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You are on page 1of 53

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.

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 trans- mission 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.

Page 2 A.Sarkar,ECE,JGEC,

chapter 1 digital communication

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.

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,

Page 3 A.Sarkar,ECE,JGEC,

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.

The following three sinusoidal functions form the basis functions for the Fourier analysis

Page 4 A.Sarkar,ECE,JGEC,

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

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.

Page 5 A.Sarkar,ECE,JGEC,

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

Page 6 A.Sarkar,ECE,JGEC,

chapter 1 digital communication

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

Page 7 A.Sarkar,ECE,JGEC,

Associated with the complex exponential function

e jwot

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:

Page 8 A.Sarkar,ECE,JGEC,

Page 9 A.Sarkar,ECE,JGEC,

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.

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

Page 10 A.Sarkar,ECE,JGEC,

Page 11 A.Sarkar,ECE,JGEC,

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.

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.

Page 12 A.Sarkar,ECE,JGEC,

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

Page 13 A.Sarkar,ECE,JGEC,

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:

Example:- in the process of modulation

2>Time shifting property

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.,

Page 14 A.Sarkar,ECE,JGEC,

Page 15 A.Sarkar,ECE,JGEC,

Page 16 A.Sarkar,ECE,JGEC,

This theorem states that convolution in time domain is equivalent to multiplication in frequency domain.

f1(t)F1(w)

and

Proof:- F[f1(t)*f2(t)]=

hence

f 1(t ) f 2(t )

f 1(t )e jw d F 2( w) = F1( w) F 2( w)

f1(t)F1(w)

and

Page 17 A.Sarkar,ECE,JGEC,

chapter 1 digital communication

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 =

Page 18 A.Sarkar,ECE,JGEC,

chapter 1 digital communication

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 ) =

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

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

Page 19 A.Sarkar,ECE,JGEC,

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)

Page 20 A.Sarkar,ECE,JGEC,

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

S ( w) =

lim

FT ( w)

FT ( w)

T

may approach a finite value. Let us denote this finite value by S(w) i.e.

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

Page 21 A.Sarkar,ECE,JGEC,

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)

Page 22 A.Sarkar,ECE,JGEC,

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

For a real signal the autocorrelation R() is given by R( ) =

Page 23 A.Sarkar,ECE,JGEC,

R( ) =

R( ) =

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)

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

Page 24 A.Sarkar,ECE,JGEC,

Page 25 A.Sarkar,ECE,JGEC,

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.

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

and we thereby obtain two relations for the joint probability, namely

Or we could eliminate P(AB) to get Bayes' theorem

Total probability:-

Page 26 A.Sarkar,ECE,JGEC,

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.

Page 27 A.Sarkar,ECE,JGEC,

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

Page 28 A.Sarkar,ECE,JGEC,

Gaussian PDF

Page 29 A.Sarkar,ECE,JGEC,

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,

Page 30 A.Sarkar,ECE,JGEC,

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

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.

Page 31 A.Sarkar,ECE,JGEC,

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Page 37 A.Sarkar,ECE,JGEC,

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Page 39 A.Sarkar,ECE,JGEC,

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.

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.

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

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 .,

Page 40 A.Sarkar,ECE,JGEC,

Discrete Time sinusoidal signal:

x[n]=Acos(wn+) -<n< w=2f i.e x[n]=Acos(2fn+)

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

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+)

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.

Page 41 A.Sarkar,ECE,JGEC,

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<

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.

Page 42 A.Sarkar,ECE,JGEC,

chapter 1 digital communication

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.

Page 43 A.Sarkar,ECE,JGEC,

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:

Page 44 A.Sarkar,ECE,JGEC,

example:-

Page 45 A.Sarkar,ECE,JGEC,

Sampling process:-

xs(t)=x(t)g(t)

C e

g(t)=

jn 2f s t

1

Where Cn=

T

T /2

n =

T / 2

Page 46 A.Sarkar,ECE,JGEC,

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 =

Xs( f ) = xs (t )e j 2ft dt =

C x(t )e

n =

jn 2f s t j 2ft

dt

Xs( f ) =

Cn x(t )e j 2 ( f nf s )dt

n =

Page 47 A.Sarkar,ECE,JGEC,

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.

Page 48 A.Sarkar,ECE,JGEC,

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

Cn =

1 0

e = 1/ T = f s

T

Xs ( f ) = f s

X ( f nf )

n =

Page 49 A.Sarkar,ECE,JGEC,

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 =

s

Page 50 A.Sarkar,ECE,JGEC,

Page 51 A.Sarkar,ECE,JGEC,

Discrete info

source

Source

encoder

Channel

encoder

modulator

noise

channel

Inf

sink/destinatio

ns

Source

decoder

Channel

decoder

demodulator

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

Page 52 A.Sarkar,ECE,JGEC,

chapter 1 digital communication

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.

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.

Page 53 A.Sarkar,ECE,JGEC,

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