Statistical and Transform Methods

for Seismic Signal Processing

M. D. Sacchi

Department of Physics
University of Alberta

M.D.Sacchi / STMSSP

Contact:
Dr M.D.Sacchi
Department of Physics,
University of Alberta,
Edmonton, Canada, AB, T6G 2J1
Sacchi@phys.ualberta.ca
www-geo.phys.ualberta.ca/~ sacchi

M.D.Sacchi / STMSSP

Review of basic tools

M.D.Sacchi / STMSSP

1.1

Fourier Series

Let us assume that we want to approximate a function

f (t) by a superposition of n orthogonal functions:

f (t)

N
X

ci i (t)

i=1

Z t2
t1

i (t)j (t) dt = ij

The coefficients ci , i = 1 . . . N can be obtained by

minimizing the means square error defined as:
N
X
1 Z t2
(f (t)
i (t))2 dt
M SE =
t 2 t 1 t1
i=1

ci =

M.D.Sacchi / STMSSP

R t2
t1

f (t)i (t) dt
.
2 dt
(t)
t1

R t2

Consider the orthogonal set given by

ejn0 t ,

n = 0, 1, 2, 3, . . .

this set is orthogonal in t [t0 , t0 +

2
].
0

When a signal is expanded in terms of exponential we

have a Fourier series:
f (t) =

Fn ejn0 t

n=

2 Z t0 +2/0
f (t)ejn0 t dt
Fn =
0 t0

Remark: Line Spectrum

M.D.Sacchi / STMSSP

1.2

Fourier Transform
F () =

f (t)ejt dt

1 Z
f (t) =
F ()ejt d .
2

F () = |F ()|ej()

Remark: Continuous spectrum

M.D.Sacchi / STMSSP

1.3

Properties of the FT

We define the transform pair as:

f (t) F ()

Symmetry.
F (t) 2f ()

Linearity.
f1 (t) F1 ()
f2 (t) F2 ()

f1 (t) + f2 (t) F1 () + F2 ()

Scale.
f (at)

M.D.Sacchi / STMSSP

F( )
|a| a

Convolution. If
f1 (t) F1 ()
f2 (t) F2 ()
Z

f1 (u)f2 (t u)du F1 ()F2 ()

Convolution in frequency.
1 Z
f1 (t).f2 (t)
F1 (v)F2 ( v)dv
2

Time delay.
f (t ) F ()ejt0

Modulation.
f (t)ej0 t F ( 0 )

Time derivatives.
df (t)
jF ()
dt

M.D.Sacchi / STMSSP

1.4

The FT of some signals

A Boxcar

f (t) =

|t| < T /2

otherwise

R T /2

F () =
=

T /2

1.ejt dt

1
(ejT /2
j

ejT /2 )

T sinc(T /2)

Boxcar T=10 sec

1.5

f (t)

0.5

0.5
80

....

.... +

60

40

20

0
time [sec]

20

40

60

80

10
8

F ()

2
0

....

.... +

2
10

0
[rad/sec]

10

Figure 1: Boxcar.

M.D.Sacchi / STMSSP

Delta function:
f (t) = (t)
Z

g(u)(u)du = g(0)

F (w) =

(t)ejt dt = 1

(t) 1

(t ) 1.ej
A complex sinusoid:
(t ) 1.ej

F (t) 2f ()

ej0 t 2( 0 )

M.D.Sacchi / STMSSP

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1.5

Truncation in time

Given f (t) t (, ), with f (t) F (), how do we

obtain the FT of the signal when the signal is recorded in
a finite interval [T /2, T /2]?.
fT (t) = f (t).bT (t)
Using the frequency convolution theorem:
1 Z
1
FT () =
F (v)BT ( v)dv =
F () BT ()
2
2
BT () = T sinc(T /2)
.
Signal observed in [10,10]

Real[f(t)]

1
0
1
80

60

40

20

0
time [sec]

20

40

60

80

60

40

20

0
time [sec]

20

40

60

80

0
[rad/sec]

Imag[f(t)]

1
0
1
80

|F()|

30
20
10
0
4

Figure 2: Truncation in time.

M.D.Sacchi / STMSSP

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1.6

Symmetries

If the signal f (t) is a real signal, we can write:

F () = R() + iG()
where
R() =

f (t)cos(t)dt and G() =

f (t)sin(t)dt

R() = R()

G() = G()
Amplitude and Phase:
F () = |F ()|ei() .
|F ()| = |F ()|

() = () .

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1.7

Living in a discrete World

fs (t) = f (t)

(t kt) .

k=

X
1
F () 0
Fs () =
( k0 ) ,
2
k=

0 =

2
T

1 X
Fs () =
F ( n0 )
T k=

If one wants to compute Fs () in such a way that F ()

can be completely recovered, the signal f (t) must be a
band-limited signal. This is a signal where the spectral
components outside the interval [max , max ] are zero. If
the following condition is satisfied
0 2max
1
2
2 2fmax then, T
.
T
2fmax
The last equation is also designated as the sampling or
Nyquist Theorem.

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Fourier transform of the continuous signal

0.8

F()

0.6

max

= 10 rad/sec

0.4

0.2

0
100

80

60

40

20
0
20
Frequency (rad/sec)

40

60

80

100

Figure 3: The Fourier transform of a continuous signal.

Fourier transform of the discretized signal
4.5

3.5

2.5

F ()

.....

.....

1.5

0 = 30 rad/sec
max = 10 rad/sec

0.5

t = 0.2094 sec
0
100

80

60

40

20
0
20
Frequency (rad/sec)

40

60

80

100

Figure 4: The Fourier transform the continuous signal after

being discretized, in this case max = 10 and 0 = 30

M.D.Sacchi / STMSSP

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Fourier transform of the discretized signal

3

2.5

.....

.....

F ()

1.5

1
0 = 20 rad/sec
max = 10 rad/sec

0.5

t = 0.3142 sec
0
100

80

60

40

20
0
20
Frequency (rad/sec)

40

60

80

100

Figure 5:

The Fourier transform the continuous signal after being discretized,

in this case max = 10 and 0 = 20. The Fourier transform of the continuous
signal is perfectly represented in the interval [wmax , wmax ].
Fourier transform of the discretized signal

.....

.....

F ()

1.5

= 15 rad/sec

max = 10 rad/sec

0.5

t = 0.4189 sec
0
100

80

60

40

20
0
20
Frequency (rad/sec)

40

60

80

100

Figure 6:

The Fourier transform the continuous signal after being discretized,

in this case max = 10 and 0 = 15. The signal is aliased. Note that Nyquist
theorem is not satisfied. The Fourier transform of the continuous signal cannot
be recovered from the Fourier transform of the sampled signal.

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Linearity, z-Transform and

Convolution

M.D.Sacchi / STMSSP

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2.1

Linear Systems

Input/output analysis of a linear system

x(t) y(t)
If the system is linear the following properties must be
satisfied:
P1 :
x(t) y(t) .
P2: If
x1 (t) y1 (t)
x2 (t) y2 (t)
then
x1 (t) + x2 (t) y1 (t) + y2 (t) .
P3: Properties P1 and P2 can be combined in a single
property:
x1 (t) + x2 (t) y1 (t) + y2 (t) .

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2.2

Time Invariant Systems

Time invariance means:

x(t T ) y(t T )
We will represent our linear system as follows:
H[x(t)] = y(t) .

(1)

General expression for a linear system

y(t) = H[x(t)] =

h(t, )x( )d .

(2)

If the system is also time invariant

y(t T ) = H[(x(t T )] .

y(t) =

h(t )x( )d

(3)

(4)

In the freq. domain:

Y () = H() .X() .

M.D.Sacchi / STMSSP

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Time Invariant Linear System

x(t)

h (t)

y(t)

(t)

h (t)

h(t)

x(t) : Input,

y(t): Output,

h(t): Impulse response

Figure 7: A continuous linear time invariant system.

Time Invariant Linear System (discrete case)
x0,x1,x2,x3,...

h0,h1,h2,h3,....

y0,y1,y2,y3,...

1,0,0,0,0,...

h0,h1,h2,h3,....

h0,h1,h2,h3,....

x : Input,

y: Output,

h: Impulse response

Figure 8: A discrete time invariant linear system.

M.D.Sacchi / STMSSP

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2.3

Discrete Convolution

If the system is discrete, this is a system where the input

and output signals are discrete signals, the convolution
integral becomes a summation:

yk =

hn xkn

(5)

n=

Finite length signal:

xn , n = 0 : N X 1 is a signal of length N X
yn , n = 0 : N Y 1 is a signal of length N Y
hn , n = 0 : N H 1 is a signal of length N H

yk =

p2
X

hkn xn ,

(6)

n=p1

where p1 and p2 indicate the finite summation limits.

Smart Conv:
y(1:NX+NH-1) = 0
for i=1:NX
for j=1:NH
y(i+j-1) = y(i+j-1) + x(i)*h(j)
end
end

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Assuming that x = [x0 , x1 , x2 , x3 , x4 ] and h = [h0 , h1 , h2 ],

and after carrying out the convolution sum (Eq.6):
y0 = x 0 h 0
y1 = x1 h0 +x0 h1
y2 = x2 h0 +x1 h1 +x0 h2
y3 = x3 h0 +x2 h1 +x1 h2
y4 = x4 h0 +x3 h1 +x2 h2
y5 =

x 4 h1

y6 =

+x3 h2
x 5 h2

or

M.D.Sacchi / STMSSP

y0

y1

y2
y3
y4
y5
y6

x0

x1 x0

x2 x1 x0
x3 x2 x1
x4 x3 x2
0

x 4 x3

x4

h0

h1

h2

21

2.4

The z transform

A digital signal:
x0 , x 1 , x 2 , x 3 , . . .
can be represented using the Z-transform:
X(z) =

xn z n

k=0

For a finite length time series xk , k = 0, . . . , N 1 we write

X(z) =

N
1
X

xn z n

k=0

Example 1:
x = 4, 12, 1, 3 ,

X(z) = 4 + 12z 1z 2 + 3z 3 .
Example 2:
x = 1, 3, 4, 3, 5, 6, 10

X(z) = z 3 + 3z 2 + 4z 1 + 3 + 5z + 6z 2 10z 3 .

M.D.Sacchi / STMSSP

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2.5

Convolution and the Z-transform

We have two times series, x = [x0 , x1 , x2 , x3 , x4 ] and

h = [h0 , h1 , h2 ]. The Z-transforms of these series are:
X(z) = x0 + x1 z + x2 z 2 + x3 z 3 + x4 z 4

H(z) = h0 + h1 z + x2 z 2
Now, let us compute the product of the above
polynomials:
X(z).H(z) = x0 h0 +
(x1 h0 + x0 h1 )z +
(x2 h0 + x1 h1 + x0 h2 )z 2 +
(x3 h0 + x2 h1 + x1 h2 )z 3 +
(x4 h0 + x3 h1 + x2 h2 )z 4 +
(x4 h1 + x3 h2 )z 5 +
(x5 h2 )z 6
One can see that the coefficient of this new polynomial are
the samples of the time series y = [y0 , y1 , . . . , y6 ] obtained
by the convolution of x and h, in other words, X(z) .H(z)
is the also the Z-transform of the time series y:
Y (z) = X(z) .H(z) .

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2.6

Elementary Signals: Dipoles

Minimum Phase Dipoles

A simple manner of visualizing the properties of a time
series in the z domain is by decomposing the Z-transform
into dipoles or elemental function of the type
1 + az
As an example, we compute the Z-transform of the series
x = [4, 12, 1, 3]:
1
1
X(z) = 4 + 12z 1z 2 + 3z 3 = 4(1 + z)(1 z)(1 + 3z) .
2
2
We have already seen that two multiply the Z-transform
of two time series is equivalent to convolve the time series
in the time domain,
1
1
4, 12, 1, 3 = 4[ (1, ) (1, ) (1, 3z)] .
2
2

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Inversion of Minimum Phase Dipoles

Let us assume that the dipole, which I will call D(z), is
given by
D(z) = 1 + az .
The inverse of the dipole F (z) must satisfy
F (z)D(z) = 1 then, F (z) =

1
1
=
,
D(z)
1 + az

if |a| < 1 the denominator can be expanded according to

the following expression a :
F (z) = 1 az + (az)2 (az)3 + (az)4 . . .
Since |a| < 1 the above series is a convergent series. F (z)
is the z transform of the following series
1, a, a2 , a3 , a4 , . . .

The convolution of the dipole with the filter yields

(1, a) (1, a, a2 , a3 , a4 , . . .) = 1, 0, 0, 0, 0, 0, . . .

A geometric series.

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Let us compute the inverse of the following dipoles:

(1, 0.9) and (1, 0.01).
Example: (1, 0.9)
F (z) =

1
= 10.9z+0.81z 2 0.729z 3 +0.6561z 4 . . . .
1 + 0.9z

(1, 0.9)(1, 0.9, 0.81, 0.729, 0.6561) = (1, 0.0, 0.0, 0.0, 0.59)
Example: (1, 0.01)
F (z) =

1
= 10.1z+0.01z 2 0.001z 3 +0.0001z 4 . . . .
1 + 0.1z

(1, 0.1)(1, 0.1, 0.01, 0.001, 0.0001) = (1, 0.0, 0.0, 0.0, 0.0) .
It is clear that the truncation is negligible when a = 0.1.
This is not true when a 1. In this case a long filer is
needed to properly invert the dipole.

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Dipole, d = (1,0.9)

Truncated inverse filter, f

Output, d f

Dipole, d = (1,0.5)

Truncated inverse filter, f

Output, d f

Dipole, d = (1,0.1)

Truncated inverse filter, f

Output, d f

M.D.Sacchi / STMSSP

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Minimum Phase Signals

A signal can be factorized in dipoles
X(z) = x0 +x1 z+x2 z 2 +x3 z 3 . . . = A(1+a1 z)(1+a2 z)(1+a3 z) . . . .
If |ai | < 1, i, the signal is a minimum phase signal. In
this case all the zeros lie outside the unit circle
X() = 0 i =

1
|ai | < 1 |i | > 1 .
ai

The inverse filter F (z) of X(z) can be calculated from

X(z)F (z) = 1
(1 + a1 z)(1 + a2 z)(1 + a3 z) . . . . F (z) = 1 .
From the above equation where we can write
F (z) = (1 + a1 z)1 (1 + a2 z)1 (1 + a3 z)1 . . .
= [(1 a1 z + (a1 z)2 (a1 z)3 . . .][(1 a2 z + (a2 z)2 (a2 z)3 . . .]
[(1 a3 z + (a3 z)2 (a3 z)3 . . .] . . . .
The final inverse operator can be written as
f0 , f1 , f2 , . . . = (1, a1 , a21 , a31 . . .)(1, a2 , a22 , a32 . . .). . . . . .

M.D.Sacchi / STMSSP

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Maximum Phase Dipoles

Elementary signals of the form (1, b), |b| > 1 are called
maximum phase dipoles
D(z) = 1 + bz D() = 1 + b = 0 = 1/b .
Since |b| < 1, it is easy to see that || < 1. The inverse of
a maximum phase dipole is a non-causal sequence.
F (z)D(z) = 1 F (z) =

1
1
=
D(z)
1 + bz

If last equation is expanded in a series of positive powers

of z we have
1
= 1 bz + (bz)2 (bz)3 . . .
1 + bz
The last series does not converge. Try this
F (z) =

1
1
=
1 + bz
bz(1 + (bz)1 )

F (z) = (bz)1 (1 (bz)1 + (bz)1 (bz)3 . . .) .

Now the inverse is stable but non-causal, the associated
operator is given by f = . . . , b3 , b2 , b1 , 0

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Example, (1, 2)

f = (0.0156, 0.0312, 0.0625, 0.125, 0.25, 0.5, 0 )

Convolution of f with the maximum phase dipole:

d f = (0.0156, 0.0312, 0.0625, 0.125, 0.25, 0.5, 0 ) (1, 2)

= (0.0156, 0, 0, 0, 0, 0, 1, 0)

Dipole, d = (1,2)

Truncated inverse filter (noncasual) , f

Output, d f

Figure 9: Maximum phase dipole, its non-causal truncated

inverse, f , and the output d f .

M.D.Sacchi / STMSSP

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Autocorrelation of Dipoles
The autoccorelation function of a sequence with
z-transform X(z) is defined as
R(z) = X(z)X (z 1 )
We will consider two dipoles of the form:
Dmin (z) = 1 + az Min Phase

Dmax (z) = a + z Max Phase

Autocorrelations
Rmin (z) = a z 1 + (1 + |a|2 ) + a z ,

Rmax (z) = a z 1 + (1 + |a|2 ) + a z .

The two dipoles have the same autocorrelation function!!
Rmax (z) = Rmin (z) = R(z)

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Amplitude and Phase of Dipoles

Minimum phase case
Dmin (z) = 1 + az z = ei Dmin () = 1 + aeiw .
Maximum phase case
Dmax (z) = a + z z = ei Dmax () = a + 1eiw

RDmin () =

1 + 2a cos() + a2

min () = arctan(

RDmax () =

a sin()
)
1 + a cos()

1 + 2a cos() + a2

max () = arctan(

M.D.Sacchi / STMSSP

sin()
)
a + cos()

32

Amplitude of 1+0.5z

Ampliture of 0.5+z)

1.3

1.3

1.2

1.2

1.1

1.1

0.9

0.9

0.8

50

100

150

0.8

Phase of 1+0.5z)

50

100

150

Phase of 0.5+z

0
0.5

0.05

1
0.1

1.5

0.15

2
2.5

0.2
0.25

3
0

50

100

150

3.5

50

100

150

Figure 10: Amplitude and phase spectrum of a minimum

phase dipole 1 + az and a maximum phase dipole a +z,
|a| < 1.

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2.7

Least Squares Inversion of Minimum

Phase Dipoles

We have already solved:

D(z)F (z) = 1 .
The solution of last problem yields a filter with an infinity
number of coefficients (Not a good idea!).
Now, our task is to construct a finite length filter with the
following property
D(z)FN (z) 1 ,
where FN (z) denotes the z transform of the finite length
operator. The above equation can be written in the time
domain as follows (assuming N = 3),
(1, a) (f0 , f1 , f2 ) (1, 0, 0, 0) .

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In matrix form

1
1 0 0

f0

0
a 1 0

.
f1

0 a 1
0

f2
0
0 0 a

The last system of equations corresponds to an

over-determined system that can be solved using the
method of least squares
Cf b ,
the solution vector is the one that minimized the mean
squared error
= ||C f b||2 .
The least squares solution of this system is found by
solving the following system of normal equations
CT C f = C T b .
The resulting filter is
f = R1 CT b ,

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Examples:
Dipole, d = (1,0.9)

Output, d f

Least squares inverse filter , f

Figure 11: Inversion of a minimum phase dipole using least

squares.
Dipole, d = (1,0.5)

Output, d f

Least squares inverse filter , f

Figure 12: Inversion of a minimum phase dipole using least

squares.

M.D.Sacchi / STMSSP

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2.8

Least Squares Inversion of a Wavelet

Given a signal (wavelet) w, we want to find f such that:

(w0 , w1 , . . . wN W 1 ) (f0 , f1 , . . . , fN F 1 ) (1, 0, . . . , )
In matrix form we can write the following expression
(assuming N W = 7 and N F = 4)

w0

w1 w0

w2 w1 w0

w3 w2 w1 w0
w4 w3 w2 w1
w5 w4 w3 w2
w6 w5 w4 w3
0

w 6 w5 w4

M.D.Sacchi / STMSSP

w 6 w5
0

w6

f0

f1

f2

f3

0
0
0
0
0
0
0
0

37

Again, this system is written in matrix form as Cf d.

We will compute the inverse filter by minimizing the error
function (mean squared error) :
= ||e||2 = ||C f b||2 ;
The Euclidean norm of the error vector e = C f b can
be written down as
= eT e = (C f b)T (C f b) .
The mean squared error is minimized when the following
condition if satisfied:
d
= 0,
df

CT C f = C T b .

f = (CT C )1 CT b .

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2.9

Stability

Pre-whitening (zero order quadratic regularization):

J = + ||f ||2 ,
The solution is now given by the penalized least squares
estimator where the parameter is also called the
regularization parameter ( ridge regression parameter or
pre-whitening parameter).
The condition
dJ
= 0,
df

f = (R + I)1 CT d .
When is large the energy of the filter will be small and
the misfit function will be large. In this case we have a
matching filter of the form
f = 1 CT d .

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Trade-off curve
Output, wf

Filter, f

=1.e6

=1.e4

Figure 13: A minimum phase wavelet inverted using different tradeoff parameters ().
Tradeoff curve
1

=0.1

0.9

0.8

0.7

Misfit,

0.6

0.5

0.4

0.3

0.2
=100000.

0.1

0.1

0.2

0.3
0.4
2
Filter norm, ||f||

0.5

0.6

0.7

Figure 14: Trade-off curve for the previous example. The

vertical axis indicated the misfit and the horizontal the
norm of the filter.

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2.10

The z transform and the Discrete

Fourier Transform (DFT)

We have already defined the Z-transform of a time series

as follows:
X(z) =

N
1
X

xn z n .

n=0

The z-transform provides a representation of our time

series in terms of a polynomial. Let us introduce the
following definition:
z = ei
in this case the z-transform becomes the DFT:
X() =

N
1
X

xn ein

n=0

We have mapped the Z-transform into the unit circle

(z = ei is a complex variable of unit magnitude and
phase given by ). The phase is also the frequency
variable in units of radians.

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The function X() is discretized according to

k =

2
k , k = 0, . . . , N 1
N

Xk = X(k ) =

N
1
X

xn eik n

n=0

4=

10

7
6

3 4= 5 6 7

21

25
0
26
27

3 4=

Figure 15: Distribution of frequencies in the unit circle.

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42

DFT in Matrix form:

X = F.x

F=

... 1

1 ei2/N

ei22/N

...

ei24/N
..
.

...
..
.

1
..
.

ei22/N
..
.

ei2(N 1)/N

i22(N 1)/N

..

1 ei2(N 1)/N ei22(N 1)/N . . . ei2(N 1)

/N

It can be shown that F is an orthogonal matrix

FH F = N I N ,

x=

1 H
F X.
N

The IDFT (inverse DFT) is given by

1
1 NX
xn =
Xk ei2kn/N
N k=0

M.D.Sacchi / STMSSP

n = 0, . . . , N 1 .

43

Deconvolution

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44

3.1

Modeling Normal Incidence

Seismograms

Incident wave: in medium 1

Reflected wave: in medium 1
Transmitted wave: in medium 2
1

I1
I2

r: reflection coefficient
t: transmition coefficient
1+r=t

Figure 16: P-wave normal incidence. The incident wave is

propagating downwards.

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45

Let us assume that the amount of incident wave is equal

to 1, the amount of reflected wave is given by r, and the
amount of transmitted wave is denoted by t. At the
boundary the following condition should be satisfied
(continuity of displacements)
1+r =t
This equation has two unknowns, to compute the r and t
we need an extra equation. Conservation of energy leads
to the following equation:
I1 1 + I 1 r 2 = I 2 t 2 .
The quantities I1 and I2 are called acoustic impedances
I 1 = 1 v1
I 2 = 2 v2
where 1 and 2 are the densities of the material above
and below the interface and v1 and v2 the P-velocities,
respectively.
Combining the two boundary conditions leads to:
r =

t =

M.D.Sacchi / STMSSP

I2 I 1
Reflection coefficient
I2 + I 1

2I1
Transmition coefficienct
I2 + I 1

46

Lets consider the case of an incident wave propagating

upwards:
Incident wave: in medium 2
Reflected wave: in medium 2
Transmitted wave: in medium 1

t
I1
I2

r: reflection coefficient
t: transmition coefficient
1+r=t

r = -r

Figure 17: P-wave normal incidence. The incident wave is

propagating upwards.
r0 =

I1 I 2
I2 + I 1

t0 =

2I2
I2 + I 1

From the above equations it is clear that

r0 = r

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47

3.2

Impulse Response

Lets assume that we run a zero offset experiment in a

stratified earth composed of 4 layers plus a half-space.
The impedances are given by I1 , I2 , I3 , I4 , I5 .
z=0
1

(1+r1)(1+r2)(1+r3)r4(1-r3)(1-r2)(1-r1)

I1
Interface 1

z1
1+r1

(1+r1)(1+r2)(1+r3)r4(1-r3)(1-r2)

(1+r1)(1+r2)

(1+r1)(1+r2)(1+r3)r4(1-r3)

(1+r1)(1+r2)(1+r3)

(1+r1)(1+r2)(1+r3) r4

I2
z2

I3
z3

I4

Reflection at interface 4

z4
Interface 4

I5
Ii = Acoustic impedance of the layer i
1+ri = Transmition coef.
1-ri = Transmition coef.

Figure 18: Amplitude of a wave plane wave propagating

in a layered medium. Analysis of the wave reflected in the
interface 4.

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48

The amplitude of the wave when reaches the layer 4 is

1 t1 t2 t3 = (1 + r1 )(1 + r2 )(1 + r3)
when the wave is reflected in layer 4 the total amplitude
at that point (last expression) needs to be multiplied by
the reflection coefficient of interface 4,
1 t1 t2 t3 r4 = (1 + r1 )(1 + r2 )(1 + r3)t4
Using
1 + r0 = 1 r
the final amplitude after propagating the wave to the
surface of the Earth is given by
(1 + r1 )(1 + r2 )(1 + r3 )

{z

Transmition

4
|{z}

Reflection

(1 r1 )(1 r2 )(1 r3 )
|

{z

Transmition

The final expression for the amplitude of the wave

reflected in interface 4 can be written down as follows

(1 r12 )(1 r22 )(1 r32 )r4 .

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49

It is clear that reflections occur at all the layers:

Amplitude of the reflection generated at interface 1
A1 = r1
Amplitude of the reflection generated at interface 2
A2 = (1 r12 )r2
Amplitude of the reflection generated at interface 3
A3 = (1 r12 )(1 r22 )r3
Amplitude of the reflection generated at interface 4
A4 = (1 r12 )(1 r22 )(1 r32 )r4
Important: The actual impulse response of the system
should include all the multiples.

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3.3

The Seismogram

The source is not 1, we need to consider a source wavelet

w(t):
s(t) = A1 w(tt1 )+A2 w(tt2 )+A3 w(tt3 )+A4 w(tt4 )
where t1 , t2 , t3 and t4 are the arrival times of each
reflection a
Notice that if transmission effects are neglected, the
amplitude Ai can be replaced by the reflection
coefficient ri .
Microlayers (a discrete Earth model using layers of
equal travel-time) leads to the convolutional model
sn = w n q n .
qn is the reflectivity sequence.

Notice that w(t ) is w(t) after being delayed seconds.

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3.4

Deconvolution of Reflectivity Series

Goals:
> Improve resolution
> Consistency of sources (All seismograms after
deconvolution should have the same residual wavelet)
> Guarrantee phase stability
Mathematical model:
In general we will assume that deterministic noise
(multiples and ground roll) has been attenuated and
therefore what is left is random noise
sn = w n q n + n n
It is clear from the above equation that one has a problem
with one equation (one observable) and two unknowns
(the wavelet and the reflectivity). Therefore, the seismic
deconvolution problem involves the solution of two
subproblems:
Wavelet Estimation
Operator design

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52

3.5

The White Reflectivity Assumption

If the reflectivity is a zero-mean white noise process, the

following remarkable property is true:
rjs = Pq rjw .
rjs : autocorrelation of the seismic trace (observable)
Pq energy of the white reflectivity
rjw : autocorrelation of the wavelet

IMPORTANT:
1 The autocorrelation function of the trace is an estimate
(within a scale factor) of the autocorrelation of the
wavelet.
2 It is clear that now we can estimate the autocorrelation
of the wavelet from the autocorrelation of our observable:
the seismic trace.
3 Where is the phase???

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53

3.6

What to do with the noise?

We start with our noisy seismogram:

sn = w n q n + n n
The goal of the deconvolution process is to recover qn
from the data, sn . Assume that we have compute and
inverse filer fk and we apply the filter to the seismogram
qk = ak qk + fk nk
= qk + (ak k ) qk + fk nk .
ak is the residual wavelet, ak = wt ft
It is clear that we need to satisfy the following two
equations:
ak = wk f k k ,

fk n k 0

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54

The last two expression can also be written in matrix

form
Cw f d

Cn f 0
Both equations are honored when we minimize the
following objective function:
J = ||Cw f d||2 + ||Cn f ||2 ,

f = (Rw + I)1 CTw d .

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55

Deconvolved reflectivity =0.001

seismogram (No noise)

Reflectivity, q

Figure 19: Deconvolution of a clean seismogram.

Deconvolved reflectivity =0.001

Noisy seismogram

Reflectivity, q

Figure 20: Deconvolution of a noisy seismogram. Small

Deconvolved reflectivity =0.05

Noisy seismogram

Reflectivity, q

Figure 21: Deconvolution of a noisy seismogram. Large .

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56

3.7

Deconvolution in the Frequency

Domain

A procedure similar to the one outlined in the previous

section can be used to deconvolve data in the frequency
domain.
k = Qk + (Ak 1)Qk + Fk Nk .
Q
k : DFT of the Reflectivity estimate
Q
Qk : DFT of True the Reflectivity
Ak : DFT of the Residual wavelet
Fk : DFT of the Inverse filter
The inverse filter should be a good approx. to the inverse
of the wavelet:
WK Fk = Ak 1 , and avoid noise amplification: Fk Nk 0
We combine the last two equations into a single cost
function
J=

X
k

|Ak 1|2 +

|Fk Nk |2 .

Minimizing the cost function with respect to the filter

coefficients leads to

W
k
.
Fk =
|Wk |2 + |Nk |2

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57

Inversion of Impedance Profiles

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58

4.1

Norms for Sparse Deconvolution

Quadratic regularization (Wiener shaping filter)

sk =

wj qkj ,

k = 1, ny

error function
k = s k

wj qkj ,

k = 1, ns .

J =

X
k

1 (

k
),
k

1
1 (u) = u2 .
2
X
J

=
( )wkl = 0 ,
ql
n
k
XX
k

wkj wkl qj =

l = 1, nq

wkl sk ,

or in matrix form
Rq = g .

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59

Stabilization (quadratic constraint)

J = J + Jq ,

J1 1 =

X
k

1 (

X
qi
k
)+
1 ( ) .
n
q
i

The minimum of the last equation is reached at the point

= (R + I)1 g ,
q

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60

Modifying Jq (non-quadratic regularization)

Introduce a new regularization (Huber norm) term

2 (u) =

J1 2 =

u2 /2

if |u| a
2

a |u| a /2 if |u| > a

X
k

X
k
qi
1 ( ) +
2 ( ) .
n
q
i

Alternatively (Cauchy Regularization):

u2
3 (u) = ln( + 1) .
2

J1 3 =

X
k

X
k
qi
1 ( ) +
3 ( ) .
n
q
i

Sacchi, M.D., 1997, Re-weighting strategies in seismic deconvolution: Geophysical Journal International, 129, 651-656.

M.D.Sacchi / STMSSP

61

Iterative solution
(R + Q)q = g .
when Huber norm is adopted:

Q2 i i =

if | qiq | a

1
a

q
| i
q

if | qiq | > a

when the Cauchy norm is adopted:

Q3 i i =

1
1+

qi2
2q2

Algorithm:
1. Start with an initial reflectivity sequence x0
2. Select the hyperparameters of the problem n , q , and
a (Huber criterion) or n and q (Cauchy criterion).
3. Compute = n2 /q2 , Q(0) , and the source
autocorrelation matrix R.
4. Iteratively solve above system
q(k) = (Q(k1) + R)1 g

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62

Remarks
Quadratic regularizations tends to smooth or damp
the solution; this is why we cannot recover spikes.
Non-quadratic norms like the Huber and Cauchy
norns have the ability to retrieve spikes. This is
consistent with the assupmtion of a sparse reflectivity
sequence

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63

Cost and Influence Functions for Sparse Deconvolution

Figure 22: Cost functions. a) 1 (u), b) 2 (u) a = 1 , c)

2 (u) a = 2 , and d) 3 (u).

Figure 23: Influence functions. a) 1 (u), b) 2 (u) a = 1 ,

c) 2 (u) a = 2 ,and d) 3 (u).

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64

Examples

Figure 24:

a) Synthetic impulse response (left) , seismogram (center), and

source wavelet (right). The seismogram was contaminated with Gaussian noise.
b) Deconvolution using zero order quadratic regularization: estimated impulse
response (left), reconstructed seismogram (center), and residuals (original minus
reconstructed data). c) Deconvolution by means of the Huber criterion (a = 1).
d) Deconvolution by means of the Cauchy criterion.

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65

Figure 25: (a) Original seismic section. (b) Deconvolved seismic section using the Cauchy criterion to regularize the inversion. The source wavelet was retrieved using a combined
cepstrum-cumulant approach.

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66

4.2

Bayesian Inversion of Impedance

Impedance constraints (zk ) for a layered Earth model

(vertical incidence)
Nk
X
1
k = ln(zk /z0 ) =
qi ,
2
i=1

(7)

Wq = s + n

(8)

Cq = + ,

(9)

Data constraint

Impedance constraint

General cost function

1
1 1
J = |{z}
Jx + || (Wq s)||2 + ||S1 (Cq )||2 (10)
|2
{z
} |2
{z
}
1

1 - The solution must be sparse.

2 - The solution must honour the seismic trace.

3 - The solution must honour a set of impedance
constraints.

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67

Example: Huber Norm

Data and Reflectivity Estimate

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68

Impedance

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69

4.3

Linear Programming Impedance

Inversion

Sparse reflectiovity inversion via LP methoda

sk = w k q k .
Data constraints
Wq = s + e
Constrained minimization using linear programming
The cost function of the problem is defined as
J = |q|1 + |e|1
Constraint minimization problem

Minimize J = |q|1 + |e|1

subject to
and impedance constraints

Wq = s + e
l < Cq < u .

Oldenburg, D. W., Scheuer, T. and Levy, S., 1983, Recovery of the acoustic
impedance from reflection seismograms: Geophysics, 48, 1318-1337.

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70

Higher order spectra and

non-minimum phase wavelet
estimation

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71

5.1

Non-minimum Phase Wavelet

Estimation

Autocorrelation based method do not preserve phase

information
Higher order statistics do preserve phase information
when the reflectivity is a white non-Gaussian process
Model:
x k = qk wk
s k = x k + nk

W (z) = |W (z)|ei(z) .
1. If qk is Gaussian and W (z) is minimum phase,
autocorrelation based methods will correctly identify both
the amplitude and the phase of the system.
2. If qk is Gaussian and W (z) is nonminimum phase, no
technique will correctly identify the phase of the system.
3. If qk is non-Gaussian and W (z) is nonminimum phase,
true magnitude and phase of the system transfer function
can be recovered by knowing the actual distribution of qk .
For MA processes of order one, it has been demonstrated
that a L1 optimization provides an estimate of the
amplitude and phase of the system when the driving noise
of the process is non-Gaussian.

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72

3-rd order statistics

If W (z) is stable and if qk is non-Gaussian, white, iid.,
with skewness 6= 0 then the bispectrum of xk is given by
Bx (z1 , z2 ) =

XX
m

rx(3) (m, n)z1n z2m

where
rx(3) (m, n) = E[xk xk+m xk+n ] ,
is the third order moment of the data. Since the third
order moment of a Gaussian signal vanishes, we can write
Bs (z1 , z2 ) = Bx (z1 , z2 ) . The bispectrum can be written in
terms of the transfer function W (z) as follows (Nikias and
Raghuveer, 1987)
Bx (z1 , z2 ) = W (z1 )W (z2 )W (z11 z21 ) .
Important: Phase is preserved!

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73

4-th order statistics

Defining the trispectrum as
Tx (z1 , z2 , z3 ) =

XXX
m

rx(4) (n, m, l)z1n z2m z3l

where rx(4) (n, m, l) is now the fourth order cumulant of xk .

The trispectrum can be written in terms of the transfer
function of the system as follows
Tx (z1 , z2 , z3 ) = W (z1 )W (z2 )W (z3 )W (z11 )W (z11 z21 z31 )
is the Kurtosis. This is valid for a non-gaussian
symmetric distribution ( =
6 0).

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74

4-th order statistics: Wavelet estimation procedure

Cumulant Matching
Find a wavelet (optimization problem) that
matches the cumulant of the data a
Kolmogoroff factorization

retrieve the wavelet from the cumulant using a

procedure similar to the Hilbert Transform
Advantages:
Phase can be estimated if assumptions are satisfied.
Problems:
It is quite difficult to satisfy assumptions

Velis D. R., and Ulrych T. J., 1996, Simulated annealing wavelet estimation
via fourth-order cumulant, Geophysics, vol.61, 1939-1948.
b
Sacchi, M.D., and Ulrych, T.J., 2000, Non-minimum phase wavelet estimation using higher order statistics: The Leading Edge, 19, 1, 80-83.

M.D.Sacchi / STMSSP

75

Example: 4-th order Cumulant factorization

Figure 26: Synthetic wavelet.

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76

Figure 27: Wavelet estimation using the tricepstrum. The

fourth order cumulant was estimated from 4 segments of
1000 samples each. The figures correspond to 20 realizations of the process.

M.D.Sacchi / STMSSP

77

Figure 28: Wavelet estimation using the tricepstrum. The

fourth order cumulant was estimated from 4 segments of
500 samples each. The figures correspond to 20 realizations
of the process.

M.D.Sacchi / STMSSP

78

Figure 29: Wavelet estimation using the tricepstrum. The

fourth order cumulant was estimated from 4 segments of
250 samples each. The figures correspond to 20 realizations
of the process.

M.D.Sacchi / STMSSP

79

Real data example

Figure 30: Segment of seismic section pre-processed for

impedance inversion.

Figure 31: Wavelet estimates computed using the tricepstrum (left) and cumulant matching plus non-linear optimization (right).

M.D.Sacchi / STMSSP

80

5.2

Minimum entropy deconvolution

In MED a we exploit, again, the assumption of a white

non-Gaussian reflectivity (sparse reflectivity assumption)
Find ft (a filter)
q t = s t ft
such that
N
X
1
V (q) =
qi0 .F (qi0 ), is maximized
N F (N ) i=1

where
qi0

yi 2
.
=P
2
k yk /N

F (q 0 ) = ln(q 0 ) For a logarithmic norm

Solution involves an iterative algorithm of the form
Rf k = gk1 that can be interpreteted as a shaping filter
with a non-linear (sparse) desired output!!
a

Wiggins, R. A., 1978, Minimum entropy deconvolution: Geoexpl., v.16, p.21-

35.

M.D.Sacchi / STMSSP

81

Frequency domian MED algorithm

The MED is extended to deal with the problem of

band-limited extrapolation (recovery of low/high
freqs.)
Can be used to improve the freq. content of the
residual wavelet obtained after conventional
deconvolution
a

Sacchi,M.D., Velis D., & Cominguez, A.H., 1994, Frequecny domain Minimum Entropy deconvolution: Geophysics, vol. 59, No. 6.

M.D.Sacchi / STMSSP

82

FX Processing

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83

6.1

FX filters

Remark: Predictability of linear eventsa

The signal model:
S(f, x) = A(f ) ei2f x ,
where A(f ) indicates the source spectrum, f the temporal
frequency, x the spatial variable or offset and the
apparent slowness along x. We will assume that the
spatial variable x is regularly discretized according to
x = (k 1)x , k = 1 : N . For any temporal frequency, f ,
we can write
Sn = A ein ,

n = 1, N

where = 2f x. The following recursion is obtained

by combining Sn and Sn1
Sn = a1 Sn1 .
where a1 = exp(i). The last equation is a first order
difference equation that allows us to recursively predict
the signal along the spatial variable x.
a

Canales, L. , 1984, Randon noise reduction: 54th Annual SEG Meeting,

p.525-527.

M.D.Sacchi / STMSSP

84

Model for p linear events

Sn = a1 Sn1 + a2 Sn2 + . . . ap Snp .

The last equation can be written in prediction error form
as follows
p
X

gk Snk = 0 ,

k=0

where
g0 = 1, gk = ak ,

k = 1, p .

In real applications, however, additive noise will corrupt

the data
Yn = S n + W n ,
The model for the noisy signal becomes
Pp

k=0 gk Ynk =

Pp

k=0

gk Wnk

= en .

The latter is an ARMA(p,p) process in which the AR and

MA components are identical.

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85

6.2

AR FX Filters

Rather than trying to solve the ARMA equations one can

replace the ARMA model by a long AR (Autoregressive)
model:
Yn f1 Yn1 + f2 Yn2 . . . + fp Ynp = Wn .
The last equation can be written in matrix form as follows
(assume p = 3),

y1

y2 y1
y3 y2
y4 y3
0

y4

y1

y2

y3

y4

f1

f2

f3

y2

y3

y4

w2

w3

w4

Solve for the filter that minimizes: Yf d = w

J = ||Y f d||2 , J = 0, => f = (Y H Y)1 YH d ,

= Yf .
Clean signal d

M.D.Sacchi / STMSSP

86

The convolution matrix:

We have adopted a very simple convolution matrix in
order to design our filter via the Levinsons recursion. But
bear in ming that other data matrices can be used to
estimate the data prediction filter.
Canales (1984) original formulation uses the following
model

y1

y2 y1
y3 y2
y4 y3
0

y4

y1

y2

y3

f1

f2
=
f3

y4

y2

y3

y4

Ulrych and Clayton (1976) proposed the transient-free

convolution matrix. In our simple example (p = 3) the
transient-free matrix formulation is given by

M.D.Sacchi / STMSSP

y4
y3 y2 y1

f1
y5
y4 y3 y2

=
f2

y5 y4 y3
y6

f3
y7
y6 y5 y4

87

Algorithm
1. Transform the data into the F X domain
Data(t, x) Data(f, x)
2. for each frequency f solve the 1D-AR prediction
problem
3. Apply the filter to the data (convolution of the filter
with the data).
4. Transform back to T X
Data(f, x) Data(t, x)
5. end

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88

1D Example
Prediction of harmonics using AR filters p=20
5

50

100

150
n, number of samples

200

250

300

Figure 32: Prediction of a single harmonic (no noise) using

AR filters
Prediction of harmonics using AR filters p=20
6

50

100

150
n, number of samples

200

250

300

Figure 33: Prediction of a single harmonic (noise = 0.2)

using AR filters

M.D.Sacchi / STMSSP

89

Input, n=0.2
0
20
40
60
80
100
120
0

10

15

20

25

30

25

30

Output after FX filtering, p=20

0
20
40
60
80
100
120
0

10

15

20

Figure 34: F X filtering of a single linear event immersed

in noise
Input, n=0.2
0
20
40
60
80
100
120
0

10

15

20

25

30

25

30

Output after FX filtering, p=20

0
20
40
60
80
100
120
0

10

15

20

Figure 35: F X filtering of a single linear hyperbolic event

immersed in noise

M.D.Sacchi / STMSSP

90

6.3

FX Gap filling and the recovery of

near offset traces

Original idea published in seismologya . Assume a gap in

the interval n1, n2. The prediction error for the AR model
in terms of the AR coefficients is given by
En = g0 Yn + g1 Yn1 + g2 Yn2 + . . . gp Ynp
Since the gap comprises the traces Yn1 , Yn1 +1 , . . . Yn2 , we
will minimize the squared sum of the prediction error
within the gap
n2
X

Ek Ek

k=n1

Taking derivatives with respect to the samples in the gap

we end up with the following system
Rg (Yn1 Yn1 +1 . . . Yn2 )T = b
The missing data in the gap can be recovered by knowing
the PEF.
a

Wiggins, R. A, and Miller, S. P., 1972, New noise-reduction technique applied

tolong-period oscillations from the Alaskan earthquake: SSA Bull., 62, 417-479

M.D.Sacchi / STMSSP

91

(m)
0

50

(m)
100

150

50

100

150

Time (sec)

0.5

Time (sec)

0.5

1.0

1.0

1.5

1.5
Original Data

Data after Gap Filling

Figure 36: Predictability of linear events. The AR gap

filling techniques is used to estimate a PEO wich is latter
utilized to fill the gap.

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92

Source-Receiver Distance (m)

0
2000

2
Time (sec)

Time (sec)

-2000

-2000

Source-Receiver Distance (m)

0
2000

Original Data

Data after Gap Filling

Figure 37: oz01.dat Original (left), after gap filling with a

PEO(3)

M.D.Sacchi / STMSSP

93

6.4

F X Projection Filters

In this section we will study the problem of

quasi-predictability and the solution via projection filters

Space domain formulation

s = s0 + n Additive noise

As0 = i Predictability .
In this case the vectors denote the spatial series of
observations s and the unknowns s0 and n. The operation
As0 also corresponds to convolution with the PEO.
Soubaras proposed to minimize the following cost
function:
J = (s s0 )T (s s0 ) + s0 T AT As0
taking derivatives yields to
s = (I + AT A)s0
since s = s0 + n
a

Soubaras R., 1994, Signal preserving random noise attenuation by the f-x
projection: 64 Annual SEG Mtg. 1576-1579.

M.D.Sacchi / STMSSP

94

s = (I + AT A)(s + n) then n = (I + AT A)1 AT As

After using the identity
AT (I + AAT )1 = (I + AAT )1 AT ,
n = AT (1 I + AAT )1 As .
The projection filter deconvolves the PEO filter. To prove
this assessment suppose that we know the filter A.
Unfortunately is not true that we know the s0 , but
suppose that we apply the filter to s
As = A(s0 + n) = As0 + An .
It is clear from the last expression that the filter output has
two sources of errors: The color introduced by the PEF and
the noise. The color introduced by the PEF can be partially
removed by deconvolution:

n = AT (1 I + AAT )1 (As)
|

{z
2

}| {z }
1

The operation 1 does the filtering of the signal and the noise,
meanwhile the operation 2 removes the color that the filter
has introduced into the noise (deconvolution of the PEO).

M.D.Sacchi / STMSSP

95

6.5

ARMA formulation of Projection

filters

It is important to note that underlying signal model in

f X is an ARMA model.
Pp

k=0 gk Ynk =

Pp

k=0

gk Wnk

= en .

The problem now can be summarized as follows, given the

ARMA representation of the noise signal:
1. How do we estimate the prediction error filter gk ?
2. How do we use gk to estimate the additive noise
sequence Wk .?
These two fundamental points are discussed in the
following pages.

M.D.Sacchi / STMSSP

96

Estimation of the ARMA prediction error filter

ARMA system in matrix form:
Yg = Wg
= e,

Y=

Y0

Y1 Y0
Y2 Y1
Y3 Y2
0

Y3

Y0

Y1

Y2

Y3

After some mathematical manipulations we can arrive to

RY g = P W g .
The matrix RY = YH Y is the Toeplitz correlation matrix
of the noisy data.
the PEF is the eigen-vector associated to the min.
eigenvalue of the data covariance matrix
the minimum eigenvalue is an estimate of the power
of the noise

M.D.Sacchi / STMSSP

97

Noise estimation
Re-write the ARMA system as follows (Convolution can
commute)
Gy = Gw .
The matrix G is the convolution matrix of the prediction
error filter, y and w are vectors containing the
observations and the white noise sequence, respectively.
Minimize
J = [G(y w)]H [G(y w)]
subject to
w H w = Pw .

M.D.Sacchi / STMSSP

98

6.6

Noise estimation

The constrained minimization problem is solved by

introducing a Lagrange multiplier to combine the last two
equations into a new objective function
J 0 = [G(y w)]H [G(y w)] + (wH w PW ) .
The objective function J 0 is minimized by the following
estimator of the noise sequence
= (GH G + I)1 GH Gy .
w
can be estimated as follows:
The clean signal, s = y w,
s = [I (GH G + I)1 GH G]y .
Remarks:
= y.
If = 0, w
= 0.
If is too large, w

M.D.Sacchi / STMSSP

99

1D - Projection filter - Examples

4

Wk

20

40

60
Samples

80

100

120

Figure 38:

D synthetic example. A sinusoid of normalized wavenumber k 0 =

0.05 is used to test the ARMA filtering method described in the paper. The data
are contaminated with white noise with standard error = 0.5.
6

=0.0001
4
=0.01

=1

=100

=10000

20

40

60
Samples

80

100

120

Figure 39:

The eigenfilter estimated from the noisy data is used to estimate

the noise. In this figure we portray the estimator of the noise sequence versus the
tradeoff parameter . Large values of will completely annihilate the noise. Small
values of will overestimate the noise (the signal leaks into the noise sequence.)

M.D.Sacchi / STMSSP

100

=0.0001

4
=0.01
3

=1
2

=100
1

=10000
0

20

40

60
Samples

80

100

120

Figure 40:

Estimator of the signal versus the tradeoff parameter . Note that

for = 0.01 the signal is recovered.
=0.0001
0

=0.01

Amplitude Spectrum [dB]

20

=1
40

=100
60

=10000
80

100

120

0.02

0.04

0.06

0.08
0.1
0.12
Normalized wavenumber

0.14

0.16

0.18

0.2

Figure 41:

Amplitude response of the projection filter utilized to estimat e

the noise. Note that large values of attenuates both signal and noise. On the
other h and, small value of does not properly attenuate the signal.

M.D.Sacchi / STMSSP

101

ARMA (Projection filter) vs AR Filtering (Canales

Method)
0

Time (sec)

0.1

0.2

0.3

FX ARMA(3,3)

Figure 42:

Left: The original data (3 waveforms immersed in spatially uncorrelated noise). Center: Filtered data using the ARMA representation. Right:
Estimate of the noise.
0

Time (sec)

0.1

0.2

0.3

FX AR(15)

Figure 43:

Left: The original data (3 waveforms immerse in spatially uncorrelated noise). Center: Filtered data using the AR representation (Conventional
f x r andom noise attenuation). Right: Estimate of the noise.

M.D.Sacchi / STMSSP

102

Example: SNR of a Common Offset Section

0.2

0.4
Time (sec)

0.6

Figure 44:

Left: A window of a common offset gather. Center: The filtered

section using the ARMA representation with a 3 points eigenfilter (p = 2). Right:
Conventional f x random noise attenuation (p = 15).

M.D.Sacchi / STMSSP

103

Eigenimages

M.D.Sacchi / STMSSP

104

7.1

Decomposition in Eigenimages

Consider seismic data as a matrix. The matrix can be

decomposed in r eigen-vectors ui , vi :
X=

r
X

i ui viT .

i=1

Reconstructed data matrix using p-eigenimages

Xp =

p
X

i ui viT .

i=1

p<r
Xp retains the information that is coherent trace to
trace

M.D.Sacchi / STMSSP

105

Eigen-images: Example
Input, =0.2
n

0
20
40
60
80
100
120
0

10

15

20

25

30

25

30

Reconstruction with 1 Eigenimage

0
20
40
60
80
100
120
0

10

15

20

Figure 45: A flat event immersed in nose and the reconstruction by means of the first eigenimage
25

20

15

10

10

15
20
Singular value index, i

25

30

35

Figure 46: Spectrum of singular values.

M.D.Sacchi / STMSSP

106

Input, n=0.2

Reconstruction with 1 Eigenimage

Reconstruction with 2 Eigenimages

Reconstruction with 3 Eigenimages

Figure 47: A Parabolic event immersed in nose and the

reconstruction by means of the 1,2 and 3 eigenimages
16

14

12

10

10

15
20
Singular value index, i

25

30

35

Figure 48: Spectrum of singular values.

M.D.Sacchi / STMSSP

107

7.2

Eigen-structure based Velocity

Analysis

We use the eigen-decomposition of the data covariance

matrix as a coherence measure.a
In velocity analysis the covariance matrix is computed
from windows (gates) extracted from the data along a
hyperbolic trajectory (NMO curve).
SNR Coherence measure

1 PN

1
i=2 i /(N 1)
b
.
C=
PN
N
i=2 i /(N 1)
i are estimators of the eigenvalues for a given gate of

analysis.

Key S.C. and Smithson S. B., 1990, New approach to seismic-reflection event
determination: Geophysics, 55, 1057-1069.

M.D.Sacchi / STMSSP

108

Figure 49: Left: Semblance of a CMP gather. Right: High

resolution coherence analysis (SNR measure).

Figure 50: Left: Average SNR measure obtained via bootstrapping individual realizations. Right: Frequency distribution of
the peak that maximizes the coherence after 50 bootstrap realizations.

M.D.Sacchi / STMSSP

109

Radon Processing

M.D.Sacchi / STMSSP

110

8.1

Slant Stacks

Let u(h, t) represent a seismic signal in offset-time. We

define the slant stack or Linear Radon transform as
follows:

v(p, ) = (Lu)(p, ) =

u(h, t = + h p)dh .

(11)

Where p and denote the slope or ray parameter and the

intercept time, respectively. v(p, ) is used to designate
the signal in the p domain.
The adjoint transform L is given by

u(h, t) = (L v)(p, ) =

M.D.Sacchi / STMSSP

v(p, t = hp)dp .

(12)

111

Slant stacks in the freq. domain:

In the frequency domain, the pair of transformations are
given by,

V (p, ) =

U (h, )eiph dh,

(13)

U (h, ) =

V (p, )eiph dp,

(14)

Last two equations can be combined:

U (h, ) =

U (h0 , )

eip(hh ) dp dh0

(15)

which may be written as follows

U (h, ) = U (h, ) (h, ) ,

(16)

where denotes convolution and the function is given

by
(h, ) =

eiph dp ,

(17)

making the substitution z = p equation (17) becomes

M.D.Sacchi / STMSSP

112

(h, ) =

2
1 ihz
e dz =
(h) .
||
||

(18)

The convolution operator is a delta function with respect

to the variable h. Using the property of the function,
U (h, ) =
=

2
U (h, )
||
2
U (h, )
||

(h)

(19)

the inversion formula becomes,

U (h, ) =

M.D.Sacchi / STMSSP

||
U (h, ) .
2

(20)

113

8.2

The Inverse Slant Stack Operator

The definition of the forward slant stack operator and its

adjoint may be changed to construct another slant stack
pair,

u(h, t) = (L v)(p, ) =

v(p, ) = (Lu)(h, t) =

v(p, t = hp)dp

(21)

u(h, t = + hp)dh,

(22)

the pair of transformations can be posed in the

frequency-offset domain,
U (h, ) =

V (p, )eiph dp,

(23)

V (p, ) =

U (h, )eiph dh .

(24)

||
V (p, ) .
2

(25)

After some math:

V (p, ) =

M.D.Sacchi / STMSSP

114

8.3

Discrete Slant Stacks

Discrete versions of equations the continuous Radon pair

are obtained by replacing integrals by summations and
imposing finite limits.
Lf
X

v(p, ) = (Lu)(p, ) =

u(hl , + hl p)hl ,

(26)

l=Ln

u(h, t) = (L v)(, p) =

JX
max

v(h, t hp)pj

(27)

j=Jmin

Taking the Fourier transform of the above equations

yields
V (p, f ) =

Lf
X

U (hl , f )e2if hl p hl

(28)

V (p, f )e2if hpj pj .

(29)

l=Ln

U (h, f ) =

JX
max

j=Jmin

Using matrix notation

m = LH d

= Lm
d

M.D.Sacchi / STMSSP

115

Inversion formula Discrete Slant Stacks

= L LH d .
d

(30)

d = (L LH )1 d
.
= G1 d

(31)

m = Ld
1

d = G Lm .

M.D.Sacchi / STMSSP

(32)

116

8.4

Slant Stacks: The Least Squares

Solution

Assume that the data is the result of applying a Radon

operator (slant stack) to a m.
d = Lm

(33)

The idea is to find m such that the following objective

function is minimized
J = ||d Lm||2

(34)

The solution to this problem is the least squares solution

m = (LH L)1 LH d

(35)

In general the inverse needs to be stabilized using a

damping parameter.
m = (LH L + I)1 LH d

M.D.Sacchi / STMSSP

(36)

117

Example: LS Linear Radon

(a)

(b)
0

0.2

0.2

0.4

0.4

t (s)

(s)

0.6

0.6

0.8

0.8

0.05

0
h (km)

0.05

(c)

0
p (s/km)

(d)

0.2

0.2

0.4

0.4
t (s)

t (s)

0.6

0.6

0.8

0.8

0.05

0
h (km)

0.05

0.05

0
h (km)

0.05

Figure 51: Linear RT synthetic example. (a) Events in t x.

(b) p panel computed using the LS linear RT. (c) and (d)
Separated waveforms using the p panel.

M.D.Sacchi / STMSSP

118

8.5

Parabolic Radon Transform

This is a simple modification to the slant stack, instead of

integrating along curves of the form
t = + ph
we use curves of the type
t = + qh2
This is a good approximation to process data containing
hyperbolic events after NMO correction.

M.D.Sacchi / STMSSP

119

Residual Moveout and the Parabolic Radon Transform

Traveltime for a multiple and a primary
Tp =

Tm =

(T02 + h2 /vp2 )

2 ).
(T02 + h2 /vm

Assume that vp > vm . Traveltimes after NMO with vN M P :

TN M O = T0

2
(T02 + h2 /vN
MO) ,

Tp (Af ter) = Tp + TN M O
if vN M O = vp
Tp (Af ter) = T0 .

Tm (Af ter) = Tm + T0

2
(T02 + h2 /vN
MO) ,

or after replacing Tm
Tm (Af ter) = T0 +

M.D.Sacchi / STMSSP

(T02

2 )
h2 /vm

2
(T02 + h2 /vN
MO)

120

Expanding in Taylor series

Tm (Af ter) = T0 +

(T02

Tm (Af ter) T0 +

2 )
h2 /vm

2
(T02 + h2 /vN
MO)

1
1
2
2
h

h
2
2
2T0 vm
2T0 vN
MO

Tm (Af ter) T0 + qh2

q=

1
1 1
( 2 2
).
2 T0 v
vN M O

A Radon transform with the following integration path is

capable of collapsing parabolas (multiples) to points in
the , q space.
h2
t= +q 2 ,
hmax

M.D.Sacchi / STMSSP

121

Example: LS Parabolic Radon Transfrom

(a)

(b)
0

0.2

0.2

0.4

0.4

t (s)

(s)

0.6

0.6

0.8

0.8

0.01

0.02 0.03
h (km)

0.04

0.05

0.05
q (s)

(c)

0.1

(d)

0.2

0.2

0.4

0.4
t (s)

t (s)

0.6

0.6

0.8

0.8

0.01

0.02 0.03
h (km)

0.04

0.05

0.01

0.02 0.03
h (km)

0.04

0.05

Figure 52: Least squares Parabolic Radon Transform synthetic

example. (a) Events in t x. (b) q panel computed using
the LS parabolic RT. (c) and (d) Separated waveforms using
the isolate the waveforms.

M.D.Sacchi / STMSSP

122

8.6

Radon De-multiple

Working flow:
NMO the CMP gather with the vN M O vp
Compute the Radon Gather
Mute primaries
Come back to CMP domain > This is your
estimate of multiples
Subtract multiples from data
Problems:
Not enough focusing of primaries and multiples
(smearing)
Alias might hampered the reconstruction and
therefore the substraction of the model of multiples
from the data.
One solution to these problems is to design a high
resolution Radon Transform

M.D.Sacchi / STMSSP

123

Least squares Parabolic Radon Transform

d(xj , t) =

M
X

m(qk , = t qk x2j ) , j = 1, N ,

k=1

d(xj , t): CMP gather, m(qk , ): Radon panel

In the frequency domain:
d(xj , f ) =

M
X

m(qk , f ) ei2f qk xj , j = 1, . . . , N .

k=1

The calculations can be carried out independently for

each frequency f .
d(f ) = L(f ) m(f ) .
To avoid notational clutter we will drop the frequency
dependency and write d = L m . The least squares Radon
operator is estimated by minimizing the following cost
function.
J = ||d L m ||2 + ||m||2 .

m = (LH L + I)1 madj

=

(R + I)1 madj .

Hampson, D., 1986, Inverse velocity stacking for multiple elimination: J.

Can. Soc. Expl. Geophys., 22, 44-55.

M.D.Sacchi / STMSSP

124

Least squares Parabolic Radon Transform

FAST ALGORITHM, Levinson alg. a
Elements of R
{R + I}l,m =

N
X

ei2f q(lm)xk + l,m .

k=1

This is a Toeplitz matrix, solving this equation using the

Levinson recursion requires approximately 4M 2 + 7M
operations, and storage of only the first row of the
Toeplitz matrix.

Kostov, C., 1990, Toeplitz structure in slant-stack inversion: 60th Annual

Internat. Mtg., Soc. Expl. Geophys., Expanded Abstracts, 1618-1621.

M.D.Sacchi / STMSSP

125

8.7

High Resolution Radon

Solve the following problem

(R + WH W)m = madj .

{W}l,m = wl l,m , l, m = 1, . . . , M .

(37)

The elements of W are chosen to increase the resolution

of the operator.
The operator is not a Toeplitz form; therefore the solution
of the system of equations is proportional to M 3 (Too bad
for Marine data sets!).

wk2

M.D.Sacchi / STMSSP

100.

if qk
/Q

0.0001

if qk Q ,

126

High Resolution Parabolic Radon

FAST VERSION, CG+FFT a

Use CONJUGATE GRADIENTS to solve the problem

iteratively!
We want to solve (R + D)m = madj , where D = WH W.
Start with an initial solution m0 , set
p0 = r0 = madj (R + D)m0 ,
i+1 = (ri , ri )/(pi , (R + D)pi )
mi+1 = mi + i+1 pi ri+1 = ri i+1 (R + D)pi
i+1 = (ri+1 , ri+1 )/(ri , ri )
pi+1 = ri+1 + i+1 pi
where i = 0, 1, 2, . . . K denotes the iteration number.

The product (R + D)x can be decomposed into two

products: Rx + Dx. The first product can be efficiently
computed using the Fast Fourier Transform (FFT), the
second product involves only 2M operations (M products
plus M additions) and does not substantially increase the
computational cost of the inversion.
a

Sacchi, M.D., and Porsani, M.J., 1999, Fast high resolution Radon transform:
69th Annual Intern. Mtg. Soc. Expl. Geophys., Expanded Abstracts, 1657-1660.

M.D.Sacchi / STMSSP

127

Example
1. Lev: Classical least squares parabolic Radon transform
implemented via the Levinson recursion (valid for a
constant damping).
2. Chol: High resolution Radon transform implemented via
the Cholesky decomposition.
3. CG+FFT: High resolution parabolic Radon transform
implemented via conjugate gradients plus matrix times
vector multiplication using the FFT.

N M

Lev

Chol

CG+FFT

128 128

256 256

42

12

Table 1: CPU times in seconds for the 3 algorithms tested

in this study. N denotes the number of traces and M the
number of q parameters, Origin 2000.

M.D.Sacchi / STMSSP

128

0.4

offset(m)
500

q(s)
1000

0.4

0.1

0.2

0.6

t(s)

tau(s)

0.6

0.8

0.8

1.0

1.0

CMP gather

tau-q (Lev.)

q(s)
0.4

0.1

q(s)
0.2

0.4

0.1

0.2

tau(s)

0.6

tau(s)

0.6

0.8

0.8

1.0

1.0

tau-q (Chol.)

tau-q (CG+FFT)

Figure 53:

A synthetic CMP gather composed of 4 parabolic events is used to

test 3 different algorithms to compute the Radon transform. Lev. indicates the
classical solution using least squares with a constant damping term; the Levinson
algorithm is used to invert the resulting Toeplitz form. Chol. indicates the high
resolution solution using non-constant damping (8)), this solution is computed
by means of the Cholesky decomposition. CG+FFT indicates the proposed fast
algorithm to compute the high resolution Radon transform. In this example the
size of the Radon operator is 256 256.

M.D.Sacchi / STMSSP

129

LS versus HR: Comparison

(a)

(b)
0

0.2

0.2

0.4

0.4

t (s)

(s)

0.6

0.6

0.8

0.8

0.01

0.02 0.03
h (km)

0.04

0.05

0.05
q (s)

(c)

0.1

(d)

0.2

0.2

0.4

0.4
t (s)

t (s)

0.6

0.6

0.8

0.8

0.01

0.02 0.03
h (km)

0.04

0.05

0.01

0.02 0.03
h (km)

0.04

0.05

Figure 54:

PRT synthetic example. (a) Events in t x. (b) q panel

computed using the LS parabolic RT. (c) and (d) Separated waveforms using the
q panel.
(a)

(b)
0

0.2

0.2

0.4

0.4

t (s)

(s)

0.6

0.6

0.8

0.8

0.01

0.02 0.03
h (km)

0.04

0.05

0.05
q (s)

(c)

0.1

(d)

0.2

0.2

0.4

0.4
t (s)

t (s)

0.6

0.6

0.8

0.8

0.01

0.02 0.03
h (km)

0.04

0.05

0.01

0.02 0.03
h (km)

0.04

0.05

Figure 55:

High resolution parabolic RT synthetic example. (a) Events in

t x. (b) Radon panel computed using the high resolution parabolic RT and (d)
Separated waveforms using the q panel (to be compared with Fig. 54).

M.D.Sacchi / STMSSP

130

8.8

Determination of weights for the HR

Radon Tranform

Non-linear problem: D = D(m)

(R + D)m = madj .
If the Cauchy regularization is adopted: Dii =
Classical Solution: IRLS

(1+|mi |2 / 2 )

For all freqs f solve:

(R + Dk1 )mk = madj .

Diik1 =
|2 / 2 )
(1 + |mk1
i
end
Non-iterative solution: k > f

For all freqs f solve:

(R + Df 1 )mf = madj .

Diif 1 =
(1 + |mfi 1 |2 / 2 )
end
with f 1 f f
a

Sacchi, M.D., and Ulrych, T.J., 1995, High resolution velocity gathers and
offset space reconstruction: Geophysics, 60, 1169-1177.
b
Herrmann, P., Mojesky, T., Magesan, M. and Hugonnet, P., 2000, De-aliased,
high-resolution Radon transforms: 70th Ann. Internat. Mtg: Soc. of Expl.
Geophys., 1953-1956.

M.D.Sacchi / STMSSP

131

Two passes solution:

i) Use the LS algorithm to estimate m
ii) Compute D from m for all freqs.
Diif =

(1 + |mfi |2 / 2 )

iii) For all freqs f solve:

(R + Df )mf = madj .
end

Remarks:
First pass with Levinsons recursion , second with
CG+FFT.
Does not give the highest resolution but attenuates
quite well alias.

M.D.Sacchi / STMSSP

132

Alias artifacts
0

Max Offset = 1910m

1

Residual Moveout, q (Seconds)

2

Residual Moveout, q (Seconds)

2

20

40

Frequency (Hz)

t (Seconds)

tau (Seconds)

60

80
5

100
6

120

(a)

(b)

(c)

Figure 56:

a) CMP gather. b) Least squares (Low res.) Radon Transform. c)

f q spectrum of the data. Notice that spite of the aliasing artifacts corrupting
the Radon panel, the f q spectrum clearly shows the prescence of 4 well-defined
modes.

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133

Max Offset = 1910m

1

3
t (Seconds)

t (Seconds)

t (Seconds)

(a)

(b)

(c)

Figure 57:

a) Data. b) Multiples estimated from the Radon panel in Fig. 56.

c) Primaries estimated via substraction of the multiple model from the original
data.

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Max Offset = 1910m

1

Residual Moveout, q (Seconds)

2

Residual Moveout, q (Seconds)

2

20

40

Frequency (Hz)

t (Seconds)

tau (Seconds)

60

80
5

100

120

(a)

(b)

(c)

Figure 58:

a) CMP gather. b) High resolution Radon Transform. c) f q

spectrum of the data. Notice that spite of the aliasing artifacts corrupting the
Radon panel, the f q spectrum clearly shows the prescence of 4 well-defined
modes. De-aliasing was achieved by the two passes algorithm.

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Max Offset = 1910m

1

3
t (Seconds)

t (Seconds)

t (Seconds)

(a)

(b)

(c)

Figure 59:

a) Data. b) Multiples estimated from the Radon panel in Fig. 58.

c) Primaries estimated via substraction of the multiple model from the original
data.

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Resolution enhancement: Impulse response

2.0

60

65

70

75

80

2.0

2.5

2.5

3.0

3.0

60

65

70

75

80

Figure 60: Left: Least squares. Right: High resolution, two

passes algorithm.

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8.9

Hyperbolic Radon Transform

Parabolic Radon Transform operates on NMO

corrected gathers (Bad at far offsets)
Hyperbolic Radon Transform operates directly on
CMP gathers
Hyperbolic Radon Transform are used to compute Time
Variant Velocity Gathers (Velocity Gathers or Velocity
Stacks).
d(t, h) =

m( =

t2 h2 /v 2 , v) dv ,

d(t, h) : CMP gather

m(, v) : Velocity gather
Remarks:
The operator is time variant, as a consequence there
are no frequency domain algorithms.
The operator can be packed in a matrix or applied in
the flight

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138

Discrete case
d, m: data and velocity gather in lexicographic
arrangement
d = Lm .
size of d: nt nh
size of m: n nv
size of the operator L: (nt nh) (n nv)
The problem
M inimize { = ||Lm d||22 }

m
= (LT L)1 LT d = (LT L)1 m0 ,
Question How do we invert (LT L) to obtain the Velocity
gather?
Answer Direct inversion: NO WAY
Example: nv = nh = 48, nt = n = 1000 then LT L is
24000 24000
Solution Use semi-iterative technique for large sparse
operators (i.e., Conjugate Gradients)

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Hyperbolic Radon Transform: CG solution

To apply CG we need to know how to do the following
two operations:
Lx Apply L to a vector in model space
LT y Apply LT to a vector in data space
Remark: L is never stored as a matrix; we treat it as an
operator

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The Algorithm (CGLS)

To solve the problem ||Lx y||2 with an initial solution
x0 , we use the following algorithm:
Set initial values: r = y L x0 , g = LT r , s = g
0 = ||g||2
for i = 1:IT ERM AX
ss = Ls

do it

= ||ss||2
= i1 /
x = x+s
r = r ss
g = LT r

undo it

i = ||g||2
= i /i1
s=g+s
end

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Hyperbolic Radon Transform: Example

0

200

400

Offset (m)
600

800

1000

1000
0

0.4

0.4

V (m/s)
2000
2500

3000

Time (s)

0.2

Time (s)

0.2

1500

0.6

0.6

0.8

0.8

(a)

1000
0

1500

V (m/s)
2000
2500

(b)

3000

0.4

0.4

400

Offset (m)
600

800

1000

Time (s)

0.2

Time (s)

0.2

200

0.6

0.6

0.8

0.8

(c)

(d)

Figure 61:

(a) Synthetic data. (b) Velocity gather obtained using the adjoint
operator. (c) Velocity gather computed using the least-squares inversion. (d)
Recovered data (primary) obtained after the de-multiple process.

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Artifacts
1000
0

1500

V (m/s)
2000
2500

3000

Near Offset Artifacts

0.2

Alias

Time (s)

0.4

0.6
Far Offset Artifacts

0.8

(c)

Figure 62: Velocity gather after clipping

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143

Hyperbolic Radon Transfrom: The operator

CGLS needs to call the following operator - Matrices are
not needed!!
. . . . . . . .
! c is a flag c=a : adjoint operator
!
c=f : forward operator
! strange stuff is for linear interpolation!!
if(c.eq.a) call clean(m,nv,nt)
if(c.eq.f) call clean(d,nh,nt)

! initialize m with zeros

! initialize d with zeros

do ih =1,nh
do iv=1,nv
do itau=1,nt
ttt=(itau-1)*dt
time=sqrt(ttt**2+(h(ih)/v(iv))**2)
it1 = int(time/dt)
a = time/dt - float(it1)
!Coeff. of the linear interp.
it2 = it1 + 1
if(it1.lt.nt.and.it1.ge.1)
if(c.eq.a) m(iv,itau) =
if(c.eq.f) d(ih,it1) =
if(c.eq.f) d(ih,it2) =
endif
enddo
enddo
enddo

then
m(iv,itau)+(1.-a)*d(ih,it1)+a*d(ih,it2)
d(ih,it1)+(1.-a)*m(iv,itau)
d(ih,it2)+
a *m(iv,itau)

! end offset loop

! end velocity loop
! end tau loop

return
end

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144

High Resolution Hyperbolic Radon Transf.

a b

We can contruct a solution m that consists on a few

isolated spikes in velocity space.
Minimize
J = ||Lm d||22 +

ln(1 + m2k /b)

LT L m L T d + Q m = 0
where Q is a diagonal matrix with elements given by
Qi =

2
b + m2i

It is clear that the system needs to be solved in an

iterative manner (Q depends on the unknown model m).
We can rewrite our solution as follows:
mk = (LT L + Qk1 )1 LT d .
a

Thorson, J. R., and Claerbout, J. F., 1985, Velocity-stack and slant stack
stochastic inversion: Geophysics, 50, 2727-2741.
b
Trad D., Ulrych J. and Sacchi M.D., 2003, Latest view of sparse Radon
transforms: Geophysics, 68, 386-399

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Example: data set from the Gulf of Mexico

-1.4

Offset (ft)
-1.2

x10 4
-1.0

Time (s)

Figure 63: CMP gather # 1000 from a data set from the Gulf
of Mexico.

M.D.Sacchi / STMSSP

146

LS Hyperbolic Radon Transf.

4000

Velocity (ft/s)
6000

8000

Time (s)

Figure 64: Velocity panel obtained by inversion of the Hyperbolic Radon transform using least-squares. CMP gather # 1000
from a data set from the Gulf of Mexico.

M.D.Sacchi / STMSSP

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High Resolution Hyperbolic Radon Transf.

4000

Velocity (ft/s)
6000

8000

Time (s)

Figure 65: Velocity panel obtained by inversion of the Hyperbolic Radon transform using sparse inversion. CMP gather #
1000 from a data set from the Gulf of Mexico.

M.D.Sacchi / STMSSP

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Figure 66: Gulf of Mexico data set. Stack section before Demultiple.

M.D.Sacchi / STMSSP

149

Figure 67: Gulf of Mexico data set. Stack section after multiple
removal.

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150

Sparseness in time v and the wavelet

You might need to solve
LW m = d
Sparseness in time is not modelled by the operator L,
therefore, W (the wavelet) is needed to make the
algorithm more stable.
A small modification to the CGLS code is all that it is
needed:
LW = Convolve with wavelet, then model
W T LT = Apply the adjoint, then cross-correlate with the
wavelet

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151

Fourier Interpolation

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152

9.1

Interpolation

Iterpolation in Shot-Rec. coordinates (or Midpoint-Offset)

using the NU-DFT (NU: Non-uniform)
2D NU-DFT Interpolation:
1
1 N
X
1 MX
U (ks (m), kr (n), )ejks (m)xs ejkr (n)xr ,
u(xs , xr , ) =
M N m=0 n=0
(39)
where xs and xr are the spatial variables along source and
receiver coordinates, ks and kr are the corresponding
wave-numbers. In matrix form:

u = AU

Amn =

1 jks (m)xs jkr (n)xr

,
e
e
MN

(40)

(41)

u and U denote the known data and unknown coefficients

of the DFT, respectively
J =k AU u k22 + k U k22

U = (AT A + I)1 AT u,

M.D.Sacchi / STMSSP

153

Band-Limited Signal Interpolation

Solutions like
U = (AT A + DI)1 AT u,

(42)

can include information about the spectral support of the

signal. The diagonal matrix D can be computed to
constraint wavenumbers not needed to reproduce the data.
Note the similarity with the High Res. Radon transform
where we exclude q-modes to achieve high resolution.

ks

k s,r

k r =f/Vmin
kr
fmax
f

k s =-f/Vmin

Figure 68: Region of support for Fourier interpolation.

a

Duijndam A.J.W., Schonewille M., and Hindriks K., 1999,Reconstruction of

seismic signals, irregularly sampled along on spatial coordinate: Geophysics, 64,
524-538.

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154

Should we use sparse inversion in Fourier Interpolation ?

a b

Here we have to be carefull. Fourier spectra is not

necesarilly sparse like the q-spectra in the parabolic
Radon transform.
There are, however, cases where the sparse
assumption is valid. See the next example:
0.2

0.4

Time [s]

0.6

0.8

1.2

1.4

500

550

600

650
700
Depth [m]

750

800

850

Figure 69: Vertical seismic profile.

a

Sacchi, M.D., and Ulrych, T.J, 1996, Estimation of the discrete Fourier transform: a linear inversion approach: Geophysics, 61, 4, 1128-1136.
b
Sacchi, M.D., Ulrych, T.J, and Walker, C., 1998, Interpolation and extrapolation using a high resolution discrete Fourier transform: IEEE Trans. on Signal
Processing, 46, No. 1, 31-38.

M.D.Sacchi / STMSSP

155

Normalized frequency

0.05

10

0.1

20

0.15

30

0.2

40

0.25

50

0.3

60

0.35

70

0.4

80

0.45

90

0.5
0.5

100
0.4

0.3

0.2

0.1
0
0.1
Normalized wavenumber

0.2

0.3

0.4

0.5

Db

Figure 70: VSP data: 2D periodogram or classical Fourier

spectrum.

Normalized frequency

0.05

10

0.1

20

0.15

30

0.2

40

0.25

50

0.3

60

0.35

70

0.4

80

0.45

90

0.5
0.5

100
0.4

0.3

0.2

0.1
0
0.1
Normalized wavenumber

0.2

0.3

0.4

0.5

Db

Figure 71: VSP data: 2D high resolution spectrum using

sparse inversion (Cauchy norm)

M.D.Sacchi / STMSSP

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Sparse Reconstruction of Fourier Spectra

(a)
2
1
0
1
2
0

20

40

60

80

100
(b)

120

140

160

180

200

20

40

60

80

100
(c)

120

140

160

180

200

20

40

60

80

100
Samples

120

140

160

180

200

2
1
0
1
2

2
1
0
1
2

Figure 72: (a) Input data (b) Reconstructed data (c) Error
(a)

Normalized Power

1
0.8
0.6
0.4
0.2
0

0.05

0.1

0.15

0.2
0.25
0.3
Normalized Frequency

0.35

0.4

0.45

0.5

0.35

0.4

0.45

0.5

(b)

Normalized Power

1
0.8
0.6
0.4
0.2
0

0.05

0.1

0.15

0.2
0.25
0.3
Normalized Frequency

Figure 73: (a) High resolution spectrum (b) Periodogram

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157

9.2

Minimum weighted norm

interpolation

This is an alternative to sparseness and band-limited

constraints:a
Minimize
J = ||u

AU ||22

Uk Uk
+
2
kK Pk
X

where Pk is an inital estimate of the power spectrum of

the data, and K the wavenumber region where Pk 6= 0.
The problem is solved with the CG method and efficiency
can be gainned by using the FFT (after data binning)
rather the NU-DFT, in that case:
Minimize J = ||W (u

AU )||22

Uk Uk
+
2
kK Pk
X

W is used to penalize non-recorded data. Remark:

Pk needed at frequency slice f is computed from the
solution at frequency f f (see, non-iterative
Radon Transform)
Helps to mitigate alias since the weigths are derived
from the interpolation of the non-aliased low freq.
portion of the data.
a

Liu B., and Sacchi M.D., 2003, 2-D/3-D seismic wavefield reconstruction for
AVA imaging: 73rd Annual Mtg. Soc. Expl. Geophys., Expanded Abstracts, 4
pages, CDROM.

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MWNI - Examples
(a)

3.0

-4800

-4600

-4400

-4200

-4000

Offset (m)
-3800 -3600

-3400

-3200

-3000

-2800

-4000

Offset (m)
-3800 -3600

-3400

-3200

-3000

-2800

3.2

Time (s)

3.4

3.6

3.8

4.0

4.2

4.4

(b)

3.0

-4800

-4600

-4400

-4200

3.2

Time (s)

3.4

3.6

3.8

4.0

4.2

4.4

Figure 74:

M.D.Sacchi / STMSSP

Before and After MWNI (Gap filling + resampling)

159

Marmousi s-r interpolation

3600

Source positions (m)

3500

3400

3300

3200

3100

Known
Unknown

3000
500

1000

1500
2000
Receiver positions (m)

2500

3000

3500

Figure 75:

Source and receiver position map where the x-mark indicates the
position of the available traces; the dots indicate the positions to be interpolated.
(a)

Trace Number
100
200

(b)

Trace Number
100
200

Time (s)

Time (s)

(c)

Trace Number
100
200

Time (s)

Figure 76:

(a) Decimated shot gathers at 3075 m, 3100 m and 3125 m. (b)

The reconstructed shot gathers. (c) The reconstruction error.

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