ANALOG & DIGITAL COMMUNICATION

LECTURE NOTES
B.TECH (II- YEAR, IV
SEM) (2024-25)

Prepared By: Dr. Deepak Bhatia

Department of Electronics Engineering

Rajasthan Technical University

Rawat Bhata Road, Kota-324010
 Syllabus:

Analog and Digital Communication

(EC)

B. Tech. IV Semester Max. Marks: 150

3L+0T

Handout/ Lecture Plan:

4EC407 DCC Analog and Digital MM:150 3L:0T:0P 3

Communication credit

Review of signals and systems, Frequency domain representation of

signals, Principles of Amplitude Modulation Systems -DSB, SSB and
VSB modulations. Angle Modulation, Representation of FM and PM
signals, Spectral characteristics of angle modulated signals.

Total: 8 Lecture

Review of probability and random process. Gaussian and white noise

characteristics, Noise in amplitude modulation systems, Noise in
Frequency modulation systems. Pre -emphasis and De -emphasis,
Threshold effect in angle modulation. Total: 7 Lecture

Pulse modulation. Sampling process. Pulse Amplitude and Pulse code

modulation (PCM), Differential pulse code modulation. Delta modulation,
Noise considerations in PCM, Time Division multiplexing, Digital
Multiplexers. Total: 8 Lecture

Elements of Detection Theory, Optimum detection of signals in noise,

Coherent communication with waveforms - Probability of Error
evaluations. Baseband Pulse Transmission - Inter symbol Interference
and Nyquist criterion. Pass band Digital Modulation schemes - Phase
Shift Keying, Frequency Shift Keying, Quadrature Amplitude Modulation,
Continuous Phase Modulation and Minimum Shift Keying.
Total: 9 Lecture

Digital Modulation tradeoffs. Optimum demodulation of digital signals

over band -limited channels -Maximum likelihood sequence detection
(Viterbi receiver). Equalization Techniques. Synchronization and Carrier
Recovery for Digital modulation. Total: 8 Lecture

Text/ Reference Books:

1. Haykin S., "Communications Systems", John Wiley and Sons,

2001.

2. Taub H. and Schilling D.L., "Principles of Communication Systems”,

Tata McGraw Hill, 2001.

3. Proakis J. G. and Salehi M., "Communication Systems

Engineering",Pearson Education, 2002.

4. Wozencraf t J. M. and Jacobs I. M., ``Principles of Communication

Engineering'', John Wiley, 1965.

5. Barry J. R., Lee E. A. and Messerschmitt D. G., ``Digital

Communication'', Kluwer Academic Publishers, 2004.

6. Proakis J.G., ``Digital Communications'', 4th Edition, McGraw Hill,

2000.
 Course Outcome:

Course Course Course

Code Name Outco
Details
me

CO 1 Analyze and compare different analog

modulation schemes for their efficiency
and
Analog and Digital Communication

bandwidth

CO 2 Analyze the behavior of a communication

system in presence of noise

4EC4-07

CO 3 Investigate pulsed modulation system and

analyze their system performance

CO 4 Analyze different digital modulation

schemes and can compute the bit error
performance

CO 5 Design a communication system

comprised of both analog and digital
modulation techniques
 CO-PO Mapping:

Course
Subject

Out
PO 1 PO PO PO PO PO PO PO PO PO PO PO
comes 2 3 4 5 6 7 8 9 10 11 12

CO 1 3 3 3 1 1
Analog & Digital
Communication

CO 2 3 2 3 1
4EC4 -02

CO 3 3 2 3 2

CO 4 3 3 3 2 1

CO 5 3 2 3 3 3 2 2

3: Strongly 2: Moderate 1: Weak

Content delivery method:

1. Chalk and Duster

2. PPT
 Lecture Plan:

Lecture No. Content to be taught

Lecture 1 Introduction to the COURSE

Lecture 2 Review of signals and systems, Frequency domain

representation

of signals

Lecture 3 Principles of Amplitude Modulation Systems - DSB, SSB

and VSB

modulations

Lecture 4 Principles of Amplitude Modulation Systems - DSB, SSB

and VSB

modulations

Lecture 5 Principles of Amplitude Modulation Systems - DSB, SSB

and VSB

modulations

Lecture 6 Angle Modulation, Representation of FM and PM signals

Lecture 7 Angle Modulation, Representation of FM and PM signals

Lecture 8 Spectral characteristics of angle modulated signals.

Lecture 9 Review of probability and random process

Lecture 10 Review of probability and random process

Lecture 11 Noise in amplitude modulation systems

Lecture 12 Noise in amplitude modulation systems

Lecture 13 Noise in Frequency modulation systems

Lecture 14 Pre-emphasis and De emphasis

Lecture 15 Threshold effect in angle modulation

Lecture 16 Pulse modulation. Sampling

Lecture 17 Pulse Amplitude and Pulse code modulation (PCM)

Lecture 18 Pulse Amplitude and Pulse code modulation (PCM)

Lecture 19 Differential pulse code modulation

Lecture 20 Delta modulation

Lecture 21 Noise considerations in PCM

Lecture 22 Time Division multiplexing, Digital Multiplexers

Lecture 23 Elements of Detection Theory

Lecture 24 Optimum detection of signals in noise

Lecture 25 Coherent communication with waveforms- Probability of

Error

evaluations

Lecture 26 Coherent communication with waveforms- Probability of

Error

evaluations

Lecture 27 Baseband Pulse Transmission - Inter symbol Interference

and Nyquist criterion

Lecture 28 Baseband Pulse Transmission - Inter symbol Interference

and

Nyquist criterion

Lecture 29 Pass band Digital Modulation schemes

Lecture 30 Phase Shift Keying

Lecture 31 Frequency Shift Keying

Lecture 32 Quadrature Amplitude Modulation

Lecture 33 Continuous Phase Modulation and Minimum Shift Keying.

Lecture 34 Digital Modulation tradeoffs

Lecture 35 Optimum demodulation of digital signals over band-limited

channels

Lecture 36 Optimum demodulation of digital signals over band-limited

channels

Lecture 37 Maximum likelihood sequence detection (Viterbi receiver)

Lecture 38 Equalization Techniques

Lecture 39 Synchronization and Carrier Recovery for Digital modulation

Lecture 40 Synchronization and Carrier Recovery for Digital modulation

 Assignments:

Assignment 1 Q1. Design Modulator and Demodulator of SSB-SC

Modulation based on its mathematical expression.

Q2. Derive the figure of merit in

a) FM Receiver b) PM Receiver

Q3. A Carrier signal c(t) = 20cos (2 n10 6t) is

modulated by a message signal having three
frequencies 5 KHz, 10 KHz & 20 KHz. The
corresponding modulation indexes are 0.4, 0.5 & 0.6.
Sketch the spectrum. Calculate bandwidth, power and
efficiency.

Assignment 2 Q1. Derive the expression for probability of error in ASK,

FSK and PSK systems and compare them.

Q2. With block diagrams explain about DPCM & DM.

also compare them.

Q3. A message signal m(t) = 4 cos (2n10 3t) is sampled

at nyquist rate and transmitted through a channel
using 3 -bit PCM system.

i. Calculate all the parameters of the PCM.

ii. If the sampled values are 3.8, 2.1, 0.5, - 1.7, -3.2 & -4
then determine the quantizer output, encoder output and
quantization error per each sample.

iii. Sketch the transfer characteristics of the quantizer.

 UNIT-I

Introduction to Communication System

Communication is the process by which information is exchanged between
individuals through a medium.

Communication can also be defined as the transfer of information from one point
in space and time to another point.

The basic block diagram of a communication system is as shown in Fig.1.

Fig.1 Block Diagram of a Communication System

 Transmitter: Couples the message into the channel using high frequency signals.
 Channel: The medium used for transmission of signals

 Modulation: It is the process of shifting the frequency spectrum of a signal to a

frequency range in which more efficient transmission can be achieved.
 Receiver: Restores the signal to its original form.

 Demodulation: It is the process of shifting the frequency spectrum back to the original
baseband frequency range and reconstructing the original form.

Modulation:
Modulation is a process that causes a shift in the range of frequencies in a signal.

• Signals that occupy the same range of frequencies can be separated.

  Modulation helps in noise immunity, attenuation - depends on the
physical medium. The below figure shows the different kinds of analog
modulation schemes that are available

Modulation is operation performed at the transmitter to achieve efficient and reliable

information transmission.

For analog modulation, it is frequency translation method caused by changing the

appropriate quantity in a carrier signal.

It involves two waveforms:

 A modulating signal/baseband signal – represents the message.
 A carrier signal – depends on type of modulation.

Once this information is received, the low frequency information must be removed from
the high frequency carrier. This process is known as “Demodulation”.

Need for Modulation:

 Baseband signals are incompatible for direct transmission over the medium so,
modulation is used to convey (baseband) signals from one place to another.
 Allows frequency translation:
o Frequency Multiplexing
o Reduce the antenna height
o Avoids mixing of signals
o Narrowbanding
 Efficient transmission

 Reduced noise and interference

 Types of Modulation:
Three main types of modulations:
Analog Modulation
 Amplitude modulation
Example: Double sideband with carrier (DSB-WC), Double- sideband suppressed
carrier (DSB-SC), Single sideband suppressed carrier (SSB-SC), vestigial sideband
(VSB)
 Angle modulation (Frequency modulation & Phase modulation)
Example: Narrow band frequency modulation (NBFM), Wideband frequency
modulation (WBFM), Narrowband phase modulation (NBPM), Wideband phase
modulation (NBPM)

Pulse Modulation
 Carrier is a train of pulses

 Example: Pulse Amplitude Modulation (PAM), Pulse width modulation (PWM) , Pulse
Position Modulation (PPM)

Digital Modulation
 Modulating signal is analog
o Example: Pulse Code Modulation (PCM), Delta Modulation (DM), Adaptive Delta
Modulation (ADM), Differential Pulse Code Modulation (DPCM), Adaptive
Differential Pulse Code Modulation (ADPCM) etc.
 Modulating signal is digital (binary modulation)
o Example: Amplitude shift keying (ASK), frequency Shift Keying (FSK), Phase Shift
Keying (PSK) etc

Frequency Division Multiplexing

Multiplexing is the name given to techniques, which allow more than one
message to be transferred via the same communication channel. The channel in this
context could be a transmission line, e.g. a twisted pair or co-axial cable, a radio system
or a fiber optic system etc.

FDM is derived from AM techniques in which the signals occupy the same
physical ‘line’ but in different frequency bands. Each signal occupies its own specific
band of frequencies all the time, i.e. the messages share the channel bandwidth.
FDM – messages occupy narrow bandwidth – all the time.
Multiplexing requires that the signals be kept apart so that they do not interfere
with each other, and thus they can be separated at the receiving end. This is
accomplished by separating the signal either in frequency or time. The technique of
separating the signals in frequency is referred to as frequency division multiplexing
(FDM), whereas the technique of separating the signals in time is called time- division
multiplexing (TDM).
Fig.2 shows the block diagram of FDM System. As shown in the figure, input
message signals, assumed to be of the low-pass type are passed through input low-pass
filters. This filtering action removes high-frequency components that do not contribute
significantly to signal representation but may disturb other message signals that share
the common channel.
The filtered message signals are then modulated with necessary carrier
frequencies with the help of modulators. The most commonly used method of
modulation in FDM is single sideband modulation, which requires a bandwidth that is
approximately equal to that of original message signal. The band pass filters following
the modulators are used to restrict the band of each modulated wave to its prescribed
range. The outputs of BPF are combined in parallel which form the input to the common
channel.
At the receiving end, BPF connected to the common channel separate the
message signals on the frequency occupancy basis. Finally, the original message signals
are recovered by individual demodulators.

Fig.2. Frequency Division Multiplexing

 Amplitude Modulation (AM)
Amplitude Modulation is the process of changing the amplitude of a relatively
high frequency carrier signal in accordance with the amplitude of the modulating signal
(Information).

The carrier amplitude varied linearly by the modulating signal which usually consists of
a range of audio frequencies. The frequency of the carrier is not affected.

Application of AM –
 Radio broadcasting,
 TV pictures (video),
 Facsimile transmission

 Frequency range for AM - 535 kHz – 1600 kHz

 Bandwidth - 10 kHz

Various forms of Amplitude Modulation

• Conventional Amplitude Modulation (Alternatively known as Full AM or Double
Sideband Large carrier modulation (DSBLC) /Double Sideband Full Carrier (DSBFC)

• Double Sideband Suppressed carrier (DSBSC) modulation

• Single Sideband (SSB) modulation

• Vestigial Sideband (VSB) modulation

Time Domain and Frequency Domain Description

It is the process where, the amplitude of the carrier is varied proportional to that
of the message signal.

Let m (t) be the base-band signal, m (t) ←→ M (ω) and c (t) be the carrier, c(t)
= Ac cos(ωc t). fc is chosen such that fc >> W, where W is the maximum frequency
component of m(t). The amplitude modulated signal is given by

𝑠(𝑡) = 𝐴𝑐 [1 + 𝑘𝑎 𝑚 (𝑡 )]cos⁡(2𝜋𝑓𝑐 𝑡)

Fourier Transform on both sides of the above equation

S(ω) = π Ac/2 (δ(ω − ωc) + δ(ω + ωc)) + k aAc/ 2 (M(ω − ωc) + M(ω + ωc))
ka is a constant called amplitude sensitivity.
kam(t) < 1 and it indicates percentage modulation.

Fig.3. Amplitude modulation in time and frequency domain

Single Tone Modulation:

Consider a modulating wave m(t ) that consists of a single tone or single frequency
component given by

m(t) =Am cos(2πfmt) --------------- (1)

where Am is peak amplitude of the sinusoidal modulating wave
fm is the frequency of the sinusoidal modulating wave
Let Ac be the peak amplitude and fc be the frequency of the carrier signal. Then the
corresponding single tone AM wave is given by
s(t)=Ac [1+ m cos(2πfmt)]cos(2πfct)--------------(2)
m is the modulation index
Let Amax and Amin denote the maximum and minimum values of the envelope of the
modulated wave. Then from the above equation we get
𝐴𝑚𝑎𝑥 𝐴𝑐(1 + 𝑚)
=
𝐴𝑚𝑖𝑛 𝐴𝑐(1 − 𝑚)
𝐴𝑚𝑎𝑥 − 𝐴𝑚𝑖𝑛
𝑚=
𝐴𝑚𝑎𝑥 + 𝐴𝑚𝑖𝑛
Expanding the equation (2), we get
1 1
𝑠(𝑡 ) = 𝐴𝑐 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡) + 𝑚𝐴𝑐 𝑐𝑜𝑠[2𝜋 (𝑓𝑐 + 𝑓𝑚 )𝑡] + 𝑚𝐴𝑐 𝑐𝑜𝑠[2𝜋 (𝑓𝑐 − 𝑓𝑚 )𝑡]
2 2
The Fourier transform of s(t) is obtained as follows
1 1
𝑆 (𝑓 ) = 𝐴𝑐 [𝛿 (𝑓 − 𝑓𝑐 ) + 𝛿 (𝑓 + 𝑓𝑐 )] + 𝑚𝐴𝑐 [𝛿 (𝑓 − 𝑓𝑐 − 𝑓𝑚 ) + 𝛿 (𝑓 + 𝑓𝑐 + 𝑓𝑚 )]
2 4
1
+ 𝑚𝐴𝑐 [𝛿 (𝑓 − 𝑓𝑐 + 𝑓𝑚 ) + 𝛿 (𝑓 + 𝑓𝑐 − 𝑓𝑚 )]
4
Thus the spectrum of an AM wave, for the special case of sinusoidal modulation
consists of delta functions at ±⁡𝑓𝑐, 𝑓𝑐 ± 𝑓𝑚⁡𝑎𝑛𝑑 − 𝑓𝑐 ± 𝑓𝑚. The spectrum for
positive frequencies is as shown in the figure

Fig.4. Frequency Domain characteristics of single tone AM

Power relations in AM waves:

Consider the expression for single tone/sinusoidal AM wave

1 1
𝑠(𝑡) = 𝐴𝑐 𝑐𝑜𝑠(2𝜋𝑓𝑐 𝑡) + 𝑚𝐴𝑐 𝑐𝑜𝑠[2𝜋 (𝑓𝑐 + 𝑓𝑚 )𝑡 ] + 𝑚𝐴𝑐 𝑐𝑜𝑠[2𝜋 (𝑓𝑐 − 𝑓𝑚 )𝑡]---------
2 2

(1)
This expression contains three components.They are carrier component, upper
side band and lower side band. Therefore Average power of the AM wave is the sum of
these three components.
Therefore the total power in the amplitude modulated wave is given by
2
𝑉𝑐𝑎𝑟 2
𝑉𝐿𝑆𝐵 2
𝑉𝑈𝑆𝐵
𝑃𝑡 = + + ------------ (2)
𝑅 𝑅 𝑅
Where all the voltages are rms values and R is the resistance, in which the power is
dissipated.
𝐴
2
𝑉𝑐𝑎𝑟 ( 𝑐⁄ )2 𝐴2
𝑃𝑐 = = √2 = 𝑐
𝑅 𝑅 2𝑅

2
𝑉𝐿𝑆𝐵 𝑚𝐴𝑐 2 1 𝑚 2 𝐴2𝑐 𝑚 2
𝑃𝐿𝑆𝐵 = =[ ] = = 𝑃
𝑅 2√2 𝑅 8𝑅 4 𝑐
 2
𝑉𝑈𝑆𝐵 𝑚𝐴𝑐 2 1 𝑚 2 𝐴2𝑐 𝑚 2
𝑃𝑈𝑆𝐵 = =[ ] = = 𝑃
𝑅 2√2 𝑅 8𝑅 4 𝑐

Therefore total average power is given by

𝑃𝑡 = 𝑃𝑐 + 𝑃𝐿𝑆𝐵 + 𝑃𝑈𝑆𝐵
𝑚2 𝑚2
𝑃𝑡 = 𝑃𝑐 + 𝑃𝑐 + 𝑃
4 4 𝑐
𝑚2 𝑚2
𝑃𝑡 = 𝑃𝑐 (1 + + )
4 4
𝑚2
⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡𝑃𝑡 = 𝑃𝑐 (1 + ) ------------- (3)
2

The ratio of total side band power to the total power in the modulated wave is
given by
2
𝑃𝑆𝐵 𝑃𝑐 (𝑚 ⁄2)
= 2
𝑃𝑡 𝑃𝑐 (1 + 𝑚 ⁄2)

𝑃𝑆𝐵 𝑚2
= ------------- (4)
𝑃𝑡 2+𝑚 2

This ratio is called the efficiency of AM system Generation of AM waves:

Two basic amplitude modulation principles are discussed. They are square law
modulation and switching modulator.

Square Law Modulator

When the output of a device is not directly proportional to input throughout the
operation, the device is said to be non-linear. The Input-Output relation of a non-linear
device can be expressed as
2 3 4
𝑉𝑂 = 𝑎0 + 𝑎1 𝑉𝑖𝑛 + 𝑎2 𝑉𝑖𝑛 + 𝑎3 𝑉𝑖𝑛 + 𝑎4 𝑉𝑖𝑛 + ⋯ … … … … … ..

When the input is very small, the higher power terms can be neglected. Hence
the output is approximately given by
2
𝑉𝑂 = 𝑎0 + 𝑎1 𝑉𝑖𝑛 + 𝑎2 𝑉𝑖𝑛
When the output is considered up to square of the input, the device is called a
square law device and the square law modulator is as shown in the figure 5
 Fig.5. Square Law Modulator
Consider a non-linear device to which a carrier c(t)=Ac cos(2πfct) and an
information signal m(t) are fed simultaneously as shown in figure 4. The total input to
the device at any instant is
𝑉𝑖𝑛 = 𝑐(𝑡) + 𝑚(𝑡)

𝑉𝑖𝑛 = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) + 𝑚(𝑡)

As the level of the input is very small, the output can be considered upto square of the
input i.e.,
2
𝑉𝑂 = 𝑎0 + 𝑎1 𝑉𝑖𝑛 + 𝑎2 𝑉𝑖𝑛

2
𝑉𝑂 = 𝑎0 + 𝑎1 [𝐴𝑐 cos(2𝜋𝑓𝑐𝑡) + 𝑚(𝑡)] + 𝑎2 [𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) + 𝑚(𝑡)]

𝑎2 𝐴2𝑐
𝑉𝑂 = 𝑎0 + 𝑎1 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) + 𝑎1 𝑚(𝑡)] + (1 + 𝑐𝑜𝑠4𝜋𝑓𝑐 𝑡) + 𝑎2 [𝑚(𝑡)]2
2
+ 2𝑎2 𝑚(𝑡)𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡)

𝑎2 𝐴2𝑐 𝑎2 𝐴2𝑐
𝑉𝑂 = 𝑎0 + 𝑎1 𝐴𝑐 cos(2𝜋𝑓𝑐𝑡) + 𝑎1𝑚(𝑡)] + (𝑐𝑜𝑠4𝜋𝑓𝑐 𝑡) + + 𝑎2 [𝑚(𝑡 )]2
2 2
+ 2𝑎2 𝑚(𝑡)𝐴𝑐 cos(2𝜋𝑓𝑐𝑡)
Taking Fourier Transform on both sides , we get
𝑎2 𝐴2𝑐 𝑎1 𝐴𝑐
𝑉𝑜 (𝑓) = (𝑎0 + ) 𝛿 (𝑓 ) + [𝛿 (𝑓 − 𝑓𝑐 ) + 𝛿 (𝑓 + 𝑓𝑐 )] + 𝑎1 𝑀(𝑓)
2 2
𝑎2 𝐴2𝑐
+ [𝛿 (𝑓 − 2𝑓𝑐 ) + 𝛿 (𝑓 + 2𝑓𝑐 )] + 𝑎2 𝑀(𝑓)
4
+ 𝑎2 𝐴𝑐 [𝑀(𝑓 − 𝑓𝑐 + 𝑀 (𝑓 + 𝑓𝑐 ))]
Therefore the square law device output 0 V consists of the dc component at f = 0. The
information signal ranging from 0 to W Hz and its second harmonics are signal at fc and
2fc.

Frequency band centered at 𝑓𝑐 with a deviation of ± W Hz.The required AM

signal with a carrier frequency 𝑓𝑐 can be separated using a BPF at the output of the
square law device. The filter should have a lower cut off frequency ranging between 2W
and (𝑓𝑐 − 𝑊) and upper cut-off frequency between (𝑓𝑐 + 𝑊) and 2𝑓𝑐 .
Therefore the filter output is
𝑠(𝑡) = 𝑎1 𝐴𝑐 cos(2𝜋𝑓𝑐𝑡) + 2𝑎2 𝐴𝑐𝑚(𝑡) cos(2𝜋𝑓𝑐𝑡)
𝑎2
𝑠(𝑡) = 𝑎1 𝐴𝑐[1 + 2 𝑚(𝑡)] cos(2𝜋𝑓𝑐𝑡)
𝑎1
If 𝑚(𝑡) = 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡), we get
𝑎2
𝑠 (𝑡) = 𝑎1 𝐴𝑐[1 + 2 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡)] cos(2𝜋𝑓𝑐𝑡)
𝑎1
Comparing this equation with the standard representation of AM Signal,

𝑠(𝑡) = 𝐴𝑐[1 + 𝑘𝑎 𝑚(𝑡)] cos(2𝜋𝑓𝑐 𝑡)

Therefore modulation index of the output signal is given by

𝑎2
𝑚=2 𝐴𝑚
𝑎1

The output signal is free from distortion and attenuation only when (𝑓𝑐 − 𝑊)˃2𝑊⁡𝑜𝑟⁡𝑓𝑐 >
3𝑊

Spectrum is as shown below

Switching Modulator
Consider a semiconductor diode used as an ideal switch to which the carrier
signal
𝑐 (𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡)
and information signal 𝑚(𝑡) are applied simultaneously as shown in the Fig.6

Fig.6. Switching Modulator

The total input for the diode at any instant is given by
𝑣1 = 𝑐(𝑡) + 𝑚(𝑡)
𝑣1 = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡 ) + 𝑚(𝑡)

When the peak amplitude of c(t) is maintained more than that of information
signal, the operation is assumed to be dependent on only c(t) irrespective of m(t).
When c(t) is positive,⁡𝑣2 = 𝑣1 since the diode is forward biased. Similarly, when
c(t) is negative, v2=0 since diode is reverse biased. Based upon above operation,
switching response of the diode is periodic rectangular wave with an amplitude unity
and is given by
∞
1 1 (−1)𝑛−1
𝑝 (𝑡 ) = + ∑ cos⁡(2𝜋𝑓𝑐 𝑡(2𝑛 − 1))
2 𝜋 2𝑛 − 1
𝑛=−∞
1 2 2
𝑝 (𝑡 ) = + cos(2𝜋𝑓𝑐 𝑡) − cos(6𝜋𝑓𝑐 𝑡) + ⋯ … … ..
2 𝜋 3𝜋

Therefore the diode response 𝑉𝑜 is a product of switching response⁡𝑝(𝑡) and input 𝑣1 .

𝑣2 = 𝑣1 ∗ 𝑝(𝑡)

1 2 2
𝑉2 = [𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) + 𝑚(𝑡)][ + cos(2𝜋𝑓𝑐𝑡) − cos(6𝜋𝑓𝑐𝑡) + ⋯ … … . ]
2 𝜋 3𝜋

Applying the Fourier Transform we get

 𝐴𝑐 𝐴𝑐 𝑀(𝑓)
𝑉2 (𝑓) = ( ) 𝛿 (𝑓) + [𝛿 (𝑓 − 𝑓𝑐 ) + 𝛿 (𝑓 + 𝑓𝑐 )] +
𝜋 4 2
𝐴𝑐 1
+ [𝛿 (𝑓 − 2𝑓𝑐 ) + 𝛿 (𝑓 + 2𝑓𝑐 )] + [𝑀(𝑓 − 𝑓𝑐 + 𝑀(𝑓 + 𝑓𝑐 ))]
2𝜋 𝜋
𝐴𝑐 𝐴𝑐
− [𝛿 (𝑓 − 4𝑓𝑐 ) + 𝛿 (𝑓 + 4𝑓𝑐 )] − [𝛿 (𝑓 − 2𝑓𝑐 ) + 𝛿 (𝑓 + 2𝑓𝑐 )]
6𝜋 3𝜋
1
− [𝑀(𝑓 − 3𝑓𝑐 + 𝑀(𝑓 + 3𝑓𝑐 ))]
3𝜋

The diode output 𝑣2 consists of a dc component at 𝑓 = 0, information signal

ranging from 0 to w Hz and infinite number of frequency bands centered at
𝑓𝑐 , 2𝑓𝑐 , 3𝑓𝑐 , 4𝑓𝑐 … … … …

The required AM signal centred at fc can be separated using band pass filter. The
lower cut off-frequency for the band pass filter should be between w and fc-w and the
upper cut-off frequency between fc+w and 2fc. The filter output is given by the equation

𝐴𝑐 4 𝑚 (𝑡 )
𝑠 (𝑡 ) = [1 + ]cos⁡2𝜋𝑓𝑐 𝑡
2 𝜋 𝐴𝑐
For a single tome information, let 𝑚 (𝑡) = 𝐴𝑚 cos(2𝜋𝑓𝑚 𝑡 )

𝐴𝑐 4 𝐴𝑚
𝑠 (𝑡 ) = [1 + 𝑐𝑜𝑠2𝜋𝑓𝑚 𝑡]cos⁡2𝜋𝑓𝑐 𝑡
2 𝜋 𝐴𝑐
Therefore modulation index,
4 𝐴𝑚
𝑚=
𝜋 𝐴𝑐

The output AM signal is free from distortions and attenuations only when
𝑓𝑐 − 𝑤 > 𝑤⁡𝑜𝑟⁡𝑓𝑐 > 2𝑤

Detection of AM waves
Demodulation is the process of recovering the information signal (base band)
from the incoming modulated signal at the receiver. There are two methods, they are
Square law Detector and Envelope Detector

Square Law Detector

Consider a non-linear device to which the AM signal s(t) is applied. When the level of
s(t) is very small, output can be considered upto square of the input.
 Fig .7. Demodulation of AM using square law device

Therefore
2
𝑉𝑂 = 𝑎0 + 𝑎1 𝑉𝑖𝑛 + 𝑎2 𝑉𝑖𝑛

If 𝑚(𝑡) is the information signal (0-wHz) and 𝑐(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) is the carrier, input
AM signal to the non-linear device is given by
𝑠(𝑡) = 𝐴𝑐[1 + 𝑘𝑎 𝑚(𝑡)] cos(2𝜋𝑓𝑐 𝑡)

𝑉𝑂 = 𝑎0 + 𝑎1 𝑠(𝑡) + 𝑎2 [𝑠(𝑡)]2

𝑉𝑂 = 𝑎0 + 𝑎1 𝐴𝑐 cos(2𝜋𝑓𝑐𝑡) + 𝑎1 𝐴𝑐𝑘𝑎 𝑚(𝑡) cos(2𝜋𝑓𝑐𝑡)

2
+ 𝑎2 [𝐴𝑐 cos(2𝜋𝑓𝑐𝑡) + 𝐴𝑐𝑘𝑎𝑚(𝑡) cos(2𝜋𝑓𝑐𝑡)]

Applying Fourier Transform on both sides, we get

The device output consists of a dc component at f =0, information signal ranging

from 0-W Hz and its second harmonics and frequency bands centered at fc and 2fc. The
required information can be separated using low pass filter with cut off frequency
ranging between W and fc-w. The filter output is given by
The dc component (first term) can be eliminated using a coupling capacitor or a
transformer. The effect of second harmonics of information signal can be redu ced by
maintaining its level very low. When 𝑚(𝑡) is very low, the filter output is given by
𝑚 ′ (𝑡) = 𝑎2 𝐴2𝑐 𝑘𝑎 𝑚(𝑡)

When the information level is very low, the noise effect increases at the receiver, hence
the system clarity is very low using square law demodulator.

Envelope Detector

It is a simple and highly effective system. This method is used in most of the
commercial AM radio receivers. An envelope detector is as shown below.

Fig.8. Envelope Detector

During the positive half cycles of the input signals, the diode D is forward biased
and the capacitor C charges up rapidly to the peak of the input signal. When the input
signal falls below this value, the diode becomes reverse biased and the capacitor C
discharges through the load resistor RL.

The discharge process continues until the next positive half cycle. When the
input signal becomes greater than the voltage across the capacitor, the diode conducts
again and the process is repeated.

The charge time constant (rf+Rs)C must be short compared with the carrier
period, the capacitor charges rapidly and there by follows the applied voltage up to the
positive peak when the diode is conducting.That is the charging time constant shall
satisfy the condition,
1
(𝑟𝑓 + 𝑅𝑠 )𝐶 ≪ ⁡
𝑓𝑐
 On the other hand, the discharging time-constant RLC must be long enough to
ensure that the capacitor discharges slowly through load resistor R L between positive
peaks of the carrier wave, but not so long that the capacitor voltage will not discharge at
the maximum rate of change of the modulation wave.
That is the discharge time constant shall satisfy the condition,
1 1
≪ ⁡ 𝑅𝐿 𝐶 ≪ ⁡
𝑓𝑐 𝑊
Where ‘W’ is band width of the message
signal. The result is that the capacitor voltage or detector output is nearly the same as the
envelope of AM wave.

Advantages and Disadvantages of AM: Advantages of AM:

 Generation and demodulation of AM wave are easy.

 AM systems are cost effective and easy to build.

Disadvantages:
 AM contains unwanted carrier component, hence it requires more transmission power.
 The transmission bandwidth is equal to twice the message bandwidth.
To overcome these limitations, the conventional AM system is modified at the
cost of increased system complexity. Therefore, three types of modified AM systems
are discussed.
DSBSC (Double Side Band Suppressed Carrier) modulation: In DSBC
modulation, the modulated wave consists of only the upper and lower side bands.
Transmitted power is saved through the suppression of the carrier wave, but the channel
bandwidth requirement is the same as before.
SSBSC (Single Side Band Suppressed Carrier) modulation: The SSBSC
modulated wave consists of only the upper side band or lower side band. SSBSC is
suited for transmission of voice signals. It is an optimum form of modulation in that it
requires the minimum transmission power and minimum channel band width.
Disadvantage is increased cost and complexity.

VSB (Vestigial Side Band) modulation: In VSB, one side band is completely
passed and just a trace or vestige of the other side band is retained. The required channel
bandwidth is therefore in excess of the message bandwidth by an amount equal to the
width of the vestigial side band. This method is suitable for the transmission of wide
band signals.
 DSB-SC MODULATION

DSB-SC Time domain and Frequency domain Description:

DSBSC modulators make use of the multiplying action in which the modulating
signal multiplies the carrier wave. In this system, the carrier component is eliminated
and both upper and lower side bands are transmitted. As the carrier component is
suppressed, the power required for transmission is less than that of AM.

If 𝑚(𝑡) is the message signal and 𝑐 (𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡 ) is the carrier signal,
then DSBSC modulated wave 𝑠(𝑡) is given by
𝑠(𝑡) = 𝑐 (𝑡)𝑚(𝑡)

𝑠(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) 𝑚(𝑡)

Consequently, the modulated signal s(t) under goes a phase reversal , whenever the
message signal m(t) crosses zero as shown below.

Fig.1. (a) DSB-SC waveform (b) DSB-SC Frequency Spectrum

The envelope of a DSBSC modulated signal is therefore different from the

message signal and the Fourier transform of s(t) is given by
𝐴𝑐
𝑆 (𝑓 ) = [𝑀(𝑓 − 𝑓𝑐 ) + 𝑀(𝑓 + 𝑓𝑐 )]
2
 For the case when base band signal 𝑚(𝑡)is limited to the interval −𝑊 < 𝑓 < 𝑊
as shown in the figure below, we find that the spectrum 𝑆 (𝑓) of the DSBSC wave 𝑠(𝑡)
is as illustrated below. Except for a change in scaling factor, the modulation process
simply translates the spectrum of the base band signal by𝑓𝑐 . The transmission bandwidth
required by DSBSC modulation is the same as that for AM

Fig.2. Message and the corresponding DSBSC spectrum

Generation of DSBSC Waves:

Balanced Modulator (Product Modulator)

A balanced modulator consists of two standard amplitude modulators arranged

in a balanced configuration so as to suppress the carrier wave as shown in the
following block diagram. It is assumed that the AM modulators are identical, except for
the sign reversal of the modulating wave applied to the input of one of them. Thus, the
output of the two modulators may be expressed as,
𝑠1 (𝑡) = 𝐴𝑐[1 + 𝑘𝑎𝑚(𝑡)] cos(2𝜋𝑓𝑐𝑡)

𝑠2 (𝑡) = 𝐴𝑐[1 − 𝑘𝑎 𝑚(𝑡)] cos(2𝜋𝑓𝑐 𝑡)

Fig.3. Balanced Modulator

Subtracting 𝑠2 (𝑡 ) from 𝑠1 (𝑡 )
𝑠(𝑡) = 𝑠1 (𝑡) − 𝑠2(𝑡)

𝑠(𝑡) = 2𝑘𝑎 𝑚(𝑡) 𝐴𝑐cos(2𝜋𝑓𝑐𝑡)

Hence, except for the scaling factor 2ka, the balanced modulator output is equal
to the product of the modulating wave and the carrier.

Ring Modulator

Ring modulator is the most widely used product modulator for generating DSBSC wave
and is shown below.

Fig.4. Ring Modulator

The four diodes form a ring in which they all point in the same
direction. The diodes are controlled by square wave carrier c(t) of frequency fc, which
is applied longitudinally by means of two center-tapped transformers. Assuming the
diodes are ideal, when the carrier is positive, the outer diodes D1 and D2 are forward
biased where as the inner diodes D3 and D4 are reverse biased, so that the
modulator multiplies the base band signal m(t) by c(t). When the carrier is negative,
the diodes D1 and D2 are reverse biased and D3 and D4 are forward, and the
modulator multiplies the base band signal –m(t) by c(t).
Thus the ring modulator in its ideal form is a product modulator for square
wave carrier and the base band signal m(t). The square wave carrier can be expanded using
Fourier series as
∞
4 (−1)𝑛−1
𝑐 (𝑡 ) = ∑ cos⁡(2𝜋𝑓𝑐 𝑡(2𝑛 − 1) )
𝜋 2𝑛 − 1
𝑛=1

Therefore the ring modulator output is given by

 𝑠(𝑡) = 𝑐 (𝑡)𝑚(𝑡)
∞
4 (−1)𝑛−1
𝑠 (𝑡 ) = 𝑚(𝑡 )[ ∑ cos(2𝜋𝑓𝑐 𝑡(2𝑛 − 1))]
𝜋 2𝑛 − 1
𝑛=1

From the above equation it is clear that output from the modulator consists
entirely of modulation products. If the message signal m(t) is band limited to the
frequency band − w < f < w, the output spectrum consists of side bands centred at fc.

Detection of DSB-SC waves:

Coherent Detection:

The message signal m(t) can be uniquely recovered from a DSBSC wave s(t)
by first multiplying s(t) with a locally generated sinusoidal wave and then low pass
filtering the product as shown.

Fig.5 Coherent Detector

It is assumed that the local oscillator signal is exactly coherent or synchronized,

in both frequency and phase, with the carrier wave c(t) used in the product modulator to
generate s(t). This method of demodulation is known as coherent detection or
synchronous detection.
Let 𝐴′𝑐 cos⁡(2𝜋𝑓𝑐 𝑡 + 𝜙) be the local oscillator signal, and𝑠(𝑡) =
𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) 𝑚(𝑡) be the DSBSC wave. Then the product modulator output 𝑣(𝑡) is
given by
𝑣 (𝑡 ) = 𝐴𝑐 𝐴′𝑐 cos(2𝜋𝑓𝑐 𝑡) cos(2𝜋𝑓𝑐 𝑡 + 𝜙 )𝑚(𝑡)
𝐴𝑐 𝐴′𝑐 𝐴𝑐 𝐴′𝑐
𝑣 (𝑡 ) = cos(4𝜋𝑓𝑐 𝑡 + ∅) 𝑚 (𝑡) + cos(∅) 𝑚(𝑡)
4 2
The first term in the above expression represents a DSBSC modulated signal
with a carrier frequency2𝑓𝑐 , and the second term represents the scaled version of
message signal. Assuming that the message signal is band limited to the interval– 𝑤 <
𝑓 < 𝑤, the spectrum of 𝑣 (𝑡) is plotted as shown below.
 Fig.6.Spectrum of output of the product modulator

From the spectrum, it is clear that the unwanted component (first term in the
expression) can be removed by the low-pass filter, provided that the cut-off frequency
of the filter is greater than W but less than 2fc-W. The filter output is given by
𝐴𝑐 𝐴′𝑐
𝑣𝑜 (𝑡) = cos(∅) 𝑚(𝑡)
2
The demodulated signal vo(t) is therefore proportional to m(t) when the phase error ϕ
is constant.
Costas Receiver (Costas Loop):
Costas receiver is a synchronous receiver system, suitable for demodulating
DSBSC waves. It consists of two coherent detectors supplied with the same input
signal, that is the incoming DSBSC wave 𝑠(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) 𝑚(𝑡) but with
individual local oscillator signals that are in phase quadrature with respect to each other
as shown in Fig.7

Fig.7. Costas Receiver

 The frequency of the local oscillator is adjusted to be the same as the carrier
frequency fc. 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.

These 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. Suppose

the local oscillator signal is of the same phase as the

carrier c(t) = Accos(2πfct) wave used to generate the incoming DSBSC wave. Then we
find that the I-channel output contains the desired demodulated signal m(t), where as the
Q-channel output is zero due to quadrature null effect of the Q-channel.
Suppose that the local oscillator phase drifts from its proper value by a small angle ϕ
radians. The I-channel output will remain essentially unchanged, but there will
be some signal appearing at the Q-channel output, which is
proportional to sin(𝜙)⁡≈⁡𝜙 for small ϕ.

This Q-channel output will have same polarity as the I-channel output for one
direction of local oscillator phase drift and opposite polarity for the opposite direction of
local oscillator phase drift. Thus by combining the I-channel and Q-channel outputs in a
phase discriminator (which consists of a multiplier followed by a LPF), a dc control
signal is obtained that automatically corrects for the local phase errors in the VCO.

Radio Transmitters

There are two approaches in generating an AM signal. These are known as low
and high level modulation. They're easy to identify: A low level AM transmitter
performs the process of modulation near the beginning of the transmitter. A high level
transmitter performs the modulation step last, at the last or "final" amplifier stage in the
transmitter. Each method has advantages and disadvantages, and both are in common
use.
Low-Level AM Transmitter:

Fig.8. Low-Level AM Transmitter Block Diagram

There are two signal paths in the transmitter, audio frequency (AF) and radio
frequency (RF). The RF signal is created in the RF carrier oscillator. At test point A the
oscillator's output signal is present. The output of the carrier oscillator is a fairly small
AC voltage, perhaps 200 to 400 mV RMS. The oscillator is a critical stage in any
transmitter. It must produce an accurate and steady frequency. Every radio station is
assigned a different carrier frequency. The dial (or display) of a receiver displays the
carrier frequency. If the oscillator drifts off frequency, the receiver will be unable to
receive the transmitted signal without being readjusted. Worse yet, if the oscillator drifts
onto the frequency being used by another radio station, interference will occur. Two
circuit techniques are commonly used to stabilize the oscillator, buffering and voltage
regulation.

The buffer amplifier has something to do with buffering or protecting the

oscillator. An oscillator is a little like an engine (with the speed of the engine being
similar to the oscillator's frequency). If the load on the engine is increased (the engine is
asked to do more work), the engine will respond by slowing down. An oscillator acts in
a very similar fashion. If the current drawn from the oscillator's output is increased or
decreased, the oscillator may speed up or slow down slightly.

Buffer amplifier is a relatively low-gain amplifier that follows the oscillator. It

has a constant input impedance (resistance). Therefore, it always draws the same
amount of current from the oscillator. This helps to prevent "pulling" of the oscillator
frequency. The buffer amplifier is needed because of what's happening "downstream" of
the oscillator. Right after this stage is the modulator. Because the modulator is a
nonlinear amplifier, it may not have a constant input resistance -- especially when
information is passing into it. But since there is a buffer amplifier between the oscillator
and modulator, the oscillator sees a steady load resistance, regardless of what the
modulator stage is doing.

Voltage Regulation: An oscillator can also be pulled off frequency if its power
supply voltage isn't held constant. In most transmitters, the supply voltage to the
oscillator is regulated at a constant value. The regulated voltage value is often between 5
and 9 volts; zener diodes and three-terminal regulator ICs are commonly used voltage
regulators. Voltage regulation is especially important when a transmitter is being
powered by batteries or an automobile's electrical system. As a battery discharges, its
terminal voltage falls. The DC supply voltage in a car can be anywhere between 12 and
16 volts, depending on engine RPM and other electrical load conditions within the
vehicle.

Modulator: The stabilized RF carrier signal feeds one input of the modulator
stage. The modulator is a variable-gain (nonlinear) amplifier. To work, it must have an
RF carrier signal and an AF information signal. In a low-level transmitter, the power
levels are low in the oscillator, buffer, and modulator stages; typically, the modulator
output is around 10 mW (700 mV RMS into 50 ohms) or less.

AF Voltage Amplifier: In order for the modulator to function, it needs an

information signal. A microphone is one way of developing the intelligence signal,
however, it only produces a few millivolts of signal. This simply isn't enough to operate
the modulator, so a voltage amplifier is used to boost the microphone's signal.

The signal level at the output of the AF voltage amplifier is usually at least 1
volt RMS; it is highly dependent upon the transmitter's design. Notice that the AF
amplifier in the transmitter is only providing a voltage gain, and not necessarily a
current gain for the microphone's signal. The power levels are quite small at the output
of this amplifier; a few mW at best.

RF Power Amplifier: At test point D the modulator has created an AM signal

by impressing the information signal from test point C onto the stabilized carrier signal
from test point B at the buffer amplifier output. This signal (test point D) is a complete
AM signal, but has only a few milliwatts of power. The RF power amplifier is normally
built with several stages. These stages increase both the voltage and current of the AM
signal. We say that power amplification occurs when a circuit provides a current gain. In
order to accurately amplify the tiny AM signal from the modulator, the RF power
amplifier stages must be linear. You might recall that amplifiers are divided up into
"classes," according to the conduction angle of the active device within. Class A and
class B amplifiers are considered to be linear amplifiers, so the RF power amplifier
stages will normally be constructed using one or both of these type of amplifiers.
Therefore, the signal at test point E looks just like that of test point D; it's just much
bigger in voltage and current.

Antenna Coupler: The antenna coupler is usually part of the last or final RF
power amplifier, and as such, is not really a separate active stage. It performs no
amplification, and has no active devices. It performs two important jobs: Impedance
matching and filtering. For an RF power amplifier to function correctly, it must be
supplied with a load resistance equal to that for which it was designed.

The antenna coupler also acts as a low-pass filter. This filtering reduces the
amplitude of harmonic energies that may be present in the power amplifier's output. (All
amplifiers generate harmonic distortion, even "linear" ones.) For example, the
transmitter may be tuned to operate on 1000 kHz. Because of small nonlinearities in the
amplifiers of the transmitter, the transmitter will also produce harmonic energies on
2000 kHz (2nd harmonic), 3000 kHz (3rd harmonic), and so on. Because a low-pass
filter passes the fundamental frequency (1000 kHz) and rejects the harmonics, we say
that harmonic attenuation has taken place.

High-Level AM Transmitter:

Fig.9. Low-Level AM Transmitter Block Diagram

The high-level transmitter of Figure 9 is very similar to the low-level unit. The
RF section begins just like the low-level transmitter; there is an oscillator and buffer
amplifier. The difference in the high level transmitter is where the modulation takes
 place. Instead of adding modulation immediately after buffering, this type of transmitter
amplifies the unmodulated RF carrier signal first. Thus, the signals at points A, B, and D
in Figure 9 all look like unmodulated RF carrier waves. The only difference is that they
become bigger in voltage and current as they approach test point D.

The modulation process in a high-level transmitter takes place in the last or final
power amplifier. Because of this, an additional audio amplifier section is needed. In
order to modulate an amplifier that is running at power levels of several watts (or more),
comparable power levels of information are required. Thus, an audio power amplifier is
required. The final power amplifier does double-duty in a high-level transmitter. First, it
provides power gain for the RF carrier signal, just like the RF power amplifier did in the
low-level transmitter. In addition to providing power gain, the final PA also performs
the task of modulation. The final power amplifier in a high-level transmitter usually
operates in class C, which is a highly nonlinear amplifier class.

Comparison:
Low Level Transmitters
 Can produce any kind of modulation; AM, FM, or PM.

 Require linear RF power amplifiers, which reduce DC efficiency and increases

production costs.

High Level Transmitters

 Have better DC efficiency than low-level transmitters, and are very well suited for
battery operation.
 Are restricted to generating AM modulation only.
 UNIT-II
Introduction to SSB-SC

Standard AM and DSBSC require transmission bandwidth equal to twice the

message bandwidth. In both the cases spectrum contains two side bands of width
W Hz, each. But the upper and lower sides are uniquely related to each other by the
virtue of their symmetry about the carrier frequency. That is, given the amplitude and
phase spectra of either side band, the other can be uniquely determined. Thus if only one
side band is transmitted, and if both the carrier and the other side band are suppressed at
the transmitter, no information is lost. This kind of modulation is called SSBSC and
spectral comparison between DSBSC and SSBSC is shown in the figures 1 and 2.

Frequency Domain Description

Consider a message signal 𝑚(𝑡) with a spectrum 𝑀(𝑓) band limited to the
interval– 𝑤 < 𝑓 < 𝑤⁡as shown in Fig.3, the DSBSC wave obtained by multiplexing
𝑚(𝑡) by the carrier wave 𝑐 (𝑡) = 𝐴𝑐 cos⁡(2𝜋𝑓𝑐 𝑡) and is also shown in figure 4. The upper
side band is represented in duplicate by the frequencies above 𝑓𝑐 and those below -𝑓𝑐 ,
and when only upper side band is transmitted; the resulting SSB modulated wave has the
spectrum shown in figure 6. .
 Similarly, the lower side band is represented in duplicate by the
frequencies below fc and those above -fc and when only the lower side band is
transmitted, the spectrum of the corresponding SSB modulated wave shown in
figure 5.Thus the essential function of the SSB modulation is to translate the spectrum
of the modulating wave, either with or without inversion, to a new location in the
frequency domain.
The advantage of SSB modulation is reduced bandwidth and the elimination
of high power carrier wave. The main disadvantage is the cost and complexity of its
implementation.
Generation of SSB wave:
Frequency discrimination method
Consider the generation of SSB modulated signal containing the upper side
band only. From a practical point of view, the most severe requirement of SSB
generation arises from the unwanted sideband, the nearest component of which is
separated from the desired side band by twice the lowest frequency component of the
message signal. It implies that, for the generation of an SSB wave to be possible,
the message spectrum must have an energy gap centered at the origin as shown in
figure 7. This requirement is naturally satisfied by voice signals, whose energy gap
is about 600Hz wide.

The frequency discrimination or filter method of SSB generation consists of a

product modulator, which produces DSBSC signal and a band-pass filter to extract the
desired side band and reject the other and is shown in the figure 8.

Application of this method requires that the message signal satisfies two conditions:

1. The message signal m(t) has no low-frequency content. Example: speech, audio,
music.
2. The highest frequency component W of the message signal m(t) is much less
than the carrier frequency fc.
 Then, under these conditions, the desired side band will appear in a non-
overlapping interval in the spectrum in such a way that it may be selected by an
appropriate filter.

In designing the band pass filter, the following requirements should be satisfied:

1. The pass band of the filter occupies the same frequency range as the
spectrum of the desired SSB modulated wave.

2. The width of the guard band of the filter, separating the pass band from
the stop band, where the unwanted sideband of the filter input lies, is
twice the lowest frequency component of the message signal.

When it is necessary to generate an SSB modulated wave occupying a frequency

band that is much higher than that of the message signal, it becomes very difficult to
design an appropriate filter that will pass the desired side band and reject the other. In
such a situation it is necessary to resort to a multiple-modulation process so as to ease
the filtering requirement. This approach is illustrated in the following figure 9 involving
two stages of modulation.

The SSB modulated wave at the first filter output is used as the modulating wave
for the second product modulator, which produces a DSBSC modulated wave with a
spectrum that is symmetrically spaced about the second carrier frequency f2. The
frequency separation between the side bands of this DSBSC modulated wave is
effectively twice the first carrier frequency f1, thereby permitting the second filter to
remove the unwanted side band.

Hilbert Transform & its Properties:

The Fourier transform is useful for evaluating the frequency content of an
energy signal, or in a limiting case that of a power signal. It provides mathematical basis
for analyzing and designing the frequency selective filters for the separation of signals
on the basis of their frequency content. Another method of separating the signals is
based on phase selectivity, which uses phase shifts between the appropriate signals
(components) to achieve the desired separation. In case of a sinusoidal signal, the
simplest phase shift of 180o is obtained by “Ideal transformer” (polarity reversal). When
the phase angles of all the components of a given signal are shifted by 90 o, the resulting
function of time is called the “Hilbert transform” of the signal.
Consider an LTI system with transfer function defined by equation 1

−𝑗, 𝑓 > 0
𝐻 (𝑓) = { 0, 𝑓 = 0 … … … … … … … … . (1)
𝑗, 𝑓 < 0

and the Signum function given by

1, 𝑓 > 0
𝑠𝑔𝑛 (𝑓) = { 0, 𝑓 = 0
−1, 𝑓 < 0

The function 𝐻(𝑓) can be expressed using Signum function as given by equ (2)
𝐻 (𝑓) = −𝑗𝑠𝑔𝑛 (𝑓) … … … … … … … … . (2)

We know that
−𝑗𝜋⁄ 𝑗𝜋⁄
⁡𝑒 2 = −𝑗, ⁡𝑒 2 = 𝑗⁡𝑎𝑛𝑑⁡𝑒 ±𝑗𝜃 = cos⁡(𝜃) ± 𝑗𝑠𝑖𝑛(𝜃)
Therefore,
−𝑗𝜋
⁡𝑒 ⁄2 , 𝑓 > 0
𝐻(𝑓) = { 𝑗𝜋
⁡𝑒 ⁄2 , 𝑓 < 0

Thus the magnitude |𝐻(𝑓)| = 1, for all 𝑓, and angle

−𝜋⁄ , 𝑓 > 0
⦟𝐻 𝑓 = { 𝜋 2
( )
⁄2 , 𝑓 < 0

The device which possesses such a property is called Hilbert transformer.

Whenever a signal is applied to the Hilbert transformer, the amplitudes of all frequency
components of the input signal remain unaffected. It produces a phase shift of -90o for
all positive frequencies, while a phase shifts of 90 o for all negative frequencies of the
signal.
 If x(t) is an input signal, then its Hilbert transformer is denoted by xˆ(t ) and
shown in the following diagram.

To find impulse response ℎ(𝑡) of Hilbert transformer with transfer

function⁡𝐻(𝑓). Consider the relation between Signum function and the unit step
function.
𝑠𝑔𝑛 (𝑡) = 2𝑢 (𝑡) − 1 = 𝑥(𝑡)

Differentiating both sides with respect to t,

𝑑
{𝑥(𝑡)} = 2𝛿(𝑡)
𝑑𝑡
Applying Fourier transform on both sides,
2 1
𝑠𝑔𝑛 (𝑡) ↔ ⁡ → 𝑠𝑔𝑛(𝑡) ↔
𝑗𝜔 𝑗𝜋𝑓
Applying duality property of Fourier transform,
1
−𝑆𝑔𝑛(𝑓) ↔
𝑗𝜋𝑡
We have,
𝐻 (𝑓) = −𝑗𝑠𝑔𝑛 (𝑓)

1
𝐻 (𝑓 ) ↔
𝜋𝑡

Therefore the impulse response ℎ(𝑡) of an Hilbert transformer is given by the equation
(3),
1
ℎ (𝑡 ) = … … … … (3)
𝜋𝑡
Now consider any input x(t) to the Hilbert transformer, which is an LTI system.
Let the impulse response of the Hilbert transformer is obtained by convolving the input
x(t) and impulse response h(t) of the system.
𝑥̂(𝑡) = 𝑥(𝑡) ∗ ℎ(𝑡)
1
𝑥̂(𝑡) = 𝑥(𝑡) ∗
𝜋𝑡
 1 +∞ 𝑥(𝜏)
𝑥̂(𝑡) = ∫ 𝑑𝜏 … … … … … … . (4)
𝜋 −∞ (𝑡 − 𝜏)
This equation gives the Hilbert transform of 𝑥(𝑡).
The inverse Hilbert transform 𝑥(𝑡) is given by
−1 +∞ 𝑥̂(𝜏)
𝑥 (𝑡 ) = ∫ 𝑑𝜏 … … … … … … . (5)
𝜋 −∞ (𝑡 − 𝜏)

We have 𝑥̂(𝑡) = 𝑥(𝑡) ∗ ℎ(𝑡)

The Fourier transform 𝑋̂(𝑓) of 𝑥̂(𝑡) is given by
𝑋̂(𝑓) = 𝑋(𝑓)𝐻(𝑓)
𝑋̂(𝑓) = −𝑗𝑠𝑔𝑛 (𝑓)𝑋(𝑓) … … … … … . (6)

Properties:
1. A signal 𝑥 (𝑡) and its Hilbert transform 𝑥̂(𝑡) have the same amplitude spectrum. The
magnitude of −𝑗𝑠𝑔𝑛(𝑓) is equal to 1 for all frequencies f. Therefore x(t) and 𝑥̂(𝑡) have the
same amplitude spectrum. That is |𝑋̂(𝑓)| = |𝑋(𝑓)|⁡𝑓𝑜𝑟⁡𝑎𝑙𝑙⁡𝑓
2. If 𝑥̂(𝑡) is the Hilbert transform of x(t), then the Hilbert transform of 𝑥̂(𝑡 )𝑖𝑠 − 𝑥(𝑡). To
obtain its Hilbert transform of x(t), x(t) is passed through LTI system with a transfer
function equal to −𝑗𝑠𝑔𝑛(𝑓). A double Hilbert transformation is equivalent to passing 𝑥 (𝑡)
through a cascade of two such devices. The overall transfer function of such a cascade is equal
to
[−𝑗𝑠𝑔𝑛 (𝑓)]2 = −1⁡𝑓𝑜𝑟⁡𝑎𝑙𝑙⁡𝑓
The resulting output is −𝑥 (𝑡). That is the Hilbert transform of 𝑥̂(𝑡)is equal to
−𝑥(𝑡).

Time Domain Description:

The time domain description of an SSB wave s(t) in the canonical form is given
by the equation 1.
𝑠(𝑡) = 𝑠𝐼 (𝑡) cos(2𝜋𝑓𝑐 𝑡) − 𝑠𝑄 (𝑡)𝑠𝑖𝑛 (2𝜋𝑓𝑐 𝑡) … … … … … … … … (1)
where 𝑆𝐼 (𝑡) is the in-phase component of the SSB wave and 𝑆𝑄 (𝑡) is the
quadrature component. The in-phase component 𝑆𝐼 (𝑡)except for scaling factor, may be
derived form 𝑆 (𝑡) by first multiplying 𝑆 (𝑡) by cos⁡(2𝜋𝑓𝑐 𝑡) and then passing the product
through a low pass filter.Similarly, the quadrature component 𝑆𝑄 (𝑡), except the scaling
factor, may be derived from 𝑆 (𝑡) by first multiplying 𝑆(𝑡) by sin⁡(2𝜋𝑓𝑐 𝑡) and then
passing the product through an identical filter.
The Fourier transformation of 𝑆𝐼 (𝑡) and 𝑆𝑄 (𝑡 ) are related to that of SSB wave as
follows, respectively
𝑆(𝑓 − 𝑓𝑐 ) + 𝑆(𝑓 + 𝑓𝑐 ), −𝑤 ≤ 𝑓 ≤ 𝑤
𝑆𝐼 (𝑓) = { … … … … … … … . (2)
0, 𝑒𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒

𝑗[𝑆(𝑓 − 𝑓𝑐 ) − 𝑆 (𝑓 + 𝑓𝑐 )], −𝑤 ≤ 𝑓 ≤ 𝑤
𝑆𝑄 (𝑓) = { … … … … … … … . (3)
0, 𝑒𝑙𝑠𝑒𝑤ℎ𝑒𝑟𝑒
Where −𝑤 ≤ 𝑓 ≤ 𝑤 defines the frequency band occupied by the message signal 𝑚(𝑡)
Consider the SSB wave that is obtained by transmitting only the upper side band
as shown in figure 10. Two frequency shifted spectra 𝑆(𝑓 − 𝑓𝑐 )⁡𝑎𝑛𝑑⁡𝑆 (𝑓 + 𝑓𝑐 ) are
shown in figure 11 and figure 12 respectively. Therefore, from equations 2 & 3, is
follows that the corresponding spectra of the in-phase component 𝑆𝐼 (𝑡) and the
quadrature component 𝑆𝑄 (𝑡) are as shown in figure 13 & 14 respectively.
 From the fig.13, it is found that
1
𝑆𝐼 (𝑓) = 𝐴 𝑀(𝑓)
2 𝑐
Where 𝑀(𝑓) is the Fourier transform of the message signal 𝑚(𝑡). Accordingly in-phase
component 𝑆𝐼 (𝑡 )⁡is defined by the equation 4.
1
𝑆𝐼 (𝑡) = 𝐴 𝑚 (𝑡 ) … … … … … … … … . . (4)
2 𝑐
Now on the basis of fig.14, it is found that
−𝑗
𝐴 𝑀 (𝑓 ), 𝑓 > 0
2 𝑐
𝑆𝑄 (𝑓) = 0, 𝑓 = 0
𝑗
( )
{ 2 𝐴𝑐 𝑀 𝑓 , 𝑓 < 0
−𝑗
𝑆𝑄 (𝑓) = 𝐴 𝑠𝑔𝑛 (𝑓)𝑀(𝑓) … … … … … … (5)
2 𝑐

Where 𝑠𝑔𝑛 (𝑓) is the Signum Function

But from the discussions on Hilbert transforms, it is shown that
̂ (𝑓) … … … … … … … . (6)
−𝑗𝑠𝑔𝑛 (𝑓)𝑀 (𝑓) = 𝑀
̂ (𝑓) is the Fourier transform of the Hilbert transform of 𝑚(𝑡).
Where 𝑀
Hence, substituting equ.(6) in equ.(5), we get
1
𝑆𝑄 (𝑓) = 𝐴 𝑀̂ (𝑓) … … … … … … … . . (7)
2 𝑐
Therefore the quadrature component 𝑆𝑄 (𝑡) is defined by equation (8)
1
𝑆𝑄 (𝑡) = ̂ (𝑡) … … … … … . . (8)
𝐴 𝑚
2 𝑐
Therefore substituting equ.(4) and (8) in equ.(1), we find that the canonical
representation of an SSB wave s(t) obtained by transmitting only the USB is given by
equ (9)

1 1
𝑆𝑈 (𝑡) = 𝐴𝑐 𝑚 (𝑡) cos(2𝜋𝑓𝑐 𝑡) − 𝐴𝑐 𝑚
̂ (𝑡)⁡𝑠𝑖𝑛 (2𝜋𝑓𝑐 𝑡) … … … … … … … … … (9)
2 2

Following the same procedure, we can find the canonical representation for an SSB
Wave s(t) obtained by transmitting only the lower side band is given by
1 1
𝑆𝐿 (𝑡) = 𝐴𝑐𝑚(𝑡) cos(2𝜋𝑓𝑐 𝑡) + 𝐴𝑐𝑚
̂ (𝑡)⁡𝑠𝑖𝑛 (2𝜋𝑓𝑐 𝑡) … … … … … … … … … (10)
2 2

Phase discrimination method for generating SSB wave:

Time domain description of SSB modulation leads to another method of SSB

generation using the equations 9 or 10. The block diagram of phase discriminator is
as shown in figure 15.

The phase discriminator consists of two product modulators I and Q,

supplied with carrier waves in-phase quadrature to each other. 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.

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.
 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.

Demodulation of SSB Waves:

Demodulation of SSBSC wave using coherent detection is as shown in fig. 16. The SSB
wave 𝑠(𝑡) together with a locally generated carrier 𝑐 (𝑡) = 𝐴′𝑐 cos(2𝜋𝑓𝑐 𝑡 + ∅)⁡is applied
to a product modulator and then low-pass filtering of the modulator output yields the
message signal.

Fig.16. Coherent Detector of SSBSC

The product modulator output 𝑣0 (𝑡) is given by

𝑣 (𝑡) = 𝐴′𝑐 cos(2𝜋𝑓𝑐 𝑡 + ∅) 𝑠(𝑡), 𝑃𝑢𝑡⁡∅ = 0

1
𝑣 (𝑡 ) = 𝐴 cos(2𝜋𝑓𝑐 𝑡) [𝑚 (𝑡) cos(2𝜋𝑓𝑐 𝑡) ± 𝑚
̂ (𝑡)sin⁡(2𝜋𝑓𝑐 𝑡)]
2 𝑐

1 1
𝑣 (𝑡 ) = 𝐴𝑐 𝑚 (𝑡) + 𝐴𝑐 [𝑚(𝑡) cos(4𝜋𝑓𝑐 𝑡) ± 𝑚
̂ (𝑡) sin(4𝜋𝑓𝑐 𝑡 )] … … … … … . . (1)
4 4

The first term in the above equ.(1) is desired message signal. The other term
represents an SSB wave with a carrier frequency of 2𝑓𝑐 as such; it is an unwanted
component, which is removed by low-pass filter.
 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.

Frequency Domain Description

The following Fig illustrates the spectrum of VSB modulated wave s (t) with respect to
the message m (t) (band limited)

Fig.1. Spectrum of message signal

Fig.2. Spectrum of VSB wave containing the vestige of the LSB

Assume that the Lower side band is modified into the vestigial side band. The
vestige of the lower sideband compensates for the amount removed from the
upper sideband. The bandwidth required to send VSB wave is
𝐵 = 𝑤 + 𝑓𝑣
where 𝑓𝑣 is the width of the vestigial side band.
Similarly, if USB is modified into the VSB then,

Fig.3. Spectrum of VSB wave containing the vestige of the 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

VSB modulated wave is obtained by passing DSBSC through a sideband shaping filter
as shown in below fig.

Fig.4. Block Diagram of VSB Modulator

The exact design of this filter depends on the spectrum of the VSB waves. The
relation between filter transfer function H (f) and the spectrum of VSB waves is given
by
𝐴𝑐
𝑆 (𝑓 ) = [𝑀(𝑓 − 𝑓𝑐 ) + 𝑀 (𝑓 + 𝑓𝑐 )]𝐻(𝑓) … … … … … … … (1)
2
 Where M(f) is the spectrum of Message Signal. Now, we have to determine the
specification for the filter transfer function H(f) It can be obtained by passing s(t) to a
coherent detector and determining the necessary condition for undistorted version of the
message signal m(t). Thus, s (t) is multiplied by a locally generated sinusoidal wave cos
(2πfct) which is synchronous with the carrier wave Accos(2πfct) in both frequency and
phase, as in the below fig.,

Fig.5 Block Diagram of VSB Demodulator

Then,
𝑣 (𝑡) = 𝑠(𝑡). cos(2𝜋𝑓𝑐 𝑡) … … … … … … … … … (2)

In frequency domain equ (2) becomes,

1
𝑉 (𝑓 ) = [𝑆(𝑓 − 𝑓𝑐 ) + 𝑆(𝑓 + 𝑓𝑐 ) … … … … … . . (3)
2
Substituting equ.1 in equ. 3, we get

1 𝐴𝑐
𝑉 (𝑓 ) = [ ⁄2 [𝑀(𝑓 − 2𝑓𝑐 ) + 𝑀(𝑓)]𝐻(𝑓 − 𝑓𝑐 )]
2
1 𝐴
+ [ 𝑐⁄2 [𝑀(𝑓 + 2𝑓𝑐 ) + 𝑀(𝑓)]𝐻 (𝑓 + 𝑓𝑐 )]
2

𝐴𝑐
𝑉 (𝑓 ) = 𝑀(𝑓)[𝐻(𝑓 − 𝑓𝑐 ) + 𝐻(𝑓 + 𝑓𝑐 )]
4
𝐴𝑐
+ [𝑀(𝑓 − 2𝑓𝑐 )𝐻 (𝑓 − 𝑓𝑐 )
4
+ 𝑀 (𝑓 + 2𝑓𝑐 )𝐻(𝑓 + 𝑓𝑐 )] … … … … … … . . (4)

The spectrum of 𝑉 (𝑓) as shown in fig below

 Fig.6. Spectrum of the product modulator output⁡𝑣(𝑡)
Pass 𝑣(𝑡) to a LPF to eliminate VSB wave corresponding to 2𝑓𝑐
𝐴𝑐
𝑉𝑜 (𝑓) = 𝑀(𝑓)[𝐻(𝑓 − 𝑓𝑐 ) + 𝐻 (𝑓 + 𝑓𝑐 )] … … … … … . (5)
4
The spectrum of 𝑉𝑜 (𝑓) is in fig below,

Fig.7. Spectrum of demodulated signal 𝑣𝑜 (𝑡 )

For a distortion less reproduction of the original signal⁡𝑚(𝑡), 𝑉𝑜 (𝑓) is to be a

scaled version of⁡𝑀(𝑓). Therefore, the transfer function 𝐻(𝑓) must satisfy the condition
𝐻(𝑓 − 𝑓𝑐 ) + 𝐻 (𝑓 + 𝑓𝑐 ) = 2𝐻(𝑓𝑐 ) … … … … … … … (6)
Where 𝐻(𝑓𝑐 ) is a constant
Since m(t ) is a band limited signal, we need to satisfy eqn. (6) in the interval
 w  f  w. The requirement of eqn. (6) is satisfied by using a filter whose transfer
function is shown below.

Fig.8. Frequency response of sideband shaping filter

Note: H ( f ) is shown for positive frequencies only
The response is normalized so that H ( f ) at 𝑓𝑐 is 0.5. Inside this interval 𝑓𝑐 − 𝑓𝑣 ≤
𝑓 ≤ 𝑓𝑐 + 𝑓𝑣 response exhibits odd symmetry .i.e., sum of the values of H ( f ) at any two
frequencies equally displaced above & below is Unity
Similarly, the transfer function H ( f ) of the filter for sending Lower sideband
along with the vestige of the Upper sideband is shown in fig below,

Fig.9. Frequency response of sideband shaping filter

Note: H ( f ) is shown for positive frequencies only

Time Domain Description:

Time domain representation of VSB modulated wave, procedure is similar to
SSB Modulated waves. Let s(t) denote a VSB modulated wave and assuming that s(t)
containing Upper sideband along with the Vestige of the Lower sideband. VSB
modulated wave s(t) is the output from Sideband shaping filter, whose input is DSBSC
wave. The filter transfer function H(f) is of the form as in fig below,

Fig.10. H ( f ) of sideband shaping filter

The DSBSC modulated wave is

𝑆𝐷𝑆𝐵𝑆𝐶 (𝑡) = 𝐴𝑐 𝑚 (𝑡) cos(2𝜋𝑓𝑐 𝑡) … … … … … . (1)
It is a band pass signal and has in-phase component only. Its low pass complex envelope
is given by
𝑆̃𝐷𝑆𝐵𝑆𝐶 (𝑡) = 𝐴𝑐 𝑚 (𝑡) … … … … … . (2)
The VSB modulated wave is a band pass signal.
Let the low pass signal ~
s (t ) denote the complex envelope of VSB wave s (t ), then
𝑠 (𝑡) = 𝑅𝑒[𝑠̃ (𝑡 ) exp(𝑗2𝜋𝑓𝑐 𝑡)] … … … … … … … … . . (3)
 To determine ~
s (t ) we proceed as follows

1. The side band shaping filter transfer function H ( f ) is replaced by its equivalent
̃(𝑓) as shown in fig below
complex low pass transfer function denoted by 𝐻

Fig.11. Low pass equivalent to H ( f )

̃(𝑓) as the difference between two components 𝐻
We may express 𝐻 ̃𝑢 (𝑓) and 𝐻
̃𝑣 (𝑓) as
̃(𝑓 ) = 𝐻
𝐻 ̃𝑢 (𝑓) − 𝐻
̃𝑣 (𝑓) … … … … … … … . (4)
These two components are considered individually as follows
i. ̃𝑢 (𝑓) pertains to a complex low pass filter equivalent to a
The transfer function 𝐻
ban pass filter design to reject the lower side band completely as

̃(𝑓)
Fig.12. First component of 𝐻

1
̃𝑢 (𝑓) = { ⁄2 [1 + 𝑠𝑔𝑛 (𝑓)],⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡0 < 𝑓 < 𝑤 … … … … … … … … (5)
𝐻
0,⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒
ii. ̃𝑣 (𝑓) accounts for the generation of vestige and removal
The transfer function⁡⁡𝐻
of a corresponding portion from the upper side band.

̃(𝑓)
Fig.13. Second component of 𝐻
 Substitute eqn.5 in eqn. 4, we get,

1 ̃
̃ (𝑓) = { ⁄2 [1 + 𝑠𝑔𝑛 (𝑓) − 2𝐻𝑣 (𝑓)],⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡𝑓𝑣 < 𝑓 < 𝑤 … … … … … … … … (6)
𝐻
0,⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒

̃𝑣 (𝑓) are both odd functions of frequency. Hence both have purely
The sgn(f) and 𝐻
imaginaryInverse Fourier transform (FT). Accordingly, the new transfer function is

𝐻𝑄 (𝑓) = 1⁄ ̃𝑣 (𝑓)] … … … … … … . . (7)

𝑗[𝑠𝑔𝑛 (𝑓) − 2𝐻

It has purely Inverse FT and ℎ𝑄 (𝑡)⁡𝑑𝑒𝑛𝑜𝑡𝑒⁡𝐼𝐹𝑇⁡𝑜𝑓⁡𝐻𝑄 (𝑓)

Fig.14. Transfer function of the filter 𝑗𝐻𝑄 (𝑓)

Rewrite eqn. (6) in terms of ⁡𝐻𝑄 (𝑓) as

1
̃(𝑓) = { ⁄2 [1 + 𝑗𝐻𝑄 (𝑓)],⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡𝑓𝑣 < 𝑓 < 𝑤⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡ … … … … … … . (8)
𝐻
0,⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡⁡𝑜𝑡ℎ𝑒𝑟𝑤𝑖𝑠𝑒

2. The DSBSC modulated wave is replaced by its complex envelope as

𝑆̃𝐷𝑆𝐵𝑆𝐶 (𝑓) = 𝐴𝑐 𝑀(𝑓) … … … … … . (9)
3. The desired complex envelope 𝑠̃ (𝑡)is determined by evaluating IFT of the
̃(𝑓)𝑆̃𝐷𝑆𝐵𝑆𝐶 (𝑓). i.e.,
product⁡𝐻
𝑆̃(𝑓) = 𝐻
̃ (𝑓)𝑆̃𝐷𝑆𝐵𝑆𝐶 (𝑓) … … … . . (10)
𝐴
𝑆̃ (𝑓) = 𝑐⁄2 [1 + 𝑗𝐻𝑄 (𝑓)]𝑀(𝑓) … … … . . (11)
Take IFT of eqn. 11, we get,
𝐴𝑐⁄
𝑠̃ (𝑡) = 2 [𝑚(𝑡) + 𝑗𝑚𝑄 (𝑡)] … … … … … . . (12)
 where 𝑚𝑄 (𝑡) is the response produced by passing the message through a low pass filter
of impulse response ℎ𝑄 (𝑡).
Substitute eqn. 12 in eqn.3 and simplify, we get
𝐴𝑐⁄ 𝐴𝑐
𝑆 (𝑡 ) = 2 𝑚 (𝑡 ) cos(2𝜋𝑓𝑐 𝑡) − ⁄2 𝑚𝑄 (𝑡)sin⁡(2𝜋𝑓𝑐 𝑡) … … … … … . . (13)
𝐴𝑐⁄
Where 2 𝑚 (𝑡) … … … . . 𝑖𝑛 − 𝑝ℎ𝑎𝑠𝑒⁡𝑐𝑜𝑚𝑝𝑜𝑛𝑒𝑛𝑡
𝐴𝑐⁄
2 𝑚𝑄 (𝑡) … … … 𝑄𝑢𝑎𝑑𝑟𝑎𝑡𝑢𝑟𝑒⁡𝑐𝑜𝑚𝑝𝑜𝑛𝑒𝑛𝑡
Note:
1. If vestigial side band is increased to full side band, VSB becomes DSCSB, i.e., mQ(t) =
0.
̂(t).Where the 𝑚
2. If VSB is reduced to zero, VSB becomes SSB.i.e., mQ(t) =𝑚 ̂ (t) is the
Hilbert transform of m(t)
Similarly if the VSB containing a vestige of the upper sideband, then s(t) is
given by
S(t)= Ac/2 m(t) cos (2πfc t) + Ac/2 mQ(t) sin (2πfct)------(14)
 Envelope detection of a VSB Wave plus Carrier
To make demodulation of VSB wave possible by an envelope detector at the
receiving end it is necessary to transmit a sizeable carrier together with the modulated
wave. The scaled expression of VSB wave by factor k a with the carrier component
𝐴𝑐 cos⁡(2𝜋𝑓𝑐 𝑡) can be given by

𝐴𝑐 𝐴𝑐
𝑠(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) + 𝑘𝑎 𝑚(𝑡) cos(2𝜋𝑓𝑐 𝑡) − 𝑘𝑎 𝑚𝑄 (𝑡)sin⁡(2𝜋𝑓𝑐 𝑡)
2 2

𝑘𝑎 𝐴𝑐 𝑘 𝑎
𝑠(𝑡) = 𝐴𝑐 [1 + 𝑚 (𝑡)] cos(2𝜋𝑓𝑐 𝑡) − 𝑚𝑄 (𝑡) sin(2𝜋𝑓𝑐 𝑡) … … … … … … . (1)
2 2

Where 𝑘𝑎 is the modulation index; it determines the percentage modulation. When

above signal 𝑠(𝑡) is passed through the envelope detector, the detector output is given
by

1⁄
2 2 2
𝑘𝑎 𝑘𝑎
𝑎 (𝑡) = 𝐴𝑐 [(1 + 𝑚(𝑡)) + ( 𝑚𝑄 (𝑡)) ]
2 2

1⁄
2 2
𝑘𝑎
𝑘𝑎 𝑚𝑄 (𝑡 )
= ⁡ 𝐴𝑐 [1 + 𝑚(𝑡)] [1 + ( 2 ) ] … … … … … … … . . (2)
2 𝑘
1 + 𝑎 𝑚(𝑡)
2

The detector output is distorted by the quadrature component 𝑚𝑄 (𝑡) as indicated by

equation (2)

Methods to reduce distortion

 Distortion can be reduced by reducing percentage modulation⁡𝑘𝑎
 Distortion can be reduced by reducing 𝑚𝑄 (𝑡) by increasing the width of the vestigial
sideband.
Comparison of AM Techniques:

S.No Parameter Standard AM SSB DSBSC VSB

1 Power High Less Medium Less than
DSBSC but
greater than
SSB
2 Bandwidth 2fm fm 2fm fm<BW< 2fm
3 Carrier No Yes Yes No
Suppression
4 Receiver Simple Complex Complex Simple
complexity
5 Modulation Non-Linear Linear Linear Linear
6 Sideband No One side band No One
Suppression completely sideband
suppressed
parlty
7 Transmission Minimum Maximum Moderate Moderate
efficiency
8 Application Radio Point to Point Point to Point TV
Communication Communication Communication Broadcasting

Applications of different AM systems:

 Amplitude Modulation: AM radio, Short wave radio broadcast

 DSB-SC: Data Modems, Color TV’s color signals.

 SSB: Telephone
 VSB: TV picture signals
 UNIT III
ANGLE MODULATION

 Basic concepts
 Frequency Modulation
 Single tone frequency modulation
 Spectrum Analysis of Sinusoidal FM Wave
 Narrow band FM, Wide band FM, Constant Average Power
 Transmission bandwidth of FM Wave
 Generation of FM Waves:
o Indirect FM, Direct FM: Varactor Diode and Reactance Modulator
 Detection of FM Waves:
o Balanced Frequency discriminator, Zero crossing detector, Phase locked
loop
 Comparison of FM & AM
 Pre-emphasis & de-emphasis
 FM Transmitter block diagram and explanation of each block
Instantaneous Frequency

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

𝑥(𝑡) = cos⁡(𝜔c +θ0)

is equal to c since it is a constant with respect to t, and the phase of the cosine is the constant
0. The angle of the cosine (t) = ct +0 is a linear relationship with respect to t (a straight
line with slope of c and y–intercept of 0). However, for other sinusoidal functions, the
frequency may itself be a function of time, and therefore, we should not think in terms of the
constant frequency of the sinusoid but in terms of the INSTANTANEOUS frequency of the
sinusoid since it is not constant for all t. Consider for example the following sinusoid

𝑦(𝑡) = cos⁡(θ(t))

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

As a checkup for this definition, we know that the instantaneous frequency of x(t) is equal to
its frequency at all times (since the instantaneous frequency for that function is constant) and is
equal to c. Clearly this satisfies the definition of the instantaneous frequency since (t) = ct
+0 and therefore i(t) = c.
If we know the instantaneous frequency of some sinusoid from – to sometime t, we can find
the angle of that sinusoid at time t using
𝑡
θ(t))=∫−∞ 𝜔(⍺⁡)d⍺⁡

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

Phase Modulation (PM)

In this type of modulation, the phase of the carrier signal is directly changed by the message
signal. The phase modulated signal will have the form
𝑔𝑝𝑚 (𝑡) = 𝐴𝑐𝑜𝑠[𝜔𝐶 𝑡 + 𝑘𝑝 (𝑡)]

where A is a constant, c is the carrier frequency, m(t) is the message signal, and kp is a
parameter that specifies how much change in the angle occurs for every unit of change of m(t).
The phase and instantaneous frequency of this signal are

θ𝑝𝑚 (𝑡) = 𝜔𝐶 𝑡 + 𝑘𝑝 (𝑡)

 𝜔𝑖 (𝑡) == 𝜔𝐶 + 𝑘𝑝 𝑑/𝑑𝑡(𝑚(𝑡)

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

Frequency Modulation (FM)

This type of modulation changes the frequency of the carrier (not the phase as in PM) directly
with the message signal. The FM modulated signal is
𝑡
𝑔𝑓𝑚 (𝑡) = 𝐴𝑐𝑜𝑠[𝜔𝐶 𝑡 + 𝑘𝑓 (𝑡) ∫−∞𝑚(⍺)𝑑⍺⁡]

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

𝑡
θ𝑓𝑚 (𝑡) = 𝜔𝐶 𝑡 + 𝑘𝑓 (𝑡) ∫−∞𝑚(⍺)𝑑⍺

𝑑 𝑡
𝜔𝑖 (𝑡) = 𝜔𝐶 𝑡 + 𝑘𝑓 𝑑𝑡 [∫−∞𝑚(⍺)𝑑⍺]

Relation between PM and FM

PM and FM are tightly related to each other. We see from the phase and frequency
t
relations for PM and FM given above that replacing m(t) in the PM signal with  m ( )d 

dm (t )
gives an FM signal and replacing m(t) in the FM signal with gives a PM signal. This is
dt
illustrated in the following block diagrams.

Frequency Modulator (FM)

t  m (t )d  Phase
 ()d 

m(t) Modulator gFM(t)
 (PM)

Phase Modulator (PM)

dm (t )
d () dt Frequency
m(t) Modulator gPM(t)
dt (FM)
Frequency Modulation

In Frequency Modulation (FM) the instantaneous value of the information signal

controls the frequency of the carrier wave. This is illustrated in the following diagrams.

Notice that as the information signal increases, the frequency of the carrier increases,
and as the information signal decreases, the frequency of the carrier decreases.

The frequency fi of the information signal controls the rate at which the carrier
frequency increases and decreases. As with AM, fi must be less than fc. The amplitude of the
carrier remains constant throughout this process.
When the information voltage reaches its maximum value then the change in frequency
of the carrier will have also reached its maximum deviation above the nominal value. Similarly
when the information reaches a minimum the carrier will be at its lowest frequency below the
nominal carrier frequency value. When the information signal is zero, then no deviation of the
carrier will occur.
The maximum change that can occur to the carrier from its base value 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.
Construction of Narrowband Frequency and Phase Modulators

The above approximations for narrowband FM and PM can be easily used to construct
modulators for both types of signals

kf<<1
t a(t)
m(t)

 ()d  X kf

sin(ct)

–/2  A gFM (NarrowBand)(t)

cos(ct)

Narrowband FM Modulator

kp<<1

m(t) X kp

sin(ct)

–/2  A gPM (NarrowBand)(t)

cos(ct)

Narrowband PM Modulator

Generation of Wideband FM Signals

Consider the following block diagram

Narrowband
m(t)
FM ( . )P gFM (WB) (t)
Modulator

gFM (NB) (t)

Assume a BPF is included in this
block to pass the signal with the
highest carrier freuqnecy and
reject all others

A narrowband FM signal can be generated easily using the block diagram of the narrowband
FM modulator that was described in a previous lecture. The narrowband FM modulator
generates a narrowband FM signal using simple components such as an integrator (an OpAmp),
oscillators, multipliers, and adders. The generated narrowband FM signal can be converted to a
wideband FM signal by simply passing it through a non–linear device with power P. Both the
carrier frequency and the frequency deviation f of the narrowband signal are increased by a
factor P. Sometimes, the desired increase in the carrier frequency and the desired increase in f
are different. In this case, we increase f to the desired value and use a frequency shifter
(multiplication by a sinusoid followed by a BPF) to change the carrier frequency to the desired
value.

SINGLE-TONE FREQUENCY MODULATION

Time-Domain Expression

Since the FM wave is a nonlinear function of the modulating wave, the frequency
modulation is a nonlinear process. The analysis of nonlinear process is the difficult
task. In this section, we will study single-tone frequency modulation in detail to
simplify the analysis and to get thorough understanding about FM.

Let us consider a single-tone sinusoidal message signal defined by

m(t)=𝐴𝑚 cos(2π𝑓𝑚 t)

The instantaneous frequency is then

ƒ(t) = ƒc + kƒAm cos(2πƒnt) = ƒc + ∆ƒ cos(2nƒnt)

where

∆ƒ = kƒAn

t
⁡θ(t) = 2𝝅𝒇c t + 2𝜋𝑘f ∫0 Am cos⁡(2𝝅𝒇𝒎 𝒕)dt
A
=2𝝅𝒇c t + 2𝜋𝑘f ⁡ 2𝝅𝒇m ⁡ sin(2𝝅𝒇𝒎 𝒕)
𝒎
Am
=2𝝅𝒇c t + 𝑘f ⁡ ⁡ sin(2𝝅𝒇𝒎 𝒕)
𝒇𝒎
Δf
=𝒇 ⁡ sin(2𝝅𝒇𝒎 𝒕)+⁡2𝝅𝒇c t
𝒎

⁡θ(t) = 2𝝅𝒇c t + ⁡ βf sin(2𝝅𝒇𝒎 𝒕)

Δf
Where βf = 𝒇
𝒎

is the modulation index of the FM wave. Therefore, the single-tone FM wave is expressed by
sFM(t) = Ac cos[2πƒct + βf sin(2πƒmt)]
 This is the desired time-domain expression of the single-tone FM wave

Similarly, single-tone phase modulated wave may be determined from Eq.as

sPM(t) = Ac cos[2nƒct + kpAn cos(2nƒnt)]

or, sPM(t) = Ac cos[2nƒct + þp cos(2nƒnt)]

where
þp = kpAn
is the modulation index of the single-tone phase modulated wave.

The frequency deviation of the single-tone PM wave is

𝑆𝐹𝑀 (𝑡) = 𝐴𝑐 cos⁡[2𝜋𝑓𝑐 𝑡 + 𝛽𝑓 sin(2𝜋𝑓𝑚 𝑡)]

Spectral Analysis of Single-Tone FM Wave

The above Eq. can be rewritten as
sFM(t) = Re{Acej2nƒctejþ sin(2nƒnt)}

For simplicity, the modulation index of FM has been considered as þ instead of þƒ

afterward. Since sin(2nƒnt) is periodic with fundamental period T = 1⁄ƒn, the
complex expontial ejþ sin(2nƒnt) is also periodic with the same fundamental period.
Therefore, this complex exponential can be expanded in Fourier series representation
as
∞

𝑒 𝑗𝛽sin⁡(2𝜋𝑓𝑚𝑡) = ∑ 𝑐𝑛 𝑒 𝑗2𝜋𝑛𝑓𝑚 𝑡
𝑛=−∞

where the Fourier series coefficients cn are obtained as

 TRANSMISSION BANDWIDTH OF FM WAVE

The transmission bandwidth of an FM wave depends on the modulation index þ. The

modulation index, on the other hand, depends on the modulating amplitude and modulating
frequency. It is almost impossible to determine the exact bandwidth of the FM wave. Rather,
we use a rule-of-thumb expression for determining the FM bandwidth.

For single-tone frequency modulation, the approximated bandwidth is determined by

the expression
1
𝐵 = 2(∆𝑓 + 𝑓𝑚 ) = 2(𝛽 + 1)𝑓𝑚 = 2∆𝑓(1 + )
𝛽
This expression is regarded as the Carson’s rule. The FM bandwidth determined
by this rule accommodates at least 98 % of the total power.

For an arbitrary message signal n(t) with bandwidth or maximum frequency W, the
bandwidth of the corresponding FM wave may be determined by Carson’s rule as
1
𝐵 = 2(∆𝑓 + 𝑊) = 2(𝐷 + 1)𝑊 = 2∆𝑓(1 + )
𝐷

GENERATION OF FM WAVES
FM waves are normally generated by two methods: indirect method and direct method.

Indirect Method (Armstrong Method) of FM Generation

In this method, narrow-band FM wave is generated first by using phase modulator and
then the wideband FM with desired frequency deviation is obtained by using
frequency multipliers.
𝑡
𝑠(𝑡) = 𝐴𝑐 cos⁡[2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝑡) 𝑑𝑡]
0

𝑠(𝑡) = 𝐴𝑐 cos⁡[2𝜋𝑓𝑐 𝑡 + ∅(𝑡)]

𝑡
∅(𝑡) = 2𝜋𝑘𝑓 ∫ 𝑚(𝑡) 𝑑𝑡
0

𝑠(𝑡) = 𝐴𝑐 cos(2𝜋𝑓𝑐 𝑡) cos[∅(𝑡)] − 𝐴𝑐 sin(2𝜋𝑓𝑐 𝑡) sin⁡[∅(𝑡)]

The above eq is the expression for narrow band FM wave
In this case

Fig: Narrowband FM Generator

The frequency deviation ∆ƒ is very small in narrow-band FM wave. To produce

wideband FM, we have to increase the value of ∆ƒ to a desired level. This is achieved by
means of one or multiple frequency multipliers. A frequency multiplier consists of a nonlinear

device and a bandpass filter. The nth order nonlinear device produces a dc component and n
number of frequency modulated waves with carrier frequencies ƒc, 2ƒc, … nƒc and frequency
deviations ∆ƒ, 2∆ƒ, … n∆ƒ, respectively. If we want an FM wave with frequency deviation

of 6∆ƒ, then we may use a 6th order nonlinear device or one 2nd order and one 3rd order
nonlinear devices in cascade followed by a bandpass filter centered at 6ƒc. Normally, we may
require very high value of frequency deviation. This automatically increases the carrier
frequency by the same factor which may be higher than the required carrier frequency. We
may shift the carrier frequency to the desired level by using mixer which does not change the
frequency deviation.

The narrowband FM has some distortion due to the approximation made in deriving
the expression of narrowband FM from the general expression. This produces some amplitude
modulation in the narrowband FM which is removed by using a limiter in frequency
multiplier.
Direct Method of FM Generation

In this method, the instantaneous frequency ƒ(t) of the carrier signal c(t) is varied directly
with the instantaneous value of the modulating signal n(t). For this, an oscillator is used in
which any one of the reactive components (either C or L) of the resonant network of the
oscillator is varied linearly with n(t). We can use a varactor diode or a varicap as a voltage-
variable capacitor whose capacitance solely depends on the reverse-bias voltage applied
across it. To vary such capacitance linearly with n(t), we have to reverse-bias the diode by
the fixed DC voltage and operate within a small linear portion of the capacitance-voltage
characteristic curve. The unmodulated fixed capacitance C0 is linearly varied by n(t) such that
the resultant capacitance becomes

C(t) = C0 − kn(t)

where the constant k is the sensitivity of the varactor diode (measured in

capacitance per volt).

fig: Hartley oscillator for FM generation

The above figure shows the simplified diagram of the Hartley oscillator in
which is implemented the above discussed scheme. The frequency of oscillation for
such an oscillator is given
is the frequency sensitivity of the modulator. The Eq. (5.42) is the required expression for the
instantaneous frequency of an FM wave. In this way, we can generate an FM wave by direct
method.
Direct FM may be generated also by a device in which the inductance of the resonant
circuit is linearly varied by a modulating signal n(t); in this case the modulating signal being
the current.
The main advantage of the direct method is that it produces sufficiently high
frequency deviation, thus requiring little frequency multiplication. But, it has poor frequency
stability. A feedback scheme is used to stabilize the frequency in which the output frequency
is compared with the constant frequency generated by highly stable crystal oscillator and the
error signal is feedback to stabilize the frequency.

DEMODULATION OF FM WAVES
The process to extract the message signal from a frequency modulated wave is known
as frequency demodulation. As the information in an FM wave is contained in its
instantaneous frequency, the frequency demodulator has the task of changing frequency
variations to amplitude variations. Frequency demodulation method is generally categorized
into two types: direct method and indirect method. Under direct method category, we will
discuss about limiter discriminator method and under indirect method, phase-locked loop
(PLL) will be discussed.
Limiter Discriminator Method

Recalling the expression of FM signal,

𝑡
𝑠(𝑡) = 𝐴𝑐 cos⁡[2𝜋𝑓𝑐 𝑡 + 2𝜋𝑘𝑓 ∫ 𝑚(𝑡) 𝑑𝑡]
0

In this method, extraction of n(t) from the above equation involves the three steps:
amplitude limit, discrimination, and envelope detection.
A. Amplitude Limit

During propagation of the FM signal from transmitter to receiver the

amplitude of the FM wave (supposed to be constant) may undergo changes due to
fading and noise. Therefore, before further processing, the amplitude of the FM
signal is limited to reduce the effect of fading and noise by using limiter as discussed
in the section 5.9. The amplitude limitation will not affect the message signal as the
amplitude of FM does not carry any information of the message signal.

B. Discrimination/ Differentiation

In this step we differentiate the FM signal as given by

Here both the amplitude and frequency of this signal are modulated.
In this case, the differentiator is nothing but a circuit that converts change in
frequency into corresponding change in voltage or current as shown in Fig. 5.11. The
ideal differentiator has transfer function

H(jw) =⁡j2nƒ
 Figure : Transfer function of ideal differentiator.

Instead of ideal differentiator, any circuit can be used whose frequency

response is linear for some band in positive slope. This method is known as slope
detection. For this, linear segment with positive slope of RC high pass filter or LC
tank circuit can be used. Figure 5.13 shows the use of an LC circuit as a
differentiator. The drawback is the limited linear portion in the

slope of the tank circuit. This is not suitable for wideband FM where the peak frequency
deviation is high.

Figure : Use of LC tank circuit as a differentiator.

A better solution is the ratio or balanced slope detector in which two tank
circuits tuned at ƒc + ∆ƒ and ƒc − ∆ƒ are used to extend the linear portion as shown in
below figure.
 Figure : Frequency response of balanced slope detector.

Another detector called Foster-seely discriminator eliminates two tank circuits but still
offer the same linear as the ratio detector.

C. Envelope Detection

The third step is to send the differentiated signal to the envelope detector to recover the
message signal.

Phase-Locked Loop (PLL) as FM Demodulator

A PLL consists of a multiplier, a loop filter, and a VCO connected together to form a
feedback loop as shown in Fig. 5.15. Let the input signal be an FM wave as defined
by

s(t) = Ac cos[2πƒct +⁡∅1(t)]

Fig: PLL Demodulator

Let the VCO output be defined by

vVCO(t) = Av sin[2nƒct + ∅2(t)]

where
t

∅2(t) = 2nkv ƒ v(t)dt

0

The high-frequency component is removed by the low-pass filtering of the loop

filter. Therefore, the input signal to the loop filter can be considered as

The difference ∅2(t) −⁡∅1(t) =⁡∅e(t) constitutes the phase error. Let us assume that
the PLL is in phase lock so that the phase error is very small. Then,
Since the control voltage of the VCO is proportional to the message signal, v(t) is the
demodulated signal.
 We observe that the output of the loop filter with frequency response H(ƒ) is the desired
message signal. Hence the bandwidth of H(ƒ) should be the same as the bandwidth W of the message
signal. Consequently, the noise at the output of the loop filter is also limited to the bandwidth W. On
the other hand, the output from the VCO is a wideband FM signal with an instantaneous frequency that
follows the instantaneous frequency of the received FM signal.

PREEMPHASIS AND DEEMPHASIS NETWORKS

In FM, the noise increases linearly with frequency. By this, the higher frequency components
of message signal are badly affected by the noise. To solve this problem, we can use a preemphasis
filter of transfer function Hp(ƒ) at the transmitter to boost the higher frequency components before
modulation. Similarly, at the receiver, the deemphasis filter of transfer function Hd(ƒ)can be used
after demodulator to attenuate the higher frequency components thereby restoring the original
message signal.
The preemphasis network and its frequency response are shown in Figure 5.19
(a) and (b) respectively. Similarly, the counter part for deemphasis network is shown in Figure
5.20.

Figure ;(a) Pre-emphasis network. (b) Frequency response of pre-emphasis network.

 Figure (a) Deemphasis network. (b) Frequency response of Deemphasis network.

In FM broadcasting, ƒ1 and ƒ2 are normally chosen to be 2.1 kHz and 30 kHz

respectively.

The frequency response of preemphasis network is

Comparison of AM and FM:
S.NO AMPLITUDE MODULATION FREQUENCY MODULATION
1. Band width is very small which is one of It requires much wider channel ( 7 to 15
the biggest advantage times ) as compared to AM.
2. The amplitude of AM signal varies The amplitude of FM signal is constant
depending on modulation index. and independent of depth of the
modulation.
3. Area of reception is large The are of reception is small since it is
limited to line of sight.
4. Transmitters are relatively simple & Transmitters are complex and hence
cheap. expensive.
5. The average power in modulated wave is The average power in frequency
greater than carrier power. This added modulated wave is same as contained in
power is provided by modulating source. un-modulated wave.
6. More susceptible to noise interference and Noise can be easily minimized amplitude
has low signal to noise ratio, it is more variations can be eliminated by using
difficult to eliminate effects of noise. limiter.
7. it is not possible to operate without it is possible to operate several
interference. independent transmitters on same
frequency.
8. The maximum value of modulation index No restriction is placed on modulation
= 1, other wise over-modulation would index.
result in distortions.

FM Transmitter
The FM transmitter is a single transistor circuit. In the telecommunication, the frequency
modulation (FM)transfers the information by varying the frequency of carrier wave according to the
message signal. Generally, the FM transmitter uses VHF radio frequencies of 87.5 to 108.0 MHz to
transmit & receive the FM signal. This transmitter accomplishes the most excellent range with less
power. The performance and working of the wireless audio transmitter circuit is depends on the
induction coil & variable capacitor. This article will explain about the working of the FM transmitter
circuit with its applications.
The FM transmitter is a low power transmitter and it uses FM waves for transmitting the sound,
this transmitter transmits the audio signals through the carrier wave by the difference of frequency. The
carrier wave frequency is equivalent to the audio signal of the amplitude and the FM transmitter
produce VHF band of 88 to 108MHZ.Plese follow the below link for: Know all About Power
Amplifiers for FM Transmitter
 Block Diagram of FM Transmitter

Working of FM Transmitter Circuit

The following circuit diagram shows the FM transmitter circuit and the required electrical and
electronic components for this circuit is the power supply of 9V, resistor, capacitor, trimmer capacitor,
inductor, mic, transmitter, and antenna. Let us consider the microphone to understand the sound signals
and inside the mic there is a presence of capacitive sensor. It produces according to the vibration to the
change of air pressure and the AC signal.

FM Transmitter circuit

The formation of the oscillating tank circuit can be done through the transistor of 2N3904 by using the
inductor and variable capacitor. The transistor used in this circuit is an NPN transistor used for general
purpose amplification. If the current is passed at the inductor L1 and variable capacitor then the tank
circuit will oscillate at the resonant carrier frequency of the FM modulation. The negative feedback
will be the capacitor C2 to the oscillating tank circuit.

To generate the radio frequency carrier waves the FM transmitter circuit requires an oscillator.
The tank circuit is derived from the LC circuit to store the energy for oscillations. The input audio
signal from the mic penetrated to the base of the transistor, which modulates the LC tank circuit carrier
frequency in FM format. The variable capacitor is used to change the resonant frequency for fine
modification to the FM frequency band. The modulated signal from the antenna is radiated as radio
waves at the FM frequency band and the antenna is nothing but copper wire of 20cm long and 24
gauge. In this circuit the length of the antenna should be significant and here you can use the 25-27
inches long copper wire of the antenna.

Application of Fm Transmitter
 The FM transmitters are used in the homes like sound systems in halls to fill the sound with the
audio source.
 These are also used in the cars and fitness centers.
 The correctional facilities have used in the FM transmitters to reduce the prison noise in common
areas.
Advantages of the FM Transmitters

 The FM transmitters are easy to use and the price is low

 The efficiency of the transmitter is very high
 It has a large operating range
 This transmitter will reject the noise signal from an amplitude variation.
 UNIT IV
NOISE
 Noise in communication System,
 White Noise
 Narrowband Noise –In phase and Quadrature phase components
 Noise Bandwidth
 Noise Figure
 Noise Temperature
 Noise in DSB& SSB System
 Noise in AM System
 Noise in Angle Modulation System
 Threshold effect in Angle Modulation System
Noise in communication system

 Noise is unwanted signal that affects wanted signal

 Noise is random signal that exists in communication systems
Effect of noise

 Degrades system performance (Analog and digital)

 Receiver cannot distinguish signal from noise
 Efficiency of communication system reduces
Types of noise

 Thermal noise/white noise/Johnson noise or fluctuation noise

 Shot noise
 Noise temperature
 Quantization noise

Noise temperature
Equivalent noise temperature is not the physical temperature of amplifier, but a theoretical construct,
that is an equivalent temperature that produces that amount of noise power

𝑇𝑒 = (𝐹 − 1)

White noise
One of the very important random processes is the white noise process. Noises in many
practical situations are approximated by the white noise process. Most importantly, the white noise
plays an important role in modelling of WSS signals.

A white noise process w(t) is a random process that has constant power spectral density at all
frequencies. Thus

𝑁0
𝑆𝑊 (𝜔) = -∞< 𝜔<∞
2
where 𝑁0 is a real constant and called the intensity of the white noise. The corresponding
autocorrelation function is given by

𝑁
𝑅𝑊 (𝜏) = δ (𝜏 ) where δ(𝜏) is the Dirac delta
2

The average power of white noise

1 ∞ 𝑁
𝑃𝑎𝑣𝑔 = 𝐸𝑊 2 (t)=2𝜋 ∫−∞ 2 𝑑 𝜔→∞

The autocorrelation function and the PSD of a white noise process is shown in Figure 1 below.

fig: auto correlation and psd of white noise

NARROWBAND NOISE (NBN)

In most communication systems, we are often dealing with band-pass filtering of signals. Wideband
noise will be shaped into band limited noise. If the bandwidth of the band limited noise is relatively
small compared to the carrier frequency, we refer to this as narrowband noise.

the narrowband noise is expressed as as

n(t)=x(t)cos(2𝜋𝑓𝑐 t)-y(t)sin (2𝜋𝑓𝑐 t)

where fc is the carrier frequency within the band occupied by the noise. x(t) and y(t) are known
as the quadrature components of the noise n(t). The Hibert transform of n(t) is

Proof.
The Fourier transform of n(t) is

1 1 1 1
N(f)= 2 𝑋 (𝑓 − 𝑓𝑐 ) + 2 𝑋 (𝑓 + 𝑓𝑐 ) + 2 𝑗𝑌(𝑓 − 𝑓𝑐 ) − 2 𝑗𝑌 (𝑓 + 𝑓𝑐 )

Let N^ ( f ) be the Fourier transform of n^ ( t). In the frequency domain, N^ (f) =

N(f)[-j sgn(f)]. We simply multiply all positive frequency components of N(f) by -j and all negative
frequency components of N(f) by j. Thus
 𝑛̂(t)=H[n(t)]=x(t)sin(2𝜋𝑓𝑐 t)+y(t)cos (2𝜋𝑓𝑐 t)
̂ (f)=−𝑗 1 𝑋(𝑓 − 𝑓𝑐 ) + 𝑗 1 𝑋(𝑓 + 𝑓𝑐 ) − 𝑗 1 𝑗𝑌(𝑓 − 𝑓𝑐 ) − 𝑗 1 𝑗𝑌 (𝑓 + 𝑓𝑐 )
𝑁 2 2 2 2

̂ (f)=−𝑗 1 𝑋(𝑓 − 𝑓𝑐 ) + 𝑗 1 𝑋(𝑓 + 𝑓𝑐 ) + 1 𝑌(𝑓 − 𝑓𝑐 ) + 1 𝑌(𝑓 + 𝑓𝑐 )

𝑁 2 2 2 2

̂ (f) is
And the Inverse Fourier transform of 𝑁

𝑛̂(t)=x(t)sin(2𝜋𝑓𝑐 t)+y(t)cos (2𝜋𝑓𝑐 t)

The quadrature components x(t) and y(t) can now be derived from equations

x(t) = n(t)co2 fct + n^(t)sin 2 fct \

and
y(t) = n(t)cos 2 fct - n^(t)sin 2 fct

Fig: Generation of narrow band noise

Fig: Generation of quadrature components of n(t).

 Filters at the receiver have enough bandwidth to pass the

desired signal but not too big to pass excess noise.
 Narrowband (NB) fc center frequency is much bigger that the bandwidth.
 Noise at the output of such filters is called narrowband noise (NBN).
 NBN has spectral concentrated about some mid-band frequency fc
 The sample function of such NBN n(t) appears as a sine wave of frequency fc which modulates
slowly in amplitude and phase

Noise figure

The Noise figure is the amount of noise power added by the electronic circuitry in the receiver to the
thermal noise power from the input of the receiver. The thermal noise at the input to the receiver passes
through to the demodulator. This noise is present in the receive channel and cannot be removed. The
noise figure of circuits in the receiver such as amplifiers and mixers, adds additional noise to the
receive channel. This raises the noise floor at the demodulator

𝑠𝑖𝑔𝑛𝑎𝑙 𝑡𝑜 𝑛𝑜𝑖𝑠𝑒 𝑟𝑎𝑡𝑖𝑜 𝑎𝑡 𝑖𝑛𝑝𝑢𝑡

Noise figure =
𝑠𝑖𝑔𝑛𝑎𝑙 𝑡𝑜 𝑛𝑜𝑖𝑠𝑒 𝑟𝑎𝑡𝑖𝑜 𝑎𝑡 𝑜𝑢𝑡𝑝𝑢𝑡

Noise Bandwidth

A filter’s equivalent noise bandwidth (ENBW) is defined as the bandwidth of a perfect rectangular
filter that passes the same amount of power as the cumulative bandwidth of the channel selective filters
in the receiver. At this point we would like to know the noise floor in our receiver, i.e. the noise power
in the receiver intermediate frequency (IF) filter bandwidth that comes from kTB. Since the units of
kTB are Watts/ Hz, calculate the noise floor in the channel bandwidth by multiplying the noise power
in a 1 Hz bandwidth by the overall equivalent noise bandwidth in Hz.

NOISE IN DSB-SC SYSTEM:

Let the transmitted signal is

u(t)=𝐴𝑐 𝑚(𝑡) cos(2𝜋𝑓𝑐 t)

The received signal at the output of the receiver noise- limiting filter : Sum of this signal and filtered
noise .A filtered noise process can be expressed in terms of its in-phase and quadrature components as

where nc(t) is in-phase component and ns(t) is quadrature component

Received signal (Adding the filtered noise to the modulated signal)

r(t)=u(t)+n(t)
=𝐴𝑐 𝑚(𝑡) cos(2𝜋𝑓𝑐 t)+ 𝑛𝑐 (𝑡)cos(2π𝑓𝑐 t)- 𝑛𝑠 (𝑡)sin(2π𝑓𝑐 t )

Demodulate the received signal by first multiplying r(t) by a locally generated sinusoid cos(2𝜋fct + ∅),

where is the phase of the sinusoid.Then passing the product signal through an ideal lowpass filter
having a bandwidth W.

The low pass filter rejects the double frequency components and passes only the low pass components.
1 1
y(t)=2 𝐴𝑐 𝑚(𝑡) cos(∅)+2 [𝑛𝑐 (𝑡)cos(∅ )+ 𝑛𝑠 (𝑡)sin(∅ )]

the effect of a phase difference between the received carrier and a locally generated carrier at the
receiver is a drop equal to 𝑐𝑜𝑠 2 (∅) in the received signal power.

Phase-locked loop

The effect of a phase-locked loop is to generate phase of the received carrier at the receiver.

If a phase-locked loop is employed, then = 0 and the demodulator is

called a coherent or synchronous demodulator.
In our analysis in this section, we assume that we are employing a coherent demodulator.

With this assumption, we assume that =0

1
Y(t)=2 [𝐴𝑐 𝑚(𝑡) + 𝑛𝑐 (t)]

Therefore, at the receiver output, the message signal and the noise components are additive and we are
able to define a meaningful SNR. The message signal power is given by

𝐴2𝐶
𝑃0 = 𝑃
4 𝑀
Power PM is the content of the message signal

The noise power is given by

1 1
𝑃𝑛 0 =4 𝑃𝑛 𝑐 =4 𝑃𝑛

The power content of n(t) can be found by noting that it is the result of passing nw(t) through a filter
with bandwidth Bc.Therefore, the power spectral density of n(t) is given by

which is identical to baseband SNR.

In DSB-SC AM, the output SNR is the same as the SNR for a baseband system. DSB-SC AM does not
provide any SNR improvement over a simple baseband communication system.
NOISE IN SSB-SC SYSTEM:

Let SSB modulated signal is

̂(t) sin (2𝜋𝑓𝑐 t)
u(t)=𝐴𝑐 𝑚(𝑡) cos(2𝜋𝑓𝑐 t)±𝑚

Input to the demodulator

Assumption : Demodulation with an ideal phase reference.

Hence, the output of the lowpass filter is the in-phase component (with a coefficient of
½) of the preceding signal.

The signal-to-noise ratio in an SSB system is equivalent to that of a DSB system.

Noise in Conventional AM

Where a is the modulation index

mn(t) is normalized so that its minimum value is -1
If a synchronous demodulator is employed, the situation is basically similar to the DSB
case, except that we have 1 + amn(t) instead of m(t).
1
y(t)= [𝐴𝑐 𝑎𝑚𝑛 (t)] + 𝑛𝑐 (t)]
2

 In practical applications, the modulation index a is in the range of 0.8-0.9.

 Power content of the normalized message process depends on the message source.

 Speech signals : Large dynamic range, PM is about 0.1.

 The overall loss in SNR, when compared to a baseband system, is a factor of

0.075 or equivalent to a loss of 11 dB.

The reason for this loss is that a large part of the transmitter power is used to send the carrier
component of the modulated signal and not the desired signal. To analyze the envelope-detector
performance in the presence of noise, we must use certain approximations.

This is a result of the nonlinear structure of an envelope detector, which makes an exact analysis
difficult

In this case, the demodulator detects the envelope of the received signal and the noise process.
The input to the envelope detector is

r(t)= [𝐴𝑐 [1 + 𝑎𝑚𝑛 (t)] + 𝑛𝑐 (t)] ) cos(2𝜋𝑓𝑐 t)- 𝑛𝑠 (𝑡)sin(2𝜋𝑓𝑐 t)

Therefore, the envelope of r ( t ) is given by

Now we assume that the signal component in r ( t ) is much stronger than the noise
component. Then
Therefore, we have a high probability that

After removing the DC component, we obtain

r(t)= 𝐴𝑐 𝑎𝑚𝑛 (t) + 𝑛𝑐 (t)

which is basically the same as y(t) for the synchronous demodulation without the ½
coefficient.
This coefficient, of course, has no effect on the final SNR. So we conclude that, under the
assumption of high SNR at the receiver input, the performance of synchronous and envelope
demodulators is the same.

However, if the preceding assumption is not true, that is, if we assume that, at the receiver input, the
noise power is much stronger than the signal power, Then

We observe that, at the demodulator output, the signal and the noise components are no longer
additive. In fact, the signal component is multiplied by noise and is no longer distinguishable. In this
case, no meaningful SNR can be defined. We say that this system is operating below the threshold.
 The subject of threshold and its effect on the performance of a communication system will be covered
in more detail when we discuss the noise performance in angle modulation.

Effect of threshold in angle modulation system:

FM THRESHOLD EFFECT FM threshold is usually defined as a Carrier-to-Noise ratio at

which demodulated Signal-to-Noise ratio falls 1dB below the linear relationship . This is the effect
produced in an FM receiver when noise limits the desired information signal. It occurs at about 10 dB,
as earlier stated in 5 the introduction, which is at a point where the FM signal-to-Noise improvement is
measured. Below the FM threshold point, the noise signal (whose amplitude and phase are randomly
varying) may instantaneously have amplitude greater than that of the wanted signal. When this
happens, the noise will produce a sudden change in the phase of the FM demodulator output. In an
audio system, this sudden phase change makes a “click”. In video applications the term “click noise” is
used to describe short horizontal black and white lines that appear randomly over a picture

An important aspect of analogue FM satellite systems is FM threshold effect. In FM systems where the
signal level is well above noise received carrier-to-noise ratio and demodulated signal-to-noise ratio
are related by:

The expression however does not apply when the carrier-to-noise ratio decreases below a certain point.
Below this critical point the signal-to-noise ratio decreases significantly. This is known as the FM
threshold effect (FM threshold is usually defined as the carrier-to-noise ratio at which the demodulated
signal-to-noise ratio fall 1 dB below the linear relationship given in Eqn 9. It generally is considered to
occur at about 10 dB).

Below the FM threshold point the noise signal (whose amplitude and phase are randomly varying),
may instantaneously have an amplitude greater than that of the wanted signal. When this happens the
noise will produce a sudden change in the phase of the FM demodulator output. In an audio system this
sudden phase change makes a "click". In video applications the term "click noise" is used to describe
short horizontal black and white lines that appear randomly over a picture, because satellite
communications systems are power limited they usually operate with only a small design margin above
the FM threshold point (perhaps a few dB). Because of this circuit designers have tried to devise
techniques to delay the onset of the FM threshold effect. These devices are generally known as FM
threshold extension demodulators. Techniques such as FM feedback, phase locked loops and frequency
locked loops are used to achieve this effect. By such techniques the onset of FM threshold effects can
be delayed till the C/N ratio is around 7 dB.

Noise in Angle Modulated Systems

Like AM, noise performance of angle modulated systems is characterized by parameter γ

Note if bandwidth ratio is increased by a factor 2.Then

Increases by a factor 4

This exchange of bandwidth and noise performance is an important feature of FM

 UNIT-V

Receivers

Introduction:

In radio communications, a radio receiver is an electronic device that receives radio

waves and converts the information carried by them to a usable form.The antenna intercepts
radio waves (electromagnetic waves) and converts them to tiny alternating currents which are
applied to the receiver and the receiver extracts the desired information. The receiver uses
electronic filters to separate the desired radio frequency signal from all the other signals picked
up by the antenna, an electronic amplifier to increase the power of the signal for further
processing, and finally recovers the desired information through demodulation. The information
produced by the receiver may be in the form of sound, moving images (television), or data.
Radio receivers are very widely used in modern technology, as components of communications,
broadcasting, remote control, and wireless networking systems.

Types of Receivers:

The component technology, and in particular semiconductor technology has surged

forwards enabling much higher levels of performance to be achieved in a much smaller
space.Many of the different radio receiver types have been around for many years. Some of them
are

1. Tuned radio frequency, TRF:

This type of radio receiver was one of the first that was used. The very first radio
receivers of this type simply consisted of a tuned circuit and a detector. Crystal sets were early
forms of TRF radios. Later amplifiers were added to boost the signal level, both at the radio
frequencies and audio frequencies. There were several problems with this form of receiver. The
main one was the lack of selectivity. Gain / sensitivity were also a use.

2.Regenerative receiver:
The regenerative radio receiver significantly improved the levels of gain and selectivity
obtainable. It used positive feedback and ran at the point just before oscillation occurred. In this
way a significant multiplication in the level of "Q" of the tuned circuit was gained. Also major
improvements in gain were obtained this way.
3. Super regenerative receiver:
The super regenerative radio receiver takes the concept of regeneration a stage further.
Using a second lower frequency oscillation within the same stage, this second oscillation
quenches or interrupts the oscillation of the main regeneration – typically at frequencies of
around 25 kHz or so above the audio range. In this way the main regeneration can be run so that
the stage is effectively in oscillation where it provides very much higher levels of gain. Using the
second quench oscillation, the effects of running the stage in oscillation are not apparent to the
listener, although it does emit spurious signals which can cause interference locally.
4. Super heterodyne receiver:
The super heterodyne form of radio receiver was developed to provide additional levels
of selectivity. It uses the heterodyne or mixing process to convert signals done to a fixed
intermediate frequency. Changing the frequency of the local oscillator effectively tunes the radio.

5. Direct conversion receiver:

This type of radio format converts the signal directly down to the baseband frequency.
Initially it was used for AM, Morse (CW) and SSB transmissions. Now it is widely used for
digital communications where IQ demodulators are used to take advantage of the variety of
phase shift keying, PSK, and quadrature amplitude modulation, QAM signals.
Many of these different types of radio receiver are in widespread use today. Each type of
radio has its own characteristics that lend its use to particular applications.

Tuned Radio Frequency Receiver(TRF)

The tuned radio frequency receiver is one in which the tuning or selectivity is provided at
the radio frequency stages.Tuning is provided by a tuned coil / capacitor combination, and then
the signal is presented to a simple crystal or diode detector where the amplitude modulated signal
is recovered. This is then passed straight to the headphones.

The tuned radio frequency receiver was used in the early days of wireless technology but
it is rarely used today as other techniques offering much better performance are available.

Fig.Block diagram of Tuned radio frequency receiver

Operation:

TRF receiver consists of two or three stages of RF amplifiers, detector, audio amplifier
and power amplifier. The RF amplifier stages placed between the antenna and detector are used
to increase the strength of the received signal before it is applied to the detector. These RF
amplifiers are tuned to fix frequency, amplify the desired band of frequencies. Therefore they
provide amplification for selected band of frequencies and rejection for all others. As selection
and amplification process is carried out in two or three stages and each stage must amplify the
same band of frequencies, the ganged tuning is provided.
The amplified signal is then demodulated using detector to recover the modulating signal.
The recovered signal is amplified further by the audio amplifier followed by power amplifier
which provides sufficient gain to operate a loud speaker. The TRF receivers suffered from
number of annoying problems.
The tuned radio frequency receiver was popular in the 1920s as it provided sufficient gain
and selectivity for the receiving the broadcast stations of the day. However tuning is difficult in
which as each stage in the early radios needed to be adjusted separately. The TRF receiver has
largely been disregarded in recent years. Other receiver topologies offer far better levels of
performance, and with integrated circuit technology, the additional circuitry of other types of
receiver is not an issue.Later ganged tuning capacitors were introduced, but by this time the
superheterodyne receiver was becoming more widespread.

Disadvantages of TRF receiver

• Poor selectivity and low sensitivity in proportion to the number of tuned amplifiers used.
• Selectivity requires narrow bandwidth, and narrow bandwidth at a high radio frequency implies
high Q or many filter sections.
• An additional problem for the TRF receiver is tuning different frequencies. All the tuned
circuits need to tune together to the same frequency or track very closely. Another problem is to
keep the narrow bandwidth tuning. Keeping several tuned circuits aligned is difficult.
• The bandwidth of a tuned circuit doesn’t remain constant and increases with the frequency
increase.
• The need to have all RF stages track one another
• Instability due to large number of RF stages.
• Received bandwidth increases with frequency (varies with center frequency)
• Gain is non-uniform over a wide range of frequencies

Superheterodyne Receiver:

To solve basic problem of TRF receivers, first all the incoming RF frequencies are
converted to fix lower frequency called Intermediate Frequency (IF).Then this fix intermediate
frequency is amplified and detected to reproduce the original information. Since the
characteristics of the IF amplifier are independent of the frequency to which the receiver is
tuned, the selectivity and sensitivity of superheterodyne receivers are fairly uniform throughout
its tuning range.
The basic concept and theory behind the superheterodyne radio involves the process of
mixing. This enables signals to be translated from one frequency to another. The input frequency
is often referred to as the RF input, whilst the locally generated oscillator signal is referred to as
the local oscillator, and the output frequency is called the intermediate frequency as it is between
the RF and audiofrequencies.
 Fig .Block diagram of a Superheterodyne receiver

Operation:

Signals enter the receiver from the antenna and are applied to the RF amplifier where
they are tuned to remove the image signal and also reduce the general level of unwanted signals
on other frequencies that are not required.
The signals are then applied to the mixer along with the local oscillator where the wanted
signal is converted down to the intermediate frequency. Here significant levels of amplification
are applied and the signals are filtered. This filtering selects signals on one channel against those
on the next. It is much larger than that employed in the front end.The advantage of the IF filter as
opposed to RF filtering is that the filter can be designed for a fixed frequency. This allows for
much better tuning. Variable filters are never able to provide the same level of selectivity that
can be provided by fixed frequency ones.
Once filtered the next block in the superheterodyne receiver is the demodulator. This
could be for amplitude modulation, single sideband, frequency modulation, or indeed any form
of modulation. It is also possible to switch different demodulators in according to the mode being
received.
The final element in the superheterodyne receiver block diagram is shown as an audio
amplifier, although this could be any form of circuit block that is used to process or amplified the
demodulated signal.
Another important circuit in the superheterodyne receiver is AGC and AFC circuit. AGC is
used to maintain a constant output voltage level over a wide range of RF input signal levels.
It derives the dc bias voltage from the output of detector which is proportional to the
amplitude of the received signal.This dc bias voltage is feedback to the IF amplifiers to control
the gain of the receiver. As a result, it provides a constant output voltage level over a wide range
of RF input signal levels. AFC circuit generated AFC signal which is used to adjust and stabilize
the frequency of the local oscillator.
Advantages of the superheterodyne receiver

• IF stage permits use at very high frequencies.

• Because many components operate at the fixed IF, they can be optimized.
• Less expensive.
• Better selectivity
• Improved circuit stability.
• Uniform gain over a wide range of frequencies

Receiver Sections:

RF tuning & amplification:

RF amplifier provides initial gain and selectivity. RF amplifier is a simple class A circuit.
This RF stage within the overall block diagram for the receiver provides initial tuning to remove
the image signal. If noise performance for the receiver is important, then this stage will be
designed for optimum noise performance. This RF amplifier circuit block will also increase the
signal level so that the noise introduced by later stages is at a lower level in comparison to the
wanted signal. A typical bipolar circuit as (a) and FET circuit as (b) is shown below.

Values of resistors R1 and R2 in the bipolar circuit are adjusted such that the amplifier
works as a class A amplifier.The antenna is connected through coupling capacitor to the base of
the transistor.This makes the circuit very broad band as the transistor will amplify virtually any
signal picked up by the antenna. The collector is tuned with a parallel resonant circuit to provide
the initial selectivity for the mixer input.
FET circuit Fig.(b) is more effective than the transistor circuit. Their high input
impedance minimizes the loading on tuned circuits, thereby permitting the Q of the circuit to be
higher and selectivity to be sharper.

Local oscillator:

The local oscillator circuit block can take a variety of forms. Early receivers used free
running local oscillators. Today most receivers use frequency synthesizers, normally based
around phase locked loops. These provide much greater levels of stability and enable frequencies
to be programmed in a variety of ways.
Mixer or Frequency Changer or Converter:

Real-life mixers produce a variety of other undesired outputs, including noise and they
may also suffer overload when very strong signals are present.
Although very basic non-linear devices can actually perform a basic RF mixing or multiplication
process, the performance will be far from the ideal, and where good receiver performance is
required, the specification of the RF mixer must be match this expectation.
The basic process of RF mixing or multiplication where the incoming RF signal and a
local oscillator are mixed or multiplied together to produce signals at the sum and difference
frequencies is key to the whole operation of the superheterodyne receiver.
There are a number of considerations when looking at the receiver design and topology
with respect to the RF mixer.There are many different forms of mixer that can be used, and the
choice of the type depends very much upon the receiver and the anticipated performance.

Separately Excited Mixer

Fig. Separately Excited Mixer

In Separately excited mixer, one device acts as a mixer while the other supplies the
necessary oscillations. Bipolar transistor T2 forms the Hartley oscillator circuit and oscillates
with local frequency. FET T1 is a mixer whose gate s fed with the output of local oscillator and
its bias is adjusted. The local oscillator varies the gate bias of the FET to vary its
transconductance resulting intermediate frequency at the output. Output is taken through double
tuned transformer in the drain of the mixer and fed to the IF amplifier.
Self Excited Mixer

Fig: Self Excited Mixer

Self Excited circuit oscillates and the transconductance of the transistor is varied in a nonlinear
manner at the local oscillator gate. This variable transconductance is used by the transistor to
amplify the incoming RF signal.
IF amplifier & filter:

Tracking
The Superheterodyne receiver has number of tunable circuits which must all be tuned
correctly if any given station is to be received. The ganged tuning is employed which
mechanically couples all tuning circuits so that only one tuning control is required. Usually there
are three tuned circuits: Antenna or RF tuned circuit, Mixer tuned circuit and local oscillator
tuned circuit.
All these circuits must be tuned to get proper RF input and to get IF frequency at the
output of the mixer. The process of tuning circuit to get the desired output is called Tracking.
Tracking error will result in incorrect frequency being fed to the IF amplifier and these must be
avoided.
To avoid tracking errors,ganged capacitors with identical sections are used.
A different value of inductance and capacitors called trimmers and padders are used to adjust the
capacitance of the oscillator to the proper range. Common methods used for tracking are
 Padder Tracking
 Trimmer Tracking
 Three-Point Tracking

Intermediate IF amplifier:

Figure shows two Stage IF Amplifier. Two stages are transformer coupled and all IF
transformers are single tuned i.e, tuned for single frequency.
IF amplifiers are tuned voltage amplifiers which istuned for the fixed frequency. Its
function is to amplify only tuned frequency signal and reject all others .Most of the receiver gain
is provided by the IF amplifiers and the required gain is obtained usually by two or more stages
of IF amplifiers are required.

Fig. Two Stage IF Amplifier

Automatic Gain Control, AGC:

An automatic gain control is incorporated into most superheterodyne radios. Its function
is to reduce the gain for strong signals so that the audio level is maintained for amplitude
sensitive forms of modulation, and also to prevent overloading. It is a system in which the
overall gain of a radio receiver is varied automatically with the variations in the strength of the
receiver signal to maintain the output substantially constant.
When the average signal level increases, the size of the AGC bias increases and the gain
is deceased. When there is no signal, there is a minimum AGC bias and the amplifiers produce
maximum gain. There are two types of AGC circuits. They are Simple AGC and Delayed AGC.

Simple AGC
In Simple AGC, the AGC bias starts to increase as soon as the received signal level
exceeds the background noise level.As a result receiver gain starts falling down, reducing the
sensitivity of the receiver.
In the circuit, the dc bias produced by half wave rectifier is used to control the gain of RF
or IF amplifier. The time constant of the filter is kept at least 10 times longer than the period of
the lowest modulation frequency received. If the time constant is kept longer, it will give better
filtering. The recovered signal is then passes through capacitor to remove dc. The resulting ac
signal is further amplified and applied to the loud speaker.
 Fig. Simple AGC circuit

Delayed AGC
In simple AGC, the unwanted weak signals (noise signals) are amplified with high gain.
To avoid this, in delayed AGC circuits, AGC bias is not applied to amplifiers until signal
strength has reached a predetermined level, after which AGC bias is applied as with simple
AGC, but more strongly.

Fig. Delayed AGC circuit

AGC output is applied to the difference amplifier. It gives dc AGC only when AGC
output generated by diode detector is above certain dc threshold voltage. This threshold voltage
can be adjusted by adjusting the voltage at the positive input of the operational amplifier.

Fig. Response of receiver with various AGC

Demodulator:

The superheterodyne receiver block diagram only shows one demodulator, but in reality
radios may have one or more demodulators dependent upon the type of signals being receiver.

Audio amplifier:

Once demodulated, the recovered audio is applied to an audio amplifier block to be

amplified to the required level for loudspeakers or headphones. Alternatively the recovered
modulation may be used for other applications whereupon it is processed in the required way by
a specific circuit block.

Receiver Characteristics

The performance of the radio receiver can be measured in terms of following receiver
characteristics
 Selectivity
 Sensitivity
 Fidelity
 Image frequency and its rejection
 Double Spotting

Selectivity
The ability of the receiver to select the wanted signals among the various incoming
signals is termed as Selectivity. It rejects the other signals at closely lying frequencies.
Selectivity of a receiver changes with incoming signal frequency and are poorer at high
frequencies.
Selectivity in a receiver is obtained by using tuned circuits. These are LC circuits tuned to
resonate at a desired signal frequency. The Q of these tuned circuits determines the selectivity.
Selectivity shows the attenuation that the receiver offers to signals at frequencies near to the one
to which it is tuned. A good receiver isolates the desired signal in the RF spectrum and
eliminates all other signals.
Sensitivity
The sensitivity of a radio receiver is its ability to amplify weak signals. It is often defined
in terms of the voltage that must be applied to the receiver input terminals to give a standard
output power, measured at the output terminals.The most important factors determining the
sensitivity of a superheterodyne receiver are the gain of the IF amplifier(s) and that of the RF
amplifier .The more gain that a receiver has, the smaller the input signal necessary to produce the
desired output power. Therefore sensitivity is a primary function of the overall receiver gain.
Good communication receiver has a sensitivity of 0.2 to 1 µV

Fidelity
Fidelity refers to the ability of the receiver to reproduce all the modulating frequencies
equally. Figure shows the typical fidelity curve for radio receiver.
 Fig.3. Typical Fidelity curve

The fidelity at the lower modulating frequencies is determined by the low frequency response of
the IF amplifier and the fidelity at the higher modulating frequencies is determined by the high
frequency response of the IF amplifier. Fidelity is difficult to obtain in AM receiver because
good fidelity requires more bandwidth of IF amplifier resulting in poor selectivity.
Image frequency and its Rejection
In a standard broadcast receiver the local oscillator frequency is made higher than the
incoming signal frequency for reasons that will become apparent .It is made equal at all times to
the signal frequency plus the intermediate frequency.Thus f0=fs+fi or f0=fs−fi, no matter what
the signal frequency may be. When f0 and fs are mixed, the difference frequency, which is one
of the by-products, equal to fi is passed and amplified by the IF stage.If a frequency fsi manages
to reach the mixer, such that fsi=fo+fi, that is , fsi=fs+2fi then this frequency will also
produce fi when mixed with f0.
Unfortunately, this spurious intermediate-frequency signal will also be amplified by the
IF stage and will therefore provide interference.This has the effect of two stations being received
simultaneously and is naturally undesirable.The term fsi is called image frequency and is defined
as the signal frequency plus twice the intermediate frequency.
The rejection of an image frequency by a single –tuned circuit, i.e., the ratio of the gain at
the signal frequency to the gain at the image frequency, is given by

α= √1+Q2ρ2
where

Q=loaded Qoftuned circuit

If the receiver has an RF stage, then there are two tuned circuits , both tuned to fs .The
rejection of each will be calculated by the same formula , and the total rejection will be product
of the two.
Image rejection depends on the front-end selectivity of the receiver and must be achieved
before the IF stage.Once the spurious frequency enters the first IF amplifier, it becomes
impossible to remove it from the wanted signal.
Double spotting
This is well-known phenomenon, which manifests itself by the picking up of the same
shortwave station at two nearby points on the receiver dial.It is caused by poor front-end
selectivity, i.e., inadequate image-frequency rejection .That is to say, the front end of the receiver
does not select different adjacent signals very well , but the IF stage takes care of eliminating
almost all of them .
As a matter of interest, double spotting may be used to calculate the intermediate
frequency of an unknown receiver, since the spurious point on the dial is precisely 2f, below the
correct frequency .An improvement in image-frequency rejection will produce a corresponding
reduction in double spotting.

FM Receiver

Fig. Block diagram of FM receiver

FM receiver is a Superheterodyne receiver and the basic block diagram is same as AM

receiver. The basic differences are as follows.

1. Generally much higher operating frequencies in FM

2. Need for limiting and de-emphasis in FM.
3. Totally different methods of demodulation.
4. Different methods of obtaining AGC.

Comparison with AM Receiver

Different stages of FM receiver are explained below.

RF Amplifier

Since FM signal has a larger bandwidth it is likely to encounter more noise. Hence to
reduce the noise figure of the receiver, an RF amplifier is used. The RF amplifier stage matches
the antenna to the receiver. For this purpose and to avoid neutralization, grounded-base or
grounded gate circuits are employed for this stage. Both circuits have low input impedance,
suitable for matching with antenna impedance and nether require neutralization.

Fig. Grounded-gate FET RF amplifier

In figure, since gate terminal is grounded, the input and output sides are isolated for RF
purposes. There is no feedback and hence no instability. Therefore circuit does not require
neutralization. The low impedance of the FET amplifier is matched to antenna through a tuned
RF transformer. Both the input and output tank circuits are tuned to carrier frequency.

Oscillators and Mixers

The oscillator circuit may be Clapp and Colpitts which is suited in VHF operation
.Tracking is not normally much of a problem in FM broadcast receivers. This is because the
tuning frequency is only 1.25:1.much less than in AM broadcasting.
The mixer stage uses a tuned circuit as its load. The circuit is tuned to Intermediate
frequency of 10.7 MHz and hence selects the difference between incoming carrier frequency and
locally generated oscillator frequency.

Intermediate Frequency and IF amplifier

The types and operation do not differ much from their AM counterparts. But the
intermediate frequency and bandwidth required are far higher than in AM broadcast receivers.
For receivers operating in the 88 to 108 MHz band is an IF of 10.7 MHz and a bandwidth of 200
KHz. Due to large bandwidth, gain per stage may be low. Two IF amplifier stages are often
provided, in which case the shrinkage of bandwidth as stages are cascaded must be taken into
account.
Amplitude Limiter:

Frequency modulation is developed to provide a communication system which is less

noise sensitive than AM system.The most of the nose accompanying the desired signal
accompanies it as AM. Thus the intelligence is contained in the frequency variation of signal. So
at the receiver remove all the amplitude variations without loss of the information content of the
desired signal. To remove the amplitude variations of the signal is the main function of the
limiter. At the output of the limiter stage, a constant amplitude signal is obtained even though the
amplitude of input signal may be varying.

Fig. Amplitude Limiter

Fig. Amplitude limiter transfer characteristics

Limiter is basically a clipper circuit which clips off the undesired amplitude variations of
the input signal. The input signal provides the bias for the FET circuit. Negative bias increases as
input increases and hence it lowers the gain of the amplifier for high amplitude of the input
signal and output voltage remains constant.
The basic function of the limiter is flat topping(Squaring off) the upper and lower
extremities of the signal.
Although the signal is distorted, it makes no difference as far as FM is considered, since
the information is contained in frequency variation and not in amplitude variation.

Use of AGC and Double Limiting

Sometimes it is quite practicable that average input signal amplitude may lie outside the
limiting range. As a result, further limiting becomes necessary. The solution for this is the use of
double limiter consisting two amplitude limiters in cascade. This gives satisfactory limiting
range.
An alternative to the used of second limiter is automatic gain control. The AGC ensures,
by reducing the gain for higher signal strengths, that the signal applied to the limiter is within the
limiting range of the limiter. This also prevents the overloading of the last IF amplifier stage.
 PULSE MODULATION

Introduction:

Pulse Modulation

 Carrier is a train of pulses

 Example: Pulse Amplitude Modulation (PAM), Pulse width modulation (PWM) ,
Pulse Position Modulation (PPM)

Types of Pulse Modulation:

 The immediate result of sampling is a pulse-amplitude modulation (PAM) signal

 PAM is an analog scheme in which the amplitude of the pulse is proportional to the
amplitude of the signal at the instant of sampling

 Another analog pulse-forming technique is known as pulse-duration modulation

(PDM). This is also known as pulse-width modulation (PWM)

 Pulse-position modulation is closely related to PDM

Pulse Amplitude Modulation:

In PAM, amplitude of pulses is varied in accordance with instantaneous value of

modulating signal.

PAM Generation:

The carrier is in the form of narrow pulses having frequency fc. The uniform
sampling takes place in multiplier to generate PAM signal. Samples are placed Ts sec
away from each other.
 Fig.12. PAM Modulator

 The circuit is simple emitter follower.

 In the absence of the clock signal, the output follows input.
 The modulating signal is applied as the input signal.
 Another input to the base of the transistor is the clock signal.
 The frequency of the clock signal is made equal to the desired carrier pulse train
frequency.
 The amplitude of the clock signal is chosen the high level is at ground level(0v) and
low level at some negative voltage sufficient to bring the transistor in cutoff region.

 When clock is high, circuit operates as emitter follower and the output follows in the
input modulating signal.
 When clock signal is low, transistor is cutoff and output is zero.
 Thus the output is the desired PAM signal.

PAM Demodulator:
 The PAM demodulator circuit which is just an envelope detector followed by a
second order op-amp low pass filter (to have good filtering characteristics) is as
shown below

Fig.13. PAM Demodulator

Pulse Width Modulation:
 In this type, the amplitude is maintained constant but the width of each pulse is varied
in accordance with instantaneous value of the analog signal.

 In PWM information is contained in width variation. This is similar to FM.

 In pulse width modulation (PWM), the width of each pulse is made directly
proportional to the amplitude of the information signal.

Pulse Position Modulation:

 In this type, the sampled waveform has fixed amplitude and width whereas the
position of each pulse is varied as per instantaneous value of the analog signal.

 PPM signal is further modification of a PWM signal.

PPM & PWM Modulator:

Fig.14. PWM & PPM Modulator

• The PPM signal can be generated from PWM signal.

• The PWM pulses obtained at the comparator output are applied to a mono stable multi
vibrator which is negative edge triggered.
 • Hence for each trailing edge of PWM signal, the monostable output goes high. It
remains high for a fixed time decided by its RC components.

• Thus as the trailing edges of the PWM signal keeps shifting in proportion with the
modulating signal, the PPM pulses also keep shifting.

• Therefore all the PPM pulses have the same amplitude and width. The information is
conveyed via changing position of pulses.

Fig.15. PWM & PPM Modulation waveforms

PWM Demodulator:

Fig.16. PWM Demodulator

  Transistor T1 works as an inverter.

 During time interval A-B when the PWM signal is high the input to transistor T2 is
low.

 Therefore, during this time interval T2 is cut-off and capacitor C is charged through
an R-C combination.

 During time interval B-C when PWM signal is low, the input to transistor T2 is high,
and it gets saturated.

 The capacitor C discharges rapidly through T2.The collector voltage of T2 during B-

C is low.

 Thus, the waveform at the collector of T2is similar to saw-tooth waveform whose
envelope is the modulating signal.

 Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.

PPM Demodulator:

Fig.17. PPM Demodulator

 The gaps between the pulses of a PPM signal contain the information regarding the
modulating signal.

 During gap A-B between the pulses the transistor is cut-off and the capacitor C gets
charged through R-C combination.

 During the pulse duration B-C the capacitor discharges through transistor and the
collector voltage becomes low.

 Thus, waveform across collector is saw-tooth waveform whose envelope is the

modulating signal.

 Passing it through 2nd order op-amp Low Pass Filter, gives demodulated signal.
 UNIT-1
Digital Pulse Modulation
Elements of Digital Communication Systems:

Fig. 1 Elements of Digital Communication Systems

1. Information Source and Input Transducer:

The source of information can be analog or digital, e.g. analog: audio or video
signal, digital: like teletype signal. In digital communication the signal produced by
this source is converted into digital signal which consists of 1′s and 0′s. For this we
need a source encoder.
2. Source Encoder:
In digital communication we convert the signal from source into digital signal
as mentioned above. The point to remember is we should like to use as few binary
digits as possible to represent the signal. In such a way this efficient representation
of the source output results in little or no redundancy. This sequence of binary digits
is called information sequence.

Source Encoding or Data Compression: the process of efficiently converting

the output of whether analog or digital source into a sequence of binary digits is
known as source encoding.

1
 3. Channel Encoder:
The information sequence is passed through the channel encoder. The
purpose of the channel encoder is to introduce, in controlled manner, some
redundancy in the binary information sequence that can be used at the receiver to
overcome the effects of noise and interference encountered in the transmission on
the signal through the channel.
For example take k bits of the information sequence and map that k bits to
unique n bit sequence called code word. The amount of redundancy introduced is
measured by the ratio n/k and the reciprocal of this ratio (k/n) is known as rate of
code or code rate.
4. Digital Modulator:
The binary sequence is passed to digital modulator which in turns convert the
sequence into electric signals so that we can transmit them on channel (we will see
channel later). The digital modulator maps the binary sequences into signal wave
forms , for example if we represent 1 by sin x and 0 by cos x then we will transmit sin
x for 1 and cos x for 0. ( a case similar to BPSK)
5. Channel:
The communication channel is the physical medium that is used for
transmitting signals from transmitter to receiver. In wireless system, this channel
consists of atmosphere , for traditional telephony, this channel is wired , there are
optical channels, under water acoustic channels etc.We further discriminate this
channels on the basis of their property and characteristics, like AWGN channel etc.
6. Digital Demodulator:
The digital demodulator processes the channel corrupted transmitted
waveform and reduces the waveform to the sequence of numbers that represents
estimates of the transmitted data symbols.
7. Channel Decoder:
This sequence of numbers then passed through the channel decoder which
attempts to reconstruct the original information sequence from the knowledge of
the code used by the channel encoder and the redundancy contained in the received
data

Note: The average probability of a bit error at the output of the decoder is a
measure of the performance of the demodulator – decoder combination.

8. Source Decoder:
At the end, if an analog signal is desired then source decoder tries to decode
the sequence from the knowledge of the encoding algorithm. And which results in
the approximate replica of the input at the transmitter end.

2
 9. Output Transducer:
Finally we get the desired signal in desired format analog or digital.

Advantages of digital communication:

3
Introduction to Pulse Modulation

What is the need for Pulse Modulation?

 Many Signals in Modern Communication Systems are digital

 Also, analog signals are transmitted digitally.
 Reduced distortion and improvement in signal to noise ratios.
 PAM, PWM, PPM, PCM and DM.
 In CW modulation schemes some parameter of modulated wave varies continuously with
message.
 In Analog pulse modulation some parameter of each pulse is modulated by a particular
sample value of the message.
 Pulse modulation is of two types
o Analog Pulse Modulation
 Pulse Amplitude Modulation (PAM)
 Pulse width Modulation (PWM)
 Pulse Position Modulation (PPM)
o Digital Pulse Modulation
 Pulse code Modulation (PCM)
 Delta Modulation (DM)

Pulse Code Modulation:

Three steps involved in conversion of analog signal to digital signal

 Sampling
 Quantization
 Binary encoding

4
 Fig. 2 Conversion of Analog Signal to Digital Signal

Note: Before sampling the signal is filtered to limit bandwidth.

Elements of PCM System:

Fig. 3 Elements of PCM System

Sampling:

 Process of converting analog signal into discrete signal.

 Sampling is common in all pulse modulation techniques

5
  The signal is sampled at regular intervals such that each sample is proportional to
amplitude of signal at that instant
 Analog signal is sampled every 𝑇𝑠 𝑆𝑒𝑐𝑠, called sampling interval. 𝑓𝑠=1/𝑇𝑆 is called
sampling rate or sampling frequency.
 𝑓𝑠=2𝑓𝑚 is Min. sampling rate called Nyquist rate. Sampled spectrum (𝜔) is repeating
periodically without overlapping.
 Original spectrum is centered at 𝜔=0 and having bandwidth of 𝜔𝑚. Spectrum can be
recovered by passing through low pass filter with cut-off 𝜔𝑚.
 For 𝑓𝑠<2𝑓𝑚 sampled spectrum will overlap and cannot be recovered back. This is
called aliasing.

Sampling methods:

 Ideal – An impulse at each sampling instant.

 Natural – A pulse of Short width with varying amplitude.
 Flat Top – Uses sample and hold, like natural but with single amplitude value.

Fig. 4 Types of Sampling

Sampling of band-pass Signals:

 A band-pass signal of bandwidth 2fm can be completely recovered from its samples.

Min. sampling rate =2×𝐵𝑎𝑛𝑑𝑤𝑖𝑑𝑡ℎ

=2×2𝑓𝑚=4𝑓𝑚

 Range of minimum sampling frequencies is in the range of 2×𝐵𝑊 𝑡𝑜 4×𝐵𝑊

Instantaneous Sampling or Impulse Sampling:

 Sampling function is train of spectrum remains constant impulses throughout

frequency range. It is not practical.

6
Natural sampling:

 The spectrum is weighted by a sinc function.

 Amplitude of high frequency components reduces.

Flat top sampling:

 Here top of the samples remains constant.

 In the spectrum high frequency components are attenuated due sinc pulse roll off.
This is known as Aperture effect.
 If pulse width increases aperture effect is more i.e. more attenuation of high
frequency components.

Sampling Theorem:

7
8
9
10
11
 Fig. 7 (a) Sampled version of signal x(t)
(b) Reconstruction of x(t) from its samples

12
PCM Generator:

13
Transmission BW in PCM:

14
PCM Receiver:

Quantization
 The quantizing of an analog signal is done by discretizing the signal with a number of
quantization levels.

15
  Quantization is representing the sampled values of the amplitude by a finite set of
levels, which means converting a continuous-amplitude sample into a discrete-time
signal
 Both sampling and quantization result in the loss of information.
 The quality of a Quantizer output depends upon the number of quantization levels
used.
 The discrete amplitudes of the quantized output are called as representation levels
or reconstruction levels.
 The spacing between the two adjacent representation levels is called a quantum or
step-size.
 There are two types of Quantization
o Uniform Quantization
o Non-uniform Quantization.
 The type of quantization in which the quantization levels are uniformly spaced is
termed as a Uniform Quantization.
 The type of quantization in which the quantization levels are unequal and mostly the
relation between them is logarithmic, is termed as a Non-uniform Quantization.

Uniform Quantization:

• There are two types of uniform quantization.

– Mid-Rise type
– Mid-Tread type.
• The following figures represent the two types of uniform quantization.

• The Mid-Rise type is so called because the origin lies in the middle of a raising part of
the stair-case like graph. The quantization levels in this type are even in number.
• The Mid-tread type is so called because the origin lies in the middle of a tread of the
stair-case like graph. The quantization levels in this type are odd in number.
• Both the mid-rise and mid-tread type of uniform quantizer is symmetric about the
origin.

16
Quantization Noise and Signal to Noise ratio in PCM System:

17
18
19
Derivation of Maximum Signal to Quantization Noise Ratio for Linear Quantization:

20
Non-Uniform Quantization:

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

21
Companding PCM System:
• Non-uniform quantizers are difficult to make and expensive.
• An alternative is to first pass the speech signal through nonlinearity before
quantizing with a uniform quantizer.
• The nonlinearity causes the signal amplitude to be compressed.
– The input to the quantizer will have a more uniform distribution.
• At the receiver, the signal is expanded by an inverse to the nonlinearity.
• The process of compressing and expanding is called Companding.

22
23
Differential Pulse Code Modulation (DPCM):

Redundant Information in PCM:

24
25
26
Line Coding:

In telecommunication, a line code is a code chosen for use within a communications

system for transmitting a digital signal down a transmission line. Line coding is often used
for digital data transport.

The waveform pattern of voltage or current used to represent the 1s and 0s of a

digital signal on a transmission link is called line encoding. The common types of

27
line encoding are unipolar, polar, bipolar and Manchester encoding. Line codes are used
commonly in computer communication networks over short distances.

28
Time Division Multiplexing:

29
 TDM is immune to nonlinearities in the channel as a source of crosstalk. The reason
for this behaviour is that different message signals are not simultaneously applied to the
channel.

Introduction to Delta Modulation

30
31
32
33
34
35
Condition for Slope overload distortion occurrence:
Slope overload distortion will occur if

36
Expression for Signal to Quantization Noise power ratio for Delta
Modulation:

37
38
39
 UNIT-5

ELEMENTS OF DETECTION THEORY & DIGITAL MODULATION TECHNIQUES

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 ones.

There are many types of digital modulation techniques and we can even use a combination of these
techniques as well. In this chapter, we will be discussing the most prominent digital modulation
techniques.

if the information signal is digital and the amplitude (lV of the carrier is varied proportional to
the information signal, a digitally modulated signal called amplitude shift keying (ASK) is
produced.
If the frequency (f) is varied proportional to the information signal, frequency shift keying (FSK) is
produced, and if the phase of the carrier (0) is varied proportional to the information signal,
phase shift keying (PSK) is produced. If both the amplitude and the phase are varied proportional to
the information signal, quadrature amplitude modulation (QAM) results. ASK, FSK, PSK, and
QAM are all forms of digital modulation:

a simplified block diagram for a digital modulation system.

Amplitude Shift Keying

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.

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.

Following is the diagram for ASK modulated waveform along with its input.

1
 Any modulated signal has a high frequency carrier. The binary signal when ASK is modulated,
gives a zero value for LOW input and gives the carrier output for HIGH input.
Mathematically, amplitude-shift keying is

where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts)
A/2 = unmodulated carrier amplitude (volts)
ωc= analog carrier radian frequency (radians per second, 2πfct)

In above Equation, the modulating signal [vm(t)] is a normalized binary waveform, where + 1 V =
logic 1 and -1 V = logic 0. Therefore, for a logic 1 input, vm(t) = + 1 V, Equation 2.12 reduces to

Mathematically, amplitude-shift keying is (2.12) where vask(t) = amplitude-shift keying wave

vm(t) = digital information (modulating) signal (volts) A/2 = unmodulated carrier amplitude (volts)

2
ωc= analog carrier radian frequency (radians per second, 2πfct) In Equation 2.12, the modulating
signal [vm(t)] is a normalized binary waveform, where + 1 V = logic 1 and -1 V = logic 0.
Therefore, for a logic 1 input, vm(t) = + 1 V, Equation 2.12 reduces to and for a logic 0 input, vm(t)
= -1 V,Equation reduces to

Thus, the modulated wave vask(t),is either A cos(ωct) or 0. Hence, the carrier is either "on “or
"off," which is why amplitude-shift keying is sometimes referred to as on-off keying (OOK).
it can be seen that for every change in the input binary data stream, there is one change in the ASK
waveform, and the time of one bit (tb) equals the time of one analog signaling element (t,).
B = fb/1 = fb baud = fb/1 = fb

Example :
Determine the baud and minimum bandwidth necessary to pass a 10 kbps binary signal using
amplitude shift keying. 10Solution For ASK, N = 1, and the baud and minimum bandwidth are
determined from Equations 2.11 and 2.10, respectively:

B = 10,000 / 1 = 10,000
baud = 10, 000 /1 = 10,000
The use of amplitude-modulated analog carriers to transport digital information is a relatively low-
quality, low-cost type of digital modulation and, therefore, is seldom used except for very low-
speed telemetry circuits.
ASK TRANSMITTER:

3
The input binary sequence is applied to the product modulator. The product modulator amplitude
modulates the sinusoidal carrier .it passes the carrier when input bit is ‘1’ .it blocks the carrier when
input bit is ‘0.’

Coherent ASK DETECTOR:

FREQUENCYSHIFT KEYING
The frequency of the output signal will be either high or low, depending upon the input data
applied.

Frequency Shift Keying (FSK) is the digital modulation technique in which the frequency of the
carrier signal varies according to the discrete digital changes. FSK is a scheme of frequency
modulation.

Following is the diagram for FSK modulated waveform along with its input.

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.

FSK is a form of constant-amplitude angle modulation similar to standard frequency modulation

(FM) except the modulating signal is a binary signal that varies between two discrete voltage levels
rather than a continuously changing analog waveform.Consequently, FSK is sometimes called
binary FSK (BFSK). The general expression for FSK is

4
where

vfsk(t) = binary FSK waveform

Vc = peak analog carrier amplitude (volts)

fc = analog carrier center frequency (hertz)

f=peak change (shift)in the analog carrier frequency(hertz)

vm(t) = binary input (modulating) signal (volts)

From Equation 2.13, it can be seen that the peak shift in the carrier frequency ( f) is proportional to
the amplitude of the binary input signal (vm[t]), and the direction of the shift is determined by the
polarity.

The modulating signal is a normalized binary waveform where a logic 1 = + 1 V and a logic 0 = -1
V. Thus, for a logic l input, vm(t) = + 1, Equation 2.13 can be rewritten as

For a logic 0 input, vm(t) = -1, Equation becomes

With binary FSK, the carrier center frequency (fc) is shifted (deviated) up and down in the
frequency domain by the binary input signal as shown in Figure 2-3.

FIGURE: FSK in the frequency domain

5
As the binary input signal changes from a logic 0 to a logic 1 and vice versa, the output frequency
shifts between two frequencies: a mark, or logic 1 frequency (fm), and a space, or logic 0 frequency
(fs). The mark and space frequencies are separated from the carrier frequency by the peak frequency
deviation ( f) and from each other by 2 f.

Frequency deviation is illustrated in Figure 2-3 and expressed mathematically as

f = |fm – fs| / 2 (2.14)

where f = frequency deviation (hertz)

|fm – fs| = absolute difference between the mark and space frequencies (hertz)

Figure 2-4a shows in the time domain the binary input to an FSK modulator and the corresponding
FSK output.

When the binary input (fb) changes from a logic 1 to a logic 0 and vice versa, the FSK output
frequency shifts from a mark ( fm) to a space (fs) frequency and vice versa.

In Figure 2-4a, the mark frequency is the higher frequency (fc + f) and the space frequency is the
lower frequency (fc - f), although this relationship could be just the opposite.

Figure 2-4b shows the truth table for a binary FSK modulator. The truth table shows the input and
output possibilities for a given digital modulation scheme.

6
FSK Bit Rate, Baud, and Bandwidth

In Figure 2-4a, it can be seen that the time of one bit (tb) is the same as the time the FSK output is a
mark of space frequency (ts). Thus, the bit time equals the time of an FSK signaling element, and
the bit rate equals the baud.

The baud for binary FSK can also be determined by substituting N = 1 in Equation 2.11:

baud = fb / 1 = fb
The minimum bandwidth for FSK is given as
B= |(fs – fb) – (fm – fb)|

=|(fs– fm)| + 2fb

and since |(fs– fm)| equals 2 f, the minimum bandwidth can be approximated as
B= 2( f + fb) (2.15)

where
B= minimum Nyquist bandwidth (hertz)
f= frequency deviation |(fm– fs)| (hertz)
fb = input bit rate (bps)

Example 2-2

Determine (a) the peak frequency deviation, (b) minimum bandwidth, and (c) baud for a binary
FSK signal with a mark frequency of 49 kHz, a space frequency of 51 kHz, and an input bit rate of
2 kbps.

Solution

a. The peak frequency deviation is determined from Equation 2.14:

f= |149kHz - 51 kHz| / 2 =1 kHz

b. The minimum bandwidth is determined from Equation 2.15:
B = 2(100+ 2000)
=6 kHz

7
c. For FSK, N = 1, and the baud is determined from Equation 2.11 as
baud = 2000 / 1 = 2000

FSK TRANSMITTER:
Figure 2-6 shows a simplified binary FSK modulator, which is very similar to a conventional FM
modulator and is very often a voltage-controlled oscillator (VCO).The center frequency (fc) is
chosen such that it falls halfway between the mark and space frequencies.

A logic 1 input shifts the VCO output to the mark frequency, and a logic 0 input shifts the VCO
output to the space frequency. Consequently, as the binary input signal changes back and forth
between logic 1 and logic 0 conditions, the VCO output shifts or deviates back and forth between
the mark and space frequencies.

FIGURE 2-6 FSK modulator

A VCO-FSK modulator can be operated in the sweep mode where the peak frequency deviation is
simply the product of the binary input voltage and the deviation sensitivity of the VCO.

8
With the sweep mode of modulation, the frequency deviation is expressed mathematically as
f = vm(t)kl (2-19)

vm(t) = peak binary modulating-signal voltage (volts)

kl = deviation sensitivity (hertz per volt).
FSK Receiver
FSK demodulation is quite simple with a circuit such as the one shown in Figure 2-7.

FIGURE 2-7 Noncoherent FSK demodulator

The FSK input signal is simultaneously applied to the inputs of both bandpass filters (BPFs)
through a power splitter.The respective filter passes only the mark or only the space frequency on to
its respective envelope detector.The envelope detectors, in turn, indicate the total power in each
passband, and the comparator responds to the largest of the two powers.This type of FSK detection
is referred to as noncoherent detection.

Figure 2-8 shows the block diagram for a coherent FSK receiver.The incoming FSK signal is
multiplied by a recovered carrier signal that has the exact same frequency and phase as the
transmitter reference.

However, the two transmitted frequencies (the mark and space frequencies) are not generally
continuous; it is not practical to reproduce a local reference that is coherent with both of them.
Consequently, coherent FSK detection is seldom used.

FIGURE 2-8 Coherent FSK demodulator

9
PHASESHIFT KEYING:
The phase of the output signal gets shifted depending upon the input. These are mainly of two
types, namely BPSK and QPSK, according to the number of phase shifts. The other one is DPSK
which changes the phase according to the previous value.

Phase Shift Keying (PSK) is the digital modulation technique in which the phase of the carrier
signal is changed by varying the sine and cosine inputs at a particular time. PSK technique is widely
used for wireless LANs, bio-metric, contactless operations, along with RFID and Bluetooth
communications.

PSK is of two types, depending upon the phases the signal gets shifted. They are −

Binary Phase Shift Keying (BPSK)

This is also called as 2-phase PSK (or) Phase Reversal Keying. In this technique, the sine wave
carrier takes two phase reversals such as 0° and 180°.

BPSK is basically a DSB-SC (Double Sideband Suppressed Carrier) modulation scheme, for
message being the digital information.

Following is the image of BPSK Modulated output wave along with its input.

10
Binary Phase-Shift Keying
The simplest form of PSK is binary phase-shift keying (BPSK), where N = 1 and M =
2.Therefore, with BPSK, two phases (21 = 2) are possible for the carrier.One phase represents a
logic 1, and the other phase represents a logic 0. As the input digital signal changes state (i.e., from
a 1 to a 0 or from a 0 to a 1), the phase of the output carrier shifts between two angles that are
separated by 180°.

Hence, other names for BPSK are phase reversal keying (PRK) and biphase modulation. BPSK
is a form of square-wave modulation of a continuous wave (CW) signal.

FIGURE 2-12 BPSK transmitter

BPSK TRANSMITTER:

Figure 2-12 shows a simplified block diagram of a BPSK transmitter. The balanced modulator acts
as a phase reversing switch. Depending on the logic condition of the digital input, the carrier is
transferred to the output either in phase or 180° out of phase with the reference carrier oscillator.

Figure 2-13 shows the schematic diagram of a balanced ring modulator. The balanced modulator
has two inputs: a carrier that is in phase with the reference oscillator and the binary digital data. For
the balanced modulator to operate properly, the digital input voltage must be much greater than the
peak carrier voltage.

This ensures that the digital input controls the on/off state of diodes D1 to D4. If the binary input is
a logic 1(positive voltage), diodes D 1 and D2 are forward biased and on, while diodes D3 and D4

11
are reverse biased and off (Figure 2-13b). With the polarities shown, the carrier voltage is
developed across transformer T2 in phase with the carrier voltage across T

1. Consequently, the output signal is in phase with the reference oscillator.

If the binary input is a logic 0 (negative voltage), diodes Dl and D2 are reverse biased and off,
while diodes D3 and D4 are forward biased and on (Figure 9-13c). As a result, the carrier voltage is
developed across transformer T2 180° out of phase with the carrier voltage across T 1.

FIGURE 9-13 (a) Balanced ring modulator; (b) logic 1 input; (c) logic 0 input

12
 FIGURE 2-14 BPSK modulator: (a) truth table; (b) phasor diagram; (c) constellation
diagram
BANDWIDTH CONSIDERATIONS OF BPSK:

In a BPSK modulator. the carrier input signal is multiplied by the binary data.
If + 1 V is assigned to a logic 1 and -1 V is assigned to a logic 0, the input carrier (sin ωct) is
multiplied by either a + or - 1 .

The output signal is either + 1 sin ωct or -1 sin ωct the first represents a signal that is in phase with
the reference oscillator, the latter a signal that is 180° out of phase with the reference
oscillator.Each time the input logic condition changes, the output phase changes.
Mathematically, the output of a BPSK modulator is proportional to

BPSK output = [sin (2πfat)] x [sin (2πfct)] (2.20)

where

13
fa = maximum fundamental frequency of binary input (hertz)
fc = reference carrier frequency (hertz)

Solving for the trig identity for the product of two sine functions,

0.5cos[2π(fc – fa)t] – 0.5cos[2π(fc + fa)t]

Thus, the minimum double-sided Nyquist bandwidth (B) is

fc + fa fc + fa
-fc + f a
-(fc + fa) or

2fa
and because fa = fb / 2, where fb = input bit rate,

where B is the minimum double-sided Nyquist bandwidth.

Figure 2-15 shows the output phase-versus-time relationship for a BPSK waveform. Logic 1 input
produces an analog output signal with a 0° phase angle, and a logic 0 input produces an analog
output signal with a 180° phase angle.

As the binary input shifts between a logic 1 and a logic 0 condition and vice versa, the phase of the
BPSK waveform shifts between 0° and 180°, respectively.

BPSK signaling element (ts) is equal to the time of one information bit (tb), which indicates that the
bit rate equals the baud.

FIGURE 2-15 Output phase-versus-time relationship for a BPSK modulator

14
Example:

For a BPSK modulator with a carrier frequency of 70 MHz and an input bit rate of 10 Mbps,
determine the maximum and minimum upper and lower side frequencies, draw the output spectrum,
de-termine the minimum Nyquist bandwidth, and calculate the baud..

Solution

Substituting into Equation 2-20 yields

output = [sin (2πfat)] x [sin (2πfct)]; fa = fb / 2 = 5 MHz

=[sin 2π(5MHz)t)] x [sin 2π(70MHz)t)]

=0.5cos[2π(70MHz – 5MHz)t] – 0.5cos[2π(70MHz + 5MHz)t]
lower side frequency upper side frequency

Minimum lower side frequency (LSF):

LSF=70MHz - 5MHz = 65MHz

Maximum upper side frequency (USF):

USF = 70 MHz + 5 MHz = 75 MHz

Therefore, the output spectrum for the worst-case binary input conditions is as follows: The
minimum Nyquist bandwidth (B) is

B = 75 MHz - 65 MHz = 10 MHz

and the baud = fb or 10 megabaud.

BPSK receiver:.
Figure 2-16 shows the block diagram of a BPSK receiver.

15
The input signal maybe+ sin ωct or - sin ωct .The coherent carrier recovery circuit detects and
regenerates a carrier signal that is both frequency and phase coherent with the original transmit
carrier.
The balanced modulator is a product detector; the output is the product d the two inputs (the BPSK
signal and the recovered carrier).
The low-pass filter (LPF) operates the recovered binary data from the complex demodulated signal.

FIGURE 2-16 Block diagram of a BPSK receiver

Mathematically, the demodulation process is as follows.

For a BPSK input signal of + sin ωct (logic 1), the output of the balanced modulator is
output = (sin ωct )(sin ωct) = sin2ωct (2.21)
or sin2ωct = 0.5(1 – cos 2ωct) = 0.5 - 0.5cos 2ωct
filtered out
leaving output = + 0.5 V = logic 1
It can be seen that the output of the balanced modulator contains a positive voltage (+[1/2]V) and a
cosine wave at twice the carrier frequency (2 ωct ).

The LPF has a cutoff frequency much lower than 2 ωct, and, thus, blocks the second harmonic of
the carrier and passes only the positive constant component. A positive voltage represents a
demodulated logic 1.

For a BPSK input signal of -sin ωct (logic 0), the output of the balanced modulator is

output = (-sin ωct )(sin ωct) = sin2ωct

or
sin2ωct = -0.5(1 – cos 2ωct) = 0.5 + 0.5cos 2ωct

filtered out

16
leaving
output = - 0.5 V = logic 0
The output of the balanced modulator contains a negative voltage (-[l/2]V) and a cosine wave at
twice the carrier frequency (2ωct).
Again, the LPF blocks the second harmonic of the carrier and passes only the negative constant
component. A negative voltage represents a demodulated logic 0.

QUADRATURE PHASE SHIFT KEYING (QPSK):

This is the phase shift keying technique, in which the sine wave carrier takes four phase reversals
such as 0°, 90°, 180°, and 270°.

If this kind of techniques are further extended, PSK can be done by eight or sixteen values also,
depending upon the requirement. The following figure represents the QPSK waveform for two bits
input, which shows the modulated result for different instances of binary inputs.

QPSK is a variation of BPSK, and it is also a DSB-SC (Double Sideband Suppressed Carrier)
modulation scheme, which sends two bits of digital information at a time, called as bigits.
Instead of the conversion of digital bits into a series of digital stream, it converts them into bit-pairs.
This decreases the data bit rate to half, which allows space for the other users.

QPSK transmitter.
A block diagram of a QPSK modulator is shown in Figure 2-17Two bits (a dibit) are
clocked into the bit splitter. After both bits have been serially inputted, they are simultaneously
parallel outputted.

17
The I bit modulates a carrier that is in phase with the reference oscillator (hence the name "I" for "in
phase" channel), and theQ bit modulate, a carrier that is 90° out of phase.

For a logic 1 = + 1 V and a logic 0= - 1 V, two phases are possible at the output of the I balanced
modulator (+sin ωct and - sin ωct), and two phases are possible at the output of the Q balanced
modulator (+cos ωct), and (-cos ωct).

When the linear summer combines the two quadrature (90° out of phase) signals, there are four
possible resultant phasors given by these expressions: + sin ωct + cos ωct, + sin ωct - cos ωct, -sin
ωct + cos ωct, and -sin ωct - cos ωct.

Example:
For the QPSK modulator shown in Figure 2-17, construct the truthtable, phasor diagram, and
constellation diagram.

Solution

For a binary data input of Q = O and I= 0, the two inputs to the Ibalanced modulator are -1 and sin
ωct, and the two inputs to the Q balanced modulator are -1 and cos ωct.

Consequently, the outputs are

18
I balanced modulator =(-1)(sin ωct) = -1 sin ωct

Q balanced modulator =(-1)(cos ωct) = -1 cos ωct and the output of the linear summer is
-1 cos ωct - 1 sin ωct = 1.414 sin(ωct - 135°)

For the remaining dibit codes (01, 10, and 11), the procedure is the same. The results are shown in
Figure 2-18a.

FIGURE 2-18 QPSK modulator: (a) truth table; (b) phasor diagram; (c) constellation
diagram
In Figures 2-18b and c, it can be seen that with QPSK each of the four possible output phasors has
exactly the same amplitude. Therefore, the binary information must be encoded entirely in the
phase of the output signal

19
Figure 2-18b, it can be seen that the angular separation between any two adjacent phasors in QPSK
is 90°.Therefore, a QPSK signal can undergo almost a+45° or -45° shift in phase during
transmission and still retain the correct encoded information when demodulated at the receiver.

Figure 2-19 shows the output phase-versus-time relationship for a QPSK modulator.

FIGURE 2-19 Output phase-versus-time relationship for a PSK modulator

Bandwidth considerations of QPSK

With QPSK, because the input data are divided into two channels, the bit rate in either the I or the Q
channel is equal to one-half of the input data rate (fb/2) (one-half of fb/2 = fb/4).

QPSK RECEIVER:

The block diagram of a QPSK receiver is shown in Figure 2-21

The power splitter directs the input QPSK signal to the I and Q product detectors and the carrier
recovery circuit. The carrier recovery circuit reproduces the original transmit carrier oscillator
signal. The recovered carrier must be frequency and phase coherent with the transmit reference
carrier. The QPSK signal is demodulated in the I and Q product detectors, which generate the
original I and Q data bits. The outputs of the product detectors are fed to the bit combining circuit,
where they are converted from parallel I and Q data channels to a single binary output data stream.
The incoming QPSK signal may be any one of the four possible output phases shown in Figure 2-
18. To illustrate the demodulation process, let the incoming QPSK signal be -sin ωct + cos ωct.
Mathematically, the demodulation process is as follows.

20
 FIGURE 2-21 QPSK receiver

The receive QPSK signal (-sin ωct + cos ωct) is one of the inputs to the I product detector. The
other input is the recovered carrier (sin ωct). The output of the I product detector is

Again, the receive QPSK signal (-sin ωct + cos ωct) is one of the inputs to the Q product detector.
The other input is the recovered carrier shifted 90° in phase (cos ωct). The output of the Q product
detector is

21
The demodulated I and Q bits (0 and 1, respectively) correspond to the constellation diagram and
truth table for the QPSK modulator shown in Figure 2-18.

DIFFERENTIAL PHASE SHIFT KEYING (DPSK):

In DPSK (Differential Phase Shift Keying) the phase of the modulated signal is shifted relative to
the previous signal element. No reference signal is considered here. The signal phase follows the
high or low state of the previous element. This DPSK technique doesn’t need a reference oscillator.
The following figure represents the model waveform of DPSK.

It is seen from the above figure that, if the data bit is LOW i.e., 0, then the phase of the signal is not
reversed, but is continued as it was. If the data is HIGH i.e., 1, then the phase of the signal is
reversed, as with NRZI, invert on 1 (a form of differential encoding).

22
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.

The word binary represents two-bits. M simply 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.
DBPSK TRANSMITTER.:
Figure 2-37a shows a simplified block diagram of a differential binary phase-shift keying
(DBPSK) transmitter. An incoming information bit is XNORed with the preceding bit prior to
entering the BPSK modulator (balanced modulator).

For the first data bit, there is no preceding bit with which to compare it. Therefore, an initial
reference bit is assumed. Figure 2-37b shows the relationship between the input data, the XNOR
output data, and the phase at the output of the balanced modulator. If the initial reference bit is
assumed a logic 1, the output from the XNOR circuit is simply the complement of that shown.

In Figure 2-37b, the first data bit is XNORed with the reference bit. If they are the same, the XNOR
output is a logic 1; if they are different, the XNOR output is a logic 0. The balanced modulator
operates the same as a conventional BPSK modulator; a logic I produces +sin ωct at the output, and
A logic 0 produces –sin ωct at the output.

23
 FIGURE 2-37 DBPSK modulator (a) block diagram (b) timing diagram

BPSK RECEIVER:
Figure 9-38 shows the block diagram and timing sequence for a DBPSK receiver. The received
signal is delayed by one bit time, then compared with the next signaling element in the balanced
modulator. If they are the same. J logic 1(+ voltage) is generated. If they are different, a logic 0 (-
voltage) is generated. [f the reference phase is incorrectly assumed, only the first demodulated bit is
in error. Differential encoding can be implemented with higher-than-binary digital modulation
schemes, although the differential algorithms are much more complicated than for DBPS K.

The primary advantage of DBPSK is the simplicity with which it can be implemented. With
DBPSK, no carrier recovery circuit is needed. A disadvantage of DBPSK is, that it requires
between 1 dB and 3 dB more signal-to-noise ratio to achieve the same bit error rate as that of
absolute PSK.

24
 FIGURE 2-38 DBPSK demodulator: (a) block diagram; (b) timing sequence

COHERENT RECEPTION OF FSK:

The coherent demodulator for the coherent FSK signal falls in the general form of coherent
demodulators described in Appendix B. The demodulator can be implemented with two correlators
as shown in Figure 3.5, where the two reference signals are cos(27r f t) and cos(27r fit). They must
be synchronized with the received signal. The receiver is optimum in the sense that it minimizes the
error probability for equally likely binary signals. Even though the receiver is rigorously derived in
Appendix B, some heuristic explanation here may help understand its operation. When s 1 (t) is
transmitted, the upper correlator yields a signal 1 with a positive signal component and a noise
component. However, the lower correlator output 12, due to the signals' orthogonality, has only a
noise component. Thus the output of the summer is most likely above zero, and the threshold
detector will most likely produce a 1. When s2(t) is transmitted, opposite things happen to the two
correlators and the threshold detector will most likely produce a 0. However, due to the noise nature
that its values range from -00 to m, occasionally the noise amplitude might overpower the signal
amplitude, and then detection errors will happen. An alternative to Figure 3.5 is to use just one
correlator with the reference signal cos (27r f t) - cos(2s f2t) (Figure 3.6). The correlator in Figure
can be replaced by a matched filter that matches cos(27r fit) - cos(27r f2t) (Figure 3.7). All

25
implementations are equivalent in terms of error performance (see Appendix B). Assuming an
AWGN channel, the received signal is

where n(t) is the additive white Gaussian noise with zero mean and a two-sided power spectral
density A',/2. From (B.33) the bit error probability for any equally likely binary signals is

where No/2 is the two-sided power spectral density of the additive white Gaussian noise. For
Sunde's FSK signals El = Ez = Eb, pI2 = 0 (orthogonal). thus the error probability is

where Eb = A2T/2 is the average bit energy of the FSK signal. The above Pb is plotted in Figure 3.8
where Pb of noncoherently demodulated FSK, whose expression will be given shortly, is also
plotted for comparison.

26
 Figure: Pb of coherently and non-coherently demodulated FSK signal.

NONCOHERENT DEMODULATION AND ERROR PERFORMANCE:

Coherently FSK signals can be noncoherently demodulated to avoid the carrier recovery.
Noncoherently generated FSK can only be noncoherently demodulated. We refer to both cases as
noncoherent FSK. In both cases the demodulation problem becomes a problem of detecting signals
with unknown phases. In Appendix B we have shown that the optimum receiver is a quadrature
receiver. It can be implemented using correlators or equivalently, matched filters. Here we assume
that the binary noncoherent FSK signals are equally likely and with equal energies. Under these
assumptions, the demodulator using correlators is shown in Figure 3.9. Again, like in the coherent
case, the optimality of the receiver has been rigorously proved (Appendix B). However, we can
easily understand its operation by some heuristic argument as follows. The received signal
(ignoring noise for the moment) with an unknown phase can be written as

27
The signal consists of an in phase component A cos 8 cos 27r f t and a quadrature component A sin
8 sin 2x f,t sin 0. Thus the signal is partially correlated with cos 2s fit and partiah'y correlated with
sin 27r fit. Therefore we use two correlators to collect the signal energy in these two parts. The
outputs of the in phase and quadrature correlators will be cos 19 and sin 8, respectively. Depending
on the value of the unknown phase 8, these two outputs could be anything in (- 5, y). Fortunately
the squared sum of these two signals is not dependent on the unknown phase. That is

This quantity is actually the mean value of the statistics I? when signal si (t) is transmitted and noise
is taken into consideration. When si (t) is not transmitted the mean value of 1: is 0. The comparator
decides which signal is sent by checking these I?. The matched filter equivalence to Figure 3.9 is
shown in Figure 3.10 which has the same error performance. For implementation simplicity we can
replace the matched filters by bandpass filters centered at f and fi, respectively (Figure 3.1 1).
However, if the bandpass filters are not matched to the FSK signals, degradation to

various extents will result. The bit error probability can be derived using the correlator demodulator
(Appendix B). Here we further assume that the FSK signals are orthogonal, then from Appendix B
the error probability is

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 UNIT-6

Digital Modulation Tradeoff

BASE BAND SIGNAL RECEIVER:

Consider that a binary encoded signal consists of a time sequence of voltage levels +V or –V.
if there is a guard interval between the bits, the signal forms a sequence of positive and
negative pulses. in either case there is no particular interest in preserving the waveform of the
signal after reception .we are interested only in knowing within each bit interval whether the
transmitted voltage was +V or –V. With noise present, the receives signal and noise together
will yield sample values generally different from ±V. In this case, what deduction shall we
make from the sample value concerning the transmitted bit?

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PROBABILITY OF ERROR

Since the function of a receiver of a data transmission is to ditinguish the bit 1 from the bit 0
in the presence of noise, a most important charcteristic is the probability that an error will be
made in such a determination.we now calculate this error probabilty Pe for the integrate and
dump receiver of fig 11.1-2

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The probability of error pe, as given in eq.(11.2-3),is plotted in fig.11.2-2.note that pe decreases
rapidly as Es./η increases. The maximum value of pe is ½.thus ,even if the signal is entirely lost in
the noise so that any determination of the receiver is a sheer guess, the receiver cannot bi wrong
more than half the time on the average.

THE OPTIMUM FILTER:

In the receiver system of Fig 11.1-2, the signal was passed through a filter(integrator),so that at the
sampling time the signal voltage might be emphasized in comparison with the noise voltage. We are
naturally led to risk whether the integrator is the optimum filter for the purpose of minimizing the
probability of error. We shall find that the received signal contemplated in system of fig 11.1-2 the
integrator is indeed the optimum filter. However, before returning specifically to the integrator
receiver.

We assume that the received signal is a binary waveform. One binary digit is represented by
a signal waveformS1 (t) which persists for time T, while the4 other bit is represented by the
waveform S2(t) which also lasts for an interval T. For example, in the transmission at baseband, as
shown in fig 11.1-2 S1(t)=+V; for other modulation systems, different waveforms are transmitted.
for example for PSK signaling , S1(t)=Acosw0t and S2(t)=-Acosw0t;while for FSK,
S1(t)=Acos(w0+Ω)t.

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Hence probability of error is

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 In general, the impulsive response of the matched filter consists of p(t) rotated about t=0 and
then delayed long enough(i.e., a time T) to make the filter realizable. We may note in passing, that
any additional delay that a filter might introduce would in no way interfere with the performance of
the filter ,for both signal and noise would be delayed by the same amount, and at the sampling time
(which would need similarity to be delayed)the ratio of signal to noise would remain unaltered.

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From parseval’s theorem we have

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COHERENT RECEPTION: CORRELATION:

(11.6-1)

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 (11.6-2)
Where s1(t) is either s1(t) or s2(t),and wthere π is the constant of the integrator(i.e.,the integrator
output is 1/π times the integral of its input).we now compare these outputs with the matched filter
outputs.

Fig:11.6-1 Coherent system of signal reception

If h(t) is the impulsive response of the matched filter ,then the output of the matched filter v0(t) can
be found using the convolution integral. we have

(11.6-3)

The limits on the integral have been charged to 0 and T since we are interested in the filter response
to a bit which extends only over that interval. Using Eq.(11.4-4) which gives h(t) for the matched
filter, we have

(11.6-4)

(11.6-5)

sub 11.6-5 in 11.6-3

(11.6-6)

(11.6-7)

(11.6-8)

Thus s0(T) and n0(T), as calculated from eqs.(11.6-1) and (11.6-2) for the correlation receiver, and
as calculated from eqs.(11.6-7) and (11.6-8) for the matched filter receiver, are identical .hence the
performances of the two systems are identical. The matched filter and the correlator are not simply

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two distinct, independent techniques which happens to yield the same result. In fact they are two
techniques of synthesizing the optimum filter h(t)

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ERROR PROBABILITY OF BINARY PSK:

To realize a rule for making a decision in favor of symbol 1 or symbol 0,we partition the signal
space into two regions:

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