EP1576847B1 - Audio playback system and method for playing back an audio signal - Google Patents

Abstract

An audio playback system is divided into a central wavefield synthesis module (10) and a multitude of decentrally arranged loudspeaker modules (12a-12e). Synthesis signals for the individual loudspeakers and corresponding items of channel information, which are assigned to the synthesis signals, are calculated in the central wavefield synthesis module. The synthesis signals for a loudspeaker together with associated items of channel information are then transmitted to corresponding loudspeaker modules via a transmission link (16a-16e). Each loudspeaker module receives the synthesis signals and associated items of channel information that are intended for the loudspeaker assigned to the loudspeaker module. A decentralized audio rendering and digital-to-analog conversion takes place inside the loudspeaker modules in order to decentrally generate the actual analog loudspeaker signals in spatial proximity to each loudspeaker. The division into a central wavefield synthesis module and a multitude of decentralized loudspeaker modules enables the production of audio playback systems that can be scaled with regard to price in order to offer different size systems, which can be scaled in terms of price, for, in particular, cinema playback spaces that vary greatly in size.

Description

The present invention relates to audio playback systems and, more particularly, to audio-reproduction systems suitable for practical use for variable size rendering rooms, such as cinemas, the audio display systems being based on wave-field synthesis.

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 conventional techniques have the disadvantage that both the location of the speakers 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, Wave Field Synthesis (WFS), were researched at the TU Delft and first introduced in the late 1980s (Berkhout, AJ, de Vries, D .; Vogel, P.). : Acoustic Control by Wavefield Synthesis JASA 93, 1993).

Due to the enormous demands of this method on computer performance and transmission rates, wave field synthesis has 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 that is detected by a wave is the starting point of an elementary wave that propagates 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 loudspeaker must be fed with a time delay and amplitude scaling so that the radiated sound fields of the individual loudspeakers are correctly superimposed. With multiple sound sources, the contribution to each source is made Speaker calculated separately and the resulting signals added. If the sources to be reproduced are in a room with reflective walls, reflections must also be reproduced as additional sources via the loudspeaker array. The cost of the calculation therefore depends heavily on the number of sound sources, the reflection characteristics of the recording room and the number of speakers.

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

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

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

This will be explained in more detail with reference to the following example. It is assumed that a loudspeaker emits a sound signal against a wall whose reflection is undesirable. For this simple example, the space compensation using wavefield synthesis would be to first determine the reflection of that wall to determine when a sound signal reflected from the wall will return to the loudspeaker and what amplitude this reflected sound signal will be Has. If the reflection from this wall is undesirable, then exists with the Wave Field Synthesis the ability to eliminate the reflection from this wall by the speaker is injected to the reflection signal in phase with the corresponding 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 that 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 mirror source, that is, a sound source reflecting incident sound.

First, the impulse response of this environment is measured and then the compensation signal is calculated, which must be impressed on the audio signal superimposed on the speaker, there will be a cancellation of the reflection of this wall, such that a listener in this environment has the sound that this impression 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 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. Details can be found in 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 wave field synthesis, an audio signal that emits a virtual source at a virtual position is used to compute a synthesis signal for each loudspeaker of the loudspeaker array, the synthesis signals being shaped in amplitude and phase such that a wave resulting from the superposition of the individual ones 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, yields then the playback signal actually sent from the speaker.

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

In 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, it will be appreciated that due to the calculation of the synthesis signals, the calculation of the channel information and the generation of the playback 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 available speakers. This means that the audio reproduction quality becomes better and more realistic as more speakers 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 provided, for example, via two-wire lines from the wave field synthesis central processing unit be transferred to the individual speakers. Although this would have the advantage that it is almost ensured that all speakers work in sync, so that no further action would be required here for synchronization purposes. 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, particularly with regard to many loudspeakers or many virtual sources ,

However, especially with regard to audio reproduction systems intended for cinemas, there is the problem that the reproduction spaces in cinemas vary considerably in size. For example, cinemas sometimes have a very large cinema hall and / or several small cinemas for films that do not experience as many audiences as films that are to be played in large movie theaters. However, different cinemas have different sized playback rooms, especially when an audio playback not only in cinemas but z. B. in concert halls, may possibly vary up to a factor of 100.

To equip such different audio playback rooms with an audio playback system based on the wave field synthesis, z. For example, a separate wave field synthesis central unit can be built for each playback room, which is unacceptable in terms of price due to the single production.

On the other hand, a maximum equipped wave field synthesis central unit could be constructed, which is controllable with regard to the connectable speakers, so in terms of the number of analog signal outputs, but internally comprises arithmetic processors, which is designed for the maximum number of analog outputs, so connectable speakers.

Such a system would also mean that smaller-room audio playback systems would cost almost the same price as very large-format audio playback systems, which would be unacceptable to the small-playback room operators. In particular, the medium to small playback rooms are interesting for providers of audio playback systems, at which point the "smallest" playback rooms should be mentioned, the z. B. domestic living room or smaller restaurants.

The above-described possibilities are therefore disadvantageous in that a thorough market acceptance is not to be expected immediately.

The object of the present invention is to provide an audio reproduction concept that has 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 claim 20.

The present invention is based on the recognition that audio playback systems that are to achieve market acceptance must be scalable. However, the scalability must not only take place in terms of the computing power provided, but must also be reflected in the price of the audio playback system impact. In other words, this means that an audio reproduction system for a large reproduction room may cost more than an audio reproduction system for a small reproduction room. In other words, an audio reproduction system for a small reproduction room must cost considerably less than an audio reproduction system for a large reproduction room.

In the conceivable concepts described above, the price differences were insignificant, since price differences were due only to the number of individual speakers, but due to the fact that a very large number of loudspeakers are provided, and due to novel integration concepts offered in the building that includes the playback room, low priced can be.

According to the invention, the audio reproduction system is divided into a central wave field synthesis module and many individual speaker modules connected to the central wave field synthesis module in a decentralized manner. The central wave-field synthesis module receives an audio signal having a plurality of audio tracks and calculates, on the one hand, the synthesis signals and, on the other hand, the channel information for the channels from the virtual positions to the real speaker positions.

The central wave-field synthesis module is further configured to supply to each loudspeaker one or more synthesis signals to be reproduced by the affected loudspeaker and channel information for the audio channels from the virtual positions of the virtual sources from which the one or more synthesis signals originate to the affected one To deliver speakers. In this case, a considerable data rate transmission restriction can already be achieved, since experience shows that the case occurs very seldom that each loudspeaker receives synthesis signals whose Energy content is greater than a certain threshold. The central wavefield synthesis module according to the invention thus already has the option of supplying only a decentralized loudspeaker module with the synthesis signals 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 to be decentralized and directly coupled to the loudspeaker or preferably arranged in spatial proximity to the loudspeaker. Each speaker module includes a receiver for receiving the one or more synthesis signals for the affected speaker and the channel information associated with the synthesis signals. Further, each speaker module includes a rendering device for calculating a reproduction signal for the speaker using the synthesis signals and the channel information for the supplied synthesis signals. Finally, each loudspeaker module also comprises a signal processing device possibly having a digital amplifier, a further digital signal processing device and finally a digital-to-analogue converter for generating an analogue loudspeaker signal to be supplied to the relevant loudspeaker on the basis of the playback signal. For connecting the central wave field synthesis module and the decentralized loudspeaker modules, a plurality of transmission links are provided, wherein a transmission path extends from the central wave field synthesis module to the individual loudspeaker.

Highly computationally expensive is the operation of rendering, which adds significantly to the cost of the required circuit hardware such as DSP or hardwired circuitry, especially when considering the multiplier provided for each individual loudspeaker. Preferably the rendering device uses channel impulse responses as channel information and thus performs computationally intensive convolution that is either directly executable in the time domain, or performed in the frequency domain, requiring frequency domain transforms and frequency domain transformations together with the actual multiplication operation in the frequency domain lead to a considerable effort. In particular, it is contemplated that a rendering unit must render not only a single synthesis signal, but always a large number of synthesis signals that normally correspond to the number of virtual sources.

The concept according to the invention results in decentralized operations being shifted 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 wavefronts synthesis module are carried out, which are equally important for all loudspeakers, during all operations which only cover one loudspeaker, or several loudspeakers connected to a loudspeaker module, can also be executed decentrally in the loudspeaker module.

Thus, the cost of the central wave synthesis module can be significantly reduced, but at the expense of the speaker modules, whose price is now no longer negligible, due to the operation of the audio rendering, which is mainly carried out in the loudspeaker modules.

However, the audio playback system according to the invention is now scalable in terms of both performance and price. It offers the possibility of offering a central wave field synthesis module for a large number of reproduction rooms at a reduced price, such that the cost of the overall system, which results from the cost of the central unit and the distributed loudspeaker modules, now strongly correlates with the number of loudspeakers established and thus the size of the playback room.

In other words, an operator of a large rendering room will still have to pay a certain price for a rendering system for his large rendering room. On the other hand, however, an operator of a smaller playback room will be able to purchase an audio playback system at a significantly lower price because the number of speakers, and thus the number of expensive and expensive speaker modules, is significantly reduced compared to the large display room.

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

In a preferred embodiment of the present invention, the central wave field synthesis unit is designed to be able to process cinematographic films recorded in the conventional audio format for motion picture films, conventional recording formats being, for example, the 5.1 surround format or 7.1.format or 10.2 format. Such a movie includes the example of the 5.1 format six audio tracks, ie audio tracks for the channel "left behind", "right back", "front left", "front right" and "front center", and the bass channel (subwoofer channel ). A reproduction of such a conventional audio film in the audio playback system according to the invention can be achieved by the Audio tracks are placed as virtual sources at virtual positions, which can be selected as desired by the Tonmeister or the operator of the playback room. The ability of compatible playback for a scalable-price audio playback system therefore provides a contribution that audio playback systems based on wave-field synthesis are already propagating at a time when there are still few cinema / video films with fully wave-field synthesis-ready audio tracks along with the appropriate ones Meta information about the recording setting available.

Fig. 1

ein Konzeptionsdiagramm des erfindungsgemäßen Audiowiedergabesystems;

Fig. 2

ein Blockschaltbild des erfindungsgemäßen zentralen Wellenfeldsynthesemoduls;

Fig. 3

ein Blockschaltbild eines erfindungsgemäßen dezentralen Lautsprechermoduls;

Fig. 4

ein Blockschaltbild einer bevorzugten Ausgestaltung der Audio-Renderingeinheit in einem dezentralen Lautsprechermodul;

Fig. 5

eine Prinzipdarstellung einer kompatiblen Wiedergabe mit großem Sweet Spot;

Fig. 6

eine Prinzipskizze für das Zustandekommen von mehreren Synthesesignalen für einen Lautsprecher, die jeweils mit Kanalinformationen zu beaufschlagen sind, um das Wiedergabesignal für den Lautsprecher LSi zu erhalten; und

Fig. 7

eine Prinzipdarstellung eines Kanals von einer virtuellen Quelle zu einem realen Lautsprecher mit Darstellung der Größen, die einen Einfluß auf den Kanal haben können.

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

Fig. 1

a conceptual diagram of the audio playback system according to the invention;

Fig. 2

a block diagram of the central wave field synthesis module according to the invention;

Fig. 3

a block diagram of a decentralized loudspeaker module according to the invention;

Fig. 4

a block diagram of a preferred embodiment of the audio rendering unit in a decentralized speaker module;

Fig. 5

a schematic representation of a compatible playback with a large sweet spot;

Fig. 6

a schematic diagram for the formation of several synthesis signals for a speaker, each to be acted upon with channel information to obtain the playback signal for the loudspeaker LSi; and

Fig. 7

a schematic representation of a channel from a virtual source to a real speaker with representation of the sizes that may have an influence on the channel.

The audio playback system according to the invention is divided, as shown in Fig. 1, basically in two parts. One part is the central wave field synthesis module 10. The other part is composed of individual speaker modules 12a, 12b, 12c, 12d, 12e which are connected to actual physical speakers 14a, 14b, 14c, 14d, 14e as shown in FIG. 1 is shown. 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 connection, it is arbitrary whether a loudspeaker module connected to four loudspeakers, for example, feeds the four loudspeakers with the same playback signal, or whether corresponding different synthesis signals are calculated for the four loudspeakers, so that such a loudspeaker module is 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 link 16a-16e, each transmission link being coupled to the central wave field synthesis module and to a separate loudspeaker module.

As a data transfer mode for transferring 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. 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 correspondingly 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 resample the individual loudspeaker modules with respect to their reproduction, ie ultimately to the digital-to-analogue conversion for obtaining the analogue loudspeaker signal and the sampling provided for this purpose. to synchronize. It is preferred that the central wave-field synthesis module operate as a master, and that all loudspeaker modules operate as clients, with the individual data streams over the various links 16a-16e all receiving the same synchronization information from the central module 10. This ensures that all the loudspeaker modules operate synchronously, synchronized by the master 10, which is important to the present audio playback system so as not to suffer any loss of audio quality, so that the synthesized signals computed by the wavefronts synthesis module are not delayed in time from the individual loudspeakers corresponding audio rendering are emitted. An advantage of this concept is that the individual speaker modules do not have to be synchronized with each other. They are automatically synchronized with each other as they all run in sync with the master. A connection of the individual loudspeaker modules with each other will be unfavorable for the present invention, since the modular concept of scalability with the speaker modules in terms of the playback room size requires a simple addition of modules, without corresponding wiring must be achieved under the modules.

2 shows a block diagram of a central wave-field synthesis module according to a preferred embodiment of the present invention. The central wave-field synthesis module first 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. B. 5.1 to achieve a compatible playback. In this case, the audio signal would have 5 + 1 = 6 audio tracks. Alternatively, the audio signal may have a larger number of audio tracks already present as wave-field synthesis-capable signals and representing audio sources in a real recording position which are mapped to the audio signal reproduction as virtual sources in the playback room using wave-field synthesis.

The input device 20 is further used in a preferred embodiment of the present invention as a 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 display device 20 is also embodied as an MPEG-4 decoding module which already supplies audio tracks 21 intended for wave field synthesis and corresponding audio source information 22. In particular, the audio tracks 21 respectively 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 respect to the size of the audio object or the density with respect to the acoustic properties of the audio object ,

Furthermore, it is preferable to also transfer properties of the recording space or the recording environment in addition to the audio tracks 21 in order to be able to take them into account in the wave field synthesis, if appropriate. The information about the recording room or the recording environment should serve to give the listener not only a visual but also an audiomäßiger impression of the recording situation. So the visitor should also remember the reproduced sound, whether a recording scene of a movie in the open air, for example, plays or z. In a small room, such as a submarine. For example, while an outdoor recording scenario provides relatively "dry" audio as the recording environment shows little or no reflection, this situation in a submarine, for example, will be completely different. Here, the recording setting is represented by a very reflective room or a very reflective audio environment. In this case, it is preferred to record the audio tracks as dry as possible, ie without the room acoustics in the recording room and to describe the room acoustics with regard to their properties by additional meta information, as they can be transmitted in the standardized data stream according to the MPEG 4 standard.

The central wave field synthesis module further comprises means 24 for determining channel information, on the one hand, and wave field synthesis 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 arranged to determine audio channel information for each audio channel from a virtual position to a speaker position, the virtual position depending on the audio source position associated with the audio track (means 25), so that for each channel of each virtual channel to each speaker, audio channel information is present. Further, the device 24 is configured to compute synthesis signals from the virtual locations for the speakers using the principles of wave field synthesis as illustrated and known above.

The central wave-field synthesis module in Fig. 2 further comprises means 26 for providing synthesis signals to one or more speakers. The device 26 is further configured to transmit channel information for the transmitted synthesis information from the central wave-field synthesis module via the corresponding transmission links 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 for each synthesis signal relating to a channel from a virtual position to a concrete loudspeaker. That is, in a preferred embodiment of the present invention, means 24 for each synthesis signal also provides channel information and interpolates from calculated channel information and means 26 for To initiate transmission to the individual loudspeaker modules. Preferably, the means 26 is arranged to filter out non-significant synthesis signals and thus to transmit neither the non-significant synthesis signals nor the associated channel information in order to save data transmission capacities. Thus, the case often occurs that a virtual source leads to significant synthesis signals only for some speakers, while for all other speakers in the speaker array, although synthesis signals can be calculated due to the theory of wave field synthesis, but z. B. in terms of their performance in a given period of time are relatively small and can therefore be neglected in view of a reduced amount of data transmission.

More specifically, the device 24 includes functionality to be used to pre-process the audio signals. In addition, the device 24 controls the individual loudspeaker modules in particular to the effect that it brings synchronization information either directly or in connection with the device 26 in the data streams transmitted to the individual loudspeaker modules and thus achieves a central synchronization of all loudspeaker modules to the central wave field synthesis module.

In particular, the central wave-field synthesis module is designed to perform all processing operations that are the same for all reproduction channels, while according to the inventive concept the processing operations are performed decentralized, which are different for the individual loudspeakers or the individual reproduction channels.

The device 24 is further configured to simulate wave field synthesis information for stereo signals, 5.1 signals, 7.2 signals, 10.2 signals, etc. with a view to a 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 below to FIG. 5. FIG. 5 shows a playback room 50, a speaker array 52 extending around the playback room, and a plurality of virtual sources 53a-53e positioned at virtual positions, as seen in FIG Playroom 50 are located. The device 24 is designed in conjunction with the device 25 of FIG. 1 in order to calculate virtual positions from the audio source information, ie the standard position information for such a 5.1 signal, for example, which are manually controllable. Depending on the embodiment, it is preferred that the virtual positions z. B. to infinity, so that the speaker array 52 sonicates the playback room 50 with even waves. As a result, the so-called sweet spot, that is, the area in a reproduction room in which an optimal sound impression is obtained, is considerably increased as compared with a conventional situation where real 5.7 speakers are placed in the reproduction room.

Alternatively, the virtual sources may also be placed at finite virtual positions and modeled as point sources, this option having the advantage that the sound impression is more pleasing to the cinema viewer / listener. Level waves have the property that the listener has the impression that he sits in a very large room, which in particular leads to an unpleasant sensory perception when, for example, a submarine scene is taking place on the screen. In this context, it should be noted that usual movies with example 5.1 audio tracks no information about acoustic Include recording settings. Therefore, in such a case, it is preferable to find a compromise between the plane waves, that is, the infinite position virtual sources or the virtual sources at a finite position. In this connection, the audio playback system according to the invention also provides the possibility of varying the virtual positions of the virtual loudspeakers 53a-53e depending on the scene of the movie. If, for example, an outdoor scene takes place, the speakers can be positioned at infinity. Conversely, if a scene takes place in a small room, the speakers can be positioned closer to the playback room 50.

In the context of compatible playback, in a preferred embodiment of the present invention, the input device 20 is configured to sample the audio tracks associated with the video signal to sample a certain time "delay" from the video signals, such that after processing in the wave field synthesis module in the individual loudspeaker modules, the sound belonging at a time is sampled simultaneously with the video signal belonging at a time. The negative "delay" must at least be such that in the audio playback system according to the invention sound and image are radiated to each other. If the negative delay is dimensioned somewhat larger, the signals can already be completely calculated and output from the loudspeaker modules to the loudspeakers, for example, by means of a corresponding synchronization signal which ensures synchronicity of image and sound.

Both in the case of the compatible reproduction and in the case where the input audio signal already includes prepared wave field synthesis information about sound sources in the recording setting, it is preferable to provide information about the reproduction room via a line 27 of FIG Channel information computation device 24 supply so that the synthesis signals can be prepared using the information about the playback room to z. B. to achieve an elimination of the acoustic properties of the playback room.

Information about the playback room can either be determined on the basis of the geometric nature of the playback room, or measured in the playback room using the speakers and special microphone arrays, wherein an activation and evaluation thereof can take place via an adaptation module 28 for the playback room. Thus, in one embodiment of the present invention, it is preferred to determine the acoustic characteristics of the playback room during playback and, accordingly, to readjust the information about the playback room, so that even for a cinema filled, for example, cinema acoustics are suppressed optimally. It should be noted that, especially in smaller, well-filled playback rooms, the acoustic properties of the playback room significantly different from those when there are no people in the playback room.

The reproduction room adaptation module 28 further comprises a microphone array which can be used to measure the characteristics of the reproduction. Further, the playback room adaptation module 28 includes algorithms for finding the location of speaker arrays in the reproduction room. Furthermore, a preprocessing of measurement results is carried out here in order to carry out an optimal inversion of the room and loudspeaker characteristics, wherein the adaptation module 28 is preferably controlled by the device 24 for this purpose.

Depending on the embodiment, the adaptation module 28 for the reproduction room is needed only for system construction. However, if a continuous adaptation to a changed situation in the reproduction room is desired, the adaptation module 28 can also be used continuously during operation.

When the channel information calculator 24 is used to process WFS-specific signals inputted 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 via a WFS information line 29 of 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 intended to achieve the synchronization between image and sound, for which purpose, as has been explained, time codes are introduced in the preferably serial data streams to the individual loudspeaker modules. Finally, as previously stated, the channel information calculator 24 is also responsible for driving the adaptation module 28 to control the measurement of the acoustic characteristics 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, by the control device 20 or in the device 26 itself into the data streams to the loudspeaker modules, which are further provided for the individual speakers required synthesis signals and necessary channel information to be supplied.

It should also be noted that the channel information computation means 24 and the synthesis signal computation means 24 must further be provided with the speaker locations in the special playback room to compute the individual synthesis signals and the individual channel information for the individual speakers. This is shown symbolically in FIG. 2 by a line 30.

Hereinafter, referring to FIG. 3, a preferred embodiment for a speaker module will be discussed. The speaker module first includes a receiver / decoder block 31 for receiving the data stream from the selection device and for extracting from the same synthesis signal 31a, associated channel information 31b and synchronization information 31c. The loudspeaker module shown in Fig. 3 further comprises as a central unit an audio rendering device 32 for calculating a reproduction signal for the loudspeaker using the one or more synthesis signals and using the channel information associated with the synthesis signals. Finally, a loudspeaker module comprises a signal processing device 33 with a digital / analogue converter for generating an analogue loudspeaker signal, which is supplied to the relevant 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 / analog converter, is supplied via the synchronization information (31c) extracted from the data stream by the receiver 31 in order synchronously with the central wave field synthesis module and thus synchronously with all other loudspeaker modules that of the device 24 of Fig. 1 calculated on the Superimposed loudspeakers superimposed and acted upon with channel information synthesis signals.

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-analog converter, wherein in the signal processing device 33 in particular a digital amplifier can also be provided. Alternatively, however, the signal may be amplified even after the digital-to-analog conversion, although the digital gain is preferred because of the more accurate possibility of synchronization. Furthermore, it is preferred to couple the loudspeaker 34 to the signal processing device 33 via a short analogue line. However, if it is not possible that the line from the signal processor 33 to the loudspeaker 34 is short, it is preferred that the respective lines of all the loudspeakers have the same length or have differences in length which are within a predetermined tolerance because the synchronization is preferably on digital side is performed so that at very different line lengths between the speaker modules and the speaker desynchronization could occur, the already audible artifacts or a loss of sound impression, which should be created by the wave field synthesis, could result.

In a preferred embodiment of the present invention, channel impulse responses in the time domain or in the frequency domain are transmitted as channel information. In this case, the audio rendering device 32 is designed to perform a convolution of the individual synthesis signals with the channel information associated with the synthesis signals. This convolution can actually be implemented in the time domain as convolution, or can be implemented as needed in the time domain Frequency domain by multiplying the analysis signal in the frequency domain with the channel transfer function are performed. An optimized with regard to the processing effort embodiment is shown in Fig. 4. 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 for each branch a multiplier 35a, 35b, 35c for multiplying the transform of one Synthesis signal with the transform of a channel impulse response H _ji (f), a summer 36 and a final frequency-time conversion means 37, which are connected in such a way as shown in Fig. 4. The arrangement shown in Fig. 4 is characterized in that it is reduced in terms of processing complexity by the summation of the synthesis signals, which are already subjected to the corresponding channel transfer functions, takes place in the frequency domain, so that for each speaker module, regardless of the number the synthesis signals only a single frequency-time conversion device is present. Depending on the embodiment, the time-frequency transformation of the synthesis signals S _ji can be carried out completely in parallel, or, if sufficient time is available, also serial / parallel or completely serial.

As has been stated, the preferred audio rendering device 32 shown in Figure 4 is characterized by having only a single frequency-to-time translator 37, regardless of the number of synthesis signals applied to a loudspeaker module. which is preferably implemented as inverse FFT, in which case the devices 34a, 34b, 34c are implemented as FFT (fast Fourier transform).

The audio rendering device 32 shown in FIG. 3 is further configured to receive special program information from the central wave field synthesis module shown in FIG. 2. To this end, the multiplexer / transmitter stage 26 includes a dedicated output to provide the program information to the speaker modules. Depending on the application, the program information can 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 transmitting program information to a speaker module. If the channel information is described as channel impulse responses and transmitted to the individual loudspeaker modules, it is preferred not to transmit the entire impulse response in terms of data rate saving, but only impulse response samples located in a forward portion of the impulse response whose envelope is still an amount above a threshold. At this point it should be noted that impulse responses typically have large values at small times and gradually take on smaller values and finally have a so-called "reverberation tail", which is important for the sound impression but whose samples are no longer particularly large, and whose special phase relationships are no longer strongly perceived by the ear. In this case, it is preferable to no longer transmit the reverberation tail, whose envelope lies below the threshold value, on the basis of its sampled values, but merely to transmit base values for the envelope. Reverberation tail samples needed by the audio rendering device 32 are then generated according to the invention by the audio rendering device generating a random sequence of zeros and ones whose amplitude is weighted with the transmitted envelope 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 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 that is up to a certain value actual samples and then only 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 of the present invention further includes a WFS mixing console, not shown in FIG. 2, which includes an authoring system to generate WFS sound descriptions.

Referring now to Fig. 6, the procedure underlying the generation of synthesis signals is discussed. Consider a system with three virtual sources at three virtual locations 60, 61, 62 and a loudspeaker LSi 63 at a real speaker 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 from being fed in a WFS-edited input signal or from being derived by the virtual position calculation means 25 using audio source positions. The synthesis signals s _2i, s _2i and s _3i are the signals which the loudspeaker 63 must emit and which go back to the respective virtual positions 60, 61, 62. It will thus be seen that, as has been stated, each speaker will emit the superposition of multiple synthesis signals.

Between each virtual position and each speaker is further defined a channel j _i , for example may be described by an impulse response, a transfer function, or any other channel information, as illustrated with reference to FIG. In the channel description, all the desired properties can be packaged 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 in the form of an impulse response describing the channel, then the imposition is a convolution. If the signals are present in the frequency domain, the application is multiplied. Alternative channel information may also be used depending on the embodiment.

In the following, it is shown with reference to FIG. 7 through 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 impulse response. Furthermore, properties of the virtual source are included, such as: B. size, density, etc. So z. For example, a small triangle must be described and modeled differently than a large timpani. Further, as shown in Fig. 7, the characteristics of the accommodating space enter into 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-idealities of the loudspeakers are contained. Information about the reproduction space is also input to the channel information in order to achieve a compensation of the acoustic properties of the reproduction space. For example, if it is known from the playback room that it has a wall facing the front of a loudspeaker, which reflects, but whose reflection is to be suppressed, then the corresponding loudspeaker becomes submerged Considering this information is controlled so that it contains a signal which is 180 degrees out of phase with the reflected signal and has a corresponding amplitude, so that an extinguishing reflection occurs and the wall is acoustically transparent, ie no longer for a listener due to the reflections is identifiable.

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

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

Claims (20)

1. An audio reproduction system for a reproduction room, wherein a plurality of loudspeakers (14a - 14e) is disposed at defined loudspeaker positions, by using an audio signal with a plurality of audio tracks, wherein an audio source position is associated to every audio track, comprising:

   a central wave-field synthesis module (10), formed to
   determine audio channel information for every audio channel from a virtual position to a loudspeaker position, wherein the virtual position depends on the audio source position associated to the audio track, so that audio channel information is present for every channel from every virtual position to every loudspeaker,
   calculate (24) synthesis signals from the virtual positions for the loudspeakers, and
   supply (26) one or several synthesis signals to every loudspeaker to be reproduced by the respective loudspeaker, as well as channel information for the one or the several synthesis signals;

   a plurality of loudspeaker modules (12a - 12e), wherein a loudspeaker module is associated to a loudspeaker and wherein every loudspeaker module comprises:

   a receiver (31) for receiving the one or several synthesis signals for the respective loudspeakers as well as the channel information;

   a rendering means (32) for calculating a reproduction signal for the loudspeaker by using the one or several synthesis signals and the channel information for the respective loudspeaker; and

   a signal processing means (33) for generating an analog loudspeaker signal, which can be supplied to the respective loudspeaker due to the reproduction signal; and

   a plurality of transmission lines (16a - 16e) from the central wave-field synthesis module to every loudspeaker, wherein every transmission path is coupled to the central wave-field synthesis module on the one hand and to an individual loudspeaker module on the other hand.
2. The audio reproduction system according to claim 1, wherein every loudspeaker module is combined with the loudspeaker to which the same is associated, so that a spatial distance between the loudspeaker and the loudspeaker module is smaller than a spatial distance between the loudspeaker module and the central wave-field synthesis module.
3. The audio reproduction system according to claim 1 or 2, wherein the audio channel information is impulse responses for the audio channels.
4. The audio reproduction system of claim 3, wherein the rendering means for calculating a reproduction system has a convolution means to perform one or several convolutions by using the one or several synthesis signals with the respective impulse responses.
5. The audio reproduction system of claim 4, wherein the rendering means (32) comprises:

   a time-domain frequency-domain conversion means (34a, 34b, 34c) for every synthesis signal;

   a multiplication means (35a, 35b, 35c) for every synthesis signal;

   a summation means (26) for summing synthesis signals provided with respective channel impulse responses present in the frequency domain; and

   a single frequency-domain time-domain conversion means (37) for converting the sum signal into the time domain to obtain the reproduction signal.
6. The audio reproduction system of claim 1, wherein the signal processing means (33) in the loudspeaker module has a digital amplifier.
7. The audio reproduction system of claim 4, wherein the central wave-field synthesis module is formed to transmit a first part of the channel impulse response sample by sample and a second part merely by using envelope support values, and
   wherein the rendering means (32) is formed to reconstruct the second part of the channel impulse response by using the supporting values.
8. The audio reproduction system of claim 7, wherein the rendering means (32) is formed to generate the second part of the channel impulse response by a noise generator or pseudo-noise generator, wherein noise values or pseudo noise values are weighted in amplitude with the support values and/or auxiliary values interpolated from the support values.
9. The audio reproduction system of one of the previous claims, wherein the audio tracks are standardized multi channel tracks and the audio source positions are standard positions relating to a positioning of reproduction loudspeakers in a reproduction room, wherein the number of standard positions is equal to the number of standardized multi channel tracks.
10. The audio reproduction system of claim 9, wherein the wave-field synthesis module is formed to calculate the virtual positions for calculating (25) the audio channel information from the standard position (22).
11. The audio reproduction system of claim 10, wherein the wave-field synthesis module is formed to place the virtual positions in infinity, so that the plurality of loudspeakers together emit plane sound waves.
12. The audio reproduction system of claim 10, wherein the wave-field synthesis module is formed to simulate virtual reproduction loudspeakers at defined virtual positions as point-shaped sound sources, which are so far away from the plurality of loudspeakers that an optimum reproduction region generally comprises the whole reproduction room.
13. The audio reproduction system of one of claims 9 to 12, wherein the audio tracks are part of a video or cinema film, wherein the wave-field synthesis module is formed to sample the audio tracks of the video or cinema film shifted by a time period prior to a video reproduction, wherein the time period is chosen to obtain a simultaneous reproduction of image and sound under consideration of a processing time in the wave-field synthesis module and the loudspeaker module.
14. The audio reproduction system of one of claims 1 to 13, wherein the audio signal for audio objects in a recording environment comprises as audio track an audio signal of the object as well as a position of the audio object in the recording environment, one or several characteristics of the audio objects, such as size or density and/or information about acoustic characteristics of a recording environment.
15. The audio reproduction system of claim 14, wherein the wave-field synthesis module is formed to determine the virtual positions from positions of the audio objects in the recording environment.
16. The audio reproduction system of one of the previous claims, wherein the wave-field synthesis module is formed to obtain information about acoustic characteristics of the reproduction room and consider them when determining the channel information, so that the sound waves reproduced by the plurality of loudspeakers are formed such that the acoustic influences of the reproduction room are reduced.
17. The audio reproduction system of one of the previous claims, wherein the wave-field synthesis module is formed to perform an adaptation to an acoustic of the reproduction room prior or during a reproduction of the audio signal, by
    calculating a plurality of room impulse response between the loudspeaker and microphones positioned in the reproduction room,
    interpolating an overall impulse response of the reproduction room from the plurality of room impulse responses, and
    considering the overall impulse response when calculating the channel information to reduce acoustic characteristics of the reproduction room.
18. The audio reproduction system of one of the previous claims, wherein the central wave-field synthesis module is formed to generate synchronization information and to embed it into data streams to the loudspeaker modules, and wherein the plurality of loudspeaker modules is formed to receive the synchronization information from the central wave-field synthesis module and to use it for synchronization, so that the loudspeaker modules are synchronized to the central wave-field synthesis module.
19. A method for reproducing an audio signal in a reproduction room, wherein a plurality of loudspeakers are disposed at defined loudspeaker positions, wherein the audio signal has a plurality of audio tracks, wherein a audio source position is associated to every audio track, comprising:

    centrally determining audio channel information for every audio channel from a virtual position to a loudspeaker position, wherein the virtual position depends on the audio source position associated to the audio track, so that audio channel information is present for every channel from every virtual position to every loudspeaker;

    centrally determining synthesis signals from the virtual positions for the loudspeakers;

    transmitting one or several synthesis signals as well as associated channel information to a plurality of loudspeaker modules;

    decentrally calculating a reproduction signal for the loudspeaker by using one or several synthesis signals and the associated channel information for a respective loudspeaker;

    performing signal processing by using a digital/analog conversion to generate an analog loudspeaker signal; and

    collectively retrieving the analog loudspeaker signals through the plurality of loudspeakers.
20. A computer program with a program code for performing the method of claim 19 when the program runs on a computer.

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