EP2000002B1 - Method and device for efficient binaural sound spatialization in the transformed domain - Google Patents

Abstract

The method involves filtering through equalization-delay, and a sub band signal by applying gain and delay on the signal to generate an equalized and delayed component from each of encoded channels. A subset of equalized and delayed signals is added to create a number of filtered signals in a transformed domain. Each of the filtered signals is synthesized by a synthesis filter to obtain a set comprising reproduction sound channels of a number higher than or equal to two sound reproduction channels in time domain. Independent claims are also included for the following: (1) a device for sound spatialization of an audio scene (2) a computer program for executing filter, addition and synthesizing steps.

Description

The invention relates to the spatialization, known as 3D rendering, of compressed audio signals.

Such an operation is for example performed during the decompression of a compressed 3D audio signal for example, represented on a number of channels, to a number of different channels, two for example, to allow the reproduction of the 3D audio effects on a headphones.

Thus, the term "binaural" refers to the reproduction on a stereophonic headphones of a sound signal with nevertheless spatialization effects. The invention is however not limited to the aforementioned technique and applies, in particular, to techniques derived from the "binaural", such as the so-called technical rendering techniques TRANSAURAL ^® , that is to say on top of remote speakers. TRANSAURAL ^® is a registered trademark of COOPER BAUCK CORPORATION. Such techniques can then use a "cross-talk cancellation", which consists in canceling the crossed acoustic paths, so that a sound, thus processed and then emitted by the loudspeakers , can be perceived only by one of the two ears of a listener.

Accordingly, the invention also relates to the transmission and reproduction of multichannel audio signals and their conversion to a rendering device, transducer, imposed by the equipment of a user. This is for example the case for the reproduction of a 5.1 sound stage by an audio headset, or by a pair of loudspeakers.

The invention also relates to the reproduction, in the context of a game or video recording for example, of one or more samples sound stored in files, with a view to their spatialization.

Among the known techniques in the field of binaural sound spatialization, different approaches have been proposed.

A sound spatialization method of the kind indicated in the preamble of claim 1 below is described in the patent application. FR 2 851 879 A .

In particular, bi-binaural synthesis consists, with reference to the figure 1a , to filter the signal of the different sound sources S _i that it is desired to position, at the restitution, at a position in space, via acoustic transfer functions left HRTF-I and right HRTF-r in the frequency domain corresponding to the appropriate direction, defined in polar coordinates (θ ₁ , φ ₁ ). The transfer functions HRTF, for "Head Related Transfer Functions" in English, are the acoustic transfer functions of the head of the listener between the positions of the space and the auditory canal. In addition, "HRIR" for "Head Related Impulse Response" is referred to as their temporal form. These functions may further include a room effect.

For each sound source S _i, two left and right signals are obtained which are then added to the left and right signals from the spatialization of the other sound sources, to finally give the signals L and R diffused to the left and right ears of the listener. .

The number of necessary filters or transfer functions is then 2.N for a static binaural synthesis and 4.N for a dynamic binaural synthesis, N designating the number of sound sources or audio streams to be spatialized.

Work titled "A model of head-related transfer based on principal components analysis and minimum - phase reconstruction" conducted by D. Kistler and FL Wightman, published in J. Acoust. Soc. Am. 91 (3): p 1637-1647 (1992) ) and by A. Kulkami 1995 "IEEE ASSP Workshop on IEEE Catalog Number: 95TH8144 , made it possible to verify that the phases of the HRTF can be decomposed into the sum of two terms, one corresponding to the interaural delay and the other equal to the minimum phase associated with the HRTF module.

$$\varphi \left(f\right)=\mathit{\varphi \; retard}\left(f\right)+\mathit{\varphi }\min \left(f\right)$$

ϕ retard (f) = 2πfτ correspond au retard interaural ;
ϕmin(f)= H(log(|H(f)|)) est la phase minimale associée au module du filtre H.Thus, for an HRTF transfer function expressed as form: $$H\left(f\right)=\left|H\left(f\right)\right|e^{-\mathit{j\phi }\left(f\right)}$$

$$\phi \left(f\right)=\mathit{\phi \; delay}\left(f\right)+\mathit{\phi }\min \left(f\right)$$

φ delay (f) = 2πfτ corresponds to the interaural delay;
φmin (f) = H ( log (| H (f ) |)) is the minimum phase associated with the H filter module.

The binaural filter implementation is generally in the form of two minimal phase filters and a pure delay, corresponding to the difference of the left and right delays applied to the ear furthest away from the source. This delay is usually implemented using a delay line.

The minimum phase filter is a finite impulse response filter and can be executed in the time or frequency domain. Infinite impulse response filters can be searched to approximate the minimum phase HRTF filter module.

As far as binauralisation is concerned, reference is made to the figure 1b , in the non-limiting frame of a sound stage spatialized in 5.1 mode, in view of the restitution thereof on the audio headset of a human being HB.

Five loudspeakers C: between C, Lf: L is NLOR f, R f: R f ight NLOR, SI: S urround l eft, Sr: S urround r ight, each produce a sound that is perceived by the human HB on the two receivers that are his ears. The transformations undergone by the sound are modeled by a filtering function representing the modification that this sound undergoes during its propagation between the speaker which reproduces this sound and a given ear.

In particular, the sound emanating from the loudspeaker Lf affects the left ear LE through an HRTF filter A but this same sound reaches the right ear RE modified by a HRTF filter B.

The position of the speakers relative to the aforementioned HB individual may be symmetrical or not.

$$\mathrm{O}\mathrm{r}\mathrm{e}\mathrm{i}\mathrm{l}\mathrm{l}\mathrm{e}\mathrm{d}\mathrm{r}\mathrm{o}\mathrm{i}\mathrm{t}\mathrm{e}\mathrm{R}\mathrm{E}:\mathrm{B}\mathrm{r}=\mathrm{A}\mathrm{r}\mathrm{f}+\mathrm{C}\mathrm{C}+\mathrm{B}\mathrm{L}\mathrm{f}+\mathrm{D}\mathrm{S}\mathrm{r}+\mathrm{E}\mathrm{S}\mathrm{l}$$

où BI est le signal binauralisé pour l'oreille gauche LE et Br est le signal binauralisé pour l'oreille droite RE.Each ear therefore receives the contribution of the 5 loudspeakers in the form modeled below: $$\mathrm{O}\mathrm{r}\mathrm{e}\mathrm{i}\mathrm{l}\mathrm{l}\mathrm{e}\mathrm{boy\; Wut}\mathrm{at}\mathrm{u}\mathrm{vs}\mathrm{h}\mathrm{e}\mathrm{The}\mathrm{E}:\mathrm{AT}\mathrm{The}\mathrm{f}+\mathrm{VS}\mathrm{VS}+\mathrm{B}\mathrm{R}\mathrm{f}+\mathrm{D}\mathrm{S}\mathrm{I}+\mathrm{E}\mathrm{S}\mathrm{r},$$

$$\mathrm{O}\mathrm{r}\mathrm{e}\mathrm{i}\mathrm{l}\mathrm{l}\mathrm{e}\mathrm{d}\mathrm{r}\mathrm{o}\mathrm{i}\mathrm{t}\mathrm{e}\mathrm{R}\mathrm{E}:\mathrm{B}\mathrm{r}=\mathrm{AT}\mathrm{r}\mathrm{f}+\mathrm{VS}\mathrm{VS}+\mathrm{B}\mathrm{The}\mathrm{f}+\mathrm{D}\mathrm{S}\mathrm{r}+\mathrm{E}\mathrm{S}\mathrm{l}$$

where BI is the binauralized signal for the left ear LE and Br is the binauralized signal for the right ear RE.

The filters A, B, C, D and E are modeled, most often, by linear digital filters and it is therefore necessary, in the configuration represented in FIG. figure 1b , 10 filtering functions to be applied, which can be reduced to 5, taking into account the symmetries.

In a manner known per se, the aforementioned filtering operations can be performed in the frequency domain, for example by virtue of a fast convolution performed in the Fourier domain. A Fast Fourier Transform (FFT) Fourier Transform is then used to perform binauralization effectively.

The HRTF filters A, B, C, D and E can be simplified as a frequency equalizer and a delay. The HRTF filter A can be realized as a simple equalizer, since it is a direct path, while the HRTF filter B includes an additional delay. Conventionally the HRTF filters can be decomposed into a minimum phase filter and a pure delay. The delay for the ear closest to the source can be taken as zero.

The spatial decoding reconstruction operation of a 3D audio sound scene, from a reduced number of transmitted channels, as represented in FIG. figure 1c , is also known from the state of the art. The configuration represented in figure 1c is that relating to the decoding of a coded sound path having location parameters in the frequency domain, in order to reconstruct a spatialized sound scene 5.1.

The aforementioned reconstruction is carried out by a frequency subband sub-frequency decoder, as represented in FIG. figure 1c . The coded audio signal m undergoes 5 spatialization processing steps, which are controlled by parameters or complex coefficients of spatialization CLD and ICC calculated by the encoder and which, by means of decorrelation operations and gain correction, to realistically reconstruct the sound stage composed of six channels, the five channels represented in figure 1b , to which is added a low frequency effect channel Ife.

When it is desired to binauralize the sound channels coming from a spatial decoder as represented in FIG. figure 1c , we are actually forced, at present, to implement a treatment according to the diagram represented in figure 1d .

With reference to the aforementioned scheme, it appears necessary to perform the transformation of the sound channels available in the time domain before proceeding to the binauralization of the signal. This return operation in the time domain is symbolized by the synth synthesizer blocks that execute the frequency-time transformation operation for each of the channels coming from the spatial decoder (SD). The filtering by HRTF filters can then be performed by the filters A, B, C, D, E, with or without applying the equalized scheme, corresponding to a conventional filtering.

A binauralization variant of the sound channels of a spatial decoder can also consist, as represented in FIG. figure 1e , converting each sound channel delivered by the audio decoder into the time domain by a synthesizer "Synth" and then performing the spatial decoding operation and binauralization, or spatialization, in the frequency domain Fourier after transformation by FFT.

In this case, each OTT module corresponding to a matrix of decoding coefficients, must then be converted into the Fourier domain, at the cost of an approximation, because the operations are not performed in the same domain. In addition, the complexity is further increased because the synthetic operation "Synth" is followed by three FFT transformations.

• soit 6 transformations temps-fréquence, si l'on veut réaliser la binauralisation en dehors du décodeur spatial ;
• soit une opération de synthèse suivie de 3 transformations de Fourier, FFT, si l'on veut réaliser l'opération dans le domaine FFT.

Thus, to binauralise a sound scene from a spatial decoder, there is hardly any other possibility than to achieve:

• or 6 time-frequency transformations, if binauralization is to be carried out outside the space decoder;
• either a synthesis operation followed by 3 Fourier transforms, FFT, if one wishes to carry out the operation in the FFT domain.

Another solution may be to perform HRTF filtering directly in the subband domain, as shown in FIG. figure 1f .

However, in this case, the HRTF filterings are complex to achieve because they require the use of subband filters, the minimum length of which is fixed and which must take into account the phenomenon of spectral folding of the subbands.

The economy introduced by the reduction of transformation operations is negatively offset by the explosion in the number of operations required for filtering, because of the execution of these operations in the PQMF domain for P seudo Q uadrature M irror F Ilter in English.

The object of the present invention is to overcome the numerous drawbacks of the above-mentioned prior art of sound spatialization of 3 D audio scenes, in particular transauralisation or binauralization of 3 D audio scenes.

In particular, an objective of the present invention is the execution of a specific filtering of spatially coded audio signals or channels in the frequency subband domain of a spatial decoding, in order to limit the number of transformations two by two, while reducing the filtering operations to a minimum, but maintaining a good quality of source spatialization, including transauralisation or binauralization.

According to a particularly remarkable aspect of the present invention, the execution of the aforementioned specific filtering is based on the equalizer-delay form of the spatialization filters, transaural or binaural, for a direct application of filtering by equalization-delay in the domain of the sub-bands.

Another objective of the present invention is to obtain a 3D rendering quality very close to that obtained from modeling filters such as original HRTF filters, by the sole addition of a transaural spatial processing of very low complexity, following a classical spatial decoding in the transformed domain.

Finally, an objective of the present invention is a new source spatialization technique applicable not only to the transaural or binaural rendering of a monophonic sound, but also to several monophonic sounds and in particular to the multiple channels of 5.1, 6.1, 7.1, 8.1 or 5.1 stereo sounds. higher.

The subject of the present invention is thus a method for sound spatialisation of an audio scene comprising a first set comprising a number greater than or equal to the unit of audio channels coded spatially over a number of sub-bands of determined frequencies, and decoded in a transformed domain, in a second set comprising a number greater than or equal to two of sound reproduction channels in the time domain, from acoustic propagation modeling filters of the audio signals of the first set of channels.

• un filtrage par égalisation-retard du signal en sous-bande, par application d'un gain respectivement d'un retard sur le signal en sous-bande, pour engendrer à partir des canaux codés spatialement, une composante égalisée et retardée d'une valeur déterminée dans la sous-bande fréquentielle considérée,
• une addition d'un sous-ensemble de composantes égalisées et retardées, pour créer un nombre de signaux filtrés dans le domaine transformé correspondant au nombre du deuxième ensemble, supérieur ou égal à deux, de canaux sonores de restitution dans le domaine temporel,
• une synthèse de chacun des signaux filtrés dans le domaine transformé par un filtre de synthèse, pour obtenir le deuxième ensemble de nombre supérieur ou égal à deux de signaux sonores de restitution dans le domaine temporel.

According to the invention, this method is remarkable in that, for each modeling filter converted into at least one gain and a delay applicable in the transformed domain, it consists in performing at least, for each sub-band frequency of the transformed domain:

• a filtering by equalization-delay of the signal in sub-band, by applying a gain respectively of a delay on the signal in sub-band, to generate from the spatially coded channels, an equalized and delayed component of a value determined in the frequency subband considered,
• an addition of a subset of equalized and delayed components, to create a number of filtered signals in the transformed domain corresponding to the number of the second set, greater than or equal to two, of sound channels of restitution in the time domain,
• a synthesis of each of the filtered signals in the transformed domain by a synthesis filter, to obtain the second set of numbers greater than or equal to two of sound signals of restitution in the time domain.

The method which is the subject of the invention is also remarkable in that the filtering by equalization-delay of the signal in sub-band includes at least the application of a phase shift and, if appropriate, a pure delay by storage, for the at least one of the frequency sub-bands.

The method which is the subject of the invention is also remarkable in that it includes filtering by equalization-delay in a hybrid transformed domain, comprising an additional step of frequency cutting into additional subbands, with or without decimation.

The method which is the subject of the invention is finally remarkable in that to convert each modeling filter into a gain value or a delay value in the transformed domain, it consists at least in associating as a gain value with each subband a real value. defined as the average of the modeling filter module in this sub-band and to associate as delay value with each sub-band a delay value corresponding to the reception delay between the left ear and the right ear for different positions.

The subject of the present invention is correspondingly to a sound spatialization device of an audio scene comprising a first set comprising a number, greater than or equal to one, of audio channels coded spatially over a number of sub-bands of determined frequencies, and decoded in a transformed domain into a second set comprising a number greater than or equal to two of time domain rendering sound channels, from sound propagation modeling filters audio signals of the first subset of channels.

• un module de filtrage par égalisation-retard du signal en sous-bande par application d'un gain respectivement d'un retard sur le signal en sous-bande, pour engendrer à partir de chacun des canaux audio-codés spatialement une composante égalisée et retardée d'une valeur de retard déterminée dans la sous-bande de fréquences considérée,
• un module d'addition d'un sous-ensemble de composantes égalisées et retardées pour créer un nombre de signaux filtrés dans le domaine transformé correspondant au nombre du deuxième ensemble supérieur ou égal à deux des canaux sonores de restitution dans le domaine temporel,
• un module de synthèse de chacun des signaux filtrés dans le domaine transformé pour obtenir le deuxième ensemble comprenant un nombre supérieur ou égal à deux des canaux sonores de restitution dans le domaine temporel.

According to the invention this device is remarkable in that, for each frequency sub-band of a spatial decoder in the transformed domain, this device comprises in addition to this spatial decoder:

• a filtering module by equalizing-delaying the signal in the sub-band by applying a gain respectively of a delay on the signal in sub-band, for generating from each of the spatially audio-coded channels an equalized and delayed component a delay value determined in the sub-frequency band considered,
• a module for adding a subset of equalized and delayed components to create a number of filtered signals in the transformed domain corresponding to the number of the second set greater than or equal to two of the time domain rendering sound channels,
• a synthesis module of each of the filtered signals in the transformed domain to obtain the second set comprising a number greater than or equal to two of the sound reproduction channels in the time domain.

The method and the device which are the subject of the invention are applicable to the electronic industry of audio and / or video hi-fi equipment, to the audio-video game industry, which is executed locally or online.

• la figure 2a représente un organigramme illustratif des étapes de mise en oeuvre du procédé de spatialisation sonore objet de l'invention ;
• la figuré 2b représente à titre illustratif, une variante de mise en oeuvre du procédé objet de l'invention représenté en figure 2a, obtenu par création de sous-bandes supplémentaires, en l'absence de décimation ;
• la figure 2c représente à titre illustratif, une variante de mise en oeuvre du procédé objet de l'invention représenté en figure 2a obtenu par création de sous-bandes supplémentaires, en présence de décimation ;
• la figure 3a représente, à titre illustratif, un étage, pour une sous-bande de fréquences d'un décodeur spatial, d'un dispositif de spatialisation sonore objets de l'invention ;
• la figure 3b représente, à titre illustratif, un détail de mise en oeuvre d'un filtre par égalisation-retard permettant la mise en oeuvre du dispositif objet de l'invention représenté en figure 3a ;
• la figure 4 représente à titre illustratif, un exemple de mise en oeuvre du dispositif objet de l'invention dans lequel le calcul des filtres d'égalisation retard est délocalisé.

They will be better understood by reading the description and by observing the drawings below, in which, in addition to the Figures 1a to 1f relating to the prior art,

• the figure 2a represents an illustrative flow diagram of the implementation steps of the sound spatialization method which is the subject of the invention;
• FIG. 2b represents, by way of illustration, an alternative embodiment of the method which is the subject of the invention represented in FIG. figure 2a obtained by creating additional subbands, in the absence of decimation;
• the Figure 2c represents, by way of illustration, an alternative embodiment of the process which is the subject of the invention represented in figure 2a obtained by creating additional subbands in the presence of decimation;
• the figure 3a represents, by way of illustration, a stage, for a frequency sub-band of a spatial decoder, of a sound spatialization device which is the subject of the invention;
• the figure 3b represents, by way of illustration, a detail of implementation of a filter by equalization-delay allowing the implementation of the device object of the invention represented in figure 3a ;
• the figure 4 represents for illustrative purposes, an example of implementation of the device according to the invention in which the calculation of the delay equalization filters is delocalized.

A more detailed description of the sound spatialization method of an audio scene in accordance with the subject of the present invention will now be given in connection with the figure 2a and the following figures.

The method according to the invention applies to an audio scene such as an audio scene 3 D represented by a first set comprising an N number of audio channels coded spatially greater than or equal to unity, N ≥ 1, on a number of frequency subbands determined and decoded in a transformed domain.

The transformed domain is a transformed frequency domain such as Fourier domain, PQMF domain or any hybrid domain derived from them by creating additional frequency subbands, whether or not subjected to a temporal decimation process.

Consequently, the spatially coded audio channels constituting the first set N of channels, are represented in a nonlimiting manner by the channels Fl, Fr, Sr, Sl, C, Ife previously described in the description and corresponding to a decoding mode of a 3 D audio scene in the corresponding transformed domain, as previously described in the description. This mode is none other than the 5.1 mode previously mentioned.

k désigne le rang de la sous-bande considérée.In addition, these signals are decoded in the aforementioned transformed domain according to a determined number of sub-bands suitable for decoding, the set of sub-bands being noted.

k denotes the rank of the subband considered.

The method which is the subject of the invention makes it possible to transform all the spatially encoded audio channels mentioned above into a second set comprising a number, greater than or equal to two, of sound reproduction channels in the time domain, the sound reproduction channels being noted Bl and Br for the left binaural channels respectively right, without limitation in the context of the figure 2a . It is understood, in particular, that instead of two binaural channels, the method which is the subject of the invention applies to any number of channels greater than two, allowing, for example, real-time sound reproduction of the 3D audio scene, as represented and described in the description in connection with the figure 1 b.

According to a remarkable aspect of the method which is the subject of the invention, this is implemented using acoustic propagation modeling filters of the audio signals of the first set of spatially coded audio channels, taking into account a conversion in the form of at least one gain and delay applicable in the transformed domain, as will be described later in the description. Without limitation, the modeling filters will be designated HRTF filters in the following description.

The aforementioned conversion is noted for each HRTF filter considered for a sub-band SB _k of rank k to establish a gain value g _k and corresponding delay d _k , the previous conversion being then noted, as represented in FIG. figure 2a HRTF Ξ (g _k , d _k ).

Given the aforementioned conversion, the method according to the invention consists, for each frequency sub-band of the transformed domain of rank k, to perform a filtering in step A by equalization-delay of the signal in subband by application a gain g _k respectively of a delay d _k on the sub-band signal, to generate from the spatially-referenced coded channels, that is to say the channels Fl, C, Fr, Sr, Sl and Ife, an equalized and delayed component of a determined delay value in the frequency sub-band SB _k of rank k.

On the figure 2a , the filtering operation by equalization-delay is noted symbolically CED _kx = {Fl, C, Fr, Sr, Sl, Ife} (g _kx , d _kx ).

In the aforementioned symbolic relation, FEB _kx denotes each equalized and delayed component obtained by applying the gain g _kx and the delay d _kx to each of the spatially coded audio channels, ie the channels Fl, C, Fr, Sr , Sl, Ife.

Accordingly, and in the aforementioned symbolic relationship, x, for the corresponding rank k sub-band, can actually take the values Fl, C, Fr, Sr, Sl, Ile.

Step A is then followed in the transformed domain of a step B of adding a subset of equalized and delayed components to create a number of filtered signals in the transformed domain corresponding to the number N 'of the second set, greater than or equal to 2, sound channels of restitution in the time domain.

In step B of the figure 2a , the addition operation is given by the symbolic relation: $$\mathrm{F}\left\{\mathrm{fl},\mathrm{VS},\mathrm{Fr},\mathrm{Sr},\mathrm{sl},\mathrm{Ife}\right\}\mathrm{=}{\mathrm{\Sigma CED}}_{\mathrm{kx}}\mathrm{.}$$

In the aforementioned symbolic relation, F {F1, C, Fr, Sr, S1, Ife} denotes the subset of the filtered signals in the transformed domain obtained by summation of a subset of equalized and delayed components CED _kx .

By way of nonlimiting example and to fix the ideas, for a first set comprising a number of spatially coded audio channels N = 6, corresponding to a mode 5.1, the subset of equalized and delayed components may consist of adding five of these components equalized and delayed for each ear to obtain the number N 'equal to 2 of filtered signals in the transformed domain, as will be described in more detail later in the description.

The aforementioned addition step B is then followed by a step C of synthesizing each of the filtered signals in the transformed domain by a synthesis filter to obtain the second set of number N 'greater than or equal to two of sound signals of restitution in the time domain.

At step C of the figure 2a , the corresponding synthesis operation is represented by the symbolic relation: $$\mathrm{bl},\mathrm{Br}=\mathrm{Synth}\left(\mathrm{F}\left\{\mathrm{fl},\mathrm{VS},\mathrm{Fr},\mathrm{Sr},\mathrm{sl},\mathrm{Ife}\right\}\right)$$

In general, it is indicated that the method that is the subject of the invention can be applied to any 3D audio scene composed of N varying from 1 to infinity of audio channels or channels coded spatially to N 'varying from 2 to the infinity of sound channels of restitution.

With regard to the summation step represented in step B of the figure 2a , it is indicated that this consists more specifically of adding a subset of components delayed differently by the different delays to generate the N 'components for each sub-band.

More specifically, it is indicated that the filtering by equalization-delay of the signal in sub-band includes at least the application of a phase shift supplemented if necessary by a pure delay by storage, for at least one sub-band. frequency bands.

The notion of applying a pure delay is symbolized in step A of the figure 2a by the relation g _Ex = 1, which represents the absence of equalization for all the audio channels of index x in the subband of rank k = E, the value 1 indicating a transmission without modification of the amplitude of each of the spatially coded audio channels.

The transformed domain may, as previously mentioned in the description, correspond to a hybrid transformed domain as will be described in connection with the figure 2b in the case where no frequency decimation is applied in the corresponding sub-band.

With reference to the figure 2b above, the delay equalization filtering represented in step A of the figure 2a is then executed in three sub-steps A1, A2, A3 represented at figure 2b .

Under these conditions, the step A comprises an additional step of frequency-cutting in additional sub-bands without decimation, to increase the number of applied gain values and thus the frequency accuracy, followed by a subgrouping step. additional bands to which the aforementioned gain values have been applied.

Frequency cutting and regrouping operations are represented in substeps A ₁ and A ₂ of the figure 2b .

The step of frequency cuts is represented in substep A ₁ by the relation: $$\mathrm{HRTF}\equiv {\left\{{\mathrm{boy\; Wut}}_{\mathrm{kz}},\mathrm{dkz}\right\}}_{z=1}^{z=\mathrm{Z}}\mathrm{.}$$

The grouping step is represented in substep A ₂ by the relation: $${{\left[{\mathrm{GAEIB}}_{\mathrm{kz}}\right]}_1}^{\mathrm{z}}\mathrm{x}=\left\{\mathrm{fl},\mathrm{VS},\mathrm{fl},\mathrm{Sr},\mathrm{sl},\mathrm{Ife}\right\}\left({\mathrm{boy\; Wut}}_{\mathrm{kz}}\right)$$

In the substep A _{1. it} is understood that the gain and delay values for the subband of rank k considered are subdivided into Z corresponding gain values, a gain value g _kz for each additional subband and at sub-step 1 _{2 it} is understood that the grouping of additional subbands is performed from the corresponding coded audio channels for the corresponding index x which has been applied the gain value g _kz in the additional subband considered.

désigne le regroupement des sous-bandes supplémentaires auxquelles ont été appliquées les valeurs de gain pour les sous-bandes supplémentaires considérées.In the previous relationship

means the grouping of additional subbands to which gain values have been applied for the additional subbands considered.

The sub-step A ₂ is then followed by a sub-step A ₃ consisting of applying the delay to the grouped additional subbands and in particular to the spatially coded audio channels of corresponding index x via the delay d _kx of similar to Step A of Fig. 2a.

The corresponding operation is noted by the relation: $${\mathrm{CED}}_{\mathrm{kz}}\mathrm{x}={\left[{\mathrm{CGOL}}_{\mathrm{kz}}\right]}_{\mathrm{z}=1}^{\mathrm{z}\mathrm{=}\mathrm{Z}}\times \left({\mathrm{dk}}_{\mathrm{x}}\right)\mathrm{.}$$

In addition, the method which is the subject of the invention can also consist in performing a filtering by equalization-delay in a hybrid transformed domain comprising an additional step of frequency cutting into additional subbands with decimation, as represented in FIG. Figure 2c .

In this case, step A ' ₁ of the Figure 2c is identical to step A ₁ of the figure 2b , to execute the creation of additional subbands with decimation.

In this case, the decimation operation in step A ' ₁ of the Figure 2c is executed in the time domain.

Step A ' ₁ is then followed by a step A' ₂ corresponding to a grouping of the additional subbands to which the above-mentioned gain values have been applied in view of the decimation.

Grouping step A ' ₂ is itself preceded or followed by the application of the delay dkx thus represented by the double reversing arrow of steps A' ₂ and A ' ₃ .

It is understood, in particular, that when the application of the delay is performed prior to the grouping, the delay is applied directly to the signals of the additional subbands prior to the grouping.

With regard to the conversion of each HRTF filter into a gain and delay value in the transformed domain, this operation may advantageously consist in associating, as a gain value with each subband of rank k, a real value defined as the average of the corresponding HRTF filter module and to associate, as a delay value with each subband of rank k, a delay value corresponding to the delay of propagation between the left ear and the right ear of a listener for different positions.

Thus, from an HRTF filter, it is possible to automatically calculate the gains and delay times applied in subband. From the frequency resolution of the HRTF filter bank, each subband SB _k is associated with a delay value corresponding to the propagation delay between the left ear and the right ear of a listener for different positions.

Thus, from an HRTF filter, one can automatically calculate the gains and delay times to be applied in subband.

From the frequency resolution of the filter bank, each band is associated with a real value. By way of nonlimiting example, it is possible from the HRTF filter module, to calculate, for each sub-band, the average of the module of the aforementioned HRTF filter. Such an operation is similar to an octave band or Bark analysis of HRTF filters. Similarly, the delay to be applied for the indirect channels, that is to say the delay values which are more particularly applicable to the channels whose delay is not minimum, is determined. There are many ways to automatically determine the remaining interaural delays. ITD designated for "I nteraural T ime D ifference" and correspond to the delays between the left and right ears, for different positions of the listener. The threshold method described by S. Busson in the doctoral dissertation of the Université de la Mediterranée Est-Marseille II, 2006 entitled " Individualization of acoustic indices for synthesis" can be used as a non-limiting example. binaural ". The principle of methods for estimating threshold-type interaural delay is to determine the arrival time, or the initial delay of the wave on the right ear Td and on the left ear Tg. Interaural delay is given by the relation ITD threshold = Td - Tg.

The most common method estimates the arrival time as the time when the HRIR time filter exceeds a given threshold. For example, the arrival time may correspond to the time for which the response of the HRIR filter reaches 10% of its maximum.

An example of a specific implementation in the PQMF transformed domain will now be given below.

In general, it is indicated that the application of a gain in the complex PQMF domain consists in multiplying the value of each sample of the subband signal, represented by a complex value, by the gain value formed by a real number.

Indeed, it is well known that the use of a complex PQMF transformed domain makes it possible to apply the gains while avoiding the problems of aliasing caused by the subsampling inherent in the filterbanks. Each sub-band SB _k of each channel is thus assigned a determined gain.

In addition, the application of a delay in the PQMF transformed domain consists, for each sample of the subband signal represented by a complex value, of introducing a rotation in the complex plane by multiplication of this sample by a complex exponential value depending on the rank of the sub-band considered, the sub-sampling rate in the sub-band considered and a delay parameter related to the interaural delay difference of a listener.

The rotation in the complex plane is then followed by a pure time delay of the sample after rotation. This pure time delay is a function of the difference in the interaural delay of a listener and the sub-sampling rate in the subband considered.

In practice, it is indicated that the aforementioned delays are applied to the resulting signals, ie the equalized signals and in particular to the subsets of these signals or channels which do not benefit from a direct path.

et par un retard pur implémenté par une ligne à retard, par exemple réalisant l'opération : $$\mathrm{y}\left(\mathrm{k},\mathrm{n}\right)\mathrm{=}\mathrm{x}\left(\mathrm{k}\mathrm{,}\mathrm{n}\mathrm{-}\mathrm{D}\right)$$

In particular, the rotation is performed in the form of a complex multiplication by an exponential value of the form: $$\exp \left(\mathrm{-}\mathrm{j}\mathrm{*}\mathrm{pi}\mathrm{*}\left(\mathrm{k}\mathrm{+}\mathrm{0}\mathrm{,}\mathrm{5}\right)\mathrm{*}\mathrm{d}\mathrm{/}\mathrm{M}\right)$$

and by a pure delay implemented by a delay line, for example carrying out the operation: $$\mathrm{there}\left(\mathrm{k},\mathrm{not}\right)\mathrm{=}\mathrm{x}\left(\mathrm{k}\mathrm{,}\mathrm{not}\mathrm{-}\mathrm{D}\right)$$

• exp est la fonction exponentielle ;
• j est tel que j*j = -1 ;
• k le rang de la sous-bande SBk considérée ;
• M est le taux de sous-échantillonnage dans la sous-bande considérée, M veut être pris égal à 64, par exemple ;
• y(k, n) est la valeur de l'échantillon de sortie après application du retard pur sur l'échantillon temporel de rang n de la sous-bande SB_k de rang k, c'est-à-dire l'échantillon x (k,n) auquel est appliqué le retard B.
• d et D dans les relations précédentes sont tels qu'ils correspondent à l'application d'un retard de D*M + d dans le domaine temporel non sous-échantillonné. Le retard D*M + d correspond au retard interaural calculé précédemment. d peut prendre des valeurs négatives ce qui permet de simuler une avance de phase en lieu et place d'un retard.

In previous relationships:

• exp is the exponential function;
• j is such that j * j = -1;
• k the rank of the SBk sub-band considered;
• M is the sub-sampling rate in the sub-band considered, M wants to be taken equal to 64, for example;
• y (k, n) is the value of the output sample after application of the pure delay on the n-rank time sample of sub-band SB _k of rank k, i.e. sample x (k, n) to which the delay B is applied.
• d and D in the previous relations are such that they correspond to the application of a delay of D * M + d in the non-subsampled time domain. The delay D * M + d corresponds to the interaural delay calculated previously. d can take negative values which makes it possible to simulate a phase advance instead of a delay.

The operation thus performed induces an approximation which is suitable for the desired effect.

In terms of calculation operations, the processing implemented therefore consists of performing a complex multiplication between an exponential complex and a subband sample formed by a complex value.

A possible delay, if the total delay to be applied is greater than the value M, is to be inserted, but this operation does not involve any arithmetic operation.

The method which is the subject of the invention can also be implemented in a hybrid transformed domain. This hybrid transformed domain is a frequency domain in which the PQMF bands are advantageously redécoupées by a bank of filters decimated or not.

If the filter bank is decimated, the decimation means a decimation in time, so the introduction of a delay advantageously follows the procedure including a pure delay and a phase shifter.

If the filter bank is not decimated, then the delay may be applied only once during the synthesis. It is indeed useless to apply the same delay on each of the branches because the synthesis is a linear operation, without subsampling.

The application of the gains remains the same, these being simply more numerous, as described previously in connection with the figure 2b for example, and thus allow to follow the more precise cutting in frequency. A real gain is then applied per additional subband.

Finally, according to an alternative embodiment, the method according to the invention is repeated for at least two equalization-delay pairs and the signals obtained are summed to obtain the sound channels in the time domain.

A more detailed description of a sound spatialization device of an audio scene comprising a first set comprising a number greater than or equal to the unit of audio channels spatially coded on a number of frequency subbands determined and decoded in a domain converted, into a second set comprising a number greater than or equal to 2 of sound reproduction channels in the time domain, according to the subject of the present invention, will now be described in connection with the figures 3a and 3b .

As mentioned above, the device the invention is based on the principle of conversion in the form of at least one gain and a delay applicable in the transformed domain of modeling filters of the acoustic propagation of the audio signals of the first set of channels mentioned above. The device according to the invention allows the sound spatialization of an audio scene, such as a 3D audio scene, into a second set comprising a number, greater than or equal to two, of sound reproduction channels in the time domain.

The device which is the subject of the invention represented in figure 3a relates in a stage of this device specific to each subband SB _k of rank k decoding in the transformed domain.

It is understood in particular that the stage, for each subband of rank k represented in figure 3a is, in fact, replicated for each of the subbands to finally constitute the sound spatialization device according to the subject of the present invention.

By convention, the floor represented in figure 3a will be designated hereinafter sound spatialization device object of the invention.

With reference to the above-mentioned figure, the device that is the subject of the invention as represented on the figure 3a comprises, in addition to the spatial decoder shown, comprising the modules OTT ₀ to OTT ₄ substantially corresponding to a spatial decoder SD of the prior art as represented in FIG. figure 1c , but in which, in a manner known as such, of the state of the art, a summation of the front channel C and the low frequency channel Ife by an adder S is carried out, an equalization filtering module 1 delay of the signal in subband by applying a gain respectively a delay on the signal in subband.

On the figure 3a , the application of a gain is represented on each of the spatially coded audio channels, represented by amplifiers 1 _0a to 1 ₈ , the latter generating an equalized component which may or may not be delayed by means of delay elements noted 1 ₉ to 1 ₁₂ for generating from each of the spatially coded audio channels an equalized and delayed component of a determined delay value in the frequency subband SB _k .

With reference to the figure 3a , the gains of the amplifiers 1 ₀ to 1 ₈ have arbitrary values A, B, B, A, C, D, E, E, D respectively. In addition, the delay values applied by the delay modules ₁₉ to ₁₂ have the values Df, Bf, Ds, Ds. In the aforementioned figure, the structure of the gains and delays introduced is symmetrical. A non-symmetrical structure can be implemented without departing from the scope of the subject of the invention.

The device according to the invention also comprises a module 2 for adding a subset of equalized and delayed components to create a number of filtered signals in the transformed domain corresponding to the number N 'of the second set greater than or equal to two sound channels of restitution in the time domain.

Finally, the device which is the subject of the invention comprises a module 3 for synthesizing each of the filtered signals in the transformed domain to obtain the second set comprising a number N 'greater than or equal to two of sound reproduction signals in the time domain. The synthesis module 3 thus comprises, in the embodiment of the figure 3a , A synthesizer 3 ₀ 3 and ₁ which each can deliver a sound signal recovery in the time domain B ₁ louse left binaural signal, respectively B _r for right binaural signal.

• A[k] désignant le gain des amplificateurs 1₀, 1₃ pour la sous-bande SB_k de rang k,
• B[k] désigne le gain de l'amplificateur 1₁, 1₂ représenté en figure 3a,
• C[k] désigne le gain de l'amplificateur 1₄,
• D[k] désigne le gain des amplificateurs 1₅ 1_8,
• E[K] désigne le gain des amplificateurs 1₆ 1₇.

Equalized and delayed components in the embodiment of the figure 3a are obtained in the following manner with:

• A [k] denoting the gain of the amplifiers 1 ₀ , 1 ₃ for the sub-band SB _k of rank k,
• B [k] denotes the gain of the amplifier 1 ₁ , 1 ₂ represented in figure 3a ,
• C [k] denotes the gain of the amplifier 1 ₄ ,
• D [k] denotes the gain of the amplifiers 1 ₅ 1 _8,
• E [K] denotes the gain of amplifiers 1 ₆ 1 ₇ .

• A[k]*Fl[k][n],
• B[k]*Fl[k][n],
• B[k]*Fr[k][n],
• A[k]*Fr[k][n],
• C[k]*Fc[k][n],
• D[k]*Sl[k][n],
• E[k]*Sl[k][n],
• E[k]*Sr[k][n],
• D[k]*Sr[k][n].

with regard to the spatially coded audio channels and in particular these channels F1, Fr, Clfe, SI and Sr for the sub-band SB _k , denotes by FI [k] [n], Fr [k] [n], Fc [k] [n], Ife [k] [n], Sl [k] [n], Sr [k] [n], the ith sample of the subband SB _k . Thus each amplifier, 1 ₀ to 1 ₈ delivers the following equalized components successively:

• A [k] * Fl [k] [n]
• B [k] * Fl [k] [n]
• B [k] * En [k] [n]
• A [k] * En [k] [n]
• C [k] * Fc [k] [n],
• D [k] * Sl [k] [n]
• E [k] * Sl [k] [n],
• E [k] * Sr [k] [n],
• D [k] * Sr [k] [n].

The foregoing operations, as previously mentioned in the description, are realized in the form of a real multiplication acting in this case on complex numbers.

The delays introduced by the delay elements 1 ₉ , 1 ₁₀ , 1 ₁₁ and 1 ₁₂ are applied to the aforementioned equalized components to generate the equalized and delayed components.

In the example shown in figure 3a these delays are applied to the subset that does not benefit from a direct trajectory. These are, in the description of the figure 3a , the signals which have undergone the multiplications by the gains B [k] and E [k] applied by the amplifiers or multipliers 1 ₁ 1 ₂ and 1 ₆ and 1 ₇ .

A more detailed description of a filter or filtering element by equalization-delay constituted for example by a multiplier amplifier 1 ₁ and a delay element 1 ₉ will now be given in connection with the figure 3b .

As regards the application of the gain, it is indicated that the filtering element, corresponding, represented in FIG. figure 3b , comprises a numerical multiplier, that is to say one of the multipliers or amplifiers 1 ₀ to 1 ₈ and represented by the gain value g _kx in FIG. 3b, this multiplier allowing the multiplication of any complex sample of each encoded audio channel of index x corresponding to the channels Fl, Fr, Clfe, Sl, or Sr by a real value, that is to say the gain value previously mentioned in the description.

In addition, the filter element represented in figure 3b includes at least one complex numerical multiplier for introducing a complex-valued rotation of any sample of the subband signal by a complex exponential value, the exp (-j φ (k, SS _k )) value where φ (k , SS _k ) denotes a phase value which is a function of the sub-sampling rate of the sub-band considered and the rank of the sub-band considered k.

In one embodiment φ (k, SS _k ) = φ * ( k +0.5) * d / M.

The complex numerical multiplier is followed by a delay line denoted by LAR introducing a pure delay of each sample after rotation, making it possible to introduce a pure time delay as a function of the difference in the interaural delay of a listener and the subsampling rate. M in the sub-band SB _k considered.

Thus, the delay line L.A.R. allows to introduce the delay on the complex sample after rotation of the form y (k, n) = x (k, n-D).

Finally, it is indicated that the values of d and D are such that these values correspond to the application of a delay D * M + d in the non-sampled time domain and that the delay D * M + d corresponds to the interaural delay previously mentionned.

For the implementation of the device which is the subject of the invention, as represented in figure 3a it can be observed that the signal Fr [k] [n] is multiplied by the gain B [k] and then delayed, which, according to one of the remarkable aspects of the object of the invention, amounts to multiplying this signal by a complex gain. The product of the gain B [k] and the complex exponential can be realized once and for all thus avoiding a complementary operation for each successive sample Fr [k] [n]. The left equalized and delayed components are referenced L ₀ to L ₄ and straight lines R ₀ to R ₄ and represented in the drawing grouped by the somerator modules 2 ₀ respectively 2 ₁ , then verify the following relations: <u> Table T </ u> L0 [k] [n] = A [k] F1 [k] [n] R0 [k] [n] = B [k] F1 [k] [n] delayed from Df samples R1 [k] [n] = A [k] Fr [k] [n] L1 [k] [n] = B [k] Fr [k] [n] delayed from Df samples L2 [k] [n] = R2 [k] [n] = C [k] (Fc [k] [n] + 1fe [k] [n]) L3 [k] [n] = D [k] S1 [k] [n] R3 [k] [n] = E [k] S1 [k] [n] delayed from samples R4 [k] [n] = D [k] Sr [k] [n] L4 [k] [n] = E [k] Sr [k] [n] delayed from samples

• L0[k][n]+L1[k][n]+L2[k][n]+L3[k][n]+L4[k][n] pour le module sommateur 2₀, et
• R0[k][n]+R1[k][n]+R2[k][n]+R3[k][n]+R4[k][n] pour le module sommateur 2₁.

For audio output channels in the time domain, ie _the left channel B, respectively B _r right represented figure 3a i.e. binauralized signals in the embodiment of the figure 3a , we add for each sample of rank n the equalized and delayed components, that is to say the addition of the components:

• L0 [k] [n] + L1 [k] [n] + L2 [k] [n] + L3 [k] [n] + L4 [k] [n] for the summing module ₂₀ , and
• R0 [k] [n] + R1 [k] [n] + R2 [k] [n] + R3 [k] [n] + R4 [k] [n] for the summing module 2 ₁ .

The resulting signals output by the summation modules 2 ₀ and 2 ₁ are subsequently passed through the synthesis filter banks 3 3 ₀ respectively ₁ to obtain the binauralisés signals in the time domain B _l B _r respectively.

The aforesaid signals can then feed a digital-to-analog converter, in order to allow the listening of sounds left B _l and right B _r on an audio headset for example.

The synthesis process performed by the synthesis modules 3 ₀ 3 ₁ includes, where appropriate, the hybrid synthesis process as described above in the description.

The method which is the subject of the invention may advantageously consist in dissociating the equalization and delay operations, which may relate to frequency sub-bands in a different number. Alternatively, the equalization can for example be performed in the hybrid domain and the delay in the PQMF domain.

It will be understood that the method and the device that are the subject of the invention, although described for binauralising six channels to a headset, can also be applied to effect the trans-scaling, ie the rendering of a 3d sound field on a pair of tops or to convert in an uncomplicated manner a representation of N audio channels or sound sources from a spatial decoder or from several monophonic decoders to N 'available audio channels at the rendering level. The filtering operations can then be multiplied if necessary.

By way of nonlimiting complementary example, the method and the device which are the subject of the invention can be applied to the case of an interactive 3D game in the sounds emitted by the different objects or sound sources, which can then be spatialized as a function of their relative position in relation to the listener. Sound samples are then compressed and stored in different files or memory areas. To be played and spatialised, they are partially decoded in order to remain in the coded domain and are filtered in the coded domain by suitable binaural filters advantageously using the writing method according to the object of the present invention.

Indeed, by grouping the decoding and spatialization operations, the overall complexity of the process is greatly reduced without causing loss of quality.

The invention finally covers a computer program comprising a sequence of instructions stored on a storage medium for execution by a computer or a dedicated sound spatialization device, which during this execution performs the addition filtering and as described in connection with the Figures 2a to 2c and 3a, 3b previously in the description.

It is understood in particular that the operations shown in the above figures can advantageously be implemented on complex digital samples via a central processing unit, a working memory and a program memory, not shown. to the drawing of the figure 3a .

Finally, the calculation of the gains and delays constituting the equalization-delay filters can be performed externally to the device of the invention represented in FIG. figure 3a and 3b , as will be described below in liaison with the figure 4 .

With reference to the above-mentioned figure, a first spatial coding and rate reduction coding unit I is considered, including a device that is the subject of the invention as represented in FIG. figure 3a , 3b , making it possible to operate the above-mentioned spatial coding from an audio scene in 5.1 mode for example and the coded audio transmission, on the one hand, and spatial parameters, on the other hand, to a decoding and decoding unit spatial II.

The calculation of the delay equalization filters can then be performed by a separate unit III, which from the modeling filters, HRTF filters, calculates the gain and delay equalization values and transmits them to the coding unit I. spatial and spatial decoding unit II.

Spatial coding can thus take into account the HRTFs that will be applied to correct its spatial parameters and improve 3D rendering. Similarly, the rate reduction encoder can use these HRTFs to measure the perceptual effects of frequency quantization.

On the decoding side, it is the transmitted HRTFs that will be applied in the space decoder, and will enable the reconstruction of the restored channels if necessary.

• projection des 3 canaux reçus sur un ensemble de canaux virtuels (supérieur aux 5 de sortie) en utilisant les informations spatiales (upmix) ;
• réduction des canaux virtuels aux 5 canaux de sortie en utilisant les HRTF.

As in the previous examples, there are 2 ways from 5 that will be restored, but other cases may include the construction of 5 channels from 3 as illustrated above. The spatial decoding method will then proceed as follows:

• projection of the 3 received channels on a set of virtual channels (greater than the 5 of output) by using the spatial information (upmix);
• reducing the virtual channels to the 5 output channels using the HRTFs.

If the HRTFs have been applied to the encoder, then their contribution before upmix may be removed to achieve the above scheme.

linéairement ou logarithmiquement.The HRTF after conversion in their gain / delay form, can be quantified in a privileged way in the following form: differential coding of their values then quantification of their differences: if we call G [k] the values of the gains of the equalizer, then we will transmit the quantified values: $$\mathrm{e}\left[\mathrm{k}\right]\mathrm{=}\mathrm{BOY\; WUT}\left[\mathrm{k}\mathrm{+}\mathrm{1}\right]\mathrm{-}\mathrm{BOY\; WUT}\left[\mathrm{k}\right]\mathrm{,}$$

linearly or logarithmically.

More specifically with reference to the figure 4 above, the process implemented by the device and method of the invention thus makes it possible to carry out a sound spatialization of an audio scene in which the first set comprises a determined number of spatially coded audio channels and the second set comprises a lower number of sound reproduction channels in the time domain. It also allows decoding to perform an inverse transformation of a number of spatially coded audio channels to a set having a greater or equal number of time domain rendering sound channels.

Claims (17)

1. Method of sound spatialization of an audio scene comprising a first set, having a number, greater than or equal to unity, of audio channels spatially coded on a determined number of frequency sub-bands and decoded in a transformed domain, into a second set having a number greater than or equal to two of sound reproduction channels in the time domain, on the basis of filters for modelling the acoustic propagation of the audio signals of said first set of channels, characterized in that, for each modelling filter converted into the form of at least one gain and one delay which are applicable in said transformed domain, said method includes at least, for each frequency sub-band of said transformed domain:

   - the filtering by equalization-delay of the sub-band signal by applying a gain respectively a delay to said sub-band signal, so as to produce, on the basis of the spatially coded channels, an equalized component delayed by a determined delay value in the frequency sub-band considered;

   - the addition of a subset of equalized and delayed components, so as to create a number of filtered signals in the transformed domain corresponding to the number of said second set greater than or equal to two of sound reproduction channels in the time domain;

   - the synthesis of each of the filtered signals in the transformed domain by a synthesis filter, so as to obtain said second set in number greater than or equal to two of sound reproduction channels in the time domain.
2. Method according to Claim 1, characterized in that said filtering by equalization-delay of the sub-band signal includes at least the application of a phase shift for one at least of the frequency sub-bands.
3. Method according to Claim 2, characterized in that said filtering by equalization-delay furthermore includes a pure delay by storage for one at least of the frequency sub-bands.
4. Method according to one of Claims 1 to 3, characterized in that said filtering by equalization-delay in a hybrid transformed domain, comprises an additional step of frequency splitting into additional sub-bands without decimation, so as to increase the number of gain values applied, followed by a step of grouping said additional sub-bands to which said gain values have been applied, and then of applying said delay.
5. Method according to one of Claims 1 to 3, characterized in that said filtering by equalization-delay in a hybrid transformed domain comprises an additional step of frequency splitting into additional sub-bands with decimation, so as to increase the number of gain values applied, followed by a step of grouping said additional sub-bands to which said gain values have been applied, said grouping step itself being preceded or followed by the application of said delay.
6. Method according to one of the preceding claims, characterized in that, to convert each modelling filter into a value of gain respectively of delay in the transformed domain, the latter consists at least in:

   - associating as gain value with each sub-band a real value defined as the mean of the modulus of the modelling filter;

   - associating as delay value with each sub-band a delay value corresponding to the propagation delay between the left ear and the right ear for various positions.
7. Method according to one of Claims 1 to 3 or 6, with the exclusion of Claims 4 or 5, characterized in that the application of a gain in the PQMF domain consists in multiplying the value of each sample of the sub-band signal, represented by a complex value, by the gain value formed by a real number.
8. Method according to one of Claims 1 to 3 or 6 or 7, with the exclusion of Claims 4 or 5, characterized in that the application of a delay in the PQMF transformed domain consists at least, for each sample of the sub-band signal, represented by a complex value, in:

   - introducing a rotation in the complex plane by multiplying this sample by a complex exponential value dependent on the rank of the sub-band considered, on the rate of sub-sampling in the sub-band considered, and on a delay parameter related to the difference in interaural delay of a listener;

   - introducing a pure time delay of the sample after rotation, said pure time delay being a function of the difference of the interaural delay of a listener and of the rate of sub-sampling in the sub-band considered.
9. Method according to one of Claims 1 to 8, characterized in that for a binaural sound spatialization of an audio scene in which the first set comprises a number of spatially coded audio channels equal to N=6, in 5.1 mode, said second set comprises two sound reproduction channels in the time domain, for playback by an audio headset.
10. Method according to one of Claims 1 to 9, characterized in that the method is repeated for at least two equalization-delay pairs and the signals obtained are summed so as to obtain the sound channels in the time domain.
11. Method according to one of Claims 1 to 9, characterized in that for a sound spatialization of an audio scene in which the first set comprises a determined number of spatially coded audio channels and the second set comprises a lesser number of sound reproduction channels in the time domain, this method consists, on decoding, in performing an inverse transformation of a number of spatially coded audio channels to a set comprising a higher or equal number of sound reproduction channels in the time domain.
12. Method according to one of the preceding claims, characterized in that the gain and delay values associated with the modelling filter are transmitted in quantized form.
13. Device for the sound spatialization of an audio scene comprising a first set, having a number, greater than or equal to unity, of audio channels spatially coded on a determined number of frequency sub-bands and decoded in a transformed domain, into a second set having a number greater than or equal to two of sound reproduction channels in the time domain, on the basis of filters for modelling the acoustic propagation of the audio signals of said first set of channels, characterized in that, for each frequency sub-band of a spatial decoder, in the transformed domain, said device comprises, in addition to this spatial decoder:

    - means for the filtering by equalization-delay of the sub-band signal by applying at least one gain respectively one delay to said sub-band signal, so as to produce, on the basis of each of the spatially coded audio channels an equalized component delayed by a determined delay value in the frequency sub-band considered;

    - means for adding a subset of equalized and delayed components, so as to create a number of filtered signals in the transformed domain corresponding to the number of said second set greater than or equal to two of sound reproduction channels in the time domain;

    - means for the synthesis of each of the filtered signals in the transformed domain, so as to obtain said second set having a number greater than or equal to two of sound playback signals in the time domain.
14. Device according to Claim 13, characterized in that said means for filtering by applying a gain comprise a digital multiplier of any complex sample of each spatially coded audio channel by a real value.
15. Device according to Claim 13 or 14, characterized in that said means for filtering by applying a delay comprise at least one complex digital multiplier, making it possible to introduce a rotation in the complex plane of any sample of the sub-band signal by a complex exponential value, dependent on the rank of the sub-band considered, on the rate of sub-sampling in the sub-band considered and on a delay parameter related to the difference in interaural delay of a listener.
16. Device according to Claim 15, characterized in that said filtering means furthermore comprise a pure delay line of each sample after rotation, making it possible to introduce a pure time delay dependent on the difference of the interaural delay of a listener and of the sub-sampling rate in the sub-band considered.
17. Computer program comprising a series of instructions stored on a storage medium for execution by a computer or a dedicated device, characterized in that during this execution, said program executes the filtering, addition and synthesis steps according to one of Claims 1 to 12.

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