Gabor Deconvolution
Gabor Deconvolution
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Gabor deconvolution
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Gabor deconvolution
ABSTRACT
The Gabor transform decomposes a 1-D temporal signal onto a time-frequency
plane. Temporal localization is accomplished by windowing the signal with a
Gaussian analysis window translated to any particular time. The Fourier transform of
the windowed signal provides frequency information for the time of the window
centre. The inverse Gabor transform is an integration over the time-frequency plane
using an optional synthesis window. The discrete Gabor transform has an approximate
realization based on the fact that a shifted suite of Gaussians can sum to unity. Gabor
filters are nonstationary filters that can be implemented by multiplying the Gabor
transform of a signal by a time-frequency filter specification. Extreme cases of Gabor
filters are shown to correspond to normal or adjoint pseudodifferential operators.
Gabor deconvolution is developed for exploration seismology based on a
nonstationary convolutional model of a seismic trace. This model predicts that the
Gabor transform of a seismic signal decomposes as the Fourier spectrum of the
source signature times the symbol of a forward Q-filter times the Gabor transform of
the reflectivity. Gabor deconvolution requires the spectral factorization of the Gabor
transform of the seismic signal into a reflectivity part times a propagating wavelet
part. This factorization can be done on magnitude spectra with phase being
determined from a minimum-phase condition. Testing on synthetic data shows the
nonstationary Gabor deconvolution is an effective tool that combines the processes of
stationary deconvolution, inverse Q-filtering, and gain adjustment.
INTRODUCTION
The deconvolution of seismic data is a fundamental part of the construction of
high-resolution seismic images. Deconvolution addresses the challenging task of
separating a seismic trace (i.e. the recording of a single sensor) into two components,
both unknown, representing the seismic waveform and the reflectivity (the latter is a
time series of reflection coefficients). That this can be done at all is remarkable and
the first solutions were the achievement of the mathematician Norbert Wiener and his
students though the application was radar not seismology. Enders Robinson and Sven
Treitel (1967) were pioneers in bringing the technique to geophysics.
This separation of a seismic trace into two parts exploits reasonable assumptions
about the nature of the seismic waveform and reflectivity and another assumption
about the seismic trace itself. The seismic waveform is postulated to be a temporally
short pulse, implying a smooth Fourier transform, the amplitude and phase spectra of
which are linked by the minimum-phase condition. The latter is a consequence of
linearity and causality and says that the phase spectrum can be calculated as the
Hilbert transform of the logarithm of the amplitude spectrum (e.g. Claerbout, 1976
and other basic texts). The reflectivity is postulated to be a random time-series whose
autocorrelation therefore is nonzero only at zero lag and whose power spectrum is a
constant at all frequencies.
The final assumption is that the seismic signal is assumed to have the property of
stationarity. Though intuitively rather obvious, stationarity is difficult to define in a
truly useful way. A common definition requires that the seismic trace, s (t ) , be
regarded as one realization of a random process. If many other realizations of the
same random process are available (perhaps from geophones placed in parallel
quantum universes) then the expectation value at any time is defined as the average
taken over the entire, possibly infinite, ensemble of realizations at that time. If this
expectation value is found to be independent of time, then the signal is said to be
stationary. Of course, such ensembles are not often encountered in practice so an
alternative test for stationarity is useful. Commonly, the ensemble average is replaced
by a time average of the signal multiplied by some localizing window. (Signals with
the property that ensemble averages and time averages are equal are called ergodic.)
Practically then, a signal is stationary if such time averages are independent of time.
Among the many drawbacks of this definition is the fact that it depends upon the
localizing window. A signal may be stationary with respect to one window and
nonstationary with respect to another.
A more common test of stationarity is to require that the power spectrum of the
localized signal is independent of the time of localization. This is also a difficult
criterion to apply because real signals are finite in length and, if such a signal is
stationary by the ensemble average test, it will generally still show variations in local
power spectra. Therefore, some level of power fluctuation is generally tolerated in
signals that are called stationary.
physical grounds. We then derive an expression for the Gabor transform of this trace
model. This expression guides our Gabor deconvolution algorithm. We argue that the
Gabor transform of a seismic signal can be processed in such a way that the
contribution from reflectivity is suppressed to produce an estimate of the propagating
wavelet. When the unaltered Gabor transform of the signal is divided by the estimate
of the propagating wavelet, the result estimates the Gabor transform of the
reflectivity. We close our paper with a numerical example.
The Gabor transform can be developed for a continuous or discrete set of window
positions. The continuous Gabor transform follows easily from the theory of the
Fourier transform. The signal can obviously be reconstructed from its Gabor
transform since it can be constructed from a single local Fourier spectrum by inverse
Fourier transformation and demodulation. This makes it apparent that the Gabor
transform is a highly redundant representation of a signal. Nevertheless, the inverse of
the continuous Gabor transform is constructed from the entire Gabor spectrum by
superposition of the local spectra after inverse Fourier transformation without
demodulation.
The discrete Gabor transform is much more difficult to develop. The redundancy
of the continuous Gabor transform suggests a discrete set of window positions should
suffice; however, this was not proven until Bastiaans (1980). The theory of the
discrete Gabor transform that has developed since then is, like the wavelet transform,
built upon the concept of an expansion frame. Similar to an orthogonal basis, an
expansion frame allows an arbitrary function to be written as a scalar-weighted
superposition of elementary functions. However, while the scalar weights for the
orthogonal basis are formed by the inner product of the orthogonal basis function
with the signal, for the frame expansion the inner product is taken not against the
frame expansion function but rather against the corresponding function from the dual
frame. The calculation of this dual frame is a major technical challenge. For the
Gabor expansion, the expansion frame may be the Gaussian-modulated complex
exponentials; although the theory generalizes to windows other than Gaussians. The
dual frame that is required to compute the expansion coefficients is a complex
exponential modulated by the dual window. A complete overview of the theory of the
discrete Gabor transform is found in Feichtinger and Strohmer (1998).
In this paper, we present the theory of the continuous Gabor transform and only an
approximate theory for the discrete transform. Our approximate discrete transform
does not exactly recreate the original signal, but we show that the loss can be made as
small as desired.
where γ (t ) is the Gabor synthesis window. The analysis and synthesis windows must
satisfy the condition
∞
∫ g (t )γ (t ) dt = 1 . (3)
−∞
∞ ∞ ∞ ∞ ∞
∫ Vg s (τ , f )γ (t − τ ) e dfdτ = ∫ ∫ ∫ s (t ′) g (t ′ − τ )e dt ′γ (t − τ ) e 2π if t dfdτ
2π if t −2π if t ′
∫ .
−∞ −∞ −∞ −∞ −∞
(4)
Next we assume that the integrands are sufficiently well behaved to allow the order of
integration to be exchanged and perform the f integral first. Since f dependence occurs
only in the complex exponentials, this becomes
2π i[t −t ′] f
∫e df = δ (t − t ′ )
f integral = (5)
∞ ∞ ∞ ∞
Now consider the τ integral. Since τ dependence occurs only in the window
functions, this integral becomes
∞ ∞
∫ g (t ′ − τ )γ (t − τ ) dτ = ∫ g (u )γ (t − t ′ + u ) du = Λ (t − t ') (7)
−∞ −∞
where Λ (t − t ′ ) simply symbolizes the result of the integration; but we note that,
according to equation (3) Λ ( 0 ) = 1 . Thus we write equation (6) as
∞ ∞ ∞
The last step, that completes the proof, follows from the defining property of the
Dirac delta function and the fact that Λ ( 0 ) = 1 .
In closing this section, we note several distinctive things about the Gabor transform
pair. First, the forward transform maps a one-dimensional signal to a two-dimensional
spectrum whose properties are determined by the nature of the signal and of the
analysis window. Second, the inverse transform is a two-dimensional operation that
collapses the spectrum into a one-dimensional signal. As with the Fourier transform,
there is an integration over frequency; but additionally, there is an integration over
window position. Finally, given an analysis window, the synthesis window is not
uniquely determined. In particular, the windows g ≡ 1 or γ ≡ 1 are always possible
provided that the nontrivial window is suitably normalized. The Gabor transform
reduces to the Fourier transform if both the analysis and synthesis windows are unity.
sample interval. We can regard the DFT as an end member case of the DGT with a
single window position and an analysis window g ≡ 1 . In this case, ∆τ = N ∆t , that is
the spacing of the temporal window position is equal to the length of the signal. Then,
for the DFT, ∆ f ∆τ = ( N ∆t ) N ∆t = 1 . If we now choose a smaller ∆τ , it would be
−1
S g s (t ) = ∑ s, g m , n g m , n ( t )
m ,n∈Z . (11)
For example, the DFT, using the orthonormal basis hn (t ) = e2π itn∆ f , recovers a signal
exactly from the expansion s (t ) = ∑ s, hn hn (t ) . Since the g m,n form a frame and
n∈Z
not an orthonormal basis, for the DGT, the recovery of the signal from equation (11)
requires the inversion of the Gabor frame operator. That is
where γ m,n (t ) = S g−1 g m,n (t ) is called the dual Gabor frame. So, if the Gabor frame
operator can be inverted then the inverse DGT (equation 12) can exactly recover the
signal.
∑ g (t − k ∆τ ) ≈ 1
k∈" (13)
∆τ −[t −k ∆τ ]2 T −2
g (t − k ∆τ ) = e
where T π (14)
with T being the Gaussian (half) width. More precisely, in Appendix A, we show that
Under the assumption that a set of Gaussian windows has been chosen that makes
the sum in equation (13) as close to unity as desired, we decompose a seismic signal
into Gaussian slices as
s (t ) = s (t ) ∑ g (t − k ∆τ ) = ∑ s (t ) g (t − k ∆τ ) = ∑ sk (t )
k∈" k∈" k∈" (16)
where the Gaussian slice is defined to be sk (t ) = s (t ) g (t − k ∆τ ) . Next we apply a
forward Fourier transform
∞
sˆ ( f ) = ∑ ∫ sk (t ) e−2π if t dt = ∑ s$k ( f )
k∈" −∞ k∈" (17)
where ŝ is the Fourier transform of s and
∞
s$k ( f ) ≡ ∫ sk (t ) e
−2π if t
dt
−∞ (18)
is our approximate Gabor transform. As written, it is discrete in the window position
coordinate, as indexed by k, but continuous in frequency f. Of course, in a computer
implementation, we would replace the integral Fourier transform with the DFT as
The recovery of the original signal from s$k ( f ) follows by simply taking the
inverse Fourier transform of equation (17)
∞
s (t ) = ∫ k∑ s$k ( f ) e 2π if t
df
−∞ ∈" . (21)
Thus s$k ( f ) is simply summed over k and then inverse Fourier transformed. In
comparison with the theory of the continuous Gabor transform given previously, we
have presented a Gabor methodology that uses a special discrete set of analysis
windows, such that equation (13) is satisfied, and uses a synthesis window of γ ≡ 1 .
We have avoided the difficult issue of inverting the Gabor frame operator by using a
special set of windows and accepting an approximate transform. In the next section,
in the context of Gabor filtering, we will introduce a non-trivial synthesis window.
As a test of this approximate Gabor transform, a seismic signal was forward and
inverse Gabor transformed and is compared to the original signal in Figure 3. The
difference plot shows that the most significant differences occur at the ends of the
signal. This is due to the fact that the Gaussian summation of equation (13) must be
conducted over a finite line segment and has significant departures from unity near
the ends of the segment (Figure 1). The inherent error in the transform can be reduced
by a simple normalization procedure in which the Gabor transform is divided by the
summation curve. That is, let h (t ) = ∑ k∈" g (t − k ∆τ ) be the actual summation curve
employed in a Gabor transform. The Gabor transform of equation (18) can be
normalized by
∫ sk ( t ) e
−2π if t
dt
s$k ( f ) ≡ −∞
h ( k ∆τ ) . (18a)
Figure (4) is a recreation of the test of Figure (3) using the normalized transform of
equation (18a). The end effects are clearly reduced, though not eliminated, by the
normalization.
The Gabor transform of a signal is not unique and depends strongly on the
Gaussian half-width T and the window increment ∆τ . Figure 5 shows eight versions
of the Gabor transform of the nonstationary signal of Figures 3 and 4. The larger T is,
the greater is the frequency resolution but at the expense of temporal resolution. As T
decreases, frequency resolution is lost but temporal resolution increases. When
temporal resolution is low, a small ∆τ results in a highly redundant transform;
conversely, when ∆τ is large and temporal resolution is high, the transform can
actually be undersampled in τ. This tradeoff in time-frequency resolution is often
called the uncertainty principle and has been known since at least the early days of
Quantum mechanics. The Gaussian window is unique in that, for a given window
width, it is known to give the best possible simultaneous resolution in both time and
frequency compared to all other possible windows.
GABOR FILTERING
A filter can be implemented using the Gabor transform by simply modifying the
Gabor spectrum of a signal prior to the reconstruction. We describe the filter by its
nonstationary transfer function, α (τ , f ) , and illustrate its application to a signal s(t)
using the continuous Gabor transform as
∞ ∞
sα (t ) = ∫ ∫ α (τ , f )Vg s (τ , f )γ (t − τ ) e
2π if t
dfdτ
−∞ −∞ . (22)
We now examine two limiting forms of this general Gabor filter. First consider the
case when γ (t − τ ) ≡ 1 . In this case, we refer to a Gabor filter of type 1, or an analysis
window filter, and write
∞ ∞
sα 1 (t ) = ∫ ∫ α (τ , f )Vg s (τ , f ) e
2π if t
dfdτ
−∞ −∞ . (23)
This filter is achieved by forming the product of the Gabor transform with the
nonstationary transfer function, averaging over τ, and inverse Fourier transforming.
Second, consider the case when g (t − τ ) ≡ 1 and note that in this case the Gabor
transform reduces to the Fourier transform. We call this a Gabor filter of type 2, or a
synthesis window filter, and it is given by
∞ ∞
sα (t ) = ∫ ∫ α (τ , f ) sˆ ( f )γ (t − τ ) e
2π if t
dfdτ
−∞ −∞ . (24)
This filter begins with the product of the Fourier transformed signal and the
nonstationary transfer function, then applies an inverse Fourier transform, and finally
applies the synthesis window and averages over τ. In the limit as γ (t − τ ) → δ (t − τ ) ,
equation (24) becomes a standard-form Kohn-Nirenberg pseudodifferential operator.
Similarly, in the limit g (t − τ ) → δ (t − τ ) , equation (23) becomes an adjoint-form (or
dual-form) Kohn-Nirenberg pseudodifferential operator. Thus, the general Gabor
filter of equation (22) becomes a standard form Kohn-Nirenberg pseudodifferential
operator if g ≡ 1 and γ ≡ δ while it reduces to an adjoint form if g ≡ δ and γ ≡ 1 . In
this sense, Gabor filters generalize pseudodifferential operators.
We now present formulae for the two limiting forms of Gabor filters based on our
approximate semi-discrete Gabor transform. For this purpose, the nonstationary
transfer function must be specified in a semi-discrete fashion as α k ( f ) . The analysis
window filter is then
∞
sα 1 (t ) = ∫ k∑
∈"
α k ( f ) s$k ( f )e2π if t df
(25)
−∞
∞
s (t ) = ( r • w )(t ) ≡ ∫ w (t − τ ) r (τ ) dτ
−∞ (28)
where w(t) is the seismic wavelet and r(t) is the reflectivity. The term reflectivity
refers to a time series the samples of which are the normal incidence P-wave
reflection coefficients of subsurface reflectors positioned at the normal incidence
traveltimes to each reflector. This model is often justified on physical grounds using
the Green’s function approach to solving partial differential equations. It is argued
that, if the impulse response (i.e. solution for a Dirac delta-function source δ (t ) ) of
the wave equation is known, then the solution for a temporally extended source is
given by the convolution of the source waveform (wavelet) with the impulse
response. However, this argument only implies equation (28) if r(t) is interpreted to
be the impulse response, a much more complex time series than the reflectivity. In
addition to normal incidence reflection coefficients representing primary reflections,
the impulse response also contains all possible multiply-scattered waves. The
viewpoint that r(t) is the impulse response and w(t) is the source signature leads to
gapped predictive deconvolution algorithms such as described in Peacock and Treitel
(1969) that are intended to estimate and subtract the multiple sequence. In this case,
the goal of deconvolution is not only to estimate and remove the source signature but
also to somehow transform the impulse response into the simpler reflectivity. An
alternative viewpoint is to argue that the multiples can be considered as part of the
seismic wavelet so that r(t) can still be reflectivity. Then the deconvolution technique
seeks to estimate r(t) by removing w(t). Regardless of which perspective, impulse
response or reflectivity, is taken, the goal of a deconvolution process is to estimate
reflectivity not impulse response.
On another level, even if the medium is not dissipative the effective seismic
wavelet still evolves as it propagates. This is due to the effect of many local interbed
multiples that build up the tail of the wavelet as was first explained by O’Doherty and
Anstey (1971). Called stratigraphic filtering, the effect of these multiples is to
attenuate high frequencies in a manner similar to dissipation though the applicability
of the minimum-phase condition is in doubt. In practice, it is essentially impossible to
separate the effects of stratigraphic filtering from those of dissipation. Instead, a
single effective Q is estimated that is assumed to roughly account for both effects.
where r(τ) is explicitly the reflectivity and the constant-Q transfer function is
α Q (τ , f ) = e −π f τ / Q +iH (π f τ / Q )
(30)
where H denotes the Hilbert transform over f at constant τ. Equation (29) can be
understood as a nonstationary convolution by noting that the f integral can be written
as
∞
2π if [t −τ ]
aQ (τ , t − τ ) = ∫ α Q (τ , f ) e df
−∞ (31)
so that
∞
sQ = ∫ aQ (τ , t − τ ) r (τ ) dτ (32)
−∞
∞
sˆ ( f ) = wˆ ( f ) ∫ αQ (τ , f ) r (τ ) e
−2π if τ
dτ
−∞ (33)
where ŵ and ŝ are the Fourier transforms of the source signature and the
nonstationary seismic trace respectively. Equation (33) is our nonstationary trace
model proposed as a replacement for the stationary convolution model of equation
(28). We prefer to express it in the Fourier domain for simplicity.
Finally, since we apply the Gabor transform in our deconvolution method, the
Gabor transform of s(t), whose Fourier transform is given by equation (33), is desired.
In Appendix B, we derive an asymptotic result for this as
Vg s (τ , f ) ≈ wˆ ( f )α Q (τ , f )Vg r (τ , f )
. (34)
In words, the Gabor transform of our nonstationary trace is approximately equal to
the product of the Fourier transform of the source signature, the constant Q transfer
function, and the Gabor transform of the reflectivity. Since, for fixed τ, the Gabor
transform is just a Fourier transform, this is a convolutional model in some local
sense.
wˆ ( f )α Q (τ , f ) ≈ Vg s (τ , f ) eiϕ (τ , f )
sep
(35)
where the phase ϕ (τ , f ) is given by the Hilbert transform expression
∞ ln Vg s (τ , f ′ )
ϕ (τ , f ) = ∫ df ′
sep
−∞
f − f′ (36)
Vg s (τ , f )
Vg r (τ , f )est = e− ( )
iϕ τ , f
Vg s (τ , f ) (37)
sep
Vg s (τ , f ) − iϕ (τ , f )
Vg r (τ , f )est = e
Vg s (τ , f ) + µ Amax (38)
sep
An interesting variant of the Gabor deconvolution algorithm can be done using the
Burg spectral estimation (Claerbout 1976). Since the Burg method is often preferred
on small data segments, it has obvious appeal when short temporal windows are
desired. Our approach is to compute Vg s (τ , f ) using the approximate discrete Gabor
transform as discussed above and, simultaneously to compute a time-variant Burg
Figure 14 shows a Gabor spectrum of the synthetic signal of Figure 13. This was
computed with T = 0.1 s and ∆τ = 0.01 s. The Gabor spectrum reveals that the signal
contains the greatest power at early times and low frequencies. Next, Figure 15 is the
result of smoothing the Gabor spectrum of Figure 14 by convolving it with a boxcar
of dimensions 0.1 s by 20 Hz. This is taken to be an estimate of wˆ ( f ) α Q (τ , f ) .
Then, Figure 16 is the deconvolved Gabor spectrum computed using equation (38).
Since this is an estimate of the Gabor spectrum of the reflectivity, it is interesting to
compare it to the correct result shown in Figure 17. The Gabor algorithm has
achieved a very strong, broadband whitening but there are important differences in
amplitudes between the deconvolution result and the answer. In Figure 18, the final
deconvolved trace is shown in the time domain compared to the reflectivity. All of
the major reflectors have been resolved but there are differences in amplitude. These
amplitude differences are a consequence of the rather short temporal smoother that
was used in this case. Such short smoothers tend to give strongly whitened results but
also induce a kind of AGC (automatic gain correction).
CONCLUSIONS
The Gabor transform decomposes a signal onto a time-frequency plane by
windowing the signal with a sliding Gaussian and Fourier transforming. The Gaussian
window used in the forward transform is called the analysis window. The inverse
Gabor transform recreates the signal as a 2D integration over the time-frequency
plane using a synthesis window. The theory of the Gabor transform can be developed
for continuous or discrete window positions though the discrete theory is more
difficult to develop precisely. The discrete Gabor transform can be more easily
developed in an approximate way by using a special set of Gaussians whose width is
related to their spacing in a controlled way. The discrete Gabor transform is lossy (the
result of a forward and inverse transform does not exactly replicate the signal) but the
error is easily controlled to lie below any given tolerance.
ACKNOWLEDGEMENTS
We benefited from a series of valuable discussions with Jeff Grossman, Victor
Iliescu, Kris Vasudevan, and Rita Aggarwala. These discussions occurred in a weekly
seminar on Gabor Analysis, led by the POTSI project, which the authors have
participated in over the past spring and summer sessions at the University of Calgary.
(POTSI, or Pseudodifferential Operator Theory in Seismic Imaging, is a collaborative
research project between the University of Calgary departments of Mathematics and
Geophysics (supported by MITACS), Imperial Oil, CREWES, and NSERC.)
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∆τ −[t −k ∆τ ]2 T −2
g (t − k ∆τ ) = e
T π (A-1)
and we desire an expression for ∑ g (t − k ∆τ ) .
k
This sum can be written as the
convolution of an unshifted Gaussian with a Dirac comb
h (t ) ≡ ∑ g (t − k ∆τ ) = ( g • c )(t )
k∈" (A-2)
gˆ ( f ) = ∆τ e [ ]
2
−π fT
(A-3)
and
1 k
cˆ ( f ) =
∆τ
∑ δ f − ∆τ
k . (A-4)
Therefore, by the convolution theorem
ˆ ˆ ) = F −1 #δ f +
1 −[π fT ]2 −[π fT ] 1 −[π fT ]2
h = F −1 ( gc ( )
2
e + δ f e + δ f − e + #
∆τ ∆τ
(A-5)
where F −1 is the inverse Fourier transform. Using the delta function to evaluate the
transforms shows the central term is unity while the two neighbouring terms are
= 2 cos ( 2π t / ∆τ ) e [
− π T / ∆τ ]
2
(A-6)
Thus we conclude
h (t ) = 1 + 2 cos ( 2π t / ∆τ ) e [
− π T / ∆τ ]
2
+#
. (A-7)
Furthermore, because the higher order terms are integrated against delta functions
with higher frequency arguments (larger values of k / ∆τ ) they are exponentially
smaller with each value of k.
∞
sˆ ( f ) = wˆ ( f ) ∫ α (t , f ) r (t ) e
−2π if t
dt
−∞ (B-2)
where the hat denotes the Fourier transform. Thus
∞ ∞
s (t ) = ∫ ˆ
w ( f ) ∫ α (u , f ) r (u ) e
−2π if u
du e 2π if t df
−∞ −∞
2π if [ t −u ]
= ∫∫ wˆ ( f )α (u , f ) r (u ) e dfdu
(B-3)
This may seem a bit lacking in justification but is, in fact, quite well established. We
now want to show that a time-frequency decomposition of s (t ) , call it Ts (τ , ν ) , has
the approximate factorization: Ts (τ , ν ) ∼ wˆ (ν ) α (τ , ν ) Tr (τ , ν ) . Here Tr (τ , ν ) is the
time-frequency decomposition of the reflectivity. This will be demonstrated using
Gabor spectra as the time-frequency decomposition.
∞
Vg s (τ ,ν ) = ∫ s (t ) g (t − τ ) e
−2π iν t
dt
−∞ (B-4)
−2π it[ν − f ]
I = ∫ g (t − τ )e dt
. (B-6)
Let t ′ = t − τ and so
−2π iτ [ν − f ]
Vg s (τ ,ν ) = ∫∫ wˆ ( f )α (u, f ) r (u ) e−2π ifu e gˆ (ν − f ) dfdu
(B-8)
Now, let f ′ = ν − f so that
∂wˆ
Vg s (τ ,ν ) = ∫∫ wˆ (ν ) − (υ ) f ′ + …
∂ν
∂α −2π i[ν − f ′]u −2π iτ
α (u ,ν ) − ∂ν (u ,ν ) f ′ + … r (u ) e gˆ ( f ′ ) df ′du
f′
e
(B-10)
or
2π i[u −τ ] f ′
Vg s (τ ,ν ) = wˆ (ν ) ∫ α (u ,ν ) r (u ) e−2π iν u ∫ e gˆ ( f ′ )df ′ du + #
(B-11)
or
Vg s (τ ,ν ) = wˆ (ν ) ∫ α (u,ν ) r (u ) g (u − τ ) e−2π iν u du + #
(B-12)
let u ′ = τ − u
−2π iν [τ −u′]
Vg s (τ ,ν ) = wˆ (ν ) ∫ α (τ − u′,ν ) r (τ − u ′ ) g ( −u ′ ) e du′ + #
(B-13)
expand α (τ − u ′, ν ) in a Taylor series
−2π iν [τ −u′]
Vg s (τ ,ν ) = wˆ (ν )α (τ ,ν ) ∫ r (τ − u ′ ) g ( −u ′ ) e du′ + #
(B-14)
now, back to u = τ − u ′
Vg s (τ ,ν ) = wˆ (ν )α (τ ,ν ) ∫ r (u ) g (u − τ ) e−2π iν u du + #
(B-15)
or
Vg s (τ ,ν ) = wˆ (ν )α (τ ,ν )Vg r (τ ,ν ) +#
(B-16)
These last several manipulations seem almost circular but they amount to a
justification that α (u , v ) can be pulled out of the integral in equation (B-12) where it
becomes α (τ , v ) . This analysis shows the leading-order behaviour fits the
nonstationary model but so far provides no estimate of the next asymptotic term.
FIGURES
Summation curve
FIG. 1. The summation of a set of Gaussian windows is illustrated. The window half-width is
0.1 seconds and the window increment is 0.05 seconds.
0.2
0.15
0.1
0.05
0.025
0.01
FIG. 2. The summation curves for a suite of different Gaussian summations are shown for a
window half-width of .1 seconds. Each summation curve is labelled with its window
increment. When the window increment significantly exceeds the half-width, then the
summation is no longer close to unity.
Difference
Seismic signal
FIG. 3. The result of a forward and inverse Gabor transform using the approximate algorithm
(equations (18) and (21)) is compared with the original signal. The largest differences are
found at the ends of the signal where the Gaussian summation has the largest departure
from unity.
Difference
Seismic signal
FIG. 4. The comparison of Figure 3 is repeated but the forward Gabor transform uses the
normalized expression of equation (18a) instead of (18). The end effects are clearly reduced.
T ∆τ 0.01 S 0.04 S
0.8 S
0.4 S
0.2 S
0.1 S
FIG. 5. The Gabor transform of the signal of Figure 3 is shown for a variety of window widths
and increments. The highest frequency resolution and lowest temporal resolution occurs at
the left while the highest temporal resolution and lowest frequency resolution is on the right.
Nonstationary synthetic
Reflectivity
time (seconds)
FIG. 7. The magnitude of the Gabor transform of the nonstationary time series of Figure 1.
time (seconds)
frequency (Hz)
FIG. 8. The magnitude of the Gabor transform of the reflectivity of Figure 7.
time (seconds)
frequency (Hz)
time (seconds)
frequency (Hz)
FIG. 10. The pointwise product of the Gabor transform of the reflectivity (Figure 7) and the
forward-Q operator (Figure 8).
time (seconds)
frequency (Hz)
FIG. 11. A stationary wavelet represented on a time-frequency plane.
time (seconds)
frequency (Hz)
FIG. 12. The pointwise product of the Gabor transform of the reflectivity (Figure 8) with the
forward-Q operator (Figure 9) and with the wavelet (Figure 10). Comparison with Figure 7
shows that this is a good representation of the Gabor spectrum of the nonstationary
synthetic.
reflectivity
FIG 13. A pseudo-random reflectivity is show (bottom) and a nonstationary seismic signal
created from the reflectivity (top). The nonstationary signal has a minimum phase source
signature and a minimum-phase forward-Q filter applied.
FIG. 14. The magnitude of the Gabor transform of the nonstationary seismic signal of Figure
13 is shown. The Gaussian half-width was 0.1s and the increment between Gaussian
windows was 0.01s.
FIG. 15. The magnitude of the Gabor transform of Figure 14 after smoothing by convolution
with a 2-D boxcar of dimensions 0.1s by 20Hz.
FIG 16. The result of dividing the Gabor transform of Figure 14 by that of Figure 15 using the
procedure described in the text. This is an estimate of the Gabor transform of the reflectivity.
FIG. 17. This is the magnitude of the Gabor transform of the reflectivity signal shown in
Figure 13. It was computed with the same Gabor parameters as Figure 14. Figure 16 is an
estimate of this transform produced by Gabor deconvolution.
reflectivity
FIG 18. On top is a Gabor deconvolution of the nonstationary seismic signal of Figure 13. It is
the inverse Gabor transform of the Gabor spectrum of Figure 16. In comparison with the true
reflectivity, there is a general correspondence of major peaks.