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I. Introduction
Analog Signal
In this chapter we present some of the fundamental concepts of sampling theory that will be used later for explaining and understanding some of the basic issues related to
the problem of seismic acquisition.
Sampling theory provides us with the basic elements for
the successful design and implementation of a seismic survey. Indeed, seismic surveying, as will be seen later, has to
be with the sampling of a seismic wave eld that propagates
through an earthen formation.
First, we will discuss some basic concepts on signals and
their representations. Then, the sampling process and its
eects will be studied from a mathematical point of view
in order to better understand the phenomenon of aliasing
and the problem of information preservation as well as the
problem of reconstruction. Finally, some related topics are
presented and briey discussed.
f n
10
10
10
10
Discretevariable Signal
6
Discreteamplitude Signal
6
5
Digital Signal
represented by means of dierent types of signals. For example, an audio signal can be represented by an equivalent
electric signal and vice versa { this is the base of telephony.
Of special interest nowadays is the representation of analog signals by means of digital ones. This is because, although digital computers are powerful tools for handling
and processing information, most of the signals we are generally interested in deal with are analog.
Although apparently trivial, the problem of representing
analog signals with digital ones is not a simple one. This
is basically because of two reasons: rst, it is a non-linear
process and second, it is an irreversible process. However,
as seen latter, some restrictions can be applied and some
considerations taken into account in order to eliminate the
ambiguities inherent to the digitization process.
To obtain a digital representation of an analog signal,
two steps must be performed. First, the analog signal
must be discretized in its variable dimension. This step
is called sampling. Second, the resulting discrete-variable
signal must be discretized in its amplitude dimension. This
step is called quantization. Figure 2 illustrates the digitization process of an audio signal. As seen from the gure,
an additional codication step is implemented in order to
put the quantized samples into the binary format used by
computers.
( )
f x
]
f x
( )=
f x
f ]g
f n
n=;1
W
] ;j 22nx
(1)
g n e
Discrete Signal
Digital Signal
g n
Z1
( )=
dx :
R x
r y
Encoded Signal
f x
r y
Coding
C
1
X
g n
Quantization
f n
One of the most revolutionary concepts in communication theory has been the Nyquist-Shannon sampling theorem, 1] and 2]. According to this theorem, a band limited
signal has a restricted number of degrees of freedom per
variable interval. In other words, what the theorem states
is that a band limited analog signal has an inherent redundancy of the information it contains. In this way, it is
possible to represent the analog signal by using samples of
it without lossing information. However, there is a limit
on the sampling period to be used which is related to the
band-width of the signal.
A. Theorem Postulate
The Nyquist-Shannon sampling theorem establishes the
limits for preserving information when sampling band limited analog signals. It can be stated as follows: A band
limited signal with maximum spectral content W, can be
sampled at a minimum rate of 2W without lossing any information.
B. Mathematical Motivation
Consider a periodic function ( ) with period 2 . As
known, it admits as Fourier series representation given by
Analog Signal
Sampling
f n
;1
( ) j2yx
R x e
(3)
dx :
f x
R x
f x
r y
n= W
:::
:::
r y
ZW
j 22nx
W
(5)
2 = ;W ( )
Notice that the integral in (5) is the same, except for
3 However, quantization eects must be taken into account and dealt
a
scaling
factor, as the one in (2). Indeed, it follows that
with when recursive procedures, susceptible to error accumulation,
2 ] = ].
are involved.
]=
r n
W g n
r n
f x
dx :
At this point we can see that, since there is a unique reBy using the sifting property of the Dirac's delta funclationship between the function ( ) and its Fourier trans- tion, equation (6) can be equivalently written as
form ( ), and between the function ( ) and its Fourier
1
X
series coecients ] we will be able to recover ( ) from
(
)
=
( s) ( ; s)
(7)
s
the values of ] if, and only if, we are able to obtain ( )
n
=;1
from ( ). This ideas are illustrated in Figure 3
from which the conversion to a discrete sequence follows
immediately,
] = ( s)
(8)
Notice that this conversion process implicitly includes the
normalization of the variable dimension by the sampling
period value s . As will be seen in the next subsection,
this normalization is also carried over to the spectrum of
the sequence ].
The mathematical model for the analog to discrete conversion process is illustrated in Figure 4.
r y
R x
f x
g n
r y
g n
R x
f nx
x
nx
f x
15
1.5
10
0.5
f n
f nx
f n
5
20
30
1.5
20
10
0.5
10
20
10
10
10
20
s (x)
f n
As described in the previous section, the NyquistShannon theorem provides a basis for sampling properly
analog signals in order to preserve information. This section presents a mathematical representation4 of the sampling process and discusses it in detail.
A. Analog to Discrete Conversion
The mathematical representation of the sampling process is achieved by taking advantage of the Dirac's delta
function. Suppose that the signal to be sampled, ( ), is
multiplied by a periodic unit impulse train, the resulting
signal will be another periodic impulse train, s ( ), with
the amplitude values of the original signal at the impulse
locations
f x
Sequencer
1
X
n=;1 (x ; nxs )
Multiplier
10
20
P1
( )
f x
Discrete-variable Signal
1
s () =
()
1
X
k=;1
( ; 2
x
s)
=x
(9)
f x
( ; s)
x
nx
f x
f n
=x
=x
is actually dierent to the mathematical approach. For more information on sampler circuits, the reader can refer to 6]
s k=;1
Equation (10) reveals that the spectrum of the modu- related synthesis formula can be obtained from equation
lated impulse train s ( ) is actually given by the sum- (2). It is given by
mation of scaled copies of the original signal's spectrum
Z
located at integer multiples of the sampling angular fre ] = 21
( j! ) j!n
(13)
quency s . The spectrum of the related discrete sequence,
;
], can be obtained by using the same normalization that
Equations (12) and (13) constitute the analysis and syntook place in the conversion step presented in (8). In this
thesis
Fourier formulas for discrete sequences 4].
way, the angular frequency axis, , is normalized as
V. The Reconstruction Problem
(11)
= s= 2 s
This section studies in detail the problem of reconstructFigure 5 shows a continuous signal ( ), a related im- ing an analog signal from a discrete-variable representation
pulse train s ( ) and its corresponding discrete sequence of it. First, the discrete to analog conversion process is
] together with their corresponding amplitude spectra. presented. Afterwards, the phenomenon of aliasing and its
implications in the problem of preserving information is
discussed.
A. Discrete to Analog Conversion
Regardless of the origin of a discrete-variable signal ],
an analog representation of it, ^( ), can be always obtained. The problem of nding such an analog representation is indeed an interpolation problem, in which a
continuous-variable signal is generated from the discretevariable one. This interpolation can be performed by following the two-step process illustrated in Figure 6, which
is described next.
First, the discrete sequence must be converted into a
periodic impulse train of the form
f
f n
F e
d! :
f n
f x
f n
Analog Signal
Amplitude Spectrum
100
1500
50
f n
1000
f x
500
50
100
0.05
0.1
0.15
0.2
0
1500 1000 500
0.25
500
1000
1500
Amplitude Spectrum
100
600
500
50
400
1
^s ( ) = X ] ( ; p )
300
200
50
100
n=;1
f n x
nx
(14)
0.05
0.1
0.15
0.2
0
1500 1000 500
0.25
Discrete Sequence
500
1000
1500
Amplitude Spectrum
100
600
f x
500
50
400
300
200
50
f x
100
100
10
20
30
40
50
10
g x
g x
10
f n
sin x=x
x=x
F e
F e
f n e
f x
f n
f n
Sequencer
^s ( )
Filter
^( )
f x
800
800
600
600
400
400
200
200
10
0
1500 1000 500
10
Reconstruction Filter
Analog Signal
500
1000 1500
Recovered Spectrum
1500
3
1000
2
500
f n
f n
f x
x
f x
<
=
500
1000 1500
0
1500 1000 500
500
1000 1500
< =x
0
1500 1000 500
f x
f n
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0
300
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0
f n
f n
D (ejw )
"
-D
f
D
-U
U (ejw )
f n
6 This refers to the fact that the spectral content is in alias due to
a decient sampling in time. This must not be confused with the
concept of aliasing in the time domain (variable domain), which will
be discussed later.
f n
x =U
D=U
f n
VII. Summary
The Nyquist-Shannon sampling theorem presents the basis for properly sampling analog signals in order to obtain
discrete representations without loosing any of the information contained in them. According to the theorem, a
signal with a maximum frequency content of must be
sampled above the minimum sampling frequency of 2 in
order to preserve information.
The analysis of the sampling process in the frequency
domain give us an important insight of how the information is preserved or lost, being this latter the case in which
spectral overlapping, or aliasing, occurs. When an analog
signal has been properly sampled, it can be perfectly recovered from its related sequence by a process of interpolation.
It is important, in order to avoid confusion, to have it
clear the dierences among the dierent concepts that were
presented. An analog signal ( ), with Fourier transform
(), is sampled to obtain a discrete representation of it,
the sequence ], with frequency response ( jw ), which
happens to be a periodic, scaled and frequency-normalized
version of (). It is also possible to obtain an analog
representation ^( ) for a discrete sequence ]. If the sequence ] was obtained from an original analog signal
( ), the sampling period used is known and the NyquistShannon theorem conditions were met, it is possible to obtain an analog representation ^( ) such that ^( ) = ( ).
All concepts considered here were based on the assumption that all samples are uniformly spaced. Multirate signal
processing deals with discrete signals resulting from a nonuniform sampling process. For information on this topic,
the reader must refer to 3].
W
C. Spectral Sampling
As was already seen from equation (12), the frequency
response F (ejw ) of a discrete sequence f n] is a periodic
continuous-variable function. In many computational applications of signal processing, operations are carried over
on the frequency domain. Then, it is required to handle
f x
frequency responses in a discrete way.
F
f n
F e
This problem can be visualized as the problem of sampling the spectrum content of discrete sequences. Here too,
F
the conditions of the Nyquist-Shannon theorem must be
met in order to preserve information. By a similar analf x
f n
ysis to the one performed when sampling signals in their
f n
variable domain, it can be seen that the risk of aliasing is f x
also present but, in this case, it will occur as overlapping of
replicas of the sequence in the variable domain. This phef x
f x
f x
nomenon is referred to as aliasing in the variable domain.
When the sequence under consideration is of nite
length7 N , the use a sampling period of 2=N is a sufcient condition for properly sampling its spectrum. In
other words, given a sequence f n] of N samples, N samples of its frequency response F (ejw ) are enough to recover
it. In fact that would be the critical sampling period esVIII. Discussion
tablished by the sampling theorem.
The following problems or exercises are strongly recomFrom a mathematical point of view, the result of this mended
order to develop a better understanding of the
process is a couple of discrete periodic sequences of length conceptsindiscussed
before. Some of them might require
~
~
N , f n] and F k ], which are related to each other through
further
reading
or
research.
the following two equations
1.- Sampling Band-pass Signals
N ;1
~ k] = X f~n] e;j 2Nk n
F
(18)
The postulate of the Nyquist-Shannon theorem for samn=0
pling analog signals actually imposes a sucient condition
but not a necessary one. In fact, when dealing with bandNX
;1
pass signals, it is often possible to nd a sampling period
1
2
k
~n] =
~ k] ej N n
f
F
(19)
that is larger than the one established by the theorem and
N
k=0
still preserve the signal's information.
Suppose that we are dealing with a continuous-time
which are known as the discrete Fourier transform (DFT)
band-pass
signal with spectral content between 15 H z and
analysis and synthesis formulas respectively. The tilde is
25
H
z
.
What
are all the possible sampling period values
often used in the standard notation in order to emphasize
such
that
aliasing
does not occurs? How should the frethe periodic character of both sequences.
quency
response
of
the ideal reconstruction lter look like
~
~
The two new sequences, f n] and F k], are related to the
in
this
case?
jw
original sequence f n] and its frequency response F (e )
Suppose the general case of a band-pass signal with specas follows: f~n] is a periodic version of f n], and F~ k] is a
jw
tral
content between fmin and fmax. Under what condisampled version of F (e ).
tions is it possible to sample it properly by using a sampling
period larger than 1=fmin ? Can you postulate a general7 Which is always the case in practical applications since it is not
ized sampling theorem?
possible to handle innite-length sequences in a digital computer.
Consider again the band-pass signal with spectral content between 15 and 25 . Additionally, suppose that
it is a zero phase signal and its amplitude spectrum is symmetric around 20 . In this particular case, what is the
maximum possible sampling period?
Hz
Hz
Hz
;1
k n
1 NX
Integer
;j 2N
= 10 Otherwise.
k=0
n=N
(20)
= f1 1g and 2 = f1 ;1g. Compute the frequency responses ( jw ), 1 ( jw ) and 2 ( jw ) and make a sketch of
their respective amplitudes. Notice that 1 ] and 2 ] are
low-pass and high-pass lters respectively with frequency
responses very far from being ideal.
Now compute the discrete convolutions 1 = 1 and
2 . Multiply the resulting sequences 1 and 2
2 =
by the comb sequence ] = f 1 0 1 0 1 0 1 0 g to
obtain the new sequences ^1 and ^2 . Sketch the frequency
responses of this two new sequences. Are their spectra
aected by aliasing?
Compute two new discrete convolutions ~1 = ^1 ^1 and
~2 = ^2 ^2, where the new lters ^1 and ^2 are reversetime versions of the original lters i.e. ^1 = f1 1g and
^2 = f;1 1g. Finally, add the resulting sequences in order
to get ~ = ~1 + ~2 . Can you nd any relationship between
the original sequence ] and the obtained one ~ ]? Can
you nd an explanation for that?
6.- Unavoidable Aliasing
Although for practical purposes aliasing can be considered to be avoidable by means of an appropriate sampling
rate, from a mathematical point of view, aliasing only can
be avoided if innite length sequences are considered. In
practice, due to computational restrictions, nite length
sequences are always considered. Take some time to think
about the implications of these ideas and try to justify the
validity of practical discrete signal processing applications.
Consider now the DFT analysis, it seems that in this case
aliasing is always going to be present either in the frequency
domain or in the variable domain. What justication can
you give?
f1
G e
:::
:::
g n
1]
2]
3]
4]
5]
c n
References
g n