Communication Basics

Syllabus
EE-206-F COMMUNICATION SYSTEMS

L T P Class Work mark: 50
3 1 0 Theory marks: 100
Total marks: 150
Duration of Exam: 3 hr
NOTE: For setting up the question paper, Question No. 1 will be set up from all the four
sections
which will be compulsory and of short answer type. Two questions will be set from each of
the four
sections. The students have to attempt first common question, which is compulsory, and
one question
from each of the four sections. Thus students will have to attempt 5 questions out of 9
questions.
SECTION-A
INTRODUCTION TO COMMUNICATION SYSTEMS:
The essentials of a Communication system, modes and medias of Communication,
Classification of signals
and systems , Fourier Analysis of signals. Analog Communication & Digital Communication.
Basic
concepts of Modulation, Demodulators, Channels,Multiplexing & Demultiplexing.
SECTION-B
AMPLITUDE MODULATION:
Amplitude modulation, Generation of AM waves, Demodulation of AM waves, DSBSC,
Generation of
DSBSC waves, Coherent detection of DSBSC waves, single side band modulation, generation of
SSB
waves, demodulation of SSB waves, vestigial sideband modulation (VSB).
ANGLE MODULATION :
Basic definitions: Phase modulation (PM) & frequency modulation(FM), narrow band frequency
modulation, wideband frequency modulation, generation of FM waves, Demodulation of FM
waves.
SECTION C
PULSE ANALOG MODULATION: Sampling theory, sampling and hold circuits. Time
division (TDM)
and frequency division (FDM) multiplexing, pulse amplitude modulation (PAM), pulse time
modulation.
World Institute Of Technology
8km milestone ,Sohna Palwal Road , NH-71 B ,Sohna , Gurgaon ,Haryana.
Website : www.wit.net.in
E-mail : info@wit.net.in
PULSE DIGITAL MODULATION : Coding & Decoding techniques, Elements of pulse code
modulation, noise in PCM systems, Measure of information, channel capacity, channel capacity
of a PCM
system, differential pulse code modulation (DPCM). Delta modulation (DM)
SECTION D
DIGITAL MODULATION TECHNIQUES: ASK, FSK, BPSK, QPSK, M-ary PSK.
PC-PC data Communication
INTRODUCTION TO NOISE: External noise, internal noise, S/N ratio, noise figure.
TEXT BOOKS:
1. Communication systems (4th edn.): Simon Haykins; John Wiley & sons.
2. Communication systems: Singh & Sapre; TMH.
REFERENCE BOOKS:
1. Electronic Communication systems: Kennedy; TMH.
2. Communication Electronics: Frenzel; TMH.
3. Communication system: Taub & Schilling; TMH.
LECTURE NOTES OF COMMUNICATION SYSTEM

SECTION A
LECTURE1
Essentials of communication system modes and medias of Communication -:
Communication is the process of establishing connection or link between two points for
information exchange. The essential components of basic communication system are:
1) Information source
2) Transmitter
3) Channel
4) Reciever
Information
o/p
source
I/P
transducer
transmitter
channel
reciever
o/p
transducer
input message
Distortion
or noise
1) Information Source: The message or information originates in information source.There

can be various messages in form of words, group of words , codes, symbols, sound signal
etc. However out of these messages only desired message is selected and conveyed.
2) Input Transducer: A transducer is a device which converts one form of energy into
another . The message from information source may or may not be electrical in nature.
3) Transmitter: Modulation is a function of transmitter. In modulation message signal is
super imposed upon high frequency carrier signal. In short we can say that inside the
transmitter signal processing such as restriction of range of audio frequencies,
amplification and modulation are achieved.
4) Channel and Noise: The term channel means medium through which message travels
from transmitter and receiver .Noise is an unwanted signal which tend to interfere with
required signal . Noise signal is always random.
5) Reciever: The reproduction of original signal is accomplished by process known as
demodulation or detection .
6) Destination: it is the final stage which is used to convert an electrical message signal into
its original form.

LECTURE2
Classification of signals and system-:
DEFINATION-:
Signal in the present context means electrical manifestation of a physical process. Usually an
essence of information will be associated with a signal. Mathematical representation or
abstraction should also be possible for a signal such that a signal and its features can be classified
and analyzed.
Examples of a few signals-:
(a) Electrical equivalent of speech/voice as obtained at the output of a microphone.
(b) Electrical output of a transducer used to sense the temperature of a furnace.
(c) Stream of electrical pulses (digital) generated by a computer.
(d) Electrical output of a TV camera (video signal).
(e) Electrical waves received by the antenna of a radio/TV/communication receiver.
(f) ECG signal.
When a signal is viewed as electrical manifestation of a process, the signal is a function of
one or more independent variables. For all the examples cited above, the respective signals may
commonly be considered as function of time. So, a notation like the following may be used to
represent a signal:
s(a, b, c, t,..), where a, b, are the independent variables.
However, observe that a mere notation of a signal, say s(t), does not reveal all its features
and behavior and hence it may not be possible to analyze the signal effectively. Further,
processing and analyses of many signals may become easy if we can associate them, in some
cases even approximately, with mathematical functions that may be analyzed by well-developed
mathematical tools and techniques. The approximate representations, wherever adopted, are
usually justified by their ease of analysis or tractability or some other evidently rewarding
reason. For example, the familiar mathematical function
s(t) = A cos(t +)
is extensively used in the study, analysis and testing of several principles of

communication theory such as carrier modulation, signal sampling etc. However, one can very
well contest the fact that
s(t) = A cos(t +) hardly implies a physical process because of the following reasons:
(i)
No range oft is specified and hence, mathematically the range may be from - to
+. This implies that the innocent looking function s(t) should exist over the infinite
range of t, which is not true for any physical source if t represents time. So, some
range for t should be specified.
(ii) S (t), over a specified range oft, is a known signal in the sense that, over the range of t,
if we know the value of s(t) at say t = t0, and the values of A, and we know the value of s(t)
at any other time instant t. We say the signal s(t) is deterministic. In a sense, such a
mathematical function does not carry information.
While point (i) implies the need for rigorous and precise expression for a signal, point (ii)
underlines the usage of theories of mathematics for signals deterministic or non-deterministic
(random).
To illustrate this second point further, let us consider the description of s(t) = A cos t,
where t indicates time and = 2f implies angular frequency:
(a) Note that s(t) = A cos t , - < t < is a periodic function and hence can be expressed
by its exponential (complex) Fourier series. However, this signal has infinite energy
E, E= s2 (t) dt
Hence, theoretically, can not be expressed by Fourier Transformation.
(b) Let us now consider the following modified expression for s(t) which
may be a
closer representation of a physical signal:
s(t) = A cos t , 0 t <
= A.u (t). Cos t where u (t) is the unit step function, u(t) = 0, t < 0 and
u(t) = 1, t 0
If we further put an upper limit to t, say, s(t) = A cos t , t1 t t2, such a signal can be
easily generated by a physical source, but the frequency spectrum of s(t) will now be different
compared to the earlier forms. For simplicity in notation, depiction and understanding, we will,
at times, follow mathematical models for describing and understanding physical signals and
processes. We will, though, remember that such mathematical descriptions, while being elegant,
may show up some deviation from the actual behavior of a physical process. Henceforth, we will
mean the mathematical description itself as the signal, unless explicitly stated otherwise.
Now, we briefly introduce the major classes of signals that are of frequent interest in the
study of digital communications. There are several ways of classifying a signal and a few types
are named below.
TYPES OF SIGNAL-:
Energy signal: If, for a signal s(t)=
i.e. the energy of the signal is finite, the signal is called an energy signal. However, the
same signal may have large power. The voltage generated by lightning (which is of short
duration) is a close example of physical equivalent of a signal with finite energy but very large
power.
Power signal: A power signal, on the contrary, will have a finite power but may have
finite or infinite energy. Mathematically,
Note: While electrical signals, derived from physical processes are mostly energy signals, several
mathematical functions, usually deterministic, represent power signals.
Deterministic and random signals: If a signal s(t), described at t = t1 is sufficient for
determining the signal at t = t2 at which the signal also exists, then s(t) represents a deterministic
signal.
Example: s(t) = A cos t , T1 t T2
There are many signals that can best be described in terms of a probability and one may
not determine the signal exactly.
Example: (from real process) Noise voltage/current generated by a resistor.
Such signals are labeled as non-deterministic or random signals.
Continuous time signal: Assuming the independent variable t to represent time, if s(t) is
defined for all possible values of t between its interval of definition (or existence), T1 t T2.
Then the signal s(t) is a continuous time signal.
If a signal s(t) is defined only for certain values of t over an interval T1 t T2, it is a
discrete-time signal. A set of sample values represent a discrete time signal.
Continuous time signal: Assuming the independent variable t to represent time, if s(t) is
defined for all possible values of t between its interval of definition (or existence), T1 t T2.
Then the signal s(t) is a continuous time signal.
If a signal s(t) is defined only for certain values of t over an interval T1 t T2, it is a
discrete-time signal. A set of sample values represent a discrete time signal.
Periodic signal: If s(t) = s(t + T), for entire range of t over which the signal s(t) is
defined and T is a constant, s(t) is said to be periodic or repetitive. T indicates the period of the
signal and 1T is its frequency of repetition.
Example: s(t) = A cos t , - t , where T = 2/.
Analog signal: If the magnitudes of a real signal s(t) over its range of definition, T1 t
T2, are real numbers (there are infinite such values) within a finite range, say, Smin S(t)
Smax, the signal is analog.
A digital signal s(t), on the contrary, can assume only any of a finite number of values.
Usually, a digital signal implies a discrete-time, discrete-amplitude signal.
LECTURE 3-:
FOURIER ANALYSIS OF SIGNAL-:
Fourier analysis takes a signal and represents it either as a series of cosines (real part)
and sines (imaginary part) or as a cosine with phase (modulus and phase form). As an illustration
we will look at Fourier analysing the sum of the two sine waves shown below. The resultant
summed signal is shown in the third graph.


If we now carry out a Fourier Analysis of the combined signal then

we obtain the following result.
.
The amplitude shown is exactly half of the original constituent sine waves.
This is to make so called half range analysis compatible with full range analyses.This is because
Fourier analysis uses cosines and sines. It is cosines, not the sines, which
are the basic reference.
Signal Duration Effects
If we have data taken over a longer period then the frequency spacing will be narrower. In many
cases this will assist the problem but if there is no exact match the same phenomenonwill arise.
Fourier nalysis tells us the amplitude and phase of that set of cosines which have the same
duration as the original signal. A Fourier analysis shows the (half) amplitudes and phases of the
constituent cosine waves that exist for the whole duration of that part of the signal that has been
analysed then so does the Fourier transformed signal. Fourier analysis by itself does nothing to
remove or minimise the effects of noise. Thus simple Fourier analysis is not suitable for random
data, but it is for signals such as transients and complicated or simple periodic signals such as
those generated by an engine running at a constant speed.
Mathematically,
We will not go into all the mathematical niceties except to see that a Fourier series could be
written in the forms below. In real and imaginary terms we have
The above forms are a slightly unusual way of expressing the Fourier expansion. For instance
is in degrees. More significantly the product n k f t is shown explicitly.
However the point of using n k f t explicitly above is to indicate that nothing in the Fourier
expansion inhibits the choice of actual frequency at which we evaluate the Fourier coefficients.
Fourier Representation of continuous time signals

Any periodic signal f(t) can be represented with a set of complex
exponentials as shown below.
The exponential terms are orthogonal to each other because
the energy of these signals is unity since

Representing a signal in terms of its exponential Fourier series
components is called Fourier Analysis.
The weights of the exponentials are calculated as Extending this representation to aperiodic
signals:
Extending this representation to aperiodic signals:

When T and 0 0, the sum becomes an integral and 0
becomes continuous.
The resulting represention is termed as the Fourier Transform
(F()) and is given by
The signal f(t) can recovered from F() as
Some Important Functions

Delta function is a very important signal in signal analysis. It is
defined as

The Diac delta function is also called the Impulse function.

This function can be represented as the limiting function of a
number of sampling functions:
1. Gaussian Pulse
2. Triangular Pulse
3. Exponential Pulse
4. Sampling Function

5. Sampling Square function
The unit step function is another important function signal

processing. It is defined by
The Fourier transform of the unit step can be found only in the
limit.

LECTURE 4-:
Analog Communication & Digital Communication. Basic Concepts of Modulation,
Demodulators-:
Communication is the process of establishing connection or link between two points for
information exchange. The essential components of basic communication system are:
1)Information source
2) Transmitter
3) Channel
4)Reciever
Information
o/p
source
I/P
transducer
transmitter
input message
msg
channel
reciever
o/p
transducer
Distortion
or noise
1) Information Source: The message or information originates in information source.There

can be various messages in form of words, group of words , codes, symbols, sound signal etc.
However out of these messages only desired message is selected and conveyed.
2) input Transducer: A transducer is a device which converts one form of energy into
another . The message from information source may or may not be electrical in nature.
3) Transmitter: Modulation is a function of transmitter. In modulation message signal is
super imposed upon high frequency carrier signal. In short we can say that inside the
transmitter signal processing such as restriction of range of audio frequencies, amplification
and modulation are achieved.

4) Channel and Noise: The term channel means medium through which message travels
from transmitter and receiver .Noise is an unwanted signal which tend to interfere with
required signal . Noise signal is always random.
5) Reciever: The reproduction of original signal is accomplished by process known as
demodulation or detection .
6)Destination: it is the final stage which is used to convert an electrical message
signal
into its original form.
DIGITAL COMMUNICATION-:
Block Schematic Description of a Digital Communication System
In the simplest form, a transmission-reception system is a three-block system, consisting of a) a
transmitter,
b) a transmission medium
c) a receiver.
If we think of a combination of the transmission device and reception device in the form of a
transceiver and if (as is usually the case) the transmission medium allows signal both ways, we
are in a position to think of a both-way (bi-directional). In a simple form, again consists of three
different entities, viz. a transmitter, a communication channel and a receiver. A digital
communication system has several distinguishing features when compared with an analog
communication system. Both analog (such as voice signal) and digital signals (such as data
generated by computers) can be communicated over a digital transmission system.
A key feature of a digital communication system is that a sense of information with appropriate
unit of measure, is associated with such signals.The overall purpose of the digital communication
system is to collect information from the source and carry out necessary electronic signal
processing such that the information can be delivered to the end user (information sink) with
acceptable
quality.

It is possible to expand our basic three-entity description of a digital communication

system in multiple ways. For example, Fig. shows a somewhat elaborate block diagram explicitly
showing the important processes of modulation-demodulation source coding-decoding and channel
encoding-decoding
To elaborate this potentially useful style of representation, let us note that we have hardly discussed
about the third entity of our model, viz. the channel. One can define several types of channel. For
example, the channel in Fig should more appropriately be called as a modulation channel with an
understanding that the actual transmission medium (called physical channel), any electromagnetic (or
other wise) transmission reception operations, amplifiers at the transmission and reception ends and
any other necessary signal processing units are combined together to form this modulation channel.
BASIC CONCEPT OF MODULATORS AND DEMODULATORS

The design of the modulator can trade resolution in time for resolution in amplitude in such a
way that imprecise analog circuits can be tolerated. The use of high frequency
Modulation and demodulation eliminates the need for abrupt cutoffs in the analog
antialiasing filter at the input to the A/D converter, as well as in the filters that smooth
the analog output of the D/A converter. A digital filter smoothes the output of the modulator,
attenuating noise, interference, and high-frequency components of the signal before they can
alias into the signal band when the code is re sampled at the Nyquist rate. Another digital filter
interpolates the code in the decoder to a high word rate before it is demodulated to analog form.
Oversampling converters make extensive use of digital signal processing taking
advantage of the fact that fine-line VLSI is better suited for providing fast digital circuits
than for providing precise analog circuits. Because their sampling rate usually needs to be
several orders of magnitude higher than the Nyquist rate, oversampling methods are best suited
for relatively low-frequency signals.
Applications
1) Digital audio
2) Digital telephony
3) Instrumentation.
4) Video
5) Radar systems
An important difference between conventional converters and oversampling ones
Involve testing and specifying their performance. With conventional converters there is a
One-to-one correspondence between input and output sample values and hence one can describe
their accuracy by comparing the values of corresponding input and output samples. In contrast
there is no similar correspondence in oversampling converters because they inherently include
digital low-pass filters, and hence each input sample value contributes to a whole train of output
samples. Consequently, it has been useful to borrow techniques from communication technology
to describe the performance of oversampling converters. Thus we measure their root-meansquare (rms) noise under various conditions, the distortion they introduce into sinusoidal signals,
and their frequency responses. An important task in designing an oversampling converter is
therefore the calculation of rms values of modulation noise and its spectral density.
Some Alternative Modulator Structures

1 .Error Feedback-: Noise-shaping quantization was first introduced
using the structure shown. In this circuit, the difference between the input
Quantizer with error feedback

and the output of the quantizer is a measure of the quantization error, which is fed back
and subtracted from the next input sample. The circuit is algebraically equivalent to the
AI circuit in, but it has the serious practical disadvantage that inaccuracies in
that the circuit can be used as a demodulator because there the processing is performed digitally.
The circuit can be generalized by replacing the delay with a prediction filter.
2 Cascaded Modulators-: The performance of a modulator can be improved
by taking a measure of its noise digitizing that measure in a second modulator,
and combining the output of the two modulators in a way that cancels the noise of the first
modulator. The output of the integrator in the first modulator is fed to the second modulator. Its
output is digitally differentiated and subtracted from the output of the first modulator to provide
the net output of the circuit. We have used e' to denote the quantization error in the first
modulator and e that of the second one. When scaling factors are ignored, it can be shown that
the net output of the circuit may be expressed in the form
where g is a measure of the accuracy of the error cancellation. It depends on a number of

parameters, including the precision of component values and the low-frequency gain of the first
integrator. Ideally, g is unity; then the noise of the first modulator does not contribute to the
output. The remaining noise is the second difference of the quantization error.

from the second modulator: It is in the same form as the noise of a second-order AL modulator
given .An ideal second-order modulator oversampling by a factor of 120. At this sampling rate
the noise from the first-order modulation is given by the ordinate of point . It is at -57 dB. In
practice this requirement is tightened by needs to scaling signal amplitudes in practical circuits.
The need for such precision is alleviated by raising the sampling rate: Because of the difficulty in
obtaining adequate precision, the noise from these cascaded circuits is often dominated by the
noise from the first stage.
Demodulating Signals -:
This circuit a digital filter interpolates sample values of the input signal in order to raise the word
rate well above the Nyquist rate. A demodulator then truncates the words and
Converts them to analog form at the high sample rate. In most applications it is advantageous to
raise the word rate of the signal in stages at the encoder. The output of the filter resembles a
PCM encoding of the signal at 32 kHz. The next stage is a linear interpolation that inserts three
new values between each adjacent pair of 32-kHz samples, raising the word rate to 128 kHz. The
words enter a register from which they feed the demodulator at 1 MHz; each word repeats eight
times. The demodulator rounds off the code to single-bit words, converts them to analog levels,
and smoothes these with an analog filter. The single-bit quantization occurs in a feedback circuit
that shapes the spectrum of the quantization noise, moving most of the power far above the
signal band. The 1 MHz demodulation rate is sufficiently high that a very simple analog filter
will smooth the noise. The filtering actions that are inherent in the interpolation that raises the
word rate from 8 kHz to 1 MHz smooth out sampling images of the signal, leaving only those
adjacent to the new sampling rate 1 MHz and its harmonics. Figure illustrates this action:
(a) Represents the spectral density of the baseband signal,
(b) Is spectral density when sampled
at the Nyquist rate,
(c) Is the frequency response of the low-pass filter including the
Sine response of the holding register,
Both (d) and (e) represent the output spectrum of the
Low-pass filter,
(f) is the sine2 response of linear interpolation,
(g) is the result of this interpolation,
(h) is the frequency response of the final holding register, and
(i) is the spectral density of its output.
The filter requirements for attenuating sampling images of the signal at the decoder
are usually less stringent than are the requirements for preventing aliasing at the encoder.

FIG shows-:Spectral densities of signals, and the frequency response of filters

used for interpolating sample values: (a) spectral density of
the signal; (b) spectral density of the sampled signal; (c) low pass
filter characteristic; (d, e) spectral density of the filter output
on different frequency scales (f) frequency response of linear
interpolation (g) spectral density of the interpolated signal;
(h) frequency response of the holding register (i) spectral density
of the held signal.

LECTURE-5
CHANNEL MULTIPLEXING & DEMULTIPLEXING
Detailed explaination later in lecture 17

LECTURE-6
REVISION & ASSIGNMENT
Q1) Give three examples of types of signals that a source may generate.
Q2) Explain basic block diagram of analog communication & digital communication.
Q3) Classify signals into various categories.
Q4) Explain Fourier analysis of signal.

LECTURE7-:
Amplitude Modulation, Generation of AM waves&Demodulation of AM -:
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.
Figure shows the different kinds of analog modulation schemes
that are available
Amplitude Modulation- It is the process where, the amplitude of

the carrier is varied proportional to that of the message signal.Amplitude Modulation with
carrier.
Let m(t) be the base-band signal, m(t)-----(M())
and c(t)be the carrier, c(t) = Ac cos( ct).
fc is chosen such that fc >> W, where W is the maximum frequency component of m(t). The
amplitude modulated signal is given by

Amplitude modulation
Figure shows the spectrum of the Amplitude Modulated
signal.
Ka is constant called amplitude sensitivity. K am(t) < 1 and

it indicates percentage modulation. Modulation in AM: A product modulator is used for
Generating the modulated signal as shown in Figure
Modulation using product modulator

Demodulation in AM: An envelope detector is used to get
the demodulated signal as shown in fig.
Demodulation using Envelope detector

The voltage Vm(t) across the resistor R gives the message
signal m(t).

LECTURE-8
Double Side Band - Suppressed Carrier
( DSB-SC) Modulation
In AM modulation, transmission of carrier consumes lot of
Power. Since, only the side bands contain the information
About the message, carrier is suppressed. This results in a
DSB-SC wave.
A DSB-SC wave s(t) is given by
Modulation in DSB-SC: Here also product modulator is used as

Shown in Figure, but the carrier is not added. Figure shows
the spectrum of the DSB-SC signal.

Demodulation in DSB-SC: A coherent demodulator is used.The local oscillator present in the

demodulator generates a Carrier which has same frequency and phase in
a as that of the carrier in the modulated signal .

If, the demodulator has constant phase, the original

signal is reconstructed by passing v(t) through an LPF.
lECTURE 9
SINGLE SIDE BAND MODULATION-:
In DSB-SC it is observed that there is symmetry in thebandstructure. So, even if one half is
transmitted, the other half can be recovered at the received. By doing so, the
bandwidth and power of transmission is reduced by half.
Depending on which half of DSB-SC signal is transmitted,
there are two types of SSB modulation
1. Lower Side Band (LSB) Modulation
2. Upper Side Band (USB) Modulation

Fig(1) SSB signals from orignal signal

Fig(2) Frequency analysis of SSB signals
From Figure 2 and the concept of the Hilbert Transform,
But, from complex representation of signals,

So,
Similarly,
Generation of SSB signals A SSB signal is represented by:
Fig(3) Generation of SSB signals
As shown in Figure 3, a DSB-SC modulator is used for SSB signal generation. Coherent
Demodulation of SSB signals SSB(t) is multiplied with cos(ct) and passed through low pass
filter to get back the orignal signal.

Figure 4: Demodulated SSB signal

The demodulated signal is passed through an LPF to remove
unwanted SSB terms.
LECTURE-10
VESTIGIAL SIDEBAND MODULATION-:
The following are the drawbacks of SSB signal generation:
1. Generation of an SSB signal is difficult.
2. Selective filtering is to be done to get the original signal
back.
3. Phase shifter should be exactly tuned to 900.
To overcome these drawbacks, VSB modulation is used. It can viewed as a compromise between
SSB and DSB-SC. Figure 5 shows all the three modulation schemes.

Figure 5: VSB Modulation

In VSB
1. One sideband is not rejected fully.
2. One sideband is transmitted fully and a small part (vestige)
of the other sideband is transmitted.The transmission BW is BWv = B + v. where, v is the
vestigial frequency band. The generation of VSB signal is
shown in Figure 6
Figure 6: Block Diagram - Generation of VSB signal

Here, Hi() is a filter which shapes the other sideband.
To recover the original signal from the VSB signal, the VSB
signal is multiplied with cos(ct) and passed through an LPF
such that original signal is recovered.

The Hilbert Transform

The Hilbert Transform on a signal changes its phase by _900. The
Hilbert transform of a signal g(t) is represented as ^g(t).
We, say g(t) and ^g(t) constitute a Hilbert Transform pair. If we observe the above equations, it
is evident that Hilbert transform is nothing but the convolution of g(t) with 1.The Fourier
Transform of ^g(t) is computed from signum function sgn(t).
Where,

Properties of Hilbert Transform

1. g(t) and ^g(t) have the same magnitude spectrum.
2. If ^g(t) is HT of g(t) then HT of ^g(t) is -g(t).
3. g(t) and ^g(t) are orthogonal over the entire interval - to
+.
Complex representation of signals

If g(t) is a real valued signal, then its complex representation g+(t)
is given by
Therefore,
g+(t) is called pre-envelope and exists only for positive frequencies.

For negative frequencies g(t) is defined as follows:

Therefore,
Essentially the pre-envelope of a signal enables the suppression of one of the sidebands in signal
transmission. The pre-envelope is used in the generation of the SSB-signal.

LECTURE-11,12&13
ANGLE MODULATION-:
In this type of modulation, the frequency or phase of carrier is varied in proportion to the
amplitude of the modulating signal.
Figure 1: An angle modulated signal

If s(t) = Accos(i(t)) is an angle modulated signal, then
1. Phase modulation:
2. Frequency Modulation
Phase Modulation If m(t) = Am cos(2fmt) is the message

signal, then the phase modulated signal is given by
Here, kp is phase sensitivity or phase modulation index.

Frequency Modulation If m(t) = Am cos(2fmt) is the message
signal,then the Frequency modulated signal is give
here,
k fAm/2 is called frequency deviation (f) and
f/ fmis called modulation index (). The Frequency modulated signal is
given by
Depending on how small is FM is either Narrowband

FM( << 1) or Wideband FM( >> 1).
Narrow-Band FM (NBFM)
In NBFM << 1, therefor s(t) reduces as follows:
Since, is very small, the above equation reduces to

The above equation is similar to AM. Hence, for NBFM the bandwidth is same as that of AM
i.e.,
2 *message bandwidth(2 *B).
A NBFM signal is generated shown in Figure as
Generation of NBFM signal
Wide-Band FM (WBFM)
A WBFM signal has theoretically infinite bandwidth. Spectrum calculation of WBFM signal is a
tedious process. For, practical applications however the Bandwidth of a
WBFM signal is calculated as follows:
Let m(t) be band limited to Hz and sampled adequately at 2BHz. If time period T = 1/2B is too
small, the signal can be approximated by sequence of pulses as shown in Figure

Approximation of message signal
If tone modulation is considered, and the peak amplitude of the sinusoid is mp, the minimum and
maximum frequency deviations will be c- K fm p and c+ K fm p respectively. The spread of
pulses in frequency domain will be 2 /T=4B as shown in fig.

Bandwidth calculation of WBFM signal
Therefore, total BW is 2 K fm p + 8B and if frequency deviation is considered
The bandwidth obtained is higher than the actual value. This is due to the staircase
approximation of m(t). The bandwidth needs to be readjusted. For NBFM, k f is
very small an d hence f is very small compared to B.
This implies
But the bandwidth for NBFM is the same as that of AM which is 2B.A better bandwidth estimate
is therefore:
This is also called Carson's Rule.

Demodulation of FM signals-:
Let f m(t) be an FM signal.
This signal is passed through a differentiator to get

If we observe the above equation carefully, it is both amplitude and frequency modulated.
Hence, to recover the original signal back an envelope
detector can be used. The envelope takes the form
FM signal - both Amplitude and Frequency Modulation

The block diagram of the demodulator is shown in Figure
Demodulation of an FM signal

The analysis for Phase Modulation is identical. Analysis of bandwidth in PM
The difference between FM and PM is that the bandwidth is independent of signal bandwidth in
FM while it is strongly dependent on signal bandwidth in PM.
EXAMPLE OF ANGLE MODULATION-:

An angle-modulated signal with carrier frequency c = 2*106 is described by the equation:
1. Determine the power of the modulating signal.

2. What is f?
3. What is ?
4. Determine the phase deviation.
5. Estimate the bandwidth of EM(t)?
1. P = 122/2= 72 units
2. Frequency deviation f, we need to estimate the instantaneous frequency:
The deviation of the carrier is

When the two sinusoids add in phase, the maximum value will be 7500 + 20000
Hence,
3)
4)
The angle
The maximum angle deviation is 15, which is the phase deviation.

5)
LECTURE -14
REVISION &ASSIGNMENT-:
Q1) Explain Amplitude modulation in detail.
Q2) Explain SSB and its generation &demodulation.
Q3) Write a short note on VSB.
Q4) Explain Frequency modulation and its generation.
Also explain its demodulation.


LECTURE-15
PULSE ANALOG MODULATION-:
1) Sampling Theorem and its Importance
Sampling Theorem:
A band limited signal can be reconstructed exactly if it is sampled at a rate at least twice the
maximum frequency component in it.
Figure shows a signal g(t) that is band limited.
Spectrum of band limited signal g(t)

The maximum frequency component of g (t) is fm
To recover the signal g (t) exactly from its samples it has to be sampled at a rate
fs= 2fm. The minimum required sampling rate fs = 2fm is called Nyquist rate.
Proof: Let g(t) be a band limited signal whose bandwidth is fm and

(=2fm)

Original signal g(t)
Spectrum G()
Let gS(t) be the sampled signal. Its Fourier Transform GS() is given by

If S = 2m, i.e., T = 1/2fm. Therefore, Gs () is given by

To recover the original signal G ():

1. Filter with a Gate function, H 2m () of width 2 m
2. Scale it by T.
Recovery of signal by filtering with a filter of width 2m

Aliasing
Aliasing is a phenomenon where the high frequency components of the sampled signal interfere
with each other.
because of inadequate sampling s< 2fm

.
Aliasing due to inadequate sampling
Aliasing leads to distortion in recovered signal. This is the reason why sampling frequency
should be at least twice the bandwidth of the signal.
Oversampling-:
In practice signal are oversampled, where fs is significantly higher than Nyquist rate to
avoid aliasing.

Oversampled signal-avoids aliasing.

LECTURE-16
SAMPLE AND HOLD CIRCUITSample-and-hold (S/H) is an important analog building block with many applications,
Including analog-to-digital converters (ADCs) and switched-capacitor filters. The Function of
the S/H circuit is to sample an analog input signal and hold this value over a Certain length of
time for subsequent processing. Taking advantages of the excellent properties of MOS capacitors
and switches, traditional switched capacitor techniques can be used to realize different S/H
circuits. The simplest S/H circuit in MOS technology is shown in Figure 1, where Vin is the
inputsignal, M1 is an MOS transistor operating as the sampling switch, Ch is the hold capacitor,
ck is the clock signal, and Vout is the resulting sample-and-hold output signal.
Figure 1: Simplest sample-and-hold circuit in MOS technology.
As depicted by Figure 1, in the simplest sense, a S/H circuit can be achieved using only
one MOS transistor and one capacitor. The operation of this circuit is very straightforward.
Whenever ck is high, the MOS switch is on, which in turn allows Vout to track Vin. On the other
hand, when ck is low, the MOS switch is off. During this time, Ch will keep Vout equal to the
value of Vin at the instance when clk goes low.

LECTURE-17
Time division (TDM) and frequency division (FDM) multiplexing

Since only samples of a message signal are transmitted, the channel is occupied only for a short
time slot in pulse modulation systems. Consequently, samples of N message signals may be
transmitted over the same channel Message signals 1; 2; : : : ;N are separated in the time domain
.Note, the multiplexed signal is the input to the pulse modulator.
TDM: CONCEPT OF FRAMING AND SYNCHRONIZATION

Consider a multiplexed PAM wave generated by the commutator. The time
interval TF containing one sample from each message signal is called a frame Synchronization
must be established and maintained between the commutator and decommutator. Generally, an
extra pulse (called marker) or a special sequence of pulses are transmitted at the beginning of
each frame to help the clock recovery circuit to establish the synchronization
FDM:- Frequency-division multiplexing (FDM) is a form of signal multiplexing which involves

assigning non-overlapping frequency ranges to different signals or to each "user" of a medium.
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.
FDM is widely used in radio and television systems (e.g. broadcast radio and TV) and
was widely used in multichannel telephony (now being superseded by digital techniques
and TDM).
The multichannel telephone system illustrates some important aspects and is considered
below. For speech, a bandwidth of 3kHz is satisfactory.
The physical line, e.g. a co-axial cable will have a bandwidth compared to speech as
shown next
3kHz
freq
GHz
From AM we have noted:
m(t)
freq
m(t)
DSBSC
carrier
cos( c t )
DSBSC
freq
fc
In order to use bandwidth more effectively, SSB is used i.e

SSB
Filter
m(t)
SSBSC
carrier
cos( c t )
freq
fc

We have also noted that the message signal m(t) is usually band limited
Band
Limiting
Filter
Speech
m(t)
300Hz 3400Hz
SSB
Filter
SSBSC
cos( c t )
The Band
Limiting Filter (BLF) is usually a band pass filter with a pass band 300Hz to
3400Hz for speech. This is to allow guard bands between adjacent channels.
f
300H z
3400H z
f
300H z
3400H z
10kH z
m (t)
S peech
C o n v e n tio n
For telephony, the physical line is divided (notionally) into 4kHz bands or channels, i.e.
the channel spacing is 4kHz. Thus we now have:
Guard Bands
Bandlimited
Speech
f
4kHz
Note, the BLF does not have an ideal cut-off the guard bands allow for filter roll off
in order to reduce adjacent channel crosstalk.

Consider now a single channel SSB system
DSBSC
m(t)
SSBSC
SSB
Filter
BLF
fc
300Hz
3400Hz
The spectra will be

m(t)
freq
DSBSC
freq
fc
freq
fc
Consider now a system with 3 channels

m 1(t)
SSB
F ilte r
BLF
fc1
f1
m 2(t)
SSB
F ilte r
BLF
FD M
S ig n a l
M (t)
fc2
f2
SSB
F ilte r
BLF
m 3(t)
f c3
f3
F D M T r a n sm itte r
or E n cod er
B a n d lim ite d
Each carrier frequency, fc1, fc2 and fc3 are separated by the channel spacing
frequency, in this case 4 kHz, i.e. fc2 = fc1 + 4kHz, fc3 = fc2 + 4kHz.
The spectrum of the FDM signal, M(t) will be:
4kHz
4kHz
M(t)
4kHz
Shaded areas are to

show guard bands.
f1
fc1
f3
f2
fc2
freq
fc3
Note that the baseband signals m1(t), m2(t), m3(t) have been multiplexed into adjacent
channels, the channel spacing is 4kHz. Note also that the SSB filters are set to select

the USB, tuned to f1, f2 and f3 respectively.

SSB
Filter
f1
M(t)
FDM
Signal
LPF
fc1
SSB
Filter
f2
Band
Limited
LPF
m2(t)
Back to
baseband
fc2
SSB
Filter
f3
m1(t)
LPF
m3(t)
fc3
The diagram below illustrates the FDM principle for 12 channels (similar to 3 channels)
to a form a basic group.
m1(t)
m2(t)
m3(t)
Multiplexer
freq
60kHz
12kHz
m12(t)
i.e. 12 telephone channels are multiplexed in the frequency band 12kHz 60 kHz in
4kHz channels basic group.
LECTURE-18&19
Pulse amplitude modulation (PAM), pulse time modulation

Introduction
The purpose of the modulator is to convert discrete amplitude serial symbols (bits in a binary system) ak
to analogue output pulses which are sent over the channel.
The demodulator reverses this process
Introduction
Possible approaches include

Pulse width modulation (PWM)
Pulse position modulation (PPM)
Pulse amplitude modulation (PAM)
We will only be considering PAM in these lectures
PAM is a general signalling technique whereby pulse amplitude is used to convey the
messageFor example, the PAM pulses could be the sampled amplitude values of an
analoguesignal. We are interested in digital PAM, where the pulse amplitudes are
constrained to chosen from a specific alphabet at the transmitter

PAM
In binary PAM, each symbol ak takes only two values, say {A1 and A2}.In a multilevel,
i.e., M-ary
ary system, symbols may take M values {A1, A2 ,... AM}.Signalling
Signalling period, T
Each transmitted pulse is given by
ak hT (t kT )
Where hT(t) is the time domain pulse shape
To generate the PAM output signal, we may choose to represent the input to the transmit
filter hT(t) as a train of weighted impulse functions
xs (t ) =
a (t kT )
k =
Consequently, the filter output x(t) is a train of pulses, each with the required shape
xs (t ) =
ak (t kT )
x (t ) =
k =
a h (t kT )
k =
k T

8km milestone ,Sohna Palwal Road , NH
NH-71
71 B ,Sohna , Gurgaon ,Haryana.
E-mail
mail : info@wit.net.in
Filtering of impulse train in transmit filter

filter.To
To generate the PAM output signal, we may
choose to represent the input to the transmit filter hT(t) as a train of weighted impulse
functions
x s (t ) =
a (t kT )
k =
Consequently, the filter output x(t) is a train of pulses, each with the required shape hT(t)
x(t ) =
a h (t kT )
k =
k T
Hence the signal at the receiver filter output is
y (t ) =
a h(t kT ) + v(t )
k =
Where h(t) is the inverse Fourier transform of H(w) and v(t) is the noise signal at the receive
filter output. Data detection is now performed by the Data Slicer.Sampling
Slicer Sampling y(t), usually at the
optimum instant t=nT+td when the pulse magnitude is the greatest yields
Where vn=v(nT+td) is the sampled noise and td is the time delay required for optimum sampling
8km milestone ,Sohna Palwal Road , NH
NH-71
71 B ,Sohna , Gurgaon ,Haryana.
E-mail
mail : info@wit.net.in
yn is then compared with threshold(s) to determine the recovered data symbols.
LECTURE-20&21-:
Elements of pulse code modulation
In pulse-code modulation (PCM), an analog message signal is represented by a
Sequence of coded pulses, which is accomplished by representing the signal in
Discrete form in both time and amplitude
PCM is the most basic form of digital pulse modulation
A PCM system contains three main blocks:
PCM transmitter
Transmission path
Receiver
PCM transmitter

T.1: Sampling
To avoid aliasing, a pre-alias (low-pass) filter is used to limit the bandwidth of message signal
Sampling rate must be greater than the Nyquist rate 2W
T.2: Quantization
To reduce quantization noise, a non uniform quantizer is used
T.3: Encoding contains two encoding processes
Encoder maps quantized samples into v-length code word
A line encoding converts the digital signal (codeword) into an analog waveform.
PCM transmission path
TP.1: Regeneration
Regenerative repeater performs three tasks:
Equalization means a compensation for the effects of amplitude and
phase distortion produced by the channel
Timing circuitry assigns the decision time instants when the

probability of making of a wrong decision is minimum
Decision making regenerates the PCM wave
The most important feature of PCM systems is the ability to control the effects
of distortion and noise. A PCM signal may be reconstructed from the distorted
and noisy input by means of regenerative repeaters placed sufficiently close to
each other along the transmission route
PCM receiver
R.1: Regeneration circuit

Reshapes and cleans up the received noise and distorted signal. Its three tasks
were discussed in the previous transparency
R.2: Decoding
First the code word is recovered from the data sequence and then a pulse is
generated, where the amplitude of pulse is determined by the code word
R.3: Reconstruction filter
The cut-off frequency of low-pass reconstruction filter is equal to the message

bandwidth W. It recovers the analog message signal
Transmission bandwidth of a PCM wave

Each encoded message sample is represented by a v-digit code word. Consequently, the
signaling rate becomes
The transmission bandwidth required by the PCM wave is
Main feature of pulse-code modulation:

If the regenerators are well placed then they cancel the effect of channel distortion and noise. In
this case the only source of distortion and noise is the quantization error

LECTURE-22
DPCM-:
The standard sampling rate for pulse code modulation (PCM) of telephone grade speech signal is
fs = 8 Kilo samples per sec with a sampling interval of 125 sec. Samples of this band limited
speech signal are usually correlated as amplitude of speech signal does not change much within
125 sec. A typical auto correlation function R ( ) for speech samples at the rate 8 Kilo samples
per sec is shown in Fig . R ( = 125 sec) is usually between 0.79 and 0.87. This aspect of
speech signal is exploited in differential pulse code modulation (DPCM) technique. A schematic
diagram for the basic DPCM modulator is shown in Fig Note that a predictor block, a summing
unit and a subtraction unit have been strategically added to the chain of blocks of PCM coder
instead of feeding the sampler output x (kTs) directly to a linear quantizer.
The error sample is given by the following expression
is a predicted value for
and is supposed to be close to
small in magnitude
such that
is called as the prediction error for the n-th sample.

Typical normalized auto-correlation coefficient for speech signal
Schematic diagram of a DPCM modulator

If we assume a final enclosed bit rate of 64kbps as of a PCM coder, we envisage smaller step
size for the linear quantizer compared to the step size of an equivalent PCM quantizer. As a
result, it should be possible to achieve higher SQNR for DPCM codec delivering bits at the same
rate as that of a PCM code.
There is another possibility of decreasing the coded bit rate compared to a PCM system if an
SQNR as achievable by a PCM codec with linear equalizer is sufficient. If the predictor output
can be ensured sufficiently close to then we can simply encode the quantizer output sample
v(kTs) in less than 8 bits. For example, if we choose to encode each of by 6 bits, we achieve a
serial bit rate of 48 kbps, which is considerably less than 64 Kbps. This is an important feature of
DPCM, when the coded speech signal will be transmitted through wireless propagation channels.
A block schematic diagram of a DPCM demodulator is shown in Fig The scheme is

straightforward and it tries to esteem using a predictor unit identical to the one used in the
modulator. We have already observed that is very close to within a small quantization error of .
The analog speech signal is obtained by passing the sample through an appropriate low pass
filter. This low pass filter should have a 3 dB cut off frequency at 3.4kHz.
Schematic diagram of a DPCM demodulator; note that the demodulator is very similar to a
portion of the modulator

LECTURE -23
Delta modulation (DM)
If the sampling interval Ts in DPCM is reduced considerably, i.e. if we sample a band limited
signal at a rate much faster than the Nyquist sampling rate, the adjacent samples should have
higher correlation Fig.. The sample-to-sample amplitude difference will usually be very small.
So, one may even think of only 1-bit quantization of the difference signal. The principle of Delta
Modulation (DM) is based on this premise. Delta modulation is also viewed as a 1-bit DPCM
scheme. The 1-bit quantizer is equivalent to a two-level comparator (also called as a hard
limiter). Fig. Shows the schematic arrangement for generating a delta-modulated signal.
The correlation increases when the sampling interval is reduced

Block diagram of a delta modulator

Note Some interesting features of Delta Modulation
No effective prediction unit the prediction unit of a DPCM coder is eliminated and
replaced by a single-unit delay element.
A 1-bit quantizer with two levels is used. The quantizer output simply indicates whether
the present input sample x(kTs) is more or less compared to its accumulated
approximation (skTx.)
Output (skTx) of the delay unit changes in small steps.
The accumulator unit goes on adding the quantizer output with the previous accumulated
version (skTx).
u(kTs), is an approximate version of x(kTs)
Performance of the Delta Modulation scheme is dependent on the sampling rate. Most of
the above comments are acceptable only when two consecutive input samples are very
close to each other.
This diagram indicates the output levels of 1-bit quantizer. Note that if is the step size, the
two output levels are s
Now, assuming zero initial condition of the accumulator, it is easy to see that
Fig shows that is essentially an accumulated version of the quantizer output for the error signal
e. also gives a clue to the demodulator structure for DM. Fig. shows a scheme for demodulation.
The input to the demodulator is a binary sequence and the demodulator normally starts with no
prior information about the incoming sequence.

Demodulator structure for DM

Now, let us recollect from our discussion on DPCM that, u(kTs) closely represents the input
signal with small quantization error
Next, from the close loop including the delay-element in the accumulation unit in the Delta
modulator structure, we can write
Hence, we may express the error signal as,
That is, the error signal is the difference of two consecutive samples at the input except the
quantization error (when quantization error is small).
LECTURE-25
Amplitude Shift Keying (ASK) Modulation:

Amplitude shift keying (ASK) is a simple and elementary form of digital modulation in which
the amplitude of a carrier sinusoid is modified in a discrete manner depending on the value of a
modulating symbol. Let a group of m bits make one symbol. Hence one can design M = 2m
different baseband signals, dm(t), 0 m M and 0 t T. When one of these symbols
modulates the carrier, say, c(t) = cosct, the modulated waveform is:
sm(t) = dm(t).cosct 5.23.1
This is a narrowband modulation scheme and we assume that a large number of carrier cycles are
sent within a symbol interval, i.e. cT2 is a large integer. It is obvious that the
information is embedded only in the peak amplitude of the modulated signal. So, this is a kind of
digital amplitude modulation technique. From another angle, one can describe this scheme of
modulation as a one-dimensional modulation scheme where one basis function 1(t) =
tTccos.2 , defined over 0 t T and having unit energy is used and all the baseband signals
are linearly dependent.
Ex. #5.23.1 Let m = 2 and d0 = 0, d1 = 1, d2 = 2 and d3 = 3. It is simple to generate such distinct
and fixed levels in practice. Further, let us arbitrarily assume the following information to signal
mapping: d0 (1,1), d1 (1,0), d2 (0,1) and d3 (0,0). So, we have four symbols and the
modulated waveforms are:
s0(t) = d0(t). tTccos.2 = 0, s1 (t) = d1(t). tTccos.2 = tTccos.2, s2 (t) = d2(t). tTccos.2 = 2.
tTccos.2 and s3(t) = d3(t). tTccos.2 = 3. tTccos.2
The signal constellation consists of four points on a straight line. The distances of the points
from the origin (signifying zero energy) are 0, 1, 2 and 3 respectively. Note that in this example,
no-transmission indicates that d0, i.e. the symbol (1,1) is transmitted. This is not surprising
and it also should not give an impression that we are able to transmit information without
spending any energy. In fact, it is a bad practice to assign zero energy to a symbol for any good
quality carrier modulation scheme because, demodulation at the receiving end and that ultimately
leads to poor SER and BER. Another interesting feature to note is that the modulated symbols
have different energy levels, viz. 0, 1, 4 and 9 units. This feature does not make the highest
energy symbol d3 more immune to thermal noise.
On the contrary, the large range of energy level, namely, from 0 to 9 implies that the power
amplifier in the transmitter has to have a large linear range of operation sometime a costly
proposition. If the power amplifier goes into its non-linear range while amplifying s3(t),
harmonics of the carrier sinusoid will be generated which will rob some power from s3(t) away
and may interfere with other wireless transmissions in frequency bands adjacent to 2c, 3c,
etc. The point to note is that, the Euclidean distance of s3(t) from the nearest point s2(t) in the
receiver signal space decreases because of amplifier nonlinearity and it means that the receiver
will confuse more between s3(t) and s2(t) while trying to detect the symbols in presence of noise.

Assuming that all the symbols are equally likely to appear at the input of the modulator, we see
that the average energy per symbol (sE) is 14/4 = 3.5 unit. This is an important parameter for
transmission of digital signals because it is ultimately proportional to the average transmission
power. A system designer would always try to ensure low transmission power to save cost and to
enhance reliability of the system. So, we see the simple example of ASK modulation of four
symbols could be cited in such a way that the signal points were better placed in the constellation
diagram such that sE is minimum.
Now, ASK being a form of amplitude modulation, we can say that the bandwidth of the
modulated signal will be the same as the bandwidth of the baseband signal. The baseband signal
is a long and random sequence of pulses with discrete values. Hence, ASK modulation is not
bandwidth efficient. It is implemented in practice when simplicity and low cost are principal
requirements.
On-off keying
On-Off Keying (OOK) is a particularly simple form of ASK that represents binary data as the
presence or absence of a sinusoid carrier. For example, the presence of a carrier over a bit
duration Tb may represent a binary 1 while its absence over a bit duration Tb may represent a
binary 0. This form of digital transmission (OOK) has been commonly used to transmit Morse
Codes over a designated radio frequency for telegraph services. As mentioned earlier, OOK is
not a spectrally efficient form of digital carrier modulation scheme as the amplitude of the carrier
changes abruptly when the data bit changes. So, this mode of transmission is suitable for low or
moderate data rate. When the information rate is high, other bandwidth efficient phase
modulation schemes are preferable.

LECTURE-26
FSK(Frequency Shift Keying)
Frequency Shift Keying (FSK) modulation is a popular form of digital modulation used in lowcost applications for transmitting data at moderate or low rate over wired as well as wireless
channels. In general, an M-ary FSK modulation scheme is a power efficient modulation scheme
and several forms of M-ary FSK modulation are becoming popular for spread spectrum
communications and other wireless applications. In this lesson, our discussion will be limited to
binary frequency shift keying (BFSK).
Two carrier frequencies are used for binary frequency shift keying modulation. One frequency is
called the mark frequency (f2) and the other as the space frequency ( f1). By convention, the
mark frequency indicates the higher of the two carriers used. If Tb indicates the duration of one
information bit, the two time-limited signals can be expressed as :
The binary scheme uses two carriers and for special relationship between the two frequencies
one can also define two orthonormal basis functions as shown below.
If T1 = 1/f1 and T2 = 1/f2 denote the time periods of th = n.T2 = Tb, where m and n are
positive integers, the two carriers are orthogonal over e carriers and if we choose m.T1 = = n.T2
= Tb, where m and n are positive integers, the two carriers are orthogonal over the bit
duration Tb. If Rb = 1/Tb denotes the data rate in bits/second, the orthogonal condition implies,
f1 = m.Rb and f2 = n.Rb. Let us assume that n > m, i.e. f2 is the mark frequency. Let the
separation between the two carriers be, f = f2 - f1 = (n-m).Rb.
Now, the scalar coefficients corresponding to Eq.
So, the two signal victors can be expressed as:
Please note that one can generate an FSK signal without following the above concept of
orthogonal carriers and that is often easy in practice. Fig shows the possible FSK modulated
waveform. Notice the waveform carefully and verify if the two carriers are orthogonal. An
obvious feature of an FSK modulated signal, analogous to analog FM signal is that envelop of
the modulated signal is constant. All modulation schemes which exhibit constant envelope, are
preferable for high power digital transmission because, operation of the power amplifier in a
non-linear region may not produce considerable harmonics.

Fig. shows the constellation diagram for binary FSK. Fig. shows a conceptual diagram for
generating binary FSK modulated signal. Note that the input random binary sequence is
represented by 1 and 0 where 0 represents no voltage at the input of the multipliers. A 0
input to the inverter results in a 1 at its output. That is, the inverter, along with the two
multipliers and the summing unit, may be thought to behave as a switch which selects output of
one of the two oscillators.
Signal constellation for binary FSK. The diagram also shows the two decision zones, Z1
and Z2.
A schematic diagram for BFSK modulation

In practice, however, this scheme will not work reliably because, the two oscillators being
independent, it will be difficult to maintain the orthogonal relationship between the two carrier
frequencies. Any relative phase shift among the two oscillators, which may even occur due to
thermal drift, will result in deviation from the orthogonality condition. Another disadvantage of
the possible relative phase shift is random discontinuity in the phase of the modulated signal
during transition of information bits. A better proposition for physical implementation is to use a
voltage controlled oscillator (VCO) instead of two independent oscillators and drive the VCO
with an appropriate baseband modulating signal, derived from the serial bit stream. The VCO
free-running
frequency ( ffree ) should be chosen as:
Fig. shows the form of a coherent FSK demodulator, based on the concepts of correlation
receiver as outlined in Module #4. The portion on the LHS of the dotted line shows the
correlation detector while the RHS shows that the vector receiver reduces to a subtraction unit.
Output of the subtraction unit is compared against a threshold of zero to decide about the
corresponding transmitted bit.

Schematic diagram of a coherent BFSK demodulator

Fig. gives a scheme for non-coherent demodulation of BFSK signal using matched filters. It is
often easier to follow this approach than the coherent demodulation scheme without sacrificing
error performance.
General scheme for non-coherent demodulation of BFSK signal using matched filters
When the issues of performance and bandwidth are not critical and the operating frequencies are
low or moderate, a low complexity realization of the demodulator is also possible. Two bandpass
filters, one centered at f1 and the other centered at f2 may replace the matched filters .
Power Spectra of BFSK

Power spectrum of an FSK modulated signal depends on the choice of f1 and f2, i.e. on m and
n. When (n-m) is large, we may visualize BFSK as the sum of two OOK signals (see
Fig.5.23.3) with carriers f1 and f2. However, such choice of (n-m) does not a result in bandwidth
efficiency.
In the following, = (m+1), i.e. f1 = m.Rb = m/Tb
Fig. 5.23.5 shows a sketch (approximate) of the power spectrum of binary FSK.we will discuss
about another form of FSK, known as Minimum Shift Keying (MSK), which is operates with
minimum possible separation between two frequencies f1 and f2.
Sketch of the power spectrum of binary FSK signal


LECTURE-27
BPSK, QPSK
Quaternary Phase Shift Keying (QPSK)
This modulation scheme is very important for developing concepts of two-dimensional I-Q
modulations as well as for its practical relevance. In a sense, QPSK is an expanded version from
binary PSK where in a symbol consists of two bits and two orthonormal basis functions are
used. A group of two bits is often called a dibit. So, four dibits are possible. Each symbol
carries same energy.
Let, E: Energy per Symbol and T: Symbol Duration = 2. Tb, where Tb: duration of 1 bit. Then, a
general expression for QPSK modulated signal, without any pulse shaping
On simple trigonometric expansion, the modulated signal si (t) can also be expressed as:
The two basis functions are:
The four signal points, expressed as vectors, are:
Fig. shows the signal constellation for QPSK modulation. Note that all the four points are
equidistant from the origin and hence lying on a circle. In this plain version of QPSK, a symbol
transition can occur only after at least T = 2Tb sec. That is, the symbol rate Rs = 0.5Rb. This is
an important observation because one can guess that for a given binary data rate, the
transmission bandwidth for QPSK is half of that needed by BPSK modulation scheme.
Signal constellation for QPSK. Note that in the above diagram has been considered to be
zero. Any fixed non-zero initial phase of the basis functions is permissible in general.
Now, let us consider a random binary data sequence: 10111011000110 Let us designate the
bits as odd (bo) and even (be) so that one modulation symbol consists of one odd bit and the
adjacent even bit. The above sequence can be split into an odd bit sequence (1111001) and an
even bit sequence (0101010). In practice, it can be achieved by a 1-to-2 DEMUX. Now, the
modulating symbol sequence can be constructed by taking one bit each from the odd and even
sequences at a time as {(10), (11), (10), (11), (00), (01), (10), }. We started with the odd
sequence. Now we can recognize the binary bit stream as a sequence of signal points which are
to be transmitted: {1s, 4s, 1s, 4s, 2s, 3s, 1s, }.

Table summarizes the features of QPSK signal constellation

Fig shows the QPSK modulated waveform for a data sequence 101110110001. For better
illustration, only three carrier cycles have been shown per symbol duration.
QPSK modulated waveform

Generation of QPSK modulated signal
Let us recall that the time-limited energy signals for QPSK modulation can be expressed

The QPSK modulated wave can be expressed in several ways such as:
For narrowband transmission, we can further express s(t) as:
where
is the complex low-pass equivalent representation of s(t).
One can readily observe that, for rectangular bipolar representation of information bits and
without any further pulse shaping,

Block schematic diagram of a QPSK modulator
Another schematic diagram of a QPSK modulator
QPSK modulated signal can indeed be viewed as consisting of two independent BPSK
modulated signals with orthogonal carriers.
The structure of a QPSK demodulator, following the concept of correlation receiver, is shown in
Fig. . The received signal r(t) is an IF band pass signal, consisting of a desired modulated signal
s(t) and in-band thermal noise. One can identify the I- and Q- path correlators, followed by two
sampling units. The sampling units work in tandem and sample the outputs of respective
integrator output every T = 2Tb second, where Tb is the duration of an information bit in
second. From our understanding of correlation receiver, we know that the sampler outputs, i.e. r1
and r2 are independent random variables with Gaussian probability distribution. Their variance is
same and decided by the noise variance while their means are 2E, following our style of
representation. Note that the polarity of the sampler output indicates best estimate of the
corresponding information bit. This task is accomplished by the vector receiver, which consists
of two identical binary comparators as indicated in Fig. The output of the comparators are
interpreted and multiplexed to generate the demodulated information
Correlation receiver structure of QPSK demodulator

Spectrum of QPSK modulated signal
To determine the spectrum of QPSK modulated signal, we follow an approach similar to the one
we followed for BPSK modulation in the previous lesson. We assume a long sequence of random
independent bits as our information sequence. Without Nyquist filtering, the shaping function in
this case can be written as:
After some straight forward manipulation, the single-sided spectrum of the equivalent complex
baseband signal can be expressed as:
Here E is the energy per symbol and T is the symbol duration. The above expression can also
be put in terms of the corresponding parameters associated with one information bit
Normalized base band bandwidth of QPSK and BPSK modulated signal

LECTURE-28
M-ary PSK
This is a family of two-dimensional phase shift keying modulation schemes. Several bandwidth
efficient schemes of this family are important for practical wireless applications.
As a generalization of the concept of PSK modulation, let us decide to form a modulating
symbol by grouping m consecutive binary bits together. So, the number of possible modulating
symbols is, M = 2m and the symbol duration T = m. Tb. Fig. 5.25.6 shows the signal
constellation for m = 3. This modulation scheme is called as 8-PSK or Octal Phase Shift
Keying. The signal points, indicated by *, are equally spaced on a circle. This implies that all
modulation symbols si(t), 0 i (M-1), are of same energy E. The dashed straight lines are
used to denote the decision zones for the symbols for optimum decision-making at the receiver.
Signal space for 8-PSK modulation

The two basis functions are similar to what we considered for QPSK, viz.,
The signal points can be distinguished by their angular location
The time-limited energy signals si(t) for modulation can be expressed in general as

Considering M-ary PSK modulation schemes are narrowband-type, the general form of the
modulated signal is
FIG shows a block schematic for an M-ary PSK modulator. The baseband processing unit
receives information bit stream serially (or in parallel), forms information symbols from groups
of m consecutive bits and generates the two scalars si1 and si2 appropriately. Note that these
scalars assume discrete values and can be realized in
Block schematic diagram of M-ary PSK modulator
Normalized scalars for 8-PSK modulation

Without any pulse shaping, the uI(t) and uQ(t) of Eq. are proportional to si1 and si2 respectively.
Beside this baseband processing unit, the M-ary PSK modulator follows the general structure of
an I/Q modulator.
Fig. shows a scheme for demodulating M-ary PSK signal following the principle of correlation
receiver. The in-phase and quadrature-phase correlator outputs
Structure of M-ary PSK demodulator

LECTURE-29&30
Noise Analysis
The following assumptions are made:
Channel model
distortionless
{Additive White Gaussian Noise (AWGN)
Receiver Model (see Figure 1)

Ideal band pass filter
Ideal demodulator

BPF (Band pass filter) - bandwidth is equal to the message Bandwidth B midband frequency is fc.
Power Spectral Density of Noise
N0/2 and is defined for both positive and negative frequency .N0 is the average power/(unit
BW) at the front-end of the receiver in AM and DSB-SC.
The filtered signal available for demodulation is given by:
Input SNR:
(SNR)I =Average power of modulated signal s(t)/Average power of noise
output SNR:
(SNR)O =Average power of demodulated signal s(t)/Average power of noise
The Output SNR is measured at the receiver.
Channel SNR:
(SNR)C =Average power of modulated signal s(t)/Average power of noise in message
bandwidth


Communication Basics

Hochgeladen von

Dokumentinformationen

Originaltitel

Copyright

Verfügbare Formate

Dieses Dokument teilen

Dokument teilen oder einbetten

Freigabeoptionen

Stufen Sie dieses Dokument als nützlich ein?

Sind diese Inhalte unangemessen?

Copyright:

Verfügbare Formate

Communication Basics

Hochgeladen von

Copyright:

Verfügbare Formate

Syllabus

EE-206-F COMMUNICATION SYSTEMS

LECTURE NOTES OF COMMUNICATION SYSTEM

1) Information Source: The message or information originates in information source.There

World Institute Of Technology

is extensively used in the study, analysis and testing of several principles of

World Institute Of Technology

World Institute Of Technology

If we now carry out a Fourier Analysis of the combined signal then

Fourier Representation of continuous time signals

The exponential terms are orthogonal to each other because

the energy of these signals is unity since

Extending this representation to aperiodic signals:

The signal f(t) can recovered from F() as

Some Important Functions

World Institute Of Technology

The Diac delta function is also called the Impulse function.

World Institute Of Technology

5. Sampling Square function

The unit step function is another important function signal

World Institute Of Technology

1) Information Source: The message or information originates in information source.There

World Institute Of Technology

World Institute Of Technology

It is possible to expand our basic three-entity description of a digital communication

BASIC CONCEPT OF MODULATORS AND DEMODULATORS

Some Alternative Modulator Structures

Quantizer with error feedback

where g is a measure of the accuracy of the error cancellation. It depends on a number of

World Institute Of Technology

World Institute Of Technology

FIG shows-:Spectral densities of signals, and the frequency response of filters

World Institute Of Technology

Detailed explaination later in lecture 17

World Institute Of Technology

World Institute Of Technology

Amplitude Modulation- It is the process where, the amplitude of

World Institute Of Technology

Ka is constant called amplitude sensitivity. K am(t) < 1 and

Modulation using product modulator

Demodulation using Envelope detector

World Institute Of Technology

Modulation in DSB-SC: Here also product modulator is used as

World Institute Of Technology

Demodulation in DSB-SC: A coherent demodulator is used.The local oscillator present in the

World Institute Of Technology

If, the demodulator has constant phase, the original

World Institute Of Technology

Fig(1) SSB signals from orignal signal

World Institute Of Technology

Fig(2) Frequency analysis of SSB signals

From Figure 2 and the concept of the Hilbert Transform,

But, from complex representation of signals,

World Institute Of Technology

Generation of SSB signals A SSB signal is represented by:

Fig(3) Generation of SSB signals

World Institute Of Technology

Figure 4: Demodulated SSB signal

World Institute Of Technology

Figure 5: VSB Modulation

Figure 6: Block Diagram - Generation of VSB signal

World Institute Of Technology

The Hilbert Transform