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You are on page 1of 119

ON

DIGITAL COMMUNICATIONS

UNIT I

ELEMENTS OF DIGITAL COMMUNICATION SYSTEMS

from a source to a user destination via a communication channel.

Transmitter,

Receiver and

Channel.

The Overall purpose of this system is to transfer information from one point (called Source)

The message produced by a source, normally, is not electrical. Hence an input transducer

is used for converting the message to a time varying electrical quantity called message

signal. Similarly, at the destination point, another transducer converts the electrical

waveform to the appropriate message.

The transmitter is located at one point in space, the receiver is located at some other point

separate from the transmitter, and the channel is the medium that provides the electrical

connection between them.

The purpose of the transmitter is to transform the message signal produced by the source

of information into a form suitable for transmission over the channel.

The received signal is normally corrupted version of the transmitted signal, which is due

to channel imperfections, noise and interference from other sources.The receiver has the

task of operating on the received signal so as to reconstruct a recognizable form of the

original message signal and to deliver it to the user destination.

Communication Systems are divided into 3 categories:

1. Analog Communication Systems are designed to transmit analog information using

analog modulation methods.

2. Digital Communication Systems are designed for transmitting digital information using

digital modulation schemes, and

3. Hybrid Systems that use digital modulation schemes for transmitting sampled and

quantized values of an analog message signal.

The figure 1.2 shows the functional elements of a digital communication system. Source of Information:1. An

2. Digital Information Sources.

Analog Information Sources Microphone actuated by a speech, TV Camera scanning a scene, contin

Digital Information Sources These are teletype or the numerical output of computer which consists of a se

An Analog information is transformed into a discrete information through the process of sampling and quant

Digital Communication System

SOURCE ENCODER / DECODER:

The Source encoder ( or Source coder) converts the input i.e. symbol sequence

into a binary sequence of 0s and 1s by assigning code words to the symbols in the

input sequence. For eg. :-If a source set is having hundred symbols, then the

number of bits used to represent each symbol will be 7 because 2 7=128 unique

combinations are available. The important parameters of a source encoder are

block size,

code word lengths, average data rate and theefficiency of the coder (i.e. Actual

output data rate compared to the minimum achievable rate)

At the receiver, the source decoder converts the binary output of the channel

decoder into a symbol sequence. The decoder for a system using fixed length

code words is quite simple, but the decoder for a system using variable

words will be very complex.

thatbandwidthrequiredfortransmis

transmittinginformation,so

minimized. Based on the probability of the symbol code word is assigned. Higher the

probability, shorter is the codeword. Ex: Huffman coding.

CHANNEL ENCODER / DECODER:

Error control is accomplished by the channel coding operation that consists of systematically addin

These extra bits do not convey any information but helps the receiver to detect and /

or correct some of the errors in the information bearing bits. There are two

methods of channel coding:

1. Block Coding: The encoder takes a block of k information bits from

the source encoder and adds r error control bits, where r is dependent on k and

error control capabilities desired.

2. Convolution Coding: The information bearing message stream is encoded in

a continuous fashion by continuously interleaving information bits and error

control bits.

The Channel decoder recovers the information bearing bits from the coded binary

stream. Error detection and possible correction is also performed by the channel

decoder.

The important parameters of coder / decoder are: Method of coding, efficiency,

error control capabilities and complexity of the circuit.

MODULATOR:

The Modulator converts the input bit stream into an electrical waveform

suitable for transmission over the communication channel. Modulator can be

effectively used to minimize the effects of channel noise, tomatch the

frequency

spectrumoftransmittedsignalwithchannel characte

capability to multiplex many signals.

DEMODULATOR:

The extraction of the message from the information bearing waveform produced by the modulation

is bit stream. The important parameter is the method of demodulation.

CHANNEL:

The Channel provides the electrical connection between

The different channels are: Pair of wires, Coaxial cable, Optical fibre,

Radio channel, Satellite channel or combination of any of these.

response, the signal often suffers amplitude and phase distortion as it travels over

the channel. Also, the signal power decreases due to the attenuation of the

channel. The signal iscorrupted by unwanted, unpredictable electrical signals

referred to as noise.

The important parameters of the channel are Signal to Noise power Ratio(SNR),

usable bandwidth, amplitude and phase response and the statistical properties of

noise.

Sometimes, a comparison between two digital transmission systems is needed.

There are many parameters that can be used to compare between digital

transmission systems, but some of the most important parameters of a digital

transmission system are:

Transmission

Rate (measured in bits per second): This is a measure of the

number of bits that can be transmitted over the communication channel per unit time.

Bandwidth Requirements

(measured in Hz): This is a measure of the

spectrum that the communication system requires to transmit the information at the desired transm

in percentages): This represents the percentage of

bits that are in error relative to the overall number of bits that are transmitted by the communicat

T

ransmission Power (or Bit Energy) (measured in Watts (or Jules/bit)): This

represents the amount of power of the transmitted signal that would be required to

achieve a particular desired error probability.

represents that amount of money that a manufacturer will have to spend to build

the system and the amount of money that a user will have to pay to use the

system.

the following factors:

1. Amount of energy in each digital bit (or pulse): Generally, the more energy a

digital bit (or pulse) has, the better the performance that the system will have.

2. The distance between the transmitter and receiver: Because energy is spread or

attenuated as it travels over the channel and more noise is added due to the

existence of more noise sources over long channels, generally the longer the path

that the digital transmitted signal has to travel, the worse the performance that the

system will have. However, you do not always have control over the distance

between the transmitted and receiver.

3. Amount of noise that is added to the signal: Certainly, the less the noise that is

added to the transmitted signal, the better the performance of the communication

system. We usually have limited control over the added noise.

4. Bandwidth of the transmission channel: By using larger bandwidth, we can

either transmit at a higher transmission bit rate while keeping the same probability o

bit error, or we can transmit at the same transmission bit rate but reduce the probab

communication system, the better the performance it will have.

The effect of distortion, noise and interference is less in a digital communication system. This is be

Regenerative repeaters can be used at fixed distance along the link, to identify and

regenerate a pulse before it is degraded to an ambiguous state.

3. Digital circuits are more reliable and cheaper compared to analog circuits.

4. The Hardware implementation is more flexible than analog hardware because of

the use of microprocessors, VLSI chips etc.

5. Signal processing functions like encryption, compression can be employed to

maintain the secrecy of the information.

6. Error detecting and Error correcting codes improve the system performance by

reducing the probability of error.

7. Combining digital signals using TDM is simpler than combining analog signals

using FDM. The different types of signals such as data, telephone, TV can be

treated as identical signals in transmission and switching in a digital

communication system.

8. We can avoid signal jamming using spread spectrum

technique. Disadvantages of Digital Communication:

1. Large System Bandwidth:- Digital transmission requires a large system

bandwidth to communicate the same information in a digital format as compared

to analog format.

2. System Synchronization:- Digital detection requires system synchronization

whereas the analog signals generally have no such requirement.

The modulation and coding used in a digital communication system depend on the

BANDWIDTH and POWER. In addition the other character

interference.

channels, coaxial cables, optical fibers, microwave radio, and satellite channels.

Telephone channel: It is designed to provide voice grade communication. Also good for data comm

about 30db, and approximately linear response.

For the transmission of voice signals the channel provides flat amplitude

response. But for thetransmission of data and image transmissions, since the

phase delay variations are important an equalizer is used to maintain the flat

amplitude response and a linear phase response over the required frequency band.

Transmission rates upto16.8 kilobits per second have been achieved over the

telephone lines.

Coaxial Cable: The coaxial cable consists of a single wire conductor centered

inside an outer conductor, which is insulated from each other by a dielectric. The

main advantages of the coaxial cable are wide bandwidth and low

external

interference. But closely spaced repeaters are required. With repeaters spaced at

1km intervals the data rates of 274 megabits per second have been achieved.

Optical Fibers: An optical fiber consists of a very fine inner core made of silica

glass, surrounded by a concentric layer called cladding that is also made of glass.

The refractive index of the glass in the core is slightly higher than refractive index

of the glass in the cladding. Hence if a ray of light is launched into an optical fiber

at the right oblique acceptance angle, it is continually refracted into the core by

the cladding. That means the difference between the refractive indices of the core

and cladding helps guide the propagation of the ray of light inside the core of the

fiber from one end to the other.

Compared to coaxial cables, optical fibers are smaller in size and they offer

higher transmission bandwidths and longer repeater separations.

Microwave radio: A microwave radio, operating on the line-of-sight link, consists basically of a trans

30 GHz.

reliable and prov

an uplink from ground station, and a down link to another ground station. Both

link operate at microwave frequencies, with uplink the uplink frequency higher

than the down link frequency. In general, Satellite can be viewed as repeater in

the sky. It permits communication over long distances at higher bandwidths and

relatively low cost.

Bandwidth:

Bandwidth is simply a measure of frequency range. The range of frequencies

contained in a composite signal is its bandwidth. The bandwidth is normally a

difference between two numbers. For example, if a composite signal contains

frequencies between 1000 and 5000, its bandwidth is 5000 - or 4000. If a range of

2.40 GHz to 2.48 GHz is used by a device, then the bandwidth would be 0.08

GHz (or more commonly stated as 80MHz).It is easy to see that the bandwidth we

define here is closely related to the amount of data you can transmit within it - the

more room in frequency space, the more data you can fit in at a given moment.

The term bandwidth is often used for something we should rather call a data rate,

as in my Internet connection has 1 Mbps of bandwidth, meaning it can

transmit data at 1

Sampling

A message signal may originate from a digital or analog source. If the message signal is analog in n

Sampling operation is performed in accordance with the sampling theorem.

Part - I If a signal x(t) does not contain any frequency component beyond W Hz,

then the signal is completely described by its instantaneous uniform samples with

sampling interval (or period ) of Ts< 1/(2W) sec.

Part II The signal x(t) can be accurately reconstructed (recovered) from the set

of uniform instantaneous samples by passing the samples sequentially through an

ideal (brick-wall) lowpass filter with bandwidth B, where W B < fs W and fs

= 1/(Ts).

As the samples are generated at equal (same) interval (Ts) of time, the process of

sampling is called uniform sampling. Uniform sampling, as compared to any non-

uniform sampling, is more extensively used in time-invariant systems as the

theory of uniform sampling (either instantaneous or otherwise) is well developed

and the techniques are easier to implement in practical systems.

no practical sampling device can actually generate truly instantaneous samples (a

sampling pulse should have non-zero energy). However, this is not a deterrent in

instantaneous sampling is sufficient for most practical systems. To contain our discussion on Nyquis

uniform samples {x(nTs)} may be represented as,

where x(nTs) = x(t) t =nTs , (t) is a unit pulse singularity fun

Conceptually, one may think that the continuous-time signal x(t) is multiplied by

an (ideal) impulse train to obtain {x(nTs)} as in equation(1) can be rewritten as,

x(t), i.e.

+

X ( f )= x (t ).exp( j 2 ft )dt . (3)

Now, from the theory of Fourier Transform, we know that the F.T of (t- nTs),

the impulse train in time domain, is an impulse train in frequency domain:

If Xs(f) denotes the Fourier transform of the energy signal xs(t), we can write

using Eq. (1.2.4) and the convolution property:

= fs.X(f)* (f- nfs)

+

= fs. nfs )d = fs. X().(f- nfs-)d = fs. X(f- nfs)

X(). (f .

(5)

[By sifting property of (t) and considering (f) as an even function, i.e. (f) = (-

f)]

theorems as stated above and also helps to appreciate their practical implications.

Let us note that while writing Eq.(5), we assumed that x(t) is an energy signal so

that its Fourier transform exists.

Further, the impulse train in time domain may be viewed as a periodic singularity

function with almost zero (but finite) energy such that its Fourier Transform [i.e. a

train of impulses in frequency domain] exists.

With this setting, if we assume that x(t) has no appreciable frequency component

greater than W Hz and if fs > 2W, then Eq.(1.2.5) implies that Xs(f), the Fourier

Transform of the sampled signal xs(t) consists of infinite number of replicas of

X(f),

centered at discrete frequencies n.fs, - < n < and scaled by a constant fs= 1/Ts

(Fig. 1.3).

xs(t) is infinite while the spectrum of x(t) appears in a periodic manner, c

Now, Part I of the sampling theorem is about the condition fs > 2.W i.e. (fs W) >

W and ( fs + W) < W. As seen from Fig. 1.3, when this condition is satisfied, the spectra of xs(t),

of x(t) is present in xs(t) without any distortion. This implies that xs(t), the

appropriately sampled version of x(t), contains all information about x(t) and

thus represents x(t).

The second part of Nyquists theorem suggests a method of recovering x(t) from

its sampled version xs(t) by using an ideal lowpass filter. As indicated by dotted

lines in Fig. 1.3, an ideal lowpass filter (with brick-wall type response) with a

bandwidth W

B < (fs W), when fed with xs(t), will allow the portion of Xs(f), centered at f

= 0 and will reject all its replicas at f = n fs, for n 0. This implies that the shape

of the continuous-time signal xs(t), will be retained at the output of the ideal

filter.

Hartley Shannon Law

The theory behind designing and analyzing channel codes is called Shannons

noisy channel coding theorem. It puts an upper limit on the amount of information

you can send in a noisy channel using a perfect channel code. This is given by the

following equation:

where C is the upper bound on the capacity of the channel (bit/s), B is the bandwidth

of the channel (Hz) and SNR is the Signal-to-N ise ratio (unitless).

Bandwidth-S/N Tradeof

The expression of the channel capacity of the Gaussian channel makes intuitive sense:

changes in the information signal, thereby increasing the information rate.

As S/N increases, one can increase the information rate while stillpreventing errors due to noise.

rate is possibleirrespective of bandwidth.

Thus we may trade off bandwidth for SNR. For example, if S/N = 7 and B =

4kHz, then the channel capacity is C = 12 10 3 bits/s. If the SNR increases to S/N

= 15 and B is decreased to 3kHz, the channel capacity remains the same.

However, as B tends to 1, the channel capacity does not become infinite since,

with an increase in bandwidth, the noise power also increases. If the noise power

spectral density is /2, then the total noise power is N = B, so the Shannon-

Hartley law becomes

Pulse Code Modulation

Introduction

In the simplest model of a telephone speech communication there is a direct, dedicated, physical c

In Pulse Amplitude Modulation (PAM), the unmodified electrical signal is not sent on to the connect

Because each sample is very short (~4s) there is a lot of time between samples

(~121s). Samples from other conversations are put into this spare time.

Usually the samples from 32 separate conversations are put on to a single line.

This process is called Time Division Multiplexing (TDM).

communications network. In order to reconstruct the original analogue signal the

only information the receiver needs to have about a sample is its amplitude, but if

this is distorted then all information about the sample has been lost. To overcome

this problem, the pulse is not transmitted directly, instead its amplitude is

measured and converted into an 8 binary number - a sequence of 1s and 0s. At the

receiver end, the receiver merely needs to detect if a 1 or a 0 has been received so

that it can still recover the amplitude of a PAM pulse even if the 1s and 0s used to

describe it have been distorted.

The process of converting the amplitude of each pulse into a stream of 1s and 0s

is called Pulse Code Modulation (PCM)

Note that the process of PAM and PCM (but without the use of TDM) is

essentially used to store music and speech on CDs, but with a higher sample rate,

more bits per sample and complex error correction mechanisms.

discrete instants. It converts a continuous-time signal to a discrete-

time signal.

Quantizing Representing the sampled values of the amplitude by a finite set of

levels. It converts a continuous-amplitude sample to a discrete-

amplitude sample.

Encoding Designating each quantized level by a (binary) code.

Sampling and quantizing operations transform an analogue signal to a digital signal.

modulation methods.

The quantizing and encoding operations are usually performed in the same circuit

at the transmitter, which is called an Analogue to Digital Converter (ADC). At the

receiver end the decoding operation converts the (8 bit) binary representation of

the pulse back into an analogue voltage in a Digital to Analogue Converter (DAC)

Pulse Code Modulation

Pulse Code Modulation (PCM) is an extension of PAM wherein each analogue

sample value is quantized into a discrete value for representation as a digital code

word.

Thus, as shown in Fig. 2.1 a PAM system can be converted into a PCM system by

adding a suitable analogue-to-digital (A/D) converter at the source and a digital-

to- analogue (D/A) converter at the destination.

M o d u la to r

A n a lo

PCM

gue P a ra lle l to S e rDiaigl ita l P u ls e

A to D B in a C o n v e r te r G e n e ra to r u tp u t

O

In p uS a m p le r C o n v e r te r

ry C o

t

der

D e m o d u la to r

P C M S e r ia l to P a ra lle l A n a lo g u e

In p uCt o n v e r te r D to A

C o n v e r te r O u tpt

LPF

PCM is a true digital process as compared to PAM. In PCM the speech signal is

converted from analogue to digital form.

PCM is standardised for telephony by the ITU-T (International Telecommunications Union - Telecoms

frequency response of the handset microphone has a sharp roll-off from 3.4 kHz.

In quantization the levels are assigned a binary codeword. All sample values

falling between two quantization levels are considered to be located at the centre

of the quantization interval. In this manner the quantization process introduces a

certain amount of error or distortion into the signal samples. This error known as

quantization noise, is minimised by establishing a large number of small

quantization intervals. Of course, as the number ofquantization intervals increase,

so must the number or bits increase to uniquely identify the quantization intervals.

For example, if an analogue voltage level is to be converted to a digital system

with 8 discrete levels or quantization steps three bits are required. In the ITU-T

version there are 256 quantization steps, 128 positive and 128 negative, requiring

8 bits. A positive level is represented by having bit 8 (MSB) at 0, and for a

negative level the MSB is 1.

Quantization

analog samples to a PCM code. In quantization, an analog sample with an

amplitude that may take value in a specific range is converted to a digital sample

with an amplitude that takes one of a specific predefined set of quantization

values. This is performed by dividing the range of possible values of the analog

samples into L different levels, and assigning the center value of each level to any

sample that falls in that quantization interval. The problem with this process is

that it approximates the value of an analog sample with the nearest of the

quantization values. So, for almost all samples, the quantized samples will differ

from the original samples by a small amount. This amount is called the

quantization error. To get some idea on the effect of this quantization error,

quantizing audio signals results in a hissing noise similar to what you would hear

when play a random signal.

Assume that a signal with power Psis to be quantized using a quantizer with L =

2n levels ranging in voltage from mp tomp as shown in the fig. 2.2

value mp

v

0T Ts 3 Ts 4 Ts 5 Ts

s 2

L = 2n

L levels t

0

n bits

mp

Q uantizer Input Sam ples x

Q uantizer O utput Sam ples x q

Fig. 2.2

We can define the variable v to be the height of the each of the L levels of the

quantizer as shown above. This gives a value of v equal to

2m

v

p

.

L

Therefore, for a set of quantizers with the same mp, the larger the number of levels

of a quantizer, the smaller the size of each quantization interval, and for a set of

quantizers with the same number of quantization intervals, the larger mp is the

larger the quantization interval length to accommodate all the quantization range.

similar to the red line in the following figure. Note that as long as the input is

within the quantization range of the quantizer, the output of the quantizer

represented by the red line follows the input of the quantizer. When the input of

the quantizer exceeds the range of mp tomp, the output of the quantizer starts to

deviate from the input and the quantization error (difference between an input

and the corresponding output sample) increases significantly.

Quan

tizer

Outp

ut xq

x

xq

v/2

v/2

v/2

Quantizer

v/2

Input x

v v v

v v v v v v

/2

v/2

v/2

v/2

mp

Fig. .2.3

Now let us define the quantization error represented by the difference between the

input sample and the corresponding output sample to be q, or

q x

x q.

Plotting this quantization error versus the input signal of a quantizer is seen next.

Notice that the plot of the quantization error is obtained by taking the difference

between the blow and red lines in the above Fig. 2.3

Quantization Error q

v/2

Quantizer

v/2 v v v Input x

v v v v v

mp

Fig. 2.4

It is seen from the Fig 2.4 that the quantization error of any sample is restricted

between v/2 andv/2 except when the input signal exceeds the range of

quantization of mp to mp.

Quantization confined to the range (-mp, +mp ). This range (2mp) is

divided into L levels, each of step size , given by

O u tp u t

= 2 mp / L

-m p

midpoint of the interval in which it lies. The

+m p input/output characteristics of a uniform quantizer is

In p u t

shown in Fig. 2.5

Fig. 2.5

Companding

-High amplitude analog signals are compressed prior to txn. and then expanded in

the receiver

-Higher amplitude analog signals are compressed and Dynamic range is improved

-Early PCM systems used analog companding, where as modern systems use

digital companding.

Analog companding

2.7 PCM system with analog companding

--In the transmitter, the dynamic range of the analog signal is compressed, and

then converted o a linear PCM code.

--In the receiver, the PCM code is converted to a PAM signal, filtered, and then

expanded back to its original dynamic range.

-- There are two methods of analog companding currently being used that closely

approximate a logarithmic function and are often called log-PCM codes.

2) A-law

-law companding

ln 1 inV

mV

ax V

max

V

ln 1

out

= parameter used tio define the amount of compression (unitless) Vout = compressed output am

A-law companding

where y =

A | x| 1

, 0 | x | x = Vin

Vout y 1 log A /

A

Vmax 1 log( A | x |)

| x | 1

1

,

1 log A A

Digital Companding:

--With digital companding, the analog signal is first sampled and converted to a

linear PCM code, and then the linear code is digitally compressed.

-- In the receiver, the compressed PCM code is expanded and then decoded back

to analog.

-- The most recent digitally compressed PCM systems use a 12- bit linear PCM

code and an 8-bit compressed PCM code.

%erro

12-bit

The output from the analogue to digital converter (ADC) has n parallel bits. In the case of telephony

the signed bit is 1. The remaining 7 bits are used to code the sample value. The

ITU- T define a look up table which allocates a particular binary code to each

quantified A-law value.

The line coding which is used assigns opposite polarities to successive 1s. This

eliminates any DC voltage on the line, and reduces the inter symbol interference

if adjacent bits are 1. If there is silence on the PCM channel then the measured

samples will be 0 Vrms and the output of the DAC will be 1000 0000. A stream of

all zeros is not desirable on an active channel because

it is difficult to recover the clock signal from the incoming signal.

The coding system HDB3 is used and was developed to eliminate all zeros, and to

assign opposite polarities to successive 1s.

positive and negative pulses).

In AMI positive and negative pulses (of equal amplitude) are used for alternative

symbols 1. No pulse is used for symbol 0. In either case the pulse returns to 0

before the end of the bit interval. This eliminates any DC on the line.

HDB3 encoding rules follow those for AMI, except that a sequence of four

consecutive 0's are encoded using a special "violation" bit. The 4 th 0 bit is given

the same polarity as the last 1-bit which was sent using the AMI encoding rule.

This prevents long runs of 0's in the data stream which may otherwise prevent a

receiver from tracking the centre of each bit. By introducing violations, extra

"edges" are introduced, enabling a Digital PLL to reliably reconstruct the clock

signal at the receiver. The HDB3 is transparent to the sequence of bits being

transmitted (i.e. whatever data is sent, the Digital PLL can reconstruct the data

and extract the bits at the receiver).

than four zeros encodes as B00V. The value of B is assigned + or - alternately

throughout the bit stream.

B BBBBBBB

1010 1010 = + 0 - 0 + 0 - 0

B 0 B 0 B 0 B 0

1000 0001 + 0 0 0 + 0 0 -

= B 0 0 0 V 00B

1000 0110 = + 0 0 0 + - +0

= B 0 0 0 V BB0

PCM Timing and Synchronisation

The PCM receiver must be able to identify the start and finish of each full

sampling sequence and to identify each bit position. The sampling clock needs to

be either sent to, or regenerated at, the receiving side to determine when each full

sequence of sampling begins and ends. The data clock is also needed to determine

exactly when to read each bit of information.

d a ta c lo c k 6 4 k b i t / s

15.625 s

fra m e c lo c k 125

A PCM channel is sampled at 8,000 Hz or

s

once every 125 s. If there is one channel or

30 TDM channels the sampling period is

fixed at 125 s and this period is known as a

frame.

B1 B2

1 0

B3 B4 B5 B6 B7 B8 B1 Therefore the frame clock must have a

1 0 0 1 1 1 ?

period of 125 s. The rising edge of the

frame clock

Fig. 2.9 informs the receiver that the next bit will be

Bit 1 of a new sample. The falling edge of

the

data clock informs the receiver that it must read the data bit.

When the bit stream is transmitted along a line the pulses become distorted and

the rise and fall times become significant. Ideally, a 1 will be high for 15.625

s. In practice the pulse may only be above the high threshold for a few s so it

is very important that the bit is read within a certain time limit of the clock pulse.

The simplest way to synchronise a PCM sender to a PCM receiver is to send the

clock signals on different circuits to the data This would be done in a self-

contained system such as private branch exchange (PBX). Telephony is full

duplex so that there is a coder and a decoder at each port, but each would use the

same clock.

which allows the receiver to extract the clocks from the PCM signal. In this case

the receiver will have free running clocks that lock (using a PLL) to the phase

and frequency of the transitions in the data stream. The line-coding scheme

ensures that there is a transition for every data bit.

PCM transmits the absolute value of the signal for each frame. Instead we can

transmit information about the difference between each sample. The two main

differential coding schemes are:

Delta Modulation

Differential PCM and Adaptive Differential PCM (ADPCM)

Delta Modulation

Delta modulation converts an analogue signal, normally voice, into a digital

signal.

g ra n u la r n

1 the PCM process. Then the sample is

1 0 0 1

1 1 0

compared with the previous sample.

0 0

1

The result of the comparison is

0 0 0

quantified using a one bit coder. If the

0

sample is greater than the previous

sample a 1 is generated. Otherwise a 0

is generated.

The advantage of delta modulation

over

PCM is its simplicity and lower cost. But the noise performance is not as

Fig. 2.10

good as PCM.

To reconstruct the original from the quantization, if a 1 is received the signal is increased by a step

dx(t)/dt q /T = q * fs

where: x(t) = input signal, q = step size, T = period between samples, fs = sampling frequency

Assume that the input signal has maximum amplitude A and maximum frequency

F. The most rapidly changing input is provided by x(t) = A * sin (2 * * F * t).

For this dx(t)/dt = 2 * * F * A * sin (2 * * F *

t). This slope has a maximum value of 2 * * F * A

Overload occurs if 2 * * F * A> q

* fs To prevent overload we require q * fs> 2 * * F

*A

Example A = 2 V, F = 3.4 kHz, and the signal is sampled 1,000,000 times

per second, requires q > 2 * 3.14 * 3,400 * 2 /1,000,000 V > 42.7 mV

Granular noise occurs if the slope changes more slowly than the step size. The

reconstructed signal oscillates by 1 step size in every sample. It can be reduced by

decreasing the step size. This requires that the sample rate be increased. Delta

Modulation requires a sampling rate much higher than twice the bandwidth. It

requires oversampling in order to obtain an accurate prediction of the next input,

since each encoded sample contains a relatively small amount of information.

Delta Modulation requires higher sampling rates than PCM.

D iffe re n t ia Encod

A n a lo g to r Q u a n t is e d D iffe

B a n d L im itin

u e In erEnod re n c e S a

g F i l te r

put er m p le s

+

- ADC

Acc um

u la to r D

AC

Fig. 2.11

waveform. In DPCM the differences between samples are quantized with fewer

bits that would be used for quantizing an individual amplitude sample. The

sampling rate is often the same as for a comparable PCM system,

T recommendations G.721 and G.726. The method uses 32,000 bits/s per voice

channel, as compared to standard PCMs 64,000 bits/s. Four bits are used to

describe each sample, which represents the difference between two adjacent

samples. Sampling is 8,000 times a second. It makes it possible to reduce the bit

flow by half while maintaining an acceptable quality. While the use of ADPCM

(rather than PCM) is imperceptible to humans, it can significantly reduce the

throughput of high speed modems and fax transmissions.

The principle of ADPCM is to use our knowledge of the signal in the past time to

predict the signal one sample period later, in the future. The predicted signal is

then compared with the actual signal. The difference between these is the signal

which is sent to line - it is the error in the prediction. However this is not

done by making

comparisons on the incoming audio signal - the comparisons are done after PCM

coding.

produce a code word. This code word is manipulated to produce the predicted

code word for the next sample. The new predicted code word is compared with

the code word of the second sample. The result of this comparison is sent to line.

Therefore we need to perform PCM before ADPCM.

The ADPCM word represents the prediction error of the signal, and has no

significance itself. Instead the decoder must be able to predict the voltage of the

recovered signal from the previous samples received, and then determine the

actual value of the recovered signal from this prediction and the error signal, and

then to reconstruct the original waveform.

ADPCM is sometimes used by telecom operators to fit two speech channels onto a

single 64 kbit/s link. This was very common for transatlantic phone calls via satellite up until a few

Delta modulation, like DPCM is a predictive waveform coding technique and can

be considered as a special case of DPCM. It uses the simplest possible quantizer,

namely a two level (one bit) quantizer. The price paid for achieving the simplicity

of the quantizer is the increased sampling rate (much higher than the Nyquist rate)

and the possibility of slope-overload distortion in the waveform reconstruction, as

explained in greater detail later on in this section.

In DM, the analog signal is highly over-sampled in order to increase the adjacent

sample correlation. The implication of this is that there is very little change in two

adjacent samples, thereby enabling us to use a simple one bit quantizer, which like

in DPCM, acts on the difference (prediction error) signals.

In its original form, the DM coder approximates an input time function by a series

of linear segments of constant slope. Such a coder is therefore referred to as a

Linear (or non-adaptive) Delta Modulator (LDM). Subsequent developments have

resulted in delta modulators where the slope of the approximating function is a

variable. Such coders are generally classified under Adaptive Delta Modulation

(ADM) schemes. We use DM to indicate either of the linear or adaptive variety.

Deltamodulation principleofoperation

Deltamodulationwasintroducedinthe1940sasasimplifiedformofpulsecodemodul

atio n(PCM),whichrequiredadifficult-to-implementanalog-to-

digital(A/D)converter.

Theoutputofadeltamodulatorisabitstreamofsamples,atarelatively

highrate(eg,100kbit/sor

moreforaspeechbandwidthof4 kHz)thevalueofeachbitbeing

determinedaccordingas

towhethertheinputmessagesampleamplitudehasincreasedordecreasedrelativetoth

epr evioussample.Itisan exampleofdifferentialpulsecodemodulation(DPCM).

Blockdiagram

Theoperationofadeltamodulatoristoperiodicallysampletheinputmessage,tomak

eac omparisonofthecurrentsamplewiththatprecedingit,andtooutputasinglebit

would requirea sample-and-hold type circuit.

DeJager(1952)hitonanideafordispensingwiththeneedforasampleandholdcircuit.

Her

easonedthatifthesystemwasproducingthedesiredoutputthenthisoutputcouldbese

ntb acktotheinputandthetwoanalogsignalscomparedinacomparator.Theoutput

isa delayedversionoftheinput,andsothecomparison

isineffectthatofthecurrentbitwiththepreviousbit,asrequiredbythedeltamodulatio

npri nciple.

Figure2.13illustratesthebasicsysteminblockdiagramform,andthiswillbethemod

ulat oryouwill bemodelling.

Thesystemisintheformofafeedbackloop.Thismeansthatitsoperationisnotn ecessaril

yobvious,anditsanalysisnon-

trivial.Butyoucanbuildit,andconfirmthatitdoesbehaveinthe manner

adelta

modulatorshould.

Thesystemisacontinuoustimetodiscretetimeconverter.Infact,itisaformofanalogt

odi gitalconverter,andis thestarting pointfrom which

more sophisticateddeltamodulatorscanbe

developed.

Thesamplerblockisclocked.Theoutputfromthesamplerisabipolarsignal,intheblockd

iagrambeingeither Vvolts.Thisisthedeltamodulatedsignal,thewaveformof

whichisshowninFigure 2.Itisfedback,inafeedbackloop,viaanintegrator,toasummer.

Theintegratoroutputisasawtooth-likewaveform,alsoillustratedinFigure

Figure 2.15:integrator output superimposed on the messagewith the delta modulated signal below

Thesawtoothwaveformissubtractedfromthemessage,alsoconnectedtothesum

mer, andthe difference-anerror signal-isthe signalappearingatthe

summeroutput.

Anamplifierisshowninthefeedbackloop..Thiscontrolstheloopgain.Inpracticeitma

ybe

aseparateamplifier,partoftheintegrator,orwithinthesummer.Itisusedotonctrolthe

size

oftheteethofthesawtoothwaveform,inconjunctionwiththeintegratortimeconsta

nt.

WhenanalysingtheblockdiagramofFigure

2.13itisconvenienttothinkofthesummerhavingunitygainbetweenbothinputsandth

eou tput.Themessagecomes in

at

afixedamplitude.Thesignalfromtheintegrator,whichisasawtoothapproximationto

the message,isadjustedwiththeamplifiertomatchitasclosely aspossible.

stepsizecalculation

InthedeltamodulatorofFigure2.13theoutputoftheintegratorisasawtooth-

likeapproximationtotheinputmessage.Theteethofthesaw

mustbeabletorise(orfall)fastenoughtofollowthemessage.Thustheintegratortimec

ons tantisanimportantparameter.

Foragivensampling(clock)ratethestepslope(volt/s)determinesthesize(volts)ofth

este p withinthesamplinginterval.

Supposetheamplitudeof

therectangularwavefromthesamplerisV volt.Forachangeofinputsampleto

theintegratorfrom(say)negativeto

positive,thechangeofintegrator output will be,

afteraclockperiodT:

AnswerTutorialQuestions1and

2beforeattemptingtheexperiment.Youcanlatercheckyouranswerbymeasurem

ent.

slopeoverloadandgranularity

ThebinarywaveformillustratedinFigure2.15isthesignaltransmitted.Thisisthedelt

amo dulatedsignal.

Theintegralofthebinarywaveformisthesawtoothapproximationto themessage.

IntheexperimententitledDeltademodulation(inthisVolume)youwillseethatthissa

wto othwave istheprimaryoutputfromthedemodulatoratthereceiver.

Lowpassfilteringofthesawtooth(fromthedemodulator)givesabetterapproximatio

ntot

hemessage.Buttherewillbeaccompanyingnoiseanddistortion,productsoftheappr

oxi mationprocessatthemodulator.

Theunwantedproductsofthemodulationprocess,observedatthereceiver,areof

two kinds.Thesearedue toslopeoverload, and granularity.

slopeoverload

Thisoccurswhenthesawtoothapproximationcannotkeepupwiththerate

-of- changeoftheinput signalinthe regionsofgreatestslope.

Thestepsizeisreasonableforthosesectionsofthesampledwaveformofsmallslope,

butt heapproximationispoorelsewhere.Thisisslopeoverload,duetotoosmalla

step.

Slopeoverloadisillustrated inFigure2.16.

slo p e o v e rlo a d

Figure2.16:slopeoverload

Toreducethepossibilityofslopeoverloadthestepsizecanbeincreased(forthesamesa

mp ling rate).This isillustratedin

Figure

2.17.Thesawtoothisbetterabletomatchthemessageinthe regionsofsteep slope.

An

alternativemethodofslopeoverloadreductionis

toincreasethesamplingrate.ThisisillustratedinFigure

2.18,wheretheratehasbeenincreasedbyafactorof2.4times, but thestep isthe same

size asinFigure2.15.

tim e

1.4 Granularnoise

ReferbacktoFigure 2.16.Thesawtoothfollowsthemessagebeingsampledquitewellintheregionsofsmalls

Thedegradationshowsup,a

ty.

1.5 noiseanddistort

Thereisaconflictbetweentherequirementsforminimizationofslopeoverloadandthe

gra

nularnoise.Theonerequiresanincreasedstepsize,theotherareducedstepsize.You

shouldrefertoyourtextbook

formorediscussion

ofwaysandmeansofreachingacompromise.Youwillmeetanexampleintheexperime

nte

ntitledAdaptivedeltamodulation(inthisVolume).Anoptimumstepcanbedetermine

dby minimizingthequantizingerroratthesummer output, or

thedistortionatthe

demodulatoroutput.

The Operation Theory of ADM Modulation

when the input audio signal frequency exceeded the limitation of delta modulator,

i.e.

Then this situation will produce the occurrence of slope overload and cause signal

distortion. However, the adaptive delta modulation (ADM) is the modification of

delta modulation to improve the disadvantage of the occurrence of slope overload.

Figure 2.20 is the block diagram of ADM modulator. In figure 2.20, we can see

that the delta modulator is comprised by comparator, sampler and integrator, then

the slope controller and the level detect algorithm comprise a quantization level

adjuster, which can control the gain of the integrator in the delta modulator. ADM

modulator is the modification of delta modulator, therefore, due to the delta

modulator has the problem of slope overload at low and high frequencies. The

reason is the magnitude of the (t) of delta modulator is fixed, i.e. the increment

of or - is unable to follow the variation of the slope of the input signal. When

the variation of the slope of the input signal is large, the magnitude of (t) still

can increase by following the variation, then this situation will not occur the

problem of slope overload. On the other hand, there is another technique, which is

known as continuous variable slope delta (CVSD) modulation. This technique is

commonly used in Bluetooth application. CVSD modulator is also the

modification of delta modulator, use to improve the occurrence of slope overload.

The different between the CVSD and ADM modulators are the quantization level

adjuster A. ADM modulator is discrete values and the quantization level adjuster

of CVSD modulator is continuous. Simply, the quantization value of ADM

modulator is the variation of digital, such as the quantization values of +1, +2, +3,

-2, -3 and so on. As for CVSD modulator, the quantization value is the variation

of analog, such as the quantization values of +1,

+1.1, +1.2, -1.5, -0.3, -0.9 and so on.

Fig. 2.20 The Operation Theory of ADM Modulation

UNIT - II

Modulation is defined as the process by which some characteristics of a carrier

is varied in accordance with a modulating wave. In digital communications, the

modulating wave consists of binary data or an M-ary encoded version of it and

the carrier is sinusoidal wave.

Different Shift keying methods that are used in digital modulation techniques are

Frequency shift keying [FSK]

Phase shift keying [PSK]

Fig 3.1 Different modulations

1. ASK[Amplitude Shift Keying]:

S (t) 2E b

Cos2 f t for symbol 1

1

Tb 1

S (t) 2Eb Cos2f t for symbol 1

1 1

Tb

S 2(t) 2Eb Cos2f 2t for symbol 0

Tb

3. PSK[Phase Shift Keying]:

In a binary PSK system the pair of signals S1(t) and S2(t) are

used to represent binary symbol 1 and 0 respectively.

S1 (t) 2Eb Cos2fc t --------- for Symbol 1

Tb

2Eb 2Eb

S2 (t) Cos(2fc t ) Cos2fc t ------- for Symbol 0

Tb Tb

Digital Modulation Technique

(m) = 2 (m) = 2

* FSK M-ary FSK M-ary QAM * FSK M-ary FSK

* PSK M-ary PSK * DPSK M-ary DPSK

(QPSK)

Coherent Binary PSK:

Non Return to

Zero Level Product

Encoder Modulator

Binary Binary PSK Signal

Data Sequence

(t) 2 Cos2f t

1 c

Tb

Tb

x(t) dt x1 Decision Device Choose 1 if x1>0

0

Choose 0 if x1<0

Correlator

1 (t) Threshold = 0

In a Coherent binary PSK system the pair of signals S1(t) and S2(t) are used

to represent binary symbol 1 and 0 respectively.

2Eb

S1 (t) Cos2fc t --------- for Symbol 1

Tb

2Eb 2Eb

S2 (t) Cos(2fc t ) Cos2fc t ------- for Symbol 0

Tb Tb

E E

b0 b1

Where Eb= Average energy transmitted per bit Eb

2

In the case PSK, there is only one basic function of Unit energy which is given

by

(t) 2 Cos2f t 0tT

1

T c b

b

Symbol

S (t) E (t) 0 t T for 1

1 b 1 B

T

2 b 1

B

is one dimensional (N=1) with two message points (M=2)

Tb

Eb

S11 S1 (t)1 (t) dt

0

Tb

0

The message point corresponding to S1(t) is located at S11 Eb and S2(t) is located at S21 Eb

To generate a binary PSK signal we have to represent the input binary sequence in

polar form with symbol 1 and 0 represented by constant amplitude levels of

Eb &respectively.

E

b This signal transmission encoding is performed by a NRZ

level encoder.The resulting binary wave [in polar form] and a sinusoidal carrier 1 (t)

nc

[whose frequency] fare

c applied to a product modulator. The desired BPSK wave

Tb

To detect the original binary sequence of 1s and 0s we apply the noisy PSK

signal x(t) to a Correlator, which is also supplied with a locally generated coherent

reference signal 1 (t) as shown in fig (b). The correlator output x1 is compared with a

threshold of zero volt.

If x1 > 0, the receiver decides in favour of

symbol 1. If x1 < 0, the receiver decides in

favour of symbol 0.

Coherent Binary FSK

1 1

Tbb

2Eb

S 2(t) Cos2f 2t for symbol 0

Tb

1 T 1

b

2 2 b

Tb

message points i.e. N=2 and m=2.

(a)

(b)

Fig. 3.4: (a) FSK transmitter (

A binary FSK Transmitter is as shown in fig. (a). The incoming binary data

sequence is applied to on-off level encoder. The output of encoder is Eb volts for symbol 1 and 0 vo

switched on with oscillator frequency f1, for symbol 0, because of inverter the lower

channel is switched on with oscillator frequency f2. These two frequencies are combined using an a

The detector consists of two correlators. The incoming noisy BFSK signal x(t) is

common to both correlator. The Coherent reference signal 1 (t) and 2 (t) are supplied

The correlator outputs are then subtracted one from the other and resulting

a

random vector l (l=x1 - x2). The output l is compared with threshold of zero

volts.

If l > 0, the receiver decides in favour of

symbol 1. l < 0, the receiver decides in

favour of symbol0

BINARY ASK SYSTEM:-

Data ON-OFF Product Modulator Signal

Sequence Level

Encode

r

2

(t) Tb Cos2f t

1 e

Tb

dt Decision Device

x(t) X 0 If x > choose symbol 1

1 (t)Thresho

Fig. 3.6 Coherent binary ASK demodulator In Coherent binary ASK system the basic function is give

(t) 2 Cos2f t0 t T

1 e b

Tb

The transmitted signals S1(t) and S2(t) are given by

S1 (t) Eb 1 (t) for Symbol 1

The BASK system has one dimensional signal space with two messages (N=1,

M=2)

Region E2 Region E1

Message

Point 2

Eb

1 (t)

E

0 b

Message

2 Point 1

Fig. 3.7 Signal Space representation of BASK signal

In transmitter the binary data sequence is given to an on-off encoder. Which gives

an

output Eb volts for symbol 1 and 0 volt for symbol 0. The resulting binary wave [in unipolar

form] and sinusoidal carrier 1 (t) are applied to a product modulator. The desired BASK wave is ob

In demodulator, the received noisy BASK signal x(t) is apply to correlator with coher

reference signal 1 (t) as shown in fig. (b). The correlator output x is compared with

If x > the receiver decides in favour of symbol 1. If x < the receiver decides in fa

Incoherent detection:

Incoherent detection as used in analog communication does not require carrier for

reconstruction. The simplest form of incoherent detector is the envelope detector as shown in

Fig. 3.8. The output of envelope detector is the baseband signal. Once the baseband signal is

recovered, its samples are taken at regular intervals and compared with threshold.

If Z(t) is greater than threshold ( ) a decision will be made in favour of symbol 1

If Z(t) the sampled value is less than threshold ( ) a decision will be made in favour

of

symbol 0.

Fig. 3.9 shows the block diagram of incoherent type FSK demodulator. The detector

consists of two band pass filters one tuned to each of the two frequencies used to

communicate 0s and 1s., The output of filter is envelope detected and then baseband detected u

sinusoids is stronger at the receiver. If we take the difference of the outputs of the two envelope

detectors the result is bipolar baseband.

The resulting envelope detector outputs are sampled at t = kTb and their values are

compared with the threshold and a decision will be made infavour of symbol 1 or 0.

(a) DPSK Transmitter

A DPSK system may be viewed as the non coherent version of the PSK. It eliminates the need for co

Differential encoding of the input binary wave and

Phase shift keying

Hence the name differential phase shift keying [DPSK]. To send symbol 0

0

we phase advance the current signal waveform by 180 and to send symbol 1 we

leave the phase of the current signal waveform unchanged.

The differential encoding process at the transmitter input starts with an

arbitrary first but, securing as reference and thereafter the differentially encoded

sequence{dk} is generated by using the logical equation.

d k d k 1 bk d k 1 bk

Where bk is the input binary digit at time kTb and dk-1 is the previous value of

the

differentially encoded digit. Table illustrate the logical operation involved in the

generation of DPSK signal.

Differentially Encoded 1 1 0 1 1 0 1 1 1

sequence {dK}

Transmitted Phase 0 0 0 0 0 0 0

Received Sequence 1 0 0 1 0 0 1 1

(Demodulated Sequence)

A DPSK demodulator is as shown in fig(b). The received signal is first passed through a BPF centere

to the cosine of the difference between the carrier phase angles in the two correlator

inputs. The correlator output is finally compared with threshold of 0 volts . If correlator output is +

COHERENT QUADRIPHASE SHIFT KEYING

In case of QPSK the carrier is given by

i

T c

i

T c

T c

Fig. 3.11(c) QPSK Waveform

in the phase. The transmitted signals are given by

n

Where the carrier frequency f C

C

for some fixed integer nc

7

T = Symbol duration.

2

1 (t) T cos 2 fc

t

0 t T

2

t

b

2(t) T b sin 2 f c 0 t T

There are four message points and the associated signal vectors are defined by

E cos

2i 1

Si

4 i 1,2,3,4

E sin 2i

4

The table shows the elements of signal vectors, namely Si1 & Si2

Table:-

Input dibit Phase of Coordinates of message

QPSK points

signal(radians) Si1 Si2

10 E E

4 2 2

00 3 E E

4 2 2

01 5 E E

4 2 2

11 7 E E

4 2 2

Unit III

BASEBAND:

Pulse Shaping for Optimum Transmissions

Base Band Reception Techniques

Receiving Filter:

Correlative receiver

For an AWGN channel and for the case when the transmitted signals are

equally likely, the optimum receiver consists of two subsystems

correlators 1(t) ,2(t) .M(t) orthonormal function

2) The bank of correlator operate on the received signal x(t) to produce

observation vector x

on observationvector x to produce an estimate of the transmitted symbol m ii = 1

to M, in a way that would minimize the average probability of symbol error.

The N elements of the observation vector x are first multiplied by the

corresponding N elements of each of the M signal vectors s1, s2 sM , and

the resulting products are successively summed in accumulator to form the

corresponding set of

Inner products {(x, sk)} k= 1, 2 ..M. The inner products are corrected for the

fact that the transmitted signal energies may be unequal. Finally, the largest in

the resulting set of numbers is selected and a corresponding decision on the

transmitted message made.

The optimum receiver is commonly referred as a correlation receiver

MATCHED FILTER

Science each of the orthonormal basic functions are 1(t) ,2(t) .M(t) is

assumed to be zero outside the interval 0<t<T. we can design a linear filter with

impulse response hj(t), with the received signal x(t) the fitter output is given by

the convolution integral

yj(t) = xj

input signal is said to be matched to xj (t)correspondingly, the optimum

receiver based on this isreferred as the matched filter receiver.

causal.

MATCHED FILTER

(t) = input signal

W(t) =white noise

version of the input signal

PROPERTY 1

The spectrum of the output signal of a matched filter with the matched signal as inpu

PROPERTY 2

autocorrelation function of the input signal to which the filter is matched.

PROPERTY 3

The output Signal to Noise Ratio of a Matched filter depends only on the ratio

of the signal energy to the power spectral density of the white noise at the filter

input.

PROPERTY 4

The Matched Filtering operation may be separated into two matching

conditions; namely spectral phase matching that produces the desired output

peak at time T, and the spectral amplitude matching that gives this peak value its

optimum signal to noise density ratio.

EYE PATTERN

The quality of digital transmission systems are evaluated using the bit

error rate. Degradation of quality occurs in each process modulation,

transmission, and detection. The eye pattern is experimental method that

contains all the information concerning the degradation of quality. Therefore,

careful analysis of the eye pattern is important in analyzing the degradation

mechanism.

applied to the vertical deflection plates of an oscilloscope and the

sawtooth wave at a rate equal to transmitted symbol rate is applied to the

horizontal deflection plates, resulting display is eye pattern as it

resembles human eye.

shown below.

The width of the eye opening defines the time interval over which the

received wave can be sampled without error from ISI

of the margin over channel noise.

The sensitivity of the system to timing error is determined by the rate of closure

of the eye as the sampling time is varied.

squinted eye. When the effected of ISI is excessive, traces from the upper

portion of the eye pattern cross traces from lower portion with the result that the

eye is completely closed.

INFORMATION THEORY

Although it is in principle a very old concept, entropy is generally credited to Shannon

because it is the fundamental measure in information theory. Entropy is often defined as an

expectation:

where 0 log(0) = 0. The base of the logarithm is generally 2. When this is the case, the units

of entropy are bits.

aeasure of the average length of a message that would have to be sent to describe a sample.

Recall our fair coin from 1. Its entropy is:0.5log0.5 + 0.5log0.5= 1; that is, thereis one

bit of information in the random variable.This means that on average we need to sendone bit per tr

Even if I flip this coin 100 times, it doesnt matter because the outcome is always heads. I dont need to

There are other possibilities besides being completely random and completely deter-mined.

Imagine a weighted coin, such that heads occurr 75% of the time. The entropy would be:

0.75log0.75 + 0.25log0.25= 0.8113. After 100 trials, Id only need a message of about 82 bits

on average to describe the sample. Shannon showed that there exists a coder that can construct

messages of length H(X)+1, nearly matching this ideal rate.

Just as with probabilities, we can compute joint and conditional entropies. Joint

entropy is the randomness contained in two variables, while conditional entropy is a measure

of the randomness of one variable given knowledge of another. Joint entropy is defined as:

while the conditional entropy is:

There are several interesting facts that follow from these definitions. For example, two

random variables, X and Y, are considered independent if and only if HY| X=

HY

or HXY= HX+HY It is also the case that HY|XHY (knowing more information

can never increase our uncertainty). Similarly, HXYHX+HY It is alsothe case that

HXY=HY|X+HX=HX Y+HY These relations hold in thegeneral case of more than

two variables.

There are several facts about discrete entropy, H(), that do not hold for continuous

ordifferential entropy, h(). The most important is that while H X 0 h() can actually be negative.

of randomness, it is still that case that if h X h Y then X has more randomness than Y.

Mut

Although co

not an adeq

us a great deal about Y or that H(Y) is small to begin with. Thus, we measure dependence

using mutual information:

IXY= HYHY|X

knowledge of another variable. Using properties of logarithms, we can derive several equiva-

lent definitions:

IXY= HYHY | X

= HXHX | Y

=HX+HYHXY

= IYX

particular case of the Kullback-Leibler divergence. The KL divergence is defined as:

KL divergence measures the difference between two distributions. It is sometimes called the

relative entropy. It is always non-negative and zero only when p=q; however, it is not a

distance because it is not symmetic.

In other words, mutual information is a measure of the difference between the joint

probability and product of the individual probabilities. These two distributions are equivalent

only when X and Y are independent, and diverge as X and Y become more dependent.

Shannon-Fano Code

ShannonFano coding, named after Claude Elwood Shannon and Robert Fano, is a technique

for constructing a prefix code based on a set of symbols and their probabilities. It is

suboptimal in the sense that it does not achieve the lowest possible expected codeword length

like Huffman coding; however unlike Huffman coding, it does guarantee that all codeword

lengths are within one bit of their theoretical ideal I(x) =log P(x).

In ShannonFano coding, the symbols are arranged in order from most probable to least

probable, and then divided into two sets whose total probabilities are as close as possible to

being equal. All symbols then have the first digits of their codes assigned; symbols in the first

set receive "0" and symbols in the second set receive "1". As long as any sets with more than

one member remain, the same process is repeated on those sets, to determine successive

digits of their codes. When a set has been reduced to one symbol, of course, this means the

symbol's code is complete and will not form the prefix of any other symbol's code.

The algorithm works, and it produces fairly efficient variable-length encodings; when the two

smaller sets produced by a partitioning are in fact of equal probability, the one bit of

information used to distinguish them is used most efficiently. Unfortunately, ShannonFano

does not always produce optimal prefix codes.

For this reason, ShannonFano is almost never used; Huffman coding is almost as

computationally simple and produces prefix codes that always achieve the lowest expected code w

performance and minimum requirements for programming.

Sha

A Shannon

code table

For

counts so that each symbols relative frequency of occurrence is known.

Sort the lists of symbols according to frequency, with the most frequently occurring

symbols at the left and the least common at the right.

Divide the list into two parts, with the total frequency counts of the left part being as

close to the total of the right as possible.

The left part of the list is assigned the binary digit 0, and the right part is assigned the

digit 1. This means that the codes for the symbols in the first part will all start with 0,

and the codes in the second part will all start with 1.

Recursively apply the steps 3 and 4 to each of the two halves, subdividing groups and

adding bits to the codes until each symbol has become a corresponding code leaf on the tree.

Example:

The source of information A generates the symbols {A0, A1, A2, A3 and A4} with the

corresponding probabilities {0.4, 0.3, 0.15, 0.1 and 0.05}. Encoding the source symbols using

binary encoder and Shannon-Fano encoder gives:

A0 0.4 000 0

A1 0.3 001 10

A2 0.15 010 110

A3 0.1 011 1110

A4 0.05 100 1111

Lavg H = 2.0087 3 2.05

Shannon-Fano code is a top-down approach. Constructing the code tree, we get

Source Coding

random sequence models. Let us consider the example of a discrete memory less

source (DMS), which is a simple random sequence model.

A DMS is a source whose output is a sequence of letters such that each letter is

.ak. The letters in the source output sequence are assumed to be random and

statistically

independent of each other. A fixed probability assignment for the occurrence of

each

letter is also assumed. Let us, consider a small example to appreciate the importance of

probability assignment of the source letters.

, aa34 P(a1)=0.5,

P(a2)=0.25, P(a3)= 0.13, P(a4)=0.12. Let us decide to go for binary coding of these four

source letters. While this can be done in multiple ways, two encoded representations are shown be

has not been considered and all letters have been represented by two bits each.

However in

the second method only a1 has been encoded in one bit, a2 in two bits and the

remaining two in three bits. It is easy to see that the average number of bits to be used

per source letter for the two methods are not the same. ( a for method #1=2 bits per

letter and a for

method #2 < 2 bits per letter). So, if we consider the issue of encoding a long sequence of

letters we have to transmit less number of bits following the second method. This is

an important aspect of source coding operation in general. At this point, let us note the

following:

a) We observe that assignment of small number of bits to more probable letters and

assignment of larger number of bits to less probable letters (or symbols) may lead to

efficient source encoding scheme.

b) However, one has to take additional care while transmitting the encoded letters. A

careful inspection of the binary representation of the symbols in method #2 reveals

that it may lead to confusion (at the decoder end) in deciding the end of binary

representation of a letter and beginning of the subsequent letter.

The average number of coded bits (or letters in general) required per source letter is as

small as possible and

The source letters can be fully retrieved from a received encodedsequence.

In the following we discuss a popular variable-length source-coding scheme satisfying the above tw

{a1,a2, .aK} with

probabilities P(a 1), P(a2),. P(aK). Each source letter is to be encoded into a

codeword made of elements (or letters) drawn from a code alphabet containing D

symbols. Often for ease of implementation a binary code alphabet (D = 2) is chosen.

As

we observed earlier in an example, different codeword may not have same number of

code symbols. If nk denotes the number of code symbols corresponding to the source

letter ak , the average number of code letters per source letter ( n ) is:

k =1

Intuitively, if we encode a very long sequence of letters from a DMS, the

number of code letters per source letter will be close to n .

finite length, the sequence of code letters corresponding to that source sequence is

different from the sequence of code letters corresponding to any other possible source

sequence.

as

prefix condition code. Let the code word in a code be represented as

(when D=2, these are 1 or 0). Any sequence made up of an initial part of x k

that is xk,1,xk,2,......,xk,i for i nk is called a prefix of xk .

any other codeword.

Example: consider the following table and find out which code out of the four shown is

/are prefix condition code. Also determine n for each code.

Source letters:- a4

P(ak) :- P(a1)=0.5, P(a2)=0.25, P(a3)= 0.125, P(a4)=0.125

A prefix condition code can be decoded easily and uniquely. Start at the beginning of

a

sequence and decode one word at a time. Finding the end of a code word is not a

problem as the present code word is not a prefix to any other codeword.

theprevious example. See that the corresponding source letter sequence is a1 ,a4 , a2 , a1 .

withoutproof.

In Binary Huffman Coding each source letter is converted into a binary code

word. It is a prefix condition code ensuring minimum average length per source letter in bits.

Let the source letters a1, a 2, .aK have probabilities P(a1), P(a2),.

P(aK) and let us assume that P(a1) P(a2) P(a 3). P(aK).

We now consider a simple example to illustrate the steps for Huffman coding.

Example Let us consider a discrete memoryless source with six letters having

Arrange the letters in descending order of their probability (here they are

arranged).

Consider the last two probabilities. Tie up the last two probabilities. Assign,

say, 0 to the last digit of representation for the least probable letter (a6) and

1 to the

last digit of representation for the second least probable letter (a5). That is,

lower arm.

1

P(a5)=0.1

2 0.2

P(a6)=0.08 0

(3) Now, add the two probabilities and imagine a new letter, say b1, substituting

for

a6 and a5. So P(b1) =0.2. Check whether a4 and b1are the least likely letters. If

not, reorder the letters as per Step#1 and add the probabilities of two least

letters.likely

For our example, it leads to:

P(a1)=0.3, P(a2)=0.2, P(b1)=0.2, P(a3)=0.15 and P(a4)=0.15

(4) Now go to Step#2 and start with the reduced ensemble consisting of a1 , a2 , a3 ,

P(a3)=0.15 1

0.3

0

P(a4)=0.15

t Continue till the first digits of the most reduced ensemble of two letters are

assigned a 1 and a 0.

P(b1)=0.2.

Now we consider the last two probabilities:

P(a2)=0.2

0.4

P(b1)=0.2

So, P(b3)=0.4. Following Step#2 again, we get, P(b3)=0.4, P(a1)=0.3 and

P(b2)=0.3.

to: 1

P(a1)=0.3

0.6

P(b2)=0.3 0

P(b4)=0.6 1

1.00

P(b3)=0.4

0

6. Now, read the code tree inward, starting from the root, and construct the

codewords. The first digit of a codeword appears first while reading the code

tree inward.

Hence, the final representation is: a1=11, a2=01, a3=101, a4=100, a5=001, a6=000.

Prefix condition says that representation of any letter should not be a part of

any other letter.

3. Average length/letter (in bits) after coding is

i

4. Note that the entropy of the source is: H(X)=2.465 bits/symbol. Average length

per source letter after Huffman coding is a little bit more but close to the source

entropy. In fact, the following celebrated theorem due to C. E. Shannon sets the limiting value of av

CONVOLUTIONAL CODES

Convolutional codes are commonly described using two parameters: the code

rate and the constraint length. The code rate, k/n, is expressed as a ratio of the number

of bits into the convolutional encoder (k) to the number of channel symbols output by

the convolutional encoder (n) in a given encoder cycle. The constraint length

parameter, K, denotes the "length" of the convolutional encoder, i.e. how many k-bit

stages are available to feed the combinatorial logic that produces the output symbols.

Closely related to K is the parameter m, which indicates how many encoder cycles an

input bit is retained and used for encoding after it first appears at the input to the

convolutional encoder. The m parameter can be thought of as the memory length of

the encoder.

communication systems for error correction. The encoded bits depend on the current k

input bits and a few past input bits. The main decoding strategy for convolutional

codes is based on the widely used Viterbi algorithm. As a result of the wide

acceptance of convolutional codes, there have been several approaches to modify and

extend this basic coding scheme. Trellis coded modulation (TCM) and turbo codes are

two such examples. In TCM, redundancy is added by combining coding and

modulation into a single operation. This is achieved without any reduction in data rate

or expansion in bandwidth as required by only error correcting coding schemes.

A simple convolutional encoder is shown in Fig. 7.1. The information bits are

fed in small groups of k-bits at a time to a shift register. The output encoded bits are

obtained by modulo-2 addition (EXCLUSIVE-OR operation) of the input information

bits and the contents of the shift registers which are a few previous information bits.

bits, the code rate R is commonly defined as R = k/n. In Fig. 7.1, k = 1 and n = 2.

The number, K of elements in the shift register which decides for how many

codewords one information bit will affect the encoder output, is known as the

constraint length of the code. For the present example, K = 3.

operation starts. It is easy to verify that encoded sequence is 00 11 10 00 01

.for an input message sequence of 01011.

Fig. 7.1A convolutional encoder with k=1, n=2 and r=1/2

equivalent ways such as, by a) state diagram representation, b) tree diagram

representation and c) trellis diagram representation.

A convolutional encoder may be defined as a finite state machine. Contents of

the rightmost (K-1) shift register stages define the states of the encoder. So, the

encoder in Fig. 7.2has four states. The transition of an encoder from one state to

another, ascaused by input bits, is depicted in the state diagram. Fig. 7.2 shows the

state diagram of the encoder in Fig. 7.1. A new input bit causes a transition from

one state to another. The

path information between the states, denoted as b/c1c2 , represents input information bit

b and the corresponding output bits (c1c2). Again, it is not difficult to verify from the

state diagram that an input information sequence b = (1011) generates an encoded

sequence c = (11, 10, 00, 01).

Fig. 7.2 State diagram representation for the encoder in Fig. 7.1

b) Tree Diagram Representation

The tree diagram representation shows all possible information and encoded

sequences for the convolutional encoder. Fig. 7.3 shows the tree diagram for the

encoder in Fig. 7.1. The encoded bits are labeled on the branches of the tree. Given an

input sequence, the encoded sequence can be directly read from the tree. As an

example, an input sequence (1011) results in the encoded sequence (11, 10, 00, 01).

c) Trellis Diagram Representation

The trellis diagram of a convolutional code is obtained from its state diagram.

All state transitions at each time step are explicitly shown in the diagram to retain the

time dimension, as is present in the corresponding tree diagram. Usually, supporting

descriptions on state transitions, corresponding input and output bits etc. are labeled in

the trellis diagram. It is interesting to note that the trellis diagram, which describes the

operation of the encoder, is very convenient for describing the behavior of

thecorresponding decoder, especially when the famous Viterbi Algorithm (VA) is

followed. Figure 7.4 shows the trellis diagram for the encoder in Figure 7.1.

Fig. 7.4Trellis diagram, used in the decoder corresponding to the encoder in

Fig.7.1

Hard-Decision and Soft-Decision Decoding

Hard-decision and soft-decision decoding are based on the type of

quantization used on the received bits. Hard-decision decoding uses 1-bit quantization

on the received samples. Soft-decision decoding uses multi-bit quantization (e.g. 3

bits/sample) on the received sample values.

The Viterbi Algorithm (VA) finds a maximum likelihood (ML) estimate of a

transmitted code sequence c from the corresponding received sequence r by

maximizing the probability p(r|c) that sequence r is received conditioned on the

estimated code sequence c. Sequence c must be a valid coded sequence.

The Viterbi algorithm utilizes the trellis diagram to compute the path metrics. The channel is assum

metric of all branches, associated with all the states are calculated similarly Now, at each depth of

process, the accumulated path metric is updated by adding the metric of the

incoming branch with the accumulated path metric of the state from where the

branch originated. No decision about a received codeword is taken from such

operations and the decoding decision is deliberately delayed to reduce the possibility

of erroneous decision.

The basic operations which are carried out as per the hard-decision Viterbi

Algorithm after receiving one codeword are summarized below:

All the branch metrics of all the states are determined;

.

Accumulated metrics of all the paths (two in our example code) leading to a

state are calculated taking into consideration the accumulated path metrics of

the states from where the most recent branches emerged;

Only one of the paths, entering into a state, which has minimum accumulated

path metric is chosen as the survivor path for the state (or, equivalently

node);

So, at the end of this process, each state has one survivor path. The history

of a survivor path is also maintained by the node appropriately ( e.g. by storing

the codewords or the information bits which are associated with the branches

making the path);

(5) Steps a) to d) are repeated and decoding decision is delayed till sufficient number

of codewords has been received. Typically, the delay in decision making = Lx k codewords where L

codeword.

The above procedure is repeated for each received codeword hereafter. Thus, the decision for a cod

In soft-decision decoding, the demodulator does not assign a 0 or a

1 to each received bit but uses multi-bit quantized values. The soft-decision Viterbi

algorithm is very similar to its hard-decision algorithm except that squared Euclidean

distance is used in the branch metrics instead of simpler Hamming distance. However,

the performance of a soft-decision VA is much more impressive compared to its HDD

(Hard Decision Decoding) counterpart [Fig. 7.6 (a) and (b)]. The computational

requirement of a Viterbi decoder grows exponentially as a function of the constraint

length and hence it is usually limited in practice to constraint lengths of K = 9.

Fig. 7.6 (a) Decoded BER vs input BER for the rate half convolutional

codes withViterbi Algorithm ; 1) k = 3 (HDD),2) k = 5 (HDD),3) k = 3

(SDD), and 4) k= 5 (SDD). HDD: Hard Decision Decoding; SDD: Soft

Decision Decoding.

Fig. 7.6 (b) Decoded BER vs Eb/No(in dB) for the rate half convolutional codes withViterbi

Algorithm ; 1) Uncoded system; 2) with k = 3 (HDD) and 3) k = 3 (SDD). HDD: Hard

Decision Decoding; SDD: Soft Decision Decoding.

Introduction:

Initially developed for military applications during II world war, that was less sensitive to

intentional interference or jamming by third parties.

Spread spectrum technology has blossomed into one of the fundamental building blocks in

current and next-generation wireless systems

Narrow band can be wiped out due to interference

Solution

A spread spectrum system is therefore designed to make these tasks as difficult as possible.

signal should have a low probability of intercept (LPI).

Secondly, the signal should be difficult to disturb with a jamming signal, i.e.,

thetransmitted signal should possess an anti-jamming (AJ) property

Remedy

Spread the narrow band signal into a broad band to protect against interference

In a digital communication system the primary resources are Bandwidth andPower. The

study of digital communication system deals with efficient utilization ofthese two resources,

but there are situations where it is necessary to sacrifice their efficient utilization in order to

meet certain other design objectives.

For example to provide a form of secure communication (i.e. the transmitted signal is

not easily detected or recognized by unwanted listeners) the bandwidth of the transmitted

signal is increased in excess of the minimum bandwidth necessary to transmit it. This

requirement is catered by a technique known as Spread SpectrumModulation.

ability to reject Interference whether it be the unintentional or the

intentionalinterference.

BW (Bandwidth) in excess of the minimum BW necessary to transmit it.

code that is independent of the data sequence. The Same code is used in the receiver

to despread the received signal so that the original data sequence may be recovered.

s(t) wide band r(t) wide band b(t) + Noise

b(t) ..... ... . Narrow Wide

Band Band

= Wide Band code

s(t) = c(t) * b(t) (wide Band)

Code division multiple access

technologies.

One of the basic concepts in data communication is the idea of allowing several

transmitters to send information simultaneously over a single communication channel. This

allows several users to share a bandwidth of frequencies. This concept is called multiplexing.

CDMA employs spread-spectrum technology and a special coding scheme (where each

transmitter is assigned a code) to allow multiple users to be multiplexed over the same

physical channel. By contrast, time division multiple access (TDMA) divides access by time,

while frequency-division multiple access (FDMA) divides it by frequency. CDMA is a form

of "spread-spectrum" signaling, since the modulated coded signal has a much higher data

bandwidth than the data being communicated.

wish to communicate with each other. To avoid confusion, people could take turns speaking

(time division), speak at different pitches (frequency division), or speak in different

languages (code division). CDMA is analogous to the last example where people speaking

the same language can understand each other, but not other people. Similarly, in radio

CDMA, each group of users is given a shared code. Many codes occupy the same channel,

but only users associated with a particular code can understand each other.

Technical details

code runs at a much higher rate than the data to be transmitted. Data for transmission is

simply logically XOR (exclusive OR) added with the faster code. The figure shows how

spread spectrum signal is generated. The data signal with pulse duration of Tb is XOR added

with the code signal with pulse duration of Tc. (Note: bandwidth is proportional to 1 / T

where T = bit time) Therefore, the bandwidth of the data signal is 1 / Tb and the bandwidth of

the spread spectrum signal is 1 / Tc. Since Tc is much smaller than Tb, the bandwidth of the

spread spectrum signal is much larger than the bandwidth of the original signal.

Fig. 8.3

CDMA uses Direct Sequence spreading, where spreading process isdone by directly

combining the baseband information to high chip rate binary code. The Spreading Factor is

the ratio of the chips (UMTS = 3.84Mchips/s) to baseband information rate. Spreading

factors vary from 4 to 512 in FDD UMTS. Spreading process gain can in expressed in dBs

(Spreading factor 128 = 21dB gain).

Fig. 8.4

Each user in a CDMA system uses a different code to modulate their signal. Choosing

the codes used to modulate the signal is very important in the performance of CDMA

systems. The best performance will occur when there is good separation between the signal

of a desired user and the signals of other users. The separation of the signals is made by

correlating the received signal with the locally generated code of the desired user. If the

signal matches the desired user's code then the correlation function will be high and the

system can extract that signal. If the desired user's code has nothing in common with the

signal the correlation should be as close to zero as possible (thus eliminating the signal); this

is referred to as cross correlation. If the code is correlated with the signal at any time offset

other than zero, the correlation should be as close to zero as possible. This is referred to as

auto-correlation and is used to reject multi-path interference.

Fig. 8.5

PSUEDO-NOISE SEQUENCE:

Generation of PN sequence:

Clock

Shift Shift Shift Output

Register1 Register2 Register3

S0 S3

Logic Circuit

A feedback shift register is said to be Linear when the feed back logic consists of

entirely mod-2-address ( Ex-or gates). In such a case, the zero state is not permitted. The period of

Example1: Consider the linear feed back shift register as shown in fig 2involve

three flip-flops. The input so is equal to the mod-2 sum of S1 and S3. If the initial state of the shift re

100,110,011,011,101,010,001,100 . . . . . .

3

Which repeats itself with period 2 1 = 7 (n=3)

m

N = 2 -1

Fig.8.7

(7) Contents of required stages are modulo 2 added and fed back to input.

Fig. 8.8 Initial stages of Shift registers1000 Let initial status of shift register be 1000

1 0 0 0

0 1 0 0

0 0 1 0

1 0 0 1

1 1 0 0

0 1 1 0

1 0 1 1

0 1 0 1

1 0 1 0

1 1 0 1

1 1 1 0

1 1 1 1

0 1 1 1

0 0 1 1

0 0 0 1

1 0 0 0

We can see for shift Register of length m=4.

.At each clock the change in state of flip-flop is shown.

After 15 clock pulses the sequence repeats.

Output sequence is

000100110101111

Properties of PN Sequence

Randomness of PN sequence is tested by following properties

u Balance property

v Run length property

w Autocorrelation property

1. Balance property

In each Period of the sequence , number of binary ones differ from binary zeros by

at most one digit .

Consider output of shift register 0 0 0 1 0 0 1 1 0 1 0 1 1 1 1 Seven

zeros and eight ones -meets balance condition.

Among the runs of ones and zeros in each period, it is desirable that about one half the runs

of each type are of length 1, one- fourth are of length 2 and one-eighth are of length 3 and so-

on.

Number of runs =8

0 0 0 1 0 0 1 1 0 1 01 1 1 1

3 1 2 2 1 1 1 4

output of shift register for l=1

Yields PN autocorrelation as

Range of PN Sequence Lengths

7 127

8 255

9 511

10 1023

11 2047

12 4095

13 8191

17 131071

19 524287

A Notion of Spread Spectrum:

protection against externally generated interfacing signals with finite power. Protection

against jamming (interfacing) waveforms is provided by purposely making the information

bearing signal occupy a BW far in excess of the minimum BW necessary to transmit it.

This has the effect of making the transmitted signal a noise like appearance so as to blend

into the background. Therefore Spread Spectrum is a method of camouflaging the

information bearing signal.

V

b(t) m(t). . r(t) z(t) Tb

Decisi on Device

dt

0

<----Tran

{ cK } denotes a PN sequence.

The desired modulation is achieved by applying the data signal b(t) and PN signal c(t) to a product

For base band transmission, the product signal m(t) represents the transmitted

The received signal r(t) consists of the transmitted signal m(t) plus an additive

interference noise n(t), Hence

= c(t).b(t) + n(t)

+1

-1

+1

0 -1

+1

0 -1

To recover the original message signal b(t), the received signal r(t) is applied to a

demodulator that consists of a multiplier followed by an integrator and a decision device. The

multiplier is supplied with a locally generated PN sequence that is exact replica of that used

in the transmitter. The multiplier output is given by

Z(t) = r(t).c(t)

2

5.c (t).b(t) + c(t).n(t)

The data signal b(t) is multiplied twice by the PN signal c(t), where as unwanted signal

2

n(t) is multiplied only once. But c (t) = 1, hence the above equation reduces to

Now the data component b(t) is narrowband, where as the spurious component c(t)n(t)

is wide band. Hence by applying the multiplier output to a base band (low pass) filter most of

the power in the spurious component c(t)n(t) is filtered out. Thus the effect of the interference

n(t) is thus significantly reduced at the receiver output.

The integration is carried out for the bit interval 0 t T b to provide the sample

value V. Finally, a decision is made by the receiver.

modulated carrier is widened by changing the carrier frequency in a pseudo random

manner. The type of spread spectrum in which the carrier hops randomly form one

frequency to another is called FrequencyHop (FH) Spread Spectrum.

Since frequency hopping does not covers the entire spread spectrum

instantaneously. We are led to consider the rate at which the hop occurs. Depending

upon this we have two types of frequency hop.

1. Slow frequency hopping:- In which the symbol rate Rs of the MFSK signal is an

integer multiple of the hop rate Rh. That is several symbols are transmitted on each

frequency hop.

2. Fast Frequency hopping:- In which the hop rate Rh is an integral multiple of the

MFSK symbol rate Rs. That is the carrier frequency will hoop several times during

the transmission of one symbol.

ary frequency shift keying (MFSK).

Slow frequency hopping:-

Fig. 8.12 a) Shows the block diagram of an FH / MFSK transmitter, which involves

The incoming binary data are applied to an M-ary FSK modulator. The resulting

modulated wave and the output from a digital frequency synthesizer are then applied to a

mixer that consists of a multiplier followed by a band pass filter. The filter is designed

to select the sum frequency component resulting from the multiplication process as the

transmitted signal. An k bit segments of a PN sequence drive the frequencysynthesizer,

n

which enables the carrier frequency to hop over 2 distinct values. Since frequency

synthesizers are unable to maintain phase coherence over successive hops, most frequency

hops spread spectrum communication system use non coherent M-ary modulation system.

Fig 8.12:- Frequency hop spread M-ary Frequency shift keying

In the receiver the frequency hoping is first removed by mixing the received signal

with the output of a local frequency synthesizer that is synchronized with the transmitter. The

resulting output is then band pass filtered and subsequently processed by a non coherent M-

ary FSK demodulator. To implement this M-ary detector, a bank of M non coherent matched

filters, each of which is matched to one of the MFSK tones is used. By selecting the largest

filtered output, the original transmitted signal is estimated.

An individual FH / MFSK tone of shortest duration is referred as a chip. The chip rate

Rc for an FH / MFSK system is defined by

Rc = Max(Rh,Rs)

In a slow rate frequency hopping multiple symbols are transmitted per hop. Hence

each symbol of a slow FH / MFSK signal is a chip. The bit rate R b of theincoming binary

data. The symbol rate Rs of the MFSK signal, the chip rate Rc and the hop rate Rn are related by

Rc = Rs = Rb /k Rh

where k= log2M

A fast FH / MFSK system differs from a slow FH / MFSK system in that there

are multiple hops per m-ary symbol. Hence in a fast FH / MFSK system each hop is a chip.

Fig. illustrates the variation of the frequency of a slow FH/MFSK signal with time for one

4

complete period of the PN sequence. The period of the PN sequence is 2 -1 = 15. The

FH/MFSK signal has the following parameters:

Number of bits per MFSK symbol K = 2.

K

Number of MFSK tones M=2 =4

k

Total number of frequency hops 2 =8

Fig. illustrates the variation of the transmitted frequency of a fast FH/MFSK signal with time.

The signal has the following parameters:

K

Number of MFSK tones M=2 =4

k

Total number of frequency hops 2 =8

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