Method for transforming the characteristics of an audio signal and related apparatus

Abstract

A method and associated device for transforming multiple characteristics of an audio signal is disclosed. The present invention relates to a method and associated device for combined transformation of multiple characteristics of an audio signal. The transformation allows the signal to be typified based on a profile selected by a control unit. The method and device according to the present invention are particularly intended for the field of loudspeakers.

Description

[Technical Field]

[0001] The present invention relates to a method and related device for transforming in a combined manner several characteristics of an audio signal for a loudspeaker, the device comprising a processor and an amplifier for all or part of the band, the processor being connected to a control module allowing the selection of a transformation mode of the signal characteristics. [Background technology]

[0002] Loudspeakers generally refer to all types of electrical and mechanical-acoustic transducers.

[0003] From DE 10 2004 013 14 A1 an acoustic loudspeaker system with digital signal processing is known.

[0004] The system utilizes a sensor to compare an output signal to an input signal, which is used to make corrections so that the output signal conforms to the input signal.

[0005] This equalizer device allows the modification of the gain (dB) of a signal in a specific frequency band by a coefficient adapted to the respective bandwidth of the loudspeaker to be compensated.

[0006] The main drawback of this device is that it only works with respect to the gain parameter (dB). This correction allows to achieve linearity in the gain / frequency ratio, but remains unsatisfactory in relation to all other parameters that characterize the complex structure of the signal, such as phase and time. In fact, the nonlinearities of phase and time do not allow a faithful reproduction of the original.

[0007] From patent document 2, a frequency response correction device for loudspeakers is known, and from patent document 3, a related method is known. These allow the modification of the gain (dB) and phase of the signal over the entire frequency spectrum. A digital auto-adaptive system intervenes at each frequency to linearize the amplitude / frequency curve and the phase / frequency curve. The device continuously corrects the signal with the aid of sensors.

[0008] The main drawback of continuous correction is the processing delay, and as a result it does not work for signals whose play time is less than the processing time.

[0009] Additionally, spurious signals such as noise in the room can interfere with the process.

[0010] From US Pat. No. 5,629,999 an analog signal processing device is known which corrects harmonic and phase inaccuracies generated by transduction, recording and live playback of audio signals.

[0011] Corrections are applied automatically and continuously to restore realism to the reproduced audio signal.

[0012] Permanent and constant correction does not allow adaptation to the type of music being listened to, which requires different processing.

[0013] From patent document 5, a method is known which allows access to a content stream distributed to a playback device and identifies the content for which a determined profile allows provision.

[0014] Depending on the identified profile, the method allows for modification of equalization parameters related to the playback of the content stream.

[0015] This method allows adaptation of the equalization in relation to the information available in the audio support, identified at the time of playback, linked to the user's profile or according to user settings.

[0016] The main drawback of this method is that it only provides an equalization correction, i.e. a correction of the gain (expressed in dB) as a function of frequency, which remains unsatisfactory in relation to all the other parameters characterizing the complex structure of the signal, such as phase and time. [Prior art documents] [Patent documents]

[0017] [Patent Document 1] U.S. Patent No. 6,697,492 [Patent Document 2] Patent No. 2571091 [Patent Document 3] Patent No. 2530474 [Patent Document 4] Canadian Patent No. 2098319 [Patent Document 5] US Patent Application Publication No. 2015 / 073574 Summary of the Invention

[0018] The present invention therefore aims to remedy these drawbacks. More specifically, it aims to gain, phase, time, distorted, Bandwidth, bandwidth distribution by loudspeakers, Dynamics compression / expansion, Directionality, sampling, the absolute phase corresponding to the electrical polarity of the loudspeaker group in the impulse response, A shift in the reference point where all frequencies are in phase, The present invention aims to provide a method and related apparatus that allows for the modification of all characteristics of complex structures of signals such as:

[0019] The combination of these modifications allows for the typification, compensation or improvement of the sound in a precise and instantaneous manner as a function of the normal profile.

[0020] Typification generally refers to the attribution of particular characteristics to an audio signal.

[0021] The method allows the transformation of several properties of an audio signal in a combined way and is divided into a series of actions that can be performed in one or several stages.

[0022] The first action is to generate corrections aimed at linearizing the output signal by taking into account the imperfections inherent in the loudspeaker's components and architecture. By loudspeaker we mean a grouping of one or more loudspeakers installed in a closed or open structure.

[0023] Then, depending on the determined profile, the second action is to apply a modification relating to the overall signal characteristics.

[0024] These two signal transformation actions can be performed in a single step, allowing for straightforward application of all selected transformations.

[0025] These modifications can also be applied in several steps, which allows for the separation of corrective actions to neutralize the signal from modification actions to add classification, compensation, or improvement, thus making it relatively easier to control each of the actions, while allowing for the standardization of modification formulas, since they are applied in the neutral state of the signal.

[0026] The present invention relates to a method for transforming in a combined way several characteristics of an audio signal intended for a loudspeaker, the method comprising the following actions:

[0027] The first correction action is to measure the output signal of one or more loudspeakers to determine the defects to be corrected as a function of a reference template, and then generate a correction formula. This correction formula is then applied to linearize all characteristics such as gain, phase, time equalization, and distortion minimization. The corrections applied in this way may therefore vary depending on the loudspeaker used.

[0028] The second action consists of modifying the previously obtained neutral signal to adapt it to a given profile. The modification can be performed through one or more criteria such as gain, phase, time, distortion, bandwidth, bandwidth distortion / loudspeaker, dynamic range compression / expansion, directivity, sampling, a reference phase corresponding to the polarity of the group of loudspeakers along with the impulse response, and a displacement of the reference point where all frequencies are in phase.

[0029] According to advantageous, but not essential, aspects of the invention, such a method may comprise one or more of the following features included in any technically acceptable combination:

[0030] The control module can be manually operated by a user.

[0031] The control module can automatically adjust by selecting a normal profile based on the music style information contained on the music track.

[0032] The control module can recognize the signal and automatically adapt based on information contained in the remote service that identifies the normal profile. The control module can automatically adapt according to the user preferences identified by the device.

[0033] The control module can automatically adapt according to information received from sensors present within the device or at a remote site that measure weather conditions such as air temperature, atmospheric pressure, or humidity.

[0034] The invention also relates to a related device for transforming in a combined manner several characteristics of an audio signal for a loudspeaker having a signal transformation module for all or part of the frequency band, the transformation module being connected to a control module for selecting a transformation mode of the signal characteristics depending on the determined profile.

[0035] According to advantageous, but not essential, aspects of the invention, such an apparatus may comprise one or more of the following features included in any technically permissible combination:

[0036] The conversion of the signal can be achieved according to digital methods using a processor.

[0037] The conversion of the signal can be achieved according to analog methods using electrical and / or electronic components.

[0038] The transformation of the signal can be achieved by one or more mechanical means using tuned structures, acoustic lenses, and / or transformation of the geometric properties of the device.

[0039] Further features and advantages of the present invention will become apparent from the following detailed description, the understanding of which is based on the accompanying drawings. [Brief explanation of the drawings]

[0040] [Figure 1] FIG. 1 is a schematic diagram of an apparatus according to the present invention. [Figure 2] FIG. 2 shows the steps of a general signal conversion method. [Figure 3] FIG. 3 illustrates the transformation of the frequency response of an audio signal using the method of FIG. [Figure 4] FIG. 4 illustrates the transformation of the phase characteristics of an audio signal using the method of FIG. [Figure 5] FIG. 5 illustrates the transformation of the time characteristics of an audio signal using the method of FIG. [Figure 6] FIG. 6 illustrates the transformation of the bandwidth characteristics of an audio signal using the method of FIG. [Figure 7] FIG. 7 illustrates the transformation of the compression / expansion characteristics of an audio signal using the method of FIG. [Figure 8] FIG. 8 shows the transformation of the distortion characteristics of an audio signal using the method of FIG. [Figure 9] FIG. 9 illustrates the transformation of the directional characteristics of an audio signal using the method of FIG. [Figure 10] FIG. 10 illustrates the conversion of the sampling characteristics of an audio signal using the method of FIG. [Figure 11] FIG. 11 shows the transformation of the absolute phase characteristic of an audio signal using the method of FIG. [Figure 12] FIG. 12 shows the transformation of the reference point characteristics of all frequencies of an audio signal using the method of FIG. [Figure 13] FIG. 13 illustrates the transformation of an audio signal with a change in cutoff frequency using the method of FIG. DETAILED DESCRIPTION OF THE INVENTION

[0041] With reference to Figure 1, the device according to the invention comprises a processor 1, such as a digital or analog signal processor 1 (for example in the form of a separate filter), receiving an audio signal, which may be analog or digital, in a wired or wireless manner, for at least one frequency band. In Figure 1, this acquired audio signal is marked IN.

[0042] 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 module. The power of this signal is amplified in an analog or digital manner by an amplifier 2. In the case of analog-to-digital domain conversion, a converter, not shown in the figure, must be added to convert the signal from an analog signal to a digital signal.

[0043] This electrical signal is finally converted into an acoustic signal by an electro-acoustic transducer, also called a mechano-acoustic transducer, such as a loudspeaker 3 .

[0044] As in the example of Figure 1, according to an example implementation, the device may include a signal processing chain including such a processor 1, such an amplifier 2, and such a transducer 3 for each frequency band B1, Bn.

[0045] It will therefore be appreciated that in this case the device includes a dedicated processor 1, amplifier 2 and transducer 3 for each frequency band B1, Bn.

[0046] Alternatively, the device includes a common processor 1, amplifier 2 and transducer 3 for all frequency bands.

[0047] The device is completed by a control module 4, also called a mode decoder, for automatically or manually selecting and obtaining or overriding the modification of the signals applied to the device. The user selection can be carried out, for example, through a selection module 7 with a man-machine interface.

[0048] In automatic mode, the device can receive a profile from a remote service 5 such as Gracenoto® or Shazam® or any equivalent service, see US Patent Application Publication No. 2015 / 073574, or can select a profile using an internal database or through a recognition system due to artificial intelligence.

[0049] Optionally, the device may be completed by a mechanical or acoustic system 6 for modifying the physical properties of the device. This modification system 6 may be realized, for example, by modifying the volume of an acoustic load, by applying an acoustic lens made up of one or more deflectors, or by modifying the properties of a resonator, or by any equivalent means.

[0050] Generally, the system 6 may include a mechanical-acoustic processor 6-1 and a mechanical-acoustic actuator 6-2.

[0051] In general, the device according to the invention allows the combined transformation of several characteristics of the audio signal, chosen in a non-limiting way from the following characteristics: -gain, -phase, -time, -distorted, -bandwidth, -Bandwidth distribution / loudspeaker, - Dynamics compression / expansion, -directional, -sampling, - the absolute phase corresponding to the electrical polarity (connection polarity) of the loudspeaker group in the impulse response, - Displacement of the reference point where all frequencies are in phase

[0052] The combination of several of these modifications of the characteristics of the audio signal makes it possible to typify, compensate or improve the corresponding sound precisely and instantly according to a typical profile. By "typifying" we mean giving a particular characteristic to the audio signal.

[0053] The flow diagram of FIG. 2 illustrates a general method for transforming a signal incorporating corrective and other modifying actions according to one embodiment of the present invention.

[0054] For example, the execution of the steps of the conversion method is controlled by a control module 4 of the device according to the invention.

[0055] The method begins in step 100 by measuring the loudspeaker output signal. This measurement can be performed in a laboratory at the time of device design with the aid of a system consisting of a generator, a microphone, and a signal processing system connected to a computer, where the computer is running information acquisition and processing software.

[0056] The defects to be corrected are then defined in step 102 by analyzing the differences between the input signal and a reference template, which represents ideal curves of relevant characteristics such as gain, phase, time, and distortion.

[0057] Based on this analysis and the selected criteria, a correction formula is then generated in step 104. Depending on the type of processing selected, this may include the application of an algorithm for digital processing, an analog processing scheme made up of a set of electrical and / or electronic components, or an algorithm to control the mechanical system 6.

[0058] The system then applies, in step 106, correction formulas to linearize all characteristics of the signal in order to restore its original neutrality. Depending on the type of processing selected, the formulas 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 a mechanical system 6 capable of transforming the geometric characteristics of the device.

[0059] Once the signal has been rendered in a linear state, modification formulas are applied in step 108 to categorize the characteristics according to a selected profile. These formulas have been previously generated by feedback depending on the respective profile desired, e.g., type of music, type of sound recording, type of reproduction or ambience, etc. These formulas have been selected, e.g., after a previous acquisition of a profile according to a profile selected by the user in manual mode (step 110), 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).

[0060] Then, in step 114, the power of this signal is amplified by one or more of the amplifiers 2 in an analog or digital manner.

[0061] Finally, in step 116, this electrical signal is converted into an acoustic signal by the loudspeaker 3 or by any equivalent transducer.

[0062] Optionally, the control module 4 adjusts automatically as a function of information received by sensors present within the device or at a remote site that measure weather conditions such as air temperature, atmospheric pressure or humidity.

[0063] FIG. 3 shows, for a measured audio signal given as an example, a curve illustrating the transformation of the signal amplitude curve (vertical axis) as a function of frequency (horizontal axis) at various stages of this transformation.

[0064] Inset (a) of Figure 3 shows an example of the signal measured in step 100 above. For example, this signal is not ideal due to the inherent characteristics of the equipment components. Even with state-of-the-art technology, all speakers distort the signals they process.

[0065] Inset (b) of Figure 3 shows this same corrected curve, for example, after applying step 106, which is defined by the goal of flattening all amplitudes as equally as possible as a function of frequency. In the case of analog processing, the correction would be applied by a function such as a filter, for example, a tank circuit. In the case of digital processing, the correction would be applied by a digital signal processor, such as a DSP, which would correct the gain of the signal for each frequency being processed. In the case of mechanical processing, tuned structures such as cavities, resonators, baffles, and / or absorbers would be used.

[0066] Inset (c) of Figure 3 is an example of a modified curve after applying step 108. This amplitude modification map originates from feedback in the world of sound recording or reproduction. In the case of analog processing, the modification would be applied by a function such as a filter, e.g., a tank circuit. In the case of digital processing, the correction would be applied by a digital signal processor, e.g., a DSP, which would correct the gain of the signal for each frequency being processed. In the case of mechanical processing, tuned structures such as cavities, resonators, baffles, and / or absorbers would be used.

[0067] FIG. 4 shows a curve representing the signal of FIG. 3, thereby illustrating the transformation steps of the phase curve (vertical axis) of this signal as a function of frequency (horizontal axis) at the various steps of the transformation described above.

[0068] Inset (a) of Figure 4 shows the signal measured in step 100. Again, this signal is not ideal due to the inherent characteristics of the equipment components: even with state-of-the-art technology, all speakers distort the signals they process.

[0069] Inset (b) of Figure 4 shows this same curve after step 106, after correction, which is defined with the goal of flattening all phases as equally as possible as a function of frequency. In the case of analog processing, the correction would be applied by a function such as a filter, e.g., a phase circuit. In the case of digital processing, the correction would be applied by a digital signal processor, e.g., a DSP, which would correct the phase of the signal for each frequency being processed. In the case of mechanical processing, tuned structures such as cavities, resonators, baffles, and / or absorbers would be used.

[0070] Inset (c) of Figure 4 is an example of a modified curve after step 108. This phase modification map is defined to approximate the phase variations of the studio or playback speakers. In the case of analog processing, the modification would be applied by a function such as a filter, e.g., a phase circuit. In the case of digital processing, the correction would be applied by a digital signal processor, e.g., a DSP, which would correct the phase of the signal for each frequency being processed. In the case of mechanical processing, tuned structures such as cavities, resonators, deflectors, and / or absorbers would be used.

[0071] FIG. 5 shows, for a measured audio signal given as an example, a curve illustrating the transformation of the signal's time curve (vertical axis) as a function of frequency (horizontal axis) at various steps of this transformation.

[0072] Inset (a) of Figure 5 shows an example of the signal measured in step 100 above. For example, this signal is not ideal due to the inherent characteristics of the device components. Even with state-of-the-art technology, all transducers distort the signals they process.

[0073] Inset (b) of Figure 5 shows this same corrected curve after applying step 106, for example. This is defined by the goal of flattening time as a function of frequency as evenly as possible. In the case of analog processing, the correction would be applied by a function such as a filter, for example a phase circuit with its variation with time. In the case of digital processing, the correction would be applied by a digital signal processor such as a DSP, which would correct the time of the signal for each frequency being processed. In the case of mechanical processing, physical shifting of the loudspeaker in space and possibly tuned structures such as cavities, resonators, baffles, and / or absorbers would be used.

[0074] Inset (c) of Figure 5 is an example of a modified curve after applying step 108. This modification map is defined to approximate the time variation of the studio or playback speakers. In the case of analog processing, the modification would be applied by a function such as a filter, e.g., a phase circuit. In the case of digital processing, the modification would be applied by a digital signal processor, e.g., a DSP, which would time-correct the signal for each frequency being processed.

[0075] More precisely, the purpose of the processing is to correct the time for each of the bands in the frequency decomposition (or analysis) of the signal.

[0076] In the case of mechanical treatment, physical shifting of the loudspeaker in space and possibly tuned structures such as cavities, resonators, baffles and / or absorbers would be used.

[0077] Figure 6 shows a frequency response signal curve of an audio signal given as an example to illustrate bandwidth curve conversion using the method of Figure 2. The solid line represents the first response signal corresponding to the frequency response normally provided by a transducer due to its inherent performance.

[0078] In contrast, the dotted lines represent two modified signals corresponding to shortened or lengthened response curves, respectively.

[0079] On the other hand, this curve can be shortened (narrowed) at bass and treble levels to protect the loudspeaker and limit mechanical distortions that contaminate the rest of the spectrum. In the case of analog processing, the bandwidth shortening will be applied by a function such as a filter, e.g., a high-pass and / or low-pass circuit. In the case of digital processing, the correction will be applied by a digital signal processor, e.g., a DSP, that executes high-pass and / or low-pass filtering algorithms. In the case of mechanical processing, tuned structures such as cavities, resonators, acoustic shorts, and / or absorbers will be used.

[0080] On the other hand, this curve can be extended (widened) as much as possible to improve the recovery of the sound signal. In the case of analog processing, the bandwidth extension will be applied by a function such as a resonant circuit. In the case of digital processing, the correction will be applied by a digital signal processor such as a DSP running a filtering algorithm with gain. In the case of mechanical processing, tuned structures such as cavities, resonators and / or acoustic horns will be used.

[0081] Figure 7 shows, by way of example, schematic curves illustrating the transformation of the compression or expansion characteristics of a given signal using the method shown in Figure 2. In these curves, the output signal OUT (vertical axis) is represented as a function of the input signal IN (horizontal axis).

[0082] The inset (a) of Figure 7 shows the compression curve obtained after compressing the measured signal. In compression mode, the amplification ratio of the circuit under consideration decreases as a function of the increasing input signal until it becomes negative. Thus, there is a very pronounced level control effect. In the case of analog processing, signal compression is applied by a function such as a compressor circuit, like an amplifier with a variable gain depending on the input level. In the case of digital processing, signal compression is applied by a digital signal processor, such as a DSP, which runs a compression algorithm.

[0083] The inset (b) of Figure 7 shows the expansion curve obtained after expansion of the measured signal. In expansion mode, the amplification ratio of the circuit under consideration increases as the input signal increases. This therefore has the effect of restoring the dynamics of the compressed signal to improve its snappiness. In the case of analog processing, signal expansion is applied by a function such as an expander circuit, e.g., an amplifier with a variable gain according to the input level. In the case of digital processing, signal expansion is applied by a digital signal processor, e.g., a DSP, running an expansion algorithm.

[0084] FIG. 8 shows curves illustrating the signal transformation obtained by modifying the distortion characteristics using the method of FIG. 2 for a given measured audio signal as an example.

[0085] The inset (a) of Figure 8 shows a spectral analysis consisting of a fundamental frequency F and its harmonics Hn, which induce large distortion rates. Large distortion rates represent the addition of unwanted signals not present in the original signal. These large distortion rates are primarily due to electrical and mechanical imperfections in the playback system or to the system's phase and time nonlinearities. It is also possible to increase the distortion rate of a signal to simulate imperfections not present in the original signal in order to color the sound. Coloration generally refers to imparting a specific characteristic to an audio signal. Controlled distortion makes it possible, for example, to approach the harmonic distortion characteristics of high-performance loudspeakers. In analog processing, increased distortion is achieved by adding multiple frequencies to a selected fundamental. In digital processing, increased distortion is achieved by a digital signal processor (DSP) running an algorithm to generate harmonic frequencies.

[0086] The inset (b) of Figure 8 shows a spectral analysis consisting of the fundamental and its harmonics that induce a weakened distortion rate after conversion. A small distortion rate indicates a reproduced signal that is relatively close to the original. In the case of analog processing, the weakened distortion is obtained by suppressing undesired frequencies due to filtering functions or phase and time correction. In the case of digital processing, the reduction in distortion is obtained by a digital signal processor such as a DSP that runs filtering and / or phase and time correction algorithms.

[0087] Figure 9 shows the various directions of sound from loudspeakers with different directional characteristics.

[0088] Inset (a) of Figure 9 shows an open horizontal directivity diagram that highlights the scattering of sound on wall M, thereby increasing the percentage of reflected sound interfering with the direct sound.

[0089] Insets (b) and (c) of Figure 9 show a relatively closed directivity pattern to limit reflections on wall M. Listener A will hear more of the direct sound than the reflected sound. This result is achieved by a combination of mechano-acoustic and electrical solutions, such as the addition of loudspeakers and waveguides and / or controlled time and phase variations between them.

[0090] 10 shows curves S1, S2 showing the amplitude (vertical axis) of the sampled signal as a function of time (horizontal axis), with the reference S indicating the corresponding analog signal before sampling.

[0091] The inset (a) of Figure 10 shows the curve S1 of coarse sampling in time and quantization, for example the CD standard, characterized by a 16-bit format with a sampling frequency of 44.1 kHz.

[0092] The inset (b) of Figure 10 shows the curve S2 of relatively fine sampling in time and quantization. This conversion is performed by increasing the number of bits, for example, from 16 bits to 24 bits, and by increasing the sample / time unit, for example, by changing the sampling frequency from 44.1 kHz to 192 kHz. This conversion allows for a reduction in the rate of distortion by adding signals through interpolation, which reduces the size of the increments, thus increasing listening comfort. This conversion is performed digitally by an asynchronous sample rate converter, abbreviated as ASRC.

[0093] In the method depicted in FIG. 11, absolute phase positioning is shown, which corresponds to the electrical polarity of the loudspeaker group relative to the impulse response, thereby modifying the perception of depth in the sound scene.

[0094] Inset (a) of Figure 11 shows the negative impulse response I- for the perception of the proximity of the sound (position P1).

[0095] Inset (b) of Figure 11 shows the positive impulse response I+ for increased perception of scene depth (position P2).

[0096] Switching from one to the other can be achieved by reversing the polarity of the speaker group connections.

[0097] The method depicted in FIG. 12 shows the positioning of the reference phase.

[0098] 12 shows several possible positions C1, C2, C3 of the reference phase. The reference phase is a line at 0 degrees (deg), depending on the desired position relative to a device such as a loudspeaker HP. For example, this position can be located at a large or small negative distance for the perception of increased scene depth. It can also be located at a large or small positive distance to give a sense of scene proximity.

[0099] This conversion can be performed digitally by a processor such as a DSP that recalculates the correct phase at selected distances.

[0100] FIG. 13 shows different cases of bandwidth distribution / loudspeaker corresponding to one or more cutoff frequency displacements.

[0101] Inset (a) of FIG. 13 shows the crossover frequency FC1 shifted towards bass (low frequencies) which increases the distribution rate and decreases the directivity of the device.

[0102] The inset (b) of Figure 13 shows a uniformly distributed bandwidth (cut-off frequency FC2 is essentially located in the middle of the frequency band) to balance the area of ​​use between different loudspeakers by taking into account mechanical and electrical power handling and / or directivity limitations.

[0103] The inset (c) of Figure 13 then shows the crossover frequency FC3 shifted towards the higher frequencies of the audio band to protect the loudspeakers intended to receive these frequencies, so that they receive relatively less energy, while at the same time increasing the directivity of the device.

[0104] In all three cases, the shifts in crossover frequency and slope are achieved by changing the filter type and its parameterization, both analog and digital.

[0105] In many embodiments, the control module automatically adapts the selection of the normal profile as a function of information about the specific musical style of the track. In other words, the control module is configured to automatically recognize the musical genre of the reproduced signal. As a result, the control module is able to determine the type of music being reproduced and automatically adjust its settings to suit the recording conditions and the type of work being reproduced. This description is particularly applicable to the case where the system includes two separate active multi-channel speakers (right / left).

[0106] For example, music recognition is performed by sampling the signal and then analyzing it by one or more possible means, such as online services or applications like Shazam or Gracenote® or others, and / or by detecting the music sample and comparing it with criteria stored in a remote or local database via an Internet connection. Also, determining the type of music can be performed via information contained in the music file (e.g., ID3 tag in MP3 format) or by any other determining means, such as a determining algorithm based on one or more characteristics of the music (tempo, harmonic content, etc.).

[0107] For example, the recognition method may differ depending on whether the recognition is performed in the receiver (speaker) or in the transmitter. In a wireless link, if the recognition is performed in the receiver, synchronization must exist between the receivers to avoid any mismatch in settings between them. The model to be used is preferably master / slave, where the "master" device is responsible for determining the type of music and settings to be applied and sharing the results with the "slave" devices, which apply the requested settings program, which are stored in each of them. Analysis can also be performed in the transmitter, which then assumes the "master" status. Once the music genre is identified, the control module selects a normal profile corresponding to the identified music genre. The normal profile can be a set of settings or "formulas" for one or more characteristics of the signal, and the combination of these settings modifies the loudspeaker's behavior. Thus, a single loudspeaker can behave acoustically as if it were another one designed differently or intended for different types of music. The loudspeaker may be supplied with several (e.g., four) basic settings that are predefined by the loudspeaker manufacturer and that can be updated later by the user.

[0108] In practice, the settings may include some or all of the following elements: gain, phase, time, distortion, bandwidth, bandwidth distribution / speaker, dynamics compression, directivity, absolute phase, equalization.

[0109] For example, a typical profile corresponding to a music genre called current music may have the following settings: - Gain: The gain of the signal in the high frequency channel is increased. - Phase: Phase rotations induced by the various filters are maintained (these are not to be corrected). The cutoff frequencies of the filters between the bass and treble signals are shifted so that adjustments to the phase curve are required to maintain the desired energy in the connection. - Time: The time step specific to acoustic loading and filtering is also maintained (uncorrected). Distortion: Filtering, slope or type options allow to control or limit the mechanical distortion of the loudspeaker as well as phase and time distortion. - Bandwidth: A high-pass filter cuts off signals at frequencies below 60Hz. - Bandwidth distribution / loudspeakers are selected in such a way as to produce an overlap of bass and mid-range signals at their connection frequency. For example, with a connection frequency selected at 150 Hz, the bass transducer will be cut off at frequencies above 200 Hz and the mid-range transducer will start at 100 Hz. - Compression: The difference in dynamics between peak and average amplitude is limited. -Directivity: The cutoff frequency between the midrange and treble is shifted up by one octave. - Absolute Phase: The speaker polarity is not inverted. -Equalization: At 42.5Hz, it is +2.5dB and the Q factor is 3.4. At 200Hz, it is -0.5dB and the Q factor is 2.2. At 3400Hz, it is +1.5dB and the Q factor is 0.71. At 20000Hz, it is +5.0dB and the Q factor is 0.50.

[0110] Other examples are possible.

[0111] For example, a typical profile corresponding to a music genre called Acoustic may include the following settings: Gain: The gain setting of the signal is selected so that there are no amplitude differences between the frequency bands. Phase: Phase rotations generated by the load and various filters are removed using correction (eg, by DSP). - Time: The signal processing delay is adjusted for each frequency band so that all of these signals are emitted by their respective transducers with the same overall delay. Distortion: The filtering options and other characteristics (type, slope, etc.) will make it possible to limit the rate of mechanical distortion of the transducer as much as possible and to eliminate phase and time distortions. -Bandwidth: No bandwidth limit. The distribution of frequency bands assigned to each transducer is guided by the trade-off between array directivity, distortion, and the mass of the mobile device. -Compression: No dynamic range limits are applied. -Directivity: Directivity is controlled in on and off axis states. -Sampling: Oversampling at maximum during digital processing. The polarity of the transducer is inverted so that the impulse response is positive. - Reference point: the phase and time curves are straight from the moment the signal is emitted (front of the speaker). -Equalization has been chosen to linearize the frequency response amplitude curve as much as possible.

[0112] Other examples can be considered.

[0113] The invention is in no way limited to the embodiments described and shown, and those skilled in the art will know how to modify the invention at their will. The inventions disclosed herein include the following: [Aspect 1] 1. A method for converting an audio signal (IN) for an electro-acoustic transducer, comprising: the signal is modified (106, 108) in a combined manner using a plurality of signal characteristics as a function of a nominal profile selected (110, 112) by a control module to provide the audio signal with a particular characteristic, the signal characteristics being selected from a list including gain, phase, time, distortion, bandwidth, bandwidth dispersion / speaker, dynamics compression / expansion, directivity, sampling, absolute phase corresponding to the electrical polarity of a group of loudspeakers in the impulse response, displacement of a reference point where all frequencies are in phase, and the control module automatically adapts the selection of the nominal profile as a function of information on the determined musical style of the music track. [Aspect 2] 2. The method of claim 1, wherein the transformation of the signal is performed in one or more steps consisting of at least one correction action for linearizing the signal to match recording data and a modification action for typifying the signal as a function of a selected type profile. [Aspect 3] 3. The method of any one of claims 1 to 2, wherein the conversion of the signal is performed digitally using a processor. [Aspect 4] 3. The method of any one of aspects 1 to 2, wherein the conversion of the signal is performed according to an analog method using electrical and / or electronic components. [Aspect 5] 3. The method of claim 1 or 2, wherein the transformation of the signal is performed according to one or more mechanical means using a tuned structure, an acoustic lens, and / or a transformation of a geometric property of the device. [Aspect 6] Aspect 6. The method of any one of aspects 1 to 5, wherein the control module is manually operated by a user. [Aspect 7] 7. The method of any one of aspects 1 to 6, wherein the control module is automatically adapted as a function of information contained on the remote service for recognizing the signal and for identifying a normal profile. [Aspect 8] 8. The method of any one of aspects 1 to 7, wherein the control module is automatically adapted as a function of user preferences identified by the device. [Aspect 9] A device (1, 2, 3, 4, 6) for converting audio signals for an acoustic transducer (3), comprising: the device is configured to modify (106, 108) the audio signal in a combined manner using a plurality of signal characteristics as a function of a nominal profile selected (110, 112) by a control module to impart a particular characteristic to the audio signal, the signal characteristics being selected from a list including gain, phase, time, distortion, bandwidth, bandwidth dispersion / speaker, dynamics compression / expansion, directivity, sampling, absolute phase corresponding to the electrical polarity of a group of speakers in an impulse response, displacement of a reference point where all frequencies are in phase, and the control module automatically adapts the selection of the nominal profile as a function of information on the determined musical style of the music track.

Claims

1. 1. A method for converting an audio signal (IN) for an electro-acoustic transducer, comprising: the audio 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 the audio signal with a particular characteristic, the signal characteristics including gain, phase, time, distortion, bandwidth, bandwidth dispersion per speaker, dynamics compression / expansion, directivity, sampling, absolute phase corresponding to the electrical polarity of a group of loudspeakers in an impulse response, displacement of a reference point where all frequencies are in phase, and the control module automatically adapts a selection of a type profile as a function of information of the determined musical style of the music track, the selected type profile including settings of the electro-acoustic transducer for the signal characteristics.

2. 2. The method of claim 1, wherein the transformation of the audio signal is performed in one or more steps consisting of at least one corrective action for linearizing the audio signal to match a recording and a modifying action for typifying the audio signal as a function of a selected type profile.

3. 3. The method according to claim 1, wherein the conversion of the audio signal is performed according to a digital method using a processor.

4. 3. The method according to claim 1 or 2, wherein the conversion of the audio signal is performed according to analog methods using electrical and / or electronic components.

5. 3. The method of claim 1, wherein the transformation of the audio signal is performed according to one or more mechanical means using tuned structures, acoustic lenses, and / or transformation of geometric properties of the device.

6. The method of any one of claims 1 to 5, wherein the control module is manually operated by a user.

7. 3. The method according to claim 1, wherein the control module is adapted automatically as a function of information contained on a remote service for recognizing the audio signal and for identifying a normal profile.

8. 3. The method according to claim 1 or 2, wherein the control module is adapted automatically as a function of user preferences identified by the device.

9. A device (1, 2, 3, 4, 6) for converting an audio signal for an acoustic transducer (3), comprising: the device is configured to modify (106, 108) the audio 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 impart a particular characteristic to the audio signal, the signal characteristics including gain, phase, time, distortion, bandwidth, bandwidth dispersion per speaker, dynamics compression / expansion, directivity, sampling, absolute phase corresponding to the electrical polarity of a group of speakers in an impulse response, displacement of a reference point where all frequencies are in phase, and the control module automatically adapts the selection of a type profile as a function of information of the determined musical style of the music track, the selected type profile including settings of the acoustic transducer for the signal characteristics.

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