EP2754151A1

EP2754151A1 - Device, method and electro-acoustic system for prolonging the reverberation period

Info

Publication number: EP2754151A1
Application number: EP12756411.0A
Authority: EP
Inventors: René RODIGAST
Original assignee: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Current assignee: Fraunhofer Gesellschaft zur Forderung der Angewandten Forschung eV
Priority date: 2011-09-07
Filing date: 2012-08-23
Publication date: 2014-07-16
Anticipated expiration: 2032-08-23
Also published as: EP2754151B1; US9355632B2; JP5995973B2; WO2013034444A1; CN103907151A; US20140185817A1; DE102011082310A1; EP2754151B2; JP2014529251A; CN103907151B

Abstract

A device for prolonging the reverberation period is provided. The device comprises a module (110) for calculating wave field synthesis information, and a signal processor (120) for generating a plurality of audio output signals for a plurality of loudspeakers based on the audio input signals received by a plurality of microphones, and based on the wave field synthesis information. The device also comprises an operating unit (130) for determining a virtual position of one or more virtual walls. The module (110) for calculating wave field information is designed to calculate the wave field synthesis information based on the virtual position of the one or more virtual walls. Further, the virtual position can be adjusted by the operating unit (130) for at least one of the virtual walls.

Description

Apparatus, method and electroacoustic system for

Reverberation time extension

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. The basic idea of WFS is based on the application of the 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 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 represents the speaker signal immediately if only one virtual source is present, or that contributes to addition of other component signals for the considered speaker from other virtual sources then to the speaker signal for the considered speaker.

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. 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 various sound sources are detected via microphone or direct input and artificially reverberated via a Reverbmatrix. The output signals of this system are given to distributed speakers and thus create an artificial reverberation in the room. Similarly, 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 loudspeakers. "US005142586A describes an approach by Berkhout a signal recorded in a room is folded and reproduced via a reconstructed wave field.

In the patent literature, there are other various systems for reverberation time extension such as, for example, US 3614320 A and WO 2006092995 AI. 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 detected room signals can now be positioned as desired and can be moved as "virtual walls", thus creating individual room geometries The recorded room signals are typically represented as plane waves and thus correspond to the acoustic effect of a wall Virtual wall can not only be moved, but also be changed in your angle and thus directly influences 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 on 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 a speaker the virtual walls is.

In another embodiment, the wave field synthesis information calculation module is configured to provide the delay value and the amplitude factor value for 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 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 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 so 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 a further embodiment, the reverberation time extension apparatus may comprise 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.

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

1 shows a block diagram of a device for reverberation time extension according to an embodiment, FIG. 10 is a block diagram showing the interaction of a wave field synthesis information calculation module and a signal processor; FIG. 3 illustrates an electroacoustic WFS system for reverberation time extension, illustrating another embodiment of an electroacoustic WFS system, illustrating a middle conference room (5mxl 8xl 5m) Featuring 5 ceiling microphones, 40 ceiling loudspeakers, and a recirculating horizontal band of conventional loudspeakers in a reduced WFS arrangement according to one embodiment, the invention features an array of loudspeakers, virtual walls, and microphones according to one embodiment, and shows an array of loudspeakers and a virtual loudspeaker Wall according to another embodiment.

1 shows a device for reverberation time extension according to an embodiment. The apparatus comprises a wave field synthesis information calculation module 1 10. The apparatus further comprises a signal processor 120 for generating a plurality of audio output signals y _ls y ₂ , y _n for a plurality of loud speakers (not shown) based on a plurality of audio input signals si, 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 10 is configured to calculate the wave field synthesis information WS _mf 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). Referring now to Figure 2, an embodiment of the interaction of the wave field synthesis information calculation module and the signal processor is set forth. In Fig. 2, 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 wave field synthesis information calculation module 210 from an operation unit, first, delay information (delay information) Vi and amplitude factors (scaling factors) SF; calculated from the position information and the position of the currently considered loudspeaker, e.g. B. the speaker with the ordinal number j, so LS _j depend. The calculation of a delay information Vj and a scaling factor SF; due to the position information of a virtual source (virtual wall) and the location of the considered loudspeaker j is done by known algorithms implemented in devices 300, 302, 304, 306. On the basis of the delay information Vi (t) and SFj (t) and on the basis of the individual virtual source associated audio signal ASj (t) is for a current time t _A, a discrete value AWi (t _A ) for the component signal K _y in an ultimately obtained loudspeaker signal. This is done by means 310, 312, 314, 316, as shown schematically in FIG. Fig. 2 also shows, as it were, a "flash shot" at time t _A for the individual component signals The individual component signals are then summed by a summer 320 at nodes to determine the discrete value for the current time t _A of the loudspeaker signal loudspeaker j which can then be fed to the speaker for output.

As can be seen in FIG. 2, first, for each virtual source, a value valid on the basis of a delay information (delay delay) and a scaling with an amplitude factor (scaling factor) at a current instant is calculated, after which all the component signals for a loudspeaker due to the various virtual walls (virtual sources) are summed up. For example, if only one virtual source were present, the summer would be omitted and the signal applied to the output of the summer in FIG. 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. Such an arrangement would be provided in principle for each speaker, unless that 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 exemplary embodiment.

According to the embodiment of FIG. 3, four microphones 350 Im are installed uniformly suspended in a room 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 comprises a wave field synthesis information calculation module according to the embodiment of FIG. 1 and a signal processor according to the embodiment of FIG. 1. In one embodiment, the operation unit 375 is an operation unit according to the embodiment of FIG.

In one embodiment, unit 355 in FIG. 3 represents a filter that serves to filter resonant 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. 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 41 1, 412 input audio input signals to microphone preamplifiers 416, 417. The microphone 41 1 is a microphone which is close to a sound source, e.g. located at a lectern. The microphone 412 is a room microphone located in the room but farther from the sound source than the microphone 41 1. 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 pre-amplified audio input signals. The microphone preamplifiers 416, 417 may be conventional microphone preamplifiers. The preamplified 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 in FIG. 4 comprises a central operating unit 435 and a module for calculating wave field synthesis information 440. The central operating unit 435 may correspond to the operating unit in FIG. 1. 4 may be provided with a GUI ("Graphical User Interface") The wave field synthesis information calculation module 440 may correspond to the wave field synthesis information calculation module of FIG.

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 handle 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 wavelet field sync 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 by 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. In one embodiment, for example, 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, the audio signals are output from either WFS-10 loudspeakers 481 or ceiling loudspeakers 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 (5mxl 8xl5m), with 5 ceiling microphones

15 51 1, 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 space proportion and 0 speaker filtering as virtual sound sources 521, 522, 523 , 524, 525 is replayed via a WFS array and evenly distributed ceiling speakers. The structure is shown in 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 5 virtual sound sources 521, 522, 523, 524, 525, also directly microphones are taken into the room simulation and as a result of their positioning also for display in one artificially 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 account of 5 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 comprising, as shown in FIG. 6, 12 speakers 611, 612, 613. In actual embodiments, the number of loudspeakers will often be significantly larger, e.g. 60, 100, 200, 300 or more speakers. Also shown are four virtual walls 621, 622, 623, 624. In the following, the speaker 61 1 and the virtual wall 621 will be considered in greater detail. These form a speaker-virtual wall pair (61 1, 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, a plurality of microphones 631, 632, 633 are also provided. For the sake of simplicity, it is assumed that a microphone 631 generates an audio signal by recording sound waves to be reproduced through the speaker 61 1. In this case, the signal reproduced by the loudspeaker 61 1 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 may 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 virtual wall 621 and loudspeaker 61 1, the larger the time delay, ie the delay value with which the signal recorded by the microphone 631 is to be reproduced on the loudspeaker 61 1. FIG. 6 shows, by the broken line 629, a displacement of the virtual wall 621, wherein the distance of the virtual wall from the loudspeaker 61 1 increases from d to 2 d. The delay value will increase accordingly.

In a specific embodiment, one may use the delay value according to the formula:

Calculate delay = (d + c) * pi, where d is the distance between the loudspeaker and the virtual wall of the speaker-virtual wall pair, c is a constant value, and pi 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 larger the distance between the virtual wall and the loudspeaker, the smaller is the amplitude factor to choose in an embodiment, since the amplitude of a real sound source becomes smaller the farther away from a sound source, the virtual sound source here being the 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.

In a specific embodiment, the amplitude factor can be calculated according to the formula:

Amplitude factor = [1 / (d + h)] * p 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 proportional constant greater 0 is. 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 shows another embodiment in which the current position 729 of the virtual wall is changed such 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 71 1 is indicated by arrow e, the distance of the new position of the virtual wall from the speaker 71 1 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 exemplary embodiment of the method according to the invention is therefore a data carrier (or a digital storage medium or a computer-readable medium) which has 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

claims

A reverberation time extension apparatus comprising: a wave field synthesis information calculating module (1 10); 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 picked up by a plurality of microphones and based on the wave field synthesis information, and a manipulation unit (130) for determining a virtual position of one or more virtual walls, wherein the wave field synthesis information calculation module (110) is adapted for generating the wave field synthesis information based on the virtual position of the one or more virtual walls, and wherein for at least one of the virtual walls, the virtual position is adjustable by the operating unit (130).

The apparatus of claim 1, wherein the wave field synthesis information calculation module (110) is adapted to calculate delay values and amplitude factor values as wave field synthesis information, the delay value indicating the delay to delay one of the audio input signals at one the loudspeaker is reproduced, and wherein the amplitude factor 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 loudspeakers.

The apparatus of claim 2, wherein the wave field synthesis information calculation module (110) is adapted to calculate a delay value and an amplitude factor value for each loudspeaker-virtual wall pair for a time, one loudspeaker virtual Wall pair, a pair of one of the speakers and one of the virtual walls is.

The apparatus of claim 3, wherein the wave field synthesis information calculation module (110) is adapted to calculate the delay value and the amplitude factor value for a speaker-virtual wall pair based on the distance from the speaker and the virtual wall of the speaker-virtual wall pair.

5. The apparatus of claim 3 or 4, wherein the module (1 10) for calculating wave field synthesis information is adapted to set the delay value of a speaker virtual wall pair the larger, the greater the distance between the speaker and the virtual wall is.

6. Device according to one of claims 3 to 5, wherein the module (1 10) for calculating wave field synthesis information is designed to set the amplitude value of a speaker-virtual wall pair the smaller, the greater the distance between the speaker and the virtual

Wall is.

7. Device according to one of the preceding claims, wherein the operating unit (130) is adapted for that at least one of the virtual walls from a first virtual position to a second virtual position is displaceable, so that the virtual wall displaceable in any direction parallel to its first position is.

8. Device according to one of the preceding claims, wherein the operating unit (130) is adapted for that at least one of the virtual walls from a first virtual position to a second virtual position is displaceable, so that the virtual wall displaceable rotatable relative to its first position is.

9. Device according to one of the preceding claims, wherein for all of the virtual walls, the virtual position by the operating unit (130) is adjustable.

10. The device of claim 1, wherein the operating unit is configured such that each of the virtual walls is displaceable from a first virtual position to a second virtual position such that each virtual wall is arbitrarily parallel and rotatable relative to its first position is displaceable.

A device according to any one of the preceding claims, wherein the device further comprises a parametric filter for filtering resonance frequencies.

12. A method of reverberation time extension, comprising:

Definition of a virtual position of one or more virtual walls,

Receiving a plurality of audio input signals received from a plurality of microphones,

Calculation of wave field synthesis information, and

Generating a plurality of audio outputs for a plurality of loudspeakers based on the audio input signals and based on the wavefield synthesis information, wherein the wavefield synthesis conversation is calculated based on the virtual position of the one or more virtual walls, and wherein for at least one of the virtual walls the virtual position is adjustable ,

13. Computer program with a program code for carrying out the method according to claim 12, when the computer program runs on a computer.

14. An electroacoustic reverberation time extension system comprising: a plurality of microphones (350, 41 1, 412), a reverberation time extension apparatus according to any one of claims 1 to 12, and a loudspeaker array comprising a plurality of loud speakers (370, 481, 482); wherein the plurality of microphones (350; 41 1, 412) are adapted to generate a plurality of audio input signals fed to the reverberation time extension apparatus, and wherein the plurality of loudspeakers (370; 481, 482) of the loudspeaker array are arranged therefor are the audio output signals from the reverberation time extension device fed and play the input audio output signals.

A method for reverberation time extension by means of an electroacoustic system, comprising:

Definition of a virtual position of one or more virtual walls,

Generating a plurality of audio input signals received from a plurality of microphones,

Calculation of wave field synthesis information,

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

Outputting the audio output signals by means of a loudspeaker array comprising a plurality of loudspeakers, the wavefield synthesis information being calculated based on the virtual position of the one or more virtual walls, and wherein for at least one of the virtual walls the virtual position is adjustable.