Synchronized seismic signal acquisition method and device

Method and device intended for synchronized acquisition of seismic signals by one or more acquisition units suited for seismic signal digitizing, allowing to obtain, for each signal, a series y[n] of samples of these signals readjusted from a reference time on, from a first series x[n] of digitized samples of this seismic signal produced from any initial time prior to the reference time. The method essentially comprises detecting a synchronization signal indicative of this reference time (T.sub.R), measuring the effective time difference (D) between the reference time and the initial time, determining coefficients of a digital filter (F) suited to compensate for the fractional part (d) of the measured effective time difference, and applying this compensation digital filter to the first series of samples, which allows to obtain a series of digitized samples readjusted from the reference time. In order to accelerate determination of the filter coefficients depending on the difference D observed, the coefficients of a certain number of intermediate filters corresponding to determined fractions of the sampling interval are preferably precalculated. The method can be applied for seismic prospecting or monitoring, earthquake detection, etc.

FIELD OF THE INVENTION
 The present invention relates to a synchronized seismic signal acquisition
 method allowing resynchronization of seismic signal acquisition on an
 exterior event, and to a device for implementing it.
 The method according to the invention finds applications in many fields
 where various measured signals are to be sampled for example, by means of
 different acquisition chains, by imposing that the series of samples taken
 are substantially synchronous with an exterior event. This is notably the
 case in the field of seismic exploration where an initial time from which
 significant seismic signals are recorded is to be fixed for an acquisition
 system.
 The initial reference time generally selected is the time of triggering of
 a source of seismic waves. The waves emitted are propagated in the subsoil
 and received by seismic pickups distributed at the ground surface for
 example. The signals delivered by these pickups are transmitted to a
 central control and recording station, generally by means of acquisition
 devices distributed in the field. Each one of them is suited to amplify,
 filter, digitize and store all the signals picked up after each source
 triggering. The stored data are transmitted to a central station from each
 acquisition device at fixed time intervals (after each emission-reception
 cycle for example) or "with the stream", as soon as a transmission time
 interval is available. Seismic acquisition systems are for example
 described in patents FR-2,511,772 (U.S. Pat. No. -4,583,206) or
 FR-2,538,194 (U.S. Pat. No. -4,628,494).
 In each acquisition device, the seismic signals are applied to an
 acquisition chain. A conventional acquisition chain structure comprises a
 steady-gain preamplifier, a high-pass filter, an anti-aliasing low-pass
 filter and an analog-to-digital (ADC) converter. The converters deliver
 for example 24-bit numerical words. They are for example (sigma-delta
 type) oversampling analog-to-digital converters associated with digital
 filters (FIR).
 Oversampling converters produce numerical words of reduced format in
 relation to conventional converters, but with a much higher frequency. The
 normal dynamic range is restored by applying to the signals coming from
 the converter a digital filter referred to as decimation filter which,
 besides its anti-aliasing filtering functions, is suited to stack a
 determined number of samples with appropriate weightings as it is
 well-known to specialists.
 An analog-to-digital converter digitizes series of analog samples taken
 from a signal at times fixed by an internal clock. This is no drawback
 when the converter works in isolation. It becomes a drawback in all the
 cases where it is desired to precisely fix an initial reference time in
 relation to which a sequence of events is located, and especially when
 signal acquisitions are to be carried out by a series of different
 converters.
 In seismic prospecting operations notably, the seismic waves coming from
 the subsoil as a result of an emission by a source of seismic waves are
 picked up by a multiplicity of receivers and converted to digitized
 samples by an often considerable number of different acquisition chains
 provided each with an analog-to-digital converter. A reference time is
 selected, generally the time of triggering of the seismic source, and one
 tries to adjust in relation to this time the first significant sample
 taken by the various converters on each signal picked up by the receivers.
 If the sampling time of each converter only depends on an internal clock,
 there is no reason to be synchronized with the exterior event selected as
 the reference. A certain random delay or jitter follows therefrom, which
 is generally different from one acquisition chain to the next. The
 consequence thereof is a lack of synchronization that is very disturbing
 when signals received and acquired by different acquisition chains have to
 be combined, as it is generally the case in conventional seismic
 processing.
 BACKGROUND OF THE INVENTION
 Patent FR-2,666,946 (U.S. Pat. No. 5,245,647) filed by the applicant
 discloses a signal sampling device comprising in combination a sigma-delta
 type oversampling converter associated with a FIR type digital filter
 performing decimation of successive series of oversamples and a device for
 synchronizing the samples delivered with an exterior event such as the
 time of triggering of a seismic source for example. The solution used in
 this prior device essentially consists in a memory inserted between the
 sigma-delta converter and the decimation filter, wherein a series of
 oversamples is permanently stored. On reception of an exterior reference
 signal, the device is suited to find in the inserted memory the
 oversamples formed before reception of this signal and to command transfer
 thereof in the decimation filter so as to produce the first of the
 resynchronized samples.
 Although this solution is perfectly operational, it has the drawback of
 requiring complex and expensive electronic components inserted between the
 delta-sigma modulator and the FIR anti-aliasing filter, i.e. a memory and
 relatively complex means for managing it.
 There are also well-known fractional (less than one unit) delay processing
 techniques notably described by: Laakso T. I. et al : Splitting the Unit
 Delay; in IEEE Signal Processing Magazine; 1996, allowing to carry out, by
 means of calculations, time readjustment of the signal sampling. Certain
 principles thereof, useful for better understanding of the method, are
 reminded hereafter.
 x[n] denotes a series of digitized samples S.sub.k, S.sub.k+1, S.sub.k+2 .
 . . S.sub.k+p, etc, taken (FIG. 1) from a measuring signal from an initial
 time t.sub.0 on, with a sampling interval .DELTA.t, by an
 analog-to-digital converter, and y[n] denotes a series of samples
 S'.sub.1, S'.sub.2, S'.sub.3 . . . S'.sub.p+1, etc, taken with the same
 interval from the same measuring signal but readjusted in time from a
 reference time T.sub.R after t.sub.0. The readjustment time difference D
 is a positive real number.
 This number can generally be written as follows: D=int(D)+d, where int(D)
 corresponds to a whole number of sampling periods and d is a fraction of a
 period.
 We must have: y[n]=x[n-D].
 In order to obtain a delay int(D), it is sufficient to delay the initial
 signal x[n] by a simple translation. The samples of y[n] are those of x[n]
 whose index is simply delayed (renumbered) by int(D). The sample bearing
 number k in the first series for example becomes the sample bearing number
 1 in the second series, with k..DELTA.t=int(D). For the fractional part of
 this time difference, the readjusted samples y[n] will be somewhere
 between the values of x[n] at two successive sampling positions by the
 local clock and they must best correspond to the effective amplitudes of
 the sampled signals at these intermediate positions. This delay with
 readjustment can be obtained by applying a digital filtering F (FIG. 2).
 With the notations specific to the z transform, this delay by digital
 filtering can be expressed as follows:
EQU Y(z)=X(z).z.sup.-D.
 The frequency response of the ideal filter H.sub.ID is:
EQU H.sub.id =z.sup.-D =e.sup.-j.omega.D
 with
EQU z=e.sup.j.omega..
 The amplitude and phase responses of the ideal filter for any .omega. are
 therefore:
EQU .vertline.H.sub.id (e.sup.j.omega.).vertline.=1
 and
EQU arg[H.sub.id (e.sup.j.omega.)]=.theta..sub.id (.omega.)=-D.omega..
 The phase is often represented as a phase lag defined by:
 ##EQU1##
 a lag that is here D.
 The corresponding impulse response is obtained by inverse Fourier
 transform:
 ##EQU2##
 for any n, hence:
 ##EQU3##
 for any n.
 This ideal filter cannot be implemented because its impulse response is
 infinitely long. There are however several methods allowing to approximate
 to this ideal solution close enough for the readjustment precision to
 remain compatible with the precision expected in practice. Selection of
 the method to be used depends on the specific criteria to be observed
 within the scope of the application.
 The filtering method to be implemented must correspond to certain
 requirements linked with the means used:passband of the signals to be
 acquired, sampling frequency, technical limitations of the available
 digital filtering application means (calculation means) and expected
 precision of the readjusted sample calculation.
 Within the scope of an application to seismic data acquisition for example,
 it is imposed that the passband of the filter is compatible with all the
 useful signals carrying seismic information and therefore contains for
 example the [0 Hz, 375 Hz] frequency interval, as well as a 1000 Hz
 sampling frequency for the seismic signals. Real-time sample readjustment
 can be imposed if the acquisition units comprise powerful DSP type signal
 processors for example, as described in the aforementioned patents, which
 also contributes to facilitating implementation of digital filtering.
 SUMMARY OF THE INVENTION
 The synchronized seismic signal acquisition method according to the
 invention allows to obtain a series of digitized samples of each signal
 readjusted in time from at least one reference time on, from a first
 series of digitized samples of these seismic signals produced from any
 initial time prior to the reference time, by an acquisition unit, with a
 definite sampling interval.
 It is characterized in that it comprises:
 detecting a synchronization signal indicative of this reference time
 (produced in response to the detection of an event),
 measuring the effective time difference (D) between the reference time and
 the initial time,
 determining coefficients of a digital filter suited to compensate for the
 measured effective time difference, the coefficients being sufficient in
 number to obtain a fixed time compensation precision, and
 applying to the first series of samples the digital compensation filter,
 which allows to obtain a series of digitized samples readjusted from the
 reference time.
 Calculation of the fractional filter coefficients advantageously comprises
 direct determination of the ripple margins of the amplitude and of the
 phase respectively according to the maximum time error (E.sub.M) affecting
 the signal delayed by the fractional filter, which allows the coefficients
 of this filter to be calculated.
 According to a preferred embodiment,
 a) the values of the coefficients of N intermediate digital filters suited
 to compensate for N time difference values spread over the duration of the
 seismic signal sampling interval by the converter are first determined,
 then these filtering coefficients are stored,
 b) the coefficients of the fractional-delay digital filter suited to
 compensate for the measured effective time difference are calculated by
 interpolation (Lagrange type interpolation for example) between the
 coefficients of the series of filtering coefficients associated with the
 closest difference values, and
 c) the fractional compensation filter is applied to the first series of
 samples.
 The synchronization signal is generated for example upon detection of a
 seismic signal which can be the first-break wave emitted when a seismic
 source (impulsive source or vibrator(s)) is triggered, or upon
 identification (by correlation) with a known signature or upon detection
 of a certain seismic energy level, in the case where it is used for
 detecting an earthquake or a nuclear explosion, etc.
 The method according to the invention can be used notably in a device
 intended for synchronized acquisition of seismic signals picked up by
 seismic receivers in response to an emission of elastic waves in the
 ground at reference times by one or more seismic sources, these receivers
 being in contact with the ground and connected to at least one seismic
 acquisition unit at a distance from a control station. It comprises
 acquisition, with a definite sampling interval, of at least one series of
 digitized samples of these seismic signals by an analog-to-digital
 converter in each acquisition unit from an initial time prior to the
 reference time, detection by each acquisition unit of a synchronization
 signal indicative of the reference time, measurement of the effective time
 difference between the initial time and the reference time (possibly
 taking account of the time of propagation of each synchronization signal
 to each acquisition unit), determination of the coefficients of a
 fractional digital filter suited to compensate for the measured effective
 time difference, the coefficients being sufficient in number to obtain a
 fixed compensation precision, and application of the compensation filter
 to the first series of samples, allowing to form a series of digitized
 samples readjusted from said reference time.
 According to an embodiment applicable to seismic prospecting operations
 wherein elastic waves are emitted by several vibrators, the method
 comprises detection, by each acquisition unit, of synchronization signals
 emitted in response to the emission of waves by each vibrator and
 measurement of the corresponding effective time differences, determination
 of the coefficients of the various fractional digital filters suited to
 compensate for the various measured effective time differences, the
 coefficients being sufficient in number to obtain a fixed compensation
 precision, and application of the various compensation filters allowing to
 form series of digitized samples readjusted from the corresponding
 reference time.
 The digitizing method as defined above is advantageous on several accounts.
 It can be implemented from analog-to-digital converters of a standard type,
 non-modified and therefore readily available, in cooperation with a
 precise measuring means that is possibly included in the processing unit.
 Since digital filtering is carried out by software means, the method is
 flexible and readily adaptable to any type of application.
 Narrowing the interval between whose intermediate boundaries interpolation
 is performed in order to find the exact coefficients of the suitable
 digital filter, in cases where the coefficients of the digital filters
 applicable to several well-defined fractions of the sampling interval are
 precalculated, considerably reduces the calculation time.
 The method of readjusting series of samples according to the invention
 allows to let the (or each) converter "run" permanently and to form the
 series of samples readjusted from the reference time as soon as delay D is
 known.
 The synchronization signal that can be used is no longer only the
 triggering signal (TB) but, as we have seen above, the first-break wave,
 the time of recognition of an acoustic signature or of detection of an
 energy threshold, etc.
 In the case where several vibrators vibrating more or less in synchronism
 are used, with continuous seismic acquisition, it is necessary, in order
 to process the signals received (to correlate them with the signals
 emitted), to delay the reference time selected for readjustment to the
 beginning of the corresponding vibration, and in this respect the method
 according to the invention is particularly flexible.
 The synchronized seismic signal acquisition device according to the
 invention allows to obtain, from each seismic signal, at least one series
 of digitized samples readjusted from at least one reference time on, from
 a first series of digitized samples taken from this seismic signal,
 produced from any initial time prior to the reference time.
 It comprises at least one signal acquisition unit including at least one
 analog-to-digital converter producing this first series of samples, and it
 is characterized in that it comprises, in each acquisition unit, a means
 for detecting a synchronization signal, a processing unit associated with
 means for storing series of samples produced by said converter, a metering
 means for measuring the effective time interval elapsed between the
 initial time and the reference time, the processing unit being programmed
 to determine coefficients of a digital filter suited to compensate for the
 measured effective time difference (consisting of a whole part and of a
 fractional part) and to apply, to the first series of samples, the digital
 fractional compensation filter.
 The device is advantageously used in a seismic signal acquisition system
 where each acquisition unit comprises means allowing data exchange, by
 means of a transmission channel, with a distant station provided with
 means for emitting the synchronization signal in response to detection of
 an event, and a control system including the processing unit, suited to
 process the seismic data before they are transmitted to the central
 station. The processing unit is then programmed to determine coefficients
 of a fractional digital filter suited to compensate for the effective time
 difference measured by the metering means, by taking account of the time
 of propagation of the synchronization signal on the transmission channel
 connecting each acquisition unit to the central station.
 According to a preferred embodiment, the device comprises means for storing
 coefficients of N intermediate digital filters suited to compensate for N
 time difference values spread over the duration of the seismic signal
 sampling interval by the converter, and the processing unit is suited to
 calculate the coefficients of the digital filter suited to compensate for
 the measured effective time difference, by interpolation between the
 series of filtering coefficients associated with the closest difference
 values, and to apply the compensation filter to the first series of
 samples.
 According to an embodiment suited for seismic prospecting operations using
 vibrators as emission means, the device comprises metering means, in each
 acquisition unit, for detecting synchronization signals emitted in
 response to the emission of waves by each vibrator, and for measuring each
 effective time difference, the processing unit comprising means for
 determining the coefficients of the various fractional digital filters
 suited to compensate for the various measured effective time differences,
 and means for applying the various compensation filters allowing to form
 series of digitized samples readjusted from said reference time.

DETAILED DESCRIPTION OF THE INVENTION
 An important point of the method is that it has been possible to establish
 that, if .DELTA.D and a are respectively the ripple margins within the
 passband concerning the phase and the amplitude respectively, the maximum
 time error E.sub.M point by point between the delayed signal calculated by
 the filter and the ideal delayed signal (for which E.sub.M =0) can be
 expressed by the relation:
EQU E.sub.M.apprxeq.[(.DELTA.D+L ).sup.2 +L +(.alpha.).sup.2 +L ](1)
 provided that the signal is in the frequency band below the filter cutoff
 frequency.
 Knowing this relation between the maximum error and the filtering
 characteristics, a known FIR (Finite Impulse Response) filter design
 method is selected, that gives the coefficients of the appropriate filter.
 When an analog-to-digital converter running on 24 bits, including a sign
 bit, is used for example, one can impose that the error is at most equal
 to one quantification interval, i.e. 2.sup.-23.apprxeq.10.sup.-7.
 For a 10.sup.-7 error, a has to be of the order of 10.sup.-7, which
 requires a ripple margin .vertline.H.vertline. of the filtering function
 such that: 1-a.ltoreq..vertline.H.vertline..ltoreq.1+a. Since a is very
 small, it can be easily shown that this inequality is equivalent to:
 -10.sup.-6.ltoreq..vertline.H.vertline..sub.dB.ltoreq.10.sup.-6. The same
 relation (1) shows that .vertline..DELTA.D.vertline. must also be of the
 order of 10.sup.-7.
 With the specifications given above, suitable for seismic prospecting
 applications:
 cutoff frequency characterized by two values, 375 Hz and 420 Hz,
 oscillations in the [0, 375 Hz] frequency band below 0.05 dB,
 oscillations in the [0, 420 Hz] frequency band below 0.2 dB,
 phase cutoff frequency of 375 Hz,
 oscillations in the [0, 375 Hz] frequency band below a delay corresponding
 to a time delay of 4 .mu.s, the phase specification corresponding to 4
 .mu.s for a sampling frequency of 1000 Hz (therefore a time period T.sub.e
 =10.sup.-3 s=1 ms) gives a fractional delay:
 ##EQU4##
 These amplitude and phase constraints are thus greatly fulfilled if an
 error below 10.sup.-7 is imposed.
 The value of the filter coefficients then has to be determined, considering
 the imposed limitations, by selecting, as mentioned above, a method from
 the filter design methods known to specialists. The Lagrange interpolation
 method can be used, which provides very good frequency response and very
 flat amplitude response for the low frequencies, but the resulting filter
 has a very narrow passband that does not vary much with the order of the
 filter. There are also well-known time windowing methods (Kaiser window,
 Dolph-Tchebichev window, Blackman window, Hamming window, etc). Another
 known method, referred to as least-squares method, essentially consists in
 minimizing the frequency error between the ideal filter and the FIR filter
 used. The computing programmes required for calculation of the filtering
 coefficients according to these various methods are most often available
 from known signal processing software libraries such as Matlab.TM. for
 example.
 For each particular delay value, a resampling allowing approximation with a
 precision of the order of 10.sup.-7 requires calculation of a filter
 defined by several ten filtering coefficients. 60 are for example required
 with the generalized least-squares method to reach a precision of this
 order as shown in the table of FIG. 6.
 The time required fo such a calculation depends of course on the available
 computing means or resources. There are cases where, because of the
 conditions imposed: relatively wide passband of the signals to be acquired
 and/or performance of the available computing means, the desired
 readjustment of the sampled signals to a reference time cannot be
 performed in real time.
 A solution consists in this case in precalculating the filtering
 coefficients for well-defined fractions of the period or sampling interval
 .DELTA.t. Interval .DELTA.t being subdivided by N points I.sub.1, I.sub.2,
 . . . I.sub.k, I.sub.N (FIG. 3) into N+1 parts (N being 10 for example), N
 precalculated filters F.sub.1 to F.sub.N are thus defined. The various
 series of coefficients are then stored in memories of the computing unit,
 before the acquisition operations start. During operation, the time
 subinterval (between I.sub.k and I.sub.k+1 in the figure) in which the
 fractional time difference d between the initial sampling time and the
 reference time lies is determined, and interpolation is performed between
 the stored corresponding series of coefficients so as to calculate the
 coefficient of the required readjustment filter. The closer approximation
 to fractional delay d allowed by these previous calculations considerably
 reduces the time required for calculation of the appropriate digital
 filter. A precision of the order of 10.sup.-7 can for example be reached
 with 10 series of precalculated intermediate coefficients for an
 interpolation of the 5.sup.th order and with 20 series of intermediate
 coefficients for an interpolation of the 4.sup.th order, as shown in the
 table of FIG. 7.
 The method described can be implemented in a seismic acquisition system as
 described in the aforementioned patents, notably in patents
 FR-A-2,720,518; EP-A-594,477 or in patent application FR-97/09,547, suited
 to acquire the signals picked up by seismic receivers R (FIG. 4)
 distributed over a zone to be explored, according to a layout suited for
 the 2D or 3D type prospection to be performed, these receivers R picking
 up the seismic waves reflected by underground discontinuities, and to
 transmit them to a distant station CS such as a central control and
 recording station where all the seismic signals collected are eventually
 centralized, either directly or by means of intermediate stations LS
 fulfilling more or less complex functions:concentration, organization and
 sequencing of the exchanges between acquisition units A and central
 station CS.
 Source S can be impulsive (an explosive charge for example, or an air gun)
 or consist of one or more vibrators. This source can be coupled with the
 formations of the zone to be explored and connected by radio link or
 control cable to central station CS or, in the case of coastal zone
 exploration, possibly towed, while immersed, by a shooting boat connected
 to central station CS by radio link.
 Each acquisition box is suited (FIG. 5) for acquisition of a number k
 (k.gtoreq.1)) of seismic receivers R.sub.1, R.sub.2, . . . R.sub.k,
 providing each a seismic "trace". It therefore comprises for example k
 acquisition chains CA.sub.1, to CA.sub.k receiving respectively the k
 signals and comprising each, for example, a low-pass filter F.sub.11,
 F.sub.12, . . . F.sub.k, a preamplifier PA.sub.1, PA.sub.2, . . .
 PA.sub.k, a high-pass filter F.sub.21, F.sub.22, . . . F.sub.2k, and an
 analog-to-digital (ADC) converter AD.sub.1, AD.sub.2, . . . AD.sub.k for
 converting the amplified and filtered analog signals to 24-bit numerical
 words for example. The converters are for example oversampling sigma-delta
 type converters. All the acquisition chains are connected to a
 microprocessor 2 processing 16 to 32-bit numerical words for example,
 programmed to manage acquisition and exchanges with the distant station
 (not shown). A working memory M.sub.1 and a memory Mp for the programmes
 are associated with microprocessor 2. Processor 2 is connected to an
 emission-reception unit 3 suited to the transmission channel used for
 communication with the distant station. If it is a Hertzian channel, unit
 3 comprises a radio transmitter RE and a radio receiver RR that
 communicate with an antenna 4. An interface unit 5 described in patent
 FR-A-2,608,780 mentioned above also allows infrared communication with an
 initialization box 6 by means of which an operator can possibly transmit
 to management processor 2 addressing and selection instructions concerning
 the working parameters of the acquisition chains.
 Each acquisition box Ai also preferably comprises a processor 7 specialized
 in signal processing, such as for example a DSP 96002 type floating point
 32-bit processor, that is associated with a DMA type device for
 accelerating data block transfers between the two processors 2 and 7. A
 working memory M.sub.3 is associated with the latter processor. Each
 acquisition box also comprises a self-contained power supply unit 8.
 The function of general processor 2 is to perform decoding of the orders
 transmitted by the distant station and to manage acquisition of the
 signals of receivers R.sub.1 to R.sub.k by the various acquisition chains,
 transmissions in connection with transmission unit 3, memory M.sub.1 for
 temporary data storage, inputs/outputs, interrupts between programmes,
 exchanges with DSP computing processor 7, etc.
 DSP computing processor 7 is particularly well-suited for high-speed
 operations such as format conversions, complex number multiplications, FFT
 type Fourier transforms, correlations between received signals and emitted
 signals, digital filtering, successive shot stacking with suppression of
 disturbance noises of non-seismic nature, combination of the signals
 delivered by multi-axis seismic receivers such as three-axis geophones for
 example, etc. Preprocessing performed locally prior to transmission
 contributes to appreciably reducing the number of tasks set to the distant
 station.
 Each acquisition box can also comprise a flash-type high-capacity storage
 memory 9 for example, capable of absorbing a certain data volume that can
 be transmitted later to the central station.
 Processing unit (2, 7) in each acquisition unit preferably comprises means
 for storing (in working memory M.sub.3 for example) series of coefficients
 defining a certain number of intermediate precalculated filters F.sub.1,
 to F.sub.N and metering means C for determining precisely the time
 interval D=int(D)+d between the time when sampling of the signals produced
 by the seismic receivers has started by order of the local clock and the
 precise time of arrival of the reference signal. This time interval takes
 account of the time of emission of the signal by the seismic source (TB)
 and also of the effective time of propagation of this signal to the
 acquisition unit concerned, through the transmission channel (cable or
 radio link) connecting it to central station CS, which may vary on account
 of its position in the field.
 This time interval D being measured, signal processor 7 is programmed to a)
 renumber the samples taken before the reference time according to the
 value of int(D), as described above, b) calculate the coefficients of the
 fractional digital filter suited to compensate for the measured
 difference, and c) apply the suitable delay filter.
 The coefficients of the suitable digital filter are preferably calculated
 as mentioned above by interpolation between the series of coefficients of
 the N precalculated filters F.sub.1 to F.sub.N that bound the fraction d
 of time interval D, stored in memories M of processing unit 2, 7.
 The extremely precise readjustment that is performed between the signal
 emitted by the or each seismic source allows to improve the processing
 results such as "trace" stacking or correlations that are performed by
 each acquisition device in the field prior to repatriation of the seismic
 data to the central station.
 The method is particularly advantageous in cases where, for example,
 seismic operations are carried out with vibrators working simultaneously
 or with a time lag in relation to one another. Acquisition of the seismic
 signals is then performed continuously. The correlation that is
 conventionally carried out between each vibrational signal and the
 acquired signals first requires resynchronization thereof so as to take
 account of the time lags in relation to the various TB. This operation is
 performed without any difficulty considering the software means selected
 to do this.
 When using for example four sets of sweep-frequency vibrators with a sweep
 time of 16 s, an acquisition window or listening period of 6 s and a
 displacement time interval of 30 s, a mean slip time of (16+6+30)/4=13 s
 is determined. The signals intended to be correlated must first be
 readjusted by using the beginning of each period of time equal to this
 slip time as the reference. The processing unit of each acquisition device
 is readily adaptable in order to carry out the desired adjustments with
 the signals of the various vibrators.