EP1671516B1 - Device and method for producing a low-frequency channel - Google Patents

Abstract

The invention relates to a method for producing a low-frequency channel for a low-frequency speaker which is disposed in a predetermined low-frequency speaker position. According to said method, a plurality of audioobjects is provided (900), and an object position and an object description is associated with every audioobject. A calculation (906) of an audioobject calibration value for every audioobject is carried out in accordance with the object description so that an actual amplitude status at a reference playback position at least approximates a desired amplitude status. Every object signal is scaled with an associated audioobject scaling value (910) and the scaled object signals are summed up (914). The sum signal thereby obtained is used to derive a low-frequency channel for the low-frequency speaker and the corresponding low-frequency speaker is provided (918). The scaling of the individual object signals of the audioobjects renders the method independent of an actual situation of a multichannel projection system with respect to the number and density of the speakers and with respect to the size of the actually present projection zone.

Description

The present invention relates to generating one or more low frequency channels, and more particularly to generating one or more low frequency channels associated with a multi-channel audio system, such as a wave field synthesis system.

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, known as Wave Field Synthesis (WFS), was researched at the TU Delft and first introduced in the late 1980s (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. The first professional products are expected next year. In a few years, the first wave field synthesis applications for the consumer sector will be launched.

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 effort in the calculation therefore depends heavily on the number of strongly depending 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 leads to the space being designed, i. H. the designed scene is perceived as less authentic.

In the majority of cases, a concept is pursued which involves obtaining an overall acoustic impression of the visually depicted scenery. This works well with the term originating from the image design rewrite the "totals". This "total" sound impression usually remains constant over all the settings in a scene, although the optical view of things usually changes greatly. Thus, optical details are highlighted by appropriate settings or placed in the background. Also counter shots in the cinematic dialogue design are not reconstructed by the sound.

Therefore, there is a need to acoustically embed the viewer in an audiovisual scene. Here, the canvas or image surface forms the viewing direction and the perspective of the viewer. This means that the sound should track the image in the form that it always coincides with the viewed image. This becomes even more important for virtual studios, as there is typically no correlation between the moderation sound, for example, and the environment in which the presenter is currently located. In order to get an overall audiovisual impression of the scene, it is necessary to simulate a spatial impression matching the rendered picture. An essential subjective characteristic of such a sonic concept in this context is the location of a sound source, as perceived by a viewer, for example, a movie screen.

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 at 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 loudspeakers is used according to the WFS principle with an audio signal from a virtual source that has a specific delay and a has certain level, driven. 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 and then to compute a component signal for that single loudspeaker that this loudspeaker ultimately has to radiate to allow the listener to superpose the loudspeaker signal from one loudspeaker to the loudspeaker signals from the other active ones Speaker 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 wave-field 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. In the case of, for example, three virtual sources, the superimposition of the loudspeaker signals of all active loudspeakers on the listener would result in the listener not feeling 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 calculation of the component signals is in practice usually characterized by the fact that the audio signal associated with a virtual source, depending on the position of the virtual source and position of the speaker at a given time with a delay and a scaling factor is applied to a delayed and / or scaled audio signal of the to obtain a 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 loudspeakers are present in the loudspeaker array. 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.

Furthermore, various scenarios exist in that the loudspeaker array is only viewed when viewing a movie theater, e.g. B. is arranged on the side of the movie screen. In this case, the wave-field synthesis module would generate loudspeaker signals for these loudspeakers, the loudspeaker signals for these loudspeakers normally being the same as for corresponding loudspeakers in a loudspeaker array extending not only over the side of a cinema, for example, on which the screen is arranged, but also the left, right and behind the audience room is arranged. Of course, this "360 °" speaker array will provide a better approximation to an exact wave field than just a one-sided array, for example, in front of the viewers. However, the loudspeaker signals for the loudspeakers in front of the audience are the same in both cases. This means that a wave-field synthesis module typically does not receive feedback on how many speakers are present or whether it is a one-sided or multi-sided or even a 360 ° array or not. In other words, a wave field synthesizer calculates a loudspeaker signal for a loudspeaker based on the position of the loudspeaker and regardless of which other loudspeakers still exist or are absent.

Although this is an essential strength of the wave field synthesis algorithm to the extent that it is optimally modular adaptable to different circumstances by simply given the coordinates of the existing speakers in completely different screening rooms. The disadvantage, however, is that in addition to the tolerable under certain circumstances worse reconstruction of the current wave field significant level artifacts. Thus, for a real impression, it is not only crucial in which direction the virtual source is located with respect to the listener, but also how loudly the listener hears the virtual source, ie what level "arrives" at the listener due to a special virtual source. The incoming at a listener level, which is related to a considered virtual source, results from the superposition of the individual signals of the speakers.

For example, consider the case where there is a loudspeaker array of 50 loudspeakers in front of the listener, and that the virtual source audio signal by the wave field synthesizer into component signals for the 50 loudspeakers such that the audio signal is radiated from the 50 loudspeakers simultaneously with different delay and different scaling, a listener of the virtual source senses a level of the source resulting from the individual levels of the component signals of the virtual source individual loudspeaker signals.

If the same wavefield synthesizer is now used for a reduced array in which, for example, only 10 loudspeakers are in front of the listener, then it is readily apparent that the level of the signal from the virtual source resulting at the ear of the listener has decreased, since, so to speak, 40 component signals of the now missing speakers are "missing".

It may also occur the alternative case in which z. B. first left and right of the listener speakers are driven in a particular constellation in opposite phase, so that cancel the speaker signals from two opposite speakers due to a specific calculated by the wave field synthesis device delay. Is now in a reduced system z. For example, if you omit the speakers on one side of the listener, the virtual source suddenly seems much louder than it should be.

While static sources for level correction could still be thought of as constant factors, this solution is no longer sustainable if the virtual sources are not static but move. This is just an essential feature of wave field synthesis, that it can handle and especially moving virtual sources. Correction with a constant factor would be too short here, since the constant factor would be correct for one position, but would be artifact enhancing for another position of the virtual source.

Wave field synthesis devices are also capable of replicating several different types of sources. A prominent source form is the point source, where the level decreases proportionally 1 / r, where r is the distance between a listener and the position of the virtual source. Another source form is a source that emits plane waves. Here, the level remains constant regardless of the distance to the listener, since plane waves can be generated by point sources, which are arranged at an infinite distance.

According to wave field synthesis theory, in two-dimensional loudspeaker arrangements, the level change agrees with the natural level change depending on r up to a negligible error. Depending on the position of the source, however, different, sometimes considerable errors in the absolute level may result, resulting from the use of a finite number of loudspeakers instead of the theoretically required infinite number of loudspeakers, as stated above.

A further problem that exists in multi-channel playback systems, and in particular in wave field synthesis systems, which not only use, for example, five or seven speakers, but use a much higher number of speakers, is that the speakers can lead to considerable costs due to their high number. To reduce the cost of the speakers, the so-called subwoofer principle is used in such existing five-channel systems or seven-channel systems. The subwoofer principle is used in multi-channel playback systems to save expensive and large woofers. A low-frequency channel is used, which contains only music signals with frequencies lower than a cut-off frequency of about 120 Hz. This low-frequency channel drives a woofer with a large diaphragm area, which achieves high sound pressure, especially at low frequencies.

The subwoofer principle makes use of the fact that the human ear can localize low-frequency sounds in the direction very difficult to locate. In current systems, an extra woofer channel is already mixed in the sound mixing for a special speaker arrangement (spatial arrangement). Examples of such multi-channel playback systems are Dolby Digital, Sony SDDS and DTS. In these multi-channel formats, the subwoofer channel can be mixed regardless of the room size to be sounded, since the spatial relationships change only to scale. The speaker assembly remains the same to scale.

With wave field synthesis (WFS) a large audience area can be sonicated. Sound events can be reproduced in their spatial depth. For this purpose, the entire sound field of the individual sound events is reproduced in the audience area. This is done by a large number of speakers. For large installations, about 500 or more speaker systems are needed. If you wanted to equip each speaker system with a powerful woofer, very high costs would arise.

It has been mentioned that existing multi-channel formats require a special speaker arrangement to mix a particular subwoofer channel. However, the speaker assembly can be scaled without having to change the corresponding mixture. The ratio of the distances between the individual speakers to each other remains. However, all this is not possible with the WFS, as the number of speaker channels depends on the amount of WFS playback system surface to be sounded. Therefore, not the individual speaker channels can be stored, which would also be quite memory-consuming, if one considers systems with 500 or more audio channels. Therefore, only the virtual sound events to be simulated are stored. Only when playing The individual speaker channels are calculated using the WFS algorithm.

On the one hand, therefore, the number of speaker channels is related to the size of the audience area. In addition, the number of loudspeaker channels is determined by how densely the loudspeakers are distributed over the circumference of the surface to be sounded. The quality of the WFS playback system depends on this density. The volume is related to the number of loudspeaker channels and the density of the loudspeakers, since all loudspeaker channels add up to a wave field. The volume of a WFS system is therefore not readily predetermined. The volume of the subwoofer channel, however, is predetermined with the known parameters of the electric amplifier and the loudspeaker. It is therefore not possible to transfer a mixture of a subwoofer channel from a WFS system without error to a WFS system with different speaker density and number of speakers. The volume levels of the low frequency system on the one hand and the mid / high frequency system on the other hand would not match.

The object of the present invention is to provide a concept for generating a woofer channel in a multi-channel reproducing system which enables reduction of level artifacts.

This object is achieved by a device for generating a woofer channel according to claim 1 or a method for generating a woofer channel according to claim 25 or by a computer program according to claim 26.

The present invention is based on the finding that the woofer channel for a woofer or that several woof channels for several woofers in a multi-channel system is not already generated in a sound mixing process, which is independent of an actual Playback room takes place, but that reference is made to the actual playback room by the predetermined position of the woofer on the one hand and properties of audio objects, which are typically virtual sources, on the other hand also taken into account to produce the bass channel. In particular, audio objects are assumed, wherein an audio object is assigned an object description on the one hand and an object signal on the other hand. Depending on the object description, an audio object scaling value is calculated for each audio object signal, which is then used to scale each object signal, and then summed up the scaled object signals to obtain a summed signal. From the sum signal then the bass channel is derived, which is supplied to the woofer.

In the case of sources that emit plane waves, thus assuming a position at infinity, the virtual position of the source on the one hand and a reference reproduction position on the other, for which a reference volume is required, are irrelevant. However, this is not the case with conventional punctiform assumed sources, such as occur in a film setting when dialogues, etc. take place. In this case, scaling of the audio object signal originating from a virtual source located at a virtual position is made such that an actual volume or an actual amplitude state due to this virtual source at the reference reproduction position corresponds to a target amplitude state. The target amplitude state depends on the volume of the audio object signal associated with the virtual source and the distance between the virtual position and the reference playback position. This calculation of audio object scaling values is done for all virtual sources to then scale the audio object signals of each virtual source with the corresponding scaling value.

The scaled audio object signals are then summed to obtain a summed signal. From this summation signal is then derived again in the case in which only a single woofer, the bass channel. This can be done by simple low-pass filtering.

It should be noted that the low-pass filtering can already be performed with the still unscaled audio object signals, so that only low-pass signals are processed further, so that the sum signal is already the low-frequency channel itself.

According to the invention, however, it is preferred to carry out the extraction of the bass channel only after the summation of the scaled object signals in order to obtain the best possible approximation of the volume of the bass signals in the screening room on the one hand and the volume of the mid and high signals in the screening room on the other hand.

According to the invention, therefore, a subwoofer channel is not already mixed in the sound mixing process from the virtual sources, ie the sound material for the wave field synthesis. Instead, the mix automatically occurs when playing in the wave-field synthesis system, regardless of the size of the system and the number of speakers. The volume of the subwoofer signal depends on the number and on the circumference of the fringed area of the wave field synthesis system. Even prescribed loudspeaker arrangements no longer have to be complied with, since loudspeaker position and loudspeaker number are included in the generation of the bass channel.

The present invention is not limited to wave field synthesis systems, but can be generally applied to any multichannel reproduction system in which the mixing and generation, that is, the rendering, of the reproduction channels, that is, the speaker channels themselves, only at the actual playback takes place. Systems of this type are, for example, 5.1 systems, 7.1 systems, etc.

Preferably, the inventive low frequency channel generation is combined with a level artifact reduction to perform level corrections in a wave field synthesis system not only for low frequency channels but for all loudspeaker channels to be independent of the number and position of the speakers used with respect to the wave field synthesis algorithm used.

In preferred embodiments of the present invention in which only a single woofer channel and thus a single woofer is provided, the woofer will not be located in a reference display position for which optimum level correction is performed. In this case, according to the present invention, the sum signal is scaled by taking into account the position of the woofer using a speaker scale value to be calculated. This scaling will preferably be only amplitude scaling and no phase scaling, taking into account the fact that the ear does not have good localization at the low frequencies present in the low frequency channel, but only shows accurate amplitude / volume perception. Alternatively or additionally, a scaling can be used as scaling, if such is desired in an application scenario.

In the case of positioning multiple woofers, a separate woofer channel is created for each woofer. The bass channels of the individual woofers preferably differ only in their amplitude, but not in the signal itself. All woofers thus emit the same sum signal, but with different amplitude scaling, the amplitude scaling for a single one Subwoofer depending on the distance of the individual woofer to the reference reproduction point is done. In addition, the invention ensures that the overall volume of all superimposed low-frequency channels at the reference playback position is equal to the volume of the sum signal or the loudspeaker of the sum signal corresponds at least within a predetermined tolerance range. For this purpose, a separate loudspeaker scaling value is calculated for each individual low-frequency channel, with which the summation signal is then scaled accordingly in order to obtain the individual low-frequency channel.

The use of a subwoofer channel is particularly advantageous in that it leads to a significant price reduction, since the individual speakers z. B. a Wellenfeldsynthesesystems can be constructed much cheaper, since they must have no low-frequency characteristics. In contrast, only one or a few, such as three to four, Subwooferlautsprecher enough to realize the very low frequencies with high sound pressure through a correspondingly large membrane area.

The present invention is further advantageous in that the one or more low-frequency channels for any speaker assemblies and multi-channel formats can be generated automatically, which requires only a small overhead, especially in the context of a wave field synthesis system, since the wave field synthesis system anyway performs a level correction.

With respect to the necessary number of woofers as well as the optimal positioning of one or more low-frequency loudspeakers, reference is made to the technical literature, in particular from the Welti, Todd, "How Many subwoofer are Enough", 112 ^th AES Conv. Paper 5602, May 2002, Munich, Germany, Martens, "The impact of decorrelated low-frequency reproduction on auditory spatial imagery: are two subwoofers better than one? ", 16 ^th AES Conf. Paper, April 1999, Rovaniemi, Finland.

In a preferred embodiment of the present invention, in which only a single woofer is used, first the single volume and preferably also the delay of each virtual source, ie each sound object or audio object, is calculated relative to the reference playback position. Thereafter, the audio signal of each virtual source is scaled and delayed accordingly, and then summed up all the virtual sources. This will calculate the overall volume and delay of the subwoofer, depending on its distance from the reference point, if the subwoofer is not already in the reference point.

In the case of several subwoofers, it is preferred to first determine the individual volumes of all subwoofers depending on their distances to the reference point. In this case, it is preferable to adhere to the boundary condition that the sum of all subwoofer channels is equal to the reference volume at the reference reproduction position, which preferably corresponds to the midpoint of the wave field synthesis system. Thus, corresponding scaling factors are calculated per subwoofer, but initially again individual volume and delay of each virtual source relative to the reference point are determined. Then each virtual source is again scaled accordingly and optionally delayed, then summing all the virtual sources to the sum signal, which is then scaled by the individual scaling factors for each subwoofer channel to obtain the individual bass channels for the different woofers.

Fig. 1

ein Blockschaltbild der erfindungsgemäßen Vorrichtung zum Pegel-Korrigieren in einem Wellenfeldsynthesesystem;

Fig. 2

ein Prinzipschaltbild einer Wellenfeldsyntheseumgebung, wie sie für die vorliegende Erfindung einsetzbar ist;

Fig. 3

eine detailliertere Darstellung des in Fig. 2 gezeigten Wellenfeldsynthesemoduls;

Fig. 4

ein Blockschaltbild einer erfindungsgemäßen Einrichtung zum Ermitteln des Korrekturwerts gemäß einem Ausführungsbeispiel mit Nachschlagtabelle und gegebenenfalls Interpolationseinrichtung;

Fig. 5

ein weiteres Ausführungsbeispiel der Einrichtung zum Ermitteln von Fig. 1 mit Sollwert/Istwert-Ermittlung und anschließendem Vergleich;

Fig. 6a

ein Blockschaltbild eines Wellenfeldsynthesemoduls mit eingebetteter Manipulationseinrichtung zur Manipulation der Komponentensignale;

Fig. 6b

ein Blockschaltbild eines weiteren Ausführungsbeispiels der vorliegenden Erfindung mit einer vorgeschalteten Manipulationseinrichtung;

Fig. 7a

eine Skizze zur Erläuterung des SollAmplitudenzustands an einem Optimal-Punkt in einem Vorführbereich;

Fig. 7b

eine Skizze zur Erläuterung des Ist-Amplitudenzustands an einem Optimal-Punkt in dem Vorführbereich;

Fig. 8

ein prinzipielles Blockschaltbild eines Wellenfeldsynthesesystems mit Wellenfeldsynthesemodul und Lautsprecherarray in einem Vorführbereich;

Fig. 9

ein Blockschaltbild einer erfindungsgemäßen Vorrichtung zum Erzeugen eines Tieftonkanals;

Fig. 10

eine bevorzugte Ausgestaltung der Einrichtung zum Bereitstellen des Tieftonkanals für mehrere Tieftonlautsprecher; und

Fig. 11

eine schematische Darstellung eines Vorführbereichs mit einer Mehrzahl von Einzellautsprechern sowie zwei Subwoofern.

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

Fig. 1

a block diagram of the inventive device for level correction in a wave field synthesis system;

Fig. 2

a schematic diagram of a wave field synthesis environment, as it is applicable for the present invention;

Fig. 3

a more detailed representation of the wave field synthesis module shown in Figure 2;

Fig. 4

a block diagram of a device according to the invention for determining the correction value according to an embodiment with look-up table and optionally interpolation device;

Fig. 5

a further embodiment of the device for determining Figure 1 with setpoint / actual value determination and subsequent comparison.

Fig. 6a

a block diagram of a wave field synthesis module with embedded manipulation device for manipulating the component signals;

Fig. 6b

a block diagram of another embodiment of the present invention with an upstream manipulation device;

Fig. 7a

a sketch for explaining the target amplitude state at an optimum point in a demonstration area;

Fig. 7b

a sketch for explaining the actual amplitude state at an optimum point in the demonstration area;

Fig. 8

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

Fig. 9

a block diagram of an apparatus according to the invention for generating a bass channel;

Fig. 10

a preferred embodiment of the device for providing the woofer channel for multiple woofers; and

Fig. 11

a schematic representation of a demonstration area with a plurality of individual speakers and two subwoofers.

As already stated, the wave field synthesis algorithm calculates both volume and delay for each loudspeaker channel and each virtual source. The position of the individual loudspeaker must be known. For this purpose, it is preferred according to the invention to scale the overall volume of all loudspeakers in a reference point of the wave field synthesis reproduction system to an absolute reference volume, that is to say the desired amplitude state. This scaling of the individual audio object signals for the individual wave field synthesis system loudspeakers, ie the individual loudspeakers of the array, is based on the finding that the shortcomings of a wave field synthesis system with a (practically realizable) finite number of loudspeakers can at least be mitigated if a level correction is carried out, in that either the pre-wave field synthesis audio signal or the component signals for different loudspeakers originating from a virtual source are manipulated after wavefield synthesis using a correction value to detect a deviation between a desired amplitude state in a demonstration area and an actual area. Amplitude state in the demonstration area to reduce. The desired amplitude state results from the fact that depending on the position of the virtual source, and z. B. depending on a distance of a listener or an optimal point in a demonstration area to the virtual source and optionally taking into account the wave a target level is determined as an example of a desired amplitude state, and further that an actual level as an example an actual amplitude state is determined at the listener. While the target amplitude state is determined independently of the actual grouping and type of individual loudspeakers only on the basis of the virtual source or its position, the actual situation is calculated taking into account the positioning, type and control of the individual loudspeakers of the loudspeaker array.

Thus, the sound level at the ear of the listener can be determined at the optimum point within the performance area due to component signals of the virtual source radiated through a single speaker. Correspondingly, for the other component signals going back to the virtual source and being emitted via other loudspeakers, the level at the ear of the listener can also be determined at the optimum point within the presentation area, and then, by combining these levels, the actual actual level at the ear of the listener To receive the listener. For this purpose, the transfer function of each individual loudspeaker as well as the level of the signal on the loudspeaker and the distance of the listener in the considered point within the presentation area to the individual loudspeaker can be taken into account. For simpler embodiments, the transmission characteristic of the loudspeaker can be assumed to work as an ideal point source. For more complex implementations, however, the directional characteristic of the individual loudspeakers can also be taken into account.

A significant advantage of this concept is that in one embodiment, where sound levels are considered, only multiplicative scalings occur, in that for a quotient between the desired level and the actual level that gives the correction value, not the absolute level the listener or the absolute level of the virtual source is required. Instead, the correction factor depends only on the position of the virtual source (and hence the positions of the individual speakers) and the optimal point within the demonstration area. However, these quantities are fixed with respect to the position of the optimum point and the positions and transmission characteristics of the individual speakers and are not dependent on a piece being played.

Therefore, the concept can be implemented computationally efficiently as a look-up table, in that a look-up table is generated and used that includes position correction factor value pairs for all or a substantial portion of possible virtual positions. In this case, then no on-line setpoint determination, actual value determination and setpoint / actual value comparison algorithm is to be carried out. These algorithms, which can be computationally expensive, can be dispensed with if the look-up table is accessed on the basis of a position of a virtual source, in order to determine therefrom the correction factor valid for this position of the virtual source. In order to further increase the computational and storage efficiency, it is preferred to store only relatively rough-screened support value pairs for positions and associated correction factors in the table and correction factors for position values lying between two support values, one-sided, two-sided, linear, cubic etc. to interpolate.

Alternatively, it may also be useful in one case or another to use an empirical approach, in that level measurements are performed. In such a case, a virtual source with a particular calibration level would be placed at a particular virtual location. Then, for a real wave field synthesis system, a wave field synthesis module would compute the loudspeaker signals for the individual loudspeakers to eventually measure on the listener the actual level due to the virtual source. A correction factor would then be determined such that it at least reduces the deviation from the desired level to the actual level, or preferably brings it to zero. This correction factor would then be stored in the look-up table in association with the position of the virtual source so as to gradually generate, for many positions of the virtual source, the entire look-up table for a particular wave-field synthesis system in a particular presentation room.

For manipulation based on the correction factor, there are several possibilities. In one embodiment, it is preferable to manipulate the audio signal of the virtual source, such as recorded in an audio track coming from a recording studio, with the correction factor, and then to feed the manipulated signal into a wave field synthesis module. In a sense, this automatically means that all component signals going back to this manipulated virtual source are also weighted accordingly, compared to the case where no correction has been made in accordance with the present invention.

Alternatively, for certain applications, it may also be beneficial not to interfere with the source audio signal of the virtual source, but intervene in the component signals generated by the wave field synthesis module to manipulate these component signals all preferably at the same correction factor. It should be noted at this point that the correction factor is not necessarily identical for all component signals got to. However, this is largely preferred so as not to overly affect the relative scaling of the component signals required to reconstruct the actual wave situation.

One advantage is that with relatively simple measures, at least during operation, a level correction can be made to the effect that the listener, at least with regard to perceived by him volume of a virtual source not notice that not the actually required infinitely many speakers are present but only a limited amount of speakers.

Another advantage is that even if a virtual source moves at a constant distance (eg, from left to right) with respect to the viewer, this source for the viewer sitting in the middle in front of the screen, for example, always the same is loud and not even louder and once quieter, which would be the case without correction.

A further advantage is that it provides the option of offering lower cost wave field synthesis systems with fewer speakers, yet without any level artifacts, particularly for moving sources, thus acting as well for a listener as to the level problem more complex wave field synthesis systems with a high number of speakers. Also for holes in the array may be corrected to low levels according to the invention.

Before discussing the above-described preferred manner for level artifact correction in detail, the concept according to the invention for generating a low-frequency channel, which is either stand-alone, that is to say without level correction, will first be illustrated with reference to FIG Single loudspeaker can be used, or preferably combined with the concept of level artifact correction described later with reference to Figures 1-8, to also use the correction values used for level artifact correction of the individual speakers as audio object scaling values to be used in woofer channel generation.

Fig. 9 shows an apparatus for generating a woofer channel for a woofer located at a predetermined loudspeaker position. The device shown in FIG. 9 initially comprises a device 900 for providing a plurality of audio objects, an audio object having an audio object signal 902 and an audio object description 904 associated therewith. The audio object description will typically include an audio object position and possibly also the audio object type. Depending on the embodiment, the audio object description may also directly include an indication of the audio object volume. If this is not the case, then the audio object volume is easily calculated from the audio object signal itself, for example by sample-by-sample squaring and summation over a certain period of time. If the transfer function, frequency response, etc. of the individual loudspeakers considered or of the woofer are to be taken into consideration, this will likewise be possible by means of a simple table look-up or a correction factor, since in a reproduction system the electrical behavior of the loudspeaker or the signal / sound characteristic of the speaker is a stationary size.

The object description of the audio signal is supplied to a means 906 for calculating an audio object scaling value for each audio object. The individual audio object scaling values 908 are then applied to means 910 for scaling the object signals, as shown in FIG. 9. The means 906 for calculating the audio object scaling value is configured to calculate an audio object scaling value for each audio object depending on the object description. If it is a source that emits plane waves, then the audio object scaling value or correction factor will be equal to 1 because for such plane-wave audio objects, a spacing between the position of that object and the optimal reference playback position is insignificant because the virtual position in this case is adopted at infinity.

On the other hand, when the audio object is a point-source virtual source positioned at a virtual position, the audio object scale value becomes dependent on the volume of the object either in the object description or derived from the object signal and the distance between the virtual object The position of the audio object and the reference playback position is calculated.

In particular, it is preferable to calculate the audio object scaling value to be considered to be based on a target amplitude state in the demonstration area, wherein the target amplitude state depends on a position of the virtual source or a kind of the virtual source the correction value is further based on an actual amplitude state in the demonstration area based on the component signals for the individual speakers due to the considered virtual source. The correction value is thus calculated so that a deviation between the desired amplitude state and the actual amplitude state is reduced by a manipulation of the audio signal assigned to the virtual source using the correction value. After scaling the object signals performed by the device 910 to obtain the scaled object signals 912, they become one device 914 for summing to produce a sum signal 916.

As has been stated, it is preferred to also consider, prior to summation by means 914, a delay possibly due to different virtual positions, so that the individual audio object signals, which are present as sequences of samples, are shifted in respect of a time reference, to sufficiently account for propagation time differences of the sound signal from the virtual position to the reference playback position. After scaling and taking into account the delay, the scaled and correspondingly delayed object signals are then sampled by means 914 to obtain a sum signal having a sequence of sum signal samples, designated 916 in FIG. This sum signal 916 is supplied to a device 918 for providing the woofer channel for the one or more subwoofers, which outputs the subwoofer signal or the woof channel 920 on the output side.

As has been pointed out, the sound signal emitted by a woofer is not a full bandwidth sound signal but with an upwardly limited bandwidth. In one embodiment, it is preferred that the cut-off frequency of the sound signal emitted by a woofer be less than 250 Hz, and preferably even only 125 Hz. The band limitation of this sound signal can be done at different locations. A simple measure is to provide the woofer with a full bandwidth excitation signal which is then band limited by the woofer itself, as it only translates low frequencies into sound signals but suppresses high frequencies.

Alternatively, however, the band limitation may also be performed in the means 918 for providing the bass channel, by low-pass filtering the signal there prior to digital-to-analog conversion, this low-pass filtering, since it can be performed on the digital side, is preferred so that clear conditions exist independently of the actual implementation of the subwoofer. Alternatively, however, the low-pass filtering may already occur prior to means 910 for scaling the object signals, so that the operations performed by means 910, 914, 918 are now performed with low-pass signals rather than full-bandwidth signals.

According to the invention, however, it is preferred to perform the low-pass filtering in the device 918 so that the calculation of the audio object scaling values, the scaling of the object signals and the summation with full bandwidth signals is performed to best match the loudspeakers between woofer on the one hand and midtone and treble on the other hand sure. In other words, it is preferred to carry out as many operations as possible in parallel to the determination of the actual loudspeaker signals for the loudspeakers in the wave field synthesis array and only very late to perform a "splitting off" of the low-frequency channel.

Fig. 10 shows a preferred embodiment of the means 918 for providing now multiple low frequency channels for multiple subwoofers. Before discussing FIG. 10 in detail, the geometric situation will first be illustrated with reference to FIG. 11. FIG. 11 schematically shows a wave field synthesis system with a plurality of individual loudspeakers 808. The single speakers 808 form an array 800 of single speakers enclosing the demonstration area. Preferably within the demonstration area is the reference reproduction position or reference point 1100.

In Fig. 11 is also schematically an audio object 1102, which is referred to as a "virtual sound object". The virtual sound object 1102 includes an object description representing a virtual position 1104. On the basis of the coordinates of the reference point 1100 and the coordinates of the virtual position 1104, which can be optionally converted accordingly, the distance D of the virtual sound object 1102 from the reference reproduction position 100 can be determined. On the basis of this distance D, a simple audio object scaling value calculation can already be carried out, namely on the basis of the law which will be explained in detail later in FIG. 7a. FIG. 11 further shows a first woofer 1106 at a first predetermined loudspeaker position 1108 and a second woofer 1110 at a second woofer position 1112. As shown in FIG. 11, the second subwoofer 1110 and each other are not shown in FIG additional subwoofer optional. The first subwoofer 1106 has a distance d1 from the reference point 1100, while the second subwoofer 1110 has a distance d2 from the reference point. Similarly, a subwoofer n (not shown in FIG. 11) has a distance dn from the reference point 1100.

Referring again to Fig. 10, the means 918 for providing the bass channel is adapted to include, in addition to the sum signal 916, denoted by s in Fig. 10, also the distance d1 of the woofer 1, designated 930, to the distance d2 of the woofer 2 labeled 932 and the distance dn of the woofer n designated 934. On the output side, means 918 provides a first woofer channel 940, a second woofer channel 942, and an nth woofer channel 944. From Fig. 10 it can be seen that all woof channels 940, 942, 944 are weighted versions of the sum signal 916, the respective weighting factors being a ₁ , a ₂ , ..., a _{n are} designated. The individual weighting factors a ₁ , a ₂ , a _n depend on the one hand on the distances 930-934 and on the other hand from the general boundary condition that the volume of the bass channels at the reference point 100 is equal to the reference volume, that is, the target amplitude state for the bass channel at the reference playback position 1100 (FIG. 11). After all subwoofers are away from the reference point 1100, the sum of the loudspeaker scaling values a ₁ , a ₂ , a _{n will be} greater than 1 to account for the attenuation of the bass channels on the way from the corresponding subwoofer to the reference point. If only a single woofer (eg, 1106) is provided, the scaling factor a _{1 will} also be greater than 1, while no further scaling factors are to be calculated, as there is only a single woofer.

Referring now to FIGS. 1-8, there is shown a level artifact correction device for the loudspeaker array 800 in FIGS. 8 and 11, respectively, which are preferably combined with the inventive low frequency channel calculation as illustrated in FIGS. 9-11 can.

Before discussing the present invention in detail, the basic structure of a wave field synthesis system is shown below with reference to FIG. The wave field synthesis system has a speaker array 800 placed relative to a demonstration area 802. Specifically, the loudspeaker array shown in Fig. 8, which is a 360 ° array, includes four array sides 800a, 800b, 800c, and 800d. If the demonstration area 802 z. As a cinema, it is assumed that the cinema screen is on the same side of the screening area 802 on which the sub-array 800c 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 802, would see to the front, ie to the screen. Behind the viewer would then be the sub-array 800a, while left of While the partial array 800d would be located by the viewer and the sub-array 800b would be to the right of the viewer. Each loudspeaker array consists of a number of different individual loudspeakers 808, each of which is driven by its own loudspeaker signals provided by a wave-field synthesis module 810 via a data bus 812 shown only schematically in FIG. 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 demonstration area 802, so of speaker information (LS information), and optionally with other inputs to calculate loudspeaker signals for the individual speakers 808, 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 therefore be located anywhere in the demonstration area 802. 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 region 802, it is preferred to use the optimum point or the optimal line in the center or center of gravity of the wave field synthesis system which is passed through the loudspeaker sub-arrays 800a, 800b, 800c , 800d is defined to assume.

A more detailed representation of the wave field synthesis module 800 is given below with reference to FIGS. 2 and 3 with reference to the wave field synthesis module 200 in FIG. 2 or to the arrangement shown in detail in FIG. 3.

Fig. 2 shows a wave field synthesis environment in which the present invention can be implemented. At the center of a wave field synthesis environment is a wave field synthesis module 200 comprising various inputs 202, 204, 206 and 208 as well as various outputs 210, 212, 214, 216. Via inputs 202 to 204, different audio signals for virtual sources are supplied to the wave field synthesis module. Thus, the input 202 receives z. B. an audio signal of the virtual source 1 and associated position information of the virtual source. For example, in a cinema setting, the audio signal would be 1 z. For example, the language of an actor moving from a left side of the screen to a right side of the screen, and possibly additionally away from the viewer or toward the viewer. The audio signal 1 would then be the actual language of this actor, while the position information as a function of time represents the current position of the first actor in the recording setting. By contrast, the audio signal n would be the language of, for example, another actor moving in the same way or different from the first actor. The current position of the other actor to whom the audio signal n is assigned is notified to the wave field synthesis module 200 by position information synchronized with the audio signal n. In practice, various virtual sources exist depending on the recording setting, wherein the audio signal of each virtual source is supplied to the wave field synthesis module 200 as a separate audio track.

As stated above, a wave field synthesis module feeds a plurality of loudspeakers LS1, LS2, LS3, LSm by outputting loudspeaker signals via the outputs 210 to 216 to the individual loudspeakers. The wave field synthesis module 200 is informed via the input 206 of the positions of the individual speakers in a playback setting, such as a movie theater. In the cinema hall, there are many individual loudspeakers grouped around the cinema viewer, preferably in arrays are arranged so that both in front of the viewer, so for example behind the screen, as well as behind the viewer and right and left of the viewer speakers are located. Furthermore, the Wellenfeldsynthesemodul 200 still other inputs can be communicated, such as information about the room acoustics, etc., to be able to simulate the actual prevailing during the recording setting room acoustics in a cinema.

Generally speaking, the loudspeaker signal supplied to the loudspeaker LS1 via the output 210, for example, will be a superposition of component signals of the virtual sources, in that the loudspeaker signal for the loudspeaker LS1 is a first component originating in the virtual source 1, a second one Component that originates from the virtual source 2 and an nth component that goes back to the virtual source n. The individual component signals are superimposed linearly, that is to say added according to their calculation, in order to simulate the linear superposition at the ear of the listener, who in a real setting will hear a linear superimposition of the sound sources that he can perceive.

Referring now to FIG. 3, a more detailed embodiment of the wave-field synthesis module 200 will be set forth below. The wave field synthesis module 200 has a highly parallel construction in that, starting from the audio signal for each virtual source and based on the position information for the corresponding virtual source, first delay information V _i and scale factors SF _i calculated from the position information and the position of the currently considered one Speaker, z. B. the speaker with the ordinal number j, so LSj depend. The calculation of a delay information V _i and a scaling factor SF _i on the basis of the position information of a virtual source and the position of the considered loudspeaker j occurs by known algorithms implemented in devices 300, 302, 304, 306. On the basis of the delay information V _i (t) and SF _i (t) and on the basis of the individual virtual source associated audio signal AS _i (t) is for a current time t _A a discrete value AW _i (t _A ) for the _Computed signal K _ij calculated in a final received speaker signal. This is done by means 310, 312, 314, 316, as shown schematically in FIG. Fig. 3 also shows a kind of "flash photography" at time t _A for the individual component signals. The individual component signals are then summed by a summer 320 to determine the discrete value for the current time t _{A of} the loudspeaker signal for the loudspeaker j, which is then used for the output (for example the output 214 if the loudspeaker j is the loudspeaker LS3). to which speaker can be supplied.

As can be seen in Fig. 3, first each value is calculated for each virtual source individually, based on a delay and scaling with a scaling factor at a current time, after which all the component signals for a loudspeaker are summed due to the different virtual sources. For example, if only one virtual source were present, the summer would be omitted and the signal applied to the output of the summer in FIG. For example, correspond to the signal output from the device 310 when the virtual source 1 is the only virtual source.

It should be noted that at the output 322 of Figure 3, the value of a loudspeaker signal is obtained which is a superposition of the component signals for that loudspeaker due to the various virtual sources 1, 2, 3, ..., n. An arrangement as shown in FIG. 3 would, in principle, be provided for each loudspeaker 808 in the wave field synthesis module 810, unless that, which is preferred for practical reasons, always z. B. 2, 4 or 8 speakers are driven together with the same speaker signal.

Fig. 1 is a block diagram of the level correction apparatus of the present invention in a wave field synthesis system set forth with reference to Fig. 8; The wave field synthesis system includes the wave field synthesis module 810 and the speaker array 800 for sounding the presentation area 802, the wave field synthesis module 810 configured to receive source signal information associated with a virtual sound source and source position information associated with the virtual sound source, and component signals for the speakers based on the virtual speaker To calculate source. The apparatus according to the invention comprises first a means 100 for determining a correction value based on a target amplitude state in the demonstration area, wherein the target amplitude state depends on a position of the virtual source or a type of the virtual source, and wherein the correction value is further on a Actual amplitude state is based in the demonstration area, which depends on the component signals for the speakers due to the virtual source.

The device 100 has an input 102 for obtaining a position of the virtual source, when e.g. B. has a point source characteristic, or to obtain information about a type of source when the source z. B. is a source for generating plane waves. In this case, the distance of the listener from the source to determine the actual state is not necessary, since the source is anyway infinitely distant from the listener due to the generated plane waves in the model and has a position independent level. The device 100 is designed to output on the output side a correction value 104, which is assigned to a device 106 for manipulating manipulating an audio signal associated with the virtual source (obtained via an input 108) or manipulating component signals for the speakers due to a virtual source (obtained via an input 110). If the alternative of manipulating the audio signal provided via the input 108 is performed, an engineered audio signal results at an output 112, which is then fed to the wave field synthesis module 200 instead of the original audio signal provided at the input 108 to generate the individual loudspeaker signals 210, 212, ..., 216.

If, on the other hand, the other alternative was used for manipulation, namely the somewhat embedded manipulation of the component signals obtained via the input 110, manipulated component signals are obtained on the output side which still need to be summed loudspeaker wise (device 116), if appropriate manipulated component signals from other virtual sources provided via further inputs 118. On the output side, the device 116 again provides the loudspeaker signals 210, 212, ..., 216. It should be noted that the alternatives of the upstream manipulation (output 112) or the embedded manipulation (output 114) shown in FIG. 1 are used alternatively to each other can. Depending on the embodiment, however, there may also be cases in which the weighting factor or correction value provided via the input 104 into the device 106 is split, as it were, so that an upstream manipulation and partly an embedded manipulation is partially performed.

Thus, with reference to Figure 3, the upstream manipulation would be to manipulate the audio signal of the virtual source fed to a device 310, 312, 314, and 316, respectively, prior to its injection.

The embedded manipulation, on the other hand, would be that the component signals output by means 310, 312, 314, and 316, respectively, are manipulated prior to their summation to obtain actual loudspeaker signal.

These two possibilities, which can be used either alternatively or cumulatively, are shown in FIGS. 6a and 6b. Thus, Fig. 6a shows the embedded manipulation by the manipulation device 106, which is drawn in Fig. 6a as a multiplier. A wave field synthesis device, which consists for example of the blocks 300, 310 and 302, 312, and 304, 314 and 306 and 316 of FIG. 3, provides the component signals K ₁₁ , K ₁₂ , K ₁₃ for the loudspeaker LS1 and the component signals K _n1 , K _n2 and K _n3 for the loudspeaker LSn.

In the notation selected in Figure 6a, the first index of K _ij indicates the speaker, and the second index indicates the virtual source from which the component signal originates. The virtual source 1, for example, is expressed in the component _signal K _l1 , ..., K _nl . In order to selectively influence the level of the virtual source 1 depending on the position information of the virtual source 1 (without affecting the levels of the other virtual sources), in the embedded manipulation shown in Fig. 6a, a multiplication of the component signals belonging to the source 1 becomes , So the component signals whose index j points to the virtual source 1, take place with the correction factor F ₁ . In order to carry out a corresponding amplitude or level correction for the virtual source 2, all the component signals which go back to the virtual source 2 are multiplied by a correction factor F ₂ determined for this purpose. Finally, the component signals going back to the virtual source 3 are also weighted by a corresponding correction factor F ₃ .

It should be noted that the correction factors F ₁ , F ₂ and F ₃ , if all other geometric parameters equal are only dependent on the location of the corresponding virtual source. Would thus all three virtual sources z. For example, if point sources (that is, of the same kind) are at the same position, the correction factors for the sources would be identical. This law will be explained in more detail with reference to FIG. 4, since it is possible to use a lookup table with position information and respective associated correction factors, which must be created at some point, but which can be accessed quickly during operation, without having to In operation, a setpoint / actual value calculation and comparison operation must be performed constantly, which is also possible in principle.

Fig. 6b shows the inventive alternative to source manipulation. The manipulation device is connected upstream of the wave field synthesis device and is operative to correct the audio signals of the sources with the corresponding correction factors to obtain manipulated audio signals for the virtual sources, which are then supplied to the wave field synthesis device to obtain the component signals, which are then output from the sources respective component summation means are accumulated to obtain the loudspeaker signals LS for the respective loudspeakers, such as the loudspeaker LS _i .

In a preferred embodiment of the present invention, the means 100 for determining the correction value is formed as a look-up table 400 which stores position correction factor value pairs. The device 100 is preferably also provided with an interpolator 402 in order, on the one hand, to keep the table size of the lookup table 400 in a limited frame and, on the other hand, also for current positions of a virtual source, which are fed into the interpolator via an input 404, at least below Using one or more adjacent position correction factor value pairs stored in the look-up table, which are supplied to the interpolator 402 via an input 406 to produce an interpolated current correction factor at an output 408. However, in a simpler version, the interpolator 402 may also be omitted so that the means 100 for determining FIG. 1 performs direct access to the look-up table using position information supplied at an input 410 and provides an appropriate correction factor at an output 412. If the current position information associated with the audio track of the virtual source does not correspond exactly to position information found in the look-up table, then the look-up table may still be assigned a simple round-up / round-down function to the nearest one stored in the table Support value instead of the current base value.

It should be noted here that different tables can be created for different types of sources, or that a position is assigned not just one correction factor but several correction factors, each correction factor being linked to a source type.

Alternatively, instead of the look-up table or "populating" the look-up table in FIG. 4, the means for determining may be configured to actually perform a setpoint-actual value comparison. In this case, the device 100 of FIG. 1 includes a desired amplitude state determination device 500 and an actual amplitude state determination device 502 to provide a desired amplitude state 504 and an actual amplitude state 506, which are supplied to a comparison device 508 For example, a quotient of the desired amplitude state 504 and the actual amplitude state 506 is calculated to produce a correction factor 510 that is applied to the device 106 for manipulating, which is shown in Fig. 1, is supplied for further use. Alternatively, the correction value can also be stored in a lookup table.

The desired amplitude state calculation is designed to determine a target level at the optimum point for a virtual source configured at a specific position or in a specific manner. For the target amplitude state calculation, the target amplitude state determination device 500 does not need any component signals, of course, since the target amplitude state is independent of the component signals. Component signals are, however, as shown in FIG. 5, the actual amplitude determining means 502 which further depending on the embodiment also information about the speaker positions and information about speaker transfer functions and / or information on directional characteristics of the speakers can get to a To determine the actual situation as well as possible. Alternatively, the actual amplitude state determination device 502 can also be embodied as an actual measurement system in order to determine an actual level situation at the optimum point for specific virtual sources at specific positions.

Hereinafter, referring to FIGS. 7a and 7b, reference will be made to the actual amplitude state and the target amplitude state, respectively. FIG. 7 a shows a diagram for determining a desired amplitude state at a predetermined point, which is designated "optimal point" in FIG. 7 a, and which lies in the demonstration area 802 of FIG. 8. In FIG. 7a, merely by way of example, a virtual source 700 is shown as a point source, which generates a sound field with concentric wavefronts. Further, due to the audio signal for the virtual source 700, the level L _{v of} the virtual source 700 is known. The target amplitude state or, when the amplitude state is a level state, the target level at the point P in the demonstration area becomes easily obtained by the level L _P at the point P is equal to the quotient of L _v and a distance r has the point P to the virtual source 700. The desired amplitude state can thus be easily determined by calculating the level L _{v of} the virtual source and by calculating the distance r from the optimal point to the virtual source. Typically, to compute the distance r, a coordinate transformation of the virtual coordinates into the coordinates of the presentation space or a coordinate transformation of the presentation space coordinates of the point P into the virtual coordinates must be performed, as is known to those skilled in the art of wave field synthesis.

If, on the other hand, the virtual source is an infinitely distant virtual source which generates plane waves at the point P, then the distance between the point P and the source is not needed to determine the desired amplitude state, since this point goes to infinity anyway. In this case, only information about the type of source is needed. The desired level at point P is then equal to the level associated with the planar wave field generated by the infinitely distant virtual source.

Fig. 7 is a diagram for explaining the actual amplitude state. In particular, in Fig. 7b different speakers 808 drawn, all of which are fed with its own speaker signal, the z. B. generated by the wave field synthesis module 810 of FIG. 8. Further, each speaker is modeled as a point source that outputs a concentric wave field. The law of the concentric wave field is again that the level decreases according to 1 / r. Thus, in order to calculate the actual amplitude state (without measurement), the signal generated by the speaker 808 directly on the speaker diaphragm or the level of this signal may be determined on the basis of the speaker characteristics and the component signal in the loudspeaker signal LSn, which goes back to the considered virtual source. Further, based on the coordinates of the point P and the location information on the position of the loudspeaker LSn, the distance between P and the loudspeaker diaphragm of the loudspeaker LSn can be calculated so that a level for the point P can be obtained on the basis of a component signal corresponding to the virtual source under consideration goes back and has been sent from the speaker LSn.

A corresponding procedure may also be performed for the other loudspeakers of the loudspeaker array, so that the point P results in a number of "sub-level values" representing a signal contribution of the considered virtual source which has passed from the individual loudspeakers to the listener at point P. , By summing these sub-level values, the entire actual amplitude state is then obtained at the point P which, as stated, can then be compared to the desired amplitude state by a correction value, which is preferably multiplicative, but in principle additive or subtractive could get.

According to the invention, the desired level for one point is thus calculated on the basis of certain source forms, that is to say the desired amplitude state. It is preferred that the optimal point or point in the demonstration area being viewed is usefully located in the middle of the wave field synthesis system. It should be noted at this point that an improvement is already achieved even if the point which was used to calculate the setpoint amplitude state does not coincide directly with the point which was used to determine the actual amplitude state. Since the aim is to achieve the best possible artifact reduction for the largest possible number of points in the projection area, it is sufficient in principle for a desired amplitude state to be present for any point in the projection area is determined, and that an actual amplitude state is also determined for any point in the demonstration area, but it is preferred that the point to which the actual amplitude state is related, in a zone around the point for which the Desired amplitude state has been determined, this zone is preferably less than 2 meters for normal cinema applications. For best results, these points should essentially coincide.

Thus, according to the present invention, after calculating the individual levels of the loudspeakers, according to conventional wave field synthesis algorithms, the level practically generated by superposition is calculated at this point, which is called the optimum point in the demonstration area. The levels of the individual speakers and / or sources are then corrected according to the invention with this factor. For computational time-efficient applications, it is particularly preferred to compute and store correction factors once for all positions in a particular array, and then access the table in operation to achieve computational time savings.

Attention is drawn in particular to FIG. 6b, in which the means 914 for summation is drawn in order to deliver the sum signal 916 on the output side, while on the input side the scaled object signals 912 are obtained which, as can be seen from FIG Scaling of the source signals of the sources 1, 2, 3 with the corresponding audio object scaling values or correction values F1, F2, F3 are obtained. It should also be noted that for the present invention of woofer generation, the version shown in Fig. 6b is preferred, in which already a scaling or manipulation or correction on audio object signal level and not on component level, as shown in Fig. 6a, is carried out. Nevertheless, the concept of component-level correction shown in FIG According to the concept of low-frequency channel generation according to the invention, at least the calculation of the audio object scaling values F1, F2,..., Fn must be performed only once.

According to the invention, the scaling of the subwoofer channel is thus similar to the scaling of the overall volume of all loudspeakers in the reference point of the wave field synthesis display system. The inventive method is thus suitable for any number of subwoofer loudspeakers, which are all scaled so that they reach a reference level in the center of the wave field synthesis system. The reference volume depends only on the position of the virtual sound source. With the known dependencies of distance of the sound object to the reference point and the associated attenuation of the volume, the single volume of the respective sound object for each subwoofer channel is preferably calculated. The delay of each source is calculated from the distance of the virtual source to the reference point of the volume scaling. Each subwoofer speaker reflects the sum of all converted sound objects. How the single subwoofer speakers add up depends on their position. The preferred positioning of subwoofer speakers and the selection of the number of required subwoofers are the already mentioned specialist publications Welti, Todd, "How Many Subwoofers are Enough", 112 ^th AES Conv. Paper 5602, May 2002, Munich, Germany, Martens, "The impact of decorrelated low-frequency reproduction on auditory spatial imagery: are two subwoofers better than one?", 16 ^th AES Conf. Paper, April 1999, Rovaniemi, Finland.

Depending on the circumstances, the method according to the invention for generating a bass channel, as illustrated with reference to FIG. 9, can be implemented in hardware or in software.

Depending on the circumstances, the level correction method according to the invention, as shown in FIG. 1, can be implemented in hardware or in software. The implementation may be on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals, which may interact with a programmable computer system such that the method is executed. In general, the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out the method for level correction 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 (26)

1. An apparatus for generating a low-frequency channel (940, 942, 944) for a low-frequency loudspeaker (1106, 1110), comprising:

   a means (900) for providing a plurality of audio objects, an audio object having an object signal and an object description associated with it;

   a means (906) for calculating an audio object scaling value for each audio object in dependence on the object description (904);

   a means (910) for scaling each object signal with an associated audio object scaling value (908) so as to obtain a scaled object signal (912) for each audio object;

   a means (914) for summing the scaled object signals so as to obtain a composite signal (916); and

   a means (918) for providing the low-frequency channel (920, 940, 942, 944) for the low-frequency loudspeaker (1106, 1110) on the basis of the composite signal (916).
2. The apparatus as claimed in claim 1, wherein the low-frequency loudspeaker is arranged at a predetermined loudspeaker position (1108, 1112), the predetermined loudspeaker position (1108) differing from a reference playback position (100), and
   wherein the means (918) for providing the low-frequency channel is configured to calculate a loudspeaker scaling value for the low-frequency loudspeaker in dependence on the predetermined loudspeaker position (1108), so that a low-frequency signal at the reference playback position (1100) has a loudness which corresponds to a loudness of the composite signal (916) within a predetermined tolerance range, and
   wherein the means (918) for providing is further configured to scale the composite signal (916) with the loudspeaker scaling value so as to generate the low-frequency channel (920, 940, 942, 944).
3. The apparatus as claimed in claims 1 or 2, wherein each object signal is a low-frequency signal having an upper cutoff frequency smaller than or equal to 250 Hz.
4. The apparatus as claimed in claims 1 or 2, wherein the composite signal (916) has an upper cutoff frequency higher than 8 kHz, and
   wherein the means (918) for providing the low-frequency channel is configured to conduct a low-pass filtering at a cutoff frequency smaller than or equal to 250 Hz.
5. The apparatus as claimed in any of the previous claims,
   wherein an audio object of the plurality of audio objects includes an object description which includes an audio object position, and
   wherein the means (906) for calculating an audio object scaling value for the audio object is configured to perform the audio object scaling value in dependence on the audio object position of the audio object and on a reference playback position (1100), and in dependence on an object loudness associated with the audio object.
6. The apparatus as claimed in any of the previous claims,
   wherein a plurality of low-frequency channels for a plurality of low-frequency loudspeakers may be generated at predetermined low-frequency loudspeaker positions, and
   wherein the means (918) for providing is configured to calculate a loudspeaker scaling value for each low-frequency loudspeaker in dependence on the position of a low-frequency loudspeaker and in dependence on a number of further low-frequency loudspeakers,
   so that a low-frequency signal which is superposition of output signals of all low-frequency loudspeakers at the reference position (1100) has a loudness which corresponds to a loudness of the composite signal (916) within a predetermined tolerance range.
7. The apparatus as claimed in any of the previous claims,
   wherein the means (906) for calculating audio object scaling values is further configured to calculate an audio object delay value for each audio object, the former depending on an object position and a reference playback position, and
   wherein the means (914) for summing is configured to delay each object signal or each scaled object signal by the respective audio object delay value prior to summing.
8. The apparatus as claimed in any of the previous claims,
   wherein the means (918) for providing is configured to calculate, for a low-frequency loudspeaker, a low-frequency loudspeaker delay value which depends on a distance of the low-frequency loudspeaker from the reference playback position, and
   wherein the means (918) for providing is further configured to take into account the low-frequency loudspeaker delay value when providing the low-frequency channel.
9. The apparatus as claimed in claim 2, wherein several low-frequency loudspeakers are provided, and wherein the means (918) for providing is further configured to calculate the loudspeaker scaling values such that for each low-frequency loudspeaker, a loudspeaker scaling value in accordance with the following equation is obtained: $$\mathrm{(}{\mathrm{a}}_{\mathrm{1}}\mathrm{+}{\mathrm{a}}_{\mathrm{2}}\mathrm{+}\dots \mathrm{+}{\mathrm{a}}_{\mathrm{n}}\mathrm{)}\mathrm{\cdot }\mathrm{s}\mathrm{=}\mathrm{LSref}\mathrm{,}$$

   

   
   wherein LSref is a reference loudness at a reference playback position (1100), wherein s is the composite signal (916), wherein a₁ is the loudspeaker scaling value of a first low-frequency loudspeaker, wherein a₂ is a loudspeaker scaling value of a second low-frequency loudspeaker, and wherein a_n is a loudspeaker scaling value of an n^th low-frequency loudspeaker.
10. The apparatus as claimed in claim 9, wherein the loudspeaker scaling value of a low-frequency loudspeaker depends on a distance of the low-frequency loudspeaker from the reference playback position (1100).
11. The apparatus as claimed in any of the previous claims, which is configured to operate in a wave-field synthesis system with a wave-field synthesis module (810) and an array (800) of loudspeakers (808) for exposing a presentation area (802) to sound, the wave-field synthesis module being configured to receive an audio signal associated with a virtual sound source, as well as source position information associated with the virtual sound source, and to calculate component signals for the loudspeakers due to the virtual source while taking into account loudspeaker position information, and
    wherein the means (906) for calculating the audio object scaling values (908) includes a means (100) for determining a correction value as an audio object scaling value, the means (100) for determining being configured to calculate the audio object scaling value such that it is based on a target amplitude state in the presentation area, the target amplitude state depending on a position of the virtual source or a type of the virtual source, and which audio object scaling value is further based on an actual amplitude state in the presentation area which is based on the component signals for the loudspeakers due to the virtual source.
12. The apparatus as claimed in claim 11, wherein the means (100) for determining the correction value (104) is configured to calculate (500) the target amplitude state for a predetermined point in the presentation area, and to determine (502) the actual amplitude state for a zone in the presentation area which equals the predetermined point or extends around the predetermined point within a tolerance range.
13. The apparatus as claimed in claim 12, wherein the predetermined tolerance range is a sphere having a radius smaller than 2 meters around the predetermined point.
14. The apparatus as claimed in any of claims 11 to 13,
    wherein the virtual source is a source for plane waves, and wherein the means (100) for determining the correction value is configured to determine a correction value wherein an amplitude state of the audio signal associated with the virtual source equals the target amplitude state.
15. The apparatus as claimed in any of claims 11 to 14,
    wherein the virtual source is a point source, and wherein the means (100) for determining the correction factor is configured to operate on the basis of a target amplitude state which equals a quotient of an amplitude state of the audio signal associated with the virtual source, and the distance between the presentation area and the position of the virtual source.
16. The apparatus as claimed in any of claims 11 to 15,
    wherein the means (100) for determining the correction value is configured to operate on the basis of an actual amplitude state, the determination of which takes into account a loudspeaker transmission function of the loudspeaker (808).
17. The apparatus as claimed in any of claims 11 to 16,
    wherein the means (100) for determining the correction factor is configured to calculate, for each loudspeaker, a damping value which depends on the position of the loudspeaker and on a point to be contemplated in the presentation area, and wherein the means (100) for determining is further configured to weight the component signal of a loudspeaker with the damping value for the loudspeaker so as to obtain a weighted component signal and so as to further sum component signals or component signals, weighted accordingly, from other loudspeakers so as to obtain the actual amplitude state at the point contemplated on which the correction value (104) is based.
18. The apparatus as claimed in any of claims 11 to 17,
    wherein the means (106) for manipulating is configured to use that correction value (104) as a correction factor which equals a quotient of the actual amplitude state and the target amplitude state.
19. The apparatus as claimed in claim 18, wherein the means (106) for manipulating is configured to scale the audio signal associated with the virtual source with the correction factor prior to the calculation of the component signals by the wave-field synthesis module (810).
20. The apparatus as claimed in any of claims 11 to 19,
    wherein the target amplitude state is a target sound level, and wherein the actual amplitude state is an actual sound level.
21. The apparatus as claimed in claim 20, wherein the target sound level and the actual sound level are based on a target sound intensity and an actual sound intensity, respectively, the sound intensity being a measure of an energy falling onto a reference area within a period of time.
22. The apparatus as claimed in claims 20 or 21, wherein the means (100) for determining the correction value is configured to calculate the target amplitude state in that samples of the audio signal associated with the virtual source are squared sample by sample, and in that a number of squared samples, the number being a measure of an observation time, are summed, and
    wherein the means (100) for determining the correction value is further configured to calculate the actual amplitude state in that each component signal is squared sample by sample, and in that a number of squared samples, which equals the number of summed squared samples for calculating the target amplitude state, are added up, and wherein addition results from the component signals are further added up so as to obtain a measure of the actual amplitude state.
23. The apparatus as claimed in any of claims 11 to 22,
    wherein the means (100) for determining the correction value (104) comprises a look-up table (400) which has position/correction-factor value pairs stored therein, a correction factor of a value pair depending on an arrangement of the loudspeakers in the array of loudspeakers, and on a position of a virtual source, and the correction factor being selected such that a deviation between an actual amplitude state due to the virtual source at the associated position and a target amplitude state is at least reduced when the correction factor is used by the means (106) for manipulating.
24. The apparatus as claimed in claim 23, wherein the means (100) for determining is further configured to interpolate (402) a current correction factor for a current position of the virtual source from one or several correction factors from position/correction-factor value pairs, whose position(s) is/are located adjacent to the current position.
25. A method for generating a low-frequency channel (940, 942, 944) for a low-frequency loudspeaker (1106, 1110), comprising:

    providing (900) a plurality of audio objects, an audio object having an object signal and an object description associated with it;

    calculating (906) an audio object scaling value for each audio object in dependence on the object description (904);

    scaling (910) each object signal with an associated audio object scaling value (908) so as to obtain a scaled object signal (912) for each audio object;

    summing (914) the scaled object signals so as to obtain a composite signal (916); and

    providing (918) the low-frequency channel (920, 940, 942, 944) for the low-frequency loudspeaker (1106, 1110) on the basis of the composite signal (916).
26. A computer program having a program code for performing the method as claimed in claim 25, when the program runs on a computer.

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