EP1972181B1 - Device and method for simulating wfs systems and compensating sound-influencing wfs characteristics - Google Patents

Abstract

An aliasing correction in a wave field synthesis system is performed by determining the aliasing filter characteristic specific to a virtual source. Said aliasing filter characteristic, which can be the aliasing frequency for example, is determined by means of the source position information. The aliasing filter characteristic is used for an adaptive anti-aliasing filter for the adaptive filtering of the audio signal associated with the source or the component signals associated with the source.

Description

The present invention relates to wave field synthesis systems, and more particularly to aliasing correction in wave field synthesis systems.

There is an increasing demand for new technologies and innovative products in the field of consumer electronics. It is an important prerequisite for the success of new multimedia systems to offer optimal functionalities and capabilities. This is achieved through the use of digital technologies and especially computer technology. Examples of these are the applications that offer an improved, realistic audiovisual impression. In previous audio systems, a significant weakness lies in the quality of the spatial sound reproduction of natural, but also of virtual environments.

Methods for multi-channel speaker reproduction of audio signals have been known and standardized for many years. All the usual techniques have the disadvantage that both the installation site of the loudspeakers and the position of the listener are already impressed on the transmission format. If the speakers are arranged incorrectly with respect to the listener, the audio quality suffers significantly. An optimal sound is only possible in a small area of the playback room, the so-called sweet spot.

A better natural spatial impression as well as a stronger envelope in the audio reproduction can be achieved with the help of a new technology. The basics of this technology, the so-called wave field synthesis (WFS), were researched at the TU Delft and first presented in the late 80s ( Berkhout, AJ; de Vries, D .; Vogel, P .: Acoustic control by wave-field synthesis. JASA 93, 1993 ).

Due to the enormous demands of this method on computer performance and transmission rates, wave field synthesis has rarely been used in practice. Only the advances in the areas of microprocessor technology and audio coding allow today the use of this technology in concrete applications. In a few years, the first wave field synthesis applications for the consumer sector will be launched.

• Jeder Punkt, der von einer Welle erfasst wird, ist Ausgangspunkt einer Elementarwelle, die sich kugelförmig bzw. kreisförmig ausbreitet.

The basic idea of WFS is based on the application of Huygens' principle of wave theory:

• Every point, which is detected by a wave, is the starting point of an elementary wave, which spreads in a spherical or circular manner.

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. In the simplest case, a single point source to be reproduced and a linear arrangement of the speakers, 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. With multiple sound sources, the contribution to each speaker is calculated separately for each source and the resulting signals added together. In a room with reflective walls, 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. In contrast to the known techniques, the direction and distance of sound sources are reproduced very accurately. To a limited extent, virtual sound sources can even be positioned between the real speaker array and the listener.

Although 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.

However, the technique of wave field synthesis can also be used advantageously to supplement a visual perception with a corresponding spatial audio perception. Until now, 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.

In the technical publication " Subjective experiments on the effects of combining spatialized audio and 2D video projection in audio-visual systems ", W. de Bruijn and M. Boone, AES convention paper 5582, 10-13 May 2002, Münch In addition, subjective experiments on the effects of combining spatial audio and two-dimensional video projection in audiovisual systems are presented. In particular, it is emphasized that two in one Speakers standing at different distances from one another standing close to one another can be better understood by a viewer when the two consecutive persons are understood and reconstructed as different virtual sound sources with the help of wave field synthesis. In this case, subjective tests have shown that a listener can better understand and distinguish between the two speakers speaking at the same time.

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. As has been pointed out, 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.

As already stated, 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.

For multiple virtual sources in a wavefield synthesis setting, the contribution from 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., is calculated to then add up the component signals finally get the actual speaker signal. For example, in the case of three virtual sources, 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.

Due to loudspeaker array effects, below an aliasing frequency, the bass frequencies of, for example, 3dB per octave are summed. This gain is a consequence of the soundwave overlays for low tones in WFS playback. Therefore, for the WFS rendering below the aliasing frequency, a static filter is calculated which corrects the low frequency part, i. lowers. This filter is calculated as a function of the speaker distance and the adjustment of the aliasing frequency is currently done manually according to the sound impression of the Tonmeister.

It has been found that the manual adjustment is subjective and therefore expensive and has also led to strong quality fluctuations of the perceived sound.

The technical publications of E. Corteel, U. Horbach, RS Pellegrini: "Multichannel Inverse Filtering of Multiexciter Distributed Mode Loudspeakers for Wave Field Synthesis", AES convention paper 5611, 10-13 May, Münch and, U. Horbach, E. Corteel, D. de Vries: "Spatial Audio Reproduction Using Distributed Mode Loudspeaker Arrays", AES conference paper, June 1 - 3, St. Petersburg , as well as the patent DE 103 21 986 refer to amplitudes or Frequency manipulation to quality improvements in wave field synthesis.

The publication " Caractérisation et Extensions de la Wave Field Synthesis en conditions réelles ", E. Corteel, December 9, 2004, Doctoral thesis at the University of Par is, describes filters that cause an equalization of the sound signal above the aliasing frequency.

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.

This object is achieved by a device according to claim 1, a method according to claim 12 or a computer program according to claim 13.

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.

In one embodiment of the present invention, 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. Using the sound propagation laws, 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.

An advantage of embodiments according to the invention is that 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. In an implementation of 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.

Another advantage of the present invention is that 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.

According to the invention, it is thus considered that the characteristic of 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.

Fig. 1a

ein Blockschaltbild der erfindungsgemäßen Vor- richtung zum Aliasing-Filtern in einem Wellen- feldsynthesesystem, wobei die Komponentensignale gefiltert werden;

Fig. 1b

ein Blockschaltbild der erfindungsgemäßen Vor- richtung zum Aliasing-Filtern in einem Wellen- feldsynthesesystem, wobei die Audiosignale, die einer virtuellen Quelle zugeordnet sind, gefil- tert werden;

Fig. 2

ein Prinzipschaltbild in einer Wellenfeldsynthe- seumgebung, wie sie für die vorliegende Erfindung einsetzbar ist;

Fig. 3a

ein Blockschaltbild einer erfindungsgemäßen Ein- richtung zum Ermitteln der Aliasing-Frequenz;

Fig. 3b

eine Skizze zur Erläuterung des Ausbreitungsver- zögerungs- und Ausbreitungsskalierungswerts von den Lautsprechern zu dem Abhörpunkt;

Fig. 3c

ein Beispiel von 10 Lautsprechern, wo die Skalie- rungs- und Verzögerungswerte der einzelnen Laut- sprecher zu einem Zeitsignal am Abhörpunkt kombi- niert werden, aus welchem man nach der spektralen Darstellung die Aliasing-Frequenz ermittelt;

Fig. 4

ein Blockschaltbild zur Ermittlung der Aliasing- Frequenzen, die verschiedenen virtuellen Quellen entsprechen;

Fig. 5

ein Blockschaltbild zur Mittelung der Aliasing- Filtereigenschaften für verschiedene Abhörpunkte;

Fig. 6

ein Blockschaltbild für ein adaptives Filter für mehrere virtuelle Quellen; und

Fig. 7

ein prinzipielles Blockschaltbild eines Wellen- feldsynthesesystems mit Wellenfeldsynthesemodul und Lautsprecherarray in einem Vorführbereich.

Preferred embodiments of the present invention will be explained below in detail with reference to the accompanying drawings. Show it:

Fig. 1a

a block diagram of the inventive device for aliasing filtering in a wave field synthesis system, wherein the component signals are filtered;

Fig. 1b

a block diagram of the inventive device for aliasing filtering in a wave field synthesis system, wherein the audio signals that are assigned to a virtual source, be filtered;

Fig. 2

a schematic diagram in a Wellenfeldsynthe- seumgebung, as can be used for the present invention;

Fig. 3a

a block diagram of a device according to the invention for determining the aliasing frequency;

Fig. 3b

a sketch for explaining the propagation delay and propagation scaling value from the speakers to the listening point;

Fig. 3c

an example of 10 loudspeakers, where the scaling and delay values of the individual loudspeakers are combined into a time signal at the interception point, from which the aliasing frequency is determined after the spectral representation;

Fig. 4

a block diagram for determining the aliasing frequencies corresponding to different virtual sources;

Fig. 5

a block diagram for averaging the aliasing filter properties for different Abhörpunkte;

Fig. 6

a block diagram for an adaptive filter for multiple virtual sources; and

Fig. 7

a schematic block diagram of a wave field synthesis system with wave field synthesis module and speaker array in a demonstration area.

Before going into the present invention in detail, will be described below with reference to Fig. 7 the basic structure of a wave field synthesis system shown. The wave field synthesis system has a speaker array 700 placed with respect to a demonstration area 702. In detail, 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. As a movie theater, it is assumed that 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. Behind the viewer would then be the sub-array 700a, while to the left of the viewer would be the sub-array 700d, and to the right of the viewer would be the sub-array 700b. 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. type and location of the speaker with respect to the screening area 702, so of speaker information (LS information), and optionally with other inputs to calculate loudspeaker signals for the individual speakers 708, each of the audio tracks for virtual sources, which further assigned position information are derived according to the known wave field synthesis algorithms. 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. 1a 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. At the center of a wave field synthesis environment is 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. In one output, the wave-field synthesis module 100 provides the component signals 110 as well as the corresponding delay and scale values for the individual speakers. These data serve as input data to the device 120 for determining a source-specific aliasing filter characteristic (AFE) 130, which optionally also receives the information about the position of the interception point 125. 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.

In 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.

Like it out Fig. 2 As can be seen, the wave field synthesis module 100 receives an audio signal and position information from each virtual source. By way of example, 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. Using the data on the position of the speakers 106 as well as other inputs such. For example, in room acoustics 108, 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). From this, 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.

In Fig. 3b different speakers 708 are 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. 3c 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. 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. In this case, 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. Analogously, 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 Abhörpunkt 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. For each additional listening point, 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.

In the technical implementation, 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.

In order to achieve a sound improvement in WFS systems, 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.

Depending on the circumstances, 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. Generally, 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. In other words, 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.

Claims (13)

1. Device for aliasing correction in a wave field synthesis system with a wave field synthesis module (100) and an array of loudspeakers (708) for sound supply of a show area (702), wherein the wave field synthesis module (100) is configured to receive an audio signal (102) associated with a virtual sound source and source position information (104) associated with the virtual sound source, and to calculate, taking into account loudspeaker position information (106), component signals (110) for the loudspeakers (708) due to the virtual sound source, comprising:

   means for ascertaining (120) an aliasing filter property (130) specific for a virtual sound source using the source position information (104), wherein the means for ascertaining is configured to obtain, for the loudspeakers (708) in the array, wave field synthesis scaling values and wave field synthesis delay values (310) associated with the loudspeakers, and to ascertain the aliasing filter property (130) based on a listening point (P) in the show area (702) and the wave field synthesis scaling values and wave field synthesis delay values (310);

   characterized in that the device comprises an adaptive anti-aliasing filter (140) for adaptive filtering of the audio signal (102) associated with the virtual sound source or the component signals (110) associated with the virtual sound source, wherein the adaptive anti-aliasing filter (140) is adjusted according to the aliasing filter property (130) specific for the virtual sound source to effect an aliasing correction.
2. Device according to claim 1, wherein the means for ascertaining (120) is configured to calculate the aliasing filter property (130) using an impulse response for a channel between the virtual sound source and a listening point (P) in the reproduction space (702).
3. Device according to claim 1 or claim 2, wherein the means for ascertaining (120) is configured to ascertain propagation delay values and propagation scaling values (340) between the loudspeakers (708) and the listening point (P) to combine the wave field synthesis delay value and the propagation delay value for each loudspeaker to obtain a total delay value, to combine the wave field synthesis scaling value and the propagation scaling value for each loudspeaker to obtain a total scaling value, and to ascertain an impulse response to the virtual sound source and the listening point (P) using the total scaling values and the total delay values for the loudspeakers (708).
4. Device according to claim 3, wherein the means for ascertaining (120) is configured to translate a time signal with time values the time coordinates of which are defined by the total delay values, and the amplitudes of which are defined by the total scaling values, into a spectral representation and to ascertain, from the spectral representation, an aliasing filter frequency (390) as aliasing filter property (130).
5. Device according to claim 2, wherein the means for ascertaining (120) is configured to ascertain, from a spectral representation of the impulse response, an aliasing filter frequency (390) as aliasing filter property (130).
6. Device according to claim 4 or 5, wherein the means for ascertaining (120) is configured to ascertain, as aliasing filter frequency (390), a frequency which is in a range limited, towards low frequencies, by an increase of the spectral representation, and limited, towards higher frequencies, by a fluctuation of the spectral representation.
7. Device according to claim 6, wherein the means for ascertaining (120) is configured to select, as aliasing filter property (130), a frequency deviating by less than ± 25 % from a frequency value corresponding to a transitional value between an increase of the spectral representation and a fluctuation of the spectral representation.
8. Device according to claim 4 or 5, wherein the means for ascertaining (120) is configured to ascertain, for a virtual sound source, aliasing filter properties (130) for different listening points in the reproduction space (702), and to average the different aliasing filter properties to obtain the aliasing filter property specific for the virtual sound source.
9. Device according to one of claims 1 to 7, wherein the means for ascertaining (120) is configured to calculate different aliasing filter properties for virtual sound sources at different virtual positions, and wherein the adaptive anti-aliasing filter (140) is configured to filter the audio signals (102) associated with the virtual sound sources or the component signals (110) associated with the virtual sound sources using the different aliasing filter properties.
10. Device according to claim 9, wherein the adaptive anti-aliasing filter (140) is configured to filter the audio signals (102) associated with the virtual sound sources separately using the different aliasing filter properties to obtain aliasing-filtered audio signals, and wherein the wave field synthesis module (100) is configured to calculate the component signals (110) for each virtual sound source using the filtered audio signals, and to combine component signals belonging to a loudspeaker to obtain a loudspeaker signal (160) for the loudspeaker.
11. Device according to claim 9, wherein the adaptive anti-aliasing filter (140) is configured to filter component signals (110) calculated for a first virtual source using the anti-aliasing filter property (130) specific for the first virtual source so as to obtain first aliasing-filtered component signals for the first virtual source and to obtain, for a second virtual source, second aliasing-filtered component signals for the second virtual source, wherein the wave field synthesis module (100) is further configured to combine component signals (110), belonging to a loudspeaker, of the first aliasing-filtered component signals and the second aliasing-filtered component signals to obtain a loudspeaker signal (160) for the loudspeaker.
12. Method for aliasing filter correction in a wave field synthesis system with a wave field synthesis module (100) and an array of loudspeakers (708) for sound supply of a show area (702), wherein the wave field synthesis module (100) is configured to receive an audio signal (102) associated with a virtual sound source and source position information (104) associated with the virtual sound source, and to calculate, taking into account loudspeaker position information (106), component signals (110) for the loudspeakers due to the virtual sound source, comprising:

    ascertaining aliasing filter properties (130) specific for a virtual sound source using the source position information (104), wherein the step of ascertaining comprises obtaining wave field synthesis scaling values and wave field synthesis delay values (310) associated with the loudspeakers, so that the aliasing filter property (130) is ascertained based on a listening point (P) in the show area (702) and the wave field synthesis scaling values and wave field synthesis delay values (310); and

    characterized by the step of:

    adaptive filtering of the audio signals (102) associated with the virtual sound source or the component signals (110) associated with the virtual sound source, wherein the adaptive filtering is performed according to the aliasing filter property (130) specific for the source to effect an aliasing correction.
13. Computer program with a program code for performing the method according to claim 12, when the computer program runs on a computer.

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