EP2754151B1 - Device, method and electro-acoustic system for prolonging a reverberation period - Google Patents

Description

The present invention relates to a device, a method and an electroacoustic system for reverberation time extension.

A room is not optimal for different applications from an acoustic point of view. So a musical performance usually requires some reverb to sound good. On the other hand, speakers are sometimes incomprehensible when the room is too reverberant. An adjustment of the reverberation time with the help of a public address system is therefore useful.

For example, in theaters, congress centers, planetariums, seminar rooms, multifunctional rooms, different acoustic conditions will be required for different occasions, and in particular, different reverberation time requirements will be required. To influence the reverberation time, electroacoustic systems for reverberation time extension can be used. Such systems can either be e.g. be retrofitted into an existing concert hall. Likewise, however, it may be useful to begin with the construction and construction of corresponding buildings and halls, e.g. in trade fair construction, to provide an electroacoustic system for reverberation time extension and to include it in the planning of the building. Also in the context of audio playback for entertainment purposes, a reverberation time extension may be desirable.

In the following, the technique of wave field synthesis is explained in more detail. Wave Field Synthesis (WFS) was researched at the TU Delft and first presented in the late 1980s ( Berkhout, AJ; de Vries, D .; Vogel, P .: Acoustic control by wave-field synthesis. JASA 93, 1993 ).

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

• Jeder Punkt, der von einer Welle erfasst wird, ist Ausgangspunkt einer Elementarwelle, die sich kugelförmig bzw. kreisförmig ausbreitet. Angewandt auf die Akustik kann durch eine große Anzahl von Lautsprechern, die nebeneinander angeordnet sind (einem so genannten Lautsprecherarray), jede beliebige Form einer einlaufenden Wellenfront nachgebildet werden. Im einfachsten Fall, einer einzelnen wiederzugebenden Punktquelle und einer linearen Anordnung der Lautsprecher, müssen die Audiosignale eines jeden Lautsprechers mit einer Zeitverzögerung und Amplitudenskalierung so gespeist werden, dass sich die abgestrahlten Klangfelder der einzelnen Lautsprecher richtig überlagern. Bei mehreren Schallquellen wird für jede Quelle der Beitrag zu jedem Lautsprecher getrennt berechnet und die resultierenden Signale addiert. In einem Raum mit reflektierenden Wänden können auch Reflexionen als zusätzliche Quellen über das Lautsprecherarray wiedergegeben werden. Der Aufwand bei der Berechnung hängt daher stark von der Anzahl der Schallquellen, den Reflexionseigenschaften des Aufnahmeraums und der Anzahl der Lautsprecher ab.

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

• Every point, which is detected by a wave, is the starting point of an elementary wave, which spreads in a spherical or circular manner. Applied to the acoustics can be simulated by a large number of speakers, which are arranged side by side (a so-called speaker array), any shape of an incoming wavefront. In the simplest case, a single point source to be reproduced and a linear arrangement of the speakers, the audio signals of each speaker must be fed with a time delay and amplitude scaling so that the radiated sound fields of each speaker properly overlap. With multiple sound sources, the contribution to each speaker is calculated separately for each source and the resulting signals added together. In a room with reflective walls, reflections can 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.

The technique of wave field synthesis (WFS) can thus achieve a good spatial sound for a large listener area. As stated above, wave field synthesis is based on the principle of Huygens, according to which wavefronts can be formed and built up by superimposing elementary waves. After mathematically exact theoretical description, infinitely many sources in infinitely small distance would have to be used for the generation of the elementary waves. Practically, however, many speakers are finally used at a finite distance from each other. Each of these speakers is driven according to the WFS principle with an audio signal from a virtual source having a particular delay and a certain level. Levels and delays are usually different for all speakers.

As stated above, a wave field synthesis system operates on the Huygens principle and reconstructs a given waveform, for example, of a virtual source located at a certain distance from a listener by a plurality of single waves. The wave-field synthesis algorithm thus obtains information about the actual position of a single loudspeaker from the loudspeaker array and then calculates a component signal for this single loudspeaker that this loudspeaker ultimately has to emit, so that the listener can superimpose the loudspeaker signal from one loudspeaker to the loudspeaker signals from the other active loudspeaker Speaker performs a reconstruction in that the listener has the impression that he is not "sonicated" by many individual speakers, but only from a single speaker at the position of the virtual source.

For multiple virtual sources in a wave-field synthesis setting, the contribution from each virtual source for each loudspeaker, that is, the component signal of the first virtual source for the first loudspeaker, the second virtual source for the first loudspeaker, etc., is calculated to then add up the component signals finally get the actual speaker signal. In the case of, for example, three virtual sources, superimposing the loudspeaker signals of all the active loudspeakers on the listener would mean that the listener does not feel that he is being sonicated by a large array of loudspeakers, but that the sound he hears is merely from three sound sources positioned at specific positions, which are equal to the virtual sources.

The calculation of the component signals usually takes place in practice by applying a delay and a scaling factor to the audio source assigned to a virtual source, depending on the position of the virtual source and the position of the loudspeaker, at a specific point in time in order to obtain a delayed and / or scaled audio signal to obtain a virtual source that directly represents the loudspeaker signal when there is only one virtual source, or that after addition of other component signals for the considered loudspeaker from other virtual sources then contributes to the loudspeaker signal for the considered loudspeaker.

Typical wave field synthesis algorithms operate regardless of how many speakers are present in the speaker array. The underlying theory of Wave Field Synthesis is that any sound field can be accurately reconstructed by an infinite number of individual speakers, with the individual individual speakers arranged infinitely close to each other. In practice, however, neither the infinite number nor the infinitely close arrangement will be realized. Instead, there are a limited number of speakers, which are also arranged at certain predetermined distances from each other. Thus, in real systems, only an approximation to the actual waveform that would occur if the virtual source were actually present, would be a real source.

Wave field synthesis devices are also capable of replicating several different types of sources. A prominent source form is the point source, where the level decreases proportionally 1 / r, where r is the distance between a listener and the position of the virtual source. Another source form is a source that emits plane waves. Here, the level remains constant regardless of the distance to the listener, since plane waves can be generated by point sources, which are arranged at an infinite distance.

• In US005109419A beschreibt Griesinger ein elektroakustisches System zur Nachhallzeitverlängerung in dem verschiedene Schallquellen über Mikrofon oder Direkteingang erfasst und über eine Reverbmatrix künstlich verhallt werden. Die Ausgangssignale dieses Systems werden an verteilte Lautsprecher gegeben und erzeugen so einen künstlichen Nachhall im Raum.

Following the above excursus on existing wave field synthesizers, we now turn to the prior art systems for reverberation time extension:

• In US005109419A Griesinger describes an electroacoustic system for reverberation time extension in which different sound sources are detected via microphone or direct input and artificially reverberated via a reverb matrix. The output signals of this system are given to distributed speakers and thus create an artificial reverberation in the room.

Likewise Poletti describes in "Reverberators for use in wide band assisted reverberation systems" US000000039189E an electroacoustic system for reverberation time extension based on the detection of spatial signals and processing them in a delay matrix which in turn drives a number of speakers.

In US005142586A Berkhout describes an approach in which a signal recorded in a room is folded and reproduced via a reconstructed wave field.

In G. Theile, H. Wittek and M. Reisinger: "Wave Field Synthesis, New Possibilities of Spatial Sound Recording and Reproduction", FKT Television and Cinema Technology, Fachverlag Schiele & Schon GmbH, Berlin, DE, Vol. 57, No. 4 , April 1, 2003, pages 735-739, ISSN: 1430-9947 , Wave Field Synthesis is referred to the problems of rendering stereophonic sound sources and integrating different signal and test configurations to give a comprehensive insight and overview of the sound recording and reproduction process.

In Marinus M. Boone: "Acoustic rendering with wave field synthesis", ACM SIGGRAPH AND EUROGRAPHICS CAMPFIRE: Acoustic Rendering for Virtual Environments, Snowbird, Utah, May 26-29, 2001; Pages 1 - 9 , an overview of wave field synthesis, its theory, and implementation is given as a laboratory demonstration system, with application examples and methods provided.

In the patent literature, there are other various systems for reverberation time extension such as. US 3614320 A and WO 2006092995 A1 ,

However, none of the systems enables flexible, dynamic adaptation to different and changing acoustic conditions and users' wishes for reverberation time extension.

The object of the present invention is therefore to provide improved concepts for devices, methods and electroacoustic systems for reverberation time extension. The object of the present invention is achieved by a device according to claim 1, a method according to claim 12, a computer program according to claim 13, an electroacoustic system according to claim 14 and a method according to claim 15.

The invention provides a device for reverberation time extension. The apparatus comprises a wave field synthesis information calculation module and a signal processor for generating a plurality of audio output signals for a plurality of loud speakers based on a plurality of audio input signals, and based on the wave field synthesis information, the audio signals being received from a plurality of microphones. Furthermore, the device comprises an operating unit for determining a virtual position of one or more virtual walls. The wave field synthesis information calculation module is configured to calculate the wave field synthesis information based on the virtual position of the one or more virtual walls. Furthermore, the virtual position can be set by the operating unit for at least one of the virtual walls.

By shifting the virtual walls to the outside, one thus achieves an acoustic spatial enlargement. The acoustic amplification is also achieved by a regenerative effect, which consists in that the output via speakers generated audio output signals are picked up again by the microphones and thus enter into the generation of the audio output signals at a later date.

Thus, an apparatus and method for generating an acoustic spatial magnification is provided, wherein distributed microphones detect relevant sound sources and the acoustic environment and reproduce this with respect to fixed or dynamic virtual source positions via a wave field synthesis system.

The invention is based on the concept that the virtual sources are generated in an algorithm based on wave field synthesis. In this case, the invention describes a method in which by means of distributed microphones in the room to be sounded, the acoustics of the room is detected with the sources to be amplified and a AD converter Processing system is supplied. The processing system may consist of software in which the signal is first processed via filters, and then processed in a wave field synthesis algorithm to an object-based sound source, which in turn is processed via filters, to then be played over a wave field synthesis system. Due to the possibilities of wave field synthesis, the acquired room signals can now be positioned as desired and can be moved as "virtual walls" as desired. As a result, individual room geometries can be generated. The recorded space signals are typically represented as plane waves and thus correspond to the acoustic effect of a wall. Not only can this virtual wall be moved, it can also be changed in angle, directly affecting the reflection patterns of the sound sources.

In one embodiment, the wave field synthesis information calculation module is configured to calculate delay values and amplitude factor values as wave field synthesis information. The delay value indicates the delay by which one of the audio input signals is delayed at one of the speakers. The amplitude factor value indicates by what factor the amplitude of one of the audio input signals is modified to obtain a modified signal output at one of the speakers.

Further, the wave field synthesis information calculation module may be configured to calculate a delay value and an amplitude factor value for each speaker-virtual wall pair at a time, wherein a speaker-virtual wall pair, a pair of one of the speakers and one of the virtual walls is.

In another embodiment, the wave field synthesis information calculation module is configured to apply the delay value and the amplitude factor value to a speaker-virtual wall pair based on the distance from the speaker and the virtual wall of the speaker-virtual wall pair to calculate.

Further, the wave field synthesis information calculation module may be configured to set the delay value of a speaker virtual wall pair the larger the distance between the speaker and the virtual wall.

Further, the wave field synthesis information calculation module may be configured to set the amplitude factor value of a speaker-virtual wall pair the smaller the distance between the speaker and the virtual wall.

In a further embodiment, the operating unit is designed so that at least one of the virtual walls is displaceable from a first virtual position to a second virtual position, so that the virtual wall can be displaced in any desired parallel position relative to its first position. Furthermore, the operating unit can be designed such that at least one of the virtual walls is displaceable from a first virtual position to a second virtual position, so that the virtual wall can be displaced in any rotationally opposite its first position.

In a further embodiment, the operating unit is designed so that the virtual position can be set by the operating unit for all of the virtual walls. The operating unit can be designed such that each of the virtual walls can be displaced from a first virtual position to a second virtual position. so that each virtual wall is arbitrarily parallel and rotatable relative to its first position.

In another embodiment, the reverberation time extension apparatus may include a parametric filter for filtering resonant frequencies.

Further, there is provided an electroacoustic reverberation time extension system comprising a plurality of microphones, a reverberation time extension apparatus according to any one of the above-described embodiments, and a loudspeaker array of a plurality of loud speakers. The plurality of microphones are configured to generate a plurality of audio input signals fed to the reverberation time extension apparatus, and wherein the plurality of loudspeakers of the loudspeaker array are adapted to receive and feed the audio output signals from the reverberation time extension apparatus Play audio output signals.

Fig. 1

stellt ein Blockschaltbild einer Vorrichtung zur Nachhallzeitverlängerung gemäß einem Ausführungsbeispiel dar,

Fig. 2

stellt ein Blockschaltbild dar, das das Zusammenwirken eines Moduls zur Berechnung von Wellenfeldsyntheseinformation und eines Signalprozessors zeigt,

Fig. 3

stellt ein elektroakustisches WFS System zur Nachhallzeitverlängerung gemäß einem Ausführungsbeispiel dar,

Fig. 4

illustriert ein weiteres Ausführungsbeispiel eines elektroakustischen WFS Systems,

Fig. 5

stellt einen mittleren Konferenzraum (5mx18x15m) dar, der mit 5 Deckenmikrofonen, 40 Deckenlautsprechern und einem umlaufenden horizontalen Band aus konventionellen Lautsprechern in einer reduzierten WFS Anordnung gemäß einer Ausführungsform ausgestattet ist,

Fig. 6

zeigt eine Anordnung von Lautsprechern, virtuellen Wänden und Mikrofonen gemäß einem Ausführungsbeispiel, und

Fig. 7

zeigt eine Anordnung von Lautsprechern und einer virtuellen Wand gemäß einem weiteren Ausführungsbeispiel.

Preferred embodiments of the invention will be explained below with reference to the accompanying drawings.

Fig. 1

FIG. 12 is a block diagram of a reverberation time extension apparatus according to an embodiment; FIG.

Fig. 2

FIG. 12 is a block diagram showing the interaction of a wave field synthesis information calculation module and a signal processor; FIG.

Fig. 3

illustrates an electroacoustic WFS system for reverberation time extension according to an embodiment,

Fig. 4

illustrates another embodiment of an electroacoustic WFS system,

Fig. 5

presents a medium conference room (5mx18x15m) equipped with 5 overhead microphones, 40 ceiling loudspeakers and a rotating horizontal band of conventional loudspeakers in a reduced WFS arrangement according to one embodiment,

Fig. 6

shows an arrangement of speakers, virtual walls and microphones according to an embodiment, and

Fig. 7

shows an arrangement of speakers and a virtual wall according to another embodiment.

Fig. 1 shows a device for reverberation time extension according to one embodiment. The apparatus includes a wave field synthesis information calculation module 110. Further, the apparatus includes a signal processor 120 for generating a plurality of audio output signals y ₁ , y ₂ ,..., Y _n for a plurality of loud speakers (not shown) based on a plurality of audio input signals s ₁ , s ₂ , ..., s _n picked up by a plurality of microphones (not shown) and based on the wave field synthesis information. Furthermore, the device comprises an operating unit 130 for determining a virtual position of one or more virtual walls. The wave field synthesis information calculation module 110 is configured to calculate the wave field synthesis information WS _inf based on the virtual position of the one or more virtual walls. In this case, its virtual position can be set by the operating unit for at least one of the virtual walls. In one embodiment, the wave field synthesis information calculation module 110 and the signal processor 120 may be implemented in one module (a wave field synthesis module).

Subsequently, reference will be made to Fig. 2 an embodiment of the interaction of the module for calculating wave field synthesis information and the signal processor set forth. In Fig. 2 For example, the wave field synthesis information calculation module 210 and the signal processor 220 are shown by dashed lines.

The wave field synthesis information calculation module 210 and the signal processor 220 have a highly parallel construction in that, starting from the audio signal supplied to the signal processor for each virtual wall (a virtual source) and from the position information for the corresponding virtual wall (virtual source), which has received the module for calculating wave field synthesis information 210 from an operation unit, first delay information (delay information) V _i and amplitude factors (scaling factors) SF _i are calculated by the position information and the position of the currently considered loudspeaker, z. B. the speaker with the ordinal number j, so LS _j depend. The calculation of a delay information V _i and a scaling factor SF _i on the basis of the position information of a virtual source (virtual wall) and the position of the considered loudspeaker j is done by known algorithms implemented in devices 300, 302, 304, 306.

On the basis of the delay information V _i (t) and SF _i (t) and on the basis of the assigned for each virtual source audio signal AS _i (t) is for a current time t _A a discrete value AW _i (t _A) for the _Computed signal K _ij calculated in a final received speaker signal. This is done by means 310, 312, 314, 316, as they are in Fig. 2 are shown schematically. Fig. 2 also shows a kind of "flash photography" at time t _A for the individual component signals. The individual component signals are then summed by a summer 320 in nodes to determine the discrete value for the current time t _{A of} the loudspeaker signal for the speaker j, which can then be supplied to the speaker for the output.

Like it out Fig. 2 1, a value valid on the basis of a delay information and a scale factor (scaling factor) at a current time is first calculated for each virtual source, after which all the component signals for a loudspeaker due to the different virtual walls (virtual sources ) are summed up. For example, if only one virtual source were present, then the summer would be omitted, and that at the output of the summer in Fig. 2 applied signal would z. B. the signal output from the device 310 when the virtual source 1 is the only virtual source.

It should be noted at this point that the value of a loudspeaker signal which is a superimposition of the component signals for this loudspeaker due to the various virtual sources 1, 2, 3,..., N is obtained at the respective output. One - such arrangement would in principle be provided for each speaker, unless what is preferred for practical reasons, always z. B. 2, 4 or 8 speakers are driven together with the same speaker signal.

Fig. 3 illustrates an electroacoustic WFS system for reverberation time extension according to an embodiment.

According to the embodiment of Fig. 3 are installed in a room evenly four microphones 350 1m suspended from the ceiling. The microphone signals are processed in a WFS algorithm 360 to a virtual source, which is reproduced as a plane wave via a WFS sound system 370 in the same room. The WFS system includes a user interface for moving the 4 microphone sources. With this control unit, the 4 microphone signals are detected and pulled outwards. The result is an acoustic enlargement of the room.

Thus, arbitrarily large rooms can now be created. The positioning of the virtual walls changes the reverberation time of the room in relation to the position and arrangement (angle) of the walls.

In one embodiment, the wave field synthesis module 360 includes a wave field synthesis information calculation module according to the embodiment of FIG Fig. 1 and a signal processor according to the embodiment of FIG Fig. 1 ,

In one embodiment, the operating unit 375 is an operating unit according to the embodiment of FIG Fig. 1 ,

In one embodiment, unit 355 sets in Fig. 3 a filter which serves to filter resonance frequencies. A sound wave output at one of the speakers 370 is resumed by the microphones 350 and re-considered in the generation of the later audio signal output via the speakers. In order to avoid unwanted resonances occurring during this process, filter 355 can be used to suppress these resonances.

In one embodiment, filter 365 may be a conventional filter used, for example, to adapt the speakers.

Fig. 4 illustrates another embodiment of an electroacoustic WFS system. The signals of the ceiling microphones are processed in a central processing unit and processed after filtering in a matrix to virtual sources, which is played back after level adjustment, control of the spatial proportion and speaker filtering as virtual sound sources via a WFS array and evenly distributed ceiling speakers.

In Fig. 4 Microphones feed 411, 412 audio input signals into Microphone Preamplifier 416, 417. The microphone 411 is a microphone that is close to a sound source such as a lectern. The microphone 412 is a room microphone located in the room but farther from the sound source than the microphone 411. Typically, multiple room microphones and / or multiple microphones are used close to the sound source.

Microphone preamplifiers 416, 417 amplify the audio input signals received from the microphones 411, 412 to obtain preamplified audio input signals. The microphone preamplifiers 416, 417 may be conventional microphone preamplifiers. The pre-amplified audio input signals are fed to an analog-to-digital converter 420, which converts the audio input signals, which are initially in analog form, into digital audio signals. The analog-to-digital converter 420 may be a conventional analog-to-digital converter.

The analog-to-digital converter 420 feeds the digital audio signals into the absorption filter 425. Absorption filter 425 performs filtering to match the wall material. In one embodiment, absorption filter 425 filters such that when highly reflective walls are to be replicated, the digital audio signals pass absorption filter 425 almost unfiltered. On the other hand, if strongly attenuating walls are to be simulated, absorption filter 425 in one embodiment filters the digital audio signals to a great extent.

Filter 430 is a filter for feedback compensation and sound adjustment. If a signal is reproduced by a loudspeaker, the sound waves of this signal are in turn detected by the microphone and this leads to a feedback. In one embodiment, filter 430 may be used to provide this feedback entirely or partially compensate. In addition, Filter 430 can be used for sound adjustment. In one embodiment, the feedback compensation and / or the sound adjustment may be performed in a conventional manner.

Furthermore, the system includes in Fig. 4 a central operating unit 435 and a module for calculating wave field synthesis information 440. In this case, the central operating unit 435 of the operating unit in Fig. 1 correspond. The central operating unit in Fig. 4 can be equipped with a GUI (Graphical User Interface). The wave field synthesis information calculation module 440 may be adapted to the wave field synthesis information calculation module Fig. 1 correspond.

The wave field synthesis information calculation module 440 passes the calculated wave field synthesis parameters to module 445. These wave field synthesis parameters may be e.g. to act on delay values and amplitude values, such as amplitude factor values.

Module 445 builds a Delay Amplitude Matrix from the values passed by Module 440. For example, in one embodiment, the delay amplitude matrix may include a delay value and an amplitude factor value for a particular point in time for each speaker-virtual wall pair.

Module 445 performs audio scaling based on the wave field synthesis parameters obtained from the wave field synthesis information calculation module 440. For example, if a delay value and an amplitude factor value were obtained for a loudspeaker-virtual wall pair, the signal originating from the virtual wall (eg, apparently reflected from the virtual wall) will be the received delay value is delayed, and the amplitude factor value obtained from module 440 is modified to the amplitude of the signal to be output by the amplitude factor value, for example by multiplying the amplitude factor value by the amplitude of the signal to be output.

Subsequently, filter 450 filters the module 445 modified audio signals to achieve speaker matching. In a master gain module 455, the audio signals are modified to adjust the overall volume. This can be done in the usual way. In a gain-space-share module 460, the ratio of volume proportion to original signal is adjusted. For example, in one embodiment, the ratio of audio signals generated from audio signals of room microphones were adjusted to audio signals that were generated from audio signals of microphones near the lectern, for example by adjusting the amplitudes of the respective signals.

The modified digital audio signals are then fed to a digital-to-analog converter 465 which converts the modified digital audio signals to analog audio output signals. The analog audio output signals are then amplified by power amplifiers 471, 472 and output from loudspeakers 481, 482. In the embodiment of Fig. 4 For example, the audio signals are output from either WFS speakers 481 or ceiling speakers 482. It is understood that in a real system a variety of WFS speakers and / or ceiling speakers can be used.

Fig. 5 shows a medium conference room (5mx18x15m) equipped with 5 ceiling microphones 511, 512, 513, 514, 515, 40 ceiling loudspeakers and a circulating horizontal band of conventional loudspeakers in a reduced WFS arrangement 530. The signals of the ceiling microphones 511, 512, 513, 514, 515 are processed in a central processing unit and processed after filtering in a matrix to virtual sources, which after level adjustment, control of the spatial proportion and speaker filtering as virtual sound sources 521, 522, 523, 524, 525 via a WFS array and evenly distributed ceiling speakers is played again. The structure shows FIG. 4 , The microphone signals are represented by the respectively opposite virtual sources in order to avoid feedback. Through the use of a flexible matrix and the ability to represent any inputs (microphone inputs / line in) as virtual sound sources 521, 522, 523, 524, 525, directly microfoned signals are taken into the room simulation and due to their positioning also for visualization in an artificial reverberated room used. However, these signals must be considered as little regenerative, since they contain little space. Additional microphones can be added to capture and reproduce a complex distribution of reflections. Likewise, the representation of virtual sound sources 521, 522, 523, 524, 525 on the ceiling is possible, which is to be regarded as an essential quality requirement in the representation of a real room. In addition to a filter unit with narrow-band filters for feedback suppression, the input branch of the matrix also contains a filter unit which takes into account different spatial materials in order to incorporate various absorption and reflection parameters into the space to be reverberated. The detected microphone signals are reproduced in freely positionable sources as described imaged and acted upon by the existing room characteristics again captured by the microphone, which leads to a regeneration of the room acoustics.

Fig. 6 illustrates a basic concept of certain embodiments. Shown is a speaker array that, as in Fig. 6 12 speakers 611, 612, 613. In actual embodiments, the number of loudspeakers will often be significantly larger and include, for example, 60, 100, 200, 300 or more loudspeakers. Also shown are four virtual walls 621, 622, 623, 624.

In the following, the speaker 611 and the virtual wall 621 will be considered in more detail. These form a speaker-virtual wall pair (611, 621). Also, any other combination of one of the speakers and one of the virtual walls forms a speaker-virtual wall pair. The distance between the speaker and the virtual wall is indicated by an arrow d. In Fig. 6 In addition, a plurality of microphones 631, 632, 633 is provided. For the sake of simplicity, it is assumed that a microphone 631 generates an audio signal to be reproduced through the speaker 611 by recording sound waves. In this case, the signal reproduced by the loudspeaker 611 should correspond to a reflection of the sound waves recorded by the microphone 631 on the virtual wall 621. This means that the signal picked up by the microphone can only be reproduced with a time delay by the loudspeaker 611, which depends on the distance between the loudspeaker and the virtual wall: the greater the distance between the virtual wall 621 and the loudspeaker 611, the greater the temporal Delay, ie the delay value with which the signal picked up by the microphone 631 is to be reproduced on the loudspeaker 611. Fig. 6 shows by the dashed line 629 a displacement of the virtual wall 621, wherein the distance of the virtual wall of the speaker 611 increases from d to 2d. The delay value will increase accordingly.

berechnen, wobei d der Abstand zwischen dem Lautsprecher und der virtuellen Wand des Lautsprecher-virtuelle Wand-Paares ist, c ein konstanter Wert ist, und p₁ eine Proportionalitätskonstante größer 0 ist. Der Delay-Wert wird somit umso größer, je größer der Abstand zwischen Lautsprecher und virtueller Wand ist.In a specific embodiment, one may use the delay value according to the formula: $$\mathrm{delay}=\left(\mathrm{d}+\mathrm{c}\right)*{\mathrm{<figure-callout\; id="1"\; label="p"\; filenames="imgf0001.png,imgf0002.png"\; state="\{\{state\}\}">p</figure-callout>}}_{\mathrm{1}}$$

where d is the distance between the speaker and the virtual wall of the speaker-virtual wall pair, c is a constant value, and p _{1 is} a proportionality constant greater than 0. The greater the distance between the speaker and the virtual wall, the greater the delay value becomes.

The greater the distance between the virtual wall and the loudspeaker, the smaller the amplitude factor is to choose in one embodiment, since the amplitude of a real sound source becomes smaller, the farther one is from a sound source, the virtual sound source here represents a virtual wall that seems to reflect a sound wave. The amplitude factor is the factor with which the amplitude of one of the output signals is to be modified in order to obtain a modified signal to be output at one of the loudspeakers.

berechnen, wobei d der Abstand zwischen dem Lautsprecher und der virtuellen Wand des Lautsprecher-virtuelle Wand-Paares ist, h ein konstanter Wert ist, und p₂ eine Proportionalitätskonstante größer 0 ist. In bevorzugten Ausführungsformen wird die Proportionalitätskonstante p₂ und so gewählt, dass der Amplitudenfaktor immer einen Wert größer 0 und kleiner 1 annimmt.In a specific embodiment, the amplitude factor can be calculated according to the formula: $$\mathrm{amplitudes}-\mathrm{factor}=\left[\mathrm{1}/\left(\mathrm{d}+\mathrm{H}\right)\right]*{\mathrm{p}}_{\mathrm{2}}$$

where d is the distance between the speaker and the virtual wall of the speaker-virtual wall pair, h is a constant value, and p _{2 is} a proportionality constant greater than 0. In preferred embodiments, the proportionality constant p ₂ and is chosen so that the amplitude factor always assumes a value greater than 0 and less than 1.

In principle, an increase in the delay value can bring about a reverberation time extension.

Fig. 7 FIG. 12 shows another embodiment in which the current position 729 of the virtual wall is changed so that the current position 729 of the virtual wall has been rotatably changed from its old position 721. The distance of the old position of the virtual wall of speaker 711 is indicated by arrow e, the distance of the new position of the virtual wall from the speaker 711 is shown by arrow f.

Although some aspects have been described in the context of a device, it is apparent that these aspects also constitute a description of the corresponding method, wherein a block or device corresponds to a method step or feature of a method step. Similarly, aspects described in the context of a method step also represent a description of a corresponding block or element or feature of a corresponding device.

A computer program or signal according to the invention can be stored on a digital storage medium or can be transmitted on a transmission medium be such as a wireless transmission medium or a wired transmission medium, such as the Internet.

Depending on certain implementation requirements, embodiments of the invention may be implemented in hardware or in software. The implementation may be done using a digital storage medium, such as a digital storage medium. a floppy disk, a DVD, a CD, a ROM, a PROM, an EPROM, an EEPROM, or a FLASH memory which stores electronically readable control signals that cooperate (or are able to work together) with a programmable computer system so that the appropriate procedure is carried out.

Some embodiments according to the invention comprise a non-transitory data carrier having electronically readable control signals capable of cooperating with a programmable computer system to perform one of the methods described herein.

In general, embodiments of the present invention may be implemented as a computer program product having program code, wherein the program code is operative to perform one of the methods when the computer program product is executed on a computer. The program code may e.g. be stored on a machine-readable carrier.

Other embodiments include the computer program for executing one of the methods described herein stored on a machine readable carrier.

In other words, an embodiment of the method according to the invention is therefore a computer program with a program code for carrying out one of the methods described herein when the computer program is executed on a computer.

A further embodiment of the inventive method is therefore a data carrier (or a digital storage medium or a computer-readable medium) recorded thereon the computer program for carrying out one of the methods described herein.

A further embodiment of the method according to the invention is therefore a data stream or a series of signals representing the computer program for carrying out one of the methods described herein. The data stream or the Signal series can be configured, for example, to be transmitted via a data communication connection, eg via the Internet.

Another embodiment comprises processing means, e.g. a computer, or programmable logic device configured or adapted to perform one of the methods described herein.

Another embodiment includes a computer on which the computer program is installed to perform one of the methods described herein.

In some embodiments, a programmable logic device (e.g., a field programmable gate array) may be used to perform some or all of the functionality of the methods described herein. In some embodiments, a field programmable gate array may cooperate with a microprocessor to perform any of the methods described herein. In general, the methods are preferably performed by any hardware device.

The embodiments described above are merely illustrative of the principles of the present invention. It will be understood that modifications and variations of the arrangements and details described herein will be apparent to others of ordinary skill in the art. Therefore, it is intended that the invention be limited only by the scope of the appended claims and not by the specific details presented in the description and explanation of the embodiments herein.

Claims (15)

1. Apparatus for reverberation time extension, comprising:

   a module (110) for calculating wave field synthesis information,

   a signal processor (120) for generating a plurality of audio output signals for a plurality of loudspeakers based on a plurality of audio input signals that have been recorded by a plurality of microphones, and based on the wave field synthesis information, and

   an operating unit (130) for determining a virtual position of one or several virtual walls,

   wherein the module (110) for calculating wave field synthesis information is implemented to calculate the wave field synthesis information based on the virtual position of the one or several virtual walls, and

   wherein, for at least one of the virtual walls, the virtual position is adjustable by the operating unit (130).
2. Apparatus according to claim 1, wherein the module (110) for calculating wave field synthesis information is implemented to calculate delay values and amplitude factor values as wave field synthesis information, wherein the delay value indicates the delay by which one of the audio input signals is reproduced at one of the loudspeakers in a delayed manner, and wherein the amplitude factor indicates by what factor the amplitude of one of the audio input signals is modified to acquire a modified signal which is output at one of the loudspeakers.
3. Apparatus according to claim 2, wherein the module (110) for calculating wave field synthesis information is implemented to calculate a delay value and an amplitude factor for each loudspeaker/virtual wall pair for a specific time, wherein a loudspeaker/virtual wall pair is a pair of one of the loudspeakers and one of the virtual walls.
4. Apparatus according to claim 3, wherein the module (110) for calculating wave field synthesis information is implemented to calculate the delay value and the amplitude factor value for a loudspeaker/virtual wall pair based on the distance of the loudspeaker and the virtual wall of the loudspeaker/virtual wall pair.
5. Apparatus according to claim 3 or 4, wherein the module (110) for calculating wave field synthesis information is implemented to set the delay value of a loudspeaker/virtual wall pair the greater the greater the distance between the loudspeaker and the virtual wall is.
6. Apparatus according to one of claims 3 to 5, wherein the module (110) for calculating wave field synthesis information is implemented to set the amplitude value of a loudspeaker/virtual wall pair the smaller the greater the distance between the loudspeaker and the virtual wall is.
7. Apparatus according to any one of the preceding claims, wherein the operating unit (130) is implemented such that at least one of the virtual walls can be shifted from a first virtual position to a second virtual position, such that the virtual wall can be shifted arbitrarily in parallel to its first position.
8. Apparatus according to any one of the preceding claims, wherein the operating unit (130) is implemented such that at least one of the virtual walls can be shifted from a first virtual position to a second virtual position, such that the virtual wall can be shifted arbitrarily in a rotatable manner with respect to this first position.
9. Apparatus according to any one of the preceding claims, wherein the virtual position for all of the virtual walls is adjustable by the operating unit (130).
10. Apparatus according to any one of the preceding claims, wherein the operating unit (130) is implemented such that each of the virtual walls can be shifted from a first virtual position to a second virtual position, such that each virtual wall can be shifted arbitrarily in parallel and in a rotatable manner with respect to its first position.
11. Apparatus according to any one of the preceding claims, wherein the apparatus further comprises a parametric filter for filtering resonance frequencies.
12. Method for reverberation time extension, comprising:

    determining a virtual position of one or several virtual walls;

    receiving a plurality of audio input signals that have been recorded by a plurality of microphones,

    calculating wave field synthesis information, and

    generating a plurality of audio output signals for a plurality of loudspeakers based on the audio input signals and based on the wave field synthesis information,

    wherein the wave field synthesis information is calculated based on the virtual position of the one or several virtual walls, and

    wherein the virtual position is adjustable for at least one of the virtual walls.
13. Computer program including a program code for performing the method according to claim 12, when the computer program runs on a computer.
14. Electroacoustic system for reverberation time extension, comprising:

    a plurality of microphones (350; 411, 412),

    an apparatus for reverberation time extension according to one of claims 1 to 11, and

    a loudspeaker array comprising a plurality of loudspeakers (370; 481, 482),

    wherein the plurality of microphones (350; 411, 412) is implemented to generate a plurality of audio input signals fed into the apparatus for reverberation time extension, and wherein the plurality of loudspeakers (370; 481, 482) of the loudspeaker array is implemented to have the audio output signals fed in by the apparatus for reverberation time extension and to reproduce the fed-in audio output signals.
15. Method for reverberation time extension by means of an electroacoustic system, comprising:

    recording a plurality of audio input signals by a plurality of microphones,

    performing the method for reverberation time extension according to claim 12 for generating a plurality of audio output signals, wherein receiving the plurality of audio input signals comprises that that plurality of audio input signals that have been recorded by the plurality of microphones is received, and

    outputting the plurality of audio output signals by means of a loudspeaker array comprising a plurality of loudspeakers.

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