Lecture Notes, Mr.

Chibole, EEENG 483, Chuka University, Module-1 1

 EEE 483 Communication

Systems Modulation
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 1

To understand what modulation is all about in general, I will use a transport example. There are at least
three ways you came to this class today: 1. You walked from your hostel, 2. boarded a bodaboda
(motorbike) or 3. used a vehicle. The most determinant of the mode of transport was the distance from
your home to the classroom. Those who live nearby come on foot – they just walked, those somehow far,
maybe 5k or more, they used a motorbike or vehicle. Definitely, because I am many kms away from the
university, I used a vehicle.

The above concept is rightly applicable to the general human communication. If you want to talk to your
deskmate, you will simply turn around and talk. If you are to talk to someone, a km away, you will have to
shout, perhaps. If you have to talk to someone, in Chuka town from your classroom, however hard you
shout your voice cannot reach the desired destination.

Welcome to modulation. The natural voice you use to talk to your deskmate or shout is what is known as
the baseband signal. Human voice is at lower frequencies, and just like I had to use a vehicle to come to
class, lest I get late, baseband cannot be used to communicate far, they need to be transported over
another signal called the carrier signal. The process of transmitting a baseband signal over a carrier signal
for long distance communication is known as modulation. The reverse process is known as demodulation.
Carrier signals are high frequency signals used for long distance communication, while baseband signals
are low frequency signals and need to be transported over high-frequency signals for them to reach far
destinations before they are converted back to baseband signals. Oftentimes, the baseband signal is
sometimes referred to as the information signal or the intelligent signal.

In the modulation process, the baseband voice, video, or digital signal modifies another, higher-frequency
signal called the carrier, which is usually a sine wave. A sine wave carrier can be modified by the
intelligence (baseband) signal through amplitude modulation, frequency modulation, or phase
modulation. We focus on amplitude modulation (AM) first.

Amplitude Modulation (AM)

As the name suggests, in 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. Fig. 3-1 shows a single frequency sine wave
intelligence signal modulating a higher-frequency carrier.

The carrier frequency remains constant during the modulation process, but its amplitude varies in
accordance with the modulating signal. An increase in the amplitude of the modulating signal causes the
amplitude of the carrier to increase. Both the positive and the negative peaks of the carrier wave vary
with the modulating signal. An increase or a decrease in the amplitude of the modulating signal causes a
corresponding increase or decrease in both the positive and the negative peaks of the carrier amplitude.
An imaginary line connecting the positive peaks and negative peaks of the carrier waveform (the dashed
line in Fig. 3-1) gives the exact shape of the modulating information signal. This imaginary line on the
carrier waveform is known as the envelope.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 2

Because complex waveforms such as that shown in Fig. 3-1 are difficult to draw, they are often simplified
by representing the high-frequency carrier wave as many equally spaced vertical lines whose amplitudes
vary in accordance with a modulating signal, as in Fig. 3-2:

The signals illustrated in Figs. 3-1 and 3-2 show the variation of the carrier amplitude with respect to time
and are said to be in the time domain. Time-domain signals— voltage or current variations that occur
over time—are displayed on the screen of an oscilloscope.
Using trigonometric functions, we can express the sine wave carrier with the simple expression:
𝑣𝑐 = 𝑉𝑐𝑠𝑖𝑛2𝜋𝑓𝑐𝑡
In this expression, 𝑣𝑐 represents the instantaneous value of the carrier sine wave voltage at some specific
time in the cycle; Vc represents the peak value of the constant unmodulated carrier sine wave as
measured between zero and the maximum amplitude of either the positive-going or the negative-going
alternations (Fig. 3-1); fc is the frequency of the carrier sine wave; and t is a particular point in time during
the carrier cycle.
A sine wave modulating signal can be expressed with a similar formula
𝑣𝑚 = 𝑉𝑚𝑠𝑖𝑛2𝜋𝑓𝑚𝑡
where 𝑣𝑚 = instantaneous value of information signal
Vm = peak amplitude of information signal fm
= frequency of modulating signal

In Fig. 3-1, the modulating signal uses the peak value of the carrier rather than zero as its reference point.
The envelope of the modulating signal varies above and below the peak carrier amplitude. That is, the
zero reference line of the modulating signal coincides with the peak value of the unmodulated carrier.
Because of this, the relative amplitudes of the carrier and modulating signal are important. In general, the
amplitude of the modulating signal should be less than the amplitude of the carrier. When the amplitude
of the modulating signal is greater than the amplitude of the carrier, distortion will occur, causing
incorrect information to be transmitted. In amplitude modulation, it is particularly important that the
peak value of the modulating signal be less than the peak value of the carrier. Mathematically,
𝑉𝑚 > 𝑉𝑐
Values for the carrier signal and the modulating signal can be used in a formula to express the complete
modulated wave. First, keep in mind that the peak value of the carrier is the reference point for the
modulating signal; the value of the modulating signal is added to or subtracted from the peak value of the
carrier. The instantaneous value of either the top or the bottom voltage envelope 𝑣1 can be computed by
using the equation:
𝑣1 = 𝑉𝑐 + 𝑣𝑚 = 𝑉𝑐 + 𝑉𝑚𝑠𝑖𝑛2𝜋𝑓𝑚𝑡
Which expresses the fact that the instantaneous value of the modulating signal algebraically adds to the
peak value of the carrier. Thus, we can write the instantaneous value of the complete modulated wave 𝑣2
by substituting 𝑣1for the peak value of carrier voltage Vc as follows:
𝑣2 = 𝑣1𝑠𝑖𝑛2𝜋𝑓𝑐𝑡
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 3

Now substituting the previously derived expression for v1 and expanding, we get the following:
𝑣2 = (𝑉𝑐 + 𝑉𝑚𝑠𝑖𝑛2𝜋𝑓𝑚𝑡)𝑠𝑖𝑛2𝜋𝑓𝑐𝑡 = 𝑉𝑐𝑠𝑖𝑛2𝜋𝑓𝑐𝑡 + (𝑉𝑚𝑠𝑖𝑛2𝜋𝑓𝑚𝑡)(𝑠𝑖𝑛2𝜋𝑓𝑐𝑡)
Where 𝑣2 is the instantaneous value of the AM wave (or 𝐴𝑀), 𝑉𝑐𝑠𝑖𝑛2𝜋𝑓𝑐𝑡 is the carrier waveform, and
(𝑉𝑚𝑠𝑖𝑛2𝜋𝑓𝑚𝑡)(𝑠𝑖𝑛2𝜋𝑓𝑐𝑡) is the carrier waveform multiplied by the modulating 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 circuit used for producing AM is called a modulator. Its two inputs, the carrier and the modulating
signal, and the resulting outputs are shown in Fig. 3-3. Amplitude modulators compute the product of the
carrier and modulating signals. Circuits that compute the product of two analog signals are also known as
analog multipliers, mixers, converters, product detectors, and phase detectors. A circuit that changes a
lower-frequency baseband or intelligence signal to a higher-frequency signal is usually called a modulator.
A circuit used to recover the original intelligence signal from an AM wave is known as a detector or
demodulator. Mixing and detection applications are discussed in detail in later chapters.

3-2 Modulation Index and Percentage of Modulation

As stated previously, for undistorted AM to occur, the modulating signal voltage 𝑉𝑚 must be less than the
carrier voltage 𝑉𝑐. Therefore, the relationship between the amplitude of the modulating signal and the
amplitude of the carrier signal is important. This relationship, known as the modulation index m (also
called the modulating factor or coefficient, or the degree of modulation), is the ratio

These are the peak values of the signals, and the carrier voltage is the unmodulated value. Multiplying the
modulation index by 100 gives the percentage of modulation. For example, if the carrier voltage is 9V and
the modulating signal voltage is 7.5 V, the modulation factor is 0.8333 and the percentage of modulation is
0.8333𝑥100 = 83.33. Overmodulation and Distortion
The modulation index should be a number between 0 and 1. If the amplitude of the modulating voltage is
higher than the carrier voltage, m will be greater than 1, causing distortion of the modulated waveform. If
the distortion is great enough, the intelligence signal becomes unintelligible. Distortion of voice
transmissions produces garbled, harsh, or unnatural sounds in the speaker. Distortion of video signals
produces a scrambled and inaccurate picture on a TV screen.
Simple distortion is illustrated in Fig. 3-4. Here a sine wave information signal is modulating a sine wave
carrier, but the modulating voltage is much greater than the carrier voltage, resulting in a condition called
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 4

overmodulation. As you can see, the

waveform is flattened at the zero line. The received signal will produce an output waveform in the shape
of the envelope, which in this case is a sine wave whose negative peaks have been clipped off. If the
amplitude of the modulating signal is less than the carrier amplitude, no distortion will occur. The ideal
condition for AM is when 𝑉𝑚 = 𝑉𝑐, or m = 1, which gives 100 percent modulation. This results in the
greatest output power at the transmitter and the greatest output voltage at the receiver, with no
distortion.
Preventing overmodulation is tricky. For example, at different times during voice transmission voices will
go from low amplitude to high amplitude. Normally, the amplitude of the modulating signal is adjusted so
that only the voice peaks produce 100 percent modulation. This prevents overmodulation and distortion.
Automatic circuits called compression circuits solve this problem by amplifying the lower-level signals and
suppressing or compressing the higher-level signals. The result is a higher average power output level
without overmodulation.
Distortion caused by overmodulation also produces adjacent channel interference. Distortion produces a
nonsinusoidal information signal. According to Fourier theory, any nonsinusoidal signal can be treated as
a fundamental sine wave at the frequency of the information signal plus harmonics. Obviously, these
harmonics also modulate the carrier and can cause interference with other signals on channels adjacent
to the carrier.
Percentage of Modulation
The modulation index can be determined by measuring the actual values of the modulation voltage and
the carrier voltage and computing the ratio. However, it is more common to compute the modulation
index from measurements taken on the composite modulated wave itself. When the AM signal is
displayed on an oscilloscope, the modulation index can be computed from Vmax and Vmin, as shown in
Fig. 3-5.

The peak value of the modulating signal 𝑉𝑚 is one-half the difference of the peak and trough values:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 5

As shown in Fig. 3-5, Vmax is the peak value of the signal during modulation, and Vmin is the
lowest value, or trough, of the modulated wave. The Vmax is one-half the peak-to-peak value
of the AM signal, or Vmax (p 2p)/2. Subtracting Vmin from Vmax produces the peakto- peak
value of the modulating signal. One-half of that, of course, is simply the peak value.
The peak value of the carrier signal Vc is the average of the Vmax and Vmin values:

The modulation index is

The values for 𝑉max (𝑝−𝑝) and 𝑉min( 𝑝−𝑝) can be read directly from an oscilloscope screen and
plugged directly into the formula to compute the modulation index.
The amount, or depth, of AM is more commonly expressed as the percentage of modulation rather than
as a fractional value. In Example 3-1, the percentage of modulation is 100 𝑋 𝑚, or 66.2 percent. The
maximum amount of modulation without signal distortion, of course, is 100 percent, where Vc and Vm
are equal. At this time, Vmin = 0 and Vmax = 2Vm, where Vm is the peak value of the modulating signal.
Example 3-1
Suppose that on an AM signal, the 𝑉max( 𝑝−𝑝) value read from the graticule on the oscilloscope screen is
5.9 divisions and 𝑉min( 𝑝−𝑝) is 1.2 divisions.
a. What is the modulation index?

b. Calculate 𝑉𝑐, 𝑉𝑚 and m if the vertical scale is 2V per division. (Hint: Sketch the signal.)

𝑉𝑚 = 2.35 × 2𝑉 = 4.7𝑉
𝑉 47

Double Sideband (DSB)

In the process of Amplitude Modulation, the modulated wave consists of the carrier wave and
two sidebands. The modulated wave has the information only in the sidebands. Sideband is
nothing but a band of frequencies, containing power, which are the lower and higher frequencies
of the carrier frequency.
The transmission of a signal, which contains a carrier along with two sidebands can be termed as
Double Sideband Full Carrier system or simply DSBFC. It is plotted as shown in the following
figure.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 6

However, such a transmission is inefficient. Because, two-thirds of the power is being wasted in
the carrier, which carries no information.
If this carrier is suppressed and the saved power is distributed to the two sidebands, then such a
process is called as Double Sideband Suppressed Carrier system or simply DSBSC. It is plotted as
shown in the following figure.

Mathematical Expressions
Let us consider the same mathematical expressions for modulating and carrier signals as we have
considered in the earlier chapters. i.e., Modulating signal
𝑚(𝑡) = 𝐴𝑚cos (2𝜋𝑓𝑚𝑡)
Carrier signal
𝑐(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐𝑡)
Mathematically, we can represent the equation of DSBSC wave as the product of modulating and
carrier signals.
𝑠(𝑡) = 𝑚(𝑡)𝑐(𝑡)
⇒ 𝑠(𝑡) = 𝐴𝑚𝐴𝑐cos (2𝜋𝑓𝑚𝑡) cos(2𝜋𝑓𝑐𝑡)
Bandwidth of DSBSC Wave
We know the formula for bandwidth (BW) is
𝐵𝑊 = 𝑓𝑚𝑎𝑥 − 𝑓𝑚𝑖𝑛 Consider
the equation of DSBSC modulated wave.

The DSBSC modulated wave has only two frequencies. So, the maximum and minimum
frequencies are 𝑓𝑐 + 𝑓𝑚 and 𝑐 − 𝑓𝑚 respectively. i.e.,
𝑓𝑚𝑎𝑥 = 𝑓𝑐 + 𝑓𝑚 and 𝑚𝑖𝑛 = 𝑓𝑐 − 𝑓𝑚
Substitute, 𝑓𝑚𝑎𝑥 and 𝑓𝑚𝑖𝑛 values in the bandwidth formula.
𝐵𝑊 = 𝑓𝑐 + 𝑓𝑚 − (𝑓𝑐 − 𝑓𝑚)
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 7

⇒ 𝐵𝑊 = 2𝑓𝑚
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.
Power Calculations of DSBSC Wave
Consider the following equation of DSBSC modulated wave.

Power of DSBSC wave is equal to the sum of powers of upper sideband and lower sideband
frequency components.
𝑃𝑡 = 𝑃𝑈𝑆𝐵 + 𝑃𝐿𝑆𝐵
We know the standard formula for power of cos signal is

First, let us find the powers of upper sideband and lower sideband one by one. Upper
sideband power

Similarly, we will get the lower sideband power same as that of upper sideband power.

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)
In the previous section, we have discussed DSBSC modulation. The DSBSC modulated signal has
two sidebands. Since, the two sidebands carry the same information, there is no need to transmit
both sidebands. We can eliminate one sideband.
The process of suppressing one of the sidebands along with the carrier and transmitting a single
sideband is called Single Sideband Suppressed Carrier system or simply SSBSC. It is plotted as
shown in the following figure.

In the above figure, the carrier and the lower sideband are suppressed. Hence, the upper sideband
is used for transmission. Similarly, we can suppress the carrier and the upper sideband while
transmitting the lower sideband.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 8

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.
Mathematical Expressions
Let us consider the same mathematical expressions for the modulating and the carrier signals as
we have considered in the previous section. i.e., Modulating signal
𝑚(𝑡) = 𝐴𝑚cos (2𝜋𝑓𝑚𝑡)

Carrier signal
𝑐(𝑡) = 𝐴𝑐cos( 2𝜋𝑓𝑐t)
Mathematically, we can represent the equation of SSBSC wave as

for the upper sideband, or

for the lower sideband

Bandwidth of SSBSC Wave
We know that the DSBSC modulated wave contains two sidebands and its bandwidth is 2𝑓𝑚. Since
the SSBSC modulated wave contains only one sideband, its bandwidth is half of the bandwidth of
DSBSC modulated wave.
i.e., Bandwidth of SSBSC modulated wave =
Therefore, the bandwidth of SSBSC modulated wave is 𝑓𝑚 and it is equal to the frequency of the
modulating signal.
Power Calculation of SSBSC Wave
Consider the following equation of SSBSC modulated wave.

for the upper sideband, or

for the lower sideband

Power of SSBSC wave is equal to the power of the power of any one sideband frequency
components.
𝑃𝑡 = 𝑃𝑈𝑆𝐵 + 𝑃𝐿𝑆𝐵 We
know the standard formula for power of cos signal is

In this case, the power of the upper sideband is.

Similarly, we will get the lower sideband power same as that of upper sideband power.

Therefore, the power of SSBSC wave is

Advantages
• Bandwidth or spectrum space occupied is lesser than AM and DSBSC waves.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 9

• Transmission of more number of signals is allowed.

• Power is saved.
• High power signal can be transmitted.
• Less amount of noise is present.
• Signal fading is less likely to occur.
Disadvantages
• The generation and detection of SSBSC wave is a complex process.
• The quality of the signal gets affected unless the SSB transmitter and receiver have an
excellent frequency stability.
Applications
• For power saving requirements and low bandwidth requirements.
• In land, air, and maritime mobile communications.
• In point-to-point communications.
• In radio communications.
• In television, telemetry, and radar communications.
• In military communications, such as amateur radio, etc.
Vestigial Sideband (VSB)
In case of SSB modulation, when a sideband is passed through the filters, the band pass filter
may not work perfectly in practice. As a result of which, some of the information may get lost.
Hence to avoid this loss, a technique is chosen, which is a compromise between DSB-SC and
SSB, called as Vestigial Sideband (VSB) technique. The word vestige which means “a part” from
which the name is derived. Vestigial Sideband
Both of the sidebands are not required for the transmission, as it is a waste. But a single band if
transmitted, leads to loss of information. Hence, this technique has evolved.
Vestigial Sideband Modulation or VSB Modulation is the process where a part of the signal called
as vestige is modulated, along with one sideband. A VSB signal can be plotted as shown in the
following figure.

Along with the upper sideband, a part of the lower sideband is also being transmitted in this
technique. A guard band of very small width is laid on either side of VSB in order to avoid the
interferences. VSB modulation is mostly used in television transmissions.
Transmission Bandwidth
The transmission bandwidth of VSB modulated wave is represented as –
𝐵 = (𝑓𝑚 + 𝑓𝑣)𝐻𝑧
where,
𝑓𝑚 = Message bandwidth
𝑓𝑣 = Width of the vestigial sideband
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 10

VSB Modulation − Advantages

Following are the advantages of VSB −
Highly efficient.
• Reduction in bandwidth.
• Filter design is easy as high accuracy is not needed.
• The transmission of low frequency components is possible, without difficulty.
Possesses good phase characteristics. VSB Modulation − Disadvantages
Following are the disadvantages of VSB −
• Bandwidth when compared to SSB is greater.
• Demodulation is complex.
VSB Modulation − Application
The most prominent and standard application of VSB is for the transmission of television signals.
Also, this is the most convenient and efficient technique when bandwidth usage is considered.

Superhetrodyne AM Receiver

The antenna present at the beginning of the receiver section, receives the modulated wave. First
let us discuss the requirements of a receiver.
Requirements of a Receiver
AM receiver receives AM wave and demodulates it by using the envelope detector. Similarly, FM
receiver receives FM wave and demodulates it by using the Frequency Discrimination method.
Following are the requirements of both AM and FM receiver.
• It should be cost-effective.
• It should receive the corresponding modulated waves.
• The receiver should be able to tune and amplify the desired station.
• It should have an ability to reject the unwanted stations.
• Demodulation has to be done to all the station signals, irrespective of the carrier signal
frequency.
For these requirements to be fulfilled, the tuner circuit and the mixer circuit should be very
effective. The procedure of RF mixing is an interesting phenomenon.
RF Mixing
The RF mixing unit develops an Intermediate Frequency (IF) to which any received signal is
converted, so as to process the signal effectively.
RF Mixer is an important stage in the receiver. Two signals of different frequencies are taken
where one signal level affects the level of the other signal, to produce the resultant mixed output.
The input signals and the resultant mixer output is illustrated in the following figures.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 11

𝑓1 𝑓2Let the first and second signal

frequencies be and . If these two signals are applied as inputs of RF mixer, then it produces
an output signal, having frequencies of 1 + 𝑓2 and 1 − 𝑓2. If this is observed in the frequency
domain, the pattern looks like the following figure.

In this case, 𝑓1 is greater than 𝑓2. So, the resultant output has frequencies 𝑓1 + 𝑓2and 𝑓1 − 𝑓2.
Similarly, if 𝑓2 is greater than 𝑓1, then the resultant output will have the frequencies 𝑓1 +
2and 1 − 2.

AM Receiver
The AM super heterodyne receiver takes the amplitude modulated wave as an input and produces
the original audio signal as an output. Selectivity is the ability of selecting a particular signal, while
rejecting the others. Sensitivity is the capacity of detecting RF signal and demodulating it, while
at the lowest power level.
Radio amateurs are the initial radio receivers. However, they have drawbacks such as poor
sensitivity and selectivity. To overcome these drawbacks, super heterodyne receiver was
invented. The block diagram of AM receiver is shown in the following figure.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 12

RF Tuner Section
The amplitude modulated wave received by the antenna is first passed to the tuner circuit
through a transformer. The tuner circuit is nothing but a LC circuit, which is also called as resonant
or tank circuit. It selects the frequency, desired by the AM receiver. It also tunes the local
oscillator and the RF filter at the same time.
RF Mixer
The signal from the tuner output is sent to the RF-IF converter, which acts as a mixer. It has a local
oscillator, which produces a constant frequency. The mixing process is done here, having the
received signal as one input and the local oscillator frequency as the other input. The resultant
output is a mixture of two frequencies [(𝑓1 + 𝑓2),(𝑓1 − 𝑓2)] produced by the mixer, which is called
as the Intermediate Frequency (IF).
The production of IF helps in the demodulation of any station signal having any carrier frequency.
Hence, all signals are translated to a fixed carrier frequency for adequate selectivity.
IF Filter
Intermediate frequency filter is a band pass filter, which passes the desired frequency. It
eliminates all other unwanted frequency components present in it. This is the advantage of IF
filter, which allows only IF frequency.
AM Demodulator
The received AM wave is now demodulated using AM demodulator. This demodulator uses the
envelope detection process to receive the modulating signal.
Audio Amplifier
This is the power amplifier stage, which is used to amplify the detected audio signal. The
processed signal is strengthened to be effective. This signal is passed on to the loudspeaker to get
the original sound signal. Carrier Acquisition
In suppressed carrier communication, the demodulation process requires an identical local carrier
at the demodulator. This local carrier needs to have same frequency and phase as the transmitted
signal. which is acquired using carrier acquisition.
Amplitude Modulated Suppressed Carrier signal needs a locally generated signal having the same
phase and frequency as the received signal (carrier signal).
If the frequency or phase of the signal is different than the received signal, then the demodulated
message signal we get will be distorted or may be completely destroyed. To avoid such problems
and get a clear message signal, we need an identical carrier.
Suppose we receive a DSB-SC signal which is 𝑚(𝑡)𝑐𝑜𝑠(𝜔𝑐𝑡 + 𝛳𝑖) and the carrier generated by
demodulator has a frequency (ωc+Δω) and phase ϴo i.e. the local carrier is 2cos((ωc+Δω)t+ϴo).
Thus the demodulated signal e(t) will be:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 13

e(t) = m(t) cos(ωct + ϴi) 2cos((ωc+Δω)t+ ϴo) e(t)

= 2m(t) cos(ωct + ϴi) cos((ωc+Δω)t+ ϴo)
e(t) = m(t) [cos {(ωc+ωc+Δω) t +ϴi+ϴo} + cos(Δωt+ ϴi-ϴo)] e(t)
= m(t) cos {(ωc+ωc+Δω) t +ϴi+ϴo} + m(t)cos(Δωt+ ϴi-ϴo)
By passing through Low-pass filter e(t) = m(t)cos(Δωt+ ϴi-ϴo)
If the frequency difference Δω = 0 and phase difference (ϴi-ϴo) = 0. Then e(t)
= m(t)
Which means the message signal is successfully received.
However, if the frequency difference Δω = 0 and phase difference (ϴi-ϴo) ≠0. Then e(t)
= m(t)cos(ϴi-ϴo)
This means that the message signal is attenuated by factor cos(ϴi-ϴo). if (ϴi-ϴo)= π/2 , then e(t)
= 0 and the message signal is completely destroyed.
Another case is if the phase difference (ϴi-ϴo) =0 and frequency difference Δω ≠ 0. Then e(t)
= m(t)cos(Δωt)
This equation implies that the same message signal is multiplied with a sinusoid of frequency
Δω. Δω is usually very small. which means that the message signal will go from maximum to
zero at the rate of two times its frequency. This is called beating effect. This beating effect
distorts the original signal even if the Δω is very small.
Techniques Of Carrier Acquisition
There are few different techniques used for carrier acquisition. Some of them are given below.
Phase-Locked Loop (PLL)
Phase locked loop, commonly known as PLL is one of the most widely used circuit for carrier
acquisition. It tracks the phase and frequency of the incoming/reference signal and generates a
stable frequency signal.
A PLL is made up of 3 components
• VCO
• Phase detector
• Loop filter

VCO
VCO stands for the voltage controlled oscillator. It generates frequency signal, which is
controlled by an external voltage signal. The frequency signal produced by VCO is ω(t)
= ωc + ceo(t)
ωc is free running frequency when external voltage eo(t) is zero. C is constant of VCO & eo(t) is the
external voltage signal. The frequency is increased or decreased according to this voltage signal
eo(t).
Phase Detector
A Multiplier is used as a phase detector. It has 2 input signals, a reference signal & the output of
VCO.
It generates a signal proportional to the phase difference between the two signals.
Suppose the input signal is Asin(ωct + ϴi) & output is Bcos(ωct + ϴo),then its product is
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 14

e(t) = Asin(ωct + ϴi) Bcos(ωct + ϴo) e(t) = AB/2 sin(2ωct + ϴi + ϴo) + AB/2 sin(ϴi – ϴo)
The high-frequency term is filtered the loop filter discussed below.
Loop Filter
This loop filter is actually a narrow band low pass filter. It blocks any high-frequency components
from its input signal (output of multiplier) and generates a dc voltage. which is supplied as input
to the VCO.
The signal after passing through loop filter becomes
eo(t) = AB/2 sin(ϴi – ϴo)
If the phase difference (ϴi – ϴo) is not zero then the signal eo(t) will generate DC voltage & supplies
to the VCO. This voltage leads to increment in the VCO frequency.
The process is repeated until the frequency & phase matches the input signal. Such case is called
in phase lock or phase coherent state.
Carrier Acquisition In DSB-SC
In DSB-SC scheme the level carrier can be regenerated using two methods discussed below.
Signal Squaring Method:
This method is used for carrier acquisition in DSB-SC communication. The
block diagram of signal-squarer is given below.

The received DSB-SC signal x(t) is first passed through a squarer, which takes the square of the
signal.
The received signal x(t) is:
x(t) = m(t)cos ωct The
output y(t) of squarer is:
y(t) = x2(t)
y(t) = (m(t)cos ωct)2
y(t) = m2(t)cos2 ωct
y(t) = ½ m2(t)(1+cos 2ωct) y(t)
= ½ m2(t)+ ½ m2(t)cos 2ωct
2
As we can see, m (t) is a non-negative signal i.e. it is positive for every value of t. Therefore, it has
positive average (DC) value.
Let suppose the average value of m2(t)/2 is k then
½ m2(t) = k + ϕ(t) Now the
signal y(t) can be expressed as: y(t) = ½ m2(t)+ (k + ϕ(t))cos 2ωct y(t)
= ½ m2(t)+ k cos 2ωct + ϕ(t)cos 2ωct
After passing through the narrow-band band-pass filter, it will block m2(t) completely because of
its ω=0. k cos 2ωct will flow through. However, some parts of ϕ(t)cos 2ωct will also flow out
because it has almost no power at 2ωc. Thus the signal y0(t) becomes: y0(t) = k cos 2ωct + ϕ(t)cos
2ωct
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 15

The next stage is PLL. The PLL will block any residual frequencies & produce a stable frequency
signal z(t), which is : z(t) = k cos 2ωct
The last stage of signal squarer is the divider. The divider divides the frequency of the input signal
by two. Thus the output signal becomes a pure sinusoidal wave of frequency ωc.
The output r(t) of signal squarer is: r(t) = k cos ωct
COSTAS Loop
John P.Costas was an Electrical engineer. In 1950, he invented the method to use a modified PLL
to regenerate the carrier signal in suppressed carrier communication. This circuit is known as
Costas loop.
Costas loop is used to acquire the carrier signal in DSB-SC communication.
Block Diagram
The block diagram of Costas loop is given below:

This diagram shows the received signal DSB-SC signal m(t)cos(ωct+ϴi) is multiplied with local
carriers cos(ωct+ϴo) & sin(ωct+ϴo) separately to get x1(t) and x2(t) respectively.
The VCO generates the local carrier cos(ωct+ϴo), which is phase shifted by –π/2 to generate
sin(ωct+ϴo).
The signal x1(t) and x2(t) is given by: x1(t) = m(t)cos(ωct+ϴi)
cos(ωct+ϴo)
x1(t) = ½ m(t){cos(ϴi– ϴo) +cos(2ωct+ ϴi +ϴo)} x1(t)
= ½ m(t)cos(ϴi– ϴo) +½ m(t)cos(2ωct+ ϴi +ϴo)
x2(t) = m(t)cos(ωct+ϴi) sin(ωct+ϴo)
x2(t) = ½ m(t){sin(ϴi– ϴo) +sin(2ωct+ ϴi +ϴo)} x2(t)
= ½ m(t)sin(ϴi– ϴo) +½ m(t)sin(2ωct+ ϴi +ϴo)
The signal x1(t) & x2(t) is then passed through low pass filter, it blocks high frequency
components & allow low frequency components. Thus producing y1(t)
& y2(t) for the signal x1(t) & x2(t) respectively. y1(t) = ½ m(t)cos(ϴi– ϴo) y2(t)
= ½ m(t)sin(ϴi– ϴo)
These two signals y1(t) & y2(t) are then multiplied to produce z(t) as:
z(t) = ½ m(t)cos(ϴi– ϴo) ½ m(t)sin(ϴi– ϴo)
z(t) = ⅛ m2(t){sin(0) + sin2(ϴi– ϴo)} z(t) =
⅛ m2(t) sin2(ϴi– ϴo)
Thus the signal z(t) will produce a DC voltage depending on the phase difference (ϴi– ϴo).
If there is any phase difference, then this signal will produce DC voltage.
The narrowband low-pass filter will suppress any frequency components and produce a pure DC
signal. This DC signal will either increase or decrease the frequency of the VCO.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 16

When the frequency and phase of the input signal and matches the VCO output, then the phase
difference (ϴi– ϴo) = 0 and the DC output of Narrowband LPF becomes 0. In such case, the VCO
output remains unchanged.
The output of the VCO is the acquired carrier we need & the signal y1(t) is the demodulated
message signal.
y1(t) = ½ m(t)cos(ϴi– ϴo)
y1(t) = ½ m(t)cos(0) y1(t)
= ½ m(t)
Carrier Acquisition in SSB
In single sideband (SSB) communication, the methods of carrier acquisition do not work as it did
in the DSB-SC. The signal-squaring method & Costas loop does not work. The reason is that after
squaring SSB signal, the product terms does not contain a pure sinusoid of the carrier frequency
as in DSB-SC. So extracting the carrier through such method does not work.
However, if we transmit a carrier signal of low power with SSB signal, it can be extracted using a
narrowband band-pass filter. The said signal is then amplified, in such way the demodulator will
know the frequency & phase of the carrier signal.
Vestigial Sideband (VSB) has the same situation as SSB and it also needs a separate carrier with
the transmitted signal.

Angle Modulation
In the previous section, we studied the different AM technique in which the amplitude of some
carrier signal is modified according to the message signal. The frequency and phase of the carrier
of the carrier signal in all AM modulation techniques were constant. In this section, we will study
a different method for transmitting information by changing the phase or frequency (changing
the angle) of the carrier signal and keeping its amplitude constant.
Angle Modulation is the process in which the frequency or the phase of the carrier signal varies
according to the message signal. The standard equation of the angle modulated wave is:
𝑠(𝑡) = 𝐴𝑐𝑐𝑜𝑠𝜃𝑖(𝑡)
where,
𝐴𝑐 is the amplitude of the modulated wave, which is the same as the amplitude of the carrier
signal, and 𝜃𝑖(𝑡) is the angle of the modulated wave.

Instantaneous Frequency

The frequency of a cosine function x(t) that is given by

is equal to 𝜔𝑐 since it is a constant with respect to t, and the phase of the cosine is the constant
0. The angle of the cosine 𝜃(𝑡) = 𝜔𝑐𝑡 + 𝜃0 is a linear relationship with respect to t (a straight
line with slope of 𝜔𝑐 and y–intercept of 𝜃0). 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
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 17

𝑦(𝑡) = cos[ 𝜃(𝑡)]

where 𝜃(𝑡) is a function of time. The frequency of y(t) in this case depends on the function of
𝜃(𝑡) and may itself be a function of time. The instantaneous frequency of y(t) given above is
defined as

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 𝜔𝑐. Clearly this satisfies the definition of the instantaneous frequency since 𝜃(𝑡) = 𝜔𝑐𝑡
+ 𝜃0 and therefore 𝜔𝑖(𝑡) = 𝜔𝑐.

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 𝜃(𝑡) of some sinusoid is the bases for the two types of anglemodulation:
Phase and Frequency modulation techniques.

Phase Modulation (PM)

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

where A is a constant, 𝜔𝑐 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
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 17

So, the frequency of a PM signal is proportional to the derivative of the message signal.

Frequency Modulation (FM)

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

where kf is a
parameter that specifies how much change in the frequency occurs forevery unit change of m(t).
The phase and instantaneous frequency of this FM are

Relation between PM and FM

PM and FM are tightly related to each other. We see from the phase and frequency relations for
PM and FM given above that replacing m(t) in the PM signal with

Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 18

gives an FM signal and replacing m(t) in the FM signal with gives a PM signal. This is
illustrated in the following block diagrams.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 20

Bandwidth of FM/PM
The bandwidth, sideband formation and
spectrum of a frequency modulated signal are
not as straightforward as they are for an
amplitude modulated signal.

Nevertheless the sidebands and bandwidth of

the FM signal are still very important and used
within the planning, design and even the
maintenance of radio broadcast and radio
communication systems.
Frequency
Using a well know rule called Carson's Rule it is modulation sideband levels
possible to provide a good estimate of the
It may also be helpful to have some tabulated
bandwidth of an FM signal. This estimate is
values - from this it can be seen that for a
sufficiently good for virtually all requirements
modulation index of 2.41, the carrier falls to zero,
and as a result Carson's rule is widely used.
and all the power is contained within the
Knowing the levels of the sidebands and the sidebands.

signal bandwidth is very important for broadcast It can also be seen that for low levels of
transmitters and receivers as well as those sued modulation index, the only sidebands that have
for radio communication applications. any significant levels of power within them are
the first, and possibly the second sidebands.
Frequency modulation sidebands
RELATIVE AMPLITUDES OF FM SIDEBANDS FOR DIFFERENT M
The modulation of any carrier in any way
produces sidebands. For amplitude modulated
signals, the way in which these sidebands are RELATIVE SIDEBAND AM
created and their bandwidth and amplitude are
quite straightforward. The situation for
MOD 0 1 2 3
frequency modulated signals is rather different.
INDEX
The FM sidebands are dependent on both the
level of deviation and the frequency of the 0.00 1.00
modulation. In fact the total spectrum for a
frequency modulated signal consists of the 0.25 0.98 0.12
carrier plus an infinite number of sidebands
spreading out on either side of the carrier at 0.5 0.94 0.24 0.03
integral multiples of the modulating frequency.
1.0 0.77 0.44 0.11 0.02
From the diagram it can be seen that the values
for the levels of the sidebands rise and fall with 2.0 0.22 0.58 0.35 0.13
varying values of deviation and modulating
frequency. PLIT
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 21

2.41 0.00 0.52 0.43 0.20 Where:

𝛥𝑓 = 𝑑𝑒𝑣𝑖𝑎𝑡𝑖𝑜𝑛
In theory the sidebands of a frequency 𝐵𝑇 =
modulated signal extent out for ever. 𝑡𝑜𝑡𝑎𝑙 𝑏𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ (𝑓𝑜𝑟 98% 𝑝𝑜𝑤𝑒𝑟)
Fortunately outside the main signal 𝑓𝑚 = 𝑚𝑜𝑑𝑢𝑙𝑎𝑡𝑖𝑛𝑔 𝑓𝑟𝑒𝑞𝑢𝑒𝑛𝑐𝑦
area itself, the level of the sidebands To take the example of a typical broadcast FM signal that has a
falls away and for practical systems deviation of ±75kHz and a maximum modulation frequency of 15
filtering all but removes them kHz, the bandwidth of 98% of the power approximates to 2 (75 +
without any main detriment to the 15) = 180kHz. To provide conveniently spaced channels 200 kHz is
signal. allowed for each station.

For small values of modulation index, when using The rule is also very useful when determining the
narrow-band FM, NBFM, radio communication bandwidth of many two-way radio
systems, the signal consists of the carrier and the communications systems. These use narrow band
two sidebands spaced at the modulation FM, and it is particularly important that the
frequency either side of the carrier. The sidebands do not cause interference to adjacent
sidebands further out are minimal and can be channels that may be occupied by other users.
ignored. On a spectrum analyzer the signal looks
very much like the spectrum of an AM signal. The Equations & calculation for FM sideband levels
difference is that the lower sideband is out of
phase by 180°. Whilst it is very useful to have an understanding
of the broad principles of the generation of
As the level of the modulation index is increased
sidebands within an FM signal, it is sometimes
other sidebands at twice the modulation
necessary to determine the levels
frequency start to appear. Further increases in
mathematically.
modulation index result in the level of other
sidebands increasing in level. The calculations are not nearly as simple as they
are for amplitude modulated signals and they
Carson's Rule for FM bandwidth
involve some long equations. It is for this reason
that rules like Carson's rule are so useful as they
The bandwidth of an FM signal is not as provide workable approximations that are simple
straightforward to calculate as that of an AM and straightforward to calculate, whist being
signal. sufficiently accurate for most radio
communications applications.
A very useful rule of thumb used by many
engineers to determine the bandwidth of an FM The sideband levels can be calculated for a carrier
signal for radio broadcast and radio modulated by a single sine wave using Bessel
communications systems is known as Carson's functions of the first kind as a function of
Rule. This rule states that 98% of the signal power modulation index.
is contained within a bandwidth equal to the
deviation frequency, plus the modulation The basic Bessel function equation is described below:
frequency doubled. Carson's Rule can be
expressed simply as a formula:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 22

Where: signal bandwidth and the way in which the sidebands are
α is an arbitrary complex number produced is useful for these systems.
In terms of the format of the
equation, α and -α produce the It is worth summarizing some of the highlight points about
same differential equation, but it frequency modulation sidebands, FM spectrum & bandwidth.
is conventional to define • The bandwidth of a frequency modulated
different Bessel functions for signal varies with both deviation and
these two values in such a way modulating frequency.
that the Bessel functions are • Increasing modulating frequency increases
mostly smooth functions of α. the frequency separation between sidebands.
Solving the Bessel equations to • Increasing modulating frequency for a given
determine the levels of the individual level of deviation reduces modulation index.
sidebands can be quite complicated, As a result, it reduces the number of
but is ideal for solution using a sidebands with significant amplitude. This has
computer. the result of reducing the bandwidth.

By manipulating the mathematics, it is possible to • The frequency modulation bandwidth

solve the basic Bessel function equation and increases with modulation frequency but it is
express it in the format: not directly proportional to it.

Frequency modulation bandwidth is a key issue as

it is very important to ensure that these
transmissions stay within their allocated channel.
Accordingly, FM signals need to be carefully
tailored to ensure all the significant sidebands
remain within the channel allocation.

The way the series has expanded shows how the

various sidebands are generated and how they
extend out to infinity.

Summary of frequency modulation bandwidth

& sidebands

Frequency modulation is still in widespread use,

both for broadcasting and for two-way radio
communications. As a result, a knowledge of the

Bandwidth of PM
 Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 23

Phase modulation (PM) is a method of analog modulation where the phase of a carrier wave is varied in
accordance with the instantaneous amplitude of the modulating signal. This modulation technique is
commonly used in various communication systems, including radio broadcasting, satellite
communication, and more. The bandwidth of a phase-modulated signal depends on several factors,
including the modulation index and the modulation frequency.
Here are some detailed notes on the bandwidth of phase modulation:
1. Phase Modulation Basics:
 In phase modulation, the instantaneous phase of the carrier signal is varied in response to the
modulating signal's amplitude variations.
 Mathematically, phase modulation is expressed as: 𝜃(𝑡) = 𝜃𝑐 + 𝐾𝑝 ⋅ 𝑚(𝑡)
Where:
 𝜃(𝑡) is the instantaneous phase at time t.
 𝜃𝑐 is the carrier phase.
 𝐾𝑝 is the phase sensitivity (modulation index).
 𝑚(𝑡) is the modulating signal.
2. Bandwidth in Phase Modulation:
 The bandwidth of a phase-modulated signal is related to the frequency content of the
modulating signal and the modulation index.
 The bandwidth of the PM signal can be approximated as: 𝐵𝑃𝑀 ≈ (1 + 𝛽). 𝑊𝑚
Where:
 𝐵𝑃𝑀 is the bandwidth of the phase modulated signal.
 𝛽 is the modulation index,
𝛽 = 𝐾𝑝 ⋅ 𝐴𝑚 .
 𝑊𝑚 is the bandwidth of the modulating signal.
 The modulation index 𝛽 is a crucial factor in determining the bandwidth. If the modulation index
is small (close to 0), the bandwidth is approximately equal to the bandwidth of the modulating
signal. As the modulation index increases, the bandwidth of the PM signal also increases.
3. Bandwidth Calculation Example:
 Suppose you have a sinusoidal modulating signal with a bandwidth of 10 kHz, and you apply
phase modulation with a modulation index of 2. This results in a bandwidth of approximately
30 kHz for the phase-modulated signal.
4. Importance of Bandwidth Control:
 Managing bandwidth is essential in communication systems because it directly affects the
allocation of frequency spectrum and system capacity.
 In some cases, bandwidth efficiency can be improved by limiting the modulation index to
reduce the signal's bandwidth while maintaining the necessary information content.
5. Comparison with Frequency Modulation (FM):
 Phase modulation is closely related to frequency modulation (FM). In FM, the frequency of the
carrier wave is varied in response to the modulating signal, and the bandwidth calculation is
different.
 In FM, the bandwidth is directly proportional to the frequency deviation of the carrier due to
modulation. It is given by Carson's rule:
𝐵𝐹𝑀 ≈ 2(∆𝑓 + 𝑊𝑚 )
Where:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 24

 𝐵𝐹𝑀 is the bandwidth of the FM signal.

 ∆𝑓 is the frequency deviation.
6. Trade-offs:
 In practice, the choice between PM and FM depends on specific system requirements and trade-
offs. FM is often preferred when signal-to-noise ratio (SNR) and immunity to amplitude variations
are more critical, while PM is favored for certain applications where bandwidth efficiency is
important.
In summary, the bandwidth of a phase-modulated signal is determined by the modulation index and the
bandwidth of the modulating signal. By controlling these parameters, you can adjust the bandwidth of
the phase-modulated signal to suit the requirements of the communication system.
Generation of FM/PM
Frequency Modulation (FM) and Phase Modulation (PM) are two common methods of analog modulation
used in communication systems to convey information by varying the carrier signal. Generation of
Frequency Modulation (FM):
1. Carrier Signal: Start with a high-frequency carrier signal, typically a sinusoidal waveform. The
frequency of this carrier signal is denoted as 𝑓𝑐 , and it's usually in the radio frequency (RF) range.
2. Modulating Signal: You need a modulating signal that contains the information you want to transmit.
This signal can be an audio waveform, a digital data stream, or any other source of information. Let's
denote the modulating signal as 𝑚(𝑡).
3. Frequency Deviation: Determine the maximum frequency deviation, denoted as ∆𝑓. This parameter
is related to the extent to which the carrier frequency will change based on the modulating signal. It
is typically given in Hertz (Hz).
4. FM Signal Generation:
 Use a voltage-controlled oscillator (VCO) or a phase-locked loop (PLL) to create the FM signal.
 The output of the VCO or the PLL is your FM signal, and it's given by:
𝑓𝐹𝑀 (𝑡) = 𝑓𝑐 + 𝐾𝑓 . 𝑚(𝑡)
Where:
 𝑓𝐹𝑀 (𝑡) is the instantaneous frequency of the FM signal at time 𝑡.
 𝐾𝑓 is the frequency sensitivity or modulation index (sometimes denoted as 𝛽, which determines
the sensitivity of the carrier frequency to changes in the modulating signal.
5. Frequency Deviation Control: Adjust the modulation index (𝐾𝑓 ) to control the amount of frequency
deviation and, consequently, the bandwidth of the FM signal.
Generation of Phase Modulation (PM):
1. Carrier Signal: As with FM, start with a high-frequency carrier signal with frequency 𝑓𝑐 .
2. Modulating Signal: Have your modulating signal, denoted as 𝑚(𝑡), ready.
3. Phase Modulation:
 Phase modulation is generated by directly varying the phase of the carrier signal in response to the
modulating signal. The phase-modulated signal: 𝜃(𝑡) can be expressed as: 𝜃(𝑡) = 𝜃𝑐 + 𝐾𝑝 ⋅ 𝑚(𝑡)
Where:
 𝜃(𝑡) is the instantaneous phase of the PM signal at time t.
 𝜃𝑐 is the carrier phase.
 𝐾𝑝 is the phase sensitivity or modulation index, which determines the sensitivity of the phase to
changes in the modulating signal.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 25

4. Phase Modulation Control: Adjust the modulation index (𝐾𝑝) to control the extent of phase
variation, and by extension, the bandwidth of the PM signal.
5. Conversion to Complex Signal (Optional): In some cases, you may want to convert the phase-
modulated signal to a complex signal using a mixer or a similar technique, where the real part of
the complex signal represents the PM signal's instantaneous amplitude, and the imaginary part
represents its phase.
It's important to note that while FM varies the frequency of the carrier, PM directly manipulates the
carrier's phase. The choice between FM and PM depends on the specific requirements of the
communication system and the characteristics of the information signal being transmitted
Demodulation of FM/PM
Demodulation of Frequency Modulation (FM) and Phase Modulation (PM) is the process of recovering the
original modulating signal from the modulated carrier signal. Demodulation is essential in communication
systems to extract information accurately.
Demodulation of Frequency Modulation (FM):
Demodulating an FM signal involves extracting the original modulating signal (often an audio signal) from
the carrier signal. There are several methods to achieve this, with the most common one being the
frequency discriminator method.
1. Frequency Discriminator Method:
 A frequency discriminator is used to demodulate FM signals. This method takes advantage of the fact
that the frequency of the FM signal is directly proportional to the instantaneous phase of the
modulating signal.
 A simple frequency discriminator circuit can be implemented using a resonant tank circuit (an LC
circuit) tuned to the carrier frequency, as shown in Figure 1.

 As the FM signal passes through the resonant tank circuit, it causes the tank circuit's resonance
frequency to vary, depending on the instantaneous frequency of the FM signal.
 The output of the tank circuit contains the demodulated signal, which is the original modulating signal.
2. Phase-Locked Loop (PLL) Demodulation:
 Another method for FM demodulation is the use of a Phase-Locked Loop (PLL). The PLL can be used
to track and reproduce the modulating signal by controlling the frequency of a voltage-controlled
oscillator (VCO). This is a more sophisticated approach and is often used in practice.
 Figure 2 illustrates a basic block diagram of a PLL-based FM demodulator.

 In this setup, the PLL's VCO is adjusted to track the instantaneous frequency of the FM signal, and
the control voltage generated by the PLL represents the demodulated signal.
3. Zero-Crossing Detector:
 A simpler approach to FM demodulation is to use a zero-crossing detector, which can convert the
FM signal into a pulse train. The pulse train carries the information, and it can be filtered to obtain
the modulating signal.
Demodulation of Phase Modulation (PM):
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 26

Demodulating a PM signal is the process of extracting the original modulating signal from the phase-
modulated carrier. There are various methods to demodulate PM signals, with the most common being
the phase-locked loop (PLL) and the discriminator methods.
1. Phase-Locked Loop (PLL) Demodulation:
 A PLL can be used for PM demodulation by having the VCO track the carrier phase. The error signal
generated by the PLL is proportional to the modulating signal's amplitude.
 Figure 3 shows a block diagram of a PLL-based PM demodulator.

 A phase detector is nothing but a comparator here. It performs a comparison of two frequency
component fed at its input and generates a dc voltage. This generated voltage is proportional to the
difference in phase of the two frequencies.

 Usually, the frequency applied to the PLL is digital frequencies.

 Now as we can clearly see in the figure shown above that through a feedback path the output of the
VCO is provided to the phase detector. This fed back signal acts as the second input of the
comparator.

 The phase detector performs a comparison of the frequency of actually applied digital input signal
with the frequency of feedback signal. The output generated is a dc voltage (also called error
voltage, Ve) whose amplitude is proportional to the phase difference of the two signals applied at
the input.

 Low-pass filter: The output of the phase detector is provided to a low pass filter, that eliminates the
high-frequency component and noise from the output of the comparator.
 The phase detector gives the sum and difference frequency of two input signals as its output.
 The sum component of two frequencies (i.e., fi + fo) is a high-frequency component thus is
eliminated by the LPF. While the difference (i.e., fi – fo) is a low-frequency component which is
passed by the filter.
 This low-frequency dc voltage signal is then provided to a dc amplifier which amplifies the signal
level. This amplified signal is then provided to the VCO.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 27

Input and output waveform Commented [FO1]:

 Basically the PLL tries to adjust the frequency of the output signal. Thus this adjustment process
includes 3 major stages, which are free-running, capture and phase lock stage.

 When the difference of frequencies of the two inputs is 0, showing a constant phase difference then
it is said that the loop is locked.

 In case there exist a phase shift of 180⁰ between the two signals, then the output voltage will be
maximum.

 In the absence of an input signal, the generated output voltage will be zero, allowing the VCO to
operate at a set frequency. This frequency is known as the free-running frequency of the oscillator.
 The error signal generated by the PLL represents the modulating signal, and it can be used as the
demodulated output.
2. Phase Discriminator Method:
 Another method for PM demodulation is the phase discriminator. This method directly measures
the phase difference between the incoming PM signal and a locally generated carrier.
 Figure 4 illustrates the basic concept of a phase discriminator.

 The phase discrimination circuit delays the modulating signal by 90 degree, upconverts the signal and
its delayed version by CK(t) and CK(t − 1 4fCK ) respectively, then sums up the mixed signals. The
mixing of the modulating signal with the carriers are performed by balanced mixers.
 The output of the phase discriminator is proportional to the modulating signal, making it suitable for
PM demodulation.
 Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 28

In both FM and PM demodulation, the recovered signal may require additional filtering and processing to
obtain the original modulating signal. The choice of demodulation method depends on the specific
application and the characteristics of the modulated signal.
Signal Noise: Mathematical Representation
Noise in a signal is typically represented mathematically as an additive component that introduces random
variations or disturbances to the signal. Common mathematical representations of noise include white
noise, Gaussian noise, and other statistical models. Here are some of the key mathematical
representations of signal noise:
1. White Noise:
 White noise is a type of noise where the amplitude of the noise is constant across all frequencies. It is
characterized by statistical properties, such as zero mean and constant variance.
 In the time domain, white noise can be represented as a sequence of uncorrelated random variables
with a probability density function (PDF) that is uniform over a specified range.
 Mathematically, white noise can be represented as: 𝑛(𝑡) = 𝐴 ⋅ 𝑤(𝑡)
where:
 𝑛(𝑡)is the noisy signal.
 A is the amplitude of the noise.
 𝑤(𝑡) is a random process that is often modeled as a zero-mean Gaussian white noise process.
2. Gaussian Noise:
 Gaussian noise is a type of noise where the probability density function of the noise values follows a
Gaussian distribution (also known as the normal distribution).
 In the time domain, Gaussian noise is represented as a random process with a Gaussian PDF.
 Mathematically, Gaussian noise can be represented as:
𝑛(𝑡) = 𝐴 ⋅ 𝑁(0, 𝜎 2 )
where:
 𝑛(𝑡) is the noisy signal.
 𝐴 is the amplitude of the noise.
 𝑁(0, 𝜎 2 ) represents a Gaussian distribution with a mean of 0 and a variance of 𝜎 2 .
3. Impulse Noise:
 Impulse noise, also known as "spike noise" or "salt-and-pepper noise," is characterized by the
presence of occasional spikes or impulsive disturbances in the signal.
 In the time domain, impulse noise can be represented as a sequence of random impulses
superimposed on the original signal.
4. Colored Noise:
 Colored noise is a type of noise where the amplitude of the noise varies with frequency. Unlike
white noise, it does not have a constant power spectral density (PSD).
 Colored noise is often characterized by a specific PSD, such as pink noise (1/f noise) or brown
noise (random walk noise).
 Mathematically, colored noise can be represented by specifying its power spectral density in the
frequency domain.
5. Additive Noise Model:
 In practice, noise is often modeled as an additive component that is superimposed on the clean
signal. This is a common way to mathematically represent the presence of noise in a signal.
 Mathematically, the noisy signal can be represented as:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 29

𝑥(𝑡) = 𝑠(𝑡) + 𝑛(𝑡)

where:
 𝑥(𝑡) is the noisy signal.
 𝑠(𝑡) is the clean signal (the signal of interest).
 𝑛(𝑡) is the noise component, which can be represented using one of the noise models described
above.
The choice of noise model depends on the specific characteristics of the noise in a given signal and the
modeling assumptions made for analysis and processing. In many real-world scenarios, noise is a complex
mixture of various noise sources and may require more sophisticated statistical models for accurate
representation and analysis.
Signal to Noise Ratio
The Signal-to-Noise Ratio (SNR) is a fundamental concept in signal processing and communications. It is a
measure of the relative power or strength of a desired signal compared to the background noise in a given
communication or measurement system. A higher SNR indicates a stronger and more reliable signal, while
a lower SNR indicates a weaker signal relative to the noise. SNR is typically expressed in decibels (dB) for
convenience. The SNR can be calculated as follows:
𝑺𝑵𝑹 (𝒊𝒏 𝒅𝑩) = 𝟏𝟎 ∗ 𝒍𝒐𝒈𝟏𝟎(𝑷𝒔 / 𝑷𝒏 )
Where:
 𝑆𝑁𝑅 (𝑖𝑛 𝑑𝐵) is the Signal-to-Noise Ratio in decibels.
 𝑃𝑠 is the power of the signal (the desired signal component).
 𝑃𝑛 is the power of the noise (unwanted background noise).
Here are some key points about SNR:
1. Importance of SNR:
 A high SNR is desirable in communication systems because it means that the signal is strong
compared to the noise, leading to better signal quality and less likelihood of errors in data
transmission.
 A low SNR can result in signal degradation, making it difficult to distinguish the signal from the
noise, and can lead to data errors.
2. Measurement of SNR:
 To measure SNR in practice, you would typically take power measurements of the signal and the
noise components. These measurements are usually performed in the frequency domain.
3. SNR in Digital Systems:
 In digital communication systems, SNR plays a critical role in determining the bit error rate (BER). A
higher SNR generally leads to a lower BER, which is important for reliable data transmission.
4. Application in Analog Systems:
 In analog systems, SNR is used to describe the quality of an analog signal, such as an audio or video
signal. A high SNR in audio systems means clear and low-noise sound, while a high SNR in video
systems means clear and sharp images.
5. Dynamic Range:
 SNR is also related to the dynamic range of a system. The dynamic range represents the range
between the weakest and strongest signals that can be accommodated without distortion. A higher
SNR results in a larger dynamic range.
6. Trade-offs:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 30

 In many real-world scenarios, achieving a very high SNR may not be feasible or cost-effective.
Engineers must balance SNR with other system constraints, such as bandwidth and power
consumption.
7. Improving SNR:
 There are several techniques to improve SNR, including signal processing methods, better analog-to-
digital converters (ADCs), noise reduction methods, and the use of more efficient antennas or
transmitters in communication systems.
SNR is a crucial metric in the design and evaluation of communication systems, audio systems, image
processing, and many other applications where signal quality is of paramount importance. It helps
engineers and researchers assess the performance and reliability of such systems in the presence of noise.
Noise in AM, FM, and PM
Noise is an inherent part of any communication system, and it can affect signals modulated using various
techniques, including Amplitude Modulation (AM), Frequency Modulation (FM), and Phase Modulation
(PM). Here's a brief overview of how noise impacts these modulation schemes:
1. Noise in AM (Amplitude Modulation) Systems:
In AM systems, the noise typically affects the amplitude of the modulated signal. This noise can be due to
various sources, such as electromagnetic interference, thermal noise, and quantization noise (in digital
AM systems). The impact of noise in AM systems includes:
 Amplitude Noise: External noise sources can alter the amplitude of the modulated signal. As a result,
the received signal can suffer from variations in amplitude, leading to distortion and reduced signal
quality.
 Signal-to-Noise Ratio (SNR): The SNR of an AM signal is critical. A higher SNR indicates a stronger
signal relative to the noise. A lower SNR can result in distorted audio or poor reception in AM radio
broadcasts, for example.
 Demodulation: AM demodulation is relatively straightforward, but noise can lead to distortion. In the
presence of significant noise, the demodulated signal may require filtering and post-processing to
recover the original message signal.
2. Noise in FM (Frequency Modulation) Systems:
In FM systems, noise primarily affects the phase of the modulated signal, which, in turn, can impact the
frequency. The key points related to noise in FM systems are:
 Phase Noise: Noise sources, such as thermal noise and interference, can introduce phase variations
in the FM signal. These phase variations lead to frequency deviations, affecting the quality of the
received signal.
 Threshold Effect: FM systems exhibit a threshold effect, meaning that small variations in signal
strength have a minimal impact on signal quality. However, once the noise becomes significant enough,
it can cause severe distortion.
 Bandwidth and Noise Trade-off: FM systems often require a larger bandwidth to maintain signal
quality, especially in the presence of noise. A larger bandwidth allows for greater frequency deviations
and, thus, better noise immunity.
 SNR Considerations: FM systems often require a higher SNR compared to AM systems for similar
signal quality, making them more resilient to noise but requiring cleaner reception.
3. Noise in PM (Phase Modulation) Systems:
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 31

Noise in PM systems primarily impacts the phase of the modulated signal. The following points are
relevant to PM systems:
 Phase Variations: Noise sources introduce phase variations in the PM signal. These phase variations
can affect the timing and synchronization of the signal.
 Demodulation Challenges: Demodulating PM signals in the presence of noise can be challenging. The
receiver must accurately track the carrier phase to recover the modulating signal. Noise-induced phase
errors can lead to signal distortion.
 SNR and Phase Noise: PM systems are sensitive to phase noise. The SNR, especially the phase
component of the SNR, plays a crucial role in the demodulation process.
 Carrier Recovery: PM demodulation often involves carrier recovery techniques to mitigate the impact
of phase noise and improve the quality of the recovered signal.
In all these modulation schemes, the noise performance is critical for determining the quality and
reliability of signal transmission. Engineers often use various techniques, such as filtering, error correction
coding, and noise reduction methods, to mitigate the effects of noise and improve the overall
performance of the communication system.

Pulse Modulation
Pulse modulation is a type of modulation in which the signal is transmitted in the form of pulses.
It can be used to transmit analogue information. In pulse modulation, continuous signals are
sampled at regular intervals.

Sampling
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 32

A continuous time varying signal can be represented into its samples form and can be recovered
back when sampling frequency 𝑓𝑠 is greater than or equal to the twice the highest frequency
component of message signal. 𝑓𝑠 ≥ 2𝑓𝑚
Types of Sampling techniques:
 Ideal – An impulse at each sampling instant.
 Natural – A pulse of short width with varying amplitude.
 Flat Top – Uses sample and hold, like natural but with single amplitude value

Quantization
The quantizing of an analog signal is done by discretizing the signal with a number of quantization
levels. Quantization is representing the sampled values of the amplitude by a finite set of levels,
which means converting a continuous-amplitude sample into a discrete-time signal. The discrete
amplitudes of the quantized output are called as representation levels or reconstruction levels.
The spacing between the two adjacent representation levels is called quantum or step-size.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 33

Here we are taking 2 bits i.e. 4 levels in quantization.

𝑋𝑚𝑎𝑥 − 𝑋𝑚𝑖𝑛
𝑄𝑢𝑎𝑛𝑡𝑖𝑧𝑒𝑑 𝑠𝑡𝑒𝑝 𝑠𝑖𝑧𝑒𝑑 =
2𝑛
Where 𝑛 is number of bits. So
1 − (−1)
𝑄𝑢𝑎𝑛𝑡𝑖𝑧𝑒𝑑 𝑠𝑡𝑒𝑝 𝑠𝑖𝑧𝑒𝑑 = = 0.5
22
Types of Quantization: There are two types of Quantization
1. Uniform Quantization
2. Non-uniform Quantization.
Uniform Quantization: In uniform quantization, the step size is fixed.
Types of Uniform Quantization
a) Mid-Rise type
b) Mid-Tread type
a) Mid-Rise type: The Mid-Rise type is so called because the origin lies in the middle of a
raising part of the stair-case like graph.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 34

b) Mid-Tread type: The Mid-tread type is so called because the origin lies in the middle of a
tread of the stair-case like graph.

Non-uniform Quantization: In non-uniform quantization, the step size is not fixed. It varies
according to certain law or as per input signal amplitude. The following fig shows the
characteristics

There are two forms of Pulse Modulation: Analog Pulse Modulation and Digital Pulse Modulation.
1. Analog Pulse Modulation
Analog Pulse Modulation techniques
 Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 35

a) Pulse Amplitude Modulation (PAM)

It is the simplest form of pulse modulation. In this type of modulation, each sample is made
proportional to the amplitude of the signal at the instant of sampling. The PAM signal follows the
amplitude of the original signal, as the signal traces out the path of the whole wave. Here, a signal
which is sampled at the Nyquist rate can be reconstructed by passing it through an efficient Low
Pass Filter (LPF) with an exact cutoff frequency. It is very easy to generate and demodulate PAM.
This technique transmits the data by encoding the amplitude of a series of signal pulses.
There are two types of PAM:
i). Single Polarity PAM: A fixed DC level is added to the signal so that the signal is always positive.
ii). Double Polarity PAM: Here, the pulses are both positive and negative.
PAM is illustrated in the figure below:

From the figure, it is clear that the pulse amplitude modulated signal follows the amplitude of the
message signal.
Advantages of PAM
Both modulation and demodulation are simple.
Easy construction of transmitter and receiver circuits.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 36

Disadvantages of PAM
A large bandwidth is required for transmission.
More noise.
Here, the amplitude varies. Therefore, the power required will be more.
Applications of PAM
Mainly used in ethernet communication.
Many microcontrollers use this technique in order to generate control signals.
It is used in photo-biology.
It acts as an electronic driver for LED circuits.
b) Pulse Width Modulation (PWM)
Pulse width modulation is also known as pulse duration modulation (PDM). Here, as the name
suggests, the width of the pulse is varied in proportion to the amplitude of the signal. Since the
width is changing, the power loss can be reduced when compared to PAM signals. In this
modulation, the amplitude of the signal is constant. Amplitude limiters are used to achieve this
requirement. Since clipping of amplitude at desired levels take place, this modulation produces
less noise.
c) Pulse Position Modulation (PPM)
In this type of modulation, both the amplitude and width of the pulse are kept constant, but we
change the position of each pulse with reference to a particular pulse. Here, a single pulse is
transmitted with the required number of phase shifts. So, we can say that pulse position
modulation is an analogue modulation scheme where the amplitude and width of the pulse are
kept constant, while the position of the pulse with respect to the position of a reference pulse is
varied according to the instantaneous value of the message signal.
PPM can be obtained from PWM. This is done by getting rid of the leading edge and bodies of
PWM pulses. The main advantage of pulse position modulation is that it requires constant
transmitter power output, while the major disadvantage is that it depends upon transmitter-
receiver synchronization.
The waveforms of PPM are given below:

Advantages of PPM
As it has constant amplitude, noise interference is less.
We can easily separate a signal from a noisy signal.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 37

Among all three types, it has the most power efficiency.

It requires less power when compared to pulse amplitude modulation.
Disadvantages of PPM
The system is highly complex.
The system requires more bandwidth.
Applications of PPM
It is used in the air traffic control system and telecommunication systems.
Remote-controlled cars, planes, and trains use pulse code modulations.
It is used to compress data, and hence it is used for storage.
2. Digital Pulse Modulation
a) Pulse Code Modulation (PCM)
b) Delta Modulation (DM) – (We will not cover this one in class)
Pulse Code Modulation (PCM)
This type of modulation is different from all modulations learnt so far. It is clear from the block
diagram given at the top that it is a type of digital modulation. That is, the signals here are
sampled and sent in pulse form. A common feature among other techniques is that pulse code
modulation also uses the sampling technique. In this case, instead of sending a pulse train which
is capable of continuously varying parameters, this type of generator produces a series of
numbers or digits. Each digit in it represents the appropriate length of the sample at a particular
instant.
Advantages of PCM
It is mainly used in long distant communication.
Transmitter efficiency is more.
It has higher noise immunity when compared to other methods.
Disadvantages of PCM
More bandwidth is required when compared to analogue systems.
In this method, encoding, decoding and quantization of the circuit have to be done. This makes
it more complex.
Applications of PCM
It is used in the satellite transmission system.
It is also used in space communication.
It is used in telephony.
One of the recent applications is the compact disc.
Quantization Error
Quantization error is the difference between the analog signal and the closest available digital
value at each sampling instant from the A/D converter Quantization error also introduces noise,
called quantization noise. The difference between an input value and its quantized value is called
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 38

a Quantization Error. A Quantizer is a logarithmic function that performs Quantization (rounding

off the value). An analog-to- digital converter (ADC) works as a quantizer.
This can be explained further as when an information signal is pulse amplitude modulated, it
becomes discrete in time only. It remains analogue in amplitudes since all the values within the
specified range are allowed. PAM signal is said to be quantized when each pulse of the PAM signal
is adjusted in amplitude to coincide with the nearest level within a finite set. It is clear from the
figure below, quantization error (noise) can be reduced by increasing the number of quantization
levels (𝐿), i.e. decreasing the intervals (𝑞) between the levels. For any system, during its
functioning, there is always a difference in the values of its input and output.

Quantization Noise It is a type of quantization error, which usually occurs in analog audio
signal, while quantizing it to digital. For example, in music, the signals keep changing
continuously, where a regularity is not found in errors. Such errors create a wideband noise
called as Quantization Noise.
Signal to Quantization Noise Ratio
Signal power and quantization noise should be discussed in unison. Most importantly, the
signal power should be sufficiently more than the power of the quantization noise. The
signal-to-quantization noise ratio (SNQR) is used to equate this ratio.
As discussed in the previous section, quantization noise is added when quantization is used
to digitize a signal. Some level of error is introduced with this method of analog-to-digital
conversion. The SQNR is used as a quick error metric in regard to quantization noise.
A higher SQNR value indicates a lower value of quantization noise.
2.Digital Pulse Modulation
Pulse Code Modulation
Pulse Code Modulation (PCM) is a digital modulation technique. A signal is pulse code modulated to
convert its analog information into a binary sequence, i.e., 1s and 0s. The output of a PCM will resemble
a binary sequence. The following figure shows an example of PCM output with respect to instantaneous
values of a given sine wave.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 39

Instead of a pulse train, PCM produces a series of numbers or digits, and hence this process is called as
digital. Each one of these digits, though in binary code, represent the approximate amplitude of the signal
sample at that instant. In Pulse Code Modulation, the message signal is represented by a sequence of
coded pulses. This message signal is achieved by representing the signal in discrete form in both time and
amplitude.
Basic Elements of PCM The transmitter section of a Pulse Code Modulator circuit consists of Sampling,
Quantizing and Encoding, which are performed in the analog-to-digital converter section. The low pass
filter prior to sampling prevents aliasing of the message signal. The basic operations in the receiver section
are regeneration of impaired signals, decoding, and reconstruction of the quantized pulse train. Following
is the block diagram of PCM which represents the basic elements of both the transmitter and the receiver
sections.

➢ Low Pass Filter This filter eliminates the high frequency components present in the input analog signal
which is greater than the highest frequency of the message signal, to avoid aliasing of the message signal.
➢ Sampler This is the technique which helps to collect the sample data at instantaneous values of
message signal, so as to reconstruct the original signal. The sampling rate must be greater than twice the
highest frequency component W of the message signal, in accordance with the sampling theorem.
➢ Quantizer Quantizing is a process of reducing the excessive bits and confining the data. The sampled
output when given to Quantizer reduces the redundant bits and compresses the value.
➢ Encoder Encoder assigns code words to quantized sampled values. This coding techniques uses bits 0
and 1. If number of quantized levels are 16 then each sample is assigned with 4 bit code word.
Regenerative repeater: The PCM has an ability to control the distortion and noise caused by the
transmission of bits along the channel. This ability is accomplished by several regenerative repeaters
located at sufficient placing along channel. Regenerative repeaters have three functions. 1. Equalizing 2.
Timing circuits 3. Decision making device Equalizer shapes the received pulse so as to compensate
amplitude and phase distortion caused by the channel. Timing circuits provides periodic pulse trains.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 40

 Decision making device compares amplitude of equalized pulse plus noise to the pre-defined
threshold levels to make decisions whether the pulse is present or not.
 If the pulse is present (i.e. decision is yes), clean new pulse is generated and transmitted through
channel to next regenerative pulse. If the pulse is not present (i.e. the decision is no), it generates
clean base line to next regenerative repeater, provided the noise too large caused bit error by taking
the wrong decision

➢ Decoder Decoder reboots all the received bits to make more words then it decodes as quantized PAM
signals.
➢ Reconstruction Filter: All coded words are passed through low pass filter so that analog signal can be
reconstructed from quantized PAM signal.The cut off frequency of low pass filter is 𝑓𝑚 Hz which is equal
to band width of message signal. ➢ Destination It receives the signal from the reconstructive filter output
is analog signal.

Bit rate and bandwidth requirements of PCM :

➢ The bit rate of a PCM signal can be calculated form the number of bits per sample × the sampling rate.
𝐵𝑖𝑡 𝑟𝑎𝑡𝑒 = 𝑛𝑏 × 𝑓𝑠 The bandwidth required to transmit this signal depends on the type of line encoding
used.
➢ A digitized signal will always need more bandwidth than the original analog signal. Price we pay for
robustness and other features of digital transmission.
Important Relations
• Quantization Noise (𝑁𝑞) = ∆2 /2
•𝑆𝑖𝑔𝑛𝑎𝑙 𝑡𝑜 𝑁𝑜𝑖𝑠𝑒 𝑟𝑎𝑡𝑖𝑜 (𝑆𝑄𝑁𝑅) = 32.22𝑛 𝑜𝑟 𝑆𝑄𝑁𝑅 𝑖𝑛 𝑑𝐵 = 1.76 + 6.02𝑛 ≅ (1.8 + 6𝑛)𝑑𝐵
•𝐵𝑖𝑡 𝑟𝑎𝑡𝑒 = 𝑁𝑜. 𝑜𝑓 𝑏𝑖𝑡𝑠 𝑝𝑒𝑟 𝑠𝑎𝑚𝑝𝑙𝑒 × 𝑠𝑎𝑚𝑝𝑙𝑖𝑛𝑔 𝑟𝑎𝑡𝑒 = 𝑛𝑓𝑠
• 𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ 𝑓𝑜𝑟 𝑃𝐶𝑀 𝑠𝑖𝑔𝑛𝑎𝑙 = 𝑛. 𝑓𝑚 Where,
𝑛 – No. of bits in PCM code Fm – signal bandwidth 𝑓𝑠 – sampling rate
1. Delta Modulation (DM):
Definition: Delta Modulation is a type of analog-to-digital conversion technique where the analog signal
is approximated by a sequence of discrete values. It's a form of differential pulse-code modulation.
Delta Modulation is a form of pulse modulation where a sample value is represented as a single bit. This
is almost similar to differential PCM, as the transmitted bit is only one per sample just to indicate
whether the present sample is larger or smaller than the previous one. The encoding, decoding and
quantizing process become extremely simple but this system cannot handle rapidly varying samples.
This increases the quantizing noise.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 41

DM Wave for AC Input Signal

Operation:
 At each sampling instant, the difference between the current sample and the predicted value is
quantized.
 This quantized difference, often called the delta or error, is then transmitted.
 The receiver uses this information to reconstruct the signal by integrating the received delta
values.
Advantages:
 Simple implementation.
 Suitable for transmission over low-bandwidth channels.
Disadvantages:
 Can suffer from slope overload and granular noise issues.
 Limited signal-to-noise ratio.
2. Frequency Shift Keying (FSK):
Definition: Frequency Shift Keying is a modulation technique where the carrier signal frequency is
shifted between two or more discrete values based on the digital input.
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.
Following is the diagram for FSK modulated waveform along with its input.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 42

The output of a FSK modulated wave is high in frequency for a binary HIGH input and is low in frequency
for a binary LOW input. The binary 1s and 0s are called Mark and Space frequencies.

It is used in the voice frequency telegraph system and for wireless telegraphy in the high-frequency
bands.
Types:
 Binary FSK (BFSK): Two frequencies represent binary values (0 and 1).
 Multiple FSK (MFSK): More than two frequencies, allowing for the encoding of multiple bits per
symbol.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 43

Operation:
 The carrier frequency changes according to the input digital data.
 In BFSK, each bit corresponds to one of two frequencies.
 MFSK extends this concept to represent more than two symbols.
Applications:
 Commonly used in data communication systems, RFID, and wireless communication.
3. Phase Shift Keying (PSK):
Phase Shift Keying is a modulation technique where the phase of the carrier signal is varied based on the
digital input. The phase of the output signal gets shifted depending upon the input. These are mainly of
two types, namely BPSK and QPSK, according to the number of phase shifts. The other one is DPSK
which changes the phase according to the previous value.
Phase Shift Keying (PSK) is the digital modulation technique in which the phase of the carrier signal is
changed by varying the sine and cosine inputs at a particular time. PSK technique is widely used for
wireless LANs, bio-metric, contactless operations, along with RFID and Bluetooth communications.
PSK is of two types, depending upon the phases the signal gets shifted. They are −
Binary Phase Shift Keying (BPSK)
This is also called as 2-phase PSK (or) Phase Reversal Keying. In this technique, the sine wave carrier
takes two phase reversals such as 0° and 180°.
BPSK is basically a DSB-SC (Double Sideband Suppressed Carrier) modulation scheme, for message being
the digital information.
Following is the image of BPSK Modulated output wave along with its input.

Quadrature Phase Shift Keying (QPSK)

This is the phase shift keying technique, in which the sine wave carrier takes four phase reversals such as
0°, 90°, 180°, and 270°.
If this kind of techniques are further extended, PSK can be done by eight or sixteen values also,
depending upon the requirement. The following figure represents the QPSK waveform for two bits
input, which shows the modulated result for different instances of binary inputs.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 44

QPSK is a variation of BPSK, and it is also a DSB-SC (Double Sideband Suppressed Carrier) modulation
scheme, which send two bits of digital information at a time, called as bigits.
Instead of the conversion of digital bits into a series of digital stream, it converts them into bit-pairs. This
decreases the data bit rate to half, which allows space for the other users.
Differential Phase Shift Keying (DPSK)
In DPSK (Differential Phase Shift Keying) the phase of the modulated signal is shifted relative to the
previous signal element. No reference signal is considered here. The signal phase follows the high or low
state of the previous element. This DPSK technique doesn’t need a reference oscillator.
The following figure represents the model waveform of DPSK.

It is seen from the above figure that, if the data bit is LOW i.e., 0, then the phase of the signal is not
reversed, but is continued as it was. If the data is HIGH i.e., 1, then the phase of the signal is reversed, as
with NRZI, invert on 1 (a form of differential encoding).
If we observe the above waveform, we can say that the HIGH state represents an M in the modulating
signal and the LOW state represents a W in the modulating signal.
Applications:
 Used in various communication systems, including Wi-Fi, satellite communication, and digital
television.
Advantages of PSK:
 More bandwidth-efficient compared to FSK.
 Improved error performance compared to ASK (Amplitude Shift Keying).
Disadvantages:
 Susceptible to phase noise and synchronization issues.
Lecture Notes, EENG 438, Angle Modulation, Instructor: Mr. Chibole 45