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In the previous three lessons, we discussed the

Fourier Series, which is for periodic signals.
This lesson will cover the Fourier Transform
which can be used to analyze aperiodic signals.
(Later on, we'll see how we can also use it for
periodic signals.)
The Fourier Transform is another method for
representing signals and systems in the
frequency domain
Chapter-5 Fourier Transform
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On the -axis, distance between two consecutive aks is now 0=2/T, the
fundamental frequency.
3
Bridge Between Fourier Series and
Transform
Consider the periodic signal x(t) below:
We know that the Fourier coefficients for x(t) will be:

=
=
0
0
) sin(
,
2
1 0
1
k
k
k
T k
T
T
a
k

x(t)
t
-T
1
0 T
1 T -T
Bridge Between Fourier Series and Transform
(contd)
Now, sketch a
k
on the -axis:
On the -axis, distance between two consecutive a
k
s is
now
0
=2/T, the fundamental frequency.
a
k

-2
0
-
0
0
0
2
0
2T
1
/T
Bridge Between Fourier Series and Transform
(contd)
As the period T, the fundamental frequency
0
0.
So, the distance between the two consecutive a
k
s
becomes zero, and the sketch of a
k
becomes continuous,
what is called as Fourier Transform.
At the other side, as T, the signal x(t) becomes
aperiodic and takes the form:
This means the Fourier Transform can represent an
aperiodic signal on the frequency-domain.
x(t)
t
-T
1
0 T
1
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Bridge to Fourier Transform
Periodic Signals Fourier Series
Aperiodic Signals Fourier Transform
Consider an aperiodic signal x(t). Now, consider
periodic version of x(t), x
p
(t), where x(t) is repeated
over a period T.
We can say that x
p
(t) is the limiting case of x(t) as T
approaches to infinity, i.e.
Since x
p
(t) is periodic, we can write Fourier series
representation for it:
where
0
=2/T.
) ( lim ) ( t x t x
p
T
=

=
= =
2 /
2 /
0 0
) (
1
) (
T
T
t jk
p k
k
t jk
k p
dt e t x
T
a e a t x

Bridge to Fourier Transform
(contd)
Since x(t)=0 outside of the interval (-T/2,T/2):
Define an envelope function:
So,
Now, substitute that in the Fourier series representation
of x
p
(t):

= dt e t x j X
t j
) ( ) (

= dt e t x
T
a
t jk
k
0
) (
1

) (
1
0
j X
T
a
k
=

=
=
k
t jk
p
e j X
T
t x
0
) (
1
) (
0

Bridge to Fourier Transform

(contd)
Since 2/T=
0
:
As T, this sum will approach to an integral because
there will a continuum of Fourier terms in the Fourier
series.
Finally, the Fourier Transform of x(t):
And, the Inverse Fourier Transform of x(t):

= dt e t x j X
t j
) ( ) (

d e j X t x
t j
) (
2
1
) (

=
=
k
t jk
p
e j X t x
0 0
0
) (
2
1
) (

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Convergence of Fourier Transform
Similar to Fourier Series, convergence
conditions apply for Fourier Transform:
The signal must be absolutely integrable
Over a finite interval of time, the signal must
have finite number of maxima and minima (or
variations)
Over a finite interval of time, the signal must
have finite number of discontinuities. Also,
those discontinuities must be finite.
<

dt t x ) (
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Inverse Fourier Transform:
It is always possible to move back from the frequency-domain to
time-domain, by either summing the terms of the Fourier Series or
by Inverse Fourier Transform
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Properties of the Fourier
Transform
The point of this lesson is that knowledge
of the properties of the Fourier Transform
can save you a lot of work. We will cover
some of the important Fourier Transform
properties here
Linearity
Because the Fourier Transform is linear,
we can write:
F[a x1(t) + bx2(t)] = aX1() + bX2()
where X1() is the Fourier Transform of x1(t)
and X2() is the Fourier Transform of x2(t).
Time Scaling
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Time Shifting
Duality
In general, the Duality property is very useful because it can
enable to solve Fourier Transforms that would be difficult to
compute directly (such as taking the Fourier Transform of a sinc
function). The Duality Property tells us that if x(t) has a Fourier
Transform X(), then if we form a new function of time that has
the functional form of the transform, X(t), it will have a Fourier
Transform x() that has the functional form of the original time
function (but is a function of frequency). Mathematically, we can
write:
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Notice that the second term in the last line is simply the Fourier Transform
integral of the function X(t), i.e.
Example 1 Using the Fourier Transform integral
equation, directly find the Fourier Transform of
x(t) = e
-at
u(t), a> 0
Example 2 Using the results of Example 1 and the Duality
Property, find the Fourier Transform of
The Convolution Property
It tells us that convolution in time
corresponds to multiplication in the
frequency domain. Therefore, we can
avoid doing convolution by taking Fourier
Transforms! In many cases, this will be
much more convenient than directly
performing the convolution.
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The convolution property states that:
Let us show that. To start, let
The Convolution Property
Then we can take the Fourier Transform of y(t) and plug
in the convolution integral for y(t) (notice how we've
marked the integrals with dt and d to keep track of them):
Therefore,
We've just shown that the Fourier Transform of the convolution of two functions is
simply the product of the Fourier Transforms of the functions. This means that for
linear, time-invariant systems, where the input/output relationship is described by a
convolution, you can avoid convolution by using Fourier Transforms. This is a very
powerful result.
Multiplication of Signals
It states that the Fourier Transform of the product of
two signals in time is the convolution of the two
Fourier Transforms.
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Proof
Therefore,
Example 2 Find the Fourier Transform of x(t) = sinc
2
(t)
(Hint: use the Multiplication Property).
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Example 1 Find the inverse Fourier Transform of
Here is a plot of this function:
This tells us that modulation (such as multiplication in
time by a complex exponential, cosine wave, or sine
wave) corresponds to a frequency shift in the
frequency domain.
DC Level
Example Find the Fourier Transform of the constant 1. Use duality and
the fact that the transform of (t) is 1
We know ej
t
2(-
0
)
=0
ee
Differentiation in time
Transform of derivatives
Suppose that f
(n)
is piecewise continuous, and absolutely integrable on R. Then
In particular
and
) ( ) ( )} ( {
) (
w F iw t f F
n n
=
) ( )} ( {
'
w iwF t f F = ) ( )} ( {
2 ' '
w F w t f F =
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but we can not directly calculate this integral because it does not converge.
For many periodic signals, the integral for Fourier transform may not work. Luckily,
there is an easier way of determining Fourier transforms of periodic signals
So we will generalize the Fourier Transform to include impulses in the frequency
domain.
We can use either the Duality or Modulation Properties to show that. So we will
generalize the Fourier Transform to include impulses in the frequency domain.
We can use either the Duality or Modulation Properties to show that:
We can check this by taking Inverse Fourier Transform (and using the sifting property)
Now let's consider a general periodic signal x(t) that we
can represent as a Fourier Series:
Since we know that:
by linearity of the Fourier Transform, we get that:
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Example Find the Fourier Transform of cos(
0
t)
Pulsed Cosine
=
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Fourier Transforms of Sampled Signals
In this lesson, we will discuss sampling of
continuous time signals. Sampling a
continuous time signal is used, for
example, in A/D conversion, such as
would be done in digitizing music for
storage on a CD, digitizing a movie for
storage on a DVD, or taking a digital
picture.
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To start, we define the continuous time impulse train as:
As you just saw, p(t) is an infinite train of continuous
time impulse functions, spaced T
s
seconds apart.
Now, x(t) is the continuous time signal we wish to sample. We
will model sampling as multiplying a signal by p(t
Let x
s
(t) = x(t)p(t) be the sampled signal. Then
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Now, we will derive the Sampling Theorem . To do this, we will
examine our signals in the frequency domain.
To start, let p(t) have a Fourier Transform P(), x(t) have a
Fourier Transform X(), and x
s
(t) have a Fourier Transform
X
s
().
Then, because x
s
(t) = x(t)p(t), by the Multiplication
Property,
Now let's find the Fourier Transform of p(t). Because
the infinite impulse train is periodic, we will use the
Fourier Transform of periodic signals:
where Ck are the Fourier Series coefficients of the periodic signal.
Let's find the Fourier Series coefficients C
k
for the periodic
impulse train p(t):
Therefore
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Now, to finish our derivation of the Sampling Theorem,
we will go back and determine X
s
().
We saw that:
Therefore,
or we get replicated, scaled versions of X(), spaced every

0
apart in frequency:
From this development and observing the above figure, we can
derive our Sampling Theorem. This is one of the most important
results.
As you can see from the figure if
0
-
c
<
c
, we would get overlap of the
replicates of X() in frequency. This is known as "aliasing." Therefore, to
avoid aliasing, we require 0 - c > c or 0 > 2c. If we avoid aliasing,
we can recover x(t) from its samples. (Usually, we choose a sampling rate a
bit higher than twice the highest frequency since filters are not ideal.)
We hear music up to 20 kHz and CD sampling rate is 44.1 kHz. (Dogs
would need a higher quality CD since they hear higher frequencies than
humans.)
We can recover x(t) from its sampled version xs (t) by
using a low pass filter to recover the center island:
After low pass filtering
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Sampling Theorem Conditions
Sampling Theorem: A continuous-time signal x(t) can be
uniquely reconstructed from its samples x
s
(t) with two
conditions:
x(t) must be band-limited with a maximum frequency
M
Sampling frequency
s
of x
s
(t) must be greater than 2
M
, i.e.

s
>2
M
.
The second condition is also known as Nyquist Criterion.

s
is referred as Nyquist Frequency, i.e. the smallest
possible sampling frequency in order to recover the
original analog signal from its samples.
Aliasing (Under-sampling)
What happens when sampling frequency is less than
Nyquist Frequency, i.e.
s
<2
M
?
The original signal x(t) cannot be recovered from x
s
(t)
since there will be unwanted overlaps in X
s
().
This will cause the recovered signal x
r
(t) to be different
than x(t), also called as aliasing or under-sampling.
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Example 1 Given a signal x(t) with Fourier Transform with cutoff
frequency &omega;
c
as shown:
you are given three different pulse trains with periods
Draw the sampled spectrum in each case. Which case(s)
experiences aliasing?
Example 2 The inverse Fourier Transform of the signal in the
previous example is
Draw the sampled signals using the sampling trains of the
previous example
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Applications of the Fourier Transform:
Ideal Filters
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Example 1 Given two functions x1(t) = cos(500t) and x2(t) =
cos(1000t), form a new signal x
3
(t) = x1(t) x
2
(t).
Draw the frequency response of a low pass filter that passes the low
frequency component of the signal and blocks the high frequency
component of the signal.
Sinusoidal Amplitude Modulation
Modulation is multiplying a signal in time by
complex exponentials (such as sine waves and
cosine waves) to shift the signal to a desired
frequency band. This is done, for example, by
radio (or television) to assign different parts of
the frequency spectrum to different radio (or
television) stations. Remember that the
Modulation Theorem told us that:
We will use the Modulation Theorem in an
example of AM radio.
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Example 2 AM radio - Double-sideband, suppressed
carrier, amplitude modulation
Let x(t) be a music signal with
Fourier Transform X().
To modulate this signal, we'll form and transmit
Your radio demodulates the signal by multiplying the
received signal y(t) by another cosine wave to form a third
signal z(t):
To recover the original music signal x(t) from the demodulated
signal z(t), you filter z(t) with a low-pass filter by multiplying by a
rect function in frequency. This corresponds to convolving with a
sinc function in time.