Digital Communications

Computer communications
engineering
Third Class
2015-2016
Assistance Professor,
Dr. Mahmood Farhan
Reference: Digital Communications
Fundamentals and Applications, 2nd
Addition, by Fernard Sklar
 Chapter One

Introduction:

Digital communication systems are becoming increasingly attractive because of the

ever-growing demand for data communication and because digital transmission
offers data processing options and flexibilities not available with analog
transmission. The principal feature of a digital communication system (DCS) is that
during a finite interval of time, it sends a waveform from a finite set of possible
waveforms, in contrast to an analog communication system, which sends a
waveform from an infinite variety of waveform shapes with theoretically infinite
resolution. In a DCS, the objective at the receiver is not to reproduce a transmitted
waveform with precision; instead, the objective is to determine from a noise-
perturbed signal which waveform from the finite set of waveforms was sent by the
transmitter. An important measure of system performance in a DCS is the
probability of error (PE).

Advantages of Digital Communication:

There are many reasons. The primary advantage is the ease with which digital
signals, compared with analog signals, are regenerated. Figure 1 illustrates an ideal
binary digital pulse propagating along a transmission line.

During the time that the transmitted pulse can still be reliably identified (before it is
degraded to an ambiguous state), the pulse is amplified by a digital amplifier that

DR. MAHMOOD 2
recovers its original ideal shape. The pulse is thus “reborn” or regenerated. Circuits
that perform this function at regular intervals along a transmission system are called
regenerative repeaters.

Digital circuits are less subject to distortion and interference than are analog
circuits. Because binary digital circuits operate in one of two states—fully on or
fully off—to be meaningful, a disturbance must be large enough to change the
circuit operating point from one state to the other.

With digital techniques, extremely low error rates producing high signal fidelity are
possible through error detection and correction but similar procedures are not
available with analog.

Security is another priority of messaging services in modern days. Digital

communication provides better security to messages than the analog
communication. It can be achieved through various coding techniques available in
digital communication.

Digital circuits are more reliable and can be produced at a lower cost than analog
circuits. Also, digital hardware lends itself to more flexible implementation than
analog hardware.

Disadvantages of Digital Communications:

Digital communications require greater bandwidth than analogue to transmit the

same information.

The detection of digital signals requires the communications system to be

synchronised, whereas generally speaking this is not the case with analogue
systems.

The noise of sampling error.

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When the signal-to-noise ratio drops below a certain threshold, the quality of
service can change suddenly from very good to very poor. In contrast, most analog
communication systems degrade more gracefully.

Typical Block Diagram and Transformations

The functional block diagram is shown in Fig.2. The upper blocks—format, source
encode, encrypt, channel encode, multiplex, pulse modulate, bandpass modulate,
frequency spread, and multiple access—denote signal transformations from the
source to the transmitter (XMT). The lower blocks denote signal transformations
from the receiver (RCV) to the sink, essentially reversing the signal processing
steps performed by the upper blocks. The modulate and demodulate/detect blocks
together are called a modem.

Figure 2 Block diagram of a typical digital communication system.

Information source. This is the device producing information to be communicated

by means of the DCS. Information sources can be analog or discrete. Analog
information sources can be transformed into digital sources through the use of

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sampling and quantization. Sampling and quantization techniques called formatting
and source coding.

Binary digit (bit).This is the fundamental information unit for all digital systems.
The term bit also is used as a unit of information content.

Bit stream. This is a sequence of binary digits (ones and zeros). A bit stream is
often termed a baseband signal, which implies that its spectral content extends from
(or near) dc up to some finite value, usually less than a few megahertz.

Symbol (digital message). A symbol is a group of k bits considered as a unit. We

refer to this unit as a message symbol mi (i=1, . . . , M) from a finite symbol set or
alphabet. The size of the alphabet, M ,is M =2k , where k is the number of bits in the
symbol. For baseband transmission, each mi symbol will be represented by one of a
set of baseband pulse waveforms g1 ( t ) , g 2 ( t ) ,… … g M (t). When transmitting a sequence
of such pulses, the unit baud is sometimes used to express pulse rate (symbol rate).
For typical bandpass transmission, each gi (t) pulse will then be represented by one
of a set of bandpass waveform s ( t ) , s 2 ( t ) , … … s M (t ).

Digital waveform. This is a voltage or current waveform (a pulse for baseband

transmission, or a sinusoid for bandpass transmission) that represents a digital
symbol.

K 1
Data rate. This quantity in bits per second (bits/s) is given by R= T = T log 2 M ()
bits/s, where k bits identify a symbol from an M =2k -symbol alphabet, and T is the
k-bit symbol duration.

Classification of Signals:

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 1- Deterministic and Random Signals: Deterministic signals or waveforms
are modeled by explicit mathematical expressions, such as x(t) =5 cos 10t.
For a random waveform it is not possible to write such an explicit
expression. However, when examined over a long period, a random
waveform, also referred to as a random process, may exhibit certain
regularities that can be described in terms of probabilities and statistical
averages.
2- Periodic and Nonperiodic Signals: A signal x(t) is called periodic in time if
there exists a constant T0 >0 such that
x ( t )=x ( t+T 0 ) for−∞< t<∞

where t denotes time. The smallest value of T0 that satisfies this condition is
called the period of x(t). The period T0 defines the duration of one complete
cycle of x(t). A signal for which there is no value of T 0 that satisfies above
Equation is called a nonperiodic signal.
3- Analog and Discrete Signals: An analog signal x(t) is a continuous function
of time; that is, x(t) is uniquely defined for all t. An electrical analog signal
arises when a physical waveform (e.g., speech) is converted into an electrical
signal by means of a transducer. By comparison, a discrete signal x(kT) is
one that exists only at discrete times; it is characterized by a sequence of
numbers defined for each time, kT, where k is an integer and T is a fixed
time interval.
4- Energy and Power Signals: An electrical signal can be represented as a
voltage v(t) or a current i(t) with instantaneous power p(t) across a resistor R
defined by
v 2 (t)
P (t)=
R
or P ( t ) =i2 (t )R

In communication systems, power is often normalized by assuming R to be

1Ω, although R may be another value in the actual circuit. Therefore,

DR. MAHMOOD 6
 regardless of whether the signal is a voltage or current waveform, the
normalization convention allows us to express the instantaneous power as
P ( t ) =x2 (t )
The energy dissipated during the time interval (−T/2, T/2)
T
2

ETx = ∫ x 2( t) dt
−T
2

and the average power dissipated by the signal during the interval is
T
2
1 T 1
PTx = E = ∫ x 2 (t)dt
T x T −T
2

5- The Unit Impulse Function: A useful function in communication theory is

the unit impulse or Dirac delta function δ (t). The impulse function is an
abstraction—an infinitely large amplitude pulse, with zero pulse width, and
unity weight (area under the pulse), concentrated at the point where its
argument is zero. The unit impulse is characterized by the following
relationships:
∞

∫ δ2 (t) dt=1
−∞

δ ( t )=0 for t ≠ 0
δ ( t ) is unbounded at t=0
∞

∫ x (t) δ ( t−t0 ) dt =x (t 0)
−∞

Digital Coding:
DR. MAHMOOD 7
 If the data consist of alphanumeric text, they will be character encoded with
one of several standard formats; examples include the American Standard
Code for Information Interchange (ASCII).

The textual message is word “THINK” using 6-bit ASCII character coding yeldes a
bit stream comprising 30 bits. The symbol set size, M, has been chosen to be 8
(each symbol represents an 8- ary digit). The bits are therfore partitioned into
groups of three (k =log 2 8). The transmitter must have a repertoire of eight
waveforms Si ( t ) , where i=1, … .. , 8. to represent the possible symbols, any one of
which may be transmitted during a symbol time.

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 Fourier Transform: It is a technique used to transform nonperiodic and
periodic signal from time domain to frequency domain and vise versa.
Foureir Transform:
∞
X ( w )= ∫ x ( t ) e− jωt dt
−∞

Or
∞
X ( f )= ∫ x ( t ) e− j 2 πft dt sinc w=2 πf
−∞

Invers Fourier Transform:

DR. MAHMOOD 9
 ∞
1
x (t)= ∫ X (w) e jωt dw
2 π −∞
∞
x (t)= ∫ X (f )e j 2 πt df
−∞

Example: Obtain the Fourier transform of rectangular pulse of duration T and

amplitude A shown below:

The rectangular pulse represented by:

T T
rect
t
T
=
{
A for− <t <
2
0 elsewhere
2

x ( t )= A rect ( Tt )
FT for x (t )
∞
X ( f )= ∫ x ( t ) e− j 2 πft dt
−∞

T
2 T
− j 2 πft A
¿∫ Ae dt= [ e− j2 πft ]−T
2

−T − j 2 πf 2
2

− jπfT jπfT
A
¿ [ e− jπfT −e jπfT ]= A e −e [ ]
− j 2 πf πf 2j

A sin ( πfT )
¿ sin ( πfT )= AT
πf πfT

∴ X (f )=ATsinc (πfT )

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DR. MAHMOOD 11
Sampling theorem:

Sampling of the signals is the fundamental operation in digital communication. A

continuous time signal is first converted to discrete time signal by sampling
process. Also it should be possible to recover or reconstruct the signal completely
from its samples.

The sampling theorem state that:

1- A band limited signal of finite energy, which has no frequency components

higher than W Hz, is completely described by specifying the values of the
signal at instant of time separated by 1/2W second and
2- A band limited signal of finite energy, which has no frequency components
higher than W Hz, may be completely recovered from the knowledge of its
samples taken at the rate of 2W samples per second.

Proof of sampling theorem:

Let x(t) the continuous time signal shown in figure below, its band width does
not contain any frequency components higher than W Hz. A sampling function
samples this signal regularly at the rate of fS sample per second.

Assume an analog waveform, x (t ) with a Fourier transform, X ( f ) , which is zero

outside the interval (−f m <f < f m). The sampling of x (t ) can viewed as the product
of x (t ) with periodic train of unit impulse function x δ (t) defined as
∞
x δ ( t ) = ∑ δ (t−n T s )
n=−∞

The sifting property of unit impulse state that

x ( t ) δ ( t−t 0 )=x ( t 0 ) δ (t−t 0 )

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 Using this property so that:
∞
x s ( t ) =x ( t ) x δ ( t )= ∑ x ( t ) δ ( t−nT s )
n=−∞

∞
¿ ∑ x ( n T s ) δ (t−n T s)
n=−∞

Notice that the Fourier transform of an impulse train is another impulse train.

∞
1
X δ ( f )= ∑ δ (f −n f s)
T s n=−∞

Convolution with an impulse function simply shifts the original function:

X ( f )∗δ ( f −n f s )

We can now solve for the transform X s (f ) of the sampled waveform:

X ( f )∗δ ( f −n f s )= X (f −n f s )

So that

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 ∞ ∞
1 1
X s ( f )= X ( f )∗X δ ( f )=X ( f )∗[ ∑ δ ( f −n f s ) ]= ∑ X ( f −n f s)
T s n=−∞ T s n =−∞

When the sampling rate is chosen f s=2 f m each spectral replicate is separated from
each of its neighbors by a frequency band exactly equal to f s hertz, and the analog
waveform ca theoretically be completely recovered from the samples, by the use of
filtering. It should be clear that if f s >2 f m , the replications will be move farther apart
in frequency making it easier to perform the filtering operation.

When the sampling rate is reduced, such that f s <2 f m , the replications will overlap,
as shown in figure below, and some information will be lost. This phenomenon is
called aliasing.

Sampled spectrum f s >2 f m

Sampled spectrum f s <2 f m

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 A bandlimited signal having no spectral components above f m hertz can be

1
determined uniquely by values sampled at uniform intervals of Ts≤ sec .
2f m

1
The sampling rate is f s= T
s

So that f s ≥2 f m. The sampling rate f s=2 f m is called Nyquist rate.

Example: Find the Nyquist rate and Nyquist interval for the following signals.

sin ⁡(500 πt)

i- m ( t )=
πt
1
ii- m ( t )= cos ( 4000 πt ) cos ( 1000 πt )
2π

Solution:

i- wt =500 πt ∴2 πf =500 π → f =250 Hz

1 1
Nyquist interval ¿ 2 f = =2 msec .
max 2 ×250

Nyquist rate =2 f max =2 ×250=500 Hz

1 1
ii- m ( t )=
2π 2[{ cos ( 4000 πt−1000 πt ) +cos ( 4000 πt +1000 πt ) } ]
1
¿ {cos ( 3000 πt ) +cos ( 5000 πt ) }
4π
Then the highest frequency is 2500Hz
1 1
Nyquist interval ¿ 2 f = =0.2 msec .
max 2 ×2500

Nyquist rate =2 f max =2 ×2500=5000 Hz

H. W:

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Find the Nyquist interval and Nyquist rate for the following:

1
i- cos ( 400 πt ) . cos ( 200 πt )
2π
1
ii- sinπt
π

Example:

A waveform [20+20sin(500t+30o] is to be sampled periodically and

reproduced from these sample values. Find maximum allowable time interval
between sample values, how many sample values are needed to be stored in
order to reproduce 1 sec of this waveform?.
Solution:
x ( t )=20+ 20 sin ( 500 t +300 )
w=500 →2 πf =500 → f =79.58 Hz
Minimum sampling rate will be twice of the signal frequency:
f s (min )=2× 79.58=159.15 Hz
1 1
T s(max)= = =6.283 msec .
f s (min ) 159.15
1
Number of sample in 1 sec= 6.283 msec =159.16 ≈160 sample

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