EP1972181B1 - Dispositif et procédé de simulation de systèmes wfs et de compensation de propriétés wfs influençant le son - Google Patents

Dispositif et procédé de simulation de systèmes wfs et de compensation de propriétés wfs influençant le son Download PDF

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Publication number
EP1972181B1
EP1972181B1 EP07700228A EP07700228A EP1972181B1 EP 1972181 B1 EP1972181 B1 EP 1972181B1 EP 07700228 A EP07700228 A EP 07700228A EP 07700228 A EP07700228 A EP 07700228A EP 1972181 B1 EP1972181 B1 EP 1972181B1
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Prior art keywords
aliasing
wave field
aliasing filter
field synthesis
virtual sound
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EP07700228A
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German (de)
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EP1972181A1 (fr
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Joachim Deguara
René RODIGAST
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
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    • 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
    • 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 
    • H04S2420/00—Techniques used stereophonic systems covered by H04S but not provided for in its groups
    • H04S2420/13—Application of wave-field synthesis in stereophonic audio systems

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  • the present invention relates to wave field synthesis systems, and more particularly to aliasing correction in wave field synthesis systems.
  • WFS wave field synthesis
  • Applied to the acoustics can be simulated by a large number of speakers, which are arranged side by side (a so-called speaker array), any shape of an incoming wavefront.
  • a so-called speaker array any shape of an incoming wavefront.
  • the audio signals of each speaker must be fed with a time delay and amplitude scaling so that the radiated sound fields of each speaker properly overlap.
  • the contribution to each speaker is calculated separately for each source and the resulting signals added together.
  • reflections can also be reproduced as additional sources via the loudspeaker array. The cost of the calculation therefore depends heavily on the number of sound sources, the reflection characteristics of the recording room and the number of speakers.
  • the advantage of this technique is in particular that a natural spatial sound impression over a large area of the playback room is possible.
  • the direction and distance of sound sources are reproduced very accurately.
  • virtual sound sources can even be positioned between the real speaker array and the listener.
  • wavefield synthesis works well for environments whose characteristics are known, irregularities occur when the texture changes, or when wave field synthesis is performed based on environmental conditions that do not match the actual nature of the environment.
  • the technique of wave field synthesis can also be used advantageously to supplement a visual perception with a corresponding spatial audio perception.
  • production in virtual studios focused on providing an authentic visual impression of the virtual scene.
  • the matching to the image acoustic impression is usually impressed by manual operations in the so-called post-production subsequently the audio signal or classified as too complex and time-consuming in the realization and therefore neglected. This usually leads to a contradiction of the individual sense sensations, which results in that the designed space, d. H. the designed scene is perceived as less authentic.
  • Wave Field Synthesis In the audio field, the technique of Wave Field Synthesis (WFS) can be used to achieve a good spatial sound for a large listener area.
  • wave field synthesis is based on the principle of Huygens, according to which wavefronts can be formed and built up by superimposing elementary waves. After mathematically exact theoretical description, infinitely many sources in infinitely small distance would have to be used for the generation of the elementary waves. Practically, however, many speakers are finally used at a finite distance from each other. Each of these speakers is driven according to the WFS principle with an audio signal from a virtual source having a particular delay and a certain level. Levels and delays are usually different for all speakers.
  • the wave field synthesis system operates on the basis of the Huygens principle and reconstructs a given waveform of, for example, a virtual source located at a certain distance from a demonstration area or a listener in the show area by a plurality of single waves .
  • the wave field synthesis algorithm thus obtains information about the actual position of a single loudspeaker from the loudspeaker array, in order then to calculate a component signal for this single loudspeaker, which then ultimately has to radiate this loudspeaker so that the listener can overlay the loudspeaker signal from the loudspeaker signal a speaker with the speaker signals of the other active speakers performs a reconstruction in that the listener has the impression that he is not "sonicated" by many individual speakers, but only from a single speaker at the position of the virtual source.
  • each virtual source for each loudspeaker that is, the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc.
  • the contribution from each virtual source for each loudspeaker is calculated to then add up the component signals finally get the actual speaker signal.
  • superimposing the loudspeaker signals of all active loudspeakers on the listener would mean that the listener does not feel that he is being sonicated by a large array of loudspeakers, but that the sound he hears is merely from three sound sources positioned at specific positions, which are equal to the virtual sources.
  • the computation of the component signals usually takes place in practice by applying a delay and a scaling value to the audio source assigned to a virtual source, depending on the position of the virtual source and the position of the loudspeaker, at a particular time, a delayed and / or scaled audio signal the virtual source that directly represents the loudspeaker signal when there is only one virtual source, or that adds to other component signals for the considered loudspeaker from other virtual sources then to the loudspeaker signal for the considered loudspeaker.
  • Typical wave field synthesis algorithms work regardless of how many speakers are present in the speaker array are.
  • the underlying theory of Wave Field Synthesis is that any sound field can be accurately reconstructed by an infinite number of individual speakers, with the individual individual speakers arranged infinitely close to each other. In practice, however, neither the infinitely high number nor the infinitely close arrangement can be realized. Instead, there are a limited number of speakers, which are also arranged at certain predetermined distances from each other. Thus, in real systems, only an approximation to the actual waveform that would occur if the virtual source were actually present, would be a real source.
  • the object of the present invention is to provide a concept for aliasing correction in a wave field synthesis system which reduces quality variations in the perceived sound.
  • the present invention is based on the finding that aliasing correction in a wave field synthesis system is improved by determining the virtual source specific aliasing filter property using the source position information.
  • This aliasing filter property which is e.g. B. the aliasing frequency can be determined using the source position information.
  • This aliasing filter property is used for an adaptive anti-aliasing filter for adaptively filtering the audio signal assigned to the sources or the component signals assigned to the source.
  • a listening point in the playback room is selected, and the wave-field synthesis module provides scaling and delay values for the individual speakers corresponding to a virtual source.
  • the amplitude value and the time value of the arrival of the pulse at the listening point are calculated for a given pulse.
  • the individual impulses of the individual loudspeakers do not arrive simultaneously at the listening point and instead supply time signals and time values. These time signals are transformed into a spectral representation, from which the aliasing frequency is determined.
  • This aliasing frequency marks the area between a fluctuating one Behavior of the spectral representation and an increasing behavior to lower frequencies.
  • This aliasing frequency is now used as input for an anti-aliasing filter, which corrects the level below the aliasing frequency, eg attenuates with 3dB per octave.
  • each virtual source is assigned an aliasing frequency. This makes it possible to dynamically filter even moving virtual sources and thus discoloration due to the movement is suppressed. In previously used static filters, this is not possible and as a result these static filters lead to a distortion of the sound during a movement of the virtual sources.
  • the aliasing filter in a computer system one can perform the filtering in real time with the movement of the virtual sources. In order to save computing time, in a further embodiment it is not possible to calculate the aliasing frequency continuously for all possible positions of the virtual source, but instead to determine it only for discrete points. These obtained aliasing frequencies may be e.g. be included in a table so that further calculations are omitted. The quality achieved is given by the density of the discrete points.
  • aliasing filtering can also be performed with respect to different listening points. By averaging these different aliasing frequencies associated with a virtual source, one can obtain an average aliasing frequency for the entire listening room. This averaged aliasing frequency, in turn, changes as the position of the virtual source changes, and can be corrected as previously described, depending on the position of the virtual source.
  • this bass tone boost is dynamic and depends on different factors. These are z. B. the speaker density and the angle of incidence of the virtual sound sources.
  • the aliasing frequency changes with the positioning of the virtual sound sources and is therefore dynamic. This dynamic is not taken into account in the current calculation.
  • a major disadvantage of previous WFS systems is that swelling movements are perceivable as tone color changes. These are the result of the static filter and the dynamic change of the aliasing frequency and the bass boost. These tone color changes are particularly significant when the virtual source is parallel to the speakers.
  • Another disadvantage of the existing technology is that different speaker setups (with different speaker distances) affect the aliasing frequency and the bass boost, which until now had to be adjusted manually on the respective setup.
  • the wave field synthesis system has a speaker array 700 placed with respect to a demonstration area 702.
  • a demonstration area 702. this includes in Fig. 7 shown speaker array which is a 360 ° array, four array sides 700a, 700b, 700c and 700d. If the demonstration area 702 z.
  • the cinema screen is on the same side of the screening area 702 on which also the sub-array 700 c is arranged with respect to the conventions front / back or right / left. In this case, the observer who is sitting at the so-called optimal point P in the demonstration area 702 would see to the front, ie to the screen.
  • Each loudspeaker array consists of a number of different individual loudspeakers 708, each of which is driven by its own loudspeaker signals transmitted by a wave field synthesis module 710 via an in-line loudspeaker signal Fig. 7 only schematically shown data bus 712 are provided.
  • the wave field synthesis module is configured to use the information about e.g. B.
  • the wave field synthesis module can also receive further inputs, such as information about the room acoustics of the demonstration area, etc.
  • the following embodiments of the present invention may in principle be performed for each point P in the demonstration area.
  • the optimum point can thus lie anywhere in the demonstration area 702. It can also be several optimal points, z. B. on an optimal line, give. However, in order to obtain the best possible ratios for as many points as possible in the demonstration area 702, it is preferable to use the optimum point or the optimal line in the middle or at the center of gravity of the wave field synthesis system, which is defined by the speaker sub-arrays 700a, 700b, 700c, 700d.
  • FIG. 12 is a block diagram of the inventive apparatus for aliasing correction in a wave-field synthesis system, referring to FIG Fig. 7 has been set out.
  • a wave field synthesis module 100 having an input for the audio signals 102 of the virtual sources, an input for the position data 104 of the virtual sources, an input for the position data of the speakers 106 and optionally other inputs 108, the z. B. provide information about the room acoustics has.
  • the wave-field synthesis module 100 provides the component signals 110 as well as the corresponding delay and scale values for the individual speakers.
  • AFE source-specific aliasing filter characteristic
  • the aliasing filter property 130 as well as the component signals 110 serve as inputs to the adaptive anti-aliasing filter 140 for the virtual sources. After filtering the component signals 110, the corresponding loudspeaker signals 160 are created in a means for combining the component signals 150.
  • Fig. 1b For example, an apparatus according to the invention is shown in which the component signals 110 are not filtered by the adaptive anti-aliasing filter 140, but the audio signals 102 are filtered in the virtual source adaptive anti-aliasing filter 140.
  • the filtered audio signal 165 is input to the wave field synthesis module 100 to generate filtered component signals and to generate the corresponding loudspeaker signals 160 in the component signal combining unit 150.
  • the wave field synthesis module 100 receives an audio signal and position information from each virtual source.
  • this figure shows the audio signal of the first source 212 and the position of the first source 214, the audio signal of the second source 222 and the position information of the second source 224 as well as the audio signal of the last source 232 and the position information of the last source 234.
  • the wave field synthesis module 100 determines for each virtual source the component signals for each speaker.
  • the component signals of the first virtual source KS11 to KSn 240, the second virtual source KS21 to KS2n 250 and the component signals of the last virtual source KSm1 to KSmn 260 are shown by way of example.
  • Fig. 3a shows a block diagram of a preferred apparatus according to the invention for determining the aliasing frequency.
  • the wave field synthesis module 100 generates a wave field synthesis scaling value (WFS-SW) and a wave field synthesis delay value (WFS-VW) 310 for a virtual source. From the position of the listening point 320 and the position information of the speakers 330, a propagation delay value (AVZW) is generated in the device 340. and a spread scaling value (ASKW). These values, along with the WFS-SW and the WFS-VW 310, serve as input to the device 350, which determines a total scaling value (GSW) as well as a total delay value (GVW).
  • WFS-SW wave field synthesis scaling value
  • WFS-VW wave field synthesis delay value
  • a time signal and corresponding time values are determined in the device 360, which is converted into a spectral representation in the device 370. Finally, this spectral representation is evaluated in the device 380 and a corresponding aliasing frequency 390 is determined.
  • each speaker 708 is shown, all of which are fed with their own loudspeaker signal, generated by the wave field synthesis module 100. So each speaker can be modeled as a point wave that outputs a concentric wave field. Following the law of the concentric wave field, the level of the sound field decreases with the distance r to the loudspeakers by a factor of 1 / r 2 . This results in a dependence of 1 / r on the signal. Taking into account the propagation velocity of the sound wave, it can thus be determined when (propagation delay value) with respect to the loudspeaker which signal arrives in which scaling (propagation scaling value) at the listening point P.
  • FIG. 11 shows a concrete example of a 10-speaker demonstration area 702, from which the loudspeakers 4 to 7 transmit a virtual source signal having a particular scaling value and delay 392.
  • a total delay and a total scaling value at the listening point 394 After taking into account the time delay and the attenuation due to the propagation from the loudspeakers to the listening point P, one obtains for each loudspeaker a total delay and a total scaling value at the listening point 394. If these total scaling values are plotted according to the total delay values as the time coordinate, the time signal results at the bottom left in Fig. 3c , which is called IR (impulse response) at the listening point.
  • IR impulse response
  • the first signal with the smallest time value corresponds to the signal emitted by loudspeaker 6, which according to table 392 has a scaling value of 0.8 and a delay value of 10 ms.
  • the second signal in 394 is the signal from loudspeaker 5, which according to Table 392 has a scaling value of 0.7 and a delay value of 12ms.
  • the signals then follow from the loudspeaker 4 and from the loudspeaker 7, whose scaling and delay values are also indicated in table 392.
  • This time signal is converted into a spectral representation 396 characterized by two regions. At high frequencies, the spectral representation shows a fluctuating behavior, and too low Frequencies an increasing behavior. In the transition area between the areas is the aliasing frequency.
  • This aliasing frequency then serves as an input signal for a corresponding correction filter 398. This filter serves to bring about a lowering of the low frequency components by, for example, 3dB per octave.
  • Fig. 4 shows a block diagram in which the determination of the aliasing frequencies for different virtual sources is shown.
  • Wave field synthesis module 100 provides scaling and delay values for each virtual source and speaker. In the example shown here, both the scaling and delay values of the first virtual source 402 and the scaling and delay values of the last virtual source 404 are shown. By combining these values with the propagation delay values and the propagation scaling values, a set of data is thus obtained for each virtual source, which in turn serves as inputs to the means 350 for determining the total scaling values and the total delay values. From this, the device 360 determines separately for each virtual source corresponding time signals and time values, which in turn are converted into a spectral representation in the device 370. These spectral representations will be evaluated in the device 380 so as to obtain aliasing frequencies 410 for each virtual source.
  • Fig. 5 shows a block diagram in which aliasing frequencies are determined for each Ab stealddling and then averaging an average aliasing frequency is determined.
  • the scaling values and delay values 310 for a virtual source serve as inputs to a means 510 for determining a source-specific aliasing filter characteristic for a first listening point, and as inputs to a means for determining a source-specific aliasing filter characteristic for a second listening point 520.
  • the scaling and delay values are also determined in a corresponding means for determining a source-specific aliasing filter property.
  • the filter characteristics thus obtained for each listening point are averaged in the device 530 over all the listening points. This gives the entire listening area 702 an aliasing filter property for each virtual source.
  • This averaged aliasing filter property may be e.g. B. be an average aliasing filter frequency.
  • Fig. 6 shows a block diagram of an adaptive filter for virtual sources.
  • the inputs to this virtual source adaptive filter 140 are both the aliasing frequencies f 1 through f n and the component signals 110, those with KS11 through KS1n for the first virtual source, with KS21 through KS2n for the second virtual source, and with KSm1 through KSmn for the last virtual source are designated.
  • the output signals of the adaptive filter 140 are modified component signals 610, which in turn serve as input to the means 150 for combining the component signals to finally provide the loudspeaker signals 160.
  • the aliasing frequency determined in this algorithm is the dynamically changing frequency below which WFS playback produces a bass boost of, for example, 3dB per octave. Above this frequency, aliasing artifacts lead to frequency cancellations and comb filter effects. As already stated, a dynamic filter is calculated by analyzing this frequency, which compensates the bass boost source dependent. Depending on the speaker setup used, this boost does not always correspond to the theoretical value of 3dB per octave. This dynamic correction filter is constantly updated during source movements. The result is the optimal bass correction for the respective source position.
  • the source position-dependent scaling and delay values of the signal are constantly determined. From knowledge of the current aliasing frequency, a correction filter is calculated and constantly updated (source position-dependent). The speaker signals for this source are calculated by this correction filter. According to the invention, an optimal sound for different speaker setups including the source-position-dependent aliasing frequency is thus achieved in the calculation of the loudspeaker signals. In addition, this results in correction possibilities of the loudspeaker frequency response by including the loudspeaker parameters in the calculation. It is also possible to integrate it as a plugin into conventional simulation tools (eg in EASE). Likewise, real sound field calculations involving the entire transmission chain (source position, WFS algorithm, loudspeaker parameters, room parameters, listening position) can take place.
  • a complex impulse response is thus calculated in a preferred embodiment with knowledge of the position of a virtual sound source, as well as the loudspeaker and room parameters. Simulations and auralizations of WFS sound fields are possible with this impulse response.
  • the system also provides information on the dynamic control of the compensation filter (3dB filter) for the WFS.
  • An optimized filter improves the sound quality of a WFS system.
  • the scheme according to the invention can also be implemented in software.
  • the implementation may be on a digital storage medium, in particular a floppy disk or a CD with electronically readable control signals, which may interact with a programmable computer system such that the corresponding method is executed.
  • the invention thus also consists in a computer program product with program code stored on a machine-readable carrier for carrying out the method according to the invention, when the computer program product runs on a computer.
  • the invention can thus be realized as a computer program with a program code for carrying out the method when the computer program runs on a computer.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Stereophonic System (AREA)
  • Obtaining Desirable Characteristics In Audible-Bandwidth Transducers (AREA)

Abstract

Une correction de repliement dans un système de champ d'ondes est effectuée en déterminant la propriété de filtre repliement du spectre, spécifique d'une source virtuelle. Cette propriété de filtre repliement du spectre, qui peut être, par exemple, la fréquence de repliement, est déterminée au moyen de l'information de position des sources. Cette propriété de filtre repliement du spectre est utilisée pour un filtre anti-repliement, pour le filtrage adaptatif du signal audio associé à la source ou des signaux de composants associés à la source.

Claims (13)

  1. Dispositif de correction de repliement pour un système de synthèse de champ d'ondes avec un module de synthèse de champ d'ondes (100) et une rangée de haut-parleurs (708) pour la sonorisation d'une zone de représentation (702), le module de synthèse de champ d'ondes (100) étant réalisé pour recevoir un signal audio (102) associé à une source sonore virtuelle ainsi que l'information de position de source (104) associée à la source sonore virtuelle et pour calculer, en tenant compte des informations de position de haut-parleurs (106), des signaux à composantes (110) pour les haut-parleurs (708) sur base de la source sonore virtuelle, aux caractéristiques suivantes:
    un moyen pour déterminer (120) une propriété de filtre de repliement (130) spécifique à une source sonore virtuelle à l'aide des informations de position de source (104), le moyen pour déterminer étant réalisé pour obtenir, pour les haut-parleurs (708) dans la rangée, les valeurs de modulation de synthèse de champ d'ondes et les valeurs de retard de synthèse de champ d'ondes (310) associées aux haut-parleurs, et pour déterminer, sur base d'un point d'écoute (P) dans la zone de représentation (702) et des valeurs de modulation de synthèse de champ d'ondes et des valeurs de retard de synthèse de champ d'ondes (310), la propriété de filtre de repliement (130);
    caractérisé par le fait que le dispositif présente un filtre anti-repliement adaptatif (140) destiné à filtrer de manière adaptative le signal audio (102) associé à la source sonore virtuelle ou les signaux à composantes (110) associés à la source sonore virtuelle, le filtre anti-repliement adaptatif (140) étant réglé selon la propriété de filtre de repliement (130) spécifique à la source sonore virtuelle, afin de provoquer une correction de repliement.
  2. Dispositif selon la revendication 1, dans lequel le moyen pour déterminer (120) est réalisé pour calculer la propriété de filtre de repliement (130) à l'aide d'une réponse impulsionnelle pour un canal entre la source sonore virtuelle et un point d'écoute (P) dans la salle de reproduction (702).
  3. Dispositif selon la revendication 1 ou la revendication 2, dans lequel le moyen pour déterminer (120) est réalisé pour déterminer des valeurs de retard de propagation et des valeurs de modulation de propagation (340) entre les haut-parleurs (708) et le point d'écoute (P), pour combiner, pour chaque haut-parleur, la valeur de retard de synthèse de champ d'ondes et la valeur de retard de propagation, pour obtenir une valeur de retard totale, pour combiner, pour chaque haut-parleur, la valeur de modulation de synthèse de champ d'ondes et la valeur de modulation de propagation, pour obtenir une valeur de modulation totale, et pour déterminer une réponse impulsionnelle à la source sonore virtuelle et au point d'écoute (P) à l'aide des valeurs de modulation totale et des valeurs de retard total pour les haut-parleurs (708).
  4. Dispositif selon la revendication 3, dans lequel le moyen pour déterminer (120) est réalisé pour convertir un signal temporel à valeurs temporelles dont les coordonnées temporelles sont définies par les valeurs de retard total et dont les amplitudes sont définies par les valeurs de modulation totale en une représentation spectrale, et pour déterminer, comme propriété de filtre de repliement (130), une fréquence de filtre de repliement (390) à partir de la représentation spectrale.
  5. Dispositif selon la revendication 2, dans lequel le moyen pour déterminer (120) est réalisé pour déterminer, comme propriété de filtre de repliement (130), une fréquence de filtre de repliement (390) à partir d'une représentation spectrale de la réponse impulsionnelle.
  6. Dispositif selon l'une des revendications 4 ou 5, dans lequel le moyen pour déterminer (120) est réalisé pour déterminer, comme fréquence de filtre de repliement (390), une fréquence qui se situe dans une plage qui est limitée vers les basses fréquences par une augmentation de la représentation spectrale, et qui est limitée vers les fréquences supérieures par une fluctuation de la représentation spectrale.
  7. Dispositif selon la revendication 6, dans lequel le moyen pour déterminer (120) est réalisé pour choisir, comme propriété de filtre de repliement (130), une fréquence qui s'écarte de moins de ± 25 % d'une valeur de fréquence correspondant à une valeur de transition entre une augmentation de la représentation spectrale et une fluctuation de la représentation spectrale.
  8. Dispositif selon la revendication 4 ou 5, dans lequel le moyen pour déterminer (120) est réalisé pour déterminer, pour une source sonore virtuelle, des propriétés de filtre de repliement (130) pour différents points d'écoute dans la salle de reproduction (702) et pour établir la moyenne des différentes propriétés de filtre de repliement, pour obtenir la propriété de filtre de repliement spécifique à la source sonore virtuelle.
  9. Dispositif selon l'une des revendications 1 à 7, dans lequel le moyen pour déterminer (120) est réalisé pour calculer, pour des sources sonores virtuelles à différentes positions virtuelles, différentes propriétés de filtre de repliement, et dans lequel le filtre anti-repliement adaptatif (140) est réalisé pour filtrer les signaux audio (102) associés aux sources sonores virtuelles ou les signaux à composantes (110) associés aux sources sonores virtuelles à l'aide des différentes propriétés de filtre de repliement.
  10. Dispositif selon la revendication 9, dans lequel le filtre anti-repliement adaptatif (140) est réalisé pour filtrer séparément les signaux audio (102) associés aux sources sonores virtuelles à l'aide des différentes propriétés de filtre de repliement, pour obtenir des signaux audio filtrés en repliement, et dans lequel le module de synthèse de champ d'ondes (100) est réalisé pour calculer les signaux à composantes (110) pour chaque source sonore virtuelle à l'aide des signaux audio filtrés, et pour combiner les signaux à composantes appartenant à un haut-parleur, pour obtenir un signal de haut-parleur (160) pour le haut-parleur.
  11. Dispositif selon la revendication 9, dans lequel le filtre anti-repliement adaptatif (140) est réalisé pour filtrer les signaux à composantes (110) calculés pour une première source virtuelle à l'aide de la propriété de filtre de repliement (130) spécifique à la première source virtuelle, pour obtenir des premiers signaux à composantes filtrés en repliement pour la première source virtuelle, et pour obtenir, pour une deuxième source virtuelle, des deuxièmes signaux à composantes filtrés en repliement pour la deuxième source virtuelle, le module de synthèse de champ d'ondes (100) étant par ailleurs réalisé pour combiner les signaux à composantes (110) appartenant à un haut-parleur parmi les premiers signaux à composantes filtrés en repliement et les deuxièmes signaux à composantes filtrés en repliement, pour obtenir un signal de haut-parleur (160) pour le haut-parleur.
  12. Procédé de correction de filtre de repliement pour un système de synthèse de champ d'ondes avec un module de synthèse de champ d'ondes (100) et une rangée de haut-parleurs (708) pour la sonorisation d'une zone de représentation (702), le module de synthèse de champ d'ondes (100) étant réalisé pour recevoir un signal audio (102) associé à une source sonore virtuelle ainsi que les informations de position de source (104) associée à la source sonore virtuelle et pour calculer, en tenant compte des informations de position de haut-parleurs (106), des signaux à composantes (110) pour les haut-parleurs (708) sur base de la source sonore virtuelle, avec l'étape suivante consistant à:
    déterminer des propriétés de filtre de repliement (130) spécifiques à une source sonore virtuelle à l'aide des informations de position de source (104), l'étape de détermination comprenant l'obtention des valeurs de modulation de synthèse de champ d'ondes et les valeurs de retard de synthèse de champ d'ondes (310) associées aux haut-parleurs, de sorte que soit déterminée, sur base d'un point d'écoute (P) dans la zone de représentation (702) et des valeurs de modulation de synthèse de champ d'ondes et des valeurs de retard de synthèse de champ d'ondes (310), la propriété de filtre de repliement (130);
    et caractérisé par l'étape suivante consistant à:
    filtrer de manière adaptative les signaux audio (102) associés à la source sonore virtuelle ou les signaux à composantes (110) associés à la source sonore virtuelle, la filtration adaptative étant réalisée selon la propriété de filtre de repliement (130) spécifique à la source, afin de provoquer une correction de repliement.
  13. Programme d'ordinateur avec un code de programme pour réaliser le procédé selon la revendication 12 lorsque le programme d'ordinateur est exécuté sur un ordinateur.
EP07700228A 2006-03-06 2007-01-17 Dispositif et procédé de simulation de systèmes wfs et de compensation de propriétés wfs influençant le son Expired - Fee Related EP1972181B1 (fr)

Applications Claiming Priority (2)

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DE102006010212A DE102006010212A1 (de) 2006-03-06 2006-03-06 Vorrichtung und Verfahren zur Simulation von WFS-Systemen und Kompensation von klangbeeinflussenden WFS-Eigenschaften
PCT/EP2007/000385 WO2007101498A1 (fr) 2006-03-06 2007-01-17 Dispositif et procédé de simulation de systèmes wfs et de compensation de propriétés wfs influençant le son

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EP1972181A1 EP1972181A1 (fr) 2008-09-24
EP1972181B1 true EP1972181B1 (fr) 2010-12-22

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EP (1) EP1972181B1 (fr)
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WO (1) WO2007101498A1 (fr)

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DE102005033239A1 (de) * 2005-07-15 2007-01-25 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Steuern einer Mehrzahl von Lautsprechern mittels einer graphischen Benutzerschnittstelle
DE102005033238A1 (de) * 2005-07-15 2007-01-25 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Ansteuern einer Mehrzahl von Lautsprechern mittels eines DSP
DE102006053919A1 (de) * 2006-10-11 2008-04-17 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Erzeugen einer Anzahl von Lautsprechersignalen für ein Lautsprecher-Array, das einen Wiedergaberaum definiert
KR101268779B1 (ko) * 2009-12-09 2013-05-29 한국전자통신연구원 라우드 스피커 어레이를 사용한 음장 재생 장치 및 방법
JP2013051643A (ja) * 2011-08-31 2013-03-14 Nippon Hoso Kyokai <Nhk> スピーカアレイ駆動装置およびスピーカアレイ駆動方法
EP2777301B1 (fr) 2011-11-10 2015-08-12 SonicEmotion AG Procédé d'implémentations pratiques de reproduction de champs sonores basé sur des intégrales de surface en trois dimensions
DE102012200512B4 (de) 2012-01-13 2013-11-14 Fraunhofer-Gesellschaft zur Förderung der angewandten Forschung e.V. Vorrichtung und Verfahren zum Berechnen von Lautsprechersignalen für eine Mehrzahl von Lautsprechern unter Verwendung einer Verzögerung im Frequenzbereich
WO2013149867A1 (fr) * 2012-04-02 2013-10-10 Sonicemotion Ag Procédé pour reproduction efficace de son 3d haute qualité
WO2014007724A1 (fr) * 2012-07-06 2014-01-09 Dirac Research Ab Conception de commande de pré-compensation audio avec similitude par paires entre canaux de haut-parleurs
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CN101406075B (zh) 2010-12-01
WO2007101498A1 (fr) 2007-09-13
CN101406075A (zh) 2009-04-08
US20090220111A1 (en) 2009-09-03
DE102006010212A1 (de) 2007-09-20
DE502007006021D1 (de) 2011-02-03
JP2009529262A (ja) 2009-08-13
JP4977720B2 (ja) 2012-07-18
US8363847B2 (en) 2013-01-29
EP1972181A1 (fr) 2008-09-24

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