Basics of Digital Comm

Basics of Digital Communication
chapter 1 digital communication
An introduction to Communication basics and Concepts

A communication system conveys information from its source to a destination some distance away. Recognizing that
all communication systems have the same basic function of information transfer, we'll seek out and isolate the
principles and problems of conveying information in electrical form.
A Historical perspective in the development of digital communications

It is remarkable that the earliest form of electrical communication was a digital communication, namely telegraphy. It
was developed by Samuel Morse in 1837. He used variable length binary code in which English letters are represented
by a sequence of dots and dashes. In this code more frequently used letters are represented by short code words and less
frequently used letters are represented by long code words. Morse code was the precursor of the variable length source
coding.
Nearly 40 years later Emilie Baudot devised a code for telegraphy known as Baudot code which is a binary code of
fixed length 5. Binary code elements are designated as mark and space.
The beginning of modern digital communication was started from the work of Nyquist (1924), who investigated the
problem of determining of maximum signaling rate that can be used over a telegraph channel of a given bandwidth
without inter symbol interference. He formulated a model of a telegraph system in which a transmitted signal has the
general form s(t ) = a n g (t nT ) where g(t) represents a basic pulse shape and {an} is the binary data sequence of
n
{1} transmitted at a rate of 1/T bits/sec. He also determines the optimum pulse shape that was band limited to W Hz
and maximized the bit rate under the constraint that the pulse cause no Inter Symbol Interference at the sampling time.
His studies led him to conclude that the maximum pulse rate is 2W pulses/s. The rate is called Nyquist rate. Moreover
this pulse rate can be achieved by using the pulses g(t)=sinc(t). Nyquist result is equivalent to a version of the
sampling theorem of band limited signals. The sampling theorem states that a signal of bandwidth W can be
reconstructed from samples taken at Nyquist rate of 2W samples/s using interpolation formula.
In light of Nyquist work, Hartley considered the issue of the amount of the data that can be transmitted when multiple
amplitude levels are used.
Another significant advancement of communication was the work of Wiener who considered the problem of estimating
a desired signal waveform s(t) in the presence of additive noise n(t), based on the observation of the received signal
r(t)=s(t)+n(t). Wiener determined the linear filter whose output is the best mean-square approximation of the desired
signal. The resulting filter is called Optimum filter.
Shanon (1948) formulated the basic problem of reliable transmission of information in statistical terms.
The fundamental limitations of information transmission by electrical means are bandwidth and noise.
Taking both limitations into account, Shannon (1948)i stated that the rate of information transmission cannot
exceed the channel capacity. C = B log (1 + S/N)
This relationship, known as the Hartley-Shannon law, sets an upper limit on the performance of a communication
system with a given bandwidth and signal-to- noise ratio.
If the information rate is less than C, then it is theoretically possible to achieve reliable(error free) transmission through
the channel by appropriate coding. On the other hand regardless of signal processing used at the transmitter and
receiver if information rate > c then reliable transmission is not possible.
Hamming in his classic work devised error-detecting and error-correcting codes to combat channel noise.
The increase in demand for data transmission coupled with development of VLSI , has led to development of very
efficient and more reliable digital communication system.
Distortion and Interference
Distortion is waveform perturbation caused by imperfect response of the sys-tem to the desired signal itself. Unlike
noise and interference, distortion disappears when the signal is turned off. If the channel has a linear but distorting
response, then distortion may be corrected, or at least reduced, with the help of special filters called equalizers.
Page 1 A.Sarkar,ECE,JGEC,
Interference is contamination by extraneous signals from human sources- other transmitters, power lines and
machinery, switching circuits, and so on. Interference occurs most often in radio systems whose receiving antennas
usually intercept several signals at the same time. Radio-frequency interference (WI) also appears in cable systems if
the transmission wires or receiver circuitry pick up signals radiated from nearby sources. Appropriate filtering removes
interference to the extent that the interfering signals occupy different frequency bands than the desired signal.
One-way or simplex (SX) transmission. Two-way communication, of course, requires a transmitter and receiver at
each end. A full-duplex (FDX) system has a channel that allows simultaneous transmission in both directions. A halfduplex (HDX) system allows transmission in either direction but not at the same time.
Why is noise unavoidable? Rather curiously, the answer comes from kinetic theory. At any temperature above absolute
zero, thermal energy causes microscopic particles to exhibit random motion. The random motion of charged particles
such as electrons generates random currents or voltages called thermal noise. There are also other types of noise, but
thermal noise appears in every communication system.
Modulation Benefits
The primary purpose of modulation in a communication system is to generate a modulated signal suited to the
characteristics of the transmission channel. Actually, there are several practical benefits and applications of modulation
briefly discussed below.
Modulation for Efficient Transmission: Signal transmission over appreciable distance always involves a traveling
electromagnetic wave, with or without a guiding medium. The efficiency of any particular transmission method
depends upon the frequency of the signal being transmitted. By exploiting the frequency-translation property of CW
Modulation, message information can be impressed on a carrier whose frequency has been selected for the desired
transmission method.
As a case in point, efficient Line-of-sight ratio propagation requires antennas whose physical dimensions are at
least 1/10 of the signal's wavelength. Un-modulated transmission of an audio signal containing frequency components
down to 100 Hz would thus call for antennas some 300 km long. Modulated transmission at 100 MHz,
as in FM broadcasting, allows a practical antenna size of about one meter. It Likewise follows that signals with large
bandwidth should be modulated on high-frequency carriers. Since information rate is proportional to bandwidth,
according to the Hartley-Shannon law, we conclude that a high information rate requires a high carrier
frequency. For instance, a 5 GHz microwave system can accommodate 10,000 times as much information in a given
time interval as a 500 Hz radio channel. Going even higher in the electromagnetic spectrum, one optical laser beam has
a bandwidth potential equivalent to 10 million TV channels.
Modulation to Reduce Noise and Interference: A brute-force method for combating noise and interference is to increase
the signal power until it overwhelms the contaminations. But increasing power is costly and may damage equipment.
This property is called wideband noise reduction because it requires the trans-mission bandwidth to be much greater
than the bandwidth of the modulating signal. Wideband modulation thus allows the designer to exchange increased
bandwidth for decreased signal power, a trade-off implied by the Hartley-Shannon law. Note that a higher carrier
frequency may be needed to accommodate wideband modulation.
Modulation for Frequency Assignment: When you tune a radio or television set to a particular station, you are selecting
one of the many signals being received at that time. Since each station has a different assigned carrier frequency, the
desired signal can be separated from the others by filtering. Were it not for modulation, only one station could
broadcast in a given area; otherwise, two or more broadcasting stations would create a hopeless jumble of interference.
Modulation for Multiplexing: Multiplexing is the process of combining several signals for simultaneous transmission
on one channel. Frequency-division multiplexing (FDM) uses CW modulation to put each signal on a different carrier
frequency, and a bank of filters separates the signals at the destination. Time-division multiplexing (TDM) uses pulse
modulation to put samples of different signals in non-overlapping time slots.
The gaps between pulses could be filled with samples from other signals. A switching circuit at the destination then
separates the samples for signal reconstruction.

A variation of multiplexing is multiple accesses (MA). Whereas multiplexing involves a fixed assignment of the
common communications resource (such as frequency spectrum) at the local level, MA involves the remote sharing of
the resource. For example, code-division multiple accesses (CDMA) assign a unique code to each digital cellular user,
and the individual transmissions are separated by correlation between the codes of the desired transmitting and
receiving parties. Since CDMA allows different users to share the same frequency band simultaneously, it provides
another way of increasing communication efficiency.
Phasors and Line Spectra
Fourier analysis
Abstract: The Fourier Series and its applications to the Communication are discussed. The tutorial is written in a colloquial
style to avoid intimidating readers who are of a lesser level of intelligence
History:
Jean Baptiste Joseph Fourier (1768-1830) studied the mathematical theory of heat conduction in his major work, The Analytic
Theory of Heat. He established the partial differential equation governing heat diffusion and solved it using an infinite series of
trigonometric functions. The description of a signal in terms of elementary trigonometric functions had a profound effect on the
way signals are analyzed. The Fourier method is the most extensively applied signal-processing tool. The Fourier transform of a
signal lends itself to easy interpretation and manipulation, and leads to the concept of frequency analysis.
Fourier was interested in heat propagation, and presented a paper in 1807 to the Institute de France on the use of sinusoids to
represent temperature distributions. The paper contained the controversial claim that any continuous periodic signal could be
represented as the sum of properly chosen sinusoidal waves. Among the reviewers were two of history's most famous
mathematicians, Joseph Louis Lagrange (1736-1813), and Pierre Simon de Laplace (1749-1827).
While Laplace and the other reviewers voted to publish the paper, Lagrange adamantly protested. For nearly 50 years, Lagrange
had insisted that such an approach could not be used to represent signals with corners, i.e., discontinuous slopes, such as in square
waves. The Institute de France bowed to the prestige of Lagrange, and rejected Fourier's work. It was only after Lagrange died
that the paper was finally published, some 15 years later.
Who was right? It's a split decision. Lagrange was correct in his assertion that a summation of sinusoids cannot form a signal with
a corner. However, you can get very close. So close that the difference between the two has zero energy. In this sense, Fourier was
right, although 18th century science knew little about the concept of energy. This phenomenon now goes by the name: Gibbs
Effect,
Objective & Applications:

The applications of the Fourier transform include filtering, telecommunication, music processing, pitch modification, signal
coding, signal synthesis, feature extraction for pattern identification as in speech or image recognition, spectral analysis in physics
and radar signal processing.
The objective of signal transformation is to express a signal as a weighted combination of a set of relatively simple elementary
signals, known as the basis functions. The transform output should lend itself to convenient signal analysis, interpretation and
application. Transformation is an effort to represent an arbitrary signal with a set of signals (called basis functions) with known
characteristics like
amplitude, frequency and phase. The sum of all components in the set provides an approximation of the arbitrary signal. The
difference between the original and the replica is an error, measured in terms of the mean squared error (MSE).
In the Fourier transform the basic elementary signals are sinusoidal signals with different periods giving rise to the concept of
frequency. In Fourier analysis a signal is decomposed into its constituent sinusoids. The amplitudes of sinusoids of various
frequencies form the so-called frequency spectrum of the signal. In inverse Fourier transform a signal can be synthesized by
adding up its constituent frequencies.
The power of the Fourier transform in signal analysis and pattern recognition is in its ability to reveals spectral structures that can
be used to characterize a signal. This is illustrated in Fig. 3.1 for the two extreme cases of a sine wave and a purely random signal.
For a periodic signal, such as a sine wave, the power is concentrated in extremely narrow bands of frequencies indicating the
existence of structure and the predictable character of the signal. In the case of a pure sine wave as shown in Fig. 3.1.a the signal
power is concentrated in one frequency. For
a purely random signal as shown in Fig 3.1.b the signal power is spread equally in the frequency domain indicating the lack of
structure in the signal.
Fourier Series: Representation of Periodic Signals:

The following three sinusoidal functions form the basis functions for the Fourier analysis
why sinusoid is chosen as basis function?

The following properties make the sinusoids the ideal choice as the elementary building block
basis functions for signal analysis and synthesis:
(i) Orthogonality ; two sinusoidal functions of different frequencies have the following orthogonal property:
For harmonically related sinusoids the integration interval can be taken over one period. Similar equations can be derived for the
product of cosines, or sine and cosine, of different frequencies. Orthogonality implies that the sinusoidal basis functions are
independent and can be processed independently. For example in a graphic equalizer we can change the relative amplitudes of one
set of frequencies, such as the bass, without affecting other frequencies
e jwot
(ii) The sine and cosine components of

have only a relative phase difference of /2 or equivalently a relative time delay
of a quarter of one period i.e. T/4. This allows decomposition of a signal in terms of orthogonal cosine and sine components.
(iii) Sinusoidal functions are infinitely differentiable. This is a useful property, as most signal analysis, synthesis and processing
methods require the signals to be differentiable. A useful consequence of transforms, such as the Fourier and the Laplace
transforms, is that differential analysis on the time domain signal become simple algebraic operations on the transformed signal.
Why are sinusoids used instead of, for instance, square or triangular waves?
Remember, there are an infinite number of ways that a signal can be decomposed. The goal of decomposition is to end up with
something easier to deal with than the original signal. For example, impulse decomposition allows signals to be examined one
point at a time, leading to the powerful technique of convolution. The component sine and cosine waves are simpler than the
original signal because they have a property that the original signal does not have: sinusoidal fidelity. A sinusoidal input to a
system is guaranteed to produce a sinusoidal output. Only the amplitude and phase of the signal can change; the frequency and
wave shape must remain the same. Sinusoids are the only waveforms that have this useful property. While square and triangular
decompositions are possible, there is no general reason for them to be useful.
Concept behind Fourier series with mathematical details:

Linear Algebra:A key idea in Linear Algebra is that a vector can be any abstract object. Take two arbitrary vectors v and w:
v = (v1, v2, ..., vn )
w = (w1,w2, ...,wn)
The dot product of two vectors is then the sum of the products of the individual elements making up the two vectors.
v.w = (v1w1 + v2w2 + ... + VnWn)
Central to the Fourier Series (and consequently the Discrete Fourier Transform) is the fact that continuous functions can be
thought of as vectors. Take two sinusoidal functions f and g:
f(x) = sin(x)
g(x) = cos(x)
When the dot product of these two continuous function vectors is taken, it becomes an inner product with infinitely many
terms...thus an integral.
The integrals of certain trigonometric functions (evaluated from zero to twice pi, the period of sinusoidal functions) turn out to be
zero. (their inner products are zero).
Thus by the omnipotent laws of Linear Algebra, the function vectors are orthogonal and can span a space of periodic functions.
This means that any element in the function space can be written as a linear combination of these basis vector functions.
Now that we have established the concept of a set of functions that are orthogonal to one another we are ready to approach the
Fourier Series. It is a fundamental concept in linear algebra that a given set of orthogonal vectors span a space that has a
dimension equal to the number of vectors in the set. For instance, if we have three mutually orthogonal vectors we can think of
them as spanning Cartesian 3-space. From this follows the concept that any object in Cartesian 3-space can be described as a
linear combination of these basis vectors. The coefficients of this linear combination are regarded as that objects coordinates. The
set of orthogonal functions that compose the Fourier series can be thought of as spanning the space of periodic functions. Thus,
analogous to the vectors that span 3-space, this means that a periodic function can be described as a linear combination of the
Fourier basis functions.
It should be noted that not all periodic functions are included in this span. A periodic
function must satisfy three criteria known as the Dirichlet conditions. These conditions are as follows:
1. f(t) is piecewise continuous.
2. f(t) has isolated maxima and minima.
3. f(t) is absolutely integrable over a period.
Summary:Fourier Series
It is a representation of a function f(t) by the linear combination of elements of infinite mutual orthogonal functions.
Mutual orthogonal function:Two real time functions are said to be mutual orthogonal over an interval t1 and t2, if the integral of their product over
this integral is zero .i.e. f(t) is orthogonal to h(t) if
t2
t1
f (t )h(t ) = 0

in terms of vector it is their dot product is zero, which means that has zero component in the direction of other and they
nothing in common. If a function f1(t) does not contain any component that is in the form of any other function f2(t);
then the functions are orthogonal. For example rectangular function f(t) ha s a component sin(t) and hence they are not
orthogonal. The examples of orthogonal functions: Legendre polynomials, Jacobi Polynomials, trigonometric and
exponential functions. Only the last two will be discussed considering their significance in communication systems.
Trigonometric Fourier Series:
Any periodic function can be expressed as the sum of a series of sines and cosines (of varying amplitudes)
a0
f ( x) =
+ a n cos(nx) + bn sin( nx)
2 n =1
n =1
a0 =
an =
bn =
f ( x)dx
f ( x) cos(nx)dx
f ( x) sin(nx)dx
Fourier Series:
Any periodic function can be expressed as the sum of a series of sines and cosines (of varying amplitudes)
f ( x) =
a0 =
an =
bn =
a0
+ a n cos(nx) + bn sin( nx)
2 n =1
n =1
f ( x)dx
f ( x) cos(nx)dx
f ( x) sin(nx)dx
Exponential Fourier series:

Associated with the complex exponential function
e jwot
is a set of harmonically related complex exponential of the form
The set of exponential signals in above equation are periodic with a fundamental frequency
is the fundamental frequency. These signals form the set of basis functions for the
Fourier analysis. Any linear combination of these signals of the form
is also periodic with a period of T0. Conversely any periodic signal x(t) can be synthesized from a linear combination of
harmonically related exponentials.
The Fourier series representation of a periodic signal are given by the following synthesis and analysis equations:
Fourier Synthesis of Discontinuous Signals : Gibbs Phenomenon :

The sinusoidal basis functions of the Fourier transform are smooth and infinitely differentiable. In the vicinity of a discontinuity
the Fourier synthesis of a signal exhibits ripples as shown in the Fig 3.5. The peak amplitude of the ripples does not decrease as
the number of harmonics used in the signal synthesis increases. This behavior is known as the Gibbs phenomenon. For a
discontinuity of unity height, the partial sum of the harmonics exhibits a maximum value of 1.09 (that is an overshoot of 9%)
irrespective of the number of harmonics used in the Fourier series. As the number of harmonics used in the signal synthesis
increases, the ripples become compressed toward the discontinuity but the peak amplitude of the ripples remains constant.
Fourier Transform: Representation of Non-periodic Signals:

The Fourier representation of non-periodic signals can be developed by considering a non-periodic signal as a special case of a
periodic signal with an infinite period. If the period of a signal is infinite, then the signal does not repeat itself and is non-periodic
or Non-periodic.
The Fourier synthesis and analysis equations for non-periodic signals, known as the Fourier transform pair, are given by
The spectrum of an Impulse Function:
The Spectrum of a Rectangular Function: Sinc Function:

The rectangular pulse is a particularly important signal in digital signal analysis and digital Communication. A rectangular pulse is
inherent whenever a signal is segmented and processed Frame by fame, each frame is the result of multiplication of the signal and
a rectangular window. Furthermore, the spectrum of a rectangular pulse can be used to calculate the bandwidth required by pulse
radar systems or by digital communication systems that transmit binary pulses.
Comparison of the Laplace and the Fourier Transforms:

The Fourier transform assumes that the signal is stationary; this implies that the frequency content of the signal (the number of
frequency components and their magnitude and phase) does not change over time. Hence the signal is transformed into a
combination of stationary sine waves of various frequencies, magnitudes and phase. In contract the Laplace transform can model a
non-stationary a signal as a combination of rising, steady and decaying sine waves.
The Laplace transform is particularly useful in solving linear ordinary differential equations as it can transform relatively difficult
differential equations into relatively simple algebraic equations. The Laplace transform of x(t) is given by the integral
Different categories of Fourier analysis:

The general term: Fourier transform, can be broken into four categories, resulting from the four basic types of signals that can be
encountered.
1.)
non-periodic-Continuous
This includes, for example, decaying exponentials and the Gaussian curve. These signals extend to both positive and negative
infinity without repeating in a periodic pattern. The Fourier Transform for this type of signal is simply called the Fourier
Transform.
2.)
Periodic-Continuous
Here the examples include: sine waves, square waves, and any waveform that repeats itself in a regular pattern from negative to
positive infinity. This version of the Fourier transform is called the Fourier Series.
3.)
Non-periodic-Discrete
These signals are only defined at discrete points between positive and negative infinity, and do not repeat themselves in a periodic
fashion. This type of Fourier transform is called the Discrete Time Fourier Transform(DTFT).
4.)
Periodic-Discrete
These are discrete signals that repeat themselves in a periodic fashion from negative to positive infinity. This class of Fourier
Transform is sometimes called the Discrete Fourier Series, but is most often called the Discrete Fourier Transform(DFT).
Two most important property of Fourier transform for communication:1> frequency shifting property:
If f(t)F(w) Then f(t)ejw0tF(w-w0)

Example:- in the process of modulation
f(t)cosw0t1/2[F(w+w0)+F(w-w0)] which is frequently used in modulation

2>Time shifting property
If f(t)F(w) Then f(t-t0)F(w) e-jwt0

We may state that a shift of t0 in the time domain is equivalent to multiplication by e-jwt0 in the frequency domain.
Convolution:- The convolution f(t) of two time functions f1(t) and f2(t), is defined by the following integral
f (t ) =
f 1( ) f 2(t )d
It is a mathematical operation and is useful for describing the input/output relationship in a linear time invariant
system.
Steps to perform convolution graphically
1> f1() is the first function , where an independent variable (t) is replaced by a dummy variable .
2> f2(-) is the mirror image of f2().
3> f2(t-) represents the function f2(-) shifted to the right by t seconds.
4> For a particular value of t=b , integration of the product f1()f2(b-) represents the area under the product
curve(common area). This common area represents the convolution of f1(t) and f2(t) for a shift of t=b
5> The procedure is repeated for different values of t to evaluate the convolution.
6> The value of convolution obtained at different values of t may be plotted on a graph.,
Time convolution theorem:

This theorem states that convolution in time domain is equivalent to multiplication in frequency domain.
f1(t)F1(w)
f2(t)F2(w) then f1(t)*f2(t)F1(w)F2(w)
and
Proof:- F[f1(t)*f2(t)]=
f 1( ) f 2(t )d e jwt dt = f 1( ) e jwt f 2(t )dt d

but we know from time shifting theorem
e jwt f 2(t )dt = F 2( w)e jw
hence
f 1(t ) f 2(t )
f 1(t )e jw d F 2( w) = F1( w) F 2( w)
Frequency convolution theorem

f1(t)F1(w)
and
f2(t)F2(w) then 2f1(t) f2(t)F1(w) *F2(w)
basic important terms in signals & systems

Causal Signal:- The signals having zero value for t<0 is called causal signal. Otherwise they are non-causal.
System:- A system is a set of function that associates an output time function for every input time function.
Linear System:- A system which obeys superposition called Linear system.
a1f1(t)+a2f2(t)a1r1(t)+a2r2(t)
Time Invariant system:-if a time shift in the input causes a equal time shift in the output then the system is time
invariant.
Causal System:- A system where the response does not begin before the input function is applied is known as
causal system. In other words , the value of the output ,r(t) at any instant t=t0 depends only on the values on the
input f(t) for tto.
The unit impulse response h(t) of a causal system is also causal i.e. h(t)=0 for t<0.
Causal system is physically realizable and they are operating in the real world. A physically realizable system
cannot have a response before the driving function or excitation applied. Pauley-wiener criterion gives the physical
realizable properties in frequency domain.
System function or transfer function:-The response r(t) of a linear system to given input f(t) can be determined by taking
advantage of superposition theorem.
1> Decomposition :- resolve the input function f(t) in terms of simpler function such as exponential, impulse for which the
response can easily be evaluated.
2> Determine individually the response of a linear system for the simple input function..
3> Synthesis: - Find the sum of the individual responses which become the output response.
Representation of a function f(t) as continuous sum of impulse function:Let us consider an arbitrary excitation f(t) a shown in figure. In the limit t0, nth element area may be constructed as
a rectangle of width t and height f(nt). This delta function is symbolically represented as
f(nt)( t)(t-nt).
The function is continuous sum of such impulse functions
f (t ) =
lim
f(nt )( t ) (t nt )
t 0 n =

Superposition Theorem to obtain response r(t)
Let h(t) be the unit impulse response of a linear system. i.e. h(t) is the response of a linear system when input is an
impulse. Located at t=0 and of unit strength. Obviously the response of the system for an impulse function of strength
f(nt)( t) located at t= nt will be given as f(nt)( t)h(t-nt). According to superposition theorem , the response r(t)
of the system for input function f(t) will be given by
lim
r (t ) =
f(nt )( t )h(t nt ) Smaller the t the better the approximation and the limit t0, the summation
t 0 n =
becomes integration
Yielding r(t) as
r (t ) =
f ( )h(t )d = f (t ) h(t ) = h(t ) f (t )
R ( w) = F ( w) H ( w)
H(w) is known as transfer function or system function.
Energy Signal & Power signal:A useful parameter of a signal f(t) is its normalized energy. We define the normalized energy ( or simply the energy) E
of a signal as the energy dissipated by a voltage f(t) applied to a 1 ohm resistor( or by passing a current f(t) through 1
ohm resistor). Thus
E=
f 2 (t )dt
The energy of a signal exists only if the integral is finite. The signals for which E is finite is called energy signal.
Non-periodic signals are examples of Energy signals.
The energy may be infinite for many signals and for such signals w used to define another quantity called average
power. Average power may exist if its energy is infinite. Such signals with finite average power is called power signal.
Periodic signals are examples of power signals.
Parsevals theorem for energy signal:- Energy can be calculated in terms of Fourier transformation. This theorem is
very useful as it helps for evaluating the energy of signal without knowing its time domain nature.
1
1
1
E = f 2 (t )dt = f (t ) f (t )dt = f (t ) F ( w)e jwt dwdt =

F ( w) f (t )e jwt dt dw =
F ( w) F ( w)dw
2
2
2
1
2
2
E = f 2 (t )dt =
F ( w) dw = F ( w) df
2
Energy Spectral Density:-Consider a signal f(t) is applied to an ideal narrowband filter. The transfer function of the
filter is H(w). The response of the system is given by R(w)=H(w)F(w).
The energy E of the output signal is given by
1
1
2
2
E=
R( w) dw =
R( w) F ( w) dw
2
2
Now it is obvious that H(w)=0 except for narrow band wm to wm for which it is unity. F(w) is constant with frequency
for a narrowband filter(w0). Hence energy over this narrowband w=2wm is given as
1 wm
1
2
2
(
)
=
(
)
(2 wm )
E=
F
w
dw
F
w
2 wm
2
let 2 wm = w
E=
1
2
2
F ( w) (w) = F ( w) (f )
2
E
2
= F ( w) = Energy Spectral Density = ( w)
f
Thus E represents the contribution of energy due to bandwidth w of the signal. Hence energy contribution per unit
bandwidth is given as energy spectral density.The total energy E is given by
1
E=
2
1
F ( w) dw =
2
2
( w)dw
2
Energy contribution includes both negative and positive frequency and the contribution is equal. F ( w) = F ( w)
For real functions , the spectrum is symmetrical .Hence
E=
( w)dw
The relationship between energy densities of input and response can be derived as follows
R(w)=H(w)F(w)
Hence
2
R( w) = H ( w) F ( w) = H ( w) F ( w)
r ( w) = H ( w) f ( w)
Average Power:- The average power dissipated by a voltage f(t) applied across 1-ohm resistor is defined as the average
power or simply power of the signal f(t). This is same as the power dissipated by a current f(t) passing through a 1 ohm
resistor.
2
lim 1 T / 2
f
(
t
)
dt = f 2 (t )
P=
T T T / 2
The above equation represents the mean square value of the signal f(t) and hence the average power is same as the
mean square value.
The power Density Spectrum:We will drive an expression for the power density spectrum assuming power signal as a limiting case of energy signal.
Consider a power signal f(t) (extending to infinity) . Let us truncate this signal so that it is zero outside the interval
T/2.
f (t ) | t |< T / 2
Let us call FT (t ) =
0 otherwsie
Signal FT(t) is of finite duration T and hence it is an energy signal with energy E given by
ET=
f T (t ) dt =
2
FT ( w) df
Since f(t) over the integral (-T/2,T/2) is same as fT(t) over the interval(-,) we have
f T (t ) dt =
2
T /2
T / 2
f (t ) dt
2
1 T /2
1
f
(
t
)
dt
=
F
(
w
)
df
T
T T / 2
T
In the limit T , the left hand side represents the average power P of the function f(t).
2
lim FT ( w)
P=
df
T
T
2
In the limit T , the ratio
S ( w) =
lim
FT ( w)
FT ( w)
T
may approach a finite value. Let us denote this finite value by S(w) i.e.
.Average power P of the function f(t) is given as
P = f 2 (t ) = S ( w)df =
1
2
S ( w)dw
According to above equation the total power is obtained by multiplying S(w) with bandwidth w and integrating
over the entire bandwidth. Therefore S(w) may be thought as average power per unit bandwidth and hence known as
power spectral density.
2
FT ( w) = FT ( w) the contribution of +ve and ve frequencies are equal. Hence average power may be written as
1
P = 2 S ( w)df = S ( w)dw
0
0
PSD of input and the response:Let us apply a power signal f(t) at the input of a linear system with transfer function H(w) and let the output signal be
r(t).
The signals fT(t) and rT(t) represent signals f(t) and r(t) respectively truncated |t|=T/2.
If we apply this truncated signal at the input the response will not be rT(t), it will extend beyond t=T/2. However since
the input is zero for t>T/2, for a stable system the response for t>T/2 must decay with time. In the limit as T this
contribution (beyond t=T/2) will be of no significance.
Hence for T the response for fT(t) may be considered to be rT(t0 without much error.
lim
f T (t ) rT (t )
T
lim
RT ( w) = H ( w) FT ( w)
T
lim 1
2
S r ( w) =
RT ( w)
T T
lim 1
2
S r ( w) =
H ( w) FT ( w)
T T
lim 1
2
2
S r ( w) = H ( w)
FT ( w)
T T
2
S r ( w) = H ( w) S f ( w)
Correlation:
the application of correlation to signal detection in radar, where a signal pulse is transmitted in order to detect a
suspected target. If a target is present, the pulse will be reflected by it. If there is no pulse, there will be no pulse, just a
noise. By detecting the presence or absence of the reflected pulse we confirm the presence or absence of a target. By
measuring the time delay between the transmitted and received pulses we determine the distance of the target.
For some value of there is a strong correlation, then we not only detect the target also can detect relative time shift of
transmitted signal with received signal. The cross correlation between two functions can be given by
Consider
R1, 2 ( ) =
f 1(t ) f 2(t + )d =
f 1(t ) f 2(t )d
The searching parameter introduced in the expressions of correlations is needed to find out the maximum possible
correlation between waveforms. It may happen when =0 two waveforms have no correlation but they may have
significant correlation with suitable value of . For example two non overlapping signals correlation is zero but
correlation increases as increases.
Difference between Convolution and correlation:Correlation is a function of the delay parameter , where as convolution is a function of t. In convolution delay plays
the role of a dummy variable and it disappears after solution of an integral. Where as in correlation physical time t
plays the role of dummy variable.
Convolution does not depend on which function is shifted and which direction it is shifted but correlation does. So
convolution is commutative.
Autocorrelation:It is a special form of cross-correlation.
R( ) =
f (t ) f (t + )dt =
f (t ) f (t )dt
Autocorrelation is a measure of similarity of a function with its delayed replica. An analogy comparison of your
photo with the photo five year back
Time -autocorrelation and ESD

For a real signal the autocorrelation R() is given by R( ) =
f (t ) f (t + )dt setting x=t+ yields
R( ) =
R( ) =
f ( x) f ( x )dx in this equation x is dummy variable and could be replaced by t. thus

f (t ) f (t )dt
This shows the for a real f(t), the autocorrelation function is an even function of , that is R()=R(-)
We now show that ESD (w)=|F(w)|2 is the Fourier transform of the autocorrelation function R().
F [ R ( )] = f (t ) f (t + )dt e jw d = f (t ) f (t + )e jw d dt
The inner integral is the Fourier transform of f(t+) which is f() left shifted by t seconds and by time shifting property
it is in frequency domain F(w)ejwt . Therefore
F [ R ( )] = F ( w)
f (t )e jwt dt = F ( w) F ( w) =| F ( w) |2 = ( w)
Time - Autocorrelation and PSD

The time autocorrelation function R() of real power signal is given by
lim 1
R ( ) =
f (t ) f (t + ) dt
T T
like energy signal we can show R() is an even function of .
lim 1
lim R ( )
R ( ) =
f
t
f
t
dt
(
)
(
)
+
=
T
T
T T
T T
2
now recall that R()|FT(w)| hence the Fourier transform of the preceding equation gives
2
lim FT ( w)
R( )
= S ( w)
T
T
Probability
we'll deal with random signals whose exact behavior cannot be described in advance. Random signals occur
in communication both as unwanted noise and as desired information-bearing waveforms. Lacking detailed
knowledge of the time variation of a random signal, we must speak instead in terms of probabilities and
statistical properties.
Random Experiment:- Whose outcome cannot be predicted with certainty like flipping a coin, throwing a
die.
Outcome:- elementary result of experiment
Sample space:- set of all possible outcome
Event:- subset of sample space, collection of outcome. Example: for the experiment throwing a die event
outcome is odd {1,3,5}
Disjoint event:- if their intersection is empty.
For this purpose, let's identify a specific event A as something that might be observed on any trial of a chance
experiment. We repeat the experiment N times and record NA the number of times A occurs. The ratio NA / N then
equals the relative frequency of occurrence of the event A for that sequence of trials.
as N becomes very large and if every sequence of trials yields the same limiting value. Under these conditions we take
the probability of A to be P(A)=NA/N
N
The union event A + B (also symbolized by A U B) stands for the occurrence of A or B or both, so its subset consists
of all events either A or B.
The intersection event AB (also symbolized by A B) stands for the occurrence of A and B, so its subset consists
only of those events in both A and B.
P(A1+A2)=P(A1)+P(A2) IF A1A2=0
A,A, = 0 means that they are mutually exclusive.
Consider events A and B that are not mutually exclusive, P(A+B)=P(A)+P(B)-P(AB)
1.
0P(A)1
2.
P(SS)=1 where SS= sample space 3. P(EC)=1-P(E) where EC is the complement of E 4.
Conditional probability:Conditional probabilities are introduced here to account for event dependence and also to define statistical
independence. Let us assume two events A and B have probabilities P(A) and P(B) . I an observer knows that the event
A has occurred , the the probability that event B will occur will not be P(B).
We measure the dependence of B on A in terms of the conditional probability. P(B|A)=P(AB)/P(A)
. If P(A/B)=P(A) then the knowledge of B does not change the probability of occurrence of A. In this case A and B are
said to be independent. for independent events
P(A B)=P(A) P(B)
The notation B/A stands for the event B given A, and P(B/A) represents the probability of B conditioned by the
knowledge that A has occurred. If the events happen to be mutually exclusive, then P(AB) = 0 and confirms that
P(B(A) = 0 as expected. With P(AB) 0, as N
P(B|A)=NAB/N|NA/N= NAB/ NA
Interchanging B and A yields
and we thereby obtain two relations for the joint probability, namely
Or we could eliminate P(AB) to get Bayes' theorem
Total probability:-
Random Variable:It is a mapping from sample space to a set of real numbers. I.e. assignment of real numbers to the outcome of a random
experiment. Despite the name, a random variable is neither random nor a variable. Instead, it's a function that generates
numbers from the outcomes of a chance experiment.
Although a mapping relationship underlies every RV, we usually care only about the resulting numbers. We'll therefore
adopt a more direct viewpoint and treat X itself as the general symbol for the experimental outcomes. This viewpoint
allows us to deal with numerical-valued events such as X = a or X a, where a is some point on the real line.
Furthermore, if we replace the constant a with the independent variable x, then we get probability functions that help us
calculate probabilities of numerical-valued events.
Since the CDF represents a probability, it must be bounded by
Probability density function

But a more common description of a continuous RV is its probability density function (or PDF), defined by
A PDF is a non negative function whose total area is unity and whose area in the range a<x<b equals the probability of
observing x in that range.
let a = x - dx and b = x. The integral then reduces to the differential area px(x) dx and we see that
Some Important Distribution:-
Gaussian PDF
STATISTICAL AVERAGES:For some purposes, a probability function provides more information about an RV than actually needed. Indeed, the
complete description of an RV may prove to be an embarrassment of riches, more confusing than illuminating. Thus, we
often find it more convenient to describe an RV by a few characteristic numbers. These numbers are the various statistical
averages presented here.
The mean of the random variable X is a constant m, that equals the sum of the values of X weighted by their probabilities.
This statistical average corresponds to an ordinary experimental average in the sense that the sum of the values observed
over
N >> 1 trials are expected to be about Nmx. For that reason, we also call mx, the
Expected value of X, and we write E[XI or X to stand for the expectation operation
that yields mx .
which expresses the mean of a discrete RV in terms of its frequency function Px(xi).
for continuous RV,
If g(X) = Xn then E[Xn] is known as the nth moment of X. The first moment, of course, is just the mean value E[X] = mx. The
2
second moment E[X2] or <X2> is called the mean-square value, as distinguished from the mean squared mx2 = X with g(X) =
X2, we have
for a discrete RV,
The mean-square value will be particularly significant when we get to random signals and noise.
The standard deviation of X, denoted by x provides a measure of the spread of observed values of X relative to m,. The square of
the standard deviation is called the variance, or second central moment, defined by
A small standard deviation therefore implies a small spread of likely values, RV concentrated around mean so less random and
vice versa.
Introduction to sampling & reconstruction:Signals can be classified depending on the chars. of the time (independent) variable and the values they take.
Continuous Time (Analog) signals: They are defined for every value of time where they take values in the
continuous interval (a,b). a can be - and b can be . Mathematically they are described by function of a continuous
variable.
X(t)=cost -<t<.
Discrete Time signals:-They are defined only at certain specific values of time. These time instants may not be
equidistant but in usual practice, they are usually taken at equally spaced intervals for computational convenience and
mathematical tractability.
X (tn)=e|-tn| n=0,1, 2,..
If we use n of the discrete time instants as the independent variable, the signal values become a function of integer
variable (i.e. a sequence of numbers). Thus a DT signal is represented by a sequence of numbers.
X (n) =x(t)|t = nTs

A DT signal may arise in two ways:
1> at discrete time instants selecting values of analog signal, known as sampling process
2> By accumulating a variable over a period of time.
For example: Counting the number of cars in a given road every hour or recording the value of gold every day results in
DT signal.
Continuous versus Discrete Valued signal:The value of a CT or DT signal can be continuous or discrete. If a signal takes on all possible values on a finite or
infinite range, it is said to be continuous valued signal.
If the signal takes on values from a finite set of possible values, it is said to be discrete-values signal.
A DT signal having a set of discrete value is called a digital signal.
Conversion form CT to DT signal is done by sampling.
Quantization is the process for converting a continuous-valued signal into a discrete-valued signal.
Concept of Frequency in CT & DT Signal:
Let a continuous time signal mathematically described as
xa(t)=Acos(t+) , -<t<, where =2F i.e xa(t)=Acos(2Ft+)=A/2

ej(t+) + A/2 e-j(t+)
xa(t)= xa(t+T) T=1/F=period
where =+ve angular freq. and - =-ve angular freq units are rad/sec .,
Frequency range for analog signals are -<F<

Discrete Time sinusoidal signal:
x[n]=Acos(wn+) -<n< w=2f i.e x[n]=Acos(2fn+)
In contrast to CT sinusoids DT sinusoids are characterized by the following properties

1> A DT sinusoid is periodic only if its frequency f is a rational number.
x[n+N]=x[n] To Achieve that cos(2fo(n+N)+)= cos(2fon+) it is true only if 2fon=2k . Therefore
fo=k/N=ratio of two integers

2> DT sinusoids whose frequencies are separated by an integer multiple of 2 are identical.
In contrast CT sine signal with distinct frequency are distinct.
cos ((wo+2)n+)=cos(won+2n+)= cos(won+)
as a result , all sinusoidal sequences xk[n]=Acos(wkn+ ), k=0,1,2,3.

Where wk= wo+2k, -wo are indistinguishable(identical)
On the other hand, the sequence at any two sinusoids with frequencies in the range - <w< or
- 1/2<f<1/2 are distinct.
Consequently ,DT sinusoidal signals with frequency |w| or |f|1/2 are unique.
Any sequence resulting from a sinusoid with a frequency |w| or |f|1/2 is identical to sequence obtained from a
sinusoid with frequency |w|<. Because of this similarity, we call the sinusoids with frequency |w| an alias of
corresponding sinusoid with frequency |w|<.
Thus we regard frequency in the range - <w< or - 1/2<f<1/2 are unique and all frequencies |w| or |f|1/2
as alias.
3>The highest rate of oscillation in a DT sinusoid is attained when w= ( or w=-) or f=1/2 or(f=-1/2)
Proof:- let x[n]=coswon To see what happens for w02. We consider sinusoid with frequency w1=w0
and w2=2- w0
as w1 varies from to 2
w2 varies from to 0
x1[n]=ACos w1n= ACos w0n
x2[n]=ACos w2n= ACos(2- w0)n= ACos (-w0n)=x1[n]
Hence w2 is an alias of w1. If you use a sine function, result will be same except for a 180 deg phase shift.
As we increase the relative frequency wo from to 2, rate of oscillation decreases.
For wo =2 , the result is a constant signal as in the case of wo =0. Obviously for wo =(f=1/2) we have
the highest rate of oscillation.
Sampling:- x[n]=x(nT) where t=nT=n/Fs
xa(t)=Acos(2Ft+)
xa(nT)=Acos(2FnT+)
xa(nT)=Acos(2nF/ Fs +)
x[n]= Acos(2nf +)
Therefore f=F/Fs Therefore w=T
-<F<
-1/2<f<1/2
-<<
-<w<
-1/2T= -Fs/2 F Fs/2=1/2T

or
- /T= - Fs Fs= /T
Fundamental difference between CT and DT signal is their range of value for F,f,w,.
Periodic sampling of CT signal implies frequency range F into a finite frequency range f. Since the highest
frequency in a DT signal w= or f=1/2.
Fmax=Fs/2=1/2T
max=Fs=/T
Therefore sampling introduces ambiguity, since the highest frequency in a CT signal that can be uniquely
distinguished when such a signal is sampled at a rate Fs=1/T is Fmax=Fs/2. To see what happens to frequency above
Fs/2 consider the following example.

Example:X1(t)=cos2 (10)t
X2(t)=cos2 (50)t
Both signals sampled at Fs=40Hz yeilds
X1(n)=cos2 (10/40)n=cos n/2
X2(n)=cos2 (50/40)n=cos 5n/2
However cos 5n/2=cos(2n + n/2)=cos(n/2)
Therefore X2(n)= X1(n) Thus they are undistinguishable from the sampled value it is ambiguous to tell whether they
belongs to X1(t) or X2(t).
Since X2(t) yields exactly the same value of X1(t) when two are sampled at Fs=40samples/sec. we say that
F2=50Hz is an alias of F1-10hz at the sample rate=40 samples /sec
Therefore it is important to note that not only F2 is an alias of F!, In fact at the sampling rate of 40 samples /sec the
frequency F3=90hz, f4=130hz etc are also aliases. cos2 (F1+40k)t where k=1,2,3,4.
Is sampled at 40 samples/sec yield identical values.
In General
-1/2fo1/2
-Fs/2FoFs/2
In This case the relationship between Fo and fo is one- to- one, and hence it is possible to identify or reconstruct the
analog signal xa(t) from samples of x[n].
On the other hand xa(t)= Acos(2Fkt+)
Where Fk=Fo+kFs,k=1, 2, 3.
Is sampled at a rate of Fs, it is clear that the frequency Fk is outside the fundamental frequency range
-Fs/2FFs/2
consequently x[n]=xa(nT)= Acos(2(Fo+kFs)/Fs.n+)= Acos(2nFo/Fs++2kn)= Acos(2fon+)
which is identical to DT signal obtained earlier.
Thus an infinite number of CT sinusoid is represented by sampling the same DT signal. Consequently an ambiguity
exists as to which CT signals these values represent.
Since Fs/2 , which corresponds to w=, the highest frequency that can be represented unique with a sampling rate
Fs; it is a simple matter to determine the mapping at any alias frequency above Fs/2(w=) into equivalent
frequency below Fs/2. We can use Fs/2 or w= as the pivotal point and reflect or fold the alias frequency to the
range 0w. The point of reflection Fs/2(w=) is called the folding frequency.
Sampling Theorem:We know that the highest frequency in an analog signal that can be unambiguously represented when the signal is
sampled at a rate Fs=1/T is Fs/2.
Any frequency above Fs/2 or below Fs/2 results in samples that are identical with corresponding frequency in the
range -Fs/2FFs/2
To avoid this ambiguity resulting from aliasing, we must select the sampling rate to be sufficiently high i.e we must
select
Fs/2>Fmax
Fs>2Fmax
With this sampling rate, any frequency component, say |Fi|<Fmax in the analog signal is mapped into a DT sinusoid
with frequency
-1/2fi=Fi/Fs1/2
-wi=2fi
|f|=1/2 or |w|= is highest frequency in a DT signal, choice of sampling rate avoids aliasing. Fs>2Fmax ensures that
all the sinusoidal components in the analog signal are mapped into corresponding DT frequency components with
frequencies in the fundamental interval.
Example:
example:-
Sampling process:-
xs(t)=x(t)g(t)
C e
g(t)=
jn 2f s t
1
Where Cn=
T
T /2
n =
Any periodic can be expressed by Fourier series.
T / 2
g (t )e jn 2f s t dt where fs=1/T=sampling rate
The Ct signal x(t) must be sampled in such a way that the original signal can be reconstructed from these samples
Otherwise the sampling process is useless. Let us obtain the condition necessary to faithfully reconstruct the original
signals from the samples of the signal. The condition can be easily obtained if the signals are analyzed in the frequency
domain.
x s (t ) = x (t ) C n e
jn 2 f s t
n =
= C n x (t ) e jn 2f s t
n =
by Fourier Transform we can obtain the Xs(f)
Xs( f ) = xs (t )e j 2ft dt =
C x(t )e
n =
jn 2f s t j 2ft
dt
interchanging the order of summation and integration
Xs( f ) =
Cn x(t )e j 2 ( f nf s )dt
n =
from the definition of Fourier transform
x(t )e j 2 ( f nf s ) tdt = X ( f nf s )
Therefore
Xs( f ) =
C X ( f nf )
n =
Spectrum of sampled CT signal =spectrum of x(t)+spectrum of x(t) translated to each harmonic of the sampling
frequency.
To avoid overlapping fs-fhfh therefore fs2fh fs=2fh is called Nyquist rate.
Sampling Theorem:- a band limited signal x(t) having no frequency components above fh Hz , is completely specified
by samples that are taken at a uniform rate greater than 2fh hz.
If X(f)=0 for |f|>fh is called band limited.

Sampling by impulse function:-
(t nT )
g (t ) =
n =
By Fourier Series
g(t)=
C n e jn2f st
n =
1 T /2
jn 2f s t
dt where fs=1/T=sampling rate
Where Cn= T / 2 (t )e
T
(t)=1 at t=0
=0 otherwise
x(t ) (t )dt = X (0)
Cn =
1 0
e = 1/ T = f s
T
Xs ( f ) = f s
X ( f nf )
n =
Signal Reconstruction:Spectrum of sampled signal has an amplitude equal to fs=1/T Therefore in order to remove this scalingh constant, the
LPF must have an amplitude response of 1/fs=T.
BW=fs/2=2fh(assuming sampling is done at fs=2fh)
h(t ) = T
fs / 2
fs / 2
e j 2ft df
T
(e jf s t e jf s t )
j 2t
sin f st
h(t ) = Tf s
= sin c( f st )
f st
h(t ) =
x(t ) =
n =
n =
x(nT )h(t nT ) = x(nT ) sin cf (t nT )

s
BLOCK DIAGRAM OF DIGITAL COMMUNICATION SYSTEM

Discrete info
source
Source
encoder
Channel
encoder
modulator
noise
channel
Inf
sink/destinatio
ns
Source
decoder
Channel
decoder
demodulator
ELEMENTS OF A DIGITAL COMMUNICATION SYSTEM

Above Figure shows the functional diagram and basic elements of a digital communication system. In a DCS the
information may emit from a discrete information source like computer or teletype machine. An analog information
source can be transformed into a discrete information source through the process of sampling and quantization.
Discrete information source are characterized by 1) source alphabet 2) symbol rate 3) source alphabet probabilities.
Using this parameters it is possible to construct a probabilistic model of the info source.
Source encoder/decoder: the input to the source encoder is a string of symbols occurring at a particular rate. The
source coder converts the symbol sequence into a binary sequence of 1s and 0s by assigning code words to each
symbol in the input signals. A practical encoder however takes into account the probability of the source symbols and
assigns variable length code words to these symbols. One of the important objectives of the source encoder is to
eliminate or reduce redundancy so as to provide efficient representation of the source. So in summary ideally we should
like to represent the source output (message) by as few binary digits as possible. In other words source output has little
or no redundancies. The process of efficiently converting the output of either an analog or digital source into a
sequence of binary digits is called source encoding or data compression.
At the receiver the source decoder converts the binary o/p of the channel decoder into a symbol sequence. This is then
presented to the info sink or destination.
Channel encoder/decoder: 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 redundancy in a
controlled manner. This redundancy is used in the receiver to overcome the effect of noise and the interference
occurred in the channel. Take k information bit at a time and map each k-bit sequence into a unique n-bit sequence
called a code word. the amount of the redundancy introduced is measured by n/k. reciprocal of that is known as code
rate k/n. in other words channel encoding is a practical method of realizing high transmission reliability and efficiency
over the channel to combat the effect of noise present in channel. Error control is accomplished by the channel coding
operation which consists of systematically adding extra bits to the output of the source coder. Theses extra bits do not
carry any information but they make it possible for the receiver to detect or/and correct some of the errors in the info
bearing bits.
The channel decoder recovers the info bearing bits from the coded binary stream. Error detection and possible
correction is performed by the channel decoder by exploiting the redundancy introduced by channel encoder. The

channel encoder operates either in a block mode or in a continuous sequential mode depending on the type of coding
used in the system.
Modulator/demodulator: the modulator accepts a bit stream as its input and converts it into an electrical waveform
suitable for transmission over the channel. Modulation can be effectively used to minimize the effect of channel noise,
to match the frequency spectrum of the transmitted signal with channel characteristics and to provide the capability to
multiplex many signal over the channel.
Communication channel: the physical channel may be a pair of wires, optical fibers or free space over which
information bearing signals are progressed. The signal is corrupted by additive noise present in the channel. The effect
of noise may be minimized by increasing the power of the transmitted signal. However equipment power ratings and
other practical constraints puts limit of the power level of the transmitted signal. Another basic limitation is that the
communication channels have finite bandwidth and information bearing signals often suffers amplitude and phase
distortion as it travels over the channel.
Advantages of digital communication system

1) Digital transmission of information has some well defined features which make it easy for measuring, analyzing and
manipulating the signals. Another advantage is it is possible to time multiplex digitals signals easily. Multiplexing is
a process which allows various users to share the channel simultaneously. In TDM the entire channel BW is shared
between different users at different times. Each user is assigned unique time slot, during that particular time slot the
entire channel BW is available for that particular user.
2) The use of digital signals simplifies detection and other signal processing equipments.
The noise which is present every where distorts the analog signal and the problem of the receiver is to estimate the
correct value of the signal. In a DCS the problem of estimated signal is changed to the simpler problem of detection as
it is necessary to detect the presence or absence of the signal.
3) Digital signals can be reconditioned or regenerated in repeater stations which are not possible for analog system.
Digital systems can defeat the effect of noise by the process of signal regeneration where the noise corrupted pulses are
reconstructed after detecting them. Such type of reconditioning the signal is not possible in case of analog systems as
the process of amplifying the signal also enhances the noise which can not be distinguished from the analog signal.
4) It is possible to introduce performance monitoring by introducing error correcting and detection techniques. By
introducing redundancies it is possible to detect and/or correct errors.
5) It is possible to encrypt the digital data easily. it is easily possible to protect the data from unauthorized users.
Analog encryption technique is difficult.
6) It is possible to introduce compression techniques easily in a DCS rather than analog system. Which enables us to
use less BW in many modern applications?
Disadvantages:
1) A major disadvantage of DCS is that it typically requires a larger system BW to communicate the same info. In
digital format as compared to an analog format.
2) A DCS requires synchronization, timing control at every stage. We have bit synchronization, frame synchronization
etc. These techniques require special coder and extra hardware.

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Basics of Digital Communication

chapter 1 digital communication

An introduction to Communication basics and Concepts

A Historical perspective in the development of digital communications