Multichannel audio data processing device for automobiles

Abstract

Multichannel audio data processing device for motor vehicles. A multichannel audio data processing device for motor vehicles includes a data storage unit (110) that receives multichannel audio data comprising rank audio data including a first audio channel and a second audio channel, a first rank digital filter, and a second rank digital filter for each rank. The first rank digital filter and a second rank digital filter for a given rank are determined from a digital filter transformation of rank correction data for each row of speakers. A computer (120) applies each first rank digital filter to all or part of the first audio channel of the rank audio data and each second rank digital filter to all or part of the second audio channel of the rank audio data. Fig. 1

Description

Title of the invention: Multichannel audio data processing device for motor vehicles

[0001] The invention relates to the field of audio processing for motor vehicles.

[0002] The sound quality of vehicle interiors is a constantly evolving field. For the past few decades, audio content has been predominantly in stereo and now includes a left and a right channel (and sometimes a center channel). With technological advancements, the number of speakers inside the vehicle has increased, to the point where they are replicated in every row of seats. Thus, these channels are generally associated with a specific row—front or rear in a conventional car, but potentially more in a van.

[0003] Improvements in sound quality have been accompanied by a growing increase in the computing power required to implement the desired audio processing. Generally, this processing is part of a suite of in-car processing systems, which also consume computing power. Consequently, in recent years, the pursuit of audio quality has been accompanied by a need to find a compromise in available computing power, as all audio processing must fit within a given available computing power envelope.

[0004] The sound design, i.e. the position of the instruments, as well as the tonality of the instruments, are currently corrected in a way that the Applicant considers unsatisfactory: background noise in the bass, spatial blur of the voice and instruments, energy gaps in the sound design, voice split in two, etc.

[0005] These problems are primarily related to the phase and timing issues of the car-cabin acoustics, which have not yet been corrected. Correcting these problems for each seat in the car simultaneously (in the case of multiple listening solutions) adds a further challenge that is very difficult to overcome.

[0006] There are hardware solutions that will not be detailed, but they are not relevant to the scope of the invention, which relates to a software solution. The Applicant has developed an algorithm called Stage that can address some of the problems mentioned by modifying the phase of certain frequency bands. However, the Applicant believes it can improve this solution.

[0007] The use of manually tuned infinite impulse response (IIR) all-pass filters can also be implemented. However, this type of solution is approximate and has very limited accuracy.

[0008] The invention improves the situation. To this end, it provides a multichannel audio data processing device for motor vehicles comprising a data storage system arranged to receive multichannel audio data including rank audio data associated with a respective speaker rank, the multichannel audio data being associated with at least one rank, and the rank audio data associated with a given speaker rank comprising a first audio channel and a second audio channel, a first rank digital filter and a second rank digital filter for each rank. The first rank digital filter and the second rank digital filter for a given rank are determined from correction data comprising rank correction data for each speaker rank associated with the multichannel audio data.the rank correction data associated with a given speaker rank being derived from the phase measurement of two signals from the reproduction of, on the one hand, the first audio channel in the given speaker rank and, on the other hand, the second audio channel in the given speaker rank, each time with one among a plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, these phase measurements being carried out, on the one hand, at a first location associated with the given speaker rank and used to determine a first set of rank phase differences in which each element is associated with the phase difference at the first location for the measurements obtained for an identical respective frequency for the first audio channel and the second audio channel,and on the other hand, in a second location associated with the given speaker rank and used to determine a second set of rank phase differences in which each element is associated with the phase difference at the second location for measurements obtained for the same respective frequency for the first and second audio channels. Each element of the rank correction data being associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, each Q element is associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, according to the formula C; - Dj + Pj + S^Fj where i is the index of a given frequency in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, Dt is calculated according to the formula t„„-1 / 52^^52^ j where Rlf is the value of the element , LdI1 \ sinÆlrsin«2; / index i in the first set of rank phase differences, and RZ is the value of the element with index i in the second set of rank phase differences, Pi is a phase correction value equal to 0 or ir and a function of the value of the sum cosZ?l7- + cosÆ2f-, S is a distribution factor between -1 and 1, and Fi is calculated according to the formula _ R2- + tt) mod 2tt — TT. The first rank digital filter and the second rank digital filter are derived from a data transformation. rank correction in a finite impulse response filter or an infinite impulse response filter. This device further includes a computer arranged to apply each first digital rank filter of a given rank to all or part of the first audio channel of the rank audio data associated with the given rank, and each second digital rank filter of a given rank to all or part of the second audio channel of the rank audio data associated with the given rank.

[0009] This device is particularly advantageous because it allows for phase alignment correction between the various frequencies for each position in the vehicle with limited computing power. Unlike the Stage algorithm or manual adjustment of all-pass filters, this solution is based on metrology; phase problems in the car are measured, and correction is performed using a measurement system. Furthermore, this correction can be customized, balanced between the first and second positions, or centered on the first or second position, albeit at the expense of the other position.

[0010] According to various embodiments, the invention may have one or more of the following features: - The first and second digital rank filters are determined from rank correction data where the Pi value is calculated according to the formula TT if Cos\j + Cos2; < h and P^ — 7T where h is a chosen hysteresis factor 0 if Cos^ + Cos2î > - h and PiA = 0 P* Æ if Cadj + Coslj< -h and 0 if Cos^ + Cos2i >h and PiA = tt in the range [0; 0,2], - the calculator is also arranged: * on the one hand, to receive as input audio data of input rank, to separate it into low-frequency audio data whose frequencies are below said threshold between 1.5 kHz and 5 kHz, and into high-frequency audio data whose frequencies are above said threshold between 1.5 kHz and 5 kHz, and * on the other hand, to return audio data of output rank comprising a first audio channel combining a signal taken from the first audio channel of the high-frequency audio data and a signal taken from the first audio channel of the low-frequency audio data corrected by the first digital filter of rank and a second audio channel combining a signal taken from the second audio channel of the high-frequency audio data and a signal taken from the second audio channel of the low-frequency audio data corrected by the second digital filter of rank, - the computer is arranged to downsample the low-frequency audio data, firstly to apply the first digital filter of rank to the first audio channel of the downsampled low-frequency audio data and upsample the resulting signal to produce the signal taken from the first audio channel of the low-frequency audio data corrected by the first digital filter of rank, and secondly, to apply the second digital filter of rank to the second audio channel of the downsampled low-frequency audio data and upsample the resulting signal to produce the signal taken from the second audio channel of the low-frequency audio data corrected by the second digital filter of rank, - the computer is configured to produce a signal from the first audio channel of high-frequency audio data and a signal from the second audio channel of high-frequency audio data with a delay chosen according to the rank correction data, and - the first digital filter of rank and the second digital filter of rank are obtained by transformation into a finite impulse response filter according to a distribution of the rank correction data such that C'first audio channel) -and Ci (second audio channel) = -(1- a)*^ where a is a real number between 0 and 1.

[0011] The invention also relates to a method for multichannel audio signal processing for motor vehicles comprising the following operations: a) receive multichannel audio data comprising rank audio data associated with a respective speaker rank, the multichannel audio data being associated with at least one rank, and the rank audio data associated with a given speaker rank comprising a first audio channel and a second audio channel, b) obtain a first rank digital filter and a second rank digital filter for each rank, the first rank digital filter and a second rank digital filter of a given rank being determined from correction data comprising rank correction data for each speaker rank associated with the multichannel audio data, the rank correction data associated with a given speaker rank being derived from the phase measurement of two signals resulting from the reproduction of, on the one hand, the first audio channel in the given speaker rank and, on the other hand, the second audio channel in the given speaker rank,each time with one among a plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, these phase measurements being carried out on the one hand at a first location associated with the given rank of loudspeakers and used to determine a first set of rank phase differences in which each element is associated with , the phase difference at the first location for measurements obtained for the same respective frequency for the first and second audio channels, and on the other hand at a second location associated with the given speaker rank and used to determine a second set of phase differences of rank in which each element is associated with the phase difference at the second location for measurements obtained for the same respective frequency for the first and second audio channels, each element of the rank correction data being associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, each element C, is associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, according to the formula C, = D) + Pj + S*F{ where i is the index of a given frequency in the plurality of frequencies below a threshold between 1,5 kHz and 5 kHz, is calculated according to the formula / cosR^-cosRii \ oq r 1; csl |has the value of the element with index i \ sinÆl^siiJK; / , in the first set of rank phase differences, and R^j is the value of the element with index i in the second set of rank phase differences, P is a phase correction value equal to 0 or ir and a function of the value of the sum cos Φ1^ + cos Φ1, S is a distribution factor between -1 and 1, and Fj is calculated according to the formula (RI - R2 + tt) mod 2tt - Φ1. The first rank digital filter and the second rank digital filter are derived from a transformation of the rank correction data into a finite impulse response filter or an infinite impulse response filter, and c) apply each first digital filter of rank of a given rank to all or part of the first audio channel of the audio data of rank associated with the given rank and each second digital filter of rank of a given rank to all or part of the second audio channel of the audio data of rank associated with the given rank.

[0012] According to various embodiments, this process may have one or more of the following characteristics: - The first and second rank digital filters of operation b) are determined from rank correction data where the value of Pi is calculated according to the formula TT if Cos 1; + Cos2; < h and - TT where h is a chosen hysteresis factor 0 if Cos + Cos2} > - h and PiA = 0 n if Cos+ Cos2; < -h and P^ = 0 0 if Cos lf + Cos2( > h and P^ = tt in the range [0; 0,2], - operation a) comprises separating the rank audio data into low-frequency rank audio data whose frequencies are below said threshold between 1.5 kHz and 5 kHz, and high-frequency rank audio data whose frequencies are above said threshold between 1.5 kHz and 5 kHz, and operation c) comprises returning rank audio data comprising a first audio channel combining a signal taken from the first audio channel of the high-frequency rank audio data and a signal taken from the first audio channel of the low-frequency rank audio data corrected by the first rank digital filter and a second audio channel combining a signal taken from the second audio channel of the high-frequency rank audio data and a signal taken from the second audio channel of the low-frequency rank audio data corrected by the second rank digital filter, - operation c) includes c) downsampling the low-frequency audio data, firstly to apply the first digital filter of rank to the first audio channel of the downsampled low-frequency audio data and upsampling the resulting signal to produce the signal taken from the first audio channel of the low-frequency audio data corrected by the first digital filter of rank, and secondly, to apply the second digital filter of rank to the second audio channel of the downsampled low-frequency audio data and upsampling the resulting signal to produce the signal taken from the second audio channel of the low-frequency audio data corrected by the second digital filter of rank,and c2) combine on the one hand the signal taken from the first audio channel of the low-frequency audio data corrected by the first digital filter of rank with the first audio channel of the high-frequency audio signal and on the other hand the signal taken from the second audio channel of the low-frequency audio data corrected by the second digital filter of rank with the second audio channel of the high-frequency audio signal to produce the returned audio data of rank, , - operation c) comprises producing a signal from the first audio channel of the high-frequency audio data and a signal from the second audio channel of the high-frequency audio data having a delay chosen according to the rank correction data, and - the first digital filter of rank and the second digital filter of rank are obtained by transformation into a finite impulse response filter according to a distribution of the rank correction data such that C'first audio channel) -and Ci (second audio channel) = -(1- a)*^ where a is a real number between 0 and 1.

[0013] The invention also relates to a computer program implemented by a computer comprising instructions for executing the process according to the invention, and a data storage medium on which such a computer program is recorded.

[0014] Other features and advantages of the invention will become more apparent upon reading the following description, taken from illustrative and non-limiting examples shown in the drawings: - Figure [1] represents a schematic top view of a vehicle implementing the device according to the invention, and - Figure 2 represents a generic block diagram of the device used in Figure 1.

[0015] The drawings and description below contain, essentially, elements of a definite nature. They may therefore not only serve to better understand the present invention, but also contribute to its definition, if necessary.

[0016] Fig. 1 represents a schematic top view of a vehicle implementing the device according to the invention.

[0017] As can be seen in this figure, a vehicle 2 has two rows of seats: a front row 4 and a rear row 6. The front row 4 comprises a driver's seat 8 and a passenger seat 10. The rear row 6 comprises a left rear seat 12 and a right rear seat 14. Alternatively, the rear row 6 could have more seats, and the vehicle could have a third or more rows. Also alternatively, there may be a single row of seats.

[0018] In the example described here, the front row 4 includes a front left speaker 16 and a front right speaker 18, while the rear row 6 includes a rear left speaker 20 and a rear right speaker 22.

[0019] The Applicant, seeking to improve the Stage algorithm, became interested in conducting measurements in vehicle interiors. Indeed, obtaining a wide and coherent soundstage at a given location relies largely on the fact that, for a given signal, the frequencies it contains reach that location simultaneously. However, sound processing and the specific shape of the vehicle interior mean that the frequencies do not reach the ear identically and do not propagate identically. This is particularly pronounced in the case of the portion of multichannel audio signals that is identical for all channels and is generally referred to as the "mono portion" or "correlated portion."

[0020] In doing so, the Applicant discovered that it is advantageous to address the phase row by row, since the seats in a given row primarily receive the sound channels from the speakers in that row. Thus, initially, the Applicant carried out separate linear frequency sweep measurements at each location in each row, for each audio channel in each row.

[0021] Thus, it was possible to obtain vectors that represent, for a large number of frequencies, the phase of the signal reaching each location. To be more explicit, these vectors can be designated by specifying the location at which they were measured, the rank to which the location belongs, and the channel from which they are derived.

[0022] For example, the vector associated with location 1 in row 1 (i.e. pilot seat 8 on the [Fig.l]) and the first audio channel (e.g. left channel) of row 1 will be denoted FL1, the vector associated with location 1 in row 1 (i.e. pilot seat 8 on the [Fig.l]) and the second audio channel (e.g. right channel) of row 1 will be denoted FRI, the vector associated with location 2 in row 1 (i.e. passenger seat 10 on the [Fig.l]) and the first audio channel (e.g. left channel) of row 1 will be denoted FL2, the vector associated with location 2 in row 1 (i.e. passenger seat 10 on the [Fig.l]) and the second audio channel (e.g. right channel) of row 1 will be denoted FR2. For rank 2, the notations will be similar, but the first letter of the vector will change to indicate this rank (e.g., "G", or any other letter).

[0023] Each vector comprises phase measurements for a plurality of frequencies below a chosen threshold. The Applicant has found that a threshold between 1.5 kHz and 5 kHz gives the best results.

[0024] Indeed, phase alignment effects are particularly noticeable at the lowest frequencies and are no longer perceptible above 5 kHz: perceiving phase effects at this frequency requires almost perfect hearing, while keeping the head within 5 cm – and the higher the frequency, the lower the accuracy of the human ear becomes, while the possible head movement decreases. Keeping the head perfectly still within 5 cm in a moving vehicle already seems highly unlikely.

[0025] As will be seen below, the filter which corresponds to the correction calculated using the measurements of the phase vectors can be implemented by a finite impulse response filter ("FIR" in English) or an infinite impulse response filter ("IIR" in English).

[0026] The threshold value determines, on the one hand, the frequencies at which phase alignment is achieved, and, on the other hand, via the Nyquist condition, the size of the filter that must be used to implement it. However, the higher the frequency, the greater the number of coefficients in the filter must be for comparable correction accuracy, and the greater the computing power required to implement it. The lower the threshold, the less computing power is required, but the correction is then less effective. The Applicant determined that with a threshold below 1.5 kHz, the correction no longer offers satisfactory quality. The range between 1.5 kHz and 5 kHz was therefore determined to offer the best compromise. for the implementation of the invention. Preferably, the threshold range may be between 2 kHz and 3 kHz, and the preferred threshold value is 3 kHz as it represents the best compromise between computing power and correction quality in the Applicant's tests. As will be seen below, the correction is implemented using a digital filter. The Applicant has found that it is preferable for the phase measurement resolution for the corrections to be greater than or equal to that of the filter.

[0027] Once the phase vectors were measured, the Applicant worked on a solution to correct the phase alignment at each location of a given rank. Its work showed that it is advantageous to work on the phase differences at each location. Thus, the vectors FL1, FRI, FL2, and FR2 were combined into two phase difference vectors, RI and R2, according to the formulas R1 = FL1 - FR1 and R2 = FL2 - FR2. This means that the vector RI contains the phase difference at location 1 of rank 1 between the phases of the first and second audio channels. Similarly, the vector R2 contains the phase difference at location 2 of rank 1 between the phases of the first and second audio channels.

[0028] It then becomes possible to correct the phase alignment of the various vectors by applying a correction based on three correction factors: a) a phase inversion factor P, which consists, for each frequency, of adding a phase correction of value Æ if the sum of the cosines of the phase difference at the two locations is less than 0, b) a symmetrization regularization factor D, and c) a correction location adjustment factor S*F.

[0029] The Applicant has discovered that the phase inversion factor P can be determined in two distinct ways: with and without hysteresis. The correction without hysteresis is based on the assumption that the measurements are perfectly reliable. In this case, the value of each element Pi of the phase inversion vector P is determined according to the following formula: - if cosR 1; + cos / ?2(- < 0, P} = tt - if cosKl; + cosR2j > 0, P{ = 0.

[0030] However, the measurement of phases, and therefore their difference, is necessarily noisy. This means that around 0 for the sum cos(π / 1, + cos(π / 2)), the value of Pi at neighboring frequencies can vary significantly, solely due to measurement noise, which is likely to generate artifacts in the correction. Even if the measurements were perfect, if the sum cos(π / 1, + cos(π / 2)) varies very rapidly around 0, this can pose a problem in the design of a digital filter. Indeed, it is complex, if not impossible, to vary a digital filter successively at such a rapid rate.

[0031] For this reason, the Applicant determined the correction with hysteresis, according to the The following formula: TT if cosR 1,-+cosÆ2; < h and P = n I l {“1 0 if cosÆl, + cosR2i> -h et= 0 7i if cosRlz- + cosÆ2; < -h and P^ = 0 0 if cosRïi + cosR2i >h and P^ = æ

[0032] This means that, around 0, and within a range of width h, the value of Pi does not change in order to preserve the stability of the phase correction. Therefore, the value of cos(θ) + cos(θ) must have an absolute value exceeding h to go from a correction of 0 to θ or from Φ to 0. Consequently, some frequencies will not be corrected when they should have been, but artifacts due to measurement noise will be avoided. The Applicant has determined that values ​​of h chosen in the range [0; 0.2] give the best results. Incidentally, the value of h equal to zero corresponds to the formula without hysteresis.

[0033] The symmetrization regularization factor D is determined according to the following formula: cosXl,-cosX2,. \ sinX l-sinXÎ, /

[0034] The regularization factor D ensures symmetrical performance between the left and right seats: the recombination of the right and left channels in both seats will be identical. Perceptually, this results in a soundstage quality and frequency response that are as similar as possible to each other in different locations within the car.

[0035] Finally, the correction location adjustment factor S*F allows the phase correction to be adjusted to favor the correction at a particular location in the rank. This is obtained first by determining the vector F whose elements Fi are calculated by the formula — R^-+ yj mod 2tt tt' and in 'c scalar adjustment S to the vector F.

[0036] If S is equal to 0, then the phase correction is perceived as the same at the first and second locations. If S is equal to 1, the phase correction will be optimal for location 1. If S is equal to -1, the phase correction will be optimal for location 2. This factor therefore allows the perceived quality of the phase correction to be modulated for a given location. Thus, when a person is driving alone, it will be very advantageous to use a value of S equal to 1. Similarly, it may be advantageous to switch between various filters corresponding to different values ​​of S depending on the situation: for example, in the event of an interruption of the audio program to provide information (e.g., traffic information on the radio or the (interruptions in GPS guides), it may be advantageous to use a filter in which S equals 1 to promote understanding by the driver.

[0037] Finally, the phase correction vector can be defined as the sum of the three vectors of the phase inversion factor, the symmetrization regularization factor D, and the correction location adjustment factor, i.e., a correction vector C, each element of which Ci is obtained by the formula Ct = D{ + P; +

[0038] Once the correction vector C has been determined for each rank and for each desired value of S, it is transformed into a digital filter for use in the motor vehicle. For this, within the framework of a finite impulse response filter, a design based on least mean squares (LMS) or an integration and windowing design can be used.

[0039] In the case of an infinite impulse response filter, the design can be based on least squares. Great care must then be taken to handle the divergence cases that characterize infinite impulse response filters. For this reason, although they consume more computing power, finite impulse response filters are preferred by the Applicant.

[0040] This results in one or more digital filters for each rank and for each value of S that is to be used. These filters are to be applied exclusively to the first audio channel, or the second audio channel.

[0041] Alternatively, the Applicant has discovered that it can be advantageous to produce digital filters for each audio channel, which allows the correction to be distributed and thus limits the risk of filler saturation in the digital filter. In this case, a correction factor a is used to define a correction filter for the first channel and a correction filter for the second channel, according to the formula G (first audio channel) = a*Ci and q (second audio channel) = - (1 - a)*Cj, with a chosen from the range between 0 and 1. The value of 0.5 optimizes the risk of saturation since half of the correction is carried by each audio channel in the resulting digital filters.

[0042] Once this research work was completed, the Applicant was therefore able to implement a multichannel audio data processing device for motor vehicle 100 as represented in [Fig.2].

[0043] The device 100 includes a data storage 110 and a calculator 120.

[0044] Data storage 100 receives two types of data: on the one hand, the data multichannel audio to be processed 104, and on the other hand the data defining the digital phase correction filters for each rank 108.

[0045] Data storage 100 can be any type of data storage suitable for receiving digital data: hard disk, hard disk with flash memory, flash memory in any form, RAM, optical disk, locally distributed storage or in the cloud, etc.

[0046] The computer (PC) 120 may include one or more processors (P) 122. In some embodiments in which the computer 120 includes a programmable processor, a computer program product (CPP) 124 may be provided to implement the phase correction processing. The CPP 124 stores a computer program (CP) 126 that includes computer readable instructions (CRI) 128. The CPP 124 computer program can be stored on a computer readable medium (CRM) 130, which can be a non-transitory computer readable medium, such as a magnetic medium (hard drive, SSD, magnetic tape, etc.), an optical medium (CD, DVD, Blu-ray, etc.), a memory (RAM, flash memory, etc.).), distributed or cloud storage, etc. Computer-readable media (CRM) 130 can also be stored in data storage 100.

[0047] The P 122 processor(s) can be any processor suitable for the calculations described below. Such a processor can be implemented in any known form, such as a microprocessor for a personal computer, laptop, tablet, or smartphone; a dedicated digital signal processor (DSP); a dedicated FPGA or SoC chip; a computing resource on a grid or in the cloud; a graphics processing unit (GPU); a microcontroller; or any other form suitable for providing the computing power necessary for the implementation described below. One or more of these elements can also be implemented as specialized electronic circuits such as an ASIC. A combination of processor(s) and electronic circuits can also be considered. Processors dedicated to machine learning may also be considered.

[0048] Since the correction filter is limited to frequencies below the threshold between 1.5 kHz and 5 kHz, the computer 120 filters the incoming multichannel audio signal into a high-frequency signal with frequencies above the threshold and a low-frequency signal with frequencies below the threshold. This means that each channel in each rank of the multichannel audio signal is separated into a low-frequency component and a high-frequency component.

[0049] Next, the computer 120 applies the corresponding digital filter to each channel of each rank of the low-frequency signal, and then it recombines the signals resulting with the high-frequency signal from each channel to produce the corrected audio signal.

[0050] Advantageously, the Applicant has discovered that two treatments can be applied to further improve the accuracy and / or the computing power consumed: - The 120 calculator can be arranged to downsample the low-frequency signal, apply digital filters to it, then upsample it before recombining it with the high-frequency signal.

[0051] This allows the use of even less expensive digital filters. - Calculator 120 can determine a delay factor for the high-frequency signal based on the C vector determined for the relevant rank. Indeed, the C vector will delay some frequencies in the low-frequency signal, and it is desirable to align the high-frequency signal as closely as possible with these delays, as well as the delay induced by the design of the digital filter.

Claims

1. Demands Multichannel audio data processing device for motor vehicles including - a data storage (110) arranged to receive * multichannel audio data comprising rank audio data associated with a respective speaker rank, the multichannel audio data being associated with at least one rank, and the rank audio data associated with a given speaker rank comprising a first audio channel and a second audio channel, * a first digital filter of rank and a second digital filter of rank for each rank, the first digital filter of rank and a second digital filter of rank of a given rank being determined from correction data comprising rank correction data for each row of loudspeakers associated with the multichannel audio data, the rank correction data associated with a given row of loudspeakers being taken from the phase measurement of two signals resulting from the reproduction of, on the one hand, the first audio channel in the given row of loudspeakers and, on the other hand, the second audio channel in the given row of loudspeakers, each time with one among a plurality of frequencies below a threshold between 1.5 kHz and 5 kHz,These phase measurements are carried out, on the one hand, at a first location associated with the given speaker row and used to determine a first set of phase differences of rank in which each element is associated with the phase difference at the first location for measurements obtained for an identical respective frequency for the first audio channel and the second audio channel, and on the other hand, in a second location associated with the given speaker rank and used to determine a second set of rank phase differences in which each element is associated with the phase difference at the second location for measurements obtained for the same respective frequency for the first and second audio channels, each element of the rank correction data being associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, each element Cf is associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, according to the formula Ci = ^t + Pi + S^Fj where * i is the index of a given frequency in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, * D{ is calculated according to the formula \ where ri. csl |a \ siuRlfSinÆS; / value of the element of index i in the first set of phase differences of rank, and is the value of the element of index i in the second set of phase differences of rank * Pi is a phase correction value equal to 0 or ir and a function of the value of the sum cosZ?l;+ cos / f2j, * S is a distribution factor between -1 and 1, and * F{ is calculated according to the formula R2- + tt) mod Zît — TV the first digital filter of rank and the second digital filter of rank being taken from a transformation of the rank correction data into a finite impulse response filter or an infinite impulse response filter, the device (100) further comprising a computer (120) arranged to apply each first digital filter of rank of a given rank to all or part of the first audio channel of the audio data of rank associated with the given rank and each second digital filter of rank of a given rank to all or part of the second audio channel of the audio data of rank associated with the given rank.;

2. Device according to claim 1, wherein the first digital rank filter and the second digital rank filter are determined from rank correction data in the value Pi is calculated according to the formula tt if Cos lr- + Cos2{ < h and P iA-jr 0 if Cos\ + Cos!:> -h and Pn = 0 p. - 1 tt if Cos\i + Coslj < -h and Pi} - 0 0 if Cos 1, + Coslj > h and PiA - tt where h is a hysteresis factor chosen in the range [0; 0,2].

3. A device according to any one of the preceding claims, wherein the computer (120) is further arranged: * on the one hand, to receive input-rank audio data as input, and to separate it into low-frequency-rank audio data whose frequencies are below said threshold between 1.5 kHz and 5 kHz, and in high frequency audio data whose frequencies are above said threshold between 1.5 kHz and 5 kHz, and * on the other hand to return audio data of output rank comprising a first audio channel combining a signal taken from the first audio channel of the high frequency rank audio data and a signal taken from the first audio channel of the low frequency rank audio data corrected by the first digital filter of rank and a second audio channel combining a signal taken from the second audio channel of the high frequency rank audio data and a signal taken from the second audio channel of the low frequency rank audio data corrected by the second digital filter of rank.

4. Device according to claim 3, wherein the computer (120) is arranged to downsample the low-frequency rank audio data, on the one hand to apply the first rank digital filter to the first audio channel of the downsampled low-frequency rank audio data and upsample the resulting signal to produce the signal taken from the first audio channel of the low-frequency rank audio data corrected by the first rank digital filter, and on the other hand, to apply the second rank digital filter to the second audio channel of the downsampled low-frequency rank audio data and upsample the resulting signal to produce the signal taken from the second audio channel of the low-frequency rank audio data corrected by the second rank digital filter.

5. Device according to claim 3 or 4, wherein the computer (120) is arranged to produce a signal taken from the first audio channel of the high-frequency rank audio data and a signal taken from the second audio channel of the high-frequency rank audio data having a delay selected according to the rank correction data.

6. A device according to any one of the preceding claims, wherein the first digital filter of rank and the second digital filter of rank are obtained by transformation into a finite impulse response filter according to a distribution of the rank correction data such that first audio channel) = a*C{ and

7. Cj (second audio channel) = - (1 - a)*Q where a is a real number between 0 and 1. A method for processing multichannel audio signals for motor vehicles comprising the following operations: a) receive multichannel audio data comprising rank audio data associated with a respective speaker rank, the multichannel audio data being associated with at least one rank, and the rank audio data associated with a given speaker rank comprising a first audio channel and a second audio channel, b) obtain a first rank digital filter and a second rank digital filter for each rank, the first rank digital filter and a second rank digital filter of a given rank being determined from correction data comprising rank correction data for each speaker rank associated with the multichannel audio data, the rank correction data associated with a given speaker rank being derived from the phase measurement of two signals resulting from the reproduction of, on the one hand, the first audio channel in the given speaker rank and, on the other hand, the second audio channel in the given speaker rank,each time with one from a plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, these phase measurements being carried out, on the one hand in a first location associated with the given speaker rank and used to determine a first set of rank phase differences in which each element is associated with the phase difference at the first location for measurements obtained for an identical respective frequency for the first audio channel and the second audio channel, and on the other hand in a second location associated with the given speaker rank and used to determine a second set of rank phase differences in which each element is associated with the phase difference at the second location for measurements obtained for the same respective frequency for the first and second audio channels, each element of the rank correction data being associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, each element Q is associated with one of the frequencies in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, according to the formula Ci = ^t + Pi + S^Fj where * i is the index of a given frequency in the plurality of frequencies below a threshold between 1.5 kHz and 5 kHz, * D{ is calculated according to the formula \ where ri.csl |a \ siuRlfSinÆS; / value of the element of index i in the first set of phase differences of rank, and Ri is the value of the element of index i in the second set of phase differences of rank * Pi is a phase correction value equal to 0 or ir and a function of the value of the sum cosZ?l; + cos / f2j, * S is a distribution factor between -1 and 1, and * F{ is calculated according to the formula R2- + tt) mod 2tt — TV the first digital filter of rank and the second digital filter of rank being taken from a transformation of the rank correction data into a finite impulse response filter or an infinite impulse response filter, c) apply each first digital filter of rank of a given rank to all or part of the first audio channel of the audio data of rank associated with the given rank and each second digital filter of rank of a given rank to all or part of the second audio channel of the audio data of rank associated with the given rank.

8. Method according to claim 7, wherein the first digital rank filter and the second digital rank filter of operation b) are determined from rank correction data in the value Pi is calculated according to the formula tt if Cosl^ + Coslf < h and PiA = tt 0 if Cos L + Cos2f > - h and Pn = 0 P.= { L ît if Coslf + Cos2( £ -h and PiA = 0 0 if Cos I,- + Cos2{ > h and P^ j = tt where h is a hysteresis factor chosen in the range [0; 0.2].

9. A method according to any one of claims 7 and 8, wherein the operation a) comprises separating the audio data into low-frequency audio data whose frequencies are below said threshold between 1.5 kHz and 5 kHz, and into high-frequency audio data whose frequencies are above said threshold between 1.5 kHz and 5 kHz. kHz and 5 kHz, and operation c) includes returning rank audio data comprising a first audio channel combining a signal taken from the first audio channel of the high-frequency rank audio data and a signal taken from the first audio channel of the low-frequency rank audio data corrected by the first rank digital filter and a second audio channel combining a signal taken from the second audio channel of the high-frequency rank audio data and a signal taken from the second audio channel of the low-frequency rank audio data corrected by the second rank digital filter.

10. A method according to claim 9, wherein operation c) comprises c) downsampling the low-frequency audio data, on the one hand to apply the first digital filter of rank to the first audio channel of the downsampled low-frequency audio data and upsampling the resulting signal to produce the signal from the first audio channel of the low-frequency audio data corrected by the first digital filter of rank, and on the other hand, to apply the second digital filter of rank to the second audio channel of the downsampled low-frequency audio data and upsampling the resulting signal to produce the signal from the second audio channel of the low-frequency audio data corrected by the second digital filter of rank,and c2) combine, on the one hand, the signal from the first audio channel of the low-frequency audio data corrected by the first digital filter of rank with the first audio channel of the high-frequency audio signal, and on the other hand, the signal from the second audio channel of the low-frequency audio data corrected by the second digital filter of rank with the second audio channel of the high-frequency audio signal to produce the returned audio data of rank.

11. A method according to claim 9 or 10, wherein operation c) comprises producing a signal taken from the first audio channel of the high-frequency rank audio data and a signal taken from the second audio channel of the high-frequency rank audio data having a delay selected according to the rank correction data.

12. A method according to any one of claims 7 to 11, wherein the first digital filter of rank and the second digital filter of rank are obtained by transformation into a finite impulse response filter according to a data distribution of 20 rank correction such that first audio channel ) - a*Ct and Ci ( second audio channel ) = -(1- a)*Ci where a is a real number between 0 and 1.

13. Computer-implemented computer program comprising instructions for carrying out the process according to any one of claims 7 to 12.

14. Data storage medium on which the computer program according to claim 13 is recorded.

Images