Classifications

• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R3/00—Circuits for transducers, loudspeakers or microphones
  • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
  • H04R1/00—Details of transducers, loudspeakers or microphones
  • H04R1/20—Arrangements for obtaining desired frequency or directional characteristics
  • H04R1/22—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired frequency characteristic only 
  • H04R1/28—Transducer mountings or enclosures modified by provision of mechanical or acoustic impedances, e.g. resonator, damping means
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S2400/00—Details of stereophonic systems covered by H04S but not provided for in its groups
  • H04S2400/09—Electronic reduction of distortion of stereophonic sound systems
• • H—ELECTRICITY
  • H04—ELECTRIC COMMUNICATION TECHNIQUE
  • H04S—STEREOPHONIC SYSTEMS 
  • H04S7/00—Indicating arrangements; Control arrangements, e.g. balance control
  • H04S7/30—Control circuits for electronic adaptation of the sound field
  • H04S7/301—Automatic calibration of stereophonic sound system, e.g. with test microphone

Definitions

• TITLE Method and associated device for transforming characteristics of an audio signal
• 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 making it possible to select a mode for transforming the characteristics of the signal.
• loudspeaker we mean, in general, all types of electro and mechanical-acoustic transducers.
• 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.
• CA2098319 is known an analog signal processing device for correcting harmonic and phase inaccuracies caused by the transduction, recording and live playback of audio signals.
• Correction is automatically and continuously applied to restore the realism of the reproduced audio signal. a permanent and constant correction not allowing to adapt to the types of music listened to which require a different treatment.
• Publication US2015073574 discloses a method making it possible to access a stream of content to be distributed to a reading device and then to identify a content allowing it to be delivered a determined profile.
• This method makes it possible to adapt the equalization with respect to the information available on the audio medium, identified during playback, linked to the profile of the user or by a setting of the user.
• the present invention therefore aims to remedy these drawbacks. 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 such as:
• the method makes it possible to transform several characteristics of an audio signal in a combined manner and is available in a series of actions which can be carried out in one or more steps.
• the first action is to create a correction aiming to linearize the output signal taking into account the defects inherent in the components and in the architecture of an enclosure.
• enclosure we mean a group of one or more loudspeakers installed in a closed or open structure.
• the second action is to apply a modification which concerns all the characteristics of the signal.
• the first corrective action consists in measuring the output signal of the loudspeaker (s) in order to determine the defects to be corrected according to a reference template, then to generate the correction formula. This correction formula is then applied to linearize all characteristics such as gain, phase, time equalization and distortion minimization. The correction thus applied may therefore be different depending on the loudspeaker used.
• the second action consists in modifying the neutral signal obtained previously in order to be able to adapt it to a determined profile. The modification can be made through one or more criteria such as: gain, phase, time, distortion, bandwidth, distribution of bandwidth per speaker, compression / expansion of dynamics, directivity, sampling, the reference phase corresponding to the polarity of the speaker group to the impulse response and the displacement of the reference point where all frequencies are in phase.
• such a method may include one or more of the following characteristics, taken in any technically acceptable combination:
• the controller can be adapted automatically by selecting a typical profile according to the musical style information contained on a music title.
• the controller can automatically adapt based on user preferences identified by the device.
• such a device may include one or more of the following characteristics, taken in any technically admissible combination:
• Signal transformation can be performed digitally using a processor.
• the transformation of the signal can be carried out according to an analog method using electrical and / or electronic components.
• the transformation of the signal can be 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.
• FIG 1 is a schematic representation of the device according to the invention.
• FIG 3 illustrates the transformation of a frequency characteristic of an audio signal using the method of Figure 2
• FIG 4 illustrates the transformation of a phase characteristic of an audio signal using the method of Figure 2,
• FIG 6 illustrates the transformation of a bandwidth characteristic of an audio signal using the method of Figure 2,
• FIG 8 illustrates the transformation of a distortion characteristic of an audio signal using the method of Figure 2,
• FIG 9 illustrates the transformation of a directivity characteristic of an audio signal using the method of Figure 2,
• FIG 10 illustrates the transformation of a sampling characteristic of an audio signal using the method of Figure 2,
• Figure 12 illustrates the transformation of a reference point characteristic of all frequencies of an audio signal using the method of Figure 2,
• FIG. 13 illustrates the transformation of an audio signal comprising the modification of the cut-off frequencies by means of the method of FIG. 2.
• 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, whether wired or not, an audio signal which may be analog or digital.
• a processor 1 such as a digital or analog signal processor 1 (for example in the form of discrete filters), which receives, whether wired or not, an audio signal which may be analog or digital.
• this acquired audio signal bears the reference IN.
• This signal processor 1 can perform the processing analogically using electrical or electronic components or digitally using a processor, such as a digital signal processing processor (DSP) or a microcontroller.
• DSP digital signal processing processor
• This signal is amplified in power in an analog or digital way by an amplifier 2.
• an amplifier 2 In the case of a change of analog-digital domain, it is necessary to add a converter, not shown in the figure, to transform the signal from an analog signal to a digital signal.
• This electrical signal is finally transformed into an acoustic signal by an electro-acoustic transducer, also called a mechanical-acoustic transducer, such as speaker 3.
• an electro-acoustic transducer also called a mechanical-acoustic transducer, such as speaker 3.
• the device may include 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.
• the device comprises a processor 1, an amplifier 2 and a transducer 3 for each frequency band B1, Bn.
• 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 from through a recognition system using an internal database, or through artificial intelligence.
• a remote service for example that of Gracenote (registered trademark), or Shazam (registered trademark), or any equivalent service, with reference to publication US2015073574, or select a profile from through a recognition system using an internal database, or through artificial intelligence.
• the device can be supplemented by a mechanical or acoustic system 6 making it possible to modify the physical characteristics of the device.
• This modification system 6 can be achieved, for example, by modification of 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.
• the system 6 can include a mechanical-acoustic processor 6-1 and a mechanical-acoustic actuator 6-2.
• the device according to the invention makes it possible to transform in a combined manner several characteristics of an audio signal, chosen without limitation from the following characteristics:
• the flow diagram of Figure 2 shows the general signal transformation process integrating a corrective action and another modification according to one embodiment of the invention.
• the execution of the steps of the transformation process is controlled by the control unit 4 of the device according to the invention.
• the method begins at a step 100 by measuring the output signal from the loudspeakers. This measurement can be carried out in the laboratory when the device is being designed using a system made up of a generator, a microphone, a signal processing system connected to a computer, the latter running a information acquisition and processing software. 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.
• a correction formula is developed on the basis of this analysis and the selected criteria.
• it may include the application of an algorithm for digital processing, of an analog processing diagram made up of a set of electrical and / or electronic components, or of a control algorithm. of the mechanical system 6.
• the system then applies, in a step 106, the correction formula to linearize all the characteristics of the signal, in order to reproduce its original neutrality.
• 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.
• modification formulas are applied in step 108 to type the characteristics according to a chosen profile.
• These formulas are created beforehand by experience feedback as a function of each desired profile, for example, a type of music, a type of sound recording, a type of reproduction or atmosphere.
• These formulas are for example chosen, after the prior acquisition of a profile (step 110), as a function of the profile selected in manual mode by the user or in automatic mode by the control module 4.
• the device can receive a profile from the remote service 5 or from an internal database (step 112).
• 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.
• FIG. 3 are represented curves showing, for a measured audio signal given by way of example, the transformation of the amplitude curve of the signal (y-axis) as a function of frequency (x-axis) for different stages of this transformation.
• the insert (a) of FIG. 3 represents an example of a signal measured during step 100 described above.
• this signal is not ideal due to the intrinsic characteristics of the components of the device. In the state of the art, all the loudspeakers distort the signal which they process.
• the insert (b) of FIG. 3 represents this same corrected curve, for example after application of step 106. It is defined by the objective of leveling all the amplitudes as equally as possible as a function of the frequencies.
• the correction will be applied by functions such as filters, for example, trap circuits.
• the correction will be applied by a digital signal processor, such as a DSP, which will correct the gain of the signal for each frequency processed.
• a digital signal processor such as a DSP, which will correct the gain of the signal for each frequency processed.
• use will be made of tuned structures such as cavities, resonators, deflectors and / or absorbers.
• the insert (c) of FIG. 3 is an example of a curve modified after application of step 108.
• This amplitude modification map arises from experience feedback in the world of sound recording or sound recording. reproduction.
• the modification will be applied by functions such as filters, for example, trap circuits.
• the correction will be applied by a digital signal processor, for example, a DSP which will correct the gain of the signal for each frequency processed.
• a digital signal processor for example, a DSP which will correct the gain of the signal for each frequency processed.
• use will be made of tuned structures such as cavities, resonators, deflectors and / or absorbers.
• FIG. 4 are represented curves showing, for the signal of FIG. 3, the stages of the transformation of the phase curve of this signal (y-axis) as a function of the frequency (x-axis) at different stages of the transformation described above.
• the insert (a) of FIG. 4 represents a signal measured during step 100. Again, this signal is not ideal due to the intrinsic characteristics of the components of the device. In the state of the art, all the loudspeakers distort the signal which they process.
• the insert (b) of FIG. 4 represents this same curve corrected after step 106. It is defined by the objective of leveling all the phases as equally as possible as a function of the frequencies.
• the correction will be applied by functions such as filters, for example, phase circuits.
• the correction will be applied by a digital signal processor, such as a DSP, which will correct the phase of the signal for each frequency processed.
• a digital signal processor such as a DSP
• use will be made of tuned structures such as cavities, resonators, deflectors and / or absorbers.
• the insert (b) of Figure 4 is an example of a curve modified after step 108.
• This phase modification map is defined to approximate the phase variations of studio or reproduction speakers.
• the modification will be applied by functions such as filters, for example, phase circuits.
• 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.
• a digital signal processor for example a DSP, which will correct the phase of the signal for each processed frequency.
• use will be made of tuned structures such as cavities, resonators, deflectors and / or absorbers.
• the insert (a) of FIG. 5 represents an example of a signal measured during step 100 described above.
• this signal is not ideal due to the intrinsic characteristics of the components of the device. In the state of the art, all the transducers distort the signal that they process.
• the insert (b) of FIG. 5 represents this same corrected curve, for example after application of step 106. It is defined by the objective of leveling the time as evenly as possible as a function of the frequencies.
• the correction will be applied by functions such as filters, for example, phase circuits with their modifications over time.
• the correction will be applied by a digital signal processor, such as a DSP, which will correct the signal time for each frequency processed.
• a physical offset of the loudspeakers in space and possibly of the tuned structures such as cavities, resonators, baffles and / or absorbers will be used.
• the insert (c) of FIG. 5 is an example of a curve modified after application of step 108.
• This modification map is defined to approximate the variations in time of the studio or reproduction speakers.
• the modification will be applied by functions such as filters, eg phase circuits.
• the correction will be applied by a digital signal processor, for example, a DSP which will correct the signal time for each processed frequency.
• the purpose of the processing is to correct the time for each of the bands of the frequency decomposition (or analysis) of the signal.
• FIG. 6 represents the curve of a frequency response signal of an audio signal given by way of example, to illustrate the transformation of the curve of the passband by means of the method of FIG. 2.
• the solid line represents a first response signal, corresponding to the frequency response typically provided by the transducers by their intrinsic performance.
• the dotted lines represent two modified signals corresponding respectively to a shortened or extended response curve.
• this curve can be shortened (narrowed) at the bass and treble levels to protect the loudspeakers and limit the mechanical distortion which pollutes the rest of the spectrum.
• the shortening of the passband will be applied by functions such as filters, for example, high pass and / or low pass circuits.
• the correction will be applied by a digital signal processor, such as a DSP, performing high pass and / or low pass filter algorithms.
• a digital signal processor such as a DSP, performing high pass and / or low pass filter algorithms.
• tuning structures such as cavities, resonators, acoustic short circuits and / or absorbers.
• this curve can be lengthened (widened) as much as possible to improve the restitution of the sound signal.
• the extension of the bandwidth will be applied by functions such as resonant circuits.
• the correction will be applied by a digital signal processor, such as a DSP, performing filtering algorithms with gain.
• a digital signal processor such as a DSP
• tuned structures such as cavities, resonators and / or acoustic horns will be used.
• FIG. 7 are schematically represented curves illustrating the transformation, by means of the method of FIG. 2, of characteristics of compression or expansion of a signal given by way of example.
• the output signal OUT (ordinate axis) is represented as a function of the input signal IN (abscissa axis).
• the insert (a) in Figure 7 represents the compression curve obtained after compression of the measured signal.
• the amplification rate of the circuit considered decreases until it becomes negative as a function of the increase in the input signal. There is therefore a very pronounced level regulation effect.
• signal compression will be applied by functions such as compressor circuits, such as amplifiers with variable gain depending on the input level.
• signal compression will be applied by a digital signal processor, such as a DSP, performing compression algorithms.
• the insert (b) in Figure 7 represents the expansion curve obtained after expansion of the measured signal.
• the amplification rate of the circuit considered increases as a function of the increase in the input signal. It therefore has the effect of restoring the dynamics of the compressed signal, in order to improve ventilation.
• the expansion of the signal will be applied by functions such as expansion circuits, such as amplifiers with variable gain depending on the input level.
• the expansion of the signal will be applied by a digital signal processor, such as a DSP, performing expansion algorithms.
• the insert (a) of FIG. 8 represents the spectral analysis made up of a fundamental frequency F and its harmonics Hn evoking a high degree of distortion.
• a high rate of distortion involves adding unwanted signals that were not present in the original signal. This high rate of distortion is mainly due to electrical and mechanical faults in reproduction systems or to phase and time nonlinearity of the system. It is also possible to raise the distortion rate of the signal to simulate faults which did not exist at the origin, to color the sound. By color, we mean, in general, to give specific characteristics to the audio signal. Controlled distortion can, for example, make it possible to approximate the harmonic distortion characteristics of high efficiency loudspeakers.
• analog processing the increase in distortion will be obtained by adding multiple frequencies at the chosen fundamental.
• the increase in distortion will be achieved by a digital signal processor, such as a DSP, executing algorithms that generate harmonic frequencies.
• FIG. 9 are shown different orientations of sounds coming from loudspeakers according to different directivity characteristics.
• the insert (a) in figure 9 represents an open horizontal directivity diagram, showing the diffusion of sounds on the walls M, thus increasing the percentage of reverberated sounds interfering with the direct sounds.
• the inserts (b) and (c) of figure 9 represent more closed directivity diagrams making it possible to limit the 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, waveguide and / or the control of a variation of time and phase between them.
• curves S1, S2 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.
• the insert (a) in Figure 10 represents the curve S1 of a coarse sampling in time and in quantization.
• this is the CD standard characterized by the 16-bit format, with a sampling frequency of 44.1 kHz.
• the insert (b) of FIG. 10 represents the curve S2 of a finer sampling in time and in quantization.
• This transformation is done by increasing the number of bits, to go for example from 16 bits to 24 bits, and increasing the number of samples per unit of time, to, for example, go from a frequency of sampling from 44.1kHz to 192kHz.
• This transformation makes it possible to decrease the distortion rate by adding signals by interpolation, which reduce the size of the increments. Listening comfort is thus increased.
• This transformation is performed digitally by an asynchronous sampling converter, better known by the acronym ASRC, from the English Asynchronous Sample Rate Converter.
• the positioning of the absolute phase is shown, which corresponds to the electrical polarity of the group of loudspeakers at the impulse response, which makes it possible to modify the sensation of depth of the sound scene.
• the insert (a) in Figure 11 represents a negative impulse response I- for a perception of sound proximity (position P1).
• the insert (b) of figure 11 represents a positive impulse response I + for an increased perception of the stage depth (position P2).
• FIG. 12 illustrates several possible positions C1, C2, C3 of the reference phase.
• the reference phase is a straight line at 0 degrees, as a function of a desired position relative to the device such as a loudspeaker.
• this position can be at a more or less distant negative distance for an increased perception of scene depth. It can also be located at a more or less distant positive distance to give a feeling of proximity to the stage.
• This transformation can be carried out digitally by a processor, such as a DSP, which recalculates the right phase at the chosen distance.
• a processor such as a DSP
• FIG. 13 are shown different cases of distribution of the passband by loudspeaker, corresponding to the displacement of the cutoff frequency (s).
• the insert (a) in Figure 13 represents a cutoff frequency FC1 shifted towards the bass (low frequencies), having the effect of increasing the distortion rate and reducing the directivity of the device.
• the insert (b) of figure 13 represents a bandwidth distributed uniformly (cut-off frequency FC2 located essentially in the middle of the frequency band), to balance the zone of use between the different high- loudspeakers, taking into consideration the mechanical, electrical, admissible power and / or directivity limits.
• the insert (c) of FIG. 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.
• the shift of the cutoff frequency and the slopes is effected by changing the type of filter and its parameterization, as well in analog as in digital.
• the controller automatically adapts the selection of a typical profile based on the determined musical style information of a music title (or piece).
• the controller is configured to automatically recognize a musical genre from 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 in the case where the system includes two separate active multi-channel speakers (left / right).
• music recognition is achieved by sampling the signal and then analyzing the signal by one or more possible means, such as online services or applications, such as Shazam or Gracenote (trademarks) or the like, 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 can 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.).
• the way of recognizing can differ depending on whether the recognition is done in the receivers (speakers) or in the transmitter.
• the recognition is done in the receivers, there will have to be a synchronization between the receivers, of the result, in order to avoid any disparity of settings between the receivers.
• the model that will preferably be used will be the master / slave: the "master" device will be responsible for determining the type of music and the setting to be applied and share 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”.
• the control unit chooses a typical profile corresponding to the identified musical genre.
• the typical profile can be a collection of settings or "formulas" relating to one or more characteristics of the signal, and the combination of these settings changes the behavior of the loudspeaker.
• a single speaker can therefore behave acoustically like another designed differently or intended for another type of music.
• the loudspeakers may be delivered with a few basic settings (four for example) predefined by the loudspeaker manufacturer and subsequently updated by the user.
• the settings may relate to one or part of the following elements: gain, phase, time, distortion, passband, distribution of passband per speaker, dynamic range compression, directivity, absolute phase, equalization.
• a typical profile corresponding to a musical genre called modern music can include the following settings:
• phase rotations induced by the different filtering are kept, (they will not be corrected).
• a high pass filter cuts signals with frequencies below 60Hz
• a typical profile corresponding to a so-called acoustic musical genre can include the following settings:
• the signal gain adjustment is chosen so that there is no difference in amplitude between the frequency bands.
• Directivity is controlled on axis and off axis.
• phase and time curves are straight when the signal is transmitted (front of the speaker)
• the equalization is chosen to linearize the frequency response amplitude curve as much as possible.

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• Physics & Mathematics (AREA)
• Engineering & Computer Science (AREA)
• Acoustics & Sound (AREA)
• Signal Processing (AREA)
• Circuit For Audible Band Transducer (AREA)
• Stereophonic System (AREA)
• Tone Control, Compression And Expansion, Limiting Amplitude (AREA)

Applications Claiming Priority (2)

Publications (1)

Family

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Family Applications (1)

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• 2020
  • 2020-01-06 FR FR2000060A patent/FR3106030B1/fr active Active
• 2021
  • 2021-01-05 EP EP21700042.1A patent/EP4088487A1/de active Pending
  • 2021-01-05 JP JP2022541809A patent/JP2023509719A/ja active Pending
  • 2021-01-05 CA CA3163814A patent/CA3163814A1/fr active Pending
  • 2021-01-05 CN CN202180012642.1A patent/CN115428475A/zh active Pending
  • 2021-01-05 WO PCT/EP2021/050058 patent/WO2021140089A1/fr unknown
  • 2021-01-05 AU AU2021205599A patent/AU2021205599A1/en active Pending
  • 2021-01-05 US US17/791,192 patent/US20230069729A1/en active Pending

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