GEOPHYSICS,

PRINCIPLES

OF

DIGITAL

E. A. ROBINSOh-*

AKD

VOL.

XXIX,

NO.

3 (JUNE,

1964).

PP.

395-404,

4 FIGS.

FILTERING?

S. TREITELS

The digital computer is a versatile tool that may be used to filter seismic traces. Conventional filtering is performed by means of analog-type electronic networks, whose behavior is ordinarily studied in the frequency domain.
Digital filtering, on the other hand, is more fruitfully treated in the time domain. A digital filter is represented by a
sequence of numbers called weighting coefficients. The output of a digital filter is obtained by convolving the digitized input trace with the filters weighting coefficients. The mechanics of digital filtering in the time domain are
described with the aid of discrete z-transform theory. These ideas are then related to the more familiar interpretation of filter behavior in the frequency domain. An important criterion for the classification of filters is the notion
of minimum phase-lag. This paper ends with a new and simple presentation of this concept.
INTRODUCTION

The use of digital

computers

to process digi-

tized seismic recordings

as a research tool in the

development

methods

tablished.

of filtering

A partial

listing

subject as applicable

is now well es-

of the papers on this

to exploration

given in the references

geophysics

is

listed at the end of this

paper.
Our purpose here is to give a systematic
tation
bring

of digital

and,

out some new techniques

tions which
part

filtering

deserve further

of this paper

ment of the theory

presents
of digital

presen-

by so doing,

to

and interpreta-

attention.

The first

an expository
filtering

treat-

We chose to do this because no truly ele-

mentary

discussion of these topics has come to

attempt
domain

In the latter

to relate
filter

part

time-domain

theory,

of the paper we
and frequency-

and are led to a considera-

tion of the concept of minimum

phase-lag,

main point here is that it is the phase-lag,

phase, that must he considered
frequency

range when classifying

DIGITAL

continous

FILTERING

may
Each

be converted
number

not the

over the positive

filters.

data versus time

into a sequence of numbers.

represents

the reading,

or ampli-

tude, of the trace at a specific time instant.

time points

are chosen to be equally

between

process of converting

a sequence

two consecutive
the same, for ex-

a continuous

of numbers

at equally

tization
Today

was carried

there are various automatic

arrows

and semiauto-

for this purpose.

of a digitized

represent

the indicated

points.

Figure

trace. The vertical

the amplitudes

time

digi-

out by a scale and the eye.

matic devices available

shows a portion

trace
spaced

time points is called digitization.Formerly,

time scale appearing

or some other con-

of geophysical

into

interval

of the trace is always

one millisec.

The

venient

OF TRACES

seismic trace,

tinuous recording

Our

the time

ample,

in the time

domain.

our attention.

that

readings

of the trace

Instead

at

of using the

on the trace, it is more con-

to use a time index t so selected that the

time increment

is one unit.

index 1 associated
number,

or integer.

data of Figure

In this way the time

with any reading

For example,

xt is a whole

a portion

of the

1 becomes:

The

spaced, so

time
time

t Manuscript received by the Editor July 15. 1963.

* Consultant for Pan American Petroleum Corporation.

in set
index 1

Trace reading
XI

j Pan American Petroleum Corporation.

1.000

1.001

t=0

l=l

xu=lO

x1=20

1.002
f=2

n=lO

1 ,003
f=3

x3=0

(time increment =O.OOl set).

395

1.004

1.005

f=4

f=5

x$=-lo

ra=O

396

E. A.

Robinson

A digital jilter is represented by a sequence of

numbers called weighting coefficients. The simplest possible case is a digital filter with a constant weighting coefficient aa, that is, a constant
Jilter. Its action is shown schematically by the
block diagram where ovals indicate input and outDigital

Input: x1
Output:yl=;xl

S. Treitel

By a tandem connection we mean that the output

from Filter One is the input to Filter Two.
Kow, Filter One produces a unit delay, so its output is ~~-1, that is,

OUT

filter

put, and a rectangular box indicates the filter.

The variable t is the time index, where t=O, 1,
2, 3, . . . Alternatively,
we may illustrate the
action of the filter a0 by the table (where we have
let aO=$):
time index .t

and

10

20

10

O-10

10

O-5

Now we use the fact that the input to Filter Two

is the output from Filter One, so that the input
to Filter Two is at-I. Because Filter Two produces
a unit delay, its output is x1_-.,that is,

In terms of the readings, we have,

time

index I

Input:

Next, we introduce the concept of a digital filter

that produces a unit delay. This is the unit-delay
filter. Let us represent such a filter by the symbol
z. Thus, we have

Output
Input
--__
output:

0
10

Zt
from Filter
to Filter
21-2

One

Two

123
20

4
10

10

20

10

O-10

10

20

10

O-10

10

20

time lag is
two units

10

0 -10

In summary, we see that two-unit-delayfilters

IN

OIJT

unit delay

lay, that is, ij the input

In terms of the readings, we have y,=~~, y2=x1,

y3=x2, . . . , or:
time

index L
21

output:
Yt =21__1

10

20

10

time lag
is one unit

10

20

in

tandem are equivalent to a jilter with a two-unit de-

Input:

67

o-10

O-10
10

5
0
__--10

is xt, then the output is

x1-2. A delay of two units is thus represented by

the mathematical operator,

(i.e., z to the second power), the exponent 2 representing the delay:

0

The symbol z used here has a special mathematical meaning, that is, z represents a mathematical
operator which produces a unit delay. We thus
call z the unit-delay operator. If we write z more
explicitly as

(i.e., z to the first power) then the exponent 1 represents the delay.
What happens when we connect two unit-delay
filters in tandem? That is, suppose we have,

By the same reasoning, it is easy to see that

?zunit-delay jilters in tandem are equivalent to a jilter with an n-unit delay, that is, ij the i?zput is xi,
then the output is n-t-n. A delay of it units is represented by the mathematical operator,

(i.e., z to the 12th power), the exponent it representing the delay:

Principles

of Digital

One may ask what happens when n=O? Because the exponent represents the delay, we see
that in this case the delay is zero, and so input is
equal to output:

Thus, the filter z represents the identityfclter, and

in keeping with ordinary algebra we let
20 = 1.
In this way, we see that the constant filter a0
(described previously) may be represented more
explicitly by

Filtering

397

Up to this point, we have connected two digital

filters in series (or tandem). Now we wish to introduce a parallel connecfio?z.Such a connection
will be illustrated in our block diagrams by the
connecting element:

This figure illustrates that a parallel connecting element, when taken by itself, yields the same
output on each line of a fork. The combination of
the filter a0 and the filter urz connectedin parallel
to the same input x1 would yield the block diagram,

The tandem (or series) combination of a constant filter al followed by the unit-delay filter z
gives the filter arz; that is, the filter u1connectedin
series-withtheJlter z is shown by:
r-----

A mixer is a device which adds (or subtracts)

two inputs to yield an output. An example of this
device is:

or (with, say, al = $) :
time

index: t

Input:

.x1

azt

10

20

10

-10
-

2.5

Output:

y1 =a1.%__1

2.5

2.5

2.5

0
2.5
0

0
-2.5

We notice that the weighting coefficient a0 is associated with the constant, zero-delay filter, while
al is associated with the unit-delay filter. It is evident that the tandem combination of the z filter
followed by the al filter gives the filter zu,; that is

Another example of a mixer is:

or (with al = i) :
time

index: t

Input:

Xt

X1-1

output:

y1 =x1__1a1

o
10

12

20

10

10

20

2.5

O-10
10
2.5

Let us now consider the delay filter uO+arz,

which is shown by the block diagram,

56
0

0
0

-10
-2.5

0
0

Hence, we see that the filter al is equivalent to

the filter zar.

398

E. A.

Robinson

The output of the subcomponent a0 is aOxt. The

output of the subcomponent alz is UI.X_~. These
two outputs are then fed as inputs to the mixer,
which adds them, and, hence, produces the output Yt= UOX~+Q~-Z.~--~.
Let us illustrate numerically
the action of this filter for uo= f and a,=+, so
that our filter is ++$z:
time

index 1

Input:
z1
aXl

I2

and

S. Treitel

time

index I

Input:
21

10

20

Timelagis
2 units

x1-2

10
10

7.5

O-10
20

20

10

10

20

Zt-l

am-,

10

15

O-10

7.5

2.5

2.5

12.5

10

2.5

O-IO
0

-7.5

* . * u,Zn.

o-5
IO

The most-general delay filter has the form

56

-10

0
10

aa + UlZ + l&z? + u:jz3 + Q4Z4+

10

2.5

2.5

For example, the delay filter (with a finite

number of components), a~+a~z+a~z* has the
block diagram,

output:
Yr

=aoxr+alxt_,

-5

It is not difficult to verify that the delay filter

a~falz may also be illustrated by the block diagram:

We

may

numerically illustrate the filter

by letting ao=:, ul=$, and u2= 2:

aO+u,z+u22

As we have seen, the series connection of two

unit-delay filters is equivalent to a filter with a
two-unit delay:

time

index: 1

Input

xt
allxl

10

20

10

-10

10

-5

2.5

aLr_,

ax,

2.5

7.5

-1

output:

yt

12.5

Ii.5

0
0
0

-2.5

15

7.5

-7.5

17.5

2.5

-2.5

-i.5

'0

where:
The dashed rectangular box is the filter z2, which
is a two-unit delay filter. We recall that zn represents an n-unit delay filter, that is

y1 = aozt +
=

UlX,-I

UzXt--2

1/2xt + 114 St-_l + 3/a xt-_2.

An equivalent block diagram

u~+u~+u~z~ is given by:
r-----------

A constant filter a, connected in series with

for

(1)

the filter

the n-unit delay filter zn yields the filter anzTL,

shown by

For example, suppose n = 2 and us= j. Then the

filter a2z2is illustrated by the table:

The fcth-order delay filter u~+u~z+u~z*+ . . .

may be illustrated by either of the two
block diagrams:
+a,,z

Principles

of Digital

O
T

or,
~

r-___--______________,l

nput q

IN

rTis?-im,
;

81

k-1

Bn

z.++

++

THE

As a matter of terminology, the fzth order polynomial in 2,

F(z)

= an + aiz + a*22 +

. . + unzn, (2)

is called the z-&~~~zsformof the gzth order delay

filter. The constants
al,

al, a-2, . . . Q,

are the weighting coefficients of the delay filter.

An alternative description of such digital delay
filters, together with their relations to the analogtype delay-line filters, is given by Smith (1958).
Let us consider the action of the nth-orderdelay filter on an input given by the equally
spaced sampled values ~0, ~1, . . . , xm. Proceeding as in (l), we write for the output at time t:
yt = uoxt + u1xt_1 + a$c-2 +

399

This expression is the discrete representation of

the hear operation commonly known as convolution. In the literature one sees this operation
represented usually by an integral rather than by
a discrete summation. This is because the analog
filters dealt with by electrical engineers operate
in continuous time which calls for an integral
representation of the convolution operation. Since
we are here dealing with discrete time data, it is
necessary to represent the convolution process by
a summation. The output ot the fzth order digitaldelay filter is thus obtainable by the discrete convolution of the input -I~, ?;I, . . .rm with the
filters weighting coefficients au, nl, . . a,.

Inputxt
IN

Filtering

. . . u,,x~-r..

AMPLITUDE

AND

PHASE

OF DIGITAL

CHARACTERISTICS
FILTERS

The filters that we have discussed in the previous section all operate in the time domain. Therefore they may be called time-domain digital jilters.
The numerical examples presented thus far illustrate digital filtering in the time domain. Many
people, engineers in particular, are more accustomed to think about filtering in the frequency
domain. One can profitably study the action of
filters either in the time or in the frequency domains, or in a combination of both. The choice of
a particular approach depends on the nature of
the problem at hand. We shall now proceed to
sketch the relation that exists between time-domain and frequency-domain filtering. Before doing
this, a brief discussion of simple harmonic motion
is in order.
Simple harmonic or sinusoidal motion at a
given frequency may be illustrated by a wheel
rotating at a constant angular velocity. The following diagram shows a wheel of unit radius rotating counterclockwise at an angular velocity of w
radians per unit time

A more compact notation for this process is:

yt = 2

usztps fort

= 0, 1, 2, . . . 92 + n (3)

S=O

when t falls outside the range 1= 0, 1,

. . WZ+?Z.

y,=O
2 ,.

1 This definition of the z-transform follows mathematical usageas originatedby Laplace. Someengineering texts use 2-r in place of our z. We prefer Laplaces
version since it leads to z-transformswhich are polynomialsrather than Laurent series.

The two vectors show that an angle of w radians

is swept out in one time unit. The lower vector
corresponds to time t=O, the upper vector to
t= 1. Instead of considering a rotating wheel, we

@I,o

E. A. Robinson and S. Treitel

400

may simply think of a vector that rotates a t a

constant angular velocity w :

a0.l

Cutput vector at
tirre t

1n t vector at
t% t

c-

cos ut+

A t time t= 0, we suppose that the vector lies in the

positive direction along the horizontal coordinate
axis. Then a t some arbitrary time t, the vector
will make an angle of ut radians with the horizontal axis. The projection of this vector on the horizontal axis is seen to be:

Since both the input vector a t time t and the output vector a t time t make the same angle ut with
the horizontal axis, we say that the input and output are in phase. Here, we have tacitly assumed
that a0 is a positive constant. If, on the other
hand, a0 were a negative constant, say ao= -4,
we would have:

horizontal component = cos w t

while its projection on the vertical axis is:
vertical component = sin wt.

the input to the various digital filters which we

have considered. First, let us consider the constant filter ac. We have:

represents the rotating vector of length one which

may be drawn in the form:

me rotating vector
at time,t 0

which shows that the output is

yt

+ i sin ut)
a0 cos wt + iao sin ut.

= aOeiot = ao(cos ut
=

Hence, the output is also a rotating vector, but

instead of having unit length, the output vector
has length ao. For example, for UO= 1/2, we have:

,flu* + ?o

me rotating vector

.i(ot

*)

at time t

At time t=O, this vector lies on the horizontal

axis in the negative direction and makes an
angle of T radians with the positive horizontal
axis. This angle ?r is called the phase of the vector:

ei(ot+r).

Principles

of Digital

Filtering

401

Returning now to our example of the filter a,,=

-3, we may say that the filter output:
_

$eiwt

i(st+r)

1
+

$ cos

(wt

T)

i$

sin (wt + 7r)

can be pictured as a rotating vector of amplitude

f+ and phase = radians. If we divide the output
+ei(wt+a)by the input eiUt, we obtain the filters
so-called tralzsjev .function,
output
___input

= 3

eiu.

This quantity, which is in general complex, may

be described in terms of its magnitude and its
phase angle. The magnitude of the transfer function is known as the filters amplitude chauacferistic, while its phase angle is called the filters
phase ckaracterisfic. Thus, the amplitude characteristic of the filter UO= -4 is +$, while its phase
characteristic is r. We notice that both the amplitude characteristic and the phase characteristic
of this filter are in.depegzdenlsf angular velocity w.
Because the phase characteristic is constant and
equal to B, we may say that this filters input and
output are out oj plzase by 7r for all w, or simply
that they are 180 degrees out of phase.
In the same way, we may compute the transfer
function of any constant filter ao. Here:

Now we recall that the amplitude characteristic of

a filter is equal to the magnitude of the filters
transfer function. Because the vector eeLw has
unit magnitude, we see that the amplitude characteristic of the unit-delay filter is one. We also
recall that the plzase characteristic of a filter is
equal to the phase angle of the transfer function
of the filter. Since the vector ePiS makes an angle
of --w with the positive horizontal axis, we see
that the phase characteristic of this filter is --w.
Hence, the phase characteristic of this filter is a
function oj angular velocity w, although its amplitude characteristic is independent of w:

Summing up, we see that the transfer function

of the filter a0 is ao, and that the transfer function
of the filter z is cPiw.For the filter alz we have the
transfer function:
aleiw(t--l)

output

ale

The transfer function is:

output
-2
Input

%W.

eiul

Input

While the filter no+alz has the transfer function:

aOeiwt
___
= a0

aOeiWt+

output
p=
Input

eiot

a,eiw(-l)
= a0 + aleCi.
eiut

which is just the constant vector aO.

The next filter that we wish to consider is the
unit-delay filter z. We have:

The transfer function of the nth-order-delay

ter :

The transfer function is now:

may now be written down by induction:

output
___-=___=
Input

e;W(f--I)

output
e-ia

eicot

fil-

>

that is, the transfer function of the unit-delay filter is the constant vector EC&, which is shown in
the diagram :

Input

aOeiWl+

aleiw(I--L) +

. . . a,beiWt+n)

ezwl

= ao + ale- ia + . .

a,e.-imn.

402

E. A.

Robinson

Our results may be tabulated in the form:

and

S. Treitel

The length, 1A(w) I, of the vector A(w) is,

Filter

[ A(w) 1 = + d(afl + al cosw)~ + (al sin w)~

= + duo + 2uour cos w + ur2.

UC +
.

This quantity is the amplitude characteristic of

the filter eo+arz. The angle 4, which is a function
of w:

UlZ
.

+(w) = tan-
uo +

UlZ

Transfer

a,,z

-u1 sln w
a0 + a1 cos w

Function
=

tan-

al sin (J
a0 + a1 cos w

UJ

yields the phase characteristic of the filter aofarz.

We see that boUzthe amplitude and phase characteristics are functions of angular frequency w.

e-iu
ale-i

~20+ arePiw

THE

. . . . . . . . . . .

Thus, the transfer function of each filter is formally obtained by the substitution ehiU= z in the
filters z-transform. We notice that, except for the
case of the constant filter ao, the transfer function
always depends on the angular velocity w.
If we write the transfer function in polar form:

A(W) =

1A(w) 1e+(

we see that the magnitude 1A(w) 1 and the angle

4(w) represent, respectively, the amplitude and
phase characteristics of the filter. For example,
the transfer function of the filter
Qo +

a12

is:
A(w)

PHASE-LAG
A DIGITAL

CHARACTERISTIC

OF

FILTER

We have seen in the previous section that the

transfer function of a digital filter can be conveniently expressed in terms of an amplitude characteristic A(w) and a phase characteristic 4(w). It is
much easier to visualize the physical significance
of the amplitude characteristic of a filter than it
is to understand the corresponding phase characteristic. Some people therefore tend to neglect
the phase characteristic of filters when solving
actual problems. It turns out, however, that the
phase characteristic is of fundamental importance
in describing filters whose amplitude characteristics are id&id.
This fact is perhaps best illustrated by a simple example.
Let us consider two filters, one with z-transform:

= a.0 + ure-iU
= (a. + al cos w) -

MINIMUM

F(Z) = 1 + 0.52,
i(ur sin w).

the other with z-transform:

Now A(w) (for a fixed value of w) is the vector

which is the sum of the vectors a~ and u~c~:

Irr(z)

= 0.5 +

lz.

In other words, the weighting coefficients of filter

F0 are (1, 0.5) while the weighting coefficients of
filter 17,are (0.5, 1). We recall that the amplitude
characteristic of the filter
F(z)

= cr.0+ arz

is:

1 A(w) 1 = + duo2 + 2U&

cos w + u12.

Principles

of Digital

403

Filtering

Setting uo= 1 and ar=O.S in this formula, we find

that the amplitude characteristic of the FO(z) filter is:

1A,(w) 1 = + &+ cos w +0.25

= + 41.25 + cosw.
Setting a,,=O.S and al = 1, we find that the amplitude characteristic of the Fr(z) filter is:

1 Al(U) 1 = + -\/0.25+
= + 41.25

cos w +1
+ cos w
FIG. 3. The phase characteristics of

which is the same as the above expression for,

1Ao(w)1. Thus, the two filters have the same
amplitude characteristic, which is shown in Figure 2 for the frequency range --r SW Ix. Negative frequencies are used for mathematical conIn
venience (see e.g. Goldman, 1948, p. 70-72).
our discussion of simple harmonic motion, a positive frequency would indicate that the wheel is
rotating in a counterclockwise direction, whereas
a negative frequency would indicate rotation in a
clockwise direction.
The question now arises: What is the relation
between the phase characteristics of the two filters 170(z) and Fr(z)? We recall that the phase
characteristic of the filter a,+~
is given by

the filters FO(z) and FI(z).

&(w)

= -

tan-r
(0.5:

ilIos a)

These two-phase characteristics are plotted in

Figure 3 for the range -r<w
<a.
Instead of considering the phase 6(w), many
people, engineers in particular, prefer to deal with
the negative phase, -c$((L~), which is called the
phase lag. The phase-lag characteristics of the
filters F(z) and PI(z) are given by
tan-

--&J(W)

-+r(w)

= tan-r

0.5 sin w
1 + 0.5 cosw

ar sin w
+(w) =

tan-r
( a0 + 6?rcos 0 ).

Thus, for the Fa(z) filter we have with no= 1 and

ar=o.s:
0.5 sin w

&(w)

tan-r

_
( 1 + 0.5 cos w)

On the other hand, letting ao=O.S and al= 1, we

have for the Fr(e) filter:

FIG. 2. The amplitude characteristic of the

filters F&) and F,(Z).

Co..5 Es

w).

Since phase as well as phase-lag curves are odd

functions of angular frequency w (that is, they
are antisymmetrical about the origin), we need
only to plot -&(w) and -+(w)
for w in the range
0 to rr. This is done in Figure 4. We thus see that
the phase-lag -&(o)
lies below the phase-lag
-+1(w). We are now in a position to state that
the filter Fe(z) has a phase-lag characteristic
which is less than the phase-lag characteristic of
the filter Fr(z) for the range O<w<a. At w=O,
the phase-lag characteristics of both filters are
zero.
Let us restrict ourselves to real weighting coefficients only, and let us consider only the delaytype digital filters, for which the output can
never precede the input in time Then, we see that
the pair of filters {Fe(z), F,(z)} represents a set
of digital delay filters each with the same amplitude characteristic. This set is complete in the

E. A. Robinson

and

S. Treitel

1961; Foster, Hicks, and Nipper, 1962; Kunetz,

1961; Rice, 1962; and many others).
CONCLUDING

w (RADIANS/UNIT

TIME)

FIG. 4. The phase-lag characteristics of

the filters F(z) and F, is).

sense that the two real weighting coefficients a0

and al can occur either in the sequence a,,, a~ or in
the sequence aI, ao. We can now say that the filter Fe(z) has the minimum phase-lug characteristic
of the filter set (PO(z), Fl(z) 1. The concept of
minimum phase-lag filters is quite general and can
be extended to sets of n&order delay filters, each
filter in the set having the same amplitude characteristic. There is in each such set one filter
whose phase-lag characteristic is minimum with
respect to the phase-lag characteristics of all
other members of that set, and this filter is the
minimum phase-lag filter. Alb filters within the
given set have the seme amplitude characteristic.
These matters are intimately related to questions of filter stability that arise when one wishes
to operate digitally on a seismic trace in order to
convert it into a continuous velocity log. This
process, known as deconvolution or inverse
convolution, has received considerable attention
in the recent geophysical literature (Backus,

REMARKS

Several treatments have appeared during the

past few years w-hich deal with end results obtained by digital means (cf. Smith, 1958; Rice,
1962). For this reason our present discussion has
been primarily concerned with the underlying
principles of digital filtering.
We have shown in some detail how digital filters operate on an arbitrary digitized input to
yield a discrete output. We found that the mechanics of this process can be visualized with
greatest ease in the time domain, but it is usually
necessary to think of filtering in terms of both the
time and the frequency domains. We have thus
attempted here to present a thorough, although
somewhat heuristic description of filter behavior
in both these domains. We have introduced the
minimum phase-lag concept for classifying digital filters.
REFERENCES

Backus, M., Deconvolution and the one-dimensional

reflection seismogram:oral presentationat the 31st

International SEG Meeting, Denver, Colo., November 1961.
Foster, M. R., Hicks, W. G., and Xipper, J. T., 1962,
Outimum inversefilters which shortenthe snacingof
velocity logs:Geophysics,v. 27,~. 317-326. _
_
Goldman,, S., 1948, Frequency analysis, modulation,
and norse:McGraw-Hill Book Co., Inc., 434 1,.
Kunetz, G., 1961, Essai danalyse de traces sismiques:
Geoph. Prosp., v. 9, p. 317-341.
MIT Geophysical Analysis Group 1952-1957, GAG
Reports No. 1 through lSo. 11: Massachusetts Institute of Technology, Cambridge, Mass.
Rice, R. B., 1962, Inverse-convolution filters: Geophysits, v. 27, p. 4418.

Robinson. E. A.. 1957. Predictive decompositionof

seismictraces: Geophysics;,v. 22, p. 767-778.
Smith, M. K., 19.58, A rewew of methods of filtering
seismic data: Geophysics, v. 23, p. 44-57.
Swartz, C. A., and Sokoloff, V. M., 1954, Filtering associated with selective sampling of geophysical data:
Geophysics, v. 19, p. 402-419.
Weld, H., 1938, A study in the analysis of stationary
time series: Uppsala, Almqvist and Wiksells.