Unit -1

Digital communication is the process of devices communicating information digitally.

Data Communication –
Data Communication refers to the sharing or transfer of collection of facts, figures, etc. between devices capable of
such exchanges using some of the others communication mediums. Whenever we communicate we simply share
facts, ideas, etc. in mutually agreed-upon language and speed with the maximum accuracy possible. The same is the
case in data communication, here the effectiveness of Data Communication is determined by correctness in delivery,
the accuracy of transfer, timeliness, and lesser variation in packet arrival times. 
Data Communication has five basic components – 
 Message – 
It is the data or the information that is to be exchanged between two points. Often in the real world, we
share messages in the form of texts, numbers, pictures, audio, and videos.
 Sender – 
It is the node (device) that is intended to send the information being transferred or communicated. It has
mechanisms of its own that make information encoded in a format that is feasible and secure to transfer on
the transmission medium accurately and timely. 
 Receiver – 
The device that holds the responsibility of receiving the encoded information and decoding it into a certain
format accurately and sending the feedback is the receiver.
 Transmission Medium – 
It is a path such as a cable that serves as traveling media on which the message is transferred from the
sender to the receiver end.  
 Protocol – 
Protocol is the rules that are agreed between sender and receiver which govern the entire exchange process.
These protocols make the communication possible between two devices without them may be connected
but won’t be communicating.

Digital Communication –
Digital communication involves the physical transfer of data and information through a suitable communication
channel. This exchange of information in the form of discrete messages can take place Point to Point or Point to
Multipoint. Conventionally analog signals have been used to establish a connection and start communication over
large distances but this made the signals suffer many losses such as distortion, interference, and even security
breaches. These problems were overcome by the usage of digital signals over analog ones. Analog signals are
digitized using different techniques. Communication using these digital signals makes it more accurate and less
vulnerable to losses or breaches.
Digital Communication has the following basic components –
 Source – 
Any point of data origin that can offer a piece of information that may be analog or digital can consider as a
source.
 Input Transducer – 
The transducer at the input end performs the task of converting a physical input from point of source into an
electrical signal. It is also equipped with an analog to a digital converter where the analog signal needs to be
digitized for further transfer or exchange. This digital signal is nothing but a sequence of binary numbers, i.e.
0s and 1s.
 Source Encoder – 
At the source encoder the compression of the digitized signal data is done to minimize the number of bits
 maintaining the correctness of the data. For the compression removal of redundancy is an important step. In
this way, the effective utilization of the bandwidth is ensured.
 Channel Encoder – 
The compressed data from the source code is encoded for error correction using a channel encoder. To
prevent the alteration of the data signal during transmission because of noise in the channel, the channel
encoder adds few redundant bits to the data being transmitted called the error-correcting bits.
 Digital Modulator – 
The transmitting channel modulates the signal to be transmitted and also the signal can be converted from
digital format to analog hence making it ready to travel along with the medium.
 Channel – 
Channel is a transmission medium through which the signal transmission takes place.
 Digital Demodulator – 
On the receiver side, the received signal is demodulated and also conversion from analog to digital takes
place using a digital demodulator. Signal reconstruction takes place here.
 Channel Decoder – 
Error corrections are done using redundancy bits after sequence detection by the channel decoder.
 Source Decoder – 
By using sampling and quantization the final signal is again converted into digital format.
 Output Transducer – 
Here the output signal is converted from the electrical form to the physical form.
 Output Signal – 
Signal that is produced after the complete process as the output is the output signal.

Below is the difference between Data Communication and Digital Communication: 

Data Communication Digital Communication

The exchange of information between a sender and Digital Communication deals with the physical transfer of
receiver is termed as Data Communication. It is information over a medium and the various conversions
mainly concerned with governing rules and the and other processes involved such as encoding, decoding,
transfer of data through a suitable medium. compression and error correction of signals.

It has several components serving different functions such

It has five basic components namely message, sender, as Analog to Digital Conversion and vice-versa, removing
receiver, transmission medium, and agreed-upon redundancy bits, add error-correcting bits, modulation and
protocols. demodulation.

Signal type or information format is not a major

concern here. The exchange is done on a common Signal type is an important factor in each step of the entire
agreed upon speed and type. process of digital communication.

Rules are not the deciding authority for the transference of

The transfer takes place according to a governing set information. Successful and accurate exchange with high
of rules. efficiency is the essential part of the process.

The communication that occurs in our day-to-day life is in the form of signals. These signals, such as sound
signals, generally, are analog in nature. When the communication needs to be established over a distance,
then the analog signals are sent through wire, using different techniques for effective transmission.
The Necessity of Digitization
The conventional methods of communication used analog signals for long distance communications, which suffer
from many losses such as distortion, interference, and other losses including security breach.
In order to overcome these problems, the signals are digitized using different techniques. The digitized signals allow
the communication to be more clear and accurate without losses.
The following figure indicates the difference between analog and digital signals. The digital signals consist
of 1s and 0s which indicate High and Low values respectively.

Advantages of Digital Communication

As the signals are digitized, there are many advantages of digital communication over analog communication, such
as −
 The effect of distortion, noise, and interference is much less in digital signals as they are less affected.
 Digital circuits are more reliable.
 Digital circuits are easy to design and cheaper than analog circuits.
 The hardware implementation in digital circuits, is more flexible than analog.
 The occurrence of cross-talk is very rare in digital communication.
 The signal is un-altered as the pulse needs a high disturbance to alter its properties, which is very difficult.
 Signal processing functions such as encryption and compression are employed in digital circuits to maintain
the secrecy of the information.
 The probability of error occurrence is reduced by employing error detecting and error correcting codes.
 Spread spectrum technique is used to avoid signal jamming.
 Combining digital signals using Time Division Multiplexing TDM is easier than combining analog signals using
Frequency Division Multiplexing FDM.
 The configuring process of digital signals is easier than analog signals.
 Digital signals can be saved and retrieved more conveniently than analog signals.
 Many of the digital circuits have almost common encoding techniques and hence similar devices can be used
for a number of purposes.
 The capacity of the channel is effectively utilized by digital signals.

Elements of Digital Communication

The elements which form a digital communication system is represented by the following block diagram for the ease
of understanding.
Following are the sections of the digital communication system.
 Source
The source can be an analog signal. Example: A Sound signal
 Input Transducer
This is a transducer which takes a physical input and converts it to an electrical signal (Example: microphone). This
block also consists of an analog to digital converter where a digital signal is needed for further processes.
A digital signal is generally represented by a binary sequence.
 Source Encoder
The source encoder compresses the data into minimum number of bits. This process helps in effective utilization of
the bandwidth. It removes the redundant bits unnecessaryexcessbits,i.e.,zeroesunnecessaryexcessbits,i.e.,zeroes.
 Channel Encoder
The channel encoder, does the coding for error correction. During the transmission of the signal, due to the noise in
the channel, the signal may get altered and hence to avoid this, the channel encoder adds some redundant bits to
the transmitted data. These are the error correcting bits.
 Digital Modulator
The signal to be transmitted is modulated here by a carrier. The signal is also converted to analog from the digital
sequence, in order to make it travel through the channel or medium.
 Channel
The channel or a medium, allows the analog signal to transmit from the transmitter end to the receiver end.
 Digital Demodulator
This is the first step at the receiver end. The received signal is demodulated as well as converted again from analog
to digital. The signal gets reconstructed here.
 Channel Decoder
The channel decoder, after detecting the sequence, does some error corrections. The distortions which might occur
during the transmission, are corrected by adding some redundant bits. This addition of bits helps in the complete
recovery of the original signal.
 Source Decoder
The resultant signal is once again digitized by sampling and quantizing so that the pure digital output is obtained
without the loss of information. The source decoder recreates the source output.
 Output Transducer
This is the last block which converts the signal into the original physical form, which was at the input of the
transmitter. It converts the electrical signal into physical output (Example: loud speaker).
 Output Signal
This is the output which is produced after the whole process. Example − The sound signal received.
What is Modulation and Different Types
What is Modulation?
Modulation is a process of changing the characteristics of the wave to be transmitted by superimposing the message
signal on the high-frequency signal. In this process; video, voice and other data signals modify high-frequency signals
– also known as the carrier wave. This carrier wave can be DC or AC or pulse chain depending on the application
used. Usually, a high-frequency sine wave is used as a carrier wave signal.
These modulation techniques are classified into two major types: analog and digital or pulse modulation.

Why Modulation is Used in Communication?

 In the modulation technique, the message signal frequency is raised to a range so that it is more useful for
transmission.
 In signal transmission, the signals from various sources are transmitted through a common channel
simultaneously by using multiplexers. If these signals are transmitted simultaneously with a certain
bandwidth, they cause interference. To overcome this, speech signals are modulated to various carrier
frequencies in order for the receiver to tune them to the desired bandwidth of his own choice within the
range of transmission.
 Another technical reason is antenna size; the antenna size is inversely proportional to the frequency of the
radiated signal. The order of the antenna aperture size is at least one by a tenth of the wavelength of the
signal. Its size is not practicable if the signal is 5 kHz; therefore, raising frequency by modulating process will
certainly reduce the height of the antenna.
 Modulation is important to transfer the signals over large distances since it is not possible to send low-
frequency signals for longer distances.
 Similarly, modulation is also important to allocate more channels for users and to increase noise immunity.

Modulating Signal
This signal is also termed as a message signal. It holds the data that has to be transmitted and so this termed as
message signal. It is considered as the baseband signal where it undergoes a modulation process to get broadcasted
or communicated. Because of this, it is the modulating signal.

Carrier Signal
This is the high range of frequency signal which is with specific amplitude, frequency, and phase levels, but it does
not hold any data. So, it is termed as carrier signal as it is an empty one. This is simply utilized to transmit the
message to the receiver section after the process of modulation.

Modulated Signal
The consequential signal that is obtained after the procedure of modulation is called a modulated signal. This is the
product of both the carrier and modulating signals.

Different Types of Modulation

The two types of modulation: analog and digital modulation techniques. In both the techniques, the baseband
information is converted to Radio Frequency signals, but in analog modulation, these RF communication (RF refers to
the frequencies that fall within the electromagnetic spectrum associated with radio wave propagation) signals are a
continuous range of values, whereas in digital modulation these are prearranged discrete states.
Analog Modulation
In this modulation, a continuously varying sine wave is used as a carrier wave that modulates the message signal or
data signal. The Sinusoidal wave’s general function is shown in the figure below, in which, three parameters can be
altered to get modulation – they are mainly amplitude, frequency, and phase, so the types of analog modulation are:
 Amplitude modulation (AM)
 Frequency modulation (FM)
 Phase modulation (PM)
In amplitude modulation, the amplitude of the carrier wave is varied in proportion to the message signal, and the
other factors like frequency and phase remain constant. The modulated signal is shown in the below figure, and its
spectrum consists of a lower frequency band, upper-frequency band, and carrier frequency components. This type of
modulation requires greater bandwidth, more power. Filtering is very difficult in this modulation.

Frequency modulation (FM) varies the frequency of the carrier in proportion to the message or data signal while
maintaining other parameters constant. The advantage of FM over AM is the greater suppression of noise at the
expense of bandwidth in FM. It is used in applications like radio, radar, telemetry seismic prospecting, and so on. The
efficiency and bandwidths depend on the modulation index and maximum modulating frequency.
In phase modulation, the carrier phase is varied in accordance with the data signal. In this type of modulation, when
the phase is changed it also affects the frequency, so this modulation also comes under frequency modulation.
Analog modulation (AM, FM, and PM) is more sensitive to noise. If noise enters into a system, it persists and gets
carried till the end receiver. Therefore, this drawback can be overcome by the digital modulation technique.
Digital Modulation
For better quality and efficient communication, the digital modulation technique is employed. The main advantages
of digital modulation over analog modulation include permissible power, available bandwidth, and high noise
immunity. In digital modulation, a message signal is converted from analog to digital message and then
modulated by using a carrier wave.
The carrier wave is keyed or switched on and off to create pulses such that the signal is modulated. Similar to the
analog, here the parameters like amplitude, frequency, and phase variation of the carrier wave decides the type of
digital modulation.
The types of digital modulation are based on the type of signal and application used such as Amplitude Shift
Keying, Frequency Shift Keying, Phase Shift Keying, Differential Phase Shift Keying, Quadrature Phase Shift Keying,
Minimum Shift Keying, Gaussian Minimum Shift Keying, Orthogonal Frequency Division Multiplexing, etc., as
shown in the figure.
Amplitude shift keying changes the amplitude of the carrier wave based on the baseband signal or message signal,
which is in digital format. It is used for low-band requirements and is sensitive to noise.
In frequency-shift keying, the frequency of the carrier wave is varied for each symbol in the digital data. It needs
larger bandwidths as shown in the figure. Similarly, the phase shift keying changes the phase of the carrier for each
symbol and it is less sensitive to noise.

Frequency Modulation
In order to create a frequency modulated wave, the frequency of the radio wave is varied in accordance with the
amplitude of the input signal.

Frequency Modulation

When the audio wave is modulated with that of the radio frequency carrier signal, then the generated frequency
signal will change its frequency level. The variation by which the wave moves upward and downward is to be noted.
This is termed as deviation and is generally represented as kHz deviation.
As an instance, when the signal has a deviation of either + or – 3kHz, then it is represented as ±3kHz. This means that
the carrier signal has up and downward deviation of 3kHz.
Broadcasting stations that need very high-frequency range in the frequency spectrum (in the range of 88.5 – 108
MHz), they need certainly a large amount of deviation which is nearly ±75 kHz. This is called wide-band frequency
modulation. The signals in this range hold the ability to assist the high quality of transmissions, whereas they require
higher bandwidth too. In general, 200kHz is permitted for every WBFM. And for narrowband FM, a deviation of ±3
kHz is enough.
While implementing an FM wave, it is more beneficial to know the effectivity range of the modulation. This stands as
the parameter in stating factors such as knowing the type of signal whether wide band or narrow band FM signal. It
also helps in making sure that the whole receivers or transmitters that are in the system are programmed to adapt to
the standardized range of modulation as this shows an impact on the factors such as the channel spacing, bandwidth
of the receiver, and others.
So, to signify the modulation level, modulation index and deviation ratio parameters are to be determined.

The different types of frequency modulation include the following.

Narrow band FM
 This is termed as the type of frequency modulation where the modulation index value is too minimal.
 When the modulation index value is < 0.3, then there will be an only carrier and corresponding sidebands
having bandwidth as twice the modulating signal. So, β ≤ 0.3 is called narrow band frequency modulation.
 The maximum range of modulating frequency is of 3 kHz
 The maximum frequency deviation value is 75 kHz

Wide band FM
 This is termed as the type of frequency modulation where the modulation index value is large.
 When the modulation index value is > 0.3, then there will be more than two sidebands having bandwidth as
twice the modulating signal. When the modulation index value increases, then the number of sidebands gets
increased. So, β > 0.3 is called narrow band frequency modulation.
 The maximum range of modulating frequencies is in between 30 Hz – 15 kHz
 The maximum frequency deviation value is 75 kHz
 This frequency modulation needs a higher bandwidth range which is almost 15 times ahead of the narrow
band frequency modulation.

The other types of modulation techniques used in the communication system are:
 Binary phase shift keying
 Differential phase-shift keying
 Differential quadrature phase shift keying
 Offset quadrature phase shift keying
 Audio FSK
 Multi FSK
 Dual-tone FSK
 Minimum shift keying
 Gaussian minimum shift keying
 Trellis coded type of modulation

Advantages of Various Types of Modulation

For transmission purposes, the size of the antenna has to be very large before the modulation technique was not
proposed. The level of communication gets restricted as there will be no long-distance communications having zero
levels of distortions.
So, with the development of modulation, there are many benefits of utilizing communication systems. And the
advantaged of modulation are:
 The size of the antenna can be lessened
 There happens no kind of signal consolidation
 The range of communication is enhanced
 There will be the possibility of multiplexing
 One can adjust bandwidth as per the requirements
 The quality of reception gets increased
 Better performance and effectivity

Applications of Various Types of Modulation

There is an extensive range of various modulation techniques and those are:
 Implemented in music mixing, and magnetic tape recording systems
 To track EEG monitoring for newly born children
 Used in telemetry
 Used in radar
 FM broadcasting techniques
What is Amplitude Modulation, Derivations, Types,and
Applications
What is Amplitude Modulation?
The amplitude modulation definition is, an amplitude of the carrier signal is proportional to (in accordance with) the
amplitude of the input modulating signal. In AM, there is a modulating signal. This is also called an input signal or
baseband signal (Speech for example). This is a low-frequency signal as we have seen earlier. There is another high-
frequency signal called carrier. The purpose of AM is to translate the low-frequency baseband signal to a higher freq
signal using the carrier. As discussed earlier, high-frequency signals can be propagated over longer distances than
lower frequency signals. The derivatives of amplitude modulation include the following.

The modulating Signal (Input Signal) Vm = Vm sin ωmt

Where Vm is the instantaneous value and Vm is the maximum value of the modulating (input) signal.
fm is the frequency of the modulating (input) signal and ωm = 2π fm
The Carrier Signal Vc = Vc sin ωct
Where Vc is the instantaneous value and Vc is the maximum value of the carrier signal, fc is the frequency of the
carrier signal and ωc = 2π fc.
The amplitude modulation equation is,
VAM = Vc +Vm= Vc + Vm sin ωmt
vAM = VAM sin θ = VAM sin ωct
= (Vc + Vm sin ωmt) sin ωct
= Vc (1+m sin ωmt) sin ωct where m is given by m = Vm/Vc

Modulation Index
Modulation Index is defined as the ratio of the amplitude of the modulating signal and the amplitude of the carrier
signal. It is denoted by ‘m’
Modulation Index m = Vm/Vc
Modulation Index is also known as Modulation factor, Modulation coefficient or degree of modulation
“m” shall have a value between 0 and 1.
“m” when expressed as a percentage is called % modulation.
Vm = Vmax-Vmin/2
Vc = Vmax-Vm
Vc = Vmax- (Vmax-Vmin/2) = Vmax + Vmin/2
Therefore, Vm/Vc = (Vmax-Vmin/ Vmax + Vmin)

Critical Modulation
It happens when modulation Index (m) =1. Note, during critical modulation Vmin =0
M = Vm/Vc = (Vmax-Vmin/ Vmax + Vmin) = (Vmax/Vmax) = 1
Substitute V m = 0 Therefore at critical modulation m = Vm/Vc
Substitute m = 1. Therefore at critical modulation Vm = Vc

What is Over Modulation and Sidebands of AM?

This can occur when m>1
That is (Vm / Vc) > 1. Therefore Vm > Vc. In other words, the modulating signal is greater than the carrier
signal.
The AM signal will generate new signals called sidebands, at frequencies other than fc or fm.
We know that VAM = (Vc + m Vm sin ωmt) sin ωct
We also know that m = Vm / Vc. Therefore Vm = m.Vc

Sidebands of AM

Therefore,
Case1: Both input signal and carrier signal are sine waves.
  VAM = (Vc + m Vc sin ωmt) sin ωct
          = Vc sin ωct + m Vc sin ωmt . Sin ωct
Recall SinA SinB = 1/2 [ cos (A-B) – cos (A + B)]
Therefore VAM = Vc sin ωct + [ mVc/2 cos (ωc – wm)t] ─ [mVc/2 cos (ωc + wm)t]
Where Vc sin ωct is carrier
mVc/2 cos (ωc – wm)t is lower side band
mVc/2 cos (ωc + wm)t I supper sideband
Therefore AM signal has three frequency components, Carrier, Upper Sideband and Lower Side Band.
Case 2: Both input signal and carrier signal are cos waves.
VAM = (Vc + m Vc cos ωmt) cos ωct
= Vc cos ωct + mVc cos ωmt. cos ωct
Recall Cos A Cos B =1/2  [cos (A ─B) + cos (A + B)]
Therefore VAM = Vc cos ωct + [mVc/2 cos (ωc – wm)t] + [mVc/2 cos (ωc + wm)t]
Where Vc cos ωct
mVc/2 cos (ωc – wm)t is lower sideband
mVc/2 cos (ωc + wm)t supper sideband
Therefore AM signal has three frequency components, Carrier, Upper Sideband and lower side Band

Bandwidth of AM
The bandwidth of a complex signal like AM is the difference between its highest and lowest frequency components
and is expressed in Hertz (Hz). Bandwidth deals with only frequencies.
As shown in the following figure
Bandwidth = (fc – fm) – (fc + fm) = 2 fm
The power levels in carrier and sidebands

Power Levels in Carrier and Sidebands

There are three components in the AM wave. Unmodulated carrier, USB & LSB.
Total Power of AM is = Power in the
Unmodulated carrier + Power in USB +Power in LSB
If R is the load, then Power in AM = V2c/R + VLSB2/R + VUSB2/2

Carrier Power
Peak carrier Power = V2c/R
Peak Voltage = Vc, therefore RMS voltage = Vc/√2
RMS carrier power =1/R [Vc/√2]2= V2c/ 2R
RMS Power in Side bands
PLSB = PUSB = VSB2/R = 1/R [mVc/2/√2]2
 = m2 (Vc)2 / 8R= m2/4 X V2c/2R
RMS Power in side Bands

We know that V2c/2R =Pc

Therefore PLSB = m2/4 x Pc
Total power = v2c/2R + m2Vc2/8R + m2Vc2/8R
v2c/2R [ 1 + (m2/4) + (m2/4)] =  Pc [ 1 + (m2/4) + (m2/4)]
PTotal = Pc [ 1 + m2/2 ]
Modulation Index in terms of Total Power (PTotal) and Carrier Power (Pc)
PTotal = Pc [1+m2/2]
PTotal/ Pc = [1+m2/2]
m2/2 = PTotal/Pc – 1
m = √2 (PTotal/Pc – 1)
Transmission Efficiency
In AM there are three power components Pc, PLSB and PUSB
Out of these Pc is an unmodulated carrier. It is wasteful as it carries no information at all.
The two sidebands carry, all the useful information and therefore useful power is spent only in Sidebands

Efficiency (η)
A ratio of transmitted power which contains the useful information (PLSB + PUSB) to the total transmitted power .
Transmission efficiency = (PLSB + PUSB) / (PTotal)
η = Pc [m2/4 + m2/4] / Pc [1 = m2/2] = m2/2+m2
η % = (m2/2+m2) X 100

 Amplitude Demodulation
The inverse of the modulator and it recovers (decodes) the original signal (what was the modulating signal at the
transmitter end) from the received AM signal.

 Envelop Detector
AM is a simple wave, and the detector is a demodulator. It recovers the original signal (what was the modulating
signal at the transmitter end) from the received AM signal. The detector consists of a simple half-wave
rectifier which rectifies the received AM signal. This is followed by a low pass filter which removes (bypasses) the
high-frequency carrier waveform the received signal. The resultant output of the low pass filter will be the original
input (modulating) signal.
 Envelop Detector

The incoming AM signal is transformer coupled HW rectifier conducts during positive cycles of AM and cuts off
negative cycles of AM. Filter capacitor C filters (bypasses) the high-frequency carrier (fc) and allows only the lower
frequency (fm). Thus, the filter output is the original input (modulating) signal.

Types of Amplitude Modulation

1) Double sideband-suppressed carrier (DSB-SC) modulation

 The transmitted wave consists of only the upper and lower sidebands
 But the channel bandwidth requirement is the same as before.

2) Single sideband (SSB) modulation

 The modulation wave consists only of the upper sideband or the lower sideband.
 To translate the spectrum of the modulating signal to a new location in the frequency domain.

3) Vestigial sideband (VSB) modulation

 One sideband is passed almost completely and just a trace of the other sideband is retained.
 The required channel bandwidth is slightly in excess of the message bandwidth by an amount equal to the
width of the vestigial sideband.

Advantages & Disadvantages of Amplitude Modulation

 The advantages of amplitude modulation include the following.
 Amplitude modulation is economical as well as easily obtainable
 It is so simple to implement, and by using a circuit with fewer components it can be demodulated.
 The receivers of AM are inexpensive because it doesn’t require any specialized components.
The disadvantages of amplitude modulation include the following.
 The efficiency of this modulation is very low because it uses a lot of power
 This modulation uses amplitude frequency several times to modulate the signal by a carrier signal.
 This declines the original signal quality on the receiving end & causes troubles in the signal quality.
 AM systems are susceptible toward the generation of noise generation.
 The applications of amplitude modulation limits to VHF, radios, & applicable one to one communication only
What is a Frequency Modulation?
The frequency modulation can be defined as; the frequency of the carrier signal is varied proportional to (in
accordance with) the Amplitude of the input modulating signal. The input is a single tone sine wave. The carrier and
the FM waveforms also are shown in the following figure.

Frequency Modulation Generation

The frequency of a carrier (fc) will increase as the amplitude of modulating (input) signal increases. The carrier
frequency will be maximum (fc max) when the input signal is at its peak. The carrier deviates maximum from its
normal value. The frequency of a carrier will decrease as the amplitude of the modulating (input) signal decreases.
The carrier frequency will be minimum (fc min) when the input signal is at its lowest. The carrier deviates minimum
from its normal value. The frequency of the carrier will be at its normal value (free-running) fc when the input signal
value is 0V. There is no deviation in the carrier. The figure shows the frequency of the FM wave when the input is at
its max, 0V, and at its min.
The frequency modulation block diagram is shown below. The message signal holds the specific data whereas the
next signal has no data known as the carrier signal.
The modulation of these signals will result in an FM modulated signal. This signal is more essential because the
frequency of this signal will flow up & down depending on the amplitude of the signal. So this frequency change can
be represented in kHz (kilohertz). For instance, once the frequency variation is 3 kHz up & down, then it is signified
like ±3 kHz.

FM History
At the time of radio, the static was the main problem &the way everybody attempted to decrease the static effects
was to decrease the bandwidth. So in this method, less noise was received through the receiver. Edwin Armstrong
was an American engineer, so he examining this problem & whether FM, rather than AM might give a benefit.
In 1928, he simply started to expand the idea with the help of FM & he increased the BW instead of decreasing it.
But, for different reasons, his ideas don’t accept by others. Immediately, he approached Radio Corporation of
America or RCA, they were impressed by his ideas, but RCA was simply focusing on TV, so they don’t want to redirect
any resource to a new type of broadcasting.
After many troubles, he launched a radio station in the year 1939 to exhibit the efficiency of FM. To hold this & other
stations following a band of frequencies among 42 MHz & 50 MHz frequencies. But after the war, the FCC within the
USA modified the assigned frequency band between 88 MHz and 108 MHz.
Even though, there was some basic pain as thousands of radios had been sold and the band was accepted worldwide
that is VHF FM band at present. In addition to this, a kind of narrowband FM became famous for UHF & VHF mobile
communications.

Frequency Deviation
 The amount of change in the carrier frequency produced, by the amplitude of the input modulating signal, is
called frequency deviation.
 The Carrier frequency swings between fmax and fmin as the input varies in its amplitude.
 The difference between fmax and fc is known as frequency deviation. fd = fmax – fc
 Similarly, the difference between fc and fmin also is known as frequency deviation. fd = fc –fmin
 It is denoted by Δf. Therefore Δf = fmax – fc = fc – fmin
 Therefore fd = fmax – fc = fc – fmin
 Frequency of Carrier
Modulating Signal Amplitude Deviation
0V 100 MHz Nil (Center frequency)

+2 V 105 MHz + 5 MHz

─2V 95 MHz – 5 MHz

Freq deviation = 105 -100 = 5 MHz (or) Freq deviation = 95-100 = -5 MHz

Frequency Modulation Equation
The FM equation include the following
v = A sin [ wct + (Δf / fm) sin wmt ]
= A sin [ wct + mf sin wmt ]
A = Amplitude of the FM signal. Δf = Frequency deviation
mf = Modulation Index of FM
mf = ∆f/fm
mf is called the modulation index of frequency modulation.
wm = 2π fm wc = 2π fc

What is the Modulation Index of Frequency Modulation?

The modulation index of FM is defined as the ratio of the frequency deviation of the carrier to the frequency of the
modulating signal
mf = Modulation Index of FM = ∆f/fm

Frequency Modulation in Communication Systems

In telecommunications, there are two types of frequency modulation techniques used like analog frequency
modulation & digital frequency modulation.
In analog frequency modulation, the data signal can be modulated through a continuously changing sine carrier
signal. This carrier signal includes different properties like frequency, amplitude & phase which are mainly used for
creating AM & PM.
The digital frequency modulation can be categorized as either FSK (Frequency Shift Key), ASK (Amplitude Shift Key),
or PSK (Phase Shift Key) which works like analog. Analog modulation technique is normally used for AM, FM & short-
wave broadcasting whereas digital modulation technique used to transmit binary signals like 0 & 1.
The techniques used in FM are the varactor diode oscillator and phase-locked loop. In the varactor diode oscillator
technique, the diode is arranged in the circuit to change the frequency. So this technique provides simply
narrowband transmissions. In the PLL technique, it provides an outstanding FM. So in this technique, phases are
constrained in the loop to change the frequency.

FM in Vibration Analysis
The process of measuring as well as analyzing the vibration signals levels, patterns otherwise machinery frequency
for noticing irregular vibration actions & estimate the complete machine’s strength along with their components.
This kind of analysis is particularly very helpful by revolving machinery, where exist fault devices that may reason for
amplitude & frequency modulation deviations.
The demodulation method directly detects these modulation frequencies, so it is used for recovering the data from
the modulated carrier signal.

The Bandwidth of Frequency Modulation Signal

Bandwidth is one of the main elements of FM signal. In FM signal, the sidebands will extend either side which will
extend to infinity; however, the strength of them drops away. Auspiciously, it is the potential to restrict the BW of an
FM signal without changing its value excessively.
Recall, the bandwidth of a complex signal like FM is the difference between its highest and lowest
frequency components, and is expressed in Hertz (Hz). Bandwidth deals with only frequencies. AM has only two
sidebands (USB and LSB) and the bandwidth was found to be 2 fm.
In FM it is not so simple. FM signal spectrum is quite complex and will have an infinite number of sidebands as shown
in the figure. This figure gives an idea, how the spectrum expands as the modulation index increases. Sidebands are
separated from the carrier by fc ± fm, fc ± 2fm, fc ± 3fm, and so on.

The bandwidth of FM Signal

Only the first few sidebands will contain the major share of the power (98% of the total power) and therefore only
these few bands are considered to be significant sidebands.
As a rule of thumb, often termed as Carson’s Rule, 98% of the signal power in FM is contained within a bandwidth
equal to the deviation frequency, plus the modulation frequency-doubled.
Carson’s rule: Bandwidth of FM BWFM = 2 [ Δf + fm ].
= 2 fm [ mf + 1 ]

FM is known as Constant Bandwidth System. Why?

The frequency modulation is known as a constant bandwidth system and an example of this system is given below.
 Δf = 75 KHz fm = 500 Hz BWFM = 2 [75 + (500/1000)] KHz = 151.0 KHz
 Δf = 75 KHz fm = 5000 Hz BWFM = 2 [75 + (5000/1000)] KHz = 160.0 KHz
 Δf = 75 KHz fm = 10000 Hz BWFM = 2 [75 + (10000/1000)] KHz = 170.0 KHz
 Although modulating frequency increased 20 times (50 Hz to 5000 Hz), deviation increased only marginally
(151 KHz to 170 KHz). Hence FM is known as a constant bandwidth system.
 Commercial FM (Carson’s Rule.)
 Max freq deviation = 75 KHz
 Max Modulating freq = 15 KHz
 BWFM = 2 [ 75 + 15 ] = 180.0 KHz

Difference between Amplitude Modulation and Frequency Modulation

Amplitude Modulation Frequency Modulation

In Amplitude modulation, the amplitude of a carrier signal In frequency modulation, the carrier wave frequency can
changed based on the data signal. AM radio broadcast be changed based on the signal that holds data. The radio
signals utilize low-carrier frequencies to travel long
distances. Sometimes, amplitude modulation signals are signals include high BW as compared to AM radio signals.
capable of bouncing off the ionosphere. As compared to These signals assist to provide good sound quality. FM
FM, the distance traveled through the AM is high. also allows sending stereo signals.
In the mid-1870s, the first audio transmission was Fm was developed in the year 1930 in the US, by Edwin
developed Armstrong.
In FM, the radio signal is known as a carrier signal,
In AM, the radio signal is known as a carrier signal & both however, the amplitude, as well as phase, remain the
the phase & frequency remain the same same
More liable to noise Less liable to noise
The sound clarity of AM is poor, however, can transmit
long distances FM has high BW including good sound quality
The FM frequency ranges from 88 MHz – 108 MHz in the
The AM frequency ranges from 535 kHz – 1705 kHz higher spectrum
The modulation index of AM ranges from 0 to 1 The modulation index of FM is higher than 1
It includes simply two sidebands It includes a number of sidebands
It has an easy circuit It has a difficult circuit
In AM, the carrier signal’s amplitude can be changed to In FM, the carrier signal’s frequency can be changed to
transmit the information. transmit the information
It has less bandwidth like 10 kHz. It has high bandwidth like 200 kHz
AM operates in the MF (medium frequency) & HF( high
frequency). FM works with very high frequency
The key differences between AM and FM include the following.
 Equation for FM: V= A sin [ wct +Δf / fm sin wmt ] = A sin [ wct + mf sin wmt ]
 Equation for AM = Vc ( 1 + m sin ωmt ) sin ωct where m is given by m = Vm / Vc
 In FM, the Modulation Index can have any value greater than 1 or less than one
 In AM, the Modulation Index will be between 0 and 1
 In FM, carrier amplitude is constant.
 Therefore transmitted power is constant.
 Transmitted power does not depend on the modulation index
 Transmitted power depends on the modulation index
 PTotal = Pc [ 1+ (m2/2) ]
 The number of significant sidebands in FM is large.
 Only two sidebands in AM
 A bandwidth of FM depends on the modulation index of FM
 Bandwidth does not depend on the modulation index of AM. Always 2 sidebands. BW of AM is 2 fm
 FM has better noise immunity.FM is rugged/robust against noise. The quality of FM will be good even in the
presence of noise.
 In AM, quality is affected seriously by noise
 The bandwidth required by FM is quite high.FM bandwidth = 2 [Δf + fm].
 The bandwidth required by AM is less (2 fm)
 Circuits for FM transmitter and receiver are very complex and very expensive.
 Circuits for AM transmitter and receiver are simple and less expensive

Advantages of Frequency Modulation

 Less noise and interference
 Service areas are well defined for specified transmitter power.
 As compared to amplitude modulation, FM includes low power consumption.
 The radiated power is less.
 Guard bands separate nearby FM channels.
 Lesser geographical interference among adjacent stations.
 Enhanced S/N (signal to noise) ratio like 25dB with respect to manmade intrusion
 Modulation technique is easily applied at a low power phase of the transmitter:
  It is the potential to employ efficient RF amplifiers including frequency modulated signals.

Disadvantages of Frequency Modulation

 High equipment cost is high
 High bandwidth
 The receiving area of the FM signal is small.
 The antennas for FM systems should be kept close for better communication
 Much more Bandwidth (as much as 20 times as much).
 More complicated receiver and transmitter.
 FM has poorer spectral efficiency than some other modulation formats:
 Requires more complicated demodulator:
 Some other modes have higher data spectral efficiency:
 Sidebands expand to infinity any side
 The spectral efficiency of FM is poor as compared to other modulation methods
 It uses a more complicated demodulator:
 Other modes include high data spectral efficiency
 Sidebands extend to infinity on either side

What is Pulse Code Modulation and Demodulation?

Pulse code modulation is a method that is used to convert an analog signal into a digital signal so that a modified
analog signal can be transmitted through the digital communication network. PCM is in binary form, so there will be
only two possible states high and low(0 and 1). We can also get back our analog signal by demodulation. The Pulse
Code Modulation process is done in three steps Sampling, Quantization, and Coding. There are two specific types
of pulse code modulations such as differential pulse code modulation(DPCM) and adaptive differential pulse code
modulation(ADPCM).

Pulse Code Modulation Block Diagram

The basic elements of PCM mainly include the transmitter section and receiver section. The pulse code modulation
steps are discussed below

Here is a block diagram of the steps which are included in PCM. In sampling, we are using a PAM sampler that is
Pulse Amplitude Modulation Sampler which converts continuous amplitude signal into Discrete-time- continuous
signal (PAM pulses). The basic block diagram of PCM is given below for better understanding.
To get a pulse code modulated waveform from an analog waveform at the transmitter end (source) of a
communications circuit, the amplitude of the analog signal samples at regular time intervals. The sampling rate or
the number of samples per second is several times the maximum frequency. The message signal converted into the
binary form will be usually in the number of levels which is always to a power of 2. This process is called
quantization.
At the receiver end, a pulse code demodulator decodes the binary signal back into pulses with the same quantum
levels as those in the modulator. By further processes, we can restore the original analog waveform.

Low Pass Filter

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 {aliasing is an effect that causes different signals to
become indistinguishable (or aliases of one another) when sampled} of the message signal.

Sampler
This is the technique which helps to collect the sample data at instantaneous values of 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 LPF (low pass filter), Sampler, and
Quantizer will act as an analog to digital converter. Encoding minimizes the bandwidth used.

Regenerative Repeater
This section increases the signal strength. The output of the channel also has one regenerative repeater circuit, to
compensate the signal loss and reconstruct the signal, and also to increase its strength.

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 given analog signal, codes it and samples it, and then transmits
it in an analog form. This whole process is repeated in a reverse pattern to obtain the original signal.

Pulse Code Modulation Theory

The above block diagram describes the whole process of PCM. The source of the continuous-time message signal is
passed through a low pass filter and then sampling, Quantization, Encoding will be done. We will see each in detail
step by step.

Sampling
Sampling is a process of measuring the amplitude of a continuous-time signal at discrete instants, converts the
continuous signal into a discrete signal. For example, conversion of a sound wave to a sequence of samples. The
Sample is a value or set of values at a point in time or it can be spaced. Sampler extract samples of a continuous
signal, it is a subsystem ideal sampler produces samples that are equivalent to the instantaneous value of the
continuous signal at the specified various points. The Sampling process generates a flat-top Pulse Amplitude
Modulated (PAM) signal.

Analog and Sampled Signal

Sampling frequency, Fs is the number of average samples per second also known as the Sampling rate. According
to the Nyquist Theorem, the sampling rate should be at least 2 times the upper cutoff frequency. Sampling
frequency, Fs>=2*fmax to avoid Aliasing Effect. If the sampling frequency is very higher than the Nyquist rate it
becomes Oversampling, theoretically a bandwidth-limited signal can be reconstructed if sampled above the
Nyquist rate. If the sampling frequency is less than the Nyquist rate it will become Undersampling.
Basically, 3 main types of techniques are used for the sampling process. Those are
1. Natural Sampling and
2. Flat- top Sampling.
3. Impulse sampling

Sampling techniques:
There are three types of sampling techniques:
 Impulse sampling.
  Natural sampling.
 Flat Top sampling.
Impulse Sampling

This is called ideal sampling or impulse sampling. You cannot use this practically because pulse width cannot be zero
and the generation of impulse train is not possible practically.
Natural Sampling
Flat Top Sampling
During transmission, noise is introduced at top of the transmission pulse which can be easily
removed if the pulse is in the form of flat top. Here, the top of the samples are flat i.e. they have
constant amplitude. Hence, it is called as flat top sampling or practical sampling. Flat top
sampling makes use of sample and hold circuit.
Nyquist Rate
It is the minimum sampling rate at which signal can be converted into samples and can be
recovered back without distortion.
Nyquist rate fN = 2fm hz

Samplings of Band Pass Signals

In case of band pass signals, the spectrum of band pass signal X[ω] = 0 for the frequencies
outside the range f1 ≤ f ≤ f2. The frequency f1 is always greater than zero. Plus, there is no aliasing
effect when fs > 2f2. But it has two disadvantages:
 The sampling rate is large in proportion with f 2. This has practical limitations.
 The sampled signal spectrum has spectral gaps.
To overcome this, the band pass theorem states that the input signal x(t) can be converted into its
samples and can be recovered back without distortion when sampling frequency f s < 2f2.
Also,

Quantization
In quantization, an analog sample with an amplitude that converted into a digital sample with an amplitude that
takes one of a specifically defined set of quantization values. Quantization is done by dividing the range of possible
values of the analog samples into some different levels and assigning the center value of each level to any sample in
the quantization interval. Quantization approximates the analog sample values with the nearest quantization values.
So almost all the quantized samples will differ from the original samples by a small amount. That amount is called
quantization error. The result of this quantization error is we will hear a hissing noise when playing a random signal.
Converting analog samples into binary numbers that are 0 and 1.
In most cases, we will use uniform quantizers. Uniform quantization is applicable when the sample values are in a
finite range (Fmin, Fmax). The total data range is divided into 2n levels, let it be L intervals. They will have an equal
length Q. Q is known as Quantization interval or quantization step size. In uniform quantization, there will be no
quantization error.
 Uniformly Quantized Signal

As we know,
L=2n, then Step size Q = (Fmax – Fmin) / L

Interval i is mapped to the middle value. We will store or send only the index value of quantized value.
An Index value of quantized value Qi (F) = [F – Fmin / Q]
Quantized value Q (F) = Qi (F) Q + Q / 2 + Fmin
But there are some problems raised in uniform quantization those are
 Only optimal for the uniformly distributed signal.
 Real audio signals are more concentrated near zeros.
 The Human ear is more sensitive to quantization errors at small values.
The solution to this problem is using Non- uniform quantization. In this process, the quantization interval is smaller
near zero.

Low Pass Filter (LPF)

This LPF is used to remove the high frequency (HF) components that are present within the input analog signal. Here
this signal is higher as compared to the highest frequency message signal so that it avoids aliasing of the message
signal.

Regenerative Repeater
The signal strength can be enhanced through this regenerative repeater. So, the channel’s output also includes a
regenerative repeater circuit to balance the signal loss, renovate the signal & also increases the signal strength.

Decoder
The main function of a decoder circuit is to decode the pulse-coded signal to repeat the actual signal. This circuit
works like a demodulator.

Reconstruction Filter
After the conversion of DAC (digital-to-analog conversion) is done with the help of the decoder and regenerative
circuit, then an LPF (low-pass filter) is used to get back the original signal. So, this is known as the reconstruction
filter
Therefore, the Pulse Code Modulator circuit (PCM) is used to digitize the specified analog signal, code it, sample it &
after that, it transmits in the form of analog. So, this entire procedure can be repeated within a reverse model to get
the actual signal.

Coding
The encoder encodes the quantized samples. Each quantized sample is encoded into an 8-bit codeword by using
A-law in the encoding process.
 Bit 1 is the most significant bit (MSB), it represents the polarity of the sample. “1” represents positive
polarity and “0” represents negative polarity.
 Bit 2,3 and 4 will defines the location of the sample value. These three bits together form a linear curve for
low-level negative or positive samples.
 Bit 5,6,7 and 8 are the least significant bits (LSB) it represents one of the segments’ quantized value. Each
segment is divided into 16 quantum levels.
Pulse code modulation is similar to PWM, PAM otherwise PPM however there is a significant disparity among them
that is they are analog pulse modulation systems but Pulse code modulation is a digital pulse modulation system.
So, the output of PCM is in the form of coded digital and it is in the form of digital signals of stable width, position &
amplitude.
The data can be transmitted in code words format. A pulse code modulation system includes a transmitter like a
PCM encoder & a receiver like a PCM decoder.
The important operations within the transmitter of pulse code modulation mainly include sampling, quantizing, and
encoding. Generally, these operations are performed within a similar circuit namely ADC.

Types of PCM
PCM is two types of Differential Pulse Code Modulation (DPCM), Adaptive Differential Pulse Code Modulation
(ADPCM) & Linear Pulse Code Modulation.

Differential Pulse Code Modulation (DPCM)

In DPCM only the difference between a sample and the previous value is encoded. The difference will be much
smaller than the total sample value so we need some bits for getting the same accuracy as in ordinary PCM. So that
the required bit rate will also reduce. For example, in 5-bit code 1 bit is for polarity, and the remaining 4 bits for 16
quantum levels.

Differential Pulse Code Modulation Advantages and Disadvantages

The advantages of differential pulse code modulation include the following.
 The requirement of bandwidth is low as compared to pulse code modulation.
 Due to the prediction filter, the quantization error can be decreased
 As compared to PCM, the number of bits that are used to represent one sample value can also be reduced
 Low signaling rate
The disadvantages of differential pulse code modulation include the following.
 Bit rate is high
 Practical usage is restricted
 A Predicator circuit needs to be used which is extremely complex.

Adaptive Differential Pulse Code Modulation (ADPCM)

ADPCM is achieved by adapting the quantizing levels to analog signal characteristics. We can estimate the values
with the preceding sample values. Error estimation is done as same as in DPCM. In the 32Kbps ADPCM method
difference between the predicted value and sample, the value is coded with 4 bits, so that we’ll get 15 quantum
levels. In this method data rate is half of the conventional PCM.

Linear Pulse Code Modulation

The term LPCM stands for “Linear pulse code modulation”. This is one kind of modulation technique, used to encode
uncompressed audio data digitally, wherever audio signals are signified through a series of amplitude values from a
model on a linear scale where these values are comparative to the amplitudes. So, the amplitude values are
quantized linearly, therefore similar to a very large set of feasible values through a quite small set of values that may
be discrete symbols or integers.
This kind of modulation is also used for audio formats like a collective reference that occurs when using the result of
this encoding technique. PCM or Pulse code modulation is a general method of encoding and the main function of
this is to describe LPCM frequently and it is capable of extremely high throughput.

Pulse Code Demodulation

Pulse Code Demodulation will be doing the same modulation process in reverse. Demodulation starts with the
decoding process, during transmission the PCM signal will be affected by noise interference. So, before the PCM
signal sends to the PCM demodulator, we have to recover the signal to the original level for that we are using a
comparator. The PCM signal is a series pulse wave signal, but for demodulation, we need a wave to be parallel.
By using a serial to parallel converter the series pulse wave signal will be converted into a parallel digital signal. After
that the signal will pass through the n-bits decoder, it should be a Digital to Analog converter. Decoder recovers the
original quantization values of the digital signal. This quantization value also includes a lot of high-frequency
harmonics with original audio signals. For avoiding unnecessary signals we utilize a low-pass filter at the final part.

Advantages
 Analog signals can be transmitted over a high-speed digital communication system.
 The probability of occurring error will reduce by the use of appropriate coding methods.
 PCM is used in Telkom system, digital audio recording, digitized video special effects, digital video, voice
mail.
 PCM is also used in Radio control units as transmitters and also a receiver for remote-controlled cars, boats,
planes.
 The PCM signal is more resistant to interference than normal signals.

Pulse Code Modulation Applications

 PCM technique is mainly used to change the signal from analog to digital signal so that an analog signal
which is changed can be broadcasted throughout the digital communication network. This modulation is
available in binary form, so the available possible states will be two types like high & low.
 Pulse-code modulation (PCM) is a technique used to represent sampled analog signals digitally. It is the
normal form of digital audio within computers, digital telephony, compact discs & other digital audio
applications.
 These modulations can be used for temperature regulation, cold or heat storage through high storage
density & thermal comfort within buildings that need a narrow range of temperature. Thus, if the solar
energy is stored efficiently, then it can be used for night cold.
The pulse code modulation refers to the utilization of a precise set of rules for changing a signal into a stream of
digits.

Limitations of PCM
The sampling theorem like Nyquist–Shannon illustrates the operating of pulse code modulation devices can be done
without establishing distortions in their frequency bands if these bands offer a sampling frequency as a minimum
twice that of the maximum frequency included within the i/p signal.
For instance, the voiceband frequency which is used mainly ranges from 300 Hz -3400 Hz. For the efficient
renovation of the voice signal, the applications of telephony normally utilize a sampling frequency of 8000 Hz which
is twice the maximum working voice frequency.
Apart from in any PCM system, there are impairment implicit possible sources like the following
Selecting a separate value that is close but not precisely at the analog signal range for every sample guides to
quantization error.
In between the samples, no signal measurement can be made; so, the sampling theorem assurances non-ambiguous
depiction & signal recovery simply if it has no energy at ‘fs/2’ frequency, high frequencies will not be properly
signified otherwise recovered & include aliasing distortion toward the signal under the Nyquist frequency.
Because samples are reliant on time, so a precise clock is necessary for precise reproduction. If any of the encodings
otherwise decoding CLK is not steady, these defects will directly influence the output of the device quantity.
Sampling
Sampling is defined as, “The process of measuring the instantaneous values of continuous-time signal in a discrete
form.”
Sample is a piece of data taken from the whole data which is continuous in the time domain.
When a source generates an analog signal and if that has to be digitized, having 1s and 0s i.e., High or Low, the signal
has to be discretized in time. This discretization of analog signal is called as Sampling.

Sampling Rate
To discretize the signals, the gap between the samples should be fixed. That gap can be termed as a sampling period
Ts.

Sampling frequency is the reciprocal of the sampling period. This sampling frequency, can be simply called
as Sampling rate. The sampling rate denotes the number of samples taken per second, or for a finite set of values.
For an analog signal to be reconstructed from the digitized signal, the sampling rate should be highly considered. The
rate of sampling should be such that the data in the message signal should neither be lost nor it should get over-
lapped. Hence, a rate was fixed for this, called as Nyquist rate.
Nyquist Rate
Suppose that a signal is band-limited with no frequency components higher than W Hertz. That means, W is the
highest frequency. For such a signal, for effective reproduction of the original signal, the sampling rate should be
twice the highest frequency.
Which means,

According to the Nyquist Theorem, the sampling rate should be at least 2 times the upper cutoff frequency.
Sampling frequency, Fs>=2*fmax to avoid Aliasing Effect. If the sampling frequency is very higher than the Nyquist
rate it becomes Oversampling, theoretically a bandwidth-limited signal can be reconstructed if sampled above the
Nyquist rate. If the sampling frequency is less than the Nyquist rate it will become Undersampling.

Sampling Theorem
The sampling theorem, which is also called as Nyquist theorem, delivers the theory of sufficient sample rate in terms
of bandwidth for the class of functions that are bandlimited.
The sampling theorem states that, “a signal can be exactly reproduced if it is sampled at the rate fs which is greater
than twice the maximum frequency W.”
To understand this sampling theorem, let us consider a band-limited signal, i.e., a signal whose value is non-
zero between some –W and W Hertz.
We need a sampling frequency, a frequency at which there should be no loss of information, even after sampling.
For this, we have the Nyquist rate that the sampling frequency should be two times the maximum frequency. It is
the critical rate of sampling.
If the signal x(t) is sampled above the Nyquist rate, the original signal can be recovered, and if it is sampled below the
Nyquist rate, the signal cannot be recovered.
The following figure explains a signal, if sampled at a higher rate than 2w in the frequency domain.

The above figure shows the Fourier transform of a signal xs(t). Here, the information is reproduced without any
loss. There is no mixing up and hence recovery is possible.
The Fourier Transform of the signal xs(t) is

The result will be as shown in the above figure. The information is replaced without any loss. Hence, this is also a
good sampling rate.
Now, let us look at the condition,
fs<2W
We can observe from the above pattern that the over-lapping of information is done, which leads to mixing up and
loss of information. This unwanted phenomenon of over-lapping is called as Aliasing.

Aliasing
Aliasing can be referred to as “the phenomenon of a high-frequency component in the spectrum of a signal, taking
on the identity of a low-frequency component in the spectrum of its sampled version.”
The corrective measures taken to reduce the effect of Aliasing are −
 In the transmitter section of PCM, a low pass anti-aliasing filter is employed, before the sampler, to
eliminate the high frequency components, which are unwanted.
 The signal which is sampled after filtering, is sampled at a rate slightly higher than the Nyquist rate.
This choice of having the sampling rate higher than Nyquist rate, also helps in the easier design of the reconstruction
filter at the receiver.

Scope of Fourier Transform

It is generally observed that, we seek the help of Fourier series and Fourier transforms in analyzing the signals and
also in proving theorems. It is because −
 The Fourier Transform is the extension of Fourier series for non-periodic signals.
 Fourier transform is a powerful mathematical tool which helps to view the signals in different domains and
helps to analyze the signals easily.
 Any signal can be decomposed in terms of sum of sines and cosines using this Fourier transform.

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

Quantizing an Analog Signal

The analog-to-digital converters perform this type of function to create a series of digital values out of the given
analog signal. The following figure represents an analog signal. This signal to get converted into digital, has to
undergo sampling and quantizing.
The quantizing of an analog signal is done by discretizing the signal with a number of quantization
levels. Quantization is representing the sampled values of the amplitude by a finite set of levels, which means
converting a continuous-amplitude sample into a discrete-time signal.
The following figure shows how an analog signal gets quantized. The blue line represents analog signal while the red
one represents the quantized signal.

Both sampling and quantization result in the loss of information. The quality of a Quantizer output depends upon the
number of quantization levels used. The discrete amplitudes of the quantized output are called as representation
levels or reconstruction levels. The spacing between the two adjacent representation levels is called
a quantum or step-size.

The following figure shows the resultant quantized signal which is the digital form for the given analog signal.

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

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

There are two types of uniform quantization. They are Mid-Rise type and Mid-Tread type. The following figures
represent the two types of uniform quantization.
Figure 1 shows the mid-rise type and figure 2 shows the mid-tread type of uniform quantization.
 The Mid-Rise type is so called because the origin lies in the middle of a raising part of the stair-case like
graph. The quantization levels in this type are even in number.
 The Mid-tread type is so called because the origin lies in the middle of a tread of the stair-case like graph.
The quantization levels in this type are odd in number.
 Both the mid-rise and mid-tread type of uniform quantizers are symmetric about the origin.

Quantization Error
For any system, during its functioning, there is always a difference in the values of its input and output. The
processing of the system results in an error, which is the difference of those values.
The difference between an input value and its quantized value is called a Quantization Error. A Quantizer is a
logarithmic function that performs Quantization rounding off the value. An analog-to-digital converter (ADC) works
as a quantizer.
The following figure illustrates an example for a quantization error, indicating the difference between the original
signal and the quantized signal.
Quantization Noise
It is a type of quantization error, which usually occurs in analog audio signal, while quantizing it to digital. For
example, in music, the signals keep changing continuously, where a regularity is not found in errors. Such errors
create a wideband noise called as Quantization Noise.

Companding in PCM
The word Companding is a combination of Compressing and Expanding, which means that it does both. This is a non-
linear technique used in PCM which compresses the data at the transmitter and expands the same data at the
receiver. The effects of noise and crosstalk are reduced by using this technique.
Companding is a portmanteau for a “compressing-expanding” mechanism. It compresses the signals before
transmitting on a band-limited channel and expands them when received. During compression, an analog signal is
quantized to create a digital signal using unequal steps in order to amplify the quiet sounds while attenuating the
loud ones. Conversely, at the receiver, the digital signal is converted back to an analog signal after expansion, in
which the low amplitude signals are amplified less when compared to higher ones.
This act would alter the dynamic range of the signal, facilitating the transmission of a signal with a large dynamic
range{{ Dynamic range (DR, DNR, or DYR) is the ratio between the largest and smallest values that a certain quantity can assume. It is often
used in the context of signals, like sound and light. It is measured either as a ratio or as a base-10 (decibel) or base-
2 (doublings, bits or stops) logarithmic value of the difference between the smallest and largest signal values }}. over a channel with smaller
dynamic range

There are two types of Companding techniques. They are −

A-law Companding Technique

 Uniform quantization is achieved at A = 1, where the characteristic curve is linear and no compression is
done.
 A-law has mid-rise at the origin. Hence, it contains a non-zero value.
 A-law companding is used for PCM telephone systems.
µ-law Companding Technique
 Uniform quantization is achieved at µ = 0, where the characteristic curve is linear and no compression is
done.
 µ-law has mid-tread at the origin. Hence, it contains a zero value.
 µ-law companding is used for speech and music signals.

https://www.allaboutcircuits.com/technical-articles/companding-logarithmic-laws-implementation-and-
consequences/
Differential PCM
For the samples that are highly correlated, when encoded by PCM technique, leave redundant information behind.
To process this redundant information and to have a better output, it is a wise decision to take a predicted sampled
value, assumed from its previous output and summarize them with the quantized values. Such a process is called
as Differential PCM (DPCM) technique.

DPCM Transmitter
The DPCM Transmitter consists of Quantizer and Predictor with two summer circuits. Following is the block
diagram of DPCM transmitter.
https://www.allaboutcircuits.com/textbook/semiconductors/chpt-8/averager-summer-circuits/

The signals at each point are named as −

The predictor produces the assumed samples from the previous outputs of the transmitter circuit. The input to this
predictor is the quantized versions of the input signal x(nTs)
Quantizer Output is represented as −
v(nTs)=Q[e(nTs)]v(nTs)=Q[e(nTs)]
=e(nTs)+q(nTs)=e(nTs)+q(nTs)
Where q (nTs) is the quantization error
Predictor input is the sum of quantizer output and predictor output,
DPCM Receiver
The block diagram of DPCM Receiver consists of a decoder, a predictor, and a summer circuit. Following is the
diagram of DPCM Receiver.

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

Differential Pulse Code Modulation Working and Application

Differential pulse code modulation is a technique of analog to digital signal conversion. This technique samples the
analog signal and then quantizes the difference between the sampled value and its predicted value, then encodes
the signal to form a digital value. Before going to discuss differential pulse code modulation, we have to know the
demerits of PCM (Pulse Code Modulation). The samples of a signal are highly correlated with each other. The signal’s
value from the present sample to the next sample does not differ by a large amount. The adjacent samples of the
signal carry the same information with a small difference. When these samples are encoded by the standard PCM
system, the resulting encoded signal contains some redundant information bits. The below figure illustrates this.
 Redundant Information Bits in PCM
The above figure shows a continuing time signal x(t) denoted by a dotted line. This signal is sampled by flat-top
sampling at intervals Ts, 2Ts, 3Ts…nTs. The sampling frequency is selected to be higher than the Nyquist rate. These
samples are encoded by using 3-bit (7 levels) PCM. The samples are quantized to the nearest digital level as shown
by small circles in the above figure. The encoded binary value of each sample is written on the top of the samples.
Just observe the above figure at samples taken at 4Ts, 5Ts, and 6Ts are encoded to the same value of (110). This
information can be carried only by one sample value. But three samples are carrying the same information means
redundant.
Now let consider the samples at 9Ts and 10Ts, the difference between these samples only due to the last bit and first
two bits are redundant since they do not change. So in order to make the process this redundant information and to
have a better output. It is an intelligent decision to take a predicted sampled value, assumed from its previous
output and summarise them with the quantized values. Such a process is called a Differential PCM (DPCM)
technique.

Principle of Differential Pulse Code Modulation

If the redundancy is reduced, then the overall bitrate will decrease and the number of bits required to transmit one
sample will also reduce. This type of digital pulse modulation technique is called differential pulse code modulation.
The DPCM works on the principle of prediction. The value of the present sample is predicted from the previous
samples. The prediction may not be exact, but it is very close to the actual sample value.

Differential Pulse Code Modulation Transmitter

The below figure shows the DPCM transmitter. The transmitter consists of a comparator, quantizer, prediction
filter, and an encoder.

Differential Pulse Code Modulator

The sampled signal is denoted by x(nTs) and the predicted signal is indicated by x^(nTs). The comparator finds out
the difference between the actual sample value x(nTs) and the predicted value x^(nTs). This is called signal error and
it is denoted as e(nTs)
e(nTs)= x(nTs)- x^( nTs) …….(1)
Here the predicted value x^(nTs) is produced by using a prediction filter(signal processing filter). The quantizer
output signal eq(nTs) and the previous prediction is added and given as input to the prediction filter, this signal is
denoted by xq(nTs). This makes the prediction closer to the actually sampled signal. The quantized error signal
eq(nTs) is very small and can be encoded by using a small number of bits. Thus the number of bits per sample is
reduced in DPCM.
The quantizer output would be written as,
eq(nTs)= e(nTs)+ q(nTs) ……(2)
Here q(nTs) is quantization error. From the above block diagram the prediction filter input xq(nTs) is obtained by sum
of x^(nTs) and the quantizer output eq(nTs).
i.e, xq(nTs) = x^(nTs)+ eq(nTs).………. (3)
by substituting the value of eq(nTs) from the equation (2) in equation (3) we get,
xq(nTs) = x^(nTs)+ e(nTs)+ q(nTs)……. (4)
Equation (1) can written as,
e(nTs)+ x^( nTs) = x(nTs)……. (5)
from the above equations 4 and 5 we get,
xq(nTs) = x(nTs)+ x(nTs)
Therefore, the quantized version of signal xq(nTs) is the sum of original sample value and quantized error q(nTs). The
quantized error can be positive or negative. So the output of the prediction filter does not depend on its
characteristics.

Differential Pulse Code Modulation Receiver

In order to reconstruct the received digital signal, the DPCM receiver (shown in the below figure) consists of a
decoder and prediction filter. In the absenteeism of noise, the encoded receiver input will be the same as the
encoded transmitter output.

As we discussed above, the predictor undertakes a value, based on the previous outputs. The input given to the
decoder is processed and that output is summed up with the output of the predictor, to obtain better output. That
means here first of all the decoder will reconstruct the quantized form of the original signal. Therefore the signal at
the receiver differs from the actual signal by quantization error q(nTs), which is introduced permanently in the
reconstructed signal.
Differential Pulse Code Modulation
S. NO Parameters Pulse Code Modulation (PCM) (DPCM)
1 Number of bits It uses 4, 8, or 16 bits per sample < PCM bits
2 Levels, step size Fixed step size. Cannot varied A fixed number of levels are used.
3 Bit redundancy Present Can permanently remove
Slope overload distortion and
Depends on the number of levels quantization noise are present but
4 Quantization error and distortion used very less as compared to PCM
Higher bandwidth has been
The bandwidth of the transmission required since the number of bits
5 channel is absent Lower than PCM bandwidth
6 Feedback No feedback in Tx and Rx Feedback exists
7 Complexity of notation Complex Simple
8 Signal to noise ratio (SNR) Good Fair
Applications of DPCM
The DPCM technique mainly used Speech, image and audio signal compression. The DPCM conducted on signals with
the correlation between successive samples leads to good compression ratios. In images, there is a correlation
between the neighboring pixels, in video signals, the correlation is between the same pixels in consecutive frames
and inside frames (which is the same as correlation inside the image).
This method is suitable for real-Time applications. To understand the efficiency of this method of medical
compression and real-time application of medical imaging such as telemedicine and online diagnosis. Therefore, it
can be efficient for lossless compression and implementation for lossless or near-lossless medical image
compression.

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

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

Features of Delta Modulation

Following are some of the features of delta modulation.
 An over-sampled input is taken to make full use of the signal correlation.
 The quantization design is simple.
 The input sequence is much higher than the Nyquist rate.
 The quality is moderate.
 The design of the modulator and the demodulator is simple.
 The stair-case approximation of output waveform.
 The step-size is very small, i.e., Δ deltadelta.
 The bit rate can be decided by the user.
 This involves simpler implementation.
Delta Modulation is a simplified form of DPCM technique, also viewed as 1-bit DPCM scheme. As the sampling
interval is reduced, the signal correlation will be higher.

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

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

From the above diagram, we have the notations as −
 x(nTs) = over sampled input
 ep(nTs) = summer output and quantizer input
 eq(nTs)= quantizer output = v(nTs)
 xˆ(nTs)= output of delay circuit
 u(nTs) = input of delay circuit
Using these notations, now we shall try to figure out the process of delta modulation.

Which means,
The present input of the delay unit
= The previous output of the delay unit  + the present quantizer output
Assuming zero condition of Accumulation,
Delay unit output is an Accumulator output lagging by one sample.
From equations 5 & 6, we get a possible structure for the demodulator.
A Stair-case approximated waveform will be the output of the delta modulator with the step-size as delta (Δ). The
output quality of the waveform is moderate.

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

From the above diagram, we have the notations as −

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

Advantages of DM Over DPCM

 1-bit quantizer
 Very easy design of the modulator and the demodulator
However, there exists some noise in DM.
 Slope Over load distortion (when Δ is small)
  Granular noise (when Δ is large)
Granular noise is the manifestation of random signals when the variation of the input signal is
smaller than the step size.

Adaptive Delta Modulation (ADM)

In digital modulation, we have come across certain problem of determining the step-size, which influences the
quality of the output wave.
A 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. The minute details get missed in the process. So, 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.
Following is the block diagram of Adaptive delta modulator.

The gain of the voltage controlled amplifier is adjusted by the output signal from the sampler. The amplifier gain
determines the step-size and both are proportional.
ADM quantizes the difference between the value of the current sample and the predicted value of the next sample.
It uses a variable step height to predict the next values, for the faithful reproduction of the fast varying values.

The delta modulation has two major drawbacks as under :

 Slope overload distortion
 Granular or idle noise

Slope Overload Distortion

This distortion arises because of large dynamic range of the input signal.

Fig.1: Quantization Errors in Delta Modulation

We can observe from fig.1 , the rate of rise of input signal x(t) is so high that the staircase signal can not approximate
it, the step size ‘Δ’ becomes too small for staircase signal u(t) to follow the step segment of x(t).
Hence, there is a large error between the staircase approximated signal and the original input signal x(t).
This error or noise is known as slope overload distortion .
To reduce this error, the step size must be increased when slope of signal x(t) is high.

Granular or Idle Noise

Granular or Idle noise occurs when the step size is too large compared to small variation in the input signal.
This means that for very small variations in the input signal, the staircase signal is changed by large amount (Δ)
because of large step size.
Fig.1 shows that when the input signal is almost flat , the staircase signal u(t) keeps on oscillating by ±Δ around the
signal.
The error between the input and approximated signal is called  granular noise.
The solution to this problem is to make the step size small .

Solution
In order to overcome the quantization errors due to slope overload and granular noise, the step size (Δ) is made
adaptive to variations in the input signal x(t).
Particularly in the steep segment of the signal x(t), the step size is increased. And the step is decreased when the
input is varying slowly.
This method is known as Adaptive Delta Modulation (ADM).
The adaptive delta modulators can take continuous changes in step size or discrete changes in step size.

Digital Communication - Techniques

There are a few techniques which have paved the basic path to digital communication processes. For the signals to
get digitized, we have the sampling and quantizing techniques.
For them to be represented mathematically, we have LPC and digital multiplexing techniques. These digital
modulation techniques are further discussed.

Linear Predictive Coding

Linear Predictive Coding LPC is a tool which represents digital speech signals in linear predictive model. This is
mostly used in audio signal processing, speech synthesis, speech recognition, etc.
Linear prediction is based on the idea that the current sample is based on the linear combination of past samples.
The analysis estimates the values of a discrete-time signal as a linear function of the previous samples.
The spectral envelope is represented in a compressed form, using the information of the linear predictive model.
This can be mathematically represented as −

s(n)=∑k=1pαks(n−k)s(n)=∑k=1pαks(n−k) for some value of p and α k

Where
 sn is the current speech sample
 k is a particular sample
 p is the most recent value
 αk is the predictor co-efficient
 sn−kn−k is the previous speech sample
For LPC, the predictor co-efficient values are determined by minimizing the sum of squared differences over a finite
interval between the actual speech samples and the linearly predicted ones.
This is a very useful method for encoding speech at a low bit rate. The LPC method is very close to the Fast Fourier
Transform FFT method.

Multiplexing
Multiplexing is the process of combining multiple signals into one signal, over a shared medium. These signals, if
analog in nature, the process is called as analog multiplexing. If digital signals are multiplexed, it is called as digital
multiplexing.
Multiplexing was first developed in telephony. A number of signals were combined to send through a single cable.
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
called as a MUX. The reverse process, i.e., extracting the number of channels from one, which is done at the receiver
is called as de-multiplexing. The device which does de-multiplexing is called as DEMUX.
The following figures represent MUX and DEMUX. Their primary use is in the field of communications.

Types of Multiplexers
There are mainly two types of multiplexers, namely analog and digital. They are further divided into FDM, WDM, and
TDM. The following figure gives a detailed idea on this classification.

Actually, there are many types of multiplexing techniques. Of them all, we have the main types with general
classification, mentioned in the above figure.

Analog Multiplexing
The analog multiplexing techniques involve signals which are analog in nature. The analog signals are multiplexed
according to their frequency FDMFDM or wavelength WDMWDM.

Frequency Division Multiplexing  FDM

In analog multiplexing, the most used technique is Frequency Division Multiplexing FDMFDM. This
technique A traditional television transmitter, which sends a number of channels through a single cable uses
various frequencies to combine streams of data, for sending them on a communication medium, as a single
signal.
Example −, 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. A prism which
can turn different wavelengths into a single line, can be used at the output of MUX and input of DEMUX.
Example − Optical fiber communications use WDM technique to merge different wavelengths into a single light for
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, by 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 the 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 a common clock is not
required. If the allotted device, for a time-slot, transmits nothing and sits idle, then that slot is allotted to
another device, unlike synchronous. This type of TDM is used in Asynchronous transfer mode networks.

Regenerative Repeater
For any communication system to be reliable, it should transmit and receive the signals effectively, without any loss.
A PCM wave, after transmitting through a channel, gets distorted due to the noise introduced by the channel.
The regenerative pulse compared with the original and received pulse, will be as shown in the following figure.

For a better reproduction of the signal, a circuit called as regenerative repeater is employed in the path before the
receiver. This helps in restoring the signals from the losses occurred. Following is the diagrammatical representation.
This consists of an equalizer along with an amplifier, a timing circuit, and a decision making
device. Their working of each of the components is detailed as follows.
Equalizer
The channel produces amplitude and phase distortions to the signals. This is due to the transmission characteristics
of the channel. The Equalizer circuit compensates these losses by shaping the received pulses.

Timing Circuit
To obtain a quality output, the sampling of the pulses should be done where the signal to noise ratio SNR is
maximum. To achieve this perfect sampling, a periodic pulse train has to be derived from the received pulses, which
is done by the timing circuit.
Hence, the timing circuit, allots the timing interval for sampling at high SNR, through the received pulses.

Decision Device
The timing circuit determines the sampling times. The decision device is enabled at these sampling times. The
decision device decides its output based on whether the amplitude of the quantized pulse and the noise, exceeds a
pre-determined value or not.

Line Codes
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 Unipolar signaling −
 Non Return to Zero NRZNRZ
 Return to Zero RZRZ

Unipolar Non-Return to Zero NRZ

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
 It is simple.
 A lesser bandwidth is required.

Disadvantages
 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 RZ

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
 It is simple.
 The spectral line present at the symbol rate can be used as a clock.
Disadvantages
 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
 It is simple.
 No low-frequency components are present.

Disadvantages
 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
 It is simple.
 No low-frequency components are present.

Disadvantages
 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 AMI. 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
 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
 No clock is present.
 Long strings of data causes loss of synchronization.
Power Spectral Density
The function which describes how the power of a signal got distributed at various frequencies, in the frequency
domain is called as Power Spectral Density PSD.
PSD is the Fourier Transform of Auto-Correlation Similarity between observations. It is in the form of a rectangular
pulse.

PSD Derivation
According to the Einstein-Wiener-Khintchine theorem, if the auto correlation function or power spectral density of a
random process is known, the other can be found exactly.
Hence, to derive the power spectral density, we shall use the time auto-correlation (Rx(τ)) of a power
signal x(t) as shown below.

Hence, we get the equation for Power Spectral Density. Using this, we can find the PSD of various line codes.
Encoding is the process of converting the data or a given sequence of characters, symbols,
alphabets etc., into a specified format, for the secured transmission of data. Decoding is the
reverse process of encoding which is to extract the information from the converted format.

Data Encoding
Encoding is the process of using various patterns of voltage or current levels to represent 1s and 0s of the digital
signals on the transmission link.
The common types of line encoding are Unipolar, Polar, Bipolar, and Manchester.

Encoding Techniques
The data encoding technique is divided into the following types, depending upon the type of data conversion.
 Analog data to Analog signals − The modulation techniques such as Amplitude Modulation, Frequency
Modulation and Phase Modulation of analog signals, fall under this category.
 Analog data to Digital signals − This process can be termed as digitization, which is done by Pulse Code
Modulation PCM. Hence, it is nothing but digital modulation. As we have already discussed, sampling and
quantization are the important factors in this. Delta Modulation gives a better output than PCM.
 Digital data to Analog signals − The modulation techniques such as Amplitude Shift Keying ASK, Frequency
Shift Keying FSK, Phase Shift Keying PSK, etc., fall under this category.
 Digital data to Digital signals − These are in this section. There are several ways to map digital data to digital
signals. Some of them are −

Non Return to Zero NRZ

NRZ Codes has 1 for High voltage level and 0 for Low voltage level. The main behavior of NRZ codes is that the
voltage level remains constant during bit interval. The end or start of a bit will not be indicated and it will maintain
the same voltage state, if the value of the previous bit and the value of the present bit are same.
The following figure explains the concept of NRZ coding.

If the above example is considered, as there is a long sequence of constant voltage level and the clock
synchronization may be lost due to the absence of bit interval, it becomes difficult for the receiver to differentiate
between 0 and 1.

There are two variations in NRZ namely −

NRZ - L NRZ–LEVEL
There is a change in the polarity of the signal, only when the incoming signal changes from 1 to 0 or from 0 to 1. It is
the same as NRZ, however, the first bit of the input signal should have a change of polarity.

NRZ - I NRZ–INVERTED
If a 1 occurs at the incoming signal, then there occurs a transition at the beginning of the bit interval. For a 0 at the
incoming signal, there is no transition at the beginning of the bit interval.
NRZ codes has a disadvantage that the synchronization of the transmitter clock with the receiver clock gets
completely disturbed, when there is a string of 1s and 0s. Hence, a separate clock line needs to be provided.

Bi-phase Encoding
The signal level is checked twice for every bit time, both initially and in the middle. Hence, the clock rate is double
the data transfer rate and thus the modulation rate is also doubled. The clock is taken from the signal itself. The
bandwidth required for this coding is greater.
There are two types of Bi-phase Encoding.

 Bi-phase Manchester
 Differential Manchester

Bi-phase Manchester
In this type of coding, the transition is done at the middle of the bit-interval. The transition for the resultant pulse is
from High to Low in the middle of the interval, for the input bit 1. While the transition is from Low to High for the
input bit 0.

Differential Manchester
In this type of coding, there always occurs a transition in the middle of the bit interval. If there occurs a transition at
the beginning of the bit interval, then the input bit is 0. If no transition occurs at the beginning of the bit interval,
then the input bit is 1.
The following figure illustrates the waveforms of NRZ-L, NRZ-I, Bi-phase Manchester and Differential Manchester
coding for different digital inputs.

Block Coding
Among the types of block coding, the famous ones are 4B/5B encoding and 8B/6T encoding. The number of bits are
processed in different manners, in both of these processes.

4B/5B Encoding
In Manchester encoding, to send the data, the clocks with double speed is required rather than NRZ coding. Here, as
the name implies, 4 bits of code is mapped with 5 bits, with a minimum number of 1 bits in the group.
The clock synchronization problem in NRZ-I encoding is avoided by assigning an equivalent word of 5 bits in the place
of each block of 4 consecutive bits. These 5-bit words are predetermined in a dictionary.
The basic idea of selecting a 5-bit code is that, it should have one leading 0 and it should have no more than two
trailing 0s. Hence, these words are chosen such that two transactions take place per block of bits.

4B/5B (four binary/five binary ) –

This coding scheme is used in combination with NRZ-I. The problem with NRZ-I was that it has a
 synchronization problem for long sequences of zeros. So, to overcome it we substitute the bit stream from 4-
bit to 5-bit data group before encoding it with NRZ-I. So that it does not have a long stream of zeros. The
block-coded stream does not have more than three consecutive zeros (see encoding table).

At the receiver, the NRZ-I encoded digital signal is first decoded into a stream of bits and then decoded again
to remove the redundancy bits.
Drawback – Though 4B/5B encoding solves the problem of synchronization,it increases the signal rate of NRZ-L.
Moreover,it does not solve the DC component problem of NRZ-L.

8B/6T Encoding
We have used two voltage levels to send a single bit over a single signal. But if we use more than 3 voltage levels, we
can send more bits per signal.
For example, if 6 voltage levels are used to represent 8 bits on a single signal, then such encoding is termed as 8B/6T
encoding. Hence in this method, we have as many as 729 3636 combinations for signal and 256 2828 combinations
for bits.
These are the techniques mostly used for converting digital data into digital signals by compressing or coding them
for reliable transmission of data.
8B/10B (eight binary/ten binary) –
This encoding is similar to 4B/5B encoding except that a group of 8 bits of data is now substituted by a 10-bit
code and it provides greater error detection capability than 4B/5B.
It is actually a combination of 5B/6B and 3B/4B encoding.The most five significant bits of a 10-bit block is fed
into the 5B/6B encoder; the least 3 significant bits is fed into a 3B/4B encoder. The split is done to simplify
the mapping table.

A group of 8 bits can have 2^8 different combinations while a group of 10 bits can have 2^10 different
combinations. This means that there are 2^10-2^8=768 redundant groups that are not used for 8B/10B
encoding and can be used for error detection and disparity check.
Thus, this technique is better than 4B/5B because of better error-checking capability and better
synchronization.

AMI (Alternate Mark Inversion)

It is a synchronous clock encoding technique that uses bipolar pulses to represent logical 1. The next logic 1 is
represented by a pulse of the opposite polarity. Hence a sequence of logical 1s is represented by a sequence of
pulses of alternating polarity. The alternating coding prevents the build-up of a d.c. voltage level down the cable.
 Example of AMI coding of NRZ data and the corresponding clock signal
AMI (Alternate Mark Inversion) encoding was used extensively in first generation pulse code modulation networks,
but suffers the drawback that a long run of 0's produces no transitions in the data stream (and therefore does not
contain sufficient transitions to guarantee lock of a DPLL).
Alternate Mark Inversion (AMI) is also known as Pseudo ternary coding

Scrambling
A computer network is designed to send information from one point to another. Data that we send can either be
digital or analog. Also signals that represent data can also be digital or analog.
Thus to send data by using signals, we must able to convert data into signals this conversion can be Analog to
Analog, Analog to Digital, Digital to Analog or Digital to Digital.
Digital to Digital conversion involves three techniques – Line Coding, Block Coding, and Scrambling. Line Coding is
always needed, whereas Block Coding and Scrambling may or may not be needed depending upon need.
Scrambling is a technique that does not increase the number of bits and does provide synchronization. Problem with
technique like Bipolar AMI(Alternate Mark Inversion) is that continuous sequence of zero’s create synchronization
problems one solution to this is Scrambling.
There are two common scrambling techniques:
1. B8ZS(Bipolar with 8-zero substitution)
2. HDB3(High-density bipolar3-zero)

B8ZS(Bipolar with 8-zero substitution) –

This technique is similar to Bipolar AMI except when eight consecutive zero-level voltages are encountered they are
replaced by the sequence,”000VB0VB”.
Note –
 V(Violation), is a non-zero voltage which means signal have same polarity as the previous non-zero voltage.
Thus it is violation of general AMI technique.
 B(Bipolar), also non-zero voltage level which is in accordance with the AMI rule (i.e.,opposite polarity from
the previous non-zero voltage).
Example: Data = 100000000

Note – Both figures (left and right one) are correct, depending upon last non-zero voltage signal of previous data
sequence (i.e., sequence before current data sequence “100000000”).
HDB3 (High-density bipolar3-zero) –
In this technique four consecutive zero-level voltages are replaced with a sequence “000V” or “B00V”.
Rules for using these sequences:
 If the number of nonzero pulses after the last substitution is odd, the substitution pattern will be “000V”, this
helps maintaining total number of nonzero pulses even.
 If the number of nonzero pulses after the last substitution is even, the substitution pattern will be “B00V”.
Hence even number of nonzero pulses is maintained again.
Example: Data = 1100001000000000

Explanation – After representing first two 1’s of data we encounter four consecutive zeros.Since our last
substitutions were two 1’s(thus number of non-zero pulses is even).So,we substitute four zeros with “B00V”.
Note – Zero non-zero pulses are also even.

Pulse Shaping ISI

After going through different types of coding techniques, we have an idea on how the data is prone to
distortion and how the measures are taken to prevent it from getting affected so as to establish a reliable
communication.
There is another important distortion which is most likely to occur, called as Inter-Symbol
Interference ISI.

Inter Symbol Interference

This is a form of distortion of a signal, in which one or more symbols interfere with subsequent signals, causing noise
or delivering a poor output.

Causes of ISI
The main causes of ISI are −
 Multi-path Propagation
 Non-linear frequency in channels
The ISI is unwanted and should be completely eliminated to get a clean output. The causes of ISI should also be
resolved in order to lessen its effect.
To view ISI in a mathematical form present in the receiver output, we can consider the receiver output.
In the above equation, the first term μai is produced by the ith transmitted bit

The second term represents the residual effect of all other transmitted bits on the decoding of the ith bit. This
residual effect is called as Inter Symbol Interference.
In the absence of ISI, the output will be −
y(ti)=μai
This equation shows that the ith bit transmitted is correctly reproduced. However, the presence of ISI introduces bit
errors and distortions in the output.
While designing the transmitter or a receiver, it is important that you minimize the effects of ISI, so as to receive the
output with the least possible error rate.

Correlative Coding
So far, we’ve discussed that ISI is an unwanted phenomenon and degrades the signal. But the same ISI if used in a
controlled manner, is possible to achieve a bit rate of 2W bits per second in a channel of bandwidth W Hertz. Such a
scheme is called as Correlative Coding or Partial response signaling schemes.
Since the amount of ISI is known, it is easy to design the receiver according to the requirement so as to avoid the
effect of ISI on the signal. The basic idea of correlative coding is achieved by considering an example of Duo-binary
Signaling.

Duo-binary Signaling
The name duo-binary means doubling the binary system’s transmission capability. To understand this, let us consider
a binary input sequence {ak} consisting of uncorrelated binary digits each having a duration Ta seconds. In this, the
signal 1 is represented by a +1 volt and the symbol 0 by a -1 volt.
Therefore, the duo-binary coder output ck is given as the sum of present binary digit ak and the previous value ak-1 as
shown in the following equation.

The above equation states that the input sequence of uncorrelated binary sequence {ak} is changed into a sequence
of correlated three level pulses {ck}. This correlation between the pulses may be understood as introducing ISI in the
transmitted signal in an artificial manner.

Eye Pattern
An effective way to study the effects of ISI is the Eye Pattern. The name Eye Pattern was given from its resemblance
to the human eye for binary waves. The interior region of the eye pattern is called the eye opening. The following
figure shows the image of an eye-pattern.
Jitter is the short-term variation of the instant of digital signal, from its ideal position, which may lead to data
errors.
When the effect of ISI increases, traces from the upper portion to the lower portion of the eye opening increases and
the eye gets completely closed, if ISI is very high.
An eye pattern provides the following information about a particular system.
 Actual eye patterns are used to estimate the bit error rate and the signal-to-noise ratio.
 The width of the eye opening defines the time interval over which the received wave can be sampled
without error from ISI.
 The instant of time when the eye opening is wide, will be the preferred time for sampling.
 The rate of the closure of the eye, according to the sampling time, determines how sensitive the system is to
the timing error.
 The height of the eye opening, at a specified sampling time, defines the margin over noise.
Hence, the interpretation of eye pattern is an important consideration.

Equalization
For reliable communication to be established, we need to have a quality output. The transmission losses of the
channel and other factors affecting the quality of the signal, have to be treated. The most occurring loss, as we have
discussed, is the ISI.
To make the signal free from ISI, and to ensure a maximum signal to noise ratio, we need to implement a method
called Equalization. The following figure shows an equalizer in the receiver portion of the communication system.

The noise and interferences which are denoted in the figure, are likely to occur, during transmission. The
regenerative repeater has an equalizer circuit, which compensates the transmission losses by shaping the circuit. The
Equalizer is feasible to get implemented.
Error Probability and Figure-of-merit
 The rate at which data can be communicated is called the data rate.
 The rate at which error occurs in the bits, while transmitting data is called the Bit Error Rate BER.
 The probability of the occurrence of BER is the Error Probability.
The increase in Signal to Noise Ratio SNR decreases the BER, hence the Error Probability also gets decreased.
In an Analog receiver, the figure of merit at the detection process can be termed as the ratio of output SNR to the
input SNR. A greater value of figure-of-merit will be an advantage.

Digital Modulation Techniques intro.

Digital-to-Analog signals is the next conversion we will discuss in this chapter. These techniques are also called
as Digital Modulation techniques.
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.

1. 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.

2. FSK – Frequency Shift Keying

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

3. 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 BPSK and Quadrature Phase Shift Keying QPSK, according to the number of phase shifts. The
other one is Differential Phase Shift Keying DPSK which changes the phase according to the previous value.

4. 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

Amplitude Shift Keying ASK is a type of Amplitude Modulation which represents the binary data in the form of
variations in the amplitude of a signal.
Any modulated signal has a high frequency carrier. The binary signal when ASK modulated, gives a zero value
for Low input while it gives the carrier output for High input.
The following figure represents ASK modulated waveform along with its input.
To find the process of obtaining this ASK modulated wave, let us learn about the working of the ASK modulator.

ASK Modulator
The ASK modulator block diagram comprises of the carrier signal generator, the binary sequence from the message
signal and the band-limited filter. Following is the block diagram of the ASK Modulator.

The carrier generator, sends a continuous high-frequency carrier. The binary sequence from the message signal
makes the unipolar input to be either High or Low. The high signal closes the switch, allowing a carrier wave. Hence,
the output will be the carrier signal at high input. When there is low input, the switch opens, allowing no voltage to
appear. Hence, the output will be low.
The band-limiting filter shapes the pulse depending upon the amplitude and phase characteristics of the band-
limiting filter or the pulse-shaping filter.

ASK Demodulator
There are two types of ASK Demodulation techniques. They are −

 Asynchronous ASK Demodulation/detection

 Synchronous ASK Demodulation/detection
The clock frequency at the transmitter when matches with the clock frequency at the receiver, it is known as
a Synchronous method, as the frequency gets synchronized. Otherwise, it is known as Asynchronous.

Asynchronous ASK Demodulator

The Asynchronous ASK detector consists of a half-wave rectifier, a low pass filter, and a comparator. Following is
the block diagram for the same.
The modulated ASK signal is given to the half-wave rectifier, which delivers a positive half output. The low pass filter
suppresses the higher frequencies and gives an envelope detected output from which the comparator delivers a
digital output.

Synchronous ASK Demodulator

Synchronous ASK detector consists of a Square law detector, low pass filter, a comparator, and a voltage limiter.
Following is the block diagram for the same.

The ASK modulated input signal is given to the Square law detector. A square law detector is one whose output
voltage is proportional to the square of the amplitude modulated input voltage. The low pass filter minimizes the
higher frequencies. The comparator and the voltage limiter help to get a clean digital output.

Frequency Shift Keying FSK

Frequency Shift Keying FSK is the digital modulation technique in which the frequency of the carrier signal varies
according to the digital signal changes. FSK is a scheme of frequency modulation.
The output of a FSK modulated wave is high in frequency for a binary High input and is low in frequency for a binary
Low input. The binary 1s and 0s are called Mark and Space frequencies.
The following image is the diagrammatic representation of FSK modulated waveform along with its input.
To find the process of obtaining this FSK modulated wave, let us know about the working of a FSK modulator.

FSK Modulator
The FSK modulator block diagram comprises of two oscillators with a clock and the input binary sequence. Following
is its block diagram.

The two oscillators, producing a higher and a lower frequency signals, are connected to a switch along with an
internal clock. To avoid the abrupt phase discontinuities of the output waveform during the transmission of the
message, a clock is applied to both the oscillators, internally. The binary input sequence is applied to the transmitter
so as to choose the frequencies according to the binary input.

FSK Demodulator
There are different methods for demodulating a FSK wave. The main methods of FSK detection are asynchronous
detector and synchronous detector. The synchronous detector is a coherent one, while asynchronous detector is a
non-coherent one.

Asynchronous FSK Detector

The block diagram of Asynchronous FSK detector consists of two band pass filters, two envelope detectors, and a
decision circuit. Following is the diagrammatic representation.
The FSK signal is passed through the two Band Pass Filters BPFs, tuned to Space and Mark frequencies. The output
from these two BPFs look like ASK signal, which is given to the envelope detector. The signal in each envelope
detector is modulated asynchronously.
The decision circuit chooses which output is more likely and selects it from any one of the envelope detectors. It also
re-shapes the waveform to a rectangular one.

Synchronous FSK Detector

The block diagram of Synchronous FSK detector consists of two mixers with local oscillator circuits, two band pass
filters and a decision circuit. Following is the diagrammatic representation.

The FSK signal input is given to the two mixers with local oscillator circuits. These two are connected to two band
pass filters. These combinations act as demodulators and the decision circuit chooses which output is more likely and
selects it from any one of the detectors. The two signals have a minimum frequency separation.
For both of the demodulators, the bandwidth of each of them depends on their bit rate. This synchronous
demodulator is a bit complex than asynchronous type demodulators.

Phase Shift Keying PSK -bpsk

Phase Shift Keying PSK is the digital modulation technique in which the phase of the carrier signal is changed by
varying the sine and cosine inputs at a particular time. PSK technique is widely used for wireless LANs, bio-metric,
contactless operations, along with RFID and Bluetooth communications.
PSK is of two types, depending upon the phases the signal gets shifted. They are −
Binary Phase Shift Keying BPSK
This is also called as 2-phase PSK or Phase Reversal Keying. In this technique, the sine wave carrier takes two phase
reversals such as 0° and 180°.
BPSK is basically a Double Side Band Suppressed Carrier DSBSC modulation scheme, for message being the digital
information.

Quadrature Phase Shift Keying QPSK

This is the phase shift keying technique, in which the sine wave carrier takes four phase reversals such as 0°, 90°,
180°, and 270°.
If this kind of techniques are further extended, PSK can be done by eight or sixteen values also, depending upon the
requirement.

BPSK Modulator
The block diagram of Binary Phase Shift Keying consists of the balance modulator which has the carrier sine wave
as one input and the binary sequence as the other input. Following is the diagrammatic representation.

The modulation of BPSK is done using a balance modulator, which multiplies the two signals applied at the input. For
a zero binary input, the phase will be 0° and for a high input, the phase reversal is of 180°.
Following is the diagrammatic representation of BPSK Modulated output wave along with its given input.

The output sine wave of the modulator will be the direct input carrier or the
inverted 180°phaseshifted180°phaseshifted input carrier, which is a function of the data signal.

BPSK Demodulator
The block diagram of BPSK demodulator consists of a mixer with local oscillator circuit, a bandpass filter, a two-
input detector circuit. The diagram is as follows.
By recovering the band-limited message signal, with the help of the mixer circuit and the band pass filter, the first
stage of demodulation gets completed. The base band signal which is band limited is obtained and this signal is used
to regenerate the binary message bit stream.
In the next stage of demodulation, the bit clock rate is needed at the detector circuit to produce the original binary
message signal. If the bit rate is a sub-multiple of the carrier frequency, then the bit clock regeneration is simplified.
To make the circuit easily understandable, a decision-making circuit may also be inserted at the 2 nd stage of
detection.

Quadrature Phase Shift Keying QPSK

The Quadrature Phase Shift Keying QPSK is a variation of BPSK, and it is also a Double Side Band Suppressed
Carrier DSBSC modulation scheme, which sends two bits of digital information at a time, called as bigits.
Instead of the conversion of digital bits into a series of digital stream, it converts them into bit pairs. This decreases
the data bit rate to half, which allows space for the other users.

QPSK Modulator
The QPSK Modulator uses a bit-splitter, two multipliers with local oscillator, a 2-bit serial to parallel converter, and
a summer circuit. Following is the block diagram for the same.

At the modulator’s input, the message signal’s even bits (i.e., 2 nd bit, 4th bit, 6th bit, etc.) and odd bits (i.e., 1st bit,
3rd bit, 5th bit, etc.) are separated by the bits splitter and are multiplied with the same carrier to generate odd BPSK
(called as PSKI) and even BPSK (called as PSKQ). The PSKQ signal is anyhow phase shifted by 90° before being
modulated.
The QPSK waveform for two-bits input is as follows, which shows the modulated result for different instances of
binary inputs.

QPSK Demodulator
The QPSK Demodulator uses two product demodulator circuits with local oscillator, two band pass filters, two
integrator circuits, and a 2-bit parallel to serial converter. Following is the diagram for the same.

The two product detectors at the input of demodulator simultaneously demodulate the two BPSK signals. The pair of
bits are recovered here from the original data. These signals after processing, are passed to the parallel to serial
converter.

Differential Phase Shift Keying DPSK

In Differential Phase Shift Keying DPSK the phase of the modulated signal is shifted relative to the previous signal
element. No reference signal is considered here. The signal phase follows the high or low state of the previous
element. This DPSK technique doesn’t need a reference oscillator.
The following figure represents the model waveform of DPSK.
It is seen from the above figure that, if the data bit is Low i.e., 0, then the phase of the signal is not reversed, but
continued as it was. If the data is a High i.e., 1, then the phase of the signal is reversed, as with NRZI, invert on 1 a
form of differential encoding.
If we observe the above waveform, we can say that the High state represents an M in the modulating signal and the
Low state represents a W in the modulating signal.

DPSK Modulator
DPSK is a technique of BPSK, in which there is no reference phase signal. Here, the transmitted signal itself can be
used as a reference signal. Following is the diagram of DPSK Modulator.

DPSK encodes two distinct signals, i.e., the carrier and the modulating signal with 180° phase shift each. The serial
data input is given to the XNOR gate and the output is again fed back to the other input through 1-bit delay. The
output of the XNOR gate along with the carrier signal is given to the balance modulator, to produce the DPSK
modulated signal.

DPSK Demodulator
In DPSK demodulator, the phase of the reversed bit is compared with the phase of the previous bit. Following is the
block diagram of DPSK demodulator.
From the above figure, it is evident that the balance modulator is given the DPSK signal along with 1-bit delay input.
That signal is made to confine to lower frequencies with the help of LPF. Then it is passed to a shaper circuit, which is
a comparator or a Schmitt trigger circuit, to recover the original binary data as the output.

M-ary Encoding
The word binary represents two bits. M represents a digit that corresponds to the number of conditions, levels, or
combinations possible for a given number of binary variables.
This is the type of digital modulation technique used for data transmission in which instead of one bit, two or more
bits are transmitted at a time. As a single signal is used for multiple bit transmission, the channel bandwidth is
reduced.

M-ary Equation
If a digital signal is given under four conditions, such as voltage levels, frequencies, phases, and amplitude, then M =
4.
The number of bits necessary to produce a given number of conditions is expressed mathematically as
N=log2M
Where
N is the number of bits necessary
M is the number of conditions, levels, or combinations possible with N bits.
The above equation can be re-arranged as
2N=M
For example, with two bits, 22 = 4 conditions are possible.

Types of M-ary Techniques

In general, Multi-level M−ary modulation techniques are used in digital communications as the digital inputs with
more than two modulation levels are allowed on the transmitter’s input. Hence, these techniques are bandwidth
efficient.
There are many M-ary modulation techniques. Some of these techniques, modulate one parameter of the carrier
signal, such as amplitude, phase, and frequency.

M-ary ASK
M-ary FSK
This is called as M-ary Frequency Shift Keying M−aryFSK
The frequency of the carrier signal, takes on M different levels.

M-ary PSK
This is called as M-ary Phase Shift Keying M−aryPSK.
The phase of the carrier signal, takes on M different levels.

Some prominent features of M-ary PSK are −

 The envelope is constant with more phase possibilities.
 This method was used during the early days of space communication.
 Better performance than ASK and FSK.
 Minimal phase estimation error at the receiver.
 The bandwidth efficiency of M-ary PSK decreases and the power efficiency increases with the increase in M.

What is the signal-to-noise ratio?

In analog and digital communications, a signal-to-noise ratio, often written S/N or SNR, is a measure of the strength
of the desired signal relative to background noise (undesired signal). S/N can be determined by using a fixed formula
that compares the two levels and returns the ratio, which shows whether the noise level is impacting the desired
signal.
The ratio is typically expressed as a single numeric value in decibels (dB). The ratio can be zero, a positive number or
a negative number. A signal-to-noise ratio over 0 dB indicates that the signal level is greater than the noise level. The
higher the ratio, the better the signal quality.
For example, a Wi-Fi signal with S/N of 40 dB will deliver better network services than a signal with S/N of 20 dB. If a
Wi-Fi signal's S/N is too low, network performance can be impacted because it becomes more difficult for devices to
distinguish the desired signal from the noise. This can result in dropped packets and data retransmissions, leading to
lower throughput and higher latency.
Noise includes any unwanted disturbance that degrades the quality of the desired signal. It can include thermal,
quantum, electronic, impulse or intermodulation noise, as well as other forms of noise. Environmental factors, such
as temperature and humidity, can also affect noise levels.
If the noise is significant enough in comparison to the desired signal -- that is, S/N is low -- it can disrupt a wide range
of data transfers, including text files, graphics, telemetry, applications, and audio and video streams.

Difference between PCM and DPCM

Pulse Code Modulation (PCM): 
PCM is the technique used for remodeling analog signal into digital signal. Pulse Code Modulation
has good signal to noise ratio. For transmission channel, Pulse Code Modulation needs high
bandwidth than DPCM. 
The PCM method is split into 3 components, initial is that the transmission at the supply finish,
second regeneration at the transmission path and also the receiving finish. 
These steps are given below through figure: 
 

Differential Pulse Code Modulation (DPCM): 

DPCM is same as the PCM technique used for remodeling analog signal into digital signal. DPCM
has moderate signal to noise ratio. 
DPCM differs from PCM as a result of it quantizes the distinction of the particular sample and
expected price. that’s the explanation it’s referred to as as differential PCM. 
The operations at DPCM transmitter and DPCM receiver are given below through figure: 
 

Let’s see the difference between PCM and DPCM, which are given below: 
 

S.NO PCM DPCM

While DPCM stands for Differential Pulse Code

1. PCM stands for Pulse Code Modulation. Modulation.

2. In PCM, feedback is not provided. While in DPCM, feedback is provided.

3. It has good signal to noise ratio. While it has moderate signal to noise ratio.

4. It is less efficient than DPCM. While it is more efficient than PCM.

For transmission channel, PCM needs high

5. bandwidth(B). Whereas DPCM needs less bandwidth(B) than PCM.

PCM is complex than DPCM in terms of

6. complexity. While DPCM is simple in terms of complexity.

In PCM, seven bits are transmitted per eight

7. sample. In DPCM, four bits are transmitted per six sample.

In PCM, for transmitting bits rate varies from fifty While in DPCM, for transmitting bits rate varies from
8. five to sixty four. thirty two to forty eight.
Difference between Delta Modulation (DM) and Differential
Pulse Code Modulation (DPCM)
Delta Modulation (DM):
Delta modulation is associate analog to digital and digital to analog signal conversion technique.
Delta modulation is utilized to appreciate high signal to noise magnitude relation. It uses one bit
PCM code to appreciate digital transmission of analog signal. With delta modulation, rather than
transmit a coded illustration of a sample only one bit is transmitted, that simply indicates whether or
not or not the sample is larger or smaller than the previous sample. it’s the most effective kind or
simplest sort of Differential Pulse Code Modulation. Delta modulation signal is smaller than Pulse
Code Modulation system.

Differential Pulse Code Modulation (DPCM):

DPCM stands for Differential Pulse Code Modulation, is same as Pulse Code Modulation technique
used for reworking analog signal into digital signal. Differential Pulse Code Modulation has
moderate signal to noise magnitude relation. Differential Pulse Code Modulation differs from Pulse
Code Modulation as a results of it quantizes the excellence of the actual sample and expected value.
that’s the reason it’s cited as as differential Pulse Code Modulation(DPCM).
DPCM transmitter and DPCM receiver operations are given below through figure:

In the above diagram, if the signal is large then the next bit in digital data is 1 otherwise next bit is
0.
Let’s see that the difference between DM and DPCM, which are given below:

Comparison based
S.NO on DM DPCM

In DM, feedback exists in Here, feedback exists in both

1. Feedback transmitter. transmitter and receiver.

DM has poor signal to noise

2. signal to noise ratio ratio. DPCM has fair signal to noise ratio.

Transmission Here, DPCM requires less bandwidth

3. bandwidth It requires lowest bandwidth. than PCM.

4. Levels, step size In DM, step size is fixed. While here, number of levels are fixed.

DM is less efficient than

5. Efficiency DPCM. DPCM is more efficient.

Here more than one but less than

In DM, only one bit is used PCM(Pulse Code Modulation) bits are
6. Number of bits per sample. used.

Quantization error Slop overload distortion and Slop overload distortion and
7. and distortion granular noise are present. quantization noise are present.

It is generally used in It is mostly used in videos and

8. Applications speeches and images. speeches.

Difference between Pulse Code Modulation (PCM) and Delta

Modulation (DM)
PCM:
It is that the technique used for reworking analog signal into digital signal. PCM has good or sensible signal to noise
ration. For transmission, Pulse Code Modulation wants high transmitter bandwidth. PCM technique is split into three
elements, initial is that the transmission at the provision end, second regeneration at the transmission path and
conjointly the receiving end.

Delta Modulation(DM):
Delta modulation is an analog to digital and digital to analog signal conversion technique. Delta modulation is
employed to realize high signal to noise ratio. It uses one bit PCM code to realize digital transmission of analog
signal. With delta modulation, instead of transmit a coded illustration of a sample solely one bit is transmitted, that
merely indicates whether or not the sample is larger or smaller than the previous sample. it’s the best type or
simplest type of Differential Pulse Code Modulation. Delta modulation signal is smaller than Pulse Code Modulation
system.

If signal is large, the next bit in digital data is 1 otherwise 0.

Difference between Pulse Code Modulation (PCM) and Delta Modulation (DM):
S.NO PCM DM

1. PCM stands for Pulse Code Modulation. DM stands for Delta Modulation.

In PCM, feedback does not exist in transmitter or While in DM, feedback exists in
2. receiver. transmitter.
 S.NO PCM DM

3. Per sample 4, 8, or 16 bits are used. Here, only one bit is used per sample.

DM requires lowest transmitter

4. PCM requires highest transmitter bandwidth. bandwidth.

PCM is complex in terms of complexity of While DM is simple in terms of complexity

5. implementation. of implementation.

6. PCM has good signal to noise ratio. While DM has poor signal to noise ratio.

7. PCM is costly. DM is cheap.

PCM may be a technique wont to digitally Digital to analog and analog to digital
8. represent sampled analog signals. converter.

In PCM, signal requires encoder and decoder both In DM, signal can modulate and
9. sides. demodulate.

PM is mostly used in video telephony and audio DM is mostly used in speeches as well as
10. telephony. images.

https://www.sbistudy.com/probability-of-error-for-pcm-system/

Additive White Gaussian Noise (AWGN)

https://byjusexamprep.com/modulation-schemes-and-decoding-ii-study-notes-i-3e707c10-a6da-11e7-b6d3-
565118e362bb
Spread Spectrum Modulation
A collective class of signaling techniques are employed before transmitting a signal to provide a secure
communication, known as the Spread Spectrum Modulation. The main advantage of spread spectrum
communication technique is to prevent “interference” whether it is intentional or unintentional.
The signals modulated with these techniques are hard to interfere and cannot be jammed. An intruder with no
official access is never allowed to crack them. Hence, these techniques are used for military purposes. These spread
spectrum signals transmit at low power density and has a wide spread of signals.

Pseudo-Noise Sequence
A coded sequence of 1s and 0s with certain auto-correlation properties, called as Pseudo-Noise coding sequence is
used in spread spectrum techniques. It is a maximum-length sequence, which is a type of cyclic code.

Narrow-band and Spread-spectrum Signals

Both the Narrow band and Spread spectrum signals can be understood easily by observing their frequency spectrum
as shown in the following figures.

Narrow-band Signals
The Narrow-band signals have the signal strength concentrated as shown in the following frequency spectrum figure.
Following are some of its features −

 Band of signals occupy a narrow range of frequencies.

 Power density is high.
 Spread of energy is low and concentrated.
Though the features are good, these signals are prone to interference.

Spread Spectrum Signals

The spread spectrum signals have the signal strength distributed as shown in the following frequency spectrum
figure.

Following are some of its features −

 Band of signals occupy a wide range of frequencies.

 Power density is very low.
 Energy is wide spread.
With these features, the spread spectrum signals are highly resistant to interference or jamming. Since multiple
users can share the same spread spectrum bandwidth without interfering with one another, these can be called
as multiple access techniques.

FHSS and DSSS / CDMA

Spread spectrum multiple access techniques uses signals which have a transmission bandwidth of a magnitude
greater than the minimum required RF bandwidth.
These are of two types.

 Frequency Hopped Spread Spectrum FHSS

 Direct Sequence Spread Spectrum DSSS
Frequency Hopped Spread Spectrum  FHSS
This is frequency hopping technique, where the users are made to change the frequencies of usage, from one to
another in a specified time interval, hence called as frequency hopping. For example, a frequency was allotted to
sender 1 for a particular period of time. Now, after a while, sender 1 hops to the other frequency and sender 2 uses
the first frequency, which was previously used by sender 1. This is called as frequency reuse.
The frequencies of the data are hopped from one to another in order to provide a secure transmission. The amount
of time spent on each frequency hop is called as Dwell time.

Direct Sequence Spread Spectrum  DSSS

Whenever a user wants to send data using this DSSS technique, each and every bit of the user data is multiplied by a
secret code, called as chipping code. This chipping code is nothing but the spreading code which is multiplied with
the original message and transmitted. The receiver uses the same code to retrieve the original message.

Comparison between FHSS and DSSS/CDMA

Both the spread spectrum techniques are popular for their characteristics. To have a clear understanding, let us take
a look at their comparisons.
FHSS DSSS / CDMA

Multiple frequencies are used Single frequency is used

Hard to find the user’s frequency at any instant of time User frequency, once allotted is always the same

Frequency reuse is allowed Frequency reuse is not allowed

Sender need not wait Sender has to wait if the spectrum is busy

Power strength of the signal is high Power strength of the signal is low

Stronger and penetrates through the obstacles It is weaker compared to FHSS

It is never affected by interference It can be affected by interference

It is cheaper It is expensive

This is the commonly used technique This technique is not frequently used

Advantages of Spread Spectrum

Following are the advantages of spread spectrum −
 Cross-talk elimination
 Better output with data integrity
 Reduced effect of multipath fading
 Better security
 Reduction in noise
 Co-existence with other systems
 Longer operative distances
 Hard to detect
 Not easy to demodulate/decode
 Difficult to jam the signals
Although spread spectrum techniques were originally designed for military uses, they are now being used widely for
commercial purpose.
Linear Block Codes
In the linear block codes, the parity bits and message bits have a linear combination, which means that the resultant
code word is the linear combination of any two code words.
Let us consider some blocks of data, which contains k bits in each block. These bits are mapped with the blocks
which has n bits in each block. Here n is greater than k. The transmitter adds redundant bits which are n−k bits.
The ratio k/n is the code rate. It is denoted by r and the value of r is r < 1.
The n−kn bits added here, are parity bits. Parity bits help in error detection and error correction, and also in
locating the data. In the data being transmitted, the left most bits of the code word correspond to the message bits,
and the right most bits of the code word correspond to the parity bits.

Systematic Code
Any linear block code can be a systematic code, until it is altered. Hence, an unaltered block code is called as
a systematic code.
Following is the representation of the structure of code word, according to their allocation.

If the message is not altered, then it is called as systematic code. It means, the encryption of the data should not
change the data.

Convolution Codes
So far, in the linear codes, we have discussed that systematic unaltered code is preferred. Here, the data of
total n bits if transmitted, k bits are message bits and n−k bits are parity bits.
In the process of encoding, the parity bits are subtracted from the whole data and the message bits are encoded.
Now, the parity bits are again added and the whole data is again encoded.
The following figure quotes an example for blocks of data and stream of data, used for transmission of information.

The whole process, stated above is tedious which has drawbacks. The allotment of buffer is a main problem here,
when the system is busy.
This drawback is cleared in convolution codes. Where the whole stream of data is assigned symbols and then
transmitted. As the data is a stream of bits, there is no need of buffer for storage.
Hamming Codes
The linearity property of the code word is that the sum of two code words is also a code word. Hamming codes are
the type of linear error correcting codes, which can detect up to two bit errors or they can correct one bit errors
without the detection of uncorrected errors.
While using the hamming codes, extra parity bits are used to identify a single bit error. To get from one-bit pattern to
the other, few bits are to be changed in the data. Such number of bits can be termed as Hamming distance. If the
parity has a distance of 2, one-bit flip can be detected. But this can't be corrected. Also, any two bit flips cannot be
detected.
However, Hamming code is a better procedure than the previously discussed ones in error detection and correction.

BCH Codes
BCH codes are named after the inventors Bose, Chaudari and Hocquenghem. During the BCH code design, there is
control on the number of symbols to be corrected and hence multiple bit correction is possible. BCH codes is a
powerful technique in error correcting codes.

Cyclic Codes
The cyclic property of code words is that any cyclic-shift of a code word is also a code word. Cyclic codes follow this
cyclic property.

Digital Communication - Error Control Coding

Noise or Error is the main problem in the signal, which disturbs the reliability of the
communication system. Error control coding is the coding procedure done to control the
occurrences of errors. These techniques help in Error Detection and Error Correction.
There are many different error correcting codes depending upon the mathematical principles
applied to them. But, historically, these codes have been classified into Linear block
codes and Convolution codes.

Channel Coding Theorem

The noise present in a channel creates unwanted errors between the input and the output sequences of a digital
communication system. The error probability should be very low, nearly ≤ 10-6 for a reliable communication.
The channel coding in a communication system, introduces redundancy with a control, so as to improve the
reliability of the system. The source coding reduces redundancy to improve the efficiency of the system.
Channel coding consists of two parts of action.
 Mapping incoming data sequence into a channel input sequence.
 Inverse Mapping the channel output sequence into an output data sequence.
The final target is that the overall effect of the channel noise should be minimized.
The mapping is done by the transmitter, with the help of an encoder, whereas the inverse mapping is done by the
decoder in the receiver.

Hence, the maximum rate of the transmission is equal to the critical rate of the channel capacity, for reliable
error-free messages, which can take place, over a discrete memoryless channel. This is called as Channel
coding theorem.

Source Coding Theorem

The Code produced by a discrete memoryless source, has to be efficiently represented, which is an important
problem in communications. For this to happen, there are code words, which represent these source codes.
For example, in telegraphy, we use Morse code, in which the alphabets are denoted by Marks and Spaces. If the
letter E is considered, which is mostly used, it is denoted by “.” Whereas the letter Q which is rarely used, is denoted
by “--.-”
Let us take a look at the block diagram.

Where Sk is the output of the discrete memoryless source and bk is the output of the source encoder which is
represented by 0s and 1s.
The encoded sequence is such that it is conveniently decoded at the receiver.
Let us assume that the source has an alphabet with k different symbols and that the kth symbol Sk occurs with the
probability Pk, where k = 0, 1…k-1.
Let the binary code word assigned to symbol Sk, by the encoder having length lk, measured in bits.
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
When we observe the possibilities of the occurrence of an event, how surprising or uncertain it would be, it means
that we are trying to have an idea on the average content of the information from the source of the event.
Entropy can be defined as a measure of the average information content per source symbol. Claude Shannon, the
“father of the Information Theory”, provided a formula for it as −

Mutual Information
Let us consider a channel whose output is Y and input is X
Let the entropy for prior uncertainty be X = Hx
This is assumed before the input is applied
To know about the uncertainty of the output, after the input is applied, let us consider Conditional Entropy, given
that Y = yk
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.