DDC REVIEWER

Lesson 1: Digital Transmission

INTRODUCTION
DIGITAL TRANSMISSION is the transmittal of digital signals between two or
more points in a communications system. The signals can be binary or any other form
of discrete-level digital pulses.
The original source information may be in digital form, or it could be analog
signals that have been converted to digital pulses prior to transmission and converted
back to analog signals in the receiver.
With digital transmission systems, a physical facility, such as a pair of wires,
coaxial cable or an optical fiber cable is required to interconnect the various points
within the system.
The pulses are contained in and propagate down the cable. Digital pulses cannot
be propagated through a wireless transmission systems, such as Earth’s atmosphere or
free space (vacuum).
AT&T developed the first digital transmission system for the purpose of carrying
digitally-encoded analog signals, such as the human voice, over metallic wire cables
between telephone offices.
Today, digital transmission systems are used to carry not only digitally-encoded
voice and video signals but also digital source information directly between computers
and computer networks. Digital transmission systems use both metallic and optical fiber
cables for their transmission medium.

ADVANTAGES OF DIGITAL TRANSMISSION

The primary advantage of digital transmission over analog transmission is NOISE
IMMUNITY.
Digital signals are also better suited than analog signals for processing and
combining using a technique called multiplexing.
In addition, digital transmission systems are more resistant to analog systems to
additive noise because they use signal regeneration rather than signal amplification.
Noise produced in electronic circuits is additive; therefore, the signal-to-noise ratio
deteriorates each time an analog signal is amplified.
Consequently, the number of circuits the signal must pass through limits the total
distance analog signals can be transported. However, digital regenerators sample noisy
signals and then reproduce an entirely new digital signal with the same signal-to-noise
ratio as the original transmitted signal.
Finally, digital signals are simpler to measure and evaluate than analog signals.
Therefore, it is easier to compare the error performance of one digital system to another
digital system. Also, with digital signals, transmission errors can be detected and
corrected more easily and more accurately than is possible with analog signals.

DISADVANTAGES OF DIGITAL TRANSMISSION

The transmission of digitally-encoded analog signals requires significantly more
bandwidth than simply transmitting the original analog signal. Bandwidth is one of the
most important aspects of any communications system because it is costly and limited.
Also, analog signals must be converted to digital pulses prior to transmission and
converted back to their original analog form at the receiver, thus, necessitating
additional encoding and decoding circuitry.
In addition, digital transmission requires precise time synchronization between
the clocks in the transmitters and receivers. Finally, digital transmission systems are
incompatible with older analog transmission systems.

PULSE MODULATION
Pulse Modulation consists essentially of sampling analog information signal and
then converting those samples into discrete pulses and transporting the pulses from a
source to a destination over a physical transmission medium.

The Four Predominant Methods of Pulse Modulation

1. Pulse Width Modulation (PWM)
2. Pulse Position Modulation (PPM)
3. Pulse Amplitude Modulation (PAM)
4. Pulse Code Modulation (PCM)

PWM is sometimes called pulse duration modulation (PDM) or pulse length

modulation (PLM), as the width (active portion of the duty cycle) of a constant amplitude
pulse is varied proportional to the amplitude of the analog signal at the time the signal is
sampled.
With PPM, the position of a constant-width pulse within a prescribed time slot is
varied according to the amplitude of the sample of the analog signal.
 With PAM, the amplitude of a constant-width, constant position pulse is varied
according to the amplitude of the sample of the analog signal.
With PCM, the analog signal is sampled and then converted into a serial n-bit binary
code for transmission.
PAM is used as an intermediate form of modulation with PSK, QAM and PCM,
although it is seldom used by itself. PWM and PPM are used in special-purpose
communications systems mainly for the military but are seldom used for commercial
digital transmission systems.
PCM is by far the most prevalent form of pulse modulation and, consequently, will be
discussed in more detail in the next topic.
PULSE CODE MODULATION
Alex H. Reeves is credited with inventing PCM in 1937 while working for AT&T at
its Paris laboratories.
In US today, PCM is the preferred method of communications within the public
switched telephone network because with PCM, it is easy to combine digitized voice
and digital data into a single, high-speed digital signal and propagate it over metallic or
optical fiber cables.
PCM is the only digitally-encoded modulation technique that is commonly used
for digital transmission. The term pulse code modulation is somewhat of a misnomer, as
it is not really a type of modulation but rather a form of digitally coding analog signals.
With PCM, the pulses are of fixed length and fixed amplitude. PCM is a binary
system where a pulse or lack of a pulse within a prescribed time slot represents either
logic 1 or a logic 0 condition. PWM, PPM and PAM are digital but seldom binary, as a
pulse does not represent a single binary digit (bit).
PCM SAMPLING
The function of a sampling circuit in a PCM transmitter is to periodically sample
the continually changing analog input voltage and convert those samples to a series of
constant-amplitude pulses that can be more easily converted to binary PCM code.
For the ADC to accurately convert a voltage to a binary code, the voltage must
be relatively constant so that the ADC can complete the conversion before the voltage
level changes. If not, the ADC would continually attempting to follow the changes and
may never stabilize on any PCM code.

Two Basic Techniques used to perform the Sampling Function

1. Natural Sampling
2. Flat-top Sampling

NATURAL SAMPLING
Natural sampling is when tops of the sample pulses retain their natural shape
during the sample interval, making it difficult for an ADC to convert the sample to a PCM
code. With natural sampling, the frequency spectrum of the sampled output is different
from that of an ideal sample.
FLAT-TOP SAMPLING
The most common method used for sampling voice signals in PCM systems is
the flat-top sampling, which is accomplished in a sample-and-hold circuit.
With flat-top sampling, the input voltage is sampled with a narrow pulse and then
held relatively constant until the next sample is taken.
The figure above shows flat-top sampling. As the figure shows, the sampling
process alters the frequency spectrum and introduces an error called aperture error,
which is when the amplitude of the sampled signal changes during the sample pulse
time.

SAMPLE AND HOLD CIRCUIT

 The FET acts as a simple analog switch. When turned on, Q1 provides a low-
impedance path to deposit the analog sample voltage across capacitor C1. The time
that Q1 is on is called aperture or acquisition time.
Essentially, C1 is the hold circuit. When Q1 is off, C1 does not have a complete
path to discharge through and, therefore, stores the sampled voltage. The storage time
of the capacitor is called the A/D conversion time because it is during this time that the
ADC coverts the sample voltage to a PCM code.
The acquisition time should be very short to ensure that a minimum change
occurs in the analog signal while it is being deposited across C1. If the input to the ADC
is changing while it is performing the conversion, aperture distortion results.

The figure above shows the input analog signal, the sampling pulse and the
waveform developed across C1. It is important that the output impedance of a voltage
follower Z1 and the on resistance of Q1 be as small as possible.
The rapid drop in the capacitor voltage immediately following each sample pulse
is due to the redistribution of the charge across C1. The inter-electrode capacitance
between the gate and the drain of the FET is placed in series with C1 when the FET is
off, thus acting as a capacitive voltage divider network.
Also, note the gradual discharge across the capacitor during the conversion time.
This is called droop and is caused by the capacitor discharging through its own leakage
resistance and the input impedance of voltage follower Z2.
Therefore it is important that the input impedance of Z2 and the leakage
resistance of C1 be as high as possible. Essentially, voltage followers Z1 and Z2 isolate
the sample-and-hold circuit (Q1 and C1) from the output and input circuitry.
SAMPLING RATE
The Nyquist Sampling Theorem establishes the minimum sampling rate, fs that
can be used for a given PCM system. For a sample to be reproduced accurately in a
PCM receiver, each cycle of the analog input signal (fa) must be sampled at least twice.
Consequently, the minimum sampling rate is equal to twice the highest audio
input frequency. If fs is less than two times fa, an impairment called alias or foldover
distortion occurs.

A sample and hold circuit is a non-linear device (mixer) with two inputs: the
sampling pulse and the analog input signal. Consequently, nonlinear mixing
(heterodyning) occurs between these two signals.

Because the sampling pulse is a repetitive waveform, it is made up of a series of

harmonically-related sine waves. Each of these sine waves is amplitude-modulated by
the analog signal and produces sum and difference frequencies symmetrical around
each of the harmonics of fs.
Each sum and difference frequency generated is separated from its respective
center frequency by fa. As long as fs is at least twice fa, none of the side frequencies
from one harmonic will spill into the sidebands of another harmonic, and aliasing does
not occur.

The side frequencies from one harmonic fold over into the sideband of another
harmonic. The frequency that folds over is an alias of the input signal(hence the names
“aliasing” and “fold over distortion”). If an alias side frequency from the first harmonic
fold over into the audio spectrum, it cannot be removed through filtering or any other
technique.

Example:
For a PCM system with a maximum audio input frequency of 4kHz, determine the
minimum sample rate and the alias frequency produced if a 5-kHz audio signal were
allowed to enter the sample-and-hold circuit.
 The input bandpass filter shown in the figure of a single channel, simplex PCM
system is called an antialiasing or antifoldover filter. Its upper frequency is chosen such
that no frequency greater than one-half the sampling rate is allowed to enter the
sample-and-hold circuit, thus, eliminating the possibility of foldover distortion occurring.
With PCM, the analog input signal is sampled, then converted to a serial binary
code. The binary code is transmitted to the receiver, where it is converted back to the
original analog signal. The binary codes used for PCM are sign-magnitude codes,
where the most significant bit (MSB) is the sign bit and the remaining bits are used for
magnitude.
Lesson 2: Introduction to Data Communications and Networking

INTRODUCTION

 Information is defined as the knowledge or intelligence.

 Data are generally defined as information that is stored in digital form.
 Information that has been processed, organized and stored is called data.
 In essence, data communications is the transmission, reception and processing
of digital information.
 The fundamental purpose of a data communications circuit is to transfer digital
information from one place to another.
 A network is a set of devices (sometimes called nodes or stations)
interconnected be media links.
 Data communications networks are systems of interrelated computers and
computer equipment and can be as simple as a personal computer connected to
a printer or two personal computers connected together through the public
telephone network.

HISTORY OF DATA COMMUNICATIONS

 If we limit the scope of data communications to methods that use binary-coded

electrical signals to transmit information, then the first successful (and practical)
data communications system was invented by Samuel F.B. Morse in 1832 and
called telegraph.
 Morse also developed the first practical data communications code which he
called Morse code.
 In 1849, the first low-speed telegraph printer was invented, but it was not until
1860 that high-speed (15bps) printers were available.
 In 1850, Western Union Telegraph Company was formed in Rochester, New
York, for the purpose of carrying coded messages from one person to another.
 In 1874, Emile Baudot invented a telegraph multiplexer, which allowed signals
from up to six different telegraph machines to be transmitted simultaneously over
a single wire.
 The telephone was invented in 1875 by Alexander Graham Bell.
 In1899, Guglielmo Marconi succeeded in sending radio (wireless) telegraph
messages. Telegraph was the only means of sending information across large
spans of water until;
 In 1920, the first commercial radio stations carrying voice information were
installed.
 Konrad Zuis, a German engineer, demonstrated a computing machine some time
in the late 1930s; however, at that time, Hitler was preoccupied trying to conquer
the rest of the world, so the project fizzled out.
 Bell Telephone Laboratories was given credit for developing the first special-
purpose computer in 1940 using electromechanical relays for performing logical
operations.
 J.Presper Eckert and John Mauchley of the University of Pennsylvania were
given credit for beginning modern-day computing when they developed the
ENIAC on February 14, 1946.
 In 1949, the US National Bureau of Standards developed the first all-electronic
diode-based computer capable of executing stored programs. The US Census
Bureau installed the machine, which is considered the first commercially
produced American computer.
 In 1950s, computers used punch cards for inputting information, printers for
outputting information and magnetic tape reels for permanently storing
information. These early computers could process only one job at a time using a
technique called batch processing.
 The first general purpose computer was an automatic sequence-controlled
calculator developed jointly by Harvard University and International Business
Machine (IBM) Corporation. The UNIVAC computer, built in 1951by Remington
Rand Corporation, was the first mass-produced electronic computer.
 In 1960s, batch-processing systems were replaced by on-line processing
systems with terminals connected directly to the computer through serial or
parallel communications lines.
 In 1970s, microprocessor-controlled microcomputers were the most dominant.
 By the 1980s, personal computers became an essential item in the home and
workplace. Since then, the number of mainframe computers, small business
computers, personal computers and computer terminals has increased
exponentially, creating a situation where more and more people have the need to
exchange digital information with each other.
 Recent developments in data communications networking, such as the Internet,
intranets and the World Wide Web (WWW), have created a virtual explosion in
the data communications industry.
 The Internet is a public data communications network used by millions of people
all over the world to exchange business and personal information. The Internet
began to evolve in 1969 at the Advanced Research Projects Agency (ARPA).
ARPANET was formed in the late 1970s to connect sites around the United
States.
 From the mid-1980s to April 30, 1995, the National Science Foundation (NSF)
funded a high-speed backbone called NSFNET.
  Intranets are private data communications networks used by many companies to
exchange information among employees and resources. Intranets normally are
used for security reasons or to satisfy specific connectivity requirements.
 The World Wide Web (WWW) is a sever-based application that allows
subscribers to access the services offered by the Web. Browsers, such as
Netscape Communicator and Microsoft Internet Explorer are commonly-used for
accessing data over the WWW.

DATA COMMUNICATIONS NETWORK ARCHITECTURE

 A data communications network is any system of computers, computer terminals

or computer peripheral equipment used to transmit and/or receive information
between two or more locations.
 Network Architectures outline the products and services necessary for the
individual components within a data communications network to operate
together.
 The primary goal of network architecture is to give the users of the network tools
necessary for setting up the network and performing data flow control. It outlines
the way in which a data communications network is arranged or structured and
generally includes the concept of levels or layers or functional responsibility
within the architecture.
 Networks and network protocols fall into three general classifications: current,
legacy and legendary.
 Current Networks include the most modern and sophisticated networks and
protocols available.
 If a network or protocol becomes a legacy, no one really wants to use it, but for
some reason it just will not go away.
 When an antiquated network or protocol finally disappears, it becomes
legendary.
 In general terms, computer networks can be classified in two different ways:
broadcast and point to point.
 With broadcast networks, all stations and devices on the network share a single
communications channel. Data are propagated through the network in relatively
short messages sometimes called frames, blocks or packets.
 Many or all subscribers of the network receive transmitted messages, and each
message contains an address that identifies specifically which subscriber is
intended to receive the message. When messages are intended for all
subscribers on the network, it is called broadcasting, and when the messages
are intended for a specific group of subscribers, it is called multicasting.
 Point-to-point networks have only two stations. Therefore, no addresses are
needed. All transmissions from one station are intended for and received by the
other station. With point-to-point networks, data are often transmitted in long,
continuous messages, sometimes requiring several hours to send.
 In more specific terms, point-to-point and broadcast networks can be subdivided
into many categories in which one type of network is often included as a
subnetwork of another.