Audio playback system and method for playing back an audio signal

The present invention relates to audio reproduction systems, and more particularly, to practical audio reproduction systems for variable size reproductions, such as cinemas, wherein the audio reproduction systems are based on wave field synthesis.

There is a growing need for new technologies and innovative products in the field of entertainment electronics. It is an important prerequisite for the success of new multimedial systems to offer optimal functionalities and abilities. This is achieved through the use of digital technologies and, in particular, computer technology. Examples include applications that provide an improved, realistic audio-visual impression. In previous audio systems, a significant weakness is the quality of the spatial sound reproduction of natural as well as of virtual environments.

Methods for the multi-channel loudspeaker reproduction of audio signals have been known and standardized for many years. All conventional techniques have the disadvantage that both the installation location of the loudspeakers and the position of the listener are already impressed on the transfer format. If the loudspeakers are incorrectly positioned with respect to the listener, the audio quality is significantly reduced. An optimal sound is only possible in a small area of ​​the playback area, the so-called sweet spot.

A better natural spatial impression as well as a stronger encapsulation in the audio reproduction can be achieved with the help of a new technology. The fundamentals of this technology, known as Wave Field Synthesis (WFS), have been researched at TU Delft and presented for the first time in the late 1980s (Berkhout, AJ, De Vries, D ,; : Acoustic control by Wavefield Synthesis, JASA 93, 1993).

As a result of the enormous demands of this method on computer performance and transmission rates, wave field synthesis has so far only rarely been applied in practice. Only the advances in the areas of microprocessor technology and audio coding now allow the use of this technology in concrete applications. First products in the professional sector are expected next year. In a few years, the first wave field synthesis applications for the consumer sector will also be launched.

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

Each point which is detected by a shaft is the starting point of an elementary wave which propagates in a spherical or circular manner.

Applied to acoustics, any form of an incoming wavefront can be simulated by a large number of loudspeakers arranged side by side (a so-called loudspeaker array). In the simplest case, a single point source to be reproduced and a linear arrangement of the loudspeakers, the audio signals of each loudspeaker must be fed with a time delay and amplitude scaling so that the radiated sound fields of the individual loudspeakers correctly overlap. In the case of several sound sources, the contribution to each loudspeaker is calculated separately for each source and the resulting signals are added. If the sources to be reproduced are in a room with reflecting walls, then reflections must also be reproduced as additional sources via the loudspeaker array. The effort involved in the calculation therefore depends strongly on the number of sound sources, the reflection characteristics of the recording space and the number of loudspeakers.

The advantage of this technique lies in particular in the fact that a natural spatial sound effect is possible over a large area of ​​the reproduction space. In contrast to the known techniques, the direction and distance of sound sources are reproduced very precisely. To a limited extent, virtual sound sources can even be positioned between the real loudspeaker array and the receiver.

Although the wave field synthesis works well for environments whose characteristics are known, irregularities occur when the texture changes, or when the wave field synthesis is performed on the basis of an environmental condition which does not match the actual nature of the environment.

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

This is explained in more detail with reference to the following example. It is assumed that a loudspeaker emits a sound signal against a wall, the reflection of which is undesirable. For this simple example, the space compensation using wave field 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 the amplitude of that reflected sound signal Has. If the reflection from this wall is undesirable, the wave field synthesis makes it possible to eliminate the reflection from this wall by impressing the loudspeaker with an amplitude corresponding to the reflection signal in addition to the original audio signal in such a way that the compensating wave, Reflection reflection wave, such that the reflection from this wall is eliminated in the environment being viewed. This can be done by first calculating the impulse response of the environment and determining the nature and position of the wall on the basis of the impulse response of this environment, the wall being interpreted as a mirror source, ie as a sound source which reflects an incident sound.

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

However, the decisive factor for optimal compensation of the reflected wave is that the pulse response of the space is precisely determined so that no over-compensation or under-compensation occurs.

The wave field synthesis thus enables a correct reproduction of virtual sound sources over a large reproduction range. At the same time, it offers the sound engineer and sound engineer new technical and creative potential for creating even more complex soundscapes. Wave field synthesis (WFS or sound field synthesis) developed at the TU Delft in the late 1980s is a holographic approach to sound reproduction. The Kirchhoff-Helmholtz integral serves as the basis for this. This means that any sound fields can be generated within a closed volume by means of a distribution of monopole and dipole sound sources (loudspeaker arrays) on the surface of this volume. For details, see MM Boone, ENG Verheijen, PF v. 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 case of 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 terms of amplitude and phase such that a wave resulting from the superimposition of the individual loudspeakers The sound output from the loudspeaker array corresponds to the wave that would originate from the virtual source at the virtual position if this virtual source at the virtual position were a real source with a real position.

Typically, several virtual sources are present at different virtual locations. The calculation of the synthesis signals is performed for each virtual source at each virtual position, so that typically a virtual source results in synthesizing signals for several loudspeakers. When viewed from a loudspeaker, this loudspeaker thus receives several synthesizing signals which are based on different virtual sources. An overlay of these sources, which is possible due to the linear superposition principle, then yields the reproduction signal actually emitted by the loudspeaker.

The larger the loudspeaker arrays, the more individual loudspeakers are provided, the better the possibilities of the wave field synthesis can be exhausted. However, this also increases the computing power which a wave field synthesis unit has to perform, since channel information must also be taken into account. This means in particular that, from each virtual source to each loudspeaker, there is in principle a separate transmission channel, and that in principle the case may be that each virtual source leads to a synthesis signal for each loudspeaker or that each loudspeaker has a number of synthesesignals Equal to the number of virtual sources.

If, in particular in the case of cinema applications, the possibilities of the wave field synthesis are to be exhausted in such a way that the virtual sources can also be mobile, it can be seen that on the basis of 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 Very considerable computing power.

Moreover, it should be noted here that the quality of the audio reproduction increases with the number of loudspeakers provided. This means that the audio reproduction quality is all the better and more realistic, the more loudspeakers are present in the loudspeaker array (s).

In the above scenario, the ready-rendered 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. This would have the advantage that almost all loudspeakers operate synchronously, so that no further measures would be necessary for synchronization purposes. On the other hand, the wave-field synthesis central unit could be produced only for a specific reproduction space or for a reproduction with a fixed number of loudspeakers. This means that for each reproduction space a separate wave-field synthesis central unit would have to be produced, 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 .

Particularly with regard to audio reproduction systems intended for cinemas, however, the problem exists that the reproduction spaces in cinemas vary considerably in size. Thus, cinemas sometimes have a very large cinema hall and / or at the same time several small cinemas for films that do not have such a high audience interest as films, which are to be played in large cinemas. However, different cinemas also have differently large playback spaces, which may vary to a factor of 100, especially when audiobooks are not only used in cinemas, for example in concert halls.

In order to equip such different audio reproduction rooms with an audio reproduction system on the basis of the wave field synthesis, for example, a separate wave field synthesis central unit would have to be built for each reproduction space, which is unacceptable in view of the price of the single production.

On the other hand, a maximum-equipped wave-field synthesis central unit could be constructed, which can be controlled with regard to the connectable loudspeakers, that is to say with regard to the number of analog signal outputs, but internally comprises computing processors which are designed for the maximum number of analog outputs, ie connectable loudspeakers.

Such a system would also result in the fact that audio reproduction systems for smaller playback rooms are priced at the same price as audiovisual systems for very large playback times, which is unlikely to be acceptable to operators of small playback rooms. In particular, the medium to small reproduction spaces are of interest to providers of audio reproduction systems, where the "smallest" reproduction spaces, eg domestic living rooms or smaller restaurants, are also mentioned.

The possibilities described above are thus disadvantageous in that a thoroughgoing market acceptance is not to be expected immediately.

The object of the present invention is to provide an audio reproduction concept which has a higher market acceptance.

This object is achieved by an audio reproduction system according to claim 1, a method for reproducing an audio signal according to claim 19 or a computer program according to patent claim 20.

The present invention is based on the recognition that audio reproduction systems which are to achieve market acceptance must be scalable. The scalability, however, must not only take place with regard to the computational capacity provided, but must also have an effect on the price of the audio reproduction system. In other words, an audio playback system for a large playback space may cost more than an audio playback system for a small playback space. In other words, an audio playback system for a small playback space must cost considerably less than an audio playback system for a large playback space.

In the conceivable concepts described above, the price differences were irrelevant, since price differences were due to the number of individual loudspeakers, which, however, were offered at low prices due to the fact that a large number of loudspeakers are provided and due to novel integration concepts Can be.

According to the invention, the audio reproduction system is divided into a central wave field synthesis module and many individual loudspeaker modules, which are connected decentrally to the central wave field synthesis module. The central wave-field synthesis module receives an audio signal with a plurality of audio tracks and in turn calculates the synthesizing signals and, on the other hand, the channel information for the channels from the virtual positions to the real loudspeaker positions.

The central wavefield synthesis module is further adapted to supply each loudspeaker with one or more synthesizing signals to be reproduced by the concerned loudspeaker as well as channel information for the audio channels from the virtual positions of the virtual sources from which the one or more syntheses signals originate to the affected one Speakers. In this case, a considerable data rate transmission limitation can already be achieved, since experience shows that very rarely is the case that each loudspeaker receives synthesesignals whose energy content is greater than a certain threshold value. The central wave-field synthesis module according to the invention thus already has the option of supplying only a synthesized signal to a decentralized loudspeaker module and, furthermore, only the channel information for the synthesis signals which are significant for the individual loudspeaker.

The loudspeaker modules according to the invention are designed decentrally and are directly coupled to the loudspeaker or preferably arranged in a spatial proximity to the loudspeaker. Each loudspeaker module comprises a receiver for receiving the one or more synthesizing signals for the affected loudspeaker as well as the channel information assigned to the synthesizing signals. Further, each loudspeaker module comprises a rendering means for calculating a reproducing signal for the loudspeaker by using the synthesizing signals and the channel information for the supplied synthesizing signals. Finally, each loudspeaker module also comprises a signal processing device with possibly a digital amplifier, a further digital signal processing device and finally a digital-to-analog converter for generating an analog loudspeaker signal which is to be supplied to the affected loudspeaker on the basis of the reproduction signal. A plurality of transmission paths is provided for connecting the central wave field synthesis module and the decentralized loudspeaker modules, wherein a transmission path extends from the central wave field synthesis module to the individual loudspeaker.

The operation of the rendering is very expensive, which in view of the required circuitry in the form of, for example, a DSP or a hard-wired circuit, contributes significantly to the costs, in particular if the multiplier intended for each individual loudspeaker is intended. Preferably, the rendering device operates as channel information using channel pulse responses and thus performs a computation-time-intensive convolution, which is either directly executable in the time domain or is performed in the frequency domain, wherein transformations into the frequency domain and transformations from the frequency domain are required With the actual multiplication operation in the frequency domain lead to a considerable effort. In particular, it is intended here that a rendering unit must not only render a single synthesis signal, but always a large number of synthesis signals, which normally correspond to the number of virtual sources.

The concept according to the invention leads to decentralized operations being transferred out of the central wave field synthesis module into the decentralized loudspeaker modules in such a way that in the best case only the operations in the central wave field synthesis module are executed which are equally important for all loudspeakers, , Which relate to only one loudspeaker, or several loudspeakers, which are connected to a loudspeaker module, can also be implemented decentrally in the loudspeaker module.

Thus, the cost for the central wave synthesis module can be significantly reduced, but at the expense of the speaker modules, the price of which is now negligible, due to the operation of the audio rendering mainly performed in the loudspeaker modules.

However, the audio reproduction system according to the invention can now be scaled in terms of both performance and price. It is possible to offer a central wave field synthesis module for a large number of reproducible rooms at a reduced price such that the cost for the overall system resulting from the cost of the central unit and the decentralized loudspeaker modules is now very high with the number of And thus the size of the playback space.

In other words, an operator of a large playback space will still have to pay a certain price for a playback system for its large playback space. On the other hand, however, an operator of a smaller playback space will be able to purchase an audio playback system at a considerably lower price, since the number of loudspeakers and thus the number of expensive and cost-intensive loudspeaker modules is considerably reduced compared to the large playback space.

The audio reproduction system according to the invention thus makes it possible to offer audio reproduction systems for smaller reproduction spaces at considerably reduced prices compared to large reproduction spaces so that a market acceptance is hoped for due to the reduced price on the very competitive market of the audio / video components.

In a preferred embodiment of the present invention, the central wavefield synthesis unit is designed to be able to process kinofilms recorded in the conventional audio format for kinofilms, customary recording formats being, for example, the 5.1 surround format or 7.1. Format or 10.2 format. In the example of the 5.1 format, such a cinema film comprises six audio tracks, ie audio tracks for the "left rear", "right rear", "front left", "front right" and "front center" channels, as well as the bass channel (subwoofer channel) ). A reproduction of such a kinofilm, which is conventional with regard to audio technology, in the audio reproduction system according to the invention can be achieved in that the audio tracks are placed as virtual sources at virtual positions which can be selected as desired by the audio master or the operator of the playback space. The possibility of compatible playback for an audio playback system with a scalable price therefore contributes to the fact that audio reproduction systems based on wave field synthesis already spread at a time when a small number of cinema / video films with completely wave field synthesis-suitable audio tracks are required together with the correspondingly necessary Metainformations about the recording setting.

Preferred exemplary embodiments of the present invention are explained in detail below with reference to the accompanying drawings. Show it:<dl id="dl0001"><dt>FIG</dt><dd>A conceptual diagram of the audio reproduction system according to the invention;</dd><dt>FIG</dt><dd>A block diagram of the central wave-field synthesis module according to the invention;</dd><dt>FIG</dt><dd>A block circuit diagram of a decentralized loudspeaker module according to the invention;</dd><dt>FIG</dt><dd>A block diagram of a preferred embodiment of the audio rendering unit in a decentralized loudspeaker module;</dd><dt>FIG</dt><dd>A schematic representation of a compatible playback with large sweet spot;</dd><dt>FIG</dt><dd>A principle diagram for the condition of a plurality of synthesizing signals for a loudspeaker each to be supplied with channel information to obtain the reproducing signal for the loudspeaker LSi; and</dd><dt>FIG</dt><dd>A schematic representation of a channel from a virtual source to a real loudspeaker, showing the quantities which can have an influence on the channel.</dd></dl>

The audio reproduction system according to the invention is basically divided into two parts, as shown in FIG. The other part is composed of individual loudspeaker modules 12a, 12b, 12c, 12d, 12e, which are connected to actual physical loudspeakers 14a, 14b, 14c, 14d, 14e, as shown in FIG. 1. It should be noted that the number of loudspeakers 14a-14e in typical applications is in the range over 50 and typically even well over 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 speak from a loudspeaker module of a small group of adjacent loudspeakers. In this connection, it is arbitrary whether a loudspeaker module which is connected to four loudspeakers, for example, feeds the four loudspeakers with the same reproducing signal, or whether different sound signals corresponding to the four loudspeakers are calculated so that such a loudspeaker module is actually composed of Several individual loudspeaker modules, which, however, are physically combined in one unit.

A separate transmission path 16a-16e is located between the wave field synthesis module 10 and each individual loudspeaker module 12a-12e, each transmission path being coupled to the central wave field synthesis module and its own loudspeaker module.

As a data transmission mode for transmitting data from the wave field synthesis module to a loudspeaker module, a serial transmission format which provides a high data rate, such as, for example, a so-called firewire transmission format or a USB data format, is preferred. Data transfer rates of more than 100 megabits per second are advantageous.

The data stream which is transmitted from the wave field synthesis module 10 to a loudspeaker module is accordingly formatted according to the selected data format in the wave field synthesis module and provided with synchronization information which is provided in conventional serial data formats. This synchronization information is extracted from the data stream by the individual loudspeaker modules and used to provide the individual loudspeaker modules with respect to their reproduction, ie ultimately to the digital-to-analog conversion for obtaining the analogue loudspeaker signal and the resampling provided therefor, To synchronize. It is preferred that the central wave field synthesis module operates as a master, and that all the loudspeaker modules operate as clients, the individual data streams receiving all the same synchronization information from the central module 10 over the various transmission paths 16a-16e. This ensures that all loudspeaker modules operate synchronously, in synchronization with the master 10, which is important for the present audio playback system so as not to suffer any loss of audio quality in order that the synthesized signals calculated by the wave field synthesis module do not depend on the individual loudspeakers Corresponding audio rendering. An advantage of this concept is that the individual loudspeaker modules need not be synchronized with one another. They are automatically synchronized with each other since they all run synchronously with the master. Connecting the individual loudspeaker modules one below the other will be unfavorable to the present invention because the modular concept of scalability with the loudspeaker modules requires a simple addition of modules with respect to the reproduction space size without the need for corresponding wiring under the modules.

FIG. 2 shows a block circuit diagram of a central wave field synthesis module according to a preferred exemplary embodiment of the present invention. The central wave field synthesis module initially comprises an input device 20, which is basically designed to receive an audio signal at an input, the audio signal having a plurality of audio tracks, each audio track being assigned an audio source position.

Depending on the application, the audio source position is an indication of the position of a loudspeaker with respect to a listener in the playback room according to a standardized audio format, such as 5.1, to achieve compatible playback. In this case the audio signal would have 5 + 1 = 6 audio tracks. Alternatively, the audio signal may have a greater number of audio tracks that are already present as waveform-suitable signals and represent audio sources in a real recording position that are imaged as virtual sources in the playback space using wave-field synthesis in view of the audio signal reproduction.

In a preferred embodiment of the present invention, the input device 20 is also used as the main control unit, which advantageously has further functionalities. In particular, it has the functionality of a decoding module, as is commonly used in cinemas. Alternatively or additionally, the input device 20 is also designed as a DVD decoder, which supplies the separate audio channels or audio tracks.

Alternatively, the reproducing device 20 is also designed as an MPEG-4 decoding module which provides audio paths 21 and corresponding audio source information 22 already intended for wave field synthesis. In particular, the audio tracks 21 each relate to audio signals of audio objects in a recording setting, to the position of the audio objects in the recording setting, to properties of audio objects, in particular with regard to the size of the audio object or the density with regard to the acoustic properties of the audio object .

Furthermore, it is also preferred to also transmit properties of the recording space or of the recording environment in addition to the audio tracks 21 in order to be able to take these into consideration in the case of wave field synthesis. The information about the recording space or the recording environment should serve to give the listener not only a visual but also an audible impression of the recording situation. Thus the visitor should also note on the reproduced sound whether a recording scene of a cinema film takes place under the open sky, for example, in a small space, such as a submarine. While a recording scenario under the open air provides relatively "dry" audio signals, since the recording environment shows little or no reflections, this situation will be completely different in a submarine. Here, the recording setting is represented by a very reflexionsreichen space or a very reflexions rich audio environment. In this case, it is preferred to record the audio tracks as dry as possible, that is, without the room acoustics in the recording space, and to describe the room acoustics with regard to their properties through additional meta information as can be transmitted in the standardized data stream in accordance with the standard MPEG 4.

The central wave field synthesis module further comprises means 24 for determining channel information on the one hand and wave field synthesizing signals on the other hand for the individual loudspeakers. For this purpose, a device 25 is also provided for converting the audio source positions 22 into virtual positions for the wave field synthesis.

Specifically, the means 24 is adapted to determine audio channel information for each audio channel from a virtual position to a speaker position, the virtual position being dependent on the audio source position associated with the audio track (means 25) such that for each channel of each Virtual position to each speaker audio channel information. Further, the means 24 is adapted to compute synthesizing signals from the virtual positions for the loudspeakers, using the principles of wave field synthesis as initially shown and known.

The central wave-field synthesis module in FIG. 2 further comprises means 26 for supplying synthesizing signals to one or more loudspeakers. The device 26 is further configured to transmit channel information for the transferred synthesis information from the central wave field synthesis module over the respective transmission paths to the individual loudspeaker modules so that an audio rendering can take place there. Depending on the embodiment, it is preferred to transmit further channel information for this channel to each synthesis signal which relates to a channel from a virtual position to a concrete loudspeaker. This means that, in a preferred embodiment of the present invention, the device 24 also provides channel information for each synthesis signal or interpolates it from computed channel information and provides the device 26 with a view to initiating a transmission to the individual loudspeaker modules. Preferably, the means 26 is adapted to filter out non-significant synthesizing signals and thus not to transmit the non-significant synthesizing signals or the associated channel information to save data transmission capacities. Thus, a case where a virtual source leads to significant synthesizing signals only for some loudspeakers, whereas, for all other loudspeakers in the loudspeaker array, it is also possible to calculate synthesis signals based on the theory of the wave field synthesis Are relatively small and can therefore be neglected with respect to a reduced data transmission amount.

More specifically, the device 24 includes functionalities to be used to preprocess the audio signals. In addition, the device 24 also controls the individual loudspeaker modules in particular in that it either directly or in conjunction with the device 26 introduces synchronization information into the data streams transmitted to the individual loudspeaker modules and thus achieves a central synchronization of all loudspeaker modules on the central wave field synthesis module.

In particular, the central wave field synthesis module is designed to carry out all processing operations which are the same for all reproduction channels, whereas according to the inventive concept the processing operations are performed decentrally, which are different for the individual loudspeakers or the individual reproduction channels.

The device 24 is further configured to perform a simulation of wave field synthesis information for stereo signals, 5.1 signals, 7.2 signals, 10.2 signals, etc., with a view to compatible reproduction. For this purpose, the standard positions of loudspeakers with respect to a reproduction space for the standardized audio format are used as audio source positions.

In this regard, reference is made to FIG. 5 below. FIG. 5 shows a reproduction space 50, a loudspeaker array 52 extending around the reproduction space, and a plurality of virtual sources 53a-53e, which, as can be seen in FIG. 5, are positioned at virtual positions outside the Playback space 50. The device 24 is designed in conjunction with the device 25 of FIG. 1 in order to calculate virtual positions, which can be controlled manually, from the audio source information, ie the standard position data for such a 5.1 signal. Depending on the embodiment, it is preferred to shift the virtual positions, for example, to infinity, so that the loudspeaker array 52 irradiates the reproduction space 50 with plane waves. This leads to the fact that the so-called sweet spot, that is to say the region in a reproduction space in which an optimum sound impression is obtained, is considerably increased in comparison with a customary situation in which real 5.7. loudspeakers are placed in the reproduction space.

Alternatively, the virtual sources can also be placed at finite virtual positions and modeled as point sources, which has the advantage that the sound impression on the cinema viewer / listener has a more pleasant effect. Level waves have the property that the listener has the impression that he is sitting in a very large room, which leads to an unpleasant sensory perception, for example, when a submarine scene takes place on the screen. In this connection, it should be pointed out that conventional cinema films with, for example, 5.1 audio tracks do not contain information on the acoustic properties of the recording setting. Therefore, in such a case, it is preferable to find a compromise between the plane waves, that is, the virtual sources at an infinite position, or the virtual sources at a finite position. In this context, the audio reproduction system according to the invention also provides the possibility of varying the virtual positions of the virtual loudspeakers 53a-53e, depending on the film scene. If, for example, a scene takes place in the open air, the loudspeakers can be positioned at infinity. If, on the other hand, a scene takes place in a small space, the loudspeakers can be positioned closer to the reproduction space 50.

In the context of compatible playback, in a preferred embodiment of the present invention, the input means 20 is adapted to scan the audio tracks associated with the video signal to sample a certain time "delay" before the video signals such that after processing in the wave field synthesis module Is sampled in the individual loudspeaker modules of the sound belonging to a time at the same time as the video signal belonging to a time. The negative "delay" must at least be dimensioned in such a way that sound and image are transmitted to one another in the audiovisual system according to the invention. If the negative delay is somewhat larger, then the signals can already be computed and, for example, output from the loudspeaker modules to the loudspeakers by means of a corresponding synchronization signal, which ensures synchronicity of the picture and sound.

Both in the case of compatible reproduction and in the case where the input audio signal comprises already prepared wave field synthesis information about sound sources in the recording setting, it is preferable to supply information on the reproduction space via a line 27 to the channel information calculating means 24 so that the synthesizing signals Can be processed using the information on the reproduction space in order to achieve, for example, an elimination of the acoustic properties of the reproduction space.

Information on the playback space can be determined either by virtue of the geometrical nature of the playback space, or in the playback space using the loudspeakers and special microphone arrays, whereby a control and evaluation for this can take place via an adaptation module 28 for the playback space. Thus, in an exemplary embodiment of the present invention, it is preferred to determine the acoustic properties of the reproduction space during playback and to adjust the information about the reproduction space accordingly, so that an ideal suppression of the cinema acoustics also takes place for a cinema which is filled, for example. At this point, it should be pointed out that, particularly in the case of smaller, fully filled reproduction spaces, the acoustic properties of the reproduction space are distinctly different from those in which no persons are present in the reproduction space.

The playback module adaptation module 28 further comprises a microphone array which can be used to measure the characteristics of the display. Further, the playback module adaptation module 28 includes algorithms to find the position of speaker arrays in the reproduction space. Furthermore, pre-processing of measurement results is carried out here in order to perform an optimal inversion of the room and loudspeaker characteristics, the adaptation module 28 being preferably controlled by the device 24 for this purpose.

Depending on the embodiment, the adaptation module 28 for the reproduction space is only required for the system configuration. If, however, a continuous adaptation to a changed situation in the reproduction space is desired, the adaptation module can also be used continuously during operation.

When the channel information calculating means 24 is used to process WFS-specific signals input to the device 20, the additional WFS information, that is, the characteristics of, for example, the audio objects and the characteristics of the recording space, are extracted from the input audio signal and transmitted via a WFS information line 29 to the device 24, so that this information can be taken into account in the channel information calculation.

In this case, the central WFS module is further configured to perform pre-processing of the WFS-processed audio signals. Furthermore, the device 24 and / or the device 26 is provided for achieving the synchronization between the picture and the sound. For this purpose, time codes have been introduced into the preferably series data streams for the individual loudspeaker modules, as has been stated. Finally, the channel information computation means 24, as already explained, is also responsible for driving the adaptation module 28 to control the measurement of the acoustic properties of the reproduction space, if desired, either before playback or during playback.

The multiplexer / transmitter stage 26 is designed to insert synchronization information generated either by the device 24 from the control device 20 or in the device 26 itself into the data streams to the loudspeaker modules, which also contain the synthesesignals required for the individual loudspeakers and the necessary loudspeakers Channel information.

At this point it should also be pointed out that the device 24 for calculating the channel information and the calculation of the synthesis signals also have to provide the loudspeaker location in the special reproduction space in order to calculate the individual synthesis signals and the individual channel information for the individual loudspeakers. This is represented symbolically in FIG. 2 by a line 30.

A preferred exemplary embodiment for a loudspeaker module is described below with reference to FIG. The loudspeaker module first comprises a receiver / decoder block 31 for receiving the data stream from the selection means and for extracting channel information 31b and synchronization information 31c from the same synthesis signal 31a. The loudspeaker module shown in FIG. 3 further comprises as a central unit an audio rendering means 32 for calculating a reproduction signal for the loudspeaker using the one or more synthesizing signals and using the channel information associated with the synthesizing signals. Finally, a loudspeaker module comprises a signal processing device 33 with a digital-to-analog converter for generating an analog loudspeaker signal, which is fed to the affected loudspeaker LSi 34 in order to generate a sound signal. The signal processing device 33 and, in particular, the resampler, which cooperates with the digital-to-analog converter, is supplied with the synchronization information (31c) extracted from the data stream by the receiver 31 in order to synchronize the synchronization data generated by the device, in synchronism with the central wave field synthesis module and thus synchronously with all other loudspeaker modules 24 of FIG. 1, which are superimposed on the loudspeakers and are supplied with channel information, in a timely manner.

The loudspeaker module shown in FIG. 3 is thus characterized by the combination of a digital receiver, a further signal processing device and a digital-to-analog converter, wherein a digital amplifier can also be provided in the signal processing device 33. Alternatively, however, the signal can also be amplified after the digital-to-analog conversion, although the digital gain is preferred because of the more precise possibility of synchronization. Furthermore, it is preferred to couple the loudspeaker 34 to the signal processing device 33 via a short analog line. If, however, it is not possible for the line from the signal processing device 33 to the loudspeaker 34 to be short, it is preferred that the corresponding lines of all loudspeakers have the same length or have a length difference which lies in a predetermined tolerance limit since the synchronization preferably takes place Digital side so that, with greatly different line lengths between the loudspeaker modules and the loudspeaker, a desynchronization could occur, which could already lead to audible artifacts or to a loss of the sound impression which is to be created by the wave field synthesis.

In a preferred exemplary embodiment of the present invention, channel pulse responses are transmitted as channel information in the time domain or in the frequency range. In this case, the audio rendering means 32 is implemented to perform a convolution of the individual synthesis signals with the channel information assigned to the synthesis signals. This convolution can actually be implemented in the time domain as a convolution, or can be performed as required in the frequency domain by multiplying the analysis signal in the frequency domain by the channel transmission function. An embodiment optimized with respect to the processing complexity is shown in FIG. FIG. 4 shows a preferred embodiment of the audio rendering device 32 and comprises, for each synthesis signal s_ji(T) a time-frequency conversion block 34a, 34b, 34c, and a multiplier 35a, 35b, 35c for multiplying the transform of a synthesizing signal by the transform of a channel pulse response H_ji(F), a summer 36, and a final frequency-time conversion means 37 connected as shown in FIG. The arrangement shown in FIG. 4 is characterized in that it is reduced in view of the processing complexity by virtue of the fact that the summation of the synthesis signals which have already been acted upon by the corresponding channel transmission functions takes place in the frequency range so that, for each loudspeaker module, Of the synthesis signals, only a single frequency-time conversion device is present. Depending on the embodiment, the time-frequency transformation of the synthesesignals S_ji Can be executed in parallel, or, if sufficient time is available, also serially / parallel or completely serially.

As has been pointed out, the preferred audio rendering device 32 shown in FIG. 4 is characterized in that it has only a single frequency-time conversion device 37, independent of the number of synthesesignals which are fed to a loudspeaker module, Which is preferably implemented as an inverse FFT, in which case the devices 34a, 34b, 34c are implemented as Fast Fourier Transform (FFT).

The audio rendering means 32 shown in FIG. 3 is further adapted to receive specific program information from the central wave field synthesis module shown in FIG. To this end, the multiplexer / send stage 26 comprises a special output to provide the program information to the loudspeaker modules. Depending on the application, the program information may also be multiplexed into the data stream with synthesis signals and channel information, although this is not absolutely necessary.

The following is an example of the transmission of program information to a loudspeaker module. If the channel information is described as channel pulse responses and is transmitted to the individual loudspeaker modules, it is preferred, in the sense of a data rate reduction, not to transmit the entire pulse response, but merely samples of the pulse response which lie in a front region of the pulse response, the envelope of which is still an amount Above a threshold. It should be pointed out at this point that impulse responses typically have large values ​​at small instants and gradually assume smaller values ​​and finally have a so-called "reverberant tail", which is important for the sound impression, but whose samples are no longer particularly large The specific phase relationships of which are no longer strongly perceived by the ear. In this case, it is preferred not to transmit the reverberant tail, whose envelope end is below the threshold value, by means of its sample values, but merely to transmit support values ​​for the envelope end. Samples for the reverberation tail required by the audio rendering means 32 are then generated according to the invention in that the audio rendering means produces a random sequence of zeros and ones whose amplitude is weighted with the transmitted support values ​​for the envelope . For further data reduction, it is preferred to transmit only a few support values ​​and to interpolate between the support values, and then to use the interpolated envelope to weight the random 0/1 sequence.

It should be noted that the random 0/1 sequence is preferably realized by positive voltage values ​​for a "1" and negative voltage values ​​for a "0". The information that the audio rendering device receives channel information which is actual samples up to a certain value, and then is merely the support values ​​for the envelope, is transmitted via the program information input shown in FIG. 3 or is fixed agreed.

The wave-field synthesis module according to the invention further comprises a WFS mixing console, not shown in FIG. 2, which comprises an authoring system to produce WFS sound descriptions.

The procedure based on the generation of synthesis signals is described below with reference to FIG. Consider a system with three virtual sources at three virtual positions 60, 61, 62 and a loudspeaker LSi 63 at a real loudspeaker position known to the central WFS module. Further, the virtual positions of the virtual sources 60, 61, 62 are known to the central wave-field synthesis module either by being input in a WFS-processed input signal or by the use of audio source positions by the virtual-position calculation means 25. The Synthesignals s_2i, s_2i and s_3i Are the signals which the loudspeaker 63 has to radiate and which return to the respective virtual positions 60, 61, From this it will be seen that, as has been stated, each loudspeaker will radiate the overlay of several synthesizer signals.

Further, between each virtual position and each speaker, a channel j_i Which can be described, for example, by an impulse response, a transmission function or any other channel information, as is illustrated with reference to FIG. All the desired properties can be packaged into the channel description in order then to apply the channel information for the corresponding channel assigned to a synthesis signal to the synthesis signals which are calculated by the wave field synthesis module. If the channel information is given in the form of an impulse response which describes the channel, the impulse is a convolution. If the signals are present in the frequency range, the load is a multiplication. Alternative channel information may also be used depending on the embodiment.

7, by means of which information a channel 70 from a virtual source 71 to a real loudspeaker 72 can be influenced. First, the virtual position of the virtual source 71 enters into the channel information, that is to say, for example, the channel pulse response. In addition, properties of the virtual source, such as size, density, etc., are entered. For example, a small triangle will have to be described and modeled differently as a large timbre. Further, as shown in FIG. 7, the characteristics of the receiving space enter the channel transfer function. Further influencing components are a system distortion of the entire audio reproduction system, in which, for example, loudspeaker distortions or non-functionalities of the loudspeakers are contained. The channel information also contains information about the playback space to achieve compensation for the acoustic properties of the playback space. If, for example, it is known from the reproduction space that it has a wall which is located on the front side of a loudspeaker and which is to be reflected and whose reflection is to be suppressed, the corresponding loudspeaker is controlled in such a way that it contains a signal which leads to the loudspeaker Is phase-shifted by 180 degrees and has a corresponding amplitude, so that an extinguishing reflection occurs and the wall becomes acoustically transparent, ie is no longer identifiable by a listener due to the reflections.

Finally, the channel information may also be used to adjust a particular target rendering acoustics. For this purpose, it is preferred first to suppress the acoustics of the reproduction space in the form of a reproduction space compensation in order then to generate channel information and to supply it to the wave field synthesis module so that an acoustics of any other reproduction space can be simulated in a reproduction space.

Depending on the circumstances, the method according to the invention can be implemented for reproducing an audio signal in hardware or in software. The implementation can be carried out on a digital storage medium, in particular a floppy disk or CD, with electronically readable control signals which can cooperate with a programmable computer system in such a way that the method is carried out. In general, the invention thus also consists in a computer program product with a program code stored on a machine-readable carrier for carrying out the method according to the invention when the computer program product runs on a computer. In other words, the invention can thus be implemented as a computer program with a program code for carrying out the method when the computer program runs on a computer.