CECS 474 Computer Network Interoperability

CHAPTER 6
Information Sources &
Signals
Tracy Bradley Maples, Ph.D.
Computer Engineering & Computer Science
California State University, Long Beach

Notes for Douglas E. Comer, Computer Networks and Internets (5th Edition)
 Analog and Digital Signals
 
Data communications deals with two types of information:
• analog
• digital
 
Defn: An analog signal is characterized by a continuous mathematical function
 
When the input changes from one value to the next, it does so by moving
through all possible intermediate values.
 
Defn: A digital signal has a fixed set of valid levels
 
When the input changes from one value to the next, each change consists of
an instantaneous move from one valid level to another.
 Periodic and Aperiodic Signals
 
Signals are broadly classified as
• periodic
• aperiodic (sometimes called nonperiodic)
 
The classification depends on whether the signals repeat or do not repeat.

Sine Waves and Signal Characteristics

 

Data communications often involves the use of sinusoidal trigonometric functions:

especially sine (abbreviated sin)
 
We are interested in sine waves that correspond to a signal that oscillates in time:

Sine waves are important because:

• natural phenomena produce
sine waves
• when a microphone picks up
an audible tone, the output is
a sine wave
• electromagnetic radiation can
be represented as a sine waves
 Sine Waves and Signal Characteristics
 

Important characteristics of signals that relate to sine waves:

 Frequency: The number of oscillations per unit time (usually seconds)
 Peak amplitude: The maximum absolute value of the signal height
 Peak-to-Peak amplitude: The difference between the maximum and minimum
signal heights
 Phase: The distance from the start of the sine wave is shifted from a reference time
 Wavelength: The length of a cycle as it propagates across a medium (determined by
the speed with which a signal propagates)
 Sine Waves and Signal Characteristics
 
The frequency can be calculated as the inverse of the time required for one cycle (also
called the period of the wave).
 
Example: Using the sine wave in Figure (a):
 
• a period T = 1 seconds
• a frequency of 1 / T = 1 / 1 = 1 Hertz
 
Example: Using the sine wave in Figure (b):
 
• a period of T = 0.5 seconds
• its frequency is 1 / 0.5 = 2 Hertz
•  
Notes:
• Both examples are considered extremely low frequencies
• Typical systems use high frequencies
(often millions /second)
• Engineers often express:
time in fractions of a second &
frequency megahertz (MHz)
 Composite Signals

Signals like the ones illustrated in this figure

are classified as simple:

…because they consist of a single sine wave

that cannot be decomposed further.

Most signals are classified as composite:

 
…because the signal can be…
• decomposed into a set
of simple sine waves
• formed by adding two
simple sine waves
 The Importance of Composite Signals and Sine Functions

Q: Why does data communications (not Computer Networking) seem obsessed with sine
functions and composite signals?
A1: Signals that result from modulation are usually composite signals
A2: The mathematician Fourier discovered that it is possible to decompose a composite
signal into its constituent parts into a set of sine functions, each with a frequency,
amplitude, and phase.
 
Time and Frequency Domain Representations

Two key methods are used to represent composite signals:

 
1) Time Domain Representation -- A graph of a signal as a function of time

2) Frequency Domain Representation -- A graph of a signal made up of every simple

frequency included.

• The y-axis gives the amplitude, and the x-axis gives the frequency

 
 Example: The function asin(2πt) is represented by a single line of height a that is
positioned at x = t

Time Domain Representation

Frequency Domain Representation

Advantages of the Frequency Domain Representation:

• It can be used with nonperiodic signals
• It is compact
 Bandwidth of an Analog Signal
 
Defn: Analog bandwidth is the difference between the highest and lowest frequencies of
the constituent parts of the signal (i.e., the highest and the lowest frequencies obtained
by Fourier analysis)

Figure 6.7 shows a Frequency Domain plot with

frequencies measured in Kilohertz (KHz)
 
• Such frequencies are in the range audible to a human ear.
• In the figure, the bandwidth is the difference between the highest and lowest
frequency (5 KHz - 1 KHz = 4 KHz)
• In analog telephone systems, the usable frequency range is 300 Hz to 3400 Hz range
 Digital Signals and Signal Levels
 
Some systems use voltage to represent digital values:
• Make a positive voltage correspond to a logical one
• zero voltage correspond to a logical zero
 

For example, +5 volts can be used for a logical one and 0 volts for a logical zero
 

If only two levels of voltage are used, each level corresponds to one data bit (0 or 1)

Some physical transmission mechanisms can support more than two signal levels,
 
When multiple digital levels are available each level can represent multiple bits.

For Example:
consider a system
that uses four
voltage levels:
-5 volts, -2 volts,
+2 volts, +5 volts

Each level can

correspond to
two bits of data as
in Figure 6.8 (b).
 Digital Signals and Signal Levels
 
The relationship between the number of levels required and the number of bits to be
sent is straightforward:
 
• There must be a signal level for each possible combination of bits
• There are 2n combinations possible with n bits
 
NOTE: Practical electronic systems cannot distinguish between signals that differ by
arbitrarily small amounts. Thus, practical systems are restricted to a few signal levels.
 
 
 Baud and Bits Per Second

Q: How much data can be sent in a given time?

A: The answer depends on two aspects of the communication system.

• the number of signal levels
• the amount of time the system remains at a given level before moving to the next
(the number of signal changes per second)
 
 Baud and Bits Per Second
 
Defn: The baud rate is how many times the signal can change per second.
 
For example: If a system requires the signal to remain at a given level for 0.001
seconds, we say that the system operates at 1000 baud

Thus, both the baud rate and the number of signal levels control the bit rate of a system.
 Synchronization and Agreement About Signals
 
The electronics at both ends of a physical medium must have circuitry to measure time
precisely and synchronize with the data as it arrives.
 
Example: If one end transmits a signal with 10 elements per second, the other end must
expect exactly 10 elements per second.
 
Building electronic systems that agree at the high speeds used in modern networks is
extremely difficult.
 
A fundamental issue that arises from the way data is represented concerns synchronization
of sender/receiver.
 
Line Coding

Several techniques are used to help with synchronization errors.

In general, there are two broad approaches:
• Before transmitting, the sender transmits a known pattern of bits (typically a set of
alternating 0s and 1s) for receiver to use in synchronization
• Data is represented in such a way that synchronization is included in the encoding
 
The term line coding is used to describe the way data is encoded in a signal.
 Manchester Encoding Used in Computer Networks
 
Manchester Encoding is used with Ethernet
 
Why? Because detecting a transition in signal level is easier
than measuring the signal level
 
So Manchester Encoding uses transitions rather than levels to define bits
 
In Manchester Encoding,
• a 1 corresponds to a transition from negative voltage level to a positive voltage level
• a 0 corresponds to a transition from a positive voltage level to a negative level
 
The transitions occur in the middle of the time slot of a bit.
 Converting an Analog Signal to Digital
 
Many sources of information are analog:
 which means they must be converted to a digital form for further processing (e.g.,
before they can be encrypted)
 
There are two basic approaches:
1. pulse code modulation (PCM)
2. delta modulation (DM)
 
In PCM, the level of an analog signal is measured repeatedly at fixed time intervals and
converted to digital form.
In DM, the changes in the levels of an analog signal are measured repeatedly at fixed
time intervals and converted to digital form. (Not discussed further here.)
 Converting an Analog Signal to Digital

The three steps used in PCM for telephony:

1. sampling -- each measurement is known as a sample
2. quantization -- a sample is quantized by converting it into a small integer
value, called a quanta. The range of the signal from the minimum to maximum
levels is divided into a set of quanta. Typically, a power of 2 values.
3. encoding -- the quantized sample is encoded into a specific format

Converting an Analog Signal to Digital Using PCM for Voice

 The Nyquist Theorem and Sampling Rate

An analog signal must be sampled in PCM (or DM).

How often should an analog signal be sampled?

 Taking too few samples (undersampling) means that the digital values only give a
crude approximation of the original signal
 Taking too many samples (oversampling) means that more digital data will be
generated, which uses extra bandwidth

The mathematician Nyquist determined the answer to the question of how much
sampling is required:
sampling rate = 2* fmax

where fmax is the highest frequency in the composite signal.

Nyquist’s Theorem provides a practical solution to the problem:

Sample a signal at least twice as fast as the highest frequency that must be preserved
 Nyquist Theorem and Telephone System Transmission

The telephone system was designed to transfer voice.

Measurements of human speech have shown that preserving frequencies between 0 -

4000 Hz provides acceptable audio quality.

Thus, when converting a voice signal from analog to digital the signal should be
sampled at a rate of 8000 samples per second.

The PCM standard used by the phone system quantifies each sample into:

• An 8 bit value for quality

 the range of input is divided into 256 possible levels so that each sample has a
value between 0 and 255

• The rate generated for a single telephone call is:

 Encoding and Data Compression

Data compression refers to a technique that reduces the number of bits required to
represent data.

Data compression is relevant to communication systems because reducing the number of

bits used to represent data reduces the time required for transmission.

A communication system can be optimized by compressing data

There are two types of compression:

• Lossy - some information is lost during compression
• Lossless - all information is retained in the compressed version
• Lossy compression is generally used with data that a human consumes, such as an
image, video/audio
 Encoding and Data Compression

The key idea is that the compression only needs to preserve details to the level of human
perception:

a change is acceptable if humans cannot detect the change

Example: JPEG (used for images) compression or MPEG-3 (abbreviated MP3 and used for audio
recordings) employ lossy compression

Lossless compression preserves the original data without any change

 lossless compression can be used for documents or in any situation where data must be
preserved exactly
 arbitrary data can be compressed by a sender and decompressed by a receiver to recover an exact
copy of the original
(Chapter 29 discusses compression in more detail)