TELE3113 NOTES

Analog & Digital Communications

 Chapter 1: Fourier Series & Fourier Transformations

Fourier Series & Definition

For any periodic function, , the function repeats itself for every , such that = .
i.e. a periodic function can be described by:

() = ( + )

Any periodic function can be described by an infinite sum of cosine and sine waves. This sum
is called the Fourier series of which can be described by the equation:

() = 0 + cos(0 ) + sin(0 )
=1

DC AC

Where and ( ) are the Fourier coefficients of the series to be solved.

Fourier Coefficients, 0 , and

The coefficients are defined as:

2
= ()cos(0 )
0

2
= ()sin(0 )
0

Where is the period of the signal function, . The DC component is defined as:

1
0 = ()
0

Complex Exponential Fourier Series

On the other hand, a signal, can be represented as the infinite sum of complex
exponentials. For all and 0 = 2/, we say:


() = 0
=

Where . The coefficient can be solved for by the following equations:

1
= () 0
0

Because is complex, it can be written in polar/phasor form. Form this, we can obtain the
magnitude spectrum (| | as a function of frequency) and phase spectrum ( as a function
of frequency) of the signal:

= | |

Example Question

Obtain the Fourier series expansion/representation of the following signal. Hence, obtain the
magnitude and phase spectrum of the signal, :

() = 1 + (0 )

Solution:

Convert any trigonometric terms to exponential terms via Eulers Formula:

0 0
() = 1 + [ ] = 1 + ( 0 ) ( 0 )
2 2 2

The signal has now been written as the complex exponential definition of the Fourier series
for 1 1 (() = =
0
). From here, we consider ONLY the coefficients of
each exponential term:

Comparing to the Fourier series definition, we assign the following coefficients, for 1
1:

1 = =
2 2

0 = 1

1 = =
2 2
Magnitude Spectrum:

The magnitude spectrum is obtained by taking the absolute value of all , 1 1.

|1 | = | |=
2 2

|0 | = 1

|1 | = | | = | | =
2 2 2

I.e. At = 0, we have a vertical line of length 1 from the origin. At = 20 , we have

two vertical lines of length 2 from the x-axis. If < 1, the spectra at = 20 are smaller
in magnitude. If > 1, the lines are larger than the line at = 0. This is shown below:

Phase Spectrum:

Again, we look the coefficients of the Fourier series representation. The goal is to express
each coefficient in terms of complex exponentials where possible:

1 = = = ( 2 )
2 2 2

0 = 1 = (0)

1 = = = ( 2 )
2 2 2

The phase spectra are shown below as a function of frequency. This is shown in the graph
below:

Fourier Transform
The Fourier transform is used to transform a function from the time () domain to the
frequency ( or ) domain. The transformation is usually written as:

() ()
Time to Frequency Domain (Fourier Transform, )

The Fourier Transform is defined, in terms of angular frequency (, rad/s) as:

() = [()] = ()

In terms of normal frequency (, Hz), we know = 2. Hence, the transform can be

defined alternatively as:

() = [()] = () 2

Frequency to Time Domain (Inverse Fourier Transform, 1)

The inverse Fourier transform is defined by the following:

1
() = 1 [()] = () +
2

In terms of normal frequency, the inverse transformation can be written as:

() = 1 [()] = () +2

For majority of Fourier transformation questions asked, the transform table is provided to
make it easier to perform the transformation than go through the algebra. The transform
table is shown on the page below
Example Questions

Find the Fourier transformation of the same signal:

() = 1 + (0 )

Solution:

() = [()]
= [1 + (0 )]
= [1] + [(0 )] ( )
= 2() + [( + 0 ) ( 0 )] ( )

Example Question:

Find the Fourier transformation of the following signal function shown below:

Solution:

The time differentiation property can be used. The signal can be written as a piecewise
function:

10, 1 < < 0

() = {
10, 0<<1

Taking the derivative, we obtain a sum of delta functions:

() = 10( + 1) 20() + 10( 1)

Taking the Fourier transform of both sides, using the differentiation property:

[ () ()] = () ()
Then:

()() = 10 20 + 10
 10( + ) 20
() =

20( + ) 20
() =
2

20(cos()) 20
() =

Hence, the final transformation of the signal function is given by:

20(cos() 1)
() =

Amplitude Modulation
Amplitude modulation is a technique of transmitting information using a carrier wave. The
reasons for the use of modulation is because:

1. Preventing Poor Reception: Transmitting a raw signal (most likely a radio wave)
means some frequencies will be attenuated while travelling through space, typically
the lower frequencies. This consequently means poorer reception at the receiving
end of the signal.

2. The use of a smaller antenna: The length of an antenna must be 1/10th of the length
of the wavelength of the signal being received. Hence, the signal to be received
should preferably have a high frequency (or small wavelength) to minimize the
antenna length. Modulation shifts the signal frequency to higher frequencies which
works in our favour.

3. The Wavelength Of The AM Signal: The amplitude modulated signal has a higher
frequency allowing it to travel a longer distance to the signal receiver.

Amplitude Modulation (Theory)

Suppose we are transmitting a message signal to a receiver. Let the message signal be
defined as (), such that:

() = cos( )

Let the carrier signal be defined as () such that:

() = cos( )

To modulate the signal, we are trying to send, the message signal varies about the mean
(average) of . This is can be written as: () = + (). The signal () is then
multiplied with the carrier sinusoid to give the following:

() = [ + ()] cos( )
= [1 + ()] cos( )

1
Where is the amplitude of the carrier wave (without modulation). The factor =

called the amplitude sensitivity factor of the modulator. The expression [1 + ()] is
the amplitude of the modulated carrier signal otherwise called the envelope of the AM
signal. = 2 , where is the carrier frequency.

This can be further expanded. Expanding the expression on the RHS, the modulated signal
can be written as:
 () = cos( ) + ()cos( )
= cos( ) + cos( )cos( )

We use the product-to-sum trigonometric identity:

1
cos() cos() = [cos( + ) + cos( )]
2

Setting = and = and multiplying both sides by the amplitude of the message
signal, we obtain:

cos( ) cos( ) = [cos{( + )} + cos{( )}]
2

Hence, the modulated signal can be written completely as:

() = cos( ) + [cos( + ) + cos( )]
2

AM Carrier Upper Side Lower Side

Wave Frequency Freq. Comp. Freq. Comp.

Spectrum of an AM Signal

Consider again the expression for an AM signal. We perform a Fourier transform on the
entire signal to consider the signal in the frequency () domain:

() = [1 + ()] cos( )

Let the message signal by sinusoidal i.e. define () = cos( ). Then, the AM signal
can be written as:

() = [1 + cos( )] cos( )
= [1 + cos( )] cos( )

Where = which is called the modulation index. Expanding the above equation
gives:

() = cos( ) + cos( )cos( )

= cos( ) + [cos{( + )} + cos{( )}]
2
Performing a Fourier Transform to both sides gives:

() = [( ) + ( + )]

+ [( + ( + )) + ( ( + ))]
2

+ [( + ( )) + ( ( ))]
2

Expressing = 2, we can rewrite the AM signal as:

() = [( ) + ( + )]
2

+ [( + ( + )) + ( ( + ))]
4

+ [( + ( )) + ( ( ))]
4

The Fourier transformed signal shows that the AM signal consists of three components:

The carrier signal

Two other sinusoids at a lower and higher frequency.

The sinusoid with frequency is the lower sideband of the AM signal. The sinusoid
with frequency + is the upper sideband of the AM signal. This highlights that:

Modulating signal shifts the frequency spectrum of the message signal

A larger frequency spectrum allows for different channels to access certain
frequencies of the AM signal.

The frequency spectrum of the AM signal is shown below:

Figure 1: Frequency Spectrum Of AM Signal Centered At = . About the carrier frequency are the upper sideband and
the lower sideband
 Figure 2: About each frequency of the carrier sinusoid are the upper sideband frequencies and the lower sideband
frequencies. These are the lower and upper bandpass bandwidths.

The specific regions where the spectra are located are the transmission (bandpass)
bandwidths of the signal. The bandpass spectra of the AM signal have a bandwidth that is
twice the frequency of the message signal, that is:

= 2

Further Details on Amplitude Modulation

Message Signal, ():

The signal to be transmitted was called the message signal, (). We assume the signal to
be a low-pass signal of bandwidth, . This is shown below:

Modulation Index, :
The constant, was defined earlier during the AM spectrum derivation to be:

This is called the modulation index. The modulation index indicates the amount of
modulation imposed on the message signal. Recall the equation for the AM signal:

() = [1 + cos( )]cos( )

The waveform for is shown in the following graph:

Recall that the amplitude of the modulated signal, otherwise called the envelope of the
signal:

= [1 + cos( )]

The amplitude of the modulated signal has a maximum and minimum that is dependent on
the cosine term in the envelope expression. Hence, we say:

= [1 + max(cos( ))] = [1 + ]

= [1 + min(cos( ))] = [1 ]

Solving both equations above for , the modulation index can be expressed as:

=
+

What does indicate?

The value of gives a percentage to which the signal has been modulated. Ideally
(theoretically), the signal should be able to achieve 100% modulation i.e. = 1. However,
this is usually not the case. Below are the AM waveforms for different values of and the
corresponding conditions for .

No Modulation 0% ( = ):

=
Under-Modulation 50% ( < ):

>

Complete Modulation 100% ( = ):

= 2
= 0

Over-Modulation 200% ( > ):

< 0
Causes envelope distortion; modulated signal has not been efficiently transmitted or
received.
To prevent over-modulation we require | ()| 1,
AM Transmission Efficiency
Generating, transmitting and modulating a message signal requires energy and hence,
power. We may want to determine the efficiency and average energy and power
consumed in an AM signal transmission.

Average Power of AM Transmission

Recall the equation for an AM signal which we will treat as a voltage that is continuous:

() = cos( ) + cos( + ) + cos( )
2 2

For power calculations, it is more useful to convert all voltages (amplitudes) to their RMS
values. Recall:

=
2

Then, we re-write the AM signal equation:

() = cos( ) + cos( + ) + cos( )
2 22 22

The power of each sinusoid in the AM signal is simply the coefficient of each sinusoid. We
2
use the equation = across a to show that:

2
( ) 2
Carrier Power: = 2
= 2

2
( ) 2 2
Upper Side-Band Power: = 22
=
8

2
( ) 2 2
Lower Side-Band Power: = 22
=
8

The total power used to transmit the AM signal is hence:

= + +
2 2 2 2 2
= + +
2 8 8
2 2 2
= (1 + + )
2 4 4
2 2 2
= (1 + ) = (1 + )
2 2 2
Modulation Efficiency,

To determine the efficiency of the modulation of the message signal, we use:

We can re-write this as:

+
=
+ +

+
= ( )
2
(1 + 2 )

2 2 2 2
+ 8
= 8
2
(1 + 2 )

2 2
= 4
2
(1 + 2 )

2 2
2 ( 2 )
=
2
(1 + 2 )

2
( 2 )
=
2
(1 + 2 )

2
=
2 + 2

The modulation efficiency for a single tone message (i.e. () = cos( )) is given by
which can be expressed as:
2
=
2 + 2
2
% = 100
2 + 2
Efficiency of AM in terms of

Assuming that over-modulation is not achieved, the maximum value of the

modulation index is = 1
This means:

12 1
= = = 33.3%
2 + 12 3

i.e. Only 1/3 of the power is contained in the sideband signals (what we intend)
Meanwhile 2/3 of the power is contained in the carrier which does not contain any
of the information being transmitted (does not contribute to the transfer of data).
This makes AM inefficient for signal transmission, especially with power-limited
applications.
Demodulation of AM Signal
At the receiving end of the signal, only () should be extracted with the carrier wave
component removed. This is called demodulation and there are two methods which are
used to achieve this:

Envelope Detection ( < 1)

Coherent Detection (Regardless of modulation index)

Envelope Detection

Envelope detection works by extracting the audio signal from the carrier wave. This is
achieved via the following circuit:

Recall that the AM signal is expressed as:

() = [1 + ()]cos( )

With reference to the above circuit, the envelope detector works as follows:

1. The AM signal from the previous stages is sent to the diode in the circuit via the
transformer

2. The diode is forward biased. This means the bottom (negative) half of the AM signal
is removed. This is shown below:
 3. The capacitor, in the form of a low-pass filter removes the frequency components of
the carrier wave which produces the audio signal intended which a reduced gain:

4. An amplifier is then used to amplify the signal to a more audible level for the receiving
end.

Advantages Disadvantages

Low frequency distortion, bad if m(t)

Very simple circuit
contains low frequency content

Low cost Poor sensitivity

Affected by selective fading

Envelope Detector That Removes DC Bias

The following circuit is an extension off of the first envelope detector utilizes a 2 nd capacitor
that removes the DC bias of the AM signal
 Coherent Detection

Coherent Detection uses a combination of product modulation and a low-pass filter to filter
out high-frequency components of the intended signal. The block diagram of its operation is
shown below:

Let the AM signal, to be demodulated, be the same equation:

() = [1 + ()]cos( )

Its operation can be described in the following steps:

1. AM signal is multiplied by a signal generated from a local oscillator. The signal must
be of the same frequency AND phase (synchronized) as the input AM signal. Let the
local oscillator signal be:

() = cos( )

2. The low-pass filter filters out the frequency component of the product modulated
signal as well as the DC component of the signal.

3. The demodulated signal is attained with a non-unity gain.

The coherent detection can be described mathematically. Let the demodulated signal be
(). The following series of equations describe the demodulation process:

() = () cos( ) ( / . )
= [1 + ()] cos 2 ( )
1
= [1 + ()] (1 + cos(2 ))
2 LPF filters frequency

LPF also filters out = [1 + ()](1 + cos(2 )) component of Signal
2
DC component
= [1 + ()] + [ + ()] ( )
2

= + ()
2

= () ( !)
2
Receiving the Signal
The theory behind amplitude modulation and detection methods (e.g. envelope detectors,
coherent detection) were explored. However, the theory behind receiving the signal is also a
process that needs to be explained.

How does a mixer work

Receiving signal requires the use of a mixer, a circuit module which essentially acts as a
product modulator (multiplier).

A mixer (or frequency mixer) is a non-linear electrical circuit which creates new frequencies
from two signals applied to it.

A mixer produces a new signal at the sum and difference of the original signal (typically the
carrier frequency and local oscillator frequency)

To understand this, consider a block diagram of two signals, () and a tuned signal from a
local oscillator, ().

The local oscillator is tuned to some frequency. Let the frequency be and let the signal
be () = ( ). Let us multiply the signals and observe the Fourier transform of
the signal:

() = () ()
= [() (0 )]

. . .

() = [() [( + 0 ) + ( 0 )]]
= [( + 0 ) + ( 0 )]

i.e. The mixer produces side-frequencies next to the frequency of the message signal.
If () = cos( ), we say that the passband of the signal is at = . Performing
the time-domain multiplication, we see how the side-frequencies form:

() = () ()
= cos( ) cos( )
= [cos( + ) + cos(0 ) ]

Heterodyning & Heterodyne Receiver

The action of the mixer on the two input signals is called heterodyning

The process with which the mixer produces a sum and difference of the frequencies of the
input signals is called heterodyning.

The process is non-linear (multiplication) and requires an intermediate (IF) frequency that is
fixed (This is required for the IF amplifier).

The entire process behind receiving the signal can be summarized with the following block
diagram.

The block diagram can be summarized via the following steps:

1. The AM signal is received at the antenna to be passed into the super-heterodyne

receiver.

2. The signal is then passed into an RF filter - The filter works by selecting a carrier
frequency within the AM freq. spectrum via tuning and amplifies the signal (It also
lets the LSB and USB pass through)

3. The local oscillator produces a signal = above the RF AM signal

frequency i.e. if the RF AM signal has a carrier frequency of 1000Hz, then the local
oscillator signal must have a frequency of 1000 + 455 = 1455kHz. In general:

= + | | =

4. The RF-AM Signal and local oscillator signal pass into mixer and are multiplied
The multiplication of two signals gives the sum and difference of the frequencies
that lie above and below the passband carrier frequency. In other words

= + ( ) 2455 = 1455 + 1000

= ( ) =

At the design level, the oscillator has a higher frequency than the RF-AM signal frequency.
i.e. We desire the signal with the difference in the oscillator and RF-AM frequency.

Image Frequency

Due to the non-linear nature of the mixer, we may also receive an undesired frequency. This
frequency is called the image frequency.

The image frequency is an undesired input frequency which is demodulated by

superheterodyne receivers along with the desired incoming signal. This results in two
stations being received at the same time, thus producing interference.

Suppose the following:

= /
=
= (455)
= ()

The desired signal has been calibrated so it is = 455 away from the local oscillator
frequency. So where is the image frequency?

The image frequency is also = 455 away from the local oscillator frequency but on
the opposite side with respect to the desired station.

Consider the following frequency spectra that represents the signal we wish to receive and
the location of the image frequency spectrum:
i.e. The image frequency is always twice of the IF away from the desired station. In an
equation, this can be expressed as:

= 2
(+) 0 >
( ) 0 <

High-Side Tuning VS Low-Side Tuning

Double Sideband Suppressed Carrier (DSB-SC)

DSB-SC is another method of modulation via multiplication. The carrier, () is multiplied by

the message signal (). This is shown in the block diagram below:

No Carrier Frequency Spectra

The use of a product modulator to multiply the carrier and message signal does NOT
produce a signal with a carrier frequency, spectra. Let the carrier signal be () =
cos( ) and the message signal be () = cos( ).

() = () ()
= cos( ) cos( )
= cos( )cos( )

1
Using the trigonometric identity: cos() cos() = 2 [cos( + ) + cos( )]

() = [cos( + ) + cos( )]
2

Upper-Side Lower-Side
Freq. Freq.
Taking the Fourier transform on both sides:

() = [( + ( + )) + ( ( + ))] + [( + ( ))
2 2
+ ( ( ))]

The Fourier transform shows that there is no frequency component for the carrier signal.
After all, the equation for () only contains cosines of the LSB and USB. The spectrum is
shown below:
Advantages of using DSB-SC:

Because there is no carrier frequency ( ) component in the spectra the power is

ONLY due to the sideband spectra
o No wastage of power
o 100% power efficient
But bandwidth of DSB-SC spectrum is the same as AM spectrum
o = 2

Alternate Method to Generate DSB-SC Signal

We may choose two utilize two AM modulator modules to generate two ordinary AM
signals. Our input will be two identical message signals:

() = cos( ) () = cos( )

i.e. one message signal is the negative

version of the other. Both message
signal are multiplied with a carrier
wave:

() = cos( ) ( )

To give the two following signals:

1 () = [1 + ()]cos( )
2 () = [1 m(t)]cos( )

The two signal are then passed into an adder to output the desired product modulated
signal:

() = 1 () 2 ()
= 2 ()cos( )
Demodulation via Coherent Detection

Coherent detection for a DSB-SC modulated signal is applied in the same fashion as the AM
signal. Consider, the following block diagram:

The steps to demodulate the DSB-SC signal are as follows:

1. Multiply the DSB-SC modulated signal () = ()cos( ) by a carrier () =

cos( ) (Amplitude is not too important). Note the condition:

The carrier wave () must have its phase and frequency synchronized to the phase and
frequency of the input signal to be demodulated.

This gives the following signal function:

() = () cos( ) cos( ) = () cos 2 ( )

1
= () [1 + cos(2 )]
2
1 1
= () + ()cos(2 )
2 2

2. A low-pass filter is then used to filter the high-frequency components of the signal.
This leaves us with the message signal scaled down by a factor of 1/2.

1
() = () + () cos(2 )
2

LPF filters this part of

()
What If Carrier Was Out Of Phase

Consider this time the input signal, () and the carrier, () with a phase offset as the
following:

() = () cos( )
() = cos( + )

i.e. the local carrier oscillator is not phase-synchronized with the input DSB-SC signal.
Multiplication gives us:

() = () cos( ) cos( + )
= () cos( ) [cos( ) cos() sin( )sin()]
.
.
.
1 1
() = () cos() + () cos() cos(2 ) ()sin()sin(2 )
2 2

LPF filters this part of

()

Hence, if the carrier has a phase offset, , the message signal () is scaled down further
by a factor of cos(). Depending on the phase offset, we get the following:

1
= 0 cos() = 1 () = () ()
2
= 90 cos() = 0 () = 0 ( )

The issue is that does not necessarily need to be constant. often varies with time (i.e.
()). This gives rise to a distortion in the signal.
Phase-Locked Loop (PLL)
When using coherent detection, the frequencies, and hence the phase, of both the carrier
signal and the signal to be demodulated must be frequency and phase synchronized.

This can be achieved by using a phase-locked loop (PLL) which consists of the following
components:

Reference Signal The carrier signal to be made synchronous to the modulated

signal to be demodulated cos[ + ()]
VCO (Voltage Controlled Oscillator) Produces a voltage signal that is
proportional to the input signal:
o () = + () ( . )
= . (/)
o () = 0 sin( + 0 ())
Phase detector Multiplier with a LPF built in
Loop Filter <Fill in the blank>

The block diagram below shows the function of the PLL:

The aim of the PLL is to minimize the phase difference either to a constant or 0. This is
achieved via the following procedure:

1. Both the carrier (reference) and signal from the VCO enter the phase detector. The
phase detector first multiplies the signals:

() = () ()
= cos[ + ()] 0 sin[ + 0 ()]
0
= [sin( + () + + 0 ()) + sin( + () 0 ())]
2
0 0
= [sin(2 + () + 0 ())] + [sin(() 0 ())]
2 2
LPF filters this part of
()

This leaves us with the error voltage which is in terms of the phase difference between the
carrier with the unknown phase and the VCO with the known configured phase:
 0
() = [sin(() 0 ())]
2

The phase error, the argument of the sinusoid is what we are concerned about:

() = () 0 ()

The carrier will be in phase with the signal to be demodulated by ensuring () is either a
constant or 0. Either case, if the phase difference is made constant, the frequencies are
synchronized.

This will be discussion in-depth in frequency modulation.

Quadrature Amplitude Modulation (QAM)

Quadrature Amplitude Modulation (QAM) is the use of two quadrature carriers (carriers
with same frequency and phase difference of 90o) to DSB-SC modulate an input signal

The block diagram for QA Modulation is shown below:

The I-channel (otherwise called the in-phase input) and the Q-channel (otherwise called the
quadrature channel) take in two signals:

1 ()
2 ()

The oscillator generates a cosine signal. One goes through a 90o phase shift. This gives rise
to two carriers for each channel:

() = ( )
() = cos(90 c t) = ( )

The carriers are then product modulated with the message signals and then passed into an
adder to give the final DSB-SC QAM signal:

() = 1 () cos( ) + 2 ()sin( )

Because it is still DSB-SC, there is still no carrier frequency spectra 100% transmission
efficiency still holds.
Performing a Fourier transformation on () allows us to see where the spectra lie. Note
that:

1
[cos( ) ()] = [( + ) + ( )]
2

[sin( ) ()] = [( + ) ( )]
2

Then, we can say:

[ ()] = [1 () cos( )] + [2 ()sin( )]

= [1 ( + ) + 1 ( )] + [2 ( + ) + 2 ( )]
2 2

This can be interpreted as the QAM signal having a real frequency spectra and an imaginary
frequency spectra. This is shown in the diagram below:

QAM Demodulation

The retrieval of the message signal from the QAM modulated signal can be achieved using
the block diagram process shown below:

Where the mixer is a product modulator with an LPF to filter the high-frequency carrier components of the
QAM input signal.

Recall that the QAM signal, () can be expressed as:

() = [1 () cos( ) + 2 () sin( )]

The demodulation process can be described in the following steps:

1. The QAM signal is split to two pathways that lead to their own mixers. A VCO is connected to
each mixer, with one of them connected to a unit that causes a 90o shift. Define:

() = 2cos( )

2. The first mixer multiplies with . After manipulating the trigonometric terms using
identities, LPF filters out terms with high-frequency components:

() = ()

= [1 () cos( ) + 2 () sin( )] 2 cos( )

= 2 1 () cos 2( ) + 2 2 ()sin( )cos( )

1
= 2 1 () [ (1 + cos(2 ))] + 2 () 2sin( )cos( )
2

= 1 ()[1 + cos(2 )] + 2 () sin(2 )

= () + 1 () cos(2 ) + 2 ()sin(2 )

THE LPF filters this part of ()

MESSAGE
The same process applies for the Q-output of the demodulator except the VCO signal used
to product modulate with the signal has been phase shifted by 90o. i.e.

() = 2 cos(90 c t) = 2sin(c t)

Hence, the Q-output signal would be:

= () 2 () cos(2 ) + 1 ()sin(2 )

THE LPF filters this part of ()

MESSAGE
SSB Modulation
DSB-SC Modulation is advantageous over AM for the fact it can suppress the carrier-
frequency component of the signal (which requires 66% of the power used to transmit the
signal). The spectra of a DSB-SC transmitted signal are shown below:

Single Sideband SC (SSB-SC) is another modulation method that further improves spectra
efficiency. The spectra for an SSC transmitted signal are shown below:

Some of the consequences of utilized SSB are:

Bandwidth of half of the DSB-SC and AM signals More power efficient

Effective when transmitting signals that are deficient of low/high-frequency content.
At the cost of more complex circuitry and difficult tuning at the receiver end.

SSB-SC modulation is not as simple as DSB-SC modulation and requires a technique called
Hilbert Transformation.
Hilbert Transform ()

The Hilbert Transform () aims to create an analytic signal, a signal with no negative
frequency components.

Notation for a Hilbert Transformed Signal & Reasons for

For a signal, (), the Hilbert Transform of is expressed with the following notation:

[(())]
() ()

Given that is Hilbert transformable, can be expressed as an analytic signal or as a

complex-valued function. Let the analytic version of be () which can be written as:

() = () + ()

Like any complex function, can be expressed in polar form. If we let:

() = [()]2 + [ ()]2
()
() = ( ) = tan1 ( )
()

Then, we can also express in polar (exponential) form as:

() = () ()

Where () is the instantaneous amplitude or envelope of and () is the instantaneous phase of

The reasons why we want to Hilbert Transform a signal are:

of a signal allows us to express a signal in terms of its envelope. We can see this in
in the graph below, where the envelope is () marked with a red outline:
 The instantaneous phase, () can be used to calculate the instantaneous
frequency, () through the following relationship:

1 ()
() =
2

Where did that come from? The phase is related to angular frequency/velocity by:

() =

But = 2() and in our case, () = (). Then:

2() =

1 ()
() =
2

Definition of The Hilbert Transform,

Let () be the signal subject to . Then in the time-domain is defined as:

1 ()
() = [()] =


1
= () ()

= () ()

Hence, the Hilbert Transform can also be defined as the linear impulse function:

1
() =

Fourier Transform of ()

Using a series of Fourier transform properties, we find that:

[()] = () = ()

1 < 0
Where () is the sign function can be defined as () = { 0 = 0 . Hence, we the
1 >0
frequency response of can be defined as:

< 0
() = { 0 = 0
>0
From, here we can say:

1 <0
|()| = {
1 >0

And, its phase response can be written as:

90 < 0
() = {
90 > 0

Because of this, we conclude two things from the magnitude and phase response of the
Hilbert Transform:

The Hilbert transform, is in fact a simple phase filter that performs phase shifts.
The Hilbert transform does not alter the magnitude of the signal to be transformed
The transform imposes a phase shift. There are two cases for this:
o 90
o 90

Block Diagram for SSB Modulation (Hartley Modulator)

Figure 3: The first instance of the -90o phase shifter performs a Hilbert Transformation on m(t). The second instance of the -
90o phase shifter is to produce a quadrature carrier signal to the original version from the VCO

Explanation:

1. The message, () is split, one for the channel, one for the channel.
2. The product modulator multiplies the message signal with the carrier from the
VCO:

() = ()cos( )

Prior to product modulator, , the first phase shifter performs a Hilbert

transformation on the message signal:
 [()]
()
()

Which is then product modulated with the carrier that has been phase shifted by 90o
to form a sine signal rather than a cosine signal. The carrier signal can also be phase
shifted by another 180o to give ( ). Hence, the channel signal is given by:

() =
()sin( )

3. Both and signals are then added together to form the SSB modulated signal:

() = ()cos( )
()sin( )

Why the need for ?

The determines whether the modulated signal corresponds to the lower or upper
sideband:

For the SSB modulated signal:

() = ()cos( )
()sin( )

The gives either the lower-sideband (+) or the upper-sideband () respectively.

SSBLSB Modulation Equations

Let the VCO signal be () = ( ) and let the message signal be single-toned and
be expressed as () = ( )

I - Channel

1. Product modulate VCO carrier and message signal:

() = cos( )cos( )

= [cos( + ) + cos( )]
2

Q - Channel

2. Hilbert Transform message signal and then product modulate with the carrier signal
which has been phase shifted by 90

cos( ) sin( )

() = sin( )sin( )

[cos( ) cos( + )]
2
SSB Modulated/Generated

3. Both I and Q signals are then added. Sinusoids corresponding to the upper sideband
cancel leaving only sinusoids with the lower frequency-spectrum corresponding to
the lower sideband

() = () + ()

= [cos( + ) + cos( ) + cos( ) cos( + )]
2

= [2cos( )]
2
= cos( )

SSBLSB Modulation Equations

I Channel: The same equation applies:

() = [cos( + ) + cos( )]
2

Q Channel: The oscillator performs a Hilbert shift first and then a further 180o shift

() = [cos( + ) cos( )]
2

SSB Modulated Signal:

() = () + ()
= cos( + )
SSB Demodulation

To demodulate the SSB modulated signal (which is just a cosine), it is enough coherent
detection can be used. The procedure is:

1. Multiply SSB signal with LO signal with carrier frequency,

2. Use a LPF to filter out high-frequency components and retrieve ()

Working Out

Let the SSB-SC wave be:

()sin( )
() = () cos( )

Product modulate with a LO that is frequency synchronized to ():

() = () cos( )
= [ () cos( )
() sin( )] cos( )
= () cos 2 ( ) + () sin( ) cos( )
() 1
= [1 + cos(2 )] () sin(2 )
2 2
() () ()
= + cos(2 ) sin(2 )
2 2 2

THE LPF filters this part of ()

MESSAGE
VSB Modulation
Vestigial Sideband or VSB is another method of modulation that gives rise to a frequency
spectra similar to SSB. An example of a VSB spectra, compared to an SSB spectra is shown
below:

Differences from VSB to SSB:

Carrier frequency is included

Instead of receiving just one sideband upon signal retrieval, one sideband + 25% of
the other sideband is received

Implications

Advantages Disadvantages

The presence of the carrier Less power efficient due to carrier

frequency means envelope
detection for demodulation can be
used
o Less complex circuitry is
required for demodulation
Low frequency components are
also present in message signal upon
retrieval
Reduction in bandwidth compared
to DSB and AM modulation. Almost
as efficient as SSB-SC
frequency
o Portion of transmission
power required to transmit
carrier frequency
Larger bandwidth compared to SSB
o Approximate larger by 25%
compared to SSB.

How to generate a VSB modulated signal?

1. Generate a DSB-SC modulated signal Gives rise to two sidebands

2. Pass DSB-SC signal into a sideband shaping filter to generate a VSB signal

The block diagram for VSB generation is

more relaxed and is shown to the right.
How does the sideband shaping filter work?

The action of the filter can be described by considering the spectra of the signal as it passes
through each stage of the block diagram. As a sidenote, the transfer function of the shaping
filter can be defined as:

() = ( ) ( )

Essentially acting as a gate function but with a slow gradual rise (i.e. slow attack)

Stage 1: Spectra of DSB-SC signal

Stage 2: VSB Shaping Filter Magnitude Spectra

Stage 3: Formation of VSB Modulated Signal

The spectra from stage 1 and 2 interact to form a spectra that follows the slop of the filter:
Angle Modulation Derivation

Angle modulation, of a signal () involves varying the frequency AND phase of a carrier
wave according to the message signal ().

For a general signal given by:

() = cos[ + ()]

We define the instantaneous phase by the argument of the cosine signal. This is denoted by
and is expressed as:

() = + ()
( () )

We can differentiate the instantaneous phase with respect to time to get the instantaneous
angular frequency,

() = = [ + ()] = + ()


( () = )

Phase Modulation (PM) Derivation

Phase modulation, is when the phase deviation of the carrier varies linearly with the
message signal, ()

In other words, can be written in two ways:

() = + () = + ()

() = ()
[/]
Frequency Modulation (FM) Derivation

Frequency modulation, is when the frequency deviation of the carrier varies linearly with
the message signal, ()

In other words, can be written in two ways:

() = + () = + ()
() = ()

By dividing both sides by 2, we attain an alternate definition for the instantaneous
frequency in terms of the message

1 1
() = [ + ()]
2 2

() = + ()

Integrating both sides with respect to time, we attain the instantaneous phase in terms of
frequency deviation:

() = [ + ()]
0 0

() = + ()

If we then substitute the phase back into our general signal, (), we attain the frequency
modulated version of () which we shall denote as (). We shall compare this also with
phase modulation. These final equations are shown on the next page:
Frequency Modulation (FM)

Frequency Modulation of ()

For a general signal, (), the frequency modulated signal is given by the following
equation:

() = [ + ()]

This shows that the FM signal is dependent on the integral of the message signal. In terms
of frequency rather than angular frequency, the FM signal can be expressed as:

() = [ + ()]

Where = . [ ] and = [ ]

Phase Modulation (PM)

Phase Modulation of ()

For a general signal, (), the phase modulated (PM) signal is given by the following
equation:

() = [ + ()]

The cosine of the signal is called the instantaneous phase and is expressed in the following
equation:

() = + () = + ()

Where = [ ]

Mutual Convertibility Between PM and FM

()| () = ()|()
Comparing Both FM and PM Modulation Equations:

Inst. Phase, () Inst. Freq, / = ()

1 ()
FM () + [ ]
2

PM 2 () = () + ()

Waveforms for AM, FM and PM

Form of Modulation & Description Waveform

The amplitude of the envelope [ +
()] changes with respect to the
message signal amplitude
Frequency of carrier remains
constant
The frequency of the carrier wave varies
with respect to the frequency of the
message signal
The amplitude of the FM signal
envelope is constant [ , ]
The zeroes (roots) of ()
correspond to the resting
frequency, of the FM signal
Maxima of () give rise to a high-
frequency section of the FM signal
( > 0 )
Minima of () give rise to a low-
frequency section of the FM signal
( < 0 )

The phase of the carrier wave varies with

respect to the phase changes from the
message signal
The modulated signal swaps its
phase in accordance to the zeroes
of ()
This is shown in the blue signal in
the diagram to the right