EP3485653A1 - Éléments de compensation pour un système de haut-parleurs - Google Patents

Éléments de compensation pour un système de haut-parleurs

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Publication number
EP3485653A1
EP3485653A1 EP17751256.3A EP17751256A EP3485653A1 EP 3485653 A1 EP3485653 A1 EP 3485653A1 EP 17751256 A EP17751256 A EP 17751256A EP 3485653 A1 EP3485653 A1 EP 3485653A1
Authority
EP
European Patent Office
Prior art keywords
signal
sound
loudspeaker
distortion
signal component
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Withdrawn
Application number
EP17751256.3A
Other languages
German (de)
English (en)
Inventor
Lorenz Betz
Maximilian Wolf
Current Assignee (The listed assignees may be inaccurate. Google has not performed a legal analysis and makes no representation or warranty as to the accuracy of the list.)
Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Original Assignee
Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Priority date (The priority date is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the date listed.)
Filing date
Publication date
Application filed by Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV filed Critical Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Publication of EP3485653A1 publication Critical patent/EP3485653A1/fr
Withdrawn legal-status Critical Current

Links

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • 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
    • H04R1/2807—Enclosures comprising vibrating or resonating arrangements
    • H04R1/283—Enclosures comprising vibrating or resonating arrangements using a passive diaphragm
    • H04R1/2834—Enclosures comprising vibrating or resonating arrangements using a passive diaphragm for loudspeaker transducers
    • 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
    • H04R1/32—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only
    • H04R1/40—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers
    • H04R1/403—Arrangements for obtaining desired frequency or directional characteristics for obtaining desired directional characteristic only by combining a number of identical transducers loud-speakers
    • 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
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/10—Applications
    • G10K2210/105—Appliances, e.g. washing machines or dishwashers
    • G10K2210/1053—Hi-fi, i.e. anything involving music, radios or loudspeakers
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
    • H04R2201/403—Linear arrays of transducers
    • 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
    • H04S2400/11—Positioning of individual sound objects, e.g. moving airplane, within a sound field

Definitions

  • Embodiments of the present invention relate to a compensation means for a loudspeaker system and a loudspeaker system, to a calculation unit and an associated method. Further embodiments relate to the control of a loudspeaker array or an array of actuators for Schallab- radiation.
  • Loudspeakers are electromechanical systems with non-linear properties.
  • the output signal eg diaphragm displacement, fast, sound pressure
  • the nonlinear behavior is expressed by nonlinear distortions.
  • Nonlinear distortions are signal components in the spectrum of the output signal that are not included in the spectrum of the input signal (eg harmonic distortion, intermodulation distortion). This creates an unwanted sound discoloration.
  • An additional signal is added to the input signal, which is compensated by the converter's own non-linear behavior.
  • the excitation signal of the loudspeaker thus consists of the input signal and a second control signal / compensation signal.
  • the added compensation signal can increase the electrical energy supplied to the loudspeaker and thus load it more heavily. As a result, the mechanical properties undergo a stronger aging process, and the thermal load is higher.
  • only those non-linearities caused by the electromechanical drive can be compensated, for example the nonlinear coupling / force factor between the electrical and mechanical side, or the effects of the nonlinear suspension (spring) of the diaphragm, but not nonlinearities other causes that only arise in the sound pressure eg Doppler distortions or nonlinearities in a resonator or high frequency horn.
  • the non-linear behavior of systems without their own drive can not be compensated with this approach z.
  • the prior art offers no approach to compensate for the distortion spectrum radiated by the passive membrane (s), since the passive membranes do not have their own drive. Therefore, there is a need for an improved approach.
  • the object of the present invention is to provide a concept for reducing the distortion signal component in the case of a sound pressure signal.
  • Embodiments of the present invention provide a speaker system compensation means comprising a first speaker group having at least one sound transducer.
  • the first loudspeaker group is designed to generate a first sound signal on the basis of an audio signal, wherein the sound signal comprises a useful signal component and a distortion signal component.
  • the distortion signal component usually results from nonlinearities in the generation of the useful signal by means of the one or more sound transducers of the first loudspeaker group.
  • the compensation means comprise at least a second group of loudspeakers having at least one sound transducer, wherein the second loudspeaker group is designed to generate a second sound signal based on a compensation signal which, when superposed with the first sound signal, contains the distortion signal component (ie the interference components or generally the unwanted components) , radiated sound components) compensated and / or reduced.
  • the compensation signal or control signal from the distortion signal component, z. B. by inversion derived.
  • Embodiments of the present invention is based on the finding that it is possible radiated sound pressure of a speaker or a group of multiple speakers, the so-called distortion signal component - ie an unwanted / unwanted radiated sound signal, such.
  • a background noise or noise - in the reproduction of a useful signal includes, can be optimized by an additional speaker or an additional group of speakers another sound signal is played, which is suitable to the distortions of the first volume Speakers or the first speaker group in the sound field to compensate in the sense of extinguish or reduce.
  • This second group of loudspeakers is controlled by a so-called compensation signal, which is dependent on the distortion spectrum of the loudspeaker or the loudspeaker group which radiates the actual audio signal.
  • a loudspeaker system which comprises the first loudspeaker group for generating the actual audio signal and the second loudspeaker group for generating the compensation signal.
  • the first and the second loudspeaker group are arranged relative to each other in accordance with exemplary embodiments in such a way that a superposition of the second sound signal relative to the first sound signal takes place in a space of the sound field generated.
  • the superposition takes place in the near field. This can be achieved, for example, by the first group of loudspeakers being separated from the second group of loudspeakers by a small distance, e.g. B. maximally 3 m or preferably at most 1 m are placed away from each other.
  • At least one sound transducer of the second loudspeaker group or the entire second loudspeaker group can be aligned with a sound field generated by the first loudspeaker group or with the sound field of the first loudspeaker group.
  • the superposition takes place in the near field, it should be mentioned that the particular advantage is that here the mode of operation of the concept is independent of the listening location, i. H. So that also in the far field with respect to the first (or on the second) speaker group an optimized listening experience takes place.
  • the particular advantage is that there is an optimization for just one listening position.
  • the loudspeaker group 2 may comprise a plurality of loud speakers and be configured to perform beamforming. In this case, it is then advantageously possible to determine the superimposition location exactly by means of the generated and aligned sound cones via the control of the loudspeakers of the second loudspeaker group.
  • a stereo sound field or a surround sound field can be generated by such a speaker system.
  • the first loudspeaker group according to this extended embodiment comprises at least two, or even more, channels. This embodiment is preferably in combination with the beam found forming approach.
  • the loudspeakers of the first loudspeaker group generate a plurality of sound fields
  • the loudspeakers of the second loudspeaker group are also formed in order to generate a plurality of sound fields.
  • a plurality of loudspeakers in the loudspeaker group 2 can be used.
  • the calculation unit also additionally comprises a signal analyzer, which is designed to analyze the first sound signal with respect to the useful signal component and the distortion signal component in order to extract the information about the distortion signal component.
  • the signal analyzer can compare the audio signal with the first sound signal.
  • the signal is usually with a pickup for the first signal, such. B. a microphone or general means for measuring the acceleration, the speed and / or the deflection of the membrane or generally the Schallabstrahl
  • the signal analyzer may also be configured to analyze the audio signal and to simulate the distortion signal component.
  • the information about the distortion signal component comprises information isolated from the useful signal. Both variants advantageously make it possible to determine the distortion signal component, with a more realistic result being obtained in the variant of analyzing the first sound signal, since the signal is analyzed among all influencing factors currently present.
  • the signal synthesizer is configured to invert the distortion signal component to obtain the compensation signal.
  • the signal synthesizer can be designed in accordance with further exemplary embodiments in order to determine the compensation signal taking into account the transfer function of the at least one sound transducer of the second loudspeaker group. This creates the advantage that any distortions of the second speaker group are already taken into account in advance.
  • Another embodiment provides the calculation unit with the signal synthesizer and optionally with the signal analyzer.
  • a method for generating a useful signal component comprises the steps of outputting the first sound signal and outputting the second sound signal, so that the second sound signal compensates or reduces the distortion signal component of the first sound signal when superposed with the first sound signal.
  • Another embodiment provides a method for calculating a compensation signal. This comprises the steps of determining a compensation signal based on information about a distortion signal component, which is included together with a useful signal component of a first sound signal that is generated by a first loudspeaker group on the basis of an audio signal. In the next step, the compensation or reduction of the distortion signal component is then performed by outputting the compensation signal component as a second sound signal in order to obtain the desired equalization / sound enhancement when superposed with the first sound signal.
  • the steps of the methods explained above or at least one or a few steps of the methods explained above can be carried out with the aid of a computer. Therefore, another embodiment provides a computer program with a program code for performing the method.
  • Fig. 1a is a schematic block diagram of a loudspeaker arrangement having a first and a second loudspeaker group according to a basic embodiment
  • Fig. 1b is a schematic flow diagram of a corresponding compensation method
  • a schematic block diagram for compensation of the non-linearity of a transducer / transducer group 1 by additional transducer / transducer group 2 according to an extended embodiment a schematic block diagram illustrating a basic configuration of a narrow band active noise control system
  • a schematic representation of simulated isobars of two antiphase converter at different distances
  • 3a, b are schematic diagrams of amplitude spectra illustrating the reduction of harmonic distortions with the aid of the compensation means according to embodiments;
  • 4a, b are schematic diagrams of amplitude spectra illustrating the reduction of the harmonic distortion by compensation means according to embodiments.
  • 1 a shows a loudspeaker system 100 with a first loudspeaker 12 and a second loudspeaker 14.
  • the first loudspeaker 12 belongs to the first loudspeaker system. group and in this embodiment comprises two sound transducers 12a and 12b, wherein the second sound transducer is optional.
  • the second loudspeaker 14 in turn comprises two sound transducers, namely the sound transducers 14a and the optional sound transducers 14b.
  • the sound transducers 14a and 14b and the loudspeaker 14 belong to the second loudspeaker group. For example, both are juxtaposed and angled in such a way that, for example, they emit the sound into a common space, which is provided with the reference numeral 16.
  • both the first loudspeaker 12 or the sound transducers 12a and 12b and the second loudspeaker 14 or sound transducers 14a and 14b forward the first or the second sound signal (including all the considered components) to the front, ie emit over the front membrane of the transducers 12a, 12b, 14a and 14b.
  • the speaker 12 emits a sound signal 12e based on an audio signal 12s.
  • the sound signal 12e comprises on the one hand a useful signal 12n and on the other hand a distortion signal 12v.
  • the loudspeaker 14 of the second loudspeaker group serves to superimpose the sound signal 12e such that the distortion signal component 12v is reduced or removed.
  • the second group of loudspeakers transmits, on the basis of a compensation signal 14, a sound signal 14e which is suitable for reducing or eliminating the distortion signal component 12v when superposed with the signal 12e, and then obtaining the undistorted signal 12e + 14e as a result, which is comparable or similar to the useful signal 12n.
  • the compensation signal 14e may, for example, be an inverse of the distortion signal 12v.
  • the control signal 14s is derived, for example, from the sound signal 12e or the audio signal 12s.
  • the control signal for the compensation signal 14e may also be determined taking into account the transmission characteristic of the compensation loudspeaker, so that additional noise is not yet generated.
  • the superimposition of the signal 14e or 12e takes place in the space 16 or, to be more precise, according to embodiments in the near field of the two speakers 12 and 14.
  • the speakers 12 and 14 with a small distance , such as B. 1 m or a maximum of 3 m spaced apart from each other.
  • the sound signals 12e and 14e can then be superimposed in the near field, so that a distortion-reduced or distortion-free signal 12e + 14e can be perceived practically at each listening position in the space 16.
  • two formulas are known with which the maximum or frequency-dependent distance of the loudspeaker groups can be determined. The background is explained below:
  • the minimization of the harmonic interference signal by the equalizer loudspeaker represents a basic basic task of ANC systems.
  • the velocity of the equalizer loudspeaker must be proportional in amplitude to the relative distance of the far field position and produce a sound pressure signal which is 180 ° out of phase with the signal of the distorted loudspeaker when it reaches the far field position.
  • the question that arises here is how large the distance d between distorted and equalizer loudspeakers may be, in order to audibly reduce the nonlinearities not only at a certain point but at every point in the far field. To solve this problem, it is first necessary to apply a far-field approximation in the above equation.
  • Equation can be simplified to:
  • a loudspeaker-dependent upper limit frequency is also known, from which aliasing effects occur.
  • the arrangement of the loudspeakers equals a spatial sampling and the upper limit frequency f Alias is given with:
  • 300Hz was chosen as the examination frequency. It is to be expected that the influence of the transducer distances will be most clearly recognizable in the case of the harmonics to be compensated up to 1500 Hz.
  • the effectiveness of the multi-position method in the room was investigated by turning the loudspeaker array using a turning device. The measurements could thus be carried out at 0 °, 45 ° and 90 °.
  • the loudspeaker array was mounted so that the respective speaker combination lay on a horizontal line.
  • the following table shows the THD values and the corresponding THD reduction at different angles and different distances. Initially, the THD reduction of the D2Z, D3E transducer combination coincides with the original measurement. The THD reduction remains stable with this converter combination on the other microphone positions; at 45 °, the harmonic distortion reduction is even significantly higher at 18.6 dB.
  • the distance criterion applies to all points in the far field, but this does not coincide with the results of the measurement.
  • the purely theoretical consideration of the complete extinction while maintaining the distance criterion is based on the reduction of the radiation impedance. If it is equal to zero, the coupling of the mem brane fast to the surrounding air is prevented and the transducers do not radiate sound pressure into the far field. This eliminates an angle-dependent consideration in the theoretical ideal case. In the real measurement, the radiation resistance is not lowered to zero and part of the sound power is still emitted into the far field.
  • Fig. 2c the simulated isobars of two transducers are shown with the above-mentioned distances.
  • the measured directivity of the two transducers was included in the simulation.
  • the excitation of the transducers was simulated so that a converter completely out of phase so inversely to the second transducer emits sound.
  • This idealized simulation thus shows the radiated sound pressure in the far field of the transducer pair with an ideal control signal on the Entzerrlaut Maschinener. Blue to orange areas show the lowering of the sound pressure of the Zer Jardiners, red areas an increase in the sound pressure. With this representation, reference can be made to the aliasing frequency in loudspeaker arrays.
  • the aliasing frequencies for 4.3 cm to 4645Hz for 8.6 cm to 2323Hz and for 12.9 cm to 1549Hz From the isobars an increase of the sound pressure from 3350Hz / 4.3cm, 2100Hz / 8.6cm and 1250Hz / 12.9cm can be seen. These values are each slightly smaller than the calculated cutoff frequencies, but the results of the equation used represent a practical guideline value.
  • the isobars show well that a pronounced interference pattern is formed above the loudspeaker-dependent aliasing frequency.
  • the geometric distance ideally has only little influence on the reduction of harmonic distortions.
  • the weak effectiveness of the real converter pairs measured at different distances can be attributed to the fact that the phase position of the overtones to be canceled was not perfectly met, or the sound field is superimposed by additional components, for example by edge reflections.
  • a loudspeaker or loudspeaker array 12 generates harmonic distortions and intermodulation distortions 12v due to the non-linear characteristics of the specific transducer principle. If one places an equalizer loudspeaker or compensating loudspeaker 14 geometrically close to the single loudspeaker or loudspeaker array 12, a calculated control signal 14e can be emitted via the compensating loudspeaker 14, which deletes the distortion products in the sound field 16.
  • the extinction of the distortion products corresponds to a lowering of the radiation resistance (real part of the complex radiation impedance Re ⁇ Z ⁇ ) of the distorted single loudspeaker or loudspeaker array 12 in the frequency range in which the distortion products lie. This reduces or eliminates the radiation of distortion artifacts into the far field.
  • the mode of action is independent of the listening location. It should be noted that it is the preferred variant in the speaker system 100 that the two speakers 12 and 14 emit the sound 12e or 14e in approximately the same direction or are screwed to each other. Although it was assumed in the above embodiments of the speaker system 100, according to other embodiments, just the compensation means for another speaker system, which includes at least the speaker 12, are created. This compensation means is essentially formed by the loudspeaker 14 or generally the second loudspeaker group having at least one sound transducer 14a.
  • FIG. 1 b shows a method 100 which comprises the two basic steps 1 10 and 120.
  • the basic step 110 relates to the outputting of the first sound signal with the aid of the first loudspeaker group on the basis of the audio signal 12s.
  • the second sound signal is then output for compensation.
  • This second sound signal is based on the interference signal 14s.
  • the interference signal 14s is dependent on the audio signal 12s.
  • the method 100 may be supplemented by the method 200, the z. B. based on the audio signal 12s preferably in combination with the first sound signal, the compensation signal 14s determined.
  • the method 200 when considered separately, comprises the step of determining the compensation signal (see step 200) based on information about the distortion signal component and the step of compensating and / or reducing the distortion signal component when the compensation signal 14s is output as the second sound signal.
  • this second step is equivalent to step 120.
  • 2a shows the two loudspeaker groups 12 and 14 which emit the sound signals 12e (starting from the audio signal 12s) and 14e (starting from the control signal 14s), so that when superimposed (see overlay function Z or reference symbols fi2e + ie ) the distortion-corrected signal 12e + 14e takes place.
  • Z represents complex radiation impedance.
  • the signal 14e reduces the radiation resistance in the frequency range of signal 12v and thus prevents the radiation of 12v in the far field.
  • the control signal 14s for driving the second loudspeaker group 14 is generated by signal synthesis means 17. There are different approaches to this, how the signal 14s is calculated.
  • the input signal u or 12s is reproduced by the loudspeaker group 1 (cf., reference numeral 12) and results in a membrane rapid signal v or 12e.
  • This signal 12e is subjected to a signal analysis, in which the useful signal 12s is separated from the interference signal 12v.
  • a signal analyzer 19 is used, which is optionally coupled to a microphone 21 or another sound receiver, so that it can receive the signal 12e of the loudspeaker group 12.
  • another type of signal recording done, for. B. by a sensor on the membrane of the speaker of the speaker group 12 or by tapping the electrical signal of the speaker group 12.
  • the extraction of the output signal from the speakers or speaker group 1 can be done in different ways .
  • Further examples are the measurement of the acceleration, the fast, the deflection of the diaphragm, an airborne or structure-borne sound measurement or an electrical measurement at the speaker terminals.
  • the result of the signal analyzer 19 is information about the noise signal 12v.
  • This information is supplied to the signal synthesizer 12v, which then processes the noise signal 12v into a control signal c or 14s, respectively.
  • the processing may include, for example, an inversion.
  • the interference signal 14s or c is then reproduced by the loudspeakers of the loudspeaker group 2 (see reference numeral 14), so that the interference component from the loudspeakers or loudspeaker group 1 (see reference numeral 12) in the sound field p or 16 is reduced or completely extinguished becomes.
  • the signal synthesis can be realized for example by an inversion of the interference signal with inclusion of the transfer function of the compensation loudspeaker or the compensation of the loudspeaker group 2 (see reference numeral 14).
  • the signal analyzer 19 is not necessary, so that the signal synthesizer 17 receives the interference signal 12 v or, in general, information about the interference signal 12 v through modulation or simulation of the loudspeaker 12.
  • the compensation means in addition to the loudspeaker 14 of the second loudspeaker group comprise the signal nalsynthetisator 17 and, alternatively to the signal analyzer 19 a signal simulator (not shown).
  • This signal simulator can, for. B. based on the signal 12s and information about the speaker group 1 (see BZ 12) simulate or predict the interference signal 12v.
  • a useful output signal diaphragm excursion, fast, sound pressure Certainly is simulated and this prediction is analyzed by its distortion component.
  • the loudspeaker group 1 may also comprise a plurality of individual loudspeakers for different channels of a stereo application or surround sound application.
  • the loudspeaker group 2 then preferably includes the correspondingly assigned loudspeakers.
  • the channels can be assigned to different loudspeakers (spatially separated units) or even several channels can be reproduced via a loudspeaker (cf. Depending on the speaker group 2 is then divided into several individual speakers or a multi-channel speaker.
  • the loudspeaker group 1 can be designed to perform beamforming.
  • the loudspeaker group 2 is designed (operated) for beamforming
  • a loudspeaker system of the loudspeaker group 1 (with a plurality of loudspeakers, for example) can likewise be operated by means of beamforming.
  • the compensation of an audio signal reproduced by means of beamforming by a single loudspeaker of the loudspeaker group 2 is also possible.
  • a beam-forming technology can also be used at the same time. The other way around: If you use beamforming technology, I can also track the compensation approach with the number of converters.
  • FIG. 2a The schematic block diagram and the signal flow diagram of this method is outlined in FIG. 2a.
  • An audio signal represented by a voltage signal, is applied to a loudspeaker.
  • the linear complex transfer function Hlin describes the transformation from voltage to membrane velocity.
  • the non-linear transfer function H (x) denotes the deflection-dependent nonlinearities of the converter.
  • the measurement of the membrane velocity is fed to a signal analysis, which can separate the interference signal - ie the distortion products - from the useful signal.
  • This interference signal continues to exist as a fast signal.
  • This fast signal can be converted back into a voltage signal.
  • This time signal is inverted and fed to a second loudspeaker, the equalizer loudspeaker.
  • This method is adjacent to a subset of Active Noise Control (ANC) Narrowband Active Noise Control.
  • Active noise control involves the electro-acoustic generation of a sound field to extinguish an existing but unwanted sound field.
  • Narrow Band Active Noise Control Systems are concerned with the regulation of periodic spurious signals often emitted by rotating mechanical components such as motors or fans. The suppression of background noise is based on the principle of superposition; a control signal with the same amplitude but opposite phase is generated by an electroacoustic or electromechanical system and its sound radiation combined with the sound field of the source of interference. This results in an extinction of both sound fields.
  • FIG. 2b the basic principle of a narrow band ANC system is shown schematically.
  • a source of interference emits a periodic harmonic signal.
  • a non-acoustic sensor records a synchronization signal which is used in a signal generator to synthesize a reference signal x (n).
  • a digital filter generates from this reference signal the control signal y (n), which is reproduced via the control loudspeaker.
  • an error microphone can be used which measures the residual sound field and feeds it as an error signal e (n) to an adaptive algorithm. This adjusts the coefficients of the digital filter.
  • an error microphone can be used which measures the residual sound field and feeds it as an error signal e (n) to an adaptive algorithm. This adjusts the coefficients of the digital filter.
  • the method is limited to single sine tones or harmonic narrowband signals
  • the signal analysis separates the interference signal from the useful signal and generates the reference signal
  • Voltage signal generates the control signal.
  • a first speaker is stimulated with a tonal signal. It is possible to adapt the amplitude and phase of the excitation signal of a second loudspeaker to extinguish the sound pressure directly in front of the diaphragm of the first loudspeaker. Likewise, the first speaker extinguishes the sound pressure of the second speaker. There is no active power, there is only a pressure equalization between the two speakers instead; an acoustic short circuit is formed. The air is pushed back and forth blindly between the two loudspeakers, which creates a local sound pressure field between the loudspeaker diaphragms, but by lowering the real part of the radiation impedance no sound pressure is radiated into the far field. The radiation impedance is frequency-dependent.
  • the interference signal is thus composed of the sum of the overtones. Is the radiation impedance only in the frequency range above the Base frequency lowered, only the coupling of the harmonics is prevented. In the ideal case, the resulting sound pressure signal now only results in a signal which is in a linear relationship to the audio signal or voltage signal.
  • FIG. 3 a shows the amplitude spectrum of a single loudspeaker with a sinusoidal excitation of, for example, 170 Hz.
  • FIG. 3 b the amplitude spectrum of the distorted loudspeaker is shown with the sine excitation (170 Hz) with a second loudspeaker, via which a compensation signal is reproduced.
  • the combination of the two loudspeakers results in a reduction of the harmonic overtones (340 Hz, 510 Hz, 680 Hz, 850 Hz), from THD 28.8% (see Fig. 3a) to THD 6.0% (see Fig. 3b ).
  • THD Total Harmonie Distorsion
  • FIG. 4 a shows an amplitude spectrum of an array of 12 individual loudspeakers with a sine excitation of, for example, 200 Hz.
  • FIG. 4 b the resulting amplitude spectrum of the distorted loudspeaker array (see BZ 12) with sinusoidal excitation (200 Hz) with an additional thirteenth loudspeaker, via which a compensation signal is reproduced.
  • the additional loudspeaker is combined with the array there is a significant reduction of harmonic overtones (400 Hz, 600 Hz, 800 Hz and 1000 Hz), from THD 10.7% to THD 3.4%.
  • equalizer loudspeaker the converter D3E was chosen.
  • the twelve distortion loudspeakers are arranged around the equalizer loudspeaker.
  • the amplifier voltage has been adjusted so that the array of twelve transducers without equalizer at 200Hz reaches a THD value of 10%. This value is achieved by halving the previous voltage per single transducer and represents a burden for the converter, which he can survive unscathed even over a long period of operation.
  • the distortion factor of the loudspeaker array can be reduced by the equalizer speaker from 10.7% to 3.4%. This corresponds to a reduction of 10 dB.
  • the maximum distance between the equalizer and the distortion loudspeakers is 7,4 cm.
  • the transducer distance has a decisive influence on the effectiveness of the driving method. If the distance between the transducers is too long, different propagation times of distorted and equalizer loudspeakers result, depending on the listening location. As a result, higher order harmonics can be reduced inferior due to their shorter wavelength compared to lower orders.
  • the THD as the sole appraisal measure, is not sufficient, since it only reproduces the sum of the overtones, but not their relationship to one another.
  • the equalizer speaker will lower the THD value, but amplify individual harmonics and degrade the sound image.
  • the spatial sound radiation of the array of twelve loudspeakers was examined with an equalizer loudspeaker in the horizontal 0 ° plane.
  • the sound pressure level of the second and third harmonics is reduced by the equalizer loudspeaker largely independent of angle by about 10 dB.
  • the fourth harmonic shows a deviating behavior. Its sound pressure level does not seem to cancel itself out by the Entzerrlaut Maschinener, but rises even slightly. Again, this behavior is largely independent of the measurement angle, suggesting that the correct phase angle necessary for extinction has not been hit accurately enough.
  • the THD at 200 Hz can be reduced from 10.7% to 3.4% by adding an equalizer loudspeaker.
  • the following is a theoretical estimate of how many speakers are needed in an array without an additional equalizer speaker in order to achieve the same sound pressure level with the same THD value.
  • the array of twelve loudspeakers plus equalization loudspeakers achieves a sound pressure level of 81, 4 dB10 at 200 Hz excitation and 3.4% THD at a distance of 1.55 m. Without equalizer, the single transducer generates a maximum of 53.4 dB SPL1 1 at the same distance and 3.4% THD.
  • FIG. 5a shows a schematic diagram of an amplitude spectrum of a single loudspeaker in a two-sine excitation with, for example, 200 Hz and 5500 Hz.
  • the resulting amplitude spectrum of the single loudspeaker with an additional compensation loudspeaker is shown in Fig. 5b.
  • the use of the compensation means causes a reduction of IMD intermodulation distortion from 4.3% to IMD 1.2%.
  • the investigations with the model-based control signals show that the loudspeaker model taken from the specialist literature is too inaccurate to effect harmonic distortion reduction when excited with discrete sine tones. In the case of harmonic distortions, a reduction of 20% to 1-3% percent or an attenuation of at least 16 dB is achieved. Possibly, the adaptive tracking, which constantly renews the model parameters, can provide greater compensation despite the simple speaker model.
  • the following table shows the THD of the measured sound velocity of the distorted loudspeaker D2z alone and together with the control signal calculated on the basis of the rapid measurement, which was reproduced via the same transducer.
  • the second overtone is reduced less at 170 Hz and 200 Hz when the control signal is reproduced via the same converter D 2Z.
  • the reduction power of both methods is almost identical.
  • the lower reduction at 170Hz and 200Hz when the control signal is reproduced through the same transducer can be explained by the high deflection of the diaphragm at lower frequencies. It can also be read that the investigated miniature converter achieves a higher membrane deflection at 170 Hz and 200 Hz than at 300 Hz.
  • the converter generates higher non-linear distortions due to the higher deflection, which can be recognized by the increasing harmonic distortion.
  • the addition of the control signal to the excitation signal will result in peaks in the time signal depending on the phase position of the harmonics, which cause an even higher deflection and form new distortion products whose compensation is not provided by the actual control signal.
  • the converter already reaches its deflection limit and no longer has sufficient capacity or latitude to precisely reproduce the additional control signal.
  • the converter At the 300Hz tone, which the converter generates with less deflection, there is still enough deflection reserve to emulate the equalizer signal sufficiently.
  • the equalization method with additional equalizer loudspeakers makes sense as soon as the distorted loudspeaker is already operated close to its deflection limit.
  • the control signal can "about the same Zerrlaut Maschinener no longer be reproduced accurately enough.
  • the Zerrlaut Anlagener is in a less non-linear region, ie at low deflection, the new method offers no advantages over the prior art. Both methods are then able distortion products
  • the influence of propagation time differences at higher frequencies can be a disadvantage compared with the prior art.
  • the values for harmonic distortion can be reduced by up to 21 dB.
  • the distortion behavior of a transducer group consisting of twelve loudspeakers could be significantly improved by an equalizer loudspeaker.
  • Applications for the above-mentioned equalization are: loudspeaker / loudspeaker groupings, sound transducers, actuators, closed loudspeakers, ventilated loudspeakers, loudspeakers with one / several active and one / more passive diaphragms, structure-borne sound exciters, bass shakers, ultrasonic transducers, exciters, sound sources of all kinds
  • the term "speaker" mentioned in the text can generally be replaced by the terms listed here.
  • aspects have been described in the context of a device, it will be understood that these aspects also constitute a description of the corresponding method, so that a block or a component of a device is also to be understood as a corresponding method step or as a feature of a method step , Similarly, aspects described in connection with or as a method step also represent a description of a corresponding block or detail or feature of a corresponding device.
  • Some or all of the method steps may be performed by a hardware device (or using a hardware device). Apparatus), such as a microprocessor, a programmable computer or an electronic circuit. In some embodiments, some or more of the most important method steps may be performed by such an apparatus.
  • embodiments of the invention may be implemented in hardware or in software.
  • the implementation may be performed using a digital storage medium, such as a floppy disk, a DVD, a Blu-ray Disc, a CD, a ROM, a PROM, an EPROM, an EEPROM or FLASH memory, a hard disk, or other magnetic disk or optical memory are stored on the electronically readable control signals that can cooperate with a programmable computer system or cooperate such that the respective method is performed. Therefore, the digital storage medium can be computer readable.
  • embodiments according to the invention include a data carrier having electronically readable control signals capable of interacting with a programmable computer system such that one of the methods described herein is performed.
  • embodiments of the present invention may be implemented as a computer program product having a program code, wherein the program code is operable to perform one of the methods when the computer program product runs on a computer.
  • the program code can also be stored, for example, on a machine-readable carrier.
  • inventions include the computer program for performing any of the methods described herein, wherein the computer program is stored on a machine-readable medium.
  • an exemplary embodiment of the method according to the invention is thus a computer program which has program code for carrying out one of the methods described herein when the computer program runs on a computer.
  • a further embodiment of the inventive method is thus a data carrier (or a digital storage medium or a computer-readable medium) on which the computer program is recorded for carrying out one of the methods described herein.
  • a further embodiment of the method according to the invention is thus a data stream or a sequence of signals, which represent the computer program for performing one of the methods described herein.
  • the data stream or the sequence of signals may be configured, for example, to be transferred via a data communication connection, for example via the Internet.
  • Another embodiment includes a processing device, such as a computer or a programmable logic device, that is configured or adapted to perform one of the methods described herein.
  • Another embodiment includes a computer on which the computer program is installed to perform one of the methods described herein.
  • Another embodiment according to the invention comprises a device or system adapted to transmit a computer program for performing at least one of the methods described herein to a receiver.
  • the transmission can be done for example electronically or optically.
  • the receiver may be, for example, a computer, a mobile device, a storage device or a similar device.
  • the device or system may include a file server for transmitting the computer program to the recipient.
  • a programmable logic device eg, a field programmable gate array, an FPGA
  • a field programmable gate array may cooperate with a microprocessor to perform one of the methods described herein.
  • the methods are performed by any hardware device. This may be a universal hardware such as a computer processor (CPU) or hardware specific to the process, such as an ASIC.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Health & Medical Sciences (AREA)
  • Otolaryngology (AREA)
  • Multimedia (AREA)
  • Circuit For Audible Band Transducer (AREA)

Abstract

L'invention concerne des éléments de compensation pour un système de haut-parleurs comprenant un premier groupe de haut-parleurs pourvu d'au moins un transducteur acoustique, le premier groupe de haut-parleurs étant conçu pour produire un premier signal sonore sur la base d'un signal audio, le signal sonore comprenant une fraction de signal utile et une fraction de signal de distorsion. Les éléments de compensation comprennent un deuxième groupe de haut-parleurs pourvu d'au moins un transducteur acoustique, le deuxième groupe de haut-parleurs étant conçu pour produire un deuxième signal sonore sur la base du signal de compensation, le deuxième signal sonore compensant et/ou réduisant la fraction de signal de distorsion en cas de superposition avec le premier signal sonore.
EP17751256.3A 2016-07-13 2017-07-12 Éléments de compensation pour un système de haut-parleurs Withdrawn EP3485653A1 (fr)

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DE102016212828 2016-07-13
DE102017200488.6A DE102017200488A1 (de) 2016-07-13 2017-01-13 Kompensationsmittel für ein lautsprechersystem und lautsprechersystem
PCT/EP2017/067603 WO2018011296A1 (fr) 2016-07-13 2017-07-12 Éléments de compensation pour un système de haut-parleurs

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JP2560618Y2 (ja) 1991-07-11 1998-01-26 株式会社ケンウッド スピーカシステム
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JP3260078B2 (ja) * 1996-04-19 2002-02-25 株式会社ケンウッド 音響再生装置
JP3533092B2 (ja) * 1998-08-05 2004-05-31 パイオニア株式会社 オーディオシステム
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DE102009010278B4 (de) 2009-02-16 2018-12-20 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Lautsprecher
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US20190147844A1 (en) 2019-05-16
DE102017200488A1 (de) 2018-01-18
US10872591B2 (en) 2020-12-22
WO2018011296A1 (fr) 2018-01-18

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