Fourier Transforms and Sampling

Samantha R Summerson
19 October, 2009
1 Fourier Transform
Recall the formulas for the Fourier transform:
F {s(t)} = S(f) =
_

s(t)e
j2ft
dt,
F
1
{S(f)} = s(t) =
_

S(f)e
j2ft
df.
Suppose s(t) is periodic. Then we can write it using the Fourier series,
s(t) =

l=
c
l
e
j2lt
T
.
We can compute the Fourier transform of the signal using its Fourier series representation.
S(f) =
_

l=
c
l
e
j2lt
T
_
e
j2ft
dt,
=

l=
c
l
_

e
j2lt
T
e
j2ft
dt,
=

l=
c
l

_
f
l
T
_
.
The function (t) is the Dirac delta function:
(t) =
_
1 t = 0
0 t = 0
.
This means that in order to nd the Fourier transform of a periodic signal, we only need to nd the Fourier
series coecients.
Example 1. Find the Fourier transform of
s(t) = cos(2f
0
t).
We can re-write the signal using Eulers formula:
s(t) =
1
2
e
j2f0t
+
1
2
e
j2f0t
.
1
Thus, the Fourier series coecients are
a
k
=
_
1
2
k = 1, 1
0 o/w
,
and so the Fourier transform is
S(f) = a
1

_
f +
1
T
_
+ a
1

_
f
1
T
_
,
=
1
2
(f + f
0
) +
1
2
(f f
0
) .
In general, for non-periodic signals, the Fourier transform has many nice properties. I recommend
looking at CTFT tables online or in the course book. Two nice properties to highlight are the operations of
dierentiation and integration in the time domain. Consider a signal s(t) and take the derivative.
y(t) =
d
dt
s(t),
=
d
dt
_

S(f)e
j2ft
df,
=
_

S(f)
d
dt
_
e
j2ft
_
df,
=
_

S(f)j2fe
j2ft
df,
=
_

Y (f)e
j2ft
df
Thus,
d
dt
s(t) j2fS(f).
Dierentiating in the time domain corresponds to mutliplying by j2f in the frequency domain. Similarly,
we can show that integrating in the time domain corresponds to dividing by j2f in the frequency domain
(if S(0) = 0).
_
t

s()d
1
j2f
S(f)
2 Sampling
We can create discrete-time signals by sampling continuous-time signals at regular intervals of length T
s
. If
we multiply a continuous-time signal s(t) with a train of Dirac delta functions, we have a sampled signal:
s(t)

n=
(t nT
s
) =

n=
s(t)(t nT
s
),
=

n=
s(nT
s
)(t nT
s
).
The sampling period, or interval, is T
s
, and the sampling frequency, or rate, is f
s
=
1
Ts
. If s(t) is band-limited,
we can prevent aliasing (overlap in the frequency domain) by selecting T
s
such that
T
s
<
1
2W
,
2
where W is the bandwidth of the signal. This is the as the Nyquist-Shannon Sampling theorem. We refer to
f =
1
2Ts
as the Nyquist frequency since it is the highest frequency at which a signal can contain energy and
remain compatible with the sampling theorem.
If we wish to lter a discrete-time signal that originates from a continuous-time signal, does it matter
the order in which we perform the operations, i.e. if we sample and then lter versus lter and then sample?
Consider the two systems shown below:
s(t) Sampler
s(n)
H(f) y
1
(n)
Figure 1: System 1: Sampling and then ltering.
s(t) H(f)
x(t)
Sampler y
2
(n)
Figure 2: System 2: Filtering and then sampling.
Does y
1
(n) = y
2
(n) for these systems? In class we showed that
F {s(n)} =

a
k
S
_
f
k
T
s
_
.
We will use this result in order to show that, in fact, the two signals are not equal. In the rst system, the
Fourier transform for s(n), the output of the sampler, is exactly the formula we have above. If we put this
signal through a LTI lter, the Fourier transform of the output is
Y
1
(f) = F{s(n)}H(f),
=

a
k
S
_
f
k
T
s
_
H(f).
For the second system, we know that
X(f) = S(f)H(f),
and then we can use the same formula above for the spectrum of the sampled signal:
Y
2
(f) = F {y
2
(n)} ,
=

b
k
X
f
f
k
T
_
,
=

b
k
S f
k
T
_
H
_
f
k
T
_
.
We can see in general the two formulas are not equal. Lets consider the above problem in pictures.
Sampling the signal creates multiples copies of the spectrum of the signal centered at dierent frequencies.
If we low-pass lter this sampled signal using a lter with passband size WHz, as in the rst system,
then we will get the original signal back and the spectrum Y
1
(f) is the same as S(f). If we use the LPF rst
on s(t), the spectrum is unchanged since it falls within the passband. If we then sample, as in the second
system, the spectrum of the output, Y
2
(f), is the spectrum of s(nT). Thus, the two signals are dierent
since their frequency content is dierent.
3
f
|S(f)|
W W
Figure 3: Magnitude of the spectrum of s(t).
f
|S(f)|
W W
. . . . . .
Figure 4: Magnitude of the spectrum of s(nT).
Example 2. What should the sampling period be for the sinc function,
s(t) = sinc(t)?
Recall that the sinc function is dened as
sinc(t) =
sin(t)
t
.
We know that
sin(2Wt)
t
S(f) =
_
1 W < f < W
0 o/w
.
Thus, our signal has bandwidth W =
1
2
. By the sampling theorem, we require
T <
1
2
_
1
2
_ = 1.
Any sampling interval less than one will suce (equivalently, any sampling frequency greater than one). If
we select T =
1
2
, then
s(n) =
sin
_

1
2
n
_

1
2
n
,
=
_

_
1 n = 0
0 n even
2
n
n = 1, 5, 9, ...
2
n
n = 3, 7, 11, ...
.
4