FR3030965A1 - METHOD AND DEVICE FOR TRANSMITTING, METHOD AND DEVICE FOR RECEIVING - Google Patents

Abstract

The invention relates to a method for transmitting data represented by symbols, comprising the steps of: - applying a modulation to a set of symbols (D) in order to obtain a sample block (xe (1), xe (M)), - applying the block samples to a filter bank (120) arranged to output samples (z (1), z (KM)) forming a distributed signal over a set of subchannels, with overlapping adjacent sub-channels in the frequency domain; - transmission on a transmission channel (300) of signals determined according to the produced samples (z (1), z (KM)). The filter bank (120) is adapted to apply to the block samples a plurality of filters adapted to introduce a phase shift of the transmitted signals into neighboring subchannels. A corresponding reception method, as well as a transmitting device and an associated receiving device, are also provided.

Description

TECHNICAL FIELD TO WHICH THE INVENTION RELATES The present invention generally relates to the transmission of multicarrier signals.

It relates more particularly to a method and an emission device, and a method and a receiving device. BACKGROUND TECHNOLOGY Data transmission systems use multicarrier modulations to achieve high efficiency and provide operational flexibility. According to a widely used approach (commonly known as OFDM for "Orthogonal Frequency Division Multiplexing"), a discrete Fourier Transform (or Fast Fourier Transform) is used to spread the subcarriers over the width of the spectrum. transmission channel band and a Quadrature Amplitude Modulation (QAM) for sub-carrier data transport. In this technique, an iFFT block (designed to output the inverse Fourier transform of the inputted signal) outputs, on the basis of symbols received at the input, a block of samples of the signal to be transmitted. The sample blocks thus produced must however be transmitted with a guard interval to avoid interferences between blocks in the presence of the channel (due to multiple propagation paths). A cyclic prefix is introduced for this purpose in the transmitted sequence, which however decreases the useful data rate that can be transmitted. Another approach has been proposed to maximize throughput, as explained in the "FBMC physical layer: a primer" publication by M. Bellanger et al., PHYDIAS, June 2010. According to this approach (commonly known as FBMC modulation for 30 "Filter Bank based MultiCarrier" or OFDM / 0QAM, where OQAM stands for "Offset Quadrature Amplitude Modulation"), filter banks are used to divide the transmission channel into subchannels and a modulation (0QAM) in which the real part and the imaginary part of a complex symbol are transmitted with a time offset equal to half the duration of a symbol. Indeed, because of the frequency overlap between two adjacent subchannels, it is sought to cancel the interference between these subchannels, which allows the OQAM modulation alternating real and imaginary data (the response of the interference filter to a real pulse canceling half symbol for the imaginary part). A detailed description of this approach is given in the article by B. Hirosaki, "An Orthogonally Multiplexed QAM System Using the Discrete Fourier Transform", in IEEE Trans. on Communications, Vol. COM-29, July 1981. This technique is attractive for future transmission systems because of its increased maximum throughput (in the absence of guard time) and partly because of the high spectral resolution provided by filtering, which gives the possibility of having independent sub-channels. Both of these characteristics are crucial for new concepts of dynamic spectrum access and cognitive radio.

However, the use of the OQAM modulation inherent in this technique imposes a double processing rate of the symbol rate, which implies a significant increase in the complexity of the transceivers. It has been considered that this doubling of the rate at the receiver could be used to implement time equalizers of the subchannels; however, it has recently been shown that equalization in the frequency domain leads to higher performance without additional delay. Furthermore, the OQAM modulation is incompatible with certain multi-antenna techniques MIMO (for "Multi Input Multi Output"), in particular the so-called "Alamouti" technique which is included in recent standards relating to wireless communications and mobile cellular networks. OBJECT OF THE INVENTION In this context, the present invention provides a method of transmitting data represented by symbols, comprising the following steps: - application of a modulation to a set of symbols to obtain a block of samples - applying the block samples to a filterbank designed to output samples forming a signal distributed over a set of subchannels, with overlapping adjacent sub-channels in the frequency domain, - transmission on a transmission channel of the signals determined according to the samples produced, characterized in that the filter bank is adapted to apply to the block samples a plurality of filters designed to introduce a phase shift of the transmitted signals into neighboring subchannels. As explained below, the phase shift introduced makes it possible to eliminate interference between neighboring channels, without having to introduce a temporal shift between the real part and the imaginary part of a complex symbol (as in the case of the OQAM modulation as recalled above. ). It is thus possible to exploit the advantages associated with the filterbanks while using, for example, QAM modulation, as in the case of the OFDM technique. The filters of the filterbank can be real coefficients and thus affect independently the real part and the imaginary part of the processed samples.

The filters of the filter bank are for example derived from prototype sinus filters. The filters of the filter bank can thus notably be of the pseudo-QMF type. In one conceivable embodiment, the modulation applied to the set of symbols is a quadrature amplitude modulation.

In practice, the filter bank may be implemented by a set of phase shifters, a set of filters and an inverse Fourier transform block. According to a possibility of embodiment presented in detail in the description which follows, the samples of the block can be real and only half of the signals generated at the output of the set of filters can then be applied to the inverse Fourier transform block. Moreover, the filter bank can use a recovery factor equal to 2. The invention also proposes a method of receiving data 30 comprising the following steps: reception of signals on a transmission channel, construction of samples in function of the received signals, - application of the samples constructed to a bank of filters designed to receive, as input, samples forming a signal distributed over a set of subchannels, with overlapping of adjacent sub-channels in the frequency domain, - demodulation of the samples output products of the filter bank, characterized in that the filter bank is adapted to apply to the constructed samples a plurality of filters adapted to introduce a phase shift of the transmitted signals into neighboring subchannels. As in the case of transmission, the filters of the filter bank can advantageously be of real coefficients, for example derived from prototype sinus filters, possibly of the pseudo-QMF type. The demodulation of the samples produced can perform, for example, quadrature amplitude modulation detection. In practice, it can be provided that the filter bank is implemented by a plurality of operations including a Fourier transformation applied to the constructed samples. In addition, after Fourier transformation, the filter bank can be implemented by a set of filters and a set of phase shifters. It is then possible to interpose an equalizer between a block performing the Fourier transform and the set of filters. Sub-channel drift correction may also be provided by means of a phase shift specific to each subchannel after Fourier transformation and an interpolation adjustment of coefficients of at least one filter of the set. filters. According to the possibility of realization described later and already mentioned, each value generated by the Fourier transform can be applied to a first input of the set of filters and, in conjugate form, to a second input of the set of filters. . The invention also provides a data transmission device represented by symbols, comprising a modulation unit adapted to apply modulation to a set of symbols in order to obtain a sample block, a bank of filters designed to receive at the input of the sample block and for outputting samples forming a signal distributed over a set of subchannels, with overlapping adjacent sub-channels in the frequency domain, and a transmission interface designed to transmit, on a transmission channel, signals determined according to the samples produced, characterized in that the filter bank is adapted to apply to the block samples a plurality of filters designed to introduce a phase shift of the transmitted signals in neighboring subchannels. The invention further provides a data receiving device comprising a reception interface adapted to receive signals on a transmission channel, a sample construction module based on the received signals, a filter bank adapted to receive input of samples forming a signal distributed over a set of subchannels, with overlapping adjacent sub-channels in the frequency domain, and a unit for demodulating the samples produced at the output of the filter bank, characterized in that the filter bank is designed to apply to the constructed samples a plurality of filters designed to introduce a phase shift between the signals transmitted in 2 of the neighboring subchannels. The optional features contemplated above in terms of method can also be applied to these devices. A transmission system is thus proposed comprising a transmitting device as described above and a receiving device as described above. DETAILED DESCRIPTION OF AN EXEMPLARY EMBODIMENT The following description with reference to the accompanying drawings, given as non-limiting examples, will make it clear what the invention consists of and how it can be achieved. In the accompanying drawings: FIG. 1 represents the main elements of a digital data transmission system comprising a transmitter and a receiver; FIG. 2 shows in detail an element of the receiver of FIG. 1; FIG. 3 represents an example of the use of frequency ranges of a frequency band by a transmission system such as that of FIG. 1; FIG. 4 represents a system in which the frequency range used is centered on a specific frequency; FIG. 5 represents an alternative embodiment that can be envisaged for an element of the transmitter of FIG. 1. FIG. 1 represents the main elements of a digital data transmission system comprising a transmitter 100 and a receiver 200. transmitter 100 comprises a modulation unit 110, a synthesis filter bank 120, a collection and summation unit 130 and an analog transmission interface 140. The data sequence to be transmitted D (or symbol sequence) is applied in accordance with FIG. input of the modulation unit 110 which generates as output blocks of M complex samples xe (1), ..., xe (M) here corresponding to a QAM modulation of the received received sequence. To do this, the modulation unit 110 performs a series-to-parallel conversion and then a modulation (here of QAM type) of the data put in parallel. By "complex sample" is meant a sample whose value is a complex number, which therefore comprises a real part and an imaginary part. The QAM modulation used here generates complex samples, with no time lag between the real part and the imaginary part. These M complex samples xe (1), ..., xe (M) are applied in parallel to a real synthesis filter bank 120, for example of the pseudo-QMF type, which generates at output KM samples to be emitted z ( 1), ..., z (KM), where K is an integer representing the recovery factor used. A system using a recovery factor of 2 (K-2) is described. The filters of the filter bank 120 are designed to distribute the signals on M subchannels, overlapping neighboring sub-channels in the frequency domain, which makes it possible to optimize the use of the available frequency range on the channel. transmission 300, and to introduce a phase shift of 2 between neighboring subchannels, in order to eliminate interference between adjacent sub-channels as explained below. It should be noted that, since the filters of the filter bank 120 have real coefficients, the real part and the imaginary part of the complex samples xe (k) received at the input are processed independently by the filters, which makes it possible to apply the suppression technique. interference between neighboring subchannels by offset of 7.1 in transmission and reception, as described in M. Bellanger's "Digital Signal Processing", VViley, 2000, pages 320-325. As already indicated, pseudo5 QMF type filters are used for example. The filter bank 120 then comprises KM (here 2M) synthesis filters and the index filter n (1511-2M) has the following M coefficients: 1 1 7r 1 7r Ts (n, k) = (n) cos ( n - -) (k) + (k - -) -], 2 2M 22 the coefficients hs (n) of the prototype filter being of the form 1 7r hs (n) = sin (n--) I. With these notations, the filter bank 120 performs the following processing: z (n Ts (n, k) .xe (k)) k = This treatment can be carried out in practice by applying, to the M samples xe (k ) received at the input, coefficients Ts (n, k) of the 2M filters according to the formula which has just been indicated, An alternative embodiment of this treatment is furthermore described below with reference to FIG. signal, this bank of filters is called "lapped transform" (or "Lapped Transform") and its properties are described in the book HS Malvar entitled "Signal Processing with Lapped Transforms", Artech House, 1992. Samples z (1),..., Z (KM) generated at the output of the filter bank 120 are applied to the overlay and summation unit 130 which generates on this basis a sequence of complex values. refer to the patent application published under the reference FR2951046 for pl Details of the operation of the overlay and summation unit 130. The complex value sequence is transmitted to the transmit interface 140. This complex value sequence can be broken down into a real sequence and a sequence. imaginary which are multicarrier signals. The transmission analog interface 140 transmits an analog signal 30 corresponding to the sequence of complex values on the transmission channel 300 (for example in the form of radio waves propagated in the earth's atmosphere), an analogue signal which is received at the of an analog reception interface 210 of the receiver 200. In addition to the analog reception interface 210, the receiver comprises a serial / parallel converter 220, an analysis filter bank 230 and a QAM detection unit 240. These various elements allow the reception of reverse operations from those performed on transmission. Specifically, the receiving analog interface 210 provides real and imaginary sequences which are applied, through the serial / parallel converter 220, to the analysis filter bank 230, for example pseudo-QMF type. The QAM detection unit 240 then restores, from the samples generated at the output of the filterbank 230, the data, converted into a serial stream. Note that the samples of the received signal are grouped in blocks of KM samples at the serial-parallel converter 220, two successive blocks having (K-1) M samples in common. The filter filters of the filter bank 230 are real filters (that is, filters with real coefficients), which are the transposed filters of the synthesis filters used on transmission (in the filter bank 120 ), here for example pseudo-QMF type filters. Thus, if we write x (1), ..., x (KM) the KM samples generated by the series-parallel converter 220 and applied at the input of the filterbank 230, and y (/), the samples produced in At the output of the filter bank 230, the treatment carried out by the filter bank 230: 2 Al y (k) = T (n, k) × (n) n = 1 where the index filter can be expressed as follows. k (among M filters) has the KM (here the 2M) coefficients: 1 41 1 27- T, k) = h (n) cock (n + -) (k -)] n <2M 1 <k <M 2 2 2 M and with, for the coefficients of the prototype filter: h (n) = - sir "n 12) 271-'11. A possible embodiment 30 for the implementation of the analysis filter bank 230 is described with reference to FIG. 2. This embodiment is based on the decomposition of the treatment carried out by the filter bank into several operators. , explained below for the case described here where K = 2. Indeed, the coefficients introduced below can be written: T (n, k) = - [e I, r I, r IMI, r 1 M 4 In) Jln n +) (k) 2 2 2M 2 2M - e 2 2M 1 [e 2 2 2 M e In developing the product, it comes z 1 71- (k 1) jel (k 1) .1 (k) jkj ink 2 2 [e 2 - 2 Az / AI] ] T (n, k) = - [e - e] - e 2 2 [e Let YF (k) be the output of a discrete Fourier transform (or FFT) of order 2M when the block x (n) is applied at the input: 2M-1 - jk 2.7 jk-211 / n jnk (k) = x (n) e 2M = e M x (n) e Ai n = 0 n = 1. By combining the above expressions we obtain the following relation between Y (k) and YF (k) I .1 J (kJ (Ic y (k) _ -He 1 2 e 2M YF (2M + 1- k) - e1) 21YF (2M + 2 - k)] 4 zi (ki (k - e 2 2 [e 2M "F (k) e 21! F oll Thus, the analysis filter bank 230 can be realized, in accordance with the FIG. 2, by the cascading of the following three operators: a discrete Fourier transform (FFT) of order 2M, carried out by an FFT block 232, a set of filters 236 with the coefficients [1 -1] in the frequency domain (the terms eJk2Ai representing the temporal offset of a half-period of the data), this filtering in the frequency domain making it possible to combine, for each subchannel, a number of signals equal to the number of frequency coefficients of the prototype filter, a set of phase shifters 238 which perform the rotations of multiple phases of and thus make it possible to cancel the interference between neighboring subchannels 2; Conferences between neighboring subchannels are eliminated by the phase shift of between adjacent subchannels to the emission and phase shift of 2 2 between neighboring subchannels upon reception, which combine into a subtraction.

As seen in the last formula above, the final result y (k) is obtained by differentiating between index values k and 2M-1-k. Note that, thanks to this decomposition, the first operator of which is a Fourier transform, it is possible to introduce an equalizer 234 into the processing, precisely at the output of the FFT block 232, as can be seen in FIG. thus be easily compensated by equalization, as in the case of OFDM. For example, in the equalizer 234, an equalization technique known as "zero forcing" may be used, according to which the equalization consists of multiplying the output of the FFT by the inverse of the quantity C (k), estimated or measured value of the response of the channel at the frequency k, 2M. The decomposition which makes it possible to introduce the equalizer 234 at the output of the FFT block 232 has been presented above in the case where K-2. However, it can be implemented with a larger recovery factor K, in order to obtain even more selective filtering. In this case, methods for calculating the prototype filter as described in the literature are used, in particular in the P.Chu article, "Quadrature mirror filter design for an arbitrary number of equal bandwidth channels", IEEE Transactions, Vol. ASSP-33, pages 203-218, 1985, or in the article by H.S.Malvar "Extended lapped transforms: properties, applications and fast algorithms", IEEE Transactions on signal processing, vol. 40, pp. 2703-2714, 1992. The coefficients of the prototype filter h (n) are then formed of several summed terms. For example, h (n) can take the following form, which corresponds to a recovery factor K-2P: h (n) = a psin ((2p-1) 2 1 KW-) P = 1 where a, denotes a real coefficient. By repeating the previous developments, the schemas of the transmitter and the receiver are preserved, but, the Fourier transform at the reception 30 takes the KM dimension. In the receiver, the equalizer 234 comprises KM terms and its KM outputs are applied to the filter 236, a first part of which provides 2M outputs (using the coefficients aP) and a second part of which provides, from these 2M outputs, M outputs. by subtraction two by two (as explained above for the case where K-2). FIG. 3 shows an example of the use of frequency ranges of a frequency band by a transmission system such as that described above with reference to FIGS. 1 and 2. For purely illustrative purposes, the frequency band represented in FIG. 3 is between 0 and 1; another frequency band could naturally be used by simple transposition. It is noted that a characteristic of an actual filter bank, such as that used in the transmission system of FIG. 1, is the symmetry of the frequency response: with a bank of M filters, the frequency band used ( assumed equal to the unit as indicated above) is divided into M sub-channels of width LM, each subchannel consisting of two sub-bands of width 1 / 2M and symmetrical with respect to the center of the band used ( here 1/2).

In wireless and mobile cellular transmission systems, a frequency band is generally shared among several users, ie each occupies part of the band. Moreover, the advantage of the technique of filter banks is that it does not impose the synchronization of the users like the OFDM, but that it allows them to be independent. With real filter banks such as those of the transmission system of FIG. 1, it is therefore possible in practice to assign to a user the frequency ranges 111, f 2 and [1712, 17M, as represented in FIG. other use of the rest of the frequency band (or spectrum); this is called "fragmented spectrum". To avoid this fragmentation of the spectrum, it is also possible to assign to a user an unfragmented frequency range of the type [1 / 2- (f2-f1), 1 / 2- (f2-f ')]. FIG. 4 shows a system in which the frequency range assigned to the user is furthermore centered on a specific frequency / 0 in the frequency shift band, ie a multiplication of the output signal of the transmitter by the factor e2 "2)". Then, on reception, the spectrum must be recentered before the multicarrier demodulation, as also shown in FIG. 4. Another advantage of the use of a transmission system as described above with reference to FIG. ability to easily and effectively compensate for frequency of carrier frequency offset (CFO) that can occur in the transmission, for example due to the Doppler effect for mobiles or oscillator imperfections in terminals. Thus, for a frequency offset (5., the received signal is of the form: x (n) = f By considering, in the receiver, a filter with 2M coefficients denoted 11 ,,, with 0 i 2M-1, and frequency response H (f), the signal after filtering 2,11-1 j2zf fy (FO = x (no - is written at a particular instant - 2M-1 is again (n0) = e .12 "fh 122 r ji x (no ,, 11 This expression shows new coefficients '' which correspond to a frequency shift in the response of the filter H (f) which becomes H, (f + f). in the receiver of the interpolated coefficients which produce the inverse shift As for the term fno, it is corrected by phase rotation at the output of the filter The interpolated coefficients for the receiver can be calculated from the original coefficients, for example [1 -11 in the case where K-2, using the Lagrange technique, as described in the article by G. Oetken, "A new approach for the design of digital interpolation filters ", IEEE Trans., ASSP-27, 1979, pp. 637-643. The quality of the interpolation increases with the number of coefficients used; 6 or 8 coefficients are usually enough. The Lagrange technique is known for its simplicity and flexibility of use but other techniques can also be applied, in particular the discrete Fourier transform. FIG. 5 represents an alternative embodiment of the synthesis filter bank 120 of FIG. 1. According to this variant, dual operations of those used in the receiver and represented in FIG. 2 are used. Thus, the samples xe (k) received at the input are applied to a set of phase shifters 122, then to a set of filters 124 which generates KM output samples, which are applied to a KM order iFFT block 126 which performs an inverse discrete Fourier transform and outputs the KM samples z (n). Indeed, by taking the expression given above from the output of the filter bank 120 z (n): 71 "1 1 7-1-1 7-1-z (n) = sin ((n-1/2 ) cos [(n - -) (k - -) - + (k - -) - I xe (k) 2M k = 1 2 2M 2 2 1 z (n) - (zl + z 2) one can write: 4j with z1 = j1k - 1, r22 in (k-11, r1A1 e_ 72m) 1k 121, - i 2 ink M e-jk 2A1 xe (k) 2 xe (k) k = 1 and z2 = -p This last term can be rewritten Yr xe (k) j (2M -1-1-A-) .7 A / - / (2A / + 1-k) / r / 221 j (221 1-k -) / 2 -j (2214-k--), fn (21V1-k) Yrhil "211.1-k), 2 , 14 2 e - xe (k) k = I By introducing the discrete inverse Fourier transform of order 2M, it comes: 1 2A1 xel (k) k = 1 with, for 2 <k <M, xe 1 (k ) = e / 4 (k) e2-xe (k-1) e-1'k-2lej (k-2M xe 1 (2M + 2-k) = - e14 [xe (k) ejk'n - xe (k -1) ej (k-11, r / 2iej (2M + 2-kl, ri2M z2 = k = 1 e and xel (1) = -xe (1) .2.sort (/ 4); xe l (M + 1) = j.xe (M) .2.sin (/ 4) It thus appears that the signal to be emitted z (n) is obtained at the output of an inverse discrete Fourier transform (iFFT) at the input of which the data xe (k) are applied to the symmetrical index inputs k and 2M-i-2-k, after filtering and phase shifts, as proposed above with reference to FIG. has just been described further allows to introduce an alternative embodiment for the transmission system, as now described. In this alternative embodiment, instead of the QAM unit 110, a modulation unit is used which produces M real samples, for example by means of a modulation of PAM type (for "Pulse Amplitude Modulation"). "). These M real samples are applied to the real filterbank (for example as described above with reference to FIG. 5), which thus produces samples to be transmitted at real value. Since the transmitted signal is then real, its spectrum has hermitian symmetry, ie, by repeating the notations used above, a component at the frequency 1-k / 2111 is the conjugate complex of a component at the frequency k / 2M. It is proposed here to use this redundancy to limit the transmission to only one of these two components. This compensates for the division of the bit rate by two, because of the use of a modulation producing real values (compared for example with the QAM modulation) and the spectral efficiency of the system is then the same as for the QAM modulation. . This results in an operation at the symbol rate which has the same spectral efficiency as a filter bank system using the OQAM modulation, which in turn operates at a rate twice that of the symbols. In practice, this results in a division by two of the rate of computations and simplifications in the manipulation of data, which no longer have to be applied alternately to the real and imaginary inputs of the system for the successive symbols, as required the OQAM. The transmission of only one of the two components can be carried out in practice in the transmitter by not applying, at the input of the block iFFT 126, any sample on the entries of index greater than M, which makes it possible not to generate the upper half of the spectrum (corresponding to the components at frequency 1- (k-1/2) / 2M mentioned above).

At the level of the receiver, constructed for example as that of FIG. 2, the components (not transmitted) at the frequencies 1-k / 2M are outputted from the FFT block 232 by taking as their value the conjugate complex of the components at the frequencies Ie2M for 0 k s-AI-1. (Recall that the conjugate of a complex number has a real part equal to the real part of said complex number and an imaginary part opposite to the imaginary part of said complex number.) Thus the signal is artificially reconstructed over the entire spectrum, have actually transmitted a signal (and thus occupied the transmission channel) on half of the spectrum only.

Claims (20)

REVENDICATIONS1. A method of transmitting data represented by symbols, comprising the steps of: - applying a modulation to a set of symbols (D) to obtain a sample block (xe (1), xe (M)) applying the block samples to a filter bank (120) arranged to output samples (z (1), z (KM)) forming a distributed signal over a set of subchannels, with overlapping sub-channels; neighbor channels in the frequency domain, - transmission on a transmission channel (300) of signals determined according to the samples produced, characterized in that the filter bank (120) is adapted to apply to the block samples a plurality of filters designed to introduce a phase shift of the transmitted signals into neighboring subchannels. 2 The transmitting method according to claim 1, wherein the filters of the filter bank (120) are derived from prototype sinus filters. 3. Transmission method according to claim 1 or 2, wherein the filters of the filter bank (120) are pseudo-QMF type. 4. Transmission method according to one of claims 1 to 3, wherein the modulation applied to the set of symbols is a quadrature amplitude modulation. 5. Transmission method according to one of claims 1 to 4, wherein the filters of the filter bank (120) are real coefficients. 6. Emission method according to one of claims 1 to 5, wherein the filter bank (120) is implemented by a set of phase shifters (122), a set of filters (124) and a transformation block Fourier inverse (126). The method according to claim 6 taken in dependence on one of claims 1 to 3, wherein the block samples are real and in which only half of the signals generated at the output of the filter set are applied to the block. inverse Fourier transform. 8. Transmission method according to one of claims 1 to 7, wherein the filter bank uses a recovery factor equal to 2. 9. A method of receiving data comprising the following steps: - receiving signals on a transmission channel (300), - building samples (x (1), x (KM)) according to the received signals, - applying the samples constructed (x (1), x (KM)) to a filter bank (230) adapted to receive as input samples forming a signal distributed over a set of subchannels, with overlapping neighboring subchannels in the frequency domain, - demodulation of the samples (y (1), (y (M)) produced at the output of the filter bank (230), characterized in that the filter bank (230) is designed to apply to the samples constructed a plurality of filters adapted to introduce a phase shift of the transmitted signals into neighboring subchannels. The reception method of claim 9, wherein the filters of the filter bank (230) are derived from prototype sinus filters. 11. The reception method according to claim 9 or 10, wherein the filters of the filterbank (230) are pseudo-QMF type. The reception method according to one of claims 9 to 11, wherein the demodulation of the produced samples performs a quadrature amplitude modulation detection. 13. The reception method according to one of claims 9 to 12, wherein the filters of the filter bank (230) are real coefficients. 14. The reception method according to one of claims 9 to 13, wherein the filter bank (230) is implemented by a plurality of operations including a Fourier transform applied to the constructed samples. 15. The reception method according to claim 14, wherein the filter bank (230) is implemented, after Fourier transform, by a set of filters (236) and by a set of phase shifters (238). 16. The reception method according to claim 15, wherein a subchannel drift correction is performed by means of a phase shift specific to each subchannel after Fourier transform and an interpolation adjustment of coefficients. at least one filter of the set of filters. The method of claim 15 or 16, claim 14 being dependent on one of claims 9 to 11, wherein each value generated by the Fourier transform is applied to a first input of the set of filters ( 236) and, in conjugate form, at a second input of the filter assembly (236). 18. The reception method according to one of claims 14 to 17, wherein an equalizer (234) is interposed between a block (232) performing the Fourier transform and the set of filters (236). A data transmission device represented by symbols, comprising: - a modulation unit (110) adapted to apply a modulation to a set of symbols to obtain a sample block (xe (1), xe ( M)), - a filter bank (120) arranged to receive the sample block at the input and to output samples (z (1), z (KM)) forming a signal distributed over a set of sub-elements. channels, with overlapping adjacent sub-channels in the frequency domain, - a transmission interface (140) arranged to transmit, on a transmission channel (300), signals determined according to the samples produced (z (1), z (KM)), characterized in that the filter bank (120) is adapted to apply to the block samples a plurality of filters arranged to introduce a phase shift of 7.1 between the transmitted signals in neighboring subchannels. 2 20. A data receiving device comprising: - a reception interface (210) designed to receive signals on a transmission channel, - a construction module (220) of samples (x (1), x (KM)) according to the received signals; - a filter bank (230) adapted to receive as input samples forming a signal distributed over a set of subchannels, with overlapping of adjacent sub-channels in the frequency domain, - a demodulation unit (240) samples produced at the output of the filterbank (230), characterized in that the filter bank (230) is adapted to apply to the constructed samples a plurality of filters designed to introduce a phase shift of 7r between the signals transmitted in neighboring subchannels. 2