CS2204 ANALOG AND DIGITAL COMMUNICATION

DEPARTMENT OF ECE

CS2204 ANALOG AND DIGITAL COMMUNICATION

Question Bank
UNIT I : FUNDAMENTALS OF ANALOG COMMUNICATION
PART-A:
1. What is the need for modulation?
It is extremely difficult to radiate low frequency signals through earths
atmosphere in the form of electromagnetic energy.
At low frequency, the antenna size required becomes impractical.
Information signals often occupy the same frequency band. Signals from two
or more sources would interfere if they are not modulated and translated to a
different frequency band.
2. With reference to AM, define modulation index (or) depth of modulation.
It is defined as the ratio of peak amplitude of the message to the carrier signal.
m=

Em
Ec , where E m = peak amplitude of modulating signal voltage

E c = peak amplitude of the unmodulated carrier voltage

3. A broadcast radio transmitter radiates 5 kW power when the modulation
percentage is 60%. How much is the carrier power?
Pt = Pc(1+m2/2)
Pc= Pt/(1+m2/2) = 5000 / (1+0.62/2) = 4237.28w
4. What is the relationship between total power in AM wave and unmodulated
carrier power?
Pt = Pc(1+m2/2)
Pc=unmodulated carrier power
Pt=total power
m=modulation index
5. What is the relationship between total current in AM wave and unmodulated
carrier current?
It =Ic(1+m2/2)
Ic= carrier current
It=total current
m=modulation index
6. An unmodulated carrier is modulated simultaneously by three modulating signals
with coefficients of modulation m1 = 0.2, m2 = 0.4, m3 = 0.5. Determine the total
coefficient of modulation.
mt = m12 +m22 +m32 = 0.22+0.42+0.52 =0.67
7. Define amplitude Modulation.
Amplitude Modulation is the process of changing the amplitude of a relatively
high frequency carrier signal in proportion with the instantaneous value of the
modulating signal.
8. Define Modulation index and percent modulation for an AM wave.
Modulation index is a term used to describe the amount of amplitude
change present in an AM waveform .It is also called as coefficient of modulation.
Mathematically modulation index is m = Em/Ec,
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DEPARTMENT OF ECE

Where m = Modulation coefficient

Em = Peak change in the amplitude of the output waveform voltage.
Ec = Peak amplitude of the unmodulated carrier voltage.
Percent modulation gives the percentage change in the amplitude of the output
wave when the carrier is acted on by a modulating signal.
9. What is Frequency modulation?
Frequency of carrier is varied in accordance with amplitude of modulating
signal.
10. What is Phase modulation?
Phase of carrier is varied in accordance with the amplitude of modulating signal.
11. What is Bandwidth of AM wave?
Band width is difference between highest upper side frequency and lowest
lower side frequency. B.W = 2fm(max).
12. What is over, under, critical modulation?
If m >1, has severe distortion. This condition is Over modulation. If m=1, has
greatest output and condition is Critical modulation. If m< 1 ,has no distortion and
condition is Under modulation.
13. Draw AM envelope with Vmax and Vmin?

14. With reference to FM, define modulation index.

Modulation index is the ratio of frequency deviation and modulating signal
frequency.
m = f / fm
f = frequency deviation in Hz
fm = modulating signal frequency in Hz
15. Define deviation ratio.
It is the worst-case modulation index which is the ratio of maximum
permitted frequency deviation and maximum modulating signal frequency.
Deviation ratio = f(max) / fm(max)
16. State Carsons rule for determining approximate Band Width of FM signal.
Carson rule states that the bandwidth required to transmit an angle modulated
wave as twice the sum of the peak frequency deviation and the highest modulating signal
frequency.
Band Width = 2 [ f + fm(max) ] Hz
f = frequency deviation in Hz
fm(max) = highest modulating signal frequency in Hz
17. A carrier is frequency is frequency modulated with a sinusoidal signal of 2 KHz
resulting in a maximum frequency deviation of 5 KHz. Find the approximate band
width of the modulated signal.
f = frequency deviation in Hz = 5 KHz
fm(max) = highest modulating signal frequency in Hz = 2 KHz
Band Width = 2 [ f + fm(max) ] Hz = 14 KHz
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18. Determine the modulation index of a FM system with a maximum frequency

deviation of 75 KHz and maximum modulating frequency of 10 KHz.
m = f / fm = 75 KHz/ 10 KHz = 7.5
19. Distinguish between narrow band FM and wide band FM.
Narrow band FM
Wide band FM
Frequency deviation in carrier Frequency deviation in carrier
frequency is very small
frequency is large
Band width is twice the highest Band width is calculated as per
modulating frequency
Carsons rule
20. What are the advantages of FM over AM?
The amplitude of FM is constant. Hence transmitter power remains
constant in FM where as it varies in AM.
Since amplitude of FM is constant, the noise interference is minimum in
FM. Any noise superimposing on modulated carrier can be removed with
the help of amplitude limiter.
The depth of modulation has limitation in AM. But in FM, the depth of
modulation can be increased to any value.
Since guard bands are provided in FM, there is less possibility of adjacent
channel interference.
Since space waves are used for FM, the radius of propagation is limited to
line of sight( LOS ) . Hence it is possible to operate several independent
transmitters on same frequency with minimum interference.
Since FM uses UHF and VHF ranges, the noise interference is minimum
compared to AM which uses MF and HF ranges.
21. What is the advantage and disadvantage of Angle modulation?
Advantages: 1. Noise Reduction
2. Improved system fidelity
3. More effective use of power
Disadvantage: 1. Require more Bandwidth
2. Use more complex circuits in both transmitter and receiver
22. Draw the FM waveform?

23. Define percent modulation?

Percent modulation = [actual frequency deviation/max allowable frequency deviation]
*(100)
24. A Transmitter supplies 8KW to the antenna when unmodulated. Determine the
total power when amplitude modulates to 30%.
Pt=Pc(1+ma2 /2)
=8x103 (1+0.32/2)=8.36kw
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25. What is the main difference b/w frequency modulation and phase modulation?
Frequency modulation: It is the form of angle modulation in which instantaneous
frequency fI(t) is varied linearly with the base band signal m(t)
Where,fI (t)=fc+kf m(t) fc unmodulated carrier
kf Frequency sensitivity of the modulator
m(t)-Base band signal
Integrating above equation with respect to time and multiplying with 2
i(t)= 2fc t+2Kf m(t) dt
s(t)=Ac cos i (t)
s(t)= Ac cos(2fc t+2Kf m(t) dt)
Phase modulation:
It is that form of Angle modulation in which angle i(t) is varied linearly with
the base band signal m(t) as as shown by
i(t)= 2fc t+Kpm(t)
s(t)=Ac cos i (t)
s(t)= Ac cos(2fc t+Kpm(t)
26. Determine the modulation depth of FM system with a maximum frequency
deviation of 75 KHz and the maximum modulating frequency of 10 KHz
=f /fM
3
3
=75 x10 /10 x10
=7.5
27. Write down the expression for FM signal with sinusoidal modulation
Frequency modulation: It is the form of angle modulation in which instantaneous
frequency fI(t) is varied linearly with the base band signal m(t)
Where,fI (t)=fc+kf m(t) fc unmodulated carrier
kf Frequency sensitivity of the modulator
m(t)-Base band signal
Integrating above equation with respect to time and multiplying with 2
i(t)= 2fc t+2Kf m(t) dt
s(t)=Ac cos i (t)
s(t)= Ac cos(2fc t+2Kf m(t) dt)
28. Define instantaneous frequency deviation.
The instantaneous frequency deviation is the instantaneous change in the frequency of the
carrier and is defined as the first derivative of the instantaneous phase deviation.

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DEPARTMENT OF ECE

PART B:
1. Obtain AM wave equation and explain each term with the help of frequency
spectrum and also obtain an expression for its bandwidth?
In amplitude modulation, amplitude of the carrier wave is changed according to the
amplitude of the signal. The technique is very much used in the transmission of radio
signals.
4 shows the process of amplitude modulation.

Note that
Only amplitude of the carrier is varied while its frequency and phase remain unchanged.
Whenthere is no signal, the amplitude of the carrier is equal to the unmodulated
amplitude. When signal is present, the amplitude of the carrier changes in accordance
with the instantaneous value of the signal.
During positive cycle of the signal, the amplitude of the carrier increases to the sum of
the amplitudes of the carrier and the signal (E + Em)
(iv) During negative cycle of the signal, the amplitude of the carrier decreases and
becomes equal to the difference of the amplitudes of the carrier and the signal.

CS2204 ANALOG AND DIGITAL COMMUNICATION

DEPARTMENT OF ECE

CS2204 ANALOG AND DIGITAL COMMUNICATION

DEPARTMENT OF ECE

Note that the equation has three parts:

First part is an unmodulated carrier wave, which remains unchanged in the process.
The maximum amplitude is E.
Second part has a maximum amplitude of mE/2 and its frequency is equal to the
difference of carrier and the signal frequencies. This is called lower side band (L.S.B.).
Recall that angular velocity of the carrier o 2rf, where f is the frequency of the carrier.
Similarly tOrn = fm
Third part has also max. amplitude of mE/2 and frequency equal to the sum of carrier and
signal frequencies. This is called upper side band (U.S.B.) (See Figure 6b).

Figure 6 shows frequency spectrum of an A.M. wave which is equivalent to three sine
waves as shown. Bandwidth (B.W.) of an A.M. wave
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Hence, in amplitude modulation, the bandwidth is twice the signal frequency.

2. Define modulator factor/Index/depth of modulation. Derive expression of AM
wave is displaced on a CR0. Explain its significance.
Ans. (a) Modulation Factor/Index (m) : The modulation factor/index, (m) can be defined
in the following ways.
1. It is the ratio of maximum value of the signal to the maximum value of the carrier, i.e.,
2.m Em/Eci or = mE
It is the ratio of the change in the amplitude of the carrier to its original amplitude, i.e.,
4. It is the ratio of minimum amplitude to the maximum amplitude to the modulated
(ratio) wave as shown below.
(b) If modulation curve is displayed on a cathode ray oscilloscope (CR0), we get the
curve as shown in figure 6. The modulation index m can be calculated as follow:
m AEC/EC
3. It is the percentage change in the amplitude of the carrier, i.e.,
m = 1LEC/EC)< 100
Hence m can be expressed as the ratio of minimum amplitude to the maximum amplitude
of the radio wave.
Note:
1. The vkof m lies between 0 and 1.
2. T){value of m depends upon the amplitudes of the signal as well as of the carrier.
Significance of m (Table 1)
The modulation factor m plays a very important role in the modulation process. This will
be made clear by calculating the values of m for different amplitudes of the signal and the
carrier.
1. Let the amplitude of signal be zero (i.e. signal is not present) and amplitude of carrier
is E. In this case, the amplitude of modulated wave 0 +E = E.
Change in the carrier amplitude E - E = 0. Modulation index 0/Er = 0 (No modulation)
2. Let the amplitude of carrier = E and amplitude of signal = E/2.

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DEPARTMENT OF ECE

Hence m depends on the amplitude of both the signal and the carrier.
The value of m decides the strength of the modulated wave and hence that of the signal.
When rn = 1(100%) the signal will be strongest, perfect and clear. In the case of
overmodulation (m = 150%) the modulated wave will be clipped..off arid huge distortion
will occur in the reception. Hence, the ideal value of modulation is I or 100%.
3) Derive AM power and obtain current relation between amplitude modulated and
unmodulated wave.
Ans. We know that the power contained in a voltage wave is proportional to the square of
its amplitude (V2). Note that an A.M. wave is a voltage wave.
The total power contained in an A.M. wave will be the sum of the powers contained in
the three parts of the wave.
Considering root mean square (R.M.S) values

Power contained in the carrier

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DEPARTMENT OF ECE

Note that power in both the sidebands is equal and at m = 1, the sidebands contain
1/3rd (33%) power and the carrier contains 66% of the total power. Hence bands
carry half the carrier power of the wave. (d) Total power contained in two side bands = 1/3 = 33%
Total power contained in each side band = 1/6 = 16.5%. V
As the signal is contained only in the sidebands, useful power is contained in
sidebands. This is the reason, that we are interested only in the sidebands. The power
in the sidebands go on increasing with the increase in the modulating index (m).
CURRENT RELATION:
Let l = Unmodulated current (carrier current)
1T = Modulated current of an A.M. wave (both in R.M.S. values) and R = Resistance
through which current flows. Assume it to be the same in both the cases.

4) Write a note on evolution of SSBAM techniques.

Ans. (a) In the theory of amplitude modulation (AM), we have seen that a
carrier and two sidebands (SBs) are required for AM transmission. But it is not necessary
to transmit all the three signals (1 carries and 2 sidebands). The carrier or one of
the sidebands may be removed (or attenuated). The SSB modulation is the fastest
spreading form of analog modulation. The greatest advantage is its ability to transmit
signals by using a very narrow band, width, very low power for the distances involved.
For 100% modulation (m = 1), only 113rd of the total power is present in one of
the sidebands, while 2/37d power is carried by the carrier, which contains no
information. Thus if the carrier and one of the sidebands is eliminated from the signal, the
transmission will need only 1/6th of the total power.
The Fig. 1 shows double sideband with full carjier (DSBFC) and (b) shows

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double sideband with supressed carrier (DSBSC) and (c) shows single
sideband transmission with supressed carrier (SSBSC). It can be noted that (c) requires
only half the bandwidth (BW) as required to (a) and (b).

(b) Evolution of SSBAM : The evolution of SSB amplitude modulation may be done in
following steps:
(i) The carrier contains no power and all the power is contained in the sidebands.
(ii) Therefore there is no need to transmit carrier. V.
(iii) The modulated wave contains three frequencies f f + f fc - fm
(iv) Two sidebands are exact images of each other; since each is affected by changes in
the modulating voltage via the exponent m EJ2. Recall that m is the
modulation index and E the carrier voltage.
(v) Therefore all the information may be transmitted by the use of one sideband only, as
the carrier is superfluous and the other sideband is redundant.
(vi) If the carrier is supressed only the two sidebands power remains and which is
equal to = Pc.m2/4 about 66% saving will be done. Recall that is the carrier
power.
(vii) It one of the sidebands is also supressed, the remaining power is Pc. m2/4 a
further saving of 50% power will be achieved.
5) Write down the expression for FM and PM waves and draw their

frequency spectrum and explain. Obtain the mathematical expressions for

AM & FM modulated waves & draw the necessary waveforms in both cases
Frequency Modulation (FM) In FM transmission, the frequency of the
carrier signal is modulated to follow the changing voltage level amplitude of the
modulating signal. The peak amplitude and phase of the carrier signal remain constant,
but as the amplitude of the information signal changes, the frequency of the carrier
changes correspondingly.
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Phase Modulation: PM is used in some systems as an alternative to frequency

modulation in PM transmission, the phase of the carrier signal is modulated to follow the
changing voltage level of the modulating signal. The peak amplitude and frequency of the
carrier signal remains constant, but as the amplitude of the information signal changes,
the phase of the carrier changes correspondingly.

Unit II : DIGITAL COMMUNICATION

PART-A
1. Define ASK and FSK.
ASK: A binary information signal directly modulates the amplitude of an analog
carrier.
FSK: The frequency of a sinusoidal carrier is shifted between two discrete values.
2. Define bit time and baud rate.
Bit time: It is the reciprocal of the bit rate
Baud rate: The rate of change of a signal on the transmission medium after
encoding and modulation have occurred.
Baud = 1/ts
3. Define DPSK.
DPSK is an alternative form of digital modulation where the binary input
information is contained in the difference between two successive signaling elements
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rather than absolute phase .It combines two basic operations namely ,differential
encoding and phase shift keying.
4. Define QPSK
.
QPSK: The two successive bits in a bit stream ar combined together to form a message
and each message is represented by a distinct value of phase shift of the carrier. Each
symbol or message contains two bits so the symbol duration Ts =2Tb.These symbols are
transmitted by the same carrier at four different phase shifts as shown below.
Symbol
Phase
00
-135
01
-45
10
135
11
45
5. What is a constellation diagram? Draw the constellation diagram and phasor
diagram for BPSK.
Constellation diagram is used to show the relative positions of the peaks of the
phasors.

Phasor diagram:
6. Draw the phasor diagram of QPSK signal.

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constellation diagram

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DEPARTMENT OF ECE

7. What is the minimum bandwidth required for an FSK system?

Bandwidth required=fm-fs+ 2/tb
1/tb =fb=bit rate,fm=mark frequency,fs=space frequency
8. What is the primary advantage of DBPSK and what is its disadvantage?
Advantage: simple implementation. No carrier recovery circuit needed for
detection.
Disadvantage: It requires between 1 dB and 3 dB more signal to noise ratio to
achieve the same BER as that of standard absolute PSK
9. What are the advantages of M-ary signaling schemes?
M-ary signaling schemes transmit multiple bits at a time.
Bandwidth requirement of M-ary signaling schemes is reduced.
10. Compare binary PSK with QPSK.
BPSK
QPSK
Binary Phase Shift Keying
Quadrature Phase Shift Keying
One bit form a symbol
Two bits form a symbol
Two possible symbols
Four possible symbols
Minimum bandwidth required = f b Minimum bandwidth required = f b / 2
where f b is bit rate
where f b is bit rate
11. What are the advantages of QPSK as compared to BPSK?
For the same bit rate, the bandwidth required by QPSK is reduced to half as
compared to BPSK.
Because of reduced bandwidth, the information transmission rate of QPSK
is higher.
12. What happens to the probability of error in M-ary PSK as the value of M
increases?
As the value of M increases, the Euclidean distance between the symbols
reduces. Hence the symbols are closer to each other. This increases the probability
of error in M-ary systems.
13. What is the minimum bandwidth required for BPSK, QPSK, 8-PSK, 8-QAM
and 16-QAM systems if the bit rate is 10 MBPS?
system
BPSK
QPSK

15

Minimum band width

required
if f b = bit rate
fb
fb / 2

Minimum band width

required
if f b = 10 Mbps
10 MHz
5 MHz

CS2204 ANALOG AND DIGITAL COMMUNICATION

8 - PSK
8- QAM
16 - QAM

fb / 3
fb / 3
fb / 4

DEPARTMENT OF ECE
3.33 MHz
3.33 MHz
2.5 MHz

14. What is difference between coherent and non coherent detection?

Coherent detection
Non- Coherent detection
Carrier which is in perfect No carrier recovery circuit needed
coherence with that used in for detection.
transmitter
is
used
for
demodulation.
Carrier recovery circuit is
needed for detection
Relatively complex
Simple implementation
15. Define Bandwidth efficiency. What is the bandwidth efficiency of BPSK and 8PSK system?
It is the ratio of the transmission bit rate to the minimum bandwidth required for a
particular modulation scheme.

For BPSK , transmission rate = f b and minimum bandwidth = f b

Band width efficiency = 1
For 8-PSK , transmission rate = f b and minimum bandwidth = f b/3
Band width efficiency = 3
16. What is the difference between probability of error P(e) and bit error rate
BER?
P(e) Probability of error is a theoretical (mathematical) expectation of the
bit error rate for a given system.
BER is an empirical record of a systems actual bit error performance.
For Example, if a system has a P(e) of 10 -5 , this mean that, you can expect one
bit error in every 100,000 bits transmitted.
If a system has a BER of 10 -5 , this mean that, there was one bit error for every
100,000 bits transmitted.
BER is measured and then compared to the expected probability of error to
evaluate the systems performance.
17. What is the probability of error for (i) non-coherent FSK and (ii) coherent FSK?
Compare their error performance.

For a given energy per bit to noise power density ratio, probability of error for
non-coherent FSK is greater than that of coherent FSK.
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18. Define ( Eb / N0 ) Energy per bit to Noise power density ratio.

Energy per bit to noise power ratio is used to compare two or more digital
modulation systems that uses different bit rates and modulation schemes.
It is the product of carrier to noise power ratio and the noise band width to bit rate
ratio. This is equivalent to signal to noise ratio.
Eb
=energyperbittonoisepowerdensityratio
N0
Eb =Energyperbit=CT b
whereC=carrierpowerinwatt
T b =bitdurationin sec
N
N 0 =Noisepowerdensity=
B
whereN =thermalnoisepowerinwatt
B=noisebandwidthinHz
Eb C B
=
N0 N f b
19. List out the advantages and disadvantages of QPSK.
Advantages: low error probability, good noise immunity, baud rate is half of the bit rate
Disadvantages: very complex to generate and detect the signal
20. Define carrier recovery.
It is the process of extracting a phase coherent reference carrier from a received signal.
PART B
1) Explain ASK system
In Binary ASK system, two binary (1, 0) symbols are represented by two different
amplitudes (A? 0) of the carrier frequency f Biiary symbol 1 is represented by the
presence of a constant amplitude of carrier for T, second where as other binary symbol o
is represented by the absence of the carrier for T second. This signal can be generated
simply by turning on and off the carrier of a sinusoidal oscillator for the prescribed
periods shown in modulating pulse train. For this, this technique is also called as on-off
keying (0 0 K).

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2) Explain BPSK Transmitter and receiver with a neat diagram.

In Binary Phase Shift Keying system, the sinusoidal carrier wave of fixed
frequency f and fixed amplitude A is used to represent binary symbols 0 and 1 except that
the phase of carrier for one (1) differs by a phase angle of 180 to the phase of carrier for
zero (0).

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Here, it may be noted that unlike ASK transmission, the P5K transmission is polar.
Polarity changes in the binary signal b(t) are used to produce 1800 changes in the carrier
phase. This may be achieved by using double sideband, suppressed carrier modulation
(DSBSC), with binary signal as a polar NRZ waveform. The carrier amplitude is
multiplied by 1, pulsed waveform. When the binary signal b(t) is + 1, the carrier
sinusoid is unchanged, and when b(t) is -1, the carrier sinusoid is changed in phase by
180. Binary phase shift keying is also known as Phase reversal keying (PRK).

3) Explain BFSK Transmitter and receiver with a neat diagram.

Frequency shift keying is the oldest and simplest method of modulation used in modems.
Two sinusoidal frequencies are used to represent binary symbols Os and is. For example,
a frequency of 1070 Hz for space in data communication is used to represent binary 0
and a frequency 1270 Hz for mark is used to represent binary 1. These two frequencies
1070 Hz and 1270 Hz are used to represent binary signal 0 and 1 respectively for
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transmission of binary data as shown in figure 34. These two frequency tones are well
within the 300 to 3400 Hz band width associated with telephone system.

Another set of frequencies are required to permit simultaneous transmit and receive
operation with a modem, known as full duplex operation. A frequency 2025 Hz is used to
represent a binary 0 or space and frequency 2225 Hz used to represent binary 1 or
mark. These two frequency tones are spaced far enough from the other frequencies (1070
Hz and 1270 Hz) as shown in Fig. 35. To distinguish between these

two sets of frequencies selective filters are used. The frequency set consists of 1070 and
1270 Hz tones is used for transmitting and frequency set of 2025 and 2225 Hz tones used
for receiving purposes.
4) Compare the difference between ASK, FSK, PSK.
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5) Explain the operation of QAM.

Quadrature Amplitude Modulation (QAM): Quadrature amplitude

modulation is a combination of ASK and P so that a maximum contrast
between each signal unit (bit, dibit, tribit and so on) is achieved.

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The number of amplitude shifts is fewer than the number of the phase shifts.
Because amplitude changes are susceptible to noise and require greater shift
difference than do phase changes, the number of phase changes used by
QAM system is always large the number of amplitude shifts.
The Quadrature Amplitude Modulation (QAM) is called Quadrature
Carrier
Multiplexing
(QCM).
Further, this scheme enables two DSBSC (Double side band with supressed
carrier) modulated signals topy the same B.W. and therefore it allows the
two signals separation at the receiver; therefore, it is also known as Band
width
conservation
Scheme
(BCS).
(i) QAM Modulator Transmitter. It consists of two separate balanced
modulators, which are supplied with two carrier waves of the sme
frequency
and
differing
by
quadrature
(i.e.
by
900).

The output of the balanced modulators is given to an circuit, and we get the
output (multiplexed) signal, which is transmitted. This signal consists of in
phase
and
the
quadrature
phase
components.
QAM detector/Receiver. The transmitted (multiplexed) signal is applied
simultaneously to two separate coherent detectors, which are supplied with
two local carriers of same frequency but differing in phase by 900. The
outputs of the detectors are given to low pass filters (LPF) which gives the
two
original
massage
signals.
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For satisfactory operation of the receiver, the phase and frequency of the
local oscillators at transmission and reception should be coherent (or
aligned)
to
each
other.
(b) Application. The QAM system is used exclusively for digital modulation
of analog carriers in data modems (modulation-demodulation) to convey
data through public telephone network (PTN). The QAM is also used for
digital satellite communication systems and in colour televisions.

UNIT III : DIGITAL TRANSMISSION

PART-A
1. State sampling Theorem.
If a finite energy signal g(t) contains no frequency component higher than W Hz, it is
completely determined by specifying its ordinates at a separation of points spaced
1/ 2W seconds apart.
2. Give the methods of Sampling
1. Ideal Sampling (or) Instantaneous Sampling
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2. Natural Sampling
3. Flat-top Sampling
3. What is Aliasing or Foldover?
When the continuous time signal g(t) is sampled at the rate less than Nyquist rate,
frequencies higher than W takes on the identity of the low frequencies in sampled signal
spectrum . This is called aliasing.
The use of a low pass reconstruction filter, with its pass band extending from W
to W will not yield an undistorted version of the original signal. Aliasing can be reduced
by sampling at a rate higher than Nyquist rate.
In other words, Aliasing occurs when the signal is sampled at a rate less than Nyguist
rate (2W samples/ sec). It is prevented by using
Guard Bands
Pre-alias Filter
4. Define Nyquist rate and Nyquist interval.
According to sampling theorem, a continuous time signal can be completely
represented in its samples and recovered back if the sampling frequency is fS 2W. Here
fS is sampling frequency and W is the highest frequency component of the signal.
Nyquist rate: The minimum sampling rate of 2W samples per second is called Nyquist
rate.
i.e., fS = 2W Nyquist rate
Nyquist interval: Reciprocal of 2W is called the Nyquist interval.
Nyquist interval = 1/2W
5. Give the practical procedure for the sampling of a signal whose spectrum is not
strictly band limited.
i) Prior to sampling, a low pass pre-alias filter ( anti-alias filter) of high enough
order is used to attenuate those frequency components of the signal that do not contribute
significantly to the information content of the signal.
ii) The filtered signal is sampled at a rate slightly higher than the Nyquist rate 2W,
where W is the cutoff frequency of the pre-alias filter.
6. Define aliasing error. Give the upper bound for the aliasing error.
Let {g(n/fS)} denote the sequence obtained by sampling an arbitrary signal g(t) at
the rate fS samples per second. Let gi(t) denote the signal reconstructed from this
sequence by interpolation;
That is, gi(t)= g( ) sinc ( fSt - n)
The absolute error = | g(t) gi(t) | is called the aliasing error.
The aliasing error is bounded as
2 | G(f) | df
7. Given the signal m(t)=10 cos (2000 t) cos (8000 t ) , what is the minimum
sampling rate based on the low pass uniform sampling theorem?
The equation shows that m(t) is generated by multiplication of two signals. We
know that cosAcosB = [cos (A-B)+cos (A+B)]
There fore, m(t) = (10/2) [ cos (6000 t) + cos (10000 t)]
The two frequencies in m(t) are 3000 Hz and 5000 Hz and the highest frequency present
in m(t) is 5000 Hz.
Minimum sampling rate is 2 (5000) = 10000 samples per second.
8. What is Inter symbol Interference (ISI) ?
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ISI arises because of imperfections in the overall frequency response of the

system.
When a short pulse of duration Tb seconds is transmitted through a band limited
system, the frequency components constituting the input pulse are differentially
attenuated and, more significantly, differentially delayed by the system. Consequently the
pulse appearing at the output of the system is dispersed over an interval longer than Tb
seconds.
Thus when a sequence of short pulses are transmitted through the system, one
pulse every Tb seconds, the dispersed responses originating from different symbol
intervals will interfere with each other, there by resulting in ISI.
9. What is eye pattern?
When the received signal in a data transmission system is applied to the vertical
deflection plates of an oscilloscope and saw tooth wave( frequency = symbol rate) is
applied to the horizontal deflection plates of an oscilloscope, the resulting display is
called an eye pattern.
The display pattern is called an eye pattern because of its resemblance to the
human eye. An eye pattern provides a great deal of information about the performance of
the data transmission system. Eye pattern is particularly useful in studying ISI problem.
10. List out the information provided by eye pattern about the system performance.
i) The width of the eye opening defines the time interval over which the received wave
can be sampled without error from ISI. The preferred time for sampling is the instant of
time at which the eye is open widest.
ii) The sensitivity of the system to timing error is determined by the rate of closure of the
eye as the sampling time is varied.
iii) The height of the eye opening , at a specified sampling time, defines the margin over
noise.
iv) Completely closed eye shows that the effect of ISI is severe.
11. State Nyquist criterion for zero ISI.
P( f n Rb ) = Tb
Tb = bit duration in seconds ; Rb = bit rate = 1/Tb
p(t ) = received pulse
P(f) = Fourier Transform of p(t)
12. What is adaptive equalization?
Equalization is the process of correcting channel induced distortion. This process
is said to be adaptive when it adjusts itself continuously during data transmission by
operating on the input signal.
In adaptive equalization, filters adapt themselves to the channel characteristics.
That is , the filter coefficients are adjusted in such a way that the distortion in the data is
reduced.
13. What is PAM?
It is a process in which amplitudes of regularly spaced pulses are varied in proportion
to the corresponding sample values of continuous message signal.
14. What do you mean by Aperture Effect?
It is nothing but amplitude distortion occurring at PAM due to the sinc function. It is
overcome by using a Equalizer whose transfer function is
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| H(f)| = T 1 sinc(fT)
15. What is PPM?
It is the process in which the position of a pulse relative to its unmodulated time of
occurrence is varied in accordance with message signals.
16. What is PWM?
It is the process in which the samples of message signal are used to vary the duration
of individual pulses in the carrier.
17. What is Quantization and sampling?
Quantization: It is the process in which the analog sample of the original signal is
converted into a digital form.
Sampling: It is the process in which the original analog signal is converted into a
discrete time and continuous amplitude signal
18. Classify Quantizers.
Uniform Quanatizer Representation levels are uniformly spaced
Non-Uniform Quanatizer Representation levels are non-uniformly spaced
19. What is Quantization Noise?
The difference between the output analog sample and the discrete output quantized
signal gives raise to an error called Quantization Noise.
20. What are the limitations of delta modulation?
The major limitations of delta modulation are:
a) Slope overload error
b) Granular noise
21. What is Slope- Overload Distortion?
When the message signal varies steeply, the quantized approximation cannot follow
the message signal and this results in slope-overload distortion. It can be reduced by
increasing the step-size. To minimize the distortion, we need to have
/ Ts max | d(x(t)) / dt |
22. What is Granular Noise?
When the input waveform has a flat segment, then the step-size is larger
when compared to the input. Therefore the approximation hunts around the segment
resulting in Granular noise. This noise can be reduced by decreasing the step-size.
23. What is meant by hunting with reference to delta modulation?
The up & down movement of the approximated signal m (t) above & below the input
signal m (t) because of the step-wise approach during delta modulation is called hunting.
24. What is the principle of adaptive delta modulation?
The principle underlying ADM:
If successive errors are of opposite polarity, then the delta modulator is operating
in its granular mode. In this case the step size is reduced.
If successive errors are of same polarity, then the delta modulator is operating in
its slope overload mode. In this case the step size is increased.
25.State the difference between DPCM and DM.
Parameter

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Number of Bits

Bits can be more than one It can use 4, 8 or 16 bits per

but less than PCM.
sample.

Levels, Step size.

Fixed numbers of levels are Step size is fixed and cannot

used.
be varied.

Bandwidth
of Bandwidth
required
transmission channel.
lesser than PCM.

is Lower
required.

bandwidth

is

26. What are the advantages of digital transmission?

The advantage of digital transmission over analog transmission is noise
immunity. Digital pulses are less susceptible than analog signals to variations caused
by noise.
Digital signals are better suited to processing and multiplexing than analog
signals.
Digital transmission systems are more noise resistant than the analog transmission
systems.
Digital systems are better suited to evaluate error performance.
27. What are the disadvantages of digital transmission?
* The transmission of digitally encoded analog signals requires significantly more
bandwidth than simply transmitting the original analog signal.
* Analog signal must be converted to digital codes prior to transmission and
converted back to analog form at the receiver, thus necessitating additional encoding and
decoding circuitry.
28. Define pulse code modulation.
In pulse code modulation, analog signal is sampled and converted to fixed length,
serial binary number for transmission. The binary number varies according to the
amplitude of the analog signal.
29. What is the purpose of the sample and hold circuit?
The sample and hold circuit periodically samples the analog input signal and
converts those samples to a multilevel PAM signal.
30. Define companding Nyquist sampling rate
Companding is the process of compressing, then expanding. With companded
systems, the higher amplitude analog signals are compressed prior to transmission,
and then expanded at the receiver.
Nyquist sampling rate: Nyquist sampling rate states that, the minimum sampling
rate is equal to twice the highest audio input frequency

PART B
1) State and prove Nyquisit theorem of sampling for low pass signals.

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Ans. (a) Sampling theorem (Nyquist Theorem) : If more no. of samples are taken, the
information can be reproduced correctly. The other side is also correct, i.e., if fewer
samples of one information are taken in between other information can also be sent. This
is similar as in our example in which one person is reading temperature of several
thermometers lesser the time the person spends reading one thermometer, more time he
has left to read other thermometers, or we can say to get other information.
Here, Nyquist theorem is to help, it says : In order to convey an information completely.
The minimum sampling frequency of a pulse modulated system, should be equal to (or
more than twice) the highest signal frequency. Mathematically.

Where f = Minimum sampling frequency to convey complete information.

fm = Maximum frequency component present in the information signal.
e.g. The minimum sampling frequency to transmit a pure sine wave of 2 kHz

(b) Proof: The Fig. 6 (a) shows continuous time signal x(t). The fig (b) shows its
frequency spectrum. The fig (c) shows impulse train. The sampling of x(t) at the rate of
f5 Hz (samples) per second can be done by multiplying it with the impulse train. The
impulse train consists of impulses repeating periodically every T (sampling time)
seconds, where T5 = 1/f5. The Fig. (d) shows the resulted sampled signal. The fig. (e)
shows spectrum of the sampled signal.

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It shows that a signal where spectrum is band limited to fm Hz (i.e. the signal has
no frequency components beyond fm or it has maximum frequency fm) can be
reconstructed from its samples taken at the rate off5> 2fm Hz.
This proves the sampling theorem.
2) What are various pulse modulation techniques. Compare them.
Ans. There are 3 pulse modulation techniques:
(1) Pulse amplitude modulation (PAM)
(2) Pulse width modulation (PWM)
(3) Pulse modulation

3. Explain the principle of delta modulation.

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As LOGIC 1s and 0s are received, the up-down counter is incremented or

decremented accordingly .consequently the output of the DAC in the decoder is
identical to the output of the DAC is transmitter.
With delta modulation, each sample requires the transmission of only one bit;
therefore, the bit rates associated with delta modulation are lower than
conventional PCM systems.

.
4. Explain various methods of analog and digital companding techniques.
(a) Analog Companding.
(b) Digital Companding
(i) It is the combined process of compressing and Expanding used for
improving the dynamic range of signal and also to increase the SNR of low level
signals.
(ii) Analog companding and description
A law and law companding
(iii) Digital companding with one example
. 5.Explain the procedure of PCM generation and detection with its block diagram.

Pulse code Modulation PCM modifies the pulses created by PAM to

create a completely digital signal. ICM first quantizes the PAM pulses.
Quantization is a method of assigning integral values in a specific range to
sampled instances.

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Fig. shows a simple method of assigning sign and magnitude to quantized

samples.
Each value is translated into its 7-bit binary equivalent. The eighth bit
indicates the sign.

The binary digits are then transformed to a digital signal by using one of the
line coding techniques. Fig shows the result of the pulse code modulation of
the original signal encoded finally into a unipolar signal. Only the first three
sampled values are shown.
PCM is made up of four separate processes
- PAM
- Quantization
- Binary encoding
- Line coding.

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UNIT IV : DATA COMMUNICATIONS

PART A
1. What is ASCII code?
ASCII has been recommended by United States as ansi AND BY THE
International Standards Organization as ISO-14962.It is a 7 bit fixed length character
set with 27 .
With ASCII code, the LSB is designated b0 and MSB as b7.
2. Define error detection.
It is defined as the process of monitoring the transmission of data and find when a
error has occurred.
3. What are the error detection techniques?
Redundancy checking ----- VRC, LRC, CRC, Checksum
Parity coding
Exact count encoding and Echoplex.
4. What is redundancy checking?
Redundancy checking is defined as the process of adding extra bits for detecting
errors at the destination.
5. What is meant by CRC?
CRC is a systematic code, where instead of adding bits together to achieve a
desired parity, a sequence of redundant bits called CRC or CRC remainder is added to
the end of the data to be transmitted. It can be written as (n,k) cyclic codes.
6. Define FCS.
The group of characters forming a message is called as block or frame of
data.Hence,the bit sequence for the LRC is called as block check sequence or frame
check sequence(FCS).
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7. Name the two error correction method.

Retransmission and forward error correction.
8. Explain briefly about retransmission.
Retransmission is the process of retransmitting the entire message to the receiver,
whenever the receiver founds that the received message is in error.
If the receiver automatically calls for a retransmission of the entire message ,this
process of retransmission is called as automatic repeat request(ARQ)
9. What is Forward Error Correction?
It is the type of error correction scheme ,where the errors are detected and
corrected without retransmission but by adding the redundant bit to the message before
transmission.
10. Define DTE and DCE.
DTE is a binary digital device where the information originates or terminates.
It contains the hardware and software necessary to establish and control
communications between end points within a data communication system.
DCE is used to interfaces data terminal equipment to a transmission channel.
11. Define parallel interface.
Parallel interface allows the user to transfer data between two devices with eight
or more bits at a same time or simultaneously. It is also called as serial by word
transmission.
12. Give the characteristics of IEEE 488 bus.
Logic level-TTL,number of data lines-8 and bidirectional, maximum number of
devices -15,connector-ribon.
13. Define modem.
It is defined as a modulator and demodulator, which is used for connecting a
computer to a telephone line.
14. Name the types of modem.
Generally it is classified as synchronous and asynchronous modem.
It can also be classified based on its features as low speed, high speed, medium
speed, wide band, internal or external, personal or commercial.
15. Define adaptive equalizer.
It is located at the receiver section of a modem, where they provide post
equalization to the received signals. This type of equalizers automatically adjusts their
gain and delay characteristics to compensate for phase and amplitude impairments
encountered on the communications channel.
16. Define UART and USRT.
UART:Universal asynchronous receiver/transmitter is used for asynchronous
transmission of data between DTE and DCE.Asynchronous transmission means that an
asynchronous data format is used and there is no clocking information transferred
between DTE and DCE.
USRT:Universal synchronous receiver/transmitter is used for synchronous data
transmission between DTE and DCE .It means that there is clocking information
transferred between USRT and modem .
17. Give the primary functions of USRT.
*To perform serial to parallel and parallel to serial conversion of data.
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* To perform error detection by inserting and checking parity bits.

*To insert and detect SYN characters.
18. What is meant by RS 232 interface?
RS 232 interface specifies a 25 wire cable with DB25P compatible connector .It
is simply a cable and two connectors, the standard also specifies limitations on the
voltage levels that the DTE and DCE can output onto or receive from the cable.0
19. What is meant by centronics parallel interface?
Centronics parallel interface was originally designed to be used for transferring
data between a microcomputer and a printer.Centronics was one of the original
companies to design printers especially for desktop computers.
20. Give the primary functions of UART.
*To perform serial to parallel and parallel to serial conversion of data.
* To perform error detection by inserting and checking parity bits.
*To insert and detect start and stop bits.

PART B
1) What is X.21? What are the various signals provided by it?
Ans. X.21 OverviewX.21 is a state-driven protocol running full duplex at
9600 bps to 64 Kbps with subscriber networks. It is a circuit-switching
Protocol using Synchronous ASCII with odd parity to connect and
disconnect a subscriber to the public- switching network.
The data-transfer phase is transparent to the network. Any data can be
transferred through the network after Call Establishment is made
successfully via the X. 21 protocol. The call control phases which are used
were defined in the CCITT (now ITU) 1988 Blue Book Recommendations
X.l-X.32.
Signals provided: The signals of the X.21 interface are presented on a
15pin connector defined by ISO Document 4903. The electrical
characteristics are defined in CCITT Recommendations X.26 and X.27,
which refer to CCITT Recommendations V.10 and V.11.
X.21 provides eight signals
Signal Ground (G): This provides reference for the logic states against the
other circuits. This signal may be connected to the protective ground (earth)
DTE Common Return (Ga): Used only in unbalanced-type configurations
(X.26), this signal provides reference ground for receiver in the DCL
interface.
Transmit (T): This carries the binary signals which carry data from the DTE
to the I)CL. this circuit can be used in data-transfer phases or in call-control
phases from the DIE to DCL (during Call Connect or Call Disconnect).
Receive (R): This carries the binary signals from DCE to DTE. It is used
during the data transfer or Call Connect/Call Disconnect phases.
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Control (C): Controlled by the DTE to indicate to the DCE the meaning of
the data sent on the transmit circuit. This circuit must be ON during datatransfer phase and can be ON or OFF during call-control phases, as defined
by the protocol.
Indication (I): The DCE controls this circuit to indicate to the DIE the type
of data sent on the Receive line. During data phase, this circuit must be ON
and it can be ON or OFF during call control, as defined by the protocol.
Signal Element Timing (S): This provides the DTE or DCE with timing
information for sampling the Receive line or Transmit line. The DTE
samples at the correct instant to determine if a binary I or 0 is being sent by
the DCL. The DCE samples to accurately recover signals at the correct
instant. This signal is always ON.
Byte Timing (B): This circuit is normally ON and provides the DTE with 8
bit byte element timing. The circuit transitions to OFF when the Signals
Element Timing circuit samples the last bit of an 8-bit byte. Call-control
characters must align with the 13 lead during call-control phases. During
data-transfer phase, the communicating devices bilaterally agree to use the B
lead to define the end of each transmitted or received byte. The C and I leads
then only monitor and record changes in this condition when the 13 lead
changes from OFF to ON, although the C and I leads may be altered by the
transitions on the S lead. This lead is frequently not used.
2) What is a modem? Explain the types.
Ans. A modem provides the communication interface. It transports device
protocols transparently over the network through a serial interface. A modem
adapts the machine to communicate over various networks in order to gain
access to the machine including
1. PSTN-a wire line dial-up network
2. GSM-.a wireless dial-up network
3. GPRS a wireless always on network.
Moderns are traditionally associated with PCs in the form of box/PC
modems,
However this technology is not suited to non-PC equipment or machines,
which have specialized needs
1. Sizethere is little space within many embedded devices for a modem
box
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2. Power consumption- some devices are battery powered and need low
power modems
3. Environmentmachines can be deployed and need wide temperature
range.
4. Integration modems need to be integrated within the machine and not
external Modems provide remote access to machines in the field to eliminate
unnecessary site visits and provide fast access to information in the machine.
However, integrating moderns.
1. Moderns are black art products, surrounded by complex compliance and
regulatory issues. Designing your own modem solutions requires specialized
skills. Using off-the-shelf modems enables designers to focus on their core
application strengths and not be drawn into solving issues that are not related
to core competency yielding potentially unreliable products.
M2M moderns are embedded with the machine and transport higher level
protocols between the machine and central location via the network.
2. Point to Point polled networks where proprietary or industry specific
protocols to communicate with central servers.
3. TCP/IP enabled remote machines such as Embedded PCs, where modems
provide the physical network connectivity.
M2M Modems are Al-command compatible at the serial
interface to the machine and common between many modems.
4. Driven and controlled by the remote machine processor using industry
standard Commands.
5. Fast time to market.

3) What are various transmission modes? What are

the advantages and disadvantage of parallel and
serial transmission?

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Ans.

Data transmission means movement of data which is in the form of

bits between two or more digital devices. The data transmission takes place
over some physical medium from one computer to the other.
There are two ways of transmitting the bits. They are
Parallel transmission
Serial transmission
Parallel Transmission In parallel transmission of data, all the bits of a byte
are transmitted simultaneously on separate wires as shown.

This type of transmission requires multiple circuits for interconnecting the

two devices. Parallel transmission is possible practically if the two devices
are close to each other.
For example parallel transmission takes place between a computer and its
printer.
Advantages :
1. The advantage of parallel transmission is that all the data bits will be
transmitted simultaneously. Therefore the time required for the transmission
of an N bit words are only one clock cycle.
2. The serial transmission will require N number of clock cycles for the
transmission of
same word. Due to this the clock frequency can be kept low without
affecting the
speed of operation. For serial transmission the clock frequency cannot be
low.
Disadvantages :
To transmit an N-bit word, we need N number of wires. With
increase in the number of users, these wires will be too many to handle. The
serial
transmission used only one wire, for connecting the transmitter to the
receiver. Hence,
practically the serial transmission is always preferred.
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Serial Transmission: In serial transmission, the bits of a byte are serially

transmitted one after the other as shown in fig.

The byte to be transmitted is first stored in a shift register. Then these bits
are shifted from MSB to LSB bit by bit in synchronization with the clock.
Bits are shifted right by one position per clock cycle. The bit which falls out
of the shift register is transmitted. Hence LSB is transmitted first. For serial
transmission only one wire is needed between the transmitter and the
receiver. Hence serial transmission is preferred for long distance data
communication. In serial transmission only one bit is transmitted per clock
cycle. It requires a time corresponding to 8 clock cycles to transmit one byte.
The time can be reduced by increasing the clock frequency. It is used for
computer to computer communication specially long distance.
Advantages of serial transmissionCommunication.
I. Only one wire is required
2. Reduction in cost due to less number of conductors.
Disadvantages
1. The speed of data transfer is low.
2. To increase the speed of data transfer, it is necessary to increase the clock
frequency
4) Explain with pin diagrams of EIA-449 interface.
Ans. EIA 449 interface Standard in case of EIA 232 the data rate is
restricted to 20kbps and the cable length to 50 feet. The EIA and ITU-T have
introduced the EIA 449 standard for users who require speed and distance.
Mechanical Specifications
1. The mechanical specification of EIA-449 defines a combination of two
connectors, one with 37 pins and with 9 pins, for a combined 46 pins as
shown in Fig.

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2. The functions related to secondary channel have been removed from the
DB 37 connector since they are very rarely used.
3. The EIA separates the secondary functions out and has puts them in the
second 9 pin connector (DB-9). In this way a second channel is available to
systems that need it.
4. To maintain compatibility with EIA-232, EIA-449 defines two categories
of pins to be used in exchanging data, control and timing information.
5. The category I pins are those pins whose functions are compatible with
those of
EIA-232. Category II pins are those that have no equivalent in
EIA-232 as shown in Table 1.

Table 2 gives the pin functions of the DB-9 connector.

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5. State and explain the various Error Detection methods

Error detection uses the concept of redundancy, which means adding extra
bits for detecting errors at the destination there ate three types of redundancy
checks are common in data communication:
(a) Parity check
(h) Cyclic Redundancy check (CRC)
(c) Checksum.
Parity check
In this technique, a redundant bit called a parity bit, is added to every data
unit so that the total number of Is in the unit becomes even (or odd).
Suppose we want to transmit 1100001. Adding the number of 1s gives us 3,
an odd number. Before transmitting, we pass the data unit through a parity
generator. The parity generator counts the 1s and appends the parity bit to
the end (al in this case)
In case of redundancy check method we have to append the data unit with
some extra bits. These extra bits are called parity.
This parity or parity hit can be even or odd.
in case of even parity we have to make number of 1s even, including the
parity hit e.g. 1110001 is the data unit where the no. of ls is already even
then we will insert 0 at the next to data unit it, 1110001. In case of odd
parity we have to make no. of ls odd, including the parity bit. e.g. 1111000
is the data unit, where the no. of 1s is even then we will insert I at the next
to data unit i.e. 11110001.
It is a most common and powerful technique for the detection of errors. In
this technique extra bits are added. But instead of repeating the entire data
stream, a shorter group of bits may be appended to the end of each unit. The
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technique is called redundancy because the extra bits are redundant to the
information. They are discarded as soon as the accuracy of transmission has
been determined.
The following fig. shows the process of using redundant bits to check the
accuracy of data unit.

Once the data stream has been generated. It passes through a device that
analyzes it and adds on appropriately coded redundancy check. The receiver
puts the entire stream through a checking function. If the received bit stream
passes the checking criteria, the data portion of the data unit is accepted and
redundant bits are discarded.
The most common and least expensive mechanism for error detection is the
parity check.
Parity checking can be simple or two-dimensional.

Simple Parity Check

In this technique, a redundant bit, called a parity bit, is added to every data
unit so that the total number of Is in the unit (including the parity bit)
becomes even (or odd). Suppose we want to transmit the binary data unit
1100001
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Transmission Mode
Adding the no. of is giving us 3 an odd number. Before transmitting we pass
the data unit through a parity generator. The parity generator counts the is
and appends the parity bit to the end. The total no. of is now 4, an even
number. The system now transmits the entire expanded unit across the
network link. When it reaches its destination, the receiver puts all 8 bits
through an even parity checking function. If the receiver sees 11000011, it
counts four is, an even number and the data unit passes. But, if instead of
11000011, the receiver sees 11001011 then when the parity checker counts
the Is it gets 5 an odd number. The receiver knows that an error has been
introduced into the data somewhere and therefore rejects the whole unit.
Two Dimensional Parity Check
A better approach is the two dimensional parity check in this method, a
block of bits is organised in a table (rows and columns). First we calculate
the parity bit for each data unit. Then we organise them into table. Shows in
fig we have four data units shown in four rows and eight columns. We then
calculate the parity hit for each column and create a new row of 8 bits. They
are the parity bits for the whole block. The first parity bit in the fifth row is
calculated based on all first bits, the second parity bit is calculated based on
all second bits, and so on. We then attach the 8 parity bits to the original data
and sent them to the receiver.

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Cyclic Redundancy Check (CRC):

Cyclic Redundancy
check method is most powerful mechanism of error detecting. Unlike the
parity check which is based on addition, CRC is based on binary division.
In CRC, instead of adding bits to achieve a desired parity, a sequence of
redundant bits, called the CRC or the CRC remainder, is appended to the end
of a data unit so that the resulting data unit becomes exactly divisible by a
second predetermined binary number. At its destination the incoming data
unit is divided by the same number. If at this step there is no remainder, the
data unit is assumed to be intact and is therefore accepted. A remainder
indicates that the data unit has been damaged in transit and therefore must be
rejected.
The redundancy bits used by CRC are derived by dividing the data unit by a
predetermined divisor, the remainder is the CRC. A CRC must have two
qualities. It must have exactly one less bit than the divisor, and appending it
to the end of the data string must make the resulting bit sequence exactly
divisible by the divisor.
CRC generator and checker
First, a string of n 0s is appended to the data unit. The number n is less than
the number of bits in the predetermined divisor, which are n + 1 bits.
Second, the newly formed data unit is divided by the divisor, using a process
called binary division the remainder resulting from this division is the CRC.
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Third, the CRC of n bits derived in step 2 replaces the appended Os at the
end of the data unit. The data unit arrives at the receiver data first followed
by the CRC. The receiver treats the whole string as a unit and divides it by
the same divisor that was used to find the CRC remainder.

If the string arrives without error, the CRC checker yields a remainder of
zero and the data unit passes. If the string has been changed in transit the
division yields a non zero remainder and the data unit does not pass.
CHECK SUM:
Checksum is the third mechanism for error detection which is also based on
the concept of redundancy.
Check sum Generator
In the sender, the check sum generator subdivides the data unit into equal
segments of n bits. These segments are added using ones complement
arithmetic in such a way

that the total is also n bits long. That total is then complemented and
appended to the and o the original data unit as redundancy bits called the
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check sum field. The extended data unit is transmitted across the network.
So if the some of data segment is T, the checksum will be T.
Check sum Checker
The receiver subdivides the data unit as above and adds all segments and
complements the result. If the extended data unit is intact, the total value
found by adding the data segments and the check sum field should be zero If
the result is not zero, the packet contains an error and the receiver rejects it.

6.State and explain the various Error Correction methods.

A mechanism that can handle correction of an error heading of error
correction code categories under the
There are two methods for error correction.
(1) Error correction by retransmission.
(2) Forward error correction.
Error Correction by Retransmission
In error correction by retransmission, when an error is discovered, the
receiver can have the sender retransmit the entire data unit.
Forward Error Correction
In forward error correction (FEC), a receiver can use an error-correcting
mode, which automatically corrects certain errors. In theory it is possible to
correct any error automatically. Error correcting codes however are more
sophisticated than error detection codes and require more redundancy bits.
e.g. To correct a single bit error in an ASCII character, the error correction
code must determine which of the 7 bits has changed In this case we have to
distinguish between eight different states no error, error in position 2, and so
on, up to the error in position 7. To do so requires enough bits to show all
eight states.
At first glance, it seems that or 3-bit redundancy code should be adequate
because 3 bits can show eight different states (000 to 111) and can therefore
indicate the locations of eight different possibilities. To calculate the no. of
redundancy bits. We should consider
Where m is the no. of bits to be transfer r stands for the no. of redundancy.
By this manner.

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There is the practical solution for this method that is Hamming Code.
The hamming code can be applied to data units of any length and uses the
relationship between data and redundancy bits.
Suppose there are 7 bits ASCII codes which requires 4 redundancy bits that
can be added to the end of the data unit or interspersed with the original data
bits. These units are position in 1, 2, 4, arid 8 (the position is in an 11 bit
sequence that are powers of 2). We prefer these bits are r1, r2, r4 and r8.

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UNIT V SPREAD SPECTRUM AND MULTIPLE ACCESS TECHNIQUES

PART-A
1. What is meant by spread spectrum?
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
2. What are the applications of spread spectrum?
It is used in military communications systems.
It allows the transmitter to transmit a message to a receiver without the message
being detected by a receiver for which it is not intended.
It decreases the transmitted power spectral density so that it lies well below the
thermal noise level of any unfriendly receiver.
It turns out not to be possible to conceal the transmission.
3. What is the primary advantage of spread spectrum communication?
The primary advantage of spread spectrum communication system is its ability to
reject interference whether it is the unintentional interference of another user
simultaneously attempting to transmit through the channel or the intentional interference
of a hostile transmitter attempting to jam the transmission.
4. Define Pseudo Noise sequence.
A Pseudo Noise (PN) sequence is coded sequence of 1s and 0s with
certain autocorrelation properties.
It is periodic a sequence of 1s and 0s repeats itself exactly with a known
period
PN sequence is a noise like high frequency signal. The sequence is not
truly random but it is generated by a well defined logic.
This is used in spread spectrum communication.
5. What are the properties of maximum length PN sequence?
Balance property: In each period of a maximum length sequence, the
number of 1s is always one more than the number of 0s.
Run property: Among the runs of 1s and 0s in each period of a
maximum length sequence, one half the runs are of length one, one
fourth are of length two, one eighth are of length three and so on.. as
long as these fractions represent meaningful number of runs
Correlation property: The auto correlation function of a maximum length
sequence is periodic and binary valued.
6. Define processing gain. What is the processing gain for DSSS?
It is defined as the ratio of spreaded signal bandwidth to unspreaded signal
bandwidth.
7. Define jamming margin.
It is the ratio of average interference power J and the signal power Ps.It is expressed in
dB as (jamming margin) dB=(processing gain)dB-10 log[Eb/No]min
where [Eb/No]min is the minimum bit energy to noise density ratio.
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8. Define chip rate and chip duration.

Chip rate is defined as the number of chips per second.
The duration of every bit in Pseudo Noise(PN) sequence is known as chip duration.
9. What is the principle of frequency hopping spread spectrum?
It is the process of randomly hopping the modulated data carrier from one frequency
to other .The data is used to modulate a carrier, due to this the spectrum of transmitted
signal sequentially spreaded rather than instantaneously.
10. Define fast frequency hopping and slow frequency hopping.
Slow frequency hopping: Symbol rate RS of the MFSK signal is an integer
multiple of the hop rate R h. i.e., several symbols are transmitted in each hop.
Fast frequency hopping: The hop rate R h is an integer multiple of the
MFSK symbol rate RS. i.e., carrier frequency will change or hop several times
during the transmission of one symbol.
11. What are the different multiple access techniques ( Fixed assignment multiple
access techniques)used in wire less communication?
TDMA Time Division Multiple Access
FDMA- Frequency Division Multiple Access
CDMA Code Division Multiple Access.
12. Compare FDMA and TDMA.
FDMA - Frequency Division Multiple TDMA - Time Division Multiple Access
Access
All users access the channel by All users occupy the same RF band width
transmitting simultaneously but using of the channel, but they transmit
disjoint frequency bands
sequentially in time
Fixed assignment multiple access Fixed assignment multiple access
technique
technique
Well suited for analog communication
Well suited for digital communication
13. List the merits of CDMA over TDMA.
CDMA does not require an external synchronization network which is an
essential feature of TDMA.
CDMA offers a gradual degradation in performance as the number of user is
increased. It is there fore relatively easy to add new users to the system.
CDMA offers an external interference rejection capability.
14. What is CDMA?
CDMA Code Division Multiple Access. In CDMA, all users transmit
simultaneously and occupy the same RF bandwidth. Each user is assigned a code
which perform the DSSS or FHSS modulation
15. List out any four features of TDMA.

TDMA shares a single carrier frequency with several users where each user makes
use of non overlapping time slots.

Data transmission for users is not continuous, but occurs in burst.

Because of discontinuous transmission, handoff process is much simpler for a

subscriber unit .
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TDMA uses different time slots for transmission and reception, thus duplexers are
not required.
16. What is DS spread spectrum?
DS spread spectrum is one in which the amplitude of an already modulated signal is AM
modulated by a very high rate NRZ binary stream of digits.
17. Define FH spread spectrum.
FH spread spectrum is an FM or FSK technique while DS is a AM or PSK technique. The
signal to be FH is usually a BFSK signal although M-ary FSK, MSK or TFM can be
employed.
18. What is meant by pseudo random noise?
The pseudo random noise is the nose which presents in the DS spread spectrum, if the DS
spread spectrum signal is V(t)=g(t)S(t)=2Psd(t)cosot. Here g(t) is a pseudo random
noise binary sequence having the values 1.
19. Write the advantages of FH over direct sequence spread spectrum.
Processing gain is more
Spread of the transmitted signal is spread sequentially rather than instantaneously
Greater transmission bandwidth and more chips per bit.
20.Compare SFH and FFH.
Slow frequency hopping -SFH
Fast frequency hopping- FFH
More than
transmitted

one

symbols

Chip rate is equal to symbol rate

are More than one hops are required to

transmit one symbol
Chip rate is equal to hop rate

Symbol rate is higher than the hop Hop rate is higher than the symbol
rate
rate
Same carrier frequency is used to One symbol is transmitted over
transmit one or more symbols
multiple carriers in different hops

PART B

1. Explain the Direct sequence spread spectrum technique with

neat diagram
In the DS SS, each bit in the original signal is represented by multiple bits in
the transmitted signal using a spreading code. The spreading code spreads
the signal across a wider frequency band in direct proportion to the number
of bits used. Therefore, a 10-bit spreading code spreads the signal across a
frequency band that is 10 times grater than a 1hit spreading code.
One technique with direct sequence spread spectrum is to combine the
digital information stream with the spreading code bit stream using an
exclusive. OR (XOR).
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The XOR obeys the following rules

The following fig. shows an example.

Note that an information bit of one inverts the spreading code bits in the
combination, while information bit of zero causes the spreading code bits to
be transmitted without inversion. The combination bit stream has the death
rate of the original spreading code sequence, so it has a wider bandwidth
than the information stream. In this example, the spreading code bit stream
is clocked t four times the information rate.
DSSS using BPSK:
Rather than represent binary data with 1 and 0, it is more convenient for our
purpose to use + I and - 1 to represent the two binary digits. In that case, a
BPSK signal can be represented as:

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2.Explain the Frequency hopping spread spectrum in detail.

FHSS is a form of spread spectrum in which the signal is broadcast
over a seemingly random series of radio frequencies, hopping from
frequency to frequency at fixed intervals.
Types of frequency hopping are
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(I) Slow frequency hopping

(2) Fast frequency hopping.
Slow-frequency hopping
In slow frequency hoping the symbol rate R5 of the MFSK signal is an
integer multiple of the hop rate R. That means several symbols are
transmitted corresponding to each frequency hop
Therefore each frequency hop several symbols
i.e. frequency hopping takes place slowly.
Fast frequency hopping
In the fast frequency hopping the hop rate R is an integer multiple of the
MFSK symbol rate R. That means during the transmission of one symbol,
the carrier frequency will hop several times.
Therefore each symbol transmission several frequency hops. Thus- the
frequency hopping takes place at a fast rate.
A common modulation tech. used is the M-ary frequency shift Keying
(MFSK). The combination of frequency hopping (FH) and MFSK is known
as FH/ MFSK.
Figure shows the Block diagram of frequency hopping system. For
transmission, binary data are fed into a modulator using some digital - to
analog encoding scheme, such as frequency shift keying (FSK) or binary
phase shift keying (BPSK). A PN source serves as an index into a table of
frequencies each K bit on PN source specifies one of the 2k carrier
frequencies.
At each successive interval a new carrier frequency is selected. This
frequency is then modulated by the signal produced from the initial
modulator to produce a new signal with the same shape.
On reception, the spread spectrum signal is demodulated using the same
sequence of PN-derived frequencies and then demodulated to produce the
output data.

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3)Explain difference between slow frequency hopping and Fast

frequency hoping?

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4) Explain CDMA system with its features and list out various problems
in CDMA systems.
CDMA refers to the Code Division Multiple Access. It is an alternative to
the FDMA and TDMA. Some times it is, also known as SSMA i.e. spread
spectrum multiple Access.
In CDMA more than one user is allowed to share a channel or sub channel
with the help of direct-sequence spread spectrum signals.
In CDMA, each user is given a unique code sequence or signature sequence.
This sequence allows the user to spread the information signal across the
assigned frequency band.
2. In CDMA, the users access the channel in random manner.
3 CDMA signals are spread in frequency.
CDMA works differently in the l)SSS environment. Following fig. shows
the working of CDBA in DSSS environment.

Above fig. shows a configuration in which there are n users, each user, the
data stream to be transmitted, d (t), is BPSK modulated to produce a signal
with a bandwidth of w, and then multiplied by the spreading code for that
user, c. (t). All of the signals, plus noise, are received at the receivers
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antenna. Suppose that the receiver is attempting to recover the data of user 1.
The incoming signal is multiplied by the spreading code of user 1 and then
demodulated. The effect of this is to narrow the bandwidth of that portion of
the incoming signal corresponding to user 1 to the original bandwidth of the
unspread signal, which is proportional to the data rate. Because the
remainder of the incoming signal is orthogonal to the spreading code of user
1, that remainder still has the bandwidth w. Thus the unwanted signal energy
remains spread over a large bandwidth and the wanted signal is concentrated
in a narrow bandwidth. The band pass filter at the demodulator can therefore
recover the desired signal.
5.How pseudo noise sequence is generated? Explain the properties.

Balance property: In each period of a maximum length sequence, the

number of 1s is always one more than the number of 0s.
Run property: Among the runs of 1s and 0s in each period of a
maximum length sequence, one half the runs are of length one, one
fourth are of length two, one eighth are of length three and so on.. as
long as these fractions represent meaningful number of runs
Correlation property: The auto correlation function of a maximum length
sequence is periodic and binary valued.

6.Explain in detail about Multipulse excited LPC and CELP

Multipulse excited LPC(4 marks):

CELP(4 marks):

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7.Write a note on TDMA (Time Division Multiple Access) .

A packet radio network is one in which communication between nodes or
terminals is accomplishe&using radio frequency. A high frequency (1-30
MHz) to the extra high frequency (3-10 GHz) region used by satellites, also
used for operation of a packet radio. The access and flow-control procedure
is one of the most interesting aspects of packet radio communication. Such
techniques are TDMA (Time Division Multiple Access), FDMA (Frequently
Division Multiple Access) and CDMA (carrier sense multiple access).
In TDMA, a radio system where each of the K user wishes to transmit over a
channel is considered fb bits/sec are generated by each user. We allow each
user, in sequence, to transmit, a single packet of b bits (b 1000 bits) using
packet switching TDMA. As shown in figure 4 and 5 each packet of bits is
stored initially in a buffer at a rate fb In order to insure that the message does
not back up in the buffer we require that when the commutator selects a
particular buffer, the bits are read out of the buffer and on to the common
channel at the bit rate kfb. Thus on the channel, bit duration is l/Kfb. The
dwell time of the commutator on each buffer is b/Kf. Since there are b bits in

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a packet and is equal to T where T is the duration of a packet.

Hence the commutator returns to each individual buffer at times separated

by the

The channel bandwidth required is equal to the bit rate on the channel, if
QPSK is employed and which is equal to B = Kfb Hz.
Fig. 4 (a) TDMA system (b) Timing patterns. The higher level of each
Waveform marks the dwell time of the commutator on each switch
position.
If the K user has a continous need to transmit, TDMA systems at that stage
are very efficient. TDMA will become inefficient when the users need varies
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from time to time due to the type of communication employed. Since a user
pays for the times-lot even when he fails to transmit it, since the time-slot is
wasted, an additional user could have been admitted to the network during
this wasted time.

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