Introduction To Communi. Sys - Module
Introduction To Communi. Sys - Module
Email: [email protected]
February, 2023
Analog signals
Examples of analog signals are electrical signals, light signals, speech signals, etc. Every
signal requires a medium to propagation. For example, Electrical signals require cables to
propagate from one place to another
Digital signals are the signal that represents the data in the form of discrete values. It takes
only two values 0 and 1, which is known as bits. The data is transmitted in the form of these
bits. For example, 01000110 is a digital signal.
Digital signal is a non-continuous electrical signal, which is used to convey (send, receive,
and process) information between the sender and receiver. In digital signal, the original
information (analog information) is converted into a string of bits (digital information) before
being transmitted.
For example, computers are digital in nature. They send, receive, store and process
information in binary form, i.e. in the combination of 0s and 1s.
An analog or digital signal which repeats itself after a specific interval of time is called
periodic signal. They are deterministic signals.
Information Source:
The information generated by the source may be in the form of sound (human speech),
picture (image source) or plain text.
Information is obtained from real life signals through the use of transducers. For example,
speech is converted into a corresponding electrical signal by a microphone and moving
picture signals are converted into the appropriate electrical signals by various cameras. The
information so obtained is called a signal that becomes a function of time which is usually
analog in nature. Signals may be described in time domain or in frequency domain. The
frequency domain description of a signal is known as spectrum that would be covered
subsequently. Data generated by the keystroke of a computer become the information when
communication is made through e-mail.
Transmitter:
• The transmitter is a device which converts the signal produced by the source into a form
that is suitable for transmission over a given channel or medium.
• Transmitters use a technique called modulation to convert the electrical signal into a
form that is suitable for transmission over a given channel or medium.
• Converts electrical signal into a form suitable for transmission through the channel
(physical medium)
• The transducer output signal cannot, in most cases, be transmitted directly (does not
match the channel)
• Transmitter converts message to a suitable form and onversion is made through
modulation
• Amplitude (AM), frequency (FM) and phase (PM) are examples of this.
•Based on the technique used for the signal transmission, we can categorize the
electronic communication system as under :
➢ Baseband transmission system
➢ Bandpass transmission system using modulation
Baseband transmission:
In baseband transmission systems, the baseband signals (original information signals) are
directly transmitted.
• The baseband transmission cannot be used with certain mediums e.g., it cannot be
used for signal transmission where the medium is free space. This is because the voice
signal cannot travel long distance in air. It gets suppressed after a short distance .
Bandpass transmission system using modulation:
• In the modulation process, some parameter of the carrier wave (such as amplitude,
frequency or phase ) is varied in accordance with the modulating signal .
Unguided signals can travel from the source to destination in several ways
Depending on the frequency, radio waves and micro waves travel in space in different
ways depending on the behavior of these waves with respect to the earth and the
atmosphere. They are:
1. Ground wave propagation
2. Sky ( ionospheric) wave propagation
3. Space (or tropospheric) wave propagation
Sky waves are the AM radio waves which are received after being reflected from
ionosphere. The propagation of radio wave signals from one point to another via
reflection from ionosphere is known as sky wave propagation.
The sky wave propagation is a consequence of the total internal reflection of radiowaves.
Higher we go in the ionosphere, free electron density increases and refractive index
decreases.
The UV and high energy radiations from the Sun are absorbed by the air molecules and
they get ionised to form the ionised layer or electrons and ions. Ionosphere extends from
80 km to 300 km in the atmosphere above the earth’s surface.
If the maximum electron density of the ionosphere is Nmax per m3, then the critical
frequency is given by:
𝟏
𝒇𝒄 = 𝟗(𝑵𝒎𝒂𝒙 ) 𝟐
Short wave radios( SW) are good example and the frequency range are from 2 to 30 MHz
A space wave travels in a straight line from transmitting antenna to the receiving antenna.
They propagate very much like electromagnetic waves in free space. In space wave
propagation there is a concept of radio horizon.
The radio horizon for space waves is about four-thirds as far as the optical horizon.This
beneficial effect is caused by the varying density of the atmosphere, and because of
diffraction around the curvature of the earth.
3. Telephones
One-to-one verbal communication is transmitted over the vast worldwide telephone
networks employing wire, fiber optics, radio, and satellites. It includes cordless
telephones, cellular phones, VoIP phones and satellite phones.
S. Short
Name of the frequency range (Band) Frequency Range
No. Form
1 Very Low Frequency VLF 3 kHz to 30 kHz
j t
X ( ) x (t ) e dt
( )
X ( ) X ( ) e X
X ( ) X ( ) eX ( )
1
X ( )e
j t
x (t ) d
2
0 | t | / 2
t
rect 0 . 5 | t | / 2
1 | t | / 2
t 0 | t | / 2
1 2 t / | t | / 2
Interpolation Function
sin t
sinc ( t )
t sinc(t)
sinc ( t ) 0 for t k
sinc ( t ) 1 for t 0
Find the FT, the magnitude, and the phase spectrum of 𝑥(𝑡) = 𝑟𝑒𝑐𝑡(𝑡/).
Answer
/2
rect
j t
X ( ) (t / )e dt sinc( / 2 )
/2
j t
X ( ) (t )e
dt 1
• Linearity:
Let and
then
• Time Scaling:
Compression in the time domain results in
Let x t X expansion in the frequency domain
1
then x at X
a a
Let x t X
j
then x (t )e 0 t
X ( 0 )
• Multiplication by a Sinusoid (Amplitude Modulation):
Let x t X
1
then x t cos 0 t X 0 X 0
2
Here 𝑐𝑜𝑠𝜔 𝑡 is the carrier, 𝑥(𝑡) is the modulating signal (message), 𝑥(𝑡) cos 𝜔0 𝑡 is the
modulated signal.
• Differentiation in the Frequency Domain:
Let xt X
n
then d
t n
x (t ) ( j ) n
X ( )
d n
Let x t X
d n
then x (t ) ( j ) n
X ( )
d t n
Integration in the Time Domain:
Let x t X
t
Then 1
x ( ) d
j
X ( ) X ( 0 ) ( )
x t X
Let
y t Y
Then
x (t ) y (t ) X ( )Y ( )
1
x1 (t ) x2 (t ) X 1 ( ) X 2 ( ) Frequency convolution
2
• Parseval’s Theorem: since x(t) is non-periodic and has FT X(), then it is
an energy signals:
1
x t X 2 d
2
E dt
2
Example
Find the energy of signal x(t) = e-at u(t) Determine the frequency so that
the energy contributed by the spectrum components of all frequencies below
is 95% of the signal energy EX.
• Duality ( Similarity) :
• Let x t X
then X (t ) 2 x ( )
Y(f)=H(f)×X(f)
H( f ) is generally referred as Frequency Response or Transfer Function of the LTI system.
• For distortionless transmission, amplitude response |H(f)| must be a constant and phase
response θh(f) must be linear function.
E
| x ( t ) | 2 dt
x * ( t ) X ( f ) e j 2 ft df dt
Reversing the order of integration, we obtain
E X ( f ) x * (t )e j 2ft dt df
*
X ( f ) x (t )e j 2ft dt df X ( f ) X * ( f ) df
| X ( f ) |2 df
Thus E | x(t) |2 dt | X ( f ) |2 df
G( f ) | X ( f ) |2
By integrating G(f) over all frequency, we obtain the total energy.
The energy and power of a signal represent the energy or power delivered by the signal
when it is interpreted as a voltage or current source feeding a 1-ohm resistor.
2
The energy content of x(t):
Ex x (t ) dt
The power content of x(t):
1 T /2 2
P x lim
T T T /2
x (t ) dt
Low pass signal - the spectrum (frequency content) of the signal is located around the
zero frequency.
Band pass signal - the spectrum is located around a frequency fc far from the zero
frequency.
The bandwidth of the band pass signal is usually much less than the frequency fc (recall that
he bandwidth of a signal is the set of the range of all positive frequencies present in the
signal).
The bandwidth of the band pass signal is usually much less than the frequency fc (recall that
he bandwidth of a signal is the set of the range of all positive frequencies present in the
signal).
The extreme case of a bandpass signal is a single frequency signal whose frequency is
equal to fc (The bandwidth of this signal is zero)
x ( t ) A cos( 2 f c t )
A cos( ) cos( 2 f c t ) A sin( ) sin( 2 f c t )
x c cos( 2 f c t ) x s sin( 2 f c t )
1) Message signal
The signal which contains a message to be transmitted to the destination is called a message
signal. The message signal is also known as a modulating signal or baseband signal.
The original frequency range of a transmission signal is called baseband signal. The message
signal or baseband signal undergoes a process called modulation before it gets transmitted
over the communication channel. Hence, the message signal is also known as the modulating
signal.
The high energy or high frequency signal which has characteristics like amplitude, frequency,
and phase but contains no information is called a carrier signal. It is also simply referred to as
a carrier. Carrier signal is used to carry the message signal from transmitter to receiver. The
carrier signal is also sometimes referred to as an empty signal.
3) Modulated signal
When the message signal is mixed with the carrier signal, a new signal is produced. This new
signal is known as a modulated signal. The modulated signal is the combination of the carrier
signal and modulating signal.
Amplitude modulation is the process of changing the amplitude of a relatively high frequency
carrier signal in proportion with the instantaneous value of modulating signal (information).
It is also a process of translating information signal from low band frequency to high band
frequency.
Signals from information sources are usually of low frequency, are called baseband signals
and baseband signals are not suitable for transmission. A technique called modulation is used
to turn a baseband message signal in to another form which is suitable for transmission. In
other words, modulation translates a base-band message signal to a band-pass signal.
The process of impressing low frequency information signals onto a high frequency carrier
signal is called modulation.
Demodulation is the reverse process where the received signal is transformed to their
original form. Band-pass signals have relatively higher frequency and are suitable for
transmission. In the process of modulation, the message signal is known as modulating
signal.
In any modulation technique, a high frequency Sin or Cosine wave is used as a carrier
signal. High quality oscillators are used to generate carrier signal.
Based on the attribute used, modulation can be classified as follows:
Amplitude Modulation (AM):
If the amplitude of the carrier signal is chosen to carry the message signal
during modulation
Frequency Modulation (FM):
If the frequency is used to carry the message signal.
Phase Modulation (PM):
If the phase is used to carry message signal.
Benefits of Modulation:-
1. Easy radiation
Antennas operate effectively when antenna size is comparable to the
wavelength. (for half-wave dipole antenna: 𝑳 = 1 2 𝝀 )
2. Frequency matching
Modulation shifts the spectral of a message signal so as to fit the frequency
band of the channel
3. Multiplexing
Accommodation for simultaneous transmission of several baseband signals
4. Less vulnerability to noise or interference
Modulation provides a mechanism for putting the information content of a
message signal into a form that may be less vulnerable to noise or interference
5. Narrow banding
For an audio signal range of 50 to 104 Hz, the ratio of the highest audio
frequency to the lowest is 200
An antenna suitable for use at one end of the range would be entirely too short or too long
for the other end
If the audio spectrum were translated from (106 + 50) to (106 + 104), then the
ratio would be only 1.01
In the modulation process, the voice, video, or digital signal modifies another signal called
the carrier. In amplitude modulation (AM) the information signal varies the amplitude of
the carrier sine wave. The instantaneous value of the carrier amplitude changes in accordance
with the amplitude and frequency variations of the modulating signal .An imaginary line
called the envelope connects the positive and negative peaks of the carrier waveform.
The AM wave is the algebraic sum of the carrier and upper and lower sideband sine waves.
Figure 2-1: Amplitude modulation. The carrier signal and message signal
In Amplitude Modulation (AM), it is particularly important that the peak value of the
modulating signal be less than the peak value of the carrier. Vm < Vc
Distortion occurs when the amplitude of the modulating signal is greater than the amplitude
of the carrier. A modulator is a circuit used to produce AM. Amplitude modulators compute
the product of the carrier and modulating signals.
where v2 is the instantaneous value of the AM wave (or vAM), Vc sin 2wfct is the carrier
waveform, and (Vm sin 2wfmt) (sin 2wfct) is the carrier waveform multiplied by the mod-
ulating signal waveform. It is the second part of the expression that is characteristic of AM.
A circuit must be able to produce mathematical multiplication of the carrier and modulating
signals in order for AM to occur. The AM wave is the product of the carrier and modulating
signals
The upper sideband fUSB and lower sideband fLSB are computed as
fUSB = fc + fm and fLSB = fc — fm
where fc is the carrier frequency and fm is the modulating frequency
The modulation index (m) is a value that describes the relationship between the amplitude of
the modulating signal and the amplitude of the carrier signal. This index is also known as the
modulating factor or coefficient, or the degree of modulation. It is a measures of the depth
of the modulation
Em
m
Ec
Percentage modulation (% m) is simply the modulation index (m) stated as a percentage.
More specifically percent modulation gives the percentage change in the amplitude of the
output wave when the carrier is acted on by a modulating signal.
𝐸
𝑚(𝑖𝑛 %) = 𝑥 100 %
𝐸
m = modulation index
Em = peak change in the amplitude output waveform (sum of voltages from upper and
lower side frequencies) and Ec = peak amplitude of the unmodulated carrier
The modulation index is also known as the modulation depth. The perfect-modulation has a
modulation depth of 100%. In perfect-modulation, the carrier level falls to zero. Perfect-
modulation causes no distortion.
Over-Modulation:
Over-modulation occurs when the maximum amplitude of the message signal or modulating
signal is greater than the maximum amplitude of the carrier signal (Am > Ac).
The modulation index is the ratio of the maximum amplitude of the message signal to the
maximum amplitude of carrier signal. For example, if the message signal maximum
amplitude is 6 volts and carrier signal maximum amplitude is 4 volts, then the ratio of
modulating signal amplitude (6 volts) to the carrier signal amplitude (4 volts) is equal to 1.5.
Therefore, the modulation index in over-modulation is greater than one (Mi > 1).
Over-modulation causes severe distortion of the waveform of the message signal which
results in data loss. Over-modulation is one of the reasons why amplitude modulation is no
longer used to transmit high-quality sound signals. At the transmitter, limiters are included
which prevent more than 100% modulation.
If the modulating signal is a pure, single-frequency sine wave and the process is
symmetrical then the modulation index can be derived as follows:
1
Em (V max V min )
2
1
Ec (V max V min )
2
1
(Vmax Vmin )
(V Vmin )
m 2 max
1
(Vmax Vmin ) (Vmax Vmin )
2
1
(V V min )
Em max
E usf E lsf 2
2 2
1
(V max V min )
4
Eusf = peak amplitude of the upperside frequency (volts)
Elsf = peak amplitude of the lower side frequency (volts)
From the modulated wave displayed in the previous equation, the maximum and
minimum values of the envelope occurs at
+Vmax = Ec + Eusb + Elsb
+Vmin = Ec – Eusb – Elsb
-Vmax = -Ec - Eusb - Elsb
-Vmin = -Ec + Eusb + Elsb
For proper AM operation, Ec > Em means that 0≤ m ≤ 1.
If Ec < Em means that m > 1 leads to severe distortion of the modulate wave.
If Vc = Vm the percentage of modulation index goes to 100%, means the maximum
information signal is transmitted. In this case, Vmax = 2Vc and Vmin = 0.
In a certain AM radio transmitter, an audio signal 𝟏𝟎𝑺𝒊n(𝟐𝝅 ×𝟓𝟎𝟎 t ) is used to modulate a
carrier wave 𝟓𝟎𝑺𝒊𝒏 𝟐𝝅 × 𝟏𝟎𝟓𝒕 .Calculate the modulation index.
Solution:
Representing both the modulating signal Vm(t) and the carrier signal Vc(t) in
trigonometric functions.
The AM DSBFC modulator must be able to produce mathematical multiplication of these
two analog signals
v m (t ) V m sin ( 2f m t )
v c (t ) Vc sin ( 2f c t )
v am ( t ) [V c mV c sin ( 2 f m t )] sin ( 2 f c t )
[1 m sin ( 2 f m t )] V c sin ( 2 f c t )
2
Pc = carrier power (watts)
(V c / 2 ) 2 V c
Pc Vc = peak carrier voltage (volts)
R 2R
R = load resistance i.e antenna (ohms)
The upper and lower sideband powers will be
2
( mVc / 2) 2 m 2Vc where mVc/2 is the peak voltage of usf and lsf.
Pus b Plsb
2R 8R
Rearranging in terms of Pc,
m2 Vc 2 m2
Pus b Plsb Pc
4 2R 4
m2 m2
Pt Pc Pc Pc
4 4
m2 m2
Pc Pc Pc [1 ]
2 2
m2 m2 m2 m2
Pt Pc Pusb Plsb Pc Pc Pc Pc Pc Pc 1
4 4 2 2
Efficiency, E is defined as the percentage of total power that conveys information i.e it is the
percentage of total transmitted power that is in the sidebands.
Example
For an AM DSBFC wave with a peak unmodulated carrier voltage Vc = 10Vp, frequency
of 100kHz, a load resistor of RL = 10 , frequency of modulating signal of 10kHz and m
= 1, determine the following
i) Powers of the carrier and the upper and lower sidebands.
ii) Total power of the modulated wave.
iii) Bandwidth of the transmitted wave.
iv) Draw the power and frequency spectrum.
2
(Vc / 2 ) 2 Vc (10) 2
Pc 5W
R 2 R 2 10
m 2 Pc
Pusb Plsb 1.25W
4
AM Current Calculations
Modulation index can be calculated by measuring the current of the carrier and the
modulated wave.
The measurement is simply by metering the transmit antenna current with and without the
presence of the modulating signal.
The relationship between the carrier current and the current of the modulated wave is
Pt It 2 R It 2 m2
1
Pc Ic 2 R Ic 2 2
It m2
1
Ic 2
m2
It Ic 1
2
Where Pt = total transmit power (watts)
Pc = carrier power (watts)
It = total transmit current (ampere)
Ic = carrier current (ampere)
R = antenna resistance (ohm)
Example
In DSB-SC, as the name implies, the carrier is removed (suppressed) from AM signal
spectrum. Only two sidebands are available for transmission and this is achieved by using
product modulator, also known as balanced modulator. Balanced modulator simply
multiplies the message signal with carrier signal.
Let, the modulating (message) signal be:
Then, using product modulator, the DSB-SC signal can be developed as:-
Thus, the bandwidth of DSBSC wave is same as that of AM wave and it is equal to twice
the frequency of the modulating signal.
Now, let us add these two sideband powers in order to get the power of DSBSC wave.
Therefore, the power required for transmitting DSBSC wave is equal to the power of both
the sidebands
Single-sideband (SSB) is a form of AM where the carrier is suppressed and one sideband
is eliminated. The process of suppressing one of the sidebands along with the carrier and
transmitting a single sideband is called as Single Sideband Suppressed Carrier system
or simply SSBSC.
The carrier is transmitted at full power but only one sideband is transmitted
– requires half the bandwidth of DSBFC AM
– Carrier power constitutes 80% of total transmitted power, while sideband
power consumes 20%
– SSBFC requires less total power but utilizes a smaller percentage of the power
to carry the information
This SSBSC system, which transmits a single sideband has high power, as the power
allotted for both the carrier and the other sideband is utilized in transmitting this Single
Sideband.
These are on different frequencies, so they are affected in slightly different ways by the
ionosphere and upper atmosphere, which have a great influence on radio signals of less than
about 50 MHz. The carrier and sidebands may arrive at the receiver at slightly different
times, causing a phase shift that can, in turn, cause them to add in such a way as to cancel
one another rather than add up to the original AM signal. Such cancellation, or selective
fading, is not a problem with SSB since only one sideband is being transmitted.
An SSB signal has some unusual characteristics. First, when no information or modulating
signal is present, no RF signal is transmitted. In a standard AM transmitter, the carrier is still
transmitted even though it may not be modulated. This is the condition that might occur
during a voice pause on an AM broadcast. But since there is no carrier transmitted in an SSB
system, no signals are present if the information signal is zero. Sidebands are generated only
during the modulation process, e.g., when someone speaks into a microphone. This explains
why SSB is so much more efficient than AM.
The main disadvantage of DSB and SSB signals is that they are harder to recover, or
demodulate, at the receiver. Demodulation depends upon the carrier being present. If the
carrier is not present, then it must be regenerated at the receiver and reinserted into the
signal. To faithfully recover the intelligence signal, the reinserted carrier must have the same
phase and frequency as those of the original carrier. This is a difficult requirement. When
SSB is used for voice transmission, the reinserted carrier can be made variable in frequency
so that it can be adjusted manually while listening to recover an intelligible signal. This is
not possible with some kinds of data signals
• VSB is similar to SSB but it retains a small portion (a vestige) of the undesired sideband
to reduce DC distortion.
• VSB signals are generated using standard AM or DSBSC modulation, then passing
modulated signal through a sideband shaping filter.
• Demodulation uses either standard AM or DSBSC demodulation.
• Also called asymmetric sideband system.
• Compromise between DSB & SSB and Easy to generate.
• Bandwidth is only ~ 25% greater than SSB signals.
• Derived by filtering DSB, one pass band is passed almost completely while just a trace
• AM wave is applied to a vestigial sideband filter, producing a modulation scheme – VSB
+C
• Mainly used for television video transmission and VSB Frequency Spectrum
Angle modulation is the process by which the angle (frequency or phase) of the carrier
signal is changed in accordance with the instantaneous amplitude of modulating or
message signal.
It results whenever the phase angle θ of a sinusoidal wave is varied with respect to time
An angle modulation results whenever the phase angle, θ of a sinusoidal wave is varied
with respect to time and can be expressed as
m ( t ) V c cos ct (t ) (1)
Then from the previous 4 terms, (3) ~ (5), PM and FM can be defined as :
• PM : an angle modulation in which θ(t) is proportional to the amplitude of the
modulating signal.
• FM : an angle modulation in which θ’(t) is proportional to the amplitude of the
modulating signal.
For a modulating signal vm(t),
θ(t) = Kvm(t) rad (6)
θ’(t) = K1vm(t) rad/s (7)
where K and K1 are constants and are the deviation sensitivities of the phase and
frequency modulators, respectively.
Then substituting a modulating signal vm(t) = Vmcos(ωmt), equation (7) and (8) into
equation (1) yields
PM :
m (t ) V c cos ct (t )
V c cos c t KV m cos( m t ) (9)
FM : as (t ) ' (t )
m (t ) Vc cos ct ' (t )
Vc cos ct K 1 Vm cos(mt ) dt
K 1Vm
V c cos c t sin( m t ) ( 10)
m
m KV m (radians ) (12)
where m = modulation index
K = deviation sensitivity (radians/volt)
Vm = peak modulating signal amplitude (volt)
Vc cosct m cos(mt )
Modulation Index and Percent Modulation for FM
The modulation index is the maximum value of phase deviation for both PM and FM and
is dimensionless (unitless).
For FM, the modulation index is directly proportional to the amplitude of the modulating
signal and inversely proportional to the frequency of the modulating signal.
(14)
K 1V m K 1V m
m (unitless )
m fm
Where K1 = deviation sensitivities (radians/second per volt or cycles/second per vol
Vm = peak modulating signal amplitude (volt)
ωm = radian frequency (radians/second)
fm = cyclic frequency (cycles/second or hertz)
Also for FM, the peak frequency deviation Δf is simply the product of the deviation
sensitivity and the peak modulating signal voltage. I.e.
f
f K 1Vm m (unitless) (15)
fm
Therefore, for FM, equation (10) can be rewritten as
K 1Vm
m (t ) Vc cos ct sin( wmt )
fm
f
V c cos ct sin( w mt )
fm
• Percent modulation for angle modulation is determined in different manner than for
amplitude modulation.
• with angle modulation, percent modulation is the ratio of frequency deviation actually
produced to the maximum frequency deviation allowed, stated in percent form
f ( actual )
Percent modulation 100 % ( 17)
f (max)
• frequency analysis of the angle-modulated wave is much more complex compared to the
amplitude modulation analysis.
• in phase/frequency modulator, a modulating signal produces an infinite number of side
frequencies pairs (i.e. it has infinite bandwidth), where each side frequency is displaced
from the carrier by an integral multiple of the modulating frequency.
Bessel Function
where Jn(m) is the Bessel function of the first kind. Then applying equation (18) to equation
(11) yields,
n
m (t ) Vc Jn ( m ) cos(ct n mt )
n 2 (19)
expanding equation (19),
m ( t ) V c J 0 ( m ) cos( c t ) J 1 ( m ) cos ( c m )t
2
J 1 ( m ) cos ( c m ) t J 2 ( m ) cos ( c 2 m ) t
2
Jn ( m )
Curves for the relative amplitudes of the carrier and several sets of side frequencies for
values of m up to 10.
BWFM 2( f fm ) 2( 1) f m Hz (20)
Depending on the value of the modulation index 𝛽; frequency modulation (FM) wave is
classified as (i) Narrowband FM (NBFM) and (ii) Wideband FM (WBFM)
(a) In narrow band FM, the modulation index 𝛽 is small as compared to one radian and the
bandwidth of the FM wave is small.
(b) In Wide band FM, the bandwidth is much larger and value of 𝛽 is very high. For larger
values of 𝛽, the FM wave ideally contains carrier and an infinite number of sidebands
located symmetrically around the carrier. Hence the BW approaches to infinity and
hence it is called wideband FM.
Carson’s Rule
o It is a general rule to estimate the bandwidth for all angle-modulated systems
regardless of the modulation index.
o The Carson’s Rule states that the bandwidth necessary to transmit an angle-
modulated wave as twice the sum of the peak frequency deviation and the highest
modulating signal frequency.
o Carson’s Rule approximate and gives a narrower bandwidth than the bandwidth
determined using Bessel function. Therefore, a system designed using Carson’s
Rule would have a narrower bandwidth but a poorer performance than system
designed using the Bessel table.
o for modulation index above 5, Carson’s Rule is a close approximation to the actual
bandwidth required.
o For FM, the bandwidth varies with both deviation and modulating frequency.
o Increasing modulating frequency reduces modulation index so it reduces the number
of sidebands with significant amplitude.
o On the other hand, increasing modulating frequency increases the frequency
separation between sidebands.
o Bandwidth increases with modulation frequency but is not directly proportional to it.
⚫ As seen in Bessel function table, it shows that as the sideband relative amplitude
increases, the carrier amplitude,J0 decreases.
⚫ This is because, in FM, the total transmitted power is always constant and the total
average power is equal to the unmodulated carrier power, that is the amplitude of the
FM remains constant whether or not it is modulated.
⚫ In effect, in FM, the total power that is originally in the carrier is redistributed
between all components of the spectrum, in an amount determined by the modulation
index, mf, and the corresponding Bessel functions.
⚫ At certain value of modulation index, the carrier component goes to zero, where in
this condition, the power is carried by the sidebands only.
⚫ The power of a sinusoidal signal depends only on the square of the amplitude and not
on the frequency. The amplitude of an FM wave is constant and therefore the total
power of an FM signal is independent of the modulation index. This contrasts with
the AM where the power of the modulated signal is a function of the modulation
index.
Vc 2
The average power in undulated carrier P ave
2
An angle-modulated signal is described by the equation:
𝑠(𝑡) = 5𝑐𝑜𝑠(2𝜋 𝑥 10 𝑡 + 20𝑠𝑖𝑛1000 𝜋𝑡 + 10𝑠𝑖𝑛 2000𝜋𝑡)
Determine:
a. the power of the angle-modulated signal
b. the frequency deviation, ∆𝑓
c. The phase deviation,∆𝜙
d. The bandwidth of the angle-modulated signal using Carson’s rule.
PM modulator : (t ) v (t )
FM modulator : (t ) v(t )
Considering the FM modulator, if the modulating signal is v(t) is differentiate before being
applied to the FM modulator, the instantaneous phase is now proportional to the modulating
signal (i.e. PM modulator).
dv (t )
Differentiator + FM modulator = (t ) (t ) v (t ) = PM modulator
dt
Meanwhile, if the modulating signal is integrated before being applied to the PM modulator,
the instantaneous phase is now proportional to the integral of the modulating signal (i.e. FM
modulator).
The most common circuits used for FM signal demodulation are slope detector, balanced
slope detector and PLL demodulator.
The slope detector and balanced slope detector are categorized as tuned-circuit frequency
discriminator.
Advantages of Angle Modulation
Noise immunity – most noise results in unwanted amplitude variations in the modulated
wave (i.e. AM noise). FM and PM receivers include limiters that remove most of the Am
noise from the received signal before the final demodulation process occurs – a process
that cannot be used with AM receivers because the information is also contained in
amplitude variations, and removing the noise would also remove the information.
Noise performance and S/N improvement – with the use of limiters, FM and PM actually
reduce the noise level and improve the S/N ratio during the demodulation process.
Capture effect - with FM and PM, a phenomenon of capture effect allows a receiver to
differentiate between two signals received with the same frequency by capturing the
stronger signal and eliminate the weaker one. With AM, both signals will be demodulated
and produce audio signals.
Vn
(t ) n cos( n t n ) rad / s
Vc
Therefore, the unwanted peak frequency deviation is
Vn Vn
peak nrad / s f peak f n Hz
Vc Vc
Sampling theorem is based on the fixed sampling rate, called Nyquist rate. Hence, sampling
theorem is also known as Nyquist theorem. It is based on the theory of the bandlimited
signals. Let's discuss the sampling theorem of the bandpass signals and baseband signals.
According to the sampling theory of the bandpass signals, a signal can be successfully
reconstructed if its sampling rate is not greater than the maximum frequency W. The samples
are spaced at sampling time 'Ts' seconds apart without zero mean square error.
𝑇𝑠 = 1/2W
According to the sampling theory of the baseband signals, a signal can be successfully
reconstructed if the samples are separated with a uniform intervals less than or equal to
1/2Fm.
𝑇𝑠 ≤ 1/2𝐹𝑚
Sampling rate
Sampling rate is defined as the number of samples taken per second from a continuous signal
for a finite set of values. We can also define it as a sampling frequency, which is the
reciprocal of the sampling time.
𝐹𝑠 = 1/𝑇𝑠
As discussed, sampling rate is an essential period for the sampler to perform sampling
process. It helps in the successful recovery of the digital signal at the receiving end. Hence, a
fixed parameter was defined for the sampling rate, known as Nyquist rate.
Nyquist rate
Suppose H is the highest selected frequency. A bandlimited signal is transmitted at the
frequency components lower than W Hz. Thus, for the replication of the original signal, the
sampling rate should be twice the highest frequency.
It is given by:
𝐹𝑠 = 2𝑊
o Ideal Sampling
o Natural Sampling
o Flat-top sampling
Ideal Sampling
Ideal sampling is also known as instantaneous sampling or impulse sampling. The sampling
process multiplies the input signal and the carrier signal, which is present in the form of train
of pulses.
The above diagram shows the waveforms of the message signal, sampling signal in the form
of train of pulses, and the sampled signal.
The above diagram shows the waveforms of the message signal, sampling signal, and the
sampled signal.
Flat-top sampling
The design and reconstruction of flat-top sampling is easy than the natural sampling process.
The pulses in the flat-top sampling method are in the flat shape at the top and are held at a
constant height. It means that the samples are flat and have constant amplitude.
Aliasing is the common effect that can arise post the sampling process. In the aliasing
process, the high frequency components in the signal override the low frequency components.
The aliasing occurs when the signal frequency exceeds half the sampling frequency (Fs/2).
𝐹𝑚 > 𝐹𝑠/2
2𝐹𝑚 > 𝐹𝑠
Similarly, the aliasing effect can be reduced when the sampling frequency exceeds twice the
signal's frequency.
It is represented as:
𝐹𝑠 > 2𝐹𝑚
The anti-aliasing filters are used to prevent the aliasing effect during the transmission
process. The cut-off frequency of such filters is equal to half the sampling rate (𝐹𝑠/2). The
function of aliasing filters is to remove and filter the high-frequency components from the
signal. It is inserted before the sampler. It is also known as low-pass anti-aliasing filter. The
signal after passing through the anti-aliasing filter is sampled at a rate higher than the Nyquist
rate. It helps in the easy recovery of the signal.
We know that the sampling process helps in the conversion of an analog signal to the digital
signal. The data transmission in the form of digital signal offers various advantages, such as
high efficiency, fast speed, low cost, low interference, low distortion, and high security.
Hence, sampling is essential to improve the quality and transmission ability of the signals
over the communication channel.
The major advantages of the sampling process are due to the conversion of the transmission
to the digital form, which has various advantages as discussed above. It converts an analog
signal to the discrete values.
o Low cost
o High accuracy
o Easy to implement
o Less time consuming
o Low signal loss
o High scope
It prevents the signal loss or any information loss by converting the incoming data to the
suitable rate for transmission. For example, if a signal contains high frequency components,
the sampling process will convert it into high rates for effective transmission. Generally, the
input signal is sampled at the frequency rate twice that the incoming signal. It is done to
preserve the full information in the signal.
Applications of sampling
Sampling describes the number of possible digital values that are used to represent a sample.
Sampling is essential because it prevents any information loss during the transmission loss. It
also increases the accuracy of the system. Sampling is used in various processes, such as
PAM, PCM, and TDM. The major applications of sampling will be discussed in detail.
Pulse Modulation
The process of transmitting signals in the form of pulses (discontinuous signals) by using
special techniques. In pulse modulation, some parameter of a pulse train is varied in
accordance with the message signal.
We may distinguish two families of pulse modulation:
1. Analog pulse modulation and
2. Digital pulse modulation.
In analog pulse modulation, a periodic pulse train is used as the carrier wave, and some
characteristic feature of each pulse (e.g., amplitude, duration, or position) is varied in a
continuous manner in accordance with the corresponding sample value of the message
signal.
In digital pulse modulation, on the other hand, the message signal is represented in a
form that is discrete in both time and amplitude, thereby permitting its transmission in digital
form as a sequence of coded pulses; this form of signal transmission has no CW counterpart.
The use of coded pulses for the transmission of analog information-bearing signals represents
a basic ingredient in the application of digital communications
In pulse modulation a periodic pulse train is used as a carrier. The following parameters of
the pulse are modified in accordance with the message signal. Signal is transmitted at discrete
intervals of time.
Pulse amplitude modulation
Pulse width modulation
Pulse position modulation
PAM is the simplest and most basic form of analog pulse modulation.In pulse-amplitude
modulation (PAM), the amplitudes of regularly spaced pulses are varied in proportion to the
corresponding sample values of a continuous message signal. In PAM: width and position
are fixed but amplitude varies.
Pulses can be of a rectangular form or some other appropriate shape.
Pulse-amplitude modulation is similar to natural sampling, where the message signal is
multiplied by a periodic train of rectangular pulses. In natural sampling the top of each
modulated rectangular pulse varies with the message signal, whereas in PAM it is maintained
flat. For minimum distortion, the sampling rate should be more than twice the signal
frequency.
Bandwidth required for transmitter of PAM signal will be equal to maximum frequency
B T f max
1
2
PAM Modulators:
The PAM modulator is a simple “Emitter Follower” circuit. The modulating signal is
applied at the input. At the base, a CLOCK signal is applied. The frequency of the clock
signal is made equal to the frequency of carrier pulse train.
When the CLOCK signal is “high”, the circuit behaves as “Emitter follower” and the output
follows the input (modulating) signal, when the CLOCK is “low”, the transistor is “cut off”
and the output is zero. In this way, at the output we get PAM signal.
Disadvantages of PAM :
It has simple transmitter and receiver designs.
It allows multiplexing, so that the sharing of the same transmission media by different
sources (or users). This is because a PAM signal only occurs in slots of time, leaving the idle
time for the transmission of other PAM signals.
It is used to carry information as well as to generate other pulse modulations.
Disadvantages of PAM :
Amplitude keeps varying so there is noise associated with it.
Due to amplitude variation peak power of receiver also varies with it.
It requires a larger transmission bandwidth (very large compare to its maximum frequency)
Interference of noise is maximum and also needed for varies transmission power.
• Modulation in which the temporal positions of the pulses are varied in accordance with
some characteristic of the information signal.
• Amplitude & width constant.
• The higher the amplitude of the sample, the farther to the right the pulse is position within
the prescribed time slot.
• The amplitude is held constant thus less noise interference and signal and noise separation
is very easy
• Due to constant pulse widths and amplitudes, transmission power for each pulse is same.
• It Require less power compare to PAM and PDM because of short duration pulses.
• As a disadvantage it require very large bandwidth compare to PAM.
PPM and PDM need a sharp rise time and fall time for pulses in order to preserve the
message information.
• Lets rise time, tr
tr« Ts
1
BT
2t r
From formula above, we know that transmission BW of PPM and PDM is higher than PAM.
Pulse duration (τ) supposed to be very small compare to the period, Ts between 2 samples
Lets max frequency of the signal, W
Fs >= 2 W
Ts =< 1/2W
T « Ts =< 1/2W
1
f max
2
Example
For PAM transmission of voice signal with W = 3kHz. Calculate BT if fs = 8 kHz and τ = 0.1
Ts
Solution 1 1 4
Ts 1.25x10 s
fs 8kHz
0.1Ts 1.25x105 s
1
2W
1
BT W
2
1
BT 40 kHz
2
For the same information as in example above, find minimum transmission BW needed for
PPM and PDM. Given tr= 1% of the width of the pulse
Solution 1
tr 1.25 x10 s
7
100
1
BT
2t r
BT 4 MHz
In PWM, the width of the carrier pulse is varied according to the instantaneous value of the
modulating signal, while the amplitude remains constant. This system is also called “Pulse
duration modulation” (PDM) or “Pulse length modulation” (PLM). PWM is more often used
for control than for communication.
Noise is less in PWM as the amplitude is kept constant. The signal and noise separation is
easy. The PWM does not require synchronization between transmitter and receiver
But as a disadvantage Large bandwidth is required for PWM communication as compared to
PAM.The transmitter should be able to handle more power (equal to the power of the
maximum width pulse).
A PCM stream is a digital representation of an analog signal, in which the magnitude of the
analog signal is sampled regularly at uniform intervals, with each sample being quantized to
the nearest value within a range of digital steps.
In PCM the sampling rate which is the number of times per second that samples are taken.
According to sampling theorem, number of Pulses per second should be twice of signal
frequency.
Let's discuss the function of a PCM system with the help of an example of an audio signal.
The audio signal is first applied to the low-pass filter, which rejects the higher range of
frequencies from the signal. The sampler performs the sampling of the left and right
channels of the audio signal based on the sampling rate of 44100 Hz or 44.1k Hz and 16/32-
bit resolution. The quantizer and encoder set the digital value based on the specified
resolution and bit rate and send it to the receiver. The digital signal passes through
the quantizer that generates the pulse according to the received positive or negative pulse.
The decoder converts the regenerated pulse back to the analog signal. Further, it sends to
the reconstruction filter, which helps in the smooth conversion of the digital signal back to
the original analog signal.
The bit depth which determines the number of possible digital values that each sample can
take.
PCM consists of three steps to digitize an analog signal:
1. Sampling
2. Quantization
3. Binary encoding
Sampling PAM:
The first step in pulse code modulation is sampling. The analog signal is sampled at equal
interval, every Ts s (sample interval). The inverse of sampling interval is sampling rate or
sampling frequency. fs= 1/Ts
The process of dividing the maximum value of the analog signal into a fixed number of levels
in order to convert the PAM into a binary code. The levels obtained are called quantization
levels. The result of PAM is a series of pulses with amplitude values between the maximum
and minimum amplitudes of the signal with real values. Quantization is a method of
assigning integer values in a specific range to sampled instances.
When a signal is quantized, an error will be introduced - the coded signal is an approximation
of the actual amplitude value.
The more zones, the smaller ∆ which results in smaller errors. But, the more zones the more
bits required to encode the samples so that higher bit rate
Thus the quantization error is the sequence eq (n) defined as the difference between the
quantized value and the actual sample value eq (n) = xq (n) - x(n)
Encoding : Encoding maps the quantized values to digital words that are n bits long. The
mapping is one-to-one so there is no distortion introduced by encoding.
The output of the quantizer is one of L possible signal levels.
If we want to use a binary transmission system, then we need to map each quantized sample
into an n bit binary word.
L 2n n log 2 L
Encoding is the process of representing each quantized sample by n bit code word. The
mapping is one-to-one so there is no distortion introduced by encoding
The spectrum of the PCM signal is not directly related to the spectrum of the input signal.
The bandwidth of (serial) binary PCM waveforms depends on the bit rate R and the
waveform pulse shape used to represent the data.
The Bit Rate R is
R=nfs
Where n is the number of bits in the PCM word (M=2n) and fs is the sampling rate.
The Minimum Bandwidth of nfs//2 is obtained only when sin(x)/x pulse is used to generate
the PCM waveform.
For PCM waveform generated by rectangular pulses, the First-null Bandwidth is:
Bpcm = R = nfs
Various forms of PCM processes are used in coding and signal processing in communication.
The following are some of the most common coding processes related to the Pulse Code
Modulation.
o LPCM
PCM converts the analog signal to the digital signal for fast and efficient transmission
by converting the analog data into binary digits 0 and 1. Linear Pulse Code
Modulation uses the linear quantization method. The data during the quantization
process is generally compressed for better transmission. But, in LPCM, the data is in
the uncompressed form. Examples include blue-ray discs, Red Book compact discs,
etc.
o DPCM
Differential Pulse Code Modulation requires fewer bits to encode the input pulse
level. It requires less bandwidth, an increased number of quantization levels, and
decreased quantization noise compared to the Pulse Code Modulation method.
o ADPCM
Adaptive Differential Pulse Code Modulation is a type of DPCM that allows the
reduction of bandwidth by varying the size of the quantization step.
o DM
Delta Modulation is a simplest type of DPCM that can convert both analog and
digital signals. It works similar to the A/D and D/A converters. It is generally used to
transmit voice signals because such signals do not require high quality at the output.
1. Analog signal can be transmitted over a high speed digital communication system.
2. Probability of occurring error will reduce by the use of appropriate coding methods.
3. PCM is used in Telkom system, digital audio recording, digitized video special
effects, digital video, voice mail.
4. PCM is also used in Radio control units as transmitter and also receiver for remote
controlled cars, boats, planes.
5. The PCM signal is more resistant to interference than normal signal.
• Pulse Code Demodulation: will be doing the same modulation process in reverse.
• Demodulation starts with decoding process
• During transmission the PCM signal will effected by the noise interference.
The following table summarizes the Pulse modulation techniques based on some basic
parameter.
Relation with
modulating Amplitude of Width of the pulse is Relative position of the
signal the pulse is proportional to pulse is proportional to
proportional to amplitude of amplitude of modulating
amplitude of modulating signal signal
modulating
signal
BW of the Depends of rise time Depends on rising time of
transmission depends on of the pulse the pulse
channel width of the
pulse
Any modulated signal has a high frequency carrier. The binary signal when ASK is
modulated, gives a zero value for LOW input and gives the carrier output for HIGH input.
In ASK, only the amplitude of the carrier signal is modified in modulation. The simplest
version is on–off keying (OOK).
Its generation and detection are easy thus facilitate simple transmitter and receiver
sections.
Disadvantages of Amplitude shift keying
ASK technique is not suitable for high bit rate data transmission.
Poor bandwidth efficiency.
Highly susceptible to noise and other external factors.
Applications of Amplitude shift keying
The frequency of the output signal will be either high or low, depending upon the input data
applied.
Frequency Shift Keying (FSK) is the digital modulation technique in which the frequency of
the carrier signal varies according to the discrete digital changes. FSK is a scheme of
frequency modulation. It is the most straightforward and efficient digital signal transmission
scheme.
The simplest form of FSK is Binary frequency shift keying (BFSK). Here, the frequency of
the carrier wave changed between discrete binary values of the modulating signal. Thus, the
frequency of the carrier shows variation according to the binary message signal.
In frequency shift keying, the carrier is modulated in such a way that high-frequency signal is
achieved for high level i.e., 1 of binary data input. Similarly, the low-frequency signal is
obtained in case of low level i.e., 0 of the message signal.
Following is the diagram for FSK modulated waveform along with its input.
It utilizes more bandwidth as compared to ASK and PSK thus is not bandwidth efficient.
Detection of the signal at the receiver is somewhat complex.
Frequency shift keying (FSK) is used in the high-frequency data transmission system. and
also extensively used in low-speed modems
Information in data communication can be in form of text, image, video, audio, etc. and
this information is represented in binary format (data). A binary digit (bit) has only two
values, 0 and 1. Text, image, audio, etc. are represented as a sequence of bits
There are the seven OSI layers. Each layer has different functions. A list of seven layers are
given below:
Application (layer 7)
Presentation (layer 6)
Session (layer 5)
Transport (layer 4)
Network (layer 3)
Data link (layer 2)
Physical (layer 1)
File transfer, access and management – allows a user to access, retrieve and
mange files in a remote computer
Mail service – email forwarding and storage
User authentication – logging to remote host
OSI-Presentation Layer
OSI-Session Layer
OSI-Transport Layer
OSI-Physical Layer
• Coordinates the functions required to carry a bit stream over a physical medium
• Defines the procedures and functions that the physical devices and interfaces have to
perform for transmission to occur
• Other functions
Representation of bits – encodes bits into electrical or optical signals
Synchronization of bits - synchronizes the sender and the receiver clocks
Transmission mode- The direction of transmission between two devices
(simplex, half- duplex, full-duplex)
o The transmission medium is used to send the signal from sender to receiver. The medium
can only have one signal at a time.
o If there are multiple signals to share one medium, then the medium must be divided in
such a way that each signal is given some portion of the available bandwidth. For
example: If there are 10 signals and bandwidth of medium is100 units, then the 10 unit is
shared by each signal.
o When multiple signals share the common medium, there is a possibility of collision.
Multiplexing concept is used to avoid such collision.
o Transmission services are very expensive
o The 'n' input lines are transmitted through a multiplexer and multiplexer combines the
signals to form a composite signal.
o The composite signal is passed through a Demultiplexer and demultiplexer separates a
signal to component signals and transfers them to their respective destinations.
In the above diagram, a single transmission medium is subdivided into several frequency
channels, and each frequency channel is given to different devices. Device 1 has a frequency
channel of range from 1 to 5
The input signals are translated into frequency bands by using modulation techniques, and
they are combined by a multiplexer to form a composite signal.
The main aim of the FDM is to subdivide the available bandwidth into different
frequency channels and allocate them to different devices.
Using the modulation technique, the input signals are transmitted into frequency bands
and then combined to form a composite signal.
The carriers which are used for modulating the signals are known as sub-carriers. They
are represented as f1,f2..fn.
FDM is mainly used in radio broadcasts and TV networks.
o Wavelength Division Multiplexing is same as FDM except that the optical signals are
transmitted through the fibre optic cable.
o WDM is used on fibre optics to increase the capacity of a single fibre.
o It is used to utilize the high data rate capability of fibre optic cable.
o It is an analog multiplexing technique.
o Optical signals from different source are combined to form a wider band of light with the
help of multiplexer.
o At the receiving end, demultiplexer separates the signals to transmit them to their
respective destinations
o It is a digital technique.
o In Frequency Division Multiplexing Technique, all signals operate at the same time with
different frequency, but in case of Time Division Multiplexing technique, all signals
operate at the same frequency with different time.
o In Time Division Multiplexing technique, the total time available in the channel is
distributed among different users. Therefore, each user is allocated with different time
interval known as a Time slot at which data is to be transmitted by the sender.
o In Time Division Multiplexing technique, data is not transmitted simultaneously rather
the data is transmitted one-by-one.
o In TDM, the signal is transmitted in the form of frames. Frames contain a cycle of time
slots in which each frame contains one or more slots dedicated to each user.
1. Modulation has a number of advantages. Which of one of the following is not correct?
a) Efficient transmission
b) Reduction in noise and interference
c) Overcomes hardware limitations
d) Requires higher power transmitter
2. For message signal 𝑚(𝑡) = 𝑐𝑜𝑠 (2𝜋𝑓 𝑡) and carrier frequency fc , which one of the
following represents a SSB signal
a) 𝑐𝑜𝑠(2𝜋𝑓 𝑡) 𝑐𝑜𝑠(2𝜋𝑓 𝑡)
b) 𝑐𝑜𝑠(2𝜋𝑓 𝑡)
c) 𝑐𝑜𝑠[2𝜋(𝑓 + 𝑓 ) 𝑡]
d) 1 + 𝑐𝑜𝑠(2𝜋𝑓 𝑡) 𝑐𝑜𝑠(2𝜋𝑓 𝑡)
20. The equation V = A sin (ωct + m sin ωmt) is the expression for
a) Amplitude modulation
b) Phase modulated signal
c) Carrier signal used for modulation
d) None of the above
21. When the modulation frequency is doubled the modulation index is halved and the
modulating index is halved and the modulation voltage remains constant. This happens
when the modulating system is
a) AM
b) PM
c) FM
d) Delta Modulation
a) i, ii and iv
b) i, ii and iii
c) i, iii and iv
d) ii, iii and iv
24. Which one of the following statement is not correct?
a) FM has an infinite number of side-bands
b) Modulation index for FM is always greater than one
c) As modulation depth increases the BW increases
d) As modulation depth increases the sideband power increases
25. Consider the following statements about frequency modulation
a. 1,2,3 and 4
b. 1 and 2 only
c. 3 only
d. 3 and 4 only
26. An angle modulated signal is described by the equation
32. What are the three steps in generating PCM in the correct sequence?
a) Sampling, quantizing and encoding
b) Encoding, sampling and quantizing
c) Sampling, encoding and quantizing
d) Quantizing, sampling and encoding
33. For a 10-bit PCM system, the signal to quantization noise ratio is 62 dB. If the number of
bits increased by 2, then how would the signal quantization noise ratio change?
a) Increase by 6 Db
b) Decrease by 6 dB
c) Increase by 12 dB
d) Decrease by 12 dB
36. The modulation technique that uses the minimum channel bandwidth and transmitted
power is
a) FM
b) DSB-SC
c) VSB
d) SSB