Compiled

 by:  Dinesh  Baniya  Kshatri  

INTRODUCTION TO DATA COMMUNICATIONS

- COMPUTER NETWORK
Interconnected collection of autonomous computers that are able to exchange information.
• No master/slave relationship between computers in the network.
- DATA COMMUNICATIONS
Transmission of signals in a reliable and efficient matter.
- COMMUNICATION MODEL (SYSTEM)
The purpose of a communications system is to exchange data between two entities.

• Source: entity that generates data; eg. a person who speaks into the phone, or a
computer sending data to the modem.
• Transmitter: a device to transform/encode the signal generated by the source.
- the transformed signal is actually sent over the transmission system.
eg. a modem transforms digital data to analog signal that can be handled by the
telephone network.

• Transmission System (Channel): medium that allows the transfer of a signal from
one point to another.
eg. a telephone network for a computer/modem.

• Receiver: a device to decode the received signal for handling by destination device.
eg. A modem converts the received analog data back to digital for the use by the
computer.
• Destination: entity that finally uses the data.
eg. Computer on other end of a receiving modem.
Data Communications
Data communications is the transfer of information that is in digital form, before it enters the
communication system.
- Basic Elements of a Communication System
Signal s(t) Channel r(t)
Transmitter Receiver

n(t)
Information
source & input
Noise Output
transducer transducer

Basic Elements of a Communication System

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• Information: generated by the source may be in the form of voice, a picture or a plain
text. An essential feature of any source that generates information is that its output is
described in probabilistic terms; that is, the output is not deterministic.
A transducer is usually required to convert the output of a source in an electrical signal
that is suitable for transmission.

• Transmitter: a transmitter converts the electrical signal into a form that is suitable for
transmission through the physical channel or transmission medium. In general, a
transmitter performs the matching of the message signal to the channel by a process
called modulation.
The choice of the type of modulation is based on several factors, such as:
- the amount of bandwidth allocated,
- the type of noise and interference that the signal encounters in transmission over the
channel,
- and the electronic devices that are available for signal amplification prior to
transmission.

• Channel: the communication channel is the physical medium that connects the
transmitter to the receiver. The physical channel may be a pair of wires that carry the
electrical signals, or an optical fibre that carries the information on a modulated light
beam or free space at which the information-bearing signal are electromagnetic waves.

• Receiver: the function of a receiver is to recover the message signal contained in the
received signal. The main operations performed by a receiver are demodulation,
filtering and decoding.
Analog and Digital Data Transmission
- Data are entries that convey information.
- Signals are electrical encoding (representation) of data.
- Signalling is the act of propagation of signals through a suitable medium.
The terms analog and digital correspond to continuous and discrete, respectively. These
two terms are frequently used in data communications.

Analog data takes on continuous values on some interval. The most familiar example of
analog data is audio signal. Frequency components of speech may be found between
3
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20 Hz and 20 kHz. The basic speech energy is concentrated between 300-3400 Hz. The
frequencies up to 4000 Hz add very little to the intelligibility of human ear.
Another common example of analog data is video. The outputs of many sensors, such as
temperature and pressure sensors, are also examples of analog data.

Digital data takes on discrete values; eg. a computer’s output.

- Analog transmission is a means of transmitting analog signals regardless of their
content. The data may be analog or digital.
- Digital transmission is the transfer of information through a medium in digital form. A
digital signal can be transmitted only for a limited distance.
- Data communications is the transfer of information that is in digital form, before it
enters the communication system.

• Two methods of sending data from computer A to computer B. both cases are
examples of data communications, because the original data is digital in nature.
Digital
Source Modem Modem

Analog Transmission

A B
Digital Transmission

• Two ways of transmitting analog information. In either cases it is not data

communications, because the original information is not digital.
Digital Transmission

ADC DAC

Analog Destination
Source

Analog Transmission

ADC: Analog-Digital-Converter
DAC: Digital-Analog-Converter
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Digital Communication System

Up to this point, we have described an electrical communication system in rather broad terms based
on the implicit assumption that the message signal is a continuous time-varying waveform. We refer
to such continuous-time signal waveforms as analog signals and to the corresponding information
sources that produce such signals as analog sources. Analog signals can be transmitted directly via
carrier modulation over the communication channel and demodulated accordingly at the receiver.
We call such a communication system an analog communication system.
Alternatively, an analog source output may be converted into a digital form and the message can be
transmitted via digital modulation and demodulated as a digital signal at the receiver. There are
some potential advantages to transmitting an analog signal by means of digital modulation. The most
important reason is that signal fidelity is better controlled through digital transmission than analog
transmission. In particular, digital transmission allows us to regenerate the digital signal in long-
distance transmission, thus eliminating effects of noise at each regeneration point. In contrast, the
noise added in analog transmission is amplified analog with the signal when amplifiers are used
periodically to boost the signal level in long-distance transmission. Another reason for choosing
digital transmission over analog is that the analog message signal may be highly redundant. With
digital processing, redundancy may be removed prior to modulation, thus conserving channel
bandwidth. Yet a third reason may be that digital communication systems are often cheaper to
implement.
In some applications, the information to be transmitted is inherently digital, e.g., in the form of
English text, computer data, etc. In such cases, the information source that generates the data is
called a discrete (digital) source.
In a digital communication system, the functional operations performed at the transmitter and
receiver must be expanded to include message signal discrimination at the transmitter and message
signal synthesis or interpolation at the receiver. Additional functions include redundancy removal,
and channel coding and decoding.
Figure 1.2 illustrates the functional diagram and the basic elements of a digital communication
system. The source output may be either an analog signal, such as audio or video signal, or a digital
signal, such as the output of a Teletype machine, which is discrete in time and has a finite number of
output characters. In a digital communication system, the messages produced by the source are
usually converted into a sequence of binary digits.

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Information
Source Channel Digital
source & input
encoder encoder modulator
transducer

Data Communication System Channel

Output Output Source Channel Digital

transducer decoder decoder demodulator
Signal

Figure 1.2. Basic elements of a digital communication system

Ideally, we would like to represent the source output (message) by as few binary digits as
possible. In other words, we seek an efficient representation of the source output that results
in little or no redundancy. The process of efficiently converting the output of either an analog
or a digital source into a sequence of binary digits is called source encoder or data
compression.
The sequence of binary digits from the source encoder, which we call the information
sequence, is passed to the channel encoder. The purpose of the channel encoder is to introduce
in a controlled manner some redundancy in the binary information sequence, which can be
used at the receiver to overcome the effects of noise and interference encountered in the
transmission of the signal through the channel. Thus, the added redundancy serves to increase
the reliability of the received data and improves the fidelity of the received signal. In effect,
redundancy in the information sequence aids the receiver in decoding the desired information
sequence. For example, a (trivial) form of encoding of the binary information sequence is
simply to repeat each binary digit m times, where m is some positive integer. More
sophisticated (nontrivial) encoding involves taking k information bits at a time and mapping
each k-bit sequence into a unique n-bit sequence, called a code word. The amount of
redundancy introduced by encoding the data in this manner is measured by the ratio n/k. The
reciprocal of this ratio, namely, k/n is called the rate of the code or, simply, the code rate.
The binary sequence at the output of the channel encoder is passed to the digital modulator,
which serves as the interface to the communications channel. Since nearly all of the
communication channels encountered in practice are capable of transmitting electrical signals
(waveforms), the primary purpose of the digital modulator is to map the binary information
sequence into signal waveforms. To elaborate on this point, let us suppose that the coded
information sequence is to be transmitted one bit at a time at some uniform rate R bits/s. The
digital modulator may simply map the binary digit 0 into a waveform s0(t) and the binary digit

1 into a waveform s1(t). In this manner, each bit from the channel encoder is transmitted
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separately. We call this binary modulation. Alternatively, the modulator may transmit b coded
information bits at a time by using M =2b distinct waveform si(t), I = 0, 1, …, m-1, one
waveform for each of the 2b possible b-bits sequences. We call this M-ary modulation (M
>2). Note that a new b-bit sequence enters the modulator every b/R seconds. Hence, when the
channel bit rate R is fixed, the amount of time available to transmit one of the M waveforms
corresponding to a b-bit sequence is b times the period in a system that uses binary
modulation.
At the receiving end of a digital communications system, the digital demodulator processes
the channel-corrupted transmitted waveform and reduces each waveform to a single number
that represents an estimate of the transmitted data symbol. For example, when binary
modulation is used, the demodulator may process the received waveform and decide on
whether the transmitted bit is a 0 or 1. In such a case, we say the demodulator has made a
binary decision. As one alternative, the demodulator may make a ternary decision; that is, it
decides that the transmitted bit is either a 0 or 1 or it makes no decision at all, depending on
the apparent quality of the received signal. When no decision is made on a particular bit, we
say that the demodulator has inserted an erasure in the demodulated data. Using the
redundancy in the transmitted data, the decoder attempts to fill in the positions where erasures
occurred. Viewing the decision process performed by the demodulator as a form of
quantization, we observe that binary and ternary decisions are special cases of a demodulator
that quantizes to Q levels, where Q ≥ 2 In general, if the digital communications system
employs M-ary modulation, where m = 0,1, … , M represent the M possible transmitted sym-

bols, each corresponding to k =log2 M bits, the demodulator may make A Q-ary decision,
where Q ≥ M .In the extreme case where no quantization is performed, Q = ∞.
When there is no redundancy in the transmitted information, the demodulator must decide
which of the M waveforms was transmitted in any given time interval. Consequently, Q = M,
and since there is no redundancy in the transmitted information, no discrete channel decoder
is used following the demodulator. On the other hand, when there is redundancy introduced
by a discrete channel encoder at the transmitter, the Q-ary output from the demodulator
occurring every k/R seconds is fed to the decoder, which attempts to reconstruct the original
information sequence from knowledge of the code used by the channel encoder and the
redundancy contained in the received data. A measure of how well the demodulator and
encoder perform is the frequency with which errors occur in the decoded sequence. More
precisely, the average probability of a bit-error at the output of the decoder is a measure of the
performance of the demodulator-decoder combination. In general, the probability of error is a
function of the code characteristics, the types of waveforms used to transmit the information
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over the channel, the transmitter power, the characteristics of the channel (i.e., the amount of
noise), the nature of the interference, etc., and the method of demodulation and decoding.
These items and their effect on performance will be discussed in detail in subsequent chapters.
As a final step, when an analog output is desired, the source decoder accepts the output
sequence from the channel decoder, and from knowledge of the source encoding method used,
attempts to reconstruct the original signal from the source. Due to channel decoding errors
and possible distortion introduced by the source encoder and, perhaps, the source decoder, the
signal at the output of the source decoder is an approximation to the original source output.
The difference or some function of the difference between the original signal and the
reconstructed signal is a measure of the distortion introduced by the digital communications
system.

Early Work in Digital Communications

Although Morse is responsible for the development of the first electrical digital
communication system (telegraphy), the beginnings of what we now regard as modem digital
communications stem from the work of Nyquist (1924), who investigated the problem of
determining the maximum signalling rate that can be used over a telegraph channel of a given
bandwidth without intersymbol interference. He formulated a model of a telegraph system in
which a transmitted signal has the general form

S (t) = ∑an g(t − nT )

n

Where g(t) represents a basic pulse shape and {an} is the binary data sequence of {±1}
transmitted at a rate of 1/T bits per second. Nyquist set out to determine the optimum pulse
shape that was bandlimited to W Hz and maximised the bit rate 1/T under the constraint that
the pulse caused no intersymbol interference at the sampling times k/T, k = 0, ±1, ±2,.... His
studies led him to conclude that the maximum pulse rate1/T is 2W pulses per second. This rate
is now called the Nyquist rate. Moreover, this pulse rate can be achieved by using the pulses
g(t) = (sin2π Wt)/2π Wt. This pulse shape allows the recovery of the data without intersymbol
interference at the sampling instants Nyquist’s result is equivalent to a version of the sampling
theorem for bandlimited signals, which was later stated precisely by Shannon (1948). The
sampling theorem states that a signal of bandwidth W can be reconstructed from samples
taken at the Nyquist rate of 2W samples per second using the interpolation formula.

In light of Nyquist's work Hartley (1928) considered the issue of the amount of data that can
be transmitted reliably over a bandlimited channel when multiple amplitude levels are used.

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Due to the presence of noise and other interference, Hartley postulated that the receiver could
reliably estimate the received signal amplitude to some accuracy, say Aδ. This investigation
led Hartley to conclude that there is maximum data rate that can be communicated reliably
over a bandlimited channel when the maximum signal amplitude is limited to Amax (fixed
power constraint) and the amplitude resolution is Aδ.
Another significant advance in the development of communications was the work of Wiener
(1942) who considered the problem of estimating a desired signal waveform s(t) in the
presence of additive noise n(t), based on observation of the received signal r(t) = s(t) + n(t).
This problem arises in signal demodulation. Wiener determined the linear filter whose output
is the best mean-square approximation to the desired signal s(t). The resulting filter is called
the optimum linear (Wiener) filter. Hartley’s and Nyquist results on the maximum
transmission rate of digital information were precursors to the work of Shannon (1948 a, b)
who established the mathematical foundations for information theory and derived the
fundamental limits for digital communication systems. In his pioneering work, Shannon
formulated the basic problem of reliable transmission of information in statistical terms, using
probabilistic models for information sources and communication channels. Based on such a
statistical formulation, he adopted a logarithmic measure for the information content of a
source. He also demonstrated that the effect of a transmitter power constraint, a bandwidth
constraint, and additive noise can be associated with the channel and incorporated into a
single parameter, called the channel capacity For example, in the case of an additive white
(spectrally flat) Gaussian noise interference, an ideal bandlimited channel of bandwidth W has
a capacity C given by
P
C = W log2(1 + ———) bits/s
WN0
where P is the average transmitted power and N0 is the power spectral density of the additive
noise. The significance of the channel capacity is as follows: If the information rate R from
the source is less than C (R < C), then it is theoretically possible to achieve reliable (error-
free) transmission through the channel by appropriate coding. On the other hand, if R > C,
reliable transmission is not possible regardless of the amount of signal processing performed
at the transmitter and receiver. Thus, Shannon established basic limits on communication of
information and gave birth to a new field that is now called information theory.
Initially the fundamental work of Shannon had a relatively small impact on the design and
development of new digital communications systems. In part, this was due to the small

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demand for digital information transmission during the 1950's. Another reason was the
relatively large complexity and, hence, the high cost of digital hardware required to achieve
the high efficiency and high reliability predicted by Shannon's theory.
Another important contribution to the field of digital communications is the work of
Kotelnikov (1947), which provided a coherent analysis of the various digital communication
systems based on a geometrical approach. Kotelnikov approach was later expanded by
Wozencraft and Jacobs (1965).
The increase in the demand for data transmission during the last three decades, coupled with
the development of more sophisticated integrated circuits, has led to the development of very
efficient and more reliable digital communications systems. In the course of these
developments, Shannon's original results and the generalization of his results on maximum
transmission limits over a channel and on bounds on the performance achieved have served as
benchmarks for any given communications system design. The theoretical limits derived by
Shannon and other researchers that contributed to the development of information theory
serve as an ultimate goal in the continuing efforts to design and develop more efficient digital
communications systems.
Following Shannon's publications name the classic work of Hamming (1950) on error
detecting and error-correcting codes to combat the detrimental effects of channel noise.
Hamming's work stimulated many researchers in the years that followed, and a variety of new
and powerful codes were discovered, many of which are used today in the implementation of
modem communication systems.

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DATA TRANSMISSION & SIGNALS

Data Transmission
Concepts and Terminology
• Transmission Terminology
Transmission from transmitter to receiver goes over some transmission medium using
electromagnetic waves.
- Guided Media: waves are guided along a physical path; twisted pair, optical fibre,
coaxial cable.
- Unguided Media: waves are not guided; air waves radio waves.
- Direct Link: signal goes from transmitter to receiver without intermediate devices,
other than amplifiers and repeaters.
- Point-to Point Link: guided media with direct link between two devices.
- Multipoint Guided Configuration: more than two devices can share the same
medium.

• Frequency, Spectrum, & Bandwidth

- Signal is generated by a transmitter and transmitted over a medium.
- Signal is a function of time or frequency.
A signal is any function that carries information. Based on the range of variation of
independent variables, signals can be divided into two classes: continuous-time (or
analog) signals and discrete-time (or digital) signals. A signal is a function of time, but
can also be expressed as a function of frequency; that is, the signal consists of components
of different frequencies.

• Analog and Digital Signals

Information can be analog or digital. Analog information is continuous. Digital
information is discrete.
Signals can be analog or digital. Analog signals can have any value in a range; while
digital signals can have only a limited number of values.

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Value Value

Analog Signal Digital Signal

• Time –Domain Concepts

- Continuous Signal: Signal intensity varies in a smooth fashion over time; no breaks
or discontinuities in the signal.
- Discrete Signals: Signal intensity can take one of two pre-specified values for any
amount of time.
A continuous time signal is defined by a continuous independent variable. A signal s(t) is
continuous if

lim sc (t) = s(t) for all a.

t →a

- Periodic Signal
A signal s(t) is periodic if

s(t + T ) = s(t)
where T is the period of the signal.

sc(t) sd(t)

A
t t

φ TT

Three important characteristics of a periodic signal are: amplitude, frequency, and phase.
Amplitude (A) is the instantaneous value of a signal at any time, and is measured in volts.
Frequency (f) is the inverse of the period (T); (f=1/T), or the number of period repetition in
one second, and is measured in cycles per second or Hertz (Hz). Phase (φ) is a measure of
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the relative position in time within a single period of a signal. Thus, we can express a
sinusoid signal as

s(t) = Asin(2πft +φ)

where A is the amplitude, f is the frequency, and φ is the phase.

• Frequency-Domain Concepts
Any signal can also be viewed as a function of frequency, for example, the signal

s(t) = sin 2πft +1/ 3sin 3(2πft) +1/ 5sin 5(2πft)

consists of three components as shown in the figure below:

sin 2πft

1/3 sin 3(2πft)

1/5 sin 5(2πft)

s(t)

The frequency components of a signal can be determined using Fourier analysis. The
following figure shows the spectrum S(f) of the signal s(t). The spectrum of a signal is the
range of frequencies that it contains. For this signal the spectrum extends from f to 5f . the
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spectrum in this case is discrete.

S(f)
1

1/3
1/5
f
f 3f 5f

Many signals, such as the one in the following figure, have continuous spectrum Sc(f) and

sd(t)

t
T

and an infinite bandwidth as shown below:

Sc(f)

1/T 2/T 3/T 4/T n/T

f

However, most of the energy in the signal is contained in a relatively narrow band of
frequencies. This band is referred to as the effective bandwidth, or just bandwidth.
If a signal includes a component of zero frequency, that component is called dc
component or constant component.

The signal s1(t) in the following figure is obtained by adding a dc component on s(t):

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s1(t)

With a dc component, it has a frequency term at f = 0 and a non-zero average amplitude.

S1(f)
1.2 1
Spectrum of s1(t)
1/3
1/5

0 f 3f 5f f

s1 (t) = 1.2 + sin (2πft) +1/ 3sin 3(2πft) +1/ 5sin 5(2πft )

• Fundamental Frequency
Base frequency such that the frequency of all components can be expressed as its integer
multiples; the period of the aggregate signal is the same as the period of the fundamental
frequency:
- Each signal can be decomposed into a set of sinusoid signals by making use of
Fourier’s analysis.
- The time-domain function s(t) specifies a signal in terms of its amplitude at each
instant of time.
- The frequency-domain function S(f) specifies the signal in terms of peak amplitude of
constituent frequencies.
Spectrum
Range of frequencies contained in a signal.
Absolute Bandwidth
Width of the spectrum.

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Effective Bandwidth
Narrow band of frequencies containing most of the energy of the signal.

DC Component
Component of zero frequency; changes the average amplitude of the signal to non-zero.

Relationship between Data Rate and Bandwidth

- Any transmitter/receiver system can accommodate only a limited range of frequencies.
* The range for FM radio transmission is 88-108 MHz
- This limits the data rate that can be carried over the transmission medium.
- Consider a square wave. Suppose that we let the positive pulse to be binary 1 and the
negative pulse to be binary 0. Then, the waveform represents the binary stream
1010… and duration (period) of each pulse is 1/2f. Thus, the data rate is equal to 2f
bits per second (bps) or the data rate is equal to twice the fundamental frequency of the
digital signal. It can be shown that the frequency-domain representation of this
waveform is:
1
s(t) = ∑ ∞
sin(2πkft)
k
k =1

- This waveform has infinite number of frequency components and infinite bandwidth.
th
- Peak amplitude of the k frequency component is 1/k, so most of the energy is
concentrated in the first few frequencies.

Ex
Consider a digital transmission system capable of transmitting signals with a bandwidth of
4 MHz.
Case 1
Approximate the square wave with a waveform of the first three sinusoidal components

sin (2πft ) +1/ 3sin(2π (3 f )t) +1/ 5sin(2π (5 f )t)

6
If f = 10 cycles per second, or 1 MHz, the bandwidth of the signal

6 6 6
[
s(t) = sin (2π ×10 t) +1/ 3sin(2π × 3 ×10 t) +1/ 5sin(2π × 5 ×10 t) ]
6 6
is 5x10 -10 = 4 MHz
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6
For f = 1 MHz, the period of the fundamental frequency is T = 1/10 = 1µs. If the
waveform is a bit string of 1’s and 0’s, then one bit occurs every 0.5 µs for a data rate of
6
2x10 bps or 2 Mbps.

Case 2
Assume a bandwidth of 8 MHz and f = 2 MHz; this gives us the signal bandwidth as

6 6
(5x2x10 )-(2x10 ) = 8 MHz
But T = 1/f = 0.5 µs, so that the time needed for one bit is 0.25 µs, giving a data rate of 4
Mbps. Other things being equal, doubling the bandwidth doubles the potential data rate.

Case 3
Let us represent the signal by the first two components of the sinusoid as

[sin (2πft ) +1/ 3sin(2π (3 f )t)]

Assume that f = 2 MHz and T = 1/f = 0.5 µs so that the time needed for one bit is 0.25 µs,
giving a data rate of 4 Mbps.
Bandwidth of the signal is
6 6
(3x2x10 )-(2x10 ) = 4 MHz
A given bandwidth can support various data rates depending on the ability of the receiver
to differentiate between 0 and 1 in the presence of noise and other impairments.

Ex
If a periodic signal is decomposed into five waves with frequencies of 100, 300, 500, 700,
and 900 Hz, what is the bandwidth of the signal?
Let fh be the highest frequency, fl be the lowest frequency, and B be the bandwidth, then
B = fh - fl
= 900-100 = 800 Hz
Digital Signals
Data can be represented by a digital signal. For example, a 1 can be encoded as a positive
voltage and a 0 as a zero voltage.
Amplitude

1 0 1 1 0 0 0 1
…
Time
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Amplitude, Period and Phase

The three characteristics of periodic signals can be redefined for a periodic digital signal

Amplitude …
Time
No phase shift

Amplitude …
Time
o
180 phase shift

Amplitude Amplitude

Time Time
¼ cycle
o o
0 phase shift 90 phase shift
(no phase shift)
Amplitude Amplitude

Time Time
½ cycle
o o
180 phase shift 270 phase shift

Bit Interval and Bit Rate

Most digital signals are aperiodic and thus terms like period or frequency are not
appropriate. Two new terms, bit interval (instead of period) and bit rate (instead of

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frequency) are used to describe a digital signal. The bit interval is the time required to
send one single bit. The bit rate is the number of bit intervals per second. This means that
the bit rate is the number of bits sent in one second, usually expressed in bits per second
(bps).
Amplitude
1 second = 8 bit intervals
bit rate = 8 bps

1 0 1 1 0 0 0 1
…
Time
bit interval

Decomposition of a Digital Signal

A digital signal can be decomposed into an infinite number of simple sine waves called
harmonics, each with different amplitude, frequency and phase. This means that when a
digital signal is sent along a transmission medium, an infinite number of simple signals is
being sent.
Harmonics of a Digital Signal

… …

a) First harmonic only b) First, third, and fifth harmonics

… …

c) First, third, fifth, and seventh harmonics d) Infinite number of harmonics

If some of the components do not pass through the medium, this results in distortion of
the signal at the receiver side. Since no practical medium (such as a cable) is capable of
transferring the entire range of frequencies, there will always be distortion.

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Amplitude

Frequency
0 Infinite bandwidth Infinity

a) Spectrum for exact replica

Amplitude

Frequency
Significant bandwidth

b) Significant spectrum
The part of the infinite spectrum whose amplitudes are significant (above an acceptable
threshold), is called the significant spectrum, and its bandwidth is called the significant
bandwidth.
When the bit rate increases, the significant bandwidth widens. For example, if the bit rate
is 1000 bps, the significant bandwidth can be around 200 Hz, depending on the level of
noise in the system. If the bit rate is 2000 bps, the significant bandwidth can be 400 Hz.

1000 bps

200 Hz

2000 bps

400 Hz
A transmission medium with a particular bandwidth is capable of transmitting only digital
signals whose significant bandwidth is less than the bandwidth of the medium.

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Channel Capacity
The maximum bit rate a transmission medium can transfer is called channel capacity of
the medium. The capacity of a channel, however, depends on the type of encoding
technique and the signal-to-noise ratio of the system. For example a normal telephone line
with a bandwidth of 3000 Hz is capable of transferring up to 20,000 bps, but other factors,
like noise, can decrease this rate.

1000 bps

Bandwidth = x

2000 bps

Bandwidth = 2x

3000 bps
Bandwidth = 3x

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Noise
In the absence of a signal, a transmission medium ideally has no electrical signal present.
In practice, however, there is what we call line noise level, because of random
perturbations on the line even when no signal is being transmitted. An important
parameter associated with a transmission medium, therefore, is the ratio of the average
power in a received signal, S , to the power in the noise level, N . The ratio S / N is known
as the signal-to-noise ratio (SNR) and normally is expressed in decibels, that is:
⎛ S ⎞
SNR = 10 log10 ⎜ ⎟dB
⎝ N ⎠
- A high SNR ratio means a good-quality signal.
- A low SNR ratio means a low-quality signal.
The theoretical maximum data rate of transmission channel is related to the SNR ratio
and we can determine this rate using a formula attributed to Shannon and Hartley. This is
known as the Shannon-Hartley Law, which states:
⎛ S ⎞
C = W log 2 ⎜1 + ⎟ bps
⎝ N ⎠

⎛ S ⎞
≈ 3.32 W log10 ⎜1 + ⎟ bps
⎝ N ⎠

where C is the data rate in bps, W is the bandwidth of the line channel in Hz, S is the
average signal power in watts and N is the random noise power in watts.
Ex
Consider a voice channel with BW of 2,800 Hz. A typical value of S/N for a telephone
line is 20 dB. What is the channel capacity?

Solution
SNR = 20 dB
20 = 10 log10 (S/N) ⇒ S/N = 100
W = 2,800 Hz
⎛ S ⎞ ⎛ S ⎞
C = W log 2 ⎜1 + ⎟ bps ≈ 3.32 W log10 ⎜1 + ⎟ bps
⎝ N ⎠ ⎝ N ⎠
≈ 3.32 (2800) log10 (1 +100)
C = 18,632 bps
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TRANSMISSION MEDIA

There are two basic categories of transmission media: guided and unguided media.

Guided transmission media use cabling system that guides the data signals along a specific
path. Data signals are bound by the cabling system. Guided media is also known as bound
media. ―Cablingǁ‖ is meant in a generic sense, and is not meant to be interpreted as copper
wire cabling only.

Unguided transmission media consists of a means for the data signals to travel but nothing
to guide them along a specific path. The data signals are not bound to a cabling media and are
therefore often called unbound media.

Transmission Media: Guided

There four basic types of guided media:

a. Open Wire
b. Twisted Pair
c. Coaxial Cable
d. Optical Fibre

Open Wire

Open wire is traditionally used to describe the electrical wire strung along power poles. There
is a single wire strung between poles. No shielding or protection from noise interference is
used. We are going to extend the traditional definition of open wire to include any data signal
path without shielding or protection from noise interference. This can include multi conductor
cables or single wires. This medium is susceptible to a large degree of noise and interference
and consequently is not acceptable for data transmission except for short distances under 20
ft.

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Twisted Pair

The wires in twisted pair cabling are twisted together in pairs. Each pair consists of a wire
used for the positive data signal and a wire used for the negative data signal. Any noise that
appears on one wire of the pair will also occur on the other wire. Since the wires have
opposite polarities, they are 180 degrees out of phase. When noise appears on both wires, it
cancels or nulls itself out at the receiving end. Twisted pair cables are most effectively used in
systems that use a balanced line method of transmission: polar line coding (Manchester
encoding) as opposed to unipolar line coding.

Unshielded Twisted Pair

The degree of reduction in noise interference is determined specifically by the number of
turns per foot. Increasing the number of turns per foot reduces the noise interference. To
further improve noise rejection, a foil or wire braid ―shieldǁ‖ is woven around the twisted
pairs. This shield can be woven around individual pairs or around a multi-pair conductor
(several pairs).

Shielded Twisted Pair

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Cables with a shield are called shielded twisted pair and are commonly abbreviated STP.
Cables without a shield are called unshielded twisted pair or UTP. Twisting the wires together
results in a characteristic impedance for the cable. Typical impedance for UTP is 100 Ohm
for Ethernet 10BaseT cable.

UTP or unshielded twisted pair cable is used in Ethernet 10BaseT and can also be used with
Token Ring. It uses the RJ line of connectors (RJ45, RJ11, etc..).

STP or shielded twisted pair is used with the traditional Token Ring cabling or ICS-IBM
Cabling System. It requires a custom connector. IBM STP (shielded twisted pair) has a
characteristic impedance of 150 Ohm.

Coaxial Cable

Coaxial cable consists of two conductors. The inner conductor is held inside an insulator with
the other conductor woven around it providing a shield. An insulating protective coating
called a jacket covers the outer conductor.

Coaxial Cable
The outer shield protects the inner conductor from outside electrical signals. The distance
between the outer conductor (shield) and inner conductor, plus the type of material used for
insulating the inner conductor determine the cable properties or impedance. Typical
impedances for coaxial cables are 75 Ohms for TV cable, 50 Ohms for Ethernet Thinnet and
Thicknet. The excellent control of the impedance characteristics of the cable allow higher
data rates to be transferred than with twisted pair cable.

Optical fibre

Optical fibre consists of thin glass fibres that can carry information at frequencies in the
visible light spectrum and beyond. The typical optical fibre consists of a very narrow strand
of glass called the core. Around the core is a concentric layer of glass called the cladding. A
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-6
typical core diameter is 62.5 microns (1 micron = 10 m). Typically Cladding has a diameter
of 125 microns. Coating the cladding is a protective coating consisting of plastic, it is called
the Jacket.

Fibre Optic Cables

Just as standard electric cables come in a variety of sizes, shapes, and types, fibre optic cables
are available in different configurations. The simplest cable is just a single strand of fibre,
whereas complex cables are made up of multiple fibres with different layers and other
elements.

The portion of a fibre optic cable (core) that carries the light is made from either glass or
plastic. Another name for glass is silica. Special techniques have been developed to create
nearly perfect optical glass or plastic, which is transparent to light. Such materials can carry
light over a long distance. Glass has superior optical characteristics over plastic. However,
glass is far more expensive and more fragile than plastic. Although the plastic is less
expensive and more flexible, its attenuation of light is greater. For a given intensity, light will
travel a greater distance in glass than in plastic. For very long distance transmission, glass is
certainly preferred. For shorter distances, plastic is much more practical.

All fibres consist of a number of substructures including:

A core, which carries most of the light, surrounded by

A cladding, which bends the light and confines it to the core, surrounded by
A substrate layer (in some fibres) of glass which does not carry light, but adds to the diameter
and strength of the fibre, covered by
A primary buffer coating, this provides the first layer of mechanical protection, covered
by A secondary buffer coating, this protects the relatively fragile primary coating.

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The cladding is also made of glass or plastic but has a lower index of refraction. This ensures
that the proper interface is achieved so that the light waves remain within the core. In addition
to protecting the fibre core from nicks and scratches, the cladding adds strength. Some fibre
optic cables have a glass core with a glass cladding. Others have a plastic core with a plastic
cladding. Another common arrangement is a glass core with a plastic cladding. It is called
plastic-clad silica (PCS) cable.
An important characteristic of fibre optics is refraction. Refraction is the characteristic of a
material to either pass or reflect light. When light passes through a medium, it "bends" as it
passes from one medium to the other. An example of this is when we look into a pond of
water.
In 1621, the Dutch mathematician Willebrard Snell established that rays of light can be traced
as they propagate from one medium to another based on their indices of refraction. Snell’s
low is stated by the equation:

Incident ray
Reflected ray
θ1 θ
Air

Water

θ2 Refracted ray

n1 sin θ2
= ;
n2 sin θ1
n1 sin θ1 = n2 sin θ2

where n1-refractive index of material 1; θ1-angle of incidence; θ2-angle of refraction; n2-

refractive index of material 2. When the angle of incidence, θ1, becomes large enough to
cause the sine of the refraction angle, θ2, to exceed the value of 1, total internal reflection
occurs. This angle is called the critical angle, θc. The critical angle, θc, can be derived from
Snell’s law as follows
n1 sin θ1 = n2 sin θ2

sin θ1 = n2 sin θ2/n1

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-1
When sin θ1 = sin θ2, then sin θ1 = n2 / n1. Therefore, the critical angle: θc = sin (n2 / n1)
Its index of refraction, however, it is typically 1% less than that of its core. This permits total
internal reflection of rays entering the fibre and striking the core-cladding interface above the
-1
critical angle of approximately 82-degree (sin (1/1.01). The core of the fibre therefore
guides the light and the cladding contains the light. The cladding material is much less
transparent than the glass making up the core of the fibre. This causes light rays to be
absorbed if they strike the core-cladding interface at an angle less than the critical angle.

If the angle of incidence is small, the light rays are reflected and do not pass into the water. If
the angle of incident is great, light passes through the media but is bent or refracted.

In the following figure, a light ray is transmitted into the core of an optical fibre. Total

Cladding

Core

Figure 2

internal reflection occurs as it strikes the lower index cladding material.

Optical fibres work on the principle that the core refracts the light and the cladding reflects
the light. The core refracts the light and guides the light along its path. The cladding reflects
any light back into the core and stops light from escaping through it.

Transmission Modes

There are three primary types of transmission modes using optical fibre. They are

a. Step Index
b. Graded Index
c. Single Mode

Step index has a large core, so the light rays tend to bounce around inside the core, reflecting
off the cladding. This causes some rays to take a longer or shorter path through the core.
Some take the direct path with hardly any reflections while others bounce back and forth

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taking a longer path. The result is that the light rays arrive at the receiver at different times.
The signal becomes longer than the original signal. LED light sources are used. Typical Core:
62.5 microns.

Graded index has a gradual change in the core's refractive index. This causes the light rays to
be gradually bent back into the core path. This is represented by a curved reflective path in
the attached drawing. The result is a better receive signal than with step index. LED light
sources are used. Typical Core: 62.5 microns.

Note: Both step index and graded index allow more than one light source to be used (different
colours simultaneously), so multiple channels of data can be run at the same time!

Single mode has separate distinct refractive indexes for the cladding and core. The light ray
passes through the core with relatively few reflections off the cladding. Single mode is used
for a single source of light (one colour) operation. It requires a laser and the core is very
small: 9 microns.

Basic Construction of Fibre-Optic Cables

There are two basic ways of classifying fibre optic cables. The first way is an indication of
how the index of refraction varies across the cross section of the cable. The second way of
classification is by mode. Mode refers to the various paths that the light rays can take in
passing through the fibre. Usually these two methods of classification are combined to define
the types of cable. There are two basic ways of defining the index of refraction variation
across a cable. These are step index and graded index. Step index refers to the fact that there is
a sharply defined step in the index of refraction where the fibre core and the cladding
interface. It means that the core has one constant index of refraction n1, while the cladding has

another constant index of refraction n2.

The other type of cable has a graded index. In this type of cable, the index of refraction of the
core is not constant. Instead, the index of refraction varies smoothly and continuously over
the diameter of the core. As you get closer to the centre of the core, the index of refraction
gradually increases, reaching a peak at the centre and then declining as the other outer edge of
the core is reached. The index of refraction of the cladding is constant.
Mode refers to the number of paths for the light rays in the cable. There are two
classifications: single mode and multimode. In single mode, light follows a single path
through the core. In multimode, the light takes many paths through the core.
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Each type of fibre optic cable is classified by one of these methods of rating the index or
mode. In practice, there are three commonly used types of fibre optic cable: multimode step
index, single mode step index and multimode graded index cables.

1. Multimode Step-Index Fibre.

This cable (see Figure 5 (a)) is the most common and widely used type. It is also the easiest to
make and, therefore, the least expensive. It is widely used for short to medium distances at
relatively low pulse frequencies.

Cross section Index profile Beam path dispersion

n1
n2
C B
A

cladding Input a)
core Input
Output
Light Output
source b)
Figure 5 c)

The main advantage of a multimode step index fibre is the large size. Typical core diameters
are in the 50-to-1000 micrometers (µm) range. Such large diameter cores are excellent at
gathering light and transmitting it efficiently. This means that an inexpensive light source
such as LED can be used to produce the light pulses. The light takes many hundreds of even
thousands of paths through the core before exiting. Because of the different lengths of these
paths, some of the light rays take longer to reach the end of the cable than others. The problem
with this is that it stretches the light pulses (Figure 5 (b). In Figure 5 ray A reaches the end
first, then B, and C. The result is a pulse at the other end of the cable that is lower in
amplitude due to the attenuation of the light in the cable and increased in duration due to the
different arrival times of the various light rays. The stretching of the pulse is referred to as
modal dispersion. Because the pulse has been stretched, input pulses can not occur at a rate
faster than the output pulse duration permits. Otherwise the pulses will essentially merge
together as shown in Figure 5 (c). At the output, one long pulse will occur and will be
indistinguishable from the three separate pulses originally transmitted. This means that
incorrect information will be received. The only core for this problem is to reduce the pulse

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repetition rate. When this is done, proper operation occurs. But with pulses at a lower
frequency, less information can be handled.

2. Single Mode Cable

In a single mode, or mono-mode, step-index fibre cable the core is so small that the total
number modes or paths through the core are minimised and modal dispersion is essentially
eliminated. The typical core sizes are 5 to 15 µm. The output pulse has essentially the same
duration as the input pulse (see Figure 6).

The single mode step index fibres are by far the best since the pulse repetition rate can be high
and the maximum amount of information can be carried. For very long distance transmission and
maximum information content, single-mode step-index fibre cables should be used.

The main problem with this type of cable is that because of its extremely small size, it is
difficult to make and is, therefore, very expensive. Handling, splicing, and making
interconnections are also more difficult. Finally, for proper operation an expensive, super
intense light source such as a laser must be used. For long distances, however, this is the type
of cable preferred.

Cross section Index profile Beam path

n1
n2

cladding
Input Output
core

Figure 6

3. Multimode graded-index fibre cables

These cables have a several modes or paths of transmission through the cable, but they are
much more orderly and predictable. Figure 7 shows the typical paths of the light beams.
Because of the continuously varying index of refraction across the core, the light rays are bent
smoothly and converge repeatedly at points along the cable.

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Cross section Index profile Beam path

n1

n2

cladding
Input Output
core

Figure 7

The light rays near the edge of the core take a longer path but travel faster since the index of
refraction is lower. All the modes or light paths tend to arrive at one point simultaneously.
The result is that there is less modal dispersion. It is not eliminated entirely, but the output
pulse is not nearly as stretched as in multimode step index cable. The output pulse is only
slightly elongated. As a result, this cable can be used at very high pulse rates and, therefore, a
considerable amount of information can be carried on it.

This type of cable is also much wider in diameter with core sizes in the 50 to 100 (µm) range.
Therefore, it is easier to splice and interconnect, cheaper, and less-intense light sources may
be used. The most popular fibre-optic cables that are used in LAN: multimode-step index
cable -65.5/125; multimode-graded index cable - 50/125. The multimode-graded index cable -
100/140 or 200/300 are recommended for industrial control applications because of its large
size. In high data rate systems single mode fibre 9/125 is used. Typical core and cladding
diameters of these cables are shown in Figure 8.

9
125

125

125

140
100
5
0

6
2
5
.

Figure 8
Specifications of the Fibre Cables

Indoor cable specifications:

• LED (Light Emitting Diode) light source

• 3.5 dB/Km Attenuation (loses 3.5 dB of signal per kilometer)

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• 850 nM - wavelength of light source

• Typically 62.5/125 (core diameter/cladding diameter)
• Multimode - can run many light sources.

Outdoor cable specifications:

• Laser Light Source

• 1 dB/Km Attenuation (loses 1 dB of signal per kilometer)
• 1170 nM - wavelength of light source
• Monomode (single mode)

Advantages of Optical Fibre:

• Noise immunity: RFI and EMI immune (RFI - Radio Frequency Interference, EMI -
Electromagnetic Interference)
• Security: cannot tap into cable.
• Large Capacity due to BW (bandwidth)
• No corrosion
• Longer distances than copper wire
• Smaller and lighter than copper wire
• Faster transmission rate

Disadvantages of optical fibre:

• Physical vibration will show up as signal noise!

• Limited physical arc of cable. Bend it too much and it will break!
• Difficult to splice

The cost of optical fibre is a trade-off between capacity and cost. At higher transmission
capacity, it is cheaper than copper. At lower transmission capacity, it is more expensive.

Media versus Bandwidth

The following table compares the usable bandwidth of the different guided transmission
media.

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Cable Type Bandwidth

Open Cable 0 - 5 MHz

Twisted Pair 0 - 100 MHz

Coaxial Cable 0 - 600 MHz

Optical Fibre 0 - 1 GHz

Transmission Media: Unguided

Unguided transmission media is data signals that flow through the air. They are not guided or
bound to a channel to follow. They are classified by the type of wave propagation.

RF Propagation

There are three types of RF (radio frequency) propagation:

• Ground Wave
• Sky Wave
• Line of Sight (LOS)

Ground wave propagation follows the curvature of the Earth. Ground waves have carrier
frequencies up to 2 MHz. AM radio is an example of ground wave propagation.

Sky wave propagation bounces off of the Earth's ionospheric layer in the upper atmosphere.
It is sometimes called double hop propagation. It operates in the frequency range of 30-85
MHz. Because it depends on the Earth's ionosphere, it changes with the weather and time of
day. The signal bounces off of the ionosphere and back to earth. Ham radios operate in this
range.
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Line of sight propagation transmits exactly in the line of sight. The receive station must be
in the view of the transmit station. It is sometimes called space waves or troposphere
propagation. It is limited by the curvature of the Earth for ground-based stations (100 km,
from horizon to horizon). Reflected waves can cause problems. Examples of line of sight
propagation are: FM radio, microwave and satellite.

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Radio Frequencies

19
The frequency spectrum operates from 0 Hz (DC) to gamma rays (10 Hz).

Name Frequency (Hertz) Examples

19
Gamma Rays 10

X-Rays 1017
15
Ultra-Violet Light 7.5 x 10
14
Visible Light 4.3 x 10
11
Infrared Light 3 x 10
9
EHF - Extremely High Frequencies 30 GHz (Giga = 10 ) Radar

SHF - Super High Frequencies 3 GHz Satellite & Microwaves

6
UHF - Ultra High Frequencies 300 MHz (Mega = 10 ) UHF TV (Ch. 14-83)

VHF - Very High Frequencies 30 MHz FM & TV (Ch2 - 13)

HF - High Frequencies 3 MHz2 Short Wave Radio

3
MF - Medium Frequencies 300 kHz (kilo = 10 ) AM Radio

LF – Low Frequencies 30 kHz Navigation

VLF - Very Low Frequencies 3 kHz Submarine Communications

VF - Voice Frequencies 300 Hz Audio

ELF - Extremely Low Frequencies 30 Hz Power Transmission

Radio frequencies are in the range of 300 kHz to 10 GHz. We are seeing an emerging
technology called wireless LANs. Some use radio frequencies to connect the workstations
together, some use infrared technology.

Microwave
Microwave transmission is line of sight transmission. The transmit station must be in visible
contact with the receive station. This sets a limit on the distance between stations depending
on the local geography. Typically the line of sight due to the Earth's curvature is only 100 km
to the horizon! Repeater stations must be placed so the data signal can hop, skip and jump
across the country.

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Microwaves operate at high operating frequencies of 3 to 10 GHz. This allows them to carry
large quantities of data due to their large bandwidth.

Advantages:

a. They require no right of way acquisition between towers.

b. They can carry high quantities of information due to their high operating frequencies.
c. Low cost land purchase: each tower occupies only a small area.
d. High frequency/short wavelength signals require small antennae.

Disadvantages:

a. Attenuation by solid objects: birds, rain, snow and fog.

b. Reflected from flat surfaces like water and metal.
c. Diffracted (split) around solid objects.
d. Refracted by atmosphere, thus causing beam to be projected away from receiver.

Satellite

Satellites are transponders (units that receive on one frequency and retransmit on another) that
are set in geostationary orbits directly over the equator. These geostationary orbits are 36,000
km from the Earth's surface. At this point, the gravitational pull of the Earth and the
centrifugal force of Earth's rotation are balanced and cancel each other out. Centrifugal force
is the rotational force placed on the satellite that wants to fling it out into space.

The uplink is the transmitter of data to the satellite. The downlink is the receiver of data.
Uplinks and downlinks are also called Earth stations because they are located on the Earth.

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The footprint is the "shadow" that the satellite can transmit to, the shadow being the area that
can receive the satellite's transmitted signal.

Iridium Telecom System

The Iridium Telecom System is a new satellite system that will be the largest private
aerospace project. It is a mobile telecom system intended to compete with cellular phones. It
relies on satellites in lower Earth orbit (LEO). The satellites will orbit at an altitude of 900 -
10,000 km in a polar, non-stationary orbit. Sixty-six satellites are planned. The user's handset
will require less power and will be cheaper than cellular phones. There will be 100%
coverage of the Earth.

Unfortunately, although the Iridium project was planned for 1996-1998, with 1.5 million
subscribers by end of the decade, it looked very financially unstable.

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ENCODING, MODULATING & TRANSMISSION CODES

ENCODING (D/D) (A/D) (D/A) (A/A)

Information must be encoded into signals before it can be transported across
communication media. We must encode data into signals to send them from one place
to another.
Digital-to-Digital Encoding
Digital-to-Digital Encoding is the representation of digital information by a digital
signal. (eg. computer-to-printer)

01011101 Digital/digital
encoding

Unipolar

Digital/ digital encoding Polar

Bipolar

Unipolar
Digital transmission systems work by sending voltage pulses along a media link,
usually a wire or a cable. In most types of encoding, one voltage level stands for
binary 0 and another level stands for binary 1. The polarity of a pulse refers to whether
it is positive or negative.
Unipolar encoding is so named because it uses only one polarity. Therefore, only one
of the two binary states is encoded, usually the 1. The other state, usually 0, is
represented by zero voltage, or an idle line.
Unipolar encoding uses only one level of value.

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Amplitude

0 1 0 0 1 1 1 0

Time

1’s encoded as positive, 0’s are idle. Unipolar encoding is straight forward and
inexpensive to implement. However, it has two problems that make it unusable: DC
component and synchronisation.
DC component
à
Average amplitude is nonzero creates a direct current (DC) component, when a
signal contains a DC component it cannot travel through media that cannot handle DC
components: e.g. microwaves or transformers.

Synchronisation
When a signal is unvarying, the receiver cannot determine the beginning and ending of
each bit. Therefore, Synchronisation problem in unipolar encoding can occur
whenever the data stream includes a long uninterrupted series of 1’s or 0’s.

Polar Encoding
Polar encoding uses two voltage levels: one positive and one negative. In most polar
encoding methods the average voltage level on the line is reduced and the DC
component problem of unipolar encoding is alleviated.

Polar

NRZ RZ Biphase

NRZ-L NRZ-I Manchester Differential

Manchester

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Non-Return-to-Zero (NRZ) Encoding

In NRZ encoding, the level of the signal is always either positive or negative. In NRZ
system if the line is idle it means no transmission is occurring at all.

• NRZ-L (Non-return-to-zero, Level)

In NRZ-L the level of the signal is dependant upon the state of the bit.
A positive voltage usually means the bit is 0, and negative voltage means the bit is a 1
(and vice versa).

• NRZ-I (Non-return-to-zero, Invert)

In NRZ-I, an inversion of the voltage level represents a 1 bit. It is the transition
between a positive and a negative voltage, not the voltages themselves that represents
a 1 bit. A 0 bit is represented by no change.
An advantage of NRZ-I over NRZ-L is that because the signal changes every time a 1
bit is encountered, it provides some synchronisation.
Each inversion allows the receiver to synchronise its timer to the actual arrival of the
transmission.

Amplitude

0 1 0 0 1 1 1 0

NRZ-L
Time

NRZ-I Time

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RZ (Return-to-zero) Encoding
To assure synchronisation, there must be a signal change for each bit. The receiver can
use these changes to built up, update, and synchronise its clock.
One solution is return to zero (RZ) encoding, which uses three Values: positive,
negative, and zero.

Amplitude

0 1 0 0 1 1 1 0

Time

The main disadvantage of RZ encoding is that it requires two signal changes to encode
one bit and therefore occupies more bandwidth. But of the three alternatives discussed
above, it is the most effective. Because a good encoded digital signal must contain a
provision for synchronisation.

Biphase Encoding
Probably the best existing solution to the problem of synchronisation is biphase
encoding. In this method, the signal changes at the middle of the bit interval but does
not return to zero. Instead it continues to the opposite pole. As in RZ, these mid-
interval transitions allow for synchronisation.
Biphase encoding is implemented in two different ways: Manchester and differential
Manchester.

• Manchester
Manchester encoding uses the inversion at the middle of each bit interval for both
synchronisation and bit representation. A negative-to-positive transition represents
binary 1 and a positive-to-negative transition represents binary 0.
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Amplitude 0 1

0 1 0 0 1 1 1 0

Time

• Differential Manchester
In this method, the inversion at the middle of the bit is used for synchronisation, but
the presence or absence of an additional transition at the beginning of the interval is
used to identify the bit. A transition means binary 0 and no transition means binary 1.
The bit representation is shown by the inversion and non-inversion at the beginning of
the bit.

Amplitude

0 1 0 0 1 1 1 0

Time

Bipolar Encoding

Bipolar encoding uses three voltage levels: positive, negative and zero. The zero level
is used to represent binary 0 positive and negative voltages represent alternating 1s. (If
st nd
1 one +ve, 2 is -ve).
* Three types of bipolar encoding are popular use by the data communications
industry: AMI, B8ZS, and HDB3.

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• Bipolar Alternate Mark Inversion (AMI)

Bipolar AMI is the simplest type of bipolar encoding. The word mark comes from
telegraphy and means 1.
AMI means alternate 1 inversion. A neutral, zero voltage represents binary 0. Binary
1s are represented by alternating positive and negative voltages

Amplitude

0 1 0 0 1 1 1 0

Time

By inverting on each occurrence of a 1, bipolar AMI accomplishes two things: first,

the DC component is zero, and second, a long sequence of 1s stays synchronised.
Two variations of bipolar AMI have been developed to solve the problem of
synchronisation sequential 0s. The first used in North America, is called bipolar 8-zero
substitution (B8ZS); the second, used in Europe and Japan, is called high-density
bipolar 3 (HDB3). Both are adaptations of bipolar AMI that modify the original
pattern only in the case of multiple consecutive 0s.

Bipolar 8-Zero Substitution (B8ZS)

B8ZS is the convention adopted in North America to provide synchronisation of long

strings of 0s. In most situations B8ZS functions identically to bipolar AMI. Bipolar
AMI changes poles with every 1 it encounters. These changes provide the
synchronisation needed by the receiver, but the signal does not change during a string
of 0s, so synchronisation is lost. The solution provided by B8ZS is to force artificial
signal changes, called violations

• In B8ZS, if eight 0s come one after another, we change the pattern in one of two ways
based on the polarity of previous 1.

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Polarity of Polarity of previous 1 bit

Previous 1 bit

+ 0 0 0 0 0 0 0 0 - 0 0 0 0 0 0 0 0

will change to

+ 0 0 0 + - 0 - + - 0 0 0 - + 0 + -

(Violation) (Violation) (Violation) (Violation)

High-Density Bipolar 3 (HDB3)

In HDB3 if four 0s come one after another, we change the pattern in one of four
ways based on the polarity of the previous 1 and the number of 1s since the last
substitution.

+ 0 0 0 0 - 0 0 0 0

th
Violation in the 4 consecutive zero

+ 0 0 0 + - 0 0 0 -

If the number of 1s since the last substitution is odd

+ 0 0 0 0 - 0 0 0 0

st th
Violation in the 1 & 4 consecutive zero

+ - 0 0 - - + 0 0 +

If the number of 1s since the last substitution is even

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Ex
Compare the bandwidth needed for unipolar encoding and RZ encoding. Assume the
worst-case scenario for both.
Solution
The worst case scenario (the situation requiring the most bandwidth) is alternating 1s
and 0s for unipolar, for RZ the worst-case is all 1s.

Unipolar encoding
Value

1 0 1 0 1 0 1 0

Time

Time

Value
RZ encoding

Time

Time

RZ needs twice the bandwidth of unipolar.

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Ex
Compare the bandwidth needed for Manchester and Differential Manchester
encoding. Assume the worst-case scenario for both.
Solution
The worst-case scenario for Manchester is consecutive 1s or consecutive 0s. There
are two transistors for each bit (one cycle per bit). For Differential Manchester the
worst – case is consecutive 0s with two transitions per each bit (one cycle per bit).
The bandwidths, which are proportional to bit rate, are the same for each.
Ex
Using B8ZS, encode the bit stream 10000000000100. Assume that the polarity of
the previous 1 is positive.
Amplitude

1 0 0 0 0 0 0 0 0 0 0 1 0 0

Time

Ex
Using HDB3, encode 10000000000100. Assume that the number of 1s so far is odd
and the previous 1 is positive.

Amplitude

1 0 0 0 0 0 0 0 0 0 0 1 0 0

Ti
me

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Analog-to- Digital Encoding

Analog/digital
encoding

A / D encoder
(Coder-decoder)

In analog-to-digital encoding, the information contained in a continuous wave form are

represented as a series of digital pulses (1s and 0s).

Pulse Amplitude Modulation (PAM)

The first step in A/D encoding is called pulse amplitude modulation (PAM). This technique
takes analog information, samples it, and generates a series of pulses based on the results of
sampling. The term sampling means measuring the amplitude of the signal at equal time
intervals.

Amplitude Amplitude

Time Time

Analog Signal PAM Signal

In PAM, the original signal is sampled at equal intervals.

PAM has some applications, but it is not used by itself in data communications. However, it is
the first step in another very popular encoding method called pulse code modulation (PCM).

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Pulse Code Modulation (PCM)

PCM modifies the pulses created by PAM to create a complete digital signal. To do so, PCM
first quantises the PAM pulses. Quantisation is a method of assigning integral values in a
specific range to sampled instances. (The result of quantisation is presented in the following
figure).
+125
+100
+75
+50
+25
000
-25
-50
-75
-100
-125
Each value is translated into its seven-bit binary equivalent. The eighth bit indicates the sign.
+24 00011000 -15 10001111 +125 01111101
+38 00100110 -80 11010000 +110 01101110
+48 00110000 -50 10110010 +90 01011010
+39 00100111 +52 00110110 +88 01011000
+26 00011010 +127 01111111 +77 01001101

The binary digits are then transformed into a digital signal using one of the digital encoding.
PCM

00011000 001 0011 0 001 100 0 0 …

Direction of transfer
The result of the PCM of the original signal encoded finally into a unipolar signal.
PCM is actually made up of four separate processes: PAM, quantisation, binary encoding, and
digital-to-digital encoding.
PCM is the sampling method used to digitize voice in T-line transmission in the North
America telecommunication system.
According to the Nyquist theorem, the sampling rate must be at least two times the highest
frequency.
Highest frequency = x Hz
Sampling rate = 2x samples/second
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Sampling interval

Quantisation
Quantising is the process of rounding-off the values of the flat-top samples to certain
predetermined levels.

u(t)

8
7
Quantising

6
5
4
3
2
1

T 2T 3T 4T 5T 6T t

Sampling

0111 0111 0100 0110 0110 0101 Binary Coding

PCM

Ex What sampling rate is needed for a signal with a bandwidth of 10,000 Hz (1000 Hz to
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11,000 Hz)? If the quantisation is eight bits per sample, what is the bit rate?

Solution
Sampling rate = 2 (11,000) = 22,000 samples/s each sample is quantised to eight bits: data
rate = (22,000 samples/s) (8 bits/sample) = 176 kbps

Digital-to-Analog Encoding

01011101 Digital/analog
encoding

Digital-to-analog encoding is the representation if digital information by an analog signal.

Digital /analog
encoding

ASK FSK PSK

QAM

Quadrature Amplitude Modulation

Bit Rate and Baud Rate

- Bit rate is the number of bits transmitted in one second.
- Baud rate refers to the number of signal units per second that are required to represent those
bits.
- For computer efficiency, bit rate is more important.
- For data transmission, baud rate is more important the fewer the signal units required,
the more efficient the system, and the less bandwidth required to transmit more bits.

Carrier signal
In analog transmission the sending device produces high - frequency signal that acts as a basis
for the information signal. The base signal is called the carrier signal or carrier frequency. The

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receiving device is tuned to the frequency of the carrier signal that it expects from the sender.
Digital information is then encoded onto the carrier signal by modifying one or more of its
characteristic (amplitude, frequency or phase). This kind of modification is called modulation
(or shift keying) and the information signal is called a modulating signal.

Amplitude Shift Keying (ASK)

In ASK the strength of the signal is varied to represent binary 1 or 0. Both frequency and
phase remain constant, while the amplitude changes.

Bit rate: 5 Baud rate: 5

Amplitude
1 bit 1 bit 1 bit 1 bit 1 bit

0 1 0 1 0

ASK Time

1 baud 1 baud 1 baud 1 baud 1 baud

1 second

0 1 0 1 1 0 0 1

b(t)

c(t) t

ASK t

Tb

1 second
Nbit = Nbaud = 8
Bit duration is the period of time that defines one bit. The peak amplitude of the signal, during
each bit duration, is constant and its value depends on the bit (0 or 1). The transmission speed
using ASK is limited by the physical characteristics of the transmission medium.

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Bandwidth for ASK

BW= (1 + d)*Nbaud

Where

BW is the bandwidth

Nbaud is the baud rate

d is a factor related to the condition of the line (with min. value of 0)

- The minimum bandwidth required for transmission is equal to the baud rate.

Amplitude
minimum bandwidth = Nbaud

Frequency
(fc-Nbaud/2) fc (fc+Nbaud/2)
Ex Find the bandwidth for an ASK signal transmitting at 2000 bps. Transmission is in half-
duplex mode.
Solution
In ASK baud rate = bit rate
Nbaud = 2,000
An ASK signal requires a bandwidth equal to its baud rate:
BW = 2,000 Hz.
Ex Given a bandwidth of 10,000 Hz (1,000 to 11,000 Hz), draw the full-duplex ASK diagram
of the system. Find the carriers and the bandwidth in each direction. Assume there is no gap
between the bands in two directions.

Solution
For full-duplex ASK the bandwidth for each direction is BW=10,000/2=5000 Hz.
The carrier frequencies can be chosen at the middle of each band
ƒc (forward) = 1,000 + 5,000/2 = 3,500 Hz
ƒc (backward) = 11,000-5,000/2 = 8,500 Hz

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Amplitude

ƒc (forward) ƒc (backward)

Frequency
1,000 3,500 6,000 8,500 11,000

Frequency Shift Keying (FSK)

In frequency shift keying (FSK), the frequency of the signal is varied to represent
binary 1 or 0. The frequency of the signal during each bit duration is constant and its
value depends on the bit (0 or 1): both peak amplitude and phase remain constant.
Bit rate: 5 Baud rate: 5
Amplitude
1 bit 1 bit 1 bit 1 bit 1 bit

0 1 0 1 0

FSK Time

1 baud 1 baud 1 baud 1 baud 1 baud

1 second

0 1 0 1 1 0 0 1

b(t)

c1(t) t

c2(t) t

FSK t

Tb
1 second
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Nbit = Nbaud = 8

FSK avoids most of the noise problems of ASK. The limiting factors of FSK are the physical
capabilities of the carrier.

Bandwidth for FSK

FSK spectrum can be considered as the combinations of two ASK spectra centred on ƒc0 and
ƒc1. The bandwidth required for FSK transmission is equal to the baud rate of the signal plus
the frequency shift (difference between the two carrier frequencies).
Amplitude
BW = Nbaud + (ƒc1 - ƒc0)

fc1 - fc0

Frequency
fc0 fc1

BW = Nbaud + (ƒc1 - ƒc0)

Ex Find the bandwidth for an FSK signal transmitting at 2,000 bps. Transmission is in half-
duplex mode and the carriers must be separated by 3,000 Hz.
Solution
BW = Nbaud + (ƒc1 - ƒc0)
= 2,000 + 3,000 = 5,000 Hz.

Ex Find the maximum bit rate for an FSK signal if the bandwidth of the medium is 12,000 Hz
and the distance between the two carriers must be at least 2,000 Hz. Transmission is in full-
duplex mode.

Solution
Because the transmission is in full-duplex, only 6,000 Hz is allocated for each direction, for
FSK, if ƒc1 and ƒc0 are the carrier frequencies,
BW = Nbaud + (ƒc1 - ƒc0)
Nbaud = BW – (ƒc1 - ƒc0)
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= 6,000 – 2,000 = 4,000

But because the baud rate is the same is bit rate, the bit rate is 4,000 bps.

Phase Shift Keying (PSK)

In the PSK, the phase is varied to represent binary 1 or 0. Both peak amplitude and frequency
remain constant as the phase changes. The phase of the signal during each bit duration, is
constant and its value depends on the bit (0 or 1).

Bit rate: 5 Baud rate: 5

Amplitude

1 bit 1 bit 1 bit 1 bit 1 bit

0 1 0 1 0

PSK Time

1 baud 1 baud 1 baud 1 baud 1 baud

1 second

0 1 0 1 1 0 0 1

b(t)

c(t) t

PSK t

DPSK t

1 second
Nbit = Nbaud = 8

DPSK eliminates the need for a coherent reference signal at the receiver by combing two
basic operations at the transmitter:

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- Differential encoding of the input data

- PSK DPSK
- To send symbol 1 we phase advance the current signal waveform by 180°.
- To send symbol 0 we leave the phase of the current signal waveform unchanged.

PSK Constellation

Bit Phase
o
0 0 1 0
1 180°
Constellation diagram

The above method is often called 2-PSK, or binary PSK, because two different phases (0° and
180°) are used in the encoding.
PSK is not susceptible (easily influenced) to the noise degradation that affects ASK, nor to the
bandwidth limitations of FSK. This means that smaller variations in the signal can be detected
reliably by the receiver. Therefore instead of utilising only two variations of a signal, each
representing one bit, we can use four variations and let each phase shift represent two bits.

Bit rate: 8 bps Baud rate: 4

Amplitude

2 bits 2 bits 2 bits 2 bits

00 01 10 11

Time
o o o o
0 90 180 270
1 baud

1 second

4-PSK (Quadrature-PSK)

This technique is called 4-PSK or Q-PSK. The pair of bits represented by each phase is called
a dibit.
Data can be transmitted two times as fast using 4-PSK as using 2-PSK.

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4–PSK characteristics 01

Dibit Phase
o
00 0
01 90 o 10 00
o
10 180
o
11 270

11
8–PSK Characteristics Constellation diagram
Tribit Phase 010
000 0o o
001 45
o
010 90
o
011 135 100 000
o
100 180
o
101 225
o
110 270o
111 315
110
Bit rate of 8-PSk is three as that of 2-PSK Constellation diagram

Bandwidth for PSK

The min. BW required for PSK transmission is the same as that required for ASK
transmission.
BW = Nbaud
Ex: Find the bandwidth for a 4-PSK signal transmitting at 2,000 bps. Transmission is in half-
duplex mode.
Solution
For 4-PSK the baud rate is half of the bit rate.
The baud rate is therefore 1,000. A PSK signal requires a bandwidth equal to its baud rate.
Therefore the bandwidth is 1,000 Hz.
Ex: Given a bandwidth of 5,000 Hz for an 8-PSK signal, what are the baud rate and bit rate?
Solution
For PSK the baud rate is the same as the bandwidth, which means the baud rate is 5,000. But
in 8-PSK, the bit rate is three times the baud rate. So the bit rate is 15,000 bps.

Quadrature Amplitude Modulation (QAM)

QAM means combining ASK and PSK in such a way that we have maximum contrast
between each bit, dibit, tribit, quadbit, and so on.

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Possible variation of QAM is numerous; theoretically any measurable number of

changes in amplitude can be combined with any measurable number of changes in
phase.
The figure below shows the constellation diagrams of 4-QAM and 8-QAM. In both
cases the number of amplitude shifts is less than the number of phase shifts. Because
amplitude changes are susceptible to noise and require greater shift differences than do
phase changes, the number of phase shifts used by a QAM system is always larger than
the number of amplitude shifts.

011

01 00 010

101 100 000 001

110
10 11
111

4-QAM 8-QAM
Constellation Diagram Constellation Diagram

Amplitude
Nbit = 24 bps, Nbaud = 8

3 bits 3 bits 3 bit 3 bits 3 bits 3 bits 3 bits 3 bits

101 100 001 000 010 011 110 111

1 baud

The output signal of the 8-QAM modem for data

b (t) = 101 100 001 000 010 011 110 111
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Three popular 16-QAM configurations are shown bellow:

4 amplitudes, 8 phase 3 amplitudes, 12 phases 2 amplitude, 8 phase

16-QAM 16-QAM 16-QAM

Since amplitude shift is more susceptible to noise, the greater the ratio of phase shifts to
amplitude, the greater the immunity to noise.
The second example, three amplitudes and 12 phases, handles noise best. The first
example, 4 amplitudes and 8 phases, is the OSI (Open Systems Interconnection)
recommendation. Several QAM designs link specific amplitudes with specific phases.
This means that even with noise problems associated with amplitude shifting, the meaning
of a shift can be recovered from phase information.

Bandwidth for QAM

The minimum bandwidth required for QAM transmission is the same as that required for
ASK and PSK transmission. QAM has the same advantages of PSK over ASK.

Bit/Baud Comparison
Assuming that an FSK signal over voice-grade phone lines can send 1200 bps, the bit rate
is 1200 bps. Each frequency shift represents a single bit; so it requires 1200 signal
elements to send 1200 bits. Its baud rate, therefore, is also 1200. Each signal variation in
8-QAM system, however, represents three bits. So a bit rate of 1200 bps, using 8-QAM,
has a baud rate of only 400.

As the figure below shows, a dibit system has a baud rate of one-half the bit rate, a tribit
system has a baud rate of one-third the bit rate, a quadbit system has a baud rate of one-
fourth the bit rate.
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Bit
Nbaud= N Nbit= N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0

Dibit
Nbaud= N Nbit= 2N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0

Tribit
Nbaud= N Nbit= 3N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0

Quadbit
Nbaud= N Nbit= 4N
0 0 1 0 1 0 0 0 1 0 1 0 1 1 1 0

Bit/ Baud Rate Comparison

Modulation Units Bits/Bauds Baud Rate Bit Rate
ASK, FSK, 2-PSK Bit 1 N N
4-PSK, 4-QAM Dibit 2 N 2N
8-PSK, 8-QAM Tribit 3 N 3N
16-QAM Quadbit 4 N 4N
32-QAM Pentabit 5 N 5N
64-QAM Hexabit 6 N 6N
128-QAM Septabit 7 N 7N
256-QAM Octabit 8 N 8N

Ex
A constellation diagram consists of eight equally spaced points on a circle. If the bit rate
is 4800 bps, what is the baud rate?
Solution
o 3
The constellation indicates 8-PSK with points 45 apart. Since 2 = 8, three bits are
transmitted with each signal element. Therefore, the baud rate is
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4800/3 = 1600 baud

Ex
Compute the baud rate for a 72,000 bps 64-QAM.
Solution
6
A 64-QAM signal means that there are six bits per signal elements since 2 = 64.
Thus, 72,000/6 = 12,000 baud

Ex
Compute the bit rate for a 1,000 baud 16-QAM signal.
Solution
4
A 16-QAM signal means that there are four bits per signal elements since 2 = 16.
Thus, (1,000)(4) = 4,000 bps

Analog-to-Analog-Encoding

Analogl/analog
encoding

Analog-to-analog encoding is the representation of analog information by an analog signal.

(eg. Radio communication).
Analog-to-analog modulation can be accomplished in three ways: amplitude modulation
(AM), frequency modulation (FM), and phase modulation (PM)

Analogl/analog
encoding

AM FM PM

Amplitude Modulation
In AM transmission, the carrier signal is modulated so that its amplitude varies with the
changing amplitudes of the modulating signal. The frequency and phase of the carrier remain
the same; only the amplitude changes to follow variations in the information. The modulating
signal becomes the envelope of the carrier.
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Amplitude
Modulating signal (audio)

Time

Carrier frequency
Amplitude

Time

AM signal
Amplitude

Time

AM Bandwidth

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The bandwidth of an AM signal is equal to twice the bandwidth of the modulating signal and
covers a range centred on the carrier frequency. The total bandwidth required for AM can be
determined from the bandwidth of the audio signal:
BWt = 2 x BWm
Amplitude

ƒc Frequency
BWm BWm

BWt = 2 x BWm
BWm = Bandwidth of the modulating signal
(audio) BWt = Total bandwidth (radio)
ƒc = Frequency of the carrier

The bandwidth of an audio signal (speech & music) is usually 5 kHz. Therefore, an AM radio
station needs a minimum bandwidth of 10 kHz. In fact, the Federal Communications
Commission (FCC) allows 10 kHz for each AM station.
AM stations are allowed carrier frequencies anywhere between 530 and 1700 kHz (1.7 MHz).
However, each station’s carrier frequency must be separated from those on either side by at
least 10 kHz (one AM bandwidth) to avoid interference.
fc fc fc fc fc
herestation

herestation

herestation
No

No

No

530 1700
kHz 10 kHz
kHz
Frequency Modulation (FM)
In FM transmission, the frequency of the carrier signal is modulated to follow the changing
voltage level (amplitude) of the modulating signal. The peak amplitude and phase of the
carrier signal remain constant, but as the amplitude of the information signal changes, the
frequency of the carrier changes correspondingly.

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Amplitude Modulating signal (audio)

Time

Carrier frequency
Amplitude

Time

Amplitude
FM signal

Time

FM Bandwidth
The bandwidth of an FM signal is equal to 10 times the bandwidth of the modulating signal
and, like AM bandwidth, covers a range centred on the carrier frequency. The total bandwidth
required for FM can be determined from the bandwidth of the audio signal:
BWt = 10 x BWm
Amplitude

ƒc Frequency
5BWm 5BWm

BWt = 10 x BWm

BWm = Bandwidth of the modulating signal

(audio) BWt = Total bandwidth (radio)
ƒc = Frequency of the carrier

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The bandwidth of an audio signal (speech & music) broadcast in stereo is almost 15 kHz.
Therefore, each FM radio station needs a minimum bandwidth of 150 kHz. The FCC allows
200 kHz (0.2 MHz) for each FM station to provide some room for guard bands.
FM stations are allowed carrier frequencies anywhere between 88 and 108 MHz. However,
stations must be separated from by at least 200 kHz to avoid overlapping.
fc fc fc fc fc

No station here
Nostationhere
No station here

108 MHz
88 MHz
200 kHz
Phase Modulation (PM)
Due to simpler hardware requirements, PM is used in some systems as an alternative to FM.
In PM transmission, the phase of the carrier signal is modulated to follow the changing
voltage level of the modulating signal. The peak amplitude and frequency of the carrier signal
remain constant, but as the amplitude of the information signal changes, the phase of carrier
changes correspondingly. The analysis and final result (modulating signal) are similar to those
of FM.

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TRANSMISSION
CODES

Transmission Codes
Binary-Coded Decimal (also called 8421 BCD)
In BCD, four bits are used to encode one decimal character.
Four bits give 16 binary combinations. Since there are 10
decimal characters, 0 through 9, only 10 of the 16 possible
combinations are necessary for encoding in BCD. The
remaining 6 combinations are said to be invalid.

Decimal BCD

0 0000
1 0001
2 0010 1010

3 0011 1011

4 0100 1100
1101 Not valid in BCD
5 0101
6 0110 1110

7 0111 1111

8 1000
9 1001

Ex
Convert 36710 to BCD
Solution
36710 = 0011 0110 0111
Ex
Convert 124910 to BCD
Solution
124910 = 0001 0010 0100
1001
Ex

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Convert 5810 to BCD

Solution
5810 = 0101 1000
BCD Addition:
Straight binary addition is performed as long as the result
does not exceed a decimal value
of 9.
Ex
Add the decimal numbers 3 and 4 in BCD
Solution
3 0011
+ 4 0100

7 0111
Ex
Add the decimal numbers 63 and 24 in BCD
Solution
63 0110 0011
+ 24 0010 0100

87 1000 0111

When the sum of two numbers exceeds 9, an invalid BCD

number is obtained. The invalid
number can be converted to a valid number by adding 0110
(6) to it.
Ex
Add the decimal numbers 9 and 6 in BCD
Solution
9 1001
+6 0110

15 1111…not valid in BCD

+ 0110…add 6 for correction
0001 0101…correct BCD number

Ex
Add the decimal numbers 46 and 79 in BCD
Solution
46 0100 0110
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+ 79 0111 1001

125 1011 1111

+ 0110 0110
0001 0010 0101
1 2 5

Excess-3 Code
Excess-3 code is very similar to 8421 BCD code. The only
difference is that 3 is added to the decimal before it is
encoded into a four-word.

Decimal BCD Excess-3

0 0000 0011
1 0001 0100
2 0010 0101 0000
3 0011 0110 0001
4 0100 0111 0010
5 0101 1000 1101 Not valid in Excess-3

6 0110 1001 1110

7 0111 1010 1111
8 1000 1011
9 1001 1100

Ex
Add the decimal numbers 9 and 7 in Excess-3
Solution
9 1100
+ 7 1010

16 1 0110

Gray Code
The disadvantage of the previous codes is that several bits
change state between adjacent counts. The Gray code is
unique in that successive counts result in only one bit
change. For example, 7 (0111) to 8 (1000) in binary, or

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BCD, all four bits change state. In Gray code, however, 7

(0100) to 8 (1100) require a single bit change.
The switching noise generated by the associated circuits
may be intolerable in some environments. The same change
with Gray code undergoes only a single bit change
consequently, less noise is generated. Shaft encoders used
for receiver tuning dials often use Gray code.
The Gray code is widely used for encoding the position of
the rotary shaft and for data transmission using PSK.

Decimal Binary Gray Code

0 0000 0000
1 0001 0001
2 0010 0011
3 0011 0010
4 0100 0110
5 0101 0111
6 0110 0101
7 0111 0100
8 1000 1100
9 1001 1101
10 1010 1111
Binary-to-Gray Conversion
- The given binary code is shifted to the right by one
bit.
- Discard the last bit (the LSB) from the obtained bits.
- Exclusive-ORing the given and obtained bits result
in the equivalent Gray code.
MSB LSB
x x x - - - - - x x

th st
16 position 1 position
Ex
Compute the Gray code for the binary number 11010
Solution
Binary code 1 1 0 10
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⊕ 1 1 01
Gray code 1 0 1 11
Ex
Compute the Gray code for the binary number 10001101
Solution
Binary code 1 0 0 0 1 1 01
⊕ 1 0 0 0 1 10
Gray code 1 1 0 0 1 0 11

Gray-to-Binary Conversion
- The first bit, the leftmost of the given Gray code,
becomes the MSB of the Binary code.
- Exclusive-ORing the second Gray code bit with the
MSB of the binary code yields the second binary bit.
- Exclusive-ORing the third Gray code bit with the
second binary code yields the third binary bit.
- Exclusive-ORing the fourth Gray code bit with the
third binary code yields the fourth binary bit. And so
on.

Ex
Compute the binary code for the Gray code 101101
Solution
Gray code 1 0 1 1 0 1

⊕ ⊕ ⊕ ⊕ ⊕

Binary code 1 1 0 1 1 0
Ex
Compute the binary code for the Gray code 11001011
Solution
Gray code 1 1 0 0 1 0 1 1

⊕ ⊕ ⊕ ⊕ ⊕ ⊕ ⊕

Binary code 1 0 0 0 1 1 0 1
Binary Numbers
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The binary numbering system provides the basis for all

computer operations. Computers work by manipulating
electrical current on and off. The binary system uses two
symbols, 0 and 1. Also called base 2.

Binary weights
Position Fifth Fourth Third Second First
4 3 2 1 0
Weight 2 (16) 2 (8) 2 (4) 2 (2) 2 (1)

Ex
1 1 0 1 digits
8 4 2 1 weights
___________________
8 4 0 1 results

13
Octal Numbers
The octal numbering system is used by computer
programmers to represent binary numbers in compact
form. Also called base 8.
Octal numbers use 8 symbols: 0,1,2,3,4,5,6,7.
Octal weights
Position Fifth Fourth Third Second First
4 3 2 1 0
Weight 8 (4096) 8 (512) 8 (64) 8 (8) 8 (1)

Ex
3 4 7 1 digits
512 64 8 1 weights
___________________
1,536 256 56 1 results

1,849
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Hexadecimal Numbers
Hexadecimal numbering system, like octal, is used by
computer programmers to represent binary numbers in
compact form. Also called base 16.
Hexadecimal uses 16 symbols:
0,1,2,3,4,5,6,7,8,9,A,B,C,D,E,F.
Hexadecimal weights
Position Fifth Fourth Third Second First
4 3 2 1 0
Weight 16 (65,536) 16 (4,096) 16 (256) 16 (16) 16 (1)

Ex
3 4 7 1 digits
4,096 256 16 1 weights
_____________________
12,288 1,024 112 1 results
+
13,425
Decimal Binary Octal Hexadecimal
0 0000 0 0
1 0001 1 1
2 0010 2 2
3 0011 3 3
4 0100 4 4
5 0101 5 5
6 0110 6 6
7 0111 7 7
8 1000 10 8
9 1001 11 9
10 1010 12 A
11 1011 13 B
12 1100 14 C
13 1101 15 D
14 1110 16 E
15 1111 17 F

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Transformations
- From Other Systems to Decimal
a) From Binary to Decimal
1 0 0 1 1 1 0 Binary
64 32 16 8 4 2 1 weights
64 0 0 8 4 2 0 weighted results

b) Hexadecimal to Decimal

4 E Hexadecimal
16 1 weights
64 14 weighted results

78
c) Octal to Decimal
1 1 6 Octal
64 8 1 weights
64 8 6 weighted results

78
- From Binary to Octal or Hexadecimal

Group by three Group by four

1 0 0 1 1 1 0 From right to from right to left 1 0 0 1 1 1 0
left

1 1 6 4 E
Octal Hexadecimal

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- From Octal or Hexadecimal to Binary

Octal Hexadecimal
1 1 6 4 E

1 0 0 1 1 1 0 1 0 0 1 11 0

Binary Binary

- From Decimal to Other Systems

a) From Decimal to Binary (divide by 2) Decimal

0 1 2 4 9 19 39 78

Remain
der
1 0 0 1 1 1 0 Binary
Code

b) From Decimal to Octal (divide by 8)

r
e
d
n
i
a
m
e
R
0 1 9 78

1 1 6 Binary
Code

c) From Decimal to Hexadecimal (divide by 16)

0 4 78
Remainde
r

4 E Binary
Code

Morse Code
Morse code is one of the oldest electrical transmission codes. The digital code
system is made up of a series of dots and dashes, representing the alphabet and
decimal numbers system. A dash is three times the duration of a dot.

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B
C

D
.
.
1
2
.
ASCII Code
The American Standard Code for Information Interchange (ASCII) is the most
widely used alphanumeric code for transmission and data processing.
ASCII is a seven-bit code that can be represented by two hexadecimal
characters for simplicity. The MS hexadecimal character in this case never
exceed 7

Ex
What is the ASCII code for the letter H (uppercase) in binary and hexadecimal?

Solution
Letter H is located in column 4 and row 8.
Binary: 100 1000
Hexadecimal: $ 48

Ex
What is the ASCII code for the letter k (lowercase) in binary and hexadecimal?

Solution
Letter k is located in column 6 and row B:
Binary: 110 1011
Hexadecimal: $ 6B

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