DIGITAL

COMMUNICATIONS
 Digital signals can be
MANIPULATED more easily
than analog signals. Digital signals can easily be
ENCRYPTED to ensure privacy
(They are easier to multiplex, for
instance.)

When an analog signal goes

through a chain of signal
processors, such as transmitters,
receivers, and amplifiers, noise
and distortion accumulate.
MUCH LESS SEVERE in digital
systems
DATA COMPRESSION can be
used with a digital signal to
reduce its bandwidth to less than
that required to transmit the
original analog signal.
Introduction to Information
Technology
 Communication systems

Digital

Analog

The block diagram on the top shows the blocks common to all communication
systems 5
We recall the components of a
communication system:
Input transducer: The device that converts a physical
signal from source to an electrical, mechanical or
electromagnetic signal more suitable for
communicating
Transmitter: The device that sends the transduced
signal
Transmission channel: The physical medium on
which the signal is carried
Receiver: The device that recovers the transmitted
signal from the channel
Output transducer: The device that converts the
received signal back into a useful quantity
6
Bandwidth
In this lecture, we will understand more deeply what signal
bandwidth is, what the meaning of channel bandwidth to a
communications engineer is, and what the limitations on
information rate are
Signal bandwidth:
We can divide signals into two categories: The pure tone signal (the
sinusoidal wave, consisting of one frequency component), and complex
signals that are composed of several components, or sinusoids of various
frequencies.

-3 -3
T=1x10 s f=1/1x10
=1000Hz=1 kHz

0 1 t (ms)

Pure signal
 The bandwidth of a signal composed of components of various frequencies (complex
signal) is the difference between its highest and lowest frequency components, and is
expressed in Hertz (Hz), the same as frequency
For example, a square wave may be constructed by adding sine waves of various
frequencies:

Pure tone

150 Hz sine wave

The resulting wave resembles
a square wave. If more sine waves
of other frequencies were added,
the resulting waveform would
Pure tone
more closely resemble a
450 Hz sine wave
square wave
Since the resulting wave
contains 2 frequency components
its bandwidth is around
450-150=300 Hz
Approaching a 150 Hz square
wave

(ms)
 Since voice signals are also
composed of several
components (pure tones) of
various frequencies, the Male voice
bandwidth of a voice signal is
taken to be the difference 3000 Hz frequency
between the highest and lowest component
frequencies which are 3000 Hz
and (close to) 0 Hz
Although other frequency
components above 3000 Hz
exist, (they are more prominent
in the male voice), an
acceptable degradation of
voice quality is achieved by
disregarding the higher
frequency components,
accepting the 3kHz bandwidth Female voice
as a standard for voice 3000 Hz frequency
communications component
 channel bandwidth:
The bandwidth of a channel (medium) is defined to be the range of
frequencies that the medium can support. Bandwidth is measured
in Hz
With each transmission medium, there is a frequency range of
electromagnetic waves that can be transmitted:
Twisted pair: 0 to 109 Hz (Bandwidth : 109 Hz)
Increasing Coax cable: 0 to 1010 Hz (Bandwidth : 1010 Hz)
bandwidth Optical fiber: 1014 to 1016 Hz (Bandwidth : 1016 -1014 = 9.9x1015 Hz)

Optical fibers have the highest bandwidth (they can support

electromagnetic waves with very high frequencies, such as light
waves)
The bandwidth of the channel dictates the information carrying
capacity of the channel
This is calculated using Shannons channel capacity formula
 SHANNONS LAW

Shannon's law is any statement defining the theoretical

maximum rate at which error free digits can be
transmitted over a bandwidth
limited channel in the presence of noise
Shannons Theorem
(Shannons Limit for Information Capacity)

Claude Shannon at Bell Labs figured out how much

information a channel could theoretically carry:
I = B log2 (1 + S/N) Note that the log
is base 2!

Where I is Information Capacity in bits per second

(bps)
B is the channel bandwidth in Hz
S/N is Signal-to-Noise ratio (SNR: unitlessdont
make into decibel:dB)
Signal-to-Noise Ratio

S/N is normally measured in dB, as a relationship

between the signal you want versus the noise that
you dont, but is in the medium
It can be thought of as a fractional relationship (that
is, before you take the logarithm):
1000W of signal power versus 20W of noise power
is either:
1000/20=50 (unitless!)
or: about 17 dB ==> 10 log10 1000/20 = 16.9897 dB
 Example #1

Consider an extremely noisy channel in which the value of the signal-to-noise ratio
is almost zero. In other words, the noise is so strong that the signal is faint. For
this channel the capacity C is calculated as

This means that the capacity of this channel is zero regardless of the bandwidth.
In other words, we cannot receive any data through this channel.
 Example #2

We can calculate the theoretical highest bit rate of a regular telephone line. A
telephone line normally has a bandwidth of 3000. The signal-to-noise ratio is
usually 3162. For this channel the capacity is calculated as

This means that the highest bit rate for a telephone line is 34.860 kbps. If we want
to send data faster than this, we can either increase the bandwidth of the line or
improve the signal-to-noise ratio.
 Example #3

The signal-to-noise ratio is often given in decibels. Assume that SNRdB = 36 and
the channel bandwidth is 2 MHz. The theoretical channel capacity can be
calculated as
For practical purposes, when the SNR is very high, we can assume that SNR + 1
is almost the same as SNR. In these cases, the theoretical channel capacity can
be simplified to

For example, we can calculate the theoretical capacity of the previous example as
 Seatwork #1:

1.We have a channel with a 1-MHz bandwidth. The SNR for this channel is 63.
What are the appropriate bit rate and signal level?

2. If it is required to transmit at 50 kbit/s, and a bandwidth of 1 MHz is used

3.If the SNR is 20 dB, and the bandwidth available is 4 kHz

Solution:

1.

2. then the minimum SNR required is given by 50 = 1000

log2(1+S/N) so S/N = 2C/W -1 = 0.035 corresponding to an
SNR of -14.5 dB. This shows that it is possible to transmit
using signals which are actually much weaker than the
background noise level.

3. C = 4 log2(1 + 100) = 4 log2 (101) = 26.63 kbit/s.

 DATA RATE LIMITS

A very important consideration in data communications is

how fast we can send data, in bits per second, over a
channel. Data rate depends on three factors:
1. The bandwidth available
2. The quality of the channel (the level of noise)
3. The level of the signals we use
Sampling

An analog signal varies continuously with time.

If we want to transmit such a signal digitally, that
is, as a series of numbers, we must first sample
the signal.
This involves finding its amplitude at discrete time
intervals. Only in this way can we arrive at a finite
series of numbers to transmit.
Sampling Rate
In 1928, Harry Nyquist showed mathematically
that it is possible to reconstruct a band-limited
analog signal from periodic samples, as long as
the sampling rate is at least twice the
frequency of the highest-frequency component
of the signal.
This assumes that an ideal low-pass filter prevents
higher frequencies from entering the sampler.
If the sampling rate is too low, a form of
distortion called aliasing
or foldover distortion is produced. In this
form of distortion, frequencies in the
sampled signal are translated downward.
Figure 3.1 shows what happens.
An aliased component will
appear as
a = s b
and low-pass filtering will not be
effective in removing it.
A digital communication system uses
sampling at 10 kilosamples per second
(kSa/s). The receiver filters out all
frequencies above 5 kHz. What
frequencies
appear at the receiver for each of the
following signal frequencies at the
input to the transmitter?
(a) 1 kHz
(b) 5 kHz
(c) 6 kHz
a.)
b = 1 kHz
s + b = 11 kHz
s b = 9 kHz
(b)
b = 5 kHz
s + b = 15 kHz
s b = 5 kHz
(c)
b = 6 kHz
s + b = 16 kHz
s b = 4 kHz
To simplify the mathematics,
we assumed that the pulse had an
amplitude of 1 V. These assumptions
yield a sample pulse whose shape follows
that of the original signal, as
shown in Figure 3.3(a). This technique is
called natural sampling.
Practical systems generally sample by
using a sample-and-hold circuit,
which maintains the signal level at the
start of the sample pulse. The results
of such a method are shown in Figure
3.3(b). This technique is known as
flat-topped sampling.
Natural and
Flat-Topped
Sampling
END