You are on page 1of 14

1

B.Sc IT 17 – 02
COMMUNICATION TECHNOLOGY

Q1. What are the advantages of digital communication? Explain briefly.

Ans:- Advantages of digital communication: -

a. The digital communication provides greater immunity to noise when compared with the Analog
communication. It also provides a various means of detecting and correcting the errors occurred during the
process of transmission of signals.
b. Digital signals, which are inherently compatible with computers, have a potential to be stored, retrieved,
processed and manipulated for signal enhancement and improved performance.
c. Using digital communication it is easy to integrate diverse. Source of information into a common format.
d. High degree of security of information can be maintained in the course of transmission by proper encryption of
the data. This is of great importance in application like military communication.

However the digital communication requires higher channel bandwidth and more
complex circuitry than that of when compared with along communication.

e. Digital communication id readily adaptive to other powerful and advanced data processing such as Digital
signal processing, image processing data compression.
f. System with higher complexities can be built in a cost- effective manner due to the advances in IC (integrated
circuitry) technology. More over they can be more reliable and compact by employing application specific ICs.

However the digital communication requires higher channel bandwidth and more complex circuitry such as
synchronization when compared with the analog communication.

Q2. What you mean by bandwidth ? what is the bandwidth occupied by


a) Voice signals
b) Audio signals
c) Video signals

Sketch the spectrum of each.

Ans:- Bandwidth is the span of frequencies within the spectrum occupied by a signal and used by the signal for
conveying information, For example Voice has a band width 3 to 4 KHz. Audio signal (Speech and music) has
15KHz. Where as video requires a bandwidth of 5 MHz.

The bandwidth is an important parameter in communication and its depends on the type of signal for type of
application, the amount of information to be communicated and the time in which the information to be
communicated.
To convey more information is short time we need more bandwidth. The same
quantity of information can be sent in a longer period using less bandwidth. Similarly to convey voice signal we
need less bandwidth and to convey video it requires more bandwidth and so on.

Q3. State and explain sampling theorem.

Ans:- Most of the times, the signals we have to communicate will be in the analog form, such as voice signals. As a
first step in digitization, the analog signal is converted to a discrete time signal by the process of sampling. While
sampling, sufficient number of samples of the signal must be taken so that the signal is completely represented in its
samples. The numbers of samples to be taken depends on the maximum frequency in the case of a low pass signal
and foe a band pass signal it depends on the bandwidth.
Sampling theorem provides complete information regarding the number of samples to be taken and
forms the basis of digitization of analog signals. In 1928 H. Nyquist showed that an analog signal can be perfectly
2
reconstructed from its samples without any loss of original information, if it sampled at least twice the rate of
maximum frequency component, or the bandwidth of the signal. This called sampling theorem or Nyquist criteria of
sampling. The rate of sampling, which is equal to twice the signal bandwidth, is called the Nyquist rate.
In other words we can state the sampling theorem as follows:---
“Any signal which is continues in time (analog ) can be completely represented by its samples can be recovered if
the sampling frequency fs ≥ 2*fm. Where fs is the sampling frequency and fm is the maximum frequency of the
signal.
This means that if the signal is of bandwidth 4000Hz. Then it must be sampled at least 8000
sample/sec. (8000Hz) or greater so as to enable the reproduction of the signal without distortion. Similarly if the
signal as 5000Hz. As the maximum frequency component (All other components as the signal as frequency less then
5000 Hz.) then it must be sampled at rate of 10,000Hz. Or more.
The sampling theorem is the fundamental principle of digital communication.
In communication sampling is done in principle by multiplying the given analog signal by a narrow pulse train
whose frequency is equal to the sampling frequency. Let us consider the message signal be m(t) band limited to fm
and the sampling signal S(t) be a pulse train of frequency fs = 1/Ts, where Ts is the time period of the sampling
signal. The message signal and the sampling signal are applied as inputs to the multiplier and output is the sampled
signal y(t)=S(t)m(t).
This product is m(t) whenever there is a pulse in S(t), and equal to zero otherwise. The figure shows the respective
waveforms.

Amplitude

Signal
m(t) m(t)

t
y(t)
Sampling
Multiplier
Signal 1
S(t)
S(t)
| Ts | t
Sampled
Output
y(t)

In the figure if we look at the output of the multiplier, the amplitude of the pulse train used for the sampling varies in
accordance with the amplitude of the message signal. This is known as Pulse Amplitude Modulation (PAM). In the
case of PAM signals the top of the pulse may follow the message signal in which case it is called Natural sampling,
or it can be made flat if so it is called flat top sampling.
The sampling signal is periodic with period Ts, and has a pulse width of dt. Using Fourier series expansion we can
express this sampling S(t) as
3
S(t) = dt / Ts + 2*dt/Ts (cos2π t / Ts + cos4π t / Ts +……… )

When Ts = 1 / 2fm, the product S(t)m(t) can be expressed as

S(t)m(t) = ((dt / Ts) * m(t)) + dt/Ts (2m(t) cos2π(2fm) t + 2m(t) cos2π(4fm)t


+ ……)

Now the first term in the series represents the signal m(t) itself with a multiplication by a constant. The second term
is product of m(t) and is a sinusiod of frequency 2fm = fs. This represents the double side band suppressed carrier
signal with carrier frequency at 2fm and has frequency components from fm(= 2fm – fm) to 3fm(= 2fm + fm).
Similarly the succeeding terms are double side band suppressed carrier signals with carrier frequency at 4fm, 6fm,
8fm and so on, the harmonics of 2fm.

Amplitude

Fm f

The figure shows the frequency spectrum of the signal m(t). A frequency spectrum is a plot of signal amplitudes
versus the frequency. For sinusoidal signals the spectrum will be what is called line spectrum, since each signal is
represented as a line at the corresponding frequency and the height of the line represents the maximum amplitude of
the signal. But for a band limited signal, that signal.
The figure shows the spectrum of sampled signal S(t)m(t) corresponding to three different cases. From
the spectrum it can be seen that when the sampling frequency fs is just equal to that of the maximum frequency of
the message signal the spectrum do not overlap. This is the minimum condition required to filter and separate the
base band signal. When fs is greater than the fm we

Case1, fs=2fm

fm 2fm 3fm 4fm

Case 2, fs < 2fm

fm fs

Case 3, fs > 2fm Guard band


4

fm fs

Have a guard band which is equal to fs – 2fm is available and makes the filtering easy. On the other hand when fs is
less then 2fm we get an overlapping spectrum and hence on filtering is possible to recover the base band signal. This
is as aliasing.
This proves the sampling theorem since we have show that the sampled signal can be recovered
exactly when Ts ≤ 1/2 fm. From the spectrum it is also clear that the minimum allowed sampling rate is known as
Nyquist rate. An increase in sampling rate is above the Nyquist rate increase the width of the guard band thereby
making other hand increase in the sampling rate increase the bandwidth required for the transmission of the sampled
signal. Therefore a compromise between the simplicity in filter design and bandwidth is adopted.
As we have seen if the actual sampling rate is below the Nyquist rate, it leads to a serious problem
called aliasing. It fails to represent the analog signal properly with all its contents in its sampled version. Therefore
care must be taken to prevent aliasing effect at all costs.

To overcome aliasing the following steps are taken while designing.

1. The sampling rate is fixed slightly above the Nyquist rate. Usually 20% more than the Nyquist rate.
2. The signal to be sampled is first band limited by passing it through a filter called anti- aliasing filter so that it
will not contain any frequency components outside the band of interest. This is vary much necessary, as it is not
possible to get strictly band-limited signals in practice. Therefore it becomes difficult to decide the sampling
frequency.

Q4. Explain the generation of PCM with the help of relevant sketches and block diagram.

Ans:- Pulse Code Modulation (PCM) is essentially analog to digital conversion of special type where the
information contained in the instantaneous samples of an analog signal is represented by digital words or codes in a
serial bit stream.
A simple PCM system is as shown in figure. The analog signal m(t) is sampled using a sampled and
hold circuit. The samples are held foe the quantizers until the next sample is ready. Some time an anti-aliasing filter
may precede the sampler so as to avoid aliasing effect. A quantizer then quantizes the samples. Therefore the output
of quantizer will be any one of the allowed levels unlike the sampled output which may take any voltage value
within the upper and lower limits. The quantizer voltage is converted into a uniquely identifiable binary code
represented code represented the quantizer value by the encoder.

Analog – to – digital converter

Digital
Analog
Encoded signal
Signal Sample and Quantizer Encoder
m(t) Hold

Channel

Analog Filter Decoder Quantizer


Signal
m(t)
The combination of the quantizer and the encoder is called an analog to digital converter. An analog to digital
converter therefore produces a succession of N-bit codes. The process of such conversion of analog signals into as
N-bit digital codes is called Pulse code Modulation.
5
For long-distance communication these PCM signals is suitably modulation using sinusoidal carries and then
transmitted.
At the receiving end the reverse operation called decoding is performed to extract
the original message. When the signal is transmitted over a channel it gets corrupted by the noise. Therefore before
performing the actual decoding the received signal is re-quantizer. Using a quantizer at the receiver, which follows
the same specifications as that at the transmitter. This quantizer has to check in each pulse interval whether the bit is
logic 1 or 0. The quantizer then applies these pulse trains to the decoder. The decoder which is essentially a digital to
analog converter produces the quantized PAM signal at the output. It is then filtered to remove any frequency
components lying outside the base band. The final output signal m(t) is a close approximate of the original analog
signal m(t).

The PCM is very popular because of the following advantages it offers.

• PCM signals derived from all types of analog sources can be merged with data signals and transmitted over a
common high-speed digital communication system. This kind of merging of multiple signals is time division
multiplexing (TDM).
• PCM has a better noise immunity. This is because the receiver knows that the signal is either a logical 1 or 0
and no intermediate level is possible. Therefore there is inherent noise immunity.
• Since the transmission format in PCM is in terms of N-bit codes, employing suitable coding techniques that
support detection and correction of errors can reduce the probability of error at the output of the system.

However the PCM has certain drawbacks. First of all PCM requires complex encoding and decoding circuitry. Since
PCM is encoded as N=-bit format the bandwidth required is very large. In fact it gets multiplied by a factor of N.
PCM is used in telephony, general data transmission involving multiplexing of signals. PCM is also used in
space communication.

Q5. What is compounding ? why it is required ? Explain.

Ans:- The quantization process described in the previous section employs uniform separation between the
quantization levels. This has one drawback. Suppose if the signal amplitude is very small smaller than the step size,
then the quantized value will be a constant even through the signal is time varying one. Similarly, if the signal
amplitude is greater than the maximum allowable range, that is, if the peak to peak value of the signal exceeds the
VH and VL, then any value greater than the highest quantization level will be limited to the highest quantized value
and if the signal has an excursion below VL, then it is limited to the lowest quantized value VL.
These two cases represent a distortion and limit the dynamic range of the signal that can be
accommodated.Moreover on the previous section we have seen that the SNR of the PCM system is dependent on the
number of levels the signal takes up for its excursion. Therefore to increase the SNR we will have the dynamic range
of the signal.
That is the signals with low amplitudes must be expended so that the awing is large compared with the
must step size. Similarly the signals with the large amplitudes must be compressed so as to bring the signal with in
the range of quantization. Accordingly before applying the signal to the quantizer it is passed through a network
which has the input – output characteristic as shown in figure.

Output V0
V0 max

Compressi
on

Vi min No Compression
6
fi Input Vi

V0 min
From the characteristics it can be readily seen that for low values of input voltage, the slope is large and for large
amplitude of the input signal the slope is small, there by compressing the signal to accommodate the entire signal
within the region of quantization. This process introduces distortion to the signal. But this distortion is
predetermined and therefore at the receiver to restore at the signals to their correct relative values, we pass the signal
through the expander. An expander network has input-output characteristic of the compressor. The combination of a
compressor and expander is called a COMPANDER (COMPressor and expANDER).
In practice while implementing the compressor characteristic, the analog signal is left
unmodified and, instead a non-uniform quantizer is deigned. In non-uniform quantization the step size is made small
so quantization levels are close together at low signal amplitude and progressively as the signal increase in
amplitude.
The use of a non-uniform quantizer is equivalent to passing baseband signal through a compressor and
then applying the compressed signal to a uniform quantizer. One from of compression law used in practices is called
μ law and is defined by

|v0| = log(1+μ|vi|) / log(1 + μ)

Where vi and v0 are normalized input voltages, and μ is a positive constant μ = 0, it corresponds to uniform
quantization (no companding). For a given value of μ the reciprocal slope of the compression curve is

d|vi| / d|vo| = (log(1 + μ ) / μ ) * (1+ μ |vi| )

It can be seen therefore that the μ law is neither strictly linear nor logarithmic but it is approximately linear at low
input voltages corresponding to μ |vi| << 1, and approximately logarithmic at high input voltages corresponding to
μ|vi| >> 1.

One more compression law used is called A- law defined by

|v0| = A|vi| / 1 + log A for 0≤ |vi| ≤ 1/A

= (1 + log (A |vi| ))/ 1+ log A for 1/A ≤ |vi| ≤ 1.

The case of uniform quantization corresponds to A = 1 the reciprocal slope of A-law is given by

d|vi| / d|v0| = (1 + log A) /A for 0 ≤ |vi| ≤ 1/A

= (1+ log A ) |vi| for 1/ A ≤ |vi| ≤ 1

The figure shows the respective curves for different values of A or μ

V0 μ = 100 V0 A = 100
μ = 10 A= 2
μ=1 A
=1
7

Vi
Vi
(a) μ – Law Characteristics, (b) A – Law
Characteristic

Q6. Explain TDM using a block diagram.

Ans:- Multiplexing is a technique for allowing more than one signal to coexist channel. In simple terms
multiplexing means transmission of more than one signal on a signal channel. By multiplexing we can achieve
higher utilization of the channel capacity.
Multiplexing can be achieved by sharing either the frequency or time. If multiplexing is by sharing the
frequency it is called Frequency Division multiplexing. On the other hand if multiplexing is done by sharing the
time it is called Time Division Multiplexing.
Time Division Multiplexing makes use of the sampling theorem, which enables us to transmit the complete
information contained in the band limited signal by using samples of the message signal taken uniform at a rate that
is usually slightly higher than the Nyquist rate. An important feature of the sampling process is a conservation of
time. That is, the message samples engages the transmission channel for only a fraction of the sampling interval on a
periodic basis. This leaves a considerable amount of unused time between the successive samples of a signal and
makes the channel free during that time interval. This free or unused time can be used for transmitting samples of
other signals. If, in this way samples of signals are transmitted through a channel on a time sharing basis it results in
Time – Division Multiplexing (TDM).
The principle of TDM is illustrated in the block diagram shown in figure. The
message signals from different sources are first band limited to the required value by passing them through Low
Pass Filters (LPF) which removes all the signal. The LPF outputs are then applied to a commutator does the
following two important functions.

1. It takes narrow samples of each of the N input massage at the sampling rate, which is slightly higher than the
Nyquist rate usually 20% more than the theoretical minimum is used.
2. It sequentially interleaves these N samples in the sampling interval Ts.

The first function is nothing but the sampling and the second one is multiplexing on a time domain.
Following the commutation process, the multiplexed signal is applied to a pulse modulator. The purpose
of which is to transform the multiplexed signal into a form suitable for transmission over the common channel. The
received signal is applied to a pulse demodulator, which performs the inverse operation of the pulse modulator. the
narrow pulses samples produced at the output of the pulse demodulator are distributed to the appropriate LPFs by
means of a demodulator. Decommutator, which demultiplexes the composite signal, operates in synchronism with
that to the commutator. To achieve synchronization normally one channel is used to transmit synchronization pulses
from the transmitter to the receiver.

The process of multiplexing increase the bandwidth requirements for the transmission. If we have N signals to be
multiplexed, each band limited to fm and sampled at a rate of fs then the bandwidth required for transmission is Nfs.
If fs = 2fm then the bandwidth required is 2Nfm. Thus multiplexing increase the bandwidth by a factor of N. this is
8
due to the fact that the system has to squeeze N samples from N independent sources into a times slot of one
sampling interval.

The TDM system is highly sensitive to dispersion in the common transmission channel, that is, to variations of
amplitude with frequency. This requires accurate equalization of both the amplitude and phase responses of the
channel is necessary to ensure a satisfactory operation of the system.

Message Message
inputs outputs
1 1
LPF LPF
Synchronized

2 2
LPF LPF
Pulse Pulse
Amplitude Amplitude
Modulator Modulator
Commutator Decomutator
Channel

N N
LPF LPF

Q7. What is PSK? Explain using suitable equations the PSK transmitter and receiver.

Ans:- In a binary shift keying (BPSK) the logic 1 and logic 0 are different by the relative phase analog of a
sinusoidal carrier of fixed amplitude and frequency. The BPSK signal has one fixed phase angle for logic 1 and
logic 0 has a phase different of 180o if the sinusoidal carrier has amplitude of A then transmitted carrier can be
expressed as

SBPSK(t) = A cos(ωot) ; for Logic 1


And SBPSK(t) = A cos(ωot) ; For Logic 0
= - A cos (ωot)

if we take the data d(t) is a bit stream of binary digits with voltages levels at + 1V and –1V we can write the
expression for BPSK signal as
SBPSK(t) = d(t) A cos (ωot )

d(t)
+1 Data input

-1
9

BPSK output

180o Phase Shift

BPSK waveform
In practice, a BPSK signal is generated by applying the waveform cos(ωot), as the carrier to a balanced modulator
and applying the base band signal d(t) as the modulation signal. The BPSK signal is therefore like an AM DSB
suppressed carrier signal. The waveform are shown in figure and the functional block diagram of BPSK modulator
is as shown in figure.
The demodulation of BPSK signal is accomplished by the use of a synchronous demodulator as shown in
the figure. A synchronous demodulation is a process of detecting the signal by multiplying the received signal by a
locally generated or recovered carrier that is synchronous with that of the transmitted carrier.

Product BPSK
d(t) Modulator output

Carrier
(A cos ωot)

BPSK modulator

Synchronous demodulation is used for suppressed carrier systems. Since there is no discrete carrier in the BPSK
signal, the carrier recovery circuits are used to synchronous the carrier from the BPSK signal.

The combination of multiplier and an integrator for a period of Tb is called correlator. The corralator output is
applied to a decision-maker, which is supplied with a threshold. If x1 > threshold, the receiver decides in favor of
symbol 1 else it is treated as 0. In this case threshold zero volts.

BPSK Integrated X1 Decision = 1 if X1 > 0;


signal (0 to Tb) Making = 0 if X1 < 0;
device
Recovered
Carrier

Block diagram of coherent BPSK receiver

Q8. What is FSK? Explain using suitable equation the FSK transmitter and receiver.
10
Ans:- In binary frequency shift keying (BFSK) a logic 1 is transmitted as a sinusoid of one frequency, and logic 0
is transmitted as a sinusoid of another frequency with fixed amplitude.

In general we can write the expression for a BFSK signal as


SBFSK (t) = A cos (ωo t + d(t) Ωt)
Here d(t) is +1 or –1 corresponding to the logic levels 1 and 0 of the data waveform. The transmitted signal is
therefore

SBFSK(t) = A cos (ωot + Ω)t ; For logic 1


SBFSK(t) = A cos (ωot + Ω)t ; For logic 0

And thus has angular frequency ω0 + Ω or ω0 – Ω with Ω a constant offset from the nominal carrier frequency ω0.
We shall call the higher frequency ωH = ω0 + Ω; and the lower frequency ωL =ω0 – Ω.

Or in terms of linear frequency f1 = (ω0 + Ω) / 2π and f2 = (ω0 + Ω) / 2π .

d(t)
+1 Data input

-1

BFSK output

Frequency f1 Frequency f2

BPSK waveform

That is in binary frequency shift keying. We switch back and forth between sinusoids of two different frequency,
depending on whether the logic 1 or 0 is being transmitted.
Conceptually the BFSK signal can be generated as shown in the figure. An alternate method of generating BFSK
signal is by making use of VCO (Voltage Controlled Oscillator). A VCO produces an output signal whose frequency
is proportional to the magnitude of the input voltage at its control input. When the control voltage applied to the
VCO is the binary waveform, the output will have only two frequencies; one for logic 1 and the other for logic 0.
Which is the required BFSK signal.

Oscillator
Frequency = f1 BFSK
Output VCO BFSK
d(t) Output

Oscillator
Frequency = f2 Control
11
Line

Binary data input d(t)

Generation of BFSK signal

One since implementation of the demodulator for the BFSK signal is straightforward. The BFSK signal consisting of
two frequency f1 and f2 is applied to two bandwidth pass filters connected in parallel as shown in the figure. One
band pass filter is centered at f1 and the other one at f2. When the input corresponds to logic 1 the frequency is f1
and hence the band pass filter with center frequency f1 will produce a output voltage where as the output of the other
band pass filter will be almost zero. Similarly when the input corresponds to logic o the band pass filter with center
frequency f2 will produce a output while the previous filter output will be zero. These outputs are applied to two
envelope detectors, which will detect the envelope or the amplitude variations at its input. The outputs of the
envelope detector are applied to a comparator. The comparator on whether the input 1 is stronger or the other.
Therefore the output of the comparator is the binary waveform as required.

Band ass
Filter Envelope
Centered detector Comparator
around f1

BFSK d(t)
In
Band ass
Filter Envelope
Centered detector
around f2

Non- coherent Demodulation of BFSK signal

Though the envelope detector (non-coherent type of detection) is simple, its performance of noise is poor. Therefore
coherent detection as shown in the figure is often employed. Since in the case of BFSK we have two frequency
components, one corresponding to logic 1 and the other supplied to logic 0, we use two correlators in parallel. The
upper correlator is supplied by a locally generated coherent reference signal Ф1(t) and the lower correlator is
supplied by another locally generated reference Ф2(t) as shown in figure. The output of upper correlator will be
greater than that of the lower one if the bit transmitted is 1. On the other hand if the lower one. With the threshold
taken as zero, the decision will be lesser than that of the lower one. With the threshold taken as zero, the decision-
maker provides an output of logic 1 if X1+ X2 is greater than zero, and zero other wise.

Integrator X1
(0 to Tb)

Ф1(t) Decision Output


BFSK Adder Making Binary
Input device wave

Integrator X2
12
(0 to Tb)

Ф2(t)

Coherent detection of BFSK

Q9. What do you mean by Inter – Symbol Interference?How to reduce ISI ?

Ans:- A rectangular pulse train can be expressed as an infinite number of sinusoidal frequency components using
Fourier series. From the bandwidth point of view this means infinite bandwidth for transmission. If we use a
bandwidth less than this rectangular pulse will not remain rectangular. Any reduction in bandwidth, as is done in
practical situation, will result in rounding off the pulses. The lesser the bandwidth the more will be the rounding off
and hence the distortion. It also means that the pulses will stretch or there will be spilling over of the energy of the
pulses to the neighboring spectrum. This causes interference and is called inter symbol interference.
In the figure the dotted line represented the ideal pulse and the solid line shows the
rounding off effect and inter and symbol interference due to the use of lower bandwidth.

Distorted Ideal
Pulse Pulse

Effect of the ISI

As more and more pulsed channels are multiplexed high frequency response and phase response become more
critical. Also when the distortion on one pulse affects the next pulse, it requires in what called cross talk between the
channels.
To overcome these problems and to make the channel response closer to the ideal
channel, the transmission lines are compensated using

networks with complimentary characteristic called to end the process is called equalization. Channel filters with
raised cosine roll off characteristic are also used with roll off factor r where 0< r ≤ 1. The amplitude and frequency
response is as shown in the figure.

| H(f)|

r=0
1
r = 0.5

0.5 r=1

Bmin 2Bmin f
13
Frequency response of the filter with raised cosine roll off
characteristics

Bmin is the absolute minimum bandwidth required. This is the case for an ideal channel.
Maximum data rates of rectangular shaped pules that can be supported with
out ISI in a channel having the transfer function H(f) is given by

fb(max) = 2Bmin / 1+r


where Bmin is a channel in Hertz and fb(max) is in pulses/sec.

Q10. What is spread spectrum? Why it is used? What are the difference
types?Explain.

Ans.: - Generally every communication system aims at achieving the following criteria:

• To limit the bandwidth used up by the signal to just that amount it needs for its data transmission.
• To confine or focus its output power only in that narrow band.
• To avoid interference with other signals in the same band.

The communication systems following above concepts are not immune to external tapping or jamming of
information. Therefore the secrecy cannot be maintained. In military communications, which requires high degree of
security to its information to be communicated, the general simple communication system will not be sufficient. So
is the case in application like cellular telephones and personal communication, where large number of users shares a
band of frequencies. In such applications w need multiple access capability because there is not enough user. The
concept of spread spectrum, which is quite different from the general communication system, gains importance in
such applications.
The main advantage of the spread spectrum communication system is its ability to
reject interference whether it to be unintentional interference whether it to be unintentional interference of another
user simultaneously attempting to transmit through the channel, or the intentional interference of a hostile
transmitter attempting to jam transmission.

Few applications where spread spectrum is extensively used are listed below.
• Military communication
• Cellular phones
• In the micro cells of LAN
• For transmitting the measured parameters such as temperature, pressure etc. from the site to the central
computing center
• In the internet
• Satellite communication

In spread spectrum systems the spectrum of signal is deliberately spread across a wide band. Only a receiver with
the corresponding characteristic will be able to receive the signal and no other receiver can detect the signal. More
over jamming of the signal is also not possible.

Spread spectrum is a techniques where by an already modulated signal is modulated second time in such a way as to
produce a waveform which interferes in binary noticeable way with any other signal operation in the same frequency
band. That is the signal power is spread over a large bandwidth and appears like noise so that unauthorized receivers
can be neither demodulated it nor jam the signal, thus a receiver turned to receiver a specific AM (Amplitude
Modulation ) or FM (Frequency Modulation ) broadcast over the same frequency band. Similarly, the receiver of
the spread spectrum would not notice the presence of an AM or FM signal.

Thus we can say that the interfering signals are transparent to the spread spectrum and the spread spectrum is
interfering signals. As an example let take an ordinary AM signal X(t) that utilities a bandwidth of 10KHz. And
14
power Ps. Let us consider another S(t) having the same carrier frequency and power but a bandwidth 1MHz due to
the process of spreading. Then, in the 10KHz. Bandwidth of AM signal X(t), the power of the second signal is Ps *
(10KHz. / 1 MHz ) = Ps/ 100. This amplitude is comparable with the noise and hence will be not contribute to signal
power and provides the transparency.

Formally we can define the spread spectrum as follows, in parts:

1. Spread spectrum is a means of transmission in which the data of interest occupies a bandwidth in excess of the
minimum bandwidth necessary to send the data. (That is, the band width of the transmitted signal say s(t) must
be much greater then that of the message signal, m(t)) .
2. The spectrum spreading is accomplished before transmission through the use of a code that is independent of
the data sequence. The same code is used at the receiver to de-spread the signal so that the original data can be
recovered.

It is worth nothing that, the very process of any kind of modulation spreads the spectrum. This satisfies the part 1 of
the definition. Still they are not spread spectrum techniques, as they don’t satisfy the second part of the definition.
That is, in the case of general modulation techniques the amount of spreading in dependent on the data where as, in
the case of spread spectrum it is independent of data.
Figure shows a generic block diagram of spread spectrum communication. The base
band signal m(t) is first modulation by conventional methods using a sinusoidal carrier. Usually a PSK or FSK is
employed. For getting a spread spectrum an additional modulation by a high rate discrete pseudorandom code is
performed. The pseudorandom code sequence, say c(t) must a rate Rc much grater than the binary message rate Rm
of the base signal m(t). That is Rc >> Rm by two or there orders of magnitude (X100 – X1000). Also the
pseudorandom code c(t) must me statistically independent of the message signal m(t). if these two conditions are
then the message signal spectrum will be spread by an amount given by

G = Rc / Rm

Where G is called the processing gain, and the resulting spread spectrum signal can be accurately demodulated only
if the receiver possesses a matched de-spreading circuit using c(t). There are two distinct of spread spectrum
systems: Frequency Hopping Spread Spectrum (FHSS) and Direct Sequence Spread Spectrum (DSSS).

Modulator
DS or FH
Carrier Spread spectrum
Spread Spectrum
Modulator
Signal

Binary Spreading
Message Code c(t)
m(t)

Generic block diagram of a spread spectrum modulation

THANK YOU

You might also like