05 - Communication All Chapters PDF

Communication Systems
1. Amplitude Modulation
01.
Sol: Given
IC = 10A
IT = 12A
Where IC & IT are the antenna currents
before and after amplitude modulation
respectively .
2
Total power is given by PT = PC 1
2
2
PT = I T R
PC = IC2R
2
2
2
IT R = IC R 1
0 .6
0.12 V-1
5
Where ka is the amplitude sensitivity
Given AC cos C t
= 50cos 104 t
Am cos mt
= 5 cos 200 t
s(t) = 50[1+0.6 cos 200 t] cos 104t
Ka =
F.T of s(t)
S(f) = F.T [Accos ct (1+ cos mt)]
=
A
+ C ( C m )
4
Where is the modulation index

2
IT
1
2
IC
I 2
2
= 2 T 1
I C
12 2
= 2 1
10
12 2 10 2
=2
2
10
0.938
Percentage Modulation = 93.80%
2
02.
Sol: Given modulation index = 0.6
For an AM signal the standard equation is
s(t) = AC 1 cos m t Cos ct
Where
KaAm = = 0.6, Am = 5 (given)
ACE Engineering Academy
AC
c c
2
A C
( C m )
4
A C
( ( C m )
4
A C
( ( C m )
4
PT = PC(1+2/2)
2
A
50 2
PC = C
2
2
PT =
50 2
2
0 .6
1 2
2
PT = 1475W
PC 2 50 2 0.6 2
PSB =
2
2
2
Hyderabad | Bhopal | New Delhi | Bengaluru | Bhubaneswar | Vijayawada | Visakhapatnam | Tirupati | Pune |Chennai
: 212 :
Postal Coaching Solutions
PSB = 225W
(2) Given
225
PUSB = PLSB =
=112.5W
2
R = 3.64 M
04.
Sol:
2
2
PC (1 )
2
Given modulation index = 1
0 .6 2
=
= 0.1525
0 .6 2 2
% efficiency = 15.25%
03.
Sol: (1) Given equation is
15[1 + 0.4cos 210 3 t ][cos 210 6 t ] --- (1)
m 2 10 3 rad/s
c 2 10 6 rad/s
Where m & c are message & carrier
frequencies respectively 0.4 ; AC = 15 V
For a envelope detector
RC
RC = 364.67 sec
PSB
PT
PC
C = 100pF
1 2
m
PT = Pc + PSB
PT = PC(1+ 2/2)
For = 1
PT = 3/2 PC -------- (1)
Where PT is the total power transmitted and
PC is the carrier power
PSB = PC / 2
PUSB = PC / 4 ---------- (2)
From (1) & (2) we get
PUSB = PLSB =
2 / 3PT
4
RC =
PUSB = PT / 6
1 2
---- (2)
m
Substituting values in (2) we get
PUSB = PLSB =
1 0 .4 2

0.4 2 10 3
PT
6
Thus at = 1 the power in sidebands is 1/6th

of the total power transmitted.
364.67 sec
= 364.67 sec
: 213 :
05.
Sol: Modulation Index = 3
Side frequencies
fc fm = 4.914 MHz
fc + fm = 4.928 MHz
2fc = 9.842
fc = 4.921 MHz
fm = 7 10-3 MHz
fm = 7 kHz
Amplitude of each side band=
A c
75V (given)
2
A
75
C
2
3
AC = 50 V &
fC = 4.921 MHz
Comparing (1) & (3) we have

AC = 30
A C
6
2
12
=
= 0.4
30
c = 400
c m = 360
c + m = 440
c = 400 rad/s
2 m = 80
m = 40 rad/s
06:
Sol: Refer Communication Systems Theory
Booklet
07.
Sol:
Given signal is x(t) = 30 cos 2 200t

+ 6 cos 2(180t)
+ 6 cos 2(220t) ------ (1)
The standard equation of single tone AM
system is
x(t) = AC (cos Ct)
[1+ cos m t ] ------ (2)
Equation (2) can be modified as
A C
A C cos C t 2 cos (C m ) t
x(t) =
----(3)
A C cos( ) t
C
m
2
PSB
2
0 .4 2
= 2
=
PT
2 2 0 .4 2
7.407 %
08.
Sol: Refer IES - 15 Previous Conventional
paper II solution book, Page number:
312, Q: 04
09.
Sol: Time domain equation for single tone AM
signal can be written as
SAM(t) = AC[1+cos2fmt] cos2fct
A
= Accos2fct + c cos2(fc + fm)t
2
(Carrier)
(USB)
A c
cos2(fc fm)t
2
(LSB)
A2
Carrier power Pc c
2R
Sideband power
A c
A c2 2
2 2
PUSB =
R
8R
: 214 :
10.
Sol: Square law modulator is a Non linear
modulation of AM generation. A simple
Diode can be used as a non-linear modulator
by restricting its operation to non-linear
region of its characteristics. The undesired
frequency terms are filtered out by a band
pass filter.
A c
2 2
2 2 Ac
PLSB =
R
8R
PT PC PUSB PLSB
A C2 A C2 2 A C2 2
2R
8R
8R
A C2 A C2 2 A C2 A C2 2
;(R=1)
2R
4R
2
4
for

1 2
PT = PC
AM
signal
V2
BPF
V0
Accos2fct
(Carrier)
m(t) represents message signal

Ac cos2fct represents carrier signal
= carrier power + sideband
The total input to the non linear system
[m(t) + A c cos 2f c t ] is kept so small that
the system operates in a non linear region of
its V-I characteristics.
power
2
P
2
Efficiency() = SB
PT
2
PC 1
2
PC
2
2 2
If 80% modulation is used, then the ratio of

the total sideband power to the total power
in the modulating signal is
PSB
2
0 .8 2
0.2424
PT 2 2 2 0.82
V1
Non
Linear
System
PC 2
2
Total power
2
PT PC 1
2
Block diagram:
m(t)
normalized power)
2
C
Thus this method is very useful only for

small signal amplitude modulation. Input
voltage is given to the non linear system
v1 = m(t) + Accos2fct
The
non
linear
characteristics
is
mathematically by a power series.
Neglecting the terms higher than the second
order.
Then the output voltage is related as
V2 = aV1 + bV12
V2 = a[m(t) + Accos2fct]
+ b[m(t) + Accos2fct]2
: 215 :
For = 1 = 1/3
= am(t) + aAccos2fct + bm2(t)
% = 33.33%
+ bAc2cos22fct + 2bm(t)Accos2fct
The expansion of this equation provides the
presence of a number of frequency
components, such as fc, 2fc, fm , 2fm, (fc
fm), (fc 2fm) and so on. All those
components, except the desired ones i.e.
fc and fc fm are filtered by a band pass filter
tuned to fc.
Thus for single tone AM (with modulation

index = 1) the efficiency is 33.33%
12.
Sol: The amplitude of a carrier signal is varied in
accordance with the instantaneous values of
the modulating signal is called amplitude
modulation.
So aAccos2fct + 2bAcm(t)cos2fct
= Ac[a + m(t)2b]cos2fct
= aA c (1
Modulation index:
The ratio of change in amplitude of
modulating signal to the amplitude of carrier
wave is known as modulation index or
modulation factor or modulation coefficient
or depth of modulation or degree of
modulation.
2b
m( t )) cos 2f c t
a
The above equation represents an AM signal

Here
2b
K = amplitude sensitivity
a
If Am is greater than Ac then distortion is

introduced into the system.
The modulating signal voltage Am must be
less than carrier signal voltage Ac for proper
amplitude modulation.
It varies between 0 and 1.
AM total spectrum:
Band pass
filter
characterist
ic
-2fc -fc+fm-fc -fc-fm -2fm-fm 0
fm2fm fc-fm fc fc+fm 2fc
Let the carrier signal c(t) = Ac cos ct

Modulating signal m(t) = Am cos mt
Condition for square law modulation fc > 3fm
The ratio
11.
Sol:
Am
Ac
is defined as modulation
index (m)
From the derivation carried out in question
number 09 ,
2
PC
P
2
Efficiency() = SB
PT
2
PC 1
2
2
2 2
(i)
The equation for the amplitude in the AM

signal is
A A c A m cos m t
s( t ) A c (1
Am
cos m t ) cos c t
Ac
(ii) The instantaneous voltage of AM wave is

Vi = V cosct
Vi = Vc [1+mcosmt] cosct
: 216 :
Representation of AM
+
V
c
+V
Vc+Vm sin mt
Vm
Vmin
Vm
Vmax
(iii) Derivation of modulation index (m)

From the above figure
V Vmin
Vm = max
2
Vc = Vmax Vm
V Vmin
= Vmax max
V
(Vc+Vm sin mt)
Vc =
13.
Sol: The amplitude of a carrier signal is varied in
accordance with the instantaneous values of
the modulating signal is called amplitude
modulation.
(i)
+
Vm
Vm
Vmax
t
V
(Vc+Vm sin mt)
75
0.75
100
512.48 W
14.
Sol: Given:
Carrier current (IC) = 8A
Total AM transmitter current (IT) = 9A
Representation of AM
Vmin
Vmax Vmin
Vmax Vmin
Pt PC 1
2
0.752
4001
2
A
v( t ) A c (1 m cos m t ) cos c t
Ac
(ii) The instantaneous voltage of AM wave is
Vi = V cosct
Vi = Vc [1+mcosmt] cosct
+V
Vm Vmax Vmin /2
Vc Vmax Vmin /2
Modulation index = 75%
A A c A m cos m t
Vc+Vm sin mt
m=
(iv) Pc = 400 W
The equation for the amplitude in the AM

signal is
Vc
Vmax Vmin
2
IT = I C 1
2
2
9 8 1
2
2
2
1.2656 2 0.5184 0.72
2
When = 0.77
1
IT = 8
0.77
1
= 8 1 0.2964 =
IT = 9.108 A
: 217 :
15. Sol:
Given Carrier signal ec = Ec sin ct
Em
E
E
sin 2 m t m sin 3 m t m sin 4 m t
2
3
4
AM signal can be represented as e E C 1 K a e m sin c t
where Ac = EC(1 + Ka em) (Carrier Amplitude)
Amplitude can be written as
E K
E K
E K
A E C [1 E m K a sin m t m a sin 2 m t m a sin 3 m m a sin 4 m t
2
3
4
E C 1 sin m t sin 2 m t sin 3 m t sin 4 m t

2
3
4
where modulation index K a E m

Now e = Ac Sinct
e E C 1 sin m t sin 2 m t sin 3 m t sin 4 m t sin c t

2
3
4
E
E
E
EC.sinc t EC sinmt.sinc t C sin2mt sinc t C sin 3 m t. sin c t C sin 4 m t sin c t
2
3
4
E
e E C sin c t C cos m c t cos m c t
2
E C
cos2 m c t cos2 m c t E C cos3 m c t cos3 m c t
4
6
E C
cos4 m c t cos4 m c t
8
From the above expression, it is clear that for every modulating frequency component, the AM
wave contains two sideband frequencies in addition to the carrier i.e., the AM wave contains the
frequency components C, C + m, C m, C + 2m, C 2m, C + 3m, C 3m, C + 4m,
C 4m,
S(f)
Modulating signal e m E m sin m t
EC
Amplitude of each
frequency component
1
E C
2
1
E C
4
1
E C
6
1
E C
8
4fm +fc 3fm +fc 2fm +fc
fm +fc
fc
fc +fm
fc +2fm
fc +3fm
fc +4fm
Frequency spectrum
: 218 :
The value of composite modulation index is t 12 22 32 24

1 = , 2= /2, 3 = /3, 4 = /4
1 1 1

1.19
c 1
4 9 16
2 3 4
2
2. Sideband
Modulation Techniques
16.
Sol: Given:
Carrier power = 10kW
Carrier current IC = 10A
Antenna current IT =11A
Total current I T I C
2
1
2
2
2 11
11 10 1
1

2
2 10
Percentage modulation = 0.64
When the total current increased to 12A
2
It I C 1
12 10
12 22
2
2
0.642
1
2
cos c t cos(i t ) sin(i t ) sin c t

is of form cosA cosB sinA sinB
using the relation
cos (A+B) = cos(A) cos(B) sin(A) sin(B)
i 1
2 0.68 5
22
0.682
10 1
2
2
11.09Amp
It I C 1
2 2
Total radiated power Pt Pc 1 1 2
2
2
i 1
V(t)= (cos( c t i t ))
0.642 22
12
1
2
2
10
cos t cos( t ) sin ( t ) sin t
2
2
2
0.68 2
0.64
101
2
2
Pt = 14.44 kW
DSB:
01
Sol :
V(t)=
The above signal is a multitone SSB signal

(USB), which has frequencies
c + 1 , c + 2 . c + n , i.e.
it has component having frequencies greater
than carrier frequencies.
It is a USB
Expression for LSB
VLSB(t) =
n
[cos t cos( t ) sin t sin( t )]

i 1
VLSB(t) =
(cos( t t ))
i 1
V(t) = VLSB(t) + VUSB(t)
: 219 :
(cos( c t i t ) (cos( c t i t ))
04.
Sol:
Refer IES - 15 Previous Conventional
310, Question number: 03
i 1
2 cos c t cosi t i
i1
03.
Sol:
(i):
We cannot detect DSBSC signal using an
envelope detector
(ii)
05.
Sol: Coherent Detection or Synchronous
Detection
Let a modulated signal reaching the receiver
signal be f(t) cosct. Assuming , a locally
generated signal with frequency and phase
error equal to f and respectively,
m(t) = cos100t + 2cos300t

c(t) = Accos( c t )
Given c = 1000 rad/s
m(t)c(t) = Accos( c t )(cos100t + 2cos300t)
= Accos(1000t)(cos100t + 2cos300t)
= Ac[cos(1000t)cos100t +2cos(1000t)cos300t]
A
c cos900t cos1100t 2cos1300t 2cos700t
2
A
A
c cos900t c cos1100t
2
2
2A c
2A c
cos1300t
cos700t
2
2
A
A
c cos900t c cos1100t
2
2
A c cos1300t A c cos700t
x(t) = x LSB (t) + x USB (t)
A
A c cos700t c cos900t
2
A
A c cos1300t c cos1100t
2
A
x LSB (t) = Ac cos700t + c cos900t
2
A
x USB (t) = A c cos1300t c cos1100t
2
s(t)
DSB
signal
v1
Multiplier
v2
LPF
Ac cos 2fct
Oscillator
Fig: coherent detector
s(t) = Ac cos (2fc t) m(t)

1(t) = Ac2 cos2 (2fc t) m(t)
1 cos 4 f c t
A c2
m(t)
2
A c2
[1 cos 4 f c t ] m(t)
2
A c2
v 2 (t)
m(t)
2
v1 (t)
If the locally generated carrier is not in

phase with the incoming signal then,
v1(t) = Ac cos 2fct m(t) . Ac cos (2fct + )
= Ac2 m(t) cos 2fc t cos (2fc t + )
: 220 :
A 2 m( t )
v1 (t ) c
[cos cos(4f c t )]
2
A2
V2 ( t ) C m( t ). cos
2
A c2 m(t)
If 0 v 2 (t)
2
information is not lost
1
if 90 0 v 2 ( t ) cos (90 ) 0
2
information is lost
The detected output is zero when
=900. This is called Quadrature null
effect, because signal is zero when the
local carrier is in phase Quadrature with
the transmitted carrier.
Hence b aK2 = 0 b = aK2

b
Gain (K) =
a
SSB & VSB

01.
Sol: Given f m1 100Hz ; f m 2 200Hz ;
f m 3 400Hz ; f c 100kHz .
Standard equation of an SSB signal is
X(t)
n
A (cos c t cos( i t ) sin c t sin i t )

i 1
06.
Sol:
X(t) =A1 (cos(2f c t ) cos(2f i t )
From the block diagram

V1 = K [m(t)Acosct]
i 1
sin( 2f c t ) sin( 2f i t ))
X(t) is applied to coherent detector.
V2 = m(t) + Acosct
V0 = a V12 bV22
=a[K(m(t)Acosct)]2+b[m(t)+Acosct]2
m(t) = Am cosmt
SSB modulator
= aK m (t) aK A cos ct
2
+2aAKm(t)cosct +bm2(t) +bA2cos2ct

+2bm(t)Acosct
= m2(t)[baK2] +2Acosctm(t) [b+aK]
+A2cos2ct [baK2]
= [baK2](m2(t)+A2cos2ct)
+ 2Acosct. m(t) (b+aK)
To yield the suppressed carrier DSB signal
coefficients of m2(t) and (Acosct)2 should
be Zero because in DSB modulation carrier
Amplitude A(t)[A cosct] is proportional to
message signal m(t).
X(t)
c(t) = Ac cos Ct
x L ( t ) = B cos 2f Lo t
y(t) = x(t) x L ( t )
3
= AB [
(cos(2f
i 1
t ) cos(2f i t )
sin( 2f c t ) sin( 2f i t )) ] [cos 2f Lo t ]
= AB
= AB
f i ) t (cos 2f Lo t )
i 1
(cos 2(f c f i ) t (cos 2f Lo t )
i 1
(cos 2(f
: 221 :
3
= AB
[cos 2(f
i 1
f i ) t cos 2f Lo t ]
AB [cos 2(f c f i f Lo ) t
i 1
2
cos 2(f c f i f Lo ) t ]
LPF allows only low frequency components

to pass through
AB 3
y 2 (t)
cos(2(f c f i f Lo )t )
2 i 1
f1 = 100Hz; f2 = 200 Hz; f3 = 400 Hz
fc = 100 kHz; fLo = 100.02 kHz
(fc + fi fLo) or fLo (fc + fi)
fA = fLo + (fc + f1)
= [100.02 + (100 + 0.1)]1000
fA = 80Hz.
fB = (fc + f2) - fLo
= (100 + 0.2) (100.2) = 0 Hz.
fC = (fC+f3) - fLo
= [(0.4 +100) (100.02)]1000
fD = 380Hz
(b) If only LSB is transmitting
3
y1 ( t ) AB (cos 2(f c f i ) t )(cos 2(f Lo t )

i 1
AB 3
[cos(2(fc fi fLo )t) cos(2(fc fi fLo )t

2 i1
fA = +fLo(fcf1) =100.2 + (100-0.1)= 120 Hz
fB = + fLo (fc f2) =100.02 (1000.2) = 220 Hz
fD = + fLo (fc f3) =100.02 (100-0.4) = 420 Hz
02.
Sol: m(t) = cos 2 1500 t + cos 2 3000 t
c(t) = cos 2 (15000 t )
m(t) c(t) = (cos 2 1500 t + cos 2 3000 t )

cos 2 (15000 t )
( t ) sin c t
SSB = m(t) cos c t m
( t ) = m t
m
2
= cos 2 1500 t + cos 2 3000 t

2
2
= sin 2 (1500 t ) + sin 2 (3000 t )
LSB
( t ) sin c t
x 1 (t) = m(t) cos c t + m
x 1 (t) = (cos 2 (1500 t ) +cos 2 (3000 t ) )
cos 2 (15000 t ) +
(sin 2 (1500 t ) +sin 2 (3000 t ) )
sin 2 (15000 t )
1
= [cos(2(16500 t )) cos 2(13500 t )]
2
1
+ [cos(2(18000 t )) cos 2(12000 t )]
2
1
+ [cos(2(13500 t )) cos 2(16500 t )]
2
1
+ [cos(2(12000 t )) cos 2(18000 t )]
2
x 1 (t) = [cos(2(13500 t )) cos 2(12000 t )]
USB
( t ) sin c t
x 2 (t) = m(t) cos c t + m
= [cos(2(16500 t )) cos 2(18000 t )]
03.(i)
Sol:
The third method of generation of SSB
retains the advantages of the phase shift
method, like its ability to produce SSB at
almost any frequency and the use of low
frequencies.
: 222 :
Block diagram
Vm = sin m t
V
BALANCED
1
MODULATOR
BM1
A cos o t
Vm
LOW PASS
FILTER
LPF1
V3
90O PHASE
SHIFTER
V5
90O PHASE
SHIFTER
A sin c t
A sin o t
AUDIO
FREQUENCY
GENERATOR
A sin o t
BALANCED
MODULATOR
BM2
BALANCED
MODULATOR
BM3
A cos c t
ADDER
FREQUENCY
GENERATOR
Vo
A sin c t
V2
LOW PASS
FILTER
LPF2
V4
BALANCED
MODULATOR
BM4
V6
Output of BM1, is
V1 = A sin m t cos o t
A
V1 = [sin( m o ) t sin( m o ) t ]
2
Output of BM2 is
V2 = A sin m t sin o t
A
V2 = [cos( m o ) t cos( m o ) t ]
2
V5 =
A2
[sin( c m o ) t sin( c ( m o )) t ]
4
Output of LPF1
A
V3 = sin( m o ) t
2
Output of LPF2
A
V4 = cos( m o ) t
2
A2
[sin(c m o )t sin(c m o )t]

4
Output voltage
Vo = V6 V5
A2
[sin( c m o ) t ]
2
A2
(or) Vo
[sin( c m o ) t ]
2
Thus the output is a SSB signal.
The output of BM3 is

A
V5 = (A cos c t ) sin( m o ) t
2
The output of BM4 is

A
V6 = (A sin c t ) cos( m o ) t
2
: 223 :
(ii)
m(t)=Amcos m t SSB
Modulator
Frequency discrimination method (or)

Filter method:
by the direction of the arrows in figure. If
this system is made completely symmetrical,
the carrier frequency will be completely
cancelled. The carrier will be heavily
suppressed rather than completely removed
.The output of the balanced modulator thus
contains the two sidebands and some of the
miscellaneous components which are taken
care of by the tuning of the output
transformers secondary winding. Thus the
final output consists only of sidebands.
A mathematical analysis of the balanced
modulator is now given.
As indicated, the input voltage will be
(V1 +V2) at the gate of T1 and (V1V2) at the
gate of T2.
Circuit diagram:
x(t)
c(t) = Ac sin t
x 1 ( t ) (A m cos m t ) A c sin t
x1 (t)
AcAm
[sin m t sin m t ]
2
x(t) USB
x(t) =
AcAm
[sin m t ]
2
Time domain representation of SSB wave
m(t)+c(t)
T1
c(t)
AF in
SSB = USB
x(t)
RF in
c(t)
V2= m(t)
I d1
s(t)
Output
V1= c(t)
I d2
04.
Sol: SSB-SC waves can be generated by two
methods
(i) Frequency discrimination method (or)
Filter method
(ii) Phase discrimination method (or) phaseshift method
(iii) Third Method (or) Weavers method.
Filter method is very popular than phase
shift method.
m(t)c(t)
T2
Fig: Balanced modulator
The two drain currents, represented will be
I d1 a bV1 V2 cV1 V2
a bV1 bV2 cV12 cV22 2cV1 V2

I d 2 a bV1 V2 cV1 V2
a bV1 bV2 cV12 cV22 2cV1 V2
: 224 :
Where,
a = dc component of the collector current
b = transconductance of the transistor
c = coefficient of non-linearity
The primary current is given by the
difference between the individual drain
currents.
Thus I1 I d1 I d 2 2bV2 4cV1V2
05.
Sol: Refer IES 15 E & T Conventional Paper
II Q.03, page: 310.
06.
Sol: Given:
Number of Voice inputs = 24
Bandwidth of each voice input fm = 4 KHz
Let the carrier voltage c(t) = Vc cosct

The modulating voltage m(t) = Vmcosmt
I1 = 2bVmcosmt + 4cVmVccosct cosmt
I1 = 2bVmcosmt +
FDM system using AM-SSB so band width

(fm) = 4 kHz
4cVmVc 1 cosc m t cosc m t

2
The output voltage V0 is proportional to this
primary current.
The proportionality constant be . Then
N = 24 ; n = 8
Transmission Bandwidth = 244 = 96 kHz

using PCM - TDM.
V0 = I1
fs = 2fm = 8 kHz
Bandwidth
So PCM TDM requires more bandwidth

than the AM-SSB-FDM system.
3. Angle Modulation
2 bVm cos m t
2 cVm Vc cos c m t cos c m t
On simplifying, let
Q = 2cVm Vc Then
P 2bVm
Lower sideband
Frequency Modulation
and
V0 P cos m t Q cos c m t Q cos c m t

Modulation
frequency
Upper sideband
Above equation shows that the carrier has

been cancelled out, leaving only the two
sidebands and the modulating frequencies.
The tuning of the output transformer will
remove the modulating frequencies from the
output, but it is also possible to suppress
them by the action of the circuit
01. (a)
Sol:
The spectrum of AM and NBFM are
identical except that the spectral component
of NBFM at frequency fc fm is 180 out of
phase. The difference in AM and NBFM can
be shown through the following spectrums:
a. Spectrum for AM
S(f)
0
R b Nnf s 24 8 2 4
768 kHz
2
2
2
A c
4
fc fm
Ac
2
fm
A c
4
fc + fm
: 225 :
b. Spectrum for NBFM
1 d (t)
fi
2 dt
S(f)
Ac
2
fc fm
0
fm
A c
4
(t) 2 f i (t) d t
A c
4
fc + fm
(t) 2 [f c K f m(t)] d t
0
(t) 2 f c t 2 K f m(t) d t
Frequency Modulation:
Changing the frequency of the carrier
according to the message signal amplitude
variations is called Frequency Modulation.
Instantaneous frequency,
fi(t) = fc + Kf m(t)
Kf = Frequency sensitivity (Hertz / Volt)
S(t)
A c cos [2 f c t 2 K f
m(t) d t ]
FM
signal
S(t) = Ac cos [ 2fc t + Kp m(t) ]
signal
PM
For a single tone frequency modulation

For single tone modulation
fi(t) = fc + Kf Am cos 2fm t
fi , max = fc + Kf Am
fi , min = fc - Kf Am
f = Kf Am = Frequency deviation
Carrier Swing (Total variation of carrier
frequency) = 2 f
m(t) = Am cos 2fm t
2K f A m
S( t ) A c cos 2f c t
sin 2f m t
2f m
K A
A c cos 2f c t f m sin 2f m t
fm
Modulation index of FM
Message (Modulating) signal

t
Carrier signal
K f A m f
fm
fm
<< 1, Narrow band FM

t
FM output
>> 1, Wide band FM
Let s(t) = Ac cos (t)

(t) = 2fi t
d(t) / dt = 2fi
be the FM wave
Narrow Band FM : ( < < 1)

S(t) = Ac cos (2fc t + sin 2fm t)
= Ac [cos 2fc t . cos ( sin 2fm t )
sin 2fc t . sin ( sin 2fm t)]
: 226 :
When < < 1
Wideband FM : ( > > 1)
cos ( sin 2fmt) 0
S(t) = Ac cos (2fc t + sin 2fm t)
sin ( sin 2fmt) sin 2fmt
Bessel function of order n is given by
Ac cos 2fc t Ac sin 2fc t sin (2 fmt)

A c cos 2f c t
Ac
cos 2(f c f m ) t cos 2 f c f m t
2
S(t) Ac cos2fct
S(f )
1
Jn(x) = e j( x sin n ) d
2
Ac
A
cos2(fc fm )t c cos2(fc fm )t
2
2
S(t) = Ac
Ac
f f c f f c
2
A c
f f c f m f f c f m
4
A c
f f c f m f f c f m
4
+ Ac J2() cos2 (fc+2fm)t

+ Ac J2() cos 2 (fc2fm)t+ ..
S( t ) A c J 0 () cos 2f c t
except that the spectral component at fc fm
A c J1 [cos 2 f c f m t cos 2f c f m t ]
is 1800 out of phase.
A c J 2 [cos 2f c 2f m t cos 2f c 2f m t ]
S(f)
Generation of NBFM Signal:

Product
Modulator
Am sin 2fmt
Ac sin (2fc t)
Oscillator
fcfm
0
NBFM
signal
fc2fm
fc
Ac cos 2fct
Ac cos 2fct
A c J 0 ()
2 A c J 1 ()
A c J 2 ()
2
2
A c J 2 ()
2
s(t)
90
nf m t
+ Ac J1() cos2 (fc+fm)t

+ Ac J1() cos2(fc fm)t
The spectrum of AM and FM are identical
Am cos 2fmt
S(t)=Ac J0() cos2fc t
B.W of NBFM = 2 fm
90
n = 0, 1, 2,
2
Pt = Pc 1
2
J cosf
fc+fm fc+2fm
A c J 1 ()
2
Theoretical bandwidth of a WBFM is .

fc = 5 kHz
kf = 10Hz/V
f(t) = 100 cos200 t
200
fm =
100Hz
2
: 227 :
Am = 100
f = kf Am =10 100 = 1000 Hz
1000
f
=
= 10
=M.I =
fm
100
BW = 2( +1)fm = 2(10+1)(100)
= 2 11 100
BW = 2.2 kHz
01. (b)
Sol:
Envelope detector
1. Used for
demodulation of
AM signal
2. Need to set
accurate time
constant (RC).
3. Synchronization
is not required.
Low pass 1/2 Accos m(t) Demofilter

dulated
signal
Cos (2fct + )
Accos
(2fct)
m(t)
Voltage
Controlled
oscillator
Phase
discriminator
90
phase shifter
sin(2fct + )
Synchronous detector
1. Used for
demodulator of
DSBSC, SSB,
VSB etc.
2. No need of time
constant term.
3. Synchronization is
required.
Costas Receiver:
This system consists of two coherent
detectors supplied with the input signal of
DSB-SC wave Ac cos (2fct) m(t) but with
individual local oscillator signals that are in
phase quadrature to each other. The
frequency of the local oscillator is adjusted
to be the same as the carrier frequency fc,
which is assumed known a priori.
The detector in the upper path is referred to
as the in-phase coherent detector or Icahnnel and in lower path is referred to as
the quadrature phase detector or Q-channel
there two detectors are coupled to form, ve
feed back system designed in such a way as
to
maintain
the
local
oscillator
synchronization with the carrier wave. We
find I-channel output contains the desired
demodulated signal m(t), where as Qchannel output is zero due to the quadrature
null effect of Q channel.
Product
modulator
Product
modulator
LPF
1/2Acsinm(t)
Suppose the local oscillator phase drifts

from its proper value by a small amount of
radians. The I-channel output will remain
essentially unchanged, but there will be
some signal appearing at the Q-channel
output. Which is proportional to sin .
This Q-channel output will have the same
polarity as the I-channel output for one
duration of local oscillator phase drift and
opposite polarity for the opposite direction
of local oscillator phase drift.
Squaring circuits (loop):
This is one method for generating a
reference carrier from DSB-SC wave
Accos
(fct)
m(t)
PLL
Narrow
squarer
band
v(t)
2
filter
H(f)
Y(t)= s (t)
LPF
e(t)
VCO
Frequency
Divider
by 2
Carrier wave
of frequency fc
: 228 :
DSB SC wave applied to the input of square

law device, we obtain
yt A c2 cos 2 2f c t m 2 t
frequency deviation and the modulating

frequency
BW = 2 ( f f m )
A c2 2
m t 1 cos 4f c t
2
This is known as Carsons rule.

Given f = 75 kHz.
BW = 2 ( +1)fm
This is y(t) is applied to the narrow-band

filter centered about 2fc.
(a) = 0.25 =
At the multiplier output we obtain two

terms, one depending on the difference
between the frequency and phase of the
VCO output and those of signal v(t) at the
PLL input, and the other depending on their
sum. This later term is removed by the loop
filter.
Finally, VCO output is frequency divided by
a factor of two, to produce the desired carrier
wave which is available for coherent
detection of DSB-SC wave.
10 3 1000
500Hz
2
2
i) Modulation Index ()
=
75
300kHz
0.25
BW for NBFM ( 1 ) is
Given as BW = 2 fm
BW = 2 300 = 600 kHz
(b) = 5.5 =
f
fm
fm =
75
5 .5
fm = 13.636
BW = 2 ( +1)fm = 2(5.5 + 1)(13.636)
= 177.268 kHz
(c) = 7 fm =
f
75
=
= 10.714 kHz
7
BW = 2 ( +1)fm = 2(7+1)(10.714)
5 10 3
f
10
fm
500
Capture range of PLL the range of frequency

over which the PLL can acquire a lock with
the input signal.
iii) Same as question 7
f
fm
fm =
02.
Sol: Given data modulating signal
(em) = Em sin 103 t
Frequency deviation (f) = 5KHz
Modulating frequency
(fm) =
BW = 171.424 kHz
ii)
03.
Sol: Carsons rule: The Bandwidth is anote
mode twice the sum of the maximum
04.
Sol: It is also known as Indirect method of FM
generation. The basic principle of this

method is to generate NBFM signal
indirectly
by
using
a
PHASE
MODULATION TECHNIQUE and then
converting this NBFM into WBFM by using
frequency multiplier.
: 229 :
modulating
signal
NBPM NBFM
signal
Integrator
m(t)
Frequency
multiplier
WBFM
signal
x FM t A c cos c t sin m t where 1
Crystal
Oscillator
NBFM Generation:
Acsin c t
/2 phase
shift
Accos c t
Crystal
oscillator
Adder
_
Balanced
modulator
Acsin c t
NBFM
signal
A c sin c t.sin m t
m( t ) sin m t
A
X NBFM t A c cos c t c cosc m t
2
A
c cosc m t
2
where 1
In Armstrong method multiplier Circuit is much
important, since It converts NBFM into
WBFM.
Frequency multiplier:
It is a nonlinear device.
Eg:- If x(t) is input & applied to multiplier
Circuit
then,
x(t)
Frequency
Multiplier
Y(t)=x2(t)
Note:
The multiplier Circuit changes the carrier

frequency (fc), modulation index ( ) and
so frequency deviation f by n time
where n is the multiplication factor.
Advantages:
Stability is high because crystal oscillator

can be used to generate are highly stable
signals with small carrier frequencies.
05. (i)
Sol: Frequency deviation f m f f m
Where m f mod ulation Index
X FM ( t ) A c cosc t sin m t
A c2
1 cos2c t 2 sin m t
X FM ( t )
2
f m mod ulation frequency
0.4f m
: 230 :
(ii)
Sol: In FM Transmitted power does not depend
upon modulation index. So on doubling
the modulation Index there will be no
change in the total transmitted power.
07.
Sol: PLL Technique is an Indirect method of
FM demodulation technique.
A Phase-Locked Loop (PLL) is primarily
used in tracking the phase and frequency of
the carrier component of an incoming FM
signal. PLL is useful for synchronous
demodulation of AM-SC signals or signals
with few cycles of pilot carrier. PLL is also
useful for demodulating FM signals in
presence of large noise and low signal
power.
Block Diagram:
e(t)
FM Wave
s(t)
Loop
Filter
V(t)
b(t)
VCO
PLL is basically a negative feedback

system.
It consists of three major
components.
These components are
multiplier, a loop filter and a voltage
controlled oscillator (VCO) connected
together in the form of a feedback loop. A
VCO is a sine wave generator whose
frequency is determined by the voltage
applied to it from an external source.
It means that any frequency modulator can
work as a VCO.
Working Operation:
The operation of a PLL is similar to any
other feedback system. In any feedback
system, the feedback signal tends to follow
the input signal. If the feedback signal is
not equal to the input signal, the error
signal will change the value of the
feedback until it is equal to the input

signal. The difference signal between s(t)
and b(t) is called an error signal. The error
signal or difference signal e(t) is adjusted
to the VCO frequency in such a way that
the instantaneous phase angle comes close
to the angle of the incoming signal s(t). At
this point, the two signals s(t) and b(t) are
in synchronism and the PLL is locked to
the incoming signal s(t).
Initially assume that VCO is adjusted so
that when the control voltage is zero, the
following two conditions are satisfied.
(i) The frequency of the VCO is precisely set
at the unmodulated carrier frequency (fc)
(ii) The VCO output has a 900 phase-shift with
respect to the unmodulated carrier wave.
Capture range: capture range is defined as
the band of frequencies centered around the
VCO natural frequency (fo) where the PLL
can initially establish or acquire frequency
lock with an external input signal from an
unlocked condition
Lock range: It is defined as the band of
frequencies centered on the VCOs natural
frequency over which a PLL can maintain
frequency lock with an external input
signal. it is the range of frequencies over
which the PLL will accurately track or
flow the frequency of the external input
signal after frequency lock had occurred.
08.
Sol: For speech processing generally we prefer
SSB type of modulation because the
bandwidth of SSB modulation is very low
compared to other modulation techniques
and the voice frequency normally ranges
from 0.3kHz to 3.4 kHz and SSB
modulation is best suitable in this
frequency range.
: 231 :
09.
Sol: Refer IES 15 E & T Conventional Paper
II Q.12, page: 323
10.
Sol: Given Audio frequency (fm) = 500Hz
Audio frequency voltage (Am) = 2.4V
Deviation (f) = 4.8KHz
We know that
Modulation index () =
f 4.8 10 3
=
500
fm
= 9.6
And deviation (f) = KfAm
4.8 103 = Kf(2.4)
Frequency sensitivity (Kf) = 2 103 Hz/V
i)
ii)
Now Am = 7.2V
Deviation (f) = KfAm
= 2 103 7.2
f =14.4KHz
f
Modulation index () =
fm
= 28.8
Am = 10V, fm = 200Hz
Deviation (f) = KfAm
= 2 103 10
f = 20103 = 20KHz
f
Modulation Index () =
fm
=
20 10 3
= 100
200
Disadvantage of FM over AM:

1. A much wider channel is required by FM,
up to 10 times as large as that needed by
2.
AM.
This is the most significant
disadvantage of FM
FM transmitting and receiving equipment
tends to be more complex, Particularly for
modulation and demodulation. So Circuit
is more complex than AM.
Threshold effect is more produced in FM
than AM.
11.
Sol: Frequency modulation is one type of angle
modulation in which the instantaneous
frequency is varied linearly with a message
or base band signal about an un-modulated
carrier frequency. This means that the
instantaneous value of the angular
frequency will be equal to the carrier
frequency plus a time-varying component
proportional to the base band signal.
Advantages of FM over AM
(i) FM receivers may be fitted
with amplitude limiters to remove
the amplitude variations caused by
noise. This makes FM reception a
good deal more immune to noise than
AM reception.
(ii) It is possible to reduce noise still
further by increasing the frequencydeviation. This is a feature which AM
does not have because it is not
possible to exceed 100 percent
modulation without causing severe
distortion.
(iii) Standard
frequency
allocations
provide a guard band between
commercial FM station. Due to this,
there is less adjacent- channel
interference than in AM
(iv) FM broadcast operate in the upper
VHF and UHF frequency ranges at
which there happens to be less noise
: 232 :
than in the MF and HF ranges

occupied by AM broad casts.
(v) The amplitude of the FM wave is
constant. It is thus independent of the
modulation depth, where as in AM,
modulation depth governs the
transmitted power. This permits the
use of low-level modulation in FM
transmitter and use of efficient class
C
amplifiers in all stages following
the modulator
(vi) In AM transmitter the maximum
power is four times the average
power. Finally in FM, all the
transmitted power is useful where as
in AM most of the power is carrier
power which does not contain any
information.
12.
Sol: Given: carrier c(t) = 10 cos ct
AC = 10
Modulating message signal m(t) =
3cosm(t)
Am = 3
fc = 100 kHz
fm = 4 kHz
Amplitude modulation
A
3
0 .3
1. Modulation index m
A C 10
2. Channel BW = 2fm = 2 4K = 8 kHz
Frequency modulation
Given sensitivity of the frequency
modulator to be 5 kHz/volt
f
fm
Frequency deviation f = Kf Am
= 51033 = 15103
15 10 3 15
3.75
4
4 10 3
As >1. This is WBFM

The channel Bandwidth B = 2(+1) fm
= 2(3.75+1)4K
= 38 kHz
13.
Sol: Given modulating signal em(t) = Amsinmt
modulating signal passed through a square
law device then the o/p em(t)= A 2m sin 2 m t
Then the instantaneous frequency expressed as
fi(t) = fc+kem(t) = fc+k E 2m sin2 mt

1 cos 2m t
f i (t) f c kE m2
Angle can be obtained as

t
i(t) = 2 f i (t)dt
0
1 cos 2 m t
2 f c kE 2m
dt
2

0
t
sin 2 m t
2f c t kE 2m t
2 m
sin 2 m t
i t 2f c t kE 2m t
2
FM signal equation Ac cos i(t)

XFM(t)
=Ac cos 2f c t kE 2m t
sin 2 m t
2 m
: 233 :
14.
Sol: fc = 105 MHz ; fmax = 105.03 MHz
20 10 3 3.2
128
2 250
fm = 5 kHz
(i)
fmax = fc +f
f = fmaxfc
= 105.03 105 = 0.03 MHz = 30 kHz
(ii) Carrier swing = 2f = 60 kHz
f 30 10 3
(iii) Modulating index
6
fm
5 10 3
(iv) Percentage of modulation
16.
Sol: FM Generation by Direct method - VCO
In the direct method of generating an FM
signal, the modulating signal directly
controls the carrier frequency. A common
method used for generating FM directly is
to vary the inductance or capacitance of a
tuned electric oscillator. Any oscillator
whose frequency is controlled by the
modulating signal voltage is called a
voltage controlled oscillator (VCO).
%f
30kHz
100
100 40%
(f ) max
75kHz
15.
Sol: Given modulating frequency f m 400Hz
Modulating voltage A m 2.4 V
Modulation index 60
From modulation index
Kf Am
Kf Am
60
2 400
m
K f A m 48 kHz
K f 20 10 3 rad / volt
When modulating frequency f m 250Hz
Modulating Voltage (Am) = 3.2V
Modulation index 1
Kf Am
2 f m
Voltage
variable
capacitor
modulating
signal m (t)
VCO
Cv
C0
Tank circuit
A fixed capacitor c0 is shunted by a voltage

variable capacitor cv commonly called as
varicap, is one whose capacitance value
depends on the dc biasing voltage across its
electrodes.
The modulating signal varies the voltage
across cv. The capacitance of cv changes
and causes a corresponding change in the
oscillator frequency. The modulating
frequency is vary small in comparison with
the oscillator frequency. Therefore the
fractional change cv may be very small
during the cause of many cycles of the
oscillator signal with this variable
capacitance, the instantaneous oscillator
1
frequency will be given by f
.
2 LC
: 234 :
v (t) = [a1 s (t) + a2 s2 (t) + . + an sn (t)]

where, a1, a2, . an are coefficients
determined by the operating point of the
device and n is the highest order of
nonlinearity.
Output
Input
V (t)
Vosc (t)
VCO
Variable
voltage
The block level relation can be written as

t
Vosc t B cos c t G 0 V d
B amplitude of the output of VCO

c = angular frequency (radians)
G0 = frequency sensitivity (rad/volt)
i.e., the rate of change of instantaneous
angular
frequency
w.r.t
frequency
controlling voltage v (t).
The instantaneous angular frequency
I (t) = c + G0 v (t)
The input s (t) is a FM wave designed by

t
S t A c cos 2f c t 2 k f m d
0
The instantaneous frequency is

f i (t) f c k f m (t)
FM wave s(t) fc
with fc &
1.
2.
Frequency
multiplier
Frequency
multiplier
NBFM
f1
X n1
f1
f2 = n1f1
f3
fLo
Xn2
Fig: Block diagram of an indirect FM

generation in the indirect method, the
message signal is first used to produce a
NBFM which is followed by frequency
multipliers.
The frequency multiplier consists of a
memory less nonlinear device followed by
a band pass filter. The input output
relation of such a device may be expressed
in the general form.
FM wave s1(t)
with carrier
frequency
fc| = nfc & n
Advantages:
The frequency modulation is achieved if v

(t) is the modulating message signal m (t)
FM generation by indirect method
Armstrongs method.
m(t)
Memory less v(t)

BPF
Nonlinear
with mid band
Device
Frequency nfc
fc
f
3.
Indirect
method
uses
the
frequency multiplication
which
increases the frequency deviation to
the desired level.
The carrier frequency stability
problem is overcome by using a
highly stable oscillator like crystal
oscillator.
The NBFM wave is multiplied
in frequency by means of a
frequency multiplier so as to produce
the desired WBFM.
18.
Sol: Given carrier signal =Ac cosct
Modulating signal = Am sinmt
A frequency modulated wave with
sinusoidal modulation can be expressed as
i(t) = c + kfAmsinmt
Where
m = 2fm
c = angular velocity of carrier wave
i = instantaneous angular velocity
The frequency deviation is proportional to
the peak amplitude of the modulating
: 235 :
signal and is independent of the

Modulating frequency.
i = c + sinmt
The phase angle of the modulated wave is
obtained by
i i dt
19.
Sol: Refer question number: 16
21.
Sol: Refer Question number . 07
sin m t dt
c t
cos m t
m
01.
i c t cos m t
Where modulation index
Phase Modulation
Sol: Given AM signal is
The FM signal is given by
FM t A cosc t cos m t
cosc t cos cosm t

FM t A
sin c t sin cosm t

The term cos (sin mt) can be expanded
into
J o 2J 2 cos2 m t
2J 4 cos4 m t .....
And sin cos m t can be exp anded as .
2J1 sin m t 2J 3 sin 3m t ...
V(t) = 5cos( 10 7 t 2 sin 500t ) ---- (1)

Standard equation for phase modulated and
frequency modulated signals is
xpm(t) = Ac cos( c t + Kp m(t))----(2)
where Bp = KpAm =
Modulation index or phase deviation of
phase modulated signals.
x FM ( t ) A c cos( c t 2K f m( t )dt )
Which can be further modified as
x FM ( t ) A c cos( c t B F sin m t ) ----- (3)

When m(t) = Amcos m t
FM t J o cos c t
And where BF = modulation index
J1 sin c m t sin c m t
f frequency deviation
fm
message frequency
J 2 cos c 2 m t cos c 2 m t
Comparing (1) & (2) we get
Bp = KpAm = Phase deviation
J 6 and so on
BWpm = 2 (Bp+1)fm = 2 (2+1)(250)
= Modulation Index = 2.
= 6 250= 1500 Hz = 1.5 kHz
: 236 :
02.
Sol: Given m(t) = 10 sin (2104)T
4
Em = 10 V
fm = 10 Hz
P = 10
f = 10
Here, modulation index f will be denoted

as mf
(i)
For FM system
f = Kf Em
mf
10
Therefore, K p
mp
Em
10
1
10
f = 1 10 104 = 105 Hz
BW = 2 (f +fm)
(Carsons rule)
5
4
= 2 (10 + 10 ) = 220 kHz
(ii) Given
fm = 2 104 Hz ,
Em = 10V
f = mf = 10, p = mp = 10
For FM System
f = Kf Em = 104 10 = 105 Hz
f K f E m
fm
fm
Bw = 2 (f + fm)
K f 10
10 4
= 2 (105 +2 104) = 240 kHz
Kf = 10 4 Hz/V
f = Kf Em = 104 10 = 105 Hz
For PM system:
f = Kp Em fm
= 1 10 2 104 = 2 105 Hz.
It has been found that for sinusoidal

modulation, 98% power transmission
occurs.
When
n = mf + 1, then BW is
BW = 2nfm = 2 (mf +1)fm

= 2 mf fm + 2fm = 2 (f + fm)
Therefore, BW = 2 (f + fm)
(Carsons Rule)
= 2 (105 + 104) = 220 kHz
For PM system:
f = Kp Emfm
mp = KpEm
BW = 2 (f + fm)
= 2 (2 105 +2 104) = 440 kHz
(iii) Given
fm = 2 104 Hz,
f = mf = 10
Em = 5V
p = mp = 10
For FM system:
f = Kf Em = 5 104 Hz
BW = 2 (f + fm) = 2 (5 104 + 2 104)
= 140 kHZ
For PM system:
f = KpEmfm = 1 5 2 104 = 105 Hz
BW = 2 (f + fm) = 240 kHz
: 237 :
are making the resonant frequency of the

tuned circuit equal to the carrier frequency
of the required channel.
4. Radio Receivers
01.
Sol: The super heterodyne receiver has a good
gain, selectivity, sensitivity and hence it
finds practical use.
TRF receiver has a non uniform selectivity

hence super heterodyne receiver is
preferred
The Block diagram of Super Heterodyne
Receiver is shown in figure.
RF
Pre
amp
selector
LS
IF amp
AF
Detector
Power
AF
Amp
Local
oscillator
1.
2.
RF Section: The RF amplifier internally

consists of a pre selector and an amplifier
stage in a single combined circuit.
The pre selector is a broad-tuned BPF
with an adjustable center frequency that is
tuned to the desired carrier frequency.
The RF amplifier must be a low noise
amplifier which reduces the noise
bandwidth of the receiver and provides the
initial step reducing the overall receiver
bandwidth to the minimum bandwidth
required to pass the information signals.
The RF amplifier determines the sensitivity
of the Receiver. It is also called as tuned
RF amplifier. By tuning arrangement, we
2 LC
Heterodyne means to mix two frequencies

together in a non linear device or to
translate one frequency to another using
non linear mixing.
IF Section
BPF
fr
The advantages of including RF amplifiers

in a Receiver are
1. Greater gain better sensitivity
2. Improved image frequency rejection
3. Better signal to noise ratio
4. Better selectivity
In Super Heterodyne Receiver the RF

signal IF signal AF signal
Mixer
fr
It consists of five sections namely, the RF

section, the mixer section, the IF section,
the audio detector and audio amplifier
section.
RF Section
Mixer section: The mixer stage is a non

linear device and its purpose is to convert
radio frequencies to intermediate frequency
(RF-to-IF)
frequency
translation.
Heterodyning takes place in the mixer
stage and radio frequencies are down
converted to intermediate frequencies.
Always the local oscillator frequency
should be greater than the signal frequency.
fl >> fs
fl-fs (down
fs
RF
Mixer
fl
IF conversion)
fl+fs (up
conversion)
Local
oscillator
: 238 :
A constant frequency difference is

maintained between the local oscillator and
the RF circuits. The down conversion is
done with respect to the tuned circuit.
Tuning means changing the local oscillator
frequency. Mixer will change the carrier
frequency from fs to fIF. The most common
intermediate frequency used in AM broad
cast receiver is 455 kHz
3.
IF Section: IF section consists of an IF

amplifier and BPF. Most of the receiver
gain and selectivity is achieved in the IF
section. IF is always lower in frequency
than the RF because it is easier and less
expensive to construct high-gain stable
amplifiers for the low-frequency signals.
IF = fl fs
Choice of IF:
1. If the IF is too high, poor selectivity
and poor adjacent channel rejection.
2. A high value of IF increases tracking
difficulties
3. If IF is very low, image frequency
rejection becomes poorer.
4. If IF is very low, the frequency
stability of the local oscillator should
be very high.
5. The IF must not fall with in the tuning
range of the Receiver.
4.
5.
Detector section: The purpose of the

detector section is to convert the IF
signals
back to the original source
information. The detector is generally
called an audio detector in the broadcast
receiver because
the
information
signals are audio-frequencies.
Audio Amplifier Section: The audio
section comprises several cascaded audio
amplifiers. The number of amplifiers used
depends upon the audio signal power.
Advantages:
a)
Greater
gain,
sensitivity
thus
imporoved
b)
Greater adjacent channel rejectivity
c)
Better image signal rejection
d) Better stability.
e) Better signal to noise ratio.
02.
Sol: The Advantages of the super heterodyne
receiver make it the most suitable type for
the great majority of radio receiver
applications, such as AM, FM, single side
band, television and even radar receivers.
Advantages:
a) Greater
gain,
thus
imporoved
sensitivity
b) Greater adjacent channel rejectivity
c) Better image signal rejection
d) Better stability.
e) Better signal to noise ratio.
Disadvantage:
a) Circuit Complexity
b) High cost
Constant IF is achieved through use of

trimmers and padders in mixer stage
RF amplifier boosts up the signal to a
high value before it encounters the noisy
mixer stage.
03.
Sol: Heterodyne means to mix two frequencies
together in a non linear device or to
translate one frequency to another using
non linear mixing.
The super heterodyne receiver has a good
gain, selectivity, sensitivity and hence it
finds practical use.
: 239 :
TRF receiver has a non uniform selectivity

hence super heterodyne receiver is
preferred
The Block diagram of Super Heterodyne
Receiver is shown in figure.
It consists of five sections namely, the RF
section, the mixer section, the IF section,
the audio detector and audio amplifier
section.
In Super Heterodyne Receiver the RF
signal IF signal AF signal
RF Section
BPF
IF
amp
AF
Detector
Power
AF
Amp
Local
oscillator
RF
Section:
The
RF
amplifier
internally consists of a pre selector and an
amplifier stage in a single combined
circuit.
The pre selector is a tuned BPF with an
adjustable center frequency that is tuned to
the desired carrier frequency.
The RF amplifier must be a low noise
amplifier which reduces the noise
bandwidth of the receiver and provides the
initial step for reducing the overall receiver
bandwidth to the minimum bandwidth
required to pass the information signals.
The RF amplifier determines the sensitivity
of the Receiver. It is also called as tuned
RF amplifier. By tuning arrangement, we
are making the resonant frequency of the
tuned circuit equal to the carrier frequency
of the required channel.
fr
fr
1
2 LC
The advantages of including RF amplifiers

in a Receiver are
1. Greater gain better sensitivity
2. Improved image frequency rejection
3. Better signal to noise ratio
4. Better selectivity
LS
Mixer
1.
IF Section
Pre
RF
selector amp
2.
Mixer section: The mixer stage is a

non linear device and its purpose is to
convert radio frequencies to intermediate
frequency (RF-to-IF) frequency translation.
Heterodyning takes place in
the mixer
stage and radio frequencies are down
converted to intermediate frequencies.
Always the local oscillator frequency
should be greater than the signal frequency.
fl >> fs
fs RF
Mixer
fl
fl-fs (down
IF conversion)
fl+fs (up
conversion)
Local
oscillator
A
constant
frequency
difference
is maintained between the local oscillator
and the RF circuits. The down
conversion is done with respect to the
tuned circuit. Tuning means changing the
local oscillator frequency. Mixer will
change the carrier frequency from fs to fIF.
: 240 :
The most common intermediate frequency

used in AM broad cast receiver is 455 kHz
3.
IF Section: IF section consists of an IF

amplifier and BPF. The receiver gain and
selectivity is achieved in the IF section. IF
is always lower in frequency than the RF
because it is easier and less
expensive
to construct high-gain stable amplifiers for
the low-frequency signals.
IF = fl fs
Choice of IF:
1. If the IF is too high, poor selectivity
and poor adjacent channel rejection.
2. A high value of IF increases tracking
difficulties
3. If IF is very low, image frequency
rejection becomes poorer.
4. If IF is very low, the frequency
stability of the local oscillator should
be very high.
5. The IF must not fall with in the tuning
range of the Receiver.
4.
Detector section: The purpose of
the detector section is to convert the IF
signals
back to the original source
information. The detector is generally
called an audio detector in the broadcast
receiver because
the
information
signals are audio- frequencies.
5. Audio Amplifier Section: The audio
section comprises several cascaded audio
amplifiers. The number of amplifiers used
depends upon the audio signal power.
04. (a)
Sol: Selectivity: Selectivity is a receiver
parameter that is used to measure the
ability of the receiver to accept a given
band of frequencies and reject all others.
Selectivity determines the adjacent-channel
rejection of a Receiver. It is also
determined by the response of the IF
section, with the mixer and RF amplifier

input circuits. It varies with the receiving
frequency if ordinary tuned circuits are
used in the IF section.
Selectivity can be measured experimentally
by attenuation factor and frequency of the
same signal.
The attenuation can be given as the ratio of
the voltage required for resonance to the
voltage required when the generator is
tuned to the receivers frequency.
Generally attenuation is expressed in
decibels (dB)
In the commercial AM broadcast band,
each stations transmitter is allocated a 10kHz band width. For a receiver to select
only those frequencies which are assigned
to a single channel, the receiver must limit
its Bandwidth to 10 kHz. If the pass band is
greater than 10 kHz, more than one channel
may be received and demodulated
simultaneously .If the pass band is less than
10 kHz a portion of the modulating signal
information of that channel is rejected or
blocked from entering the demodulator.
Experimental technique: one common
way is to simply give the bandwidth of the
receiver at the -3dB points. This
Bandwidth is not necessarily a good means
of determining how well the receiver will
reject unwanted frequencies. It is common
to give the receiver bandwidth at two levels
of attenuation .For example -3dB & -60
dB. The ratio of these two bandwidths is
called the Shape Factor
SF
Bandwidth (60dB)
Bandwidth (3dB)
SF = shape factor (unit less)

Bandwidth (-60dB) = Bandwidth 60dB
below maximum signal level
: 241 :
Bandwidth (-3dB) = Bandwidth 3dB below

maximum signal level
Ideally the Bandwidth at the -3dB and 60dB points would be equal, and shape
factor = 1. But, this is impossible, receiver
might have a-3dB bandwidth of 10kHz and
60dB bandwidth of 20kHz which gives a
SF of 2.
A radio receiver must be capable of
separating the desired channels signals
without allowing interference from an
adjacent channel to spill over into the
desired channels pass band.
Curve for selectivity of the receiver:
100
selectivity-V
Sensitivity:
It is defined as the minimum signal
strength that should be maintained at the
input of a receiver to get a standard output.
Sensitivity depends up on the over all gain
of an amplifier. If the gain of the amplifier
is high, the sensitivity is also high. The
sensitivity of a receiver is usually stated in
micro volts of received signal. The signal
to noise ratio and the power of the signal at
the output of the audio section are used to
determine the quality of a received signal.
In commercial AM broadcast band
receivers, a 10-dB or more signal to noise
ratio with 1/2 W (27 dBm) of power at the
output of the audio section is used.
For broadband microwave receivers, a
40dB or more signal to noise ratio with
approximately 5mW (7dBm) of signal
power is the minimum acceptable value.
80
60
40
20
40 30 20 10
+10 +20 +30 +40
Frequency, kHz
Receiver tuned to 950 kHz
100
Attenuation in dB
80
60
40
20
40 30 20 10
10 +20 +30 +40
Generator detuning, kHz
For a typical sensitivity of a commercial

broadcast band AM receivers is 50 v and
a two way mobile radio receivers generally
has a sensitivity bandwidth of 0.1 V and
10 V.
Curve for sensitivity of the receiver:
The receivers sensitivity is also called
receivers threshold. The sensitivity of an
AM receiver depends on the noise power
present at the input to the receiver, the
receivers noise figure, the sensitivity of
the AM detector, and the bandwidth
improvement factor of the receiver. The
best way to improve the sensitivity of
receivers is to reduce the noise level. This
can be accomplished by reducing either the
temperature or the bandwidth of the
receiver or improving the receivers noise
figure.
: 242 :
There are three forms of distortion that can

deteriorate
the
fidelity
of
a
communications system
16
15
Sensitivity-V
14
1. Amplitude
2. Phase
3. Frequency
13
(b).
Sol: Demodulation of SSB waves:
12
11
SSB
10
s(t)
100
1000
Fidelity: Fidelity is a measure of an audio

signal quality. It is defined as the ability of
the receivers to reproduce all audio
frequencies equally in the entire tuning
range at the output of receivers, High
fidelity or
Hi- fi systems are used for
high quality output.
Ex: CD, DVD players.
It will produce an exact replica of the
original source information. Any frequency
phase or amplitude variations that represent
in the demodulated waveform that were not
in the original information signal are
considered as distortions.
Curve for the Fidelity of a receiver:
Saturation
1 dB
compression
point
1 dB
Third order
distortion
output level
80
20
( t )
LPF
message
signal
1600
Ac cos 2fc t
Frequency, kHz
Output
power
(dBm)
Product
Modulator
20
The base band signal s(t) can be recovered

from the SSB wave s(t) by using coherent
detection.
The product modulator is having two
inputs. one input is the SSB modulated
wave s(t) & another input is the locally
generated carrier cos(2fct)
Thus the product modulator output is
v(t) = s(t)cos(2fct) . (1)
A
( t ) sin 2f c t
s( t ) c m( t ) cos 2f c t m
2
... (2)
substituting equation (2) in equation (1) ,
we get
Ac
sin 2f c t cos 2f c t
m(t) cos 2f c t m(t)
2
Ac
2 m( t ) cos 2f c t cos 2f c t
v(t) =
Ac m
2 ( t ) cos 2f c t sin 2f c t
v(t)
We know that,
1
cosA B cosA B
2
1
cos A. sin B sin A B sin A B
2
cos A. cos B
Input power
(dBm)
v( t )
Ac
m( t )cos2f c 2f c t cos2f c 2f c t
4
v(t) =
Ac
m( t )cos4f c t cos(0)
4
v(t) =
Ac
( t )sin 2f c 2f c t sin 2f c 2f c t
m
4
Ac
( t )sin 4f c t sin(0)
m
4
Ac
m( t )cos4f c t 1
4
v(t) =
Ac
A
m( t ) c m( t ) cos4f c t
4
4
v( t )
Ac
( t )sin 4f c t 0
m
4
Ac
( t ) sin 4f c t
m
4
Ac
A
( t ) sin 4f c t
m( t ) c m( t ) cos4f c t m
4
4
Ac
m( t ) = scaled message signal
4
( t ) sin 4f c t
m( t ) cos4f c t m
= unwanted terms
When v(t) is passed through the filter, it
will allow only the 1st term to pass through
and will reject all other unwanted terms.
Thus at the output of the filter we get the

scaled message signal thus the coherent
SSB demodulation is achieved.
A
v0(t) = c m( t )
4
The detection of SSB modulated waves is
based on the assumption that there is
perfect synchronization in between local
carrier and that in the transmitter both in
frequency and phase.
: 243 :
But in practice a phase error may arise in

the locally-generated carrier wave. Thus
the detector output is modified due to
phase error.
(c)
Sol: Multiplexing
Multiplexing
is
the
process
of
simultaneously transmitting two or more
individual
signals
over
a
single
communication
channel.
Due
to
multiplexing, it is possible to increase the
number of communication channels so that
more information can be transmitted. The
typical applications of multiplexing are in
telemetry,
telephony
satellite
communication.
The concept of multiplexing is shown in
fig. The multiplexer receives a large
number of different input signals.
Multiplexer has only one output which is
connected to the single communication
channel. The multiplexer combines all
input signals into a single composite signal
and transmits it over the communication
medium. Sometimes, the composite signal
is used to modulate a carrier before
transmission. At the receiving end of
communication link, a demultiplexer is
used to sort out the signals into their
original form.
Multiple
input
signals
MUX
DEMUX
Original
input
signals
Single
communication
channel
: 244 :
S.No.
TDM
FDM
1.
It is a technique for
transmitting
several
messages
on
one
channel by dividing
time domain slots. One
slot for each message
It is a technique to
transmit
several
messages on one
channel,
message
signals are distributed
in frequency spectrum
such that they do not
overlap.
2.
It
requires
commutator
at
the
transmitting end and a
decommutator at the
receiving end
FDM
requires
modulator, filters and
demodulators.
Perfect synchronization
between transmitter and
receiver is required.
Synchronization
between transmitter
and receiver is not
required.
4.
Crosstalk problem is
not severe in TDM.
FDM suffers from

crosstalk problem due
to imperfect band
pass filter.
5.
It is usually preferred
for
digital
signal
transmission
It is usually preferred
for analog signal
transmission.
It does not require very

Complex circuitry.
It requires complex
circuitry at transmitter
and receiver.
6.
05.
Sol: Given fs = 555 kHz
fL = 1010 kHz
fL > fs IF = fL - fs
= 1010 555 = 455 kHz
fsi = Image frequency
fsi = fs + 2 IF = 555 + 2 455 = 1465 kHz
f
f
1465 555
= 2.26
= si s =
f s f si
555 1465
IRR = 1 2 Q 2
= 1 (2.26) (40) = 90.43
2
fs = tuned frequency of circuit = 1200 kHz

fs =1.2MHz
fsi = fs + 2 IF = 1200 + 2 450 = 2.1 MHz
f
f
= si s
f s f si
2 .1 1 .2
1.178
=
15 .2.1
IRR = 1 2 Q 2
IRR = 1 (65) 2 (1.178) 2
IRR = 76.61
(b) fs = 20 MHz
=
f si f s
f s f si
fsi = fs + 2 IF
= 20 MHz + 2 0.45
fsi = 20.9 MHz
20.9 20
= 0.088
20 20.9
IRR = 5.81
=
07.
Sol:
Given IF = 450 kHz
fs = 15 MHz
image frequency fsi = fs + 2 IF
fsi = 15MHz + 2450 kHz = 15.9 MHz
f
f
si s
f s f si
15.9 15
0.1166
15 15.9
Image rejection ratio IRR = 1 2 Q 2
Q
06.
Sol:
IRR 1
2
Q = 1029.12
(a) IF = Intermediate frequency = 450 kHz
Q = 65
: 245 :
08.
Sol: Given IF1 2MHz
6. Baseband Data Transmission
IF2 = 0.2 MHz

fs = 30MHz, Q = 75
fsi = fs +2IF1 +2IF2
= 30 + 2 2 + 2 0.2
= 34.4MHz
f
f
si s
f s f si
34.4 30
0.2745
30 34.4
Image rejection ratio IRR = 1 2 Q 2
IRR 1 75 2 0.2745 2
IRR = 20.67
5. Random Variables and Noise

01.
Sol: The auto correlation function is given as
Rxx () = exp (2/22);
Fourier transform of autocorrelation function is
Power Spectral Density.
Using properties of Fourier transform
e at
2
/ 2 2
e
a
f
a
2 2
e
1
2 2
2f 2
1
2 2
; a 2
2
2
2 2
Pulse Code Modulation

01. (i) Refer IES 15 E & T Conventional
Paper II Q.01, page: 337
(ii) Bandwidth of message signalW= 4 kHz
SNR(dB) = 20 dB
Signal power Ps = 30 mW
2
Noise power (PN) =
12
3.8 3.8
2n
Let n = no. of bits
SNR(dB) = 10 log 10 SNR = 20
SNR = 100
P
SNR s
PN
30 10 3
100
PN
PN = 30 105
2
12PN
12
= 18. 97 mV
3 .8 2
18.97 mV
2n
2n = 400.63 = 8.643
n9
PN
22 e 2 f
2 2 2
PSD= 2 2 e 2 f
normalized average power = Pavg = Rxx(0)= 1W.
2 2 2
02. Refer Theory Booklet.

02.
Sol: No. of quantization levels
L = 128
n = log2 L = log 2 128
n=7
sampling frequency fs = 8 kHz
: 246 :
no. of telephone channels = N=24

BW = nNfs = 24 8 7 = 1.344 MHz
Time Division Multiplexing
03.
(a), (b) Refer Study material & Previous
conventional paper II.
01. Refer Study material p. No. 76
04 . Refer IES 15 E & T Conventional Paper

II Q.06, page: 344
02 . (i) Refer IES 15 E & T Conventional

(ii) Refer IES 15 E & T Conventional
06.
Sol: BW of message signal W = 100 Hz
Am = 10 =>Dynamic range = Am (Am)
= 10 (10)
= 20 V
(a) Minimum Sampling rate required
fs= fN= 2W = 2100 = 200 samples/sec
04.
Sol: The signals are band limited to 5 kHz, 10
kHz and 5 kHz and hence the Nyquist rates
are 10 kHz, 20 kHz, and 10 kHz
respectively. Thus, there are 10 + 20 +10
= 40 thousand samples per second.
0 .1
(b) quantization error (Qe) =

Am ----(I)
100
Am Am ----------(II)
Qe =
=
2
2 2n
From (I) & (II)
0 .1
2 Am
Am
100
2 2n
2n = 1000
n = 10
(c) minimum bit rate
Rb = nfs = 10200 = 2 kbps
(d) Minimum absolute channel BW = Rb/2
= 2/2 = 1 kHz
II Q.03, page: 342
II Q.04, page: 344
II Q.01, page: 337
Number of quantization levels L = 256

Hence, number of bits n is given by
2n = 256
Giving n = 8
Hence output bit rate = 8 40 = 320 kbps
(a) Maximum bit duration
1
10 3 s 3.125 s
320
(b) The channel bandwidth
1
= 10 20 10 8 160kHz
2
(c) There are 10000+20000+10000 = 40000
samples per second generated by the
commutator. Hence the samples generated
in one minute is 60 40000 = 24 105
The commutator takes 4 samples in one
rotation.
Hence, the commutator speed
24 10 5
=
6 10 5 rpm
4
(d) For M = 512, N comes out be equal to 9
(as 2N = M )
: 247 :
Channel bandwidth
1
= 10 20 10 9 180kHz
2
The actual value of
Ps
is given by
PN
Ps
3 R
3 64
b 192
PN 64 W
64 4
P
P
Hence, s dB 10 log10 s
PN
PN
= 10 log 10 192 = 22.83 dB

II Q.08, page: 346
II Q.02, page: 334
Which is approximately 1 dB more than

the worst-case value.
Delta Modulation
7. Bandpass Data Transmission
01. (i) Refer study material

(ii) Refer study material
II Q.02, page: 339
02. Refer Theory Booklet.
8. Noise in Analog Communication
03. (i), (ii) Refer Study material

II Q.11, page: 347
06.
Sol: Given, bit rate Rb = 64 kbps
Signal bandwidth W = 4 kHz
Hence, signal to quantization noise ratio
Ps
is
PN
3
Ps
3 f
3 R
b b
PN 80 f M 80 W
3 64
=153.6
80 4
P
P
Hence, s dB 10 log10 s
PN
PN
= 10 log10 153.6 = 21.86 dB
P
This value of s is worst case value.
PN
02.
Sol: Resistance of antenna Ra = 50
Equivelent noise resistance
Req = 30
R eq
30
f 1
1
Ra
50
f = 1.6
f (dB) = 2.04 dB
= 10 log 1.6
f (dB) = 2.04 dB
ter = To (f 1)
But To = 290 = 17oC
07.
Sol:
Given that BW = 500 106 Hz

F = 2.5 dB
10log10 F = 2.5
F = 1.778
: 248 :
Noise temperature Tn = T(F 1)

= 290 (1.7781) = 225.62 K
Internal noise power Pn = KTnB
Pn = 1.38 1023 225.62 500 106
Pn = 1.557 1012 = 1.557 pW
08.
Sol: Given: F = 4 dB
10 log10 F = 4
F = 2.5112, B = 500 kHz
Output SNR = 1
input SNR
S / Ni
F
i
output SNR So / N o
(SNR)i/p = F (SNR) o/p = F
Since, (SNR)i/p
= Si/Ni, input signal power Si is given as
Si = F Ni
Ni = KTB
= 1.38 1023 290 500 103
Therefore, Si = F Ni
= 2.512 1.38 1023 290 500103
Si = 5.026 1015 Volts
Si = 5.026 1015 Volts
As the quantizer is designed to be

equiprobable, the area under each region
should be equal and the total area under
p(x) si 1.
i.e.
px dx 1
px dx 1 3A 1
i.e.
1
1
2
X2a a
4
3
3
03.
Sol: Quantization power
pn
2
2
x px dx x px dx
a
x3
1
x dx
4
12 a
a
a
2a 3 a 3
23
4
12
6 3 6 81
Matched Filter
Modulation
9. Noise in digital Communication

01.
Sol:
Pulse Code Modulation
1
3
p(t)
3
01. Refer IES 15 E & T Conventional Paper
II Q.13, page: 349
T =1
0
p(x)
02.
Sol:
Region
1
y(t) = o/p = [h(t) * p(t)]
1/8
h(t) = I.R = p (T t)
1/4
Region
2
1 a
p(t)
Region
3
+a 1
+3
p(t)
: 249 :
h(t) = p (T t)
02.
Sol:
g(t)
1
p(t) * h(t)
0
p(t)*h(t)
h(t) = g(T t)
g(t)
1 0
1
pt 3r t 3r t 1 3u t 1
pt 3 r t r t 1 u t 1 ---(1)
h t 3u t 3r t 3r t 1
h(t) = g(T t) h(t)
h t 3 u t r t r t 1 2
1 e s e s
Ps 3 2 2
s
s
s
1 1 e s
Hs 3 2 2
s
s s
y(t) = g(t) * h(t)
P(s) H(s) = X(s)
the maximum value of the signal at t = T is

equal to the energy of the signal
1 1 es es es e2s es es e2s
3 3 4 4 3 4 4 2 3 3
s
s
s
s
s
s
s s s
t; 0 t 1
g t
1; 1 t 2
1 1 2e s e s e 2s e 2s
3 3 4 4 2 3 4
s
s
s
s
s
s
1
1
1
3
3 t 2ut t 3ut t 1 ut 1 t 1ut 1
6
3
2
1
1
2
3
t 2 ut 2 t 2 ut 3
2
6
1 1 3 1
y1 3 y
2 6 2 2
y(1) = 1
E(g(t)) =
| gt | dt
2
t 2 dt 1 dt
2
t3
4
E 1 J
3
3 0
Eg t
4
3
The maximum value of the output is
M = 4/3 and it occurs at t = T =2
: 250 :
According to optimum decision rule:

Threshold value
0 = 0
Probability of Error
01.
Sol: The pdfs of received symbol are given as
(b)
f(z/s)
f(z/s2)
1.9 0.1 0 0.1
pe
f(z/s1)
1/2
1/2
1.9
1 s1 1 s 2
p p
2 s 2 2 s1
s
p 1 probability that s2 is sent and s1 is
s2
received
s
p 2 probability that s1 is sent and s2 is
s1
received
s
1
p 2 0.1
2
s1
s
p 2
s1
1
0.1
2
1
1
1 1
p e 0 .1 0 .1
2
2
2 2
1
1
0.1 2 0.05
2
2
The optimum threshold is considered to be 0
02.
Sol: a1 = 1
a2 = 1
(a) The symbols are given to be
equiprobable
The decision threshold
a a
1 1
0 = 1 2 =
0
2
2
a a2
Pe = Q 1
2
Given Variance
2 = 0.1
= 0.3162
1 1
Pe = Q
2 0.3162
Pe = Q 3.162
03.
Ed
Sol: Pe = Q
2 No
Tb
S t S t dt
2
Ed =
Tb
A. A dt
2
= 4TbA2
4T A 2
b
Pe = Q
2 N 0
Pe 10 6
For x = 4.75
4Tb A 2
4.75
2N 0
4A 2 Tb
2
4.75
2N 0
A2 = 4.693104
A = 21.66 mV
A2 A2
Signal power =
= 0.4693 mW
2
: 251 :
04.
2E p
Sol: s1(t) = A cos ct
N0
s2(t) = A cosc t
A = 20 mV;
Rb = 1Mbps
Tb
1
1
R s 1mbps
2E p
N0
200 10 12
2 10 11
2 10 10 1011
10
2
Pe Q 10
Pe Q4.472
Tb = 1 106 s
E p E q 2E pq
Pe Q
2N0
A 2 Tb
A2
Ep
Tb ; E q
2
2
Pe= 3.9 104
10. Information Theory and coding

II Q.07, page: 291
Tb
E pq pt qt dt
0
A 2 Tb
E p
2

II Q.09, page: 292
E p E p 2E p
Pe Q
2N0
4E p
Pe Q
2 N 0

II Q.10, page: 293
II Q.22, page: 302
N0
10 11
2
N0 = 2 1011

II Q.09, page: 292
II Q.12, page: 295
2 Ep
Pe Q
11
2 10
10 . Refer study material
A 2 Tb
But E p
2
20 10 1 10
3 2
Ep
400 10 12
2
11.
Sol: Standard telephone channel BW = 3kHz
C = B log2 (1+S/N)
S
10 log10 32dB
N
S
10 3.2
N
S
1584.90
N
: 252 :
P
Also H T
I m
C = 3000 log2 1584.90

C = 9600 Hz
C = 9.6 kHz
But since m =1, Im = [1]

13. Refer IES 15 E & T Conventional Paper
II Q.21, page: 201
1
1

T
And H

1
1

11. Error Correcting Codes

02.
Sol:
1 0 0 0 1
1
0 1 0 0 1
1
(a) G ; P

0
0
0
1
1

Ik
P
note that m = 1
03.
Sol:
Data word
0 0 0
0 0 1
0 1 0
0 1 1
1 0 0
1 0 1
1 1 0
1 1 1
(b)
Data word
0
0
0
0
1
1
1
1
0
0
1
1
0
0
1
1
0
1
0
1
0
1
0
1
Code word
0
0
0
0
1
1
1
1
0
0
1
1
0
0
1
1
0
1
0
1
0
1
0
1
0
1
1
0
1
0
0
1
(c) This is a parity check code. If a single error

occurs anywhere in the code word, the
parity is violated. Therefore this code can
detect a single error.
(d) cHT = 0
Now r = c e
And rH T c e H T cH T eH T eH T
0
1
1
0
0
1
1
0
Code word
0 0 0 0
1 0 0 0
1 1 0 1
0 1 0 1
1 1 1 0
0 1 1 0
0 0 1 1
1 0 1 1
0
1
0
1
1
0
1
0
From this code we see that the distance

between any two code words is at least 3.
Hence dmin = 3
04.
Sol:
Data word
0 0 0
0 0 1
0 1 0
0 1 1
1 0 0
1 0 1
1 1 0
1 1 1
0
0
0
0
1
1
1
1
Code word
0 0 0 0
0 1 1 1
1 0 1 0
1 1 0 1
0 0 0 1
0 1 1 0
1 0 1 1
1 1 0 0
0
0
1
1
1
1
0
0
Observe that dmin = 3
If there is no error e = 0 and

rH T eH T 0
: 253 :
(c) the minimum distance between any two code

words is 3. Hence, this is a single error
correcting code. Since there are 6 single
errors and 7 syndromes, we can correct all
single errors and one double error.
05. (a)
Sol:
1
1
1 1 1
1
1 1 0
T
& H
1 0 1
1

0
P
1 0 0
0 1 0
G
0 0 1
Ik
1
1
0
0
1
0
1
0
1
0
0
(d)
E
1
0
0
0
0
0
1
(b)
Data word
0 0 0
0 0 1
0 1 0
0 1 1
1 0 0
1 0 1
1 1 0
1 1 1
Code word
0 0 0 0
0 1 1 0
1 0 1 1
1 1 0 1
0 0 1 1
0 1 0 1
1 0 0 0
1 1 1 0
0
0
0
0
1
1
1
1
0
1
0
1
1
0
1
0
0
1
0
0
0
0
0
0
0
1
0
0
0
0
0
0
0
1
0
0
1
0
0
0
0
1
0
0
0
0
0
0
0
1
0
1
1
1
1
0
0
0
S
1
1
0
0
1
0
1
1
0
1
0
0
1
1
s = eHT
s = eHT
(e)
r
1
0
1
0
0
0
1
0
1
s
1
1
0
0
1
1
0
0
0
1 1 0
1 1 0
0 0 0
e
0
0
0
1
1
0
0
0
0
06.
Sol:
Data word
0
0
0
1
1
0
1
1
0
0
1
1
Code word
0 0 0
1 1 1
0 1 0
1 0 1
c
0
0
0
0
0
0
0
0
0
1
1
1
(b) H T
0
0
1
0
1
1
1
0
1
0
0
1
0
0
1
0
1
1
1
0
0
1
0
d
1
1
0
0
1
1
0
0
0
1 1 1
0 1 0
1 0 1
0
1
0
T
and s = eH
0
0
The minimum distance between any two

code words is dmin = 4. Therefore, it can
correct all 1-error patterns. Since the code
over satisfies hamming bound it can also
correct some 2- error and possibly some
3-error patterns.
: 254 :
6 single- error patterns
7double- error patterns
2 triple- error patterns
1
0
0
0
0
0
1
1
1
1
1
0
0
0
0
0
1
0
0
0
0
1
0
0
0
0
1
1
0
0
0
0
1
0
0
0
0
1
0
0
0
1
0
0
1
s
0
0
0
1
0
0
0
0
1
0
0
0
0
1
1
0
0
0
0
1
0
0
0
0
1
0
0
1
1
0
0
0
0
0
0
1
0
0
0
0
1
0
0
1
1
1
1
1
0
0
0
0
0
1
1
1
0
1
0
1
1
0
0
1
0
0
1
1
0
1
1
0
0
1
1
1
1
0
0
1
0
0
1
1
0
1
1
0
1
0
0
1
0
0
0
1
1
0
0
0
1
1
1
1
1
The code word is 11111111
07.
Sol:
Systematic (7, 4) cyclic code
g(x) = x3 + x + 1
For data 1111d(x) = x3 + x2 + x +1
For data 1110 d(x) = x3 +x2 +x

x3 x2
x3 x 1 x6 x5 x 4
x4 x3
x6
x3 x3 x 2 x 1 x6 x5 x 4 x3
x3 x 2 1
x5
x3
x5
x3 x2
x2
x3 x 1 x6 x5 x 4 x3
x4 x3
x6
The code word is 1110100
x5
A similar procedure is used to find the
x5 x3 x2
remaining codes
x3 x2
x3 x 1
x 2 x 1
c(x) = (x3 + x+ 1) (x3 + x+ 1)
(b) From table 1

It can be seen that the minimum distance
between any two codes is 3. Hence this is a
single error correcting code.
= x6 x5 x 4 x3 x 2 x 1
: 255 :
d
1
1
1
1
1
1
1
1
0
0
0
0
0
0
0
0
1
1
1
1
0
0
0
0
1
1
1
1
0
0
0
0
1
1
0
0
1
1
0
0
1
1
0
0
1
1
0
0
1
0
1
0
1
0
1
0
1
0
1
0
1
0
1
0
1
1
1
1
1
1
1
1
0
0
0
0
0
0
0
0
1
1
1
1
0
0
0
0
1
1
1
1
0
0
0
0
1
1
0
0
1
1
0
0
1
1
0
0
1
1
0
0
c
1
0
1
0
1
0
1
0
1
0
1
0
1
0
1
0
1
1
0
0
0
0
1
1
0
0
1
1
1
1
0
0
1
0
0
1
0
1
1
0
1
0
0
1
0
1
1
0
1
0
1
0
0
1
0
1
0
1
0
1
1
0
1
0
Table: 1
(c) There are seven possible non-zero syndromes
For e = 1 0 0 0 0 0 0
x3 x 1
x3 x 1 x6
x6 x4 x3
x3 x2 x
x 1
x3
x2
s = 101
The remaining syndromes are shown in table 2.
0
1
0
0
0
0
0
0
0
1
0
0
0
0
0
0
0
0
1
0
0
0
0
0
0
0
1
0
0
0
0
0
0
0
1
Table :2
x3 x 1 x6 x5
x3 x2
x4 x3
x6
x5 x4
x2
x3 x2
x5
x4 x3
x2 x
x4
x3 x2 x
x 1
x3
x2
s(x) = x +1
s=101
from table 2
e=1000000
c=re=11011001000000
=0101100
Hence d = 0 1 0 1
x2 x
x4
1
0
0
0
0
0
0
(d) The received data is 1101100

r(x) = x6 +x5 +x3 +x2
12. Optical Fiber Communication
x4 x3
E
0
0
0
1
0
0
0
1
1
1
0
1
0
0
S
0
1
1
1
0
1
0
1
1
0
1
0
0
1
01.
03
Sol: Given data :
n1 = refractive index = 1.5
= relative refractive index = 3.0 %
n 2 n 22
= 1
2n 1
n 22 = n 12 - 2 n 12 = 1.52 [1 - 2 0.03]
= 2.115
3n 12
Rc =
4(n 12 n 22 ) 3 / 2
: 256 :
0.8 i nch
3 1 .5 2 0 .8
4(1.5 2 2.115) 3 / 2
Rc = 8.67 in
06.
Sol:
Rc =
04.
Sol:
Refer IES - 15 Previous Conventional paper II
solution book, Page number: 360, Question
number: 17
05
Sol: Advantages of single mode fibers over multi
mode fibers are:
The single mode fiber has the distinct
advantage of low intermodal dispersion
(broadening of transmitted light pulses), as
only one mode is transmitted where as with
multimode step index fiber considerable
dispersion may occur due to the differing
group velocities of the propagating modes.
This in turn restricts the maximum
bandwidth attainable with multimode step
index fibers, especially when compared with
single mode fibers.
For
lower
bandwidth
applications,
multimode fibers have several advantages
over single mode fiber mode fibers.
(a) Lower tolerance requirements on fiber
connectors
(b) The numerical apertures and the core
diameters being large, thus facilating
easier coupling to optical sources.
(c) Use of spatially incoherent optical
sources which cannot be effeciently
coupled to single mode fibers.

07.
08.
Sol:

09
Refer IES - 15 Previous Conventional paper
II solution book, Page number: 354,
Question number: 04, 05
10.
Sol:
Refer IES - 15 Previous Conventional paper
II solution book, Page number: 355,
Question number: 06
11.
12.
355,356, Question number: 08
13.
: 257 :
14.
15.
16.
17.
18.
19.
354, Question number: 05 and Page
number 359, Question number: 16
20.
Sol: NA = 0.20
2an 1 (2 )1 / 2
d
V=
=
NA
m = number of modes = 1000

V2
m=
2
2
V = 2000
V = 2000
d 0.2
850 nm
d = 60499.82 nm
d 60.50 m
V=
V2
2
d
V=
NA
At 1320 nm
60.50 10 6 0.2
V=
= 28.79
1.320 10 6
V2
m=
= 415
2
(b) m =
(c) At 1550 nm
d
V=
NA
60.50 10 6
=
0 .2
1.550 10 6
= 24.52
V2
m=
= 300
2
21
Sol: (a) = 820 nm
d = 2a = 2 25 = 50 m
n 1 = 1.48
n 2 1.46
NA = n 12 n 22 = 1.48 2 1.46 2 = 0.242
2d
V=
NA
50 10 6
=
0.242 = 46.35
820 10 9
(b) m =
V2
= 1075
2
: 258 :
50 10 6
d
0.242 = 28.79
NA =
1320 10 9
V2
m=
= 415
2
50 10 6
d
(d) V =
0.242 = 24.52
NA =
1550 10 9
V2
m=
= 300
2
(c) V =
22
Sol: n 1 = 1.48
= 0.01
Radius a = 25 m
V = normalized frequency
d
=
n 1 ( 2 )1 / 2
= 1320nm
50 10 6
V=
1.48(2 0.01)1 / 2
9
1320 10
= 24.91
V 2 24.912
m=
=
= 310.18
2
2
23
Sol: a = radius = 5 m
= 0.002
n 1 = 1.480
(a) V = normalized frequency
For 0 V 2.406 the fiber is simple mode
fiber
(b)
10 10 6
V=
1.48(2 0.002)1 / 2 = 3.586
9
820 10
As V > 2.406 the fiber is not single mode.
24
Sol: d = 62.5 m
=2;
NA = 0.275
= 850nm
V=
(b) m =
d
62.5
NA =
0.275 = 63.52
850
V2
2
63.52 2
=
=1008
2 2
22
2
m = 1008
25
Sol:
n 1 = 1.48; =2
= 0.010
= 1310nm.
V = normalized frequency
=
d
n 1 ( 2 )1 / 2
2.406 =
d 1.48
(2 0.010)1 / 2
1310 10 9
= 24.91
d = 4793 nm
: 259 :
13. Satellite Communication
21.
Sol:
(a). Path loss= 20 log ( 4 R / ) = 20 log ( 4
02. Refer IES 15 E & T conventional paper _

II Q.01, page: 362
37,500 10 / 0.04918) dB
LP= 199.6 dB

(b). Uplink power budget gives
II Q.06, page: 364
Pr = Pt + Gt + Gr - Lp dBW
= 20 + 54 + 26 199.6 = -99.6 dBW
II Q.09, page: 364,365

(c). N = k Ts BN
= -228.6 + 27 + 75.6 = -126.0 dBW
II Q.11, page: 365

(d). C/N = Pr N = -99.6 + 126.0 = 26.4 dB
(e). The gain of the transponder is 110 dB.
Output power is
II Q.16, page: 367

II Q.19, page: 369
II Q.22, page: 371,372
Pt
= Pr + G = -99.6 + 110
1.04
= 10.4 dBW or 10
= 11.0 W.
22.
Sol:
(a). G = 33,000 / ( 6 x 3)
= 1833 or 32.6 dB on axis.

II Q.24, page: 372,373
II Q.25, page: 373
Hence satellite antenna gain towards earth

station is 32.6 3 = 29.6 dB.
(b) Path loss = 20 log ( 4 R / )
= 20 log ( 4 39,000 10 / 0.07742) dB
II Q.27, page: 374

: 260 :
Lp = 196.0 dB Downlink power budget gives
(c) Margin is receiver

permitted C/N, in dB
= 10 + 29.6 + 53 196.0 = -103.4 dBW
C/N
minimum
Margin = 9.8 6.0 = 3.8 dB.

24.
Sol: Answer: Wavelengths are: 1.6 GHz, =
0.1875 m; 1.5 GHz, = 0.200 m
Path loss = 20 log ( 4 R / )
(c). N = k Ts BN
= -228.6 + 20 + 75.6 = -133.0 dBW
For 1.6 GHz, Maximum Lp = 20 log ( 4

3
2,500 10 / 0.1875) = 64.5 dB

(d). C/N = Pr N = -103.4 + 133.0 = 29.6 dB
23.
Sol:
(a). In FDMA, the output power of the transmitter
is divided equally between the channels. For
Pt = 20 W and 500 channels, power per
channels is 20 / 500 = 40 mW/ch.
(b). Each channel receiver has a noise bandwidth
of 50 kHz or 47 dBHz. Path loss at 11GHz is
For 1.5 GHz Lp
= 20 log ( 4 38,500 10 / 0.200)

= 187.7 dB
25.
Sol:
(a) Flux density is given by
2
2
F = 20 log [ Pt Gt / (4 R ) ] dBW/m
Hence for R = 38,500 km,
f = 4 GHz, = 0.075 m
206.0 dB, including atmospheric loss.

10 log Pt + Gt - 10 log (4 ) - 20
3
2
log (38,500 10 ) dBW / m
=
10.0 +30.0 - 11.0 - 151.7
2
= -122.7 dBW / m
F =
Downlink power budget for one FDMA
channel gives Pr = Pt + Gt + Gr - Lp dBW
= -14.0 + 30.0 + 40.0 206.0 = -150.0 dBW
(b) Received power can be calculated from the

The noise power at the input to the receiver is
effective area of the antenna aperture and the
N = k Ts BN = -228.6 + 21.8 + 47.0 = -159.8
incident flux density, but since the antenna
dBW Hence C/N = Pr N

= -150.0 + 159.8 = 9.8 dB.
gain is given in dB, it is better to use path
loss and the link budget.
: 261 :
Path loss Lp = 20 log (4 R / )

3
= 10 log (4 38,500 10 / 0.075)

= 196.2 dB Downlink power budget gives
27.
Sol:
(a) EIRP = Pt + Gt = Pt + 52 dBW
Flux density is given by
2
2
F = 20 log [ EIRP / (4 R ) ] dBW/m
Hence for R = 39,000 km,
f = 14.3 GHz, = 0.02010 m
= 10.0 + 30.0 + 39.0 196.2

= -117.2 dBW Alternatively, the
received power can be found from
F =
-90.0 = EIRP - 10 log (4 ) - 20

3
log (39,000 10 ) dBW / m - 90.0

Pr = F Aeff where Aeff is the effective
aperture area of the antenna. Given G = 4
2
Aeff / = 39 dB, we can find Aeff from

Aeff = G + 20 log - 11.0 dB
= 39.0 22.5 11.0 = 5.5 dB m
(c) Transponder EIRP
= Pt + Gt = 10 + 30
= 40 dBW
26.
Sol:
(a) Path loss Lp = 20 log (4 R / ) = 10 log (4
2000 10
EIRP = -90.0 + 162.8 = 72.8 dBW

(b) EIRP = Pt + Gt = 72.8 dBW.
Pr = -122.7 + 5.5 = - 117.2 dBW / m
= EIRP - 11.0 - 151.8 dBW / m
/ 0.120) = 166.4 dB
Downlink power budget gives

Pr = Pt + Gt + Gr - Lp - losses dBW
= -3.0 + 18.0 + 1.0 166.4 3.0
= -153.4 dBW
(b) The noise power at the input to the receiver is
N = k Ts BN = -228.6 + 24.1 + 43.0
= -161.5 dBW
Hence Pt = 72.8 52.0 = 20.8 dBW.

28.
Sol: System noise temperature is calculated from
Ts = Tantenna + TLNA + Tmixer / GLNA + .
Hence for GLNA = 40 dB
= 10,000 as a ratio
Ts = 50 + 100 + 1000 / 10,000
= 150.1 K
29.
Sol:
(a) Flux density is given by
2
2
F = 20 log [ EIRP / (4 R ) ] dBW/m
Hence for R = 38,000 km, f = 4.0 GHz,
= 0.0750 m, EIRP = 13.0 + 30.0
= 43.0 dBW
F = 43.0 - 10 log (4 ) - 20 log
3
(38,000 10 ) dBW / m
= 43.0 - 11.0 - 151.6 = -119.6 dBW / m
(c) C/N = Pr N = -153.4 + 161.5 = 8.1 dB.

: 262 :
(b) The effective area of the antenna is

(c) At f = 20 GHz, = 0.0150 m:
Aeff = A r = 0.65 1
= 2.042 m or 3.1 dBm
For an incident flux density of -119.6 dBW

2
-12
2
/ m or 1.10 10 W/m
Pr
= 2.042 1.10 10
= 2.24 10
-12
-12
W or 116.5 dBW
or Pr = -119.6 + 3.1 = -116.5 dBW

(c). Antenna gain for a circular aperture is given
2
by G = A ( D / )
2
G = 10 log ( 0.65 ( 2 / 0.0750) )

= 36.6 dB
(d). At a frequency of 4.0 GHz, = 0.075 m.
3
= 20 log ( 4 38,000 10 / 0.075) dB
Antenna gain for a circular aperture is given

2
by G = A ( D / )
G = 10 log ( 0.65 ( 2 / 0.0150) )

= 50.6 dB
(d) At a frequency of 20.0 GHz, = 0.015 m.
3
= 20 log ( 4 38,000 10 / 0.015) dB

Lp = 210.1 dB
Lp = 196.1 dB
= 13 + 30.0 + 36.6 196.1 = -116.5 dBW

Pr
= Pt + Gt + Gr - Lp dBW
= 13 + 30.0 + 50.6 210.1
30.
Sol:
(a) Flux density is independent of frequency, so
the result is the same as in Problem 10
above.
F = -119.6 dBW / m
= -116.5 dBW
The answer is the same as in Question 10 for
the 4 GHz satellite because the incident flux
density on the earths surface is the same,
(b) Given a flux density at the earths surface,
and the antenna effective area is the same.
an antenna of area Aeff m collects the same

power at any frequency. Hence the result is
the same as in question 10 above.
Pr = 2.042 1.10 10
= 2.24 10
-12
-12
W or 116.5 dBW

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Communication Systems

Where is the modulation index

Postal Coaching Solutions

Given modulation index = 1

Thus at = 1 the power in sidebands is 1/6th

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m(t) represents message signal

If 80% modulation is used, then the ratio of

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Postal Coaching Solutions

Thus this method is very useful only for

= am(t) + aAccos2fct + bm2(t)

Thus for single tone AM (with modulation

The above equation represents an AM signal

If Am is greater than Ac then distortion is

fm2fm fc-fm fc fc+fm 2fc

Let the carrier signal c(t) = Ac cos ct

Condition for square law modulation fc > 3fm

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The equation for the amplitude in the AM

(ii) The instantaneous voltage of AM wave is

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(iii) Derivation of modulation index (m)

Modulation index = 75%

The equation for the amplitude in the AM

E C 1 sin m t sin 2 m t sin 3 m t sin 4 m t

where modulation index K a E m

e E C 1 sin m t sin 2 m t sin 3 m t sin 4 m t sin c t

Modulating signal e m E m sin m t

4fm +fc 3fm +fc 2fm +fc

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The value of composite modulation index is t 12 22 32 24

cos c t cos(i t ) sin(i t ) sin c t

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cos t cos( t ) sin ( t ) sin t

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[cos t cos( t ) sin t sin( t )]

V(t) = VLSB(t) + VUSB(t)

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s(t) = Ac cos (2fc t) m(t)

If the locally generated carrier is not in

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SSB & VSB

A (cos c t cos( i t ) sin c t sin i t )

X(t) =A1 (cos(2f c t ) cos(2f i t )

From the block diagram

+2aAKm(t)cosct +bm2(t) +bA2cos2ct

sin( 2f c t ) sin( 2f i t )) ] [cos 2f Lo t ]

(cos 2(f c f i ) t (cos 2f Lo t )

LPF allows only low frequency components

y1 ( t ) AB (cos 2(f c f i ) t )(cos 2(f Lo t )

[cos(2(fc fi fLo )t) cos(2(fc fi fLo )t

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m(t) c(t) = (cos 2 1500 t + cos 2 3000 t )

= cos 2 1500 t + cos 2 3000 t

= sin 2 (1500 t ) + sin 2 (3000 t )