Documenti di Didattica
Documenti di Professioni
Documenti di Cultura
Raghudathesh G P
Asst Professor
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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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.
Learn from the mistakes of others, you can't live long enough to make them all yourselves.
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Raghudathesh G P
Asst Professor
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.
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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.
An analog signal is one in which both amplitudes and time vary continuously over their
respective intervals.
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Raghudathesh G P
Asst Professor
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 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
Asst Professor
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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Raghudathesh G P
Asst Professor
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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
Asst Professor
Technology
Data
transmissions
Response to
Noise
Flexibility
Uses
Applications
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
Representation
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Waves
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Signal
Digital
Power
Cost
Impedance
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Errors
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Raghudathesh G P
Asst Professor
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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
Asst Professor
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.
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
Asst Professor
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.
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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.
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.
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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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Raghudathesh G P
Asst Professor
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:
The antennas are placed on towers at sufficient height to have the transmitter and receiver
in line-of-sight of each other.
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:
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Raghudathesh G P
Asst Professor
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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second apart.
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Raghudathesh G P
Asst Professor
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.
Here,
Ts = sampling period and
fs (1/ Ts) = sampling rate
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Raghudathesh G P
Asst Professor
Proof:
The discreate time signal,
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)
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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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Raghudathesh G P
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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.
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------------ (4)
--------- (5)
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Raghudathesh G P
Asst Professor
--------- (6)
Putting equation (6) in equation (5) and fs = 2W we get,
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------- (7)
of the signal are known for all time n, then the Fourier
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The process of reconstructing a continuous time signal g(t) from its samples is called
interpolation.
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-------- (1)
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Raghudathesh G P
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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
-------------- (2)
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Raghudathesh G P
Asst Professor
The above Equation is known as Interpolation formula for reconstruction of g(t) from its
sequence of sample values
function.
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Case 1: fs > 2W
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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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Raghudathesh G P
Asst Professor
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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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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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Raghudathesh G P
Asst Professor
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.
Basic Concepts:
3. Nyquist Rate:
When the sampling rate becomes exactly equal to 2 fm samples per second, then it is
called Nyquist rate.
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fs 2fm
4. Nyquist Interval:
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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
Asst Professor
Here the signal is under-sampled and some amount of aliasing is produced in this undersampling process.
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 band- limiting 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.
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Raghudathesh G P
Asst Professor
---------- (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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Raghudathesh G P
Asst Professor
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In the previous section we studied impulse or ideal sampling. The samples were impulses
of height equal to modulating signal at sampling instant.
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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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Raghudathesh G P
Asst Professor
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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Raghudathesh G P
Asst Professor
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Here,
------- (1)
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-------- (2)
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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,
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Raghudathesh G P
Asst Professor
------- (4)
Here,
TA = pulse width =
fn = harmonic frequency = nfs.
Thus, equation (4) can be written as,
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---------- (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)
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-------- (6)
Substituting equation (6) in (1) we get the required time-domain representation for
naturally sampled signal x(t),
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------ (5)
But,
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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.
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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.
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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.
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Figure (b) below shows the general waveform of flat top samples.
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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
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----- (1)
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Raghudathesh G P
Asst Professor
Thus,
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------ (3)
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------ (4)
------- (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.
------- (9)
The above is the equation for the spectrum of flat top sampled signal.
Aperture Effect:
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------ (7)
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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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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.
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------ (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.
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Using equalizer in cascade with the reconstruction filter in the receiver as shown below
we can compensate for attenuation,
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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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Raghudathesh G P
Asst Professor
.Thus,
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------- (2)
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.
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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.
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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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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,
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Here,
-------- (1)
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h(t) = impulse response representing the action of the sample and hold circuit i.e.,
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-------- (2)
------- (3)
Here,
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Raghudathesh G P
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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
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Raghudathesh G P
Asst Professor
Waveforms
involved
Feasibility
P
This method is also
used practically
Sampling
rate
satisfies
Nyquist
criteria
Noise interference
is maximum
Frequency
domain
representatio
n
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Noise
Noise interference Noise interference is minimum
interference
is maximum
Time domain
representatio
n
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Sampling
principle
Generation
circuit
Ideal or
Natural sampling
instantaneous
sampling
It
uses It uses chopping principle
multiplication
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Sl Parameter of
No comparison
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Raghudathesh G P
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Formulas
1) Nyquist rate
2) Sampling period
3) Angular Frequency
4)
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Page No -36
Raghudathesh G P
Asst Professor
Problems:
1. Consider the cosine wave
derived by sampling g(t) at the times
(i)
(ii)
(iii)
,comment on the results.
DA
TH
ES
GH
U
RA
i)
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Page No -37
Raghudathesh G P
Asst Professor
GH
U
DA
TH
ES
RA
a)
b)
cos(200
c)
Solution:
given
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Page No -38
Raghudathesh G P
Asst Professor
Recall;
DA
TH
ES
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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Page No -39
Raghudathesh G P
Asst Professor
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
TH
ES
given
g(nT)=0
GH
U
DA
Hence ,
i)
Let n=0
Then, equation (1) becomes
RA
A 0, we get
ii)
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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
TH
ES
4. If E denotes the energy of a strictly band limited signal g(t), then prove that
DA
But
RA
GH
U
But
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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
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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
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(t) and
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Page No -42
Raghudathesh G P
Asst Professor
(t) and
(t) and
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 in- phase 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
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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
RA
GH
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DA
f+3
The spectrum of the sampled signal
The components that are appeared at the filter o/p are 180Hz, 200Hz, 300Hz, 320Hz.
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Page No -44
Raghudathesh G P
Asst Professor
= max(
Nyquist rate
=2
Nyquist interval
G
he e
= 150Hz
= 2150Hz
=300Hz
=25Hz
= 150Hz
= 50Hz.
TH
ES
= 3.3msec.
Solution:
GH
U
Given signal
DA
8. Find the Nyquist rate and Nyquist interval for the signal
RA
w.k.t
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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
= max(
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he e
= 202Hz
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Page No -46
Raghudathesh G P
Asst Professor
w.k.t
GH
U
DA
TH
ES
f-
i)
fig
RA
ii)
iii)
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Page No -47
Raghudathesh G P
Asst Professor
Solution:
TH
ES
i)
GH
U
iii)
DA
RA
iv)
= 150Hz
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Page No -48
Sampling at
Raghudathesh G P
Asst Professor
Solution:
In general form, any continuous time signal can be written as
=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
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
= 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
=0.05/3 sec
TH
ES
DA
in equation(1) , we get
(
=20)
RA
GH
U
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
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Page No -50
Raghudathesh G P
Asst Professor
= 1Hz in eq(1)
(f)
(f) =
DA
TH
ES
GH
U
RA
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Page No -51
Raghudathesh G P
Asst Professor
Things to be Known:
Signals and Their Spectrum:
TH
ES
GH
U
DA
RA
3. Box Function:
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Page No -52
Raghudathesh G P
Asst Professor
TH
ES
5. Triangular Function:
GH
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DA
5. Comb Function:
RA
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Page No -53
Raghudathesh G P
Asst Professor
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
RA
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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
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Page No -56