queue.
3. The remaining node is the root node and the tree is complete.
Tree diagram:
Pre-emphasis and De-emphasis
emphasis
As we already know that in FM, the noise has a greater effect on the higher modulating
frequencies. This effect can be reduced by increasing the value of modulation index (mf ) for
higher modulating frequencies (fm).This can be done by increasing the deviation ΔfΔ and Δf
can be increased by increasing the amplitude of modulating signal at higher modulating
frequencies.
Thus, if we boost the amplitude of higher frequency modulating signals artificially then it will
be possible to improve the noise immunity at higher
higher modulating frequencies. The artificial
boosting of higher modulating frequencies is called pre-emphasis.
pre

The process that is used at the receiver end to nullify or compensate the artificial boosting
given to the higher modulating frequencies in the proce
process of pre-emphasis
emphasis is called De-
De
emphasis.That means, the artificially boosted high frequency signals are brought to their
original amplitude using the de-emphasis
emphasis circuit.

Pre-emphasis: The noise suppression ability of FM decreases with the increase in the
frequencies. Thus increasing the relative strength or amplitude of the high frequency
components of the message signal before modulation is termed as PrePre-emphasis.
emphasis. The Fig3
below shows the circuit of pre-emphasis.
emphasis.

Fig3. Pre-emphasis circuit

 De-emphasis: In the de-emphasis

emphasis circuit, by reducing the amplitude level of the
received high frequency signal by the same amount as the increase in pre-emphasis
pre is
termed as De-emphasis.
emphasis. The Fig4. below shows the circuit of de-emphasis.
de emphasis.
 Fig4. De-emphasis circuit

 The pre-emphasis process is done at the transmitter side, while the de-emphasis
process is done at the receiver side.
 Thus a high frequency modulating signal is emphasized or boosted in amplitude in
transmitter before modulation. To compensate for this boost, the high frequencies are
attenuated or de-emphasized in the receiver after the demodulation has been
performed. Due to pre-emphasis and de-emphasis, the S/N ratio at the output of
receiver is maintained constant.
 The de-emphasis process ensures that the high frequencies are returned to their
original relative level before amplification.
 Pre-emphasis circuit is a high pass filter or differentiator which allows high
frequencies to pass, whereas de-emphasis circuit is a low pass filter or integrator
which allows only low frequencies to pass.

NOISE:
Noise is random, undesirable electrical energy that enters the communications system via the
communicating medium and interferes with the transmitted message. However, some noise is
also produced in the receiver.
(OR)
With reference to an electrical system, noise may be defined as any unwanted form of energy
which tends to interfere with proper reception and reproduction of wanted signal.
Noise may be put into following two categories.
1. External noises, i.e. noise whose sources are external. External noise may be classified
into the following three types:
· Atmospheric noises
· Extraterrestrial noises
· Man-made noises or industrial noises.
2. Internal noise in communication, i.e. noises which get, generated within the receiver
or communication system. Internal noise may be put into the following four categories.
· Thermal noise or white noise or Johnson noise
· Shot noise.
· Transit time noise
· Miscellaneous internal noise.
External noise cannot be reduced except by changing the location of the receiver or the
entire system. Internal noise on the other hand can be easily evaluated mathematically and
can be reduced to a great extent by proper design. As already said, because of the fact that
internal noise can be reduced to a great extent, study of noise characteristics is a very
important part of the communication engineering.
1. Explanation of External Noise
· Atmospheric Noise:
Atmospheric noise or static is caused by lighting discharges in thunderstorms and other
natural electrical disturbances occurring in the atmosphere. These electrical impulses are
random in nature. Hence the energy is spread over the complete frequency spectrum used for
radio communication.
Atmospheric noise accordingly consists of spurious radio signals with components spread
over a wide frequency range. These spurious radio waves constituting the noise get
propagated over the earth in the same fashion as the desired radio waves of the same
frequency. Accordingly at a given receiving point, the receiving antenna picks up not only the
signal but also the static from all the thunderstorms, local or remote.
Extraterrestrial noise:
· Solar noise
· Cosmic noise
Solar noise:
This is the electrical noise emanating from the sun. Under quite conditions, there is a steady
radiation of noise from the sun. This results because sun is a large body at a very high
temperature (exceeding 6000°c on the surface), and radiates electrical energy in the form of
noise over a very wide frequency spectrum including the spectrum used for radio
communication. The intensity produced by the sun varies with time. In fact, the sun has a
repeating 11-year noise cycle. During the peak of the cycle, the sun produces some amount of
noise that causes tremendous radio signal interference, making many frequencies unusable
for communications. During other years. The noise is at a minimum level.
Cosmic noise:
Distant stars are also suns and have high temperatures. These stars, therefore, radiate noise
in the same way as our sun. The noise received from these distant stars is thermal noise (or
black body noise) and is distributing almost uniformly over the entire sky. We also receive
noise from the center of our own galaxy (The Milky Way) from other distant galaxies and
from other virtual point sources such as quasars and pulsars.
Ø
Man-Made Noise (Industrial Noise):
By man-made noise or industrial- noise is meant the electrical noise produced by such
sources as automobiles and aircraft ignition, electrical motors and switch gears, leakage
from high voltage lines, fluorescent lights, and numerous other heavy electrical machines.
Such noises are produced by the arc discharge taking place during operation of these
machines. Such man-made noise is most intensive in industrial and densely populated areas.
Man-made noise in such areas far exceeds all other sources of noise in the frequency range
extending from about 1 MHz to 600 MHz.
2. Explanation of Internal Noise in communication:
· Thermal Noise:
Conductors contain a large number of 'free" electrons and "ions" strongly bound by
molecular forces. The ions vibrate randomly about their normal (average) positions,
however, this vibration being a function of the temperature. Continuous collisions between
the electrons and the vibrating ions take place. Thus there is a continuous transfer of energy
between the ions and electrons. This is the source of resistance in a conductor. The movement
of free electrons constitutes a current which is purely random in nature and over a long time
averages zero. There is a random motion of the electrons which give rise to noise voltage
called thermal noise.Thus noise generated in any resistance due to random motion of
electrons is called thermal noise or white or Johnson noise. The analysis of thermal noise is
based on the Kinetic theory. It shows that the temperature of particles is a way of expressing
its internal kinetic energy. Thus "Temperature" of a body can be said to be equivalent to the
statistical rms value of the velocity of motion of the particles in the body. At -273°C (or zero
degree Kelvin) the kinetic energy of the particles of a body becomes zero .Thus we can relate
the noise power generated by a resistor to be proportional to its absolute temperature. Noise
power is also proportional to the bandwidth over which it is measured. From the above
discussion we can write down.

Where Pn = Maximum noise power output of a resistor.

K = Boltzmann‘s constant = 1.38 x10-23 joules I Kelvin.

T = Absolute temperature.
B = Bandwidth over which noise is measured.
· Transit Time Noise:
Another kind of noise that occurs in transistors is called transit time noise.Transit time is the
duration of time that it takes for a current carrier such as a hole or current to move from the
input to the output.The devices themselves are very tiny, so the distances involved are
minimal. Yet the time it takes for the current carriers to move even a short distance is finite.
At low frequencies this time is negligible. But when the frequency of operation is high and the
signal being processed is the magnitude as the transit time, then problem can occur. The
transit time shows up as a kind of random noise within the device, and this is directly
proportional to the frequency of operation.
· Miscellaneous Internal Noises Flicker Noise:
Flicker noise or modulation noise is the one appearing in transistors operating at low audio
frequencies. Flicker noise is proportional to the emitter current and junction temperature.
However, this noise is inversely proportional to the frequency. Hence it may be neglected at
frequencies above about 500 Hz and it, Therefore, possess no serious problem.
· Transistor Thermal Noise:
Within the transistor, thermal noise is caused by the emitter, base
and collector internal resistances. Out of these three regions, the base region contributes
maximum thermal noise.
· Partition Noise:
 Partition noise occurs whenever current has to divide between two or more paths, and
results from the random fluctuations in the division. It would be expected, therefore, that a
diode would be less noisy than a transistor (all other factors being equal) If the third
electrode draws current (i.e.., the base current). It is for this reason that the inputs of
microwave receivers are often taken directly to diode mixers.
· Shot Noise:
The most common type of noise is referred to as shot noise which is produced by the random
arrival of 'electrons or holes at the output element, at the plate in a tube, or at the collector
or drain in a transistor. Shot noise is also produced by the random movement of electrons or
holes across a PN junction. Even through current flow is established by external bias
voltages, there will still be some random movement of electrons or holes due to
discontinuities in the device. An example of such a discontinuity is the contact between the
copper lead and the semiconductor materials. The interface between the two creates a
discontinuity that causes random movement of the current carriers.
3. Signal to Noise Ratio:
Noise is usually expressed as a power because the received signal is also expressed in terms
of power. By Knowing the signal to noise powers the signal to noise ratio can be computed.
Rather than express the signal to noise ratio as simply a number, you will usually see it
expressed in terms of decibels.

A receiver has an input signal power of l.2µW. The noise power is 0.80µW. The signal to
noise ratio is
Signal to Noise Ratio = 10 Log (1.2/0.8)
= 10 log 1.5
= 10 (0.176)
= 1.76 Db

4. Noise Figure:
Noise Figure is designed as the ratio of the signal-to-noise power at the input to the signal to
noise power at the output.The device under consideration can be the entire receiver or a
single amplifier stage. The noise figure also called the noise factor can be computed with the
expression , F = Signal to Noise power Input/Signal to noise power output .You can express
the noise figure as a number, more often you will see it expressed in decibels.

NARROW BAND NOISE:

1. Definition: A random process X(t) is bandpass or narrowband random process if its

power spectral density SX(f) is nonzero only in a small neighborhood of some high frequency
fc Deterministic signals: defined by its Fourier transform Random processes: defined by its
power spectral density.
Notes:
1. Since X(t) is band pass, it has zero mean: E[(X(t)] = 0.
2. fc needs not be the center of the signal bandwidth, or in the signal bandwidth at all.
2. Narrowband Noise Representation:
In most communication systems, we are often dealing with band-pass filtering of signals.
Wideband noise will be shaped into bandlimited noise. If the bandwidth of the bandlimited
noise is relatively small compared to the carrier frequency, we refer to this as narrowband
noise. We can derive the power spectral density Gn(f) and the auto-correlation function
Rnn(τ) of the narrowband noise and use them to analyse the performance of linear systems.
In practice, we often deal with mixing (multiplication), which is a non-linear operation, and
the system analysis becomes difficult. In such a case, it is useful to express the narrowband
noise as n(t) = x(t) cos 2πfct - y(t) sin 2πfct.
 where fc is the carrier frequency within the band occupied by the noise. x(t) and y(t) are
known as the quadrature components of the noise n(t). The Hibert transform of n(t) is n^ (t) =
H[n(t)] = x(t) sin 2πfct + y(t) cos 2πfct.
· Generation of quadrature components of n(t).
x(t) and y(t) have the following properties:
1. E[x(t) y(t)] = 0. x(t) and y(t) are uncorrelated with each other.
2. x(t) and y(t) have the same means and variances as n(t).
3. If n(t) is Gaussian, then x(t) and y(t) are also Gaussian.
4. x(t) and y(t) have identical power spectral densities, related to the power spectral density
of n(t) by Gx(f) = Gy(f) = Gn(f- fc) + Gn(f+ fc) (28.5)
for fc - 0.5B < | f | < fc + 0.5B and B is the bandwidth of n(t).
3. Inphase and Quadrature Components:
In-Phase & Quadrature Sinusoidal Components

From this we may conclude that every sinusoid can be expressed as the sum of a sine function
phase zero) and a cosine function (phase 2). If the sine part is called the ``in-phase''
component, the cosine part can be called the ``phase-quadrature'' component. In general,
``phase quadrature'' means ``90 degrees out of phase,'' i.e., a relative phase shift of ± 2. It is
also the case that every sum of an in-phase and quadrature component can be expressed as a
single sinusoid at some amplitude and phase.

FM THRESHOLD EFFECT:

In an FM receiver, the effect produced when the desired-signal gain begins to limit the desired
signal, and thus noise limiting (suppression). Note: FM threshold effect occurs at (and above) the
point at which the FM signal-to-noise improvement is measured. The output signal to noise ratio
of FM receiver is valid only if the carrier to noise ratio is measured at the discriminator input is
 high compared to unity. It is observed that as the input noise is increased so that the carrier to
noise ratio decreased, the FM receiver breaks. At first individual clicks are heard in the receiver
output and as the carrier to noise ratio decreases still further, the clicks rapidly merge in to a
crackling or sputtering sound. Near the break point eqn8.50 begins to fail predicting values of
output SNR larger than the actual ones. This phenomenon is known as the threshold effect.The
threshold effect is defined as the minimum carrier to noise ratio that gives the output SNR not
less than the value predicted by the usual signal to noise formula assuming a small noise power.
For a qualitative discussion of the FM threshold effect, Consider, when there is no signal present,
so that the carrier is unmodulated. Then the composite signal at the frequency discriminator
input is

Where nI(t) and nQ(t) are inphase and quadrature component of the narrow band noise n(t)
with respect to carrier wave Accos2 fct. The phasor diagram of fig8.17 below shows the
phase relations b/n the various components of x(t) in eqn (1).This effect is shown in fig
below, this calculation is based on the following two assumptions:

1. The output signal is taken as the receiver output measured in the absence of noise. The
average output signal poweris calculated for a sinusoidal modulation that produces a
frequency deviation Iequal to 1/2 of the IF filter bandwidth B, The carrier is thus enabled
to swing back and forth across the entire IF band.

2. The average output noise power is calculated when there is no signal present, i.e.,the carrier
is unmodulated, with no restriction placed on the value of the carrier to noise ratio.

Assumptions:

Course Outcomes:
At the end of the unit the students will be able to -Analyze the behavior of
communication systems in the presence ofnoise
 UNIT-3
PULSE MODULATION
Pre-requisite: Basic Knowledge of discrete signal, modulation and demodulation.

SAMPLING:
A message signal may originate from a digital or analog source. If the message signal is
analog in nature, then it has to be converted into digital form before it can transmit by digital
means. The process by which the continuous-time signal is converted into a discrete–time
signal is called Sampling. Sampling operation is performed in accordance with the sampling
theorem.
SAMPLING THEOREM FOR LOW-PASS SIGNALS:-
Statement: - “If a band –limited signal g(t) contains no frequency components for ‫׀‬f‫ > ׀‬W,
then it is completely described by instantaneous values g(kTs) uniformly spaced in time with
period Ts ≤ 1/2W. If the sampling rate, fs is equal to the Nyquist rate or greater (fs ≥ 2W), the
signal g(t) can be exactly reconstructed.

Proof:-
Part - I If a signal x(t) does not contain any frequency component beyond W Hz, thenthe
signal is completely described by its instantaneous uniform samples with sampling interval
(or period ) of Ts< 1/(2W) sec.
Part – II The signal x(t) can be accurately reconstructed (recovered) from the set of uniform
instantaneous samples by passing the samples sequentially through an ideal (brick-wall)
lowpass filter with bandwidth B, where W ≤ B <fs – W and fs = 1/(Ts).
If x(t) represents a continuous-time signal, the equivalent set of instantaneous uniform
samples {x(nTs)} may be represented as,

where x(nTs) = x(t)⎢t =nTs , δ(t) is a unit pulse singularity function and „n‟ is an integer.The
continuous-time signal x(t) is multiplied by an (ideal) impulse train to obtain {x(nTs)} and
can be rewritten as,

Now, let X(f) denote the Fourier Transform F(T) of x(t), i.e.

Now, from the theory of Fourier Transform, we know that the F.T of Σ δ(t- nTs), the impulse
train in time domain, is an impulse train in frequency domain:

If Xs(f) denotes the Fourier transform of the energy signal xs(t), we can write using Eq.
(1.2.4) and the convolution property:
Xs(f) = X(f)* F{Σ δ(t- nTs)}
X(f)*[fs.Σ δ(f- nfs)]
= fs.X(f)*Σ δ(f- nfs)
Aliasing and signal reconstruction:
Nyquist‟s theorems as stated above and also helps to appreciate their practical implications.

Let us note that while writing Eq.(1.4), we assumed that x(t) is an energy signal so that its
Fourier transform exists. With this setting, if we assume that x(t) has no appreciable
frequency component greater than W Hz and if fs> 2W, then Eq.(1.4) implies that Xs(f), the
Fourier Transform of the sampled signal Xs(t) consists of infinite number of replicas of X(f),
centered at discrete frequencies n.fs, -∞ < n < ∞ and scaled by a constant fs= 1/Ts.

Fig. 1.2.1 indicates that the bandwidth of this instantaneously sampled wave xs(t) is
infinitewhile the spectrum of x(t) appears in a periodic manner, centered at discrete frequency
values n.fs. Part – I of the sampling theorem is about the condition fs> 2.W i.e. (fs – W) > W
and (– fs + W) < – W. As seen from Fig. 1.2.1, when this condition is satisfied, the spectra of
xs(t), centered at f = 0 and f = ± fs do not overlap and hence, the spectrum of x(t) is present in
xs(t) without any distortion. This implies that xs(t), the appropriately sampled version of x(t),
contains all information about x(t) and thus represents x(t).
The second part suggests a method of recovering x(t) from its sampled version xs(t) by using
an ideal lowpass filter. As indicated by dotted lines in Fig. 1.2.1, an ideal lowpass filter (with
brick-wall type response) with a bandwidth W ≤ B < (fs – W), when fed with xs(t), will allow
the portion of Xs(f), centered at f = 0 and will reject all its replicas at f = n fs, for n ≠ 0. This
implies that the shape of the continuous time signal xs(t), will be retained at the output of the
ideal filter.
Quantization:
The process of transforming Sampled amplitude values of a message signal into a discrete
amplitude value is referred to as Quantization. The quantization Process has a two-fold effect:
1. The peak-to-peak range of the input sample values is subdivided into a finite set of
decision levels or decision thresholds that are aligned with the risers of the staircase, and

2. The output is assigned a discrete value selected from a finite set of representation levels
that are aligned with the treads of the staircase.
A quantizer is memory less in that the quantizer output is determined only by the value of a
corresponding input sample, independently of earlier analog samples applied to the input.

Types of Quantizers:
1. Uniform Quantizer
2. Non- Uniform Quantizer
Uniform Quantizer: In Uniform type, the quantization levels are uniformly spaced,
whereasin non-uniform type the spacing between the levels will be unequal and mostly the
relation is logarithmic.
Types of Uniform Quantizers: ( based on I/P - O/P Characteristics)
1. Mid-Rise type Quantizer
2. Mid-Tread type Quantizer
In the stair case like graph, the origin lies the middle of the tread portion in Mid –Tread type
where as the origin lies in the middle of the rise portion in the Mid-Rise type.
Mid – tread type:Quantization levels – odd number.
Mid – Rise type: Quantization levels – even number.
Quantization Noise and Signal-to-Noise:
“The Quantization process introduces an error defined as the difference between the input
signal, x(t) and the output signal, yt). This error is called the Quantization Noise.”
q(t) = x(t) – y(t)
Quantization noise is produced in the transmitter end of a PCM system by rounding off
sample values of an analog base-band signal to the nearest permissible representation levels
of the quantizer. As such quantization noise differs from channel noise in that it is signal
dependent.
Let “Δ‟ be the step size of a quantizer and L be the total number of quantization levels.
Quantization levels are 0, ± ., ± 2 ., ±3 . . . . . . . The Quantization error, Q is a random
variable and will have its sample values bounded by [-(Δ/2) < q < (Δ/2)]. If is small, the
quantization error can be assumed to a uniformly distributed random variable.
Consider a memory less quantizer that is both uniform and symmetric.
L = Number of quantization levels
X = Quantizer input
Y = Quantizer output
The output y is given by
Y=Q(x)
which is a staircase function that befits the type of mid tread or mid riser quantizer of
interest.
Advantages of Non- Uniform Quantization:
1. Higher average signal to quantization noise power ratio than the uniform quantizer
when the signal pdf is non uniform which is the case in many practical situation.
2. RMS value of the quantizer noise power of a non – uniform quantizer is
substantially proportional to the sampled value and hence the effect of the quantizer noise is
reduced.
PULSE MODULATION
Pulse modulation is furtherdivided into analog and digital modulation. The analog
modulation techniques are mainly classified into Pulse Amplitude Modulation, Pulse
Duration Modulation/Pulse Width Modulation, and Pulse Position Modulation.

Pulse Amplitude Modulation

Pulse Amplitude Modulation (PAM) is an analog modulating scheme in which the

amplitude of the pulse carrier varies proportional to the instantaneous amplitude of the
message signal.
The pulse amplitude modulated signal, will follow the amplitude of the original signal, as
the signal traces out the path of the whole wave. In natural PAM, a signal sampled at the
Nyquist rate is reconstructed, by passing it through an efficient Low Pass Frequency
(LPF) with exact cutoff frequency
The following figures explain the Pulse Amplitude Modulation.

Though the PAM signal is passed through an LPF, it cannot recover the signal without
distortion. Hence to avoid this noise, flat-top sampling is done as shown in the following
figure.
Flat-top sampling is the process in which sampled signal can be represented in pulses for
which the amplitude of the signal cannot be changed with respect to the analog signal, to be
sampled. The tops of amplitude remain flat. This process simplifies the circuit design.

Pulse Width Modulation

Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse Time
Modulation (PTM) is an analog modulating scheme in which the duration or width or time
of the pulse carrier varies proportional to the instantaneous amplitude of the message signal.
The width of the pulse varies in this method, but the amplitude of the signal remains
constant. Amplitude limiters are used to make the amplitude of the signal constant. These
circuits clip off the amplitude, to a desired level and hence the noise is limited.

The following figures explain the types of Pulse Width Modulations.

There are three variations of PWM. They are −
 The leading edge of the pulse being constant, the trailing edge varies according to the
message signal.
 The trailing edge of the pulse being constant, the leading edge varies according to the
message signal.
 The center of the pulse being constant, the leading edge and the trailing edge varies
according to the message signal.
These three types are shown in the above given figure, with timing slots.

Pulse Position Modulation

Pulse Position Modulation (PPM) is an analog modulating scheme in which the amplitude
and width of the pulses are kept constant, while the position of each pulse, with reference to
the position of a reference pulse varies according to the instantaneous sampled value of the
message signal.
The transmitter has to send synchronizing pulses (or simply sync pulses) to keep the
transmitter and receiver in synchronism. These sync pulses help maintain the position of the
pulses. The following figures explain the Pulse Position Modulation.
Pulse position modulation is done in accordance with the pulse width modulated signal.
Each trailing of the pulse width modulated signal becomes the starting point for pulses in
PPM signal. Hence, the position of these pulses is proportional to the width of the PWM
pulses.

Advantage

As the amplitude and width are constant, the power handled is also constant.

Disadvantage

The synchronization between transmitter and receiver is a must.

Comparison between PAM, PWM, and PPM

The comparison between the above modulation processes is presented in a single table.

PAM PWM PPM

Amplitude is varied Width is varied Position is varied

Bandwidth depends on the Bandwidth depends on the rise Bandwidth depends on the rise
width of the pulse time of the pulse time of the pulse

Instantaneous transmitter Instantaneous transmitter power Instantaneous transmitter

power varies with the varies with the amplitude and power remains constant with
amplitude of the pulses width of the pulses the width of the pulses
System complexity is high System complexity is low System complexity is low

Noise interference is high Noise interference is low Noise interference is low

It is similar to amplitude It is similar to frequency It is similar to phase

modulation modulation modulation

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.

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.
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.
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
message signal, 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
by a 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

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.

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.
There are two types of Companding techniques.

A-law Companding Technique

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

and there is no compression.
 A-law has mid-rise at the origin. Hence, it contains a non-zero value.
 A-law companding is used for PCM telephone systems.
 A-law is used in many parts of the world.

µ-law Companding Technique

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

there is no compression.
 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.

Differential Pulse Code Modulation (DPCM)

For the signals which does not change rapidly from one sample to next sample, the PCM
scheme is not preferred. When such highly correlated samples are encoded the resulting
encoded signal contains redundant information. By removing this redundancy before
encoding an efficient coded signal can be obtained. One of such scheme is the DPCM
technique. By knowing the past behavior of a signal up to a certain point in time, it is possible
to make some inference about the future values. The transmitter and receiver of the DPCM
scheme is shown in the below fig.respectively. Transmitter: Let x(t) be the signal to be
sampled and x(nTs) be its samples. In this scheme the input to the quantizer is a signal.

where x^(nTs) is the prediction for unquantized sample x(nTs). This predicted value is
produced by using a predictor whose input, consists of a quantized versions of the input
signal x(nTs). The signal e(nTs) is called the prediction error.
By encoding the quantizer output, in this method, we obtain a modified version of the PCM
called differential pulse code modulation (DPCM).
Quantizer output,
v(nTs) = Q[e(nTs)]
= e(nTs) + q(nTs) ---- (3.32)
Predictor input is the sum of quantizer output and predictor output, u(nTs) = x^(nTs) +
v(nTs) ---- (3.33)
Using 3.32 in 3.33,
u(nTs) = x^(nTs) + e(nTs) + q(nTs) ----(3.34)
u(nTs) = x(nTs) + q(nTs) ----(3.35)
The receiver consists of a decoder to reconstruct the quantized error signal. The quantized
version of the original input is reconstructed from the decoder output using the same
predictor as used in the transmitter. In the absence of noise the encoded signal at the receiver
input is identical to the encoded signal at the transmitter output. Correspondingly the receive
output is equal to u(nTs), which differs from the input x(nts) only by the quantizing error
q(nTs).
The sampling rate of a signal should be higher than the Nyquist rate, to achieve better
sampling. If this sampling interval in a Differential PCM (DPCM) is reduced considerably,
the sample-to-sample amplitude difference is very small, as if the difference is 1-bit
quantization, then the step-size is very small i.e., Δ (delta).

DELTA MODULATION

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

Features of Delta Modulation

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

  The quantization design is simple.
 The input sequence is much higher than Nyquist rate.
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 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.
 It requires 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.

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.

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 block diagram for delta demodulator.

Low pass filter is used for many reasons, but the prominent one 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 modulator & demodulator
However, there exists some noise in DM and following are the types of noise.

 Slope Over load distortion (when Δ is small)

 Granular noise (when Δ is large)

Adaptive Delta Modulation

In digital modulation, we come across certain problems in determining the step-size, which
influences the quality of the output wave.
The larger step-size is needed in the steep slope of modulating signal and a smaller stepsize
is needed where the message has a small slope. As a result, the minute details get missed.
Hence, it would be better if we can control the adjustment of step-size, according to our
requirement in order to obtain the sampling in a desired fashion. This is the concept
of Adaptive Delta Modulation (ADM).
LINE CODE
A line code is the code used for data transmission of a digital signal over a transmission
line. This process of coding is chosen so as to avoid overlap and distortion of signal such as
inter-symbol interference.

Properties of Line Coding

Following are the properties of line coding −

  As the coding is done to make more bits transmit on a single signal, the bandwidth
used is much reduced.
 For a given bandwidth, the power is efficiently used.
 The probability of error is much reduced.
 Error detection is done and the bipolar too has a correction capability.
 Power density is much favorable.
 The timing content is adequate.
 Long strings of 1s and 0s is avoided to maintain transparency.

Types of Line Coding

There are 3 types of Line Coding

 Unipolar
 Polar
 Bi-polar

Unipolar Signaling

Unipolar signaling is also called as On-Off Keying or simply OOK.

The presence of pulse represents a 1 and the absence of pulse represents a 0.
There are two variations in Unipolarsignaling −

 Non Return to Zero NRZNRZ

 Return to Zero RZRZ

Unipolar Non-Return to Zero NRZNRZ

In this type of unipolar signaling, a High in data is represented by a positive pulse called
as Mark, which has a duration T0 equal to the symbol bit duration. A Low in data input has
no pulse.
The following figure clearly depicts this.
Advantages
The advantages of Unipolar NRZ are −

 It is simple.
 A lesser bandwidth is required.
Disadvantages
The disadvantages of Unipolar NRZ are −
 No error correction done.
 Presence of low frequency components may cause the signal droop.
 No clock is present.
 Loss of synchronization is likely to occur (especially for long strings of 1s and 0s).

Unipolar Return to Zero RZRZ

In this type of unipolar signaling, a High in data, though represented by a Mark pulse, its
duration T0 is less than the symbol bit duration. Half of the bit duration remains high but it
immediately returns to zero and shows the absence of pulse during the remaining half of the
bit duration.
It is clearly understood with the help of the following figure.
Advantages
The advantages of Unipolar RZ are −

 It is simple.
 The spectral line present at the symbol rate can be used as a clock.
Disadvantages
The disadvantages of Unipolar RZ are −

 No error correction.
 Occupies twice the bandwidth as unipolar NRZ.
 The signal droop is caused at the places where signal is non-zero at 0 Hz.

Polar Signaling

There are two methods of Polar Signaling. They are −

 Polar NRZ
 Polar RZ

Polar NRZ

In this type of Polar signaling, a High in data is represented by a positive pulse, while a Low
in data is represented by a negative pulse. The following figure depicts this well.
Advantages
The advantages of Polar NRZ are −

 It is simple.
 No low-frequency components are present.
Disadvantages
The disadvantages of Polar NRZ are −
 No error correction.
 No clock is present.
 The signal droop is caused at the places where the signal is non-zero at 0 Hz.

Polar RZ

In this type of Polar signaling, a High in data, though represented by a Mark pulse, its
duration T0 is less than the symbol bit duration. Half of the bit duration remains high but it
immediately returns to zero and shows the absence of pulse during the remaining half of the
bit duration.
However, for a Low input, a negative pulse represents the data, and the zero level remains
same for the other half of the bit duration. The following figure depicts this clearly.
Advantages
The advantages of Polar RZ are −

 It is simple.
 No low-frequency components are present.
Disadvantages
The disadvantages of Polar RZ are −
 No error correction.
 No clock is present.
 Occupies twice the bandwidth of Polar NRZ.
 The signal droop is caused at places where the signal is non-zero at 0 Hz.

Bipolar Signaling

This is an encoding technique which has three voltage levels namely +, - and 0. Such a
signal is called as duo-binary signal.
An example of this type is Alternate Mark Inversion AMIAMI. For a 1, the voltage level
gets a transition from + to – or from – to +, having alternate 1s to be of equal polarity.
A 0 will have a zero voltage level.
Even in this method, we have two types.

 Bipolar NRZ
 Bipolar RZ
From the models so far discussed, we have learnt the difference between NRZ and RZ. It
just goes in the same way here too. The following figure clearly depicts this.

The above figure has both the Bipolar NRZ and RZ waveforms. The pulse duration and
symbol bit duration are equal in NRZ type, while the pulse duration is half of the symbol bit
duration in RZ type.

Advantages

Following are the advantages −

 It is simple.
 No low-frequency components are present.
 Occupies low bandwidth than unipolar and polar NRZ schemes.
 This technique is suitable for transmission over AC coupled lines, as signal drooping
doesn’t occur here.
 A single error detection capability is present in this.

Disadvantages

Following are the disadvantages −

 No clock is present.
 Long strings of data causes loss of synchronization.
Noise considerations in PCM:
The performance of a PCM system is influenced by two major sources of noise:
1. Channel noise, which is introduced anywhere between the transmitter output and the
receiver input. Channel noise is always present, once the equipment is switched on.
2. Quantization noise, which is introduced in the transmitter and is carried all the way 32
along to the receiver output. Unlike channel noise, quantization noise is signal dependent in
the sense that it disappears when the message signal is switched off.
The main effect of channel noise is to introduce bit errors into the received signal. In the case
of a binary PCM system, the presence of a bit error causes symbol 1 to be mistaken for
symbol 0, or vice versa.
Clearly, the more frequently bit errors occur, the more dissimilar the receiver output
becomes compared to the original message signal.
The fidelity of information transmission by PCM in the presence of channel noise may be
measured in terms of the average probability of symbol error, which is defined as the
probability that the reconstructed symbol at the receiver output differs from the transmitted
binary symbol, on the average.
The average probability of symbol error, also referred to as the bit error rate (BER), assumes
that all the bits in the original binary wave are of equal importance.
To optimize system performance in the presence of channel noise, we need to minimize the
average probability of symbol error.
For this evaluation, it is customary to model the channel noise as additive, white, and
Gaussian.
The effect of channel noise can be made practically negligible by ensuring the use of an
adequate signal energy-to-noise density ratio through the provision of short-enough spacing
between the regenerative repeaters in the PCM system.
Quantization noise is essentially under the designer's control. It can be made negligibly small
through the use of an adequate number of representation levels in the quantizer and the
selection of a companding strategy matched to the characteristics of the type of message
signal being transmitted.

MULTIPLEXING
Multiplexing is the process of combining multiple signals into one signal, over a shared
medium. If analog signals are multiplexed, it is Analog Multiplexing and if digital signals are
multiplexed, that process is Digital Multiplexing.
The process of multiplexing divides a communication channel into several number of logical
channels, allotting each one for a different message signal or a data stream to be transferred.
The device that does multiplexing can be simply called as a MUX while the one that reverses
the process which is demultiplexing, is called as DEMUX.

Types of Multiplexers
There are mainly two types of multiplexers, namely analog and digital. They are further
divided into FDM, WDM, and TDM.
Analog Multiplexing
The analog multiplexing techniques involve signals which are analog in nature. The analog
signals are multiplexed according to their frequency (FDM) or wavelength (WDM).
Frequency Division Multiplexing (FDM)
In analog multiplexing, the most used technique is Frequency Division Multiplexing FDM.
This technique uses various frequencies to combine streams of data, for sending them on a
communication medium, as a single signal.
Example: A traditional television transmitter, which sends a number of channels through a
single cable, uses FDM.
Wavelength Division Multiplexing (WDM)
Wavelength Division Multiplexing is an analog technique, in which many data streams of
different wavelengths are transmitted in the light spectrum. If the wavelength increases, the
frequency of the signal decreases.
Example: Optical fibre Communications use the WDM technique, to merge different
wavelengths into a single light for the communication.
Digital Multiplexing
The term digital represents the discrete bits of information. Hence the available data is in the
form of frames or packets, which are discrete.
Time Division Multiplexing (TDM)
In TDM, the time frame is divided into slots. This technique is used to transmit a signal over
a single communication channel, with allotting one slot for each message. Of all the types of
TDM, the main ones are Synchronous and Asynchronous TDM.
Synchronous TDM
In Synchronous TDM, the input is connected to a frame. If there are ‘n’ number of
connections, then the frame is divided into ‘n’ time slots. One slot is allocated for each input
line. In this technique, the sampling rate is common to all signals and hence same clock input
is given. The mux allocates the same slot to each device at all times.
Asynchronous TDM
In Asynchronous TDM, the sampling rate is different for each of the signals and the clock
signal is also not in common. If the allotted device, for a time-slot, transmits nothing and sits
idle, then that slot is allotted to another device, unlike synchronous.
Time Division Multiplexing (TDM)
Digital Multiplexers:

Course Outcomes:
At the end of the unit the students will be able to - Investigate pulse modulation
systems and analyze their systemperformance.
 UNIT-4
BASEBAND MODULATION TECHNIQUES
Pre-requisite:
Basic Knowledge of Digital System, modulation and demodulation process
Baseband transmission:

Baseband transmission is the simplest form for the communication of information. Discrete
information is communicated with specific symbols selected from a finite set of symbols. In
baseband transmission, symbols are simply communicated as a pulse with a discrete voltage
level and, for binary transmission, only two voltages are used. A series of pulses forms a
pulse train that carries the full message. Prior to transmission, especially in radio systems,
these pulses are shaped to limit their high frequency content so as to minimize crosstalk with
adjacent communication channels. During transmission through a bandlimited channel,
pulses are dispersed (spread) in time and can overlap with each other giving rise to
intersymbol interference (ISI). When pulses reach the receiver, dispersion and other
distortions can be partially compensated with an equalizer.

Inter symbol Interference

Generally, digital data is represented by electrical pulse, communication channel is always
band limited. Such a channel disperses or spreads a pulse carrying digitized samples passing
through it. When the channel bandwidth is greater than bandwidth of pulse, spreading of
pulse is very less. But when channel bandwidth is close to signal bandwidth, i.e. if we
transmit digital data which demands more bandwidth which exceeds channel bandwidth,
spreading will occur and cause signal pulses to overlap. This overlapping is called Inter
Symbol Interference. In short it is called ISI. Similar to interference caused by other
sources, ISI causes degradations of signal if left uncontrolled. This problem of ISI exists
strongly in Telephone channels like coaxial cables and optical fibers.
The main objective is to study the effect of ISI, when digital data is transmitted through band
limited channel and solution to overcome the degradation of waveform by properly shaping
pulse
The effect of sequence of pulses transmitted through channel is shown in fig. The Spreading
of pulse is greater than symbol duration, as a result adjacent pulses interfere. i.e. pulses get
completely smeared, tail of smeared pulse enter into adjacent symbol intervals making it
difficult to decide actual transmitted pulse.
EYE PATTERN
The quality of digital transmission systems are evaluated using the bit error rate. Degradation
of quality occurs in each process modulation, transmission, and detection. The eye pattern is
experimental method that contains all the information concerning the degradation of quality.
Therefore, careful analysis of the eye pattern is important in analyzing the degradation
mechanism.
• Eye patterns can be observed using an oscilloscope. The received wave is applied to the
vertical deflection plates of an oscilloscope and the sawtooth wave at a rate equal to
transmitted symbol rate is applied to the horizontal deflection plates, resulting display is eye
pattern as it resembles human eye.
• The interior region of eye pattern is called eye opening

We get superposition of successive symbol intervals to produce eye pattern as shown below.
• The width of the eye opening defines the time interval over which the received wave can be
sampled without error from ISI
• The optimum sampling time corresponds to the maximum eye opening
• The height of the eye opening at a specified sampling time is a measure of the margin over
channel noise.
The sensitivity of the system to timing error is determined by the rate of closure of the eye as
the sampling time is varied. Any non linear transmission distortion would reveal itself in an
asymmetric or squinted eye. When the effected of ISI is excessive, traces from the upper
portion of the eye pattern cross traces from lower portion with the result that the eye is
completely closed.
Example of eye pattern:Binary-PAM Perfect channel (no noise and no ISI)

Example of eye pattern: Binary-PAM with noise no ISI

Nyquist Criterion for Zero-ISI
Nyquist proposed a condition for pulses p(t) to have zero–ISI when transmitted through a
channel with sufficient bandwidth to allow the spectrum of all the transmitted signal to pass.
Nyquist proposed that a zero–ISI pulse p(t) must satisfy the condition

A pulse that satisfies the above condition at multiples of the bit period Tb will result in zero–
ISI if the whole spectrum of that signal is received. The reason for which these zero–ISI
pulses (also called Nyquist–criterion pulses) cause no ISI is that each of these pulses at the
sampling periods is either equal to 1 at the center of pulse and zero the points other pulses are
centered. In fact, there are many pulses that satisfy these conditions. For example, any square
pulse that occurs in the time period –Tb to Tb or any part of it (it must be zero at –Tb and Tb)
will satisfy the above condition.
Also, any triangular waveform („Δ‟ function) with a width that is less than 2Tb will also
satisfy the condition. A sinc function that has zeros at t = Tb, 2Tb, 3Tb, … will also satisfy
this condition. The problem with the sinc function is that it extends over a very long period of
time resulting in a lot of processing to generate it. The square pulse required a lot of
bandwidth to be transmitted. The triangular pulse is restricted in time but has relatively large
bandwidth.
There is a set of pulses known as raised–cosine pulses that satisfy the Nyquist criterion and
require slightly larger bandwidth than what a sinc pulse (which requires the minimum
bandwidth ever) requires. The spectrum of these pulses is given by

Where ω b is the frequency of bits in rad/s (ω b = 2 /Tb), and x is called the excess bandwidth
and it defines how much bandwidth would be required above the minimum bandwidth that is
required when using a sinc pulse. The excess bandwidth ω x for this type of pulses is
restricted between
Sketching the spectrum of these pulses we get

We can easily verify that when ωx = 0, the above spectrum becomes a rect function, and
therefore the pulse p(t) becomes the usual sinc function. For ωx = b/2, the spectrum is similar
to a sinc function but decays (drops to zero) much faster than the sinc (it extends over 2 or 3
bit periods on each side). The expense for having a pulse that is short in time is that it
requires a larger bandwidth than the sinc function (twice as much for ωx =ω b/2). Sketch of
the pulses and their spectrum for the two extreme cases of ω x =ωb/2 and ωx = 0 are shown
below
We can define a factor r called the roll–off factor to be

The roll–off factor r specifies the ratio of extra bandwidth required for these pulses compared
to the minimum bandwidth required by the sinc function.
DIGITAL MODULATION :
Digital Modulation provides more information capacity, high data security, quicker system
availability with great quality communication. Hence, digital modulation techniques have a
greater demand, for their capacity to convey larger amounts of data than analog modulation
techniques.
There are many types of digital modulation techniques and also their combinations,
depending upon the need. Of them all, we will discuss the prominent ones.

ASK – Amplitude Shift Keying

The amplitude of the resultant output depends upon the input data whether it should be a
zero level or a variation of positive and negative, depending upon the carrier frequency.

FSK – Frequency Shift Keying

The frequency of the output signal will be either high or low, depending upon the input data
applied.

PSK – Phase Shift Keying

The phase of the output signal gets shifted depending upon the input. These are mainly of
two types, namely Binary Phase Shift Keying BPSKBPSK and Quadrature Phase Shift
Keying QPSKQPSK, according to the number of phase shifts. The other one is Differential
Phase Shift Keying DPSKDPSK which changes the phase according to the previous value.

M-ary Encoding

M-ary Encoding techniques are the methods where more than two bits are made to transmit
simultaneously on a single signal. This helps in the reduction of bandwidth.
The types of M-ary techniques are −

 M-ary ASK
 M-ary FSK
 M-ary PSK
Amplitude Shift Keying (ASK):
In ASK, it requires two input signals, First input is binary sequence signal and the second
input is carrier signal. Here the most important point we need to always consider the second
input which is the carrier signal has the more amplitude/voltage range than the input binary
sequence signal.
One more important point is to consider here, the carrier signal amplitude is should be greater
than the input binary signal amplitude. Within carrier amplitude range we are going to
modulate the binary input signal amplitude. If the carrier signal amplitude is less than the
input binary signal voltage, then such a combination modulation process leads to over
modulation and under modulation effects. So to achieve perfect modulation carrier single
should have more amplitude range than input binary signal.

In amplitude shift keying theory, input binary signal amplitude varies according to the carrier
signal voltage. In ASK, the input binary signal is multiplied with the carrier signal along with
its time intervals. Between the first time interval of input binary signal multiplied with the
first time interval of carrier signal voltage and the same process continues for all time
intervals. If the input binary signal is logic HIGH for certain time interval, then the same
should be delivered at the output ports with increment in voltage level. So the main aim of the
amplitude shift keying modulation is to changing or improving the voltage characteristics of
the input binary signal concerning the carrier signal. The below diagram indicating the
Amplitude shift keying block diagram.
At Mixer Circuit Level

When the switch is closed – for all the logic HIGH time intervals i.e. when the input signal
having logic 1 during those intervals the switch is closed and it is multiplied with the carrier
signal which is generating from the function generator for the same duration.

When the switch is opened – when the input signal having logic 0, the switch is opened and
there is no output signal will be generated. Because the input binary signal logic 0 having no
voltage, so during these intervals when the carrier signal multiples with it, zero output will
come. The output is zero for all logic 0 intervals of the input binary signal. Mixer circuit
having the pulse shaping filters and band-limited filters for shaping the ASK output signal.

ASK Circuit Diagram

Amplitude shift keying modulation circuit can be designed with 555timer IC as an astable
mode. Here, the carrier signal can be varied by using the R1, R2 and C. The carrier frequency
can be instantly calculated by the formulae as 0.69*C*(R1+R2). A PIN 4 we will apply the
input binary signal and at PIN 3 the circuit will generate the ASK modulated wave.
ASK Demodulation Process

Demodulation is the process of reconstructing the original signal at the receiver level. And it
is defined as, whatever the modulated signal received from the channel at the receiver side by
implementing the proper demodulated techniques to recover/reproduce the original input
signal at the output stage of the receiver.
ASK demodulation can be done in two ways. They are,

 Coherent detection (Synchronous demodulation)

 Noncoherent Detection (Asynchronous demodulation)
We will start the demodulation process with coherent detection which is also called as
synchronous ASK detection.

1). Coherent ASK Detection

In this way of demodulation process, the carrier signal which we are using at the receiver
stage is in the same phase with the carrier signal which we are using at the transmitter stage.
It means the carrier signal at transmitter and receiver stages are the same values. This type of
demodulation is called Synchronous ASK detection or coherent ASK detection.

coherent-ask-detection-block-diagram

The receiver receives the ASK modulated waveform from the channel but here this
modulated waveform is effected with noise signal because it is forwarded from the free space
channel. So this, noise can be eliminated after the multiplier stage by the help of a low pass
filter. Then it is forwarded from the sample and hold circuit for converting it into discrete
signal form. Then at each interval, the discrete signal voltage is compared with the reference
voltage (Vref) to reconstruct the original binary signal.
2). Non-coherent ASK Detection

In this, the only difference is the carrier signal which is using at the transmitter side and
receiver side are not in the same phase with each other. By this reason, this detection is called
as Non-coherent ASK detection (Asynchronous ASK detection). This demodulation process
can be completed by using with square law device. The output signal which is generating
from the square-law device can be forwarded through a low pass filter to reconstruct the
original binary signal.

non-coherent-ask-detection-block-diagram

Amplitude shift keying is an effective technique to increase the input amplitude

characteristics in communications. But these ASK modulated waveforms are easily affected
by noise. And this leads to amplitude variations. Due to this, there will be voltage fluctuations
in the output waveforms. The second drawback of the ASK modulation technique is, it has
low power efficiency. Because ASK requires the excessive bandwidth. It leads to power loss
in the spectrum of ASK.

Whenever to modulate two input binary signals, amplitude shift keying modulation is not
preferable. Because it has to take only one input only. So, to overcome this Quadrature
Amplitude Shift Keying (ASK) is preferred. In this modulation technique, we can modulate
two binary signals with two different carrier signals. Here, these two carrier signals are in
opposite phase with 90degrees difference. Sin and cosine signals are used as carriers in
quadrature amplitude shift keying. The advantage of this is, it uses effectively the bandwidth
of the spectrum. It offers more power efficiency than the amplitude shift keying.

amplitude-shift- keying-Matlab-Simulink

Amplitude shift keying Matlab Simulink can be designed with Matlab tool. After initializing
the tool, by following the proper steps we can draw the ASK circuit on the work area. By
giving the proper signal values we can get the modulated output waveforms

ASK Applications
Modulation has an important role in communications. And amplitude shift keying
applications are mentioned below. They are:

 Low-frequency RF applications
 Home automation devices
 Industrial networks devices
 Wireless base stations
 Tire pressuring monitoring systems
Thus, Ask (amplitude shift keying) is a digital modulation technique to increase the
amplitude characteristics of the input binary signal. But its drawbacks make it so limited. And
these drawbacks can be overcome by the other modulation technique which is FSK.

Frequency Shift Keying (FSK):

The frequency shift keying is the most important digital modulation technique, and it is also
known as FSK. A signal has the amplitude, frequency, and phase as properties. Every signal
has these three properties. To increase any one of the signal property we can go for the
modulation process. Because there are various advantages of the modulation technique. In
those some of the advantages are – the antenna size reduced, avoid multiplexing of signals,
decrease the SNR, long-range communication can be possible, etc. These are the important
advantages of the modulation process. If we modulate the amplitude of the input binary
signal according to the carrier signal i.e. called as amplitude shift keying. Here, in this article,
we are going to discuss what is frequency shift keying and FSK modulation, demodulation
process along with their advantages and disadvantages.
It is defined as the changing or improving the frequency characteristics of an input binary
signal according to the carrier signal. Amplitude variation is one of the major drawbacks in
ASK. So, due to this ask modulation technique used in a few applications only. And its
spectrum power efficiency also low. It leads to wastage of power. So to overcome these
drawbacks Frequency Shift Keying is preferred. FSK is also known as Binary Frequency
Shift Keying (BFSK). The below frequency shift keying theory describes what happening
in frequency shift keying modulation.

Frequency Shift Keying Theory

This frequency shift keying theory shows how the frequency characteristics of a binary signal
changed according to the carrier signal. In FSK, the binary information can be transmitted
through a carrier signal along with frequency changes. The below diagram shows
the frequency shift keying block diagram.
FSK-block-diagram

In FSK, two carrier signals are used to produce FSK modulated waveforms. The reason
behind this, FSK modulated signals are represented in terms of two different frequencies. The
frequencies are called “mark frequency” and “space-frequency”. Mark frequency has
represented logic 1 and space-frequency has represented the logic 0. There is only one
difference between these two carrier signals, i.e. carrier input 1 having more frequency than
the carrier input 2.

Carrier input 1 = Ac Cos (2ωc+θ) t

Carrier input 2 = Ac Cos (2ωc-θ) t
The switch (s) of the 2:1 multiplexer is having the important role to generate the FSK output.
Here the switch is connected to carrier input 1 for all logic 1’s of the binary input sequence.
And switch (s) is connected to carrier input 2 for all logic 0’s of the input binary sequence.
So, the resultant FSK modulated waveforms have mark frequencies and space frequencies.

FSK-modulation-output-waveforms
Now we will see how the FSK modulated wave can be demodulated at the receiver
side. Demodulation is defined as reconstructing the original signal from the modulated signal.
This demodulation can be possible in two ways. They are
 Coherent FSK detection
 Non-coherent FSK detection
The only difference between the coherent and non-coherent way of detection is the phase of
the carrier signal. If the carrier signal we are using at the transmitter side and receiver side are
in the same phase while demodulation process i.e. called a coherent way of detection and it is
also known as synchronous detection. If the carrier signals which we are using at transmitter
and receiver side are not in the same phase then such modulation process known as Non-
coherent detection. Another name for this detection is Asynchronous detection.

Coherent FSK Detection

In this synchronous FSK detection, the modulated wave got affected by noise while reaching
the receiver. So, this noise can be eliminated from using the bandpass filter (BPF). Here at
multiplier stage, the noisy FSK modulated signal is multiplied with the carrier signal from the
local oscillator device. Then the resultant signal passes from the BPF. Here this bandpass
filter is assigned to cut off frequency which is equal to the binary input signal frequency. So
the same frequencies can be allowed to the decision device. Here this decision device gives 0
and 1 for space and mark frequencies of the FSK modulated waveforms.

coherent-FSK-detection

Non-coherent FSK Detection

The modulated FSK signal is forwarded from the bandpass filter 1 and 2 with cut off
frequencies equals to space and mark frequencies. So, the unwanted signal components can
be eliminated from the BPF. And the modified FSK signals are applied as input to the two
envelop detectors. This envelope detector is a circuit having a diode (D). Based upon the
input to the envelope detector it delivers the output signal. This envelope detector used in the
amplitude demodulation process. Based upon its input it generates the signal and then it is
forwarded to the threshold device. This threshold device gives the logic 1 and 0 for the
different frequencies. This would be equal to the original binary input sequence. So, the FSK
generation and detection can be done in this way.. In this FSK experiment, FSK can be
generated by the 555 timer IC and detection can be possible by 565IC which is known as
a phase-locked loop (PLL).

non-coherent-FSK-detection

There are few frequency shift keying advantages and disadvantages are listed below.

Advantages

 Simple process to construct the circuit

 Zero amplitude variations
 Supports a high data rate.
 Low probability of error.
 High SNR (signal to noise ratio).
 More noise immunity than the ASK
 Error-free reception can be possible with FSK
 Useful in high-frequency radio transmissions
 Preferable in high-frequency communications
 Low-speed digital applications

Disadvantages

 It requires more bandwidth than the ASK and PSK(phase shift keying)
 Due to the requirement of large bandwidth, this FSK has limitations to use only in low-
speed modems which the bit rate is 1200bits/sec.
 The bit error rate is less in AEGN channel than phase shift keying.
Thus, the frequency shift keying is one of the fine digital modulation technique to increase
the frequency characteristics of the input binary signal. By FSK modulation technique we can
achieve error-free communication in a few digital applications. But this FSK has finite data
rate and consumes more bandwidth can be overcome by the QAM, which is known as
quadrature amplitude modulation. It is the combination of amplitude modulation and phase
modulation.

Phase Shift Keying (PSK)

The term PSK or Phase shift keying is broadly used in a radio communication system. This
kind of technique is mostly compatible with data communications. It allows information in a
more efficient way to be carried over a radio communications signal compare with other
modulation forms. Data communication is rising with different forms of communication
formats like analog to digital to carry data along with different modulation forms. There are
different types of PSK where each one has its own benefits and drawbacks. An option of the
optimum format has to be prepared for every radio communication system. To make the
correct option, it is essential to have knowledge of how PSK works.

What is Phase Shift Keying (PSK)?

The Phase Shift Keying is one kind of digital modulation method. This kind of method is
used to transmit data by modulating otherwise changing the phase of the carrier signal which
is known as a reference signal. The digital data can be represented with any kind of digital
modulation method by using a limited number of separate signals. This kind of modulation
method uses a limited number of phases where each phase can be assigned with binary digits.
Generally, every phase encodes an equivalent number of bits. Every bits pattern forms the
symbol that is denoted by the exact phase.
The PSK method can be represented by a convenient method namely constellations diagram.
In this kind of communication, the points of the constellation can be selected are generally
placed by uniform angular spacing in the region of circle. So that utmost phase separation can
be offered among nearby points & therefore the best protection to corruption. These are
arranged in a circle so that they can all be transmitted by similar energy.
phase-shift-keying

Digital Modulation

The digital modulation or DM is one kind of modulation, which utilizes discrete signals to
change a carrier wave. This kind of modulation eliminates the noise of communication and
provides superior power for the signal interruption. This modulation provides additional data
capacity and security for high & easy system accessibility by huge communication quality.
So, this kind of modulation has a vast demand than analog modulation.

Types of PSK

The PSK can be classified into two types which include the following.

 BPSK – Binary Phase-Shift Keying

 QPSK – Quadrature Phase-Shift Keying
1). BPSK – Binary Phase-Shift Keying

The term BPSK stands for Binary Phase-Shift Keying. Sometimes, it is also called as PRK
(phase reversal keying) or 2PSK. This kind of phase-shift keying utilizes 2-phases which are
separated with 180 degrees. So this is the reason to call as 2-PSK.

In this method, the arrangement of constellation points is not a matter where exactly they are
placed. This type of modulation is strong to all the PSKs as it takes the maximum level of
noise otherwise to distortion to make the demodulator attain an incorrect decision. However,
it is only able to modulate at 1 bit per symbol and is not suitable for applications like high
data rate.
2). QPSK – Quadrature Phase-Shift Keying

The bit rate can be enhanced by adding more bits on one single segment. In this kind of PSK,
the bitstream can be parallelized so that every two incoming bits can be split up & phase shift
keying a carrier frequency. One carrier frequency can be phase-shifted with 90 degrees from
the other within quadrature. Then the 2 phase-shift keying signals are added to generate one
of four signal elements.

Some Other Forms of PSK

Some of the more frequently used forms of PSK mainly include the following.

 Phase-Shift-Keying (PSK)
 Binary-Phase-Shift-Keying (BPSK)
 Quadrature-Phase-Shift-Keying (QPSK)
 Offset-Quadrature-Phase-Shift-Keying (O-QPSK)
 8 Point-Phase-Shift-Keying (8 PSK)
 16 Point-Phase-Shift-Keying (16 PSK)
The above-listed forms are the main PSK forms which are frequently used in the applications
of radio communication. Each form of PSK includes advantages as well as disadvantages.
Generally, the high order modulation forms will allow high data rates to transmit in a given
bandwidth. But the problem is high data rate which needs a superior S/N ratio previous to the
error rates begin to increase & this counter works to improve the performance of data rate.
The form of modulation can be selected by the radio communications systems can depend on
the existing conditions and requirements.

Advantages and Disadvantages of Phase Shift Keying

The advantages of phase-shift keying include the following.

 This type of PSK allows information to be carried with a radio communications signal
more efficiently compare with FSK.
 QPSK is another kind of data transmits wherever 4 phase states are utilized, all in 90
degrees of one another.
 It is less vulnerable to faults when we evaluate with ASK modulation & occupies similar
bandwidth like ASK.
 By using this, the high transmission data rate can be attained with the help of high-level
PSK modulations like QPSK, 16-QAM. Here QPSK signifies 2-bits for each constellation
and 16-QAM signifies 2-bits for each constellation.
The disadvantages of phase-shift keying include the following.

 The bandwidth efficiency of this PSK is less compared with ASK type of modulation
 It is a non-coherent reference signal
 By estimating the phase states of the signal, the binary information can be decoded.
Algorithms like recovery and detection are extremely difficult.
 High-level PSK modulations like QPSK, 16-QAM is more sensitive to phase differences.
 It generates wrong demodulations as the fault can combine with time because the
reference signal for demodulation is not fixed.
Applications of Phase Shift Keying

The applications of PSK include the following.

 This method is broadly used for bio-metric, wireless LAN along with wireless
communications like Bluetooth and RFID.
 Local Oscillator
 Optical Communications
 Multi-channel WDM
 Delay & add demodulator
 Nonlinear effects for WDM transmission
This is all about Phase Shift Keying. From the above information finally, we can conclude
that this PSK is a digital modulation technique which transmits information by altering the
phase of a stable frequency carrier signal. Generally, these modulation methods are superior
to modulation techniques like FSK in terms of bandwidth. These modulation schemes
provide better efficiency. But FSK modulation methods are power-efficient at a given signal-
to-noise ratio (S/N). This method is broadly used for bio-metric, wireless LAN along with
wireless communications like Bluetooth and RFID

Quadrature Amplitude Modulation:

In the amplitude modulation scheme, we can modulate one message signal (input signal)
which is in analog form. It means we can give only one input signal and we can modulate it
and transmit to the destination level. And the effective utilization of channel bandwidth is not
up to the level. So, these can be overcome by this QAM technique. This article discusses
what is quadrature amplitude modulation, its definition, block diagram, working principle,
and it’s applications.

What is Quadrature Amplitude Modulation?

Quadrature amplitude modulation (QAM) is modulation techniques that we can utilize in

analog modulation concept and digital modulation concept. Depending upon the input signal
form we can use it in either analog or digital modulation schemes. In QAM, we can modulate
two individual signals and transmitted to the receiver level. And by using the two input
signals, the channel bandwidth also increases. QAM can able to transmit two message signals
over the same c

Quadrature Amplitude Modulation Definition

QAM can be defined as it is s a modulation technique that is used to combine two amplitude
modulated waves into a single channel to increase the channel bandwidth.

Quadrature Amplitude Modulation Block Diagram

The below diagrams show the transmitter and receiver block diagram of the QAM scheme.
QAM Modulator
qam-modulator

QAM Demodulator

qam-demodulator

QAM Working Principle

“In the QAM transmitter, the above section i.e., product modulator1 and local oscillator are
called the in-phase channel and product modulator2 and local oscillator are called a
quadrature channel. Both output signals of the in-phase channel and quadrature channel are
summed so the resultant output will be QAM.”

At the receiver level, the QAM signal is forwarded from the upper channel of receiver and
lower channel, and the resultant signals of product modulators are forwarded from LPF1 and
LPF2. These LPF’s are fixed to the cut off frequencies of input 1 and input 2 signals. Then
the filtered outputs are the recovered original signals.
The below waveforms are indicating the two different carrier signals of the QAM technique.

input-carriers-of-qam

The output waveforms of QAM is shown below.

quadrature-output-signal-waveform

Advantages of QAM

The quadrature amplitude modulation advantages are listed below. They are

 One of the best advantages of QAM – supports a high data rate. So, the number of bits can
be carried by the carrier signal. Because of these advantages it preferable in wireless
communication networks.
 QAM’s noise immunity is very high. Due to this noise interference is very less.
 It has a low probability of error value.
 QAM expertly uses channel bandwidth.
Quadrature Amplitude Modulation Applications

The applications of QAM include the following.

 The applications of QAM are mostly observed in radio communications and data delivery
applications systems.
 QAM technique has wide applications in the radio communications field because, as the
increment of the data rate there is the chance of noise increment but this QAM technique
is not affected by noise interference hence there is an easy mode of signal transmission
can be possible with this QAM.
 QAM has wide applications in transmitting digital signals like digital cable television and in
internet services.
 In cellular technology, wireless device technology quadrature amplitude modulation is
preferred.

Continuous Phase Modulation (CPM)

Continuous phase modulation (CPM) is a constant-amplitude modulation scheme that is a

generalization of continuous phase frequency shift-keying (CPFSK) or minimum shift-keying
(MSK). The lack of phase discontinuities reduces high-frequency spectral content, making
CPM a highly spectrally efficient scheme.
One form of CPM that can result in significant coding gains is multi-h phase coding,
where h stands for the modulation index. This scheme is a generalization of CPFSK schemes
because different phase changes result from the transmission of the same symbol in two
contiguous symbol intervals. A mathematical representation of the signal during the ith
interval, iT ≤ t (i + 1)T, is expressed by the following formula:

where Es is the symbol energy

T is the symbol duration
ωc is the carrier frequency in radians/second
d and ω are the sequences that represent the M-ary information sequence
The data phase terms diωi(t - iT) and φ correspond to the phase associated with the current
data symbol and the phase accumulation due to the previous data symbol such that

and
where is the angular frequency corresponding to the modulation index used
during the ith baud. The different values of h can be used between symbol intervals in a
round robin fashion.
Minimum Shift Keying (MSK)

Minimum Shift Keying (MSK) is one of the most spectrally efficient modulation schemes
available. Due to its constant envelope, it is resilient to non-linear distortion and was
therefore chosen as the modulation technique for the GSM cell phone standard.

MSK is a special case of Continuous-Phase Frequency Shift Keying (CPFSK) which is a

special case of a general class of modulation schemes known as Continuous-Phase
Modulation (CPM). It is worth noting that CPM (and hence CPFSK) is a non-linear
modulation and hence by extension MSK is a non-linear modulation as well. Nevertheless, it
can also be cast as a linear modulation scheme, namely Offset Quadrature Phase Shift Keying
(OQPSK), which is a special case of Phase Shift Keying (PSK). As a borderline case, these
relationships are illustrated in Figure 1 below.
BINARY FREQUENCY SHIFT KEYING (BFSK)

In Frequency Shift Keying (FSK), digital information is transmitted by changing the

frequency of a carrier signal. Naturally, Binary FSK (BFSK) is the simplest form of FSK
where the two bits 0 and 1 correspond to two distinct carrier frequencies F0 and F1 to be sent
over the air. The bits can be translated into symbols through the relations

0→−10→−1

1→+11→+1

This enables us to write the frequencies Fi with i ϵ {0,1}i ϵ {0,1} as

Fi=Fc+(−1)i+1⋅ΔF=Fc±ΔF

where Fc is the nominal carrier frequency and ΔF is the peak frequency deviation from this
carrier frequency. Consequently,

s(t)=Acos2πFit=Acos[2π{Fc±ΔF}t]−−−−Eq (1)

where0≤t≤Tb

Figure 2 below displays a BFSK waveform for a random stream of data at a rate of Rb=1/Tb.
Note that we are not distinguishing between a bit period and a symbol period because both
are the same for a binary modulation technique.

Figure : A Binary Frequency Shift Keying (BFSK) waveform

Course Outcomes:
At the end of the unit the students will be able to -
Comparedifferentdigitalmodulationschemesfortheirefficiency and also Compute bit
error performance of various modulationschemes.
 UNIT-5

DEMODULATION OF DIGITAL SIGNALS

Pre-requisite: Basic Knowledge of Digital modulation process and information theory,

Why digital modulation?

The move to digital modulation provides more information capacity, compatibility

with digital data services, higher data security, better quality communications, and quicker
system availability.

Developers of communications systems face these constraints:

• available bandwidth

• permissible power

• inherent noise level of the system

The RF spectrum must be shared, yet every day there are more users for that spectrum
as demand for communications services increases. Digital modulation schemes have greater
capacity to convey large amounts of information than analog modulation schemes.

Trading off simplicity and bandwidth

There is a fundamental trade off in communication systems. Simple hardware can be used in
transmitters and receivers to communicate information. However, this uses a lot of spectrum
which limits the number of users. Alternatively, more complex transmitters and receivers can
be used to transmit the same information over less bandwidth. The transition to more and
more spectrally efficient transmission techniques requires more and more complex hardware.
Complex hardware is difficult to design, test, and build. This trade off exists whether
communication is over air or wire, analog or digital.

Simple Simple
More Spectrum
Hardware Hardware

Complex Complex
Hardware Hardware
Less Spectrum

Industrytrends
Over the past few years a major transition has occurred from simple analog Amplitude
Modulation (AM) and Frequency/Phase Modulation (FM/PM) to new digital modulation
techniques. Examples of digital modulation include
• QPSK (Quadrature Phase ShiftKeying)
• FSK (Frequency ShiftKeying)
• MSK (Minimum ShiftKeying)
• QAM (Quadrature AmplitudeModulation)

TDMA, CDMA
Figure 2.Signal/SystemComplexity

Time-Variant Signals
Trends in the Industry

QAM, FSK, QPSK

Vector Signals

AM, FM

Another layer of complexity in many new systems is multiplexing. Two principal types of
multiplexing (or “multiple access”) are TDMA (Time Division Multiple Access) and CDMA
(Code Division Multiple Access).These are two different ways to add diversity to signals
allowing different signals to be separated from one another .
Band limited channels
● Analysis in previous chapters considered the channel bandwidth to beunbounded

● All physical channels are bandlimited, with C(f) = 0 for |f| > W

● Nondistorting (ideal) channel: |C(f)| = const. for | f | < W and islinear

● All other channels are nonideal (distort the signal in amplitude, phase orboth)

● Systemmodel:
x(t) = transmit pulse * channel * receive filter

● The equivalent discrete

discrete-timemodel:

● This can be expressedas:

● With :
● The equivalent discrete-timemodel:

● This can be expressedas:

● With :

● Channel is assumed to be idealband-limited

● Task: to find pulse shapes thatwill

– Result in zero ISI at thereceiver

– Allow us to achieve maximum possible transmission rate with zero

ISI (optimum utilization of the givenbandwidth)
● The condition for zeroISI:
 ● Theorem (Nyquist): The necessary and sufficient condition for x(t) tosatisfy

is that its Fourier transform X(f) satisfies

● The replicas of X(f) obtained by sampling x(t) should add up to form a flat spectrum
(a delta in timedomain)
● Three different cases of signal design are observed (with respect to signallingrate)

● Case 1: T <1/2W, or 1/T >2W

At this rate, there exists no pulse whose spectrum replicas add to form a flat spectrum →
ISI is inevitable at thisrate
● Case 2: T = 1/2W, or 1/T =2W
 ● The illustrated pulse obviously doesn't satisfy the Nyquist criterion for
zeroISI
● The only pulse which satisfies the Nyquist criterion is the sinc pulse
(rectangularspectrum)
● Difficulties with sincpulse:

– It is noncausal and thereforenonrealizable

● A possible solution: delaying it until t0 so
that is is approximately 0 fort<0

– It decays as 1/t; if there is a slightest sampling offset, the resulting

ISI is infinite (the series is not absolutelysummable)
● Case 3: T >1/2W, or 1/T <2W

● At this rate, there exist numerous pulses which satisfy the zero-
ISIcriterion.

Band limited channels with distortion:

● Channel C(f) is notideal
 ● The overall frequency response of thesystem:

whereXd(f) is selected to yield controlled ISI or zero ISI. For zeroISI, Xd(f) = Xrc(f).
● Case 1: the channel distortion is precompensated at the transmitter:

● Case 2: compensation of channel distortion is equally split between the transmitter

andreceiver:

● Although ISI is cancelled, the effect of the channel causes SNRdegradation

● It can be shown that SNR degradation is lower when the distortion compensation is
equally split between TX and RX (case2)
OPTIMUM DEMODULATION OF DIGITAL SIGNAL OVER BAND LIMITED
CHANNELS:
● The receivedsignal:

wherez(t) is white.

● The received signal is then passed through a filter matched to h(t), so combination of
transmit pulse andchannel.
● The output of the matched filter is sampled atnT.

● In discretetime:

● The matched filter output ykis affected byISI.

● If we consider that a finite number of symbols ”mixes” to form yk (channel xk has a

finite number of taps), then the system has a finite number of states between which
it passes.

● The time evolution of the system can then be described by atrellis

 ● An optimum decision on a sent information sequence after observing a received
sequence can then be made using the Viterbialgorithm.

Maximum Likelihood Decoding

Some codes need to correct as many errors as possible, since there is no possibility for
retransmission (e.g. if the data is stored and then retrieved or if the transmission comes from a
Mars craft). The standard decoding technique is called maximum likelihood decoding. In
maximum likelihood decoding, the received word is compared to all the possible code words.
The one code word that is closest (ties are being broken randomly) is then the error corrected
code word. The name describes the most important property of this decoding. If an error is
detected, then we identify the most probable error and undo this error in order to correct the
mistake. Sometimes of course we will make a mistake in guessing the error.

Figure 2: Maximum Likelihood Decoding.

As an example for maximum likelihood encoding we assume the following pairing between
data words and code words:
 00 00100

01 01110

10 10001

11 11000

Assume now that we receive the word 11111. By going through all code words in the list, we
find that none matches. Hence, a transmission error has occurred. The Hamming distances to
the code words are d(11111,00100) = 4, d(11111,01110) =2, d(11111,10001) = 3, and
d(11111,11000) = 3. We therefore conclude that the first and last bit are in error and that we
should have received 01110, which corresponds to message 01.

As you can see from this example, maximum likelihood encoding for larger message words is
difficult to implement and time consuming to execute, since we need to compare the
received, possible erroneous, code word with all possible code words. Also, to maximise
maximum likelihood decoding's capacity to find errors, we need to find code words that are
evenly spaced in the space of all possible code words.

Maximum Likelihood (ML) decoding rule

Consider a set of possible codewords (valid codewords – set ) generated by an encoder in

the transmitter side. We pick one codeword out of this set ( call it ) and transmit it via a
Binary Symmetric Channel (BSC) with probability of error At the receiver side we receive
the distorted version of ( call this erroneous codeword ).

Maximum Likelihood Decoding chooses one codeword from (the list of all possible
codewords) which maximizes the following probability.

Meaning that the receiver computes . and chooses

a codeword ( ) which gives the maximum probability. In practice we don’t know (at the
receiver) but we know . So how to compute the probability ? Maximum Likelihood
Estimation (MLE) comes to our rescue. ] The aim of maximum likelihood estimation is to find
the parameter value(s) that makes the observed data most likely. Understanding the
difference between prediction and estimation is important at this point. Estimation differs
from prediction in the following way … In estimation problems, likelihood of the parameters
is estimated based on given data/observation vector. In prediction problems, probability is
used as a measure to predict the outcome from known parameters of a model.

Examples for “Prediction” and “Estimation” :

1) Probability of getting a “Head” in a single toss of a fair coin is . The coin is tossed 100
times in a row.Prediction helps in predicting the outcome ( head or tail ) of the toss
based on the probability.

2) A coin is tossed 100 times and the data ( head or tail information) is recorded. Assuming
the event follows Binomial distribution model, estimation helps in determining the
probability of the event. The actual probability may or may not be . Maximum
Likelihood Estimation estimates the conditional probability based on the observed data (
received data – ) and an assumed model.

Example of Maximum Likelihood Decoding:

Let and . Assuming Binomial distribution model for the event

with probability of error (i.e the reliability of the BSC is ), the Hamming
distance between codewords is . For binomial model,

where =the hamming distance between the received and the sent codewords n= number of
bit sent
= error probability of the BSC.
= reliability of BSC

Substituting and , then .

Here, Hamming distance is used to compute the probability. So the decoding can be called as
“minimum distance decoding” (which minimizes the Hamming distance) or “maximum
likelihood decoding”. Euclidean distance may also be used to compute the conditional
probability.

As mentioned earlier, in practice is not known at the receiver. Lets see how to
estimate when is unknown based on the binomial model.

Since the receiver is unaware of the particular corresponding to the received, the receiver
computes for each codewordin . The which gives the maximum
probability is concluded as the codeword that was sent.

Vitterbi Decoding Algorithm

· An efficient search algorithm

· Performing ML decoding rule.
· Reducing the computational complexity.

Basic concept
· Generate the code trellis at the decoder
· The decoder penetrates through the code trellis level by level in search for the
transmitted code sequence
· At each level of the trellis, the decoder computes and compares the metrics of all
the partial paths entering a node
· The decoder stores the partial path with the larger metric and eliminates all the
other partial paths. The stored partial path is called the survivor.
Equalization techniques

Adaptive equalization

• An equalizer is a filter that compensates for the dispersion effects of a channel. Adaptive
equalizer can adjust its coefficients continuously during the transmission of data.
Pre channel equalization
· requires feed back channel
· causes burden on transmission.

Post channel equalization

Achieved prior to data transmission by training the filter with the guidance of a training
sequence transmitted through the channel so as to adjust the filter parameters to optimum
values.

Adaptive equalization

It consists of tapped delay line filter with set of delay elements, set of adjustable multipliers
connected to the delay line taps and a summer for adding multiplier outputs.
The output of the Adaptive equalizer is given by

Ci is weight of the ith tap Total number of taps are M .Tap spacing is equal to symbol
duration T of transmitted signal In a conventional FIR filter the tap weights are constant and
particular designed response is obtained. In the adaptive equaliser the Ci's are variable and
are adjusted by an algorithm.

Two modes of operation

1. Training mode
2. Decision directed mode

Mechanism of adaptation
Training mode

A known sequence d(nT) is transmitted and synchronized version of it is generated in the

receiver applied to adaptive equalizer. This training sequence has maximal length PN
Sequence, because it has large average power and large SNR, resulting response sequence
(Impulse) is observed by measuring the filter outputs at the sampling instants. The difference
between resulting response y(nT) and desired response d(nT)is error signal which is used to
estimate the direction in which the coefficients of filter are to be optimized using algorithms.
Linear equalization

● MLSE-based equalization is prohibitively complex – the complexity grows

exponentially with the channel length (ML)

● Instead of the Viterbi algorithm, a simple digital filter can be employed to perform
theequalization

● It has suboptimum performance but the complexity (length of the equalization filter)
is now linear with channellength.

● Cj are the coefficients (taps) of the equalizationfilter

● Determined by using differentcriteria
 SYNCHRONIZATION AND CARRIER RECOVERY OF DIGITAL
MODULATION:

 In a digital communication system, the output of thedemodulator must be sampled

periodically, once per symbol interval, in order to recover the transmitted
information.

 Since the propagation delay from the transmitter to the receiveris generally unknown
at the receiver, symbol timing must be derived from the received signal in order to
synchronously sample the output of the demodulator.

 The propagation delay in the transmitted signal also results ina carrier offset, which
must be estimated at the receiver if the detector is phase-coherent.

 Symbol synchronization is required in every digital communication system which

transmits information synchronously.

 Carrier recovery is required if the signal is detected coherently.

 There are two criteria that are widely applied to signal parameter estimation: the
maximum-likelihood (ML)criterion and the maximum a posteriori probability
(MAP) criterion

 In the MAP criterion, is modeled as random and

characterized by an a priori probability density
functionp().
 In the ML criterion, is treated as deterministic
but unknown.
Binary PSK signal demodulator and detector

 The carrier phase estimate is us edin generating the reference signal g(t)cos(2

fct+) for
the correlator.

 The symbol synchronizer controls the sampler and the output of the signal pulse generator.

 If the signal pulse is rectangular, then the signal generator can be eliminated.
M-ary PAM signal demodulator and detector:

 A single correlator is required, and the detector is an amplitude detector, which compares the
received signal amplitude with the possible transmitted signal amplitudes.
 The purpose of an automatic gain control (AGC) is to eliminate channel gain variations,
which would affect the amplitude detector.
 The AGC has a relatively long time constant, so that itdoes not respond to the signal
amplitude variations that occur on a symbol-by-symbolbasis.
 The AGC maintains a fixed average (signal plusnoise) power at itsoutput.

QAM signal demodulator and detector

 An AGC is required to maintain a constant averagepower signal at the input to

thedemodulator.
 The demodulator is similar to a PSK demodulator, in thatboth generate in-phase and
 quadrature signal samples (X, Y) for the detector.
 The detector computes the Euclidean distance between the received noise-corrupted
signal point and the M possible transmitted points, and selects the signal closest to the
received point.

Two basic approaches for dealing with carrier synchronization at the receiver,
 One is to multiplex, usually in frequency, a special signal, called a pilot signal,
that allows the receiver to extract and to synchronize its local oscillator to the
carrier frequency and phase of the received signal.
 When an unmodulated carrier component is transmitted along with the
information-bearing signal, the receiver employs a phase-locked loop (PLL) to
acquire and track the carrier component.
 The PLL is designed to have a narrow bandwidth so that it is not significantly
affected by the presence of frequency components from the information-bearing
signal.
 The second approach, which appears to be more prevalentin practice, is to derive
the carrier phase estimate directly from the modulatedsignal.
 This approach has the distinct advantage that the total transmitter power is
allocated to the transmission of the information-bearing signal.
Course Outcomes:
At the end of the unit the students will be able to -Comparedifferentdigital
demodulationschemesfortheirefficiency, Investigate synchronizing and carrier recovery
for digital demodulator and also concepts of equalization.