Method and device for obtaining a digital audio signal filter, method and device for implementing the filter

Abstract

A computer-implemented method is described for obtaining M filters for a digital audio signal defined in the time domain, where M is an integer greater than or equal to 1, each filter being associated with a respective sound source, called a secondary source, that emits a sound pressure field that enables minimization of the reflected sound pressure field of a set of sound sources, called primary sources, located in a room, the method comprising: - obtaining (S601) for each primary and secondary sound source an impulse response of the reflected sound pressure at N respective distinct positions in the room, where all of the primary and secondary sound sources are placed at operating positions and the N positions are the same for all of the sound sources; - determining a set of M filters by solving a regularized optimization problem formulated in the time domain based on the set of impulse responses (S602), the optimization problem including - (a) Minimizing the norm of the sum of the impulse responses at N locations, and (b) defined to introduce a non-zero action delay for each filter; Each secondary sound source is paired with an associated primary sound source known as the primary sound source.

Description

[Technical Field]

[0001] The present invention relates to a method and a device for obtaining a digital audio signal filter for use in a device for generating a signal for an audio source. The present invention also relates to a signal processing method for implementing the filter.

[0002] The present invention is used to reduce the reflection of sound waves generated by this sound source. [Background technology]

[0003] When a sound source emits sound within a closed space, the sound waves are reflected by the various walls until the waves are completely attenuated, either through absorption by the walls or (for higher frequencies) by thermal heating in the air.

[0004] At low frequencies, the attenuation of sound waves is limited and the time required for their complete absorption can be long (more than 1 second) even if absorbing material is present on the wall.

[0005] This is due to the fact that most of the absorbing materials used dissipate acoustic energy by visco-thermal friction, and their effectiveness is related to the thickness of the material in relation to the wavelength under consideration.

[0006] Note that wavelength is equal to the speed of sound in air divided by frequency. Therefore, at lower frequencies, wavelengths are longer. As an example, the wavelength associated with 50 Hz is equal to 6.8 meters.

[0007] This means that very thick absorbing material is required to attenuate sound waves in this frequency range, and that thickness is naturally greater than that of a medium-sized room (100m 2 This means that the product is not naturally suitable for use in areas below 200°C.

[0008] Furthermore, at certain frequencies whose corresponding wavelength is a multiple of the room dimension, resonance phenomena appear, characterized by highly inhomogeneous pressure fields with zones of very high acoustic energy (known as pressure antinodes) and other zones of very low acoustic energy (known as pressure nodes). These resonance phenomena, also known as "acoustic modes" or "room modes," significantly degrade the sound quality of a sound source compared to a situation emitting in a free field (e.g., outdoors).

[0009] There are different types of damage caused by acoustic modes. - Spatial: for a given frequency, areas of very high pressure coexist with areas of very low pressure, resulting in high spatial variability of sound levels. - Frequency: As a result, for a fixed position, there is a large variation in the magnitude of pressure based on frequency, commonly referred to as the "frequency response". - Time: Acoustic modes with low decay of sound energy can result in very high decay times, also known as reverberation times. This is commonly referred to as trailing.

[0010] The number of acoustic modes per frequency band, also known as "modal density," depends on the volume of the room. Larger rooms have a greater number of acoustical modes in a given frequency band compared to smaller rooms. However, low modal density is more problematic because the modes present tend to be more distinguishable (because they are further apart in the frequency domain) and more energetic.

[0011] Room modes are therefore particularly problematic in mixing and listening rooms, as well as small and medium-sized concert halls, which can strongly modify low frequencies (between 20 Hz and 100-200 Hz). For larger rooms, a higher modal density makes the room acoustics less of a problem.

[0012] The Schroeder frequency simply quantifies the transition between modal behavior, where different acoustic modes are clearly distinguishable and more energetic, and statistical behavior, where the frequencies of these modes are too close to each other to be clearly distinguishable and the modes have low energy.

[0013] This frequency is inversely proportional to the volume of the room, so in large concert halls, the frequency is low enough that no correction is needed to switch to a less problematic statistical behavior. Such control would be too complex to set up.

[0014] Certain approaches have been proposed to attenuate acoustic modes by adding so-called "secondary" sources that function to absorb the sound waves emitted by the so-called "primary" sources. These secondary sources are placed, for example, on the wall opposite the one where the primary sources are located. These include the "CABS" (Controlled Acoustic Bass System) approach by Adrian Celestinos et al. (see Reference 1 in the Appendix). Another approach is the "impedance matching" approach proposed by the Ecole Polytechnique Federale de Lausanne. A third approach is described by Franz Heuchel et al. in their paper "Compensation active d'une piece pour le renforcement du son utilisant des techniques de separation de champ sonore" ["Active room compensation for sound reinforcement using sound field separation techniques"] (Reference 2). This approach uses plane wave decomposition (PWD) to model the incident and reflected fields. The authors then solve a frequency-domain optimization problem aimed at finding the optimal filter for the secondary sources to cancel the reflected fields. This method requires a calibration step using two sets of microphones placed around the room.

[0015] A fourth approach consists in adding electronic corrections using filters to modify the frequency response curve at one or more reference positions in the room. These corrections are not intended to suppress acoustic modes, but simply to "equalize" the frequency response at specific positions in a preferential way by reducing the sound pressure level at the transmitter at frequencies corresponding to pressure antinodes, and sometimes by increasing the sound pressure level at frequencies corresponding to pressure nodes.

[0016] However, this fourth approach is unsatisfactory because reducing the sound pressure level at emission degrades the pre-reflected sound waves, also known as the "direct field," which loses precision and impact.

[0017] US Patent No. 8,660,272 describes a technique for reducing sound reflections from loudspeaker-equipped devices in a room, where the signal fed to the loudspeaker comprises an audio signal to which is added an identical filtered audio signal, the filtering comprising the application of multiple filter stages, each stage comprising a delay and a gain, the characteristics of each stage being obtained iteratively.

[0018] Sound produced by one or more sources in a room, requiring few resources during the calibration phase It is desirable to have a solution to reduce wave reflection. Summary of the Invention

[0019] A first aspect relates to a computer-implemented method for obtaining M filters for a digital audio signal defined in the time domain, where M is an integer greater than or equal to 1, each filter being associated with a respective sound source, called a secondary sound source, emitting a sound pressure field that allows minimizing the reflected sound pressure field of a set of sound sources, called primary sound sources, located in a room, the method comprising: - obtaining, for each primary and secondary sound source, N impulse responses of the reflected sound pressure at N respective distinct positions in the room, where all of the primary and secondary sound sources are placed at operating positions and the N positions are the same for all of the sound sources; determining a set of M filters by solving a regularized optimization problem formulated in the time domain based on a set of impulse responses, the optimization problem being: (a) Minimizing the norm of the sum of the impulse responses at N locations, and (b) defined to introduce a non-zero action delay for each filter; Each secondary sound source is paired with an associated primary sound source known as the primary sound source.

[0020] Each filter is intended to be applied to a signal derived from the input audio signal of the respective audio source associated with the filter, and each audio source is configured to broadcast the sum of the input audio signal and the filtered signal, with the audio input signals from the audio sources being identical, thereby reducing the reflections of the generated sound waves.

[0021] According to one or more embodiments, M is 2 or greater.

[0022] According to one or more embodiments, obtaining an impulse response of reflected sound pressure at a given one of the N locations for a given secondary sound source combined with its associated primary sound source comprises: -emitting sound waves in response to an excitation signal by a given secondary sound source in combination with its associated primary sound source; - obtaining a signal representative of the sound pressure resulting from the excitation signal at a location; - determining an impulse response of the reflected sound pressure based on the acquired signal representative of the sound pressure.

[0023] According to one or more embodiments, the sound pressure signal is acquired using a microphone placed at a given position, and determining the impulse response includes applying time windowing to the sound pressure signal to suppress direct sound waves received from a given secondary sound source combined with its associated primary sound source while retaining reflected sound waves.

[0024] According to one or more embodiments, sound pressure signals are acquired using microphone pairs positioned around a given location, and determining the impulse response includes determining the pressure and velocity of the sound waves in order to separate direct sound waves received from the sound source from reflected sound waves.

[0025] According to one or more embodiments, the action delay is substantially equal to the average propagation time of sound waves generated by the sound source between the sound source and the walls of the room.

[0026] According to one or more embodiments, the method includes determining a regularization parameter for regularizing the optimization problem, the determination taking into account a maximum amplitude threshold of the filter.

[0027] A second aspect relates to a data processing device comprising means for implementing one of the methods above.

[0028] A third aspect is an audio signal processing device, comprising: an input configured to receive a first audio signal (x(t)); a first filter for filtering the first signal to obtain a second audio signal, the first filter being a finite impulse response filter obtained by applying one of the filter obtaining methods described above; an adder for adding the first audio signal and the second audio signal to obtain a third audio signal for controlling a sound source associated with the first filter.

[0029] According to one or more embodiments, the audio signal processing device comprises a low pass filter for filtering the first audio signal, the output of which is connected to the input of the first filter.

[0030] According to one or more embodiments, the audio signal processing device comprises a downsampling circuit that downsamples the audio signal after the low pass filter and before supplying it to the first filter, and an oversampling circuit that oversamples the audio signal after filtering by the first impulse response filter and before supplying it to the summer.

[0031] According to one or more embodiments, the audio signal processing device an adjustable attenuator for applying a gain between 0 and 100% to the second audio signal; and a switch configured to connect or disconnect the second signal from the summer input.

[0032] A fourth aspect relates to an audio signal processing method implemented by a device comprising a processor, a memory, and software code, the method comprising: receiving a first audio signal; - filtering the audio signal by means of a finite impulse response filter obtained by applying one of the above filtering methods; - adding the first audio signal and the signal filtered by the finite impulse response filter to form a signal suitable for feeding to an audio source associated with the finite impulse response filter.

[0033] According to one or more example embodiments, a method for processing an audio signal includes low-pass filtering a first audio signal prior to filtering with an impulse response filter.

[0034] According to one or more example embodiments, a method for processing an audio signal includes downsampling the audio signal after low-pass filtering and before impulse response filtering, and oversampling the audio signal after impulse response filtering and before summing.

[0035] One or more embodiments relate to a storage medium readable by a device having a processor, the medium including instructions that, when executed by the processor of the device, cause the device to implement at least one of the described methods. In particular, the storage medium may be non-transitory. [Brief explanation of the drawings]

[0036] Further features and advantages of the present invention will become apparent from the following detailed description, which may be read in conjunction with the accompanying drawings.

[0037] [Figure 1] FIG. 1 is a functional block diagram of a device according to one or more embodiments. [Figure 2] FIG. 2 is a functional block diagram of a set of devices similar to those of FIG. 1 receiving the same audio signal as input. [Figure 3] 1 illustrates a schematic diagram of a room with several primary sound sources and several secondary sound sources designed to reduce reflections of waves generated by the primary sound sources, in accordance with one or more exemplary embodiments. [Figure 4] 10 is a curve showing filter magnitude based on the norm of the corrected error for different lambda regularization parameters of an optimization problem. [Figure 5] 1 illustrates a schematic representation of a room with several combined primary and secondary sound sources, according to one or more exemplary embodiments; [Figure 6] 1 is a flowchart of a method for determining one or more filters according to one or more embodiments. [Figure 7] FIG. 1 is a block diagram of an example device for determining a filter, in accordance with one or more example embodiments. DETAILED DESCRIPTION OF THE INVENTION

[0038] In the following description, identical, similar or analogous elements are referred to by the same reference numerals.

[0039] The block diagrams, flowcharts, and message sequence diagrams in the figures illustrate the architecture, functionality, and operation of systems, apparatuses, methods, and computer program products according to one or more example embodiments. Each block in the block diagrams or each step in the flowcharts may represent a module or portion of software code, comprising instructions for implementing one or more functions. According to particular embodiments, the order of the blocks or steps may be changed, or corresponding functions may be implemented in parallel. The blocks or steps of the methods may be implemented using circuits, software, or a combination of circuits and software, in a centralized or distributed manner, for all or some of the blocks or steps. The described systems, devices, processes, and methods may be modified, or may be added and / or deleted, while remaining within the scope of the present disclosure. For example, device or system components may be integrated or separated. Similarly, the disclosed features may be implemented using more or fewer components or steps, or with other components or even other steps. Any suitable data processing system may be used for implementation. A suitable data processing system or device includes, for example, a combination of software code and circuitry, such as a processor, controller, or other circuitry suitable for executing software code. When executed, the software code causes a processor or controller to prompt a system or device to implement all or part of the functionality of the blocks and / or steps of a process or method according to the exemplary embodiments. The software code may be stored in a memory or readable medium accessible by the processor or controller directly or via other modules.

[0040] One embodiment relates to a device for processing an audio signal intended to be broadcast by an audio source, the audio source typically being an electroacoustic transducer that converts an electrical signal into an acoustic signal. The transducer is for example a loudspeaker, but other transducers may also be used. The terms audio source, transducer and loudspeaker are used interchangeably in the following.

[0041] Generally, in accordance with one or more embodiments, an audio signal generated by an audio signal processing device The signal to be transmitted includes an input audio signal, known as a reference signal, e.g., music content, and a control signal that attenuates reflected waves in the room broadcast by the transducer. More precisely, the control signal corresponds to a delayed and filtered input audio signal. Thus, when the input audio signal is fed to the transducer, the processed audio signal propagates through the room and is reflected off the walls. As the waves reflect and propagate through the room, they undergo changes similar to filtering. After a certain delay relative to the reference audio signal, the transducer emits a copy of the reflected and filtered signal in antiphase. The reflected waves are then canceled, and resonance can no longer be established.

[0042] Depending on the desired implementation, the signal processing device is adapted to generate signals for either one or several transducers. The processing device may be combined into an apparatus having one or more transducers or may be in a separate apparatus.

[0043] FIG. 1 is a schematic diagram illustrating, in functional block form, the processing performed on a reference signal and the transducer fed by the processed signal. The device 100 has an input 101 for a reference audio signal x(t). This audio signal is intended to be broadcast by a loudspeaker 103, which is external to the device 100 in this example. For the purposes of this description, it is assumed that the reference signal is a digital signal; if not, a person skilled in the art would provide a circuit for digitizing the reference signal, as is well known. The reference audio signal input is connected to a first input of a summer 104, the output of which is connected to the loudspeaker 103. A reference signal bypass path, connected on the one hand to the signal input 101 and on the other hand to the second input of the summer 104, comprises a digital filter w 105 designed to filter the signal at its input and generate a control signal. According to one embodiment, the digital filter advantageously implements both delay and filtering of the signal at its input. Filter 105, in relation to its role in generating the control signal, is also referred to below as a "control filter."

[0044] According to one embodiment, the delay and filtering are introduced by separate units.

[0045] According to this embodiment, before being subjected to the filter 105, the input audio signal is subjected to a low-pass filter 106 whose role is to ensure that the filter 105 is applied only to the frequency range where reflection compensation is necessary or desired. This low-pass filtering can be achieved by downsampling the audio signal. Downsampling is performed at a frequency much lower than the frequency of the original audio signal, and filtering is performed at this new sampling frequency. For a given computational resource, this downsampling increases the efficiency of the filter (also known as increasing the filter order). Before summation by the adder 104, the filtered signal is oversampled to a sampling frequency identical to that of the reference signal.

[0046] According to one or more embodiments, the low pass filter is optional, for example, if the reference signal is already in the desired frequency band.

[0047] Reference numeral 102 denotes a signal processing block including a summer 104, a filter 105, and a low-pass filter 106. Device 100 may include other circuits such as a power amplifier. Block 102 may be an adapted signal processor that implements the functionality shown in FIG.

[0048] By way of example, in certain applications the downsampling frequency may be 500 Hz.

[0049] In a particular embodiment, the filter w(t) 105 is a finite impulse response filter that performs a convolution operation on the signal received at its input.

[0050] 1 illustrates the main signal processing operations performed and the basic concepts applied. Other components or functions may be present and additional signal processing may be performed. Furthermore, the actual implementation may of course differ from that shown in FIG. 1; in particular, the signal processing block 102 itself may be partially or fully implemented using a suitable signal processor.

[0051] Fig. 2 shows a set of several processing devices 100_i, each of which comprises the elements shown in Fig. 1. In the example shown in Fig. 2, all processing devices receive as input the same reference audio signal x(t). _i (t) can be distinguished for each processing device 100_i, where i is an integer from 1 to M, and M is the number of processing devices. In the following, we consider the general case of FIG. 2 and note that "i" can be considered to be equal to 1. For clarity, not all elements of device 100 shown in FIG. 1 are shown in FIG. 2.

[0052] According to a particular embodiment, when broadcasting separate reference audio signals on the same number of channels (a "multi-channel" context), the devices shown in FIG. 2 are reproduced for each of the reference audio signals.

[0053] The delays and filtering applied to generate the control signals are fixed in time and specific to each transducer or loudspeaker for a given loudspeaker arrangement in a given room.

[0054] According to one embodiment, the delays and filtering are not fixed in time, but are periodically or periodically re-evaluated based on one or more parameters, such as temperature changes, room occupancy, changes in transducer layout or orientation, etc.

[0055] The optimal individual filters and delays to be applied are obtained during a pre-calibration phase by a single optimization calculation for all loudspeakers, including the impulse responses between each loudspeaker and a set of N "control" microphones placed in the room. The impulse responses between each loudspeaker and the set of microphones placed in the room can be obtained either by a measurement or a simulation process. The measurement phase is followed by a calibration phase, which provides the delay and filter characteristics used to parameterize each signal processing device. According to one embodiment, the optimization problem consists in finding a set of M filters that minimizes the norm L2 of the reflected field. The formulation of the optimization problem in the time domain makes it possible to impose individual action delays for each filter, which avoids obtaining filters that cancel reference signals at low frequencies as soon as they are transmitted, as follows:

number

[0056] According to one alternative embodiment, the action time is the same for all filters. The smallest action time suitable for all filters is then taken.

[0057] The optimization problem is an ill-conditioned linear inverse problem. Therefore, the solution must be regularized. This regularization is obtained using Tikhonov's time-domain method. The regularization provides a filter whose amplitude can be controlled and limited.

[0058] In certain embodiments, if the signal processing is implemented by a signal processor, the latency of this processor is taken into account in the optimization problem. Indeed, in addition to avoiding the solution described by Equation 1 above, the possibility of imposing an action delay on the filter makes it possible to take this latency into account and avoid the truncation of the first filter samples associated with real-time digital implementations.

[0059] General case First, the location of the optimization problem for obtaining the w_i filters in the general case is described, where secondary sound sources different from the primary sound sources are used to compensate for reflections due to signals from these primary sound sources, and a control microphone is used during a calibration phase to obtain the filters to be applied to the reference audio signal and to obtain the acoustic signal broadcast by the secondary sound sources. Second, the advantageous case is described where primary and secondary sound sources are combined.

[0060] Figure 3 is a schematic diagram of a room with sound sources. This non-limiting diagram helps to explain the position of the optimization problem in the general case. a primary audio source 301 capable of broadcasting a reference audio signal; a secondary sound source 302 placed near a reflecting wall 303, from which the sound waves emitted by the primary sound source are reflected; a bank of control microphones 304; evaluation microphones 305 placed around the room.

[0061] The optimization problem to be solved to control the waves reflected by the secondary source is: n p Primary sound source and n s Secondary sound sources are considered, the latter being initially considered separate from the former. The sound field generated by the secondary sound sources, when reflected by the walls 303, is expected to cancel the field emitted by the primary sound source. Measurements obtained by an array of control microphones 304 are sent to an optimizer to obtain an electronic filter. An evaluation microphone 305 can optionally be used to check the cancellation quality.

[0062] The optimizer can be implemented using any device with one or more processors capable of solving the optimization problem and processing data to obtain a filter. FIG. 7 is a block diagram illustrating an example of such a device 700, comprising a processor 701, a working memory 702, a long-term storage memory 703 containing software code, a communication interface 704, and a communication bus 705 connecting the various components. The communication interface allows for receiving impulse response data and providing filters to the processing device 100_i, as well as controlling any necessary devices. The processor, when executing the software code, causes the device 700 to implement at least one filtering method as described. It should be noted that the device 700 may comprise other components depending on the implementation (user interface, display, etc.).

[0063] Frequency domain problem formulation The objective is to find that the sum of the primary and secondary fields generated by the primary and secondary sources, respectively, is equal to the reflected pressure, p for all pairs of control microphones. r The goal is to determine appropriate filters to apply to the secondary sources so as to minimize . Finding these filters involves solving an optimization problem that takes into account the signals from all the control microphones.

[0064] For clarity, the pressures measured by all control microphones are expressed in vector form p, so that p=[p1...p j ...p N ] T Presented in, where p j is the pressure evaluated for the jth pair of microphones, N is the total number of microphones, and [.] T is the matrix transpose. A filter vector w is also introduced, so that w=[w1w2...w M ] T , where M is the total number of secondary sources.

[0065] The complete pressure field can be expressed as the sum of the primary and secondary fields as follows:

number

number

number

[0066] Several methods for extracting the reflected pressure from the measurement of the impulse response at the control microphone are described in detail below. In the general case, the reflected pressure is assumed to be a linear function of the electronic filter, so that the pressure p r can be expressed as follows:

number

[0067] The objective is to determine the electronic filters to be applied to the secondary sources so that the reflected pressure field is minimized. This involves expressing a cost function as:

number

[0068] Considering Equation 4, Equation 5 can therefore be expressed as:

number

[0069] This is a least-squares optimization problem whose solution is a linear inversion.

number

[0070] Time Domain Formulation According to this exemplary embodiment, the optimization problem is formulated in the time domain, and therefore the filter obtained by solving this problem is a filter that is also applied in the time domain.

[0071] The time domain formulation offers the following advantages, among others: This makes it possible to take into account constraints related to the firmware of the signal processing device. It is easier to implement additional delays (compared to the action delays mentioned above) linked to the application, for example it is possible to force the output of the filter generating the control signal to zero for as many samples as necessary to introduce the desired delay. - Non-causal components due to the inverse Fourier transform of a bandwidth-limited frequency filter are avoided, since part of the impulse response energy is present at the end of the filter.

[0072] As is well known, one difference between the frequency domain and the time domain formulation is that multiplication in the frequency domain corresponds to convolution in the time domain. Consider the example of multiplying a frequency transfer function H(f) by a filter W(f) to give a pressure P(f), which can be expressed as:

number

[0073] The equivalent in the continuous time domain is

number

[0074] When applied to discrete numbers, this can be expressed in the discrete time domain as:

number

number

number

[0075] The equations for the various variables are rewritten in the time domain. P becomes:

number

number

number

number

number

number

number

number

[0076] With the time problem posed in this matrix form, the previously established cost function formulation remains valid.

number

[0077] Estimation of reflected pressure fields by plane wave decomposition The separation of incident and reflected waves can be based on the principle of plane wave decomposition, which models the sound field at each location as two plane waves moving in opposite directions: an incident component and a reflected component p i and p r and

number

[0078] Conventional microphones can only pick up pressure information, and two rows of microphones are used to obtain velocity information as well. front and p back denotes the pressure at the first and second microphones of a pair of microphones, as shown in FIG. 3, where the front microphone is the microphone in the row furthest from the reflecting wall.

[0079] The pressure and normal velocity can then be estimated at a virtual point between the two microphones of the pair separated by a distance d by defining:

number

[0080] normal velocity u n is estimated here by a finite difference approximation of the Euler equations.

[0081] Appropriate filters are then sought to be applied to the secondary sources so that the sum of the primary and secondary fields generated by the primary and secondary sources, respectively, minimizes the reflected pressure, p for all pairs of control microphones. Finding these filters involves solving an optimization problem that takes into account the signals from all control microphones.

[0082] For clarity, the pressures and velocities of all control microphones are presented in vector form as p and u, respectively. n and as a result un=[u n1 ...u nj ...u nN ] T , where u ni is the velocity estimated for the i-th pair of microphones, and p=[p1...p j ...p N ] T Here p j is the pressure evaluated for the jth pair of microphones, N is the total number of microphones, and [.] T is the matrix transpose. A filter vector w is also introduced, so that w=[w1w2...w M ] T , where M is the total number of secondary sources.

[0083] For both pressure and velocity, the complete field can be expressed as the sum of the primary and secondary fields, as follows:

number

number

number

number

[0084] Therefore, the equation for the reflected pressure field is found using a linear operator according to Equation 5 below:

number

[0085] Total Pressure Field Windowing According to an alternative embodiment, the direct and reflected fields can be separated by applying time windowing to the pressure signal measured by the control microphone, thus avoiding the use of velocity estimates.

[0086] Simple windowing can consist in removing the first instant of the pressure signal from the primary source that is identified as part of the direct field. All that remains is the reflected field. The total pressure of the primary and secondary sources to be minimized can then be written as:

number

number

number

number

number

[0087] The solution can be obtained as before by solving equation 7. In practice, the time involved (L × signal sampling frequency) is chosen to be on the order of the propagation time between the primary source and the room walls, and after this propagation delay, only reflections remain.

[0088] Special cases where a primary source can also be a secondary source According to one or more embodiments, the secondary and primary sound sources are combined in the sense that the physical sound source broadcasts both a reference signal (as the primary sound source) and a control signal (as the secondary sound source) resulting from delaying and filtering the reference signal. Thus, the sound source performs or checks itself. In other words, the sound source emits a first wave (due to the reference signal) and delays it to reduce reflections from the first wave before emitting a second wave (due to the control signal). The sum of the two signals therefore reduces the strength of the reflected pressure field.

[0089] The ability to apply a delay to the control filter makes it possible to implement this delay, and the control filter is forced to provide an output only after the delay time. This avoids the trivial solution to the optimization problem, where the source simultaneously emits a reference signal and its opposite in the form of a control signal; indeed, if a source wants to cancel the reflected pressure field, one way is to suppress the total pressure field directly at the source.

[0090] If no delay is imposed on the filter, the optimization problem has the following trivial solution:

number

number

[0091] Mathematically, one or more of the first components of each filter are forced to zero to impose a delay at which the filter does not operate.

[0092] This can be done, for example, by removing components from the optimization problem that are desired to be set to zero. In the convolution matrix formula, this amounts to removing a number of columns to the left equal to the number of samples to which it is desired to impose a start delay. For example, for a delay equal to 1 sample, we have the following convolution matrix:

number

number

[0093] Similarly, the same number of first components in each source filter are deleted w in the filter vector w defined by j is.

number

[0094] In the second step, after the optimization problem is solved, the same number of zeros are added to the beginning of each filter as were initially removed.

[0095] regularization The optimization problem of finding the optimal filter is an ill-conditioned linear optimization problem, so that, depending on the measurement noise present and the uncertain nature of the problem, the resulting filter amplitude may be much higher than necessary to control the field.

[0096] According to one alternative embodiment, the amplitude is controlled by incorporating a regularization term into the optimization problem. For example,

number

number

[0097] The function of this regularization parameter, as a multiplication factor for the filter norm, is to compensate the effort to minimize the reflected pressure field by adding another variable to be minimized, namely the filter norm. This forces the method to find a compromise between minimizing the reflected pressure field and using a large amplitude filter to achieve this. It is a good idea to limit the amplitude of the filter, and indeed, if the amplitude of the control signal is too high, the laminar pressure will be too high. This can be affected by the limiter of the amplifier in the loudspeaker.

[0098] The regularization parameter can be determined, for example, using the "L-curve" method. An example of the use of such a curve is given in Reference 3 in the Appendix. This method is based on the observation that the larger the filter norm, the smaller the error, and vice versa. In this embodiment, the error is defined as the resulting residual reflected pressure field, whose norm is required to be equal to zero. This method consists of plotting the log-log norm of the filter versus the norm of the corrected error for different values ​​of the regularization parameter λ. Figure 4 shows such a curve, with the filter norm on the vertical axis and the error norm on the horizontal axis. The larger λ is, the more the filter norm is minimized, as opposed to the norm of the reflected pressure field. The reverse is also true. A good choice for λ is its value at the inflection point (circled on the graph in Figure 4). According to Reference [3] in the Appendix, this achieves the best compromise between minimizing the error of the cost function and minimizing the norm of the control filter. Other methods for determining the regularization parameter can also be implemented.

[0099] According to one alternative embodiment, a maximum limit is imposed on the filter amplitude when selecting the regularization parameters.

[0100] Figure 5 shows a schematic representation of a room or hall, with four loudspeakers 501a-501d similar to those of Figure 1 or 2 positioned along the walls to broadcast into the room. Measurement locations are shown for illustrative purposes in three columns of eight positions (reference numeral 502).

[0101] 6 illustrates a method for determining control filters for one or more sound sources according to one embodiment. First, the reflected sound pressure impulse response of each sound source is determined for multiple control microphone positions in a room (S601). To do this, the sound source is placed at the position that will be used later. The sound source broadcasts an excitation signal, and the response to this sound source is measured and recorded for multiple microphone positions. This process is repeated for all sound sources.

[0102] In general, the signal measurements required to determine the impulse response can be performed using a single microphone that is moved to a different position in the room for each measurement, or using several microphones in parallel to obtain multiple measurements simultaneously for several positions in the room.

[0103] The filters are then determined (S602) by solving a regularized optimization problem based on the set of impulse responses, formulated in the time domain, where the optimization problem is defined to (a) minimize the norm of the sum of the impulse responses of the reflected sound pressure at N locations, and (b) introduce a non-zero action delay for each of the filters, where the delay allows reducing the effect of the control signal on the direct field.

[0104] According to one embodiment, the control signal path can be disconnected. For this purpose, device 100 shown in FIG. 1 comprises an element 107 to which a control signal 108 is applied. This element is, for example, a switch that connects or disconnects the control signal from the input of adder 104. Under the control of the control signal, device 100 operates only as a primary sound source (switch open) or as both a primary and secondary sound source (switch closed). This makes it easy to configure device 100 as part of a multi-speaker system, where flexibility of functionality is required to adapt to specific needs. This also allows the user to easily disconnect the correction.

[0105] According to another alternative embodiment, the effect of the control signal can be attenuated. , a fader that can vary the level of a control signal between 0% and 100%. The level of attenuation, for example, is controlled via a user interface, allowing the control to be adjusted to suit the user's preferences.

[0106] According to another embodiment, which can be combined with the previously described embodiments, the reference signal path can be disconnected, in which case the output signal of the block is the control signal only.

[0107] advantage One or more of the above-described embodiments have one or more of the following advantages. Since the sound source acts as both a primary and secondary sound source, according to the example shown, there is no need for an additional dedicated loudspeaker just to control low frequency resonances, resulting in a significant reduction in cost of ownership. The fact that each sound source can be both a primary and a secondary source allows for flexibility in adapting to a wide range of situations. Thus, for a given physical configuration of sound sources, it is possible to determine that the primary sources are centered and that all available sound sources (including the primary source and any other available sound sources) are secondary sources, relative to the temporal quality of the direct field. Alternatively, all sound sources can be used as both primary and secondary sources for greater directional control in the direct field without the need to add or reposition sources. Furthermore, it avoids the need for optimal trial and error placement of secondary sources, which can be a long and difficult process. The calculation of the individual filters for each source is performed in a single operation by solving an optimization problem that includes all measured impulse responses, in other words, all sources and measurements are considered simultaneously in the optimization problem. Using only the same sound source to reproduce the reference music signal and the control signal also improves the sense of sound source localization. Indeed, if secondary sound sources that are spatially different from the primary sound source are used, these, in addition to the control signal, may emit spurious sounds that may be due to the nonlinearity of the loudspeakers or may vibrate decorative elements located in their vicinity (e.g., false ceilings). This noise may be particularly noticeable, as it arrives from a different direction and at a different time than the speaker that reproduces the music signal. With the approach outlined in this description, the unwanted noise is generally masked by the music signal, since it is emitted in the same area as the music signal.

[0108] Furthermore, the reference music signal reproduced by the loudspeakers is not modified, and therefore the direct field perceived by the listener is not altered, preserving impact, precision and timbre. Resonance control is also significantly improved: by acting on the physical causes of resonance, i.e. reflections, after the passage of the direct field, there is no need to compromise between modifying the direct field and controlling resonance, as would be the case if you were to try to act on modes at a specific location in the room, for example by modifying the frequency response through equalization at the source.

[0109] References 1. A. Celestinos and S. B. Nielsen, “Controlled acoustic bass system (CABS) - A method to achieve uniform sound field distribution at low frequencies in rectangular rooms,” J. Audio Eng. Soc. 56(11), 915 - 931(2008) 2. Heuchel et al., “Active room compensation for sound reinforcement using sound field separation techniques,” The Journal of the Acoustical Society of America 143, 1346(2018) 3. Christian Hansen and Dianne Prost O’Leary, “The Use of the L-Curve in the Regularization of Discrete Ill-Posed Problems” by SIAM Journal on Scientific Computing 1993 14:6, 1487 - 1503

Claims

1. 1. A computer-implemented method for obtaining M filters for a digital audio signal defined in the time domain, where M is an integer greater than or equal to 1, each filter associated with a respective sound source, called a secondary sound source, emitting a sound pressure field that allows minimizing the reflected sound pressure field of a set of sound sources, called primary sound sources, located in a room, said method comprising: - obtaining (S601) for each primary and secondary sound source an impulse response of the reflected sound pressure at N respective distinct positions in the room, wherein all of the primary and secondary sound sources are placed at operational positions and the N positions are the same for all of the sound sources; - determining (S602) the set of M filters by solving a regularized optimization problem formulated in the time domain based on the set of impulse responses, the optimization problem being: (a) minimizing the norm of the sum of the impulse responses at the N positions; and (b) defined to introduce a non-zero action delay for each filter; A computer-implemented method in which each secondary sound source is paired with a primary sound source, known as an associated primary sound source.

2. 2. The method of claim 1, wherein M is 2 or greater.

3. Obtaining the impulse response of the reflected sound pressure for a given secondary sound source combined with its associated primary sound source at a given one of N locations comprises: - emitting, by said given secondary sound source in combination with its associated primary sound source, sound waves in response to an excitation signal; - obtaining a signal representative of the sound pressure resulting from the excitation signal at said location; - determining the impulse response of the reflected sound pressure based on the acquired signal representative of the sound pressure.

4. 4. The method of claim 3, wherein the signal representing the sound pressure is acquired using a microphone positioned at the given location, and wherein the determination of the impulse response comprises applying time windowing to the signal representing the sound pressure to suppress the direct sound wave received from the given secondary sound source combined with its associated primary sound source while retaining the reflected sound waves.

5. - the signal representative of the sound pressure is obtained using a pair of microphones arranged around the given position, - determining the impulse response by determining the pressure and velocity of the sound waves in order to separate the direct sound waves received from the sound source from the reflected sound waves; The method of claim 3.

6. The method according to any one of claims 1 to 5, wherein the action delay is substantially equal to the mean propagation time of sound waves generated by the sound source between the sound source and the wall of the room.

7. The method of any one of claims 1 to 6, comprising determining a regularization parameter for regularizing the optimization problem, said determining taking into account a maximum amplitude threshold of the filter.

8. A data processing device (700) comprising means for implementing the method according to any one of claims 1 to 7.

9. An audio signal processing device (100), comprising: an input (101) adapted to receive a first audio signal (x(t)); a first filter (106) for filtering the first signal to obtain a second audio signal, the first filter being a finite impulse response filter obtained by applying a method according to any one of claims 1 to 7; an adder (104) for adding the first audio signal and the second audio signal to obtain a third audio signal for controlling the sound source associated with the first filter.

10. 10. The device of claim 9, comprising a low-pass filter (105) for filtering the first audio signal, the output of the low-pass filter being connected to the input of the first filter.

11. a downsampling circuit for downsampling the audio signal after the low-pass filter (105) and before feeding it to the first filter (106); and an oversampling circuit for oversampling the audio signal after filtering by the first impulse response filter and before being supplied to the adder; The device of claim 10, comprising:

12. an adjustable attenuator (107) for applying a gain between 0 and 100% to said second audio signal; and a switch (107) configured to connect or disconnect said second signal from said adder input; The device according to any one of claims 9 to 11, comprising one of:

13. 1. A method of audio signal processing implemented by a device including a processor, a memory, and software code, the method comprising: receiving a first audio signal (x(t)); - filtering the audio signal using a finite impulse response filter obtained by applying the method according to any one of claims 1 to 7; - adding said first audio signal and said signal filtered by said finite impulse response filter to form a signal suitable for feeding said sound source associated with said finite impulse response filter.

14. 14. The method of claim 13, comprising low-pass filtering the first audio signal prior to filtering with the impulse response filter.

15. 15. The method of claim 14, comprising downsampling the audio signal after low-pass filtering and before impulse response filtering, and oversampling the audio signal after impulse response filtering and before summing.

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