EP1723825B1 - Apparatus and method for controlling a wave field synthesis rendering device - Google Patents

Abstract

The device has a control monitor (2) to monitor an extent of utilization situation of a wave field synthesis system with a wave field synthesis rendering device. An audio object manipulation device (3) varies a start point/end point of an audio object within a time period or an actual position of a virtual source within a local span, by the rendering device, depending on the situation of the wave field synthesis system. Independent claims are also included for the following: (1) a method of controlling a wave field synthesis rendering device arranged in a wave field synthesis system (2) a computer program with a program code for executing a method of controlling a wave field synthesis rendering device arranged in a wave field synthesis system.

Description

The present invention relates to the field of wave field synthesis, and more particularly to driving a wave field synthesis rendering device with data to be processed.

The present invention relates to wave-field synthesis concepts, and more particularly to efficient wave-field synthesis concept in conjunction with a multi-renderer system. Audio playback systems based on wave field synthesis and provided with means for providing scene description are out WO 2004/047485 known.

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

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

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

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

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

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

Applied to the acoustics can be simulated by a large number of speakers, which are arranged side by side (a so-called speaker array), any shape of an incoming wavefront. In the simplest case, a single point source to be reproduced and a linear arrangement of the speakers, the audio signals of each speaker must be fed with a time delay and amplitude scaling so that the radiated sound fields of each speaker properly overlap. With multiple sound sources, the contribution to each speaker is calculated separately for each source and the resulting signals added together. Are located If the sources to be played in a room with reflective walls, then reflections as additional sources on the speaker array must be reproduced. 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, wave field synthesis provides the ability to eliminate the reflection from that wall by giving the loudspeaker a signal in phase opposition to the reflection signal is impressed with appropriate amplitude in addition to the original audio signal, so that the outgoing compensation wave extinguishes the reflection wave, such that the reflection from this wall in the environment; which is considered 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 the compensation signal is then calculated, which must be impressed on the audio signal superimposed on the loudspeaker, then the reflection from this wall will be canceled, such that a listener in this environment will soundly have the impression that the latter 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.

In the wave field synthesis, a synthesis signal for each loudspeaker of the loudspeaker array is calculated from an audio signal which emits a virtual source at a virtual position, the synthesis signals being designed in amplitude and phase such that a wave resulting from the superposition of the individual the sound wave present in the loudspeaker array will correspond to the wave that would result from the virtual source at the virtual position if that virtual source at the virtual position were a real source with a real position.

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 larger the loudspeaker arrays are, ie the more individual loudspeakers are provided. However, this also increases the computing power that a. Wave field synthesis unit must accomplish, since typically channel information must be considered. 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.

If, in particular, in cinema applications the possibilities of wave field synthesis are to be exploited to the extent that the virtual sources can also be mobile, then it can be seen that due to the calculation of the synthesis signals, the calculation of the channel information and the generation of the reproduction signals by combining the channel information and the synthesis signals quite considerable computing power has to be mastered.

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 reproduction quality becomes better and more realistic as more loudspeakers are present in the loudspeaker array (s).

In the above scenario, the final-rendered and analog-to-digital converted reproduction signals for the individual loudspeakers could be transmitted, for example 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 German patent DE 10254404 B4 discloses a system as shown in FIG. One part is the central wave field synthesis module 10. The other part is composed individual speaker modules 12a, 12b, 12c, 12d, 12e, which are connected to actual physical speakers 14a, 14b, 14c, 14d, 14e, as shown in Fig. 7. It should be noted that the number of speakers 14a-14e in typical applications is in the range above 50 and typically even well above 100. If each loudspeaker is assigned its own loudspeaker module, the corresponding number of loudspeaker modules is also required. Depending on the application, however, it is preferred to address a small group of adjacent loudspeakers from a loudspeaker module. In this context, it is arbitrary whether a speaker module, which is connected to four speakers, for example, feeds the four speakers with the same playback signal, or whether the four speakers corresponding different synthesis signals are calculated, so that such a speaker module actually off consists of several individual speaker modules, but which are physically combined in one unit.

Between the wave field synthesis module 10 and each individual loudspeaker module 12a-12e there is a separate transmission path 16a-16e, each transmission path being coupled to the central wave field synthesis module and to a separate loudspeaker module.

As a data transmission mode for transmitting data from the wave field synthesis module to a speaker module, a serial transmission format that provides a high data rate, such as a so-called Firewire transmission format or a USB data format, is preferred. Data transfer rates in excess of 100 megabits per second are advantageous.

The data stream which is transmitted from the wave field synthesis module 10 to a loudspeaker module is thus formatted according to the selected data format in the wave field synthesis module and with synchronization information provided in common serial data formats. This synchronization information is extracted from the individual loudspeaker modules from the data stream and used to resample the individual loudspeaker modules with regard to their reproduction, that is to say finally to the analog-to-digital conversion for obtaining the analog loudspeaker signal and the purpose of resampling. to synchronize. The central wave-field synthesis module works as a master and all loudspeaker modules operate as clients, with the individual data streams across the different links 16a-16e all receiving the same synchronization information from the central module 10. This ensures that all loudspeaker modules operate synchronously, synchronized by the master 10, which is important to the audio playback system so as not to suffer any loss of audio quality, so that the synthesis signals computed by the wavefronts synthesis module are not delayed in response to the individual loudspeakers Audio rendering will be broadcast.

Although the concept described already provides considerable flexibility with regard to a wave field synthesis system that is scalable for various applications. However, it continues to suffer from the problem that the central wave field synthesis module, which performs the actual main rendering, which thus calculates the individual synthesis signals for the speakers, depending on the positions of the virtual sources and depending on the speaker positions Represents "bottleneck" for the entire system. Although in this system, the "post-rendering", ie the application of the synthesis signals with channel transfer functions, etc. already executed decentralized and thus already the necessary data transfer capacity between the central renderer module and the individual speaker modules by selecting synthesis signals with However, all virtual sources still have to be sorted out, if they have been reduced to a smaller energy than a certain threshold energy for all loudspeaker modules are rendered, so converted into synthesis signals, the Ausselektion takes place only after the rendering.

This means that the rendering still determines the total capacity of the system. Is the central rendering unit therefore z. For example, if it is able to render 32 virtual sources simultaneously, ie to compute the synthesis signals for these 32 virtual sources simultaneously, then serious capacity bottlenecks will occur if more than 32 sources are active at a time in an audio scene. This is sufficient for simple scenes. For more complex scenes, in particular with immersive sound impressions, ie when it rains and many raindrops are single sources, it is immediately obvious that the capacity with a maximum of 32 sources is no longer sufficient. A similar situation also occurs when you have a large orchestra and in fact want to process every orchestra player or at least each group of instruments as their own source in their own position. Here, 32 virtual sources can quickly become too little.

Typically, in the known wave field synthesis concept, a scene description is used in which the individual audio objects are defined together such that, using the data in the scene description and the audio data for the individual virtual sources, the complete scene is rendered by a renderer or a multi-rendering Arrangement can be processed. For each audio object, it is exactly defined where the audio object has to start and where the audio object ends. Furthermore, for each audio object, exactly the position of the virtual source is indicated at which the virtual source should be, which is to be entered into the wave field synthesis rendering device, so that for each speaker the corresponding synthesis signals are generated. As a result, by overlaying the from the individual speakers In response to the synthesis signals output sound waves for a listener an impression arises as if a sound source is positioned at a position in the playback room or outside the playback room, which is defined by the source position of the virtual source.

Typically, the capacities of the wave field synthesis system are limited. As a result, each renderer has limited computing power. Typically, a renderer is capable of processing 32 audio sources simultaneously. Furthermore, a transmission path from the audio server to the renderer has a limited transmission bandwidth, so provides a maximum transmission rate in bits per second.

For simple scenes in which z. B. only if it is thought of a dialogue, two virtual sources exist, and in addition for a background noise, another virtual source is present, the processing capacity of the renderer, the yes z. B. 32 sources can handle simultaneously, no problem. Further, in this case, the transmission volume to a renderer is so small that the capacity of the transmission line is sufficient.

However, problems will arise when playing more complex scenes, that is, scenes that have more than 32 virtual sources. For example, in such a case as to correctly render a scene in the rain or to faithfully reproduce an applause scene, the maximum computing capacity of a renderer limited to 32 virtual sources will quickly become insufficient. This is because a lot of individual virtual sources exist because, for. In general, for example, in an audience every listener who claps can be understood as his own virtual source in his own virtual position. To deal with this limitation, there are several possibilities. So there is a possibility in already creating the scene description to make sure that never a renderer has to process 32 audio objects at the same time.

Another possibility is to take into account when creating the scene description no consideration of actual wave field synthesis conditions, but to create the scene description just as it wishes the scene author.

This possibility is advantageous in view of a higher flexibility and portability of scene descriptions under different wave field synthesis systems, as this creates scene descriptions which are not designed for a specific system but are more general. In other words, the same scene description, when run on a wave field synthesis system having the high capacity renderer, will result in a better sound impression than in a system having renderers with lower computational power. In other words, the second possibility is advantageous in that a scene description does not result in a better sound impression due to the fact that it has been generated for a wave field synthesis system with a very limited capacity, even in a better capacity wave field synthesis system.

A disadvantage of the second possibility, however, is that when the wave field synthesis system is brought to its maximum capacity, performance slumps or other associated problems will occur because the renderer because of its maximum capacity, if it should process more sources, processing the In addition, it can simply deny going sources.

The object of the present invention is to provide a flexible concept for controlling a wave-field synthesis rendering device, through quality degradations be reduced at least while maintaining a high level of flexibility.

This object is achieved by a device for controlling a wave field synthesis rendering device according to claim 1, a method for controlling the wave field synthesis rendering device according to claim 13 or a computer program according to claim 14.

The present invention is based on the recognition that actual capacity limits can be extended by intercepting processing load peaks occurring in wave field synthesis by varying the beginning and / or end of an audio object or the position of an audio object within a time span or span, perhaps only one short existing overload peak intercept. This is achieved by specifying margins in the scene description rather than fixed times for certain sources where the beginning and / or the end and even the position may be variable within a certain span, and then depending on a load situation in the wave field synthesis system, the actual start and actual virtual position of an audio object are varied within that time span.

Thus, it has been found that due to the high dynamics of typical scenes to be processed, the actual number of audio sources can vary widely at a time, but overload situations, that is a very large number of virtual sources that are to be active simultaneously, are relatively short ,

According to the invention, such overload situations are thereby reduced or even completely eliminated by moving audio objects forward or backward within their time span or in the case of multi-renderer systems with respect to their position, so that one of the Due to the changed position, renderer no longer needs to generate synthesis signals for this virtual source.

Audio objects that are particularly well suited for such a duration / Ortsspannen definition are sources that have noises to the content, so z. B. gossip noise, dripping or any other background noise, such as a wind noise or z. B. also a driving noise of a approaching from far away train. Here, it will not matter to the viewer's audio impression or listening experience if a wind noise starts a few seconds earlier or later, or if the move enters the audio scene at a different virtual position than originally requested by the original author of the scene description.

However, the effects on the described very dynamically occurring overload situation can be eminent. Thus, the scheduling or scheduling of audio sources within the scope of their spatial ranges and periods of time can lead to a very short overload situation being able to be converted into a correspondingly longer situation that can still be processed. This can of course by a z. For example, within a permitted period of time, it would be conditional earlier termination of an audio object that would not have existed for a long time anyway, but because of an audio object newly transferred to the renderer, would have led to an overload situation of this renderer that would have rejected the new audio object.

It should also be noted that the rejection of an audio object has so far led to the fact that the entire audio object has not been processed, which is particularly undesirable if the old audio object might have lasted only a second and a new audio object with a Length of maybe a few minutes due to a short overload situation, which may only be due to a one-second overlap the old audio object would have been completely failed / rejected.

According to the invention, this problem is solved by z. B. the previous audio object, if a corresponding margin was specified, already ended a second earlier, or that the later audio object within a predetermined period z. B. is pushed back a second, so that the audio objects no longer overlap and thus no unpleasant rejection of the entire later audio object, which may have a length of minutes, is obtained.

According to the invention, therefore, not a concrete time but a period is defined for the start of an audio object or for the end of an audio object. This makes it possible to intercept transmission rate spikes and consequent capacity or performance issues by shifting the transmission or processing of the respective audio data forward or backward.

Fig. 1

ein Blockschaltbild der erfindungsgemäßen Vorrichtung;

Fig. 2

ein beispielhaftes Audioobjekt;

Fig. 3

eine beispielhafte Szenenbeschreibung;

Fig. 4

einen Bitstrom, in dem jedem Audioobjekt ein Header mit den aktuellen Zeitdaten und Positionsdaten zugeordnet ist;

Fig. 5

eine Einbettung des erfindungsgemäßen Konzepts in ein Wellenfeldsynthese-Gesamtsystem;

Fig. 6

eine schematische Darstellung eines bekannten Wellenfeldsynthese-Konzepts; und

Fig. 7

eine weitere Darstellung eines bekannten Wellenfeldsynthese-Konzepts.

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

Fig. 1

a block diagram of the device according to the invention;

Fig. 2

an exemplary audio object;

Fig. 3

an exemplary scene description;

Fig. 4

a bit stream in which each audio object is assigned a header with the current time data and position data;

Fig. 5

an embedding of the inventive concept in a wave field synthesis overall system;

Fig. 6

a schematic representation of a known wave field synthesis concept; and

Fig. 7

another illustration of a known wave field synthesis concept.

1 shows a device according to the invention for controlling a wave field synthesis rendering device arranged in a wave field synthesis system 0, wherein the wave field synthesis rendering device is designed to generate synthesis signals for a plurality of loud speakers within a speaker array from audio objects. In particular, an audio object comprises an audio file for a virtual source and at least one source position at which the virtual source is to be arranged inside or outside the playback room, ie with respect to the listener.

The apparatus according to the invention shown in FIG. 1 comprises a scene description providing means 1, the scene description defining a time sequence of audio data, wherein an audio object for a virtual source associated with the audio object defines a start time or an end time, the audio object for the virtual source has a period of time in which to start or end the audio object. Alternatively or additionally, the scene description is such that the audio object has a location span in which a position of the virtual source must lie.

The device according to the invention further comprises a monitoring monitor 2, which is designed to monitor a utilization of the wave field synthesis system 0, so as to determine a utilization situation of the wave field synthesis system.

Furthermore, an audio object manipulation device 3 is provided, which is configured to vary an actual start point or end point of the audio object to be taken into account by the wave field synthesis rendering device within the time span or an actual position of the virtual source within the location span, depending on a load situation of the wave field synthesis system 0. Preferably, an audio file server 4 is further provided, which can be implemented together with the audio object manipulation device 3 in an intelligent database. Alternatively, it is a simple file server which, depending on a control signal from the audio object manipulation device 3, feeds an audio file either directly via a data connection 5a to the wave field synthesis system and in particular to the wave field synthesis rendering device. Furthermore, it is preferred according to the invention to supply the audio file to the audio object manipulation device 3 via a data connection 5b, which then feeds a data stream via its control line 6a to the wave field synthesis system 0 and in particular to the individual renderer modules or the single renderer module the actual starting points and / or end points of the audio object determined by the manipulation device or comprises the corresponding position and the audio data itself includes.

Via an input line 6b, the audio object manipulation device 3 is supplied with the scene description from the device 1, while the utilization situation of the wave field synthesis system 0 is supplied from the monitoring monitor 2 via a further input line 6c. It should be noted that the individual lines described in FIG. 1 may not necessarily be implemented as separate cables etc., but merely to symbolize that corresponding data is transmitted in the system in order to implement the concept according to the invention , In this respect, the monitoring monitor 2 is also a monitoring line 7 with the wave field synthesis system 0 connected depending on the situation z. For example, to check how many sources are being processed in a renderer module and whether the capacity limit has been reached, or to check what the current data rate is, just on line 6a or data line 5a or on another Lead within the wave field synthesis system prevails.

It should be noted at this point, however, that the utilization situation does not necessarily have to be the current utilization situation, but can also be a future utilization situation. This implementation is preferred in that then the variability, such as the individual audio objects with each other in terms of avoiding overload peaks in the future can be scheduled or manipulated, for. B. by a current variation within a period of time only in some future avoid overload peak helps. The efficiency of the concept according to the invention becomes ever greater the more sources exist which do not have fixed starting points or end points, but have starting points or end points which are provided with a time span or which have no fixed source positions but source positions which provide a spatial span are.

At this point, it should be noted that there may be sources in particular, for. B. background noise, where the source position is irrelevant, so they can come from somewhere. While until now also a position had to be specified for these sources, the position information can now be replaced or supplemented by a very large explicit or implicit local span. This is especially important in multi-renderer systems. If z. If, for example, a playback room is considered that has four pages and that has a loudspeaker array on each side which is supplied by its own renderer, it can be arranged particularly well due to the arbitrary location span. For example, the situation could occur that the Front renderer is just overloaded and a source is coming, which can be at any position. Then the audio object manipulation device 3 according to the invention would position the position of this virtual source, the actual position of which is insignificant for the audio impression or for the audio scene, in such a way that it is processed by a different renderer than the front renderer. Renderer not burdened but only charged to another renderer, but anyway not at its capacity limit works.

As has already been stated, the flexibility and efficiency of the inventive concept increases the more variable the scene description is held. However, this also benefits the needs of the scene author, since it is sufficient for him that he specifies time spans and Ortsspannen and thus does not have to definitely decide for each source of points that are actually irrelevant to the listening experience. Such decisions would be a chore for the sound engineer, taken away from him by the concept of the invention and even used to increase the actual capacity by intelligently scheduling within a framework dictated by the sound engineer as compared to the capacity of a rigid field wave-field synthesis system.

Hereinafter, referring to FIG. 2, information is pointed out which an audio object should be advantageous. For example, an audio object should specify the audio file that effectively represents the audio content of a virtual source. However, the audio object does not need to include the audio file, but may have an index pointing to a defined location in a database where the actual audio file is stored.

Furthermore, an audio object preferably comprises an identification of the virtual source, which may be, for example, a source number or a meaningful file name, etc.

Further, in the present invention, the audio object specifies a period of time for the beginning and / or the end of the virtual source, that is, the audio file. Specifying only a time period for the start means that the actual starting point of the rendering of this file by the renderer can be changed within the time span. In addition, if a time limit is specified for the end, this also means that the end can also be varied within the time span, which, depending on the implementation, will generally lead to a variation of the audio file also in terms of its length. Any implementations are possible, such. For example, a definition of the start / end time of an audio file so that although the starting point may be moved, but in no case the length may be changed, so that automatically the end of the audio file is also moved. However, especially for noise, it is preferred to also keep the end variable, since it is typically not problematic whether z. For example, a wind noise starts sooner or later, or ends slightly earlier or later. Further specifications are possible or desired depending on the implementation, such as a specification, that although the starting point may be varied, but not the end point, etc.

Preferably, an audio object further comprises a location span for the position. So it will be irrelevant for certain audio objects, whether they z. B. come from the front left or front center, or if they are shifted by a (small) angle with respect to a reference point in the playback room. However, as has been said, audio objects, especially from the noise area, which can be positioned at any position and thus have a maximum spatial range, for example, by a code for "arbitrary" or by no code (implicit) in the Audio object can be specified.

An audio object may include other information, such as an indication of the type of virtual source, that is, whether the virtual source must be a point source for sound waves, or whether it must be a source of plane waves, or whether must be a source that generates sources of arbitrary wavefront, provided the renderer modules are able to process such information.

3 shows, by way of example, a schematic representation of a scene description in which the temporal sequence of different audio objects AO1,... AOn + 1 is shown. In particular, attention is drawn to the audio object AO3, for which a period of time, as shown in FIG. 3, is defined. Thus, both the start point and the end point of the audio object AO3 in Fig. 3 can be shifted by the time period. The definition of the audio object AO3, however, is that the length may not be changed, but this can be set variably from audio object to audio object.

It can thus be seen that by shifting the audio object AO3 in a positive temporal direction, a situation can be achieved in which the audio object A03 does not start until after the audio object AO2. If both audio objects are played on the same renderer, this measure avoids a short overlap 20 which otherwise might occur. If the audio object AO3 in the prior art already had the audio object that would exceed the capacity of a renderer because of all the other audio objects to be processed on the renderer, such as audio object AO2 and audio object AO1, without the present invention, a complete suppression of the Audio object AO3, although the time span 20 was only very small. According to the audio object AO3 is moved by the audio object manipulation device 3, so no capacity exceeded and thus no suppression of the audio object AO3 more takes place.

In the preferred embodiment of the present invention, a scene description is used that has relative indications. Thus, the flexibility is increased by the fact that the beginning of the audio object AO2 is no longer given at an absolute time, but in a relative time to the audio object AO1. Accordingly, a relative description of the location information is preferred, so not that an audio object is to be arranged at a certain position xy in the playback room, but z. B. is a vector offset to another audio object or to a reference object.

Thereby, the time span information or Ortsspanneninformation can be recorded very efficiently, namely simply in that the time period is set so that it expresses that the audio object AO3 z. B. in a period between two minutes and two minutes and 20 seconds after the start of the audio object AO1 can begin.

Such a relative definition of the space and time conditions leads to a database efficient representation in the form of constraints, such as. B. "Modeling Output Constraints in Multimedia Database Systems", T. Heimrich, 1st International Multimedia Modeling Conference, IEEE, January 2, 2005 to January 14, 2005, Melbourne. It shows the use of constraints in database systems to define consistent database states. In particular, temporal constraints are described using Allen relationships and spatial constraints using spatial relationships. From this, favorable output constraints can be defined for synchronization purposes. Such output constraints include a temporal or spatial condition between the objects, a response in case of a violation of a Constraints and a review time, so when such a constraint must be checked.

In the preferred embodiment of the present invention, the spatial / temporal output objects of each scene are modeled relative to one another. The audio object manipulation device achieves a translation of these relative and variable definitions into an absolute spatial and temporal order. This ordering represents the output schedule obtained at the output 6a of the system shown in Figure 1 and defines how, in particular, the renderer module in the wave-field synthesis system is addressed. The schedule is thus an output schedule that arranges the audio data according to the output conditions.

Hereinafter, a preferred embodiment of such an output schedule is set forth with reference to FIG. 4 shows a data stream which is transmitted from left to right according to FIG. 4, ie from the audio object manipulation device 3 of FIG. 1 to one or more wave field synthesis renderers of the wave field system 0 of FIG. 1. In particular, the data stream comprises for each audio object in the embodiment shown in Fig. 4, first a header H, in which the position information and the time information stand, and an audio file for the special audio object, which in Fig. 4 shows AO1 for the first audio object, AO2 for the second Audio object etc. is designated.

A wave field synthesis renderer then receives the data stream and detects z. B. to an existing and agreed synchronization information that now comes a header. Based on another synchronization information, the renderer then recognizes that the header is now over. Alternatively, a fixed length in bits can also be agreed for each header.

After receiving the header, in the preferred embodiment of the present invention shown in FIG. 4, the audio renderer automatically knows that the subsequent audio file, ie, e.g. AO1 belongs to the audio object, that is, to the source location identified in the header.

4 shows a serial data transmission to a wave field synthesis renderer. However, of course, several audio objects are played simultaneously in a renderer. Therefore, the renderer requires an input buffer preceded by a data stream reader to parse the data stream. The data stream reader will then interpret the header and store the associated audio data accordingly, so that when an audio object is to render, the renderer reads out the correct audio file and the correct source position from the input buffer. Other data for the data stream are of course possible. Also, a separate transmission of both the time / location information and the actual audio data may be used. However, the combined transfer illustrated in Figure 4 is preferred because it eliminates data consistency problems by concatenating the position / time information with the audio file, since it is always ensured that the audio data renderer also has the correct source position and not z. B. still renders audio from an earlier source, but already uses position information from the new source for rendering.

The present invention is thus based on an object-oriented approach, that is to say that the individual virtual sources are understood as objects which are distinguished by an audio file and a virtual position in space and possibly by the nature of the source, that is, if they are a point source for sound waves or a source of plane waves or a source of differently shaped sources.

As has been pointed out, the calculation of the wave fields is very computationally intensive and tied to the capacities of the hardware used, such as sound cards and computers, in conjunction with the efficiency of the calculation algorithms. Even the best-equipped PC-based solution thus quickly reaches its limits in the calculation of wave field synthesis, when many sophisticated sound events are to be displayed simultaneously. Thus, the capacity limit of the software and hardware used dictates the limitation on the number of virtual sources in the mixdown and playback.

FIG. 6 shows such a limited-capacity known wave-field synthesis concept including an authoring tool 60, a control renderer module 62, and an audio server 64, wherein the control renderer module is configured to include a speaker array 66 Supply data so that the speaker array 66 generates a desired wavefront 68 by superimposing the individual waves of the individual speakers 70. The authoring tool 60 allows the user to create scenes, edit and control the wave field synthesis based system. A scene consists of information about the individual virtual audio sources as well as the audio data. The properties of the audio sources and the references to the audio data are stored in an XML scene file. The audio data itself is stored on the audio server 64 and transmitted from there to the renderer module. At the same time, the renderer module receives the control data from the authoring tool so that the control renderer module 62, which is centrally executed, can generate the synthesis signals for the individual loudspeakers. The concept shown in Figure 6 is described in "Authoring System for Wave Field Synthesis", F. Melchior, T. Röder, S. Brix, S. Wabnik and C. Riegel, AES Convention Paper, 115th AES Assembly, 10. October 2003, New York.

If this wave field synthesis system is operated with several renderer modules, each renderer is supplied with the same audio data, regardless of whether the renderer needs this data for playback or not because of the limited number of speakers assigned to it. Since each of the current computers is capable of calculating 32 audio sources, this is the limit for the system. On the other hand, the number of renderable sources in the overall system should be significantly increased efficiently. This is one of the essential requirements for complex applications, such as movies, scenes with immersive atmospheres, such as rain or applause or other complex audio scenes.

According to the invention, a reduction of redundant data transfer operations and data processing operations in a wave field synthesis multi-renderer system is achieved, which leads to an increase in the computing capacity or the number of simultaneously computable audio sources.

To reduce the redundant transmission and processing of audio and metadata to the single renderer of the multi-renderer system, the audio server is extended by the data output device, which is able to determine which renderer needs which audio and metadata. The data output device, possibly supported by the data manager, requires a plurality of information in a preferred embodiment. This information is initially the audio data, then the source and position data of the sources, and finally the configuration of the renderers, that is, information about the connected speakers and their positions and their capacity. Using data management techniques and the definition of output conditions, an output schedule is generated by the data output device with a temporal and spatial arrangement of the audio objects. From the spatial arrangement, the time schedule and the renderer configuration, the data management module then calculates which source for which renderers are relevant at any given time.

A preferred overall concept is shown in FIG. 5. The database 22 is supplemented on the output side by the data output device 24, wherein the data output device is also referred to as a scheduler. This scheduler then generates at its outputs 20a, 20b, 20c for the various renderers 50 the renderer input signals in order to power the corresponding loudspeakers of the loudspeaker arrays.

Preferably, the scheduler 24 is still supported by a storage manager 52 in order to configure the database 42 by means of a RAID system and corresponding data organization specifications.

On the input side is a data generator 54, which may be, for example, a sound engineer or an audio engineer who is to model or describe an audio scene in an object-oriented manner. In this case, he provides a scene description that includes corresponding output conditions 56, which are then optionally stored in the database 22 together with audio data after a transformation 58. The audio data may be manipulated and updated using an insert / update tool 59.

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, particularly a floppy disk or CD, with electronically readable control signals that may interact with a programmable computer system to perform the method. 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 when the computer program product runs on a computer. In other In other words, the invention can be realized as a computer program with a program code for carrying out the method when the computer program runs on a computer.

Claims (14)

1. Apparatus for controlling a wave field synthesis rendering means arranged in a wave field synthesis system (0), wherein the wave field synthesis rendering means is formed to generate, from audio objects, wherein an audio file for a virtual source arranged at a source position is associated with an audio object, synthesis signals for a plurality of loudspeakers coupled to the wave field synthesis rendering means, comprising:

   a means (1) for providing a scene description, wherein the scene description sets a temporal sequence of audio objects, wherein an audio object defines a temporal start or a temporal end for a virtual source associated with the audio object, wherein the audio object for the virtual source comprises a time span in which the start or the end of the audio object must be, or wherein the audio object comprises a location span in which a position of the virtual source must be;

   a monitor (2) for monitoring a utilization situation of the wave field synthesis system; and

   an audio object manipulation means (3) for varying an actual starting point or end point of the audio object to be considered by the wave field synthesis rendering means within the time span or an actual position of the virtual source within the location span, depending on a utilization situation of the wave field synthesis system (0).
2. Apparatus according to claim 1, wherein the monitor is formed to monitor a utilization situation of a data connection between the audio object manipulation means (3) and the wave field synthesis rendering means; and wherein the audio object manipulation means (3) is formed to vary the actual starting point or end point of an audio object so that a utilization peak of the data connection is reduced as compared with no variation.
3. Apparatus according to claim 1 or 2, wherein the monitor (2) is formed to monitor a utilization situation of the wave field synthesis rendering means, and
   wherein the audio object manipulation means (3) is formed to vary the actual starting point or the actual end point so that a maximum number of sources to be processed at the same time given by the wave field synthesis rendering means is not exceeded at a time instant, or a number of audio objects to be processed at the same time by the wave field synthesis rendering means is reduced as compared with no variation.
4. Apparatus according to one of the preceding claims, wherein the monitor (2) is formed to predict the utilization situation of the wave field synthesis system (0) over a predetermined prediction time interval.
5. Apparatus according to claim 4, wherein the wave field synthesis rendering means (0) comprises an input buffer, wherein the predetermined prediction time interval depends on a size of the input buffer.
6. Apparatus according to one of the preceding claims, wherein the wave field synthesis rendering means comprises a plurality of renderer modules, with which loudspeakers arranged at different locations in a reproduction room are associated, and
   wherein the audio object manipulation means (3) is formed to vary a current position of the virtual source within the location span so that a renderer module is not active for the generation of the synthesis signals, although the renderer module would have been active for another position within the location span.
7. Apparatus according to one of the preceding claims, wherein the audio object manipulation means (3) is formed to choose a current time instant within a first half of the time span in a case in which the monitor detects a utilization a predetermined threshold below the maximum utilization.
8. Apparatus according to claim 7, wherein the audio object manipulation means is formed to choose an earliest time instant defined by the time span as starting point or end point in a case in which the monitor (2) signalizes a utilization lying a predetermined threshold below the maximum utilization.
9. Apparatus according to one of the preceding claims,
   wherein the means (1) for providing is formed to provide a scene description in which a temporal or spatial positioning of the audio objects relative to another audio object or relative to a reference audio object is defined, and
   wherein the audio object manipulation means (3) is formed to compute an absolute starting point or an actual absolute position of the virtual source for each audio object.
10. Apparatus according to one of the preceding claims,
    wherein the means (1) for providing is formed to provide a scene description in which a time span is indicated only for a group of sources, and in which a fixed starting point is indicated for other sources.
11. Apparatus according to claim 10, wherein the group of sources comprises a predetermined characteristic including a noise-like audio file of the virtual source.
12. Apparatus according to claim 10 or 11, wherein the group of sources includes noise sources.
13. Method for controlling a wave field synthesis rendering means arranged in a wave field synthesis system (0), wherein the wave field synthesis rendering means is formed to generate, from audio objects, wherein an audio file for a virtual source arranged at a source position is associated with an audio object, synthesis signals for a plurality of loudspeakers coupled to the wave field synthesis rendering means, comprising:

    providing (1) a scene description, wherein the scene description sets a temporal sequence of audio objects, wherein an audio object defines a temporal start or a temporal end for a virtual source associated with the audio object, wherein the audio object for the virtual source comprises a time span in which the start or the end of the audio object must be, or wherein the audio object comprises a location span in which a position of the virtual source must be;

    monitoring (2) a utilization situation of the wave field synthesis system; and

    varying (3) an actual starting point or end point of the audio object to be considered by the wave field synthesis rendering means within the time span or an actual position of the virtual source within the location span, depending on a utilization situation of the wave field synthesis system (0).
14. Computer program with program code for performing the method according to claim 13, when the program is executed on a computer.

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