Signal processing method

Abstract

The invention relates to a signal processing method that transforms X input signals into M output signals, the method comprising the following steps: a) applying a decorrelation filter to the X input signals so as to generate N decorrelated signals ei; b) for each decorrelated signal ei determined in step a), generating N delayed signals ei,k by applying a delay τk and a gain gk to each decorrelated signal ei, the delays τk and the gains gk being chosen such that, for each decorrelated signal ei, the delayed signals ei,k simulate a propagation of the decorrelated signal in a virtual space, the virtual space comprising N virtual loud speakers and a virtual listening position which are distributed according to a predetermined geometry; c) for each k, summing the delayed signals ei,k (Formula 1);<sb / > e) determining M output signals, each output signal resulting from a linear combination of the N sums of the delayed signals Ek.

Description

[0001] Description Title: Signal processing method Technical field The present invention relates to a signal processing method, in particular audio signals. Prior art There are numerous musical formats defining a structure of musical content in computer form. In particular, a musical format is characterized by a number of sound channels. Thus, depending on the musical format, the musical content is intended to be played back via a particular hardware configuration. Musical content in stereophonic format, comprising two sound channels, is for example intended to be played back on two speakers, while musical content in 5.1 format, comprising 6 sound channels, is intended to be played back on 6 speakers. When sound content of a certain format is not played back on the expected number of speakers, listening to the content is impaired.In particular, listening to stereophonic content on a home cinema system, typically comprising 6 to 8 speakers, does not provide a satisfactory immersive experience. The same is true for listening to musical content in 5.1, 7.1, Ambisonics or Dolby Atmos® format, comprising four or more channels, which would be reproduced on headphones, comprising only two speakers. Spreading algorithms designed to spread musical content across several speakers exist. These algorithms allow, for example, the diffusion of monophonic content across three or more speakers. However, even if these algorithms allow for a greater impression of diffusion of the musical content, the precision of the sound content is deteriorated by the addition of blur to the sound content.Adding reverberation is also a known method for radiating monophonic content into multiple speakers. This method involves simulating listening to the sound content in a virtual room, taking into account the acoustics of the virtual room and in particular the resonances generated by the virtual room. However, this method distorts the content by adding resonance not initially present. US 2021 / 352425 describes a method for processing a stereophonic signal, including the formation of a center channel signal. US 2021 / 352425 describes a system for simulating an acoustic space aimed at modeling the real acoustics of a venue. The article by Von Türckheim Friedrich et al, “Virtual venues - an all-pas based time-variant artificial reverberation system for automotive applications”, describes signal processing software for applications in the automotive field.This article suggests a system comprising a simulation of an acoustic space aimed at recreating a real space. US 2004 / 136554 discloses a method for processing a stereo signal intended to be broadcast via two loudspeakers. Generally speaking, the known methods do not allow for the separate definition of a number of channels of the sound content, a number of loudspeakers of the installation intended to reproduce the sound content, a resonance, a color and a sound timbre. It is not possible, starting from a monophonic sound content, to reproduce this monophonic content via all the loudspeakers of a 5.1 home cinema type installation, while maintaining a precise and detailed perception of the localization of the musical content, by introducing a sensation of sound immersion, and by preserving the integrity of the sound content, i.e. the original artistic intention. There is therefore a need to address these problems.An objective of the present invention is to meet all or part of this need. Summary of the invention The invention thus relates to a signal processing method transforming X input signals a. p , p being an integer belonging to the interval [1, X], in M ​​output signals s q , q being an integer belonging to the interval [2, M], the method comprising the following steps: a) application of a decorrelation filter to the X input signals a p so as to generate N decorrelated signals e i , i being an integer belonging to the interval [1, N]; b) for each decorrelated signal e i determined in step a), generation of N delayed signals e i,k by applying a delay τ k and a gain g k to each decorrelated signal e i, k being an integer belonging to the interval [1, N], the delays Tk and the gains gk being chosen so that for each decorrelated signal and, the delayed signals ei,k simulate a propagation of the decorrelated signal in a virtual space, the virtual space comprising N virtual loudspeakers and a virtual listening position distributed according to a predetermined geometry, c) for each k integer belonging to the interval [ 1 , N], summation of the delayed signals ei,k according to the following formula:

[0002] [Math 1] d) optionally, duplication of the delayed signals ei,k determined in step b), and for each duplicated signal ei,k,j, j being an integer greater than or equal to 1, application of a delay εi,k,j, then summation of the delayed duplicated signals ei,k(t - Si,k,j), according to the following formula: for each k integer in the interval [ 1 , N], for each j,

[0003] [Math 2] e) determination of M output signals, each output signal s q resulting from a linear combination of the N sums of the delayed signals resulting from step c), Ek, and, when a step d) is implemented, of the j*N sums of duplicated signals resulting from step d), Fk,j.

[0004] The method according to the invention makes it possible in particular to reproduce musical content comprising X input channels on an installation comprising M physical loudspeakers. Thanks to the method according to the invention, it is possible to obtain M output signals from X input signals without the latter being altered and, in particular, without the latter being compressed or there being a loss of precision.

[0005] A method according to the invention makes it possible to reproduce musical content in a format comprising X input channels on M physical speakers, without compression or addition of sound blur, M being different from X.

[0006] Preferably, no reflections or reverberations are added to the musical content.

[0007] In one embodiment, M is greater than X.

[0008] In step a), the decorrelation filter may include an allpass filter, or a whitening filter, or the application of random delays to the input signals a p , for example the application of relatively short delays, preferably the delays being less than 10 ms, or the application of a discrete cosine transform to the input signals a p. The number of decorrelated signals generated in step a) can be between 1 and the largest value between X and M, in other words N can be an integer in the interval [1, max(M, X)]. In a preferred embodiment, N is equal to the maximum of M and X, N=max(M, X). A small number of decorrelated signals makes it possible to minimize the computing resources required to implement steps a) to e), while a large number of decorrelated signals makes it possible to improve the final listening quality. A number of decorrelated signals equal to the maximum of X and N advantageously minimizes the computing resources required while having satisfactory listening quality. In step b), for each e i , we determine N delayed signals e i,k , each delayed signal e i,k which can be associated with a simulation of listening to the signal e i through a virtual speaker H kof the virtual space, for k fixed, k being an integer belonging to the interval [1; N]. For example, e 1,1 can simulate listening to a decorrelated signal e1 through the virtual loudspeaker H1 of the virtual space, e 1,1 being determined by applying a delay τ1 and a gain g1 associated with the virtual loudspeaker H1. In step b), each delay τ k can be associated with a virtual speaker H k of virtual space. Similarly, each gain g k can be associated with a virtual speaker H k of virtual space. The delays τ k can be determined based on the distance separating each virtual speaker H kof the virtual listening position in the virtual space, depending on the speed of sound propagation in the vacuum c. In a particular embodiment, a user can modify the number and / or the position of the virtual loudspeakers, and / or the virtual listening position in the virtual space. The gains g k can be calculated based on the distance separating each virtual loudspeaker from the virtual listening position in the virtual space. The further the virtual loudspeaker is from the virtual listening position, the lower the gain associated with that virtual loudspeaker will be. The virtual listening position can be modified. In one embodiment, it is pre-parameterized.

[0009] Virtual space can be a space without resonance.

[0010] In particular, the virtual space does not include any feature intended to model real acoustics. In a preferred embodiment, the predetermined geometry of the virtual space is determined so as to minimize an interaural cross-correlation coefficient (in the IACC suite).

[0011] The IACC is an index of similarity between signals from the two ears that can be used to model spatial perception of sound content. If we call g(t) and d(t) two signals from two ears, the intra-aural cross-correlation coefficient is defined, in the time domain, by:

[0012] [Math 3] as a function of the time shift δ.

[0013] The lower the IACC, the greater the feeling of immersion experienced by a user listening to the audio content.

[0014] Alternatively or additionally, the predetermined geometry is determined so as to minimize variations in frequency levels resulting from frequency responses of the delayed signals. Advantageously, such a geometry makes it possible to limit artifacts that may be present in the frequency response of the delayed signals ei,k.

[0015] Preferably, the virtual space is infinite or equivalent to a deaf chamber, that is to say, there is no resonance in this virtual space, in other words the virtual space is a space without resonance. In particular, the virtual space does not have any obstacle reflecting the signal. The absence of resonance makes it possible in particular to preserve the original artistic intention.

[0016] The method according to the invention may comprise, before step b), in particular before step a), the selection of a predetermined geometry of the virtual space. Preferably, the selection of the predetermined geometry is carried out by specifying a number of virtual loudspeakers of the virtual space. The optional step d) has the advantage of accentuating the sensation of immersion by multiplying the number of signals around the delayed signals. This is referred to as densification of the delayed signals. Preferably, J is equal to 1, thus accentuating the sensation of immersion while limiting the computing power required to generate these duplicated signals. In step e), the linear combination of the N sums of delayed signals, and, when a step d) is implemented, of the j*N sums of duplicated signals, may comprise fixed coefficients. The coefficients may vary according to the desired intensity for each of the M output signals.In particular, the coefficients may depend on an actual installation configuration comprising M physical loudspeakers through which the M output signals are intended to be transmitted. The coefficients may all be equal, with listening to sound content through the actual installation then providing an omnidirectional enveloping sensation. Alternatively, the coefficients may be higher for output signals intended to be transmitted through the physical loudspeakers of the actual installation located in front of a user compared to the coefficients of output signals intended to be transmitted through the physical loudspeakers of the actual installation located behind the user, with listening to sound content by the user through the actual installation then providing a frontal enveloping sensation.In general, the linear combination of step e) may include coefficients determined as a function of the desired enveloping sensation. The desired enveloping sensation may in particular be chosen from a lateral, frontal, rear, omnidirectional enveloping sensation, this list not being limiting. In a preferred embodiment, the coefficients are determined so that for k fixed, the coefficient applied to E. k is equal to the coefficient applied to F k,j, for all j. Step e) can include the following steps: - for all k, summation of E k and F k,j for all j, so as to obtain a vector of size (1, N), composed of the sums of the E k , F k,j; - multiplication of the vector with a matrix of size (N, M) determined according to the coefficients of the linear combination, the matrix being able to be defined by the set of coefficients αk,q, k being an integer belonging to the interval [1, N] and q being an integer belonging to the interval [1, M],

[0017] The matrix can be defined by the following coefficients:

[0018] - for all k=q, αk, q is equal to 1,

[0019] - for all k different from q, αk, q is less than 1 and greater than or equal to 0.

[0020] When N = M, the matrix can be a unitary matrix.

[0021] When N = M, the matrix can be a circulant matrix with α1,1 equal to 0, α1, q less than 1 and greater than or equal to 0, for q different from 1. In particular, the circulant matrix can include α1, 2 less than 1 and greater than 0, α1, q =M less than 1 and greater than 0 and α1,q equal to 0 for q different from 1, 2, and M, preferably α1, q =M being equal to α1, 2.

[0022] For example, for N=M=5, the matrix can be equal to:

[0023] [Math 4]

[0024] A coefficient of 1 can be replaced by applying a gain of 0 dB.

[0025] A coefficient less than 1 and greater than or equal to 0 can be replaced by applying a negative gain in decibels (dB).

[0026] The method according to the invention may comprise, before step e), in particular before step a), the selection of coefficients for the linear combination of step e), preferably by selecting a desired enveloping sensation.

[0027] Steps a), b), c), d) when implemented, and e) can be implemented by computer.

[0028] In one embodiment, M is greater than or equal to three and X is equal to two.

[0029] The method can be used to transform two input signals into at least three output signals, more preferably at least four output signals, even more preferably at least five output signals, even more preferably at least six output signals. The use of a method transforming two input signals into three or more output signals is particularly advantageous for reproducing content in stereo format on the speakers of a motor vehicle or on the speakers of a concert hall or a cinema hall or on the speakers of a home cinema installation. In an alternative embodiment, M is equal to two and X is greater than or equal to four. The method can be used to transform at least four, at least five or even more preferably at least six input signals into two output signals.Using a process that transforms four or more input signals into two output signals is particularly advantageous for reproducing content in Dolby Atmos®, Ambisonics, 5.1, 7.1 format on the two speakers of a headset.The invention also relates to a use of a method according to the invention for transmitting: - stereophonic musical content comprising two input signals via loudspeakers of a motor vehicle, or - stereophonic musical content comprising two input signals via loudspeakers of a home cinema system or a Dolby Atmos® system comprising at least six loudspeakers, better still at least eight loudspeakers, even better still at least ten loudspeakers, or - stereophonic musical content comprising two input signals in a concert hall comprising four or more loudspeakers, or - audio content comprising at least four input signals, preferably six input signals, via headphones comprising two loudspeakers. The invention also relates to a computer program comprising instructions which, when the program is executed by a computer, cause the latter to implement the signal processing method according to the invention.Preferably, the virtual space, in particular the predetermined geometry of the virtual space and / or the values ​​of the gains g. k and τ kare stored in a database. Several geometries of virtual spaces can be stored in a database. Before step a) or before step b), the predetermined geometry of the virtual space for implementing step b) can be selected from the database. Several matrices determining coefficients for the linear combination of step e) can be stored in a database. Before step e), in particular before step a), a matrix is ​​selected from the database to implement step e) according to the invention. The invention also relates to a system comprising: - a computer program according to the invention, - optionally a database, comprising at least the geometry of the virtual space for implementing step b) and at least one matrix comprising the coefficients of the linear combination for implementing step e).The system may include an interface allowing the user to configure the geometry of the virtual space and / or the coefficients of the linear combination. In particular, the user may configure the coefficients of the linear combination by selecting a desired enveloping sensation. In particular, the user may configure the geometry of the virtual space by selecting a number of virtual speakers of the virtual space. The system may include a microprocessor comprising the computer program according to the invention. The microprocessor may be embedded. The microprocessor may be included in a telephone, a television, a multimedia box such as a television box, a car radio, a computer, a tablet, a smart watch. This list is not exhaustive. A system according to the invention may further include M physical speakers, intended to transmit the M output signals.Brief description of the drawings Other features and advantages of the invention will become apparent upon reading the detailed description which follows, and upon examining the appended drawing, in which: [Fig 1] Figure 1 illustrates an example of implementation of a method according to the invention, [Fig 2a] Figure 2a shows a diagram of a top view of an example of predetermined geometry of the virtual loudspeakers and of the virtual listening position in a virtual space comprising 8 virtual loudspeakers, [Fig 2b] Figure 2b shows a diagram of a side view of the example of predetermined geometry of Figure 2a, [Fig 3] Figure 3 illustrates another example of implementation of a method according to the invention, and [Fig 4] Figure 4 shows a diagram of a system for implementing a method according to the invention.

[0030] Detailed description

[0031] Figure 1 illustrates an example of implementation of a signal processing method according to the invention, in which the number of input signals is equal to two (X=2) and the number of output signals is equal to four (M=4).

[0032] Step a)

[0033] In step a), a decorrelation filter 10 is applied to the X input signals ai, a2 so as to generate N decorrelated signals and, i being an integer belonging to the interval [1, N].

[0034] Preferably, the number of decorrelated signals generated is equal to the maximum of (X, M). In the example of Figure 1, the number of decorrelated signals is four, N being equal to the maximum of (X, M) = max(2, 4).

[0035] The decorrelation filter 10 may comprise an all-pass filter, a whitening filter, or may comprise the application of random delays to the input signals ai, a2, preferably the random delays being less than 10 ms, or may comprise the application of a discrete cosine transform to the input signals a1, a2. Preferably, the decorrelation filter 10 is an all-pass filter.

[0036] Step a) can be implemented by means of a computer module 10.

[0037] Step b)

[0038] The decorrelated signals e1, e2, e3, e4 obtained at the end of step a) are then, in step b), delayed and amplified by applying a delay Tk and a gain gk:

[0039] [Math 5]

[0040] In the example of figure 1, we obtain for example the following delayed signals ei,k: for i=1: e 1,3 (t)= g3* e1(t-τ3) e 1,4(t)= g4* e1(t-τ4) for i=2: e 2,1 (t)= g1* e2(t-τ1) e 2,2 (t)= g2* e2(t-τ2) e 2,3 (t)= g3* e2(t-τ3) e 2,4 (t)= g4* e2(t-τ4); for i=3: e 3,1 (t)= g1* e3(t-τ1) e 3,2 (t)= g2* e3(t-τ2) e 3,3 (t)= g3* e3(t-τ3) e 3,4 (t)= g4* e3(t-τ4) and for i=4: e 4,1 (t)= g1* e4(t-τ1) e 4,2 (t)= g2* e4(t-τ2) e 4,3 (t)= g3* e4(t-τ3) e 4,4 (t)= g4* e4(t-τ4). Each delay τ k can be associated with a virtual speaker H k of virtual space. Similarly, each gain g k can be associated with a virtual speaker H k of virtual space. In other words, the delayed signals e 1,1 , e 2,1 , e 3,1 , e 4,1 result from a simulation of listening to the decorrelated signals e1, e2, e3, e4, respectively through the virtual loudspeakers H1, H2, H3, H4 of the virtual space, respectively. The delays τ k and the gains g kdepend on the geometry of the virtual space. To determine the delays τ k and the gains g k , it is therefore necessary to determine the geometry of the virtual space. Virtual space The determination of the virtual space can result from the determination of the position of the virtual loudspeakers and the virtual listening position of the virtual space. The determination of the virtual space can be determined so as to minimize the IACC. The determination of the virtual space can be determined so as to minimize the variations in frequency levels resulting from the frequency responses of the delayed signals e i , k . In a preferred embodiment, the determination of the virtual space is determined so as to minimize the IACC and the variations in frequency levels resulting from a frequency response of the delayed signals e i , k. Preferably, the virtual space is a resonance-free space, i.e. the virtual space does not include any obstacle reflecting the signal. In particular, the virtual space is advantageously not a partially closed or closed virtual room. The absence of resonance makes it possible, among other things, to preserve the original artistic intention. In a particular embodiment, a user can modify the number and / or the position of the virtual loudspeakers, and / or the virtual listening position in the virtual space. Preferably, several virtual spaces are pre-configured according to the desired number of virtual loudspeakers. Before step b), in particular before step a), the user can determine the geometry of the virtual space, for example by selecting a predetermined geometry from a database.Depending on the number of decorrelated signals N, a predetermined virtual space is selected from a set of virtual spaces based on the number of virtual loudspeakers of the virtual spaces in the set of virtual spaces, each virtual space in the set of virtual spaces having a unique number of virtual loudspeakers. Delays The delays τ. k can be determined as a function of the distance separating each virtual loudspeaker from the virtual listening position in the virtual space, as a function of the speed of propagation of sound in a vacuum c, as illustrated in Figures 2a and 2b. Figures 2a and 2b schematically represent a virtual space 3 comprising eight virtual loudspeakers 32 and a virtual listening position 34. Figure 2a represents a top view of the virtual space 3 and Figure 2b represents a side view of the virtual space 3. The values ​​of the τ can be determined. k as follows: τ k = d k / c, with c the speed of sound in a vacuum. For the example in Figures 2a and 2b, we define the distances d1, d2, d3, d4 separating respectively the virtual listening position 34 and the virtual loudspeakers H1, H2, H3 and H4, as a function of the parameter y. We can deduce the following Tk:

[0041] [Math 6] τ1 being associated with the virtual loudspeaker Hi and calculated as a function of the position of the virtual loudspeaker H1 relative to the virtual listening position 34, τ2 being associated with the virtual loudspeaker H2 and calculated as a function of the position of the virtual loudspeaker H2 relative to the virtual listening position 34, τ3 being associated with the virtual loudspeaker H3 and calculated as a function of the position of the virtual loudspeaker Hg relative to the virtual listening position 34, 14 being associated with the virtual loudspeaker H4 and calculated as a function of the position of the virtual loudspeaker H4 relative to the virtual listening position 34.

[0042] Earnings

[0043] In step b), the gains gk can be calculated as a function of the distance separating each virtual loudspeaker from the virtual listening position in the virtual space.

[0044] Gains can be determined by following the following criterion: the further the virtual speaker is from the virtual listening position, the lower the gain associated with that virtual speaker will be.

[0045] Gains can be inversely proportional to the distance between the virtual speakers and the virtual listening position.

[0046] A gain of 0 dB can be associated with the virtual loudspeaker closest to the virtual listening position.

[0047] The gains can be determined by applying an affine function, depending on the distance separating each virtual loudspeaker from the virtual listening position. The gains g kcan be calculated by applying a function f respecting for example the following relation: for any distance d, f(2d) = f(d) - 6 dB, preferably a gain of 0 dB being fixed for the distance separating the virtual loudspeaker closest to the virtual listening position and the virtual listening position. At the end of step b), we obtain N times the number of virtual loudspeakers in the virtual space delayed signals. For the example in figure 1, we obtain 16 delayed signals: e 1,1, e 1,2, e 1,3, e 1,4, e 2,1, e 2,2, e 2,3, e 2,4, e 3,1, e 3,2, e 3,3, e 3,4, e 4,1, e 4,2, e 4,3, e 4,4. Step b) can be implemented by means of a computer module 12. Preferably, the geometry of the virtual space and / or the gains and delays resulting from this geometry are stored in a database. Step c) In step c), the delay signals e are summedi,k according to the formula: [Math 7] Each E k correspond to a sum of the delayed signals resulting from the simulations of listening to the decorrelated signals e i through the virtual speaker H k of the virtual space. Step c) can be implemented by means of a computer module 14. Step d) In a preferred embodiment, the method according to the invention comprises the optional step d). The optional step d) can be carried out in parallel with step c). The optional step d) is carried out after step b). The optional step d) makes it possible to accentuate the feeling of immersion by multiplying the number of signals around the delayed signals. The larger J, the greater the feeling of immersion will be. The optional step d) comprises J duplication(s) of the delayed signals e i,k , by applying a delay ε i,k,j, J being an integer greater than or equal to 1. In one embodiment, J is equal to 1, enhancing the feeling of immersion while limiting the computing power required to generate these duplicate signals.

[0048] The duplicated signals obtained by applying a delay εi,k,j to a delayed signal oscillate around said delayed signal. The applied delay can be of the order of a hundred ps.

[0049] The duplicated signals are obtained by applying the following formula: [Math 8]

[0050] Optional step d) includes, in addition to the duplication of the delayed signals ei,k, the summation of the duplicated signals fi,k,j according to the following formula:

[0051] [Math 9]

[0052] Optional step d) can be thought of as adding J virtual speakers around the virtual speakers in the virtual space.

[0053] A gain can be applied to the duplicated signals fi,k,j and / or to the sums of duplicated signals Fkj.

[0054] In one embodiment, a single gain is applied to the duplicated signals fi,kj and / or to the sums of duplicated signals Fkj, depending on the desired sound level for the duplicated signals relative to the delayed signals and in particular for the sums of duplicated signals relative to the sums of delayed signals.

[0055] When step d) is implemented, the user can determine the number of duplications J to be performed in step d), before step b), in particular before step a).

[0056] Optional step d) may be implemented by means of a computer module 16.

[0057] Step e)

[0058] In step e), the sums of signals from step c), and optionally from step d), Ek, and optionally Fk,j respectively, are combined by linear combination. The coefficients of the linear combination can be determined manually by a user. Preferably, the coefficients are predetermined according to the desired intensity for each of the M output signals. The coefficients can depend on an actual installation configuration comprising M physical loudspeakers through which the M output signals are intended to be transmitted. In one embodiment, the user selects a desired enveloping sensation, this selection of desired enveloping sensation making it possible to fix the coefficients. For example, when the coefficients are all equal, an omnidirectional enveloping sensation is provided.Alternatively, the coefficients may be higher for output signals intended to be transmitted via the physical loudspeakers of the actual installation located in front of the user compared to the coefficients of the output signals intended to be transmitted via the physical loudspeakers of the actual installation located behind the user providing a frontal envelopment sensation. In particular, for a frontal envelopment sensation, a gain of 0 dB may be applied to the output signals intended to be transmitted via the physical loudspeakers of the actual installation located in front of the user, a gain of -6 dB may be applied to the output signals intended to be transmitted via the physical loudspeakers of the actual installation located on the sides, and a gain of -12 dB may be applied to the output signals intended to be transmitted via the physical loudspeakers of the actual installation located behind the user.The desired wrapping sensation may be selected from a lateral, frontal, rear, or omnidirectional wrapping sensation. The coefficients may be preselected, the preselected coefficients defining coefficients selected by default, in the absence of a selection by the user. Before step e), in particular before step a), the user may determine the coefficients of the linear combination for implementing step e), for example by selecting the coefficients from a database. Preferably, the selection of the coefficients is performed by selecting a desired wrapping sensation, for example from a database. Step e) may be implemented by means of a computer module 18. The coefficients may be stored in a database 200.Figure 3 illustrates another example of implementation of a signal processing method according to the invention, in which the number of input signals is equal to six (X=6) and the number of output signals is equal to two (M=2). The steps are similar to those described for the example of Figure 1. In this example, N is also equal to the maximum of M and X, here equal to 6. In this example, J is equal to 1, i.e. the delayed signals are duplicated once. Figure 4 schematically shows a system for implementing a method according to the invention.The system 2 comprises: - A computer 100 comprising a computer program comprising instructions which, when the program is executed by the computer, cause the latter to implement steps a), b), c), and e) of a signal processing method according to the invention, preferably to implement steps a), b), c), d) and e) of a signal processing method according to the invention; - A database 200 comprising at least one virtual space geometry 3 for implementing step b) of a method according to the invention, and / or at least one matrix determining the coefficients of a linear combination for implementing step e) of a method according to the invention; - Optionally M physical loudspeakers 300, intended to reproduce the M output signals. The database 200 may be integrated into the computer 100. The system may comprise means 400 for communicating the database with the computer 100.The computer is understood to be any computer means allowing the execution of the instructions. A method according to the invention can advantageously be used to transmit: - stereophonic musical content comprising two input signals via loudspeakers of a motor vehicle, or - stereophonic musical content comprising two input signals via loudspeakers of a home cinema system or a Dolby Atmos® system comprising at least six loudspeakers, better still eight loudspeakers, even better still ten loudspeakers, or - stereophonic musical content comprising two input signals in a concert hall preferably comprising four or more loudspeakers, or - audio content comprising at least four input signals, preferably six input signals, via headphones comprising two loudspeakers. These examples of use are not limiting. Of course, the invention is not limited to the exemplary embodiments which have just been described.In particular, the number of input signals and the number of output signals can vary. They advantageously depend on the desired use. The invention makes it possible, among other things, to guarantee improved diffusion of the X input signals, i.e. diffusion preserving the integrity of the signals and limiting artifacts. Unlike the known state of the art, the invention does not aim to reproduce the acoustics of a room or a real place. Indeed, the addition of early reflections and / or late reverberation, although having an impact on the feeling of immersion, modifies the original artistic intention. In particular, early reflections add a color corresponding to the geometry of the room and the acoustic quality of the walls of said room. A sound to which such reflections are added can then become muffled or, on the contrary, very clear.Late reverb processing modifies the resonance and perceived depth of sound. A short, percussive sound to which late reverb is added is perceived as long and drowned in reverberation. In addition, pre- and / or post-processing can be applied to the input and / or output signals, respectively. For example, equalizer filters, Finite Impulse Response filters, or Biquad Infinite Impulse Response filters can be applied to the input signals. p and / or output s q allowing to modify the tonal color of the input signals a p and / or output s q respectively.

Claims

Demands 1. Signal processing method transforming X input signals a p , p being an integer belonging to the interval [1, X], in M ​​output signals s q , q being an integer belonging to the interval [2, M], M being different from X, the process comprising the following steps: a) application of a decorrelation filter (10) to the X input signals a pso as to generate N uncorrelated signals, i being an integer belonging to the interval [1, N]; b) for each uncorrelated signal determined in step a), generation of N delayed signals ei,k by applying a delay Tk and a gain gk to each uncorrelated signal, k being an integer belonging to the interval [1, N], the delays Tk and the gains gk being chosen so that for each uncorrelated signal, the delayed signals ei,k simulate a propagation of the uncorrelated signal in a virtual space (3), the virtual space comprising N virtual loudspeakers (32) and a virtual listening position (34) distributed according to a predetermined geometry, the virtual space being a resonance-free space, c) for each integer k belonging to the interval [1, N], summation of the delayed signals ei,k according to the following formula: d) duplication of the delayed signals ei,k determined in step b), and for each duplicated signal ei,kj, j being an integer greater than or equal to 1, application of a delay εi,k,j, then summation of the delayed duplicated signals ei,k(t-εi,k,j), according to the following formula: for each integer k in the interval [1, N], for each j, e) determination of M output signals, each output signal s q resulting from a linear combination of the N sums of delayed signals resulting from step c), Ek, and, when a step d) is implemented, of the j*N sums of duplicated signals resulting from step d), Fkj.

2. Method according to the preceding claim, the virtual space (3) being a resonance-free space.

3. A method according to any one of the preceding claims, wherein the predetermined geometry of the virtual space (3) is determined so as to minimize an internal cross-correlation coefficient.

4. A method according to any one of the preceding claims, wherein the predetermined geometry of the virtual space (3) is determined so as to minimize frequency level variations resulting from the frequency responses of the delayed signals.

5. A method according to any one of the preceding claims, wherein the decorrelation filter comprises an all-pass filter, or a whitening filter, or the application of random delays to the X input signals. p , or the application of a discrete cosine transform to the input signals a p6. A method according to any one of the preceding claims, wherein the linear combination in step e) comprises coefficients determined according to a desired surround sound sensation, the desired surround sound sensation being chosen from a frontal, lateral, rear, or omnidirectional surround sound sensation.

7. A method according to any one of the preceding claims, wherein N is an integer within the interval [1, maximum (X, M)], preferably N is equal to the maximum of X and M.

8. A method according to any one of the preceding claims, comprising, before step a), the selection of a predetermined geometry of the virtual space (3), preferably, the selection of the predetermined geometry being carried out by specifying a number of virtual loudspeakers in the virtual space (32). 9.A method according to any one of the preceding claims, comprising, before step a), the selection of coefficients for the linear combination in step e), preferably by selecting a desired immersion sensation, the linear combination in step e) comprising coefficients determined according to a desired immersion sensation, the desired immersion sensation being chosen from a frontal, lateral, rear, or omnidirectional immersion sensation.

10. Use of a method according to any one of the preceding claims for emitting: - stereophonic musical content comprising two input signals via loudspeakers of a motor vehicle, or. 22 - stereophonic music content comprising two input signals via speakers of a home cinema system or a Dolby Atmos® system comprising at least six speakers, preferably at least eight speakers, even better at least ten speakers, or - stereophonic music content comprising two input signals in a concert hall comprising four or more speakers, or - audio content comprising at least four input signals, preferably six input signals, via headphones comprising two speakers.

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