EP3844981B1 - Method for the spatial sound reproduction of a sound field that is audible in a position of a moving listener and system implementing such a method - Google Patents

Description

The invention is placed in the field of spatial audio and sound field control. The method aims to restore at least one sound field in a zone, for a listener, depending on the position of the listener. In particular, the process aims to restore the sound field by taking into account the movements of the listener.

The area is covered by an array of loudspeakers, powered by respective control signals to each emit a continuous audio signal. A respective weight is applied to each loudspeaker control signal in order to reproduce the sound field according to the listener's position. From the weights, a set of filters is determined, each filter of the filter set corresponding to each loudspeaker. The signal to be distributed to the listener is then filtered by the filter set and broadcast by the loudspeaker corresponding to the filter.

The iterative methods used use the weights calculated in the previous iteration to calculate the new weights. The set of filters therefore has a memory of previous iterations. When the listener moves, part of the sound field that was rendered in the previous iteration (or at the listener's old position) is missing from the new listener position. It is therefore no longer constrained and the part of the weights allowing this previous restitution is no longer useful but remains in memory. In other words, the sound field restored at the previous position of the listener, at the previous iteration, is no longer useful for calculating the weights at the current position of the listener, or at the current iteration, but remains in effect. memory.

It is particularly known from the document WO2012/068174 A2 a method and system for producing a localized binaural audio signal for a user. However, this document does not take into account an auditor's current position.

The present invention improves the situation.

For this purpose, it proposes a method assisted by computer means according to claim 1.

The present method is therefore based directly on the movement of the listener to vary the forgetting factor at each iteration. This helps mitigate the memory effect due to the calculation of weights in previous iterations. The precision of field restitution is greatly improved, while not requiring excessively expensive computing resources.

According to one embodiment, a plurality of points forming the respective positions of a plurality of virtual microphones are defined in the zone to estimate a plurality of respective acoustic pressures in the zone taking into account the respective weight applied to each loudspeaker, each comprising respectively a forgetting factor, and transfer functions specific to each speaker in each virtual microphone, the plurality of points being centered on the position of the listener.

In this way, the sound pressure is estimated at a plurality of points in the area surrounding the listener. This makes it possible to apply weights to each loudspeaker taking into account the differences in sound pressures that may occur at different points in the area. The estimation of acoustic pressures is therefore carried out in a homogeneous and precise manner around the listener, which makes it possible to increase the precision of the method.

• une estimation d'une pression acoustique dans la deuxième sous-zone, au moins en fonction des fonctions de transfert acoustiques, des signaux de commande respectifs des haut-parleurs, et d'un poids initial respectif des signaux de commande des haut-parleurs ;
• calcul d'une erreur entre ladite pression acoustique estimée dans la deuxième sous-zone et une pression acoustique cible, souhaitée dans la deuxième sous-zone ;
• calcul et application de poids respectifs aux signaux de commande des haut-parleurs, en fonction de ladite erreur et d'un facteur d'oubli de poids, ledit facteur d'oubli étant calculé en fonction d'un déplacement de l'auditeur, ledit déplacement étant déterminé par une comparaison entre une position précédente de l'auditeur et la position courante de l'auditeur ;
  le calcul de la pression acoustique dans la deuxième sous-zone étant mis en oeuvre à nouveau en fonction des signaux de commande respectifs, pondérés, des haut-parleurs.

According to one embodiment, the zone comprises a first sub-zone in which the chosen sound field is to be made audible and a second sub-zone in which the chosen sound field is to be made inaudible, the first sub-zone being defined dynamically as corresponding to the position of the listener and said virtual microphone, the virtual microphone being a first virtual microphone, and the second sub-zone being defined dynamically as being complementary to the first sub-zone, the second sub-zone being covered by at least one second virtual microphone whose position is dynamically defined as a function of said second sub-zone, the method further comprising iteratively:

• an estimation of a sound pressure in the second sub-zone, at least as a function of the acoustic transfer functions, of the respective control signals of the loudspeakers, and of a respective initial weight of the loudspeaker control signals;
• calculation of an error between said estimated sound pressure in the second sub-zone and a target sound pressure, desired in the second sub-zone;
• calculation and application of respective weights to the loudspeaker control signals, as a function of said error and a weight forgetting factor, said forgetting factor being calculated as a function of a movement of the listener, said displacement being determined by a comparison between a previous position of the listener and the current position of the listener;
  the calculation of the sound pressure in the second sub-zone being implemented again as a function of the respective, weighted control signals from the loudspeakers.

The method therefore makes it possible to reproduce different sound fields in the same area, using the same loudspeaker system, depending on the movement of the listener. Thus, at each iteration, the sound field actually restored in the two sub-zones is evaluated so that at each movement of the listener, the sound pressure in each of the sub-zones actually reaches the target sound pressure. The position of the listener can help determine the sub-area in which the field sound is to be made audible. The sub-zone in which the sound field is to be made inaudible is then dynamically defined each time the listener moves. The forgetting factor is therefore calculated iteratively for each of the two sub-zones, so that the sound pressure in each of the sub-zones reaches its target sound pressure.

• une estimation d'une pression acoustique dans la deuxième sous-zone, au moins en fonction des fonctions de transfert acoustiques, des signaux de commande respectifs des haut-parleurs, et d'un poids initial respectif des signaux de commande des haut-parleurs ;
• calcul d'une erreur entre ladite pression acoustique estimée dans la deuxième sous-zone et une pression acoustique cible, souhaitée dans la deuxième sous-zone ;
• calcul et application de poids respectifs aux signaux de commande des haut-parleurs, en fonction de ladite erreur et d'un facteur d'oubli de poids, ledit facteur d'oubli étant calculé en fonction d'un déplacement de l'auditeur, ledit déplacement étant déterminé par une comparaison entre une position précédente de l'auditeur et la position courante de l'auditeur ;

le calcul de la pression acoustique dans la deuxième sous-zone étant mis en oeuvre à nouveau en fonction des signaux de commande respectifs, pondérés, des haut-parleurs.According to one embodiment, the zone comprises a first sub-zone in which the chosen sound field is to be made audible and a second sub-zone in which the chosen sound field is to be made inaudible, the second sub-zone being defined dynamically as corresponding to the position of the listener and said virtual microphone, the virtual microphone being a first virtual microphone, and the first sub-zone being defined dynamically as being complementary to the second sub-zone, the first sub-zone being covered by at least one second virtual microphone whose position is dynamically defined as a function of said first sub-zone, the method further comprising iteratively:

• an estimation of a sound pressure in the second sub-zone, at least as a function of the acoustic transfer functions, of the respective control signals of the loudspeakers, and of a respective initial weight of the loudspeaker control signals;
• calculation of an error between said estimated sound pressure in the second sub-zone and a target sound pressure, desired in the second sub-zone;
• calculation and application of respective weights to the loudspeaker control signals, as a function of said error and a weight forgetting factor, said forgetting factor being calculated as a function of a movement of the listener, said displacement being determined by a comparison between a previous position of the listener and the current position of the listener;

the calculation of the sound pressure in the second sub-zone being implemented again as a function of the respective, weighted control signals from the loudspeakers.

Likewise, the position of the listener can make it possible to define the sub-zone in which the sound field is to be made inaudible. The sub-zone in which the sound field is to be made audible being defined dynamically as complementary to the other sub-zone. The forgetting factor is therefore calculated iteratively for each of the two sub-zones, so that the sound pressure in each of the sub-zones reaches its target sound pressure.

According to one embodiment, each sub-zone comprises at least one virtual microphone and two speakers, and preferably each sub-zone comprises at least ten virtual microphones and at least ten speakers.

The method is therefore capable of operating with a plurality of microphones and speakers.

According to one embodiment, a value of the forgetting factor increases if the listener moves and decreases if the listener does not move.

Increasing the forgetting factor when the listener moves makes it possible to forget the weights calculated in previous iterations more quickly. On the contrary, reducing the forgetting factor when the listener does not move makes it possible to preserve at least in part the weights calculated in previous iterations.

avec γ(n) le facteur d'oubli, n l'itération courante, γ_max le facteur d'oubli maximal, χ un paramètre défini par le concepteur égal à µ un pas d'adaptation, m une variable définie en fonction d'un déplacement de l'auditeur ayant comme maximum χ et α une variable permettant d'ajuster la vitesse d'augmentation ou de diminution du facteur d'oubli.According to one embodiment, the forgetting factor is defined by

with γ(n) the forgetting factor, n the current iteration, γ _max the maximum forgetting factor, χ a parameter defined by the designer equal to µ an adaptation step, m a variable defined as a function of a movement of the listener having as maximum χ and α a variable making it possible to adjust the speed of increase or decrease of the forgetting factor.

Thus, the forgetting factor is directly estimated as a function of a movement of the listener. In particular, the forgetting factor depends on the distance traveled in each iteration by the listener, in other words on the speed of movement of the listener. A different forgetting factor can therefore be estimated for each listener. Variable values can also be adjusted during iterations to truly account for listener movement.

• si un déplacement de l'auditeur est déterminé, m = min(m + l_u, 1)
• si aucun déplacement de l'auditeur n'est déterminé, m = max(m - l_d, 0),

avec 0< l_u <1 et 0< l_d <1, les pas de montée et de descente étant définis en fonction d'une vitesse de déplacement d'un auditeur et/ou d'une modification du champ sonore choisi à restituer.According to one embodiment, a step of rise l _u and a step of fall l _d of the forgetting factor are defined as:

• if a movement of the listener is determined, m = min(m + l _u , 1)
• if no movement of the listener is determined, m = max(m - l _d , 0),

with 0< l _u <1 and 0< l _d <1, the rise and fall steps being defined as a function of a listener's movement speed and/or a modification of the sound field chosen to be reproduced.

The definition of two distinct variables l _u and l _d makes it possible to choose the reaction speeds of the method according to the start and/or end of the listener's movement.

According to one embodiment, the forgetting factor is between 0 and 1.

Thus, this allows you to forget the previous weights in full or to keep the previous weights in full.

The present invention also relates to a spatialized sound reproduction system according to claim 10.

The present invention also relates to a storage medium for a computer program according to claim 11.

• la figure 1 représente un exemple de système selon un mode de réalisation de l'invention,
• les figures 2a et 2b illustrent, sous la forme d'un ordinogramme, les principales étapes d'un mode de réalisation particulier du procédé,
• la figure 3 illustre de façon schématique un mode de réalisation dans lequel deux sous-zones sont définies dynamiquement en fonction des données de géolocalisation d'un auditeur,
• les figures 4a et 4b illustrent, sous la forme d'un ordinogramme, les principales étapes d'un deuxième mode de réalisation du procédé.

Other advantages and characteristics of the invention will appear on reading the detailed description below of examples of embodiment of the invention, and on examining the appended drawings in which:

• there figure 1 represents an example of a system according to one embodiment of the invention,
• THE figures 2a and 2b illustrate, in the form of a flowchart, the main stages of a particular embodiment of the process,
• there Figure 3 schematically illustrates an embodiment in which two sub-zones are dynamically defined according to the geolocation data of a listener,
• THE figures 4a and 4b illustrate, in the form of a flowchart, the main stages of a second embodiment of the method.

The embodiments described with reference to the figures can be combined.

There figure 1 schematically illustrates a SYST system according to an exemplary embodiment. The SYST system comprises a network of HP loudspeakers comprising N loudspeakers (HP ₁ ,..., HP _N ), with N at least equal to 2, and preferably at least equal to 10. The loudspeaker network HP speakers cover a zone Z. The HP speakers are powered by respective control signals to each emit an audio signal continuously, with a view to spatialized sound diffusion of a sound field chosen in zone Z. More precisely , the chosen sound field is to be reproduced in a position a1 of a listener U. The speakers can be defined by their position in the zone. The position a1 of listener U can be obtained by means of a position sensor POS.

The area is additionally covered by MIC microphones. In an exemplary embodiment, the area is covered by a network of M MIC microphones, with M at least equal to 1 and preferably at least equal to 10. The MIC microphones are virtual microphones. In the remainder of the description, the term “MIC microphone” is used. MIC microphones are identified by their position in zone Z.

In an exemplary embodiment, the virtual microphones are defined as a function of the position a1 of the listener U in zone Z. In particular, the virtual microphones MIC can be defined so as to surround the listener U. In this exemplary embodiment , the position of the virtual microphones MIC changes according to the position a1 of the listener U.

As illustrated on the figure 1 , the microphone array MIC surrounds position a1 of listener U. Then, when listener U moves to position a2, the array of MIC microphones are redefined to surround the listener's A2 position. The movement of the listener U is represented schematically by the arrow F.

The SYST system further comprises a TRAIT processing unit capable of implementing the steps of the process. The TRAIT processing unit comprises in particular a memory, forming a storage medium for a computer program comprising portions of code for implementing the method described below with reference to the figures 2a and 2b . The TRAIT processing unit further comprises a PROC processor capable of executing the code portions of the computer program.

The TRAIT processing unit receives, continuously and in real time, the position of the MIC microphones, the position of the listener U, the positions of each speaker HP, the audio signal to be reproduced S(U) intended for the the listener U and the target sound field P _t to be reached at the listener's position. The TRAIT processing unit also receives the estimated sound pressure P in the position of the listener U. From these data, the TRAIT processing unit calculates the FILT filter to be applied to the signal S in order to restore the sound field target P _t . The TRAIT processing unit outputs the filtered signals S(HP ₁ ...HP _N ) to be broadcast respectively on the speakers HP ₁ to HP _N.

THE figures 2a and 2b illustrate the main steps of a process for the restitution of a chosen sound field in a position of a listener, when the listener moves. The steps of the process are implemented by the TRAIT processing unit continuously and in real time.

In step S1, the position of the listener U in the area is obtained by means of a position sensor. From this geolocation data, a network of virtual microphones MIC is defined in step S2. The network of virtual microphones MIC can take any geometric shape such as a square, a circle, a rectangle, etc. The network of virtual microphones MIC can be centered around the position of the listener U. The network of virtual microphones MIC can be centered around the position of the listener U. The network of virtual microphones MIC defined for example a perimeter of a few tens of centimeters to a few tens of meters around the listener U. The network of virtual microphones MIC comprises at least two virtual microphones, and preferably at least ten virtual microphones. The number of virtual microphones as well as their arrangement define limits in the quality of restitution of the zone.

In step S3, the position of each HP speaker is determined. Notably, the zone includes a speaker array comprising at least two HP speakers. Preferably, the speaker network includes around ten HP speakers. The HP speakers can be distributed across the area so that the entire area is covered by the speakers.

In step S4, a distance between each pair of HP loudspeaker and MIC microphone is calculated. This makes it possible to calculate each of the transfer functions Ftransf, for each HP loudspeaker/MIC microphone pair, in step S5.

More precisely, the target sound field can be defined as a vector Pt ( ω , n ) for all of the MIC microphones, at each time n for a pulsation ω = 2 πf, f being the frequency. The virtual microphones MICi to MIC _M of the network of virtual microphones are arranged at positions x _MIC = [ MIC ₁ , ..., MIC _M ] and capture a set of acoustic pressures grouped in the vector P ( ω , n ).

The sound field is reproduced by the speakers (HP ₁ ,...,HP _N ) fixed and having the respective position x _HP = [ HP ₁ ,..., HP _N ]. The loudspeakers (HP ₁ ,...,HP _N ) are driven by a set of weights grouped in the vector q ( ω , n ) = [q ₁ ( ω, n ) ,...,q _N ( ω , n )] ^T . The exponent ^T is the transposition operator.

Avec les fonctions de transfert définies comme étant : $G_{\mathit{ml}}=\frac{\mathit{j\rho ck}}{4{\mathit{\pi R}}_{\mathit{ml}}}{e}^{-{\mathit{jkR}}_{\mathit{ml}}}$

, avec R_ml la distance entre un couple haut-parleur et microphone, k le nombre d'onde, ρ la masse volumique de l'air et c la célérité du son.The sound field propagation path between each pair of HP loudspeaker and MIC microphone can be defined by a set of transfer functions G ( ω , n ) assembled in the matrix $$\mathbf{G}\left(\omega ,\mathrm{not}\right)=\left[\begin{array}{ccc}{G}_{11}\left(\omega ,\mathrm{not}\right)& \cdots & {\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0001.png,imgf0005.png"\; state="\{\{state\}\}">G</figure-callout>}}_{1,\mathrm{NOT}}\left(\omega ,\mathrm{not}\right)\\ ⋮& \ddots & ⋮\\ {\mathrm{<figure-callout\; id="1"\; label="G\; M"\; filenames="imgf0001.png,imgf0005.png"\; state="\{\{state\}\}">G</figure-callout>}}_M\left(\omega ,\mathrm{not}\right)& \cdots & G_{\mathit{M.N.}}\left(\omega ,\mathrm{not}\right)\end{array}\right]$$

With the transfer functions defined as: $G_{\mathit{ml}}=\frac{\mathit{j\rho ck}}{4{\mathit{\pi R}}_{\mathit{ml}}}{e}^{-{\mathit{jkR}}_{\mathit{ml}}}$

, with R _ml the distance between a loudspeaker and microphone pair, k the wave number, ρ the density of the air and c the speed of sound.

In step S6, the acoustic pressure P is determined in the position of the listener U. More precisely, the acoustic pressure P is determined in the perimeter defined by the network of virtual microphones MIC. Even more precisely, the acoustic pressure P is determined at each virtual microphone. The sound pressure P is the sound pressure from the signals broadcast by the speakers in the area. The sound pressure P is determined from the transfer functions Ftransf, calculated in step S5, and a weight applied to the control signals supplying each loudspeaker. The initial weight applied to the control signals of each of the loudspeakers is equal to zero. This corresponds to the weight applied to the first iteration. Then, with each new iteration, the weight applied to the control signals tends to vary, as described below.

In this example, the acoustic pressure P includes all of the acoustic pressures determined at each of the positions of the virtual microphones. Thus, the sound pressure estimated at the listener's position U is more representative. This makes it possible to obtain a homogeneous result at the end of the process.

Step S7 makes it possible to define the value of the target acoustic pressure Pt at the position of the listener U. More precisely, the value of the target acoustic pressure Pt is initialized at this step. The target sound pressure Pt can be chosen by the designer. It is then transmitted to the TRAIT processing unit in the form of the vector defined above.

In step S8, the error between the target pressure Pt and the estimated pressure P at the listener's position U is calculated. The error may be due to the fact that an adaptation step µ is applied so that the target pressure Pt is not reached immediately. The target pressure Pt is reached after a certain number of iterations of the process. This makes it possible to minimize the computational resources necessary to reach the target pressure at the position of the listener U. This also ensures the stability of the algorithm. In the same way, the adaptation step µ is also chosen so that the error calculated in step S8 has a small value, in order to stabilize the filter.

The error E(n) is calculated as follows: $$E\left(\mathrm{not}\right)=\mathbf{G}\left(\mathbf{not}\right)\mathbf{q}\left(\mathrm{not}\right)-{\mathbf{p}}_T\left(\mathrm{not}\right)=\mathbf{p}\left(\mathrm{not}\right)-{\mathbf{p}}_T\left(\mathrm{not}\right)$$

In step S12, the forgetting factor γ(n) is calculated in order to calculate the weights to be applied to each loudspeaker control signal.

The forgetting factor γ(n) has two roles. On the one hand, it makes it possible to regularize the problem. In other words, it prevents the process from diverging when it is in a stationary state.

On the other hand, the forgetting factor γ(n) makes it possible to attenuate the weights calculated in previous iterations. Thus, when the listener moves, previous weights do not influence future weights.

The forgetting factor γ(n) is determined based directly on a possible movement of the listener. This calculation is illustrated in steps S9 to S11. In step S9, the position of the listener in previous iterations is recovered. For example, it is possible to recover the position of the listener in all previous iterations. Alternatively, it is possible to recover the position of the listener only for part of the previous iterations, for example the last ten or the last hundred iterations.

From this data, a movement speed of the listener is calculated in step S10. Movement speed can be calculated in meters per iteration. The listener's speed may be zero.

avec γ le facteur d'oubli, n l'itération courante, γ _max le facteur d'oubli maximal, χ un paramètre défini par le concepteur égal à µ le pas d'adaptation, m une variable définie en fonction d'un déplacement de l'auditeur ayant comme maximum χ et α une variable permettant d'ajuster la vitesse d'augmentation ou de diminution du facteur d'oubli.In step S 11, the forgetting factor γ(n) is calculated according to the formula:

with γ the forgetting factor, n the current iteration, γ _{max x} the maximum forgetting factor, χ a parameter defined by the designer equal to µ the adaptation step, m a variable defined as a function of a displacement of the listener having as maximum χ and α a variable making it possible to adjust the speed of increase or decrease of the forgetting factor.

The forgetting factor γ is bounded between 0 and γ _{ma x} . According to this definition, γ _{ma x} therefore corresponds to a maximum percentage of weight to be forgotten between each iteration.

The choice of the value of m varies during the iterations. It is chosen such that if there is a movement of the listener, then the forgetting factor increases. When there is no movement, it decreases. In other words, when the listener's speed is positive, the forgetting factor increases and when the listener's speed is zero it decreases.

The variable α mainly influences the speed of convergence of the process. In other words, it allows you to choose the number of iterations for which the maximum value γ _{ma x} and/or minimum value of the forgetting factor is reached.

• si un déplacement de l'auditeur est déterminé, m = min(m + l_u, 1)
• si aucun déplacement de l'auditeur n'est déterminé, m = max(m - l_d , 0).

The variable m is defined as follows:

• if a movement of the listener is determined, m = min( m + l _u , 1)
• if no movement of the listener is determined, m = max( m - l _d , 0) .

The variables l _u and l _d correspond respectively to a step of rise and a step of fall of the forgetting factor. They are defined according to the speed of movement of the listener and/or according to a modification of the sound field chosen to be reproduced.

In particular, the rise step l _u has a greater value if the previous weights are to be forgotten quickly during movement (for example in the case where the speed of movement of the listener is high). The descent step l _d has a greater value if the previous weights are to be completely forgotten at the end of a movement by the listener.

The definition of two variables l _u and l _d therefore makes it possible to modulate the system. This makes it possible to take into account, in real time and continuously, the movement of the listener. Thus, at each iteration, the forgetting factor is calculated according to the actual movement of the listener, so as to restore the chosen sound field in the position of the listener.

In step S12, the forgetting factor γ is modified if necessary, depending on the result of the calculation of step S 11.

The calculation and modification of the forgetting factor in step S12 is used to calculate the weights to be applied to the control signals of the HP speakers. More precisely, at the first iteration, the weights are initialized to zero (step S13). Each loudspeaker broadcasts an unweighted control signal. Then, at each iteration, the value of the weights varies according to the error and the forgetting factor (step S14). The speakers then broadcast a weighted control signal, which can be different with each new iteration. This modification of the control signals explains in particular why the acoustic pressure P estimated at the position of the listener U can be different at each iteration.

The new weights are calculated in step S14 according to the mathematical formula: q (n + 1) = q (n)(1 - µγ ( n )) + µ G ^H ( n ) ( G ( n ) q ( n ) - Pt ( n )), with µ the adaptation step which can vary at each iteration and γ ( n ) the forgetting factor which can vary. In order to guarantee the stability of the filter, it is advantageous to prevent the adaptation step µ from being greater than the inverse of the largest eigenvalue of G ^H G .

In step S15, the FILT filters to be applied to the speakers are calculated. For example, one filter per speaker is calculated. There can therefore be as many filters as speakers. To obtain filters in the time domain from the weights calculated in the previous step, it is possible to perform a symmetry of the weights calculated in the frequency domain by taking their conjugate complex. Then, an Inverse Fourier Transform is performed to obtain the filters in the time domain. However, the calculated filters may not respect the principle of causality. A time shift of the filter, corresponding for example to half the filter length, can be carried out. Thus, a plurality of filters, for example one filter per loudspeaker, is obtained.

In step S16, the audio signal to be broadcast to the listener is obtained. It is then possible to carry out real-time filtering of the audio signal S(U) to broadcast the signal to the speakers. In particular, the signal S(U) is filtered in step S17 by the filters calculated in step S15 and broadcast by the loudspeaker corresponding to the filter in steps S18 and S19.

Then, at each iteration, the FILT filters are calculated as a function of the filtered signals S(HP ₁ ,...,HP _N ), weighted at the previous iteration and broadcast by the speakers, as perceived by the network of microphones. The FILT filters are applied to the signal S(U) to obtain new control signals S(HP ₁ ,...,HP _N ) to be broadcast respectively to each loudspeaker of the loudspeaker network.

The process is then restarted from step S6 in which the sound pressure at the listener's position is determined.

Below, another embodiment is described. The same numerical references designate the same elements.

In this embodiment, the HP loudspeaker network covers an area comprising a first sub-zone SZ1 and a second sub-zone SZ2. The HP speakers are powered by respective control signals to each emit an audio signal continuously, with a view to spatialized sound diffusion of a chosen sound field. The chosen sound field is to be made audible in one of the sub-zones, and to be made inaudible in the other sub-zone. For example, the chosen sound field is audible in the first subzone SZ1. The chosen sound field must be made inaudible in the second sub-zone SZ2. Speakers can be defined by their position in the zone.

Each subzone SZ can be defined by the position of the listener U. It is then possible to define, based on the geolocation data of the listener, the first subzone SZ1, in which the listener U hears the chosen sound field. The SZ1 subzone has, for example, predefined dimensions. In particular, the first sub-zone can correspond to a surface of a few tens of centimeters to a few tens of meters, of which the listener U is the center. The second subzone SZ2, in which the chosen sound field is to be made inaudible, can be defined as the complementary subzone.

Alternatively, the position of the listener U can define, in the same manner as described above, the second subzone SZ2. The first subzone SZ1 is defined as complementary to the second subzone SZ2.

According to this embodiment, part of the microphone network MIC covers the first sub-zone SZ1 while the other part covers the second sub-zone SZ2. Each subzone includes at least one virtual microphone. For example, the area is covered by M microphones M1 to MIC _M. The first sub-zone is covered by the microphones MICi to MIC _N , with N less than M. The second sub-zone is covered by the microphones MIC _N+1 to MIC _M.

The sub-zones being defined according to the position of the listener, they evolve as the listener moves. The position of the virtual microphones changes in the same way.

More precisely, and as illustrated in the Figure 3 , the first subzone SZ1 is defined by the position a1 of the listener U (represented in solid lines). The MIC microphone array is defined to cover the first sub-zone SZ1. The second subzone SZ2 is complementary to the first subzone SZ1. The arrow F illustrates a movement of the listener U towards a position a2. The first subzone SZ1 is then redefined around the listener U (in dotted lines). The MIC microphone array is redefined to cover the new first subzone SZ1. The rest of the zone represents the new second subzone SZ2. Thus, the first subzone SZ1 initially defined by the position a1 of the listener is found in the second subzone SZ2.

Thus, on the system illustrated Figure 3 , the TRAIT processing unit receives as input the position of the microphones MIC, the geolocation data of the listener U, the positions of each loudspeaker HP, the audio signal to be reproduced S(U) intended for the listener U and the target sound fields Pt ₁ , Pt ₂ to be achieved in each sub-zone. From this data, the TRAIT processing unit calculates the FILT filter to be applied to the signal S(U) in order to restore the target sound fields Pt ₁ , Pt ₂ in the sub-zones. The TRAIT processing unit also receives the acoustic pressures P ₁ , P ₂ estimated in each of the sub-zones. The TRAIT processing unit outputs the filtered signals S(HP ₁ ...HP _N ) to be broadcast respectively on the speakers HP ₁ to HP _N.

THE figures 4a and 4b illustrate the main stages of the process according to the invention. The steps of the process are implemented by the TRAIT processing unit continuously and in real time.

The purpose of the method is to make the sound field chosen in one of the sub-zones inaudible, for example in the second sub-zone SZ2 while following the movement of a listener whose position defines the sub-zones. The method is based on an estimation of acoustic pressures in each of the sub-zones, so as to apply a desired level of sound contrast between the two sub-zones. At each iteration, the audio signal S(U) is filtered as a function of the estimated acoustic pressures and the sound contrast level to obtain the control signals S(HP ₁ ...HP _N ) to be broadcast on the speakers.

In step S20, the position of the listener U is determined, for example by means of a position sensor POS. From this position, the two sub-zones SZ1, SZ2 are defined. For example, the first subzone corresponds to the position of the listener U. The first subzone SZ1 is for example defined as being an area of a few tens of centimeters to a few tens of meters in circumference, including the first listener U1 is the center. The second subzone SZ2 can be defined as being complementary to the first subzone SZ1.

Alternatively, it is the second subzone SZ2 which is defined by the position of the listener, the first subzone SZ1 being complementary to the second subzone SZ2.

In step S21, the network of MIC microphones is defined, at least one microphone covering each of the sub-zones SZ1, SZ2.

In step S22, the position of each HP speaker is determined, as described above with reference to the figures 2a and 2b .

In step S23, a distance between each pair of HP loudspeaker and MIC microphone is calculated. This makes it possible to calculate each of the transfer functions Ftransf, for each HP loudspeaker/MIC microphone pair, in step S4.

, pour l'ensembles des microphones MIC, à chaque instant n pour une pulsation ω = 2πf, f étant la fréquence. Les microphones MIC₁ à MIC_M sont disposés aux positions x_MIC = [MIC ₁ ,...,MIC_M ] et capturent un ensemble de pressions acoustiques regroupés dans le vecteur P(ω, n).More precisely, the target sound field can be defined as a vector $\mathbf{Pt}\left(\omega ,\mathrm{not}\right)=\left[\begin{array}{c}\mathit{P}{\mathit{t}}_1\\ \mathit{P}{\mathit{t}}_2\end{array}\right]$

, for all of the MIC microphones, at each instant n for a pulsation ω = 2 πf, f being the frequency. The microphones MIC ₁ to MIC _M are arranged at positions x _MIC = [ MIC ₁ , ..., MIC _M ] and capture a set of acoustic pressures grouped in the vector P ( ω, n ).

The sound field is reproduced by the speakers (HP ₁ ,...,HP _N ) fixed and having the respective position x _HP = [ HP ₁ ,...,HP _N ]. The loudspeakers (HP ₁ ,...,HP _N ) are driven by a set of weights grouped in the vector q ( ω , n ) = [q ₁ ( ω, n ) ,...,q _N ( ω , n )] ^T . The exponent ^T is the transposition operator.

The sound field propagation path between each pair of HP loudspeaker and MIC microphone can be defined by a set of transfer functions G ( ω , n ) assembled in the matrix $$\mathbf{G}\left(\omega ,\mathrm{not}\right)=\left[\begin{array}{ccc}{G}_{11}\left(\omega ,\mathrm{not}\right)& \cdots & {\mathrm{<figure-callout\; id="1"\; label="G"\; filenames="imgf0001.png,imgf0005.png"\; state="\{\{state\}\}">G</figure-callout>}}_{1\mathrm{NOT}}\left(\omega ,\mathrm{not}\right)\\ ⋮& \ddots & ⋮\\ {\mathrm{<figure-callout\; id="1"\; label="G\; M"\; filenames="imgf0001.png,imgf0005.png"\; state="\{\{state\}\}">G</figure-callout>}}_M\left(\omega ,\mathrm{not}\right)& \cdots & G_{\mathit{M.N.}}\left(\omega ,\mathrm{not}\right)\end{array}\right]$$

, avec R_ml la distance entre un couple haut-parleur et microphone, k le nombre d'onde, ρ la masse volumique de l'air et c la célérité du son.With the transfer functions defined as: $G_{\mathit{ml}}=\frac{\mathit{j\rho ck}}{4{\mathit{\pi R}}_{\mathit{ml}}}{e}^{-{\mathit{jkR}}_{\mathit{ml}}}$

, with R _ml the distance between a loudspeaker and microphone pair, k the wave number, ρ the density of the air and c the speed of sound.

In step S25, the acoustic pressures P ₁ and P ₂ are determined respectively in the first sub-zone SZ1 and in the second sub-zone SZ2.

According to an exemplary embodiment, the acoustic pressure P ₁ in the first sub-zone SZ1 can be the acoustic pressure resulting from the signals broadcast by the speakers in the first sub-zone. The acoustic pressure P ₂ in the second sub-zone, in which the sound signals are to be made inaudible, can correspond to the induced acoustic pressure resulting from the signals broadcast by the loudspeakers supplied by the control signals associated with the pressure P ₁ induced in the first subzone.

The acoustic pressures P ₁ , P ₂ are determined from the transfer functions Ftransf calculated in step S24, and an initial weight applied to the control signals of each loudspeaker. The initial weight applied to the control signals of each of the loudspeakers is equal to zero. Then, the weight applied to the control signals tends to vary with each iteration, as described below.

According to this exemplary embodiment, the acoustic pressures P ₁ , P ₂ each comprise all of the acoustic pressures determined at each of the positions of the virtual microphones. Thus, the estimated sound pressure in the sub-zones is more representative. This makes it possible to obtain a homogeneous result at the end of the process.

Alternatively, an acoustic pressure determined in a single position P ₁ , P ₂ is respectively estimated for the first sub-zone SZ1 and for the second sub-zone SZ2. This makes it possible to limit the number of calculations, and therefore to reduce the processing time and therefore the responsiveness of the system.

$\mathbf{G}\left(\omega ,n\right)\mathbf{q}\left(\omega ,n\right)$

More precisely, the acoustic pressures P ₁ , P ₂ in each of the sub-zones can be gathered in the form of a vector defined as: $\mathbf{p}\left(\omega ,\mathrm{not}\right)=\left[\begin{array}{c}{\mathit{P}}_1\\ {\mathit{P}}_2\end{array}\right]=$

$\mathbf{G}\left(\omega ,\mathrm{not}\right)\mathbf{q}\left(\omega ,\mathrm{not}\right)$

In step S26, the sound levels L ₁ and L ₂ are determined respectively in the first sub-zone SZ1 and in the second sub-zone SZ2. The sound levels L ₁ and L ₂ are determined at each position of the MIC microphones. This step makes it possible to convert the values of the estimated sound pressures P ₁ , P ₂ into measurable values in decibels. In this way, the sound contrast between the first and second sub-zones can be calculated. In step S27, a desired sound contrast level C _C between the first sub-zone and the second sub-zone is defined. For example, the desired sound contrast C _C between the first sub-zone SZ1 and the second sub-zone SZ2 is previously defined by a designer according to the chosen sound field and/or the perception of a listener U.

, avec p ₀ la pression acoustique de référence, c'est-à-dire le seuil de perception.More precisely, the sound level L for a microphone can be defined by $L=20{\log }_{10}\left(\frac{\left|\mathrm{<figure-callout\; id="0"\; label="P\; p"\; filenames="imgf0002.png,imgf0006.png"\; state="\{\{state\}\}">P</figure-callout>},\right|}{p_{}}\right)$

, with p ₀ the reference acoustic pressure, that is to say the perception threshold.

, avec P ^H la transposée conjuguée du vecteur de pressions acoustiques dans la sous-zone et M le nombre de microphones dans cette sous-zone.Thus, the average sound level in a sub-zone can be defined as: $L=10{\log }_{10}\left(\frac{\raisebox{1ex}{${\mathit{P}}^{\mathit{H}}\mathit{<figure-callout\; id="0"\; label="P\; M\; p"\; filenames="imgf0002.png,imgf0006.png"\; state="\{\{state\}\}">P</figure-callout>}$}\!\left/ \!\raisebox{-1ex}{$M$}\right.}{p_{}^{}}\right)$

, with P ^H the conjugate transpose of the acoustic pressure vector in the sub-zone and M the number of microphones in this sub-zone.

From the sound level L ₁ , L ₂ in the two sub-zones, it is possible to calculate the estimated sound contrast C between the two sub-zones: C = L ₁ - L ₂ .

In step S28, the difference between the estimated sound contrast between the two sub-zones and the desired sound contrast C _C is calculated. From this difference, an attenuation coefficient can be calculated. The attenuation coefficient is calculated and applied to the estimated sound pressure P ₂ in the second sub-zone in step S29. More precisely, an attenuation coefficient is calculated and applied to each of the estimated acoustic pressures P ₂ in each of the positions of the microphones MIC of the second sub-zone SZ2. The target sound pressure Pt ₂ in the second sub-zone then takes the value of the attenuated sound pressure P ₂ of the second sub-zone.

.Mathematically, the difference C _ξ between the estimated sound contrast C and the desired sound contrast C _C can be calculated as follows C _ξ = C - C _C = L ₁ - L ₂ - C _C It is then possible to calculate the coefficient of mitigation $\xi ={10}^{\frac{{\mathrm{vs}}_{\xi }}{20}}$

.

This coefficient is determined by the amplitude of the acoustic pressure to be given to each microphone so that the sound level in the second sub-zone is homogeneous. When the contrast is equivalent to that corresponding to the desired sound contrast C _C for a microphone in the second sub-zone, then C _ξ ≈ 0 therefore ξ ≈ 1. This means that the sound pressure estimated in this microphone corresponds to the pressure value target in the second subzone.

When the difference between the estimated sound contrast C and the desired sound contrast C _C is negative C _ξ < 0, this means that the desired contrast C _C is not yet achieved, and therefore that a lower pressure amplitude is required. get in this microphone.

When the difference between the estimated sound contrast C and the desired sound contrast C _C is positive C _ξ > 0, the sound pressure at this point is too low. It must therefore be increased to match the desired sound contrast in the second sub-zone.

The principle is therefore to use the pressure field present in the second sub-zone which is induced by the acoustic pressure in the first sub-zone, then to attenuate or amplify the individual acoustic pressure values estimated in each microphone , so that they match the target sound field in the second sub-zone across all microphones. For all microphones, we define the vector: ξ = [ ξ ₁ ,..., ξ _m , ..., ξ _M ] ^T.

This coefficient is calculated at each iteration and can therefore change. It can therefore be written in the form ξ ( n ).

Alternatively, in the case where a single sound pressure P ₂ is estimated for the second sub-zone SZ2, a single attenuation coefficient is calculated and applied to the sound pressure P ₂ .

The attenuation coefficients are calculated so as to meet the contrast criterion defined by the designer. In other words, the attenuation coefficient is defined so that the difference between the sound contrast between the two sub-zones SZ2 and the desired sound contrast This is close to zero.

Steps S30 to S32 make it possible to define the value of the target acoustic pressures Pt ₁ , Pt ₂ in the first and second sub-zones SZ1, SZ2.

Step S30 includes the initialization of the target acoustic pressures Pt ₁ , Pt ₂ , respectively in the first and second sub-zones SZ1, SZ2. The target sound pressures Pt ₁ , Pt ₂ characterize the target sound field to be diffused in the sub-zones. The target sound pressure Pt ₁ in the first sub-zone SZ1 is defined as being a target pressure Pt ₁ , chosen by the designer. More precisely, the target pressure Pt ₁ in the first sub-zone SZ1 is greater than zero, so that the target sound field is audible in this first sub-zone. The target sound pressure Pt ₂ in the second sub-zone is initialized at zero. The target pressures Pt ₁ , Pt ₂ are then transmitted to the TRAIT processing unit in step S31, in the form of a Pt vector.

At each iteration, new target pressure values are assigned to the target pressures Pt ₁ , Pt ₂ determined in the previous iteration. This corresponds to step S32. More precisely, the value of the target pressure Pt ₁ in the first sub-zone is that defined in step S30 by the designer. The designer can change this value at any time. The target sound pressure Pt ₂ in the second sub-zone takes the value of the attenuated sound pressure P ₂ (step S29). This makes it possible, at each iteration, to redefine the target sound field to be reproduced in the second sub-zone, taking into account the listener's perception and the loudspeaker control signals. Thus, the target sound pressure Pt ₂ of the second sub-zone is only equal to zero during the first iteration. Indeed, as soon as the speakers broadcast a signal, a sound field is perceived in the first sub-zone, but also in the second sub-zone.

Mathematically, the target pressure Pt ₂ in the second subzone is calculated as follows.

At the first iteration, Pt ₂ is equal to zero: Pt ₂ (0) = 0 .

At each iteration, the sound pressure P ₂ estimated in the second sub-zone is calculated. This sound pressure corresponds to the sound pressure induced in the second sub-zone by the radiation from the speakers in the first sub-zone. Thus, at each iteration we have: P ₂ ( ω , n ) = G ₂ ( ω , n ) q ( ω , n ), with G ₂ ( ω , n ) the matrix of transfer functions in the second subzone at iteration n .

The target pressure Pt ₂ at iteration n + 1 can therefore be calculated as Pt ₂ ( n + 1) = ξ ( n ) × P ₂ .

In step S33, the error between the target pressure Pt ₂ and the estimated pressure P ₂ in the second sub-zone is calculated. The error is due to the fact that an adaptation step µ is applied so that the target pressure Pt ₂ is not reached immediately. The target pressure Pt ₂ is reached after a certain number of iterations of the process. This makes it possible to minimize the calculation resources necessary to reach the target pressure Pt ₂ in the second sub-zone SZ2. This also ensures the stability of the algorithm. In the same way, the adaptation step µ is also chosen so that the error calculated in step S33 has a small value, in order to stabilize the filter.

The forgetting factor γ(n) is then calculated in order to calculate the weights to be applied to each loudspeaker control signal.

As described above, the forgetting factor γ(n) makes it possible to regularize the problem and to attenuate the weights calculated in previous iterations. Thus, when the listener moves, previous weights do not influence future weights.

The forgetting factor γ(n) is determined based directly on a possible movement of the listener. This calculation is illustrated in steps S34 to S36. In step S34, the position of the listener in previous iterations is recovered. For example, it is possible to recover the position of the listener in all previous iterations. Alternatively, it is possible to recover the position of the listener only for part of the previous iterations, for example the last ten or the last hundred iterations.

From this data, a movement speed of the listener is calculated in step S35. Movement speed can be calculated in meters per iteration. The listener's speed may be zero.

In step S36, the forgetting factor γ(n) is calculated according to the formula described above:

In step S33, the forgetting factor γ(n) is modified if necessary, depending on the calculation result of step S36.

The calculation and modification of the forgetting factor in step S37 is used to calculate the weights to be applied to the control signals of the HP speakers. More precisely, at the first iteration the weights are initialized to zero (step S38). Each loudspeaker broadcasts an unweighted control signal. Then, at each iteration, the value of the weights varies according to the error and the forgetting factor (step S39). The speakers then broadcast the control signal thus weighted.

The weights are calculated as described above with reference to the figures 2a and 2b , according to the formula: $$\mathbf{q}\left(\mathrm{not}+1\right)=\mathbf{q}\left(\mathrm{not}\right)\left(1-\mathit{\mu \gamma }\left(\mathrm{not}\right)\right)+\mu {\mathbf{G}}^H\left(\mathrm{not}\right)\left(\mathbf{G}\left(\mathrm{not}\right)\mathbf{q}\left(\mathrm{not}\right)-\mathbf{Pt}\left(\mathrm{not}\right)\right).$$

The FILT filters to be applied to the speakers are then determined in step S40. For example, a filter per HP speaker is calculated. There can therefore be as many filters as speakers. The type of filters applied to each speaker includes, for example, an inverse Fourier transform.

The filters are then applied to the audio signal to be reproduced S(U) which was obtained in step S41. Step S41 is an initialization step, implemented only in the first iteration of the method. The audio signal to be reproduced S(U) is intended respectively for the listener U. In step 542, the FILT filters are applied to the signal S(U), in order to obtain N control signals S(HP ₁ , ...,HP _N ) filtered to be broadcast respectively by the speakers (HP ₁ ,...,HP _N ) in step S43. The control signals S(HP ₁ ,...,HP _N ) are broadcast respectively by each loudspeaker (HP ₁ ,...,HP _N ) of the loudspeaker network in step S44. Generally speaking, HP speakers broadcast control signals continuously.

Then, at each iteration, the FILT filters are calculated based on the signals S(HP ₁ ,...,HP _N ) filtered in the previous iteration and broadcast by the speakers, as perceived by the microphone network. FILT filters are applied to the S(U) signal to obtain new control signals S(HP ₁ ,...,HP _N ) to be broadcast respectively on each loudspeaker of the loudspeaker network.

The process is then restarted from step S35 in which the acoustic pressures P ₁ , P ₂ of the two sub-zones SZ1, SZ2 are estimated.

Of course, the present invention is not limited to the embodiments described above. It extends to other variants.

For example, the method can be implemented for a plurality of listeners U ₁ to U _N. In this embodiment, an audio signal S(U ₁ , U _N ) can be provided respectively for each listener. Thus, the steps of the method can be implemented for each listener, so that the sound field chosen for each listener is returned to them in their position, and taking into account their movements. Thus, a plurality of forgetting factors can be calculated for each of the listeners.

According to another variant, the chosen sound field is a first sound field, at least a second chosen sound field being broadcast by the HP loudspeaker network. The second chosen sound field is audible in the second sub-zone for a second listener and is to be made inaudible in the first sub-zone for a first listener. The speakers are powered by the first control signals to each emit a continuous audio signal corresponding to the first chosen sound field, and are also powered by second control signals to each emit a continuous audio signal corresponding to the second sound field selected. The steps of the method as described above can be applied to the first sub-zone SZ1, so that the second chosen sound field is made inaudible in the first sub-zone SZ1 taking into account the movements of the two listeners.

According to another exemplary embodiment, the first and second sub-zones are not complementary. For example, in a zone, a first subzone can be defined with respect to a first listener U1 and a second subzone can be defined with respect to a second listener U2. The sound field is to be made audible in the first sub-zone and inaudible in the second sub-zone. The sound field in the remainder of the area may not be controlled.

Claims (11)

1. Method implemented by computational means, for spatialized audio rendering using an array of loudspeakers (HP₁, HP_N) covering a region (Z), with a view to diffusing a chosen audio field, audible in at least one position of at least one listener (U) in the region, wherein the loudspeakers (HP₁, HP_N) are fed with respective command signals (S(HP₁,...,HP_N)) so as to each continuously emit an audio signal, the method comprises, iteratively and continuously for each listener:

   - obtaining the current position of a listener (U) in the region (Z) by means of a position sensor (CAPT);

   - determining distances between at least one point of the region and the respective positions of the loudspeakers (HP₁,...,HP_N), with a view to deducing therefrom respective acoustic transfer functions of the loudspeakers at said point, said point corresponding to a position of a virtual microphone (MIC),

   - estimating an acoustic pressure (P) at said virtual microphone (MIC) at least depending on the acoustic transfer functions and on an initial respective weight of the command signals (S (HP₁, ..., HP_N)) for the loudspeakers (HP₁,...,HP_N),

   - computing an error between said estimated acoustic pressure (P) and a target acoustic pressure (Pt), desired at said virtual microphone (MIC);

   - computing and applying respective weights to the command signals (S(HP₁,...,HP_N)) for the loudspeakers (HP₁, ..., HP_N), depending on said error,

   characterized in that respective weights are furthermore computed and applied to the command signals (S(HP₁,...,HP_N)) for the loudspeakers (HP₁,...,HP_N) depending on a weight forgetting factor, said forgetting factor being computed depending on a movement of the listener, said movement being determined by comparison between a previous position of the listener and the current position of the listener;

   and in that the acoustic pressure (P) at the current position of the listener is computed again depending on the respective command signals (S(HP₁,...,HP_N)), thus weighted, for the loudspeakers (HP₁,...,HP_N),

   and in that an acoustic pressure (P) at said virtual microphone (MIC) is also estimated depending on the respective command signals (S(HP₁,...,HP_N)) for the loudspeakers (HP₁,...,HP_N),

   and in that a current position of said point is defined dynamically depending on the current position of the listener.
2. Method according to Claim 1, wherein a plurality of points forming the respective positions of a plurality of virtual microphones (MIC) is defined in the region (Z) with a view to estimating a plurality of respective acoustic pressures (P) in the region taking into account the respective weight applied to each loudspeaker (HP₁,...,HP_N), each respectively comprising a forgetting factor, and transfer functions specific to each loudspeaker (HP₁,...,HP_N) at each virtual microphone (MIC), the plurality of points being centred on the position of the listener.
3. Method according to either of Claims 1 and 2, wherein the region (Z) comprises a first sub-region (SZ1) in which the chosen audio field is to be made audible and a second sub-region (SZ2) in which the chosen audio field is to be made inaudible, the first sub-region (SZ1) being defined dynamically by the position of the listener and of said virtual microphone (MIC), the virtual microphone (MIC) being a first virtual microphone, and the second sub-region (SZ2) being defined dynamically as being complementary to the first sub-region, the second sub-region (SZ2) being covered by at least one second virtual microphone a position of which is defined dynamically depending on said second sub-region (SZ2), the method furthermore comprising, iteratively:

   - estimating an acoustic pressure (P₂) in the second sub-region, at least depending on the acoustic transfer functions, on the respective command signals (S(HP₁, ..., HP_N)) for the loudspeakers and on an initial respective weight of the command signals (S(HP₁,...,HP_N)) for the loudspeakers;

   - computing an error between said estimated acoustic pressure (P₂) in the second sub-region and a target acoustic pressure (Pt₂), desired in the second sub-region;

   - computing and applying respective weights to the command signals (S(HP₁,...,HP_N)) for the loudspeakers, depending on said error and on a weight forgetting factor, said forgetting factor being computed depending on a movement of the listener, said movement being determined by comparison between a previous position of the listener and the current position of the listener;

   - the acoustic pressure (P₂) in the second sub-region being computed again depending on the respective command signals (S(HP₁,...,HP_N)), thus weighted, for the loudspeakers.
4. Method according to either of Claims 1 and 2, wherein the region (Z) comprises a first sub-region (SZ1) in which the chosen audio field is to be made audible and a second sub-region (SZ2) in which the chosen audio field is to be made inaudible, the second sub-region (SZ2) being defined dynamically by the position of the listener and of said virtual microphone (MIC), the virtual microphone (MIC) being a first virtual microphone, and the first sub-region (SZ1) being defined dynamically as being complementary to the second sub-region (SZ2), the first sub-region (SZ1) being covered by at least one second virtual microphone (MIC) a position of which is defined dynamically depending on said first sub-region (SZ1), the method furthermore comprising, iteratively:

   - estimating an acoustic pressure (P₂) in the second sub-region, at least depending on the acoustic transfer functions, on the respective command signals (S(HP₁,...,HP_N)) for the loudspeakers and on an initial respective weight of the command signals (S(HP₁,...,HP_N)) for the loudspeakers;

   - computing an error between said estimated acoustic pressure (P₂) in the second sub-region and a target acoustic pressure (Pt₂), desired in the second sub-region (SZ2) ;

   - computing and applying respective weights to the command signals (S(HP₁,...,HP_N)) for the loudspeakers, depending on said error and on a weight forgetting factor, said forgetting factor being computed depending on a movement of the listener, said movement being determined by comparison between a previous position of the listener and the current position of the listener; the acoustic pressure in the second sub-region (SZ2) being computed again depending on the respective, weighted, command signals (S(HP₁,...,HP_N)) for the loudspeakers.
5. Method according to either of Claims 3 and 4, wherein each sub-region comprises at least one virtual microphone (MIC) and two loudspeakers (HP₁,...,HP_N), preferably each sub-region comprises at least ten virtual microphones (MIC) and at least ten loudspeakers (HP₁, ..., HP_N) .
6. Method according to one of Claims 1 to 5, wherein a value of the forgetting factor:

   - increases if the listener moves;

   - decreases if the listener does not move.
7. Method according to one of Claims 1 to 6, wherein the forgetting factor is defined by: $$\gamma \left(n\right)={\gamma }_{\mathit{max}}\times {\left(\frac{m}{X}\right)}^{\alpha }$$

   

   where γ(n) is the forgetting factor, n is the current iteration, γ_max is the maximum forgetting factor, χ is a parameter defined equal to µ, an adaptation pitch, m is a variable defined depending on a movement of the listener having χ as its maximum and α is a variable for adjusting the rate of increase or decrease of the forgetting factor.
8. Method according to Claim 7, wherein a rise pitch l_u and a fall pitch l_d of the forgetting factor are defined such that:

   - if movement of the listener is determined, $$m=\min \left(m+l_u,1\right)$$

   

   - if no movement of the listener is determined, $$m=\max \left(m-l_d,0\right),$$

   

   where 0 < l_u < 1 and 0 < l_d < 1, the rise and fall pitches being defined depending on a speed of movement of a listener and/or a change in the chosen audio field to be rendered.
9. Method according to one of Claims 1 to 8, wherein the forgetting factor is between 0 and 1.
10. System for spatialized audio rendering using an array of loudspeakers covering a region, with a view to diffusing a chosen audio field, selectively audible in a position of a listener in the region, characterized in that it comprises a position sensor (CAPT) and a processing unit that are designed for processing and implementing the method according to any one of Claims 1 to 9.
11. Medium for storing a computer program, loadable into a memory associated with a processor in a system according to Claim 10, and comprising code segments for implementing a method according to any one of Claims 1 to 9 on execution of said program by the processor.

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