Modulation

In this chapter, we are going to learn basic principles of

modulation. Modulation can be defined as the process of impressing
information from a modulating signal onto another signal called the
carrier. The resulting signal is called the modulated signal (in general,
it is a bandpass signal).

Typical modulation system

The reverse process is called demodulation. When the modulator
and the demodulator are located in the same apparatus, the system is
called a MODEM (MOdulator, DEModulator).
When we do not use modulation, the system is called a "baseband
communication" system. At that time, the baseband signal is
transmitted directly.
Before we go on in the development of modulation theory, we
have first to answer the following question: Why modulate?

1
 Historically, modulation has been introduced in order to use
reasonably sized antennas in radio communication. We know that the
physical size of an antenna is a fraction of the wavelength.
c
The wavelength is λ = , where c = 3×108 m/s is the speed of
f
light, f is the frequency of the signal. So, if we want to transmit a
baseband signal of 3 kHz by radio, the required wavelength is 100 km.
It is evident that it is very hard to build an antenna having many
kilometers of length. If we can transfer the information to a bandpass
signal with a carrier of 30 MHz, we obtain a wavelength of 10 meters.
A quarter wave antenna will be 2.5 meters. This is much more
reasonable. Modulation is also used to make the information fit the
communication channel. Sophisticated modulation schemes are
commonly used nowadays to transmit information. Techniques like
OFDM, Trellis Coding, CDMA, etc. are commonly used in everyday
communication systems.
Distortionless Communication
If we consider the whole communication system from the
baseband source signal to the baseband destination signal, all
communication systems can be considered as baseband. In this case, a
good communication system must be "distortionless". This means that
the destination signal must be a scaled (and maybe delayed) replica of
the source signal. If x(t) is the source signal and y(t) is the destination
one, we must have:
y (t ) = kx(t − τ ) . k is a constant, τ is a time delay.

2
In the frequency domain, we obtain:
Y ( f ) = ke − j 2π f τ X ( f ) . This means that the overall communication
system must behave like a filter (LTI system) with a transfer function:
Y( f )
H( f ) = = ke − j 2π f τ
X(f )
So, distortionless communication implies that the amplitude
response |H(f)| must be constant and that the phase response Arg[H(f)]
must be a linear function of the frequency. This means that all
frequencies must be delayed by the same amount. If the transfer
between the input and output signal is linear and time invariant but
without satisfying the above conditions, we say that the
communication system is subjected to "linear distortion". This
distortion can come from the amplitude response which is not constant
or from the phase response which is not linearly related to frequency
(phase or delay distortion).
This type of distortion can be cured or minimized by using a
filter called an "equalizer" at the output of the communication channel.
When the transfer function between the input and output is nonlinear,
we are in presence of "nonlinear distortion".
Harmonic distortion:
When we apply a pure sinewave at a frequency f0 to a linear
system, the output will be a sinewave at the same frequency. However,
if the system is nonlinear, the output will be a periodic waveform at
the same frequency, but it will not be sinusoidal anymore. So, we
observe harmonics at the output.

3
 Let the input be x(t ) = A cos ω0t , the output will be
∞
y (t ) = c0 + ∑ an cos ( nω0 + θ n ) , an = 2 cn and θ n = Arg [ cn ] .
n =1

The total harmonic distortion coefficient measures how far the

signal y(t) is from a sinewave. It is evaluated as:
∞

∑a 2
n
d = 100 × n=2
%
a12
It is the ratio of the rms value of all the harmonics of the signal
y(t) over the rms value of the fundamental.
Classification of modulation systems.
Depending on the modulating signal, we distinguish two different
types of modulation systems:
• Digital modulation systems: they are used to transmit digital
information through physical channels.
• Analog modulation systems: the modulating signal in this case is
a baseband analog signal.
We can also classify modulation according to the type of carrier used
(and therefore the modulated wave produced).
• Continuous wave (CW) modulation: The carrier is a sinewave
and the modulated signal is a narrow bandpass signal.
• Pulse modulation: The carrier is a periodic train of pulses. The
modulated signal will carry information about samples of the
signal.
We are going to analyze first analog CW modulation.

4
Analog CW modulation.
The modulating signal sɶ (t ) is assumed to be bounded. This
means that there exists a peak value sɶ (t ) max such

that: sɶ (t ) ≤ sɶ (t ) max for all t. We can thus define a normalized signal

sɶ (t )
s (t ) = and we have s (t ) ≤ 1.
sɶ (t ) max

The signal is also assumed to have an average value of zero. This

means that there is no delta impulse at the origin in its spectrum. It is
also assumed to be bandlimited to a maximum frequency W. In other
words, if S ( f ) = F [ s (t )] then S ( f ) = 0 for f > W .
In CW modulation, the modulated signal x(t) is a narrow
bandpass signal. This means that it can be expressed in either
quadrature form:
x(t ) = a (t )cos ω0t − b(t )sin ω0t
or in modulus/phase form:
x(t ) = r (t )cos (ω0t + ϕ (t ) )

ω0 = 2πf0 being the carrier frequency. Depending on the

modulation method, the information (signal s(t)) can affect a(t), b(t),
r(t) or ϕ(t).
I. Linear modulations
We are going to study in this part modulation methods where the
quadrature components a(t) and b(t) are linearly dependent on the
baseband signal s(t). A linear modulation method must satisfy the

5
superposition principle. If x1(t) is produced by s1(t) and x2(t) is
produced by s2(t), then a1x1(t)+a2x2(t) is produced by a1s1(t)+a2s2(t).
Before proceeding in the analysis of the different types of linear
modulation, we are going to study an "almost linear" one: The
Amplitude Modulation (AM).
Amplitude Modulation (AM)
In AM, the information s(t) is carried by the modulus r(t) of the
signal x(t). Since we have the constraint that r(t) must remain positive
all the time, we cannot simply make it proportional to s(t). We have to
add a constant in order to satisfy the above constraint.
r (t ) = A0 + ka sɶ (t )

A0 is a positive dc signal added to make r(t) ≥ 0, ka is a

proportionality constant and sɶ (t ) is the unnormalized signal. The
phase of the carrier ϕ(t) is constant and we use the value of zero. If we
introduce the normalized signal s(t), we can re-express r(t) as:
r (t ) = A0 + ka s (t ) sɶ (t ) max = A0 (1 + ms (t ) )

ka sɶ (t ) max
m=
A0
m is called the modulation index.
Since r(t) must be positive, we see that we must have 0 ≤ m ≤ 1.
If it happens that m exceeds 1, we say that we have overmodulation.
The AM signal is:
x(t ) = A0 (1 + ms (t ) ) cos ω0t

6
 Historically, amplitude modulation is the first modulation system
put into practice. It was used essentially because of the simplicity of
the receiver structure. It is easily verified that AM is not linear since it
does not satisfy the superposition principle. It is as linear as the
function f ( x) = ax + b . This function is not linear however it is
incrementally linear, i.e. an increment of the input is linearly related to
an increment of the output.

1.5

trough of modulation
1

0.5

A1 0 A2

-0.5

-1

-1.5
0 100 200 300 400 500 600 700 800 900 1000
peak of modulation

Sinewave modulated waveform

The signal s(t) can be an energy or a power type of signal. In the

first case, we can compute easily the spectrum of the modulated signal
as a function of the baseband modulating one.

7
 Starting from x(t), we obtain
x(t ) = A0 (1 + ms (t ) ) cos ω0t = A0 cos ω0t + A0 ms (t )cos ω0t
giving
A0 A A A
X(f ) = δ ( f − f 0 ) + 0 δ ( f + f0 ) + 0 mS ( f − f 0 ) + 0 mS ( f + f 0 )
2 2 2 2
This relation is shown graphically below.
S(f)

−W W f

X(f)

A0
A0
δ(f + f0) δ(f − f0)
2 2
A0
A0
m S(f + f0) m S(f − f0)
2 2

−f0 f0 − W f0 f0 + W f

The above sketch shows the transformation from the baseband

signal s(t) to the bandpass signal x(t). We can also remark that if we
do not want to have a superposition of shifted spectra (aliasing), we
must have f 0 > W . It is also apparent that the bandwidth B of the
modulated signal is twice the bandwidth of the baseband signal.
B = 2W

8
 If we consider only the positive half of the spectrum, we remark
that the Hermitian symmetry of S(f) is translated to f0. So, the positive
half is composed of two halves: the upper sideband above the carrier
frequency and the lower sideband below the carrier frequency.
Power Computation
In order to analyze power signals, we may assume that s(t) is
periodic. We can start the analysis with the simplest real periodic
signal: the sinewave. So, let us assume that s (t ) = cos ωm t where

ωm < ω0.
x(t ) = A0 (1 + m cos ωmt ) cos ω0t = A0 cos ω0t + A0 m cos ωmt cos ω0t
Using trigonometric identities, we obtain;
A0 A
x(t ) = A0 cos ω0t + m cos (ω0 − ωm ) t  + 0 m cos (ω0 + ωm ) t 
2 2
The signal in this case is composed of 3 sinewaves: the carrier with
A0
amplitude A0 and the two sidebands with amplitude m each. The
2
spectrum consists of only Dirac impulse functions. A more general
case is the one of a bandlimited periodic signal. We can express s(t) as:
N
s (t ) = ∑ ak cos ( kωmt + θ k )
k =1

The signal has a zero dc value and the number of harmonics N is

W 
given by N =   . The notation   stands for the floor (i.e. the
 fm 
integer just below) of the number written inside. The modulated signal
is now given by:

9
 N
x(t ) = A0 cos ω0t + A0 m∑ ak cos ( kωmt + θ k ) cos ω0t
k =1

A0 N
x(t ) = A0 cos ω0t + m∑ ak cos (ω0 − kωm ) t + θ k 
2 k =1
A0 N
+ m∑ ak cos (ω0 + kωm ) t + θ k 
2 k =1
The above formula is general enough to allow us to compute the
power of the modulated signal. If we assume that the different
sinewaves are independent, the total power will be given by the sum
of the power of the different components.
A02 A02 2 N 2
Px = + 2 × m ∑ ak
2 8 k =1

A02
In the above relation, we can recognize the carrier power Pc = and
2
A02 2 N 2
the sideband power Psb = m ∑ ak . So, the total power of the signal
8 k =1

is: Px = Pc + 2 Psb . The sideband power can also be expressed as a

ak2N
function of the power of the normalized baseband signal Ps = ∑ ,
k =1 2

A02 2
i.e. Psb = m Ps . So, in terms of the carrier power and the sideband
4
power, we obtain:
A02 2 A02 A02 2
Px = Pc + m Ps = + m Ps
2 2 2

10
Given that the signal s(t) is normalized with a maximum value of 1, its
1 1
Tm ∫Tm ∫
power is less than 1 ( Ps = ≤ s (t ) max dt = 1 ). The
2 2
s (t ) dt
Tm Tm

modulation index m is also less or equal to 1. This means that the

power transmitted by the two sidebands is smaller than the power used
to transmit a carrier that conveys no information. More than 50% of
the total power is used to transmit the carrier. We can evaluate the
efficiency of the system using the following efficiency coefficient:
2 Psb m 2 Ps
η= =
Pc + 2 Psb 1 + m 2 Ps
η cannot exceed the value of 1/2.
1 m2
Example: if s (t ) = cos ωm t , Ps = , then η = . If we use m = 1,
2 2 + m2
1
the efficiency becomes η = . In general, AM systems use a
3
modulation index smaller than 1. The typical value is 30%. AM is
used to transmit audio signals that have an average power Ps << 1. In
this case, the efficiency drops to a very small value.
Broadcast AM radio use carrier frequencies in the medium waves
(530 kHz to 1.71 MHz) or in short waves (3 to 30 MHz). We can
encounter stations transmitting in long waves (148.5 kHz to
283.5 kHz). The bandwidth B allocated to every channel is 10 kHz.
This means that each sideband occupies a bandwidth W of 5 kHz.
AM production:
The following block diagram reproduces the defining equation of AM.

11
 sɶ (t )

dc A cos ω0t

Another structure can be derived by manipulating the defining

equation.
x(t ) = A0 (1 + ms (t ) ) cos ω0t = A0 cos ω0t + A0 ms (t )cos ω0t

sɶ (t )

A cos ω0t

These block diagrams are essentially theoretical. In practice, radically

different methods are used. A common technique is to use a saturating
class C amplifier. The theory behind this type of amplifier will be
covered in a future course. Another method for AM production is to
use a memoriless nonlinear amplifier.
Consider a system with the following input output transfer:
z = a0 + a1w + a2 w2

Let w = s1 + s2 , then z = a0 + a1s1 + a1s2 + a2 s12 + a2 s22 + 2a2 s1s2

If s1 (t ) = A cos ω0t and s2 (t ) = sɶ (t ) , then

12
z (t ) = a0 + a1 A cos ω0t + a1sɶ (t ) + a2 A2 cos 2 ω0t + a2 sɶ 2 (t ) + 2a2 Asɶ (t )cos ω0t
1 1
Now, cos 2 θ = + cos 2θ , so z(t) is the sum of four different
2 2
components:
a2 A2
Dc component: a0 +
2
Baseband component: a1sɶ (t ) + a2 sɶ 2 (t )

Component around ω0: a1 A cos ω0t + 2a2 Asɶ (t )cos ω0t

a2 A2
Component at 2ω0: cos 2ω0t
2
If we use a bandpass filter tuned at f0, we can select the component
around ω0. This component is:
x(t ) = a1 A cos ω0t + 2a2 Asɶ (t )cos ω0t
 2a 
= a1 A 1 + 2 sɶ (t )  cos ω0t
 a1 
= A0 (1 + ms (t ) ) cos ω0t

2a2 s (t ) max
In the above expression, A0 = a1 A and m = . In order to
a1
specify the filter, we have to compute the spectrum of the signal z(t).
 a2 A 2 
Z ( f ) =  a0 + δ ( f )
 2 
+ a1Sɶ ( f ) + a2  Sɶ ( f ) ∗ Sɶ ( f ) 
aA
+ 1 [δ ( f − f 0 ) + δ ( f + f 0 ) ] + a2 A  Sɶ ( f − f 0 ) + Sɶ ( f + f 0 ) 
2
a2 A 2
+ [δ ( f − 2 f0 ) + δ ( f + 2 f0 )]
4

13
 Z(f)

a2 A2 a1 A 2
a0 + δ ( f − f0 ) a2 A
δ ( f − 2 f0 )
2 2 4
a2  Sɶ ( f ) ∗ Sɶ ( f ) 
a2 ASɶ ( f − f 0 )
a1Sɶ ( f )

W 2f0
2W f0 f
f0 − W f0 + W

Positive frequency spectrum of Z(f)

The convolution term is the Fourier transform of the square a2 sɶ 2 (t ) . If
we look at the above spectrum, we remark that the AM signal is
present around f0. It has a bandwidth B = 2W. We notice also that we
must have f0 > 3W if we want to avoid overlap of spectra. If the
nonlinear system contains a higher degree, then this condition will
change.
There exist many other methods for producing AM signals. They are
better analyzed in an electronic circuit course.
AM demodulation:
There exist several techniques for AM modulation. They fall into two
different classes: Homodyne (Synchronous, Coherent) demodulation
and Non Coherent demodulation.
Coherent demodulation:
In this technique, we multiply the received AM signal by a carrier
generated at the receiver. The local carrier must have the same
frequency and same phase as the one of the AM signal. Let the

14
received signal be: x(t ) = A0 [1 + ms (t ) ] cos ω0t and the locally generated

carrier: y (t ) = B cos (ω0 + ∆ω ) t + ϕ0  . ∆ω is a frequency error and ϕ0 is a

phase error. The result of the product is:

z (t ) = A0 B [1 + ms (t ) ] cos ω0t cos (ω0 + ∆ω ) t + ϕ0 
A0 B AB
= [1 + ms(t )] cos ( (∆ω )t + ϕ0 ) + 0 [1 + ms(t )] cos ( 2ω0t + (∆ω )t + ϕ0 )
2 2
It is composed of two components. If we use a lowpass filter with a
bandwidth W, we will recover the first term. If we want to demodulate
the signal, the frequency error ∆ω must be zero and the phase error ϕ0
must be as small as possible (far from the value of π/2). The recovered
signal will be a signal proportional to s(t) plus a dc component. A
capacitor in series is enough to eliminate the dc component. The
locally generated carrier on the other hand must be exactly
synchronized to the received carrier. In most implementations, we can
use the received carrier if the baseband signal is itself bandpass. This
is the case of most audio1 and speech2 signals.

Narrow Lowpass
bandpass filter filter
at f0 cutoff W

The above figure shows a typical synchronous demodulator.

Non coherent demodulator:

1
Audio signals are usually bandpass between 50 Hz and 15 kHz.
2
Speech signals are bandlimited between 300 and 3400 Hz.

15
The envelop detector is commonly use in AM receivers and is in fact
the first demodulator in the history of radio communication.

C R

The envelop detector is composed of a diode, a resistor and a

capacitor. It is essentially a peak detector. This means that the time
constant of the circuit (RC) must be much larger than the period of the
carrier (1/f0). On the other hand, the circuit should not distort the
information signal s(t). This implies that the time constant must be
smaller than the period of the highest frequency present in the
signal (W).
1 1
> RC >>
W f0

You will have the occasion to experiment this circuit in the lab. If the
above condition is satisfied, the signal obtained at the output will be
proportional to the envelop of the AM signal r(t). Here again a dc
blocking capacitor is needed to eliminate the dc value present in the
demodulated signal.
Double sideband suppressed carrier modulation (DSB-SC):
In AM, we spend more than half of the total power transmitting a
carrier that conveys no information. The following method transmits
just the sidebands without transmitting the carrier. The DSB-SC signal
is then:

16
 x(t ) = A0 s (t )cos ω0t

We see immediately than this method is a linear modulation scheme.

Furthermore, from the defining equation, the DSB-SC modulated
signal is a bandpass signal written in quadrature form. a(t ) = A0 s (t ) and
b(t ) = 0 . In the frequency domain, the spectrum of the DSB-SC signal

is obtained by a straightforward application of the modulation theorem.

A0
X(f )= [ S ( f − f 0 ) + S ( f + f 0 )]
2
We see that all the power is in the transmitted sidebands. There is no
transmitted carrier. The transmitted power is:
1 2
Px = 2 Psb = A0 Ps
2
Most transmitters are limited by the peak power they can transmit.
The peak power is given by the square of the maximum of the
envelop Ppeak = r (t ) max . For DSB-SC, the maximum of the envelop is A0
2

while for AM this maximum is A0(1 + m). So, for AM,

Ppeak = A02 (1 + m ) and for DSB-SC, Ppeak = A02 . The ratio of sideband
2

power over the peak power for the two modulations is given by:
 Ps
 DSB-SC
Psb  4
=
Ppeak m 2 Ps
 AM
 4 (1 + m )
2

So, for a given peak power, a DSB-SC transmitter produces more than
four times the sideband power of an AM transmitter.

17
Except for a missing impulse, the spectrum of DSB-SC and the one of
AM look alike, however, in the time domain, the is a fundamental
difference. The DSB-SC envelop and phase are given by:
0 s (t ) > 0
r (t ) = A0 s (t ) ϕ (t ) = 
π s (t ) < 0

Every time the signal s(t) changes sign, the modulated signal
undergoes a phase reversal.
1

0.8

0.6

0.4

0.2

-0.2

-0.4

Phase reversals -0.6

-0.8

-1
0 0.5 1 1.5 2 2.5

We see that we cannot demodulate a DSB-SC signal with a simple

envelop detector. We need a more sophisticated demodulator.
The most commonly used demodulator for DSB is the homodyne one.
However, since there is no carrier transmitted along with the signal,
the local carrier generation is more complex.
Let x(t ) = A0 s (t )cos ω0t . The signal s(t) is assumed to be a power
signal with zero average. If we square the signal x(t), we obtain:
A02 s 2 (t )
x (t ) = A s (t )cos ω0t =
2 2 2
0
2
(1 + cos 2ω0t ) . The signal s2(t) is
2
completely positive. This means that it has an average value that is

18
different from zero. We can express it as: s 2 (t ) =< s 2 (t ) > + s1 (t ) . The
signal s1(t) has a zero average. In the frequency domain, we obtain:
A02 < s 2 (t ) > A02
Z( f ) = δ ( f ) + S1 ( f )
2 2
A02 < s 2 (t ) >
+ [δ ( f − 2 f0 ) + δ ( f + 2 f0 )]
4
A02
+ [ S1 ( f − 2 f 0 ) + S1 ( f + 2 f 0 ) ]
4
We observe a spectrum around 2f0 that is practically the one of an AM
A02 < s 2 (t ) >
signal with a carrier area of . So, we can use a narrow
4
bandpass filter tuned at 2f0 to extract a carrier. The filter will be
followed by a frequency divider by 2 (a simple D flip-flop).

Narrow
( )² Bandpass Filter ÷2 LPF
at 2f0

Squaring loop demodulator

In the above system, there remains a small problem. When we divide a
frequency by two, we have an ambiguity of π in the phase. This is due

to the fact that

( 2ω0t + 2kπ ) = ω t + kπ . This means that we can have
0
2
a signal reversal at the demodulator. If the destination of the
demodulator is the human ear, this reversal will not be noticed by the
auditor. However, if the system is used to transmit data, and if we
assign "1" to a positive value and "0" to a negative value, then the data

19
will be negated. One way to prevent this is to send a prefix word
known to the receiver. If it is received correctly, we keep the output of
the squaring loop. Otherwise, we invert the output carrier from the
squaring loop.
One way to avoid problems in carrier recovery is to send a subcarrier
at a frequency related to the one we want to recover.
Single Sideband Modulation (SSB):
When we are transmitting real signals in DSB-SC, the two sidebands
are related and if we know one, we can deduce the other. So, this
modulation method transmits only one of the two sidebands, either the
upper sideband (USB-SSB) or the lower sideband (LSB-SSB).
Basically, an SSB modulator can be implemented using a DSB-SC
one followed by a sideband filter.

s(t)
Sideband
filter

A0 cos ω0t

SSB modulator
It is quite simple to represent the different operations in the frequency
domain. The following sketch shows a USB-SSB signal in the
frequency domain.

20
 S(f)

−W W f

X(f)

−f 0 −W −f 0 f0 f0+W f

USB-SSB Spectrum
We can observe from the above sketch that the bandwidth of the SSB
signal is the same as the one of the baseband signal. So, for the same
information, the SSB modulated signal uses half the bandwidth of the
DSB modulated signal. This is why SSB is used in crowded spectrum
environment such as amateur radio. It has been used also in Frequency
Division Multiplexing (FDM) systems to transmit different voiceband
signals3. If we observe the following figure, we can observe that the
different shifted spectra do not overlap. They can be transmitted using
a single wire. To avoid any problem in carrier recovery, a subcarrier is
usually transmitted in a separate channel.

3
A voiceband signal is a signal that conveys human speech. Its spectrum is essentially different from zero in a
band between 300 and 3400 Hz.

21
 Band Limiting Filters
SSB Filter
DSBSC
4kHz
CH1 8.6 → 15.4kHz 12.3 → 15.4kHz
m1(t)

300Hz 3400kHz
f1 = 12kHz

4kHz

12.6 → 19.4kHz 16.3 → 19.4kHz

CH2
m2(t)
300Hz 3400kHz

f1 = 16kHz

Σ
Increase in 4kHz steps

FDM OUT
12 – 60kHz
4kHz

CH12 52.6 → 59.4kHz

m12(t)
56.3 → 59.4kHz
300Hz 3400kHz

f12 = 56kHz

12 Channel FDM system

To see how SSB is demodulated, we have to express the SSB signal in
the time domain. In order to do so, it is simpler to obtain first the
corresponding analytic signal X + ( f ) and then use the fact that

x(t ) = Re [ x+ (t )] . When we observe the USB-SSB spectrum, we notice

that the analytic signal X + ( f ) is just the analytic signal S+ ( f ) shifted

and scaled in the frequency domain. So, X + ( f ) = A0 S + ( f − f 0 ) . In the

time domain, it gives x+ (t ) = A0 s+ (t ) exp [ jω0t ] . Replacing s+ (t ) by its

expression s (t ) + jsˆ(t ) , we obtain:

x(t ) = A0 Re ( s (t ) + jsˆ(t ) )( cos ω0t + j sin ω0t ) 
= A0 [ s (t )cos ω0t − sˆ(t )sin ω0t ]

22
The above expression is the one of a USB-SSB modulated signal. It is
a simple matter to show that the expression of the LSB-SSB
modulated signal is:
x(t ) = A0 [ s (t )cos ω0t + sˆ(t )sin ω0t ]
The above two expressions suggest that SSB modulators can be
implemented using the following block diagram:

A0 cos ω0t
s(t)
π ∓
2
Hilbert
transform

The minus sign is for USB-SSB while the plus is for LSB-SSB. This
method of SSB production is called the Phasing Method.
SSB Demodulation:
We are going to consider only the coherent demodulation method. A
general SSB signal is: x(t ) = A0 [ s (t )cos ω0t ∓ sˆ(t )sin ω0t ] . We

multiply this signal by a carrier y (t ) = B cos (ω0 + ∆ω ) t + ϕ0  . Let us

consider first the case of zero frequency offset (∆ω = 0).

The result of the product contains terms at low frequency and terms
around 2f0. The low frequency component is:
A0 B
[ s(t )cos ϕ0 ∓ sˆ(t )sin ϕ0 ]
2

23
So, the output of the coherent demodulator will contain a linear
combination of s (t ) and sˆ(t ) . If the end destination is the human ear,
this signal will sound exactly as s(t) alone. This is due to the fact that
the human ear is insensitive to phase shifts in the signal. In other cases,
the phase error cannot be tolerated.
The analysis of the frequency error is easier to study in the frequency
domain. Using the modulation theorem, the result of the SSB signal
multiplied by a carrier is:
1 1
F  x(t ) cos (ω0 + ∆ω ) t  = X ( f − f 0 − ∆f ) + X ( f + f 0 + ∆f )
2 2
Starting from a USB-SSB signal with the spectrum shown below:

X(f)

−f 0 −W −f 0 f0 f0+W f

We obtain the following spectrum after elimination of the components

around 2f0:

S(f)

−W − ∆f −∆f ∆f W + ∆f f

24
We remark that all the frequencies of the message are translated by a
constant shift. This constant shift does not make the speech
unintelligible. However, when ∆f is positive, it makes everybody
sound like "Donald Duck", hence the name; Donald Duck distortion.
On the other hand, music will be completely distorted since the
harmonic relations between notes will disappear.
Advantages and disadvantages of SSB:
We see that SSB is a linear modulation system that saves on
bandwidth. The transmission bandwidth is equal to the signal one.
However, in order to achieve this result we need very complex
hardware.
In the filtering method, we have to transmit completely one sideband
and eliminate completely the other. This means that the transition
region of the filter is zero. The only way to achieve reasonable filters
is to use this method for signals that have no energy around zero
frequency.

X(f) Filter
Amplitude
transfer

f 0 −W f0 f0+W f

If we want to apply the phasing method, we will encounter the same

problem. It is impossible to build a filter that phase shifts all
frequencies from zero to W. This means that the phase response of the

25
filter has also a zero transition band. We can approximate the Hilbert
transformer if the signal s(t) has the same character as above. It must
not have any energy around zero. So, SSB is useful if we want to
transmit speech. Audio signals can be transmitted at the expense of a
quite complicated hardware. Data cannot be transmitted in the shape
of a sequence of pulses. This signal possesses power at zero frequency.
If we want to transmit signals that have spectra that are different from
zero around dc, one solution is to use Vestigial Sideband.
Vestigial Sideband (VSB):
In VSB modulation, we use filter that transmit most of one sideband
and a very small amount of the other (a vestige).
In order to determine the filter characteristics, we must analyze a
complete modulation and demodulation system. The demodulation
method is always coherent. We multiply the received signal by a
carrier B cos ω0t and we lowpass filter the result to eliminate terms
around 2ω0 .

A B C D E
VSB Filter Lowpass
Filter

A0 cos ω0t B cos ω0t

In the above block diagram, we must determine the signals at different

points.
At A, we have the baseband signal xA(t) = s(t) with spectrum
XA(f) = S(f).

26
At B, we obtain the DSB-SC signal xB(t) = A0s(t)cosω0t with spectrum
A0 A
XB( f ) = S ( f − f0 ) + 0 s( f + f0 ) .
2 2
At C, we have the VSB signal obtained by filtering the DSB-SC signal.
We are going to characterize it in the frequency domain only:
A0
XC( f ) = [ H ( f ) S ( f − f0 ) + H ( f ) S ( f + f0 )]
2
At D, we use the modulation theorem of Fourier transforms and we
obtain:
A0 B
XD( f ) = [ H ( f − f0 )S ( f − 2 f0 ) + H ( f − f0 )S ( f )
4
+ H ( f + f 0 ) S ( f ) + H ( f + f 0 ) S ( f + 2 f 0 )]
The lowpass filter eliminates all the terms around ±2f0. So, the signal
A0 B
at E will be: X E ( f ) = [ H ( f − f0 )S ( f ) + H ( f + f0 )S ( f )]
4
If we want to have a distortionless transmission, this signal must be
proportional to s(t). This means that:
H ( f + f 0 ) + H ( f − f 0 ) = constante
After some manipulations, we obtain that the transfer function of the
filter must satisfy:
H ( f 0 + x) + H ∗ ( f 0 − x) = 2 Re [ H ( f 0 )] for f around f0.

If H ( f ) = R ( f ) + jX ( f ) , then
R( f 0 + x) + R ( f 0 − x ) = 2 R( f 0 )
X ( f 0 + x) = X ( f 0 − x)

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 Re[H( f )]

f
f0

Im[H( f )]

f0 f

The above graph shows the different symmetries that the real and
imaginary part of the transfer function must satisfy. The real part must
show an odd symmetry with respect to the point (f0, Re[H(f0)]) while
the imaginary part must have the vertical line passing by f0 as a
symmetry axis.
The VSB signal has been characterized in the frequency domain. We
have seen that it can be demodulated using coherent demodulation.
We can also have the expression of the VSB signal in the time domain.

28
Being a general bandpass signal, it can be expressed in quadrature
form and it is completely described by its complex envelop. The VSB
signal appears at point C in our block diagram. Its complex envelop is
given by the filtering of the complex envelop of the DSB-SC signal at
B by the lowpass equivalent filter.
The complex envelop of the signal at B is mxB (t ) = A0 s (t ) with a
spectrum M xB ( f ) = A0 S ( f ) . The lowpass equivalent filter H lp ( f ) is

the filter H ( f ) translated down. Using the symmetries derived above,

we can express the equivalent lowpass filter transfer function as:
H lp ( f ) = R ( f 0 ) + A( f ) + jX lp ( f ) = R ( f 0 ) [1 + jQ( f )]

In the above expression, A(f) is an odd function (R(f) translated down

to f = 0 and shifted down by R(f0)) while Xlp(f) is an even function.
This implies that Q(f) satisfies Q( f ) = Q∗ (− f ) . The complex envelop
of the VSB signal is then M x ( f ) = A0 R ( f 0 ) S ( f ) [1 + jQ( f )] or in the

time domain: mx (t ) = A0 R ( f 0 ) s (t ) + jA0 R ( f 0 )F -1

[Q( f ) S ( f )] . The
function Q(f)S(f) satisfies the condition of Hermitian symmetry. This
implies that its inverse Fourier transform is real. Let
q(t ) = F −1
[Q( f ) S ( f )] then the complex envelop of the VSB signal is:
mx (t ) = A0 R ( f 0 ) [ s (t ) + jq (t ) ] and the VSB signal can be written as:

x(t ) = A0 R ( f 0 ) [ s (t )cos ω0t − q (t )sin ω0t ] .

The signal q(t) is the output of the filter with transfer function Q(f).
A( f ) X lp ( f )
Q( f ) = +
jR( f 0 ) R( f 0 )

29
Two extreme cases are interesting:
If we want to keep the upper sideband and eliminate completely the
2 R( f 0 ) f > f0
lower one, we must have H ( f ) = 
 0 0 < f < f0

This implies that Q( f ) = − j sgn( f ) and q(t ) = sˆ(t ) . The VSB signal in
this case is a USB-SSB signal.
The other extreme case is when we want to keep both sidebands. At
that time, Q(f) = 0 and the signal is just a DSB-SC one.
In our analysis, we have assumed that we favor the upper sideband.
We can obtain the same results for the lower sideband. The modulated
signal bandwidth is intermediate: W < B < 2W.
Envelop demodulation of linear modulation + carrier:
If we add a large amplitude carrier to the inphase component of a
bandpass signal (DSB, SSB, VSB) we obtain:
x(t ) = B cos ω0t + A0 s (t )cos ω0t ± A0 q (t )sin ω0t
The envelop of the signal is:

A02 q 2 (t )
r (t ) = ( B + A0 s(t ) ) + A q (t ) = B + A0 s (t ) 1 +
2 2 2

( B + A0 s (t ) )
0 2

If B >> A0 , the expression inside the absolute value will always be

A02 q 2 (t )
positive and r (t ) ≈ ( B + A0 s (t ) ) + ≈ B + A0 s (t ) .This
2 ( B + A0 s (t ) )
technique is used in the transmission of analog television where the
TV signal is transmitted in VSB+Carrier.

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