Analog Communication Basics
Analog Communication Basics
COMMUNICATIONS
(19EC0408)
UNIT-I
AMPLITUDE MODULATION-I
Communication can also be defined as the transfer of information from one point in space
and time to another point.
Transmitter: Couples the message into the channel using high frequency signals.
Channel: The medium used for transmission of signals
Modulation: It is the process of shifting the frequency spectrum of a signal to a
frequency range in which more efficient transmission can be achieved.
Receiver: Restores the signal to its original form.
Demodulation: It is the process of shifting the frequency spectrum back to the
original baseband frequency range and reconstructing the original form.
Modulation:
medium. The below figure shows the different kinds of analog modulation schemes that are
available
Modulation is operation performed at the transmitter to achieve efficient and reliable
information transmission.
For analog modulation, it is frequency translation method caused by changing the appropriate
quantity in a carrier signal.
•Once this information is received, the low frequency information must be removed from the
high frequency carrier. •This process is known as “Demodulation”.
Baseband signals are incompatible for direct transmission over the medium so,
modulation is used to convey (baseband) signals from one place to another.
Allows frequency translation:
o Frequency Multiplexing
o Reduce the antenna height
o Avoids mixing of signals
o Narrowbanding
Efficient transmission
Reduced noise and interference
Types of Modulation:
Analog Modulation
Amplitude modulation
Example: Double sideband with carrier (DSB-WC), Double- sideband
suppressed carrier (DSB-SC), Single sideband suppressed carrier (SSB-SC), vestigial
sideband (VSB)
Angle modulation (frequency modulation & phase modulation)
Example: Narrow band frequency modulation (NBFM), Wideband frequency
modulation (WBFM), Narrowband phase modulation (NBPM), Wideband phase
modulation (NBPM)
Pulse Modulation
Digital Modulation
The carrier amplitude varied linearly by the modulating signal which usually consists of a
range of audio frequencies. The frequency of the carrier is not affected.
Application of AM - Radio broadcasting, TV pictures (video), facsimile transmission
Frequency range for AM - 535 kHz – 1600 kHz
Bandwidth - 10 kHz
It is the process where, the amplitude of the carrier is varied proportional to that of the
message signal.
Let m (t) be the base-band signal, m (t) ←→ M (ω) and c (t) be the carrier, c(t) = A c
cos(ωct). fc is chosen such that fc >> W, where W is the maximum frequency component of
m(t). The amplitude modulated signal is given by
S(ω) = π Ac/2 (δ(ω − ωc) + δ(ω + ωc)) + kaAc/ 2 (M(ω − ωc) + M(ω + ωc))
Two basic amplitude modulation principles are discussed. They are square law modulation
and switching modulator.
When the output of a device is not directly proportional to input throughout the
operation, the device is said to be non-linear. The Input-Output relation of a non-linear device
can be expressed as
When the output is considered up to square of the input, the device is called a square law
device and the square law modulator is as shown in the figure 4
When the peak amplitude of c(t) is maintained more than that of information signal, the
operation is assumed to be dependent on only c(t) irrespective of m(t).
When c(t) is positive, v2=v1since the diode is forward biased. Similarly, when c(t) is
negative, v2=0 since diode is reverse biased. Based upon above operation, switching response
of the diode is periodic rectangular wave with an amplitude unity and is given by
The required AM signal centred at fc can be separated using band pass filter.
The lower cut off-frequency for the band pass filter should be between w and fc-w
and the upper cut-off frequency between fc+w and 2fc. The filter output is given by
the equation
Detection of AM waves
Demodulation is the process of recovering the information signal (base band) from
the incoming modulated signal at the receiver. There are two methods, they are Square law
Detector and Envelope Detector
Consider a non-linear device to which the AM signal s(t) is applied. When the level of s(t) is
very small, output can be considered upto square of the input.
The device output consists of a dc component at f =0, information signal ranging from 0-W
Hz and its second harmonics and frequency bands centered at fc and 2fc. The required
information can be separated using low pass filter with cut off frequency ranging between W
and fc-w. The filter output is given by
When the information level is very low, the noise effect increases at the receiver, hence the
system clarity is very low using square law demodulator.
Envelope Detector
It is a simple and highly effective system. This method is used in most of the commercial AM
radio receivers. An envelope detector is as shown below.
During the positive half cycles of the input signals, the diode D is forward biased and
the capacitor C charges up rapidly to the peak of the input signal. When the input signal falls
below this value, the diode becomes reverse biased and the capacitor C discharges through
the load resistor RL.
The discharge process continues until the next positive half cycle. When the input
signal becomes greater than the voltage across the capacitor, the diode conducts again and the
process is repeated.
The charge time constant (rf+Rs)C must be short compared with the carrier period,
the capacitor charges rapidly and there by follows the applied voltage up to the positive peak
when the diode is conducting.That is the charging time constant shall satisfy the condition,
Advantages of AM:
Disadvantages:
AM contains unwanted carrier component, hence it requires more transmission
power.
The transmission bandwidth is equal to twice the message bandwidth.
Radio Transmitters
There are two approaches in generating an AM signal. These are known as low and
high level modulation. They're easy to identify: A low level AM transmitter performs the
process of modulation near the beginning of the transmitter. A high level transmitter
performs the modulation step last, at the last or "final" amplifier stage in the transmitter. Each
method has advantages and disadvantages, and both are in common use.
Low-Level AM Transmitter:
There are two signal paths in the transmitter, audio frequency (AF) and radio
frequency (RF). The RF signal is created in the RF carrier oscillator. At test point A the
oscillator's output signal is present. The output of the carrier oscillator is a fairly small AC
voltage, perhaps 200 to 400 mV RMS. The oscillator is a critical stage in any transmitter. It
must produce an accurate and steady frequency. Every radio station is assigned a different
carrier frequency. The dial (or display) of a receiver displays the carrier frequency.
If the oscillator drifts off frequency, the receiver will be unable to receive the
transmitted signal without being readjusted. Worse yet, if the oscillator drifts onto the
frequency being used by another radio station, interference will occur. Two circuit techniques
are commonly used to stabilize the oscillator, buffering and voltage regulation.
The buffer amplifier has something to do with buffering or protecting the oscillator.
An oscillator is a little like an engine (with the speed of the engine being similar to the
oscillator's frequency). If the load on the engine is increased (the engine is asked to do more
work), the engine will respond by slowing down. An oscillator acts in a very similar fashion.
If the current drawn from the oscillator's output is increased or decreased, the oscillator may
speed up or slow down slightly.
Buffer amplifier is a relatively low-gain amplifier that follows the oscillator. It has a
constant input impedance (resistance). Therefore, it always draws the same amount of current
from the oscillator. This helps to prevent "pulling" of the oscillator frequency. The buffer
amplifier is needed because of what's happening "downstream" of the oscillator. Right after
this stage is the modulator. Because the modulator is a nonlinear amplifier, it may not have a
constant input resistance -- especially when information is passing into it. But since there is a
buffer amplifier between the oscillator and modulator, the oscillator sees a steady load
resistance, regardless of what the modulator stage is doing.
Voltage Regulation: An oscillator can also be pulled off frequency if its power
supply voltage isn't held constant. In most transmitters, the supply voltage to the oscillator is
regulated at a constant value. The regulated voltage value is often between 5 and 9 volts;
zener diodes and three-terminal regulator ICs are commonly used voltage regulators. Voltage
regulation is especially important when a transmitter is being powered by batteries or an
automobile's electrical system. As a battery discharges, its terminal voltage falls. The DC
supply voltage in a car can be anywhere between 12 and 16 volts, depending on engine RPM
and other electrical load conditions within the vehicle.
Modulator: The stabilized RF carrier signal feeds one input of the modulator stage.
The modulator is a variable-gain (nonlinear) amplifier. To work, it must have an RF carrier
signal and an AF information signal. In a low-level transmitter, the power levels are low in
the oscillator, buffer, and modulator stages; typically, the modulator output is around 10 mW
(700 mV RMS into 50 ohms) or less.
Antenna Coupler: The antenna coupler is usually part of the last or final RF power
amplifier, and as such, is not really a separate active stage. It performs no amplification, and
has no active devices. It performs two important jobs: Impedance matching and filtering. For
an RF power amplifier to function correctly, it must be supplied with a load resistance equal
to that for which it was designed.
The antenna coupler also acts as a low-pass filter. This filtering reduces the amplitude
of harmonic energies that may be present in the power amplifier's output. (All amplifiers
generate harmonic distortion, even "linear" ones.) For example, the transmitter may be tuned
to operate on 1000 kHz. Because of small nonlinearities in the amplifiers of the transmitter,
the transmitter will also produce harmonic energies on 2000 kHz (2nd harmonic), 3000 kHz
(3rd harmonic), and so on. Because a low-pass filter passes the fundamental frequency (1000
kHz) and rejects the harmonics, we say that harmonic attenuation has taken place.
High-Level AM Transmitter:
The high-level transmitter of Figure 9 is very similar to the low-level unit. The RF
section begins just like the low-level transmitter; there is an oscillator and buffer amplifier.
The difference in the high level transmitter is where the modulation takes place. Instead of
adding modulation immediately after buffering, this type of transmitter amplifies the
unmodulated RF carrier signal first. Thus, the signals at points A, B, and D in Figure 9 all
look like unmodulated RF carrier waves. The only difference is that they become bigger
in voltage and current as they approach test point D.
The modulation process in a high-level transmitter takes place in the last or final
power amplifier. Because of this, an additional audio amplifier section is needed. In order
to modulate an amplifier that is running at power levels of several watts (or more),
comparable power levels of information are required. Thus, an audio power amplifier is
required. The final power amplifier does double-duty in a high-level transmitter. First, it
provides power gain for the RF carrier signal, just like the RF power amplifier did in the
low-level transmitter. In addition to providing power gain, the final PA also performs the
task of modulation. The final power amplifier in a high-level transmitter usually operates
in class C, which is a highly nonlinear amplifier class.
Comparison:
Have better DC efficiency than low-level transmitters, and are very well suited
for battery operation.
Are restricted to generating AM modulation only.
UNIT-II
AMPLITUDE MODULATION-II
VSB (Vestigial Side Band) modulation: In VSB, one side band is completely
passed and just a trace or vestige of the other side band is retained. The required channel
bandwidth is therefore in excess of the message bandwidth by an amount equal to the
width of the vestigial side band. This method is suitable for the transmission of wide
band signals.
DSB-SC Time domain and Frequency domain Description:
DSBSC modulators make use of the multiplying action in which the modulating
signal multiplies the carrier wave. In this system, the carrier component is eliminated and
both upper and lower side bands are transmitted. As the carrier component is suppressed,
the power required for transmission is less than that of AM.
Consequently, the modulated signal s(t) under goes a phase reversal , whenever the
message signal m(t) crosses zero as shown below.
Fig.1. (a) DSB-SC waveform (b) DSB-SC Frequency Spectrum
Hence, except for the scaling factor 2ka, the balanced modulator output is equal
to the product of the modulating wave and the carrier.
Ring Modulator
Ring modulator is the most widely used product modulator for generating DSBSC wave
and is shown below.
The four diodes form a ring in which they all point in the same direction. The
diodes are controlled by square wave carrier c(t) of frequency fc, which is applied
longitudinally by means of two center-tapped transformers.
Assuming the diodes are ideal, when the carrier is positive, the outer diodes D1 and
D2 are forward biased where as the inner diodes D3 and D4 are reverse biased, so
that the modulator multiplies the base band signal m(t) by c(t). When the carrier is
negative, the diodes D1 and D2 are reverse biased and D3 and D4 are forward, and the
modulator multiplies the base band signal –m(t) by c(t).
Thus the ring modulator in its ideal form is a product modulator for
square wave carrier and the base band signal m(t). The square wave carrier can be
expanded using Fourier series as
From the above equation it is clear that output from the modulator consists
entirely of modulation products. If the message signal m(t) is band limited to the
frequency band − w < f < w, the output spectrum consists of side bands centered at fc.
Detection of DSB-SC waves:
Coherent Detection:
The message signal m(t) can be uniquely recovered from a DSBSC wave s(t)
by first multiplying s(t) with a locally generated sinusoidal wave and then low pass
filtering the product as shown.
From the spectrum, it is clear that the unwanted component (first term in the
expression) can be removed by the low-pass filter, provided that the cut-off frequency
of the filter is greater than W but less than 2fc-W. The filter output is given by
The demodulated signal vo(t) is therefore proportional to m(t) when the phase error ϕ
is constant.
Introduction of SSB-SC
Consider the generation of SSB modulated signal containing the upper side band
only. From a practical point of view, the most severe requirement of SSB generation
arises from the unwanted sideband, the nearest component of which is separated from the
desired side band by twice the lowest frequency component of the message signal. It
implies that, for the generation of an SSB wave to be possible, the message spectrum
must have an energy gap centered at the origin as shown in figure 7. This requirement
is naturally satisfied by voice signals, whose energy gap is about 600Hz wide.
The frequency discrimination or filter method of SSB generation consists of a
product modulator, which produces DSBSC signal and a band-pass filter to extract the
desired side band and reject the other and is shown in the figure 8.
Application of this method requires that the message signal satisfies two conditions:
1. The message signal m(t) has no low-frequency content. Example: speech, audio, music.
2. The highest frequency component W of the message signal m(t) is much less than the
carrier frequency fc.
Then, under these conditions, the desired side band will appear in a non-overlapping
interval in the spectrum in such a way that it may be selected by an appropriate filter.
In designing the band pass filter, the following requirements should be satisfied:
1.The pass band of the filter occupies the same frequency range as the spectrum of the
desired SSB modulated wave.
2. The width of the guard band of the filter, separating the pass band from the stop
band, where the unwanted sideband of the filter input lies, is twice the lowest frequency
component of the message signal.
The SSB modulated wave at the first filter output is used as the modulating wave
for the second product modulator, which produces a DSBSC modulated wave with a
spectrum that is symmetrically spaced about the second carrier frequency f2. The
frequency separation between the side bands of this DSBSC modulated wave is
effectively twice the first carrier frequency f1, thereby permitting the second filter to
remove the unwanted side band.
The Fourier transform is useful for evaluating the frequency content of an energy signal, or in
a limiting case that of a power signal. It provides mathematical basis for analyzing and
designing the frequency selective filters for the separation of signals on the basis of their
frequency content.Another method of separating the signals is based on phase selectivity,
which uses phase shifts between the appropriate signals (components)
to achieve the desired separation.
o
In case of a sinusoidal signal, the simplest phase shift of 180 is obtained by “Ideal
transformer” (polarity reversal). When the phase angles of all the components of a given
signal are shifted by 90o, the resulting function of time is called the “Hilbert transform” of the
signal.
Consider an LTI system with transfer function defined by equation 1
The device which possesses such a property is called Hilbert transformer. Whenever a
signal is applied to the Hilbert transformer, the amplitudes of all frequency components of the
input signal remain unaffected. It produces a phase shift of -90o for all positive frequencies,
while a phase shifts of 90o for all negative frequencies of the signal.
If x(t) is an input signal, then its Hilbert transformer is denoted by xˆ(t ) and shown in
the following diagram.
Now consider any input x(t) to the Hilbert transformer, which is an LTI system. Let the
impulse response of the Hilbert transformer is obtained by convolving the input x(t) and
impulse response h(t) of the system.
Properties:
The time domain description of an SSB wave s(t) in the canonical form is given
by the equation 1.
Following the same procedure, we can find the canonical representation for an SSB
wave
s(t) obtained by transmitting only the lower side band is given by
The use of a plus sign at the summing junction yields an SSB wave with
only the lower side band, whereas the use of a minus sign yields an SSB wave with only
the upper side band. This modulator circuit is called Hartley modulator.
The following Fig illustrates the spectrum of VSB modulated wave s (t) with respect to the
message m (t) (band limited)
Assume that the Lower side band is modified into the vestigial side band. The
vestige of the lower sideband compensates for the amount removed from the
upper sideband. The bandwidth required to send VSB wave is
The vestige of the Upper sideband compensates for the amount removed from
the Lower sideband. The bandwidth required to send VSB wave is B = w+fv, where fv is
the width of the vestigial side band.
Therefore, VSB has the virtue of conserving bandwidth almost as efficiently as SSB
modulation, while retaining the excellent low-frequency base band characteristics of DSBSC
and it is standard for the transmission of TV signals.
VSB modulated wave is obtained by passing DSBSC through a sideband shaping filter as
shown in fig below.
The exact design of this filter depends on the spectrum of the VSB waves. The
relation between filter transfer function H (f) and the spectrum of VSB waves is given by
Where M(f) is the spectrum of Message Signal. Now, we have to determine the
specification for the filter transfer function H(f) It can be obtained by passing s(t) to a
coherent detector and determining the necessary condition for undistorted version of the
message signal m(t). Thus, s (t) is multiplied by a locally generated sinusoidal wave cos
(2πfct) which is synchronous with the carrier wave A ccos(2πfct) in both frequency and phase,
as in fig below,
The spectrum of Vo (f) is in fig below,
Similarly, the transfer function H (f) of the filter for sending Lower sideband along with the
vestige of the Upper sideband is shown in fig below,
Time Domain Description:
is equal to c since it is a constant with respect to t, and the phase of the cosine is the
constant 0. The angle of the cosine (t) = ct +0 is a linear relationship with respect to t
(a straight line with slope of c and y–intercept of 0). However, for other sinusoidal
functions, the frequency may itself be a function of time, and therefore, we should not think
in terms of the constant frequency of the sinusoid but in terms of the INSTANTANEOUS
frequency of the sinusoid since it is not constant for all t. Consider for example the
following sinusoid
y(t) cos (t),
where (t) is a function of time. The frequency of y(t) in this case depends on the function
of (t) and may itself be a function of time. The instantaneous frequency of y(t) given above
is defined as
d (t)
(t) .
i
dt
As a checkup for this definition, we know that the instantaneous frequency of x(t) is equal to
its frequency at all times (since the instantaneous frequency for that function is constant) and
is equal to c. Clearly this satisfies the definition of the instantaneous frequency since (t) =
ct +0 and therefore i(t) = c.
If we know the instantaneous frequency of some sinusoid from – to sometime t, we can find
the angle of that sinusoid at time t using
t
Changing the angle (t) of some sinusoid is the bases for the two types of angle modulation:
Phase and Frequency modulation techniques.
In this type of modulation, the phase of the carrier signal is directly changed by the message
signal. The phase modulated signal will have the form
g PM (t ) A cos c t k p m (t ) ,
where A is a constant, c is the carrier frequency, m(t) is the message signal, and kp is a
parameter that specifies how much change in the angle occurs for every unit of change of
m(t). The phase and instantaneous frequency of this signal are
PM (t ) ct k p m (t ),
(t )
k dm (t ) k m (t ).
i c p c p
dt
So, the frequency of a PM signal is proportional to the derivative of the message signal.
This type of modulation changes the frequency of the carrier (not the phase as in PM) directly
with the message signal. The FM modulated signal is
t
g FM (t ) A cos ct k f m ( )d ,
where kf is a parameter that specifies how much change in the frequency occurs for every
unit change of m(t). The phase and instantaneous frequency of this FM are
FM (t ) ct k t
f
m ( )d ,
d t
i (t ) c k
dt
f m ( )d c k f m (t ).
PM and FM are tightly related to each other. We see from the phase and frequency
t
relations for PM and FM given above that replacing m(t) in the PM signal with
m ( )d
t
m (t )d
Phase
gFM (t)
()d
m(t) Modulator
(PM)
dm (t )
d () Frequency
dt Modulator
m(t) dt gPM(t)
(FM)
Frequency Modulation
Notice that as the information signal increases, the frequency of the carrier increases,
and as the information signal decreases, the frequency of the carrier decreases.
The frequency fi of the information signal controls the rate at which the carrier
frequency increases and decreases. As with AM, fi must be less than fc. The amplitude of the
carrier remains constant throughout this process.
When the information voltage reaches its maximum value then the change in
frequency of the carrier will have also reached its maximum deviation above the nominal
value. Similarly when the information reaches a minimum the carrier will be at its lowest
frequency below the nominal carrier frequency value. When the information signal is zero,
then no deviation of the carrier will occur.
The maximum change that can occur to the carrier from its base value f c is called the
frequency deviation, and is given the symbol fc. This sets the dynamic range (i.e. voltage
range) of the transmission. The dynamic range is the ratio of the largest and smallest
analogue information signals that can be transmitted.
Bandwidth of FM and PM Signals
The bandwidth of the different AM modulation techniques ranges from the bandwidth
of the message signal (for SSB) to twice the bandwidth of the message signal (for DSBSC
and Full AM). When FM signals were first proposed, it was thought that their bandwidth can
be reduced to an arbitrarily small value. Compared to the bandwidth of different AM
modulation techniques, this would in theory be a big advantage. It was assumed that a signal
with an instantaneous frequency that changes over of range of f Hz would have a
bandwidth of f Hz. When experiments were done, it was discovered that this was not the
case. It was discovered that the bandwidth of FM signals for a specific message signal was at
least equal to the bandwidth of the corresponding AM signal. In fact, FM signals can be
classified into two types: Narrowband and Wideband FM signals depending on the
bandwidth of each of these signals
Narrowband FM and PM
The general form of an FM signal that results when modulating a signals m(t) is
t
g FM (t ) A cos ct k f m ( )d .
k f a(t ) 1
For FM and
k p m (t )
1
For PM, where
t
a(t ) m ( )d ,
such that a change in the message signal does not results in a lot of change in the
instantaneous frequency of the FM signal.
Starting with FM, to evaluate the bandwidth of this signal, we need to expand it using a
power series expansion. So, we will define a slightly different signal
Remember that
ct kf a (t )
gˆFM (t ) A e j A cosc t k f a(t ) jA sinc t k a(t ) ,
f
so
g FM (t ) Re gˆ FM (t ) .
a (t )
Now we can expand the term e jk f
gˆ FM (t ) , which gives
in
t
j 2k 2a2 (t ) j 3k f3a3 (t ) j 4k f4a4 (t )
gˆ (t ) A e j c 1 jk a(t ) f
FM
2! 3! 4!
f
k 2a2 (t jk 3a3 (t k 4a4 (t )
) )
t t t t t
A e j c jk a(t )e j c f e j c f e j c f e j c
f
2! 3! 4!
Since kf and a(t) are real (a(t) is real because it is the integral of a real function m(t)), and
since Re{ejct} = cos(ct) and Re{ jejct} = –sin(ct), then
g FM (t ) Re gˆ FM
(t ) k 2a2 (t ) k 3a3(t ) k 4a4 (t )
A cos( t ) k a(t ) sin( t ) f
cos( t ) f
sin( t ) f
cos( t )
c
c f c c
2! 3! c
4!
The assumption we made for narrowband FM is ( k f a(t ) 1). This assumption will result in
making all the terms with powers of k a(t greater than 1 to be small compared to the first
f
)
two terms. So, the following is a reasonable approximation
for g FM (t )
BW FM (Narrowband ) BW DSBSC 2 BW m (t )
.
We will see later that when the condition (kf << 1) is not satisfied, the bandwidth of the FM
signal becomes higher that twice the bandwidth of the message signal. Similar relationships
hold for PM signals. That is
BW PM (Narrowband ) BW DSBSC 2 BW m (t )
.
The above approximations for narrowband FM and PM can be easily used to construct
modulators for both types of signals
kf<<1
t
a(t)
m(t) ()d X kf
sin(ct)
–/2
A gFM (NarrowBand)(t)
cos(ct)
Narrowband FM Modulator
kp<<1
m(t) X kp
sin(ct)
–/2
A gPM (NarrowBand)(t)
cos(ct)
Narrowband PM Modulator
m(t)
Narrowband ( . )P
FM gFM (WB) (t)
Modulator
A narrowband FM signal can be generated easily using the block diagram of the narrowband
FM modulator that was described in a previous lecture. The narrowband FM modulator
generates a narrowband FM signal using simple components such as an integrator (an
OpAmp), oscillators, multipliers, and adders. The generated narrowband FM signal can be
converted to a wideband FM signal by simply passing it through a non–linear device with
power P. Both the carrier frequency and the frequency deviation f of the narrowband signal
are increased by a factor P. Sometimes, the desired increase in the carrier frequency and the
desired increase in f are different. In this case, we increase f to the desired value and use a
frequency shifter (multiplication by a sinusoid followed by a BPF) to change the carrier
frequency to the desired value.
Time-Domain Expression
Since the FM wave is a nonlinear function of the modulating wave, the frequency
modulation is a nonlinear process. The analysis of nonlinear process is the difficult
task. In this section, we will study single-tone frequency modulation in detail to
simplify the analysis and to get thorough understanding about FM.
∆ƒ = kƒAn
is the modulation index of the FM wave. Therefore, the single-tone FM wave is
expressed by
where
þp = kpAn (5.20)
is the modulation index of the single-tone phase modulated wave.
The frequency deviation of the single-tone PM wave is
For an arbitrary message signal n(t) with bandwidth or maximum frequency W, the
bandwidth of the corresponding FM wave may be determined by Carson’s rule as
GENERATION OF FM WAVES
FM waves are normally generated by two methods: indirect method and direct method.
In this case
device and a bandpass filter. The nth order nonlinear device produces a dc component and n
number of frequency modulated waves with carrier frequencies ƒc, 2ƒc, … nƒc and frequency
deviations ∆ƒ, 2∆ƒ, … n∆ƒ, respectively. If we want an FM wave with frequency deviation
of 6∆ƒ, then we may use a 6th order nonlinear device or one 2nd order and one 3rd order
nonlinear devices in cascade followed by a bandpass filter centered at 6ƒ c. Normally, we may
require very high value of frequency deviation. This automatically increases the carrier
frequency by the same factor which may be higher than the required carrier frequency. We
may shift the carrier frequency to the desired level by using mixer which does not change the
frequency deviation.
The narrowband FM has some distortion due to the approximation made in deriving
the expression of narrowband FM from the general expression. This produces some amplitude
modulation in the narrowband FM which is removed by using a limiter in frequency
multiplier.
Direct Method of FM Generation
In this method, the instantaneous frequency ƒ(t) of the carrier signal c(t) is varied directly
with the instantaneous value of the modulating signal n(t). For this, an oscillator is used in
which any one of the reactive components (either C or L) of the resonant network of the
oscillator is varied linearly with n(t). We can use a varactor diode or a varicap as a voltage-
variable capacitor whose capacitance solely depends on the reverse-bias voltage applied
across it. To vary such capacitance linearly with n(t), we have to reverse-bias the diode by
the fixed DC voltage and operate within a small linear portion of the capacitance-voltage
characteristic curve. The unmodulated fixed capacitance C0 is linearly varied by n(t) such that
the resultant capacitance becomes
C(t) = C0 − kn(t)
The above figure shows the simplified diagram of the Hartley oscillator in
which is implemented the above discussed scheme. The frequency of oscillation for
such an oscillator is given
is the frequency sensitivity of the modulator. The Eq. (5.42) is the required expression for the
instantaneous frequency of an FM wave. In this way, we can generate an FM wave by direct
method.
Direct FM may be generated also by a device in which the inductance of the resonant
circuit is linearly varied by a modulating signal n(t); in this case the modulating signal being
the current.
The main advantage of the direct method is that it produces sufficiently high
frequency deviation, thus requiring little frequency multiplication. But, it has poor frequency
stability. A feedback scheme is used to stabilize the frequency in which the output frequency
is compared with the constant frequency generated by highly stable crystal oscillator and the
error signal is feedback to stabilize the frequency.
DEMODULATION OF FM WAVES
The process to extract the message signal from a frequency modulated wave is known
as frequency demodulation. As the information in an FM wave is contained in its
instantaneous frequency, the frequency demodulator has the task of changing frequency
variations to amplitude variations. Frequency demodulation method is generally categorized
into two types: direct method and indirect method. Under direct method category, we will
discuss about limiter discriminator method and under indirect method, phase-locked loop
(PLL) will be discussed.
Limiter Discriminator Method
In this method, extraction of n(t) from the above equation involves the three steps:
amplitude limit, discrimination, and envelope detection.
A. Amplitude Limit
B. Discrimination/ Differentiation
Here both the amplitude and frequency of this signal are modulated.
In this case, the differentiator is nothing but a circuit that converts change in
frequency into corresponding change in voltage or current as shown in Fig. 5.11. The
ideal differentiator has transfer function
H(jw) = j2nƒ
Figure : Transfer function of ideal differentiator.
slope of the tank circuit. This is not suitable for wideband FM where the peak
frequency deviation is high.
A better solution is the ratio or balanced slope detector in which two tank
circuits tuned at ƒc + ∆ƒ and ƒc− ∆ƒ are used to extend the linear portion as shown in
below figure.
Figure : Frequency response of balanced slope detector.
Another detector called Foster-seely discriminator eliminates two tank circuits but still
offer the same linear as the ratio detector.
C. Envelope Detection
The third step is to send the differentiated signal to the envelope detector to recover the
message signal.
where
t
The difference ∅2(t) − ∅1(t) = ∅e(t) constitutes the phase error. Let us assume that
the PLL is in phase lock so that the phase error is very small. Then,
Since the control voltage of the VCO is proportional to the message signal, v(t) is
the demodulated signal.
We observe that the output of the loop filter with frequency response H(ƒ) is the
desired message signal. Hence the bandwidth of H(ƒ) should be the same as the bandwidth W
of the message signal. Consequently, the noise at the output of the loop filter is also limited to
the bandwidth W. On the other hand, the output from the VCO is a wideband FM signal with
an instantaneous frequency that follows the instantaneous frequency of the received FM
signal.
In FM, the noise increases linearly with frequency. By this, the higher frequency
components of message signal are badly affected by the noise. To solve this problem, we
can use a preemphasis filter of transfer function H p(ƒ) at the transmitter to boost the higher
frequency components before modulation. Similarly, at the receiver, the deemphasis filter
of transfer function Hd(ƒ)can be used after demodulator to attenuate the higher frequency
components thereby restoring the original message signal.
The preemphasis network and its frequency response are shown in Figure 5.19
(a) and (b) respectively. Similarly, the counter part for deemphasis network is shown
in Figure 5.20.
FM Transmitter
The FM transmitter is a single transistor circuit. In the telecommunication,
the frequency modulation (FM)transfers the information by varying the frequency of carrier
wave according to the message signal. Generally, the FM transmitter uses VHF radio
frequencies of 87.5 to 108.0 MHz to transmit & receive the FM signal. This transmitter
accomplishes the most excellent range with less power. The performance and working of the
wireless audio transmitter circuit is depends on the induction coil & variable capacitor. This
article will explain about the working of the FM transmitter circuit with its applications.
The FM transmitter is a low power transmitter and it uses FM waves for transmitting
the sound, this transmitter transmits the audio signals through the carrier wave by the
difference of frequency. The carrier wave frequency is equivalent to the audio signal of the
amplitude and the FM transmitter produce VHF band of 88 to 108MHZ.Plese follow the
below link for: Know all About Power Amplifiers for FM Transmitter
Block Diagram of FM Transmitter
FM Transmitter circuit
The formation of the oscillating tank circuit can be done through the transistor of 2N3904 by
using the inductor and variable capacitor. The transistor used in this circuit is an NPN
transistor used for general purpose amplification. If the current is passed at the inductor L1
and variable capacitor then the tank circuit will oscillate at the resonant carrier frequency of
the FM modulation. The negative feedback will be the capacitor C2 to the oscillating tank
circuit.
To generate the radio frequency carrier waves the FM transmitter circuit requires an
oscillator. The tank circuit is derived from the LC circuit to store the energy for
oscillations.
The input audio signal from the mic penetrated to the base of the transistor, which modulates
the LC tank circuit carrier frequency in FM format. The variable capacitor is used to change
the resonant frequency for fine modification to the FM frequency band. The modulated signal
from the antenna is radiated as radio waves at the FM frequency band and the antenna is
nothing but copper wire of 20cm long and 24 gauge. In this circuit the length of the antenna
should be significant and here you can use the 25-27 inches long copper wire of the antenna.
Application of Fm Transmitter
The FM transmitters are used in the homes like sound systems in halls to fill the sound
with the audio source.
These are also used in the cars and fitness centers.
The correctional facilities have used in the FM transmitters to reduce the prison noise in
common areas.
Advantages of the FM Transmitters
Noise temperature
Equivalent noise temperature is not the physical temperature of amplifier, but a theoretical
construct, that is an equivalent temperature that produces that amount of noise power
𝑇𝑒 = (𝐹 − 1)
White noise
One of the very important random processes is the white noise process. Noises in
many practical situations are approximated by the white noise process. Most importantly, the
white noise plays an important role in modelling of WSS signals.
A white noise process is a random process that has constant power spectral density at
all frequencies. Thus
where is a real constant and called the intensity of the white noise. The
corresponding autocorrelation function is given by
The autocorrelation function and the PSD of a white noise process is shown in Figure 1
below.
In most communication systems, we are often dealing with band-pass filtering of signals.
Wideband noise will be shaped into band limited noise. If the bandwidth of the band limited
noise is relatively small compared to the carrier frequency, we refer to this as narrowband
noise.
where fc is the carrier frequency within the band occupied by the noise. x(t) and y(t)
are known as the quadrature components of the noise n(t). The Hibert transform of
n(t) is
Proof.
The Fourier transform of n(t) is
^ ^
Let N ( f ) be the Fourier transform of n^ ( t). In the frequency domain, N
(f) = N(f)[-j sgn(f)]. We simply multiply all positive frequency components of N(f)
by -j and all negative frequency components of N(f) by j. Thus
The quadrature components x(t) and y(t) can now be derived from equations
Noise Bandwidth
The received signal at the output of the receiver noise- limiting filter : Sum of this signal and
filtered noise .A filtered noise process can be expressed in terms of its in-phase and quadrature
components as
Demodulate the received signal by first multiplying r(t) by a locally generated sinusoid
cos(2 f t + ), where is the phase of the sinusoid.Then passing the product signal through
c
an ideal lowpass filter having a bandwidth W.
The low pass filter rejects the double frequency components and passes only the low pass
components.
the effect of a phase difference between the received carrier and a locally generated carrier at
2
the receiver is a drop equal to os
c ) in the received signal
(
power. Phase-locked loop
The effect of a phase-locked loop is to generate phase of the received carrier at the receiver.
In our analysis in this section, we assume that we are employing a coherent demodulator.
Therefore, at the receiver output, the message signal and the noise components are additive
and we are able to define a meaningful SNR. The message signal power is given by
Power PM is the content of the messagesignal
The power content of n(t) can be found by noting that it is the result of passing n (t) through
w
a filter with bandwidth B .Therefore, the power spectral density of n(t) is given by
c
In DSB-SC AM, the output SNR is the same as the SNR for a baseband system. DSB-SC
AM does not provide any SNR improvement over a simple baseband communication system.
Noise in Conventional AM
Power content of the normalized message process depends on the message source.
The reason for this loss is that a large part of the transmitter power is used to send the
carrier component of the modulated signal and not the desired signal. To analyze the
envelope-detector performance in the presence of noise, we must use certain
approximations.
This is a result of the nonlinear structure of an envelope detector, which makes an exact
analysis difficult
In this case, the demodulator detects the envelope of the received signal and the noise
process.
The input to the envelope detector is
which is basically the same as y(t) for the synchronous demodulation without the ½
coefficient.
This coefficient, of course, has no effect on the final SNR. So we conclude that, under the
assumption of high SNR at the receiver input, the performance of synchronous and envelope
demodulators is the same.
However, if the preceding assumption is not true, that is, if we assume that, at the receiver
input, the noise power is much stronger than the signal power, Then
We observe that, at the demodulator output, the signal and the noise components are no
longer additive. In fact, the signal component is multiplied by noise and is no longer
distinguishable. In this case, no meaningful SNR can be defined. We say that this system is
operating below the threshold. The subject of threshold and its effect on the performance of
a communication system will be covered in more detail when we discuss the noise
performance in angle modulation.
The expression however does not apply when the carrier-to-noise ratio decreases below a
certain point. Below this critical point the signal-to-noise ratio decreases significantly. This is
known as the FM threshold effect (FM threshold is usually defined as the carrier-to-noise
ratio at which the demodulated signal-to-noise ratio fall 1 dB below the linear relationship
given in Eqn 9. It generally is considered to occur at about 10 dB).
Below the FM threshold point the noise signal (whose amplitude and phase are randomly
varying), may instantaneously have an amplitude greater than that of the wanted signal.
When this happens the noise will produce a sudden change in the phase of the FM
demodulator output. In an audio system this sudden phase change makes a "click". In video
applications the term "click noise" is used to describe short horizontal black and white lines
that appear randomly over a picture, because satellite communications systems are power
limited they usually operate with only a small design margin above the FM threshold point
(perhaps a few dB). Because of this circuit designers have tried to devise techniques to delay
the onset of the FM threshold effect. These devices are generally known as FM threshold
extension demodulators. Techniques such as FM feedback, phase locked loops and frequency
locked loops are used to achieve this effect. By such techniques the onset of FM threshold
effects can be delayed till the C/N ratio is around 7 dB.
Types of Receivers:
Basic principle
o Mixing
o Intermediate frequency of 455 KHz
o Ganged tuning
RF section
o Tuning circuits – reject interference and reduce noise figure
o Wide band RF amplifier
Local Oscillator
o 995 KHz to 2105 KHz
o Tracking
IF amplifier
o Very narrow band width Class A amplifier – selects 455 KHz only
o Provides much of the gain
o Double tuned circuits
Detector
o RF is filtered to ground
1. RF Amplifier:
2. Mixer
3. Tracking
4. Local Oscillator
5. IF Amplifier
Introduction:
Pulse Modulation
PAM Generation:
The carrier is in the form of narrow pulses having frequency fc. The uniform
sampling takes place in multiplier to generate PAM signal. Samples are placed Ts sec
away from each other.
Fig.12. PAM Modulator
PAM Demodulator:
The PAM demodulator circuit which is just an envelope detector followed by a
second order op-amp low pass filter (to have good filtering characteristics) is as
shown below
In pulse width modulation (PWM), the width of each pulse is made directly
proportional to the amplitude of the information signal.
In this type, the sampled waveform has fixed amplitude and width whereas the
position of each pulse is varied as per instantaneous value of the analog signal.
• The PWM pulses obtained at the comparator output are applied to a mono stable multi
vibrator which is negative edge triggered.
• Hence for each trailing edge of PWM signal, the monostable output goes high.
It remains high for a fixed time decided by its RC components.
• Thus as the trailing edges of the PWM signal keeps shifting in proportion with
the modulating signal, the PPM pulses also keep shifting.
• Therefore all the PPM pulses have the same amplitude and width. The information
is conveyed via changing position of pulses.
PWM Demodulator:
During time interval A-B when the PWM signal is high the input to transistor T2 is
low.
Therefore, during this time interval T2 is cut-off and capacitor C is charged through
an R-C combination.
During time interval B-C when PWM signal is low, the input to transistor T2 is high,
and it gets saturated.
Thus, the waveform at the collector of T2is similar to saw-tooth waveform whose
envelope is the modulating signal.
Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.
PPM Demodulator:
The gaps between the pulses of a PPM signal contain the information regarding the
modulating signal.
During gap A-B between the pulses the transistor is cut-off and the capacitor C gets
charged through R-C combination.
During the pulse duration B-C the capacitor discharges through transistor and the
collector voltage becomes low.
Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.