METHOD AND ASSOCIATED DEVICE FOR TRANSFORMING PROPERTIES OF AN AUDIO SIGNAL

Description

[0001] The present invention relates to a method and its associated device for transforming in a combined manner several characteristics of an audio signal intended for a loudspeaker. The device comprises, for all or part of the bands, a processor and an amplifier. The processor is connected to a control module for selecting a mode of transformation of the signal characteristics.

[0002] By loudspeaker we generally mean all types of electro and mechano-acoustic transducers.

[0003] From publication US6697492 an acoustic loudspeaker system with digital signal processing is known.

[0004] The system compares the output signal to the input signal using a sensor. This comparison is used to make a correction so that the output signal conforms to the input signal.

[0005] This equalizer device allows the modification of a signal in gain (dB) on certain frequency bands with coefficients adapted to each bandwidth of the loudspeaker to be corrected.

[0006] The main disadvantage of this device is that it only acts on the gain parameter (dB). This correction allows for linearity of the gain / frequency ratio, but remains unsatisfactory with regard to all the other parameters that characterize the complex structure of a signal, such as phase and time. Indeed, the non-linearity of phase and time does not allow for faithful reproduction of the original.

[0007] A device for correcting the frequency characteristics of a loudspeaker is known from publication JP2571091. The associated method is known from publication JP2530474. They allow the modification of a signal in gain (dB) and phase across the entire frequency spectrum. A self-adaptive digital system operates at each frequency to linearize the amplitude / frequency curve and the phase / frequency curve. This device, using a sensor, continuously corrects the signal.

[0008] The main disadvantage of continuous correction is the delay in processing and has the consequence of not working on signals whose reproduction time is less than the processing time.

[0009] Additionally, an extraneous signal, such as room noise, can interfere with processing.

[0010] From publication CA2098319 is known an analog signal processing device for correcting harmonic and phase inaccuracies caused by transduction, recording and live playback of audio signals.

[0011] Correction is automatically and continuously applied to restore the realism of the reproduced audio signal.

[0012] Constant and permanent correction does not allow adaptation to the types of music listened to which require different processing.

[0013] From publication US2015073574 is known a method for accessing a stream of content to be distributed to a reading device and then identifying content allowing a specific profile to be delivered to it.

[0014] Publication US 2019288657 deals with signal equalization, and describes automatic adaptation of equalization parameters by detecting the musical genre of the sound played, using remote communication to obtain data from the "cloud".

[0015] Depending on the identified profiles, the process allows the modification of the equalization parameters associated with the reading of the content stream.

[0016] This process allows the equalization to be adapted in relation to the information available on the audio medium, identified during playback, linked to the user profile or by a user setting.

[0017] The main disadvantage of this method is that it only offers an equalization correction, i.e. the correction of the gain (expressed in dB) as a function of the frequency. This correction remains unsatisfactory with regard to all the other parameters which characterize the complex structure of a signal such as phase and time.

[0018] The present invention therefore aims to overcome these drawbacks. The invention is as described in the independent claims. More particularly, it aims to provide a method and an associated device which make it possible to modify all the characteristics of the complex structure of a signal: Gain, Phase, Time, Distortion, Bandwidth, Loudspeaker Bandwidth Distribution, Dynamic Compression / Expansion, Directivity and Sampling.

[0019] The combination of these modifications makes it possible to type, compensate for or improve a sound precisely and instantly based on a typical profile.

[0020] By typing, we generally mean giving specific characteristics to the audio signal.

[0021] The process allows for the combined transformation of several characteristics of an audio signal and is broken down into a series of actions that can be carried out in one or more steps.

[0022] The first action is to create a correction aimed at linearizing the output signal, taking into account the defects inherent in the components and architecture of a speaker. By speaker we mean a grouping of one or more loudspeakers installed in a closed or open structure.

[0023] Then, depending on a determined profile, the second action is to apply a modification which concerns all the characteristics of the signal.

[0024] These two signal transformation actions can be performed in a single step, thus allowing all selected transformations to be applied directly.

[0025] These modifications can also be applied in several steps, thus making it possible to dissociate the corrective action, to make the signal neutral, from the modification action to add a typing, compensation, or improvement. This makes it easier to control each of the actions. On the other hand, this allows the modification formulas to be standardized, because they are applied on a signal-neutral basis.

[0026] The invention relates to a method for transforming in a combined manner several characteristics of an audio signal intended for a loudspeaker, the method comprises the following actions: The first correction action consists of measuring the output signal of the loudspeaker(s) in order to determine the defects to be corrected according to a reference template, then generating the correction formula. This correction formula is then applied to linearize all the characteristics such as gain, phase, time equalization and minimization of distortion. The correction thus applied may therefore be different depending on the loudspeaker used.

[0027] The second action consists of modifying the previously obtained neutral signal to adapt it to a specific profile. The modification can be carried out through the following criteria: gain, phase, time, distortion, bandwidth, bandwidth distribution per loudspeaker, dynamic compression / expansion, directivity and sampling.

[0028] According to advantageous, but not mandatory, aspects of the invention, such a method may comprise one or more of the following characteristics, taken in any technically admissible combination: The control member may be operated manually by the user.

[0029] The controller can be automatically adapted by selecting a typical profile based on the musical style information contained in a music track.

[0030] The control unit can automatically adapt based on information contained on a remote service enabling the signal to be recognized and a typical profile to be identified.

[0031] The controller can automatically adapt based on user preferences identified by the device.

[0032] The controller can automatically adapt based on information received from sensors, present in the device or at a remote site, measuring climatic conditions such as air temperature, atmospheric pressure or humidity.

[0033] The present invention also relates to an associated device for transforming in a combined manner several characteristics of an audio signal intended for a loudspeaker comprising for all the bands, or a part, a signal transformation module. The transformation module is connected to a control module making it possible to select a mode of transformation of the signal characteristics according to a determined profile.

[0034] According to advantageous, but not obligatory, aspects of the invention, such a device may comprise one or more of the following characteristics, taken in any technically admissible combination: The transformation of the signal may be carried out according to a digital method using a processor.

[0035] Signal transformation can be carried out using an analog process using electrical and / or electronic components.

[0036] The signal transformation can be achieved by one or more mechanical means using tuned structures, acoustic lenses and / or a transformation of the geometric characteristics of the device.

[0037] Other characteristics and advantages of the invention will appear on reading the detailed description which follows, for the understanding of which reference will be made to the attached drawings: [ Fig 1 ] There figure 1 is a schematic representation of the device according to the invention, [ Fig 2 ] There figure 2 is a representation of the steps of the general signal transformation process, [ Fig 3 ] There figure 3 illustrates the transformation of a frequency characteristic of an audio signal by means of the method of figure 2 , [ Fig 4 ] There figure 4 illustrates the transformation of a phase characteristic of an audio signal by means of the method of figure 2 , [ Fig 5 ] There figure 5 illustrates the transformation of a time characteristic of an audio signal by means of the method of figure 2 , [ Fig 6 ] There figure 6 illustrates the transformation of a bandwidth characteristic of an audio signal by means of the method of figure 2 , [ Fig 7 ] There figure 7 illustrates the transformation of a compression / expansion characteristic of an audio signal using the method of figure 2 , [ Fig 8 ] There figure 8 illustrates the transformation of a distortion characteristic of an audio signal by means of the method of figure 2 , [ Fig 9 ] There figure 9 illustrates the transformation of a directivity characteristic of an audio signal by means of the method of figure 2 , [ Fig 10 ] There figure 10 illustrates the transformation of a sampling characteristic of an audio signal by means of the method of figure 2 , [ Fig 11 ] There figure 11 illustrates the transformation of an absolute phase characteristic of an audio signal by means of the method of figure 2 , [ Fig 12 ] There figure 12 illustrates the transformation of a reference point characteristic of all frequencies of an audio signal by means of the method of figure 2 , [ Fig 13 ] there figure 13 illustrates the transformation of an audio signal involving the modification of the cut-off frequencies by means of the process of figure 2 .

[0038] In reference to the figure 1 , the device according to the invention comprises, for at least one frequency band, a processor 1, such as a digital or analog signal processor 1 (for example in the form of discrete filters), which receives, wired or not, an audio signal which may be analog or digital. On the figure 1 , this acquired audio signal carries the reference IN.

[0039] This signal processor 1 can perform the processing in an analog manner using electrical or electronic components or in a digital manner using a processor, such as a digital signal processor (DSP) or a microcontroller. This signal is amplified in power in an analog or digital manner by an amplifier 2. In the case of an analog-digital domain change, a converter, not shown in the figure, must be added to transform the signal from an analog signal to a digital signal.

[0040] This electrical signal is finally transformed into an acoustic signal by an electro-acoustic transducer, also called a mechano-acoustic transducer, such as a loudspeaker 3.

[0041] According to implementation examples, such as in the example of the figure 1 , the device may comprise a signal processing chain including such a processor 1, such an amplifier 2 and such a transducer 3 dedicated for each frequency band B1, Bn.

[0042] We thus understand that, in this case, the device comprises a processor 1, an amplifier 2 and a transducer 3 for each frequency band B1, Bn.

[0043] Alternatively, the device comprises a processor 1, an amplifier 2 and a transducer 3 common for all frequency bands.

[0044] The device is completed with a control module 4, also called a mode decoder, allowing signal modifications to be selected and applied to the device automatically, manually, or deactivated. The selection by the user can be made through a selection module 7, comprising for example a human-machine interface.

[0045] In automatic mode the device can either receive a profile from a remote service 5 for example that of Gracenote (registered trademark), or Shazam (registered trademark), or any equivalent service, with reference to publication US2015073574, or select a profile through a recognition system using an internal database, or thanks to artificial intelligence.

[0046] Optionally, the device may be supplemented by a mechanical or acoustic system 6 for modifying the physical characteristics of the device. This modification system 6 may be achieved, for example, by modifying a load volume, by applying an acoustic lens consisting of one or more deflectors, or by modifying the characteristics of a resonator, or by any equivalent means.

[0047] Generally, the system 6 may comprise a mechanical-acoustic processor 6-1 and a mechanical-acoustic actuator 6-2.

[0048] In general, the device according to the invention makes it possible to transform in a combined manner several characteristics of an audio signal, chosen in a non-limiting manner from the following characteristics: gain, phase, time, distortion, bandwidth, bandwidth distribution per loudspeaker, dynamic compression / expansion, directivity, sampling, optionally absolute phase, corresponding to the electrical polarity (connection polarity) of the loudspeaker group to the impulse response, optionally displacement of the reference point where all frequencies are in phase.

[0049] The combination of these changes in the characteristics of the audio signal makes it possible to type, compensate for or improve the corresponding sound precisely and instantly according to a typical profile. By "typing" we mean giving specific characteristics to the audio signal.

[0050] The flow diagram of the figure 2 represents the general method of signal transformation integrating a correction action and another modification action according to an embodiment of the invention.

[0051] For example, the execution of the steps of the transformation method is controlled by the control member 4 of the device according to the invention.

[0052] The method begins at step 100 by measuring the output signal of the loudspeakers. This measurement can be carried out in the laboratory at the time of designing the device using a system composed of a generator, a microphone, a signal processing system connected to a computer, the latter running information acquisition and processing software.

[0053] Then, the defects to be corrected are defined in a step 102 by analyzing the differences between the input signal and a reference template. The latter represents the ideal curve of the characteristic concerned such as gain, phase, time and distortion.

[0054] Then, in step 104, a correction formula is developed on the basis of this analysis and the criteria selected. Depending on the type of processing chosen, it may include the application of an algorithm for digital processing, an analog processing scheme composed of a set of electrical and / or electronic components, or an algorithm for controlling the mechanical system 6.

[0055] The system subsequently applies, during a step 106, the correction formula to linearize all the characteristics of the signal, in order to reproduce its original neutrality. Depending on the type of processing chosen, the formula can be applied directly by the processor 1 in the case of digital processing, by active or passive filtering in the case of analog processing or by the mechanical system 6 which can transform the geometric characteristics of the device.

[0056] Once the signal has been made linear, modification formulas are applied in step 108 to type the characteristics according to a selected profile. These formulas are created beforehand by feedback according to each profile sought, for example, a type of music, a type of sound recording, a type of reproduction or ambiance. These formulas are for example chosen, after the prior acquisition of a profile (step 110), according to the profile selected in manual mode by the user or in automatic mode by the control module 4. In automatic mode, the device can receive a profile from the remote service 5 or from an internal database (step 112).

[0057] Then, during step 115, this signal is amplified in power in an analog or digital manner by one or more of the amplifiers 2.

[0058] Finally, during a step 116, this electrical signal is transformed into an acoustic signal by a loudspeaker 3, or by any equivalent transducer.

[0059] Optionally, the control unit 4 adapts automatically according to the information received by sensors, present in the device or on a remote site, measuring climatic conditions such as air temperature, atmospheric pressure or humidity.

[0060] On the figure 3 are shown curves showing, for a given example measured audio signal, the transformation of the signal amplitude curve (y-axis) as a function of frequency (x-axis) for different stages of this transformation.

[0061] The insert (a) of the figure 3 represents an example of a signal measured during step 100 previously described. For example, this signal is not ideal due to the intrinsic characteristics of the components of the device. In the prior art, all loudspeakers distort the signal they process.

[0062] The insert (b) of the figure 3 represents this same corrected curve, for example after applying step 106. It is defined by the objective of leveling all amplitudes as equally as possible as a function of frequencies. In the case of analog processing, the correction will be applied by functions such as filters, for example, trap circuits. In the case of digital processing, the correction will be applied by a digital signal processor, such as a DSP, which will correct the signal gain for each processed frequency. In the case of mechanical processing, tuned structures such as cavities, resonators, deflectors and / or absorbers will be used.

[0063] The insert (c) of the figure 3 is an example of a modified curve after applying step 108. This amplitude modification map arises from feedback in the world of sound recording or its reproduction. In the case of analog processing, the modification will be applied by functions such as filters, for example, trap circuits. In the case of digital processing, the correction will be applied by a digital signal processor, for example, a DSP which will correct the signal gain for each processed frequency. In the case of mechanical processing, tuned structures such as cavities, resonators, deflectors and / or absorbers will be used.

[0064] In the figure 4 curves are shown showing, for the signal of the figure 3 , the stages of the transformation of the phase curve of this signal (ordinate axis) as a function of the frequency (abscissa axis) at different stages of the previously described transformation.

[0065] The insert (a) of the figure 4 represents a signal measured during step 100. Again, this signal is non-ideal due to the intrinsic characteristics of the device components. In the state of the art, all loudspeakers distort the signal they process.

[0066] The insert (b) of the figure 4 represents this same curve corrected after step 106. It is defined by the objective of leveling all phases as equally as possible as a function of frequencies. In the case of analog processing, the correction will be applied by functions such as filters, for example, phase circuits. In the case of digital processing, the correction will be applied by a digital signal processor, such as a DSP, which will correct the phase of the signal for each processed frequency. In the case of mechanical processing, tuned structures such as cavities, resonators, deflectors and / or absorbers will be used.

[0067] The insert (c) of the figure 4 is an example of a modified curve after step 108. This phase modification map is defined to approximate the phase variations of studio or reproduction speakers. In the case of analog processing, the modification will be applied by functions such as filters, for example, phase circuits. In the case of digital processing, the correction will be applied by a digital signal processor, for example a DSP, which will correct the phase of the signal for each processed frequency. In the case of mechanical processing, tuned structures such as cavities, resonators, baffles and / or absorbers will be used.

[0068] On the figure 5 are shown curves showing, for a given example measured audio signal, the transformation of the signal's time curve (y-axis) as a function of frequency (x-axis) for different stages of this transformation.

[0069] The insert (a) of the figure 5 represents an example of a signal measured during step 100 previously described. For example, this signal is not ideal due to the intrinsic characteristics of the components of the device. In the prior art, all transducers distort the signal they process.

[0070] The insert (b) of the figure 5 represents this same corrected curve, for example after applying step 106. It is defined by the objective of leveling time as equally as possible as a function of frequencies. In the case of analog processing, the correction will be applied by functions such as filters, for example, phase circuits with their modifications over time. In the case of digital processing, the correction will be applied by a digital signal processor, such as a DSP, which will correct the signal time for each processed frequency. In the case of mechanical processing, a physical offset of the loudspeakers in space and possibly tuned structures such as cavities, resonators, baffles and / or absorbers will be used.

[0071] The insert (c) of the figure 5 is an example of a modified curve after applying step 108. This modification map is defined to approximate the time variations of studio or reproduction speakers. In the case of analog processing, the modification will be applied by functions such as filters, for example, phase circuits. In the case of digital processing, the correction will be applied by a digital signal processor, for example, a DSP which will correct the signal time for each processed frequency.

[0072] More precisely, the processing aims to correct the time for each of the bands of the frequency decomposition (or analysis) of the signal.

[0073] In the case of mechanical treatment, physical offsetting of the loudspeakers in space and possibly tuned structures such as cavities, resonators, baffles and / or absorbers will be used.

[0074] There figure 6 represents the frequency response signal curve of an example audio signal, to illustrate the transformation of the bandwidth curve using the method of figure 2 The solid line represents a first response signal, corresponding to the frequency response typically provided by the transducers through their intrinsic performance.

[0075] For comparison, the dotted lines represent two modified signals corresponding respectively to a shortened or extended response curve.

[0076] On the one hand, this curve can be shortened (narrowed) in the bass and treble to protect the speakers and limit the mechanical distortion that pollutes the rest of the spectrum. In the case of analog processing, the shortening of the bandwidth will be applied by functions such as filters, for example, high-pass and / or low-pass circuits. In the case of digital processing, the correction will be applied by a digital signal processor, such as a DSP, running high-pass and / or low-pass filtering algorithms. In the case of mechanical processing, tuned structures such as cavities, resonators, acoustic short circuits and / or absorbers will be used.

[0077] On the other hand, this curve can be lengthened (widened) as much as possible to improve the reproduction of the sound signal. In the case of analog processing, the bandwidth extension will be applied by functions such as resonant circuits. In the case of digital processing, the correction will be applied by a digital signal processor, such as a DSP, executing filtering algorithms with gain. In the case of mechanical processing, tuned structures such as cavities, resonators and / or acoustic horns will be used.

[0078] On the figure 7 are schematically represented curves illustrating the transformation, by means of the process of figure 2 , compression or expansion characteristics of a given signal as an example. On these curves, the output signal OUT (y-axis) is represented as a function of the input signal IN (x-axis).

[0079] The insert (a) of the figure 7 represents the compression curve obtained after compressing the measured signal. In compression mode, the amplification ratio of the circuit considered decreases until it becomes negative as the input signal increases. There is therefore a very pronounced level regulation effect. In the case of analog processing, signal compression will be applied by functions such as compressor circuits, such as amplifiers with variable gain depending on the input level. In the case of digital processing, signal compression will be applied by a digital signal processor, such as a DSP, executing compression algorithms.

[0080] The insert (b) of the figure 7 represents the expansion curve obtained after expanding the measured signal. In expansion mode, the amplification ratio of the circuit in question increases as the input signal increases. It therefore has the effect of restoring the dynamics of the compressed signal, in order to improve its aeration. In the case of analog processing, signal expansion will be applied by functions such as expander circuits, such as amplifiers with variable gain depending on the input level. In the case of digital processing, signal expansion will be applied by a digital signal processor, such as a DSP, executing expansion algorithms.

[0081] On the figure 8 are shown curves showing, for a measured audio signal given as an example, the transformation of the signal obtained by modifying distortion characteristics, by means of the method of figure 2 .

[0082] The insert (a) of the figure 8 represents the spectral analysis consisting of a fundamental frequency F and its harmonics Hn evoking a high distortion rate. A high distortion rate implies the addition of unwanted signals not present in the original signal. This high distortion rate is mainly due to electrical and mechanical defects in reproduction systems or by phase and time nonlinearity of the system. The distortion rate of the signal can also be increased to simulate defects that did not originally exist, to color the sound. By coloring, we generally mean giving specific characteristics to the audio signal. Controlled distortion can, for example, make it possible to approach the harmonic distortion characteristics of high-efficiency loudspeakers. In the case of analog processing, the increase in distortion will be obtained by adding multiple frequencies to the chosen fundamental.In the case of digital processing, the increase in distortion will be achieved by a digital signal processor, such as a DSP, running algorithms that generate harmonic frequencies.

[0083] The insert (b) of the figure 8 represents the spectral analysis consisting of a fundamental and its harmonics evoking a weakened distortion rate after transformation. A low distortion rate implies a reproduced signal closer to the original. In the case of analog processing, the weakening of the distortion will be achieved by removing unwanted frequencies using filtering or phase and time correction functions. In the case of digital processing, the weakening of the distortion will be achieved using a digital signal processor, such as a DSP, executing filtering and / or phase and time correction algorithms.

[0084] On the figure 9 different sound orientations from HP speakers with different directivity characteristics are shown.

[0085] The insert (a) of the figure 9 represents an open horizontal directivity diagram, highlighting the diffusion of sounds on the walls M, thus increasing the percentage of reverberated sounds interfering with direct sounds.

[0086] Inserts (b) and (c) of the figure 9 represent more closed directivity diagrams to limit reverberations on the walls M. Listener A will thus be able to hear more direct sounds than reverberated sounds. This result is obtained by a combination of mechanical, acoustic and electrical solutions such as the addition of loudspeakers, waveguides and / or the control of a variation of time and phase between them.

[0087] On the figure 10 S1, S2 curves are shown showing the amplitude (y-axis) of a sampled signal as a function of time (x-axis). The reference S designates the corresponding analog signal before sampling.

[0088] The insert (a) of the figure 10 represents the S1 curve of a coarse sampling in time and quantization. For example, this is the CD standard characterized by the 16-bit format, with a sampling frequency of 44.1 kHz.

[0089] The insert (b) of the figure 10 represents the S2 curve of finer sampling in time and quantization. This transformation is done by increasing the number of bits, for example from 16 bits to 24 bits, and increasing the number of samples per unit of time, for example from a sampling frequency of 44.1 kHz to 192 kHz. This transformation makes it possible to reduce the distortion rate by adding interpolated signals, which reduce the size of the increments. Listening comfort is thus increased. This transformation is carried out digitally by an asynchronous sampling converter, better known by the acronym ASRC, from the English Asynchronous Sample Rate Converter.

[0090] In the process of the figure 11 The absolute phase positioning is shown, which corresponds to the electrical polarity of the speaker group to the impulse response, which allows the sensation of depth of the sound scene to be modified.

[0091] The insert (a) of the figure 11 represents a negative impulse response I- for a perception of proximity of the sound (position P1).

[0092] The insert (b) of the figure 11 represents a positive impulse response I+ for increased perception of scene depth (position P2).

[0093] You can switch from one to the other by reversing the polarity of the speaker group connection.

[0094] In the process of the figure 12 the positioning of the reference phase is represented.

[0095] There figure 12 illustrates several possible reference phase positions C1, C2, C3. The reference phase is a straight line at 0 degrees, depending on a desired position relative to the device such as a loudspeaker. For example, this position can be at a more or less distant negative distance for increased perception of stage depth. It can also be at a more or less distant positive distance to give a sensation of proximity to the stage.

[0096] This transformation can be performed digitally by a processor, such as a DSP, which recalculates the right phase at the chosen distance.

[0097] In the figure 13 different cases of bandwidth distribution per loudspeaker are represented, corresponding to the displacement of the cut-off frequency(ies).

[0098] The insert (a) of the figure 13 represents a cut-off frequency FC1 shifted towards the bass (low frequencies), having the effect of increasing the distortion rate and reducing the directivity of the device.

[0099] The insert (b) of the figure 13 represents a uniformly distributed bandwidth (FC2 cut-off frequency located essentially in the middle of the frequency band), to balance the area of ​​use between the different loudspeakers, taking into account mechanical, electrical, power handling and / or directivity limits.

[0100] The insert (c) of the figure 13 represents a cutoff frequency FC3 shifted towards the high frequencies of the audio band, to protect the loudspeaker intended to receive these frequencies, which then receives less energy. On the other hand, this increases the directivity of the device.

[0101] In all three cases, the shift of the cutoff frequency and slopes is achieved by changing the filter type and its settings, both analog and digital.

[0102] In many embodiments, the controller automatically adapts the selection of a typical profile based on the determined musical style information of a music title (or piece). In other words, the controller is configured to automatically recognize a musical genre of the signal being played. The controller can thus determine what type of music is being played and adjust its settings automatically, in order to adapt to the recording conditions and the type of work being played. The description is particularly applicable to the case where the system comprises two separate active multi-way speakers (left / right).

[0103] For example, music recognition is performed by sampling the signal, then analyzing the signal by one or more possible means, such as online services or applications, such as Shazam or Gracenote (registered trademarks) or other, and / or by detecting and comparing music samples with reference data stored in a remote database via an internet connection or a local database. The determination of the type of music may also be done via the information contained in the music file (ID3 tag for the MP3 format for example), or by any other means of determination, such as a determination algorithm based on one or more characteristics of the music (tempo, harmonic content, etc.).

[0104] For example, the way of proceeding with recognition may differ depending on whether the recognition is done in the receivers (the speakers) or in the transmitter. In a wireless connection, if the recognition is done in the receivers, there must be a synchronization between the receivers, of the result, in order to avoid any disparity of settings between the receivers. The model that will be used preferably will be the master / slave: the "master" device will be responsible for determining the type of music and the setting to be applied and sharing the result with the "slave" devices which will apply the requested setting program which will be stored in each of them. The analysis can also be done in the transmitter which will then take the status of "master". Once the musical genre is identified, the control unit chooses a typical profile corresponding to the musical genre identified.The typical profile can be a set of settings or "formulas" that affect one or more characteristics of the signal, and the combination of these settings changes the behavior of the loudspeaker. A single loudspeaker can therefore behave acoustically like another one designed differently or intended for a different type of music. Loudspeakers may be delivered with a few basic settings (four for example) predefined by the loudspeaker manufacturer and subsequently updated by the user.

[0105] In practice, the settings can relate to one or some of the following elements: gain, phase, time, distortion, bandwidth, bandwidth distribution per speaker, dynamic compression, directivity, absolute phase, equalization.

[0106] For example, a typical profile corresponding to a musical genre called current music may include the following settings: Gain: the gain of the signal on the high-frequency channel is increased. Phase: the phase rotations induced by the different filters are preserved (they will not be corrected). Since the filter cut-off frequencies between the bass signal and the mid-range signal are shifted, it will be necessary to adjust the phase curves in order to maintain the desired energy at the connection. Time: the time steps inherent to the acoustic loads and the filters are also preserved (no correction). Distortion: the choice of filtering, slope or type, allows to dose or limit the mechanical distortion rate of the loudspeakers as well as the phase and time distortion. Bandwidth: a high-pass filter cuts signals with frequencies below 60Hz. The distribution of the bandwidth per loudspeaker is chosen so as to cause an overlap of the bass and mid-range signals at their connection frequency.For example, for a crossover frequency chosen at 150Hz, the bass transducer will be cut off at frequencies above 200Hz and the midrange transducer will start at 100Hz. Compression: the dynamic differences between the peaks and the average amplitude will be limited. Directivity: the cutoff frequency between the midrange signal and the treble signal is moved up one octave. Absolute phase: the polarity of the loudspeakers is not reversed. Equalization: . at 42.5Hz, +2.5dB, Q factor=3.4 at 200Hz, −0.5dB, Q factor = 2.2 at 3400Hz, +1.5dB, Q factor = 0.71 at 20000Hz, +5.0dB, Q factor=0.50.

[0107] Other examples are possible.

[0108] For example, a typical profile corresponding to a so-called acoustic musical genre may include the following settings: Gain: The signal gain setting is chosen to avoid any amplitude differences between the frequency bands. Phase: Phase rotations caused by loads and different filtering processes are eliminated using corrections (e.g., DSP). Time: The signal processing delay is adjusted for each frequency band so that these signals are all emitted by their respective transducers with the same overall delay. Distortion: The choice of filtering and its characteristics (type, slope, etc.) will limit the mechanical distortion rate of the transducers as much as possible and eliminate phase and time distortions. Bandwidth: No bandwidth limit. The distribution of frequency bands allocated to each transducer is guided by the compromise between the directivity of the assembly, distortion, and the mass of the mobile equipment. Compression: No dynamic range limit is applied.Directivity: Directivity is controlled on-axis and off-axis. Sampling: Maximum oversampling during digital processing. Transducer polarity is reversed so that the impulse response is positive. Reference point: The phase and time curves are straight from the signal emission (front of the speaker). Equalization is chosen to linearize the frequency response amplitude curve as much as possible.

[0109] Other examples can be considered.

[0110] The present invention is in no way limited to the embodiments described and shown, but those skilled in the art will be able to make any variation in accordance with its spirit.

Claims

1. A method for transforming an audio signal (IN) for at least one electro-acoustic transducer, characterized in that said signal is modified (106, 108) in a combined manner using a plurality of signal characteristics as a function of a typical profile selected (110, 112) by a control module, to provide specific characteristics to the audio signal, said audio signal characteristics comprising: gain, phase, time, distortion, bandwidth, bandwidth distribution per speaker, dynamics compression / expansion, directivity and sampling, and wherein the control module automatically adapts the selection of a typical profile as a function of the information of the determined musical style of a music track.

2. The method according to claim 1, wherein the transformation of the signal is carried out in one or more steps consisting of at least one corrective action to linearize the signal in order to match the recording data, and a modification action to type the signal as a function of the selected type profile.

3. The method according to claim 1 or 2, wherein the transformation of the signal is carried out according to a digital method using a processor.

4. The method according to claim 1 or 2, wherein the transformation of the signal is carried out according to an analog method using electrical and / or electronic components.

5. The method according to claim 1 or 2, wherein the transformation of the signal is carried out according to one or more mechanical means using tuned structures, acoustic lenses and / or a transformation of the geometric characteristics of the device.

6. The method according to any of the preceding claims, wherein the control module is manually operated by the user.

7. The method according to any of the preceding claims, wherein the control module automatically adapts as a function of information contained on a remote service for recognizing the signal and identifying a typical profile.

8. The method according to any of the preceding claims, wherein the control module automatically adapts as a function of user preferences identified by the device.

9. The method according to any of the preceding claims, wherein said selected typical profile comprises settings of said at least one electro-acoustic transducer relative to said audio signal characteristics.

10. A device (1, 2, 3, 4, 6) for transforming an audio signal for at least one acoustic transducer (3), characterized in that the device is configured to modify (106, 108) said signal in a combined manner using a plurality of signal characteristics as a function of a typical profile selected (110, 112) by a control module, to give specific characteristics to the audio signal, said signal characteristics being chosen from the list comprising: gain, phase, time, distortion, bandwidth, bandwidth distribution per speaker, dynamics compression / expansion, directivity and sampling, and wherein the control module automatically adapts the selection of a typical profile as a function of the information of the determined musical style of a music track.