EP0719070A1 - Sound pick-up device, comprising a video system for the adjustment of parameters and process for adjusting - Google Patents

Abstract

The equipment includes an array (600) of sensors (610) which pick-up sounds from several sources, and corresp. signals are transmitted to a control unit (300) where the coeffts. of the digital filters (310) are affected by reception channel characteristic parameters evaluated and adjusted (410) individually. The variations of these parameters are represented by an image produced by video signal generators (510) and mixers (520) for superposition on the image displayed on the screen (200) from the video camera (100). The latter communicates its focal distance to the control unit for relating horizontal and vertical angles to abscissae and ordinates.

Description

The invention relates to a sound recording device with which a video pointing system has been associated.

This device can be particularly useful in certain applications and in particular during conferences, concerts or any other event deserving of perfect quality sound recording.

The device according to the invention makes it possible to simultaneously and independently capture the sounds coming from several sound sources, without having to approach the sensors close to these sources, while giving the auditory impression that the sound is picked up near each source. For this, it reduces the reverberation of the sound as well as the level of ambient noise.

Many sound recording devices have already been developed with a view to picking up sounds without having to approach the sensors close to the sources.

These devices include networks of sensors, a control unit using in particular filters for processing the signals received by the sensors, and means for adjusting the parameters characteristic of the sound recording.

However, such devices do not make it possible to independently adjust the characteristic parameters of each of the sound reception channels, in order to separately capture the sounds coming from several sound sources. Nor can one control the variations of these parameters with respect to the position and size of the sound sources from which the sounds are picked up.

Patent EP 0 381 498 further describes a sound pickup device comprising a circuit for changing the coefficients of the digital filters which makes it possible to vary, arbitrarily, the directional characteristics of the sound reception channels.

However, when changing the filter coefficients, small disturbances are audible and affect the sound quality. These disturbances are due to the sudden change in the set of parameters characteristic of the sound recording, which is governed by the change in the coefficients of the filters.

These devices do not make it possible, moreover, to obtain all the precision desired for the adjustment of the parameters characteristic of the sound reception channels.

The present invention overcomes this problem. Its object is in fact a device comprising a network of sensor elements, a control unit using in particular filters for processing the signals received by the sensors, a camera and a video screen. The camera provides a video signal to the screen corresponding to the image of the area where the sound sources are located from which the sound is captured. The video screen, for its part, makes it possible to visualize, at the same time, the sound sources, filmed by the camera, and the variations of the characteristic parameters of each of the sound reception channels. Thus, a very precise adjustment of the parameters is carried out, taking into account the position and the size of the sound sources.

A more particular subject of the invention is a device for taking sounds, comprising a network of sensors, a control unit and means for adjustment of the characteristic parameters of the sound recording, mainly characterized in that it further comprises a video camera, a video screen which displays a first video image, corresponding to the signal from the camera, and means for coupling the screen with means for adjusting the characteristic parameters of each of the sound reception channels. These coupling means make it possible to produce, for each of the sound reception channels, another video image, showing variations in the characteristic parameters, and to superimpose this image on the first video image, so as to control the adjustment of these parameters. .

• filtrage des signaux captés par les filtres à interpolation linéaire,
• modification des coefficients des filtres pour chaque modification de paramètres,
• interpolation linéaire dans le temps, à chaque instant d'échantillonnage, entre deux valeurs, correspondant au renouvellement des filtres, qui sont modifiées à cadence régulière mais plus lente que la fréquence d'échantillonnage.

The invention also relates to a method for adjusting the parameters characteristic of the sound recording, characterized in that the control unit performs processing on the signals picked up and on the signals, corresponding to the values of the parameters, supplied by the means adjustment, comprising the following steps:

• filtering of signals picked up by filters with linear interpolation,
• modification of the filter coefficients for each modification of parameters,
• linear interpolation in time, at each sampling instant, between two values, corresponding to the renewal of the filters, which are modified at a regular rate but slower than the sampling frequency.

• la figure 1, une vue d'ensemble d'un dispositif selon l'invention,
• la figure 2, un schéma plus détaillé du dispositif de la figure 1,
• la figure 3, un schéma de réalisation d'un capteur,
• la figure 4, un schéma de réalisation d'un moyen de réglage des paramètres caractéristiques d'une voie r de réception du son.

Other features and advantages of the invention will appear on reading the description given by way of illustrative and nonlimiting example with reference to the appended figures which represent:

• FIG. 1, an overall view of a device according to the invention,
• FIG. 2, a more detailed diagram of the device of FIG. 1,
• FIG. 3, a diagram of an embodiment of a sensor,
• FIG. 4, a diagram of an embodiment of a means for adjusting the parameters characteristic of a channel r for receiving sound.

An embodiment of a device according to the invention will be better understood with reference to Figure 1 which describes an overview of such a device.

First, the optical field of a camera 100 covers the entire area where there are sound sources from which the device picks up sounds. The video signal from the camera is then transmitted to a video screen 200 which displays a corresponding first video image. The concept of screen covers any type of screen such as, for example, the screen of a video monitor.

The camera supplies, on the other hand, a value of its focal length to a control unit 300, this value is useful for performing angle calculations which will be described in more detail in the following.

Secondly, adjustment means 400 make it possible to adjust the characteristic parameters of each of the sound reception channels. The signal from these adjustment means 400 is transmitted to coupling means 500 of the screen 200 to the adjustment means 400. The coupling means 500 make it possible to produce, for each of the sound reception channels, another video image. and superimpose it on the first image. This superimposition of images makes it possible to perform a precise adjustment of the characteristic parameters of each sound reception channel, and to control the variations of this adjustment with respect to the position and size of the sound sources from which the device picks up the sounds.

The signal representing the values of the adjustment parameters is also transmitted to the control unit 300 which has in particular the task of filtering the signals received by the network 600 of sensors, and of updating, periodically, the coefficients of the filters.

Figure 2 provides a more detailed diagram of a device according to the invention.

The network 600 of sensors comprises a number M of sensors 610 having the task of capturing the sounds coming from several sound sources and of transmitting the corresponding signals to a control unit 300. This control unit then processes these signals, in particular by filtering. The number M of sensors 610 is preferably at least equal to 2 and the number m associated with each sensor 610 therefore varies from 1 to M.

To be able to process the signals received by the M sensors 610, the control unit must also know the values of the characteristic parameters of each sound reception channel. This is why signals, corresponding to the values of these parameters, are sent from the network 400 of adjustment means to the control unit. This network 400 comprises a number R of adjustment means 410. Each of these adjustment means 410 of the parameters characteristic of the sound pickup corresponds to a sound reception channel. The number R of adjusting means 410, and consequently the number of sound reception channels, is preferably at least equal to 1 and the number r associated with each of these means therefore varies from 1 to R.

In addition, each channel r for receiving sound is associated with an output 710 where the signals are available.

To achieve a superimposition of video images, means 500 of the video screen 200 are introduced into the structure of the device to the adjustment means 400.

These coupling means 500 advantageously comprise, for each of the means 410 for adjusting the parameters of each of the sound reception channels, a video generator 510 and a video mixer 520. The video generator 510 makes it possible to transform the signal from of the corresponding adjustment means, into a video signal. The mixer 520 makes it possible to mix, between them, the signals coming from the video generators, and to also mix them with the signal coming from the camera. The signal from the last video mixer is then sent to the screen 200. Thus, one obtains, on the first video image corresponding to the signal from the camera, a superposition of the images showing the variations of the characteristic parameters of each of the channels of sound reception.

FIG. 3 illustrates the construction of a sensor 610. Such a sensor comprises a microphone 611, a preamplifier 612, a low-pass filter 613 and an analog-digital converter 614.

The signal picked up by the microphone 611 is injected into a preamplifier 612 then is filtered by the low-pass filter 613 to eliminate the spectral aliasing that could be introduced by the analog-digital converter 614. Each sensor receives a clock signal which regulates the frequency sampling of the converter 614. The sampled signal is quantified by the converter 614 and is transmitted, in digital form, to the control unit which will process it.

A preferred diagram of the realization of a means of adjustment, 410, of the parameters corresponding to a reception channel r is illustrated in FIG. 4.

A command 411 makes it possible to set the values of the characteristic parameters of the channel r for receiving the corresponding sound. This control can be mechanical or electronic. This will be, for example, a joystick, a rotary or linear button, a mouse acting on a potentiometer.

Each of the parameters is converted into a digital value, at a fixed rate, by an analog-digital converter 412. These digital values are advantageously between 1 and a limit value.

The sampling rate of the values will preferably be lower than the sampling rate in the sensors 610. For example, a value of 25 Hz is chosen. According to a variant, it is also possible to choose a value in the frequency range between 1Hz and 50Hz.

After sampling in the converters 412, the set of parameter values is transmitted to the control unit 300 so that it proceeds to the processing of the signals.

• l'abscisse du point visé sur l'écran vidéo,
• l'ordonnée du point visé sur l'écran,
• la largeur de la voie r de réception formée notée c(r),
• la hauteur de la voie r de réception formée, notée d(r),
• la profondeur de la voie r de réception formée notée, p(r).

The characteristic adjustment parameters of each sound reception channel r are as follows:

• the abscissa of the target point on the video screen,
• the ordinate of the target point on the screen,
• the width of the received reception channel r denoted c (r),
• the height of the reception channel r formed, denoted d (r),
• the depth of the reception channel r noted, p (r).

The abscissa of the point aimed on the screen is in one-to-one relation with the horizontal angle of sight noted a (r), and the ordinate of the point aimed on the screen is in one-to-one relationship with the vertical angle of sight noted b (r). The width and height of the video screen correspond to the value of the focal length of the camera.

As a result, the camera 100 supplies the value of its focal length to the control unit 300, so that the latter can correspond values of angles to the abscissa and to the ordinate of the point targeted on the screen, which is identified in an arbitrary system of units such as, for example, the percentage.

Thus, we set at 0%, for example, the minimum value of the abscissa, corresponding to the value of the leftmost point on the screen, and we set at 100% the maximum value of the abscissa, corresponding to the value of the rightmost point on the screen. The control unit knowing the value of the focal length of the camera, that is to say the value of the maximum opening angle corresponding to the width of the screen, defined by the value 100%; can, by a simple report, determine the value of the horizontal viewing angle, corresponding to any value of the abscissa of a target point on the screen.

Preferably, we define by A, the maximum number of values corresponding to a (r), B the maximum number of values corresponding to b (r), C, the maximum number of values corresponding to c (r), D the number maximum of values corresponding to d (r) and P the maximum number of values corresponding to p (r).

According to an embodiment of the invention, a user advantageously fixes the value of a parameter, at least, among all these parameters. The values of the parameters which are not set by the user advantageously receive a default value, or else a value deduced from another parameter. Thus, for example, if the height d (r) of the reception channel r is not adjusted by the user, the value taken can be equal to the width c (r) of the reception channel r.

According to another variant, it is considered that if one of the parameters is not relevant for the production of the device, its maximum value and therefore also its current value, are fixed at 1.

The control unit 300 makes it possible to process the signals coming from the sensors 610. It also processes the signals, coming from the adjustment means, representing the values of the parameters. These parameter values influence the calculation of the values of the coefficients of the digital filters 310, that is to say on the directional characteristics of the sound reception channels. Consequently, the values of the parameters of the reception channels play an important role in the processing of the signals coming from the sensors, since these signals will not be treated in the same way according to the directional characteristic which one fixes for each reception channel.

Firstly, the processing which must be carried out on the signals coming from the M sensors 610, consists in forming, at each instant n, the R signals at the output of the focused channels. These signals will be available at outputs 710.

The signals received by the M sensors and converted into digital signals, by the analog-digital converters 614, at the sampling instants n, are denoted x (m, n).

These signals are filtered by R digital filters having a number Q of coefficients h (q, r, m, n), where q represents the number of the coefficient and varies from 1 to Q, to give R signals denoted y (r, m, n) representing the contributions at time n of the sensor m in the channel r, according to the following equation: $${\text{y (r, m, n) = Σ}}^{\text{Q}}{}_{\text{q = 1}}\text{h (q, r, m, n) x (m, nq)}$$

In accordance with the usual structures of broadband channel formation, described by S. Haykin and T. Kesler in the article "Relation between the radiation pattern of an array and the two-dimensional Discrete Fourier Transform", published in the journal IEEE transactions on Antennas and Propagation, Volume 23, number 3, pages 419-420, 1975, each output s (r, n) in a channel r at time n is obtained by summing the M signals y (r, m, n) according to the equation: $${\text{s (r, n) = Σ}}^{\text{M}}{}_{\text{m = i}}\text{y (r, m, n)}$$

The signal s (r, n) in the channel r, is supplied, in digital form, by the control unit 300 to the corresponding output 710.

A variant would consist in supplying the corresponding output 710, the signal s (r, n) in the channel r, in analog form, after passing through a digital-analog converter.

In a second step, the processing which must be carried out on the signals coming from the R adjustment means, consists in modifying, at each instant n, the values of the coefficients of the filters in order to modify the directional characteristics of the sound reception channels.

The coefficients h (q, r, m, n) of the filter r in the path r, for the sensor m, depend on the instant n. The coefficients are updated on the basis of information, that is to say on the basis of the parameter values acquired by the control unit 300 from the R adjustment means 400 and transmitted every N samples to the unit. control 300. Thus, if the coefficients are updated at time n _o , they will be updated again at time n _o + N.

Preferably, a method of adjusting the characteristic parameters of the sound recording, furthermore consists in reconstituting, by calculation, the values of the coefficients of the filters, between these two instants n _o and n _o + N. Thus, the values coefficients can be interpolated linearly according to the equation: $${\text{h (q, r, m, n) = [(nn}}_{\text{o}}{\text{) / N] h (q, r, m, n}}_{\text{o}}{\text{+ N) + [(n}}_{\text{o}}{\text{+ Nn) / N] h (q, r, m, n}}_{\text{o}}\text{)}$$

The control unit 300 calculates at all times n the values of the coefficients h (q, r, m, n) of the filters 310 from the values of parameters received, at the sampling rate of the converters 412, of the R means of settings 410.

When the information is received at a time denoted n _o , the control unit determines, for each channel r of reception of the sound, the values of the coefficients h (q, r, m, n _o + N) of the filters, which serve at interpolate, using equation (3), the values of the coefficients h (q, r, m, n), between the present time n _o and the time n _o + N where the following information is received.

The values of the coefficients are therefore interpolated over time, at each sampling instant, between these two values, n _o and n _o + N, which are modified at a regular rate but, preferably, slower than the sampling frequency .

Alternatively, equations (1) and (2) can be applied twice. Indeed, these equations are applied for the first time for filters with coefficients h (q, r, m, n _o ), which gives the following signals: y _o (r, m, n) and s _o (r, n ). These equations are applied a second time for filters of coefficients h (q, r, m, n _o + N) which gives the following signals: y _N (r, m, n) and s _N (r, n).

The interpolation is then carried out at the level of the output signals s (r, n) according to the relation: $${\text{s (r, n) = [(nn}}_{\text{o}}{\text{) / N] S}}_{\text{NOT}}{\text{(r, n) + [(n}}_{\text{o}}{\text{+ Nn) / N] s}}_{\text{o}}\text{(r, n)}$$

Another variant of this method would consist in interpolating the values of the coefficients of the filters 310, not only in time but also in space. In this case, the filter coefficients would also be interpolated between two positions, displayed on the screen, corresponding to the renewal of the filter coefficients.

The values of the coefficients of the filters 310 are a function of the settings, given by the manipulator through the commands 411 of the adjusting means 410, and described by the parameters a (r), b (r), c (r), d (r ), p (r).

We denote by F (a, b, c, d, p) this function. It provides, for each quintuplet value (a, b, c, d, p) of parameters, a vector QxM representing the Q coefficients of the filters, corresponding to the R channels for receiving the sound of the M sensors, when the settings are (a, b, c, d, p). Thus, the coefficients h (q, r, m, n _o ) are read in the vector QxM whose components are noted f (m, q) for m varying from 1 to M and q varying from 1 to Q and we obtain: $${\text{h (q, r, m, n}}_{\text{o}}\text{) = f (m, q)}$$

The control unit applies R times this function F to obtain the values of the coefficients of the filters corresponding to the R reception channels formed.

To obtain an expression of the function used to calculate the values of the filter coefficients, we proceed in several stages.

A first step consists in determining the coordinates of the position of an actual sound source and the coordinates of the positions of fictitious sound sources taken as a reference. Thus, to find the coordinates of a real sound source, we determine, for example, the horizontal angle u _a of the beam centered on the direction defined by a, the vertical angle v _b of the beam centered on the direction defined by b , the horizontal angles, u _a1 and u _a2 , which form the horizontal limits of the beam centered on the direction defined by a and of width defined by c, and finally, the vertical angles, v _b1 and v _b2 which form the vertical limits of beam centered on the direction defined by b and of width defined by d.

To find the coordinates of the positions of fictitious sound sources, we first choose a number K of reference positions, each defined by the pair of horizontal and vertical angles (u _k , v _k ) for k varying from 1 to K.

These reference sources are advantageously distributed uniformly, in the square
[-Π, Π] x [-Π, Π] deprived of its central part
[u ₁ , u ₂ ] x [v ₁ , v ₂ ]. We then choose L reference frequencies noted f _i , for i varying from 1 to L, and a reference distance which is preferably a depth value p.

The origin in three-dimensional space is advantageously defined by the position of the camera 100. The coordinates of the positions of the reference sources are then calculated from their expression which is as follows: $${\text{[pCos (u}}_{\text{k}}{\text{) Cos (v}}_{\text{k}}{\text{), pcos (u}}_{\text{k}}{\text{) Cos (v}}_{\text{k}}{\text{), psin (u}}_{\text{k}}\text{)]}$$

For each fictitious source k and for each sensor m, the distance z (k, m) between the source and the sensor is calculated. The transfer functions are also calculated from the reference sources to the sensors, for the reference frequencies. The transfer function t (m, k, f _i ), for the sensor m, the source k and the frequency f _i , is given by equation (5) where j denotes the root of -1 and V the speed of the his : $$\text{t (m, k, fi) = 1 / z (k, m) e}{}^{{\text{-j2Πf}}_{\text{i}}\text{[z (k, m) / V]}}$$

This transfer function makes it possible, in a second step, to establish the expressions of the gains obtained, for the fictitious sounds coming from reference sound sources, and to fix the gains, which one wishes to obtain, for these same fictitious sounds. With the filter, whose coefficients are f (m, q), the sound from a source, located at a position k, will be received, for a frequency f _i , with a gain g (k, f _i ) which is determined according to the equation: $${\text{q (k, f}}_{\text{i}}{\text{) = Σ}}^{\text{M}}{}_{\text{m = i}}{\text{Σ}}^{\text{Q-1}}{}_{\text{q = o}}{\text{f (m, q) t (m, k, f}}_{\text{i}}\text{) e}{}^{{\text{-j2Πf}}_{\text{i}}\text{q}}$$

The desired gains g _s (k, f _i ) corresponding to the sounds originating from the sound sources situated at the reference positions are fixed, this for reference frequencies f _i .

In a third step, an expression of the difference between the gains obtained and the desired gains is established. This difference represents an error, which can be reduced to a threshold value, which has been fixed, using, for example, the least squares calculation method. We then obtain an expression, which represents the square of the error which we wish to reduce to a threshold value and which is written in the form: $${\text{Σ}}^{\text{K}}{}_{\text{k = o}}{\text{{Σ}}^{\text{L}}{}_{\text{i = 1}}{\text{[g (k, f}}_{\text{i}}{\text{) -g}}_{\text{s}}{\text{(k, f}}_{\text{i}}{\text{)]}}^{\text{2}}\text{}}$$

This equation (7) represents a sum of squares and double products. This means that the criterion given by equation (7) is quadratic in g (k, f _i ). Likewise the criterion given by equation (6) is quadratic in f (m, q). The reduction of the error to a threshold value leads to a system in these unknowns f (m, q), which admits a unique solution. The solution of F is obtained by derivation of equation (7) with respect to the values of the coefficients f (m, q).

If we write T (k, f), the vector containing the components of t (m, k, f) e ^-2Πfq for all the couples [m, q] listed in the same order as the vector QxM representing F (a, b, c, d, p), a solution of F is written: $${\text{F (a, b, c, d, p) = [Σ}}_{\text{k, i}}{\text{T (k, f}}_{\text{i}}{\text{) T (kf}}_{\text{i}}{\text{)}}^{\text{T}}{\text{]}}^{\text{-1}}{\text{Σ}}_{\text{k, 1}}{\text{T (k, f}}_{\text{i}}{\text{) g}}_{\text{s}}{\text{(k, f}}_{\text{i}}\text{)}$$

In a final step, the values of the filter coefficients can be determined from the expression of the function F thus found. To be able to determine the values of these coefficients, there are two possibilities.

According to a first variant, the values of the coefficients are determined, before any manipulation, from the function F and for fixed values of parameters, then they are stored in a table.

This table can, for example, be a two-dimensional table comprising QxM rows and AxBxCxDxP columns. In this case, quintuplets (a, b, c, d, p) of parameters, for example, define the indices of the columns and the numbers q of the coefficients of the filters corresponding to each sensor m define the indices of the lines. However, the dimension of the table can be higher if we decide to separate the quintuplets into 2, 3, 4, or 5 distinct parameters and if we decide to distinguish the Q coefficients and the M sensors to store them in rows and columns separated. This memorization of the values of the coefficients in a table makes it possible to change the values of the coefficients more quickly during the manipulation of sound recording, for fixed values of parameters. The coefficients will change values only when the values of the quintuplets of parameters, which are fixed and stored in the table, are reached. Between these quintuplet values, corresponding to the updating of filters, the values of the coefficients could, for example, be interpolated.

According to a second variant, the values of the coefficients of each filter are determined in real time, from the expression of the function F and for values of parameters varying continuously. In this case, the coefficients of the filters are preferably updated at regular rate and their values are interpolated according to equation (3) previously established.

The orientation of the camera and that of the sensor network must be connected, by any means, in order to avoid any discrepancy between, on the one hand, the image representing the position of the sound sources, and on the other hand, the images showing the variation of the characteristic parameters of the sound reception channels. In this way, one can visualize very precisely the variation of the parameters compared to the position and the size of the sound sources.

Another embodiment of a device according to the invention therefore relates to fixing the camera 100 relative to the network 600 of sensors. The camera 100 is advantageously fixed on the same frame as the network 600 of sensors so that its pointing is strictly invariant with respect to the position of the sensors.

A variant of this system consists in not fixing the camera 100 on the same frame as the network 600 of sensors. In this case, the sensor network must have a fixed position in space and the camera must also have a fixed position and orientation in space to obtain a pointing of the sound sources. which is invariant with respect to the positions of the sensors.

According to another alternative embodiment of the device according to the invention, it is possible to add a remote control making it possible to remotely adjust the video pointing system. However, in this case, a user does not necessarily have access to the video system, so that he cannot view the settings made. This is why, it is also preferable to equip the device with a hearing feedback system, allowing the user to make the adjustments directly, using the audible signals reaching him. Hearing feedback is for example achieved by means of an earpiece placed in the user's ear canal and connected to the device by a cable, or better, via a Hertzian channel.

Claims (9)

Sound recording device comprising a network of sensors, for picking up sound from sound sources, a control unit (300) using in particular filters (310) for processing the signals received by the sensors, and adjustment means ( 400) of the characteristic parameters of the sound recording, characterized in that it further comprises a video camera (100), a video screen (200) which displays a first video image corresponding to the signal from the camera, and means for coupling (500) the screen (200) to the means for adjusting (400) the characteristic parameters of each of the sound reception channels capable of producing, for each of the reception channels, another video image showing the variations of the characteristic parameters, and to superimpose this image on the first video image, so as to control the adjustment of these parameters. Sound pickup device according to claim 1, characterized in that the coupling means (500) comprise, for each of the adjustment means (410) of the characteristic parameters of each of the sound reception channels, a video generator (510) making it possible to transform the parameter adjustment signal and a video mixer (520) into a video signal so that the signals from the generators are mixed together and also mixed with the signal from the camera to obtain, on the displayed image, an overlay of images corresponding to each reception channel. Sound recording device according to one of claims 1 to 2, characterized in that the control unit (300) comprises filters (310) with linear interpolation. Sound pickup device according to one of claims 1 to 3, characterized in that the camera (100) is fixed on the same frame as the network (600) of sensors (620). Sound recording device according to one of claims 1 to 4, characterized in that it further comprises a remote control, able to remotely control the settings of the video pointing system, and a hearing feedback system. Method for adjusting the characteristic parameters of the sound recording, according to any one of the preceding claims, characterized in that the control unit (300) performs processing on the signals picked up and on the signals, corresponding to the values of the parameters , provided by the adjustment means, comprising the following steps: - filtering of signals picked up by filters with linear interpolation, - modification of the filter coefficients for each modification of the parameters, - linear interpolation in time, at each sampling instant, between two values, corresponding to the renewal of the filter coefficients, which are modified at a regular rate but slower than the sampling frequency. Method for adjusting the characteristic parameters of the sound recording according to claim 6, characterized in that the modification of the filter coefficients for each of the parameter modifications is effected by the determination of a function F connecting the characteristic parameters of each channel of reception of the sound at the values of the coefficients of the corresponding filters, this determination comprising the following steps: - determine the coordinates of the position of an actual sound source and of the positions of fictitious sound sources taken as reference, - establish the expression of gain obtained, for fictitious sounds coming from reference sound sources, and fix the gains, which one wishes to obtain, for these same fictitious sounds, - establish an expression of the difference between the gains obtained and the desired gains, which represents an error which can be reduced to a threshold value, - derive the expression thus established, with respect to the coefficients of the filters, to result in an expression of the function F, - determine the values of the filter coefficients from the expression of the function F thus found. Method for adjusting the characteristic parameters of the sound recording according to one of claims 6 to 7, characterized in that the values of the coefficients of each filter corresponding to each reception channel are determined for fixed values of parameters the sound of each sensor, from function F, and they are stored in a table. Method for adjusting the characteristic parameters of the sound recording according to one of claims 6 to 7, characterized in that the values of the coefficients of each filter corresponding to each reception channel are determined from function F of the sound of each sensor, at each instant n and for continuously varying parameter values.

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