Digital Communication VTU Unit 1

Intoduction & Sampling Theorem
Raghudathesh G P
Asst Professor
DIGITAL COMMUNICATION (VTU)-10EC61

UNIT - 1
Basic signal processing operations in digital communication. Sampling Principles: Sampling

Theorem, Quadrature sampling of Band pass signal, Practical aspects of sampling and signal
recovery.
7 Hours
TEXT BOOK:
1. Digital communications, Simon Haykin, John Wiley India Pvt. Ltd, 2008.
REFERENCE BOOKS:
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1. Digital and Analog communication systems, Simon Haykin, John Wildy India Lts, 2008
2. An introduction to Analog and Digital Communication, K. Sam Shanmugam, John Wiley
India Pvt. Ltd, 2008.
3. Digital communications - Bernard Sklar: Pearson education 2007
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Special Thanks To:

Faculty (Chronological): Arunkumar (STJIT), Raviteja B (GMIT).
Students: Shubham S Dhivagnya (6th sem GMIT)
PREPARED BY:
RAGHUDATHESH G P
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Asst Prof
ECE Dept, GMIT
Davangere 577004
Cell: +917411459249
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Mail: datheshraghubooks@gmail.com
Quotes:
The fragrance of flowers spreads only in the direction of the wind. But the goodness of a person
spreads in all directions.
Trust is like a paper once it crumbled it cant be perfect again.
Learn from the mistakes of others, you can't live long enough to make them all yourselves.
As soon as the fear approaches near, attack and destroy it.

Once you start a working on something, don't be afraid of failure and don't abandon it. People
who work sincerely are the happiest.
Department of ECE
datheshraghubooks@gmail.com
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Raghudathesh G P
Asst Professor
Introduction to Digital Communication

Communication:
Definition1: Communication is the process of establishing a connection or link between
two points for information exchange.
Definition2: Communication is simply the process of conveying message at a distance.
Definition3: Communication is the basic process of exchanging information.

The electronic equipments which are used for communication purpose, are called
communication equipments.
Different communication equipments when assembled together form a communication

system.
Ex: line telephony, line telegraphy, radio telephony, radio broadcasting, point-to-point
communication and mobile communication, computer communication, radar
communication, television broadcasting, radio telemetry, radio aids to navigation, radio
aids to aircraft landing etc.
Sources and Signals:
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A source of information generates a message.

Ex: man voice, television picture, teletype data, atmospheric temperature pressure.
In above examples, the message is not electric in nature, thus a transducer is used to
convert it into an electrical waveform called as message signal.
The electrical waveform or message signal is also referred to as a baseband signal.

The message signal can be of an analog or digital type.
An analog signal is one in which both amplitudes and time vary continuously over their
respective intervals.
Ex: A speech signal, a television signal, and a signal representing atmospheric

temperature or pressure at some location.
In a digital signal, on the other hand, both amplitude and time take on discrete values.
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Ex: Computer data and telegraph signals.

An analog signal can always be converted into digital form by combining three basic
operations:
1. Sampling
2. Quantizing and
3. Encoding
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The above Figure shown the block diagram of analog to digital conversion logic.
In the sampling operation, only sample values of the analog signal at uniformly spaced
discrete instants of time are retained.
In the quantizing operation, each sample value is approximated by the nearest level in a
finite set of discrete levels.
In the encoding operation, the selected level is represented by a code word that consists
of a prescribed number of code elements.
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The analog-to-digital conversion process described in figure above a segment of an

analog waveform.
The figure above shows the corresponding digital waveform, based on the use of a binary
code.
In the above example, symbols 0 and 1 of the binary code are represented by zero and
one volt, respectively.
The code word consists of four binary digits (bits), with the last bit assigned the role of a
sign bit that signifies whether the sample value in question is positive or negative.
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Raghudathesh G P
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The remaining three bits are chosen to provide a numerical representation for the absolute
value of a sample in accordance with Table below.
As a result of sampling and quantizing operations, errors are introduced into the digital
signal. These errors are nonreversible in that it is not possible to produce an exact replica
of the original analog signal from its digital representation.
However, the errors are under a designer's control. Indeed, by proper selections of the
sampling rate and code-word length (i.e., number of quantizing levels), the errors due to
sampling and quantizing can be made so small that the difference between the analog
signal and its digital reconstruction is not discernible by a human observer.
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Raghudathesh G P
Asst Professor
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The 3 Basic Operation/Steps Used To Convert Analog Signal Into Digital

Signal is Pictorically Represented As:
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Advantages Of Digital Communication:
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1. The digital communication systems are cheaper compared to analog communication

systems because of the advances made in the IC technologies.
2. In digital communication, the speech, video and other data may be merged and
transmitted over a common channel using multiplexing.
3. Using data encryption, only permitted receivers may be allowed to detect the transmitted
data. This property is most important in military applications.
4. Since the transmission is digital and the channel encoding is used, therefore the noise
does not accumulate from repeater to repeater in long distance communications.
5. Since the transmitted signal is digital in nature, therefore a large amount of noise
interference may be tolerated.
6. Since in digital communication, channel coding is used, therefore the errors may be
detected and corrected in the receivers.
7. Digital communication is adaptive to other advanced branches of data processing such as
digital signal processing, image processing and data compression etc.
8. We can avoid signal jamming using spread spectrum techniques.
Disadvantages of Digital Communication:
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Although digital communication offers, so many advantages as discussed above, it has some
drawbacks also. However, the advantages of digital communication outweigh disadvantages. The
disadvantages are as under:
1. Large System Bandwidth: - Digital transmission requires a larger system bandwidth to
communicate the same information in a digital format as compared to analog format.
2. System Synchronization:- Digital detection requires system synchronization whereas
the analog signals generally have no such requirement.
3. System Complexity:- Digital system are more complex as compared to the analog
systems.
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Raghudathesh G P
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Comparison chart For Analog and Digital System:

Analog
Technology
Data
transmissions
Response to
Noise
Flexibility
Uses
Applications
Bandwidth
Can be used in analog devices

only. Best suited for audio and
video transmission.
Thermometer
Analog signal processing can be
done in real time and consumes
less bandwidth.
PCs, PDAs
There is no guarantee that digital signal
processing can be done in real time and
consumes more bandwidth to carry out
the same information.
Stored in the form of wave signal Stored in the form of binary bit
Analog instrument draws large
Digital instrument drawS only negligible
power
power
Low cost and portable
Cost is high and not easily portable
Low
High order of 100 megaohm
Analog instruments usually have a Digital instruments are free from
scale which is cramped at lower
observational errors like parallax and
end and give considerable
approximation errors.
observational errors.
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Memory
Example
Denoted by square waves

Uses discrete or discontinuous values to
represent information
Computers, CDs, DVDs, and other
digital electronic devices.
Samples analog waveforms into a
limited set of numbers and records them.
Can be noise-immune without
deterioration during transmission and
write/read cycle.
Less affected since noise response are
analog in nature
Digital hardware is flexible in
implementation.
Best suited for Computing and digital
electronics.
Representation
Digital signals are discrete time signals

generated by digital modulation.
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Waves
Analog signal is a continuous

signal which represents physical
measurements.
Denoted by sine waves
Uses continuous range of values
to represent information
Human voice in air, analog
electronic devices.
Analog technology records
waveforms as they are.
Subjected to deterioration by
noise during transmission and
write/read cycle.
More likely to get affected
reducing accuracy
Analog hardware is not flexible.
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Signal
Digital
Power
Cost
Impedance
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Errors
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Raghudathesh G P
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Basic Signal Processing Operations In Digital Communications:
The figure shows the block diagram of a digital communication system.
In this diagram, three basic signal-processing operations are identified:

1. Source coding
2. Channel coding, and
3. Modulation.
It is assumed that the source of information is digital by nature or converted into it by

design.
Source Coding:
The encoder maps the digital signal generated at the source output into another
signal in digital form. The mapping is one-to-one, and the objective is to eliminate
or reduce redundancy so as to provide an efficient representation of the source
output.
As the source encoder mapping is one-to-one, the source decoder simply performs
the inverse mapping and thereby delivers to the user destination a reproduction of
the original digital source output.
The primary benefit thus gained from the application of source coding is a
reduced bandwidth requirement.
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Channel Coding:
The objective is for the encoder to map the incoming digital signal into a channel
input and for the decoder to map the channel output into an output digital signal in
such a way that the effect of channel noise is minimized.
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Raghudathesh G P
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That is, the combined role of the channel encoder and decoder is to provide for
reliable communication over a noisy channel.
Above provision is satisfied by introducing redundancy in a prescribed fashion in
the channel encoder and exploiting it in the decoder to reconstruct the original
encoder input as accurately as possible.
Thus, in source coding, we remove redundancy, whereas in channel coding, we introduce
controlled redundancy.
Clearly, we may perform source coding alone, channel coding alone, or the two together.
In the latter case, naturally, the source encoding is performed first, followed by channel
encoding in the transmitter as illustrated in figure.
In the receiver, we proceed in the reverse order; channel decoding is performed first,
followed by source decoding.
Whichever combination is used, the resulting improvement in system performance is

achieved at the cost of increased circuit complexity.
Modulation:
It is performed with the purpose of providing for the efficient transmission of
the signal over the channel.
In particular, the modulator (constituting the last stage of the transmitter in the
figure) operates by keying shifts in the amplitude, frequency, or phase of a
sinusoidal carrier wave to the channel encoder output.
The digital modulation technique for so doing is referred to as amplitude-shift
keying, frequency-shift keying, or phase-shift keying, respectively.
The detector (constituting the first stage of the receiver in the figure) performs
demodulation (the inverse of modulation), thereby producing a signal that follows the
time variations in the channel encoder output (except for the effects of noise).
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The combination of modulator, channel, and detector, enclosed inside the dashed
rectangle shown in the figure, is called a discrete channel. It is so called since both its
input and output signals are in discrete form.
Traditionally, coding and modulation are performed as separate operations, and the
introduction of redundant symbols by the channel encoder appears to imply increased
transmission bandwidth.
In some applications, however, these two operations are performed as one function in
such a way that the transmission bandwidth need not be increased.
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Raghudathesh G P
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Basic Concepts:
1. Source Coding:
This is done to reduce the size of the information (data compression) being transmitted
and conserve the available bandwidth.
This process reduces redundancy.
Ex. zipping files, video coding (H.264, AVS-China, Dirac) etc.
This process adds to the redundancy of data.

Ex. Turbo codes, convolution codes etc.
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2. Channel Coding:
This is done to reduce errors during transmission of data along the channel from the
source to the destination.
Channels for Digital Communications:
The modulation and coding used in a digital communication system depend on the
characteristics of the channel.
The two main characteristics of the channel are BANDWIDTH and POWER. In
addition the other characteristics are whether the channel is linear or nonlinear, and how
free the channel is free from the external interference.
Five channels are considered in the digital communication, namely:

1. Telephone channels
2. Coaxial cables
3. Optical fibers
4. Microwave radio
5. Satellite channels.
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1. Telephone channel:
It is designed to provide voice grade communication. Also good for data communication
over long distances.
The channel has a band-pass characteristic occupying the frequency range 300Hz to
3400hz, a high SNR of about 30db, and approximately linear response.
For the transmission of voice signals the channel provides flat amplitude response. But
for the transmission of data and image transmissions, since the phase delay variations
are important an equalizer is used to maintain the flat amplitude response and a linear
phase response over the required frequency band.
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Transmission rates upto16.8 kilobits per second have been achieved over the telephone
lines.
2. Coaxial Cable:
The coaxial cable consists of a single wire conductor centered inside an outer conductor,
which is insulated from each other by a dielectric.
The main advantages of the coaxial cable are wide bandwidth and low external
interference. But closely spaced repeaters are required.
With repeaters spaced at 1km intervals the data rate of 274 megabits per second have
been achieved.
3. Optical Fibers:
An optical fiber consists of a very fine inner core made of silica glass, surrounded by a
concentric layer called cladding that is also made of glass.
The refractive
index of the glass in the core is slightly higher than refractive index of the glass in the
cladding. Hence if a ray of light is launched into an optical fiber at the right oblique
acceptance angle, it is continually refracted into the core by the cladding. That means the
difference between the refractive indices of the core and cladding helps guide the
propagation of the ray of light inside the core of the fiber from one end to the other.
Compared to coaxial cables, optical fibers are smaller in size and they offer higher
transmission bandwidths and longer repeater separations.
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4. Microwave radio:
A microwave radio, operating on the line-of-sight link, consists basically of a transmitter

and a receiver that are equipped with antennas.
The antennas are placed on towers at sufficient height to have the transmitter and receiver
in line-of-sight of each other.
The operating frequencies range from 1 to 30 GHz.
Under normal atmospheric conditions, a microwave radio channel is very reliable and
provides path for high-speed digital transmission. But during meteorological variations,
a severe degradation occurs in the system performance.
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5. Satellite Channel:
A Satellite channel consists of a satellite in geostationary orbit, an uplink from ground

station, and a down link to another ground station.
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Both link operate at microwave frequencies, with uplink the uplink frequency higher than
the down link frequency.
In general, Satellite can be viewed as repeater in the sky. It permits communication over
long distances at higher bandwidths and relatively low cost.
Sampling theorem:
Necessity:
There are two types of signals:
1. Continuous time signal and
2. Discrete time signal
Due to advance development in digital technology over the past few decades, the
inexpensive, light weight, programmable and easily reproducible discrete time
systems are available. Processing of discrete-time signals is more flexible and is
also preferable to processing of continuous time signals.
For the above purpose we should be able to convert a continuous time signal into
discrete-time signal.
This problem is solved by a fundamental mathematical tool known as sampling
theorem.
The sampling theorem is extremely important and useful in signal processing.
With the help of sampling theorem a continuous time signal may be completely
represented and recovered from the knowledge of sample taken uniformly. This
means that sampling theorem provides a mechanism for representing continuous
time signal by discrete-time signal.
Therefore, sampling theorem may be viewed a bridge between continuous time
signals and discrete-time signals.
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Definition:
The statement of sampling theorem for low pass signal can be given in two parts
as:
1. A band limited signal of finite energy, which has no frequency-component
higher than fm Hz, is completely described by its sample values at
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uniform intervals less than or equal to
second apart.
2. A band limited signal of finite energy, which has no frequency

components higher than fm Hz, may be completely recovered from the
knowledge of its samples taken at the rate 2fm samples per second.
The first part represents the representation of the signal in its samples and
minimum rate required to represent a continuous time signal into its samples.
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Classification:
Ideal or Instantaneous or Impulse Sampling.
Natural or Chopper sampling.
Flat top sampling.
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The second part of the theorem represents reconstruction of the original signal
from its sample.
It gives sampling rate required for satisfactory reconstruction of signal from its
samples.
Combining the two parts the sampling theorem may be stated as under:
"A continuous-time signal may be completely represented in its samples and
recovered back if the sampling frequency is fs 2fm.
Here,
fs = sampling frequency and
fm = the maximum frequency present in the signal.
Proof of Sampling Theorem For Low Pass (LP) Signals:
Consider an analog signal g(t) that is continuous in time and amplitude.
We assume that g(t) has infinite duration, but finite energy.

Let the sample values of the signal g(t) at times t = 0, T s, 2Ts, 3T s,., be denoted by
the series{g(n Ts), n = 0, 1,2,}.
Here,
Ts = sampling period and
fs (1/ Ts) = sampling rate
A segment of the signal g(t) is as shown below
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Proof involves two parts:

1. Representation of g(t) interms of its samples.
2. Reconstruction of g(t) from its samples.
Proof:
The discreate time signal,
, resulted from the sampling process is given as below,
In the above equation delta function is located at time t = nTs. Equation 1 can be written
as,
Here,
--------- (2)
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-------------- (1)
= Dirac comb or ideal sampling function.

Applying Fourier transform to the above equation. Let G(f) and G(f) denotes the fourier
transforms of g(t) and g(t) respectively, we get,
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--------- (3)
From the above equation we see that G(f) represents a spectrum that is periodic in the
frequency f with period fs.
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The process of uniform sampling a signal in time domain results in a periodic spectrum in
the frequency domain with a period equal to the sampling rate.
Rewriting the above equation we get,
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------------ (4)
Above equation is passed through the lowpass filter we get,
--------- (5)
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Applying fourier transform to equation (2) we get,
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--------- (6)
Putting equation (6) in equation (5) and fs = 2W we get,
If the sample values
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------- (7)
of the signal are known for all time n, then the Fourier
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transform G(f) of the signal is determined uniquely by the equation (7).
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Reconstruction Of The Signal g(t) From Its Sequence Of Sample Values

:
The process of reconstructing a continuous time signal g(t) from its samples is called
interpolation.
Consider the equation below:
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-------- (1)
Appling Inverse Fourier Transform (IFT) to the above equation
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Here the limits of integration can be taken from W to +W, also interchanging the order
of integration and summation and combining the exponential terms we get,
But
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thus,
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and here
We know that
, thus above equation is reduced to,
-------------- (2)
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The above Equation is known as Interpolation formula for reconstruction of g(t) from its
sequence of sample values
where each sample is multiplied by a delayed Sinc
function.
Effect on Spectrum Due to the selection of Sampling Rate:
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Case 1: fs > 2W
Important Points Relating to Sampling:
Figure above shows the spectrum of sampled signal.
the spectrum G(W) or G(f) will repeat periodically without overlapping.

The spectrum of sampled signal extends up to infinity and the ideal bandwidth of
sampled signal is infinite.
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The original or desired spectrum is centered at W = 0 or f = 0 and is having bandwidth or

maximum frequency equal to Wm. The desired spectrum may be recovered by passing
the sampled signal with spectrum G(w) through a low pass filter with cut-off frequency
Wm.
This means that since a low-pass filter allows to pass only low frequencies up to cut-off
frequency (Wm) and rejects all other higher frequencies, the original spectrum extended
upto Wm will be selected and all other successive higher frequency cycles in the
sampled-spectrum will be rejected. Therefore, in this way, original spectrum will be
extracted out of spectrum G(w). This original spectrum can now be converted into timedomain signal.
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It may also be observed from figure that for the Case fs > 2fm, the successive cycles of
G(w) are not overlapping each other. Hence in this case, there is no problem in
recovering the original spectrum.
In this case there will be guard band between components of the spectrum components.
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Case 2: fs = 2W:
For this case the successive cycles of G(w) are not overlapping each other, but they are
touching each other.
In this case also, the original spectrum can be recovered from the sampled spectrum G(w)
using a low-pass filter with cut-off frequency Wm ideally.
Practically no filter has a sharp roll-off. Thus, practically it is not possible to recover tha
signal g(t).
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Case 3: fs < 2W:
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The successive cycles, of the sampled spectrum will overlap each other and hence in this
case, the original spectrum cannot be extracted out of the spectrum G(w).
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Thus, the output of LPF will have distortion due to the unwanted frequency component.
This is known as Aliasing. This type of sampling is known as Under Sampling.
Hence, for reconstruction without distortion, Sampling Frequency must be f s

2fm
Basic Concepts:
3. Nyquist Rate:
When the sampling rate becomes exactly equal to 2 fm samples per second, then it is
called Nyquist rate.
Nyquist rate is also called the minimum sampling rate. It is given by
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fs 2fm
4. Nyquist Interval:
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Maximum sampling interval is called Nyquist interval. It is given by
Effect of Under Sampling (Aliasing):
When a continuous rime band limited signal is sampled at a lower than Nyquist rate fs <
2fm, then the successive cycles of the spectrum G(w) of the sampled signal g(t) overlap
with each other as shown below
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Raghudathesh G P
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Here the signal is under-sampled and some amount of aliasing is produced in this undersampling process.
In fact, aliasing is the phenomenon in which a high frequency component in the

frequency-spectrum of the signal takes identity of a lower-frequency component in
the spectrum of the sampled signal.
Because of the overlap due to aliasing phenomenon, it is not possible to recover original
signal from sampled signal g(t) by low-pass filtering since the spectral components in the
overlap regions add and hence the signal is distorted.
Since any information signal contains a large number of frequencies, so, to decide a
sampling frequency is always a problem. Therefore, a signal is first passed through a lowpass filter. This low-pass filter blocks all the frequencies which are above fm Hz. This
process is known as bandlimiting of the original signal.
This low-pass filter is called prelias filter because it is used to prevent aliasing effect.
After band limiting, it becomes easy to decide sampling frequency since the maximum
frequency is fixed at fm Hz.
Steps to avoid aliasing:

1. Prealias filter must be used to limit band of frequencies of the signal to fm Hz.
2. Sampling frequency 'fs must be selected such that fs > 2 fm.
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Sampling of Bandpass Signal:
Definition: A bandpass signal say x(t) whose maximum bandwidth is 2W can be

completely represented into and recovered from its samples if it is sampled at the
minimum rate twice the bandwidth( i.e, 4W).
Generation of In and Quadrature component of a bans pass signal:
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---------- (1)
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In this scheme, the band pass signal is converted into lowpass signals by multiplying
bandpass signals with two sinusoidal signals which are phase Quadrant with each
other results in into two components, one is in-phase component and other is
quadrature component.
These two components will be lowpass signals and are sampled separately. This
form of sampling is called quadrature sampling.
Let g(t) be a band pass signal, of bandwidth 2W centered around the frequency,
fc,(fc>W). The in-phase component, gI(t) is obtained by multiplying g(t) with
cos(2fct) and then filtering out the high frequency components.
Parallelly a quadrature phase component is obtained by multiplying g(t) with
sin(2fct) and then filtering out the high frequency components..
The band pass signal g(t) can be expressed as,
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The in-phase, gI(t) and quadrature phase gQ(t) signals are lowpass signals,
having band limited to (-W < f < W). Accordingly each component may be
sampled at the rate of 2W samples per second as shown.
The spectrum of bandpass signal and inphase and quadreture component is as
shown below.
Reconstruction:
From the sampled signals gI(nTs) and gQ(nTs), the signals gI(t) and gQ(t) are
obtained.
To reconstruct the original band pass signal, multiply the signals g I(t)and gQ(t) by
cos(2fct) and sin(2fct) respectively and then add the results.
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Practical Aspects of Sampling and Signal Recovery:
In the previous section we studied impulse or ideal sampling. The samples were impulses
of height equal to modulating signal at sampling instant.
But this type of ideal sampling is not possible in practice.

Practical sampling differs from ideal sampling in following aspects:
1. The sampled pulses have finite duration rather than impulses. The amplitude of
these pulses is also finite.
2. The practical signal to be sampled are not strictly bandlimited. Therefore, there is
always problem in selecting precise sampling frequency.
3. The practical reconstruction filters are not ideal filters. The practical
reconstruction filters need a guard band or gap between the spectrums of sampled
signal.
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Flat top normally preferred because of its noise immunity.
The signals to be sampled are not strictly bandlimited, there is always chance of aliasing.
Aliasing occurs because of some high frequency spurious signals in such cases.
To avoid this problem, normally prealias filters are used before sampling. These prealias
filters are normally lowpass bandlimiting filters.
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Basically, there are two types of sampling techniques as under:

Ideal sampling:
Instantaneous sampling.
practical sampling:
Natural sampling or chopper Sampling
Flat-top sampling or rectangular pulse sampling
Natural sampling and flat top sampling are practical sampling methods.
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Natural Sampling or Chopper Sampling:

Drawback of Instantaneous Sampling:
1. The instantaneous sampling results in the samples whose width approaches zero.
Due to this, the power content in the instantaneously sampled pulse is negligible.
2. Thus, this method is not suitable for transmission purpose.
Operational Logic:
In natural sampling the pulse has a finite width equal to .
Consider an analog continuous-time signal x(t) to be sampled at the rate of fs Hz.
Here it is assumed that fs is higher than Nyquist rate such that sampling theorem
is satisfied.
Consider a sampling function c(t) which is a train of periodic pulses of width
and frequency equal to fs Hz.
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Figure shows a functional diagram of a natural sampler. With the help of this
natural sampler, a sampled signal g(t) is obtained by multiplication of sampling
function c(t) and input signal x(t).
In the figure when c(t) goes high the switch S is closed.
Therefore,
g(t) = x(t) when c(t) = A
g(t) = 0 When c(t) = 0
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Where A is the amplitude of c(t).

The waveforms of signals x(t), c(t) and g(t) have been illustrated in figure (a), (b) and (c)
below respectively.
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Now, the sampled signal g(t) may also be described mathematically as
Here,
------- (1)
c(t) = the periodic train of pulse of width and frequency fs.
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Exponential Fourier series for any periodic waveform is expressed as
-------- (2)
GH
U
Also, for the periodic pulse train of c(t). we have
RA
Therefore according to equation (2) for periodic pulse train c(t), we get
-------- (3)
Here
As c(t) is a rectangular pulse train, thus Cn for the this waveform is given as,
Department of ECE
Page No -25
Raghudathesh G P
Asst Professor
------- (4)
Here,
TA = pulse width =
fn = harmonic frequency = nfs.
Thus, equation (4) can be written as,
Substituting equation (5) in (3) we get,
GH
U
---------- (7)
In order to get the frequency- domain representation of the naturally sampled signal g(t),
we take the Fourier transform of the above signal
----- (8)
RA
-------- (6)
Substituting equation (6) in (1) we get the required time-domain representation for
naturally sampled signal x(t),
DA
TH
ES
------ (5)
But,
But, fn = harmonic frequency = nfs

Thus the spectrum of naturally sampled signal is given as below,
Department of ECE
Page No -26
Raghudathesh G P
Asst Professor
-------- (9)
Thus, the above equation shows that the spectrum x(t) i.e, X(f) are periodic in f s and are
weighted by the sinc function.
TH
ES
DA
Flat Top Sampling or Rectangular Pulse Sampling:
Flat top sampling is a practically possible sampling method.
Natural sampling is little complex whereas it is quite easy to get flat top samples.
In flat-top sampling or rectangular pulse sampling, the top of the samples remains
constant and is equal to the instantaneous value of the baseband signal x(t) at the start of
sampling.
GH
U
The duration or width of each sample is and sampling rate is equal to fs = 1/Ts.
Figure (a) below shows the functional diagram of a sample and hold circuit which is used
to generate the flat top samples.
RA
Figure (b) below shows the general waveform of flat top samples.
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Page No -27
Raghudathesh G P
Asst Professor
From figure (b), it may be noted that only starting edge of the pulse represents
instantaneous value of the baseband signal x(t).
Also the flat top pulse of g(t) is mathematically equivalent to the convolution of
instantaneous sample and a pulse h(t) as depicted in figure (c) below.
This means that the width of the pulse in g(t) is determined by the width of h(t) and the
sampling instant is determined by delta function.
In figure (b), the starting edge of the pulse represents the point where baseband signal is
sampled and width is determined by function h(t). Therefore g(t) will be expressed as
DA
----- (1)
This equation has been explained in figure (d) below.
RA
GH
U
TH
ES
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Page No -28
Raghudathesh G P
Asst Professor
The train of impulses may be represented mathematically as

------ (2)
The signal s(t) is obtained by multiplication of baseband signal x(t) and
Now, the sampled signal g(t) is given by equation (1) as,
Thus,
GH
U
DA
TH
ES
------ (3)
According to the shifting property of delta function,
RA
------ (4)
------- (5)
From equation (4) and (5),
------- (6)
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Raghudathesh G P
Asst Professor
This equation represents value of g(t) in terms of sampled value x(nT s) and function h(tnTs) for flat top sampled signal.
Now, again from equation (1), we have,
Taking Fourier transform of both sides of above equation we get,
But S(f) is given as,
Substituting equation (8) in (76) we get,
------- (9)
The above is the equation for the spectrum of flat top sampled signal.
Aperture Effect:
GH
U
------- (8)
DA
TH
ES
------ (7)
RA
The spectrum of flat top sampled signal is given by,
This equation shows that the signal g(t) is obtained by passing the signal s(t) through a
filter having transfer function H(f).
The corresponding impulse response h(t) in time domain is as shown in figure below.
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Page No -30
Raghudathesh G P
Asst Professor
The above is one pulse of rectangular pulse train where each sample of x(t)(i.e., s(t)) is
convolved with this pulse.
The spectrum of one rectangular pulse of h(t) is as shown in figure below.
The spectrum of one rectangular pulse is expressed as,
TH
ES
------ (1)
Seeing figure above, it may be observed that by using flat top samples an amplitude
distortion is introduced in the reconstructed signal x(t) from g(t).
In fact, the high frequency roll off H(f) acts like a low-pass filter and thus attenuates the
upper portion of message signal spectrum. These high frequencies of x(t) are affected.
This type of effect is known as aperture effect.
The aperture effect can be compensated by:
1. Reducing the pulse width of h(t) very small.
GH
U
DA
Using equalizer in cascade with the reconstruction filter in the receiver as shown below
we can compensate for attenuation,
RA
Equalizer used in cascade with the reconstruction filter has the effect of decreasing the
inband loss of the reconstruction filter as the frequency increases in such a way as to
compensate for the aperture effect.
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Page No -31
Raghudathesh G P
Asst Professor
Also, the transfer function of the equalizer is expressed as,

------ (1)
Here td' is known as the delay introduced by low-pass filter which is equal to
.Thus,
Which is the transfer function of an equalizer.
TH
ES
Sample and Hold Circuit for Signal Recovery:
------- (2)
Sampling process may takes the form of:

1. Natural sampling or
2. Flattop sampling.
In both cases, the spectrum of the sampled signal is scaled by the ratio T/T s .
Here,
Typically this ratio is quite small, with the result that the signal power at the output of the
lowpass reconstruction filter in the receiver is correspondingly small.
GH
U
DA
T = sampling-pulse duration and

Ts = sampling period.
Obvious remedy to this situation is the use of amplification, which can be quite large.
A more attractive approach, however, is to use a simple sample and hold circuit.
RA
The circuit, shown in the figure above, consists of an amplifier of unity gain and low
output impedance, a switch, and a capacitor; it is assumed that the load impedance is
large.
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Page No -32
Raghudathesh G P
Asst Professor
The switch is timed to close only for the small duration T of each sampling pulse, during
which time the capacitor rapidly charges up to a voltage level equal to that of the input
sample.
When the switch is open, the capacitor retains its voltage level until the next closure of
the switch.
Thus the sample-and-hold circuit, in its ideal form, produces an output waveform that
represents a staircase interpolation of the original analog signal, as shown in figure
below.
With reference to flat-top sampling, the output of a sample-and-hold circuit is defined as,
DA
TH
ES
Here,
-------- (1)
GH
U
h(t) = impulse response representing the action of the sample and hold circuit i.e.,
Spectrum of the output of the sample-and-hold circuit given by,
RA
-------- (2)
------- (3)
Here,
G(f) = the Fourier transform of the original analog signal g(t).

H(f) = transfer function of the sample and hold circuit and is given by
-------- (4)
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Page No -33
Raghudathesh G P
Asst Professor
In order to recover the original signal g(t) Without distortion, we need to pass the output
of the sample-and-hold through a low-pass filter designed to remove components of the
spectrum U(f) at multiples of the sampling rate fs and an equalizer whose amplitude
response equals 1/|H( f)|.
These operations arc illustrated by the block diagram shown in figure below
RA
GH
U
DA
TH
ES
Department of ECE
Page No -34
Raghudathesh G P
Asst Professor
Comparison of Various Sampling Techniques:
Waveforms
involved
Feasibility
P
This method is also
used practically
Sampling
rate
satisfies
Nyquist
criteria
Noise interference
is maximum
Frequency
domain
representatio
n
RA
Noise
Noise interference Noise interference is minimum
interference
is maximum
Time domain
representatio
n
DA
This is not a This method is used practically

practically
possible method
Sampling rate Sampling
rate Sampling rate satisfies Nyquist
tends to infinity
criteria
GH
U
It uses sample and

hold circuit
Sampling
principle
Generation
circuit
Flat top Sampling
Ideal or
Natural sampling
instantaneous
sampling
It
uses It uses chopping principle
multiplication
TH
ES
Sl Parameter of
No comparison
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Page No -35
Raghudathesh G P
Asst Professor
Formulas
1) Nyquist rate
2) Sampling period
3) Angular Frequency
4)
TH
ES
5)
6)
GH
U
8)
DA
7) For Band Pass Signal :Inphase and quadrature component
9)
RA
10)
11) consider 3 frequencies
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Then
Page No -36
Raghudathesh G P
Asst Professor
Problems:
1. Consider the cosine wave
derived by sampling g(t) at the times
. Plot the spectrum of discrete time signal

where n=0,
,and
(i)
(ii)
(iii)
,comment on the results.
DA
TH
ES
repeat your calculations for the sine wave

Solution:
given
Taking FT on both sides of equation (1)
GH
U
Spectrum of message signal g(t).

W.k.t
RA
i)
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Page No -37
Raghudathesh G P
Asst Professor
GH
U
DA
TH
ES
2. The signal g(t)=10 cos(20

second.
is sampled at the rate of 250 samples per
Determine spectrum of resulting sampled signal.

Specify the cutoff frequency of ideal reconstruction filter so as to recover g(t)
from its sampled version.
What is the Nyquist rate for g(t)?
RA
a)
b)
cos(200
c)
Solution:
given
Taking FT on both sides of above equation
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Page No -38
Raghudathesh G P
Asst Professor
Recall;
DA
TH
ES
The spectrum of signal g(t) is drawn using above expression
is
RA
GH
U
Using above equation spectrum is drawn as follows .it should be noted that
symmetric about vertical axis and is periodic with a period equal to
.
(b) the cutoff frequency of ideal LPF should be greater than 110HZ and less than 140HZ for
recovering g(t) from
.
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(c) Nyquist rate ,
Raghudathesh G P
Asst Professor
=2W, where W is highest frequency in g(t).thus
of the second frequency component.
Find the values of A and

Solution:
3. A signal g(t) consists of two frequency components =3.9KHZ and =4.1KHZ in such
a relationship they cancel out each other when g(t) is sampled at the instants
t=0,T,2T,..,where T=125 micro sec. The signal g(t) is defined by
Also given in problem that,
TH
ES
given
g(nT)=0
GH
U
DA
Hence ,
i)
Let n=0
Then, equation (1) becomes
RA
A 0, we get
ii)
Let n=1; equation (1) becomes
Department of ECE
Page No -40
Raghudathesh G P
Asst Professor
The above equation is satisfied only when A = 1 and +ve sign for the second term is taken. The
+ve sign can appear in the above equation, only when
in equation (1). Thus , A=1 and
TH
ES
Where W is the highest frequency component of g(t).

Solution:
If g(t) is band-limited to
, we may write g(t) as
4. If E denotes the energy of a strictly band limited signal g(t), then prove that
DA
But
RA
GH
U
Put (1) in (2)
But
Hence equation (3) becomes
Department of ECE
Page No -41
Raghudathesh G P
Asst Professor
= 75Hz.
4. The signal (t) = 10 cos(100t) and (t) =10 cos50t are both sampled with
Show that the two sequences so obtained are identical.
Solution:
given (f) = 5[ (f 50) + (f + 50)]
= 75Hz, the spectrum of sampled version of
(f) =
=
Similarly ,the ft of
(t) is
GH
U
DA
(f) =
(t) is
TH
ES
With
Letting m=l 1 in the first term of the equation (2) and m=k 1 for the second term we get
(f) = 375
(3)
+ 375
RA
Since l and k are dummy variables ,they can be replaced by m. Consequently equation (3)
becomes
Hence from equation (1) and (3) sampled version of
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(t) and
(t) are identical.
Page No -42
Raghudathesh G P
Asst Professor
(t) and
(t) are lowpass signals with a bandwidth of
The Nyquist rate for
(t) and
=2W = 20.3 = 0.6KHz.
TH
ES
Where
5. The spectrum of bandpass signal g(t) has bandwidth of 0.6KHz centered around
12KHz.find the Nyquist rate for quadrature sampling of inphase and quadrature
components of the signal g(t).
Solution:
The signal g (t) can be expressed in terms of inphase and quadrature components as
(t) is therefore
RA
GH
U
DA
6. A signal g(t) = 2cos(400t) + 6cos(640t) is ideally sampled at 500Hz. If the sampling

signal is passed through an ideal lowpass filter with a cutoff frequency of 400Hz ,what
components will appear in filter output?
Solution:
g(t) = 2cos(400t) + 6cos(640t) (1)
From eq (1)
=400 ,hence = 200Hz.
= 640, hence = 320Hz.
taking FT on both sides of the eq (2) we get
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Page No -43
Raghudathesh G P
Asst Professor
W.k.t ;
TH
ES
ii)
Using eq (3) ,the spectrum of the signal g(t) is drawn and it appears as shown below
-500n)+(f +200 -500n) ]

f-500n)+(f +320 -500n) ] } (4)
is drawn using eq (4)
RA
GH
U
DA
f+3
The spectrum of the sampled signal
The components that are appeared at the filter o/p are 180Hz, 200Hz, 300Hz, 320Hz.
7. An analog signal is expressed by the equation

. Calculate the nyquist rate and nyquist interval for this signal.
Solution:
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Page No -44
Raghudathesh G P
Asst Professor
eq(1) is in the form
= max(
Nyquist rate
=2
Nyquist interval
G
he e
= 150Hz
= 2150Hz
=300Hz
=25Hz
= 150Hz
= 50Hz.
TH
ES
comparing equation (1) and (2)

hence
= 3.3msec.
Solution:
GH
U
Given signal
DA
8. Find the Nyquist rate and Nyquist interval for the signal
RA
w.k.t
Equation (2) is in the form of
Department of ECE
Page No -45
Raghudathesh G P
Asst Professor
=1500Hz
= 2500Hz
= max(
Nyquist rate
. hence
=2
= 2500Hz
= 22500Hz
= 0.2msec.
Nyquist interval
=5000Hz
TH
ES
9. The signal
is sampled at rate of 500 samples/sec.
i) Determine the spectrum of the resulting sampled signal .
ii) What is the nyquist rate for g(t).
iii) Specify the cutoff frequency of ideal reconstruction filter .
Solution:
Given
GH
U
DA
w.k.t
from eq (2)
RA
=198Hz
= 202Hz
taking ft on both sides of eq (2) ,we get
spectrum of G(f) is as shown in below figure:
= max(
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he e
= 202Hz
Page No -46
Raghudathesh G P
Asst Professor
w.k.t
- 500n) + G(f +198 - 500n) ]

+
f-250n)+G(f +110 -250n)]}
GH
U
DA
TH
ES
f-
i)
fig
RA
ii)
iii)
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Cutoff frequency of ideal reconstruction filter is 202Hz.

Nyquist rate =2 = 2202Hz
=404Hz.
Page No -47
Raghudathesh G P
Asst Professor
11. A continuous time is given as

Determine
i)
Minimum sampling rate i.e ,Nyquist rate required to avoid aliasing.
ii)
If sampling frequency
= 400Hz . what is the discrete time signal x[n] or
x[n ] obtained after sampling?
iii)
of the sinusoidal that yields samples
What is the frequency
identical to those obtained in part ii).
Solution:
The highest frequency component of continuous time signal is f =100Hz.hence

minimum sampling rate required to avoid aliasing is called Nyquist rate and is give
Given as Nyquist rate =
=200Hz.
ii)
The continuous time signal x(t) is sampled at = 400Hz . The frequency of discrete
time signal will be
TH
ES
i)
The continuous time signal x(t) is sampled at

time signal will be
= 150Hz . the frequency of discrete
GH
U
iii)
DA
Then, the discrete time signal is given by
RA
Then the discrete time signal will be given by
iv)
For sampling rate of
= 150Hz
Then, the sinusoidal signal will be
Department of ECE
Page No -48
Sampling at
Raghudathesh G P
Asst Professor
= 150Hz, yields identical samples hence f= 100Hz is an aliasing of f= 150Hz.
12. Determine the Nyquist rate for a continuous time signal
Solution:
In general form, any continuous time signal can be written as
and the given signal is
=50/2 =25HZ
=300/2 =150HZ
=100/2 =50HZ
DA
TH
ES
On comparing the signal (1) and (2) ,we obtain the frequencies of given signal as follows
GH
U
Thus highest frequency component of given message signal will be

= 150Hz
Therefore, Nyquist rate = 2
=2 x 150 =300Hz.
RA
13. Figure shows spectrum of the arbitrary signal x(t) . This signal is sampled at the
Nyquist rate with a periodic train of triangular pulses of duration 50/3. Milliseconds
.Determine the spectrum of the sampled signal for frequencies upto 50Hz giving relevant
expression .
Solution:
from fig it may be observed that signal is bandlimited to 10Hz.
Thus
= 10Hz
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Page No -49
Raghudathesh G P
Asst Professor
So Nyquist rate is = 2 = 210 = 20Hz

Since the signal is sampled at Nyquist rate,the sampling frequency would be
= 20Hz
Given that the rectangular pulses are used for sampling i.e flat top sampling is used.
The spectrum of flat top sampled signal is given by
Value of H(f) is expressed as
=0.05/3 sec
TH
ES
Here is the value of rectangular pulse used for sampling .

The given value of rectangular sampling pulse duration is 50/3 milliseconds.
DA
Substituting the value of in eq(2) we get
in equation(1) , we get
(
=20)
RA
GH
U
Again, putting this value of H(f) and
This expression gives the spectrum up to 60Hz (since n = 3) for the sampled signal.
14. A flat top sampling system samples a signal of maximum frequency with 3.5Hz
sampling frequency. The duration of the pulse is 0.2 sec. Compute amplitude distortion due
to aperture effect at the highest signal frequency. Also determine the equal characteristic.
Solution:
Given the sampling frequency = 2.5Hz
Department of ECE
Page No -50
Raghudathesh G P
Asst Professor
Maximum signal frequency

= 1Hz and pulse width = 0.2sec.
We know that the aperture effect is expressed by transfer function H(f) as
The magnitude of this equation will be
= 1Hz in eq(1)
(f)
(f) =
DA
Putting = 0.2 sec and assuming

K =1, the last equation becomes
TH
ES
Also, equaliser characteristic is expessed as
Now, aperture effect at the highest frequency will be obtained by putting f =
(f) versus f and it represents the equalisation characteristic to
GH
U
This equation is plot of
RA
overcome aperture effect.
Department of ECE
Page No -51
Raghudathesh G P
Asst Professor
Things to be Known:
Signals and Their Spectrum:
TH
ES
1. Single Sine Wave: f(x) = Sin (x)
GH
U
DA
2. Sum of Two Sine Waves f(x) = Sin (x) + Sin (2x)
RA
3. Box Function:
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Page No -52
Raghudathesh G P
Asst Professor
4. Wider Box Function:
TH
ES
5. Triangular Function:
GH
U
DA
5. Comb Function:
RA
6. Wider Comb Function:
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Page No -53
Raghudathesh G P
Asst Professor
Commonly Used Filters for Reconstruction:
TH
ES
1. Constant Filter:
GH
U
DA
2. Linear Filter:
RA
3. Catmull-Rom Filter:
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Page No -54
Raghudathesh G P
Asst Professor
4. B-Spline Filter:
GH
U
DA
TH
ES
5. Truncated Sinc Filter:
RA
6. Full Sinc Filter:
Department of ECE
Page No -55
Raghudathesh G P
Asst Professor
TH
ES
VTU Questions
7. Lanczos Filter:
DA
1. Explain the sampling theorem for low pass signals and derive the interpolation formula. June
2013 (09 M)
2. With a neat block diagram, explain the scheme for signal reconstruction for practical
sampling. June 2013 (06 M)
3. If E denotes the energy of a strictly band limited signal g(t), then prove that
RA
GH
U
Where W is the highest frequency component of g(t). June 2013 (05 M)

4. Explain natural sampling with relevant waveforms. Give all necessary time-domain and
frequency domain equations. June 2014 (10 M)
5. What is aliasing error? Give 2 corrective measures to remove the effect of aliasing in practice.
June 2014 (04 M)
6. Consider the analog signal x(t) = 5 cos(2000t) + 10 cos(6000t).
i. What is Nyquest rate and Nyquist interval?
ii. Assume that if we sample the signal using sampling frequency Fs = 5000 Hz, what is
the resulting discrete time signal obtained after sampling?
iii. Draw the spectrum of the sapling signal. June 2014 (04 M)
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Digital Communication VTU Unit 1

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Digital Communication VTU Unit 1

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Intoduction & Sampling Theorem

DIGITAL COMMUNICATION (VTU)-10EC61

Basic signal processing operations in digital communication. Sampling Principles: Sampling

Special Thanks To:

Trust is like a paper once it crumbled it cant be perfect again.

As soon as the fear approaches near, attack and destroy it.

Intoduction & Sampling Theorem

Introduction to Digital Communication

Definition2: Communication is simply the process of conveying message at a distance.

Definition3: Communication is the basic process of exchanging information.

Different communication equipments when assembled together form a communication

Sources and Signals:

A source of information generates a message.

The electrical waveform or message signal is also referred to as a baseband signal.

Ex: A speech signal, a television signal, and a signal representing atmospheric

Ex: Computer data and telegraph signals.

Intoduction & Sampling Theorem

The analog-to-digital conversion process described in figure above a segment of an

Intoduction & Sampling Theorem

Intoduction & Sampling Theorem

The 3 Basic Operation/Steps Used To Convert Analog Signal Into Digital

Intoduction & Sampling Theorem

Advantages Of Digital Communication:

1. The digital communication systems are cheaper compared to analog communication

Disadvantages of Digital Communication:

Intoduction & Sampling Theorem

Comparison chart For Analog and Digital System:

Can be used in analog devices

Denoted by square waves

Digital signals are discrete time signals

Analog signal is a continuous

Intoduction & Sampling Theorem

Basic Signal Processing Operations In Digital Communications:

The figure shows the block diagram of a digital communication system.

In this diagram, three basic signal-processing operations are identified:

It is assumed that the source of information is digital by nature or converted into it by

Intoduction & Sampling Theorem

Whichever combination is used, the resulting improvement in system performance is

Intoduction & Sampling Theorem

This process adds to the redundancy of data.

Channels for Digital Communications:

Five channels are considered in the digital communication, namely:

Intoduction & Sampling Theorem

A microwave radio, operating on the line-of-sight link, consists basically of a transmitter

The operating frequencies range from 1 to 30 GHz.

A Satellite channel consists of a satellite in geostationary orbit, an uplink from ground

Intoduction & Sampling Theorem

uniform intervals less than or equal to

2. A band limited signal of finite energy, which has no frequency

Intoduction & Sampling Theorem

Proof of Sampling Theorem For Low Pass (LP) Signals:

Consider an analog signal g(t) that is continuous in time and amplitude.

We assume that g(t) has infinite duration, but finite energy.

A segment of the signal g(t) is as shown below

Intoduction & Sampling Theorem

Proof involves two parts:

, resulted from the sampling process is given as below,

= Dirac comb or ideal sampling function.

Intoduction & Sampling Theorem

Rewriting the above equation we get,

Above equation is passed through the lowpass filter we get,

Applying fourier transform to equation (2) we get,

Intoduction & Sampling Theorem