Chameli Devi Group of Institutions

Analog and Digital Communication

Unit I
Signals and Systems: Block diagram of a communication system, signal-definition, types of signals continuous,
discrete, deterministic, non-deterministic, periodic, non-periodic, energy, power, analog and digital signals.
Electromagnetic Spectra, Standard signals- DC, sinusoidal, unit step, ramp, signum, rectangular pulse, impulse
(delta) signal. System definition, classification of systems, linear, nonlinear, time variant, time invariant, causal,
non causal, stable and unstable systems. Fourier transforms: Time domain and frequency domain
representation of signal, Fourier Transform and its properties, conditions for existence, Transform of Gate, unit
step, constant, impulse, sine and cosine wave. Shifting property of delta function, convolution, time and
frequency convolution theorems.
Course Objective:-
The objective of this course is to study of communication systems starts with the concept of analog
communication.
Course Outcomes:-
At the end of the course student will be able to :
1. Differentiate Analog and Digital Signal and types of signals.
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 Block diagram of a communication system:

 Information Source: The source of data,

• Data could be: human voice, data storage device CD, video etc.
• Data types: Discrete: Finite set of outcomes “Digital”/ Continuous : Infinite set of outcomes “Analog”
 Transmitter
• Converts the source data into a suitable form for transmission through signal processing
• Data form depends on the channel
 Channel:
• The physical medium used to send the signal
• The medium where the signal propagates till arriving to the receiver
• Physical Mediums (Channels):
o Wired : twisted pairs, coaxial cable, fiber optics
o Wireless: Air, vacuum and water
 Receiver
• Extracting the message/code in the received signal
• Information Sink
• The final stage

 Signal and its types:

A signal is a way of conveying information. Any time varying physical phenomenon that can convey information
is called signal. Some examples of signals are human voice, electrocardiogram, sign language, videos etc.
Technically - A signal is a function of one or more independent variables. a function of time, space, or another
observation variable that conveys information
Signals are classified into the following categories:
 Continuous Time, Discrete Time Signals and Digital signals
 Deterministic and Non-deterministic Signals
 Even and Odd Signals
 Periodic and Aperiodic Signals
 Energy and Power Signals
 Real and Imaginary Signals
 Continuous Time Signal:
If the independent variable (t) is continuous, then
the corresponding signal is continuous time signal.
A finite, real-valued, smooth function x(t) of a
variable t which usually represents time.
Both s and t in X (t) are continuous. A continuous-
time signal is a signal that can be defined at every
instant of time. It is denoted by x(t). Figure shows
continuous-time signal. Figure Continuous-time signal.

 Discrete Time Signal:

If the independent variable (t) takes on only

discrete values, for example t = ±1, ±2, ±3, ...
A discrete-time signal is a bounded, continuous-
valued sequence x[n]. Alternately, it may be viewed
as a continuous-valued function of discrete index n.
Discrete time signals can be obtained by sampling a
continuous-time signal. It is denoted as x(n).Figure
Figure Discrete-time signal shows discrete-time signal.

 Digital Signal:
The signals that are discrete in time and quantized in amplitude are called digital signal. The term "digital
signal" applies to the transmission of a sequence of values of a discrete-time signal in the form of some digits in
the encoded form.

 Random (Non-deterministic) and Deterministic Signal:

A random signal cannot be described by any mathematical function, where as a deterministic signal is one that
can be described mathematically. A common example of random signal is noise. Random signal and
deterministic signal are shown in the Figure a and b respectively. A signal is said to be non-deterministic if
there is uncertainty with respect to its value at some instant of time. Non-deterministic signals are random in
nature hence they are called random signals. Random signals cannot be described by a mathematical
equation. They are modeled in probabilistic terms. A signal is said to be deterministic if there is no uncertainty
with respect to its value at any instant of time.

Figure a Random signal

Figure b Deterministic signal
 Even and Odd Signals
 Even Signals  Odd Signals
A signal is said to be even when it satisfies the A signal is said to be odd when it satisfies the
 condition x(t) = x(-t). condition x(t) = -x(-t).
Example 1: t2, t4… cos t etc. Example: t, t3 ... And sin t
Let x(t) = t2 Let x(t) = sin t
x(-t) = (-t)2 = t2 = x(t) x(-t) = sin(-t) = -sin t = -x(t)
t2 is even function ∴ sin t is odd function.

 Periodic and Aperiodic Signals

A signal is said to be periodic signal if it has a definite pattern and repeats itself at a regular interval of time.
Whereas, the signal which does not at the regular interval of time is known as an aperiodic signal or non-
periodic signal.
A signal is said to be periodic if it satisfies the condition x(t) = x(t + T) or x(n) = x(n + N).
Where T = fundamental time period, 1/T = f = fundamental frequency.

Figure Periodic signal Figure Aperiodic signal

 Energy and Power Signals

• Energy and power for Continuous time signal
A signal is said to be energy signal when it has finite energy.
∞
Energy E=∫−∞ |(𝑡)|2dt
A signal is said to be power
𝑇 signal when it has finite power.
Power P = lim 1 ∫ |(𝑡)|2 dt
𝑡→∞ 2𝑇 −𝑇
A signal cannot be both, energy and power simultaneously. Also, a signal may be neither energy nor power
signal. If 0< E< ∞, then the signal x(t ) is called an energy signal. However, there are signals where this
condition is not satisfied. For such signals we consider the power. If 0< P< ∞, then the signal is called a power
signal. Note that the power for an energy signal is zero (P = 0) and that the energy for a power signal is infinite
(E = ∞). Some signals are neither energy nor power signals.
Power of energy signal = 0
Energy of power signal = ∞

• Energy and power for discrete-time signal:

The definition of signal energy and power for discrete signals is similar definitions for continuous signals.
The signal energy in the discrete-time signal x(n ) is:
E = ∑∞ |(𝑛)|2
𝑛=−∞

The signal power in the signal x(n ) is:

1
P = lim 𝑛→∞ 2𝑁+1
∑𝑁
𝑛=−𝑁 |(𝑛)|
2

 Real and Imaginary Signals

A signal is said to be real when it satisfies the condition x(t) = x*(t)
A signal is said to be imaginary when it satisfies the condition x(t) = -x*(t)
Example:
If x(t)= 3 then x*(t)=3*=3 here x(t) is a real signal.
If x(t)= 3j then x*(t)=3j* = -3j = -x(t) hence x(t) is a imaginary signal.
Note: For a real signal, imaginary part should be zero. Similarly for an imaginary signal, real part should be
zero.
 Causal, Non-causal and Anti-causal Signal:
Signal that are zero for all negative time, that type of signals are called causal signals, while the signals that are
zero for all positive value of time are called anti-causal signal.
A non-causal signal is one that has non zero values in both positive and negative time. Causal, non-causal and
anti-causal signals are shown below in the Figure (a), (b) and (c) respectively.

Fig.(a) Causal signal Fig.(b) Non-causal signal Fig.(c) Anti-causal signal

 Standard Signal:
 DC Signal:
The “DC” or constant signal simply takes a constant
value. In continuous time it would be represented
as: s(t)=1. In discrete time, it would be s[n] =1. The
number “1” may be replaced by any constant. The
DC signal typically represents any constant offset
from 0 in real-world signals.The analog DC signal has
bounded amplitude and power and is smooth.

 Unit Step Function

The unit step: The “unit step”, also often referred to as a Heaviside function, is literally a step. It has 0 value
until time 0, at which point, it abruptly switches to 1 from 0. The unit step represents events that change
state, e.g. the switching on of a system, or of another signal. It is usually represented as u(t) in continuous time
and u[n] in discrete time.
Unit step function is denoted by u(t). It is defined as
1, 𝑡≥0 0, n < 0
u(t) ={ Discrete time: u[n]= {
0, 𝑡<0 1, n≥0
 Figure Continuous time Unit step signal
Figure Discrete time Unit step signal

 Unit Impulse Function  Ramp Signal

The unit impulse function also known as the Dirac Ramp signal is denoted by r(t), and it is defined as
Delta Function is denoted by δ(t) and it is defined 𝑡 𝑡⩾0
r(t) = {
as 0 𝑡<0
1, 𝑡=0
δ(t)={
0, 𝑡≠0

Figure Unit Impulse signal Figure Ramp signal

 Signum Function
1 𝑡>0
Signum function is denoted as sgn (t). It is defined as sgn (t) = { 0 𝑡=0
−1 𝑡<0

 Exponential Signal
Exponential signal is in the form of x(t) = eαt.
The shape of exponential can be defined by α.
Case i: if α = 0 → x(t) = e0 = 1 Case ii: if α < 0 i.e. -ve then x(t) Case iii: if α > 0 i.e. +ve then x(t)
= e−αt. The shape is called = eαt. The shape is called raising
decaying exponential. exponential.

 Rectangular Signal
Let it be denoted as x(t) and it is defined as
𝑟
x (t) = A rect [ ]
𝑟
x(t)
1

t
−𝑟 0 𝑟
2 2

Figure Rectangular function

 Triangular Signal
|𝑡|
Let it be denoted as x(t)= A [1 − ] x(t)
𝑇

-T T t

Figure Triangular function

 Sinusoidal Signal
Sinusoidal signal is in the form of x(t) = A cos (w0±ϕ) or A sin(w0±ϕ)
Where T0 = 2πw0
 1

0.5

x(t) 0
0 2 4 6 8 10 12 14
-0.5

-1
time(t)

Figure Sinusoidal function

 Sinc function:
It is denoted as sinc (t) and it is defined as
sinc(t)=sin (πt)/πt
=0 for t=±1, ±2 ,±3...

Sinc(x)

−3
−1 2

−2 0 1 3

Figure Sinc function

 Sampling Function
It is denoted as sa(t) and it is defined as
sa (t)=sin t/t
=0 for t=±π,±2π,±3π.....
Sa (x)

−3𝜋
−𝜋 2𝜋

−2𝜋 0 𝜋 3𝜋
t

Figure Sampling function

 Electromagnetic Spectra:
Electromagnetic spectrum ranges from dc to light.
The lower radio frequencies are designated mainly
by frequency. The optical ranges are referred by
wavelength.
Signal parameters: Amplitude is the height of a
wave. It is measured from a wave’s midpoint to its
peak. It is normally expressed in Volts (V).
Frequency refers to the number of times a wave
cycles past a given point each second. It is normally
expressed in Hertz (Hz).Wave Length is the distance
from the start to the end of a single wave cycle. It is
typically expressed in meters. Figure Signal Parameters

Figure Electromagnetic spectra

Figure Electromagnetic Spectrum for communication systems

 System definition and classification of systems:
System can be considered as a physical entity which manipulates one or more input signals applied to it. The
system description specifies the transformation of the input signal to the output signal.
Systems are classified into the following categories:
 Linear and Non-linear Systems
 Time Variant and Time Invariant Systems
 Linear Time variant and Linear Time invariant systems
 Static and Dynamic Systems
 Causal and Non-causal Systems
 Stable and Unstable Systems

 Linear and Non-linear Systems

A system is said to be linear when it satisfies superposition and homogeneity principles. Consider two systems
with inputs as x1(t), x2(t), and outputs as y1(t), y2(t) respectively. Then, according to the superposition and
homogenate principles,
Principle of homogeneity: T [a1*x1(t)] = a1*y1, T [a2*x2(t)] = a2*y2
Principle of superposition: T [x1(t)] + T [x2(t)] = a1*y1+a2*y2
Linearity: T [a1 x1(t)] + T[ a2 x2(t)] = a1 y1(t) + a2 y2(t)
From the above expression, is clear that response of overall system is equal to response of individual system.
Example:
y(t) = x2(t)
y1 (t) = T[x1(t)] = x12(t)
y2 (t) = T[x2(t)] = x22(t)
T [a1 x1(t) + a2 x2(t)] = [ a1 x1(t) + a2 x2(t)]2
Which is not equal to a1 y1(t) + a2 y2(t). Hence the system is said to be non linear.
 Time Variant and Time Invariant Systems
A system is said to be time variant if its input and output characteristics vary with time. If the system response
to an input signal does not change with time such system is termed as time invariant system. The behavior and
characteristics of time variant system are fixed over time.The condition for time invariant system is:
In time invariant systems if input is delayed by time t0 the output will also gets delayed by t0. Mathematically it
is specified as follows
y(t-t0) = T[x(t-t0)]
For a discrete time invariant system the condition for time invariance can be formulated mathematically by
replacing t as n*Ts is given as
y(n-n0) = T[x(n-n0)]
 Linear Time variant (LTV) and Linear Time Invariant (LTI) Systems
If a system is both linear and time variant, then it is called linear time variant (LTV) system.
If a system is both linear and time Invariant then that system is called liner time invariant (LTI) system.
 Static and Dynamic Systems
Static system is memory-less whereas dynamic system is a memory system.
Example 1: y (t) = 2 x(t)
For present value t=0, the system output is y(0) = 2x(0). Here, the output is only dependent upon present
input. Hence the system is memory less or static.
Example 2: y (t) = 2 x(t) + 3 x(t-3)
For present value t=0, the system output is y(0) = 2x(0) + 3x(-3).
Here x(-3) is past value for the present input for which the system requires memory to get this output. Hence,
the system is a dynamic system.
 Causal and Non-Causal Systems
A system is said to be causal if its output depends upon present and past inputs, and does not depend upon
future input. For non causal system, the output depends upon future inputs also.
Example 1: y(n) = 2 x(t) + 3 x(t-3)
For present value t=1, the system output is y(1) = 2x(1) + 3x(-2).
Here, the system output only depends upon present and past inputs. Hence, the system is causal.
Example 2: y(n) = 2 x(t) + 3 x(t-3) + 6x(t + 3)
For present value t=1, the system output is y(1) = 2x(1) + 3x(-2) + 6x(4) Here, the system output depends upon
future input. Hence the system is non-causal system.
 Stable and Unstable Systems
The system is said to be stable only when the output is bounded for bounded input. For a bounded input, if
the output is unbounded in the system then it is said to be unstable.
Note: For a bounded signal, amplitude is finite.
 Example 1: y (t) = x2(t)
Let the input is u(t) (unit step bounded input) then the output y(t) = u2(t) = u(t) = bounded output.
Hence, the system is stable.
 Example 2: y (t) = ∫x(t)dt
Let the input is u (t) (unit step bounded input) then the output y(t) = ∫u(t)dt = ramp signal (unbounded
because amplitude of ramp is not finite it goes to infinite when t → infinite).
Hence, the system is unstable.

 Time domain and frequency domain representation of signal

An electrical signal either, a voltage signal or a current signal can be represented in two forms: These two types
of representations are as under:
i) Time Domain representation-: In time domain representation a signal is a time varying quantity as
shown in Fig.
V(t)

0 t

Fig An arbitrary time domain signal

 ii) Frequency Domain Representation: In frequency domain, a signal is represented by its frequency
spectrum as shown in Fig
V(w)

0 w

Fig Frequency domain representation of time domain signal

 Fourier Transform and its properties

Fourier Transform pair
Fourier transform may be expressed as
∞
X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
In the above equation X(w) is called the Fourier transform of x(t). In other words X(w) is the frequency domain
representation of time domain function x(t). This means that we are converting a time domain signal into its
frequency domain representation with the help of fourier transform. Conversely if we want to convert
frequency domain signal into corresponding time domain signal, we will have to take inverse fourier transform
of frequency domain signal. Mathematically, Inverse fourier transform.
1 ∞
𝐹−1[𝑋(𝑤)] = 𝑥(𝑡) = ∫ 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞
Example
Find the fourier transform of a single-sided exponential function 𝑒−(𝑡).
Solution: 𝑒−(𝑡) is single sided function because her the main function 𝑒−𝑎𝑡 is multiplied by unit step
function u(t), then resulting signal will exist only for t>0.
u(t)= {1 for t>0
={0 for elsewhere
Now, given that x(t)= 𝑒−𝑎𝑡𝑢(𝑡)
∞
X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
∞
Or X(w)=∫−∞ 𝑒−𝑎𝑡 (𝑡) 𝑒−𝑗𝑤𝑡 dt
∞
=∫0 𝑒−𝑎𝑡 𝑒−𝑗𝑤𝑡 dt

∞
=∫0 𝑒−(𝑎+𝑗𝑤 )dt
−1 −1 1
[𝑒−∞ - 𝑒0]= [0-1]=
(𝑎+𝑗𝑤 ) (𝑎+𝑗𝑤 ) (𝑎+𝑗𝑤 )

To obtain the above expression in the proper form we write

−1 (𝑎−𝑗𝑤 )
X(w)= *
(𝑎+𝑗𝑤 ) (𝑎−𝑗𝑤 )

(𝑎−𝑗𝑤 ) 𝑎 𝑗𝑤
X(w)= (𝑎 2 +𝑤 2 ) = 2 +𝑤 2 )
-
(𝑎 (𝑎 2 +𝑤 2 )
Obtaining the above expression
𝑤
in polar form
1 −𝑗 𝑡𝑎𝑛 −1 ( )
X(w)= √𝑎 2 𝑒 𝑎
+𝑤 2
As we know that
X(w)=|𝑋(𝑤)|𝑒𝑗𝜑 (𝑗𝑤 )
On comparision amplitude spectrum
1
|(𝑤)|=
√𝑎 2 +𝑤 2

𝑤
(𝑤) = −𝑡𝑎𝑛−1( )
𝑎
 Properties of Continuous Time Fourier Transform (CTFT)
• Time Scaling Function

Time scaling property states that the time compression of a signal results in its spectrum expansion and time
expansion of the signal results in its spectral compression. Mathematically,
If x(t) X(w)
Then, for any real constant a,
1 𝑤
x(at) X( )
|𝑎| 𝑎
Proof: The general expression for fourier transform is
∞
X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
Now F[x(at)]=∫∞ (𝑎𝑡) 𝑒−𝑗𝑤𝑡 dt
−∞
Putting
At=y
𝑑𝑦
We have dt=
𝑎
Case (i): When a is positive real constant 𝑤
∞ 𝑤 𝑑𝑦 1 ∞ 1 𝑤
F[x(at)]= ∫ 𝑥(𝑦) 𝑒 −𝑗 ( 𝑎)𝑦 = ∫ 𝑥(𝑦) 𝑒−𝑗(𝑎 )𝑦 = X( )
−∞ 𝑎 𝑎 −∞ 𝑎 𝑎
Case (ii): When a is negative real constant
−1 𝑤
F[x(at)]= X( )
𝑎 𝑎
Combining two cases, we have
1 𝑤 1 𝑤
F[x(at)]= |𝑎| X( ) Or x(at) X( )
𝑎 |𝑎| 𝑎
The function x(at) represents the function x(t) compressed in time domain by a factor a. Similarly, a function
𝑤
X( ) represents the function X(w) expanded in frequency domain by the same factor a.
𝑎

• Linearity Property

Linearity property states that fourier transform is linear. This means that
If x1(t) X1(w)
And x2(t) X2(w)
Then a1 x1(t) + a2 x2(t) a1X1(w) + a2X2(w)

• Duality or Symmetry Property

If x(t) X(w)
Then X(t) 2 πx(-w)
Proof
The general expression for fourier transform is
1 ∞
𝐹−1[𝑋(𝑤)] = 𝑥(𝑡) = ∫ 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞
Therefore,
 1 ∞
x(-t)= ∫−∞ 𝑋(𝑤)𝑒−𝑗𝑤𝑡 𝑑𝑤
2𝜋

∞
2πx(-t)= ∫−∞ 𝑋(𝑤)𝑒−𝑗𝑤𝑡 𝑑𝑤
Since w is a dummy variable, interchanging the variable t and w we have
∞
2πx(-w)= ∫−∞ (𝑡)𝑒−𝑗𝑤𝑡 𝑑𝑤=F[X(t)]
Or F[X(t)]= 2πx(-w)

Or X(t) 2πx(-w)
For an even function x(-w)=x(w)

Therefore , X(t) 2πx(w)

Example (1)
1 1
The fourier transform F[𝑒−𝑡𝑢(𝑡)] is equal to . Therefore F[ ] is equal to
1+𝑗 2𝜋𝑓 1+𝑗 2𝜋𝑓
Solution:
Using Duality property of Fourier Transform, we have

If x(t) X(f)
Then X(t) x(-f)
Therefore,
1
𝑒−(𝑡)
1+𝑗 2𝜋𝑓
1
Then 𝑒−𝑓 𝑢(𝑓)
1+𝑗 2𝜋𝑡

• Time Shifting property

Time Shifting property states that a shift in the time domain by an amount b is equivalent to multiplication by
𝑒−𝑗𝑤𝑏 in the frequency domain. This means that magnitude spectrum |(𝑤)|

Remains inchanged but phase spectrum θ(w) is changed by -wb.

If x(t) X(w)
Then X(t-b) X(w) 𝑒−𝑗𝑤𝑏
∞
Proof: X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
∞
And F[x(t-b)]= ∫−∞ (𝑡 − 𝑏) 𝑒−𝑗𝑤𝑡 dt
Putting t-b=y, so that dt=dy
∞ ∞
F[x(t-b)]= ∫−∞ (𝑦) 𝑒−𝑗𝑤 (𝑏+𝑦)dy =∫ −∞
(𝑦) 𝑒−𝑗𝑤𝑏 𝑒−𝑗𝑤𝑦 dy

∞
Or F[x(t-b)]= 𝑒−𝑗𝑤𝑏 ∫−∞ (𝑦) 𝑒−𝑗𝑤𝑦 dy
Since y is a dummy variable, we have
F[x(t-b)]= 𝑒−𝑗𝑤𝑏 X(w)=X(w) 𝑒−𝑗𝑤𝑏
Or x(t-b) X(w) 𝑒−𝑗𝑤𝑏
• Frequency Shifting Property
Frequency shifting property states that the multiplication of function x(t) by 𝑒𝑗𝑤0 𝑡 is equivalent to shifting its
fourier transform X(w) in the positive direction by an amount 𝑤0 . This means that the spectrum X(w) is
translated by an amount 𝑐. hence this property is often called frequency translated theorem. Mathematically .

If x(t) X(w)
𝑡
Then 𝑒 𝑗 𝑤 0 x(t) X(w-𝑤0 )
Proof: General expression for fourier transform is
∞
X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
∞
Now, F[𝑒𝑗 𝑤0 𝑡 x(t)] = ∫−∞ (𝑡) 𝑒 𝑗𝑤 0 𝑡 𝑒−𝑗𝑤𝑡 dt
∞
Or F[𝑒𝑗 𝑤0 𝑡 x(t)] = ∫ (𝑡) 𝑒−𝑗 (𝑤−𝑤0 )dt
−∞
Or F[𝑒𝑗 𝑤0 𝑡 x(t)] = X(w − 𝑤0 )
𝑂𝑟 𝑒𝑗 𝑤0 𝑡 x(t) X(w-𝑤0 )

• Time Differentiation Property

The time differentiation property states that the differentiation of a function x(t) in the time domain is
equivalent to multiplication of its fourier transform by a factor jw. Mathematically
If x(t) X(w)
𝑑𝑥 (𝑡)
Then 𝑑𝑡 x(t) jw X(w)
Proof: The general expression for fourier transform is
1 ∞
𝐹−1[𝑋(𝑤)] = 𝑥(𝑡) = ∫ 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞
Taking differentiation, we have
𝑑𝑥 (𝑡) 1
= 𝑑 [∫∞ 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤]
𝑑𝑡 2𝜋 𝑑𝑡 −∞
Interchanging the order of differentiation and integration, we have
𝑑𝑥 (𝑡)
= 1 ∫∞ 𝑑 [𝑋(𝑤)𝑒𝑗𝑤𝑡 ]𝑑𝑤
𝑑𝑡 2𝜋 −∞ 𝑑𝑡

1 ∞
= ∫ 𝑗𝑤 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞
𝑑𝑥 (𝑡)
Or = 𝐹−1[𝑗𝑤𝑋(𝑤)]
𝑑𝑡
𝑑𝑥 (𝑡)
Or F[ ] = 𝑗𝑤𝑋(𝑤)
𝑑𝑡
𝑑𝑥 (𝑡)
Or (𝑤) Hence proved
𝑑𝑡

• Transform of Gate
A gate function is rectangular pulse. Figure shows gate function. The function or rectangular pulse shown in
𝑡
figure 1.3 is written as rect ( ).
𝑟

x(t)
1

t
−𝑟 0 𝑟
2 2
Fig A Gate Function
 𝑡
From the above figure it is clear that rect ( ) represents a gate pulse of height or amplitude unity and width 𝑟.
𝑟
𝑡 −𝑟 𝑟
x(t)= rect ( ) ={1 𝑓𝑜𝑟 < 𝑡 < 2}
𝑟 2
{0 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒}

• Sampling Function Or Interpolation Function Or Sinc function

𝑠𝑖𝑛𝑥
The functions is the “sine over argument” and denoted by sinc(x). This function plays an important role in
𝑥
signal processing. It is also known as the filtering or interpolating function. Mathematically,
𝑠𝑖𝑛𝑥
Sinc(x)=
𝑥
Or
𝑠𝑖𝑛𝑥
Sa(x)=
𝑥

Sinc(x) Or Sa (x)

−3𝜋
−𝜋 2𝜋

−2𝜋 0 𝜋 3𝜋

Fig. Sample function

From the figure, following points may be observed about the sampling function :

(i) Sa(x) or sinc(x) is an even function of x.

(ii) Sinc(x) =0 when sinx=0 except at x=0, where it is indeterminate. This means that sinc(x)=0 for
x=±nπ , here n=±1, ±2 ….
1
(iii) Sinc(x) is the product of oscillating signal sinx of period 2π and a decreasing function .
𝑥
Therefore, sinc(x) exhibits sinusoidal oscillations of period 2π with amplitude decreasing
continuously as 1/x.
Example 2: Find the fourier transform of the gate function shown in figure.

x(t)
1

t
−𝑟 0 𝑟
2 Fig.2.5 Gate fu nctio
2 n
𝑡 −𝑟 𝑟
Sol. x(t)= rect ( ) ={1 𝑓𝑜𝑟 < 𝑡 < 2}
𝑟 2
{0 𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒}
∞
X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
 ∞ 𝑡
X(w)=F[x(t)]=∫−∞ rect ( 𝑟) 𝑒−𝑗𝑤𝑡 dt

𝑟 −𝑒−𝑗𝑤𝑡 2𝑟
=∫2 1. 𝑒−𝑗𝑤𝑡 dt= [ ]
𝑗𝑤 −𝑟
−𝑟
2 2

−1 𝑤𝑟 𝑤𝑟
𝑗
= [𝑒 −𝑗 2 𝑒- 2 ]
𝑗𝑤
1 𝑤𝑟 𝑤𝑟
𝑗 −𝑗
= [𝑒 2 -𝑒 2 ] --------(1)
𝑗𝑤

We know that 𝑒𝑗𝜃 =cos𝜃 +jsin𝜃

And 𝑒−𝑗𝜃 =cos𝜃 –jsin𝜃
Hence 2cos𝜃=𝑒𝑗𝜃 +𝑒−𝑗𝜃
2jsin𝜃=𝑒𝑗𝜃 -𝑒−𝑗𝜃
𝑤𝑟
Putting 𝜃= 2 , we get

𝑤𝑟 𝑤𝑟 𝑤𝑟
2jsin =𝑒 𝑗 2 𝑒- −𝑗 2 --------(2)
2
From (1) and (2)
1 𝑤𝑟
X(w)= [2jsin ]
𝑗𝑤 2
By multiplying and dividing the equation by 𝑟
2𝑟 𝑤𝑟
= [jsin ]
𝑗𝑤𝑟 2

𝑟 𝑤𝑟
= 𝑤𝑟 [sin ]
2
2

𝑤𝑟
sin
2
=𝑟[ 𝑤𝑟 ]
2
𝑤𝑟
= 𝑟sinc( )
2
Now, since sinc(x)=0, when x=±nπ
𝑤𝑟
Therefore, sinc( ) =0, when 𝑤𝑟 =±nπ
2 2
=±2nπ
Or w= 𝑟
2π
Figure the plot of X(w)
𝑟

𝑤𝑟
𝑟 Sinc( )
2

6π 2π
− −
𝑟 𝑟 4π
𝑟

0 2π 6π
4π
− 𝑟 𝑟
𝑟
Fig.
 Impulse Functions

Unit Impulse functions:

A unit impulse function was invented by P.A.M. Diarc and so it is also called as Delta function. It is denoted by
(𝑡 ).
Mathematically,
(𝑡)=0 , t≠ 0
∞
And, ∫−∞ (𝑡)dt=1
Figure shows the graphical representation of an unit impulse function. The following points may be observed
about an unit-impulse function:

𝛿(𝑡

0 t
Fig. The Unit Impulse function
i) The width of pulse is zero. This means that pulse exist only at t=0.
ii) The height of the pulse goes to infinity
iii) The area under the pulse-curve is always is always unity.
Shifting Property of the Impulse function:
If we take the product of unit impulse function (𝑡) and any given function x(t) which is continuous at t=0,then
this product will provide the function x(t) existing only at t=0 since 𝛿(𝑡) exist only at t=0. Mathematically,
∞ ∞
∫−∞ (𝑡)(𝑡)dt=x(0) ∫ −∞ (𝑡)dt=x(0).1=x(0)
The equation is also known as shifting or sampling property of the impulse function because the impulse shifts
the value of x(t0 at t=0. This means that the value of x(t) has been sampled at t=0. to The shifting or sampling
may be also done at any, instant t=𝑡0, if we define the impulse function at the instant. Mathematically,
∞
∫−∞ (𝑡)(𝑡 − 𝑡0 )dt=x(𝑡0)
The above equation states that the product of a continuous function x(t) with an impulse function 𝛿(𝑡 − 𝑡0)
provides the sampled value of x(t) at t=𝑡0.
Q.1 Find the fourier transform of an impulse function x(t) = 𝛿(𝑡) Also draw the spectrum
Sol. Expression of the fourier transform is given by
∞ ∞
X(w)=F[x(t)]=∫−∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt=∫ −∞ (𝑡) 𝑒−𝑗𝑤𝑡 dt
Using shifting property of impulse function
X(w)=[ 𝑒−𝑗𝑤𝑡 ]at t=0
X(w)=1
(𝑡) 1
Hence the fourier transform of an impulse function is unity.
𝛿(𝑡)
(𝑤) = 1
1 1

0 t 0 w
Fig Time and frequency domain of impulse function
Figure 2.8 shows an unit impulse function and its fourier transform or spectrum. From the figure it is clear that
an unit impulse contains the entire frequency components having identical magnitude. This means that the
bandwidth of the unit impulse function is infinite. Also, since spectrum is real, only magnitude spectrum is
required. The phase spectrum (𝑤)=0, which means that all the frequency components are in the same phase.

Q.(2) Find the inverse fourier transform of 𝛿(w)

Solution. Inverse fourier transform is expressed as
1 ∞
𝐹−1[𝑋(𝑤)] = 𝑥(𝑡) = ∫ 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞

1 ∞
𝐹−1[𝛿(w)] = 𝑥(𝑡) = ∫ (w)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞
1
𝐹−1[𝛿(w)]= [ 𝑒𝑗𝑤𝑡 ] at w=0
2𝜋
1 1 1
𝐹−1[𝛿(w)]= [ 𝑒0]= .1 =
2𝜋 2𝜋 2𝜋
1
F[ ]= 𝛿(w)
2𝜋
1 (w)
2𝜋
1 2(w)

𝑋(𝑤
(𝑡) = 1
1
2(𝑤)

0 t
0 w
Fig. representation of Inverse fourier transform
This shows that the spectrum of a constant signal x(t)=1 an impulse function 2𝜋𝛿(𝑤). This can also be
interpreted as that x(t) =1 is a d.c. signal which has single frequency. W=0(dc).

Q.(3) Find the inverse fourier transform of 𝛿(w-𝑤0)

Solution. Inverse fourier transform is expressed as
1 ∞
𝐹−1[𝑋(𝑤)] = ∫ 𝑋(𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞

Or
1 ∞
− 𝑤0)] = 𝐹−1[𝛿(w
∫ (w − 𝑤0)𝑒𝑗𝑤𝑡 𝑑𝑤
2𝜋 −∞
Using shifting or sampling property of impulse function, we get
1
𝐹−1[𝛿(w − 𝑤0 )]=2𝜋 [ 𝑒 𝑗𝑤𝑡 ]at w = 𝑤0
1
𝐹−1[𝛿(w − 𝑤0 )]=2𝜋 [ 𝑒 𝑗 𝑤0 𝑡
]
1
F[ 𝑒𝑗 𝑤0 ]= 𝛿(w − 𝑤0)
2𝜋
1
𝑒𝑗 𝑤0 𝑡 (w − 𝑤0)
2𝜋
𝑂𝑟
𝑒𝑗 𝑤0 𝑡 2𝜋 (w − 𝑤0)
The above expression shows that the spectrum of an overlasting exponential 𝑒𝑗 𝑤0 𝑡 is a single impulse at w=0.
Similarly,
𝑒−𝑗𝑤0 𝑡 2𝜋 (w + 𝑤0)

 Fourier Transform of Cosine wave

Q.4 Find the fourier transform of overlasting sinusoid cos𝑤0𝑡.

Solution: We know that Euler’s identity is given by
𝑒𝑗𝜃 =cos𝜃 +jsin𝜃
And 𝑒−𝑗𝜃 =cos𝜃 –jsin𝜃
Hence 2cos𝜃=𝑒𝑗𝜃 +𝑒−𝑗𝜃 Or

𝑒 𝑗𝜃 +𝑒−𝑗𝜃
cos𝜃= 2
And
2jsin𝜃=𝑒𝑗𝜃 -𝑒−𝑗𝜃 Or

𝑗𝜃 +𝑒−𝑗𝜃
sin𝜃=𝑒
2𝑗
𝑒 𝑗 𝑤 0𝑡+𝑒−𝑗 𝑤 0𝑡
Hence, cos𝑤0𝑡= 2
We know that

𝑒𝑗 𝑤0 𝑡 2𝜋 (w − 𝑤0)
𝐴𝑛𝑑 𝑒−𝑗𝑤0 𝑡 ) 2𝜋 (w + 𝑤0
1
So that cos𝑤0𝑡 [2𝜋 (w − 𝑤0)+ 2𝜋 𝛿(w + 𝑤0]
2
Or
cos𝑤0𝑡 [𝜋 (w − 𝑤0)+ 𝜋 𝛿(w + 𝑤0]

 Fourier Transform of Periodic Function

Fourier transform of periodic function could also be found out. This means that Fourier transform may be used
as a universal mathematical tool to analyze both periodic and non-periodic waveform over the entire interval.
Let us find the fourier transform of periodic function x(t). x(t) may be expressed in terms of complex fourier
series as
x(t)=∑∞ 𝑛=−∞ 𝐶𝑛 𝑒
𝑗𝑛 𝑤0 𝑡

Taking fourier transform of both the side

∞
F[x(t)]=F[∑𝑛=−∞ 𝐶𝑛 𝑒 𝑗𝑛 𝑤0 𝑡 ] =∞
∑ 𝑛=−∞ 𝐶𝑛 . [1. 𝑒
𝑗𝑛 𝑤0 ]

Using frequency shifting shifting theorem, we can write

[1. 𝑒𝑗𝑛 𝑤0 ]=2𝜋 𝛿(w − 𝑛𝑤0)

Hence, F[x(t)]= ∑𝑛=−∞

∞ 𝐶𝑛 2𝜋 (w − 𝑛𝑤0 )=2𝜋 ∑𝑛=−∞
∞ 𝐶𝑛 (w − 𝑛𝑤0)
Hence, the fourier transform of a periodic function consist of a train of equally spaced impulses. These impulses
are located at the harmonic frequencies of the signal and the strength or area of each impulse is given by 2𝜋𝐶𝑛
.

 Convolution

Convolution is a mathematical operation used to express the relation between input and output of an LTI
system. It relates input, output and impulse response of an LTI system as
y(t)=x(t)∗h(t)
Where y (t) = output of LTI
x (t) = input of LTI
h (t) = impulse response of LTI
There are two types of convolutions:
 Continuous convolution
 Discrete convolution
Continuous Convolution

y(t)=x(t)∗h(t)
∞
= ∫−∞
∞
x(τ)h(t − τ)dτ
= ∫ x(t − τ)h(τ)dτ
−∞

A convolution is a mathematical operation that represents a signal passing through a LTI (Linear and Time-
Invariant) system or filter.

Discrete Convolution

y(n)=x(n)∗h(n)
=∑∞𝑘=−∞ 𝑥(𝑘)ℎ(𝑛 − 𝑘)
=∑∞𝑘=−∞ 𝑥(𝑛 − 𝑘)ℎ(𝑘)

Delta function, symbolized by the Greek letter delta, δ*n+ is a normalized impulse, that is, sample number zero has a value
of one, while all other samples have a value of zero. For this reason, the delta function is called the unit impulse.
Impulse response is the signal that exits a system when a delta function (unit impulse) is the input. If two
systems are different in any way, they will have different impulse responses. The input and output signals are
often called x[n] and y[n], the impulse response is usually given the symbol, h[n]. Any impulse can be
represented as a shifted and scaled delta function.
Properties of Convolution
1. Commutative Property: x1(t)∗x2(t)=x2(t)∗x1(t)
 2. Distributive Property: x1(t)∗[x2(t)+x3(t)]=[x1(t)∗x2(t)]+[x1(t)∗x3(t)]
3. Associative Property: x1(t)∗[x2(t)∗x3(t)]=[x1(t)∗x2(t)]∗x3(t)
4. Shifting Property: x1(t)∗x2(t)=y(t)
x1(t)∗x2(t−t0)=y(t−t0)
x1(t−t0)∗x2(t)=y(t−t0)
x1(t−t0)∗x2(t-t1)=y(t−t0−t1)
5. Convolution with Impulse: x1(t)∗δ(t)=x(t)
x1(t)∗δ(t−t0)=x(t−t0)
6. Convolution of Unit Steps: u(t)∗u(t)=r(t)
u(t−T1)∗u(t−T2)=r(t−T1−T2)
u(n)∗u(n)=[n+1]u(n)
7. Scaling Property: If x(t)∗h(t)=y(t)
1
then x(at)∗h(at)= |a| y(at)
8. Differentiation of Output: If y(t)=x(t)∗h(t)
dy (t) 𝑑𝑥 (𝑡)
then 𝑑𝑡 = ∗h(t)
𝑑𝑡
or
dy (t) 𝑑ℎ(𝑡)
= 𝑥(t) ∗
𝑑𝑡 𝑑𝑡
Note:
 Convolution of two causal sequences is causal.
 Convolution of two anti causal sequences is anti causal.
 Convolution of two unequal length rectangles results a trapezium.
 Convolution of two equal length rectangles results a triangle.
 A function convoluted itself is equal to integration of that function.

Limits of Convoluted Signal

If two signals are convoluted then the resulting convoluted signal has following range:
Sum of lower limits < t < sum of upper limits

Here, we have two rectangles of unequal length to convolute, which results a trapezium.
The range of convoluted signal is:
Sum of lower limits < t < sum of upper limits
−1+−2<t<2+2
−3<t<4
Hence the result is trapezium with period 7.

Area of Convoluted Signal

The area under convolved signal is given by Ay=AxAh
Where Ax = area under input signal
Ah = area under impulse response
Ay = area under output signal
∞
Proof: y(t)= = ∫−∞ x(τ)h(t − τ)dτ
Take integration on both sides
∞
∫ y(t)dt = ∫ ∫−∞ x(τ)h(t − τ)dτ dt
∞
=∫ x(τ)dτ ∫−∞ h(t − τ)dt
We know that area of any signal is the integration of that signal itself.

∴Ay=Ax Ah
DC Component
DC component of any signal is given by
DC component=area of the signal/period of the signal

NOTE:
1. The temporal output is the temporal input CONVOLVED with the Impulse Response Function.
2. The frequency domain output is the frequency domain input MULTIPLIED by the Transfer Function.
3. The frequency domain signal is the Fourier Transform of the temporal signal
𝐹𝑇
Mathematically, it must be that the FT of a convolution is a product. y (t) = x(t) ∗ h(t) − Y (ω) = X(ω)H(ω)

 Convolution Theorems:

Convolution of signals may be done either in time domain or frequency domain. So there are following two
theorems of convolution associated with Fourier transforms:
1. Time convolution theorem
2. Frequency convolution theorem

Time convolution theorem: The time convolution theorem states that convolution in time domain is equivalent
to multiplication of their spectra in frequency domain.
Mathematically, if
x1(t)↔X1(⍵)
And x2(t)↔X2(⍵)
Then x1(t) * x2(t)↔ X1(⍵)X2(⍵)
Proof:
F[x (t) * x (t)] = ∫∞ [x₁(t) ∗ x₂(t)] 𝑒 −𝑗𝑤𝑡 𝑑𝑡
1 2 −∞ ∞
We have x1(t) * x2(t) = ∫−∞∞
[x₁(τ)x₂(t
∞ − τ)]𝑑𝑟
F[x (t) * x (t)]= ∫ {∫ [[x₁(τ)x₂(t − τ)] dτ}𝑒 −𝑗𝑤𝑡 𝑑𝑡
1 2 −∞ −∞
Interchanging the order of integration, we have −
F[x (t) * x (t)]= ∫∞ x₁(τ) ∫ ∞ [x₂(t − τ)]𝑒 𝑗𝑤𝑡 𝑑𝑡] dτ
1 2 −∞ −∞
Letting t-𝑟 = p, in the second integration, we have t=p+𝑟 and dt = dp
∞ ∞
F[x1(t) * x2(t)]= ∫−∞ x₁(τ) ∫−∞[x2 (p)e−jw (p+τ)dp] dτ
F[x1(t) * x2(t)]= ∫∞ x₁(τ) ∫∞ [x2(p)e−jwp dp]𝑒 −𝑗𝑤τ dτ
−∞ −∞
F[x (t) * x (t)]= ∫ ∞ 𝑥 (τ) X (w) 𝑒 −𝑗𝑤τ dτ
1 2 −∞ 1 2
F[x (t) * x (t)]= ∫ ∞ 𝑥 (τ) 𝑒 −𝑗𝑤τ dτ X (w)
1 2 −∞ 1 2
x₁(t) * x₂(t)↔ X₁(⍵)X₂(⍵)
This is time convolution theorem
 Frequency convolution theorem:

The frequency convolution theorem states that the multiplication of two functions in time domain is
equivalent to convolution of their spectra in frequency domain.
Mathematically, if
x₁(t)↔X₁(⍵)

And x₂(t)↔X₂(⍵)

Then x₁(t) x₂(t)↔ 1/ 2𝜋 * X₁(⍵) * X₂(⍵)]

Proof:
∞ −
F*x₁(t) x₂(t)+ = ∫−∞[x₁(t) x₂(t)] 𝑒 𝑗⍵𝑡𝑑t
By definition of inverse Fourier transform

= ∫∞ [ 1 ∫∞ x (λ)ejλtdλ] x₂(t)] 𝑒 −𝑗⍵𝑡𝑑t

−∞ 2π −∞ 1
Interchanging the order of integration, we get
1 ∞ ∞ −𝑗⍵𝑡 jλt
= ∫ x (λ)[ ∫ x₂(t) 𝑒 e dλ] 𝑑t
2π −∞ 1 −∞
1 ∞ ∞
= ∫ x (λ)[ ∫ x₂(t) 𝑒 –(⍵-λ)𝑡dt] 𝑑λ
2π −∞ 1 −∞
1 ∞
= ∫ x (λ) X (⍵-λ ) 𝑑λ
2
2π −∞ 1

1
= [X1(⍵) * X2(⍵)]
2π

This is frequency convolution theorem in radian frequency in terms of frequency, we get

F*x₁(t) x₂(t)+= X₁(f) * X₂(f)

 Parseval’s Energy theorem:

For continuous time signals x(t) energy of the signal is expressed as:
∞
E=∫−∞ 𝑥2(𝑡)𝑑𝑡
According to Parseval’s theorem
1 ∞
E= ∫−∞ |(𝑗𝑤)|2𝑑𝑤
2𝜋
Let x*(t) is conjugate of x(t).
Therefore, x(t) . x*(t)= |𝑥(𝑡)|2 .................. (1)
On∞
integrating the
∞
above equation with respect to t, we get
|(𝑡)| 2
∫ 𝑑𝑡 =∫ x(t) . x∗(t)𝑑𝑡........... (2)
−∞ −∞
From the definition of inverse Fourier transform
1 ∞
(𝑡) = ∫ 𝑋(𝑗𝑤)𝑒𝑗𝑤𝑡 𝑑𝑤 ………….(3)
2𝜋 −∞
Taking conjugate to both sides
1 ∞
x*(t)= ∫ 𝑋∗(𝑗𝑤)𝑒−𝑗𝑤𝑡 𝑑𝑤……………(4)
2𝜋 −∞
Putting the above value in equation (2)
 ∞ ∞ 1 ∞
∫ |(𝑡)|2𝑑𝑡 =∫ x(t) . ∫ 𝑋∗(𝑗𝑤)𝑒−𝑗𝑤𝑡 𝑑𝑤 𝑑𝑡
−∞ −∞ 2𝜋 −∞
Changing
∞
the order
1 ∞
of integration
∞
we get
∫ |(𝑡)|2𝑑𝑡 = ∫ 𝑋∗(𝑗𝑤) ∫ x(t) 𝑒−𝑗𝑤𝑡 𝑑𝑡 𝑑𝑤
−∞ −∞ −∞
∞ 12𝜋 ∞
∫ |(𝑡)|2𝑑𝑡 = ∫ 𝑋∗(𝑗𝑤)𝑋(𝑗𝑤) 𝑑𝑤
−∞ 2𝜋 −∞

∞
1 ∞
∫ |𝑥(𝑡)|2𝑑𝑡 = ∫ |(𝑗𝑤)|2𝑑𝑡
−∞ 2𝜋 −∞
Hence proved.
 UNIT-II
Amplitude modulation: Modulation, need of modulation, types of modulation techniques, amplitude
modulation (DSB-FC), modulation index, frequency spectrum of AM wave, linear and over modulation, power
relation in AM, transmission efficiency, modulation by a complex signal, bandwidth of AM, AM modulators,
square law and switching modulator, advantages and disadvantages of AM. Demodulation of AM: Suppressed
carrier amplitude modulation systems, DSB-SC, SSB-SC, VSB-SC systems, comparison of various amplitude
modulation systems. Demodulation of AM, square law and envelope detector, synchronous detection of AM,
Low and high power AM transmitters, AM receivers, TRF and superheterodyne receivers, sensitivity, selectivity
and fidelity of receivers.
Course Objective:-
The objective of this course is to familiar with time and frequency representation of information is given.
Course Outcomes:-
At the end of the course student will be able to :
2. Understand the communication of information over the communication channel.

 Modulation:
Modulation is a technique used to convert a low frequency message signal to a higher frequency modulated
signal using a higher frequency carrier.
Definition: Modulation is the process of changing the parameters of the carrier signal, in accordance with the
instantaneous values of the modulating signal.
Signals in the Modulation Process:
1. Message or Modulating Signal
The signal which contains a message to be transmitted is called as a message signal. It is a baseband signal,
which has to undergo the process of modulation, to get transmitted. Hence, it is also called as the modulating
signal.
Baseband signal: Baseband refers to the original frequency range of a transmission signal before it is converted,
or modulated, to a different frequency range.
2. Carrier Signal
The high frequency signal which has a certain phase, frequency, and amplitude but contains no information is
called a carrier signal. It is an empty signal. It is just used to carry the signal to the receiver after modulation.
3. Modulated Signal
The resultant signal after the process of modulation is called as the modulated signal. This signal is a
combination of the modulating signal and the carrier signal.
Signal Bandwidth:
The bandwidth of a signal represents the range of its frequency components. A complex signal is made of a
range of frequencies called spectrum. The Bandwidth of a signal is calculated by subtracting the highest
frequency component from the lowest frequency component.
Demodulation: It is the reverse process of modulation, which is used to get back the original message signal.
Modulation is performed at the transmitting end whereas demodulation is performed at the receiving end.

 Need for modulation:

The baseband signals are incompatible for direct transmission. When the signal is transmitted without
modulation they cannot travel longer distances as low frequency signal get it attenuates, so its strength has to
be increased by modulating with a high frequency carrier wave, which doesn’t affect the parameters of the
modulating signal.
Modulation is needed to achieve the following basic needs:

1. Practicability of antennas: For the transmission of radio signals, the antenna height must be multiple of
λ/4 ,where λ is the wavelength.
λ = c /f
Where c: is the velocity of light
f: is the frequency of the signal to be transmitted
The minimum antenna height required to transmit a baseband signal of f = 10 kHz is 7.5 Km.
The antenna of this height is practically impossible to install.
Now, let us consider a modulated signal at f = 1 MHz The minimum antenna height is 75 meters.
This antenna can be easily installed practically. Thus, modulation reduces the height of the antenna.

2. Avoids mixing of signals

If the baseband sound signals are transmitted without using the modulation by more than one transmitter, then
all the signals will be in the same frequency range i.e. 0 to 20 kHz. Therefore, all the signals get mixed together
and a receiver cannot separate them from each other. If each baseband sound signal is used to modulate a
different carrier then they will occupy different slots in the frequency domain i.e. through different channels.
Thus, modulation avoids mixing of signals.
3. Multiplexing is possible: Multiplexing is a process in which two or more signals can be transmitted over the
same communication channel simultaneously. If transmitted without modulation, the different message signals
over a single channel will interfere with each other. So multiplexing helps in transmitting a number of messages
simultaneously over a single channel which reduces cost of installation and maintenance of more channels.
4. Narrow banding: The frequency translation through modulation converts a wideband signal to a narrowband,
which is termed as narrow banding.
Let us assume a system is radiating directly with the frequency range from 50 Hz to 10 kHz, the ratio of highest
to lowest wavelength is 200. If antenna is designed for 50 Hz, it will be too long for 10 kHz and vice versa. But if
signal is translated to higher frequency of 1 MHz range using modulation, then the ratio of lowest to highest
106 + 50
frequency will be ≈ 1 and the same antenna will be suitable for the entire band.
10 6+ 10 4
5. Improves Quality of Reception
6. Increase the Range of Communication

 Types of Modulation:
The types of modulations are broadly classified into continuous-wave modulation and pulse modulation.
Continuous-wave Modulation
In the continuous-wave modulation, a high frequency sine wave is used as a carrier wave. This is further divided
into amplitude and angle modulation.
• If the amplitude of the high frequency carrier wave is varied in accordance with the instantaneous
amplitude of the modulating signal, then such a technique is called as Amplitude Modulation.
• If the angle of the carrier wave is varied, in accordance with the instantaneous value of the modulating
signal, then such a technique is called as Angle Modulation.
The angle modulation is further divided into frequency and phase modulation.
• If the frequency of the carrier wave is varied, in accordance with the instantaneous value of the
modulating signal, then such a technique is called as Frequency Modulation.
• If the phase of the high frequency carrier wave is varied in accordance with the instantaneous value of
the modulating signal, then such a technique is called as Phase Modulation.
 Figure Types of modulation

 Pulse Modulation
In Pulse modulation, a periodic sequence of rectangular pulses is used as a carrier wave. This is further divided
into analog and digital modulation.
 In analog modulation technique, if the amplitude, duration or position of a pulse is varied in accordance
with the instantaneous values of the baseband modulating signal, then such a technique is called as Pulse
Amplitude Modulation (PAM) or Pulse Duration/Width Modulation (PDM/PWM), or Pulse Position
Modulation (PPM).
 In digital modulation, the modulation technique used is Pulse Code Modulation (PCM) where the analog
signal is converted into digital form of 1s and 0s. As the resultant is a coded pulse train, this is called as PCM.
This is further developed as Delta Modulation (DM), which will be discussed in subsequent chapters. Hence,
PCM is a technique where the analog signals are converted into a digital form.
 Amplitude modulation
Definition:
The amplitude of the carrier signal varies in accordance with the instantaneous amplitude of the modulating
signal i.e. the amplitude of the carrier signal containing no information varies as per the amplitude of the signal
containing information, at each instant.

Mathematical expression:
Let m (t) is the baseband message and C (t) = Ac Cos (ωct) is called the carrier wave. The carrier frequency, fc
should be larger than the highest spectral component in m(t).

Consider a sinusoidal carrier signal C (t) is defined as

C (t) = Ac Cos (2πfct +Φ) t
Where Ac= Amplitude of the carrier signal
fc= frequency of the carrier signal
Φ = Phase angle.
For convenience, assume the phase angle of the carrier signal is zero. An amplitude-modulated (AM) wave, S(t)
can be described as function of time is given by
S(t) = Ac [1+kam (t)+ Cos (2πfct)
Where the parameter ka is a positive constant called the amplitude sensitivity of the modulator.
Let e(t) = Ac|1 + ka m(t)| is called the envelope of the AM signal. When fc is large relative to the bandwidth of
m(t), the envelope is a smooth signal that passes through the positive peaks of S(t) and it can be viewed as
modulating the amplitude of the carrier wave in a way related to m(t) as shown in figure.

Figure. Amplitude modulation envelope in time domain

The amplitude modulated (AM) signal consists of both modulated carrier signal and un-modulated carrier signal.
There are two requirements to maintain the envelope of AM signal is same as the shape of base band signal.
1. The amplitude of the ka m(t) is always less than unity i.e., |ka m(t)|<1 for all ‘t’.
2. The carrier signal frequency fc is far greater than the highest frequency component W of the message
signal m (t) i.e., fc>>W
Assume the message signal m (t) is band limited to the interval –W <f < W

Fig..Spectrum of message signal

The spectrum of AM is shown in fig..The Fourier transform of AM signal S (t) is

Ac 𝐾 𝐴
S (f) = *δ(f-fc)+ δ (f+fc)++ 𝑎 𝑐 [M(f-fc)+ M(f+fc)]
2 2
 Fig. Spectrum of AM signal

Fig. Spectrum of AM signal representing sidebands

The AM spectrum consists of two impulse functions which are located at f c and -fc and weighted by Ac/2, two
USBs, band of frequencies from fc to fc +W and band of frequencies from -fc-W to –fc, and two LSBs, band of
frequencies from fc-W to fc and -fc to -fc+W.

The difference between highest frequency component and lowest frequency component is known as
transmission bandwidth.
B = 2W
 Figure.Amplitude modulation waveform in time domain

 Single-tone modulation
In single-tone modulation modulating signal consists of only one frequency component where as in multi-tone
modulation modulating signal consists of more than one frequency component.
Mathematical Expressions:
Following are the mathematical expressions for these waves.
Time-domain Representation of the Waves
Let the modulating signal be,
m (t) = Am Cos 2πfmt (3.1)

and the carrier signal be,

C(t)=Ac Cos(2πfct) (3.2)

Where,
Am and Ac are the amplitude of the modulating signal and the carrier signal respectively.
fm and fc are the frequency of the modulating signal and the carrier signal respectively.
The equation for the overall modulated signal is obtained by multiplying the carrier and the modulating signal
together.
S (t) = Ac [1+ka m(t)+ Cos(2πfct) (3.3)
Substituting in the individual relationships for the carrier and modulating signal in equation (3.3), the overall
signal becomes:
S (t) = Ac [1+ka Am Cos 2πfmt+ Cos (2πfct)
Replace the term ka Am by µ which is known as modulation index or modulation factor.
Or it can be written as
S (t)= [Ac+ Am Cos(2π fmt)+ Cos(2π fct) (3.4)
Modulation Index:
A carrier wave, after being modulated, if the modulated level is calculated, then such an attempt is called as
Modulation Index or Modulation Depth. Modulation index can be defined as the measure of extent of
amplitude variation about an un-modulated carrier.
Rearrange the Equation 4 as below.
S (t) = Ac[1 +(Am/Ac) Cos (2π fmt)] Cos(2π fct) (3.5)

S (t) = Ac[1 +µ Cos (2π fmt)] Cos(2π fct) (3.6)

Where, μ is Modulation index or Amplitude sensitivity of the modulator and it is equal to the ratio of Am and Ac.
Mathematically, we can write it as

μ = (Am/Ac)
Calculating the modulation index from AM envelope:
With reference to the figure 3.7 and 3.8, we can calculate the modulation index from the modulated waveform.
We know that μ = (Am/Ac)

Am = (Amax-Amin)/2 (3.8)

Ac = Amax -Am (3.9)

By substituting (3.8) equation in equation (3.9) we get

Ac =Amax - (Amax-Amin)/2 (3.10)

By diving (3.8) and (3.10) equation we get

μ = (Am/Ac)=𝐴𝑚𝑎𝑥 −𝐴𝑚𝑖𝑛 (3.11)

𝐴𝑚𝑎𝑥 +𝐴𝑚𝑖𝑛
Where
Amax = maximum amplitude of the modulated carrier signal

Amin = minimum amplitude of the modulated carrier signal

 Figure AM envelope

Modulation index µ has to be governed such that it is always less than unity; otherwise it results in a situation
known as ‘over-modulation’ (µ >1). The over-modulation occurs, whenever the magnitude of the peak
amplitude of the modulating signal exceeds the magnitude of the peak amplitude of the carrier signal. The
signal gets distorted due to over modulation. Because of this limitation on‘µ ’, the system clarity is also limited.
The AM waveforms for different values of modulation index m are as shown in figure 3.9.

If µ = 0 we haven't modulating wave, then no information is transmitted while engaging the channel with the
carrier.

If µ= 1 we have the maximum of modulation. When the modulation index is 1, i.e. a modulation depth of 100%,
the carrier level falls to zero and rise to twice its non-modulated level.
We are in optimal conditions if µ = 0.5.

If µ > 1 then we have strong crossover distortion. Any increase of the modulation index above 1.0, i.e. 100%
modulation depth causes over-modulation. The carrier experiences 180° phase reversals where the carrier level
would try to go below the zero point. These phase reversals give rise to additional sidebands resulting from the
phase reversals (phase modulation) that extend out, in theory to infinity. This can cause interference to other
users if not filtered.
 Figure AM waveforms for different values of µ

S (t) = Ac Cos (2πfct) + Acµ/2*cos2π (fc+ fm)t]+ Acµ/2*cos2π (fc-fm)t] (3.12)

• Looking at equation (3.12) we can say that 1st term represents un-modulated carrier and two additional
terms represents two sidebands
• The frequency of the lower sideband (LSB) is fc –fm and the frequency of the upper sideband (USB) is fc+
fm
Fourier transform of S (t) is
S (f) =Ac/2*δ(f-fc) + δ (f+fc)] +Acµ/4*δ (f-fc-fm) + δ (f+fc+fm)] + Acµ/4*δ (f- fc+fm ) + δ (f+fc-fm)] (3.13)

Bandwidth of AM wave:
• We know bandwidth can be measured by subtracting lowest frequency of the signal from highest
frequency of the signal
• For amplitude modulated wave it is given by
BW = fUSB - fLSB
= (fc + fm) – (fc -fm)
=2 fm
Therefore the bandwidth required for the amplitude modulation is twice the frequency of the modulating signal.

Figure Spectrum of Single tone AM signal

Power calculations of single-tone AM signal:

The standard time domain equation for single-tone AM signal is given by equation 3.12

S (t) = Ac Cos (2πfct)+Acµ/2*cos2π (fc+ fm)t]+ Acµ/2*cos2π (fc-fm)t] (3.12)

We have seen that AM wave has three components:

• Un-modulated carrier
 • Lower sideband
• Upper sideband

Therefore the total power of AM wave is the sum of the carrier power Pc and Power in the two sidebands PUSB
and PLSB. It is given as

Power of any signal is equal to the mean square value of the signal
Carrier power Pc = Ac2 /2
Upper Side Band power PUSB = Ac2 2 /8
Lower Side Band power P LSB = Ac22 /8
Total power PT = Pc + PLSB + PUSB
Total power PT = Ac2 /2 + A c2 2 /8 + A c2 2 /8
PT = Pc [1+ 2 /2]

Multi-tone modulation:
In multi-tone modulation modulating signal consists of more than one frequency component where as in single-
tone modulation modulating signal consists of only one frequency component.

Mathematical Expression
Let us consider that a carrier signal Ac Cos(2π fct) is modulated by a baseband or modulating signal m(t) which is
expressed as :

m (t) = Am1 Cos (2π fm1t) + Am1 Cos (2π fm2t) (3.14)

We know that the general expression for AM wave is

S (t) = Ac Cos (2π fct) + m(t) Cos (2π fct)

Putting the value of x(t), we get

S (t) = Ac Cos (2π fct) + [Am1 Cos (2π fm1t) + Am2 Cos (2π fm2t)] Cos (2π fct) (3.15)
or it can be written as

S (t) = Ac [1 + Ka Am1 Cos (2π fm1t) + Ka Am2 Cos (2π fm2t)] Cos (2π fct) (3.16)

Replace Ka Am1 by µ1 and Ka Am2 by µ2

So finally we get
𝐴 𝑐 µ1 𝐴 𝑐 µ1 𝐴 𝑐 µ2
S (t) = Ac Cos (2π fct) + 2
[cos 2π (fc +fm1 )t] + 2
[cos 2π(fc - fm1 )t] + 2
[cos 2π (fc +fm2 )t]
𝐴 𝑐 µ2
+ 2
[Cos 2π (fc - fm2)t] ( 3.17)

Power of multi-tone AM signal is given by:

PT = Pc [1+ µ12 /2 + µ22 /2+ .............. + µn2/2]

Where Pt = Total power
Pc = Carrier power
PT = Pc [1+ µt2 /2]
Where µt = √µ2 + µ2 + ⋯ + µ2
1 2 𝑛
Fourier 𝐴transform of S(t) is 𝐴 𝑐 µ1 𝐴 𝑐 µ1 𝐴𝑐 µ 2
S(f) = 𝑐 *δ(f-f )+ δ(f+f )] + *δ(f-fc-f )+ δ(f+f +f )] + *δ(f-f +f )+ δ(f+f -f )] + *δ(f-f -f )+
2 c c 4 m1 c m1 4 c m1 c m1 4 c m2
δ(f+f +f )]+ 𝐴𝑐 µ2 *δ(f-f +f )+ δ(f+f -f )]
c m2 4 c m2 c m2

Figure Spectrum of Multi tone AM signal

Transmission efficiency:
Transmission efficiency is defined as the ratio of total side band power to the total transmitted power.
The yield of modulation is defined therefore as the ratio between the transmitted information signal strength
content in one of the two side lines, divided by all the power you must transmit.

PLSB +PUSB
η=
𝑃𝑇
µ2
η= X 100 % (3.18)
(2+ µ 2)

The transmission efficiency (η) of AM wave is defined as the percentage of total power contributed by side
bands of the AM signal. The maximum transmission efficiency of an AM signal is 33.33%, i.e., only one third of
the total transmitted power is carried by the side bands in an AM wave. The remaining two third of the total
transmitted power gets wasted.

Advantages of Amplitude modulation:

Generation and detection of AM signals are very easy
It is very cheap to build, due to this reason it is most commonly used in AM radio broad casting

Disadvantages of Amplitude of modulation:

Amplitude modulation is wasteful of power
Amplitude modulation is wasteful of band width

Modulation by a complex signal

A complex carrier signal c(t), at a carrier frequency ωc , is described mathematically as the complex exponential
C(t) = 𝑒(𝑗 𝜔 𝑐 𝑡+ 𝛿)
For convenience we choose the initial time so that the phase (δ) is zero. Then, if m(t) is the signal or information
that is to be transmitted by the carrier, the signal m(t) is encoded onto the carrier by multiplying the carrier by
m(t)
S(t) = m(t) c(t)
S(t) = 𝑚(𝑡)𝑒(𝑗 𝜔𝑐 𝑡)
The carrier’s amplitude is modulated by the signal m(t). Now we know that multiplication in the time domain is
equivalent to convolution in the frequency domain. Thus, the Fourier transform of the signal s(t) is the
convolution of the Fourier transforms of m(t) and c(t).
S(ω) = M(jω) * C(jω)
S(ω) = 1 ∞ (𝑗𝜔)((𝜔 − 𝜔0 ))0
2𝜋 −∞
Earlier we took the Fourier transform of a complex exponential and determined it is a delta function
C(jω) = 2π δ(ω-ωc)
and upon substitution into the convolution equation we obtain
S(ω) = M(j(ω-ωc))
Thus, as a result of modulation, the transform of the signal m(t) is shifted on the frequency axis by the carrier
frequency. We can visualize the situation by considering the magnitude of M (jω). We suppose that the signal m(t)
is a real function of time and that its frequency content is bounded by some maximum frequency ωm . Hence, all
of the signal power lies in the range ± ωm, as depicted in the figure 3.12 below. The second figure depicts the
delta function at ωc and the third figure shows the result of amplitude modulation.

Figure Complex AM spectrum

Generation of AM waves
The amplitude modulator is a circuit which generates amplitude modulated signal. In the process of modulation
the frequency spectrum gets translated. The output of the modulator contains the frequencies which are
different from those present in the input signal. The amplitude modulator therefore must be time varying linear
systems such as switching or chopping circuit are a non linear time in varying system. The reason for this is that
a linear time invariant system cannot produce new frequencies in its output. Here two methods for generating
AM waves:
 1. The square law are power law modulator
2. Switching modulator.
These two methods require non linear element as active device for generating AM signals. These two methods
are use full in the low power generation of amplitude modulated waves.

Square-law modulator:
It consists of the following:
1. A non-linear device
2. A band pass filter
3. A carrier source and modulating signal
The modulating signal and carrier are connected in series with each other and their sum V 1(t) is applied at the
input of the non-linear device semi-conductor diodes and transistors are the most common nonlinear devices
used for implementing square law modulators. The filtering requirement is usually satisfied by using a single or
double tuned filters.
When a nonlinear element such as a diode is suitably biased and operated in a restricted portion of its
characteristic curve, that is ,the signal applied to the diode is relatively weak, we find that transfer characteristic
of diode-load resistor combination can be represented closely by a square law.

Figure Square law modulator

The input output relation for non-linear device is as under:
V0 (t) = a1Vi (t) + a2 V i2 (t) (3.19)
Where a1, a2 are constants now, the input voltage Vi (t) is the sum of both carrier and message signals

i.e., Vi (t) =Ac Cos (2πfct)+m (t) (3.20)

Substitute equation (3.20) in equation (3.19) we get

V0 (t) =a1Ac Cos (2πfct) +a1m (t) +a2 [Ac Cos (2πfct)+m (t)]2 (3.21)

V0 (t) =a1Ac Cos(2πfct )+a1m (t) +a2Ac 2 cos2 (2πfct)+ a2m2 (t) + 2 a2 Ac Cos (2πfct) m(t) (3.22)

The five terms in the expression for V0(t) are as under :

Term 1: a1m (t): Modulating Signal
Term 2: a1Ac Cos (2πfct): Carrier Signal
Term 3: a2m2 (t): Squared modulating Signal
Term 4: 2 a2 Ac Cos (2πfct) m(t): AM wave with only sidebands
Term 5: a2Ac 2 cos2 (2πfct) + a2m2 (t): Squared Carrier
Out of these five terms, terms 2 and 4 are useful whereas the remaining terms are not useful.

Let us combine terms 2, 4 and 1, 3, 5 as follows to get,

V0 (t) = {a1m (t) +a2Ac 2 cos2 (2πfct) + a2m2 (t)} + {a1Ac Cos(2πfct )+2 a2 Ac Cos (2πfct) m(t)} (3.23)

Now design the tuned filter /Band pass filter with center frequency fc and pass band frequency width 2W. We
can remove the unwanted terms by passing this output voltage V 0(t) through the band pass filter and finally we
will get required AM signal.
𝑎
V0 (t) =a1Ac [1+ 2𝑎2 (𝑡)+ Cos(2πfct ) (3.24)
𝑎 1
2
Where Ka= 2𝑎
1
Assume the message signal m (t) is band limited to the interval –W ≤f ≤W

Figure Spectrum of message signal

Spectrum of AM can represented a one shown in figure 3.15.The Fourier transform of output voltage VO (t) is
given by
VO (f) = a1AC/2[(f-fc) + (f+fc)] +a2 AC [M (f-fc) + M (f+fc)] (3.25)

Figure Spectrum of AM

The AM spectrum consists of two impulse functions which are located at f c & -fc and weighted by Aca1/2 &
a2Ac/2, two USBs, band of frequencies from fc to fc +W and band of frequencies from -fc-W to –fc, and two LSBs,
band of frequencies from fc-W to fc & -fc to -fc+W.

Switching Modulator:
In switching modulator the diode has to operate as an ideal switch as one shown in figure 3.16. Let the
modulating and carrier signals be denoted as m(t) and c(t)=Ac Cos(2πfct) respectively.
Working of circuit:
  The two signals i.e. modulating and carrier signals are applied as inputs to the summer (adder) block.
 Assume that carrier wave C(t) applied to the diode is large in amplitude, so that it swings right across
the characteristic curve of the diode and also the diode acts as an ideal switch, that is, it presents zero
impedance when it is forward-biased and infinite impedance when it is reverse-biased.
 We may thus approximate the transfer characteristic of the diode-load resistor combination by a
piecewise-linear characteristic. Summer block produces an output, which is the addition of modulating
and carrier signals.
 During the positive half cycle of the carrier signal i.e. if C (t)>0, the diode is forward biased, and then the
diode acts as a closed switch. Now the output voltage Vo (t) is same as the input voltage Vi (t) .
 During the negative half cycle of the carrier signal i.e. if C (t) <0, the diode is reverse biased, and then the
diode acts as an open switch. Now the output voltage V O (t) is zero i.e. the output voltage varies
periodically between the values input voltage Vi (t) and zero at a rate equal to the carrier frequency fc.

Figure Switching modulator

Mathematically, we can write it as
The input voltage applied Vi (t) applied to the diode is the sum of both carrier and message signals.

Vi (t) =Ac Cos (2πfct)+m (t) (3.26)

Vo (t) = [Ac Cos (2πfct) +m (t)] gP(t) (3.27)

Where gp(t) is the periodic pulse train with duty cycle one-half and period
Tc=1/fc and which is given by
1 2 ∞ (−1)𝑛 −1
∑
gp(t) = + 𝑛=1 2𝑛−1 Cos*2πfct(2n-1)] (3.28)
2 𝜋

Figure Pulse train

Substituting gp(t) into equation (3.27), we get
1 1 2
(2πfct) + 2𝐴𝜋 𝑐 𝑐𝑜𝑠2 (2πfct)
Vo (t) = m(t) + Ac Cos (2πfct) + 𝑚(𝑡) cos⁡ (3.29)
2 2 𝜋

The odd harmonics in this expression are unwanted, and therefore, are assumed to be eliminated. In this
expression, the first and the fourth terms are unwanted terms whereas the second and third terms together
represent the AM wave.
Combining the second and third terms together, we obtain
 𝐴𝑐 4
Vo (t) = [1 + (𝑡) ]Cos (2πfct) + unwanted terms (3.30)
2 𝜋𝐴𝑐

This is the required expression for the AM wave with µ= *4/πEc].

The unwanted terms can be eliminated using a band-pass filter (BPF). Now design the tuned filter /Band pass
filter with center frequency fc and pass band frequency width 2W.We can remove the unwanted terms by
passing this output voltage V0(t) through the band pass filter and finally we will get required AM signal.
Assume the message signal m(t) is band limited to the interval –W ≤f ≤W as one shown in figure 3.18

M(f)

Figure Spectrum of message signal

The spectrum of Am signal is shown in figure 3.19.The Fourier transform of output voltage VO (t) is given by

VO (f) = AC/4*δ(f-fc) + δ(f+fc)+ +AC/π *M (f-fc) + M (f+fc)] (3.31)

Figure Spectrum of AM signal

The AM spectrum consists of two impulse functions which are located at fc & -fc and weighted by Aca1/2 &
a2Ac/2, two USBs, band of frequencies from fc to fc +W and band of frequencies from -fc-W to –fc, and two LSBs,
band of frequencies from fc-W to fc & -fc to -fc+W.

Advantages:
1. It is very simple to design and implement
2. It can be demodulated using a circuit consisting of very few components
3. AM receivers are very cheap as no specialised components are needed.
4. AM signal are reflected back to earth from ionosphere layer. Due to this fact, AM signals can reach far places
which are thousands of miles from source. Hence AM radio has coverage wider compare to FM radio.
Disadvantage:
1. Due to large time constant, some distortion occurs which is known as diagonal clipping i.e., selection of time
constant is somewhat difficult
2. The most natural as well as man-made radio noise are of AM type. The AM receivers do not have any means to
reject this kind of noise.
 3. Weak AM signals have low magnitude compare to strong signals. This requires AM receiver to have circuitry to
compensate for signal level difference.
4. It is not efficient in terms of its use of bandwidth, requiring a bandwidth equal to twice that of the highest audio
frequency

Application:
 Broadcast transmissions: AM is still widely used for broadcasting on the long, medium and short wave bands.
 Air band radio: VHF transmissions for many airborne applications still use AM. . It is used for ground to
air radio communications as well as two way radio links for ground staff as well.

Suppressed carrier Amplitude modulation systems:

Objective: In full AM (DSB-AM), the carrier wave C (t) is completely independent of the message signal m(t),
which means that the transmission of carrier wave represents a waste of power. This points to a shortcoming of
amplitude modulation, that only a fraction of the total transmitted power is affected by m(t).Thus, the carrier
signals and one of the two sidebands may be removed or attenuated so the resulting signals will require less
transmitted power and will occupy less bandwidth, and yet perfectly acceptable communications will be
possible.

Double Sideband-Suppressed Carrier (DSBSC) Modulation

Double sideband-suppressed (DSB-SC) modulation, in which the transmitted wave consists of only the upper
and lower sidebands. Transmitted power is saved through the suppression of the carrier wave, but the channel
bandwidth requirement is same as in AM that is twice the bandwidth of the message signal.In power
calculation of AM signal, it has been observed that for single-tone sinusoidal modulation, the ratio of the total
power and carrier power is
𝑃𝑡 µ2
= [1 + ]
𝑃𝑐 2
𝑃𝑐 2
= x 100 % = 67%(for µ = 1)
𝑃𝑡 3

So for 100% modulation that is µ = 1, about 67% of the total power is wasted for transmitting carrier which
does not contain any information. So if carrier is suppressed, saving of two-third power may be achieved at
100% modulation.

Let m (t) be a band-limited baseband message signal with cutoff frequency W. The DSBSC-AM signal
corresponding to m (t) consists of the product of both the message signal m (t) and the carrier signal C (t),
as follows:

S (t) =C (t) m (t)

S (t) =Ac Cos (2πfct) m (t)

This is the same as AM except with the sinusoidal carrier component is eliminated.

The modulated signal S (t) undergoes a phase reversal whenever the message signal m (t) crosses zero. The
envelope of a DSB-SC modulated signal is different from the message signal. The transmission bandwidth
required by DSB-SC modulation can be seen from figure 4.2 which is same as that for amplitude modulation
that is twice the bandwidth of the message signal 2W.

Assume that the message signal is band-limited to the interval –W ≤f≤ W.

Figure Spectrum of message signal

Figure Spectrum of DSBSC signal

Single-tone modulation:

In single-tone modulation modulating signal consists of only one frequency component where as in multi-tone
modulation modulating signal consists of more than one frequency components.

The standard time domain equation for the DSB-SC modulation is given by

S (t) =Ac Cos (2πfct) m (t) (4.1)

m (t) =Am Cos (2πfmt) (4.2)

Substitute equation (4.2) in equation (4.1) we will get

S (t) =Ac Am Cos (2πfct) Cos (2πfmt)

𝐴𝑐 𝐴 𝑚
S (t) = *Cos 2π (f -f ) t + Cos 2π (f +f ) t] (4.3)
2 c m c m

The Fourier transform of S (t) is

𝐴𝑐 𝐴𝑚 𝐴𝑐 𝐴 𝑚
S (f) = *δ (f-f + f ) + δ(f+f +f )]
4
*δ (f-fc-fm) + δ (f+fc+fm)] + 4
c m c m
 Figure Spectrum of single tone DSBSC

Bandwidth:

The DSBSC modulated wave has only two frequencies. So, the maximum and minimum frequencies
are fc+fm and fc−fm respectively.

fmax=fc+fm and fmin=fc−fm

Substitute, fmax and fmin values in the bandwidth formula.

BW=fc+fm− (fc−fm)

BW=2fm

Power calculations of DSB-SC waves:-

Consider
𝐴 𝐴 the following equation of DSBSC modulated wave
S (t) = 𝑐 𝑚 *cos 2π (f -f ) t + Cos 2π (f +f ) t]
2 c m c m

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 cosine signal is

2
P = 𝑉𝑟𝑚𝑠
𝑅

Average power delivered to a 1ohm resistor can be calculated as,

PUSB = ( 𝑐 )2
2√2

PUSB = Am2Ac2/8
Similarly; P = ( 𝑐 )2 = A 2A 2/8
LSB m c
2√2

So total power PT =Ac2Am2/4

𝐴2 𝐴2
𝑃𝑈𝑆𝐵 𝑃𝐿𝑆𝐵 /8
= = 2 2
𝐴 𝐴/4
x 100 % = 50%
𝑃𝑇 𝑃𝑇

For the sinusoidal modulation, the average power in the lower or upper side-frequency with respect to the total
power in the DSB-SC modulated wave is 50%.

Generation of DSB-SC waves:

The generation of a DSB-SC modulated wave consists simply of the product of the message signal m(t) and the
carrier wave Ac Cos (2πfct). Devices for achieving this requirement is called a product modulator. There are two
methods to generate DSB-SC waves. They are:

 Balanced modulator
 Ring modulator
Balanced Modulator:

1. Balanced modulator consists of two identical AM modulators which are arranged in a balanced
configuration in order to suppress the carrier signal. Hence, it is called as balanced modulator as shown
in figure 4.4.
2. Assume that two AM modulators are identical, except for the sign reversal of the modulating signal
applied to the input of one of the modulators.
3. The same carrier signal C (t) = Ac Cos(2πfct) is applied as one of the inputs to these two AM modulators.
4. The modulating signal m(t) is applied as another input to the upper AM modulator. Whereas, the
modulating signal with opposite polarity, −m(t) is applied as another input to the lower AM modulator.

Figure Balanced modulator

Mathematical analysis:

The outputs of the two AM modulators can be expressed as follows:

S1 (t) = Ac [1+kam (t)+ Cos 2πfct

S2 (t) = Ac [1- ka m (t)+ Cos 2πfct

Subtracting S2 (t) from S1 (t), we obtain

S (t) = S1 (t) – S2 (t)

S (t) = 2Ac ka m (t) Cos (2πfct)

Hence, except for the scaling factor 2ka the balanced modulator output is equal to product of the modulating
signal and the carrier signal. The Fourier transform of S (t) is

S (f) =ka Ac [M (f-fc) + M (f+fc)]

Assume that the message signal is band-limited to the interval –W ≤f≤ W as shown in figure 4.5 and its DSB-SC
modulated spectrum is shown in figure 4.6.

Figure Spectrum of Baseband signal

Figure Spectrum of DSBSC wave

Ring modulator:

One of the most useful product modulator, for generating a DSBSC wave, is the ring modulator shown in figure.

1. In this diagram, the four diodes D1,D2,D3 and D4 are connected in the ring structure. Hence, this
modulator is called as the ring modulator.
 2. The diodes are controlled by a square-wave carrier C (t) of frequency fc, which applied longitudinally by
means of to center-tapped transformers. If the transformers are perfectly balanced and the diodes are
identical, there is no leakage of the modulation frequency into the modulator output.
3. The message signal m(t) is applied to the input transformer. Whereas, the carrier signals C (t) is applied
between the two centre-tapped transformers.
4. For positive half cycle of the carrier signal, the diodes D1 and D3 are switched ON and the other two
diodes D2 and D4 are switched OFF. In this case, the message signal is multiplied by +1.
5. For negative half cycle of the carrier signal, the diodes D2 and D4 are switched ON and the other two
diodes D1 and D3 are switched OFF. In this case, the message signal is multiplied by -1. This results
in 1800 phase shift in the resulting DSBSC wave.

Figure Ring modulator

Mathematical Analysis:

The square wave carrier c (t) can be represented by a Fourier series as follows:
4 (−1)𝑛 −1
C(t) = ∑∞
𝑛=1 𝑐𝑜𝑠 2𝜋(2𝑛 − 1)
𝜋 2𝑛−1

= 4/π Cos(2πfct) + higher order harmonics(n=1)

Now, the Ring modulator output is the product of both message signal m (t) and carrier signal c (t).

S (t) =c (t) m (t)

4 (−1)𝑛 −1
S (t) == ∑∞
𝑛=1 𝑐𝑜𝑠 2𝜋𝑓𝑐 (2𝑛 − 1) m (t) For n=1
𝜋 2𝑛−1

S (t) =4/π Cos (2πfct) m (t)

There is no output from the modulator at the carrier frequency i.e the modulator output consists of modulation
products. The ring modulator is also called as a double-balanced modulator, because it is balanced with respect
to both the message signal and the square wave carrier signal.

The Fourier transform of S (t) is

S (f) =2/π *M (f-fc) + M (f+fc)]

Assume that the message signal is band-limited to the interval –W ≤f≤ W as shown in figure 4.8 and its DSB-SC
modulated spectrum in figure 4.9.

Figure Spectrum of Baseband signal

Figure Spectrum of DSBSC wave

Coherent Detection of DSB-SC Waves:

The base band signal can be recovered from a DSB-SC signal by multiplying DSB-SC wave S (t) with a locally
generated sinusoidal signal and then low pass filtering the product. It is assumed that local oscillator signal is
coherent or synchronized, in both frequency and phase, with the carrier signal C (t) used in the product
modulator to generate S (t). This method of demodulation is known as coherent detection or synchronous
demodulation.
 Figure Coherent detection of DSB-SC signal

Analysis of coherent detection:

The product modulator produces the product of both input signal s(t) and local oscillator signal and the output
of the product modulator is v (t).

S (t) = Ac Cos(2πfct) m(t)

C (t) = Ac Cos(2πfct + Ø )

V (t) = C(t) S (t)

V (t) =Ac Cos (2πfct+Ø) S (t)

V (t) =Ac Cos (2πfct+Ø) Ac Cos (2πfct) m (t)

V (t) =A c2 Cos (2πf ct+Ø) Cos (2πf ct ) m (t)

𝐴2𝑐 𝐴2
V (t) = cos Ø (𝑡) + 𝑐 Cos (4πfct + Ø) m (t)
2 2

In the above equation, the first term is the scaled version of the message signal. It can be extracted by passing
the above signal through a low pass filter. Therefore, the output of low pass filter is
2
Vo (t) = 𝐴𝑐 cos Ø (𝑡)
2

The Fourier transform of Vo (t) is

2
VO (f) = 𝐴𝑐 cos Ø (𝑓)
2
 Figure DSB-SC demodulated output

The demodulated signal is proportional to the message signal m (t) when the phase error is constant. The
amplitude of this demodulated signal is maximum when Ø=0, the local oscillator signal and the carrier signal
should be in phase, i.e., there should not be any phase difference between these two signals. The demodulated
signal amplitude will be zero, when Ø=±π/2. This effect is called as quadrature null effect.

Costa’s loop detection:

1. The receiver consists of two coherent detectors supplied with same DSB-SC wave while the other input for
both product modulators is taken from Voltage Controlled Oscillator (VCO) with −90 0 phase shift to one of the
product modulator as shown in figure 4.12.

2. The frequency of the local oscillator is adjusted to be the same as the carrier frequency f c. The two detector
are coupled together to form a negative feedback system designed in such a way as to maintain the local
oscillator synchronous with the carrier wave.

3. The detector in the upper path is referred to as the in-phase coherent detector or I-channel, and that in the
lower path is referred to as the quadrature-phase coherent detector or Q-channel.

4. The output of product modulator is applied as an input of the lower low pass filter.

5. The output of lower Low pass filter has −900 phase difference with the output of the upper low pass filter.
The outputs of these two low pass filters are applied as inputs of the phase discriminator. Based on the phase
difference between these two signals, the phase discriminator produces a DC control signal.

6. This signal is applied as an input of VCO to correct the phase error in VCO output. Therefore, the carrier
signal (used for DSBSC modulation) and the locally generated signal (VCO output) are in phase.
 Figure Costa’s receiver

Mathematical Analysis: We know that the equation of DSBSC wave is

S (t) = Ac Cos (2πfct)m(t)

Let the output of VCO be c1(t)= Cos(2πfct+ϕ)

This output of VCO is applied as the carrier input of the upper product modulator. Hence, the output of the
upper product modulator is

v1 (t) = S(t) c1(t)

Substitute, S(t) and c1(t) values in the above equation.

v1(t) = Ac Cos(2πfct) m(t) Cos(2πfct+ϕ)

𝐴𝑐2 𝐴𝑐2
v1(t) = Cos ϕ 𝑚(𝑡) + Cos (4πfct + ϕ) m (t)
2 2

This signal is applied as an input of the upper low pass filter. The output of this low pass filter is

v01 (t) = Ac2cos ϕ m(t)

Therefore, the output of this low pass filter is the scaled version of the modulating signal.The output
of −900 phase shifter is

c2(t) = Cos(2πfct+ϕ−900)=sin(2πfct+ϕ)

This signal is applied as the carrier input of the lower product modulator. The output of the lower product
modulator is

v2(t) = S(t) c2(t)

Substitute, S(t) and c2(t) values in the above equation.

v2(t) = Ac Cos(2πfct)m(t)sin(2πfct+ϕ)
After simplifying, we will get v2(t) as

v2(t) = Ac2sinϕm(t)+A c2sin(4πfct+ϕ)m(t)

This signal is applied as an input of the lower low pass filter. The output of this low pass filter is

v02 (t) = Ac2 sin ϕ m(t)

The output of this Low pass filter has −900 phase difference with the output of the upper low pass filter.

Single Sideband Modulation

Single sideband modulation (SSB) is a form of amplitude modulation which uses only one sideband for a given
message signal to provide the final signal. The process of suppressing one of the sidebands along with the
carrier and transmitting a single sideband is called as Single Sideband Suppressed Carrier system or
simply SSBSC.

SSB provides a considerably more efficient form of communication when compared to ordinary amplitude
modulation in terms of the radio spectrum used a can be seen from figure 4.13, and also the power used to
transmit the signal.

Depending on which half of DSB-SC signal is transmitted, there are two types of SSB modulation

1. Lower Side Band (LSB) Modulation

2. Upper Side Band (USB) Modulation

Figure SSB-SC spectrum

Mathematical Expressions

Let us consider the mathematical expressions for the modulating and the carrier signals as follows

Modulating signal m(t) =Am Cos(2πfmt)

Carrier signal c(t) = Ac Cos(2πfct)

Mathematically, we can represent the equation of SSBSC wave as

 𝐴𝑚 𝐴𝑐
S (t) = Cos *2π(f +f )t] for the upper sideband
2 c m

Or
𝐴𝑚 𝐴𝑐
S (t) = Cos *2π (f −f )t] for the lower sideband
2 c m

Bandwidth of SSBSC Wave

As can be seen in figure 4.14, 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. Therefore, the bandwidth of SSBSC modulated wave is f m and it is equal to the frequency of
the modulating signal.

Figure Spectrums of DSBSC and SSBSC

Power Calculations of SSBSC signal:

Consider the following equation of SSBSC modulated wave.

𝐴𝑚 𝐴𝑐
s(t)= cos*2π(f +f )t] for the upper sideband
2 c m
Or
𝐴 𝑚 𝐴𝑐
s(t)= cos*2π(f −f )t] for the lower sideband
2 c m

Power of SSBSC wave is equal to the power of any one sideband frequency components.

Pt = PUSB = PLSB

We know that the standard formula for power of cosine signal is

𝑉
( 𝑚/ 2 )
𝑉2
𝑟𝑚𝑠 √2
P = =
𝑅 𝑅

In this case, the power of the upper sideband is

(𝐴 )2
PUSB=
8𝑅

Similarly, we will get the lower sideband power same as that of the upper side band power.
(𝐴 )2
PLSB= 8𝑅

Therefore, the power of SSBSC wave for 1 ohm resistance is

(𝐴 )28
Pt = P USB = PLSB =

Advantages
 Bandwidth or spectrum space occupied is lesser than AM and DSBSC waves.
 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.

Generation of SSB waves:

1. Frequency Discrimination Method
The frequency discrimination or filter method of SSB generation consists of a product modulator, which
produces DSBSC signal and a band-pass filter to extract the desired side band and reject the other and is shown
in the figure 4.15. Application of this method requires that the message signal satisfies two conditions:
1. The message signal m(t) has low or no low-frequency content. M(ω) has a “hole” at zero-frequency
Example: - speech, audio, music.
2. The highest frequency component W of the message signal m(t) is much less than the carrier frequency.
Then, under these conditions, the desired side band will appear in a non-overlapping interval in the spectrum in
such a way that it may be selected by an appropriate filter.

In designing the band pass filter, the following requirements should be satisfied:
1) The pass band of the filter occupies the same frequency range as the spectrum of the desired SSB modulated
wave.
2. The width of the guard band of the filter, separating the pass band from the stop band, where the unwanted
sideband of the filter input lies, is twice the lowest frequency component of the message signal.

Figure Filter method

2. Phase discrimination method

1. The phase discriminator consists of two product modulators I and Q, supplied with carrier waves in-
phase quadrature to each other as shown in figure 4.16.
2. The incoming base band signal m(t) is applied to product modulator I, producing a DSBSC modulated
wave that contains reference phase sidebands symmetrically spaced about carrier frequency fc.
3. The Hilbert transform mˆ(t) of m(t) is applied to product modulator Q, producing a DSBSC modulated
that contains side bands having identical amplitude spectra to those of modulator I, but with phase
spectra such that vector addition or subtraction of the two modulator outputs results in cancellation
of one set of side bands and reinforcement of the other set.
4. The use of a plus sign at the summing junction yields an SSB wave with only the lower side band,
whereas the use of a minus sign yields an SSB wave with only the upper side band. This modulator
circuit is called Hartley modulator.
 Figure Phase discrimination method

Demodulation of SSB waves:

Coherent detection: It assumes perfect synchronization between the local carrier and that used in the
transmitter both in frequency and phase. The carrier signal which is used for generating SSBSC wave is used to
detect the message signal. Hence, this process of detection is called as coherent or synchronous detection.
Following is the block diagram of coherent detector.

Figure Coherent detection

In this process, the message signal can be extracted from SSBSC wave by multiplying it with a coherent carrier
and then the resulting signal is passed through a Low Pass Filter. The output of this filter is the desired message
signal.

Mathematical Analysis:

S (t) = Am Ac/2 Cos*2π(fc−fm)t]

The output of the local oscillator is

c(t)=Ac Cos(2πfct)

From the figure, we can write the output of product modulator as

v(t) = s(t)c(t)
Substitute s(t) and c(t) values in the above equation
𝐴𝑚 𝐴𝑐
V (t) = cos*2π(f +f )t] A cos(2πf t)
2 c m c c
𝐴𝑚 𝐴2
V (t) = cos(2πf t) + 𝐴 𝐴2 cos*2π(2f −f )t]
𝑐 𝑚 𝑐
4 m 4 c m

𝐴 𝑐2
In the above equation, the first term is the scaled version of the message signal the scaling factor is . It can be
4
extracted by passing the above signal through a low pass filter.

Therefore, the output of low pass filter is

𝐴𝑚 𝐴2𝑐
V0(t)= 4
cos(2πf mt)

Vestigial side band Modulation

Vestigial sideband is a type of Amplitude modulation in which one side band is completely passed along with
trace or tail or vestige of the other side band. VSB is a compromise between SSB and DSBSC modulation. In SSB,
we send only one side band, the bandwidth required to send SSB wave is w. SSB is not appropriate way of
modulation when the message signal contains significant components at extremely low frequencies. To
overcome this VSB is used. The word “vestige” means “a part” from which, the name is derived.

VSBSC Modulation is the process, where a part of the signal called as vestige is modulated along with one
sideband. The frequency spectrum of VSBSC wave is shown in the figure 4.18. Along with the upper sideband, a
part of the lower sideband is also being transmitted in this technique. Similarly, we can transmit the lower
sideband along with a part of the upper sideband.

Figure Spectrum of VSB containing vestige of USB

The vestige of the Upper sideband compensates for the amount removed from the Lower sideband. The
bandwidth required to send VSB wave is
B = w + fv
Where fv is the width of the vestigial side band.
Therefore, VSB has the virtue of conserving bandwidth almost as efficiently as SSB modulation, while retaining
the excellent low-frequency base band characteristics of DSBSC and it is standard for the transmission of TV
signals.

Generation of VSB Modulated wave:

To generate a VSB modulated wave, we pass a DSBSC modulated wave through a sideband-shaping filter. The
modulating signal m(t) is applied to a product modulator. The output of the local oscillator is also applied to the
other input of the product modulator.

Figure VSB modulator

Mathematical Analysis:

The output of the product modulator is then given by :

P (t) =Ac Cos (2πfct) m(t)

Apply Fourier transform on both sides

P (f) =Ac/2*M(f−fc)+M(f+fc)]

The above equation represents the equation of DSBSC frequency spectrum.

Let the transfer function of the sideband shaping filter be H(f). This filter has the input p(t) and the output is
VSBSC modulated wave S(t).The Fourier transforms of p(t) and S(t) are P(f) and S(f) respectively.

S(f)=P(f)H(f)

Substitute P(f) in the above equation.

S(f)=Ac/2*M(f−fc)+M(f+fc)]H(f)

The above equation represents the equation of VSBSC frequency spectrum.

Demodulation of VSBSC
Demodulation of VSBSC wave is similar to the demodulation of SSBSC wave. Here, the same carrier signal which
is used for generating VSBSC wave is used to detect the message signal. Hence, this process of detection is
called as coherent or synchronous detection. The VSBSC demodulator is shown in the figure 4.20.

In this process, the message signal can be extracted from VSBSC wave by multiplying it with a carrier, which is
having the same frequency and the phase of the carrier used in VSBSC modulation. The resulting signal is then
passed through a Low Pass Filter. The output of this filter is the desired message signal.

Figure Demodulation of VSB-SC signal

Advantages of VSB
1. The main advantage of VSB modulation is the reduction in bandwidth. It is almost as efficient as the SSB.
2. Due to allowance of transmitting a part of lower sideband, the constraint on the filter has been relaxed.
So practically, easy to design filters can be used.
3. It possesses good phase characteristics and makes the transmission of low frequency components
possible.
Application of VSB
VSB modulation has become standard for the transmission of television signal. Because the video signal need a
large transmission bandwidth if transmitted using DSB-FC or DSB-SC techniques.

Comparison of amplitude modulation techniques:

 In commercial AM radio broadcast systems standard AM is used in preference to DSBSC or SSB

modulation.
 Suppressed carrier modulation systems require the minimum transmitter power and minimum
transmission bandwidth. Suppressed carrier systems are well suited for point –to-point
communications.
 SSB is the preferred method of modulation for long-distance transmission of voice signals over metallic
circuits, because it permits longer spacing between the repeaters.
 VSB modulation requires a transmission bandwidth that is intermediate between that required for SSB
or DSBSC.
 DSBSC, SSB, and VSB are examples of linear modulation. In Commercial TV broadcasting; the VSB
occupies a width of about 1.25MHz, or about one-quarter of a full sideband.
  In standard AM systems the sidebands are transmitted in full, accompanied by the carrier. Accordingly,
demodulation is accomplished by using an envelope detector or square law detector. On the other hand
in a suppressed carrier system the receiver is more complex because additional circuitry must be
provided for purpose of carrier recovery.
 Suppressed carrier systems require less power to transmit as compared to AM systems thus making
them less expensive.
 SSB modulation requires minimum transmitter power and maximum transmission band with for
conveying a signal from one point to other thus SSB modulation is preferred.
 VSB modulation requires a transmission band width that is intermediate of SSB or DSBSC.
 In SSB and VSB modulation schemes the quadrature component is only to interfere with the in phase
component so that power can be eliminated in one of the sidebands.
Parameter of comparison AM DSB-SC SSB-SC VSB
Carrier suppression NA Fully Fully NA
Sideband suppression NA NA One sideband One sideband
completely suppressed partially
Bandwidth 2fm 2fm fm fm <BW>2fm
Transmission efficiency Minimum Moderate Maximum moderate
Power requirement More power is Power required is less Power required is less Power required is
required for than AM than Am and DSB-SC less than DSB-SC but
transmission more than SSB-SC
Power saving (%) 0 66.67 83.33 Lies between DSB
and SSB
Applications Radio Radio broadcasting Point to point mobile TV
broadcasting communication

Demodulation of AM waves:

There are two methods to demodulate AM signals. They are:

1. Square-law detector
2. Envelope detector

Square-law detector:
Square-law detector is used to detect low level modulated signals (below 1v). A Square-law detector requires
nonlinear element and a low pass filter for extracting the desired message signal. Semi-conductor diodes and
transistors are the most common nonlinear devices used for implementing square law detectors as shown in
figure 4.21. The filtering requirement is usually satisfied by using a single or double tuned filters.

Figure Square law detector

When a nonlinear element is suitably biased and operated in a restricted portion of its characteristic curve, we
find that transfer characteristic of diode-load resistor combination can be represented closely by a square law :

V0 (t) = a1Vi (t) + a2 Vi 2 (t) (4.4)

Where a1, a2 are constants

Now, the input voltage Vi (t) is the sum of both carrier and message signals

Vi (t) = Ac [1+ka m (t)+ cos2πfct (4.5)

Substitute equation (4.5) in equation (4.4) we get

V0 (t) = a1Ac [1+kam (t)+ cos2πfct + 1/2 a 2A c2 [1+2 ka m (t) + k a2m2 (t)+ *cos4πfct] (4.6)

Now design the low pass filter with cutoff frequency f is equal to the required message signal bandwidth. We
can remove the unwanted terms by passing this output voltage V0 (t) through the low pass filter and finally we
will get required message signal.
V0 (t) = Ac2 a2m (t)
The Fourier transform of output voltage VO (t) is given by
VO (f) = Ac2 a2 M (f)

Figure Spectrum of output signal

Envelope Detector:

Envelope detector is used to detect (demodulate) high level AM wave. Following figure 4.23 is the block
diagram of the envelope detector. It is also based on the switching action or switching characteristics of a
diode. It consists of a diode and a resistor-capacitor filter.

Figure Envelope detector

The operation of the envelope detector is as follows.

1. On a positive half cycle of the AM signal, the diode is forward biased and the capacitor C charges up
rapidly to the peak value of the input signal.
 2. When the AM signal level falls below this value, the diode becomes reverse biased and the capacitor C
discharges slowly through the load resistor RL till the next positive cycle of AM signal.
3. When the input signal becomes greater than the voltage across the capacitor, the diode conducts again
and the process is repeated.
4. The component values should be selected in such a way that the capacitor charges very quickly and
discharges very slowly. As a result, we will get the capacitor voltage waveform same as that of the
envelope of AM wave as shown in figure 4.24.

Figure Input-output waveform for envelope detector

The charging time constant Rs C is very small when compared to the carrier period 1/fc, the capacitor C charges
rapidly to the peak value of the signal.
Rs C << 1/fc
Where Rs = internal resistance of the voltage source, C = capacitor, fc = carrier frequency

The discharging time constant RL C is very large when compared to the charging time constant i.e., the capacitor
discharges slowly through the load resistor.
i.e., 1/fc << RLC << 1/W
Where RL = load resistance value, W = message signal bandwidth
Distortions in the Envelope Demodulator Output
There are two types of distortions which can occur in the detector output such as:
1. Diagonal clipping
2. Negative peak clipping
Diagonal Clipping: This type of distortion occurs when the RC time constant of the load circuit is too long. Due
to this, the RC circuit cannot follow the fast changes in the modulating envelope.

Negative peak clopping: This distortion occurs due to a fact that the modulation index on the output side of the
detector is higher than that on its input side. Hence, at higher depth of modulation of the transmitted signal,
the over-modulation may takes place at the output of the detector. The negative peak clipping will take place
as a result of this over-modulation as shown in figure 4.25.
 Figure Distortion in output of envelope detector

Low and high power AM transmitters:

Transmitters that transmit AM signals are known as AM transmitters. These transmitters are used in medium
wave (MW) and short wave (SW) frequency bands for AM broadcast. The MW band has frequencies between
550 KHz and 1650 KHz, and the SW band has frequencies ranging from 3 MHz to 30 MHz The two types of AM
transmitters that are used based on their transmitting powers are:
1. High Level
2. Low Level
The basic difference between the two transmitters is the power amplification of the carrier and modulating
signals. High level transmitters use high level modulation, and low level transmitters use low level modulation.
In broadcast transmitters, where the transmitting power may be of the order of kilowatts, high level
modulation is employed. In low power transmitters, where only a few watts of transmitting power are required,
low level modulation is used.
High-Level Transmitters:
In high-level transmission, the powers of the carrier and modulating signals are amplified before applying them
to the modulator stage, as shown in figure 4.26.

1. Carrier oscillator: The carrier oscillator generates the carrier signal, which lies in the RF range. The
frequency of the carrier is always very high. Because it is very difficult to generate high frequencies with good
frequency stability, the carrier oscillator generates a sub multiple with the required carrier frequency. This
sub multiple frequency is multiplied by the frequency multiplier stage to get the required carrier frequency.
2. Buffer Amplifier: The purpose of the buffer amplifier is to match the output impedance of the carrier
oscillator with the input impedance of the frequency multiplier, the next stage of the carrier oscillator. It then
isolates the carrier oscillator and frequency multiplier.
3. Frequency Multiplier: The sub-multiple frequency of the carrier signal, generated by the carrier
oscillator, is now applied to the frequency multiplier through the buffer amplifier. This stage is also known as
harmonic generator. The frequency multiplier generates higher harmonics of carrier oscillator frequency.
4. Power Amplifier: The power of the carrier signal is then amplified in the power amplifier stage. This is
the basic requirement of a high-level transmitter. A class C power amplifier gives high power current pulses
of the carrier signal at its output.
 5. Audio Chain: The audio signal to be transmitted is obtained from the microphone. The audio driver
amplifier amplifies the voltage of this signal. This amplification is necessary to drive the audio power
amplifier. Next, a class A or a class B power amplifier amplifies the power of the audio signal.
6. Modulated Class C Amplifier: This is the output stage of the transmitter. The modulating audio signal
and the carrier signal, after power amplification, are applied to this modulating stage. The modulation takes
place at this stage. The class C amplifier also amplifies the power of the AM signal to the required
transmitting power. This signal is finally passed to the antenna, which radiates the signal into space of
transmission.
Low-Level Transmitters: In low-level modulation, the powers of the two input signals of the modulator stage
are not amplified. The required transmitting power is obtained from the last stage of the transmitter, the class
C power amplifier. The low-level AM transmitter shown in the figure 4.27 is similar to a high-level transmitter,
except that the powers of the carrier and audio signals are not amplified. These two signals are directly applied
to the modulated class C power amplifier. Modulation takes place at the stage, and the power of the
modulated signal is amplified to the required transmitting power level. The transmitting antenna then transmits
the signal.

Figure High level AM transmitter

Figure Low level AM transmitter

Coupling of Output Stage and Antenna

The output stage of the modulated class C power amplifier feeds the signal to the transmitting antenna. To
transfer maximum power from the output stage to the antenna it is necessary that the impedance of the two
sections match. For this, a matching network is required. The matching between the two should be perfect at
all transmitting frequencies. As the matching is required at different frequencies, inductors and capacitors
offering different impedance at different frequencies are used in the matching networks.
The matching network must be constructed using these passive components. This is shown in figure 4.28

Figure Double Pi matching network

AM Receiver:

Radio receivers amplify and tune the radio signals. The receiver picks up the signals from the airwaves, and
converts them to the original message signal. The radio signal that is transmitted into the air contains a carrier
wave that is much higher in frequency than message siganl.

Tuned Radio Frequency Receiver:

A TRF receiver amplifies and tunes the raw radio signal as present in the air waves by means of an RF (radio
frequency) amplifier. Some receivers will have as many 4 or 5 stages of RF amplification before the carrier
signal is stripped away leaving only the audio portion of the signal. The process of removing the carrier signal is
done by the detector circuit of a radio receiver. Afterwards the final process is amplifying the audio signal to a
level strong enough to drive a speaker.

Typically a TRF receiver would consist of three main sections:

 Tuned radio frequency stages: The tuner circuit is an LC circuit, which is also called as resonant or tank
circuit. It selects the frequency, desired by the AM receiver. This consisted of one or more amplifying
and tuning stages. In a TRF receiver a series of loosely coupled tuned circuits are used to increase
selectivity.
 Signal detector: The detector enabled the audio from the amplitude modulation signal to be extracted.
It uses envelope detection.
 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.
Drawbacks :
1. Instability
2. Poor selectivity at high frequencies
3. Bandwidth variation over the tuning range
4. Insufficient adjacent frequency rejection
5. In TRF receiver, amplification is not constant over the tuning range.
 Superheterodyne AM Receiver:
In super heterodyne receiver the incoming RF signal is combined with local oscillator signal frequency through a
mixer and converted into signal of lower fixed frequency known as intermediate frequency. It consists of RF
section, frequency converter, IF amplifier, detector, audio amplifier.

RF section:
 RF section mainly consists of a tuneable filter and an amplifier which picks up the desired station by
tuning the filter to the exact frequency band.
 The signal at the antenna has lower signal noise found anywhere in the receiver.
 Then RF amplifier provides gain to increase signal to noise ratio (SNR).
Frequency converter:
 It converts the carrier frequency fc to a fixed IF frequency of 455 KHz.
 A constant frequency difference should be maintained between the local oscillator signal and incoming
RF signal frequency. (Through capacitor tuning in which the capacitance are together and operated by a
common knob.)
 For this purpose it uses local oscillator whose frequency f co is exactly 455 KHz above the incoming carrier
frequency fc and fco=fc+455
IF amplifier:
 The intermediate frequency generated from the mixer/converter is amplified by IF amplifier. After the IF
amplifier the signal is applied at the demodulator which extract the original modulated signal.
 The reason for translating all stations to a fixed carrier frequency of 455 KHz is to obtain adequate
selectivity. The characteristics of the IF amplifier are not dependent on the incoming frequency to which
the receiver is tuned. The selectivity and sensitivity of super-heterodyne receiver are quite uniform
throughout its tuning range.
 The main function of the RF section is image frequency suppression. The mixer or converter output
consists of components of difference between the incoming fc and the local oscillator fco.
 Audio amplifier: Once demodulated, the recovered audio is applied to an audio amplifier block to be
amplified by a power amplifier to the required level for loudspeakers or headphones.

Figure Superheterodyne AM receiver

Local oscillator frequency

At design level there are two choices for the local oscillator frequency:

fLO = fRF + fIF (high-side injection) or fLO= fRF − fIF (low-side injection)

Usually for medium wave AM receivers the frequency of the oscillator is higher than the desired RF frequency
(fLO = fRF + fIF).
Image frequency

When the receiver demodulates the incoming desired signal at f RF, unfortunately it demodulates down to IF
also an unwanted signal at fRF+2fIF.This frequency is called image frequency

To reduce the design complexity of the receivers the IF frequency is chosen in such a way that the signal at fimage
= fRF+2fIF can be rejected by a simple tuneable RF band pass filter such as a tank circuit with a variable
capacitor.

Figure Concept of image frequency

Terminologies of Receiver:

1. Selectivity:

Selectivity is the measure of the ability of a radio receiver to select a particular frequency or particular band of
frequencies and rejecting all other unwanted frequencies. The receiver selectivity performance determines the
level of interference that may be experienced. It is the ability to reject unwanted signals. The signal bandwidth
should be narrow for better selectivity.

The selectivity can be aimed at rejecting signals that may reach the receiver output in a variety of ways.
 Adjacent channel selectivity: Adjacent channel selectivity of the form of selectivity that rejects signals
on nearby frequencies.
 Image rejection selectivity: When using a super heterodyne radio, it is possible for the image frequency
to reach the final stages of the receiver. Rejecting these signals is important as they can cause significant
levels of interference. The selectivity required to remove these signals is contained within the radio
frequency stages of the radio.
 Image frequency rejection ratio is the ratio of gain at the signal frequency to the gain at the image
frequency.Image frequency rejection ratio (α) is given by:
α = √𝑄2𝜌2

Where 𝜌 = fIF/fRF – fRF/fIF; Q is the quality factor of the tuned circuit

2. Sensitivity:
The ability of the radio receiver to pick up the required level of radio signals will enable it to operate more
effectively within its application.
 Sensitivity of a receiver is its ability to identify and amplify weak signals at the receiver output.
 It is often defined in terms of voltage that must be applied to the input terminals of the receiver to
produce a standard output power which is measured at the output terminals.
 The higher value of receiver gain ensures smaller input signal necessary to produce the desired output
power.
 Thus a receiver with good sensitivity will detect minimum RF signal at the input and still produce
utilizable demodulated signal.
 Sensitivity is also known as receiver threshold.
 It is expressed in microvolt or decibels.
 Sensitivity of the receiver mostly depends on the gain of IF amplifier.
 It can be improved by reducing the noise level and bandwidth of the receiver.

3. Fidelity
 Fidelity of a receiver is its ability to reproduce the exact replica of the transmitted signals at the receiver
output.
 For better fidelity, the amplifier must pass high bandwidth signals to amplify the frequencies of the
outermost sidebands, while for better selectivity the signal should have narrow bandwidth. Thus a trade
off is made between selectivity and fidelity.
 Low frequency response of IF amplifier determines fidelity at the lower modulating frequencies while
high frequency response of the IF amplifier determines fidelity at the higher modulating frequencies.
UNIT 3
Syllabus Angle modulation: Introduction and types of angle modulation, frequency modulation, frequency
deviation, modulation index, deviation ratio, bandwidth requirement of FM wave, types of FM. Phase
modulation, difference between FM and PM, Direct and indirect method of FM generation, FM demodulators-
slope detector, Foster seeley discriminator, ratio detector. Introduction to pulse modulation systems, PAM,
PPM, PWM systems, frequency and time division multiplexing.
Course Objective:-
The objective of this course is to be familiar with the basic building blocks of communication systems such as
modulator and demodulator.
Course Outcomes:-
At the end of the course student will be able to :
3. Understand how information signal of low frequency can be transmitted with the help of
modulation techniques over a long distance.

3.1.Introduction:In Frequency Modulation (FM) the instantaneous value of the information signal(message)
controls the frequency of the carrier wave. This is illustrated in the following diagrams.

Notice that as the Amplitude of information signal increases, the frequency of the carrier increases, and as the
Amplitude of information signal decreases, the frequency of the carrier decreases.
The frequency fi of the information signal controls the rate at which the carrier frequency increases and
decreases. As with AM, fi must be very much less than fc. The amplitude of the carrier remains constant
throughout this process.
When the information voltage reaches its maximum value then the change in frequency of the carrier will have
also reached its maximum deviation above the nominal value. Similarly when the information reaches a
minimum the carrier will be at its lowest frequency below the nominal carrier frequency value. When the
information signal is zero, then no deviation of the carrier will occur.
The maximum change that can occur to the carrier from its base value fc is called the frequency deviation, and
is given the symbol fc. This sets the dynamic range (i.e. voltage range) of the transmission.The dynamic range
is the ratio of the largest and smallest analogue information signals that can be transmitted.

Figure 3.1 : Frequency deviation illustration

Notice that frequency modulation looks very much like phase modulation. They are in fact very similar, and
many textbooks refer to them both as angle modulation.
Concept of Frequency Modulation:
Frequency modulation: It is the form of angle modulation in which instantaneous frequency fI(t) is varied
linearly with the information signal m(t)
fI (t)=fc+ kf m(t) .......................................................................... (1)
where fc –un-modulated carrier, kf –Frequency sensitivity of the modulator, m(t)-Information signal
Integrating above equation with respect to time limit 0 to t and multiplying with 2π
2 π ʃfi(t) dt= 2 π fc ʃ dt+2 π Kf ʃm(t) dt
Ɵi (t) = 2 π fc ʃ dt+2 π Kf ʃm(t) dt
s(t)=Ac cos (Ɵi (t))
s(t)= Ac cos(2 π fc t+2 π Kf ʃm(t) dt) ....................................................... (2)
Phase modulation:
It is that form of Angle modulation in which angle ɸi(t) is varied linearly with the base band signal m(t) as as
shown by ɸi(t) = Kpm(t)

S(t)=Ac cos (ωi (t)+ ɸi(t) )

S(t)= Ac cos(2 π fc t+Kpm(t) ).................................................................. (3)

Relationship between PM and FM

PM and FM are closely related in the sense that the net effect of both is variation in total phase angle. In PM,
phase angle varies linearly with m(t) where in FM phase angle varies linearly with the integral of m(t). In other
words, we can get FM by using PM, provided that at first, the modulating signal is integrated, and then applied
to the phase modulator. The converse is also true, i.e. we can generate a PM wave using frequency modulator
provided that m(t) is first differentiated and then applied to the frequency modulator.

Figure 3.2.Relation between Fm and PM

e  sin t   
Recall that a general sinusoid is of the form: c c

Frequency modulation involves deviating a carrier frequency by some amount. If a sine wave was used to
frequency modulate a carrier, the mathematical expression would be:
i  c  sinmt
i  instantaneous frequency
c  carrier frequency
Where
  carrier deviation
m  modulation frequency

This expression shows a signal varying sinusoidally about some average frequency. However, we cannot simply
substitute expression in the general equation for a sinusoid. This is because the sine operator acts upon angles,
not frequency. Therefore, we must define the instantaneous frequency in terms of angles. It should be noted
that the amplitude of the modulation signal governs the amount of carrier deviation, while the modulation
frequency governs the rate of carrier deviation.
d
The term is an angular velocity and it is related to frequency and angle by the following relationship:
dt
d
  2f 
dt .
 To find the angle, we must integrate ω with respect to time, we obtain:
 dt  

We can now find the instantaneous angle associated with an instantaneous frequency:
    i dt   c  sin  m t dt

 c t  cos mt   t  f cos  m t
m
c
fm
This angle can now be substituted into the general carrier signal to define FM:
 f 
e fm  sinct  cos m t  .......................................(3)
 f m 
All FM transmissions are governed by a modulation index, β, which controls the dynamic range of the
information being carried in the transmission. f
  c
fi
Tone modulation:
Tone modulation is special case when message is sinusoidal as m(t) = Am cosmt
For Phase Modulation equation become sPM(t) = A cos[ct + 0 + kPMm(t)]
= A cos[ct + 0 + kPM Amcosmt]
= A cos[ct + 0 + mp cosmt]
where mp = kPM Am is the phase modulation index, representing the maximum phase deviation .

Frequency Modulation
sFM (t)  Acos[ct   0  kFM  m(t)dt]  Acos[ct   0  kFM  Am cosmtdt]
kFM Am
 Acos[ct 0  sinmt]  Acos[ t   m sin t]
c 0 f m
m
where β = mf = kFMAm / m =  / m, i.e. the ratio of frequency deviation to the modulating frequency, is called
the frequency modulation index.
β =  / m ................................................................... (4)
The relationship between phase deviation and frequency deviation in FM is given by
 = β =  / m........................................................... (5)

Types of frequency modulation

The bandwidth of an FM signal depends on the frequency deviation. When the deviation is high, the bandwidth
will be large, and vice-versa. According to the equation  = kFMm(t)max, for a given m(t), the frequency
deviation, and hence the bandwidth, will depend on frequency sensitivity kFM. Thus, depending on the value of
kFM (or ) we can divide FM into two categories: narrowband FM and wideband FM.

Narrowband FM(NBFM (β<<1))

When kFM is small, the bandwidth of FM is narrow, this type of FM is called narrowband FM.
Since when x << 1, cosx  1, sinx  x, we have
sNBFM (t)  A cos[ct   0  kFM  m(t)dt]

 Acos(ct   0 ) cos[kFM  m(t)dt]  Asin(ct   0 ) sin[kFM  m(t)dt]

 Acos[ ct   0 ]  AkFM  m(t)dt sin[ ct   0 ]

Narrowband modulation methods:

 Figure 3.3.NBFM generation
Equation of narrowband frequency modulation (Tone modulation )
The message signal m(t) = Am cosmt
Signal waveform (assume 0 = 0 for simplicity)
sNBFM (t)  A cosct  AkFM  m(t)dt sin ct  A cosct  AkFM  Am cosmtdt sin ct
Am
 A cos t  Ak sin  t sin  t  A cos t  Am sin  t sin  t


c FM
m m c c f m c

1 1
 A cos t  Am cos(   )t  Am cos(   )t
c f c m f c m
2 2

where β = kFMAm / m is the FM modulation index.

Signal spectrum SNBFM() = A[( - c) + ( + c)] +
(1/2)Amf[( - c - m) + ( + c + m) – ( - c + m) - ( + c - m)]

Narrowband FM demodulation method

Figure 3.4.NBFM demodulation

Wideband FM (WBFM (β>>1))

When kFM is large, the bandwidth of FM is wide, this type of FM is called wideband FM.
It is usually very difficult to analyze a general FM signal, we will restrict our analysis to the wideband FM with
sinusoidal signal.
The message signal m(t) = Am cosmt
Signal waveform (assume 0 = 0 for simplicity)
sFM (t)  Acos[ct  AkFM  m(t)dt]  Acos[ct  AkFM  Am cosmtdt]
 Acos[ct  mf sinmt]  Acosct cos(mf sinmt)  Asinct sin(mf sinmt)
cos(mf sinmt) and sin(mf sinmt) can be expressed in Fourier series


cos(mf sin mt)  J0 (mf )  2 J2n (mf ) cos 2nmt

n1


sin(mf sin mt)  2 J2n1 (mf ) cos(2n 1)mt

n1
 mf

(1)m ( )2mn
where Jn (mf )   2
m0 m!(m  n)!
is the Bessel function of the first kind. Thus,
 
sFM (t)  Acos  ct[J 0 (m f )  2 J 2n (m f ) cos 2n mt]  Asin  ct[2 J 2n1 (m f ) cos(2n 1) mt]
n1 n1

by using coscos = (1/2)[cos( + ) + cos( - )],

sinsin = (1/2)[cos( - ) - cos( + )]
and the property of Bessel function J n (m f )  (1)n J n ( FM )
sFM(t) can be written in the Bessel function form


sFM (t)  A  Jn (mf ) cos[(c  nm )t]

n


The spectrum of sFM(t) SFM ()  A  J (m )[ ( c  nm )   ( c  nm)]
n
n
f
Jn(mf )

J0(mf )

J1(mf )

J2(mf )

J3(mf )

mf

mf = 0.2

f
fc - f m fc fc + fm

mf = 1

f
fc - 2f m fc fc + 2fm
mf = 5

f
fc - 6f m fc fc + 6fm
Figure 3.5.Typical plots of SFM() for different β.
The following observation can be made
 The carrier term cosct has a magnitude of J0(mf). The maximum value of J0(mf) is 1 when mf = 0, which is
equivalent to no modulation.
 Theoretically infinitely number of sidebands are produced, and the amplitude of each sideband is decided
by the corresponding Bessel function Jn(mf). The presence of infinite number of sidebands makes the ideal
bandwidth of the FM signal infinite.
 When mf is small, there are few sideband frequencies of large amplitude and, when mf is large, there are
many sideband frequencies but with smaller amplitudes. Hence, in practice, to determine the bandwidth, it is
only necessary to consider a finite number of significant sideband components.
 Thus, the sidebands with small amplitudes can be ignored. The sidebands having amplitudes more than or
equal to 1% of the carrier amplitude are known as significant sidebands. They are finite in number.

Bandwidth of a sinusoidally modulated FM signal

mf = 0.2

f
fc - fm fc fc + fm

mf = 1

f
fc - 2fm fc fc + 2fm
mf = 5

f
fc - 6fm fc fc + 6fm
Figure 3.6.Spectrum of FM with different values of mf

How many sidebands are significant in the FM?

Jn(mf) diminishes rapidly for n > mf, particularly as mf becomes large, the number of significant sideband is mf +
1, i.e. WFM = 2(mf + 1)m = 2( + m)
or
BFM = 2(mf + 1)fm = 2(f + fm)

Expressed in words, the bandwidth is twice the sum of the maximum frequency deviation and the modulating
frequency. This rule for bandwidth is called Carson’s rule.

Bandwidth of FM signal with arbitrary modulating signals

WFM = 2(DFM + 1)m = 2( + m)
or BFM = 2(DFM + 1)fm = 2(f + fm)
Where DFM is the frequency deviation ratio defined by
 f
DFM   
m fm
Power of the FM signal
Since the amplitude of FM remains unchanged, the power of FM signal is the same as that of unmodulated
carrier.
2 2

2 A2  2 A2
PFM  s FM (t)  A
 [Jn (mf ) cos(c  nm )t]  2 n
n
 Jn (mf )  2

where we used one of the properties of Bessel function J
2
n (m f )  1
n

The total power is independent of the FM modulation process, since the power is related to signal amplitude,
which is constant for FM, and not necessarily dependent on the signal’s phase.
Comparison between WBFM and NBFM.
 S. No WBFM NBFM

I. Modulating index is greater than1 Modulation index is less than 1

II. Frequency deviation =75 KHz. Frequency deviation 5 Khz.

III. Modulating frequency range Modulation frequency =3 Khz

iV. From 30 Hz-15 Khz. Bandwidth =2 FM

Bandwidth 15 times NBFM.
V. Less suppressing of noise
Noise is more suppressed.
Vi.
Use: Entertainment and Use: Mobile communication.
broadcasting

Wideband modulation methods

There are two methods for generating wideband FM signals: direct and indirect methods.

Direct method, voltage-controlled oscillator (VCO)

The direct method depends on varying the frequency of an oscillator linearly with m(t) for FM.

Figure 3.7.Direct method of FM generation

In the VCO, the modulating signal varies the voltage across the capacitor, as a consequence, the capacitance
changes and causes a corresponding change in the oscillator frequency, i.e
C = C0 + C = C0 + k0m(t)
where k0 is a constant.


1
Assume that 0 
LC0 
1 C k0m(t) 0k0
then   1 
1 1   (1 )   (1    m(t)    km(t)
 0 0 ) 0 0
C C 2C 2C 2C
 

LC LC0
LC0 (1 ) 1 0 0 0
C0 C0
where k = 0k0/2C0 is a constant, and the result: (1 + x)-1/2 = 1 – x/2, when x is small, is used.

Indirect or multiplication method

The indirect method depends on first generating a narrow FM signal and then using a multiplication technique
whereby the deviation ratio can be raised to a large value.
 Figure 3.8.WBFM signal using NBFM
The multiplier is a device that multiplies the instantaneous frequency of its input waveform by a factor N.

Wideband FM demodulation method

Apply sFM(t) to a differentiator, the output is
dsFM (t) d{Acos[ct  kFM  m(t)dt]} 
   A[ c  kFM m(t)]sin[ct  kFM  m(t)dt]
dt dt
which is similar to a standard AM signal with small deviation ratio. The deviation ratio is small, since usually 
= kFMm(t)max << c. The response of an envelope detector becomes
A[c + kFMm(t)]
Blocking the dc term Ac, the output is so(t) = AkFMm(t)

Figure 3.9.WBFM demodulator

The FM detector extracts a modulating signal from a frequency modulated carrier in two steps:
1. It converts the frequency modulated (FM) signal into a corresponding amplitude modulated (AM) signal by
using frequency dependent circuits whose output voltage depends on input frequency. Such circuits are called
frequency discriminators.
2. The original modulating signal m(t) is recovered from this AM signal by using a linear diode envelope
detector.

Comparison between AM and FM

1. Noise performance: Wideband FM has better noise performance than AM. The greater the bandwidth,
better is the noise performance. Narrowband FM has a noise performance equivalent to AM.

2. Channel bandwidth
The wideband FM has a larger bandwidth as compared to AM because wideband FM produces a larger
number of sidebands. In a typical broadcast system, each channel bandwidth in AM is 15kHz, whereas, in
FM, it is 150kHz. Therefore, FM has a disadvantage over AM.
The modulation index β, is the ratio of the frequency
deviation, fc , to the maximum information frequency,
fi , as shown below:

The diagrams opposite show examples of how the

modulation index affects the FM output, for a simple
sinusoidal information signal of fixed frequency. The
carrier signal has a frequency of ten times that of the
information signal.

The first graph shows the information signal, the second

shows the unmodulated carrier.

This graph shows the frequency modulated carrier when

the modulation index = 3.

This graph shows the frequency modulated carrier when

the modulation index = 5.

This graph shows the frequency modulated carrier when

the modulation index = 7.

Figure 3.10.FM with different Modulation indices

As the modulation index increases you should notice that the peaks of the high frequency get closer together
and low frequency get further apart. For the same information signal therefore, the carrier signal has a higher
maximum frequency.
The FM modulation index is defined as the ratio of carrier deviation to modulation frequency:
As a result, the FM equation is generally written as:
e fm  sinc t  m fm cos  mt
This is a very complex expression and it is not readily apparent what the sidebands of this signal are like. The
solution to this problem requires knowledge of Bessel’s functions of the first kind and order p. In open form, it
resembles:
 2k  p


 1k  x 
 2 
J p x  
k 0 k!k  p!
J p x  magnitude of frequency component
where p  side frequency number
x  modulation index
As a point of interest, Bessel’s functions are a solution to the following equation:
d2y dy
x  2  x  x  p 2  0
2 2
 
dx dx
Bessel’s functions occur in the theory of cylindrical and spherical waves, much like sine waves occur in the
theory of plane waves. It turns out that FM generates an infinite number of side frequencies. Each frequency is
an integer multiple of the modulation signal. It should be noted that the amplitude of the higher order sided
frequencies drops off quickly. It is also interesting to note that the amplitude of the carrier signal is also a
function of the modulation index. Under some conditions, the amplitude of the carrier frequency can actually
go to zero. This does not mean that the signal disappears, but rather that all of the broadcast energy is
redistributed to the side frequencies.

Generation of frequency Modulation

Direct FM
(a) Reactance Modulator
The reactance modulator is a voltage controlled capacitor and is used to vary an oscillator’s frequency or phase.
A simplified circuit resembles:

C i i
e
R e

Figure 3.11.Reactance modulator

Since the gate does not draw an appreciable amount of current, applying Ohm’s law in the RC branch results in:
eg  iC
e
iC 
R  jX C
e
 eg  R
R  jX C
The JFET drain current is given by:
e
id  g m eg  gm R
R  jX C
where gm is the trans-conductance.
The impedance as seen from the drain to ground is given by:
e  e 1 R  jX C 1 1  j XC
Z 
id gm e R gm gm R
Since trans-conductance is normally very large, the impedance reduces to:
XC  j
Z j 
gm R 2f Cg m R
 The term in the denominator can be thought of as an equivalent capacitance:
Ceq  Cg m R
Then
j
Z
2f Ceq
Since the equivalent capacitance is larger than the original capacitor, we have created a capacitance amplifier.
Because the value of this capacitance is a function of applied voltage, we actually have a voltage controlled
capacitor. This device can be used to control an oscillator frequency, thus producing FM.

Figure 3.12.FM transmitter

Varactor Diodes

The capacitance of a varactor diode is a function of its’ reverse bias voltage.

C0
Cd 
 1  2VR
Where C0 is the diode capacitance at zero bias, and VR is the reverse bias voltage. A typical response is:

(b) Indirect Method:

Because crystal oscillators are so stable, it is desirable to use them in modulator circuits. However, their
extreme stability makes it difficult to modulate their frequency.
Fortunately, it is possible to vary the phase of a crystal oscillator. However, in order to use this as an FM source,
the relationship between frequency and phase needs to be reexamined.
Frequency is the rate of change of angle, its first derivative:
d
 
dt
The instantaneous phase angle is comprised of two components, the number of times the signal has gone
through its cycle, and its starting point or offset:
 t    
⏟c t  ⏟
rotating offset
angle angle

The instantaneous frequency is therefore:

i  t   c t    c
d d d
 
dt dt dt
From this we observe that the instantaneous frequency of a signal is its un-modulated frequency plus a change.
This is equivalent to frequency modulation. Therefore we may write:
d
c    c   eq
dt
d
 eq  
dt
f  1 d  

eq
2 dt
This means that the output of a phase modulator is proportional to the equivalent frequency modulation.
If the angle is proportional to the amplitude of a modulation signal   k e , Then:
1 d ke
f eq  
2 dt m
and by integrating the modulation signal prior to modulation, we obtain:
1 d ke dt  k e
f eq    m
2 dt 2
m

This means that the equivalent frequency modulation is directly proportional to the amplitude of a phase
modulation signal if the modulation signal is integrated first.
This indirect modulation scheme is the heart of the Armstrong modulator.

Demodulation / Detector of Frequency Modulation:

Phase Detector [Foster-Seeley]
The Foster-Seeley detector converts the incoming frequency variation to an equivalent phase variation and
then to an equivalent amplitude variation. This is accomplished by using the phase angle shift which occurs
between the primary and secondary of a transformer tuned circuit.
It is important to recognize that the signal on the primary side gets to the secondary through two distinctly
different paths:
• through the transformer via the primary winding
• bypassing the primary winding and directly into the secondary center tap

Figure 3.13.Foster seeley detector

The voltage appearing on the secondary side of the transformer is given by:
Lp
es  e p k
Ls
e p  primary voltage
k  coupling coefficien t
Lp  primary inductance
Ls  secondary inductance
This is applied to the series resonant circuit in the transformer secondary winding. The impedance of this
resonant circuit is given by:    L
Z  R  j L  1   j  1 

 R1 
 

 
 


C    R RC  
   o L  o 1 
 R1 j 
 R   RC 
  o o 
Where ωo is the resonant frequency.
 o L 1
Since Q   , the impedance can be written as:
R o RC
    o  
Z  R1 j   Q
  
  o  
 o
Defining a new parameter: Y   , we obtain:
o 
Z  R1 jYQ
The impedance phase angle is given by:   tan YQ.1

It is interesting to observe what happens to this angle when the input frequency varies.
Let the input frequency be of the form:   o   , then:
    2    2
Y o  o
 o

o o    o 2   o 
But if the deviation is much smaller than the carrier:   o , then:
2
Y
o
Notice that the parameter Y varies directly with deviation. For small angle changes   tan  . This means that
the impedance phase angle varies directly with the frequency deviation. This in turn causes a variation in the
currents and voltages in the secondary.
The output of the transformer consists of the vector sum of two components:
• The phase shifted signal passing through the transformer
• The un-shifted signal which has bypassed the transformer
The combination of these two signals results in amplitude variations which are directly proportional to the
frequency deviation. This AM signal is then detected through a standard envelope detector.
This circuit is quite sensitive however, any amplitude variations in the signal caused by varying signal strength
are also detected.

Ratio Detector
This circuit is a slight modification of the Foster-Seeley detector:
• one diode is reversed
• the output is taken from the combined loads

Figure 3.14.Ratio detector

The limiting action of the detector has been improved and variations in signal strength are not as noticeable.
This also implies that it is less sensitive.
Phase Locked Loop
Although phase locked loops can be implemented using analog or digital circuitry, the following discussion will
be limited to linear circuits since they are much easier to analyze.
 sin  t
X

cos t

Figure 3.15.Phase lock loop

The loop achieves lock in two steps. First it acquires frequency lock, and then it acquires phase lock.
If the input signal and VCO output are completely different, the output of the multiplier is given by:
1 1
Vmult  sinit coso t   sini  o t  sini  o t
2 2
The low pass filter removes the high frequency component before passing the signal to the output amplifier.
The output, which also is the input to the VCO, is of the form:
Voutput  sin i   o t
Notice that if i  o , the output is positive, but if i  o , the output is negative.
This change in polarity can be seen by observing the sine function near the origin:
sin ( )



Figure 3.16.sinusoidal variation

This voltage can be used as an error signal to drive the VCO until i  o , and frequency lock is achieved. The
two signals however, are not necessarily in phase at this point.
Once frequency lock occurs, the multiplier output becomes:
1 1
V  sin t cos t     sin2   t  sin t
mult i i i
2 2
Again the low pass filter removes the high frequency component and the output is of the form:
Voutput  sin t
Again if   0 the output is positive, but if   0 the output is negative. This error signal is used to drive the VCO
until   0 , and phase lock is achieved. Notice that at lock, the incoming signal and the VCO output are in
quadrature.
A PLL can be used after the IF amplifier in a radio to reproduce the original modulation signal. This allows the
free running VCO oscillator frequency to be preset, thus making it easier to acquire lock.
FM Spectrum:
When the amplitude of the frequency components of FM waveform are plotted as a function of frequency, the
resulting spectrum is much more complicated than that of the AM waveform This is because there are now
multiple frequencies present in the FM signal.
Theoretically, an FM spectrum has an infinite number of
sidebands, spaced at multiples of fi above and below the
carrier frequency fc. However the size and significance of
these sidebands is very dependent on the modulation index,
β.If β<1, then the spectrum looks like this:
Figure 5.18.FM spectrum for β<1
 From the spectrum above it can be seen that there are only two significant sidebands, and thus the spectrum
looks very similar to that for an AM carrier.
Ifβ=1, then the spectrum looks like this:

Figure 3.17.FM spectrum for β =1

From the spectrum above we can see that the number of significant sidebands has increased to four.

Ifβ=3, then the spectrum looks like this:

Figure 3.18.FM spectrum for β=3

From the spectrum above we can see that the number of significant sidebands has increased to eight. It can be
deduced that the number of significant sidebands in an FM transmission is given by2(β+1). The implication for
the bandwidth of an FM signal should now be coming clear. The practical bandwidth is going to be given by the
number of significant sidebands multiplied by the width of each sideband (i.e. fi).

Bandwidth FM  2  1 fi
 fc 
 2  1 f i
 fi 
 2f c  f i 

The bandwidth of an FM waveform is therefore twice the sum of the frequency deviation and the maximum
frequency in the information.
Carson rule:
It states that the bandwidth required transmitting an angle modulated wave as twice the sum of the peak
frequency deviation and the highest modulating signal frequency.
Band Width = 2 * ∆f + fm(max) ] Hz
∆f = frequency deviation in Hz
fm(max) = highest modulating signal frequency in Hz
Additional Points to remember.
 An FM transmission is a constant power wave, regardless of the information signal or modulation index,
β because it is operated at constant amplitude with symmetrical changes in frequency.
 As β increases, the relative amplitude of the carrier component decreases and may become much
smaller than the amplitudes of the individual sidebands. The effect of this is that a much greater proportion of
the transmitted power is in the sidebands (rather than in the carrier), which is more efficient than AM.

Determination of Bandwidth for FM Radio

FM radio uses a modulation index, β> 1, and this is called wideband FM. As its name suggests the bandwidth is
much larger than AM.

In national radio broadcasts using FM, the frequency deviation of the carrier Δf c, is chosen to be 75 kHz, and the
information baseband is the high fidelity range 20 Hz to 15 kHz.
Thus the modulation index, β is 5 (i.e. 75 kHz  15 kHz), and such a broadcast requires an FM signal bandwidth
given by:
BandwidthFM Radio  2(fc  fi(max) )

 2(75 15)
 180kHz
Advantages of FM over AM
a) The amplitude of FM is constant. Hence transmitter power remains constant in FM where as it varies in
AM.
b) Since amplitude of FM is constant, the noise interference is minimum in FM. Any noise superimposing
on modulated carrier can be removed with the help of amplitude limiter.
c) The depth of modulation has limitation in AM. But in FM, the depth of modulation can be increased to
any value.
d) Since guard bands are provided in FM, there is less possibility of adjacent channel interference.
e) Since space waves are used for FM, the radius of propagation is limited to line of sight( LOS ) . Hence it is
possible to operate several independent transmitters on same frequency with minimum interference.
f) Since FM uses UHF and VHF ranges, the noise interference is minimum compared to AM which uses MF
and HF ranges.
Introduction to pulse modulation systems:

Pulse modulation is “the process in which signal is transmitted by pulses (i.e., discontinuous signals) with a
special technique”. The pulse modulation is classified as analog pulse modulation and digital pulse modulation.
The analog pulse modulation is again classified as,

1. Pulse amplitude modulation

2. Pulse width modulation and
3. Pulse position modulation

Pulse Amplitude Modulation

In Pulse Amplitude Modulation (PAM) technique, the amplitude of the pulse carrier varies, which is
proportional to the instantaneous amplitude of the message signal. The width and positions of the pulses are
constants in this modulation. There are two kinds of Pulse amplitude modulation. They are natural sampling
and flat top sampling.
 UNIT 4

Sampling of signal, sampling theorem for low pass and Band pass signal, Pulse amplitude modulation (PAM), Time
division, multiplexing (TDM). Channel Bandwidth for PAM-TDM signal Type of sampling instantaneous, Natural and flat
top, Aperture effect, Introduction to pulse position and pulse duration modulations, Digital signal, Quantization,
Quantization error, Pulse code modulation, signal to noise ratio, Companding, Data rate and Baud rate, Bit rate,
multiplexed PCM signal, Differential PCM (DPCM), Delta Modulation (DM) and Adaptive Delta Modulation (ADM),
comparison of various systems.

Course Objective:-
The objective of this course is to be familiar with the basic building blocks of communication systems such as
modulator and demodulator.
Course Outcomes:-
At the end of the course student will be able to :
4. Differentiate different modulation techniques such as AM, SSB, DSB and FM.

Sampling
The process of converting continuous time signals into equivalent discrete time signals, can be termed
as Sampling.
Sampling Techniques
Their are basically three types of Sampling techniques, namely:
1. Natural Sampling
2. Flat top Sampling
3. Ideal Sampling
1. Natural Sampling:
Natural Sampling is a practical method of sampling in which pulse have finite width equal to τ. Sampling is done
in accordance with the carrier signal which is digital in nature.

Figure. Natural Sampled Waveform

Figure. Functional Diagram of Natural Sampler

 With the help of functional diagram of a Natural sampler, a sampled signal g(t) is obtained by multiplication
of sampling function c(t) and the input signal x(t).
Spectrum of Natural Sampled Signal is given by:
G(f) = Aτ/ Ts .* Σ sin c(n fs.τ) X(f-n fs)]

2. Flat Top Sampling:

Flat top sampling is like natural sampling i.e; practical in nature. In comparison to natural sampling flat top
sampling can be easily obtained. In this sampling techniques, the top of the samples remains constant and is
equal to the instantaneous value of the message signal x(t) at the start of sampling process. Sample and hold
circuit are used in this type of sampling.

Figure. Flat top sampling

Above figure shows the general waveform of the flat top samples. It can be observed that only starting edge of
the pulse represent the instantaneous value of the message signal x(t).
Spectrum of Flat top Sampled Signal is given by:
G(f) = fs .* Σ X(f-n fs). H(f)]
Nyquist Rate
It is the minimum sampling rate at which signal can be converted into samples and can be recovered back
without distortion.
Nyquist rate fN = 2fm hz
Nyquist interval = 1/fN = 1/2fm second
Sampling theorem:

Statement: A continuous time signal can be represented in its samples and can be recovered back when
sampling frequency fs is greater than or equal to the twice the highest frequency component of message signal.

i.e. fs≥2fm.
 Figure. Illustration of ideal sampling
Here, you can observe that the sampled signal takes the period of impulse. The process of sampling can be
explained by the following mathematical expression:
Sampled signal y(t)=x(t).δ(t) ..... (1)
Possibility of sampled frequency spectrum with different conditions is given by the following diagrams:

Figure .Sampling with different sampling rates

Aliasing Effect
The overlapped region in case of under sampling represents aliasing effect, which can be removed by
 considering fs >2fm
 By using anti aliasing filters.

The pulse amplitude modulated signal will follow the amplitude of the original signal, as the signal traces out
the path of the whole wave. In natural PAM, a signal sampled at Nyquist rate can be reconstructed, by passing
it through an efficient Low Pass Filter (LPF) with exact cut-off frequency.
Generation of PAM
 Pulse amplitude modulation is the basic form of pulse modulation in which the signal is sampled at
regular and each sample is made proportional to the amplitude of the modulating signal at the sampling
instant.
 The Fig1 shows the generation of PAM signal from the sampler which has two inputs i.e. modulating
signal and sampling signal or carrier pulse.
 Thus the amplitude of the signal is proportional to the modulating signal through which information is
carried. This is Pulse amplitude modulation signal.
 Fig 5.26 shows the spectrum of pulse amplitude modulated signal along with the message signal and the
sampling signal which is the carrier train of pulses with the help of the waveform plotted in time
domain.
 Pulse Modulation may be used to transmitting analog information, such as continuous speech signal or
data.
 Figure .Spectrum of PAM signal
Demodulation of PAM
 For Demodulation of the Pulse Amplitude Modulated signal, PAM is fed to the low pass filter as shown
in Fig below.

Fig . PAM detector

 The low pass filter eliminates high frequency ripples and generates the demodulated signal which has its
amplitude proportional to PAM signal at all time instant.
 This signal is then applied to an inverting amplifier to amplify its signal level to have the demodulated
output with almost equal amplitude with the modulating signal.

Pulse Width Modulation

Pulse Width Modulation (PWM) or Pulse Duration Modulation (PDM) or Pulse Time Modulation (PTM) is an
analog modulating scheme in which the duration or width or time of the pulse carrier varies proportional to
the instantaneous amplitude of the message signal.
The width of the pulse varies in this method, but the amplitude of the signal remains constant. Amplitude
limiters are used to make the amplitude of the signal constant. These circuits clip off the amplitude, to a
desired level and hence the noise is limited.

There are three types of PWM. They are

 The leading edge of the pulse being constant, the trailing edge varies according to the message signal.
 The trailing edge of the pulse being constant, the leading edge varies according to the message signal.
 The center of the pulse being constant, the leading edge and the trailing edge varies according to the
message signal.

Generation of PWM signal:

 Figure .PWM generation by comparator

 As shown in the figure, one input of the comparator is fed by the input message or modulating signal
and the other input by a saw tooth signal which operates at carrier frequency.
 Considering both ±ve sides, the maximum of the input signal should be less than that of saw tooth
signal.
 The comparator will compare the two signals together to generate the PWM signal at its output as
shown in the third waveform of Fig.
 The rising edges of the PWM signal coincides with the falling edge of the saw tooth signal.
 When the saw tooth signal is at the minimum value which is less than the minimum of the input signal,
then the positive input of the comparator is at higher potential which gives the comparator output as
positive.
 When the saw tooth signal rises and is at the maximum value, the negative input of the comparator is at
higher potential, which will produce the comparator output to be negative.
 Thus the input signal magnitude determines the comparator output and its potential, which then
decides the width of the pulse generated at the output.
 In other words we can say that the width of the pulse generated signal is directly proportional to the
amplitude of the modulating signal.

Figure. PWM waveform

PWM demodulation:
 For PWM demodulation, put a ramp at the +ve edge which will stop at the arrival of –ve egde.
 The ramp will attain different heights in each cycle since the widths are different and the heights
attained are directly proportional to the pulse width and in turn the amplitude of the message signal.
 The waveform is the sum of a sequence of constant-amplitude and constant-width pulse generated by
demodulator. This signal is then applied to the input of clipping circuit, which cuts off the portion of
signal below the threshold voltage and outputs the reminder. Therefore the output of clipping circuit is
a PAM signal whose amplitude is proportional to the width of PWM signal.
 This is then passed through a low pass filter where it will follow the envelope i.e. the message signal,
which produces the demodulated signal at the output.

Figure. PWM demodulation

Pulse Position Modulation

Pulse Position Modulation (PPM) is an analog modulating scheme in which the amplitude and width of
the pulses are kept constant, while the position of each pulse, with reference to the position of a
reference pulse varies according to the instantaneous sampled value of the message signal.

The transmitter has to send synchronizing pulses (or simply sync pulses) to keep the transmitter and
receiver in synchronism. These sync pulses help maintain the position of the pulses. The following
figures explain the Pulse Position Modulation.

Generation of PPM signal:

PPM signal can be generated with the help of PWM as shown in Fig7 below.

Figure. PPM generation from PWM

 The PWM signal generated above is sent to an inverter which reverses the polarity of the pulses.
 This is then followed by a differentiator which generates +ve spikes for PWM signal going from High to
Low and -ve spikes for Low to High transition. The spikes generated are shown in the fourth waveform
of Fig .
 These spikes are then fed to the positive edge triggered pulse generator which generates fixed width
pulses when a +ve spike appears, coinciding with the falling edge of the PWM signal.
  Thus PPM signal is generated at the output which is shown in the fifth waveform of Figure 5.31 where
pulse position carry the message information.

Figure. PPM Waveform

PPM detection:
 For PPM demodulation, ramp is used which starts at the +ve edge of the one pulse and stops at the +ve
edge of the next pulse.
 Thus the height of the generated ramp is determined by the delay between the pulses which indirectly
follows the amplitude of the modulating signal.
 This is then passed through a low pass filter which filters the envelop information as the demodulated
signal.

Figure.PPM demodulator circuit

Difference Between PAM, PWM, and PPM
Sr. No. Parameter PAM PWM PPM
1 Type of Carrier Train of Pulses Train of Pulses Train of Pulses

Variable Characteristic of the

2 Pulsed Carrier Amplitude Width Position

3 Bandwidth Requirement Low High High

4 Noise Immunity Low High High

5 Information Contained in Amplitude Variations Width Variations Position Variations

6 Power efficiency (SNR) Low Moderate High
7 Transmitted Power Varies with amplitude of Varies with variation in Remains Constant
Need to transmit
8 Not needed Not needed Necessary
synchronizing pulses
Bandwidth depends on the Bandwidth depends on the Bandwidth depends on the
9 Bandwidth depends on
width of the pulse rise time of the pulse rise time of the pulse
Instantaneous transmitter Instantaneous transmitter Instantaneous transmitter
power varies with the power varies with the power remains constant
10 Transmitter power amplitude of the pulses amplitude and width of the with the width of the pulses
pulses
Complexity of generation and
11 Complex Easy Complex
detection
Similarity with other
12 Similar to AM Similar to FM Similar to PM
Modulation Systems

Frequency-Division Multiplexing

Frequency-division multiplexing (FDM) is an analog technique that can be applied when the bandwidth of a link
(in hertz) is greater than the combined bandwidths of the signals to be transmitted.
In FDM, signals generated by each sending device modulate different carrier frequencies. These modulated
signals are then combined into a single composite signal that can be transported by the link. Carrier frequencies
are separated by sufficient bandwidth to accommodate the modulated signal. These bandwidth ranges are the
channels through which the various signals travel.
 Figure .Frequency division multiplexing

Time Division Multiplexing (TDM)

TDM is a technique used for transmitting several message signals over a single communication channel by
dividing the time frame into slots, one slot for each message signal.
The concept of TDM is indicated in the figures. Each message signal is first restricted in bandwidth be a low pass
pre-alias filter to remove the frequencies that are not essential which helps in reducing the aliasing problem.
The outputs of these filters are then applied to a commutator.
The functions of the commutator are:

(i)Allows narrow samples of each of the N input messages at a rate of fs

(ii)Sequentially interleaves these N samples inside a sampling interval Ts.
The multiplexed signal is then applied to a pulse amplitude modulator, which transforms the multiplexed signal
into a form suitable for transmission over the communication channel. The time division scheme squeezes N
samples derived from different N independent message signals into a time slot equal to one sampling interval.
Thus the use of TDM introduces a bandwidth expansion factor N.
Figure. TDM PAM transmitter
 Unit-5
Digital modulations techniques, Generation, detection, equation and Bandwidth of amplitude shift
keying (ASK) Binary Phase Shift keying (BPSK), Differential phase shift keying (DPSK), offset and non
offset quadrature phase shift keying (QPSK), M-Ary PSK, Binary frequency Shift Keying (BFSK), M-Ary
FSK Quadrature Amplitude modulation (QAM).
Course Objective:-
The objective of this course is to study the different types of analog modulation techniques are given
in this course.
Course Outcomes:-
At the end of the course student will be able to :
5. Explain using block diagrams, modulation and demodulation techniques for digital signal
and determine bandwidth requirement.
Digital Modulation
Digital Modulation provides more information capacity, high data security, quicker system availability with
great quality communication. Hence, digital modulation techniques have a greater demand, for their
capacity to convey larger amounts of data than analog modulation techniques.
There are many types of digital modulation techniques and also their combinations, as listed below.


The amplitude of the resultant output depends upon the input data whether it should be a zero level or a
variation of positive and negative, depending upon the carrier frequency.

The frequency of the output signal will be either high or low, depending upon the input data applied.


The phase of the output signal gets shifted depending upon the input. These are mainly of two types,
namely Binary Phase Shift Keying (BPSK) and Quadrature Phase Shift Keying (QPSK), according to the
number of phase shifts. The other one is Differential Phase Shift Keying (DPSK) which changes the phase
according to the previous value.


M-ary Encoding techniques are the methods where more than two bits are made to transmit
simultaneously on a single signal. This helps in the reduction of bandwidth.
The types of M-ary techniques are M-ary ASK, M-ary FSK & M-ary PSK.

Amplitude Shift Keying (ASK) is a type of Amplitude Modulation which represents the binary data in
the form of variations in the amplitude of a signal.
Any modulated signal has a high frequency carrier. The binary signal when ASK modulated, gives a zero
value for Low input while it gives the carrier output for High input.
The figure 3.1.1 represents ASK modulated waveform along with its input.

(a) ASK Modulation

(b) ASK Modulated Wave

Figure 5.1. ASK Modulation
To find the process of obtaining this ASK modulated wave, let us learn about the working of the ASK
modulator.


The ASK modulator block diagram comprises of the carrier signal generator, the binary sequence from the
message signal and the band-limited filter. Following is the block diagram of the ASK Modulator.

Figure 5.2. ASK Modulator

The carrier generator sends a continuous high-frequency carrier. The binary sequence from the message
signal makes the unipolar input to be either High or Low. The high signal closes the switch, allowing a
carrier wave. Hence, the output will be the carrier signal at high input. When there is low input, the switch
opens, allowing no voltage to appear. Hence, the output will be low.
The band-limiting filter, shapes the pulse depending upon the amplitude and phase characteristics of the
band-limiting filter or the pulse-shaping filter.


There are two types of ASK Demodulation techniques. They are −
 Asynchronous ASK Demodulation/detection
 Synchronous ASK Demodulation/detection
The clock frequency at the transmitter when matches with the clock frequency at the receiver, it is known
as a Synchronous method, as the frequency gets synchronized. Otherwise, it is known as Asynchronous.


The Asynchronous ASK detector consists of a half-wave rectifier, a low pass filter, and a comparator.
Following is the block diagram for the same.

Figure 5.3. ASK Demodulator

The modulated ASK signal is given to the half-wave rectifier, which delivers a positive half output. The low
pass filter suppresses the higher frequencies and gives an envelope detected output from which the
comparator delivers a digital output.


Synchronous ASK detector consists of a Square law detector, low pass filter, a comparator, and a voltage
limiter. Following is the block diagram for the same.
 Figure 5.4. Synchronous ASK Demodulator
The ASK modulated input signal is given to the Square law detector. A square law detector is one whose
output voltage is proportional to the square of the amplitude modulated input voltage. The low pass filter
minimizes the higher frequencies. The comparator and the voltage limiter help to get a clean digital
output.


Frequency Shift Keying (FSK) is the digital modulation technique in which the frequency of the carrier
signal varies according to the digital signal changes. FSK is a scheme of frequency modulation. 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. The following image is the
diagrammatic representation of FSK modulated waveform along with its input.

Figure 5.5. Frequency Shift Keying (FSK)

To find the process of obtaining this FSK modulated wave, let us know about the working of a FSK
modulator.


Binary Frequency Shift Keying (BFSK)

In binary frequency-shift keying (BFSK) the binary data waveform d(t) generates a binary signal

𝑣𝐵𝐹𝑆(𝑡) = √2𝑃𝑠 cos[𝜔𝑡 + 𝑑(𝑡)Ω𝑡] …3.7.1

Here d(t) = + 1 or -1 corresponding to the logic levels 1and 0 of the data waveform. The transmitted signal
is of amplitude √2𝑃𝑠 and is either
𝑣𝐵𝐹𝑆(𝑡) = 𝑆𝐻(𝑡) = √2𝑃𝑠 cos(𝜔 + Ω) 𝑡 …3.7.2
𝑣𝐵𝐹𝑆(𝑡) = 𝑆𝐿(𝑡) = √2𝑃𝑠 cos(𝜔 − Ω) 𝑡 …3.7.3
and thus has an angular frequency ωo + Ω or ωo - Ω with Ω a constant offset from the nominal carrier
frequency ωo. We shall call the higher frequency ωH( = ωo + Ω) and the lower frequency ωL.( = ωo - Ω). We
may conceive that the BFSK signal is generated in the manner indicated in Fig. 3.7.1. Two balanced
modulators are used, one with carrier ωH and one with carrier ωL. The voltage values of PH(t) and of PL(t)
are related to the voltage values of d(t) in the following manner

d(t) PH(t) PL(t)

+1V +1V 0V
-1V 0V +1V

Thus when d(t) changes from +1 to -1 PH changes from 1 to 0 and PL from 0 to 1. At any time either PH or PL
is 1 but not both so that the generated signal is either at angular frequency ωH or at ωL.

Figure 5.6. A representation of a manner in which a BFSK signal can be generated.

The two oscillators, producing a higher and a lower frequency signals, are connected to a switch along with
an internal clock. To avoid the abrupt phase discontinuities of the output waveform during the
transmission of the message, a clock is applied to both the oscillators, internally. The binary input
sequence is applied to the transmitter so as to choose the frequencies according to the binary input.


There are different methods for demodulating a FSK wave. The main methods of FSK detection are
asynchronous detector and synchronous detector. The synchronous detector is a coherent one, while
asynchronous detector is a non-coherent one.


The block diagram of Asynchronous FSK detector consists of two band pass filters, two envelope detectors,
and a decision circuit. Following is the diagrammatic representation.
 Figure 5.7. Asynchronous FSK Detector

The FSK signal is passed through the two Band Pass Filters (BPFs), tuned to Space and Mark frequencies.
The output from these two Band Pass Filters looks like ASK signal; which is then applied to the envelope
detector. The signal in each envelope detector is modulated asynchronously.
The decision circuit chooses which output is more likely and selects it from any one of the envelope
detectors. It also re-shapes the waveform to a rectangular one.


The block diagram of Synchronous FSK detector consists of two mixers with local oscillator circuits, two
band pass filters and a decision circuit. Following is the diagrammatic representation.

Figure 5.8. Synchronous FSK Detector

The FSK signal input is given to the two mixers with local oscillator circuits. These two are connected to two
band pass filters. These combinations act as demodulators and the decision circuit chooses which output is
more likely and selects it from any one of the detectors. The two signals have a minimum frequency
separation.
For both of the demodulators, the bandwidth of each of them depends on their bit rate. This synchronous
demodulator is a bit complex than asynchronous type demodulators.


In M-ary phase-shift keying and in quadrature-amplitude shift keying, any signal could be represented as
C1u1(t) + C2u2(t). There u1(t) and u2(t) are the orthonormal vectors in signal space, that is, 𝑢1(𝑡) =
2
2 . cos(𝜔 𝑡)and 𝑢2(𝑡) = . sin(𝜔 𝑡).
√𝑇𝑠 √𝑇𝑠
The functions u1 and u2 are orthonormal over the symbol interval TS. And, if the symbol is a single bit, TS =
Tb. The coefficients C1 and C2 are constants. The normalized energies associated with C 1u1(t) and with
C2u2(t)are respectively C12 and C22 and the total signal energy is C12 + C 22.
In the present case of BFSK it is appropriate that the orthogonality should result from a special selection of
the frequencies of the unit vectors. Accordingly, with m and n integers, let us establish unit vectors
2
𝑢1(𝑡) = √ cos 2𝜋𝑚ƒ𝑏𝑡
𝑇𝑏 …1
2
and 𝑢2(𝑡) = √ cos 2𝜋𝑛ƒ𝑏𝑡
𝑇𝑏 …2
Where fb=1/Tb. The vectors U1 and U2 are the mth and nth harmonics of the
(fundamental) frequency fb. As we are aware, from the principles of Fourier
analysis, different harmonics (m ± n) are orthogonal over the interval of the
fundamental period Tb = 1/fb.
If now the frequencies fH and fL in a BFSK system are selected to be (assuming
m > n)
ƒ𝐻 = 𝑚ƒ𝑏 …3
and ƒ𝐿 = 𝑛ƒ𝑏 …4
Then corresponding signal vectors are
𝑆𝐻(𝑡) = √𝐸𝑏𝑢1(𝑡) …5
and 𝑆𝐿(𝑡) = √𝐸𝑏𝑢2(𝑡) …6

The signal space representation of these signals is shown in Fig. 3.7.4. The signals, like the unit vectors are orthogonal.
The distance between signal end points is therefore

𝑑 = √2𝐸𝑏

Note that this distance is considerably smaller than

the distance separating end points of BPSK signals,
which are antipodal.


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 −

1. 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°.
2. 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°.


The block diagram of Binary Phase Shift Keying consists of the balance modulator which has the carrier sine
wave as one input and the binary sequence as the other input. Following is the diagrammatic
representation.


PSK Wave

Career Wave
generator
Binary
Sequence Data
Figure 5.10 BPSK Modulator
The modulation of BPSK is done using a balance modulator, which multiplies the two signals applied at the
input. For a zero binary input, the phase will be 0° and for a high input, the phase reversal is of 180°.
Following is the diagrammatic representation of BPSK Modulated output wave along with its given input.
 Figure 5.11. BPSK Modulated Waveform
The output sine wave of the modulator will be the direct input carrier or the inverted (180° phase shifted)
input carrier, which is a function of the data signal.

The block diagram of BPSK demodulator consists of a mixer with local oscillator circuit, a band pass filter, a
two-input detector circuit. The diagram is as follows.

Figure 5.12 BPSK Demodulator

By recovering the band-limited message signal, with the help of the mixer circuit and the band pass filter,
the first stage of demodulation gets completed. The base band signal which is band limited is obtained and
this signal is used to regenerate the binary message bit stream.
In the next stage of demodulation, the bit clock rate is needed at the detector circuit to produce the
original binary message signal. If the bit rate is a sub-multiple of the carrier frequency, then the bit clock
regeneration is simplified. To make the circuit easily understandable, a decision-making circuit may also be
inserted at the 2nd stage of detection.


The Quadrature Phase Shift Keying (QPSK) is a variation of BPSK, and it is also a Double Side Band
Suppressed Carrier (DSBSC) modulation scheme, which sends two bits of digital information at a time,
called as digits.

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.


The QPSK Modulator uses a bit-splitter, two multipliers with local oscillator, a 2-bit serial to parallel
converter, and a summer circuit. Following is the block diagram for the same.
 Figure 5.13 QPSK Modulator
At the modulator’s input, the message signal’s even bits (i.e., 2 nd bit, 4th bit, 6th bit, etc.) and odd bits (i.e.,
1st bit, 3rd bit, 5th bit, etc.) are separated by the bits splitter and are multiplied with the same carrier to
generate odd BPSK (called as PSKI) and even BPSK (called as PSKQ). The PSKQ signal is anyhow phase shifted
by 90° before being modulated.

The QPSK waveform for two-bits input is as follows, which shows the modulated result for different
instances of binary inputs.

Figure 5.14 QPSK Waveforms

The QPSK Demodulator uses two product demodulator circuits with local oscillator, two band pass filters,
two integrator circuits, and a 2-bit parallel to serial converter. Following is the diagram for the same.

Figure 5.15 QPSK Demodulator

The two product detectors at the input of demodulator simultaneously demodulate the two BPSK signals.
The pair of bits is recovered here from the original data. These signals after processing, are passed to the
parallel to serial converter.
 
When b0 = 1 the signal so(t) = √𝑃𝑠. sin(𝜔𝑡), and so(t) = −√𝑃𝑠. sin(𝜔𝑡) when bo = -1. Correspondingly, for
be(t) = ± 1, se(t) = ±.√𝑃𝑠. (𝑡) cos(𝜔𝑡) These four signals have been represented as phasors in Fig. 3.4.4.4.
They are in mutual phase quadrature. Also drawn are the phasors representing the four possible output
signals 𝑣(𝑡) = 𝑠o(𝑡) + 𝑆e(𝑡). These four possible output signals have equal amplitude √2𝑃𝑠 and are in
phase quadrature; they have been identified by their corresponding values of b o and be. At the end of each
bit interval (i.e., after each time Tb) either bo, or be can change, but both cannot change at the same time.
Consequently, the QPSK system shown in Fig. 3.4.4.3 is called offset or staggered QPSK and abbreviated
OQPSK. After each time Tb, the transmitted signal, if it changes, changes phase by 90° rather than by 180°
as in BPSK.

Figure 5.16. Phasor diagrams for the sinusoids


Suppose that in Fig. 3.4.4.3 we introduce an additional flip-flop before either the odd or even flip-flop. Let
this added flip-flop be driven by the clock which runs at the rate f b. Then one or the other bit streams, odd
or even, will be delayed by one bit interval. As a result, we shall find that two bits which occur in time
sequence (i.e., serially) in the input bit stream b(t) will appear at the same time (i.e., in parallel) at the
outputs of the odd and even flip-flops. In this case be(t) and bo(t) can change at the same time, after each
time 2Tb, and there can be a phase change of 180° in the output signal. There is no difference, in principle,
between a staggered and non-staggered system.
In practice, there is often a significant difference between QPSK and OQPSK. At each transition time, T" for
OQPSK and 2Tb for QPSK, one bit for OQPSK and perhaps two bits for QPSK change from 1V to -1V or -1V to
1V. Now the bits be(t) and bo(t) can, not change instantaneously and, in changing, must pass through zero
and dwell in that neighborhood at least briefly. Hence there will be brief variations in the amplitude of the
transmitted waveform. These variations will be more pronounced in QPSK than in OQPSK since in the first
case both be(t) and bo(t) may be zero simultaneously so that the signal amplitude may actually be reduced
to zero temporarily.


In Differential Phase Shift Keying (DPSK) 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.
 Figure 5.17 Differential Phase Shift Keying (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 continued as it was. If the data is a 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.


DPSK is a technique of BPSK, in which there is no reference phase signal. Here, the transmitted signal itself
can be used as a reference signal. Following is the diagram of DPSK Modulator.

Figure 5.18 DPSK Modulator

DPSK encodes two distinct signals, i.e., the carrier and the modulating signal with 180° phase shift each.
The serial data input is given to the XNOR gate and the output is again fed back to the other input through
1-bit delay. The output of the XNOR gate along with the carrier signal is given to the balance modulator, to
produce the DPSK modulated signal.


In DPSK demodulator, the phase of the reversed bit is compared with the phase of the previous bit.
Following is the block diagram of DPSK demodulator.

Figure 5.19 DPSK Demodulator

From the above figure, it is evident that the balance modulator is given the DPSK signal along with 1-bit
delay input. That signal is made to confine to lower frequencies with the help of LPF. Then it is passed to a
shaper circuit, which is a comparator or a Schmitt trigger circuit, to recover the original binary data as the
output.
 The word binary represents two bits. M represents a digit that corresponds to the number of conditions,
levels, or combinations possible for a given number of binary variables.
This is the type of digital modulation technique used for data transmission in which instead of one bit, two
or more bits are transmitted at a time. As a single signal is used for multiple bit transmission, the channel
bandwidth is reduced.
If a digital signal is given under four conditions, such as voltage levels, frequencies, phases, and amplitude,
then M = 4. The number of bits necessary to produce a given number of conditions is expressed
mathematically as N=log2M. Where N is the number of bits necessary M is the number of conditions,
levels, or combinations possible with N bits.
The above equation can be re-arranged as
2N=M
For example, with two bits, 22 = 4 conditions are possible.


In general, Multi-level (M-ary) modulation techniques are used in digital communications as the digital
inputs with more than two modulation levels are allowed on the transmitter’s input. Hence, these
techniques are bandwidth efficient. There are many M-ary modulation techniques. Some of these
techniques, modulate one parameter of the carrier signal, such as amplitude, phase, and frequency.


This is called M-ary Amplitude Shift Keying (M-ASK) or M-ary Pulse Amplitude Modulation (PAM).
The amplitude of the carrier signal, takes on M different levels.


𝑆(𝑡) = 𝐴𝑚 cos(2𝜋ƒ𝑐𝑡) 𝐴𝑚 ∈(2𝑚 − 1 − 𝑀)∆, 𝑚 = 1,2, … … 𝑀 and 0 ≤ 𝑡 ≤ 𝑇𝑠
Some prominent features of M-ary ASK are −
 This method is also used in PAM.
 Its implementation is simple.
 M-ary ASK is susceptible to noise and distortion.

An M-ary FSK communications system is shown in Fig. 5.19. It is an obvious extension of a binary FSK
system. At the transmitter an N-bit symbol is presented each TS, to an N-bit D/A converter. The converter
output is applied to a frequency modulator, i.e., a piece of hardware which generates a carrier waveform
whose frequency is determined by the modulating waveform. The transmitted signal, for the duration of
the symbol interval, is of frequency f0 or f1 ...or fM-1 with M = 2N. At the receiver, the incoming signal is
applied to M paralleled band pass filters each followed by an envelope detector. The band pass filters have
center frequencies f0, f1, ... ,fM-1. The envelope detectors apply their outputs to a device which determines
which of the detector indications is the largest and transmits that envelope output to an N-bit AID
converter.
The probability of error is minimized by selecting frequencies f0, f1, ... ,fM-1 so that the M signals are
mutually orthogonal. One commonly employed arrangement simply provides that the carrier frequency be
successive even harmonics of the symbol frequency fS=1/TS. Thus the lowest frequency, say f0, is f0 = k fS,
while f1 = (k + 1) fS, f2 = (k + 2) fS etc. In this case, the spectral density patterns of the individual possible
transmitted signals overlap in the manner shown in Fig. 5.20. We observe that to pass M-ary FSK the
required spectral range is
𝐵 = 2𝑀ƒ𝑆 …1
N
Since fS= fb/N and M=2 , we have
𝐵 = 2𝑁+1ƒ𝑏/𝑁 …2
 Figure 5.20 An M-ARY Communication System
Note that M-ary FSK requires a considerably increased bandwidth in comparison with M-ary PSK.
However, as we shall see, the probability of error for M-ary FSK decreases as M increases, while for M-ary
PSK, the probability of error increases with M.

Figure 5.21. Power Spectral Density of an M-ARY FSK (Four Frequencies are shown)

Geometrical Representation of an M-ARY FSK

The case of M-ary orthogonal FSK signals is shown
in figure 5.21. We simply conceive of a coordinate
system with M mutually orthogonal coordinate
axes. The square of the length of the signal vector
is the normalized signal energy. When the
frequencies are selected to generate orthogonal
signals.
 Figure 5.22 Geometrical representation of orthogonal M-ary FSK (M = 3)

Note that this value of d is greater than the values of d calculated for M-ary PSK with the exception of the
cases M = 2 and M = 4. It is also greater than d in the case of 16-QASK.
𝑑 = √2𝐸𝑠 = √2𝑁𝐸𝑏

This is called as M-ary Frequency Shift Keying (M-ary FSK).
The frequency of the carrier signal, takes on M different levels.

𝜋
𝑆(𝑡) = √2𝐸
𝑇
𝑠
cos ( (𝑛𝑐 + )𝑡) 0 ≤ 𝑡 ≤ 𝑇𝑠 = 1,2, … … 𝑀
𝑠 𝑇𝑠
Where ƒ𝑐 = 𝑛𝑐/ 2𝑇𝑠
for some fixed integer n.
Some prominent features of M-ary FSK are −
 Not susceptible to noise as much as ASK.
 The transmitted M number of signals are equal in energy and duration.
 The signals are separated by 12Ts
 Hz making the signals orthogonal to each other.
 Since M signals are orthogonal, there is no crowding in the signal space.
 The bandwidth efficiency of M-ary FSK decreases and the power efficiency increases with the
increase in M.

This is called as M-ary Phase Shift Keying (M-ary PSK).
The phase of the carrier signal, takes on M different levels.
Representation of M-ary PSK
2𝐸
𝑆 (𝑡) = √ cos(𝜔 + ) 0 ≤ 𝑡 ≤ 𝑇 𝑎𝑛𝑑 = 1,2, … … 𝑀
𝑇 0𝑡
2
(𝑡) = where = 1,2, … … 𝑀
𝑀
Some prominent features of M-ary PSK are −
 The envelope is constant with more phase possibilities.
 This method was used during the early days of space communication.
 Better performance than ASK and FSK.
 Minimal phase estimation error at the receiver.
 The bandwidth efficiency of M-ary PSK decreases and the power efficiency increases with the
increase in M.
In BPSK we transmit each bit individually. Depending on whether b(t) is logic 0 or logic 1, we transmit one
or another of a sinusoid for the bit time Tb, the sinusoids differing in phase by 2π/2 = 1800. In QPSK we
lump together two bits. Depending on which of the four two-bit words develops, we transmit one or
another of four sinusoids of duration 2Tb the sinusoids differing in phase by amount 2π/4 = 90°. The
scheme can be extended. Let us lump together N bits so that in this N-bit symbol, extending over the time
NTb, there are 2N = M possible symbols. Now let us represent the symbols by sinusoids of duration NT b= Ts
which differ from one another by the phase 2 π / M. Hardware to accomplish such M-ary communication is
available.
Thus in M-ary PSK the waveforms used to identify the symbols are
𝑣(𝑡) = √2𝑃𝑠 cos(𝜔𝑡 + ∅𝑚) (m=0, 1, …, M-1) …1
Where phase angle is given by
𝜋 …2
∅𝑚 = (2𝑚 + 1)
𝑀
 The waveforms of Eq. are represented by the dots in Fig. 5.1 in a signal space in which the coordinate axes
are the orthonormal waveforms 𝑢1(𝑡) = √2/𝑇𝑠 cos(𝜔𝑡) and 𝑢2(𝑡) = √2/𝑇𝑠 sin(𝜔𝑡). The distance of each
dot from the origin is √𝐸𝑠 = √𝑃𝑠𝑇𝑠

From Eq. (1) we have

𝑣(𝑡) = (√2𝑃𝑠 cos ∅𝑚) cos(𝜔𝑡) − (√2𝑃𝑠 sin ∅𝑚) sin(𝜔𝑡) …3

Defining pe and po by

𝑝 = √2𝑃𝑠 cos ∅𝑚 …4
𝑝 = √2𝑃𝑠 sin ∅𝑚 …5
Equation 3 becomes

𝑣(𝑡) = 𝑝 cos(𝜔𝑡) − 𝑝 sin(𝜔𝑡) …6

Figure 5.23 Graphical representation of M-ary PSK Signals



Figure.5.24. M Ary Transmitter

The transmitter, the bit stream b(t) is applied to a serial-to-parallel converter. This converter has facility for
storing the N bits of a symbol. The N bits have been presented serially, that is, in time sequence, one after
another. These N bits, having been assembled, are then presented all at once on N output lines of the
converter, that is they are presented in parallel. The converter output remains unchanging for the duration
NTb of a symbol during which time the converter is assembling a new group of N bits. Each symbol time
the converter output is updated.
The converter output is applied to a D/A converter. This D/A converter generates an output voltage which
assumes one of 2N = M different values in a one to-one correspondence to the M possible symbols applied
to its input. That is, the D/A output is a voltage v(Sm) which depends on the symbol Sm (m = 0, 1,... ,M - 1).

Finally v(Sm) is applied as a control input to a special type of constant amplitude sinusoidal signal source whose phase
4>m is determined by v(Sm). Altogether, then, the output is a fixed amplitude, sinusoidal waveform, whose phase has a
one-to-one correspondence to the assembled N-bit symbol. The phase can change once per symbol time.


Quadrature Amplitude Modulation, QAM utilises both amplitude and phase components to provide a form
of modulation that is able to provide high levels of spectrum usage efficiency. QAM, quadrature amplitude
modulation has been used for some analogue transmissions. QAM is a signal in which two carriers shifted
in phase by 90 degrees (i.e. sine and cosine) are modulated and combined. As a result of their 90° phase
difference they are in quadrature and this gives rise to the name. Often one signal is called the In-phase or
“I” signal, and the other is the quadrature or “Q” signal.
The resultant overall signal consisting of the combination of both I and Q carriers contains of both
amplitude and phase variations. In view of the fact that both amplitude and phase variations are present it
may also be considered as a mixture of amplitude and phase modulation.

The basic way in which a QAM signal can be generated is to generate two signals that are 90° out of phase
with each other and then sum them. This will generate a signal that is the sum of both waves, which has
certain amplitude resulting from the sum of both signals and a phase which again is dependent upon the
sum of the signals.
Consider the following block diagram of a Quadrature Amplitude Modulation (QAM) and Demodulation
system:
 m1(t)cos2(ct) + m2(t)sin(ct)cos(ct)
=m1(t)/2+m1(t) cos(2ct)/2 + m2(t)sin(2ct)/2

IN-PHASE Baseband Around c Around c

modulator branch

m1(t)cos(ct)
HLPF()
m1(t) X X BW = 2B
m1(t)/2

IN-PHASE
cos(ct) cos(ct)
demodulator branch

Phase Shifter Phase Shifter
sin(ct) sin(ct) QUADRATURE
– /2 – /2
demodulator branch

HLPF()
m2(t)
X X BW = 2B
m2(t)/2
m2(t)sin(ct)
QUADRATURE
modulator branch m1(t)cos(ct) + m2(t)sin(ct) m1(t)sin(ct)cos(ct) + m 2(t)sin2( ct)
=m1(t)sin(2ct)/2 + m2(t)/2 – m2(t)cos(2ct)/2

Around c Baseband Around c

QAM Modulator/Demodulator
Figure.5.24. QAM Modulator and demodulator

The modulator/demodulator system shown above clearly is able to modulate and demodulate two
different signals without any interference. However, if the generation of the carrier at the demodulator
had even small phase or frequency errors, the demodulated signals will interfere at the outputs. The
following figure illustrate what happens when the carrier at the demodulator has a small frequency error
 (must be a small value much less than c) and/or a small phase error 
 

m1(t)cos(ct)cos[(c+t+ + m2(t)sin(ct)cos[(c+t+
=(1/2)[m1(t)cos(t+) + m1(t) cos(2ct+t+) – m2(t)sin(t+) + m2(t)sin(2ct+t+)]

Baseband Around c Baseband Around c

m1(t)cos(ct)
m (t) X X HLPF() (1/2)[m (t)cos(t+) – m (t)sin(t+)]
1 BW = 2B 1 2

cos(ct) cos[(c+t+

 
Phase Shifter Phase Shifter
sin(ct) – /2 – /2 sin[(c+t+

m (t) X X HLPF() (1/2)[m (t)sin(t+) + m (t)cos(t+)]
2
m (t)sin( t) BW = 2B 1 2
2 c

m (t)cos( t) + m (t)sin( t) m1(t)cos(ct)sin[(c+t+ + m2(t)sin(ct)sin[(c+t+

1 c 2 c =(1/2)[m (t)sin(t+) + m (t) sin(2 t+t+) + m (t)cos(t+) – m (t)cos(2 t+t+)]
1 1 c 2 2 c

Baseband Around c Baseband Around c

QAM Modulator/Demodulator with Demodulator Carrier Phase and/or Frequency Error

If the carrier at the receiver has a small frequency error  (but a phase error =0), we see that the two
output signal become

r (t) 
1

m (t) cos(t)  m (t)sin( t)
1 1 2
2
r (t) 
1

m (t)sin( t)  m (t) cos(t)
2 1 2
2 .

Clearly, in this case, the output signals are not purely either of the two message signals but a combination.
The ratio of message 1 to message 2 at the different outputs changes as a sinusoid with a frequency equal
to the frequency error .

If the carrier at the receiver has a phase error  (but a frequency error  = 0), we see that the two output
signal become

r (t) 
1
m (t) cos( )  m

(t)sin(  )
1 1 2
2
r (t) 
1
m (t)sin(  )  m (t) cos( )


2 1 2
2
