EP1872620B9 - Apparatus and method for controlling a plurality of loudspeakers by means of a graphic user interface - Google Patents

Abstract

In a reproduction area where there are at least three directional groups, each of which has speakers, triggering of the speakers is achieved in that a source path from a first directional group position to a second directional group position is initially obtained along with movement information for the source path. Subsequently, a source path parameter is calculated for different points in time on the basis of the movement information, the source path parameter indicating a position of an audio source on the source path. In addition, a path modification command is received to define a compensation path to the third directional zone, a value of the source path parameter further being stored at a location where the compensation path deviates from the source path, and being used, along with a compensation parameter, for calculating weighting factors for the speakers of the three directional groups.

Description

The present invention relates to audio engineering, and more particularly to the positioning of sound sources in systems comprising delta stereophonic systems (DSS) or wave field synthesis systems or both systems.

Typical public address systems for supplying a relatively large environment, such as in a conference room on the one hand or a concert hall in a hall or even in the open air on the other hand all suffer from the problem that due to the commonly used small number of speaker channels a faithful reproduction of the sound sources anyway eliminated , But even if a left channel and a right channel are used in addition to the mono channel, you always have the problem of the level. So, of course, the back seats, so the seats that are far away from the stage, need to be sounded as well as the seats that are close to the stage. If z. B. speakers are arranged only at the front of the auditorium or on the sides of the audience, it is inherently problematic that people who sit close to the speaker perceive the speaker as exaggeratedly loud, so that people hear something at the very back. In other words, due to the fact that individual utility loudspeakers are perceived as point sources in such a sounding scenario, there will always be people who say that it is too loud while the other people say that it is too quiet. The people who are usually too loud are the people very close to the point source type speakers, while those who are too quiet will be very far away from the speakers.

In order to at least somewhat avoid this problem, it is therefore attempted to arrange the loudspeakers higher, that is, above the persons who are sitting close to the loudspeakers, so that they at least do not fully hear the complete sound, but that a considerable amount of the sound is lost Sound of the speaker spreads over the heads of the audience and thus on the one hand not perceived by the audience in front and on the other hand still provides for the audience further back a sufficient level. Furthermore, this problem is countered by the linear array technique.

Other possibilities are that in order not to overload the people in the front rows, so close to the speakers, a low level, so of course, further back in the room, there is a risk that everything is again too quiet ,

Regarding direction perception, the whole thing is even more problematic. For example, a single monaural loudspeaker does not allow directional perception in a conference room. It only allows directional perception if the location of the loudspeaker corresponds to the direction. This is inherent in the fact that there is only one loudspeaker channel. However, even if there are two stereo channels, you can at most between the left and the right channel back and forth, so to speak panning. This may be beneficial if there is only one source. However, if there are several sources, the localization is only roughly possible in a small area of the auditorium, as with two stereo channels. You also have a sense of direction in stereo, but only in the sweet spot. In the case of several sources, this directional impression becomes more and more blurred, especially as the number of sources increases.

In other scenarios, the loudspeakers in such medium to large auditoriums, which are supplied with stereo or mono mixes, are arranged above the listeners, so that they can not reproduce any directional information of the source anyway.

Although the sound source, so z. For example, when a speaker or theater player is on stage, he is perceived from the side or center loudspeakers. On a natural direction perception is still omitted here so far. One is already satisfied when it is still loud enough for the rear viewers, and when it is not unbearably loud for the front viewers.

In certain scenarios, so-called "support loudspeakers" are also used, which are positioned near a sound source. This is an attempt to restore the natural hearing location. These support speakers are normally driven without delay, while the stereo sound is delayed through the supply speakers, so that the support speaker is first perceived and thus according to the law of the first wavefront localization is possible. However, support speakers also have the problem that they are perceived as a point source. This leads, on the one hand, to a difference to the actual position of the sound emitter and, moreover, there is the danger that everything is too loud for the front viewers, while everything is too quiet for the back viewers.

On the other hand support speakers only allow a real direction perception when the sound source, so z. B. a speaker is located directly in the vicinity of the support speaker. This would work if a support speaker is installed in the lectern, and a speaker is always on the lectern, and in this playback room it is impossible that somebody stands next to the lectern and plays something for the audience.

Thus, in the case of a local difference between the support loudspeaker and the sound source, an angle error occurs in the direction perception which leads to further uncertainty, especially for listeners who are perhaps not accustomed to supporting loudspeakers, but are used to stereo reproduction. It has been found that especially when working with the law of the first wave front and uses a support speaker, it is better to disable the support speakers when the real sound source, so the speaker z. B. has removed too far from the support speakers. In other words, this point is related to the problem that the support speaker can not be moved, so that in order not to create the above-mentioned uncertainty in the audience, the support speaker is completely disabled when the speaker has moved too far from the support speaker.

As has already been stated, support loudspeakers usually use conventional loudspeakers, which in turn have the acoustic properties of a point source-just as the supply loudspeakers-resulting in an excessive level which is often perceived as unpleasant in the immediate vicinity of the systems.

In general, therefore, the aim is to create an auditory perception of source positions for public address scenarios, as they take place in the theater / drama area, whereby conventional normal sound systems, which are only designed to provide sufficient coverage of the entire audience area with loudness directional speaker systems and their control should be added.

Typically, medium to large auditoriums are supplied with stereo or mono and occasionally with 5.1 surround technology. Typically, the speakers are located beside or above the listener and can play correct directional information of the sources only for a small audience area. Most listeners get a wrong directional impression.

In addition, however, there are also delta stereophonic systems (DSS), which produce a directional reference in accordance with the law of the first sound wave front. The DD 242954 A3 discloses a large-capacity public address system for larger rooms and areas in which the action or performance and reception or listening rooms are directly adjacent to each other or identical. The sound is made according to the principles of running time. In particular occurring misalignments and jump effects in movements, which are particularly disturbing for important solo sound sources, are avoided by a maturity graduation is realized without limited source areas and the sound power of the sources is taken into account. A control device which is connected to the deceleration or amplification means, controls this analogous to the sound paths between the source and Schallstrahlerorten. For this purpose, a position of a source is measured and used to adjust loudspeakers according to gain and delay accordingly. A playback scenario comprises a plurality of mutually delimited speaker groups, which are each controlled.

Delta stereophony means that one or more directional loudspeakers are present in the vicinity of the real sound source (eg on a stage), which detect a locator in large parts of the audience area. It is an almost natural direction perception possible. These speakers are timed to the directional speaker to realize the location reference. This is always the direction-giving Speakers perceived and thus a localization possible, this relationship is also referred to as the "law of the first wave front".

The support speakers are perceived as a point source. There is a difference to the actual position of the sound emitter, that is, the original source when e.g. a soloist is not directly in front of or next to the support speaker, but is located away from the support speaker.

If, therefore, a sound source moves between two supporting loudspeakers, then it must be dazzled between differently arranged such support loudspeakers. This affects both the level and the time. In contrast, wave direction synthesis systems can be used to achieve a real directional reference via virtual sound sources.

For a better understanding of the present invention, the wave field synthesis technique will be discussed in more detail below.

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

Due to the enormous demands of this method on computer performance and transmission rates, wave field synthesis has rarely been used in practice. Only the advances in the areas of microprocessor technology and audio coding allow today the use of this technology in concrete applications. The first professional products are expected this year. In some Years should also come first wave field synthesis applications for the consumer sector on the market.

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. If the sources to be reproduced are in a room with reflective walls, reflections must also be reproduced as additional sources via the loudspeaker array. The cost of the calculation therefore depends heavily on the number of sound sources, the reflection characteristics of the recording room and the number of speakers.

The advantage of this technique is in particular that a natural spatial sound impression over a large area of the playback room is possible. In contrast to the known techniques, the direction and distance of sound sources are reproduced very accurately. To a limited extent, virtual sound sources can even be positioned between the real speaker array and the listener.

Although wavefield synthesis works well for environments whose characteristics are known, irregularities occur when the texture changes, or when wave field synthesis is performed based on environmental conditions that do not match the actual nature of the environment.

An environmental condition can be described by the impulse response of the environment.

This will be explained in more detail with reference to the following example. It is assumed that a loudspeaker emits a sound signal against a wall whose reflection is undesirable. For this simple example, the space compensation using wavefield synthesis would be to first determine the reflection of that wall to determine when a sound signal reflected from the wall will return to the loudspeaker and what amplitude this reflected sound signal will be Has. If the reflection from this wall is undesirable, then with the wave field synthesis it is possible to eliminate the reflection from this wall by impressing the loudspeaker with a signal of opposite amplitude to the reflection signal in addition to the original audio signal, so that the traveling compensating wave is the Reflectance wave extinguished, so that the reflection from this wall in the environment that is considered, is eliminated. This can be done by first computing the impulse response of the environment and determining the nature and position of the wall based on the impulse response of that environment, the wall being interpreted as a source of mirrors, that is, a sound source reflecting an incident sound.

If the impulse response of this environment is first measured and then the compensation signal is calculated, which must be impressed on the audio signal superimposed on the loudspeaker, then the reflection from this wall is canceled take place in such a way that a listener in this environment has the sound impression that this wall does not exist at all.

Decisive for an optimal compensation of the reflected wave, however, is that the impulse response of the room is accurately determined, so that no overcompensation or undercompensation occurs.

The wave field synthesis thus allows a correct mapping of virtual sound sources over a large playback area. At the same time it offers the sound engineer and sound engineer new technical and creative potential in the creation of even complex soundscapes. Wavefield synthesis (WFS or sound field synthesis), as developed at the TU Delft in the late 1980s, represents a holographic approach to sound reproduction. The basis for this is the Kirchhoff-Helmholtz integral. This states that any sound fields within a closed volume can be generated by means of a distribution of monopole and dipole sound sources (loudspeaker arrays) on the surface of this volume. Details can be found in MM Boone, ENG Verheijen, PF. Tol, "Spatial Sound-Field Reproduction by Wave-Field Synthesis", Delft University of Technology Laboratory of Seismics and Acoustics, Journal of J. Audio Eng. Soc., Vol. 43, No. 12, December 1995 and Diemer de Vries, "Sound Reinforcement by Wavefield Synthesis: Adaptation of the Synthesis Operator to the Loudspeaker Directivity Characteristics", Delft University of Technology Laboratory of Seismics and Acoustics, Journal of J. Audio Eng. Soc., Vol. 44, No. 12, December 1996 ,

In the wave field synthesis, a synthesis signal for each loudspeaker of the loudspeaker array is calculated from an audio signal that emits a virtual source at a virtual position, the synthesis signals being designed in such a way that amplitude and phase Wave resulting from the superposition of the individual sound waves output by the loudspeakers present in the loudspeaker array corresponding to the wave that would originate from the virtual source at the virtual position, if this virtual source at the virtual position is a real source with a real position would.

Typically, multiple virtual sources exist at different virtual locations. The computation of the synthesis signals is performed for each virtual source at each virtual location, typically resulting in one virtual source in multiple speaker synthesis signals. Seen from a loudspeaker, this loudspeaker thus receives several synthesis signals, which go back to different virtual sources. A superimposition of these sources, which is possible due to the linear superposition principle, then gives the reproduced signal actually emitted by the speaker.

The possibilities of wave field synthesis can be better exploited the more closed the speaker arrays are, ie. H. the more individual speakers can be arranged as close to each other as possible. However, this also increases the computing power which a wave field synthesis unit has to accomplish, since channel information also typically has to be taken into account. This means in more detail that from each virtual source to each speaker in principle a separate transmission channel is present, and that in principle there may be the case that each virtual source leads to a synthesis signal for each speaker, or that each speaker a number of synthesis signals which equals the number of virtual sources.

In addition, it should be noted at this point that the quality of the audio playback increases with the number of speakers provided. This means that the audio playback quality is the better and more realistic is the more speakers in the loudspeaker array (s) are present.

In the above scenario, the finished and analog-to-digital converted reproduction signals for the individual loudspeakers could, for example, be transmitted via two-wire lines from the wave field synthesis central unit to the individual loudspeakers. Although this would have the advantage that it is almost ensured that all speakers work in sync, so that here for synchronization purposes, no further action would be required. On the other hand, the wave field synthesis central unit could always be made only for a special reproduction room or for a reproduction with a fixed number of loudspeakers. This means that a separate wave field synthesis central unit would have to be produced for each reproduction space, which has to accomplish a considerable amount of computing power, since the calculation of the audio reproduction signals has to be at least partially parallel and in real time, in particular with regard to many loudspeakers or many virtual sources ,

The delta stereophony is particularly problematic, since position artifacts due to phase and level errors occur when fading between different sound sources. Furthermore, phase errors and mislocalizations occur at different moving speeds of the sources. Moreover, the crossfading from one support speaker to another support speaker is associated with a great deal of programming, while at the same time there are problems maintaining the overview of the entire audio scene, especially when multiple sources are faded from different support speakers, and when In particular, many support speakers that can be controlled differently, exist.

Furthermore, on the one hand wave field synthesis on the one hand and delta stereophony on the other hand are actually opposing methods, while both systems can have advantages in different applications.

Thus, the delta stereophony is much less expensive in terms of calculating the loudspeaker signals than the wave field synthesis. On the other hand, it is possible to work without artifacts due to wave field synthesis. However, wave field synthesis arrays can not be widely used because of space requirements and the requirement for an array of closely spaced loudspeakers. Particularly in the field of stage technology, it is very problematic to arrange a loudspeaker band or a loudspeaker array on the stage, since such loudspeaker arrays can be badly hidden and thus are visible and impair the visual impression of the stage. This is especially problematic when, as is usually the case with theater / musical performances, the visual impression of a stage takes precedence over all other matters, and more particularly, from tone or tone generation. On the other hand, wave field synthesis does not predetermine a fixed grid of supporting loudspeakers, but instead a movement of a virtual source can take place continuously. A support speaker, however, can not move. However, the movement of the support speaker can be generated virtually by directional glare.

Limitations of delta stereophony are thus in particular that the number of possible support speakers, which are housed in a stage, for reasons of expense (depending on the set) and sound management reasons is limited. In addition, each support speaker, if it is to work on the principle of the first wavefront, requires additional speakers that produce the necessary volume. This is precisely the advantage of delta stereophony, which is actually a relatively small and thus well placed speaker for localization However, while many other nearby loudspeakers are designed to produce the necessary volume for the listener, who can sit quite aft in a relatively large audience room.

It is thus possible to assign all loudspeakers on the stage to different directional areas, each directional area having a localization loudspeaker (or a small group of simultaneously controlled localization loudspeakers) which is driven without or with only a slight delay, while the other loudspeakers of the directional group have the same signal, but timed to produce the required volume while the localization speaker had delivered the well-defined localization.

After you need a sufficient volume, the number of speakers in a direction group downwards is not arbitrarily reducible. On the other hand, one would like to have many directional areas in order to at least strive for a continuous sound supply. Due to the fact that each directional area next to the localization speaker also requires enough loudspeakers to generate sufficient volume, the number of directional areas is limited when a stage space is divided into contiguous non-overlapping directional areas, where each directional area is a localization speaker or a small one Group of closely adjacent Lokalisationslautsprechern is assigned.

Typical delta stereophonic concepts are based on blending between two locations when a source is to move from one location to another. This concept is problematic when z. B. should be intervened manually in a programmed set-up, or if an error correction has to take place. For example, it turns out that a singer does not follow the If the agreed route is over the stage, but rather different, there will be an increasing difference between the perceived position and the actual position of the singer, which of course is not desirable.

If a corrective intervention option is desired for such a case, a user could enter for correction purposes that the audio position should correspond at a certain time or directly to the actual position of the singer on the stage. However, this would entail a hard swap of the source, which in some circumstances would lead to even greater artifacts than the audio source and perceived audio source mismatch.

In order to avoid such a jump, one could complete the transition process that has already begun, and then, starting from a position within a directional area, that is, after a complete crossfading operation, correct the target of the next crossfading operation. This would ensure that no hard jumps occur. However, it is disadvantageous in this concept that there is no possibility of intervening within a crossfading process. So it will come to a significant delay, especially if a relatively long crossfade process is currently running, namely z. From a source on the far left of the stage to a source on the right on a large stage. As a result, there is a relatively long period of time in which the perceived one deviates from the actual position of the audio source. Furthermore, of course, the actual position, which may already be moving again, must be obtained, which can only be accomplished by a relatively fast passage of a source across the stage to the sought position. This very fast sweep, in turn, can lead to artifacts, or at least result in a user wondering why the perceived audio position moved so strongly, although the singer himself did not move or only slightly.

The object of the present invention is to provide a flexible yet artifact-reduced concept for controlling a plurality of loud speakers.

This object is achieved by a device for controlling a plurality of loudspeakers according to claim 1, a method for controlling a plurality of loudspeakers according to claim 15 or a computer program according to claim 16.

The present invention is based on the recognition that an artifact-reduced and rapid manual intervention in the course of the movement of sources is achieved by allowing a compensation path on which a source can move. The compensation path differs from the normal source path in that it does not begin at a direction group position, but begins at a connection line between two directional groups, at any point in that connection line, and extends from there to a new destination directional group. As a result, it is no longer possible to describe a source by the specification of two directional groups, but the source must be described by at least three directional groups, wherein in a preferred embodiment of the present invention, a position description of the source identification of the three directional groups involved and two blend Factors, where the first blend factor indicates where on the source path has been "bent", and where the second blend factor indicates where the source is currently on the compensation path, that is, how far the source already is from the source path or how long the source has to run until the new target direction.

The calculation of the weighting factors for the loudspeakers of the three directional areas involved takes place according to the invention based on the source path, the stored value of the source path parameter and information about the compensation path. The information about the compensation path may include the new target per se or the second blend factor. Further, for the movement of the source on the compensation path, a predefined speed may be used, which may be predetermined by the system, since the movement on the compensation path is typically a compensation movement which does not depend on the audio scene but which is there, something in a preprogrammed one Change scene or correct. For this reason, the velocity of the audio source on the compensation path will typically be relatively fast, but not so fast that problematic audible artifacts will occur.

In a preferred embodiment of the present invention, the means for calculating the weighting factors is configured to calculate weighting factors that depend linearly on the glare factors. However, alternative concepts such as non-linear dependencies in terms of a sine ² function or a cosine ² function can also be used.

In a preferred embodiment of the present invention, the device for controlling the plurality of loudspeakers further comprises a jump compensator, which preferably operates hierarchically based on various provided compensation strategies to avoid a hard swap of the source by means of a jump compensation path.

A preferred embodiment is based on having to go away from the adjoining directional areas that define the "raster" of well locatable motion points on a stage. That was due the requirement that the directional areas are not overlapping, so clear control conditions are available, the number of directional areas limited because each directional area in addition to the Lokalisationslautsprecher also needed a sufficiently large number of speakers, in addition to the first wave front, by the Lokalisationslautsprecher is generated, also to produce a sufficient volume.

Preferably, a division of the stage space is made in overlapping directional areas, thereby creating the situation that a speaker may belong not only to a single directional area, but to a plurality of directional areas, such as at least the first directional area and the second directional area and if applicable to a third or a further fourth directional area.

The affiliation of a loudspeaker to a directional area is experienced by the loudspeaker in that, if it belongs to a directional area, it is assigned a specific loudspeaker parameter which is determined by the directional area. Such a speaker parameter may be a delay which will be small for the localization speakers of the directional area and will be greater for the other speakers of the directional area. Another parameter can be a scaling or a filter curve, which can be determined by a filter parameter (equalizer parameter).

Typically, each loudspeaker on a stage will have its own loudspeaker parameter, depending on which direction it belongs to. These values of loudspeaker parameters, which depend on which directional area the loudspeaker belongs to, typically become heuristic in a sound check by a sound engineer, partly empirical for a particular room and then when the speaker is working, inserted.

However, after allowing one speaker to belong to multiple directional areas, the speaker parameter speaker has two different values. So a loudspeaker, if it belongs to directional area A, would have a first delay DA. However, if the speaker belongs to the directional area B, the speaker would have a different delay value DB.

Thus, when it is desired to go from the directional group A to a directional group B, or when to reproduce a position of a sound source that is between the directional area position A of the directional group A and the directional area position B of the directional group B, the speaker parameters are used. to use the audio signal for this speaker and for the currently viewed audio source. According to the invention, the actually indissoluble contradiction, namely that a loudspeaker has two different delay settings, scaling settings or filter settings, is eliminated by using the loudspeaker parameter values for all directional groups involved for calculating the audio signal to be output by the loudspeaker become.

Preferably, the calculation of the audio signal depends on the distance measure, that is, the spatial position between the two direction group positions, the distance measure will typically be a factor lying between zero and one, where a factor of zero determines that the speaker is at the direction group position A is, while a factor of one determines that the speaker is on the direction group position B.

In a preferred embodiment of the present invention, depending on how fast a source moving between the directional group position A and the directional group position B, real speaker parameter value interpolation or fading of an audio signal based on the first speaker parameter into a speaker signal based on the second speaker parameter. Particularly in the case of delay settings, that is to say with a loudspeaker parameter which reproduces a delay of the loudspeaker (with respect to a reference delay), special attention must be paid to whether interpolation or crossfading is used. Namely, if an interpolation is used in a very fast movement of a source, this will result in audible artifacts that will result in a rapidly rising tone or a rapidly falling tone. For fast movements of sources, therefore, a transition is preferred, which leads to comb filter effects, but which are not or barely audible due to the fast transition. On the other hand, for slow motion velocities, interpolation is preferred in order to avoid the comb filter effects associated with slow crossfades, which are also clearly audible. Furthermore, in order to avoid further artifacts, such as crackling, which would be audible in the "switching" from interpolation to crossfading, the switching is not abruptly made, ie from one sample to the next, but a transition is effected, controlled by a switching parameter within a fade range that will include multiple samples based on a fade function, which is preferably linear but which may be nonlinear, e.g. B. may be trigonometric, made.

In another preferred embodiment of the present invention, a graphical user interface is provided that graphically illustrates ways of a sound source from one directional area to another directional area. Preferably, compensation paths are also taken into account to allow rapid changes in the path of a source, or hard ones Jumps from sources, as they might occur in scene breaks, to avoid. The compensation path ensures that a path of a source can not only be changed when the source is in the directional position, but also when the source is between two directional positions. This ensures that a source can turn off its programmed path between two directional positions. In other words, this is achieved in particular by the fact that the position of a source can be defined by three (adjacent) directional areas, in particular by identification of the three directional areas as well as the indication of two glare factors.

In another preferred embodiment of the present invention, where wave field synthesis loudspeaker arrays are possible, a wave field synthesis array is mounted in the sound space, which also represents a directional region having a directional location position by indicating a virtual position (e.g., in the center of the array).

This is the decision of the user of the system, whether it is a sound source is a wave field synthesis sound source or a delta stereophonic sound source.

Thus, a user-friendly and flexible system is provided which allows a flexible division of a space into directional groups, as directional group overlaps are allowed, with loudspeakers in such an overlapping zone being supplied with loudspeaker parameters derived from the loudspeaker parameters corresponding to the directional areas, with respect to their loudspeaker parameters Derivative is preferably done by interpolation or crossfading. Alternatively, a hard decision could be made, for example, when the source is closer to the one directional area, the to take a loudspeaker parameter so as to take the other loudspeaker parameter when the source is closer to the other directional field, whereby the then occurring hard jump for artifact reduction could simply be smoothed. However, pitch-controlled blending or pitch-controlled interpolation is preferred.

Fig. 1

eine Einteilung eines Beschallungsraums in überlappende Richtungsgruppen;

Fig. 2a

eine schematische Lautsprecherparameter-Tabelle für Lautsprecher in den verschiedenen Bereichen;

Fig. 2b

eine speziellere Darstellung der für die Lautsprecherparameter-Verarbeitung nötigen Schritte für die verschiedenen Bereiche;

Fig. 3a

eine Darstellung einer linearen Zwei-Wege-Überblendung;

Fig. 3b

eine Darstellung einer Drei-Wege-Überblendung;

Fig. 4

ein schematisches Blockschaltbild der Vorrichtung zum Ansteuern einer Mehrzahl von Lautsprechern mit einem DSP;

Fig. 5

eine detailliertere Darstellung der Einrichtung zum Berechnen eines Lautsprechersignals von Fig. 4 gemäß einem bevorzugten Ausführungsbeispiel;

Fig. 6

eine bevorzugte Implementierung eines DSP zur Implementierung der Delta-Stereophonie;

Fig. 7

eine schematische Darstellung des Zustandekommens eines Lautsprechersignals aus mehreren Einzel-Lautsprechersignalen, die von verschiedenen Audioquellen herrühren;

Fig. 8

eine schematische Darstellung einer Vorrichtung zum Steuern einer Mehrzahl von Lautsprechern, die auf einer graphischen Benutzerschnittstelle basieren kann;

Fig. 9a

ein typisches Szenario der Bewegung einer Quelle zwischen einer ersten Richtungsgruppe A und einer zweiten Richtungsgruppe C;

Fig. 9b

eine schematische Darstellung der Bewegung gemäß einer Kompensationsstrategie, um einen harten Sprung einer Quelle zu vermeiden;

Fig. 9c

eine Legende für die Figuren 9d bis 9i;

Fig. 9d

eine Darstellung der Kompensationsstrategie "InpathDual";

Fig. 9e

eine schematische Darstellung der Kompensations-strategie "InpathTriple";

Fig. 9f

eine schematische Darstellung der Kompensations-strategien AdjacentA, AdjacentB, AdjacentC;

Fig. 9g

eine schematische Darstellung der Kompensations-strategien OutsideM und OutsideC;

Fig. 9h

eine schematische Darstellung eines Cader-Kompensationspfads;

Fig. 9i

eine schematische Darstellung von drei Cader-Kompensationsstrategien;

Fig. 10a

eine Darstellung zur Definition des Quellenpfads (DefaultSector) und des Kompensationspfads (CompensationSector);

Fig. 10b

eine schematische Darstellung der Rückwärtsbewegung einer Quelle mit dem Cader mit geändertem Kompensationspfad;

Fig. 10c

eine Darstellung der Auswirkung von BlendAC auf die anderen Blend-Faktoren;

Fig. 10d

eine schematische Darstellung zum Berechnen der Blend-Faktoren und damit der Gewichtungsfaktoren abhängig von BlendAC;

Fig. 11a

eine Darstellung einer Input/Output-Matrix für dynamische Quellen; und

Fig. 11b

eine Darstellung einer Input/Output-Matrix für statische Quellen.

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

Fig. 1

a division of a sound space into overlapping directional groups;

Fig. 2a

a schematic speaker parameter table for speakers in the different areas;

Fig. 2b

a more specific illustration of the steps required for speaker parameter processing for the various areas;

Fig. 3a

a representation of a linear two-way crossfade;

Fig. 3b

a representation of a three-way crossfade;

Fig. 4

a schematic block diagram of the device for driving a plurality of speakers with a DSP;

Fig. 5

a more detailed representation of the device for calculating a loudspeaker signal of Fig. 4 according to a preferred embodiment;

Fig. 6

a preferred implementation of a DSP for implementing delta stereophony;

Fig. 7

a schematic representation of the occurrence of a loudspeaker signal from a plurality of individual loudspeaker signals resulting from different audio sources;

Fig. 8

a schematic representation of an apparatus for controlling a plurality of speakers, which can be based on a graphical user interface;

Fig. 9a

a typical scenario of the movement of a source between a first directional group A and a second directional group C;

Fig. 9b

a schematic representation of the movement according to a compensation strategy to avoid a hard jump of a source;

Fig. 9c

a legend for the FIGS. 9d to 9i ;

Fig. 9d

a representation of the compensation strategy "InpathDual";

Fig. 9e

a schematic representation of the compensation strategy "InpathTriple";

Fig. 9f

a schematic representation of the compensation strategies AdjacentA, AdjacentB, AdjacentC;

Fig. 9g

a schematic representation of the compensation strategies OutsideM and OutsideC;

Fig. 9h

a schematic representation of a Cader compensation path;

Fig. 9i

a schematic representation of three Cader compensation strategies;

Fig. 10a

a representation for the definition of the source path (DefaultSector) and the compensation path (CompensationSector);

Fig. 10b

a schematic representation of the backward movement of a source with the Cader with changed compensation path;

Fig. 10c

an illustration of the effect of BlendAC on the other blend factors;

Fig. 10d

a schematic representation for calculating the blend factors and thus the weighting factors depending on BlendAC;

Fig. 11a

a representation of an input / output matrix for dynamic sources; and

Fig. 11b

a representation of an input / output matrix for static sources.

Fig. 1 shows a schematic representation of a stage space, which is divided into three directional areas RGA, RGB and RGC, each directional area comprises a geometric area 10a, 10b, 10c of the stage, the range limits are not critical. The only thing that matters is whether there are loudspeakers in the various areas that are in Fig. 1 are shown. In area I located speakers belong to the in Fig. 1 shown example only to the directional group A, wherein the position of the directional group A is designated at 11a. By definition, the directional group RGA is assigned the position 11a at which preferably the loudspeaker of the directional group A is present, which according to the law of the first wavefront has a delay which is smaller than the delays of all other loudspeakers assigned to the directional group A. In area II there are speakers, which are assigned only to the direction group RGB, the by definition has a direction group position 11b at which the support loudspeaker of the directional group RGB is located, which has a smaller delay than all other loudspeakers of the directional group RGB. In a region III, in turn, there are only loudspeakers associated with the directional group C, the directional group C having by definition a position 11c at which the supporting loudspeaker of the directional group RGC is arranged, with a shorter delay than all the other loudspeakers of the directional group RGC will send.

In addition, exists in the in Fig. 1 shown division of the stage space in direction areas an area IV, in which speakers are arranged, which are assigned to both the directional group RGA and the directional group RGB. Accordingly, a region V exists in which loudspeakers are arranged which are assigned to both the directional group RGA and the directional group RGC.

Further, there exists an area VI in which speakers are arranged, which are assigned to both the directional group RGC and the directional group RGB. Finally, there is an overlapping area between all three directional groups, this overlapping area VII comprising loudspeakers associated with both the directional group RGA and the directional group RGB as well as the directional group RGC.

Typically, each speaker in a stage setting is assigned a speaker parameter or a plurality of speaker parameters by the sound engineer or sound director. These speaker parameters include, as in Fig. 2a in column 12, a delay parameter, a scale parameter and an EQ filter parameter. The delay parameter D indicates how much of an audio signal that is output from this speaker will be relative to a reference value (which is true for another speaker, but not necessarily real) must be) is delayed. The Scale parameter indicates how much of an audio signal is amplified or attenuated by this speaker compared to a reference value.

The EQ filter parameter specifies how the frequency response of an audio signal to be output from a speaker should look like. Thus, for certain speakers, there may be a desire to amplify the high frequencies as compared to the low frequencies, which would make sense, for example, if the speaker is near a stage part that has a strong low-pass characteristic. On the other hand, for a loudspeaker that is in a stage area that does not have a low-pass characteristic, there may be a desire to introduce such a low-pass characteristic, in which case the EQ filter parameter would indicate a frequency response in which the high frequencies are attenuated with respect to the low frequencies , In general, any frequency response can be set for each speaker via an EQ filter parameter.

For all speakers arranged in the ranges I, II, III, there is only one delay parameter Dk, scale parameter Sk and Eq filter parameter Eqk. Whenever a directional group is to be active, the audio signal for a loudspeaker in the ranges I, II and III is simply calculated taking into account the corresponding loudspeaker parameter or the corresponding loudspeaker parameters.

On the other hand, if a speaker is in areas IV, V, VI, then each speaker has two associated speaker parameter values for each speaker parameter. For example, if only the speakers in the directional group RGA are active, ie if a source is located exactly at the direction group position A (11a), only the speakers of the directional group A will play for this audio source. In this case, to calculate the audio signal for the speaker uses the column of parameter values assigned to the direction group RGA.

On the other hand, if the audio source is sitting e.g. exactly in the position 11b in the directional group RGB, then, when an audio signal for the speaker is calculated, only the plurality of parameter values associated with the directional group RGB would be used.

On the other hand, if an audio source is located between the sources AB, that is, at any point on the connecting line between 11a and 11b in FIG Fig. 1 With this connecting line denoted by 12, all loudspeakers present in the area IV and III would comprise contradictory parameter values.

According to the invention, the audio signal is now calculated taking into account both parameter values and preferably taking into account the distance measure, as will be explained later. Preferably, an interpolation or cross-fading between the parameter values Delay and Scale is made. Furthermore, it is preferred to perform a mixture of the filter characteristics in order to also take into account different filter parameters associated with one and the same loudspeaker.

On the other hand, if the audio source is located at a position which is not on the connecting line 12 but, for example, below this connecting line 12, the loudspeakers of the directional group RGC must also be active. For loudspeakers located in area VII, then taking into account the three typically different parameter values for the same loudspeaker parameter, while for area V and area VI, taking into account the loudspeaker parameter values for the directional groups A and C for one and the same speaker will take place.

This scenario is in Fig. 2b summarized again. For the areas I, II, III in Fig. 1 there is no need to interpolate or mix speaker parameter values. Instead, the parameter values associated with the loudspeaker can simply be taken because a loudspeaker in unambiguous association has a single set of loudspeaker parameters. However, for areas IV, V and VI, an interpolation / blending of two different parameter values must be made to have a new loudspeaker parameter value for the same loudspeaker.

For the area VII, not only in the calculation of the new loudspeaker parameter must a consideration of two different tabular typically stored loudspeaker parameter values follow, but there must be an interpolation of three values, or a mixture of three values.

It should be noted that overlaps of a higher order can also be allowed, namely that a loudspeaker belongs to any number of directional groups.

In this case, only the requirement for the mixture / interpolation and the requirement for the calculation of the weighting factors, which will be discussed later, change.

The following will be on Fig. 3a and 9a received, where Fig. 9a the case shows that a source moves from the directional area A (11a) to the directional area C (11c). The loudspeaker signal LsA for a loudspeaker in the directional area A becomes dependent on the position of the source between A and B, ie of BlendAC in Fig. 9a S1 is decreasing linearly from 1 to 0, while at the same time the source C loudspeaker signal is attenuated less and less. This can be seen from the fact that S _{2 increases} linearly from 0 to 1. The blending factors S ₁ , S ₂ are chosen so that at any time the sum of the two factors 1. Alternative transitions, such as non-linear transitions, can also be used. It is preferred for all these blends that for each BlendAC value, the sum of the blending factors for the speakers concerned is equal to one. Such non-linear functions are, for example, a COS ² function for the factor S1, while a SIN ² function is used for the weighting factor S2. Other functions are known in the art.

It should be noted that the illustration in Fig. 3a provides a complete fading rule for all speakers in the ranges I, II, III. It should also be noted that the parameters associated with a loudspeaker are given in the table Fig. 2a from the corresponding areas already in the audio signal AS in the upper right in Fig. 3a have been included.

Fig. 3b shows next to the rule case, in Fig. 9a has been defined, in which a source is located on a connecting line between two directional areas, wherein the exact location between the start and the target direction area is described by the glare factor AC, the compensation case, which occurs, for example, when the path of a source while running Movement is changed. Then, the source of each current position, which is located between two directional areas, this position by BlendAB in Fig. 3b is displayed to be blinded to a new position. This results in the compensation path that is in Fig. 3b 15b, while the (regular) path was originally programmed between the directional areas A and B and is referred to as the source path 15a. Fig. 3b shows the case that during a movement of the source from A to B had changed something and therefore the original programming is changed so that the source is not now into the area B, but into the area C.

The under Fig. 3b The equations shown represent the three weighting factors g ₁ , g ₂ , g ₃ which provide the fading property for the loudspeakers in the directional areas A, B, C. It should again be pointed out that the directional area-specific loudspeaker parameters have already been taken into account again in the audio signal AS for the individual directional areas. For the areas I, II, III, the audio signals AS _A, AS _B can AS _c of the original audio signal AS simply by using the data stored for the corresponding speaker speaker parameter in the column 16a Fig. 2a are calculated, and then finally perform the final fading weighting with the weighting factor g ₁ . Alternatively, however, these weights do not have to be split into different multiplications but will typically take place in one and the same multiplication, in which case the scale factor Sk is multiplied by the weighting factor g _{1 in} order then to obtain a multiplier that will eventually match the audio signal is multiplied to obtain the speaker signal LS _a . For the overlapping areas, the same weighting g ₁ , g ₂ , g _{3 is} used, but in order to calculate the underlying audio signal AS _a , AS _b or AS _c, an interpolation / mixing of the loudspeaker parameter values predetermined for one and the same loudspeaker, instead of finding, as explained below.

It should be noted that the three-way weighting factors g ₁ , g ₂ , g ₃ in the two-way crossfade of Fig. 3a go over if either BlendAbC is set to zero, then still g ₁ , g ₂ remain, while in the other case, so if BlendAB is set to zero, only g ₁ and g ₃ remain.

The driving device will be described below with reference to FIG Fig. 4 explained. Fig. 4 shows a device for driving a plurality of loudspeakers, wherein the loudspeakers are grouped into directional groups, wherein a first directional group is assigned a first direction group position, wherein a second direction group is assigned a second direction group position, wherein at least one speaker of the first and the second directional group is assigned and wherein the loudspeaker is associated with a loudspeaker parameter having a first parameter value for the first directional group and a second parameter value for the second directional group. The apparatus first comprises means 40 for providing a source position between two directional group positions, that is, for example, for providing a source position between the directional group position 11a and the directional group position 11b as indicated by BlendAB in FIG Fig. 3b is specified.

The device according to the invention further comprises means 42 for calculating a loudspeaker signal for the at least one loudspeaker based on the first parameter value provided above a first parameter value input 42a which applies to the directional group RGA and based on a second parameter value corresponding to a second parameter value Parameter value input 42b is provided, and applies to the directional group RGB. Furthermore, the means 42 for calculating receives the audio signal via an audio signal input 43 in order then to supply the loudspeaker signal for the relevant loudspeaker in the range IV, V, VI or VII on the output side. The output of device 42 at output 44 will be the actual audio signal if the speaker being viewed is active only due to a single audio source. On the other hand, if the loudspeaker is active due to several audio sources, as it is in Fig. 7 for each source by means of a processor 71, 72 or 73 a component for the loudspeaker signal of the considered loudspeaker due to this one audio source 70a, 70b, 70c calculated, and then finally the in Fig. 7 N summed component signals in a summer 74. The temporal synchronization here takes place via a control processor 75, which, like the DSS processors 71, 72, 73, is preferably designed as a DSP (digital signal processor).

Of course, the present invention is not limited to implementation with application specific hardware (DSP). An integrated implementation with one or more PCs or workstations is also possible and may even be advantageous for certain applications.

It should be noted that in Fig. 7 a sample-wise calculation is shown. Summer 74 performs sample-by-sample summation, while delta stereophonic processors 71, 72, 73 also issue sample by sample, and the audio signal is further provided sample by sample. It should be noted, however, that when switching to block-by-block processing, all processing can be performed in the frequency domain as well, namely when summing 74 spectra together. Of course, with each processing by means of an up / down transformation, certain processing may be performed in the frequency domain or the time domain, depending on which implementation is more favorable for the particular application. Likewise, processing can also take place in the filter bank domain, which then requires an analysis filter bank and a synthesis filter bank.

Hereinafter, referring to Fig. 5 a more detailed embodiment of the device 42 for calculating a loudspeaker signal from Fig. 4 explained.

The audio signal, which is assigned to an audio source, is first supplied to a filter mixing block 44 via the audio signal input 43. The filter blend block 44 is configured to take into account all three filter parameter settings EQ1, EQ2, EQ3 when considering a speaker in region VII. The output of the filter blend block 44 then represents an audio signal which has been filtered in appropriate proportions, as will be described later, to some extent have influences from the filter parameter settings of all three directional regions involved. This audio signal at the output of the filter mix block 44 is then supplied to a delay processing stage 45. The delay processing stage 45 is designed to generate a delayed audio signal whose delay is now based on an interpolated delay value or, if no interpolation is possible, whose signal shape depends on the three delays D1, D2, D3. In the case of delay interpolation, the three delays associated with a loudspeaker for the three directional groups are provided to a delay interpolation block 46 to calculate an interpolated delay value D _int , which is then fed to the delay processing block 45.

Finally, a scaling 47 is performed, wherein the scaling 47 is performed using a total scaling factor that depends on the three scaling factors associated with the same loudspeaker due to the fact that the loudspeaker belongs to several directional groups. This total scaling factor is calculated in a scaling interpolation block 48. Preferably, the scaling interpolation block 48 is also given the weighting factor which describes the total fading for the directional area and in connection with Fig. 3b is also fed, as shown by an input 49, so that the scaling in block 47, the final speaker signal component is output on the basis of a source for a loudspeaker, which is at the in Fig. 5 shown embodiment may belong to three different directional groups.

All speakers of the other directional groups except for the three affected directional groups by which a source is defined will not output signals for that source, but may of course be active for other sources.

It should be noted that the same weighting factors may be used to interpolate the delay D _int or to interpolate the scaling factor S as used for fading, as indicated by the equations in FIG Fig. 5 set out in addition to the blocks 45 and 47, respectively.

Hereinafter, referring to Fig. 6 a preferred embodiment of the present invention, which is implemented on a DSP. The audio signal is provided via an audio signal input 43, wherein when the audio signal is in an integer format, an integer / floating point transformation is first performed in a block 60. Fig. 6 shows a preferred embodiment of the filter blend block 44 in FIG Fig. 5 , In particular, includes Fig. 6 Filter EQ1, EQ2, EQ3, wherein the transfer functions or impulse responses of the filters EQ1, EQ2, EQ3 are controlled by corresponding filter coefficients via a filter coefficient input 440. The filters EQ1, EQ2, EQ3 may be digital filters that convolve an audio signal with the impulse response of the corresponding filter, or there may be transform means, wherein a weighting of spectral coefficients is performed by frequency transfer functions. The signals filtered with the equalizer settings in EQ1, EQ2, EQ3, all of which go back to one and the same audio signal, as shown by a distribution point 441 then weighted in respective scaling blocks with the weighting factors g ₁ , g ₂ , g ₃ to then sum up the results of the weights in a summer. At the output of the block 44, ie at the output of the summer is then fed to a ring buffer, the part of the delay processing 45 of Fig. 5 is. In a preferred embodiment of the present invention, the equalizer parameters EQ1, EQ2, EQ3 do not become directly as shown in the table in FIG Fig. 2a is taken, but it is preferably carried out an interpolation of the equalizer parameters, which is done in a block 442.

Block 442, on the input side, actually receives the equalizer coefficient associated with a loudspeaker, as indicated by a block 443 in FIG Fig. 6 is shown. The interpolation task of the Filter Ramping block effectively performs low pass filtering of successive Equalizer coefficients to avoid artifacts due to rapidly changing Equalizer filter parameters EQ1, EQ2, EQ3.

The sources can thus be blinded over several directional areas, these directional areas are characterized by different settings for the equalizer. Between the different equalizer settings is dazzled, taking as it is in Fig. 6 is shown in block 44, all equalizers go through in parallel and the outputs are superimposed.

It should also be noted that the weighting factors g1, g2, g3, as used in block 44 for blending the equalizer settings, are the weighting factors included in FIG Fig. 3b are shown. To calculate the weighting factors, there is a weighting factor conversion block 61 that converts a position of a source into weighting factors for preferably three surrounding directional areas. The block 61 is preceded by a position interpolator 62 which typically depends on an input of a start position (POS1) and a target position (POS2) and the corresponding blending factors used in the in Fig. 3b In the scenario shown, the factors are blend AB and blend ABC, and typically calculate a current position depending on a motion velocity input at a current time. The position input takes place in a block 63. It should be noted, however, that at any time a new position can be entered, so that the position interpolator does not have to be provided. It should also be noted that the position update rate is arbitrarily adjustable. For example, a new weighting factor could be calculated for each sample. However, this is not preferred. Instead, it has been found that the weighting factor update rate also needs to be made at a fraction of the sampling frequency, even in terms of meaningful artifact avoidance.

The scaling calculation used in Fig. 5 has been illustrated by blocks 47 and 48 is in Fig. 6 only partially shown. The calculation of the total scaling factor shown in block 48 of FIG Fig. 5 has been made, does not find in the in Fig. 6 DSP instead of, but in an upstream control DSP. The overall scaling factor, as shown by "Scales" 64, is already input and interpolated in a scaling / interpolation block 65 to finally perform a final scaling in block 66a before then, as shown in block 67a is to the summer 74 of Fig. 7 is gone.

Hereinafter, referring to Fig. 6 the preferred embodiment of the delay processing 45 of Fig. 5 shown.

The device according to the invention allows two delay processing. One delay processing is the delay mix 451, while the other delay processing is the delay interpolation performed by an IIR all pass 452.

The output of the block 44 which has been stored in the ring buffer 450 is provided in the delay mix with three different delays explained below, the delays with which the delay blocks are driven in block 451 are the non-smoothed delays that in the table, based on Fig. 2a has been explained for a speaker, are specified. This fact is also clarified by a block 66b, which indicates that the direction group delays are input here, whereas in a block 67b not the direction group delays are input, but at a time only one loudspeaker a delay, namely the interpolated one Delay value Dint coming from block 46 in Fig. 5 is produced.

The audio signal with three different delays in block 451 then becomes, respectively, as shown in FIG Fig. 6 However, weighting factors are now preferably not the weighting factors generated by linear blending, as shown in FIG Fig. 3b is shown. Instead, it is preferred to perform a loudness correction of the weights in a block 453 to achieve a nonlinear three-dimensional crossfade here. It has been found that then the audio quality in the delay mixing becomes better and artifact-free, although the weighting factors g ₁ , g ₂ , g _{3 could} also be used to drive the scaler in the delay mixing block 451. The output signals of the scaler in the delay-mixing block are then summed to obtain a delay-mix audio signal at an output 453.

Alternatively, the delay processing of the invention (block 45 in FIG Fig. 5 ) also perform a delay interpolation. For this purpose, in a preferred embodiment of the present invention, an audio signal having the (interpolated) delay provided via block 67b and additionally smoothed in delay ramping block 68 is read from ring buffer 450. In addition, at the in Fig. 6 embodiment shown the same audio signal, but delayed by a sample less, also read. These two audio signals or just considered samples of the audio signals are then fed to an IIR filter for interpolation in order to obtain at an output 453b an audio signal which has been generated due to an interpolation.

As already explained, the audio signal at input 453a has little filter artifact due to the delay mix. By contrast, the audio signal at output 453b is hardly filter artifact-free. However, this audio signal may have frequency level shifts. If the delay is interpolated from a long delay value to a short delay value, the frequency shift will be a shift to higher frequencies, while if the delay is interpolated from a short delay to a long delay, the frequency shift will be a shift to lower frequencies ,

According to the invention, the output 453a and the output 453b in the crossfade block 457, controlled by a control signal which comes from the block 65 and whose calculation will be discussed, are switched back and forth.

In block 65, it is further controlled whether block 457 forwards the result of the blending or the interpolation, or in what proportion the results are blended. For this purpose, the smoothed or filtered value from the block 68 is compared with the non-smoothed one in order to be dependent thereon which is greater to make the (weighted) switching in 457.

The block diagram in Fig. 6 also includes a branch for a static source that sits in a directional area and does not need to be crossfaded. The delay for that source is the delay assigned to the speaker for that directional group.

The delay calculation algorithm therefore switches over too slow or too fast movements. The same physical speaker is present in two directional areas with different level and delay settings. With a slow movement of the source between the two directional areas, the level is blinded and the delay is interpolated by means of an Alpass filter, ie the signal is taken at the output 453b. However, this interpolation of the delay results in a pitch change of the signal which, however, is not critical in slow changes. If, on the other hand, the speed of the interpolation exceeds a certain value, such as 10 ms per second, then these pitch changes can be perceived. In the case of too high a speed, the delay is therefore no longer interpolated, but the signals with the two constant different delays are blinded, as shown in block 451. This will indeed cause comb filter artifacts. However, these will not be audible due to the high shutter speed.

As has been stated, the switching between the two outputs 453a and 453b takes place, depending on the movement of the source or, more precisely, depending on the delay value to be interpolated. If much delay has to be interpolated, the output 453a is switched through block 457. If, on the other hand, little delay has to be interpolated in a certain period of time, the output 453b is taken.

However, in a preferred embodiment of the present invention, switching through block 457 does not take place hard. The block 475 is formed such that a fade-out region exists around the threshold. Therefore, if the speed of the interpolation is at the threshold, block 457 is configured to compute the output side sample such that the current sample on output 453a and the current sample on output 453b are added and the result is divided by two , The block 457 therefore makes a smooth transition from the output 453b to the output 453a or vice versa in a fade range around the threshold. This cross-fade range can be made arbitrarily large, such that the block 457 operates almost continuously in the crossfade mode. For a rather tougher switching, the cross-fade range can be made smaller, so that block 457 mostly switches through only either output 453a or only output 453b to scaler 66a.

In a preferred embodiment of the present invention, the fade block 457 is further configured to perform jitter suppression over a low pass and a hysteresis of the delay change threshold. Due to the non-guaranteed duration of the control data flow between the configuration system and the DSP systems, jitter may occur in the control data, which may lead to artifacts in the audio signal processing. It is therefore preferred to compensate for this jitter by means of low-pass filtering of the control data stream at the input of the DSP system. This method reduces the response time of the timing. For this very large jitter fluctuations can be compensated. If, however, different threshold values are used for switching from delay interpolations to delay glare and delay glare to delay interpolation, then the jitter in the control data can alternatively be avoided for low-pass filtering without reducing the control data response time.

In another preferred embodiment of the present invention, the fade block 457 is further configured to perform control data manipulation in fanning delay interpolations to delay fade.

If the delay change changes abruptly to a value greater than the switching threshold between delay interpolations and delay glare, then in a conventional glare, part of the pitch fluctuation from the delay interpolation will still be heard. In order to avoid this effect, the fade-over block 457 is designed to keep the delay control data constant until the complete fade-over fade-over is performed. Only then will the delay control data be adjusted to the actual value. With the aid of this control data manipulation, it is also possible to realize fast delay changes with a short control data reaction time without audible tone changes.

In the preferred embodiment of the present invention, the drive system further includes a metering device 80 configured to perform a digital (imaginary) metering per directional area / audio output. This is based on the Fig. 11a and 11b explained. So shows Fig. 11a an audio matrix 1110 while Fig. 11b the same audio matrix 1110 shows, but with special consideration of the static sources, while in Fig. 11a the audio matrix is shown considering the dynamic sources.

Generally, the DSP system, part of which leads into Fig. 6 is shown, to calculate from the audio matrix at each Maxtrixpunkt a delay and a level, the level scaling value by AmP in Fig. 11a and Fig. 11b while the delay is designated by "dynamic-interval delay interpolation" or "static-source delay".

To represent these settings to the user, these settings are split into directional areas, and the directional areas are then assigned input signals. Several input signals can also be assigned to a directional area.

In order to now allow monitoring of the signals on the user side, a metering is indicated by the block 80 for the directional areas, which however is determined "virtually" from the levels of the nodal points of the matrix and the corresponding weightings.

The results are provided by the metering block 80 to a display interface, symbolically represented by a block "ATM" 82 (ATM = Asynchronous Transfer Mode).

It should be noted here that typically several sources play simultaneously in directional areas, for example when considering the case that from two different directions two separate sources "enter" in one and the same directional area. In the audience room, the contribution of a single source per directional area can never be measured. However, this is achieved by the metering 80, which is why this measurement is referred to as virtual measurement, since in a sense in the audience room all contributions of all direction groups for all sources overlap.

In addition, metering 80 may also compute the overall level of a single sound source from multiple sound sources across all of the directional areas that are active for that sound source. This result would result if the matrix points for an input source be summed up for all outputs. In contrast, a contribution of a directional group to a switching source can be achieved by summing up the outputs of the total number of outputs belonging to the considered directional group, while disregarding the other outputs.

In general, the concept according to the invention provides a universal operating concept for the representation of sources independently of the reproduction system used. Here, a hierarchy is used. The lowest hierarchy member is the single speaker. The middle hierarchy level is a directional area, and loudspeakers may also be present in two different directional areas.

The top hierarchy area are directional area presets, such that for certain audio objects / applications, certain directional areas taken together may be considered as an "over-directional area" on the user interface.

The sound source positioning system according to the invention is divided into main components comprising a system for performing a performance, a system for configuring a performance, a DSP system for calculating the delta stereophony, a DSP system for calculating the wave field synthesis and an accident. System for emergency interventions. In a preferred embodiment of the present invention, a graphical user interface is used to visually associate the actors with the stage or camera image. The system operator is presented with a two-dimensional image of the 3D space, which can be configured as shown in FIG Fig. 1 however, it may also be implemented in the way as described in the Fig. 9a to 10b is shown for only a small number of directional groups. With the help of a suitable user interface, the user arranges Directional areas and loudspeakers from the three-dimensional space of the two-dimensional illustration are accessible via a selected symbolism. This is done by a configuration setting. For the system, the two-dimensional position of the directional areas on the screen is mapped to the real three-dimensional position of the loudspeakers assigned to the corresponding directional areas. With the help of his context about the three-dimensional space, the operator is able to reconstruct the real three-dimensional position of directional areas and to realize an arrangement of sounds in the three-dimensional space.

About another user interface (mixer) and the local assignment of sounds / actors and their movements to the directional areas, the mixer according to a DSP Fig. 6 can include, the indirect positioning of the sound sources takes place in real three-dimensional space. With the help of this user interface, the user is able to position the sounds in all spatial dimensions without having to change the view, ie it is possible to position sounds in height and depth. The following is a reference to the positioning of sound sources or a concept for flexible compensation of deviations from the programmed stage sequence according to Fig. 8 shown.

Fig. 8 shows an apparatus for controlling a plurality of loud speakers, preferably using a graphical user interface, grouped into at least three directional groups, each directional group being associated with a direction group position. The apparatus first comprises means 800 for receiving a source path from a first direction group position to a second direction group position and motion information for the source path. The device of Fig. 8 further comprises means 802 for calculating a source path parameter for different times based on the motion information, wherein the Source path parameter indicates a location of an audio source on the source path.

The inventive apparatus further comprises means 804 for receiving a path change command to define a compensation path to the third directional area. Further, means 806 is provided for storing a value of the source path parameter at a location where the compensation path branches from the source path. Preferably, there is further provided a means for calculating a compensation path parameter (BlendAC) indicative of a position of the audio source on the compensation path included in Fig. 8 shown at 808. Both the source path parameter computed by means 806 and the compensation path parameter computed by means 808 are fed to means 810 for calculating weighting factors for the loudspeakers of the three directional regions.

Generally speaking, means 810 for calculating the weighting factors are configured to operate based on the source path, the stored value of the source path parameter, and information about the compensation path, wherein information about the compensation path includes either only the new destination, ie the directional area C, or wherein the information about the compensation path additionally comprises a position of the source on the compensation path, that is to say the compensation path parameter. It should be noted that this information of the position on the compensation path is not necessary if the compensation path is not yet taken, but the source is still on the source path. Thus, the compensation path parameter, which indicates a position of the source on the compensation path, is not necessarily necessary if the source does not take the compensation path, but uses the compensation path to reverse on the source path back to the starting point, to a certain extent without a compensation path directly from the starting point to transform the new goal. This option is useful if the source determines that it has traveled a short distance on the source path and the advantage of taking a new compensation path is only a small advantage. Although alternative implementations in which a compensation path is taken as an occasion to reverse and go back the source path without traversing the compensation path may occur when the compensation path would affect areas in the audience room that should not be areas for any other reason where a sound source is to be located.

The provision of a compensation path according to the invention is particularly advantageous in view of a system in which only complete paths between two directional areas are taken, since the time at which a source is in the new (changed) position, in particular when directional areas are widely spaced, significantly reduced. Furthermore, confusing or artificial paths of a source that would be perceived as strange are eliminated for the user. For example, considering the case that a source should originally move on the source path from left to right and now go to another leftmost position that is not very far from the originating position, then disallowing one Compensation paths cause the source runs almost twice over the entire stage, while according to the invention, this process is abbreviated.

The compensation path is made possible by the fact that a position is no longer determined by two directional areas and a factor, but that a position is defined by three directional areas and two factors, such that also points other than the direct connection lines between two direction group positions can be "driven" by a source.

Thus, the concept according to the invention allows any point in a reproduction room to be driven by a source, as it is directly out Fig. 3b becomes apparent.

Fig. 9a shows a rule case in which a source is located on a connecting line between the start-up area 11a and the aiming direction area 11c. The exact position of the source between the start and target directional regions is described by a blend factor Blend AC.

In addition to the rule, however, there are, as has already been stated and in connection with Fig. 3b has been explained, the compensation case that occurs when the path of a source is changed while moving. The change in the path of a source while moving can be represented by changing the destination of the source while the source is on its way to the destination. Then the source must be from its current source location on the source path 15a in Fig. 3b to their new position, namely the target 11c, blinded. This results in the compensation path 15b on which the source then runs until it has reached the new destination 11c. The compensation path 15b thus goes from the original position of the source directly to the new ideal position of the source. In the case of compensation, the source position is therefore formed over three directional areas and two glare values. The directional area A, the directional area B and the blend factor BlendAB form the beginning of the compensation path. The directional area C forms the end of the compensation path. The blend factor BlendAbC defines the position of the source between the beginning and the end of the compensation path.

When a source changes to the compensation path, the following changes are made to the positions: The directional area A remains. The directional area C becomes the directional area B and the glare factor BlendAC becomes the blend area AB, and the new destination area is written in the destination area C. In other words, therefore, the glare factor BlendAC at the time the direction change is to take place, that is to say at the time when the source is to leave the source path and swivel onto the compensation path, is stored by means 806 and for the subsequent calculation as blend used. The new destination area is written in direction area C.

It is further preferred according to the invention to prevent hard swelling. In general, source movements can be programmed so that sources jump, so they can move quickly from one place to another. This is the case, for example, when scenes are skipped, when ChannelHOLD mode is deactivated, or when a source ends up in scene 1 in a different direction than in scene 2. If source jumps are switched hard, audible artifacts would result. Therefore, according to the invention, a concept for preventing hard swelling of the source is used. For this purpose, again a compensation path is used which is selected on the basis of a specific compensation strategy. Generally, a source can be at different locations on a path. Depending on whether it is at the beginning or the end, between two or three directional areas, there are different ways in which a source comes to its desired position the fastest.

Fig. 9b shows a possible compensation strategy according to which a source located at a point of a compensation path (900) should be brought to a target position (902). Position 900 is the position a source has, for example, when a scene ends. When starting the new scene, the source should come to its initial position there, namely position 906. To get there, According to the invention, an immediate switchover from 900 to 906 is dispensed with. Instead, the source first goes to its personal destination direction area, that is, to the direction area 904, and then from there to the initial direction area of the new scene, namely 906 to run. Thus, the source is at the point where it should have been at the start of the scene. However, after the scene has already started and the source has actually already started, the source to be compensated must still run at an increased speed on the programmed path between the directional area 906 and the directional area 908 until it has recovered its nominal position 902.

Generally, below in the Fig. 9d to 9i given a representation of different compensation strategies, all of which are in Fig. 9c given notation for the directional area, the compensation path, the new ideal position of the source and the current real position of the source obey.

A simple compensation strategy is in Fig. 9d , This is called "InPathDual". The target position of the source is indicated by the same directional regions A, B, C as the source position of the source. A jump compensation device according to the invention is therefore designed to determine that the directional areas for defining the start position are identical to the directional areas for defining the target position. In this case, the in Fig. 9d chosen strategy in which simply proceeded on the same source path. So if the position to be reached by compensation (ideal position) is between the same directional areas as the current position of the source (real position), then the InPath strategies are used. These have two types, namely InPathDual, as in Fig. 9d is shown, and InPathTriple, as it is in Fig. 9e is shown. Fig. 9e also shows the case that real and ideal position the source is not between two but between three directional areas. In this case, the in Fig. 9e compensation strategy used. In particular shows Fig. 9e the case where the source is already on a compensation path and this compensation path goes back to reach a certain point on the source path.

As has been stated, the position of a source is defined over a maximum of three directional areas. If ideal position and real position have exactly one common direction, then Adjacent strategies are used Fig. 9f are shown. There are three types, with the letters "A", "B" and "C" referring to the common directional area. In particular, the current compensation means determines that the real position and the new ideal position are defined as throughputs of directional areas sharing a single directional area, which in the case of AdjacentA is the directional area A, which in the case of AdjacentB is the directional area B, and which in Case of AdjacentC the directional area C is how it looks Fig. 9f is apparent.

In the Fig. 9g Outside strategies shown are used when the real position and the ideal position have no common direction area in common. There are two types, OutsideM strategies and OutsideC strategies. OutsideC is used when the real position is very close to the position of the directional zone C. OutsideM is used when the real position of the source is between two directional areas, or when the location of the source is between three directional areas but very close to the knee.

It should also be noted that in the preferred implementation of the present invention, each directional area may be connected to each directional area, that is, the source to be from one directional area to another never to cross a third directional area, but from each directional area to every other directional area there exists a programmable source pathway.

In a preferred embodiment of the present invention, the source is moved manually, ie with a so-called cader. Thus, according to the invention, there are Cader strategies that provide different compensation paths. It is desired that the Cader strategies usually create a compensation path that connects the directional area A and the directional area C to the ideal position of the source. Such a compensation path can be seen in Fig. 9h , The newly assumed real position is the directional area C of the ideal position, where in Fig. 9h the compensation path is formed when the real-position direction area C is changed from the direction area 920 to the direction area 921.

There are three Cader strategies in total Fig. 9i are shown. The left strategy in Fig. 9i is used when the target direction area C of the real position has been changed. From the trail, Cader follows the OutsideM strategy. CaderInverse is used when the starting direction area A of the real position is changed. The resulting compensation path behaves in the same way as the compensation case in the normal case (Cader), but the calculation may differ within the DSP. CaderTriplestart is used when the real position of the source is between three directional areas and a new scene is switched. In this case, a compensation path from the real position of the source to the starting direction area of the new scene must be built.

The cader can be used to perform an animation of a source. With regard to the calculation of the weighting factors, there is no difference, which depends on whether the source moves manually or automatically becomes. A principal difference, however, is that the movement of the source is not controlled by a timer, but is triggered by a cader event that receives the means (804) for receiving a path change instruction. The cader event is therefore the path change command. A special case provided by the source animation according to the invention by means of Cader is the backward movement of sources. If the position of a source corresponds to the normal case, then the source moves either with the cadre or automatically on the intended path with the compensation case, but the backward movement of the source is subject to a special case. To describe this special case, the path of a source is divided into the source path 15a and the compensation path 15b, the default sector being a part of the source path 15a and the compensation sector in Fig. 10a represents the compensation path. The default sector corresponds to the original programmed portion of the path of the source. The compensation sector describes the path section that deviates from the programmed movement.

Moving the source backwards with the cader will have different effects, depending on whether the source is in the compensation sector or in the default sector. Assuming the source is in the compensation sector, movement of the cader to the left will result in backward movement of the source. As long as the source is still in the compensation sector, everything happens as expected. However, once the source leaves the compensation sector and enters the default sector, the following happens, the source moves normally in the default sector, but the compensation sector is recalculated so that once the cader is moved back to the right, the source does not open The default sector again runs along, but runs directly over the newly calculated compensation sector to the current target direction area. This situation is in Fig. 10b shown. By moving a source backwards and forwards then moving a source forward again will cause a changed compensation sector to be calculated if the backward move shortens a default sector.

The following is a calculation of the position of a source. A, B and C are the directional areas over which the position of a source is defined. A, B and BlendAB describe the starting position of the Compensation sector. C and BlendAbC describe the position of the source in the compensation sector. BlendAC describes the location of the source on the overall path.

Source positioning is searched for, eliminating the cumbersome entry of two BlendAB and Blend-AbC values. Instead, the source should be set directly via a BlendAC. If BlendAC is set to zero then the source should be at the beginning of the path. If BlendAC equals 1 then the source should be positioned at the end of the path. Furthermore, the user should not be "bothered" with compensation sectors or default sectors when entering them. On the other hand, setting the value for BlendAC depends on whether the source is in the compensation sector or in the default sector. Generally, the in Fig. 10c above-described equation for BlendAC.

One might come up with the idea to define the position of a source on the current path section by a clear indication of the BlendAC value. Fig. 10c shows some examples of how BlendAB and BlendAbC behave when BlendAC is set.

We will now discuss what happens when BlendAC is set to 0.5. What happens here depends on whether the source is in the compensation sector or the default sector. If the source is on the default sector, then: $$\mathrm{FadeAbC}=\mathrm{zero}\mathrm{,}$$

und $$\left(\mathrm{BlendAC}=\mathrm{BlendAB}/\mathrm{BlendAB}+1\right)\mathrm{.}$$

If, on the other hand, the source is at the end of the default sector or at the beginning of the compensation sector, then: $$\mathrm{FadeAbC}=\mathrm{zero}\mathrm{,}$$

and $$\left(\mathrm{FadeAC}=\mathrm{FadeAB}/\mathrm{<figure-callout\; id="1"\; label="FadeAB\; +"\; filenames="imgf0004.png,imgf0005.png"\; state="\{\{state\}\}">FadeAB\; </figure-callout>}+1\right)\mathrm{,}$$

Fig. 10d shows the determination of the parameters BlendAB and BlendAbC, depending on BlendAC, whereby a distinction is made in points 1 and 2, whether the source is in the default sector or in the compensation sector, and where in point 3 the values for the default sector while in point 4 the values for the compensation sector are calculated.

The according to Fig. 10d The resulting blend factors are then determined by Fig. 3b is used by the means for calculating the weighting factors to finally calculate the weighting factors g ₁ , g ₂ , g ₃ , from which in turn the audio signals and interpolations, etc., as shown by Fig. 6 has been described can be calculated.

The inventive concept can be combined particularly well with wave field synthesis. In a scenario where no wave field synthesis loudspeaker arrays can be placed on the stage for visual reasons, and instead, to achieve sound localization, delta stereophony must be used with directional groups, it is typically possible to have at least the sides of the Listening room and at the back of the auditorium wave field synthesis arrays. According to the invention, however, a user does not have to worry about whether a source is now is made audible by a wave field synthesis array or a directional group.

A corresponding mixed scenario is possible even if e.g. No wave field synthesis loudspeaker arrays are possible in a certain area of the stage, because otherwise they would disturb the visual impression, while wave field synthesis loudspeaker arrays can be used in another area of the stage. Again, a combination of delta-stereophony and wave-field synthesis occurs. However, in accordance with the present invention, the user will not have to worry about how his source is rendered since the graphical user interface also provides areas where wave field synthesis loudspeaker arrays are located as directional groups. Therefore, on the part of the system for performing a presentation, the directional area mechanism for positioning is always provided, such that in a common U-serinterface, the assignment of sources to wave field synthesis or to deltastereophonic directional sonication can take place without user intervention. The concept of the directional areas can be applied universally, whereby the user always positions sound sources in the same way. In other words, the user does not see whether to position a sound source in a directional area that includes a wave field synthesis array, or to position a sound source in a directional area that actually has a supportive loudspeaker that operates on the principle of the first wavefront.

Source movement occurs solely in that the user provides motion paths between directional areas, such user-set motion path through the source path receiving means according to FIG Fig. 8 Will be received. It is only on the part of the configuration system that a decision is made as to whether a wave field synthesis source or a delta stereophonic source is to be prepared. In particular, it will do so decided that a property parameter of the directional area is being investigated.

Each directional area may in this case contain any number of loudspeakers and always exactly one wave field synthesis source, which is held by its virtual position at a fixed position within the loudspeaker array or with respect to the loudspeaker array and in this respect the (real) position of the support loudspeaker in a delta stereophonic System corresponds. The wave field synthesis source then represents a channel of the wave field synthesis system, wherein in a wave field synthesis system, as it is known, per channel own audio object, so a separate source can be processed. The wave field synthesis source is characterized by corresponding wave field synthesis-specific parameters.

The movement of the wave field synthesis source can be done in two ways, depending on the availability of the computing power. The fix positioned wave field synthesis sources are driven by a transition. As a source moves out of a directional area, the speakers will be muted as the speakers of the directional area into which the source is entering are increasingly attenuated.

Alternatively, a new position can be interpolated from the entered fixed positions, which is then actually provided as a virtual position to a wave field synthesis renderer, so that a virtual position is generated without crossfading and true wave field synthesis, which in directional areas that are on the Base of delta stereophony work, of course, is not possible.

The present invention is advantageous in that free positioning of sources and assignments to the directional gates can occur, and in particular then, when overlapping directional areas are present, that is, when loudspeakers belong to multiple directional areas, a large number of directional areas can be achieved with a high resolution at directional location positions. In principle, due to the allowed overlap, each loudspeaker on the stage could represent its own directional area having loudspeakers arranged around it, emitting at a greater delay to meet the volume requirements. However, when other directional areas are concerned, these (surrounding) speakers will suddenly become supportive speakers and will no longer be "auxiliary speakers".

The concept according to the invention is further distinguished by an intuitive user interface which decreases the user as much as possible, and therefore enables secure operation even by users who are not proficient in all depths of the system.

Furthermore, a combination of the wave field synthesis with the delta stereophony is achieved via a common user interface, wherein in preferred embodiments, dynamic filtering is achieved in swelling movements due to the equalizer parameters and switched between two blend algorithms to produce artifact generation due to the transition from one direction region to another avoid the next directional area. In addition, according to the invention, it is ensured that no level dips occur during the dimming between the directional areas,
Furthermore, a dynamic glare is provided to reduce further artifacts. The provision of a compensation path allows for live application capability since there are now opportunities to intervene, for example, to respond to the tracking of sounds when an actor leaves the specified path that has been programmed.

The present invention is particularly advantageous in the sound in theaters, musical stages, open-air stages with mostly larger auditoriums or in concert venues.

Depending on the circumstances, the method according to the invention 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 performed. 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 according to the invention, when the computer program product runs on a computer. In other words, the invention can thus be realized as a computer program with a program code for carrying out the method when the computer program runs on a computer.

Claims (16)

1. Apparatus for controlling a plurality of speakers grouped into at least three directional groups (10a, 10b, 10c), each directional group having a directional group position (11a, 11b, 11c) associated with it, comprising:

   means (800) for receiving a source path from a first directional group position (11a) to a second directional group position (11b), and movement information for the source path;

   means (802) for calculating a source path parameter (BlendAB) for different points in time on the basis of the movement information, the source path parameter indicating a position of an audio source on the source path;

   characterized by

   means (804) for receiving a path modification command by means of which a compensation path to a third directional zone may be initiated;

   means (806) for storing a value of the source path parameter at a location where the compensation path (15b) deviates from the source path (15a); and

   means (810) for calculating weighting factors for the speakers of the three directional groups on the basis of the source path (15a), the stored value of the source path parameter (BlendAB), and information on the compensation path (15b).
2. Apparatus as claimed in claim 1, further comprising means (808) for calculating a compensation path parameter (BlendAbC) which indicates a position of the audio source on the compensation path (15b), and
   the means (810) for calculating being configured to additionally calculate the weighting factors for the speakers of the three directional groups using the compensation path parameter.
3. Apparatus as claimed in claims 1 or 2, wherein the means (802) for calculating the source path parameter is configured to calculate the source path parameters for successive points in time such that the source moves on the source path at a speed specified by the movement information.
4. Apparatus as claimed in any of the previous claims, wherein the means (808) for calculating the compensation path parameter is configured to calculate compensation path parameters for successive points in time such that the source moves on the compensation path at a predefined speed higher than a speed of a source which moves on the source path.
5. Apparatus as claimed in any of the previous claims,
   wherein the means (810) for calculating the weighting factors is configured to calculate the weighting factors as follows:

   g₁=(1-BlendAbC) (1-BlendAB);

   g₂=(1-BlendAbC) BlendAB;

   g3= BlendAbC

   wherein g₁ is a weighting factor for a speaker of the first directional group, wherein g₂ is a weighting factor for a speaker of the second directional group, wherein g₃ is a weighting factor for a speaker of the third directional group, wherein BlendAB is the source path parameter which has been stored by the means (806), and wherein BlendAbC is the compensation path parameter.
6. Apparatus as claimed in any of the previous claims, wherein the three directional groups are arranged in an overlapping manner, so that there is at least one speaker which exists in the three directional groups and which has a different parameter value for a speaker parameter associated with it for each directional group, the apparatus further comprising:

   means (42) for calculating a speaker signal for the speaker while using the parameter values and the weighting factors.
7. Apparatus as claimed in claim 6, wherein the means (42) for calculating comprises interpolation means (46, 48) for calculating an interpolated value on the basis of the weighting factors, the interpolation means being configured to perform the following interpolation: $$\mathrm{Z}\mathrm{=}{\mathrm{g}}_{\mathrm{1}}\mathrm{x}{\mathrm{a}}_{\mathrm{1}}\mathrm{+}{\mathrm{g}}_{\mathrm{2}}\mathrm{x}{\mathrm{a}}_{\mathrm{2}}\mathrm{+}{\mathrm{g}}_{\mathrm{3}}\mathrm{x}{\mathrm{a}}_{\mathrm{3}}\mathrm{,}$$

   

   
   wherein Z is the interpolated speaker parameter value, wherein g₁ is a first weighting factor, wherein g₂ is a second weighting factor, and wherein g₃ is a third weighting factor, wherein a is a speaker parameter value of the speaker corresponding to a first directional group, wherein a₂ is a speaker parameter value corresponding to a second directional group, and wherein a₃ is a speaker parameter value corresponding to a third directional group.
8. Apparatus as claimed in claim 7, wherein the interpolation means is configured to calculate an interpolated delay value or an interpolated scaling value.
9. Apparatus as claimed in any of the previous claims, wherein the means for receiving a path modification command (804) is configured to receive a manual input from a graphical user interface.
10. Apparatus as claimed in any of the previous claims, further comprising:

    jump compensation means for determining a continuous jump compensation path from a first jump position to a second jump position,

    wherein the means (810) for calculating the weighting factors is configured to calculate weighting factors for positions of the audio source on the jump compensation path.
11. Apparatus as claimed in claim 10, wherein the first jump position is predefined by three directional groups, and wherein the second jump position is predefined by three directional groups, and
    wherein the jump compensation means is configured to select, in the search for a jump compensation path, a compensation strategy which depends on whether the three directional zones defining the first jump position, and the three directional zones defining the second jump position have one or several directional zones in common.
12. Apparatus as claimed in claim 11, wherein the jump compensation means is configured to use an InpathDual compensation strategy or an InpathTriple compensation strategy when the three directional zones of the first jump position and the three directional zones of the second jump position match,
    so as to use an AdjacentA compensation strategy, an AdjacentB compensation strategy, or an AdjacentC compensation strategy when at least one directional zone of the first jump position is identical with a directional zone of the second jump position,
    or to use an OutsideM compensation strategy or an OutsideC compensation strategy when the first jump position and the second jump position have no directional zone in common.
13. Apparatus as claimed in any of the previous claims, wherein the means (804) for receiving a path modification command is configured to receive a position of the source between the first and third directional groups, and
    the means (802) for calculating the source path parameter being configured to ascertain whether at times when the path modification command is to become active, the source is located on a source path or a compensation path.
14. Apparatus as claimed in claim 13, wherein the means (802) for calculating the source path parameter or the means (808) for calculating the compensation path parameter are configured to calculate the compensation path parameters on the basis of a first calculation specification when the source is located on the compensation path, and to calculate the path parameters on the basis of a second calculation specification when the source is located on the source path.
15. Method for controlling a plurality of speakers grouped into at least three directional groups (10a, 10b, 10c), each directional group having a directional group position (11a, 11b, 11c) associated with it, the method comprising:

    receiving (800) a source path from a first directional group position (11a) to a second directional group position (11b), and movement information for the source path;

    calculating (802) a source path parameter (BlendAB) for different points in time on the basis of the movement information, the source path parameter indicating a position of an audio source on the source path;

    characterized by the steps of:

    receiving (804) a path modification command by means of which a compensation path to a third directional zone may be initiated;

    storing (806) a value of the source path parameter at a location where the compensation path (15b) deviates from the source path (15a); and

    calculating (810) weighting factors for the speakers of the three directional groups on the basis of the source path (15a), the stored value of the source path parameter (BlendAB), and information on the compensation path (15b).
16. Computer program comprising a program code for performing the method as claimed in claim 15, when the computer program runs on a computer.

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