Relation between Laplace and Fourier transform

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Basic block diagram of analog communication system

Introduction:-

Communications refer to sending, receiving and processing of information by electrical means, that is it means exchanging information between transmitter and receiver.

In early 1840’s the type of communication used was Wire telegraphy later on the forms are as telephony, Radio communication (possible with the invention of triode tube, Satellite communications and fibre optics(with the invention of transistors and IC’s and semi-conductor devices), that means communications become more advanced with increasing emphasis on computer and other data communications.

A modern communication system is concerned with

before transmission:- 

  • sorting:- sorting for the right message.
  • Processing:- processing is to make that message more suitable for transmission.
  • storing:- storing that message before transmission.

then the actual transmission of that message takes place (processing and filtering  noise)

at the receiver:-

  • decoding:-decoding the original message.
  • storage:-storing a copy of that message.
  • interpretation:-and analyzing for the correctness of that message.

the different forms of modern communication systems includes Mobile communications,Computer communications, Radio telemetry etc.

to become familiar with communication systems one needs to know about amplifiers and oscillators that means fundamentals of electronic circuits must be known, with these concepts as a background the every day communication concepts like noise, modulation and information theory as well as various types of systems may be studied.

The most general form of Communication system ( one or two blocks may differ) is shown in the figure basic terminology used in Communication systems is message signal /information/data,channel,noise,modulation, encoding and decoding. Communication system is meant for communicating messages between Transmitter and Receiver (or) source & destination.

source:-

source or information source is the primary block in communication system which generates original message / actual message. 

i.e, selecting one message (actual message) from a group of messages itself is called as sorting data (or) information. Source generates message which may be in any form like words, code , symbols, sound signal, images, videos etc.among these the desired message has been selected and conveyed.

A transducer is one which converts one form of energy into electrical energy because the message from information source may not be always in electrical form, a transducer is used in between source and transmitter as a separate block sometimes (or) may be a part of Tx r.

Transmitter:-

Txr is meant for the following tasks

  • restriction of range of audio frequencies (i.e, limiting the bandwidth of the message signal).
  • Amplification.
  • Modulation. 

In general modulation is said to be the main function of the transmitter.

Channel:-

The medium that exists between transmitter and receiver is called as channel. The function of channel is to provide connection between transmitter  and receiver, two types of channels are  there wired/point to point  and wireless/broadcasting channels.

Point to point channels are generally wired channels(i.e, a physical medium exists) like Microwave links, optical fibre links etc. 

Microwave links:- these links are used in telephone transmission.In these type of links guided EM waves are used to transmit from Txr to Rxr.

optical fibre links:- used in low-loss high speed data transmission and uses optical fibers as the medium .

Broadcast channels:- the medium or channel is wireless here, in broadcasting a single transmitter can send information to many receivers simultaneously, satellite broadcasting system is one such system.

during the process of transmission and reception, the signal gets distorted due to noise in the channel, noise may interfere with the signal at any point but noise in the channel has greatest effect on the signal.

Receiver:-

The main function of the receiver is to reproduce the message signal in electrical form from the distorted received signal. This reproduction process is called demodulation (or) detection , in general this demodulation may be assumed as the reverse process of modulation carried out in transmission. 

there are a great variety of receivers in communication systems, the type of receiver chosen depends on type of modulation, operating frequency ,its range  and type of destination required. Most common receiver is superheterodyne receiver .

                            crystal receiver with head phones
                                  Radio receiver

so many types of receivers are available from a very simple crystal receiver with headphones to radar receiver etc.

Destination:- It is the final stage of any communication system. it would be a loud speaker / a display device/simply a load etc depending up on the requirement of the system.

flooding (static)

This is another type of static algorithm.

the main concept of flooding is to sent every incoming packet on a line to every other outgoing line except the line it arrived on.

flooding generates a large no.of duplicate packets, sometimes infinite unless we may take certain measures.

the measures are as follows:-

  • one measure is use of hop count in the header of each packet and decrement this count at each hop when count reaches to zero discard the packet.
  • How to take this hop count is another problem. Generally it is set to the length of path from source to destination and in worst cases the full diameter of the subnet.

  • another way is avoid sending a packet more than once through a router this is possible by using sequence no.
  • i.e, a source router (which generates packets) can put a sequence no. to each packet and each router will maintain a list of sequence nos. and if sees a packet with same sequence no in the list that packet is discarded (not flooded).

another way of flooding is of use selective flooding.

i.e, with this the router wouldn’t send every incoming packet on every line instead the router will send packets in a particular direction only.

i.e, east bound packets are sent on east side routers and similarly on  west side by west side routers.

even flooding is cumbersome, it has some uses

i.e,

  1. used in military applications.
  2. used in distributive data base applications in which to update all data bases concurrently.
  3. used in broadcast Routing.

flooding is used rather than any other algorithm since flooding chooses shorter path between two nodes where other algorithms may not.

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Quadrature Phase Shift Keying (QPSK) Transmitter and Receiver

Quadrature Phase Shift keying:-

The designing of digital communication system requires two important goals to achieve
1. To achieve low probability of error Pe.
2. To utilize Channel Band width efficiently.
QPSK is A Band width conserving modulation scheme, which is an example of Quadrature Carrier Multiplexing.
The modulation schemes such as ASK, PSK & FSK does not meet the Band width requirements of data Communication systems since the Bit rate and Baud rate are same in these schemes. Since the channel band width depends up on the bit rate (or) signalling rate of the modulation scheme. If two (or) more bits are combined into a symbol, then the signalling rate is reduced. Therefore the frequency of the carrier is also reduced, this reduces the transmission channel band width. Thus grouping of bits into symbols reduces Channel Band width.

Meaning of QPSK:-

In Quadri Phase Shift Keying as with Binary PSK information carried by the transmitted signal is contained in the phase of the carrier. The phase of the carrier Φc takes on one of four equally spaced values such as π/4, 3π/4, 5π/4 and 7π/4 that is in QPSK two successive bits are combined into a di-bit or symbol and each possible value of the phase corresponds to a unique di-bit.
for example the foregoing set of phase values are chosen to represent the gray encoded set of di-bits 10, 00, 01 and 11 , where only a single bit is changed from one di-bit to the next.

Generation of QPSK/ QPSK transmitter:-

Consider the generation and detection of QPSK signals. The figure shows a Block diagram of a typical QPSK Transmitter.The incoming binary sequence is first transmitted into polar form by a Non-Return to zero level encoder. Thus symbols 1 and 0 are represented by
√ Es   and –√ Es
This binary wave is next divided by means of a de-multiplexer into two separate binary waves. Consisting of the odd and even numbered input bits {be(t)} and {bo(t)} represents those two binary waves.
The two bit streams be(t) and bo(t) are modulated by two ortho-normal basis functions Φ1(t) and Φ2(t).finally, the two binary PSK signals are added to produce the desired QPSK signal.
i.e, SQPSK(t) = Se(t) + So(t).
SoPSK(t)= bo(t)* √(2/Ts)* cos 2πfc t
SePSK(t)= be(t)* √(2/Ts)* sin 2πfc t

SQPSK(t)= bo(t)* √(2/Ts)* cos 2πfc t + be(t)* √(2/Ts)* sin 2πfc t.

QPSK Receiver:-

The QPSK Receiver consists of a pair of correlators  called as In-phase channel and Quadrature phase channel with a common input.  The input x(t) is supplied with a pair of coherent reference signals Φ1(t) and Φ2(t).  The two correlators produces two signals x1(t) and x2(t) in response to the received signal x(t). these signals x1(t) and x2(t) are compared with threshold voltage 0V by the decision devices in the two channels.

If x1 >0, a decision has been made in favor of symbol ‘1’ for the in-phase channel output. but if x1<0 a decision has been made in favor of ‘0’. similarly for the Q-phase channel,

x2>0—-> a symbol ‘1’ is decided.

x2<0—-> a symbol ‘0’ is decided.

finally, these two binary sequences at the I-phase and Q-phase channel outputs are combined in a multiplexer to reproduce the original binary sequence at the Receiver output with the minimum probability of symbol error in the AWGN channel.

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Reconstruction filter(Low Pass Filter)

Reconstruction filter (Low Pass Filter) Procedure to reconstruct actual signal from sampled signal:-

Low Pass Filter is used to recover original signal from it’s samples. This is also known as interpolation filter.

An LPF is that type of filter which passes only low frequencies up to cut-off frequency and rejects all other frequencies above cut-off frequency.

For an ideal LPF, there is a sharp change in the response at cut-off frequency as shown in the figure.

i.e, Amplitude response becomes suddenly zero at cut-off frequency which is not possible practically that means an ideal LPF is not physically realizable.

i.e, in place of an  ideal LPF a practical filter is used.

In case of a practical filter, the amplitude response decreases slowly to zero (this is one of the reason why we choose  f_{s}>2f_{m})

This means that there exists a transition band in case of practical Low Pass Filter in the reconstruction of original signal from its samples.

Signal Reconstruction (Interpolation function):-

The process of reconstructing a Continuous Time signal x(t) from it’s samples is known as interpolation.

Interpolation gives either approximate (or) exact reconstruction (or) recovery of CT signal.

One of the simplest interpolation procedures is known as zero-order hold.

Another procedure is linear interpolation. In linear interpolation the adjacent samples (or) sample points are connected by straight lines.

We may also use higher order interpolation formula for reconstructing the CT signal from its sample values.

If we use the above process (Higher order interpolation) the sample points are connected by higher order polynomials (or) other mathematical functions.

For a Band limited signal, if the sampling instants are sufficiently large then the signal may be reconstructed exactly by using a LPF.

In this case an exact interpolation can be carried out between sample points.

Mathematical analysis:-

A Band limited signal x(t) can be reconstructed completely from its samples, which has higher frequency component fm Hz.

If we pass the sampled signal through a LPF having cut-off frequency of  fm  Hz.

From sampling theorem  

g(t) = x(t).\delta _{T_{s}}(t).

g(t)=\frac{1}{T_{s}}\left \{ 1+2\cos \omega _{s}t+2\cos 2\omega _{s}t+2\cos 3\omega _{s}t+..... \right \}.

g(t)     has a multiplication factor  \frac{1}{T_{s}}. To reconstruct  x(t)  (or)  X(f) , the sampled signal must be passed through an ideal LPF of Band Width of  f_{m}  Hz and gain  T_{s} .

\left | H(\omega ) \right |=T_{s} \ for \ -\omega _{m}\leq \omega \leq \omega _{m}.

h(t) = \frac{1}{2\pi } \int_{-\omega _{m}}^{\omega _{m}}T_{s}e^{j\omega t}\ d\omega.

h(t) = 2f_{m}T_{s} \ sinc(2\pi f_{m}t).

If sampling is done at Nyquist rate , then Nyquist interval is  T_{s} = \frac{1}{2f_{m}}.

 therefore  h(t) = \ sinc(2\pi f_{m}t).

h(t) = 0.      at all Nyquist instants  t= \pm \frac{n}{2f_{m}}  , when    g(t)    is applied at the input to this filter the output will be  x(t)  .

Each sample in g(t)  results a sinc pulse having amplitude equal to the strength of sample. If we add all these sinc pulses that gives the original signal  x(t) .

g(t) = x(kT_{s})\delta (t-kT_{s}).

x(t) =\sum_{k} x(kT_{s})\ h (t-kT_{s}) .

x(t) =\sum_{k} x(kT_{s})\ sinc(2\pi f_{m} (t-kT_{s})).

x(t) =\sum_{k} x(kT_{s})\ sinc(2\pi f_{m}t-k\pi ) .

This is known as interpolation formula

It is assumed that the signal  x(t) is strictly band limited but in general an information signal may contain a wide range of frequencies and can not be strictly band limited this means that the maximum frequency in the signal can not be predictable.

then it is not possible to select suitable sampling frequency  fs  .

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Circuit Switched Networks

A Circuit Switched N/w consists of a set of switches connected by physical links.

A connection b/w ‘2’ stations is dedicated path made of one (or) more links. Each connection uses only one dedicated channel on each link.

i.e, each link is divided into n channels either by using TDM (or) FDM.

This circuit consists of 4 switches I, II, III and IV and Multiplexers with n=’3′ channels and one link.

In some circuits Multiplexing can be implicitly included in the switch fabric it self. In this circuit the end systems are connected to a switch for simplicity consider ‘2’ end systems A and M, connected to the switches I and III.

when A needs to communicate with M . A needs to request to a connection to M, which must be accepted by all switches and by M it self- which is called setup phase.

a channel circuit is reserved on each link and the combination of circuits forms a dedicated path. After establishing path data transfer can take place. The next phase is tear down.

i.e, after all data have been transferred. Generally circuit-switching takes place at the physical layer.

Before Communication (starting), the stations must make reservation for the resources like channels, switch buffers switch i/o ports switch processing time and are dedicated during the entire duration of data transfer until the tear down phase.

Data transferred is not packatized that is data is send as a continuous flow b/w source and destination.

there is end-to-end addressing in setup phase.

The 3 phases involved are:-

Circuit switched N/w’s requires ‘3’  setup phases

  1. Connection-setup.
  2. Data transfer.
  3. Tear down.

Setup Phase:-

A dedicated circuit is established before the ‘2’ communicating parties talk to each other.

i.e, creating  a dedicated channels b/w switches. To communicate A with M . initially a requesting process as follows

A to I, I to IV and IV to III, III to M and an acknowledgement in the reverse order after the reception of ‘ack’ a connection is established.

Data Transfer Phase:-

In this phase data transfer occurs b/w the ‘2’ devices.

Tear down phase:-

To disconnect , a signal is sent to each switch to release the resources by any one of station.

Efficiency of Circuit Switched Network:-

These are less efficient in terms of allocated resources. Since all the resources are allocated during the entire duration of the connection  and these resources are un available to other connections.

Delay in this type of N/w’s is due to establishment of connection , data transfer and disconnecting the circuit.

Switching at the physical layer in the traditional telephone N/w uses the circuit switching approach.

Broadcast Routing(dynamic)

In some applications hosts need to send messages to many (or) all other hosts like weather reports, stock market updates (or) live radio programs.

 i.e, sending a packet to all destinations simultaneously is called Broadcasting.

 Different methods of Broadcasting:-

  • first method is to send a packet to all destinations. This is a method wasteful of Band width and source needs to know the complete list of all destinations.

so this is least desirable one.

  • flooding is another way to broadcast a packet, the problem with flooding is that it generates too many packets and also consumes too much of Band width.
  • Third way is to use multi destination routing

In this technique each packet contains a list of destinations (or) a bit map for those destinations.

when a packet arrives at a router,  the router checks all the output lines it requires. The router generates a new copy of the packet for each output line after sufficient number of hops each packet will carry only one destination.

i.e, multi destination routing is like separately addressed packets (to B,C,D,E & D) must follow the same route one of them pays full fare and rest are free.

  • The fourth type of method is to use sink tree (or) spanning tree.

A spanning tree is a subset of subnet that includes all the routers but contains no loops.

if each router knows which of it’s lines belong to spinning tree then it broadcasts packet to all the lines except the one it arrived on.

This is efficient method in terms of Band width usage but problem is to maintain the knowledge of all the nodes of spanning tree at a routes.

  • Last method is to use Reverse path forwarding to approximate behavior of spanning tree.

Consider a subnet and it’s sink tree for router I as root node and how reverse path algorithm works in figure (C) 

on the first hop I sends packets to F, H, J & N. on the second hop eight packets are generated among them 5 are given to preferred paths indicated as circles (A,D,G,O,M)

of the 6 packets generated in third hop only 3 are given to preferred paths (C,E & K) the others are duplicates.

in the fourth hop to B and L after this broadcasting terminates.

advantages of reverse path forwarding:-

  • it is easy to implement.
  • it does not require routers to known about spanning trees.
  • it does not require any special mechanism to stop the process (as like flooding).

The principle is: if a packet arrives on a line if it is preferred one to reach the source it gets forwarded.

if it arrives on a line that is not preferred one that packet is discarded as a duplicate.

ex:-

 

when a packet arrives at ‘L’ the preferred paths are N and P so it forwards the packets to both N and P and if a packet arrives at ‘K’, there the preferred path is M, and N is not preferred so it forwards the packet to M and discards to N.

This is reverse path forwarding.

 

aliasing effect in Sampling

Effect of under sampling (aliasing effect):-

When a Continuous Time  band-limited signal is sampled at, then the successive cycles of the spectrum of the sampled signal overlap with each other as shown below

Some aliasing is produced in the signal this is due to under-sampling.

aliasing is the phenomenon in which a high-frequency component in the frequency spectrum of the signal takes as a low-frequency component in the spectrum of the sampled signal.

Because of aliasing, it is not possible to reconstruct x(t) from g(t) by low pass filtering.

The spectral components are in the overlapping regions and hence the signal is distorted.

Since any information signal contains a large no. of frequencies so the decision of sampling frequency always becomes a problem.

A signal is first passed through LPF  before sampling.

i.e, it is band limited by this LPF which is known as a pre-alias filter.

To avoid aliasing

  1. Pre-alias filter must be used to limit the bandwidth of the signal to f_{m}  Hz.
  2. Sampling frequency must be  f_{s}>2f_{m}.

Pre-alias filter means before sampling is passed through an LPF to make a perfect band-limited signal.

 

FSK Generator /BFSK generator

we know that the input to the FSK Generator is a binary sequence 1010…etc.

FSK generator uses two product modulators upper-modulator and lower-modulator with carriers

 and

 .

A level shifter is there in which the output of the level shifter is

when the input is a binary ‘1’ and ‘0’ volts for the input ‘0’ level shifter.

i.e, 

The working of the FSK generator is as follows when the input binary sequence is ‘1’

on the upper modulator 1 has been shifted to a voltage  so that the output of product modulator 1 is

and on the lower modulator input ‘1’ is passed through an inverter and if the output of the inverter is ‘0’ then the output of the level shifter will not change it remains at ‘0’ volt itself.

then the product modulator 2 output is 

then the overall output

similarly, when the input sequence is a binary ‘0’ 

 

Frequency Shift Keying (FSK)/BFSK

In a Binary FSK system, symbols 1 and 0 are distinguished from each other by transmitting one of two sinusoidal waves that differ in frequency by a fixed amount.

(or)

The frequency of the carrier signal shifted to two frequencies  for symbols ‘1’ and ‘0’ transmission.

The equation for FSK signal is

\[ S_{FSK}(t)= \sqrt{\frac{2E_{b}}{T_{b}}} \ \ cos (2\pi f_{c_{i}}t),\ 0\leq t\leq T_{b} , \ i=1,2 \]

\[ S_{FSK}(t)= 0 elsewhere \]

i.e, 

where 

 is generally a high frequency.

 is a low frequency and vice-versa is also true.

 

 

 

 

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Types of digital Modulation techniques (or) systems:-

Digital Modulation techniques may be classified into coherent (or) Non-coherent techniques, depending on whether the receiver is equipped with a phase-recovery circuit (or) not.

Coherent digital modulation techniques (or) systems Non-coherent digital Modulation techniques (or) systems /Envelope detection.
1. In this scheme, the local carrier generated at the receiver is phase locked with the carrier at the transmitter.

i.e, phase lock exists between Transmitter and Receiver.

1. There exists no such phase lock between Transmitter and Receiver.

 

2. This is also called  Synchronous detection. 2. This is known as Non-synchronous detection.
3. complexity increases in terms of designing of the receiver. 3. less complexity in terms of designing the receiver.

 

4. probability of error decreases.

Examples:- coherent ASK, PSK, and FSK systems.

4. error probability increases.

examples:-Non-coherent ASK, PSK, and FSK systems.

 

Base band Vs Pass band Transmission

Baseband data transmission Passband data Transmission
1. The digital data is transmitted over the channel directly, there is no carrier (or) any modulation. 1. The digital data modulates  high-frequency sinusoidal carrier. Hence it is also called as digital CW modulation .

∴carrier is required.

2. This is suitable for transmission over short distances.
Examples:- Ethernet signals operating over a LAN (Local Area Network)
The most common baseband modulation is (PAM) and PCM in local digital computer links.
2. suitable for long distances transmission.
Examples:- Microwave links, Satellite Communication links are called  Passband communication systems.
3. Baseband transmission sends the information signal as it is without modulation.
i.e, without frequency shifting.
3. passband transmission shifts the signal(information) to be transmitted in low frequency to a higher frequency.
i.e, Modulation is required.
4. baseband signals are in general low-frequency signals
i. human voice(20Hz-5KHz).
ii. video signal from a TV camera (0Hz-5.5MHz).
Examples:-
The telephone systems used for offices and homes (one room to other) transmits baseband signal as it is the system falls into baseband communication systems.
4. whereas long-distance call that is transmitted via microwave (or) satellite links uses modulation which is known as passband communication systems.
examples :- Passband Modulation techniques ASK,DPSK,FSK,QPSK,PSK,M-ary PSK etc.
 
   
   

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Drawbacks in Delta Modulation (or) errors in Delta Modulation

Delta Modulation is subject to two types of quantization error

  1. slope Overload Distortion (SOD)/ Slope Overload error
  2. Granular Noise / Granular error.

During the process of digital equivalent integration of x(t) that is approximating x(t) with   there exists an error called Quantization error as shown by

if time instance is (n-1)th instance.

the Quantization error q[n] is of two types in Delta Modulation.

  1. Slope-Overload Distortion:- if the rate of rise of  input signal is so high

i.e, the slope of the signal is so high so that the stair case signal  can not approximate to x(t) .

i.e, as            is large enough

in this case the step size     becomes to small for the stair case approximation   to follow a steep segment of the input wave form x(t) with the result that   falls behind   which can be clearly visible in the figure.

the Qunatization error that exist between x(t) and   in this condition is called as slope overload distortion.

Generaaly DM is often referred as a linear Delta Modulator because the step size  is fixed during approximation process, and also its maximum (or) minimum slopes occur along straight lines.

To avoid slope Overload Distortion, step size must be increased.

Granular Noise (or) Idle Noise:-

In contrast to slope overload distortion Granular noise occurs when the step size  is too large relative to the local slope characteristic of the input wave form  x(t)

∴ This large value of   causes that the stair case approximation  to hunt around a flat segment of the input wave form as shown in the above figure

i.e,     oscillates between   when. x(t) is almost straight.

∴ The error between    and    in this condition is called as Granular noise (or) Idle noise.

To eliminate this error is to make the step size   small.

Granular noise occurs that for a very small variations in the input signal causes a very large variations in the approximated signal  .

Thus we see that there is a need to have a large step size  to accomodate a wide dynamic range of input signal.

and a small step size is required to accurate representation of relatively low-level signals.

i.e, large step size is required to reduce slope overload distortion and small step size is required to reduce Granular noise.

∴ It is clear that the choice of the optimum step size that minimizes the Mean Square value of the Quantization error in a Linear Delat Modulator will be the result of a compromise between Slope overload Distortion and Granular Noise.

To satisfy such a requirement , we need to make the Delta modulator “Adaptive” in the sense that the step size is made to vary in accordance with the input signal x(t).

This can be further discussed in the topic called as “Adaptive Delta Modulation ” scheme (ADM).

Block Diagram of Digital Communication system

Block Diagram of Digital Communication System/Elements of DCS:-
 
A General Communication System can be viewed as a Transmitting unit and a Receiving Unit connected by a medium(Channel). Obviously, Transmitter and Receiver consist of various sub-systems (or) blocks.
Our basic aim is to understand the various modules and sub-systems in the system. If we are trying to understand the design and various features of DCS, it is plus imperative that we have to understand how we should design a transmitter and we must understand how to design a very good quality Receiver. Therefore one must know the features of the channel to design a good Transmitter as well as a receiver that is the channel and its contribution will come repeatedly in digital Communications.
Source:- the primary block (or) the starting point of a DCS is an information source, it may be an analog/digital source, for example, if the signal considered is analog in nature, then
 

the signal generated by the source is some kind of electrical signal which is random in nature. if the signal is a speech signal (not an electrical signal) that has to be converted into an electrical signal by means of a Transducer, which can be considered as a part of the source itself.
Sampling & Quantization:- the secondary block involves the conversion of analog to discrete signal this involves the following steps
Sampling:- it is the process that involves in the conversion of Continuous Amplitude Continuous Time (CACT) signal into Continuous Amplitude Discrete Time (CADT) signal.
Quantization:- it is the process that involves in the conversion of Continuous Amplitude Discrete Time (CADT) signal into Discrete Amplitude Discrete Time (DADT) signal.
Source Encoder:- An important problem in Digital Communications is the efficient representation of data generated by a Discrete Source, this is accomplished by source encoder.
” The process of representation of incoming data from a Discrete source into a more suitable form required for Transmission is known as source encoding”
Note:-The blocks Sampler, Quantizer followed by an Encoder constructs ADC (Analog to Digital Converter).
∴ the output of Source encoder is a Digital Signal, the advantages of Source coding are

  • It reduces the Redundancy.
  • Minimizes the avaerage bit rate.

Channel encoder:- Channel coding is also known as error control coding. Channel coding is a technique that reduces the probability of error by reducing Signal to Noise Ratio at the expense of Transmission Band Width. The device that performs the channel coding is known as the Channel encoder.

Channel encoding increases the redundancy of incoming data, this also involves error detection and error correction along with the channel decoder at the receiver.

Spreading Techniques:- Spread Spectrum techniques are the methods by which a signal generated with a particular Band Width is deliberately spread in the frequency domain, resulting in a signal with a wider Bandwidth.

There are two types of spreading techniques available

1. Direct Sequence Spread Spectrum Techniques.

2. Frequency Hopping Spread Spectrum Techniques.

The output of a spreaded signal is very much larger than incoming sequence. Spreading increases the BW required for transmission, which is a disadvantage even though spreading is done for high security of data.

SS techniques are used in Military applications.

Modulator:- spreaded sequence is modulated by using digital modulation schemes like ASK, PSK, FSK etc depending up on the requirement, now the transmitting antenna transmits the modulated data into the channel.

Receiver:- Once you understood the process involved in transmitter Block. One should perform reverse operations in the receiver block.

i.e the input of the demodulator is demodulated after that de- spreaded and then the channel decoder removes the redundancy added by the channel encoder ,the output of channel decoder is then source decoded and is given to Destination.

Digital Communication systems Vs Analog Communication Systems

 
Introduction:-
Communication is the process of establishing a Connection (or) link between two points (which are separated by some distance) and transporting information between those two points. The electronic equipment used for communication purposes is called Communication equipment. The equipment when assembled together forms a communication system.
Examples of different types of communications

  • Line Telephony & Telegraphy.
  • Radio Broadcasting.
  • Point-to-Point Communication.
  • Mobile Communication.
  • TV Broadcasting.
  • Radar and Satellite Communications.

Why Digital?
A General Communication system has two devices and a medium (channel) connecting those two devices. This can be understood that a Transmitter and Receiver are separated by a medium called a Communication channel. To transport an information-bearing signal from one point to another point over a communication channel either Analog or digital modulation techniques are used.
Now Coming to the point, Why Digital communication is preferred over analog Communication?
Why are communication systems, military and commercial alike, going digital?

1. There are many reasons; the primary advantage is the ease with which digital signals compared with analog signals are generated. That is the generation of digital signals is much easier compared to analog signals.
2. Propagation of Digital pulse through a Transmission line:-
When an ideal binary digital pulse propagates along a Transmission line. The shape of the waveform is affected by two mechanisms
Distortion caused on the ideal pulse because all Transmission lines and Circuits have some Non-ideal frequency Transfer function.
Unwanted electrical noise (or) other interference further distorts the pulse waveform.
Both of these mechanisms cause the pulse shape to degrade as a function of line length. During the time that the transmitted pulse can still be reliably identified (i.e. before it is degraded to an ambiguous state). The pulse is amplified by a digital amplifier that recovers its original ideal shape. The pulse is thus “re-born” (or) regenerated.
Circuits that perform this function at regular intervals along the Transmission system are called “regenerative repeaters’. This

is one of the reasons why digital is preferred over
3. Digital Circuits Vs Analog Circuits:-
Digital Circuits are less subject to distortion and Interference than are analog circuits because binary digital circuits operate in one of two states FULLY ON (or) FULLY OFF to be meaningful, a disturbance must be large enough to change the circuit operating point from one state to another. Such two-state operation facilitates signal representation and thus prevents noise and other disturbances from accumulating in transmission.
However, analog signals are not two-state signals, they can take an infinite variety of shapes with analog circuits and even a small disturbance can render the reproduced waveform unacceptably distorted. Once the analog signal is distorted, the distortion cannot be removed by amplification because accumulated noise is irrecoverably bound to analog signals, they cannot be perfectly generated.
4. With digital techniques, extremely low error rates, and high signal fidelity is possible through error detection and correction but similar procedures are not available with analog techniques.
5. Digital circuits are more reliable and can be produced at a lower cost than analog circuits also; digital hardware lends itself to more flexible implementation than analog hardware.
Ex: – Microprocessors, Digital switching, and large-scale Integrated circuits.
6. The combining of Digital signals using Time Division Multiplexing (TDM) is simpler than the combining of analog signals using Frequency Division Multiplexing (FDM).
7. Digital techniques lend themselves naturally to signal processing functions that protect against interference and

jamming (or) that provide encryption and privacy and also much data communication is from computer to computer (or) from digital instruments (or) terminal to computer, such digital terminations are normally best served by Digital Communication links.

8. Digital systems tend to be very signal-processing intensive compared with analog systems.

Apart from pros there exists a con in Digital Communications that is non-graceful degradation when the SNR drops below a certain threshold, the quality of service can change suddenly from very good to very poor. In contrast, most analog Communication Systems degrade more gracefully.

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