EP2122607B1 - Method for the active reduction of sound disturbance - Google Patents

Abstract

A method and a system for the active reduction, at a predetermined area, of the energy of a sound signal (dk(n)), also called a diffused noise signal, generated at the area by a primary signal (xk(n)), or noise signal, by the emission of a plurality of counter-noise signals (yk(n)) having an effect antagonistic to the diffused noise signal (dk(n)), each of the counter-noise signals (yk(n)) including a feedback counter-noise signal (yfbkk(n)) and a feed-forward counter-noise signal (yfwdk(n)). The method includes detecting the periodical components of diffused noise signal (dk(n)) for adjusting the feedback counter-noise signal (yfbkk(n)), and modelling the inverse of the secondary path for adjusting the feedback counter-noise (yfbkk(n)) and feed-forward counter-noise (yfwdk(n)) signals. The invention can be implemented to any type of industrial or non-industrial noise and in any location such as working places and relaxation places.

Description

The present invention relates to a method of reducing noise pollution by active control. It also relates to a system implementing the method according to the invention.

The invention aims, in particular, to reduce noise pollution in an area determined by an active reduction process. Noise can be any kind of annoying acoustic waves that can be considered noise in a certain area. These nuisances can be of all types and frequencies ranging from a few hertz to a few thousand hertz. They can be created by any device in operation. In the case, for example, of a closed enclosure, the nuisances can be generated by devices which are located inside this enclosure. They may also be caused by sources outside the enclosure, for example when the enclosure is located near sites such as an airport, highway, railway, etc.

Current systems for actively reducing noise nuisance can mitigate these nuisances by two types of processes. The first, called feedforward, requires prior notification of the noise signal which is the cause of the noise nuisance to be reduced (see for example US-A-5,978,489 and WO-A-03/015074 ). The noise signal is detected upstream of the active reduction processing zone and provides a reference signal which must be strongly correlated with the noise nuisance to be reduced. In this case, prior knowledge of the noise signal is exploited in order to minimize the error of reduction of the noise nuisance, this error being quantified by a so-called error signal measured at the determined zone. However, the prior information of a noise nuisance is not always available, hence the use of a second noise reduction method, called feedback, or closed loop control, in which no prior detection is required. is done. The reduction error signal is used to provide a control signal for minimizing this same error signal.

However, most current systems provide limited solutions to the causal constraint that is essential for the successful completion of some active control applications. This requires performing digital operations intrinsic to the active reduction process in a very short time. As a result, these systems have a limited efficiency, in reaction time, in space and in frequency.

An object of the invention is thus to provide a method of actively reducing noise pollution to better meet the constraint cited above, and therefore to achieve a better reduction of noise.

• mesure, par des moyens de mesure disposés au niveau de la zone déterminée, d'un signal, dit d'erreur, représentant une information d'efficacité de la réduction de l'énergie du signal de bruit propagé dans la zone ;
• modélisation, par au moins un premier filtre, d'un trajet acoustique direct, dit chemin secondaire, entre les moyens d'émission du signal de contre-bruit et les moyens de mesure du signal d'erreur éventuellement lors d'une étape préalable d'identification ;
• détection d'au moins une composante périodique du signal de bruit propagé par analyse d'un signal de bruit propagé estimé à partir d'une part du signal d'erreur, et d'autre part du signal de contre-bruit feedback baité par le premier filtre, ladite détection fournissant ladite composante périodique ; et
• ajustement du signal de contre-bruit feedback en fonction de la composante périodique détectée, du signal d'erreur et du chemin secondaire modélisé.

The invention proposes to remedy the aforementioned problem by an active reduction method at a given zone of the energy of a sound signal, said propagated noise signal, generated in the zone determined by a primary signal, said signal noise. The method comprises transmitting, by transmission means, at least one counter-noise signal comprising at least a first counter-noise signal, said feedback, of an antagonistic effect to the propagated noise signal, this method further comprising at least one iteration of the following operations:

• measuring, by measuring means arranged at the determined area, a signal, said error signal, representing information of efficiency of the reduction of the energy of the noise signal propagated in the area;
• modeling, by at least a first filter, of a direct acoustic path, said secondary path, between the means for transmitting the noise signal and the means for measuring the error signal, possibly during a preliminary step of 'identification ;
• detecting at least one periodic component of the propagated noise signal by analyzing an estimated propagated noise signal from a part of the error signal, and secondly the feedback noise signal baited by the first filter, said detection providing said periodic component; and
• adjusting the feedback noise signal as a function of the detected periodic component, the error signal and the modeled secondary path.

In the present application, the sources of the noise signal are called the secondary sources and the sources of the noise signal are the primary sources.

The measurement of the error signal, by a measuring means constituted for example by a control microphone, makes it possible to account for the reduction of the energy of the propagated noise signal and to adjust the noise signal of to reduce this same error signal.

The modeling of the secondary path can be carried out by transmitting, by a means of transmitting the counter-noise signal consisting for example of a loudspeaker, of a known signal, followed by a measurement of this signal at the level of the area determined by a measuring device. Thus, by knowing the transmitted signal and the measured signal, it is possible to characterize the acoustic path between the emission means of the counter-noise signal and the measurement means at the determined zone. This modeling can take place before or during any phase of emission of noise.

This determined path, and always before any reduction of the noise signal, modeling the inverse of the secondary path can be performed numerically so as not to introduce phase shift that is to say additional delay in the control chain , which would come in opposition to the main purpose of the invention. An amplitude modeling only is therefore conducted. This inverse filter makes it possible to limit the resonances inherent to the electro-acoustic equipment used as well as to the topography of the treatment zone, resonances that are found in said secondary path.

The following occurs during the actual control phase.

The detection of the periodic components of the propagated noise signal allows a better knowledge of the spectral composition of said signal and consequently makes it possible to carry out band-pass filtering operations. The counter-noise signal can thus be adjusted optimally to ensure, in greater stability, the best reduction of the energy of the propagated noise signal and thus of the nuisance caused by the noise signal at the zone level. determined, especially during rapid changes in periodic components.

The propagated noise signal is estimated from, on the one hand, the error signal, and on the other hand the feedback noise signal processed by the first filter modeling the secondary path. Indeed, by subtracting from the error signal measured in the determined zone, the counter-noise signal feedback filtered by the first filter modeling the secondary path, that is to say the acoustic path between the secondary source and the measuring means at the determined area, it is possible to make an estimate of the noise signal propagated to reduce.

The detection of the periodic components of the propagated noise signal can be carried out by a filtering of the propagated noise signal estimated by notch-type bandpass filters, performing an infinite impulse response (IIR) bandpass filtering. of constant amplitude everywhere except at the frequencies of the periodic components of the propagated noise signal where the bandwidths are practically nil. These filters are called Adaptive Notch Filters (ANF).

In addition, the method according to the invention comprises a band-pass filtering of the estimated propagated noise signal, at the frequency of all or part of the detected periodic components, said filtering providing a signal, referred to as a reference signal, consisting essentially of the periodic components of the propagated noise signal. This reference signal is then used in the adjustment of the feedback noise signal, as described below.

Indeed, the method according to the invention comprises an adjustment of at least one coefficient of a second filter, finite impulse response, provided for adjusting the feedback signal against noise as a function of the reference signal filtered by a third filter finite impulse response modeling the inverse of the secondary path in amplitude. The filtered reference signal thus obtained, composed essentially of the periodic components of the estimated propagated noise signal, thus serves as a basis for adjusting the coefficients of the second filter, the function of which is precisely to eliminate the periodic components of the propagated noise signal. . The filtering operation, by the third filter modeling in amplitude the inverse of the secondary path, makes it possible to facilitate the adjustment of the coefficients of the second filter.

Indeed, the combination, on the one hand of the first filter modeling the secondary path and, on the other hand, of the third filter modeling in amplitude the inverse of the secondary path, results, in output, a flat response in amplitude, equal This facilitates the work of the second filter which consists in finding the optimal amplitudes and phases of the signal of counter-noise feedback that minimizes the energy of the error signal and therefore the energy of the propagated noise signal. In fact, ensuring this unit amplitude makes it possible to rid the second filter of the optimal amplitude search work and to focus only on the search for the optimal phase.

Advantageously, at least one coefficient of the second filter can be adjusted by an algorithm of the Least Mean Square (LMS) minimization algorithm type as a function of the reference signal processed by the first filter, the signal of the second signal. error having undergone pass-band filtering at the frequency of all or part of the detected periodic components and of a convergence coefficient, called feedback, involved in the LMS algorithm. By performing such bandpass filtering on the error signal, it is thus possible to isolate the periodic components of the propagated noise signal that are present in the error signal so that the second filter concentrates only on them.

Advantageously, the counter-noise signal further comprises a feedback signal, called feedforward, adjusted according to the error signal and the noise signal measured by measuring means comprising for example a microphone. The feedforward counter-noise signal is intended to reduce the energy of non-periodic components of the noise signal. Thus, the method according to the invention makes it possible to implement, in a combined manner, a feedback noise signal and a feedforward counter-noise signal intended respectively to reduce the energy of the periodic components and the non-periodic components of the noise signal. .

• une modélisation en amplitude de l'inverse du chemin secondaire par au moins un quatrième filtre à réponse impulsionnelle finie et
• une modélisation par au moins un sixième filtre à réponse impulsionnelle finie du chemin secondaire,

toujours dans la perspective de faciliter le travail d'ajustement des coefficients d'un cinquième filtre défini ci-après. Le quatrième filtre peut être identique au troisième filtre et le sixième filtre identique au premier filtre.Furthermore, the method according to the invention may furthermore comprise:

• an amplitude modeling of the inverse of the secondary path by at least a fourth finite impulse response filter and
• modeling by at least one sixth finite impulse response filter of the secondary path,

always in the perspective of facilitating the work of adjusting the coefficients of a fifth filter defined below. The fourth filter may be identical to the third filter and the sixth filter identical to the first filter.

In a non-limiting exemplary embodiment, the fourth filter may be the third filter and the sixth filter may be the first filter.

The adjustment of the feedforward counter-noise signal comprises an adjustment of at least one coefficient of a fifth finite impulse response filter provided for adjusting said feedforward counter-noise signal in accordance with the noise signal previously processed by the fourth filter.

In addition, at least one coefficient of the fifth filter is adjusted by a least squares algorithm according to the error signal, the noise signal measured and previously processed by the sixth filter modeling the path. secondary and a convergence coefficient, called feedforward, involved in the algorithm in question.

As previously, the combination, on the one hand of the fourth filter modeling the secondary path and, on the other hand, of the sixth filter modeling in amplitude the inverse of the secondary path results, at the output, in a flat amplitude response, equal to 1.

• un signal de contre-bruit feedback,
• un signal de contre-bruit feedforward, ou
• un signal de contre-bruit feedback et un signal de contre-bruit feedforward.

Advantageously, the method according to the invention can be implemented for the attenuation of at least one noise signal by transmitting a plurality of counter-noise signals by a plurality of transmission means. Each of the counter-noise signals may include:

• a counter-noise feedback signal,
• a feedforward counter-noise signal, or
• a counter-noise feedback signal and a feedforward counter-noise signal.

By the emission of a plurality of counter-noise signals, and by the implementation of a plurality of measurement points of error signals, for example by control microphones, the method according to the invention makes it possible to on the one hand to increase the size of the determined zone in which it is desired to achieve a reduction of the energy of at least one propagated noise signal, and on the other hand to realize this reduction up to frequencies higher. Thus, by increasing the number of mean pairs of noise signal transmission / error signal measuring means, in other words the noise signal / error signal, it is possible to process the noise over a greater distance and in a wider frequency band.

For example, the method according to the invention can be implemented to achieve an "acoustic comfort bubble". Since the spatial extent of such a bubble of free space acoustic comfort is rather confined as the frequency increases, it is necessary to consider several sources of emission of several counter-noise signals and several microphones of control of reduction of the noise. propagated noise signal energy. For example, knowing that the inter-ear space is about 20 centimeters, and that we take an identical margin to allow any freedom for a user to move the head reasonably, we end up with a bubble of acoustic comfort to achieve 40 centimeters in diameter, an effective treatment up to 200 Hz maximum by considering only one average pair of emission of the noise signal / error signal measuring means. By multiplying the nuisance reduction points, that is to say the number of control microphones, it is possible to increase the maximum frequency of the noise signals whose energy is to be reduced. Thus, with 3 noise reduction points over this distance, noise signals up to about 700Hz can be processed in a 40 cm diameter comfort bubble. By multiplying the number of counter-noise signals and the number of reduction points and arranging them appropriately, one can also increase the size of the comfort bubble.

By reduction point, or minimization, is meant the location of a control microphone provided for measuring an error signal.

• des moyens pour émettre le signal de contre-bruit ;
• des moyens de mesure, au niveau de la zone déterminée, d'un signal, dit d'erreur, représentant une information d'efficacité de la réduction de l'énergie du signal de bruit propagé ;
• au moins un premier filtre pour modéliser un trajet acoustique direct, dit chemin secondaire, entre les moyens d'émission du signal de contre-bruit et les moyens de mesure du signal d'erreur obtenu éventuellement au terme d'une étape préalable d'identification ; - des moyens pour estimer le signal de bruit propagé à partir d'une part du signal d'erreur, et d'autre part du signal de contre-bruit feedback traité par le premier filtre, lesdits moyens fournissant un signal de bruit propagé estimé;
• des moyens pour détecter et fournir au moins une composante périodique du signal de bruit propagée par analyse dudit signal de bruit propagé estimé ; et
• des moyens pour ajuster le signal de contre-bruit feedback en fonction de la composante périodique détectée, du signal d'erreur et du chemin secondaire modélisé.

According to another aspect of the invention, there is provided an active reduction system, at a given zone, the energy of a sound signal, said propagated noise signal, generated in the area determined by a signal primary, said noise signal, by emission of at least one counter-noise signal comprising at least a first counter-noise signal, said feedback, of an antagonistic effect to the noise signal propagated at the determined zone, the system comprising:

• means for transmitting the counter-noise signal;
• means for measuring, at the level of the determined zone, a signal, called an error signal, representing information of effectiveness of the reduction of the energy of the propagated noise signal;
• at least one first filter for modeling a direct acoustic path, said secondary path, between the means for transmitting the noise signal and the means for measuring the error signal obtained possibly at the end of a prior identification step ; means for estimating the noise signal propagated on the one hand from the error signal, and on the other hand the feedback noise signal processed by the first filter, said means providing an estimated propagated noise signal;
• means for detecting and providing at least one periodic component of the propagated noise signal by analyzing said estimated propagated noise signal; and
• means for adjusting the feedback noise signal as a function of the detected periodic component, the error signal and the modeled secondary path.

Advantageously, the emission means of the counter-noise signal may comprise directional ultrasound transducers having a reduced emission beam. Indeed, one of the limitations of the current active noise reduction systems is that if the noise control contributes to reducing the noise signal in a targeted area or volume, it can quite easily increase them elsewhere. . In other words, reducing disturbances in a space does not mean reducing them in all space. In addition, means for transmitting a counter-noise signal such as loudspeakers are more directive at low frequencies than at high frequencies. Unless we can have speakers larger than the largest wavelength inherent in the spectrum of the noise signal to be treated, we can overcome this limitation, unless using ultrasonic transducers. Ultrasound, completely inaudible at emission, distorts as they propagate through the air and slide into the audible spectrum. The advantage of ultrasound transducers lies in the fact that they have a very small emission beam and the volume in which the ultrasound becomes audible is quite predictable. Another advantage to the use of such transducers lies in the fact that their directivity simplifies the multi-channel system. Indeed, the transposition to the multichannel case of the single-channel system involves considering a multitude of secondary paths: the direct secondary paths between each transducer and their associated control microphone, but also the so-called secondary paths which represent the interactions between all the transducers and the microphones. On the other hand, the contributions of each secondary source on the noise signal measuring means, called backward contributions, must likewise be considered. This requires having a high computing capacity and memory electronics. In order to minimize the often significant costs of the real-time operations inherent in the calculation of the counter-noise signals, the directivity of the ultrasound transducers presents a great advantage to design not a complex multi-channel system but a parallelization of multiple, much less complex, monovoic systems. . Indeed, in this case, the crossed paths and the rear contributions become negligible due to the directivity of the ultrasound transducers and the fact that the entities are not taken into account in the parallelized structure does not disturb the stability of the system.

The system according to the invention may further comprise means for measuring the noise signal. These means may comprise at least one microphone, said noise, appropriately placed according to the noise source.

Furthermore, the system according to the invention may comprise band-pass filtering means of the propagated noise signal estimated at the frequency of all or part of the periodic components of the propagated noise signal, and arranged to generate a reference signal, such as as described above.

The means for adjusting the feedback noise signal may advantageously comprise at least a second finite impulse response filter provided for adjusting said feedback noise signal as a function of the reference signal filtered by a third impulse response filter. finite, arranged to model in amplitude the inverse of the secondary path.

Advantageously, the counter-noise signal may comprise a second feedback signal, said feedforward, the system according to the invention further comprising means for transmitting the adjusted feedforward counter-noise signal as a function of the error signal and the noise signal.

The system may include a fourth finite impulse response filter, amplitude modeling the inverse of the secondary path, a fifth filter, provided to adjust the feedforward counter-noise signal, based on the measured noise signal processed by the fourth filter. and a sixth filter, finite impulse response, arranged to model the secondary path.

The system according to the invention can advantageously comprise a plurality of means for transmitting a plurality of counter-noise signals used for the attenuation of at least one noise signal.

• la figure 1 est une représentation schématique d'une configuration de réduction active d'un signal sonore grâce à un système monovoie selon l'invention ;
• la figure 2 est une représentation schématique d'une configuration de réduction active d'un signal sonore grâce à un système multivoies selon l'invention ;
• la figure 3 est une représentation schématique sous forme de blocs fonctionnels des opérations réalisées au niveau d'une voie d'un système multivoies selon l'invention comprenant une mesure du signal de bruit avec des microphones de mesure ;
• la figure 4 est une représentation schématique sous forme de blocs fonctionnels d'un module de détection et de filtrage de composantes périodiques d'un signal de bruit propagé au niveau d'une voie d'un système multivoies selon l'invention ;
• la figure 5 est une représentation schématique d'une carte électronique multivoies mise en oeuvre dans le système multivoies selon l'invention ;
• la figure 6 est une représentation d'un faisceau d'émission d'un transducteur ultrason utilisé dans le système selon l'invention ;
• la figure 7 est un premier exemple de réalisation du système multivoies selon l'invention pour l'obtention d'une bulle de confort ; et
• la figure 8 un deuxième exemple de réalisation du système multivoies selon l'invention pour l'obtention d'une bulle de confort.

Other advantages and characteristics will appear on examining the detailed description of a non-limiting embodiment, and the appended drawings in which:

• the figure 1 is a schematic representation of an active reduction configuration of a sound signal by means of a single-channel system according to the invention;
• the figure 2 is a schematic representation of an active reduction configuration of a sound signal by means of a multi-channel system according to the invention;
• the figure 3 is a schematic representation in the form of functional blocks of the operations performed at a channel of a multi-channel system according to the invention comprising a measurement of the noise signal with measurement microphones;
• the figure 4 is a schematic representation in the form of functional blocks of a module for detecting and filtering periodic components of a noise signal propagated at a channel of a multi-channel system according to the invention;
• the figure 5 is a schematic representation of a multi-channel electronic card implemented in the multi-channel system according to the invention;
• the figure 6 is a representation of an emission beam of an ultrasound transducer used in the system according to the invention;
• the figure 7 is a first embodiment of the multiway system according to the invention for obtaining a comfort bubble; and
• the figure 8 a second embodiment of the multi-channel system according to the invention for obtaining a comfort bubble.

The figure 1 is a schematic representation of a configuration 10 active noise reduction through a single channel system 11 according to the invention. This system 11 comprises a noise microphone for measuring a noise signal x and a transducer emitting a noise signal adjusted therein to minimize the noise nuisance caused by the noise signal x at an acoustic comfort zone 12 wherein a control microphone is provided for measuring an error signal e. In the remainder of the description, we will call the acoustic comfort zone 12 thus created "bubble of acoustic comfort".

However, the spatial extent of such an acoustic comfort bubble 12 in free space being rather confined as the frequency increases, it is necessary to consider several pairs of transducer / control microphone. For example, knowing that the inter-ear space is about 20 centimeters, and that we take an identical margin to give a user freedom to move the head reasonably, we end up with a bubble of acoustic comfort to achieve 40 centimeters in diameter, an effective treatment up to 200 Hz maximum considering only one torque transducer / microphone control. By multiplying the number of transducer / control microphone pairs, it is possible to increase the maximum frequency of treatment. Thus, with 3 noise reduction points over this distance, disturbances up to about 700 Hz can be handled in a 40 cm diameter comfort bubble. By multiplying the number of counter-noise signals and the number of reduction points and arranging them appropriately, one can also increase the size of the comfort bubble.

• K microphones de bruit permettant de mesurer K signaux de bruits x_k ,
• K microphones de contrôle mesurant K signaux d'erreur e_k , et
• K transducteurs émettant K signaux de contre-bruit y_k et produisant une bulle de confort acoustique 22 plus grande que la bulle de confort 12,

avec k compris entre 1 et K. Bien entendu, le nombre de microphones de contrôle, le nombre de microphones de bruit et le nombre de transducteurs peuvent ne pas être égaux. Cependant, pour une description plus claire des différentes opérations effectuées au niveau de chaque voie du système multivoies 21, nous admettrons ici que le système multivoies 21 comprend le même nombre de microphones de contrôle et de transducteurs et un microphone de bruit.So, the figure 2 represents a configuration 20 active noise reduction through a multi-channel system 21 according to the invention. This multi-channel system 21 comprises:

• K noise microphones for measuring K noise signals x _k ,
• K control microphones measuring K error signals e _k , and
• K transducers emitting K counter-noise signals y _k and producing an acoustic comfort bubble 22 larger than the comfort bubble 12,

with k between 1 and K. Of course, the number of control microphones, the number of noise microphones and the number of transducers may not be equal. However, for a clearer description of the different operations performed at each channel of the multi-channel system 21, we will admit here that the multi-channel system 21 comprises the same number of control microphones and transducers and a noise microphone.

The figure 3 is a block diagram representation of a channel k in the multichannel configuration 20 implementing the multichannel system 21 according to the invention making it possible to carry out the comfort bubble 22. On the figure 3 , n denotes the discretized time, that is to say the sampling time, by S _kk the secondary path between the secondary source k and the control microphone k, that is to say, the path direct acoustics between the secondary source k, and the microphone k. The module 221 P _k represents the primary path between the reference signal detection microphone x _k (n) and the control microphone k. The control microphone k makes it possible to measure the error signal e _k at the level of the comfort bubble. We will now describe the operation of the system 21 at a level k.

The system 21 comprises two parts, namely a part 211, called feedforward and a part 212, called feedback. The feedback portion 212 comprises a finite impulse response filter W ^fbk _k (z) for generating and adjusting a counter-noise feedback signal yfbk _k (n) . This feedback portion 212 also includes two FIR filters Ŝ _kk (z) digitally modeling the secondary path S _kk. A module 213, composed of a filter Ŝ _kk (z) and an adder Σ, allows an estimation of the propagated noise signal d _k (n) at the comfort bubble, from the error signal e _k (n) measured by the control microphone k and feedback signal feedback yfbk _k (n) filtered by a filter Ŝ _kk (z). This module 213 outputs an estimated propagated noise signal of _k (n). A detection and filtering module 214 makes it possible to detect the periodic components of the propagated noise signal d _k (n) from the analysis of the estimated propagated noise signal of _k (n) and outputs a signal of reference of _k (n) composed of the detected periodic components of the estimated propagated noise signal of _k (n). This module 214 comprises an ANF detection block ^C of the periodic frequencies in the estimated propagated noise signal of _k (n) and an ALE ^P ( ALE for Adaptive Line Enhancer ) band-pass filter block of the estimated propagated noise signal of _k (n) at the frequencies of the periodic components detected by the ANF detection block ^C. This module 214 will be detailed in the following description. The reference signal of _k (n) is then used by a FIR filter 1 / Ŝ _kk (z) modeling in amplitude the inverse of the modeled secondary path Ŝ _kk and then by a filter W ^fbk _k (z) to adjust the signal counter-noise feedback yfbk _k (n).

The coefficients of the filter W ^fbk _k (z) are adjusted by a minimization algorithm according to the least squares criterion, represented by the block LMS, as a function of the reference signal of _k (n) previously treated by a filter Ŝ _kk (z), and error signal e _k (n) that has been bandpass filtered by an ALE block ^P at the periodic component frequencies detected in the estimated propagated noise signal of _k (n).

The feedforward portion 211 of the system 21 comprises a FIR filter W ^fwd _k (z) for generating and adjusting a feedback signal feedback yfwd _k (n) as a function of the noise signal x _k (n) measured by means of measurement and previously filtered by a FIR filter 1 / Ŝ _kk (z) modeling in amplitude the inverse of the modeled secondary path Ŝ _kk . The coefficients of the filter W ^fwd _k (z) are adjusted by an algorithm LMS, represented by the block LMS, as a function, on the one hand, of the error signal e _k (n), and on the other hand of the noise signal measured and previously treated by a filter Ŝ _kk (z).

The counter-noise signals feedforward yfwd _k (n) and feedback yfbk _k (n) are then added by an adder Σ to obtain a signal of counter-noise y _k (n) which is transmitted to the comfort bubble by means of emission, which are in our example ultrasonic transducers.

Thus, at the level of the comfort bubble, the error signal e _k (n) for the channel k measured by a control microphone (not shown) corresponds to the sum, on the one hand, of the propagated noise signal d _k (n ), and secondly counter-noise signals corresponding to each of the channels of the system 21 and having traveled the secondary paths S _lk (z) between the secondary sources associated with each of the channels and the control microphone k, c ' that is, we can write: $${\mathit{e}}_{\mathit{k}}\left(\mathit{not}\right)\mathit{=}{\displaystyle \underset{\mathit{l}\mathit{=}1}{\overset{K}{\mathit{\Sigma }}}}{\mathit{S}}_{\mathit{lk}}\left(\mathit{z}\right){\mathit{there}}_l\left(\mathit{not}\right)\mathit{+}{\mathit{d}}_{\mathit{k}}\left(\mathit{not}\right)\mathit{.}$$

Note that in the case where ultrasonic transducers are used as secondary sources, the error signal e _k (n) for the k channel measured by a control microphone not shown corresponds this time to the sum, d part of the propagated noise signal d _k (n), and secondly of the counter-noise signal y _k (n) corresponding to the path k and having traveled the secondary path S _kk (z) , that is, ie the acoustic path between the transducer k and the control microphone k. In that case, $${\mathit{e}}_{\mathit{k}}\left(\mathit{not}\right)\mathit{=}{\mathit{there}}_{\mathit{k}}\left(\mathit{not}\right){\mathit{S}}_{\mathit{kk}}\left(\mathit{not}\right)\mathit{+}{\mathit{d}}_{\mathit{k}}\left(\mathit{not}\right)\mathit{.}$$

The figure 4 is a block diagram representation of the module 214 for detecting and filtering periodic components of the estimated propagated noise signal of _k (n). The frequency estimation method used in the present example involves an infinite impulse response bandpass filtering of constant amplitude everywhere else than the frequencies of the components of the noise signal, where the bandwidth is almost zero. These filters are called "notch" filters and are referred to as ANF (Adaptive Notch Filter). There are two types of notch filters, corresponding to two different approaches, namely the direct or lattice approach. They are both in the rational form H _i ( z, θ ) = _Ni (z, θ ) / D _i (z, θ ) . For an input signal, we search for the best set of coefficients θ that minimize the squared error defined as the filtering of this input signal by the filter H _i (z, θ ) .

avec le paramètre a_i relié directement à la fréquence recherchée par la relation a_i =-2cos(2πf_i ) et ρ un réel strictement positif proche de 1 appelé facteur de contraction et rendant compte de la bande passante autour de la fréquence coupée.The lattice formulation is in the following form: $$H_i\left(z\right)=\frac{1+\mathrm{<figure-callout\; id="2"\; label="\rho "\; filenames="imgf0002.png,imgf0003.png"\; state="\{\{state\}\}">\rho </figure-callout>}}{2}\frac{1+{\mathrm{at}}_i{z}^{-1}+z^{-2}}{1+\frac{1+\mathrm{<figure-callout\; id="2"\; label="\rho "\; filenames="imgf0002.png,imgf0003.png"\; state="\{\{state\}\}">\rho </figure-callout>}}{2}{\mathit{at}}_i{z}^{-1}+\mathrm{\rho }{z}^{-2}}$$

with the parameter a _i directly connected to the frequency sought by the relation a _i = -2cos (2π f _i ) and ρ a strictly positive real close to 1 called the contraction factor and accounting for the bandwidth around the cutoff frequency.

In order to optimize the arithmetic operations, and to limit the disturbing impact of the detection and filtering module 214 on the system 21, a cascade decomposition, represented in FIG. figure 4 , of this module 214 is chosen to determine the frequencies composing a given signal. Also, for p periodic components, we have p filters H _i (z) in series.

Note that the cascade decomposition of block 214 is indicated by a C, for cascade, in ANF ^C (see figure 3 ).

on peut déterminer chaque paramètre a_i grâce à une réécriture astucieuse de l'algorithme de minimisation selon le critère des moindres carrés récursifs (algorithme RLS : Recursive Least Squares). Pour ce faire, on fait appel à la fonction d'auto-corrélation définie récursivement par : $${\mathrm{\Phi }}_{\mathit{i}}\left(\mathit{n}\right)\mathit{=}{\mathrm{\lambda \Phi }}_{\mathit{i}}\left(\mathit{n}\mathit{-}1\right)\mathit{+}{\tilde{\mathrm{\epsilon }}}_{\mathit{i}}^2\left(\mathit{n}\mathit{-}1\right)$$

et, en posant Θ(n)=[a ₁(n)...a_p (n)] ^T , Γ(n)=[Φ₁(n)...Φ _p (n)] ^T , Ẽ(n)=[ε̃₁(n-1)...ε̃ _p (n-1)]^T et E(n)=[ε̃₁(n)...ε _p (n)] ^T , avec T signifiant transposé, on utilise la relation de récurrence suivante : $$\mathrm{\Theta }\left(n\right)=\mathrm{\Theta }\left(n-1\right)+{\Gamma }^{-1}\left(n\right)\tilde{\mathrm{E}}\left(n\right)\mathrm{E}\left(n\right)\mathrm{.}$$

By asking the following relation: $${\tilde{\mathrm{\epsilon }}}_i\left(\mathrm{not}\right)=\frac{1}{D_i}{\mathrm{\epsilon }}_{i-1}\left(\mathrm{not}\right)$$

we can determine each parameter a _i thanks to a clever rewrite of the algorithm of minimization according to the criteria of least squares recursive (algorithm RLS: Recursive Least Squares). To do this, we use the autocorrelation function defined recursively by: $${\mathrm{\Phi }}_{\mathit{i}}\left(\mathit{not}\right)\mathit{=}{\mathrm{\lambda \Phi }}_{\mathit{i}}\left(\mathit{not}\mathit{-}1\right)\mathit{+}{\stackrel{\mathit{~}}{\mathrm{\epsilon }}}_{\mathit{<figure-callout\; id="2"\; label="i"\; filenames="imgf0002.png,imgf0003.png"\; state="\{\{state\}\}">i</figure-callout>}}^2\left(\mathit{not}\mathit{-}1\right)$$

and, by putting Θ ( n ) = [ a ₁ ( n ) ... a _p ( n )] ^T , Γ ( n ) = [Φ ₁ ( n ) ... Φ _p ( n )] ^T , Ẽ ( n ) = [ε ₁ ( n -1) ... ε _p ( n -1)] ^T and E ( n ) = [ε ₁ ( n ) ... ε _p ( n )] ^T , with T meaning transposed , we use the following recurrence relation: $$\mathrm{\Theta }\left(\mathrm{not}\right)=\mathrm{\Theta }\left(\mathrm{not}-1\right)+{\Gamma }^{-1}\left(\mathrm{not}\right)\tilde{\mathrm{E}}\left(\mathrm{not}\right)\mathrm{E}\left(\mathrm{not}\right)\mathrm{.}$$

ce qui permet de commencer avec une largeur de bande élevée, de sorte à permettre à chaque section 2141 de détecter une composante périodique, puis de la resserrer afin de pouvoir préciser cette détection. C'est aussi un moyen de limiter les conflits entre sections, sachant que ceux-ci peuvent quand même se produire.Finally, λ and ρ are exponentially adapted thanks to the following recursion: $$\{\begin{array}{ll}\mathit{\lambda }\left(\mathit{not}\right)& \mathit{=}{\mathrm{\lambda }}_0\mathrm{\lambda }\left(\mathit{not}\mathit{-}1\right)\mathit{+}\left(1\mathit{-}{\mathrm{\lambda }}_0\right){\mathrm{\lambda }}_{\mathit{\infty }}\\ \mathrm{\rho }\left(\mathit{not}\right)& \mathit{=}{\mathrm{\rho }}_0\mathrm{\rho }\left(\mathit{not}\mathit{-}1\right)\mathit{+}\left(1\mathit{-}{\mathrm{\rho }}_0\right){\mathrm{\rho }}_{\mathit{\infty }}\end{array}$$

this makes it possible to start with a high bandwidth, so as to allow each section 2141 to detect a periodic component, then to tighten it in order to be able to specify this detection. It is also a way to limit conflicts between sections, knowing that they can still occur.

For reasons of stability and speed of convergence, the reference signal of _k (n) and the error signal e _k (n) are filtered by bandpass filters centered around the frequencies present in the noise signal. estimated propagation of _k (n). The complement of a notch filter, in whatever formulation is it, is a band-pass filter, denoted N ^ALE (z ^-1 ) , in which intervenes the central frequency of filtering.

• un ensemble 2142, composé de deux blocs notés 1/D_i(ρz^-1) et
  Cet ensemble 2142 est prévu pour réaliser la détection d'une composante périodique a_i du signal de bruit propagé estimé de_k(n); et
• un filtre 2143, noté et prévu pour filtrer le signal de bruit propagé estimé de_k(n) à la fréquence de la composante périodique a_i détecté par l'ensemble 2142. Ce filtre 2143 fournit en sortie un signal d'k_i(n) composé seulement de la composante périodique a_i du signal de bruit propagé estimé de_k(n).

Thus, as represented on the figure 4 , the detection and filtering module 112 is composed of as many sections 2141 in cascade as periodic components to be detected. Each section i is in the form of a filter H _i (z ^-1 ) comprising:

• a set 2142, composed of two blocks denoted 1 / D _i (ρ z ^-1 ) and
  This set 2142 is designed to perform the detection of a periodic component a _i of the estimated propagated noise signal of _k (n) ; and
• a filter 2143, noted and adapted to filter the estimated propagated noise signal of _k (n) at the frequency of the periodic component a _i detected by the set 2142. This filter 2143 outputs a signal of k _i (n ) composed only of the periodic component a _i of the estimated propagated noise signal of _k (n).

The reference signal of _k (n) is obtained by adding all the signals of _ki (n) provided by the filters of sections 2141.

Note that this addition operation is indicated by a P, as parallel, in ALE ^P (see figure 3 ).

The operations of analyzing the noise signal, generating and adjusting the counter-noise signals y _k (n) for all the channels k of the multi-channel noise reduction system 21 according to the invention can be integrated on a only electronic card.

• les voies 300-303 correspondant à quatre signaux d'erreur, respectivement e₁(n)-e₄(n), mesurés par quatre microphones de contrôle, respectivement 310-313, disposés dans de la bulle de confort 22 ;
• la voie 304 correspond au signal de bruit x(n) mesuré par un microphone de bruit ; et
• la voie 305 correspond à un signal provenant d'un potentiomètre 315 permettant d'ajuster les coefficients de convergence feedback et feedforward intervenant dans les algorithmes LMS utilisés.

The figure 5 schematically represents an example of an electronic card 30 for a multichannel noise reduction system having 6 channels 300-305 input, and 4 channels 306-309 output. In entry of this card 30:

• the channels 300-303 corresponding to four error signals, respectively e ₁ (n) -e ₄ (n), measured by four control microphones, respectively 310-313, arranged in the comfort bubble 22;
• channel 304 corresponds to the noise signal x (n) measured by a noise microphone; and
• the channel 305 corresponds to a signal coming from a potentiometer 315 making it possible to adjust the feedback and feedforward convergence coefficients involved in the LMS algorithms used.

• les voies 306-309 correspondent à quatre signaux de contre-bruit, respectivement y₁(n)-y₄(n), destinés à être émis par quatre transducteurs, respectivement 316-319, adéquatement disposés.

At the exit of this card 30:

• the channels 306-309 correspond to four counter-noise signals, respectively y ₁ (n) -y ₄ (n), intended to be emitted by four transducers, respectively 316-319, suitably arranged.

• un étage 320 de pré-amplification, réalisant une pré-amplification des signaux de chacune des voies 300 - 304, à l'aide de pré-amplificateurs 3200-3204 ;
• un étage de gain 330, disposé en sortie de l'étage 320, et appliquant un gain aux signaux de chacune des voies 300 - 304 à l'aide d'amplificateurs 3300-3304 de gain réglable ;
• un étage 340 de filtrage anti-repliement en sortie de l'étage de gain 330, et réalisant un filtrage anti-repliement des signaux de chacune des voies 300-304, à l'aide de filtres anti-repliement 3400-3404. La fréquence d'échantillonnage au niveau des filtres 3400-3404 est réglable à l'aide d'un module 3405 ;

• en sortie de l'étage 340, un multiplexeur 31 réalisant un multiplexage des signaux des voies 300 - 304 ; et
• en sortie du multiplexeur 31, un convertisseur analogique-numérique 32, réalisant une conversion analogique numérique du signal multiplexé.

For each of the channels 300-304, the card comprises:

• a pre-amplification stage 320, pre-amplifying the signals of each of the channels 300 - 304, using pre-amplifiers 3200-3204;
• a gain stage 330, disposed at the output of the stage 320, and applying a gain to the signals of each of the channels 300 - 304 using amplifiers 3300-3304 of adjustable gain;
• an anti-aliasing filtering stage 340 at the output of the gain stage 330, and performing anti-aliasing filtering of the signals of each of the channels 300-304, using anti-aliasing filters 3400-3404. The sampling frequency at the 3400-3404 filters is adjustable using a 3405 module;

• at the output of the stage 340, a multiplexer 31 performing a multiplexing of the signals of the channels 300 - 304; and
• at the output of the multiplexer 31, an analog-digital converter 32, performing an analog digital conversion of the multiplexed signal.

The multiplexed digital signal, obtained at the output of the converter 32, then enters a processor 33 of the DSP type which makes it possible to carry out for each channel the operations that we have described above and represented diagrammatically in FIG. figures 3 and 4 . The processor 33 used in the present example is an Analog Devices processor of the SHARC range in industrial finish so resistant to extreme temperatures. The implementation of the code is ensured via the interface developed by Analog Devices is the VisualDSP ++ software, which has a high-level C compiler. It is possible to work either in floating point or in fixed comma. The sampling frequency at the level of the processor is configurable, using a module 331, to respond to all cases of active reduction of the energy of a sound signal.

The DSP 33 has been sized to accommodate operations inherent to the LMS algorithms used. The DSP can accommodate more complex algorithms than those used because an external memory 34 is present on the card 30, in order to meet any additional costs in memory and calculation.

In the case of a multi-card system, a link can be made between the different cards using the connection lines 35. This possibility has been designed to be able to extend to infinity active nuisance reduction applications. sound and not to have limitations due to the processor 33.

• un lissage par un étage de lissage 350 comprenant des filtres passe-bas 3500-3503. La fréquence d'échantillonnage au niveau des filtres 3500-3503 est réglable à l'aide du module 3405 ;
• une diminution en gain par un étage de gain 360 comprenant des amplificateurs 3600-3603 de gain réglable ; et
• une amplification en puissance par un étage d'amplification de puissance 370 comprenant des amplificateurs de puissance. Cet étage d'amplification de puissance 370 peut ne pas se trouver sur la carte 30 telle que représentée en figure 5.

At the output of the processor 33, the digital signal is composed of the counter-noise signals y ₁ (n) -y ₄ (n). This digital signal is converted using a digital-to-analog converter 36. Then the analog signal obtained enters a demultiplexer 37 and demultiplexed. After the demultiplexing the different counter-noise signals y ₁ (n) -y ₄ (n) are separated and located on exit routes 306-309. Before being emitted by the transducers 316-319, the counter-noise signals undergo:

• smoothing by a smoothing stage 350 comprising low-pass filters 3500-3503. The sampling frequency at the 3500-3503 filters is adjustable using the 3405 module;
• a gain reduction by a gain stage 360 comprising amplifiers 3600-3603 of adjustable gain; and
• power amplification by a power amplification stage 370 comprising power amplifiers. This power amplification stage 370 may not be on the board as shown in FIG. figure 5 .

The adjustment signal of the feedback and feedforward convergence coefficients coming from the potentiometer 315 on the channel 305 is amplified by an amplifier 3051 and then an analog-to-digital conversion by means of a digital analog converter 3052 before entering the processor 33. convergence coefficient is a weighting factor, strictly positive and less than 1, applied at the level of the reactualization in the LMS algorithm of the coefficients of the various filters mentioned above.

Transducers 316-319 used in the present example are ultrasound transducers. These ultrasound transducers 316-319 have a transmission beam 61, represented in FIG. figure 6 , very small. In addition, ultrasound, completely inaudible to the emission, distort as they spread in the air and slide into the audible spectrum and the volume in which they become audible is quite predictable.

The figure 7 schematically represents a first exemplary embodiment of the multi-channel system according to the invention for obtaining a comfort bubble 22 using 4 ultrasound transducers 316-319 properly placed on a desk table 71. The positioning of these transducers is obviously not limited to the office alone. They can quite be arranged around an opening, a window or a door for example. The comfort bubble 22 obtained is located substantially at a level corresponding to the level of the head of a user on the desk table 71.

Another embodiment of the system according to the invention is shown schematically in figure 8 . This is a booth 80 intended to accommodate one or more users 81 to provide a noise reduction zone around their head. It is designed to be implemented both in public spaces and in factories, and can also be a medium for advertising.

The principle is as follows: a multitude of noise microphones 82 implanted at the level of the structure of the isolator 81 provide the noise signals, bases for the algorithm described above for calculating the counter-noise signals propagated by a multitude of secondary sources 83 located in the polling booth 80. A network of control microphones 84, around which is the comfort bubble, allows the real-time adaptation of the filters described above. Display panels 85 allow the display of information such as advertising. The isolator 80 includes one or more seats or buttocks 86 allowing the user 81 to land.

The advantage of such integration of the comfort bubble is that the low frequency performance of the active control is combined with the known efficiency of passive material acoustic treatments, whose structure and the booth of the booth is coated. Thus, a completely satisfactory and homogeneous attenuation is obtained over the entire noise spectrum.

Naturally, the invention is not limited to the examples of applications that we have just described and can be applied to the reduction of the energy of any sound signal in a given zone.

Claims (17)

1. Method for the active reduction in a determined zone (22) of the energy of a sound signal (d_k(n)), called propagated noise signal, generated in said zone (22) by a primary signal (x_k(n)), called noise signal, said method comprising a transmission, by transmission means, of at least one counter-noise signal (y_k(n)) comprising at least one first so-called feedback counter-noise signal (yfbk_k(n)), counteracting said propagated noise signal (d_k(n)), said method also comprising at least one iteration of the following operations:

   - measurement, by measurement means arranged in said determined zone (22), of a so-called error signal (e_k(n)), representing information on the effectiveness of the reduction of the energy of the propagated noise signal (d_k(n)) in said zone (22);

   - modelling, by at least one first filter (Ŝ_kk(z)), of a direct acoustic path (S_kk), called secondary path, between said transmission means of the counter-noise signal (y_k(n)) and said measurement means of said error signal (e_k(n));

   - detection of at least one periodic component of said propagated noise signal (d_k(n)) by analysis of a propagated noise signal (de_k(n)) estimated from, on the one hand, the error signal (e_k(n)) and, on the other hand, the feedback counter-noise signal (yfbk_k(n)) processed by the first filter (Ŝ_kk(z)), said detection providing said periodic component; and

   - adjustment of said feedback counter-noise signal (yfbk_k(n)) as a function of said detected periodic component, of said error signal (e_k(n)) and of said modelled secondary path (Ŝ_kk(z)).
2. Method according to claim 1, characterized in that it also comprises a band-pass filtering of the estimated propagated noise signal (de_k(n)), at the frequency of all or some of the detected periodic components, said filtering providing a so-called reference signal (d'_k(n)).
3. Method according to claim 2, characterized in that the adjustment of the feedback counter-noise signal (yfbk_k(n)) comprises an adjustment of at least one coefficient of a second finite impulsional response filter (W^fbk _k(z)), said second filter being provided to adjust said feedback counter-noise signal (yfbk_k(n)) as a function of the reference signal (d'_k(n)) filtered by a third finite impulsional response filter (1/Ŝ_kk(z)) amplitude modelling the inverse of the secondary path.
4. Method according to claim 3, characterized in that at least one coefficient of the second filter (W^fbk _k(z)) is adjusted by an algorithm of the minimization algorithm type according to the least mean squares (LMS) criterion as a function of the reference signal (d'_k(n)) processed beforehand by the first filter (Ŝ_kk(z)), of the error signal (e_k(n)) that has previously undergone a band-pass filtering at the frequency of all or some of the detected periodic components and of a so-called feedback convergence coefficient.
5. Method according to any one of the previous claims, characterized in that the counter-noise signal (y_k(n)) also comprises a so-called feedforward counter-noise signal (yfwd_k(n)), adjusted as a function of the error signal (e_k(n)), of the noise signal (x_k(n)) measured by measurement means (314).
6. Method according to claim 5, characterized in that it also comprises an amplitude modelling of the inverse of the secondary path (S_kk) by at least one fourth finite impulsional response filter (1/S_kk(z)).
7. Method according to claim 6, characterized in that the adjustment of the feedforward counter-noise signal (yfwd_k(n)) comprises an adjustment of at least one coefficient of a fifth finite impulsional response filter (W^fwd _k(z)), said fifth filter being provided to adjust said feedforward counter-noise signal (yfwd_k(n)) as a function of the noise signal (x_k(n)) processed beforehand by the fourth filter (1/Ŝ_kk(z)).
8. Method according to claim 7, characterized in that at least one coefficient of the fifth filter (W^fwd _k(z)) is adjusted by an algorithm of the least mean squares (LMS) algorithm type as a function of the error signal (e_k(n)), of the measured noise signal (x_k(n)) processed beforehand by a sixth filter (Ŝ_kk(z)) modelling the secondary path (S_kk) and of a so-called feedforward convergence coefficient.
9. Method according to any one of the previous claims, characterized in that it is used to attenuate at least one propagated noise signal (d_k(n)) by transmission of a plurality of counter-noise signals (y₁(n)-y₄(n)) by a plurality of transmission means (316-319).
10. System of active reduction, in a determined zone (22), of the energy of a sound signal (d_k(n)), called propagated noise signal, generated in said zone (22) by a primary signal (x_k(n)), called noise signal, by transmission of at least one counter-noise signal (y_k(n)) comprising at least one first so-called feedback counter-noise signal (yfbk_k(n)), counteracting said propagated noise signal (d_k(n)) in the determined zone (22), said system comprising :

    - means (316-319) for transmitting the counter-noise signal (y_k(n));

    - means (310-313) of measuring, in said determined zone (22), a so-called error signal (e_k(n)), representing information on the effectiveness of the reduction of the energy of said propagated noise signal (d_k(n));

    - at least one first filter (Ŝ_kk(z)) for modelling a direct acoustic path (S_kk), called secondary path, between said transmission means (316-319) of the counter-noise signal (y_k(n)) and said measurement means (310-313) of said error signal (e_k(n)).

    - means (213) for estimating the propagated noise signal (d_k(n)) from, on the one hand, the error signal (e_k(n)) and, on the other hand, the feedback counter-noise signal (yfbk_k(n)) processed by the first filter (Ŝ_kk(z)), said means (213) providing an estimated propagated noise signal (de_k(n)).

    - means (214) for detecting and providing at least one periodic component of said propagated noise signal (d_k(n)) by analysis of said estimated propagated noise signal (de_k(n)); and

    - means for adjusting said feedback counter-noise signal (yfbk_k(n)) as a function of said detected periodic component, of said error signal (e_k(n)) and of said modelled secondary path (Ŝ_kk(z)).
11. System according to claim 10, characterized in that the means (316-319) for transmitting the counter-noise signal (y_k(n)) comprise ultrasonic transducers having a reduced transmission beam (61).
12. System according to any one of claims 10 or 11, characterized in that it also comprises means for band-pass filtering (214) of the estimated propagated noise signal (de_k(n)) at the frequency of all or some of the detected periodic components, said filtering means providing a reference signal (d'_k(n)).
13. System according to claim 12, characterized in that the means for adjusting the feedback counter-noise signal (yfbk_k(n)) comprise at least one second, finite impulsional response, filter (W^fbk _k(z) provided to adjust said feedback counter-noise signal (yfbk_k(n)) as a function of the reference signal (d'_k(n)) filtered by a third filter (1/Ŝ_kk(z)) amplitude modelling the inverse of the second path.
14. System according to any one of claims 10 to 13, characterized in that the counter-noise signal (y_k(n)) comprises a second so-called feedforward counter-noise signal (yfwd_k(n)), the system also comprising means for adjusting said feedforward counter-noise signal (yfwd_k(n)) as a function of the error signal (e_k(n)) and of the noise signal (x_k(n)).
15. System according to claim 14, characterized in that it also comprises a fourth, finite impulsional response, filter (1/Ŝ_kk(z)), arranged for amplitude modelling the inverse of the secondary path.
16. System according to claim 15, characterized in that it also comprises a fifth filter (W^fwd _k(z)), provided to adjust the feedforward counter-noise signal (yfwd_k(n)), as a function of the noise signal (x_k(n)) processed by the fourth filter (1/Ŝ_kk(z)).
17. System according to any one of claims 10 to 16, characterized in that it also comprises a plurality of transmission means (316-319) of a plurality of counter-noise signals (y₁(n)-y₄(n)).

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