|
Linear transforms, especially
Fourier and Laplace transforms, are widely used in solving problems in science
and engineering. The Fourier transform is used in linear systems analysis,
antenna studies, optics, random process modeling, probability theory, quantum
physics, and boundary-value problems (Brigham,
2-3) and has been very successfully applied to restoration of astronomical data
(Brault
and White). The Fourier transform, a pervasive and versatile tool, is used
in many fields of science as a mathematical or physical tool to alter a problem
into one that can be more easily solved. Some scientists understand Fourier
theory as a physical phenomenon, not simply as a mathematical tool. In some
branches of science, the Fourier transform of one function may yield another
physical function (Bracewell,
1-2).
The Fourier
transform, in essence, decomposes or separates a waveform or function into
sinusoids of different frequency which sum to the original waveform. It
identifies or distinguishes the different frequency sinusoids and their
respective amplitudes (Brigham,
4). The Fourier transform of f(x) is defined as
F(s) = f(x) exp(-i 2 xs) dx.
Applying the same transform to F(s) gives
f(w) = F(s) exp(-i 2 ws) ds.
If f(x) is an even function of x, that is f(x) = f(-x),
then f(w) = f(x). If f(x) is an odd function of x, that is
f(x) = -f(-x), then f(w) = f(-x). When f(x) is neither even
nor odd, it can often be split into even or odd parts.
To avoid confusion, it is customary to write the Fourier transform and its
inverse so that they exhibit reversibility:
F(s) = f(x) exp(-i 2 xs) dx
f(x) = F(s) exp(i 2 xs) ds
so that
f(x) = f(x) exp(-i 2 xs) dx exp(i 2 xs) ds
as long as the integral exists and any discontinuities, usually represented
by multiple integrals of the form ½[f(x+) + f(x-)], are finite.
The transform quantity F(s) is often represented as and the Fourier
transform is often represented by the operator (Bracewell,
6-8).
There are functions for which the Fourier transform does not exist; however,
most physical functions have a Fourier transform, especially if the transform
represents a physical quantity. Other functions can be treated with Fourier
theory as limiting cases. Many of the common theoretical functions are actually
limiting cases in Fourier theory.
Usually functions or waveforms can be split into even and odd parts as
follows
f(x) = E(x) + O(x)
where
E(x) = ½ [f(x) + f(-x)]
O(x) = ½ [f(x) - f(-x)]
and E(x), O(x) are, in general, complex. In this
representation, the Fourier transform of f(x) reduces to
2 E(x) cos(2 xs) dx - 2i O(x) sin(2 xs) dx
It follows then that an even function has an even transform and that an odd
function has an odd transform. Additional symmetry properties are shown in Table
1 (Bracewell,
14).
function transform
-----------------------------------------------------------
real and even real and even
real and odd imaginary and odd
imaginary and even imaginary and even
complex and even complex and even
complex and odd complex and odd
real and asymmetrical complex and asymmetrical
imaginary and asymmetrical complex and asymmetrical
real even plus imaginary odd real
real odd plus imaginary even imaginary
even even
odd odd
An important case from Table
1 is that of an Hermitian function, one in which the real part is
even and the imaginary part is odd, i.e., f(x) = f*(-x). The
Fourier transform of an Hermitian function is even. In addition, the Fourier
transform of the complex conjugate of a function f(x) is F*(-s),
the reflection of the conjugate of the transform.
The cosine transform of a function f(x) is defined as
Fc(s) = 2 f(x) cos 2 sx dx.
This is equivalent to the Fourier transform if f(x) is an even
function. In general, the even part of the Fourier transform of f(x) is
the cosine transform of the even part of f(x). The cosine transform has a
reverse transform given by
f(x) = 2 Fc(s) cos 2 sx ds.
Likewise, the sine transform of f(x) is defined by
FS(s) = 2 f(x) sin 2 sx dx.
As a result, i times the odd part of the Fourier transform of
f(x) is the sine transform of the odd part of f(x).
Combining the sine and cosine transforms of the even and odd parts of
f(x) leads to the Fourier transform of the whole of f(x):
f(x) = CE(x) - i SO(x)
where , C, and S stand for
-i times the Fourier transform, the cosine transform, and the sine
transform respectively, or
F(s) = ½FC(s) - ½iFS(s)
(Bracewell,
17-18).
Since the Fourier transform F(s) is a frequency domain representation
of a function f(x), the s characterizes the frequency of the
decomposed cosinusoids and sinusoids and is equal to the number of cycles per
unit of x (Bracewell,
18-21). If a function or waveform is not periodic, then the Fourier transform of
the function will be a continuous function of frequency (Brigham,
4).
It is often useful to think of
functions and their transforms as occupying two domains. These domains are
referred to as the upper and the lower domains in older texts, ``as if functions
circulated at ground level and their transforms in the underworld'' (Bracewell,
135). They are also referred to as the function and transform domains, but in
most physics applications they are called the time and frequency domains
respectively. Operations performed in one domain have corresponding operations
in the other. For example, as will be shown below, the convolution operation in
the time domain becomes a multiplication operation in the frequency domain, that
is, f(x) g(x)
F(s) G(s). The reverse is also
true, F(s)
G(s) f(x) g(x). Such theorems
allow one to move between domains so that operations can be performed where they
are easiest or most advantageous.
If {f(x)} = F(s) and a is a real,
nonzero constant, then
{f(ax)} = f(ax) exp(i 2 sx) dx
= |a|-1 f( ) exp(i 2 s /a) d
= |a|-1 F(s/a).
From this, the time scaling property, it is evident that if the width
of a function is decreased while its height is kept constant, then its Fourier
transform becomes wider and shorter. If its width is increased, its transform
becomes narrower and taller.
A similar frequency scaling property is given by
{|a|-1 f(x/a)} =
F(as).
If {f(x)} = F(s) and x0
is a real constant, then
{f(x -
x0)} = f(x - x0) exp(i
2 sx) dx
= f( ) exp(i 2 s ( + x0)) d
= exp(i 2 x0s) f( ) exp(i 2 s ) d
= F(s) exp(i 2 x0s).
This time shifting property states that the Fourier transform of a
shifted function is just the transform of the unshifted function multiplied by
an exponential factor having a linear phase.
Likewise, the frequency shifting property states that if F(s)
is shifted by a constant s0, its inverse transform is
multiplied by exp(i 2 xs0)
{f(x)
exp(i 2 xs0)} =
F(s-s0).
We now derive the
aforementioned time convolution theorem. Suppose that g(x) = f(x) h(x). Then, given
that {g(x)} =
G(s), {f(x)}
= F(s), and {h(x)} = H(s),
G(s) = {f(x)
h(x)}
= { f( ) h(x - ) d }
= [ f( ) h(x - ) d ] exp(-i 2 sx) dx
= f( ) [ h(x - ) exp(-i 2 sx) dx ] d
= H(s) f( ) exp(-i 2 s ) d
= F(s) H(s).
This extremely powerful result demonstrates that the Fourier transform of a
convolution is simply given by the product of the individual transforms, that is
{f(x) h(x)} = F(s) H(s).
Using a similar derivation, it can be shown that the Fourier transform of a
product is given by the convolution of the individual transforms, that is
{f(x) h(x)} =
F(s) H(s)
This is the statement of the frequency convolution theorem (Gaskill,
194-197; Brigham,
60-65).
The correlation integral,
like the convolution integral, is important in theoretical and practical
applications. The correction integral is defined as
h(x) = f(u) g(x+u) du
and like the convolution integral, it forms a Fourier transform pair given by
{h(x)} = F(s)
G*(s).
This is the statement of the correlation theorem. If f(x) and
g(x) are the same function, the integral above is normally called the
autocorrelation function, and the crosscorrelation if they differ
(Brigham,
65-69). The Fourier transform pair for the autocorrelation is simply
f(u) f(x+u) du = |F
|2.
Parseval's Theorem states
that the power of a signal represented by a function h(t) is the same
whether computed in signal space or frequency (transform) space; that is,
h2(t) dt = |H(f) |2
df
(Brigham,
23). The power spectrum, P(f), is given by
P(f) = |H(f) |2, f .
A bandlimited signal is a
signal, f(t), which has no spectral components beyond a frequency
B Hz; that is, F(s) = 0 for |s| > 2 B. The sampling theorem states that a
real signal, f(t), which is bandlimited to B Hz can be
reconstructed without error from samples taken uniformly at a rate R >
2B samples per second. This minimum sampling frequency, s = 2B Hz, is called the
Nyquist rate or the Nyquist frequency. The corresponding sampling
interval, T = 1/2B (where t = nT), is called the Nyquist
interval. A signal bandlimited to B Hz which is sampled at less than
the Nyquist frequency of 2B, i.e., which was sampled at an
interval T > 1/2B, is said to be undersampled.
A number of practical difficulties are
encountered in reconstructing a signal from its samples. The sampling theorem
assumes that a signal is bandlimited. In practice, however, signals are
timelimited, not bandlimited. As a result, determining an adequate sampling
frequency which does not lose desired information can be difficult. When a
signal is undersampled, its spectrum has overlapping tails; that is F(s)
no longer has complete information about the spectrum and it is no longer
possible to recover f(t) from the sampled signal. In this case, the
tailing spectrum does not go to zero, but is folded back onto the apparent
spectrum. This inversion of the tail is called spectral folding or
aliasing (Lathi,
532-535).
 Figure 1: Undersampled,
oversampled, and critically-sampled unit area gaussian curves.
As an example, Figure 1 shows a unit gaussian curve sampled at three
different rates. The FFT (or Fast Fourier Transform) of the undersampled
gaussian appears flattened and its tails do not reach zero. This is a result of
aliasing. Additional spectral components have been folded back onto the ends of
the spectrum or added to the edges to produce this curve.
The FFT of the oversampled gaussian reaches zero very quickly. Much of its
spectrum is zero and is not needed to reconstruct the original gaussian.
Finally, the FFT of the critically-sampled gaussian has tails which reach
zero at their ends. The data in the spectrum of the critically-sampled gaussian
is just sufficient to reconstruct the original. This gaussian was sampled at the
Nyquist frequency.
Figure 1 was generated using IDL with the following code:
!P.Multi=[0,3,2]
a=gauss(256,2.0,2) ; undersampled
fa=fft(a,-1)
b=gauss(256,2.0,0.1) ; oversampled
fb=fft(b,-1)
c=gauss(256,2.0,0.8) ; critically sampled
fc=fft(c,-1)
plot,a,title='!6Undersampled Gaussian'
plot,b,title='!6Oversampled Gaussian'
plot,c,title='!6Critically-Sampled Gaussian'
plot,shift(abs(fa),128),title='!6FFT of Undersampled Gaussian'
plot,shift(abs(fb),128),title='!6FFT of Oversampled Gaussian'
plot,shift(abs(fc),128),title='!6FFT of Critically-Sampled Gaussian'
The gauss function is as follows:
function gauss,dim,fwhm,interval
;
; gauss - produce a normalized gaussian curve centered in dim data
; points with a full width at half maximum of fwhm sampled
; with a periodicity of interval
;
; dim = the number of points
; fwhm = full width half max of gaussian
; interval = sampling interval
;
center=dim/2.0 ; automatically center gaussian in dim points
x=findgen(dim)-center
sigma=fwhm/sqrt(8.0 * alog(2.0)) ; fwhm is in data points
coeff=1.0 / ( sqrt(2.0*!Pi) * (sigma/interval) )
data=coeff * exp( -(interval * x)^2.0 / (2.0*sigma^2.0) )
return,data
end
Because a digital
computer works only with discrete data, numerical computation of the Fourier
transform of f(t) requires discrete sample values of f(t), which
we will call fk. In addition, a computer can compute the
transform F(s) only at discrete values of s, that is, it can only
provide discrete samples of the transform, Fr. If f(kT)
and F(rs0) are the kth and rth samples of
f(t) and F(s), respectively, and N0 is the
number of samples in the signal in one period T0, then
fk = T f(kT) = T0N0-1
f(kT)
and
Fr = F(rs0)
where
s0 = 2 0 = 2 T0-1.
The discrete Fourier transform (DFT) is defined as
Fr = fk exp(-i r 0k)
where 0 = 2 N0-1. Its inverse
is
fk = N0-1
Fr exp(i r 0k).
These equations can be used to compute transforms and inverse transforms of
appropriately-sampled data. Proofs of these relationships are in Lathi
(546-548).
The Fast Fourier
Transform (FFT) is a DFT algorithm developed by Tukey
and Cooley in 1965 which reduces the number of computations from something
on the order of N02 to N0 log
N0. There are basically two types of Tukey-Cooley FFT
algorithms in use: decimation-in-time and decimation-in-frequency. The algorithm
is simplified if N0 is chosen to be a power of 2, but it is
not a requirement.
The Fourier transform, an invaluable tool in
science and engineering, has been introduced and defined. Its symmetry and
computational properties have been described and the significance of the time or
signal space (or domain) vs. the frequency or spectral domain has been
mentioned. In addition, important concepts in sampling required for the
understanding of the sampling theorem and the problem of aliasing have been
discussed. An example of aliasing was provided along with a short description of
the discrete Fourier transform (DFT) and its popular offspring, the Fast Fourier
Transform (FFT) algorithm.
Blass, William E. and
Halsey, George W., 1981, Deconvolution of Absorption Spectra, New York:
Academic Press, 158 pp.
Bracewell, Ron N., 1965, The Fourier Transform and Its
Applications, New York: McGraw-Hill Book Company, 381 pp.
Brault, J. W. and White, O. R., 1971, The analysis and
restoration of astronomical data via the fast Fourier transform, Astron.
& Astrophys., 13, pp. 169-189.
Brigham, E. Oren, 1988, The Fast Fourier Transform and Its
Applications, Englewood Cliffs, NJ: Prentice-Hall, Inc., 448 pp.
Cooley, J. W. and Tukey, J. W., 1965, An algorithm for the
machine calculation of complex Fourier series, Mathematics of
Computation, 19, 90, pp. 297-301.
Gabel, Robert A. and Roberts, Richard A., 1973, Signals and
Linear Systems, New York: John Wiley & Sons, 415 pp.
Gaskill, Jack D., 1978, Linear Systems, Fourier
Transforms, and Optics, New York: John Wiley & Sons, 554 pp.
Lathi, B. P., 1992, Linear Systems and Signals,
Carmichael, Calif: Berkeley-Cambridge Press, 656 pp.
|