Objectives:

Scope of this lecture:

Modern Communication would not have been possible without the development of sampling theory. The sampling
theory provides means and ways for processing the Continuous Time (C.T.) data in digital domain. Thus sampling
theorem provides the bridge between CT and DT signals. By sampling we mean taking the instantaneous values of
CT signal at a regular interval of time. The topics covered in this lecture are listed below:

The concept of sampling of a signal . The notion of apriori information & its use to
represent a signal economically . The most common approach towards economical signal
representation.

What is Sampling?
Sampling is a methodology of representing a signal with less than the signal itself.
We can do better than just describing a signal by specifying the value of the dependent variable
for each possible value of the independent variable. The concept is explained with the following
examples where 'x(t)' is the dependent variable and 't' is the independent variable.

Let
Here 'x(t)' is defined by a sinusoidal relation with a phase constant , amplitude and angular
frequency. Now the knowledge of these three parameters suffices to describe 'x(t)' completely.
Thus we are able to compute 'x(t)' without depending on the independent variable 't'.

Consider another example given below:

Here x(t) is a polynomial in 't' of degree 'N' and can be computed completely if we know the

coefficients .
Thus we observe that the apriori information we had that allowed us to represent these signals.
In the first case we knew that 'x(t)' is a pure sinusoid and in the second case we knew that it was
a polynomial of degree 'N'.
Thus, as a method of using Apriori information available to represent a signal economically is
one way of defining sampling.
A Common Approach for Signal Representation:
The approach most often used to economically represent a signal is to look at the values of the
dependent variable as a set of properly chosen values of the independent variable such that
these 'tuples' and the 'apriori' information can be used to reconstruct the signal completely.
Lets say we know that some signal 'x(t)' is a pure sinusoid described by the three quantities

amplitude (Ao) , angular frequency ( ),and phase constant ( ). For 't1, t2 & t3' values of 't' we
get the following three independent equations. :
From the observed values of the signal x(t1), x(t2) and x(t3) at t1, t2 and t3, the parameters of the

signal Ao, and can be determined

Consider another example:
Let x(t) be a polynomial of order 'N' which is represented mathematically as shown below. It is
further represented in the form of a matrix where the LHS is the 'apriori' information.

Thus we observe that, this system can be solved as the determinant of the square matrix on the

LHS so long as .

Thus given the 'apriori' information, the entire information about the signal is contained in its
value at N + 1 distinct points.

You have seen two examples, where 'apriori' information, and "samples" of a signal at certain
values of the independent variable help us reconstruct the signal completely.
But If you have no Apriori information you can do no better than to represent the signal as it is.
Even knowing about the continuity of a signal is 'apriori' information. Further we can talk of the
relative measure of the 'apriori' information. This can be done by observing the size of the set in
which that signal occurs. The larger the set, the lesser the 'apriori' information we have. For
example, knowing that the signal is sinusoidal is much larger an 'apriori' information than
knowing that it is continuous as the set of sine functions is much smaller than the set of
continuous functions.
The main challenge in sampling and reconstruction is to make the best use of 'apriori'
information in order to represent a signal by its samples most economically.
In the next lecture, we focus on a special class of signals those that are Band-limited (this is the
'apriori' information we shall have) and see how such signals can be reconstructed from their
samples.
Conclusion: From this lecture you have learnt : Sampling is a method of using 'apriori'
information about a signal to represent it economically.
The most common approach in sampling and reconstruction is to describe the signal by
specifying its value at selected points on the time axis ('t') such that this and the 'apriori'
information can be used to reconstruct the signal completely.
The main challenge in sampling & reconstruction is to make the best use of the apriori
information available to represent a signal most economically. Congratulations, you
have finished Lecture 21. To view the next lecture select it from the left hand side menu
of the page.

Objectives
Scope of this lecture:
If a Continuous Time (C.T.) signal is to be uniquely represented and recovered from its samples, then the signal must be band-
limited. Further we have to realize that the samples must be sufficiently close and the Sampling Rate must bear certain relation
with the highest frequency component of the original signal. In this lecture, we'll see:

A note about Band-limited signals.

The analyticity of time-limited and band-limited signals.

Reconstruction of Band-limited signals - The Shannon-Whiltaker-Nyquist Sampling Theorem

Band-limited signals:
A Band-limited signal is one whose Fourier Transform is non-zero on only a finite interval
of the frequency axis.

Specifically, there exists a positive number B such that X(f) is non-zero only in . B is
also called the Bandwidth of the signal.
To start off, let us first make an observation about the class of Band-limited signals.
Lets consider a Band-limited signal x(t) having a Fourier Transform X(f).

Let the interval for which X(f) is non-zero be -B f B.

Then, converges.
The RHS of the above equation is differentiable with respect to t any number of times as the
integral is performed on a bounded domain and the integrand is differentiable with respect to t.

Further, in evaluating the derivative of the RHS, we can take inside the integral.

In general,

This implies that band limited signals are infinitely differentiable, therefore, very smooth .
We now move on to see how a Band-limited signal can be reconstructed from its samples.
Reconstruction of Time-limited Signals
Consider first a signal y(t) that is time-limited, i.e. it is non-zero only in [-T/2, T/2].
Its Fourier transform Y(f) is given by:

Where is the periodic extension of y(t) as shown

Now, Recall that the coefficients of the Fourier series for a periodic signal x(t) are given by :
Comparing (1) and (2), you will find

That is, the Fourier Transform of the periodic signal is nothing but the samples of the
original transform.

Therefore, given that; y(t) is time-limited in [-T/2, T/2] and periodic, the entire information
about y(t) is contained in just equispaced samples of its Fourier transform! It is the dual of
this result that is the basis of Sampling and Reconstruction of Band-limited signals :-
Knowing the Fourier transform is limited to, say [-B, B], the entire information about the
transform (and hence the signal) is contained in just uniform samples of the (time) signal !
Reconstruction of Band-limited signals
Let us now apply the dual reasoning of the previous discussion to Band-limited signals.
x(t) is Band-limited, with its Fourier transform X(f) being non-zero only in [-B, B]. The dual
reasoning of the discussion in previous slide will imply that we can reconstruct X(f) perfectly in
[-B, B] by using only the samples x( n / 2B ). Let's see how.
This time, is the Fourier series co-efficient of , the periodic extension of
X(f).

(Fourier series in f -- fundamental period is 2B and

is the Fourier series

coefficient)

What is the Fourier inverse of ?

The Fourier inverse of is . Therefore, the Fourier inverse of is

Thus we see that if we multiply the original Band-limited signal with a periodic train of
impulses (period 1/2B, with impulse at the origin of strength 1/2B ) we obtain a signal
whose Fourier transform is a periodic extension of the original spectrum. So how does one

retrieve the original signal from ? We need a mechanism that will blank out the spectrum of

in , i.e: multiply the spectrum with :

In other words, we need to feed to an LSI system, the Fourier transform of whose impulse
response is the above function (recall the convolution theorem), i.e: one whose impulse response
is:

An LSI system with above type of impulse response is called an Ideal Low Pass Filter .
he Sampling Theorem
On the basis of our discussion so far, we may state formally the Sampling Theorem.
Shannon-Whiltaker-Nyquist Sampling Theorem:
A band-limited signal with band-width B may be reconstructed perfectly from its samples, if the
signal is sampled uniformly at a rate greater than 2B.
Here's and overview of the derivation of sampling theorem:

Is it essential for the sampling rate to be greater than 2B, or is it acceptable to have a sampling
rate of exactly 2B?
Top of Form

Click to find out

Bottom of Form

Conclusion: In this lecture you have learnt: Band-limited signals are infinitely differentiable
and very smooth.
Given that 'x(t)' is Band-limited with its Fourier transform 'X(f)' being non-zero only in [-B,B] ,
we can say that

has a spectrum that is the periodic

extension of 'X(f)' with period 2B. By passing through an appropriate

Ideal Low-pass filter one can obtain back
'x(t)'. Shannon-Whiltaker-Nyquist Sampling Theorem: A band-limited
signal with band-width 'B' may be reconstructed perfectly from its samples, if the signal is
sampled at a rate greater than '2B'.

Congratulations, you have finished Lecture 22. To view the next lecture select it from the left
hand side menu of the page.
Objectives: Scope of lecture:
In the previous lecture we mentioned that the ideal low pass filter can be used to recover the original continuous time
signal from its samples. In this lecture, we will derive its impulse response, and see how exactly the original signal
can be recovered. We'll be covering:

The impulse response of an ideal low pass filter. Reconstruction of a band-limited signal
by a low pass filter. Problems with the ideal low pass filter.

The Ideal Low-Pass Filter

In this lecture, we examine the Ideal low pass filter and the process of reconstruction of a Band-
limited signal.
Let us first see another way of interpreting the action of the Ideal low-pass filter.
IMPULSE RESPONSE OF IDEAL LOW PASS FILTER:
The Frequency response (the Fourier transform of the impulse response of an LSI system is also
called its frequency response) of an ideal low pass filter which allows a bandwidth B, is a
rectangle extending from -B to +B, having a constant height as shown in the figure.

Lets look at the Impulse Response of this Ideal low pass filter, taking its height in [-B, B] to be 1.
Using the formula for inverse Fourier Transform we have :

(note that )
Thus the impulse response of an ideal low pass filter turns out to be a Sinc function, which looks
like:
Reconstruction of a signal by low pass filter :
Consider a signal x(t) having bandwidth less than B.

We sample x(t) at a rate 2B and pass into an Ideal low-pass filter

of bandwidth B. .

The signal x(t) and the signal ( ) obtained by multiplying the signal by a periodic train of

impulses, separated by , having strength 1 are shown below.

What happens when is fed into the LSI system?

Lets look at the convolution of the impulse response h(t) of the Ideal low-pass filter with
where we have seen .

When is passed through a Low pass filter, the output which is the reconstructed signal is

nothing but the sum of copies of the impulse response h(t) shifted by integral multiples of and

multiplied by the value of x(t) at the corresponding integral multiple of . Also observe that the

h(t) is zero at all sample points (which are integral multiples of ) except at zero. Thus, the
reconstruction of x(t) can be visualized as a sum of the following signals :

Problems with the IDEAL LOW PASS FILTER

It is infinitely Non-Causal:

The impulse response of the ideal low pass filter extends to . If the impulse response is
denoted by h(t), the output signal y(t) corresponding to input signal x(t) is given by :
The value of y at any t depends on values of x all the way to if h(t) extends to . Thus
realization in real time is not possible for an Ideal low-pass filter. In other words, unless one

knows the entire , reconstruction cannot be done.

Note if h(t) had been finitely non-causal (say zero for all t less than some - ), then real time

realization would have been possible subject to a time-delay (of ).

It is unstable:

It can be shown that diverges.

This implies that bounded input does not imply bounded output. Thus if we build an oscillator
with Ideal Low pass Filter a bounded input may result in an unstable output.
The system is not rational:
That means, it is not exactly realizable with simple well known elements .
We will get back to how these problems are tackled a little later. In the next lecture, we move on
to the problem of impulses not being physically realizable.
Conclusion: In this lecture you have learnt: Impulse response of an ideal low pass filter turns
out to be a Sinc function. When sampled signal is passed through a low pass filter,
reconstructed output signal is nothing but the sum of copies of

impulse response h(t) shifted by integral multiples of and multiplied by the value of x(t) at the

corresponding integral multiple of .

Problems with the ideal low pass filter :

1)It is infinitely non-causal.

2)It is unstable.
3)It is not a rational system.

Congratulations, you have finished Lecture 23. To view the next lecture select it from
the left hand side menu of the page.