EP1671516A1

EP1671516A1 - Device and method for producing a low-frequency channel

Info

Publication number: EP1671516A1
Application number: EP04797996A
Authority: EP
Inventors: Michael Beckinger; Sandra Brix
Original assignee: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Current assignee: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Priority date: 2003-11-26
Filing date: 2004-11-18
Publication date: 2006-06-21
Anticipated expiration: 2024-11-18
Also published as: US8699731B2; CN1906971A; WO2005060307A1; JP2007512740A; US20060280311A1; DE10355146A1; CN100588286C; DE502004002926D1; JP4255031B2; EP1671516B1

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

Device and method for producing a bass channel

description

The present invention relates to the generation of one or more bass channels, and in particular to the generation of one or more bass channels in connection with a multi-channel audio system, such as, for example, a cellular field synthesis system.

There is an increasing need for new technologies and innovative products in the field of entertainment 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 in particular computer technology. Examples of this are the applications that offer an improved realistic audiovisual impression. With previous audio systems, a major weakness lies in the quality of the spatial sound reproduction of natural, but also of virtual environments.

Methods for multi-channel loudspeaker reproduction of audio signals have been known and standardized for many years. All common techniques have the disadvantage that both the location of the speakers and the position of the listener are already imprinted on the transmission format. If the speakers are arranged incorrectly in relation to the listener, the audio quality suffers significantly. 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 wrapping in the audio playback can be achieved with the help of a new technology. The basics of this Technology, the so-called ellenfeldsynthese (WFS; WFS = Wave-Field Synthesis), was researched at TU Delft and first introduced in the late 80s (Berkhout, AJ; de Vries, D.; Vogel, P.: Acoustic control by Wavefield Synthesis, JASA 93, 1993).

Due to the enormous demands of this method on computer performance and transmission rates, wave field synthesis has so far only rarely been used in practice. It is only the advances in the areas of microprocessor technology and audio coding that allow this technology to be used in concrete applications. The first products in the professional sector are expected next year. The first wave field synthesis applications for the consumer sector are also expected to be launched in a few years.

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

Every point that is captured by a wave is the starting point for an elementary wave that propagates in a spherical or circular manner.

Applied to acoustics, a large number of loudspeakers that are arranged next to each other (a so-called loudspeaker array) can be used to simulate any shape of an incoming wavefront. In the simplest case, a single point source to be reproduced and a linear arrangement of the loudspeakers, the audio signals of each loudspeaker have to be fed with a time delay and amplitude scaling in such a way that the emitted sound fields of the individual loudspeakers overlap correctly. If there are several sound sources, the contribution to each loudspeaker is calculated separately for each source and the resulting signals are added. In a room with reflective walls, reflections can also be reproduced via the loudspeaker array as additional sources. The effort involved in the calculation therefore depends heavily on the number of strongly depends on the number of sound sources, the reflection properties of the recording room and the number of speakers. The particular advantage of this technique is that a natural spatial sound impression is possible over a large area of the playback room. In contrast to the known techniques, the direction and distance of sound sources are reproduced very precisely. To a limited extent, virtual sound sources can even be positioned between the real speaker array and the listener.

Although wave field synthesis works well for environments whose properties are known, irregularities do occur when the nature changes or when the wave field synthesis is carried out on the basis of an environment condition that does not match the actual nature of the environment. However, the technique of wave field synthesis can also be used advantageously to complement a visual perception with a corresponding spatial audio perception. So far, the mediation was an authentic visual impression of the virtual scene ^v in the foreground in the production in virtual studios. The acoustic impression that goes with the image is usually imprinted on the audio signal by manual work steps in what is known as post-production, or is classified as too complex and time-consuming to implement and is therefore neglected. This usually leads to a contradiction of the individual sensations, which leads to the fact that the designed space, i. H. the designed scene, which is perceived as less authentic. In the majority of cases, a concept is pursued that involves getting an overall acoustic impression of the visually depicted scenery. This can be done well with the term from the image design rewrite the "total". This "total" sound impression mostly remains constant over all settings in a scene, although the optical perspective on things changes mostly strongly. Optical details are highlighted by appropriate settings or placed in the background. Even shots in filmic dialogue design are not reproduced by the sound.

There is therefore a need to acoustically embed the viewer in an audiovisual scene. The canvas or picture surface forms the viewing direction and the viewing angle of the viewer. This means that the sound should follow the picture in such a way that it always matches the picture seen. This is even more important for virtual studios, since there is typically no correlation between the tone of the moderation, for example, and the environment in which the moderator is currently located. In order to get an overall audiovisual impression of the scene, a spatial impression matching the rendered image must be simulated. In this context, an essential subjective characteristic of such a sound concept is the location of a sound source, as seen by a viewer, for example, of a cinema screen.

In the audio area, the technique of wave field synthesis (WFS) can be used to achieve a good spatial sound for a large range of listeners. As has been explained, wave field synthesis is based on the principle of Huygens, according to which wave fonts can be formed and built up by superimposing elementary waves. According to a mathematically exact theoretical description, an infinite number of sources at infinitely small distances would have to be used to generate the elementary waves. In practice, however, many loudspeakers are finally used at a finite distance apart. Each of these speakers is based on the WFS principle with an audio signal from a virtual source that has a certain delay and one has a certain level. Levels and delays are usually different for all speakers.

As has already been explained, the ellen field synthesis system works on the basis of the Huygens principle and reconstructs a given waveform, for example a virtual source, which is arranged at a certain distance from a presentation area or to a listener in the presentation area by a plurality of single waves. The wave field synthesis algorithm thus receives information about the actual position of a single speaker from the speaker array, in order to then calculate a component signal for this single speaker, which this speaker must then ultimately emit so that the listener overlays the speaker signal from one speaker with the speaker signals of the other active ones Loudspeaker carries out a reconstruction in such a way that the listener has the impression that he is not "sonicated" by many individual loudspeakers, but only borrowed from a single loudspeaker at the position of the virtual source.

For several virtual sources in a wave field synthesis setting, the contribution from each virtual source for each loudspeaker, ie the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc., is calculated in order to then add up the component signals to finally get the actual loudspeaker signal. In the case of, for example, three virtual sources, the overlaying of the loudspeaker signals of all active loudspeakers at the listener would result in the listener not having the impression that he was being exposed to a large array of loudspeakers, but that the sound he hears only comes from three sound sources positioned in special positions that are equal to the virtual sources. In practice, the component signals are usually calculated by applying a delay and a scaling factor to the audio signal assigned to a virtual source, depending on the position of the virtual source and the position of the loudspeaker, in order to produce a delayed and / or scaled audio signal To obtain a virtual source that directly represents the loudspeaker signal if only one virtual source is present, or that, after addition with other component signals for the loudspeaker under consideration, from other virtual sources then contributes to the loudspeaker signal for the loudspeaker under consideration.

Typical wave field synthesis algorithms work regardless of how many speakers are in the speaker array. The theory on which the wave field synthesis is based is that any sound field can be exactly reconstructed by an infinitely high number of individual speakers, the individual single speaker being arranged infinitely close to one another. In practice, however, neither the infinitely high number nor the infinitely close arrangement can be realized. Instead, there is a limited number of speakers, which are also arranged at certain predetermined distances from each other. This means that in real systems only an approximation to the actual waveform is achieved, which would take place if the virtual source were actually available, i.e. would be a real source.

Furthermore, there are various scenarios in that the loudspeaker array can only be viewed when viewing a cinema, e.g. B. is arranged on the side of the cinema screen. In this case, the wave field synthesis module would generate speaker signals for these speakers, the speaker signals for these speakers normally being the same as for corresponding speakers in a speaker array that extends, not just across the side of a cinema, for example, to the screen is arranged, but is also arranged on the left, right and behind the listening room. This "360 °" speaker array will of course create a better approximation to an exact wave field than just a one-sided array, for example in front of the audience. Nevertheless, the speaker signals for the speakers in front of the audience are the same in both cases This means that a wave field synthesis module typically receives no feedback as to how many loudspeakers are present or whether it is a single-sided or multi-sided or even a 360 ° array or not. In other words, a wave field synthesis device calculates a loudspeaker signal for one which other speakers still exist ^¬ According speaker based on the position of the speaker and independent of it or do not exist.

This is a major strength of the wave field synthesis algorithm in that it is optimally modularly adaptable to different circumstances by simply giving the coordinates of the existing loudspeakers in very different demonstration rooms. However, it is disadvantageous that, in addition to the poorer reconstruction of the current wave field that may be acceptable, considerable level artifacts occur. For a real impression, it is not only decisive in which direction the virtual source is with respect to the listener, but also how loud the listener hears the virtual source, ie what level the listener "receives" due to a special virtual source The level arriving at the listener, which is related to a virtual source under consideration, results from the superimposition of the individual signals of the loudspeakers.

For example, consider the case where there is a speaker array of 50 speakers in front of the listener and the audio signal from the virtual source through the wave field synthesizer in component signals. is mapped for the 50 speakers, such that the audio signal is radiated simultaneously with different delay and different scaling from the 50 speakers, so a listener of the virtual source feels a level of the source, which results from the individual levels of the component signals of the virtual source the individual loudspeaker signals.

If the same wave field synthesis device is now used for a reduced array, in which, for example, there are only 10 loudspeakers in front of the listener, it is readily apparent that the level of the signal from the virtual source that results at the listener's ear has decreased. since 40 components signals of the now missing speakers are "missing", so to speak.

The alternative case can also occur in which e.g. B. are first left and right of the listener speakers that are driven in a certain constellation in phase opposition, so that the speaker signals from two opposite speakers cancel each other due to a certain delay calculated by the wave field synthesis device. Is now in a reduced system such. B. without the speakers on one side of the listener, the virtual source suddenly appears much louder than it should be.

While constant factors could still be considered for static sources for level correction, this solution is no longer viable if the virtual sources are not static but are moving. This is an essential feature of wave field synthesis that it can also and especially process moving virtual sources. A correction with a constant factor would fall short here, since the constant factor would vote for one position, but would increase the artifact for another position of the virtual source. Wave field synthesis devices are also able to emulate 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 that are arranged at an infinite distance.

According to the wave field synthesis theory, the level change in two-dimensional loudspeaker arrangements, depending on r, corresponds to the natural level change except for a negligible error. Depending on the position of the source, however, different, sometimes considerable errors in the absolute level can result, which result from the use of a finite number of loudspeakers instead of the theoretically required infinite number of loudspeakers, as has been explained above.

Another problem that exists in multi-channel reproduction systems and in particular in wave field synthesis systems that not only use five or seven loudspeakers, for example, but also use a significantly higher number of loudspeakers, is that the loudspeakers can lead to considerable costs due to their large number. In order to reduce the costs of the loudspeakers, the so-called subwoofer principle is used in such existing five-channel systems or seven-channel systems. The subwoofer principle is used in multichannel playback systems to save expensive and large woofers. A low-frequency channel is used that only contains music signals with frequencies lower than a cut-off frequency of around 120 Hz. This low-frequency channel controls a low-frequency loudspeaker with a large diaphragm area, with which high sound pressures are achieved, especially at low frequencies. The subwoofer principle takes advantage of the fact that it is very difficult for the human ear to localize low-frequency sounds in the direction. In current systems, an extra bass channel for a special loudspeaker arrangement (spatial arrangement) is already mixed during the sound mixing. Examples of such ultimate channel playback systems are Dolby Digital, Sony SDDS and DTS. With these multichannel formats, the subwoofer channel can be mixed regardless of the size of the room to be sounded, since the spatial conditions only change in the scale sense. The loudspeaker arrangement remains the same in scale.

With the Wave Field Synthesis (WFS), a large audience area can be covered with sound. Sound events can be wiederge ^¬ in their spatial depth. For this purpose, the complete sound field of the individual sound events is reproduced in the audience area. This is done by a large number of speakers. Around 500 or more speaker systems are required for large installations. If you wanted to equip every single speaker system with a powerful woofer, very high costs would arise.

It was mentioned that a special loudspeaker arrangement is required for existing multichannel formats in order to mix a special subwoofer channel. However, the loudspeaker arrangement can be changed to scale without having to change the corresponding mix. The ratio of the distances between the individual loudspeakers remains the same. However, none of this is possible with the WFS, since the number of loudspeaker channels depends on the size of the area of the WFS playback system to be sonicated. For this reason, the individual loudspeaker channels cannot be saved, which would also be quite memory-intensive if you consider systems with 500 or more audio channels. Therefore, only the virtual sound events to be simulated are saved. Only when the The individual speaker channels are calculated using the WFS algorithm.

On the one hand, 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 area to be irradiated. The goodness of the WFS playback system depends on this density. The volume depends on the number of loudspeaker channels and the density of the loudspeakers, since all loudspeaker channels add up to form a wave field. The volume of a WFS system is therefore not predetermined. The volume of the subwoofer channel is, however, predetermined with the known parameters of the electrical amplifier and the loudspeaker. It is therefore not possible to transfer a mix of a subwoofer channel from a WFS system without errors to a WFS system with a different speaker density and number of speakers. The volumes of the bass system on the one hand and of the mid / high system on the other hand did not match.

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

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

The present invention is based on the knowledge that the low-frequency channel for a low-frequency loudspeaker or that several low-frequency channels for several low-frequency loudspeakers in a multichannel system is not already generated in a sound mixing process that is independent of an actual one Playback space takes place, but that reference is made to the actual playback space by taking into account the predetermined position of the woofer on the one hand and properties of audio objects, which typically represent virtual sources, on the other hand, in order to generate the woofer channel. In particular, audio objects are assumed, an object description on the one hand and an object signal on the other hand being assigned to an audio object. 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 to sum up the scaled object signals to obtain a sum signal. The low-frequency channel, which is fed to the low-frequency loudspeaker, is then derived from the sum signal.

In the case of sources that emit plane waves, in which a position in infinity is assumed, the virtual position of the source on the one hand and a reference playback position on the other hand, for which a reference volume is required, are irrelevant. However, this is not the case with customary sources assumed to be punctiform, such as occur in a film setting when dialogues etc. take place. In this case, a scaling of the audio object signal, which originates from a virtual source, which is arranged at a virtual position, is carried out in such a way that an actual volume or an actual amplitude state, based on this virtual source, corresponds to a desired amplitude state at the reference reproduction position , The target amplitude condition 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 carried out for all virtual sources in order to then scale the audio object signals of each virtual source with the corresponding scaling value. The scaled audio object signals are then summed up to obtain a sum signal. The low-frequency channel is then derived from this sum signal in the case where there is only a single low-frequency loudspeaker. This can be done by simple low-pass filtering.

At this point it should be pointed out that the low-pass filtering can already be carried out with the still unscaled audio object signals, so that only low-pass signals are already being 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 low-frequency channel only after summing up the scaled object signals in order to obtain the best possible approximation of the volume of the low-frequency signals in the performance room on the one hand and the volume of the mid-range and high-frequency signals in the performance room on the other hand.

According to the invention, a subwoofer channel is therefore not already mixed during the sound mixing process from the virtual sources, that is to say the sound material for the wave field synthesis. Instead, the mixing takes place automatically during playback 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 the extent of the fringed area of the wave field synthesis system. Even prescribed loudspeaker arrangements no longer have to be observed, since the loudspeaker position and number of loudspeakers are included in the generation of the low-frequency channel.

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

The low-frequency channel generation according to the invention is preferably combined with a level artifact reduction in order to carry out level corrections in a wave field synthesis system not only for low-frequency channels, but for all loudspeaker channels, in order to be independent of the number and position of the loudspeakers used with respect to the wave field synthesis algorithm used.

In preferred exemplary embodiments of the present invention, in which only a single low-frequency channel and thus a single low-frequency loudspeaker is provided, the low-frequency loudspeaker will not be arranged in a reference reproduction position for which an optimal level correction is carried out. In this case, the sum signal is scaled according to the invention, taking into account the position of the woofer using a loudspeaker scaling value to be calculated. This scaling will preferably only be an amplitude scaling and not a phase scaling, taking into account the fact that the ear has no good localization at the low frequencies present in the low-frequency channel, but only shows an exact amplitude / volume perception. Alternatively or additionally, a phase scaling can be used as scaling, if such is desired in an application scenario.

If several woofers are positioned, a separate woofer is created for each woofer. The bass channels of the individual bass speakers preferably differ only in their amplitude, but not in the signal itself. All woofers thus send the same sum signal, but with different amplitude scaling, with the amplitude scaling for a single one Woofer depends on the distance of the individual woofer to the reference playback point. In addition, it is ensured according to the invention that the overall volume of all superimposed tie tone 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 woofer channel, with which the sum signal is then scaled accordingly in order to obtain the individual woofer channel.

The use of a subwoofer channel is particularly advantageous in that it leads to a significant price reduction, since the individual speakers, for. B. a wave field synthesis system can be built much cheaper, since they do not have to have low-frequency properties. In contrast, only one or a few, such as three to four, subwoofer speakers is sufficient to realize the very low frequencies with high sound pressure through a correspondingly large membrane area.

The present invention is also advantageous in that the one or more low-frequency channels for any speaker positions and multi-channel formats can be generated automatically, this requiring only little additional effort, particularly in the context of a wave field synthesis system, since the wave field synthesis system carries out a level correction anyway.

With regard to the necessary number of woofers and the optimal positioning of one or more woofers, reference is made to the technical literature, ^th of the particular Welti, Todd, "How Many Subwoofers are Enough", 112 AES Conv. Paper 5602, May 2002, Munich, Germany , Martens, "The impact of decorrelated low-frequency reproduction on auditory spatial imagery: Are two than subwoofers better one? ", called 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, the individual volume and preferably also the delay of each virtual source, that is to say each sound object or audio object, is first calculated in relation to the reference playback position. The audio signal of each virtual source is then scaled and delayed in order to sum up all virtual sources. The total volume and delay of the subwoofer are then calculated depending on its distance from the reference point, if the subwoofer is not already located 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 from the reference point. Here it is preferred to adhere as a boundary condition that the sum of all subwoofer channels is equal to that of the reference volume at the reference playback position, which preferably corresponds to the center point of the wave field synthesis system. Corresponding scaling factors per subwoofer are thus calculated, whereby however first individual volume and delay of each virtual source are determined in relation to the reference point. Then each virtual source is scaled again accordingly and optionally delayed, in order then to sum up all virtual sources to the sum signal, which is then scaled with the individual scaling factors for each subwoofer channel in order to obtain the individual bass channels for the various bass speakers.

Preferred embodiments of the present invention are explained in detail below with reference to the accompanying drawings. Show it: 1 shows a block diagram of the device according to the invention for level correction in a wave field synthesis system;

2 shows a basic circuit diagram of a wave field synthesis environment as can be used for the present invention;

FIG. 3 shows a more detailed illustration of the wave field synthesis module shown in FIG. 2;

4 shows a block diagram of a device according to the invention for determining the correction value according to an exemplary embodiment with a look-up table and, if appropriate, interpolation device;

5 shows a further exemplary embodiment of the device for determining FIG. 1 with target value / actual value determination and subsequent comparison;

6a is a block diagram of a wave field synthesis module with an embedded manipulation device for manipulating the component signals;

6b shows a block diagram of a further exemplary embodiment of the present invention with an upstream manipulation device;

7a shows a sketch for explaining the desired amplitude state at an optimal point in a demonstration area;

7b shows a sketch to explain the actual amplitude state at an optimal point in the demonstration area; 8 shows a basic block diagram of a wave field synthesis system with a wave field synthesis module and loudspeaker array in a demonstration area;

9 shows a block diagram of a device according to the invention for generating a low-frequency channel;

10 shows a preferred embodiment of the device for providing the bass channel for a plurality of bass speakers; and

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

As has already been stated, both the volume and the delay are calculated by the wave field synthesis algorithm for each speaker 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, that is to say the individual loudspeakers of the array, is based on the knowledge that the inadequacies of a wave field synthesis system with a (practically realizable) finite number of loudspeakers can at least be alleviated if a level Correction is performed by manipulating either the audio signal associated with a virtual source prior to wave field synthesis or the component signals for various loudspeakers originating in a virtual source after wave field synthesis using a correction value to detect a deviation between a target amplitude condition in a demonstration area and an actual amplitude state in the demonstration area to reduce. The target 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 screening area to the virtual source and possibly taking into account the type of wave, a target level is determined as an example of a target amplitude state, and that an actual level as an example of an actual amplitude state of the listener is determined. While the target amplitude state is determined independently of the actual grouping and type of the individual speakers only on the basis of the virtual source or their position, the actual situation is calculated taking into account the positioning, type and control of the individual speakers of the speaker array.

In this way, the sound level at the ear of the listener can be determined at the optimum point within the demonstration area on the basis of a component signal from the virtual source, which is emitted via a single loudspeaker. Correspondingly, the level at the ear of the listener at the optimal point within the demonstration area can also be determined for the other component signals that go back to the virtual source and are emitted via other loudspeakers, in order to then summarize these levels to the actual actual level at the ear of the To receive the handset. For this purpose, the transfer function of each individual loudspeaker as well as the level of the signal at the loudspeaker and the distance of the listener at the point under consideration within the demonstration area from the individual loudspeaker can be taken into account. For simpler designs, the transmission characteristics of the speaker can be assumed to work as an ideal point source. For more complex implementations, however, the directional characteristic of the individual speaker can also be taken into account. A major advantage of this concept is that in an embodiment in which sound levels are considered, only multiplicative scaling occurs, in that for a quotient between the target level and the actual level, which gives the correction value, not the absolute level at 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 thus on the positions of the individual speakers) and the optimal point within the demonstration area. However, these values are fixed with regard to the position of the optimal point and the positions and transmission characteristics of the individual loudspeakers and do not depend on a piece being played.

Therefore, the concept can be implemented in a computationally efficient manner as a look-up table, in that a look-up table is generated and used, which includes position correction factor-value pairs, for all or a substantial part of possible virtual positions. In this case, no on-line setpoint determination, actual value determination and setpoint / actual value comparison algorithm is to be carried out. These algorithms, which can be time-consuming in some cases, 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 from there the correction factor valid for this position of the virtual source. In order to increase the computing and storage efficiency even further, it is preferred to store only relatively roughly rasterized pairs of base values for positions and assigned correction factors in the table and correction factors for position values that lie between two base values, one-sided, two-sided, interpolate linearly, cubically etc.

Alternatively, it may also make sense in one case or another to use an empirical approach in that level measurements are made. In such a case, a virtual source with a certain calibration level would be placed in a certain virtual position. Then, for a real wave field synthesis system, a wave field synthesis module would calculate the loudspeaker signals for the individual loudspeakers in order to finally measure the level actually arriving at the listener due to the virtual source. A correction factor would then be determined such that it at least reduces the deviation from the target level to the actual level or preferably brings it to 0. This correction factor would then be stored in the look-up table in association with the position of the virtual source in order to gradually generate the entire look-up table for a specific wave field synthesis system in a special demonstration room, that is to say for many positions of the virtual source.

There are several options for manipulation based on the correction factor. In one embodiment, it is preferred to manipulate the audio signal of the virtual source, as recorded, for example, in an audio track that comes from a recording studio, with the correction factor, only to then feed the manipulated signal into a wave field synthesis module. To a certain extent, this automatically leads to the fact that all component signals that originate from this manipulated virtual source are also weighted accordingly, in comparison to the case in which no correction has been made in accordance with the present invention.

Alternatively, it may also be favorable for certain applications not to intervene in the original audio signal of the virtual source, but to intervene in the component signals generated by the wave field synthesis module in order to manipulate all of these component signals preferably with the same correction factor. At this point it should be pointed out that the correction factor is not necessarily identical for all component signals got to. However, this is largely preferred in order not to impair the relative scaling of the component signals to one another, which is necessary for the reconstruction of the actual wave situation.

One advantage is that with relatively simple measures, at least during operation, a level correction can be carried out in such a way that the listener, at least in view of the volume of a virtual source that he perceives, does not notice that the infinitely many loudspeakers that are actually required are not available , but only a limited number of speakers.

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

Another advantage is that it offers the option of offering less expensive wave field synthesis systems with a smaller number of loudspeakers, which, however, do not involve any level artifacts, especially for moving sources, ie just as good for a listener with regard to the level problem act like more complex wave field synthesis systems with a large number of speakers. Levels which are too low can also be corrected according to the invention for holes in the array.

Before going into detail in the preferred manner described above for level artifact correction, the concept according to the invention for generating a low-frequency channel is first illustrated with reference to FIG. 9, which is either standing alone, ie without level correction Individual speakers can be used, or that can preferably be combined with the concept of level artifact correction described later with reference to FIGS. 1-8, in order to also use the correction values used for level artifact correction of the individual speakers as audio object scaling values, which are used in low-frequency channel generation must be used.

FIG. 9 shows a device for generating a low-frequency channel for a low-frequency loudspeaker, which is arranged 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 signal 902 and an audio object description 904 being assigned to an audio object. The audio object description will typically include an audio object position and possibly the audio object type. Depending on the embodiment, the audio object description can also directly include an indication of the audio object volume. If this is not the case, then the audio object volume can be easily calculated from the audio object signal itself, for example by squaring and summing over a certain period of time. If the transmission function, frequency response, etc. of the individual loudspeakers under consideration or the low-frequency loudspeaker are to be taken into account here, this will also be possible with a simple table lookup or a correction factor, since in a playback system the electrical behavior of the loudspeaker or the signal / sound characteristics of the speaker is a stationary size.

The object description of the audio signal is supplied to means 906 for calculating an audio object scaling value for each audio object. The individual audio object scaling values 908 are then, as shown with reference to FIG. 9, fed to a device 910 for scaling the object signals. The means 906 for computing the audio object scaling value is designed 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, the audio object scaling value or the correction factor will be 1, since for such plane-wave audio objects a spacing between the position of this object and the optimal reference playback position is irrelevant, because in this case the virtual position is assumed to be infinite.

On the other hand, if the audio object is a point-emitting virtual source that is positioned at a virtual position, the audio object scaling value becomes dependent on the object volume, which is either in the object description or can be derived from the object signal, and that Distance between the virtual position of the audio object and the reference playback position is calculated.

In particular, it is preferred to calculate the audio object scaling value or correction value such that it is taken into account that it is based on a target amplitude state in the demonstration area, the target amplitude state depending on a position of the virtual source or a type of the virtual source, where the correction value is also based on an actual amplitude state in the demonstration area, which is based on the component signals for the individual speakers on the basis of the virtual source under consideration. The correction value is thus calculated in such a way that a manipulation of the audio signal assigned to the virtual source using the correction value reduces a deviation between the desired amplitude state and the actual amplitude state. After scaling the object signals, which is carried out by the device 910 in order to obtain the scaled object signals 912, they are subjected to an input direction 914 for summing to generate a sum signal 916.

As has been explained, it is preferred to take into account a delay which may be caused by different virtual positions before the summation by means 914, so that the individual audio object signals, which are present as sequences of samples, are shifted with respect to a time reference. in order to take sufficient account of the differences in transit time of the sound signal from the virtual position to the reference reproduction position. After scaling and taking into account the delay, the scaled and correspondingly delayed object signals are then summed up by the device 914 in samples, in order to obtain a sum signal with a sequence of sum signal samples, which is designated by 916 in FIG. 9. This sum signal 916 is fed to a device 918 for providing the low-frequency channel for the one or more subwoofers, which supplies the subwoofer signal or the low-frequency channel 920 on the output side.

As has been stated, the sound signal emitted by a woofer is not a sound signal with full bandwidth, but with an upper bandwidth limit. In one embodiment, it is preferred that the cut-off frequency of the sound signal emitted by a woofer is less than 250 Hz and preferably even only 125 Hz. The band limitation of this sound signal can take place at different points. A simple measure is to supply the woofer with a full bandwidth excitation signal, which is then band-limited by the woofer itself, since it only converts low frequencies into sound signals, but suppresses high frequencies.

Alternatively, the band limitation can also be carried out in the device 918 for providing the low-frequency channel. gene by low-pass filtering the signal there before a digital / analog conversion, this low-pass filtering being preferred, since it can be carried out on the digital side, so that clear conditions exist regardless of the actual implementation of the subwoofer. Alternatively, however, the low-pass filtering can take place before the device 910 for scaling the object signals, so that the operations which are carried out by the devices 910, 914, 918 are now carried out with low-pass signals and not with signals of the entire bandwidth.

According to the invention, however, it is preferred to carry out 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 signals of full bandwidth are carried out in order for the loudspeakers to match the low tone on the one hand and the mid-tone and Ensure high tones on the other hand. In other words, it is preferred as many operations for determining the actual loudspeaker signals synthesis array parallel to the loudspeakers in the wave field to perform and very late perform a "spin-off" of the low-frequency channel. ^'' Fig. 10 a preferred embodiment of the device 918 for The geometric situation is first shown with reference to Fig. 11. Fig. 11 schematically shows a wave field synthesis system with a large number of individual speakers 808. The individual speakers 808 form an array 800 of individual speakers, which enclose the demonstration area, preferably within the demonstration area the reference playback position or reference point 1100 is located. 11 also schematically shows an audio object 1102, which is referred to as a "virtual sound object". The virtual sound object 1102 comprises an object description that represents a virtual position 1104. Using the coordinates of the reference point 1100 and the coordinates of the virtual position 1104, the if necessary, can be converted accordingly, the distance D of the virtual sound object 1102 from the reference playback position 100. 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 regularity, which will be explained in detail later in FIG. 7a. 11 also 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 or any further one in FIG 11 not shown optional additional subwoofers. The first subwoofer 1106 is at a distance d1 from the reference point 1100, while the second subwoofer 1110 is at a distance d2 from the reference point. Analogously to this, a subwoofer n (not shown in FIG. 11) is at a distance dn from the reference point 1100.

Referring back to FIG. 10, the device 918 for providing the low-frequency channel is designed to, in addition to the sum signal 916, which is denoted by s in FIG. 10, also the distance d1 of the woofer 1, which is denoted by 930, based on the distance d2 of the woofer 2, designated 932, and the distance dn of the woofer n, designated 934. On the output side, the device 918 supplies a first bass channel 940, a second bass channel 942 and an nth bass channel 944. It can be seen from FIG. 10 that all bass channels 940, 942, 944 are weighted versions of the sum signal 916, the respective weighting factors are denoted by ai, a ₂ , ..., a _n . The individual weighting factors ai, a ₂ , a _n depend on the one hand on the distances 930-934 and on the other hand on the general boundary condition that the volume of the bass channels at reference point 100 corresponds to the reference volume, that is to say the desired amplitude state for the bass channel at reference playback position 1100 (FIG. 11). After all subwoofers have been removed from reference point 1100, the sum of the loudspeaker scaling values ai, a ₂ , a _{n will be} greater than 1 in order to take into account the attenuation of the bass channels on the way from the corresponding subwoofer to the reference point. If only a single woofer (e.g. 1106) is provided, the scaling factor ai will also be greater than 1, while no further scaling factors need to be calculated, since there is only a single woofer.

1-8, a level artifact correction device for the loudspeaker array 800 in FIG. 8 and FIG. 11 is shown, which are preferably combined with the low-frequency channel calculation according to the invention, as has been illustrated with reference to FIGS. 9-11 can.

Before the present invention is discussed in detail, the basic structure of a wave field synthesis system is illustrated below with reference to FIG. 8. The wave field synthesis system has a speaker array 800 placed with respect to a demonstration area 802. Specifically, the speaker array shown in Fig. 8, which is a 360 ° array, includes four array sides 800a, 800b, 800c and 800d. Is the demonstration area 802 e.g. B. a cinema hall, it is assumed with regard to the conventions front / rear or right / left that the cinema screen is on the same side of the screening area 802 on which the sub-array 800c is also arranged. In this case, the viewer, who is sitting at the so-called optimal point P in the demonstration area 802, would see the front, that is, the screen. Sub-array 800a would then be behind the viewer, while to the left of Viewer would have sub-array 800d and sub-array 800b to the right of the viewer. Each loudspeaker array consists of a number of different individual loudspeakers 808, each of which is controlled with its own loudspeaker signals, which are provided by a wave field synthesis module 810 via a data bus 812, which is shown only schematically in FIG. 8. The wave field synthesis module is designed to use the information about e.g. B. To calculate the type and position of the loudspeakers with regard to the demonstration area 802, that is to say loudspeaker information (LS information), and, if appropriate, with other inputs, loudspeaker signals for the individual loudspeakers 808, each of which is generated by the audio tracks for virtual sources to which positive ons information are assigned, 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 explanations regarding the present invention can in principle be carried out for each point P in the demonstration area. The optimum point can thus be anywhere in the demonstration area 802. There can also be several optimal points, e.g. B. on an optimal line. However, in order to obtain the best possible conditions for as many points as possible in the demonstration area 802, it is preferred to place the optimal point or the optimal line in the middle or at the center of gravity of the wave field synthesis system, which is generated by the loudspeaker sub-arrays 800a, 800b, 800c , 800d is defined to assume.

A more detailed illustration 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 and to the arrangement shown in detail in FIG. 3. Figure 2 shows a wave field synthesis environment in which the present invention can be implemented. The center of a wave field synthesis environment is a wave field synthesis module 200, which comprises various inputs 202, 204, 206 and 208 and various outputs 210, 212, 214, 216. Various audio signals for virtual sources are fed to the wave field synthesis module via inputs 202 to 204. So the input 202 receives z. B. an audio signal of the virtual source 1 and assigned position information of the virtual source. In a cinema setting, for example, the audio signal 1 would be e.g. B. the language of an actor who moves from a left side of the screen to a right side of the screen and possibly additionally away from the viewer or towards 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 at a certain point in time. In contrast, the audio signal n would be the language of, for example, another actor who moves the same or different than the first actor. The current position of the other actor to whom the audio signal n is assigned is communicated to the wave field synthesis module 200 by position information synchronized with the audio signal n. In practice, different virtual sources exist depending on the recording setting, the audio signal of each virtual source being 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 LSI, LS2, LS3, LSm by outputting loudspeaker signals via the outputs 210 to 216 to the individual loudspeakers. The positions of the individual loudspeakers in a playback setting, such as a cinema, are communicated to the wave field synthesis module 200 via the input 206. In the cinema hall there are many individual loudspeakers grouped around the cinema audience, preferably in arrays are arranged such that there are loudspeakers both in front of the viewer, for example behind the screen, and behind the viewer and to the right and left of the viewer. Furthermore, other inputs can be communicated to the wave field synthesis module 200, such as information about the room acoustics, etc., in order to be able to simulate the actual room acoustics prevailing during the recording set-up in a cinema hall.

Generally speaking, the loudspeaker signal which is supplied to the loudspeaker LSI via the output 210, for example, will be a superimposition of component signals of the virtual sources, in that the loudspeaker signal for the loudspeaker LSI is a first component which originates from the virtual source 1, a second Component, which goes back to the virtual source 2, as well as an nth component, which goes back to the virtual source n. The individual component signals are linearly superimposed, i.e. added after their calculation, in order to simulate the linear superposition at the ear of the listener, who will hear a linear superposition of the sound sources perceivable in a real setting.

A detailed design of the wave field synthesis module 200 is set forth below with reference to FIG. 3. The wave field synthesis module 200 has a strongly parallel structure in that, starting from the audio signal for each virtual source and starting from the position information for the corresponding virtual source, delay information V _x and scaling factors SFj are first of all _. calculated from the position information and the position of the speaker under consideration, e.g. B. depend on the loudspeaker with the order number j, ie LSj. The calculation of a delay information V _x and a scaling factor SFj _. happens based on the position information of a virtual source and the position of the speaker j in question by known algorithms implemented in devices 300, 302, 304, 306. On the basis of the delay information Vi (t) and SFχ (t) and on the basis of the assigned for each virtual source audio signal ASι (t) is calculated for a current time _Ä t a discrete value AWι (t _A) for the component signal K _{i; ]} calculated in a speaker signal ultimately obtained. This is done by means 310, 312, 314, 316, as shown schematically in FIG. 3. 3 also shows, so to speak, a “flashing light recording” at time t _A for the individual component signals. The individual component signals are then summed by a summer 320 in order to determine the discrete value for the current time t _{a of} the loudspeaker signal for loudspeaker j which can then be supplied to the loudspeaker for the output (for example the output 214 if the loudspeaker j is the loudspeaker LS3).

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

At this point it should be noted that the value of a loudspeaker signal is obtained at the output 322 of FIG. 3, which is a superimposition of the component signals for this loudspeaker due to the different 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 what is preferred for practical reasons always z. B. 2, 4 or 8 lying speakers can be controlled with the same speaker signal.

FIG. 1 shows a block diagram of the device according to the invention for level correction in a wave field synthesis system, which has been explained with reference to FIG. 8. The wave field synthesis system comprises the wave field synthesis module 810 and the loudspeaker array 800 for supplying sound to the presentation area 802, the wave field synthesis module 810 being designed to receive an audio signal associated with a virtual sound source and source position information associated with the virtual sound source and component signals for them taking into account speaker position information Calculate speakers based on the virtual source. The device according to the invention first comprises a device 100 for determining a correction value which is based on a desired amplitude state in the demonstration area, the desired amplitude state depending on a position of the virtual source or a type of the virtual source, and the correction value also on Actual amplitude state is based in the demonstration area, which depends on the component signals for the loudspeakers 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 is not necessary to determine the actual state, since the source is already located infinitely far from the listener in the model due to the generated plane waves and has a position-independent level. The device 100 is designed to output a correction value 104 on the output side, which is a device 106 for manipulation an audio signal associated with the virtual source (obtained via an input 108) or for manipulating component signals for the speakers due to a virtual source (obtained via an input 110). If the alternative of manipulating the audio signal, which is provided via the input 108, is carried out, an output 112 results in a manipulated audio signal which, according to the invention, then in the wave field synthesis module 200 instead of the original audio signal which is provided at the input 108 is fed in to generate the individual loudspeaker signals 210, 212, ..., 216.

If, on the other hand, the other alternative to manipulation has been used, namely the manipulation of the component signals received via input 110 to some extent, component signals manipulated on the output side are obtained which still have to be summed up loudspeaker-wise (device 116), with or without manipulated component signals from other virtual sources, which are provided via further inputs 118. On the output side, the device 116 again delivers the loudspeaker signals 210, 212, ..., 216. It should be pointed out that the alternatives of the upstream manipulation (output 112) or the embedded manipulation (output 114) shown in FIG. 1 are used alternatively to one another can. Depending on the embodiment, however, there may also be cases in which the weighting factor or correction value, which is provided via the input 104 in the device 106, is split to a certain extent, so that partly an upstream manipulation and partly an embedded manipulation is carried out.

With regard to FIG. 3, the upstream manipulation would thus consist in manipulating the audio signal of the virtual source, which is fed into a device 310, 312, 314 or 316, before it is fed in. The embedded manipulation, on the other hand, would consist in manipulating the component signals output by devices 310, 312, 314 and 316, respectively, prior to their summation in order to obtain the actual loudspeaker signal.

These two possibilities, which can be used either alternatively or cumulatively, are shown in FIGS. 6a and 6b. 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 for example consists of blocks 300, 310 or 302, 312, or 304, 314 and 306 or 316 of FIG. 3, delivers the component signals n, Kχ ₂ , K ₁₃ for the loudspeaker LSI or Component signals K _nl , K _n2 and K _n3 for the loudspeaker LSn.

In the notation chosen in Figure 6a, the first index of Ki _j indicates the loudspeaker, and the second index indicates the virtual source from which the component signal originates. The virtual source 1 is expressed, for example, in the component signal Kn, ..., K _nl . In order to selectively influence the level of the virtual source 1 as a function of the position information of the virtual source 1 (without influencing the levels of the other virtual sources), in the embedded manipulation shown in FIG. 6a, the component signals belonging to the source 1 are multiplied , ie the component signals, the index j of which indicates the virtual source 1, take place with the correction factor F _x . In order to carry out a corresponding amplitude or level correction for the virtual source 2, all component signals which go back to the virtual source 2 are multiplied by a correction factor F ₂ determined for this. Finally, the component signals that go back to the virtual source 3 are also weighted by a corresponding correction factor F ₃ .

It should be noted that the correction factors Fi, F ₂ and F ₃ if all other geometric parameters are the same depend only on the position of the corresponding virtual source. Would all three virtual sources z. B. Point sources (ie of the same type) and be in the same position, the correction factors for the sources would be identical. This law is explained in more detail with reference to FIG. 4, since it is possible to simplify the computing time to use a look-up table with position information and respectively assigned correction factors, which must be created at some point, but which can be accessed quickly in operation without a setpoint / actual value calculation and comparison operation must be carried out continuously during operation, but this is also possible in principle.

6b shows the alternative to source manipulation according to the invention. The manipulation device is upstream of the wave field synthesis device and is effective to correct the audio signals of the sources with the corresponding correction factors in order to obtain manipulated audio signals for the virtual sources, which are then fed to the wave field synthesis device in order to obtain the component signals, which are then are summed up by the respective component summation devices in order to obtain the loudspeaker signals LS for the corresponding loudspeakers, such as, for example, the loudspeaker LSi.

In a preferred exemplary embodiment of the present invention, the device 100 for determining the correction value is designed as a look-up table 400, which stores position-correction factor-value pairs. The device 100 is preferably further provided with an interpolation device 402, on the one hand to keep the table size of the look-up table 400 within a limited framework, and on the other hand also for current positions of a virtual source that are fed into the interpolation device via an input 404, at least below Use one or more neighboring position correction factors stored in the lookup table Value pairs, which are fed to the interpolation device 402 via an input 406, to generate an interpolated current correction factor at an output 408. In a simpler version, however, the interpolation device 402 can also be omitted, so that the device 100 for determining FIG. 1 performs direct access to the lookup table using position information supplied at an input 410 and a corresponding correction at an output 412 - door factor delivers. If the current position information that is assigned to the audio track of the virtual source does not exactly correspond to a position information that can be found in the look-up table, then a simple rounding-off / rounding-up function can also be assigned to the look-up table in order to find the closest one in the Table to take the stored base value instead of the current base value.

At this point, it should be noted that different tables can be created for different source types, or that not only one correction factor is assigned to a position, but several correction factors, each correction factor being linked to a source type.

Alternatively, instead of the look-up table or to “fill up” the look-up table in FIG. 4, the device for determining can be designed to actually carry out a setpoint-actual value comparison. In this case, the device 100 of FIG. 1 comprises a setpoint amplitude State determination device 500 and an actual amplitude state determination device 502 for supplying a target amplitude state 504 and an actual amplitude state 506, which are fed to a comparison device 508 which, for example, a quotient of the target amplitude state 504 and the actual Amplitude state 506 is calculated to generate a correction factor 510, which is manipulated by the device 106, which is shown in Fig. 1 is fed for further use. Alternatively, the correction value can also be stored in a look-up table.

The target 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 type. Of course, the target amplitude state determination device 500 does not require any component signals for the target amplitude state calculation, since the target amplitude state is independent of the component signals. However, as can be seen from FIG. 5, component signals are fed to the actual amplitude determination device 502, which, depending on the embodiment, can also receive information about the speaker positions and information about speaker transmission functions and / or information about directional characteristics of the speakers by one Determine the current situation as well as possible. Alternatively, the actual amplitude state determination device 502 can also be designed as an actual measuring system in order to determine an actual level situation at the optimal point for certain virtual sources at certain positions.

In the following, reference is made to the actual amplitude state and the target amplitude state with reference to FIGS. 7a and 7b. Fig. 7a shows a diagram for determining a target amplitude state at a predetermined point, which is designated in Fig. 7a with "optimal point" and which is in the demonstration area 802 of Fig. 8. In Fig. 7a is just one example virtual source 700 is shown as a point source that generates a sound field with concentric wavefronts. Furthermore, the level L _v of virtual source 700 is known on the basis of the audio signal for virtual source 700. The desired amplitude state or when the amplitude state is a level state, the target level at point P in the demonstration area is easily determined obtained in that the level L _P at point P is equal to the quotient of L _v and a distance r that point P has from virtual source 700. The target 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. To calculate the distance r, a coordinate transformation of the virtual coordinates into the coordinates of the screening room or a coordinate transformation of the screening room coordinates of point P into the virtual coordinates must typically be carried out, which is known to those skilled in the field of wave field synthesis.

If, on the other hand, the virtual source is an infinitely distant virtual source that generates plane waves at point P, the distance between point P and the source is not required to determine the desired amplitude state, since this is in any case almost infinite. In this case, only information about the type of source is required. The target level at point P is then equal to the level which is assigned to the plane wave field which is generated by the virtual source which is infinitely distant.

Fig. 7 shows a diagram for explaining the actual amplitude state. In particular, different loudspeakers 808 are drawn in FIG. 7b, all of which are fed with their own loudspeaker signal which, for. B. has been generated by the wave field synthesis module 810 of FIG. 8. Furthermore, each loudspeaker is modeled as a point source that outputs a concentric wave field. The law of the concentric wave field is again that the level drops according to 1 / r. In order to calculate the actual amplitude state (without measurement), the signal generated by the loudspeaker 808 directly on the loudspeaker membrane or the level of this signal can be based on the loudspeaker characteristics and the component signal in the loudspeaker signal LSn, which goes back to the virtual source under consideration. Furthermore, on the basis of the coordinates of the point P and the location information relating to the position of the loudspeaker LSn, the distance between P and the loudspeaker membrane 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 which is directed to the virtual source under consideration goes back and has been emitted by the loudspeaker LSn.

A corresponding procedure can also be carried out for the other loudspeakers of the loudspeaker array, so that there is a number of "partial level values" for point P, which represent a signal contribution from the virtual source under consideration, which has reached the listener at point P from the individual loudspeakers By combining these partial level values, the entire actual amplitude state at point P is then obtained, which, as has been explained, can then be compared with the target amplitude state by a correction value, which is preferably multiplicative, but which, in principle, is additive or subtractive could be get.

According to the invention, the desired level for a 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 the point in the demonstration area that is being viewed lies in the middle of the wave field synthesis system. At this point it should be pointed out that an improvement is achieved even if the point on which the calculation of the target amplitude state is based does not correspond directly to the point which was used to determine the actual amplitude state is. After the best possible level artifact reduction for as large a number of points in the demonstration area as possible, it is in principle sufficient that a target amplitude state for any point in the demonstration area and that an actual amplitude condition is also determined for any point in the demonstration area, but it is preferred that the point to which the actual amplitude condition is related is in a zone around the point for which the Desired amplitude condition has been determined, this zone is preferably less than 2 meters for normal cinema applications. For best results, these points should essentially coincide.

According to the invention, after the individual levels of the loudspeakers have been calculated in accordance with customary wave field synthesis algorithms, the level practically generated by superimposition at this point, which is called the optimum point in the demonstration area, is calculated. The levels of the individual speakers and / or sources are then corrected with this factor according to the invention. For computation-time-efficient applications, it is particularly preferred to calculate and save correction factors once for all positions in a specific array arrangement, in order then to access the table during operation in order to achieve computation time savings.

At this point, particular reference is made to FIG. 6b, in which the device 914 for summing is drawn in in order to supply the sum signal 916 on the output side, while the scaled object signals 912 are obtained on the input side, which, as can be seen from FIG. 6b, by Scaling the source signals of sources 1, 2, 3 with the corresponding audio object scaling values or correction values F1, F2, F3 can be obtained. At this point, it should also be pointed out that for the present invention of low-frequency channel generation, the version shown in FIG. 6b is preferred, in which scaling or manipulation or correction is already carried out at the audio object signal level and not at the component level, as shown in FIG. 6a. is carried out. Nevertheless, the concept of correction at component level shown in FIG The inventive concept of low-frequency channel generation can be combined in that at least the calculation of the audio object scaling values F1, F2, ..., Fn only has to be carried out once.

According to the invention, the subwoofer channel is thus scaled in a manner similar to the scaling of the overall volume of all loudspeakers in the reference point of the wave field synthesis reproduction system. The method according to the invention is therefore suitable for any number of subwoofer loudspeakers, all of which are scaled in such a way that they reach a reference volume at the center of the wave field synthesis system. The reference volume only depends on the position of the virtual sound source. With the known dependencies on the distance of the sound object from the reference point and the associated attenuation of the volume, the individual volume of the respective sound object for each subwoofer channel is preferably calculated. The delay of each source is calculated from the distance between the virtual source and the reference point of the volume scaling. Each subwoofer speaker reproduces the sum of all sound objects converted in this way. How the individual volumes of the subwoofer speakers add up depends on their position. The preferred positioning of Subwooferlautspre- brass, and the selection of the number are far from necessary subwoofers in 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 low-frequency channel, as shown with reference to FIG. 9, can be implemented in hardware or in software. Depending on the circumstances, the method according to the invention for level correction, as shown in FIG. 1, can be implemented in hardware or in software. The implementation can take place on a digital storage medium, in particular a floppy disk or CD with electronically readable control signals, which can interact with a programmable computer system in such a way that the method is carried out. 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 implemented as a computer program with a program code for carrying out the method if the computer program runs on a computer.

Claims

claims

A device for generating a low-frequency channel (940, 942, 944) for a low-frequency loudspeaker (1106, 1110), comprising the following features: a device (900) for providing a plurality of audio objects, an object signal and an object description being assigned to an audio object; means (906) for calculating an audio object scaling value for each audio object depending on the object description (904); means (910) for scaling each object signal with an associated audio object scaling value (908) to obtain a scaled object signal (912) for each audio object; means (914) for summing the scaled object signals to obtain a sum signal (916); and means (918) for providing the woofer channel (920, 940, 942, 944) for the woofer (1106, 1110) based on the sum signal (916).

The apparatus of claim 1, wherein the woofer is located at a predetermined speaker position (1108, 1112), the predetermined speaker position (1108) being different from a reference playback position (100), and wherein the means (918) for providing the Woofer is designed to depend on a speaker scaling value for the woofer from the predetermined loudspeaker position (1108) so that a low-frequency signal at the reference playback position (1100) has a volume that corresponds to a volume of the sum signal (916) within a predetermined tolerance range, and the device (918) is further designed to provide to scale the sum signal (916) with the loudspeaker scaling value in order to generate the low-frequency channel (920, 940, 942, 944).

3. Apparatus according to claim 1 or 2, wherein each object signal is a low frequency signal with an upper cutoff frequency less than or equal to 250 Hz.

4. Apparatus according to claim 1 or 2, in which the sum signal (916) has an upper cut-off frequency which is higher than 8 kHz, and in which the device (918) for providing the low-frequency channel is designed to perform low-pass filtering with a cut-off frequency lower or 250 Hz.

5. Apparatus according to any one of the preceding claims, in which an audio object of the plurality of audio objects comprises an object description comprising an audio object position, and in which the means (906) for calculating an audio object scaling value for the audio object is designed to match the audio object scaling value depending on the audio object position of the audio object and a reference playback position (1100) and depending on an object volume assigned to the audio object.

6. Device according to one of the preceding claims, in which a plurality of woofers for a plurality of woofers can be generated at predetermined woofer positions, and in which the device (918) is designed to provide, depending on the position of a woofer and depending on one To calculate a number of additional woofers for each woofer, a speaker scaling value so that a woofer signal, which is a superposition of output signals from all woofers at the reference position (1100), has a volume that corresponds to a volume of the sum signal (916) within a predetermined tolerance range.

7. Device according to one of the preceding claims, in which the device (906) for calculating audio object scaling values is further configured to calculate an audio object delay value for each audio object, which depends on an object position and a reference playback position, and in which the device (914) is configured for summing in order to delay each object signal or each scaled object signal before summing by the corresponding audio object delay value.

8. Device according to one of the preceding claims, in which the device (918) is designed to provide for calculating a woofer delay value for a woofer which is depends on a distance of the woofer from the reference playback position, and at which the device (918) for providing is further designed to take into account the woofer delay value when providing the woofer channel.

9. The device according to claim 2, in which a plurality of woofers are provided, and in which the device (918) is also designed to be provided in order to calculate the loudspeaker scaling values such that a loudspeaker scaling value is obtained for each woofer loudspeaker in accordance with the following equation:

(a _x + a ₂ + + a _n ) ^• s = LSref, where LSref is a reference volume at a reference playback position (1100), where s is the sum signal (916), where ai is the speaker scaling value of a first woofer, where a _{2 is} a speaker scaling value of a second woofer, and wherein a _{n is} a speaker scaling value of an nth woofer. 10. The apparatus of claim 9, wherein the speaker scaling value of a woofer depends on a distance of the woofer from the reference playback position (1100).

11. Device according to one of the preceding claims, which is designed to work in a wave field synthesis system with a wave field synthesis module (810) and an array (800) of loudspeakers (808) for the sound supply of a demonstration area (802), the wave field synthesis module being formed is an audio signal assigned to a virtual sound source and a source assigned to the virtual sound source. Receiving position information and calculating component signals for the speakers based on the virtual source in consideration of speaker position information, and wherein the means (906) for calculating the audio object scaling values (908) comprises means (100) for determining a correction value as the audio object scaling value, the means (100) is configured to determine to calculate the audio object scaling value to be based on a target amplitude state in the demonstration area, the target amplitude state depending on a position of the virtual source or a kind of the virtual source, and further on an actual amplitude state in the demonstration area based on the component signals for the speakers due to the virtual source.

12. The apparatus of claim 11, wherein the means (100) for determining the correction value (104) is designed to calculate the desired amplitude state for a predetermined point in the demonstration area (500) and the actual amplitude state for one Zone 'in the demonstration area to be determined (502), which is equal to the predetermined point or extends within a tolerance range around the predetermined point.

13. The apparatus of claim 12, wherein the predetermined tolerance range is a sphere with a radius less than 2 meters around the predetermined point.

14. Device according to one of claims 11 to 13, in which the virtual source is a source for plane waves, and in which the device (100) for determining the correction value is designed to determine a correction value in which an amplitude state of the audio signal assigned to the virtual source is equal to the desired amplitude state.

15. The device according to one of claims 11 to 14, in which the virtual source is a point source, and in which the device (100) for determining the correction factor is designed to operate on the basis of a desired amplitude state which is equal to a quotient from an amplitude state of the audio signal assigned to the virtual source and the distance between the demonstration area and the position of the virtual source.

16. The device according to one of claims 11 to 15, wherein the device (100) for determining the correction value is designed to work based on an actual amplitude state, for the determination of which a loudspeaker transmission function of the loudspeaker (808) takes into account is.

17. Device according to one of claims 11 to 16, wherein the means (100) for determining the correction 'is designed to calculate an attenuation value for each speaker, which depends on the position of the speaker and a point to be considered in the demonstration area and in which the means (100) for determining is further configured to weight the component signal of a loudspeaker with the attenuation value for the loudspeaker, to obtain a weighted component signal, and to further sura component signals or correspondingly weighted component signals from other loudspeakers - Mieren in order to obtain the actual amplitude state at the point under consideration on which the correction value (104) is based.

18. Device according to one of claims 11 to 17, wherein the device (106) is designed for manipulation in order to use the correction value (104) as a correction factor, which is equal to a quotient of the actual amplitude state and the target amplitude state.

19. The apparatus of claim 18, wherein the device (106) is designed for manipulation in order to scale the audio signal associated with the virtual source before the component signals are calculated by the wave field synthesis module (810) with the correction factor.

20. Device according to one of claims 11 to 19, in which the target amplitude state is a target sound level, and in which the actual amplitude state is an actual sound level.

21. The apparatus of claim 20, wherein the target sound level and the actual sound level are based on a target sound strength and an actual sound strength, respectively, the sound strength being a measure of an energy that falls on a reference surface over a period of time. '

22. The apparatus of claim 20 or 21, wherein the means (100) for determining the correction value is designed to the desired amplitude state by sample-by-square squaring samples of the audio signal associated with the virtual source and by summing a number of squared samples, the number being a measure of an observation time, and in which the device (100) for determining the correction value is further configured to calculate the actual amplitude state by squaring each component signal and sampling an application the number of squared samples is added, which is equal to the number of added squared samples for calculating the desired amplitude state, and further addition results from the component signals are added to obtain a measure of the actual amplitude state.

23. Device according to one of claims 11 to 22, in which the device (100) for determining the correction value (104) has a look-up table (400) in which position correction factor value pairs are stored, a correction factor of a value pair of an arrangement of the loudspeakers in the array depends on loudspeakers and a position of a virtual source, and the correction factor is chosen such that a deviation between an actual amplitude state due to the virtual source at the assigned position and a desired amplitude state when using the Correction factor is at least reduced by the device (106) for manipulation.

24. The apparatus of 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 more correction factors from position correction factor value pairs Position or positions next to the current position.

25. A method for generating a woofer channel (940, 942, 944) for a woofer (1106, 1110), comprising the following steps: providing (900) a plurality of audio objects, an object signal and an object description being assigned to an audio object; Computing (906) an audio object scaling value for each audio object depending on the object description (904); Scaling (910) each object signal with an associated audio object scaling value (908) to obtain a scaled object signal (912) for each audio object; Summing (914) the scaled object signals to obtain a sum signal (916); and

Provision (918) of the bass channel (920, 940, 942, 944) for the bass speaker (1106, 1110) based on the sum signal (916).

26. Computer program with a program code for performing the method according to claim 25, when the program runs on a computer.