EP2757811B1 - Modale Strahlformung - Google Patents

Modale Strahlformung Download PDF

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EP2757811B1
EP2757811B1 EP13152209.6A EP13152209A EP2757811B1 EP 2757811 B1 EP2757811 B1 EP 2757811B1 EP 13152209 A EP13152209 A EP 13152209A EP 2757811 B1 EP2757811 B1 EP 2757811B1
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function
regularization
white noise
eigenbeam
parameter
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EP2757811A1 (de
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Markus Christoph
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Harman Becker Automotive Systems GmbH
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Harman Becker Automotive Systems GmbH
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/005—Circuits for transducers, loudspeakers or microphones for combining the signals of two or more microphones
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R2201/00—Details of transducers, loudspeakers or microphones covered by H04R1/00 but not provided for in any of its subgroups
    • H04R2201/40—Details of arrangements for obtaining desired directional characteristic by combining a number of identical transducers covered by H04R1/40 but not provided for in any of its subgroups
    • H04R2201/401—2D or 3D arrays of transducers

Definitions

  • the embodiments disclosed herein refer to sound capture systems and methods, particularly to sound capture methods that employ modal beamforming.
  • Beamforming sound capture systems comprise at least (a) an array of two or more microphones and (b) a beamformer that combines audio signals generated by the microphones to form an auditory scene representative of at least a portion of an acoustic sound field. Due to the underlying geometry, it is natural to represent the sound field captured on the surface of a sphere with respect to spherical harmonics. In this context, spherical harmonics are also known as acoustic modes (or eigenbeams) and the appending signal-processing techniques as modal beamforming.
  • the sphere may exist physically, or may merely be conceptual.
  • the microphones are arranged around a rigid sphere made of, for example, wood or hard plastic.
  • the microphones are arranged in free-field around an "open" sphere, referred to as an open-sphere configuration.
  • the rigid-sphere configuration provides a more robust numerical formulation, the open-sphere configuration might be more desirable in practice at low frequencies where large spheres are realized.
  • Beamforming techniques allow for the controlling of the characteristics of the microphone array in order to achieve a desired directivity.
  • One of the most general formulations is the filter-and-sum beamformer, which has readily been generalized by the concept of modal subspace decomposition. This approach finds optimum finite impulse response (FIR) filter coefficients for each microphone by solving an eigenvalue problem and projecting the desired beam pattern onto the set of eigenbeam patterns found.
  • FIR finite impulse response
  • Beamforming sound capture systems enable picking up acoustic signals dependent on their direction of propagation.
  • the directional pattern of the microphone array can be varied over a wide range due to the degrees of freedom offered by the plurality of microphones and the processing of the associated beamformer. This enables, for example, steering the look direction, adapting the pattern according to the actual acoustic situation, and/or zooming in to or out from an acoustic source. All this can be done by controlling the beamformer, which is typically implemented via software, such that no mechanical alteration of the microphone array is needed.
  • WO 03/061336 A1 discloses a microphone array-based audio system that supports representations of auditory scenes using second-order (or higher) harmonic expansions based on the audio signals generated by the microphone array.
  • a plurality of audio sensors are mounted on the surface of an acoustically rigid sphere. The number and location of the audio sensors on the sphere are designed to enable the audio signals generated by those sensors to be decomposed into a set of eigenbeams having at least one eigenbeam of order two (or higher). Beamforming (e.g., steering, weighting, and summing) can then be applied to the resulting eigenbeam outputs to generate one or more channels of audio signals that can be utilized to accurately render an auditory scene.
  • Beamforming e.g., steering, weighting, and summing
  • Alternative embodiments include using shapes other than spheres, using acoustically soft spheres and/or positioning audio sensors in two or more concentric patterns.
  • common beamformers fail to be directive at very low frequencies. Therefore, modal beamformers having less frequency-dependent directivity are desired.
  • a method for generating an auditory scene comprises: receiving eigenbeam outputs generated by decomposing a plurality of audio signals, each audio signal having been generated by a different microphone of a microphone array, wherein each eigenbeam output corresponds to a different eigenbeam for the microphone array and the microphones are arranged on a rigid sphere or an open sphere; generating the auditory scene based on the eigenbeam outputs and their corresponding eigenbeams, wherein generating the auditory scene comprises applying a weighting value to each eigenbeam output to form steered eigenbeam outputs and combining the steered eigenbeam outputs to generate the auditory scene.
  • Generating the auditory scene further comprises applying a regularized equalizing filter to each steered eigenbeam output, the regularized equalizing filter(s) being configured to compensate for acoustic deficiencies of the microphone array and having a regularized equalization function.
  • the regularized equalization function is a radial equalization function that comprises the quotient of a regularization function limiting the radial equalization function and a radial function describing an acoustic wave field in the vicinity of the surface of the rigid sphere or the center of the open sphere.
  • the regularization function is the quotient of the absolute value of the square of the radial function and the sum of the absolute value of the square of the radial function and a regularization parameter, the regularization parameter being set to a value greater than 0 and smaller than a maximum value that is smaller than infinity.
  • a modal beamformer system for generating an auditory scene comprises: a steering unit that is configured to receive eigenbeam outputs and to apply a weighting value to each eigenbeam output to provide steered provide steered eigenbeam outputs, the eigenbeam outputs having been generated by decomposing a plurality of audio signals, each audio signal having been generated by a different microphone of a microphone array, wherein each eigenbeam output corresponds to a different eigenbeam for the microphone array and the microphones are arranged on a rigid sphere or an open sphere.
  • the system further comprises a weighting unit that is configured to receive the steered eigenbeam outputs and to generate weighted steered eigenbeam outputs, and a summing element configured to combine the weighted steered eigenbeam outputs to generate the auditory scene.
  • the weighting unit is further configured to apply a regularized equalizing filter to each steered eigenbeam output, the regularized equalizing filter(s) being configured to compensate for acoustic deficiencies of the microphone array and having a regularized equalization function.
  • the regularized equalization function is a radial equalization function that comprises the quotient of a regularization function limiting the radial equalization function and a radial function describing an
  • the regularization function is the quotient of the absolute value of the square of the radial function and the sum of the absolute value of the square of the radial function and a regularization parameter, the regularization parameter being set to a value greater than 0 and smaller than a maximum value that is smaller than infinity.
  • FIG. 1 is a block diagram illustrating the basic structure of a beamforming sound capture system as described in more detail, for instance, in WO 03/061336 .
  • the sound capture system comprises a plurality Q of microphones Mic1, Mic2, ... MicQ configured to form a microphone array, a matrixing unit MU (also known as modal decomposer or eigenbeam former), and a modal beamformer BF.
  • modal beamformer BF comprises a steering unit SU, a weighting unit WU, and a summing element SE, each of which will be discussed in further detail later in this specification.
  • MicQ generates a time-varying analog or digital audio signal S 1 ( ⁇ 1 , ⁇ 1 ,ka), S 2 ( ⁇ 1 , ⁇ 2 ,ka) ... S Q (E Q , ⁇ Q ,ka) corresponding to the sound incident at the location of that microphone.
  • Y + ⁇ m,n ( ⁇ , ⁇ ) corresponds to a different mode for the microphone array.
  • the term auditory scene is used generically to refer to any desired output from a sound capture system, such as the system of FIG. 1 .
  • the definition of the particular auditory scene will vary from application to application.
  • the output generated by beamformer BF may correspond to one or more output signals, e.g., one for each speaker used to generate the resultant auditory scene.
  • beamformer BF may simultaneously generate beampatterns for two or more different auditory scenes, each of which can be independently steered to any direction in space.
  • microphones Mic1, Mic2, ... MicQ may be mounted on the surface of an acoustically rigid sphere or may be arranged on a virtual (open) sphere to form the microphone array.
  • weighting unit WU may be arranged upstream of steering unit SU so that the non-steered eigenbeams are weighted (not shown and not claimed).
  • FIG. 2 shows a schematic diagram of a possible microphone array MA for the sound capture system of FIG. 1 .
  • microphone array MA comprises the Q microphones Mic1, Mic2, ... MicQ of FIG. 1 mounted on the surface of an acoustically rigid sphere RS in a "truncated icosahedron" pattern.
  • Each microphone Mic1, Mic2, ... MicQ in microphone array MA generates one of the audio signals S 1 ( ⁇ 1 , ⁇ 1 ,ka), S 2 ( ⁇ 1 , ⁇ 2 ,ka)...
  • S Q (E Q , ⁇ Q ,ka) that is transmitted to matrixing unit MU of FIG. 1 via some suitable (e.g., wired or wireless) connection (not shown in FIG. 2 ).
  • the continuous spherical sensor may be replaced by a discrete spherical array, in particular when the subsequent processing is digital-signal processing.
  • beamformer BF exploits the geometry of the spherical array of FIG. 2 and relies on the spherical harmonic decomposition of the incoming sound field by matrixing unit MU to construct a desired spatial response.
  • steering unit SU generates (according to Y + ⁇ m,n ( ⁇ Des , ⁇ Des )) steered spherical harmonics Y +1 0,0 ( ⁇ Des , ⁇ Des ), Y +1 1,0 ( ⁇ Des , ⁇ Des ), ... Y + ⁇ m,n ( ⁇ Des, ⁇ Des ) from the spherical harmonics Y +1 0,0 ( ⁇ , ⁇ ), Y +1 1,0 ( ⁇ , ⁇ ), ...
  • Beamformer BF can provide continuous steering of the beampattern in 3-D space by changing a few scalar multipliers, while the filters determining the beampattern itself remain constant. The shape of the beampattern is invariant with respect to the steering direction. Beamformer BF needs only one filter per spherical harmonic (in the weighting unit WU), rather than per microphone as in known beamforming concepts, which significantly reduces the computational cost.
  • the sound capture system of FIG. 1 with the spherical array geometry of FIG. 2 enables accurate control over the beampattern in 3-D space.
  • the sound capture system can also provide multi-direction beampatterns or toroidal beampatterns giving uniform directivity in one plane. These properties can be useful for applications such as general multichannel speech pick-up, video conferencing, and direction of arrival (DOA) estimation. It can also be used as an analysis tool for room acoustics to measure, e.g., directional properties of the sound field.
  • DOA direction of arrival
  • the eigenbeams are also suitable for wave field synthesis (WFS) methods that enable spatially accurate sound reproduction in a fairly large volume, allowing for reproduction of the sound field that is present around the recording sphere. This allows for all kinds of general real-time spatial audio.
  • WFS wave field synthesis
  • FIG. 3 A circuit that provides the beamforming functionality is shown in detail in FIG. 3 .
  • the modal beamformer circuit of FIG. 3 receives the Q audio signals S 1 (S 1 , ⁇ 1 ,ka), S 2 ( ⁇ 1 , ⁇ 2 ,ka) ... S Q ( ⁇ Q , ⁇ Q ,ka) provided by microphones Mic1, Mic2, ... MicQ, transforms the audio signals S 1 ( ⁇ 1 , ⁇ 1 ,ka), S 2 ( ⁇ 1 , ⁇ 2 ,ka) ... S Q (6 Q , ⁇ Q ,ka) into the spherical harmonics Y +1 0,0 ( ⁇ , ⁇ ), Y +1 1,0 ( ⁇ , ⁇ ), ... Y + ⁇ m,n ( ⁇ , ⁇ ), and steers the spherical harmonics.
  • the circuit of FIG. 3 may be realized by hardware (and software) components that (together) build matrixing unit MU and the modal beamformer, which includes steering unit SU, modal weighting unit WU, and summing element SE.
  • Matrixing unit MU and steering unit SU include coefficient elements CE that multiply the respective input signals with given coefficients and adders AD that sum up the input signals multiplied with coefficients so that the audio signals S 1 ( ⁇ 1 , ⁇ 1 ,ka), S 2 ( ⁇ 1 , ⁇ 2 ,ka) ...
  • S Q ( ⁇ Q , ⁇ Q ,ka) are decomposed into the eigenbeams, i.e., the spherical harmonics Y +1 0,0 ( ⁇ , ⁇ ), Y +1 1,0 ( ⁇ , ⁇ ), ... Y + ⁇ m,n ( ⁇ , ⁇ ), which are then processed to provide the steered spherical harmonics Y +1 0,0 ( ⁇ Des , ⁇ Des ), Y +1 1,0 ( ⁇ Des , ⁇ Des ), ... Y + ⁇ m,n ( ⁇ Des , ⁇ Des ).
  • Modal weighting unit WU includes delay elements DE, coefficient elements CE, and adders AD, which are connected to form FIR filters for weighting. The output signals of these FIR filters are summed up by summing element SE.
  • Matrixing unit MU in the modal beamformer of FIG. 3 is responsible for decomposing the sound field, which is picked up by microphones Mic1, Mic2, ... MicQ and decomposed into the different eigenbeam outputs, i.e., the spherical harmonics Y +1 0,0 ( ⁇ , ⁇ ), Y +1 1,0 ( ⁇ , ⁇ ), ... Y + ⁇ m,n ( ⁇ , ⁇ ), corresponding to the zero-order, first-order, and second-order spherical harmonics.
  • This can also be seen as a transformation, where the sound field is transformed from the time or frequency domain into the "modal domain”.
  • the real and imaginary parts of the spherical harmonics can also work with the real and imaginary parts of the spherical harmonics.
  • weighting unit WU may be implemented accordingly.
  • Steering unit SU allows for steering the look direction by the angles ⁇ Des and ⁇ Des .
  • Weighting unit WU compensates for a frequency-dependent sensitivity over the modes (eigenbeams), i.e., modal weighting over frequency, to the effect that the modal composition is adjusted, e.g., equalized.
  • Equalizing is used to compensate for deficiencies of the microphone array, e.g., self-noise of the microphones, location errors of the microphones at the surface of the sphere, and other electrical and mechanical drawbacks.
  • the order of a modal beamformer has to be reduced toward low frequencies, leading to a gradually decreasing directivity pattern with decreasing frequency.
  • the ambisonic components up to M th order can be calculated from the Q microphone signals:
  • B W ⁇ 1 Y T Y ⁇ 1 Y T p a
  • B diag W m ⁇ 1 Y + p a
  • diag EQ m ka diagonal matrix having the radial equalizing functions EQ m ka , in which 0 ⁇ m ⁇ M .
  • FIG. 4 An arrangement for extracting the N ambisonic components B from the wave field p a is illustrated in FIG. 4 .
  • the related sound field is defined solely by the pressure distribution p a ( ⁇ q , ⁇ q ) on the sphere's surface, which can be easily measured by sound pressure sensors (microphones).
  • p a ⁇ q , ⁇ q , ⁇ q , 0
  • inner sources i.e., sources inside the measurement sphere
  • outer sources i.e., sources outside the measurement sphere
  • the outer sources serve to model the scattered field occurring at the surface of a scattered sphere.
  • a parameter called susceptibility K( ⁇ ) or its reciprocal white noise gain WNG( ⁇ ) may be used.
  • white noise gain WNG( ⁇ ) addresses most effects and problems caused by microphone noise, changes in the transfer function, and variations of the microphone positions, so that it is representative of the sensitivity of the beamformer.
  • a white noise gain WNG( ⁇ ) > 0 [dB] characterizes a sufficient suppression of uncorrelated errors and is thus indicative of a robust system behavior, while a white noise gain WNG( ⁇ ) ⁇ 0 [dB] is indicative of an amplification of the noise and is therefore indicative of an increasingly unstable system behavior.
  • the array gain G( ⁇ ) is the ratio of the energy of sound coming from the look direction of the beamformer to the energy of omnidirectionally incoming sound.
  • the array gain G( ⁇ ) is a measure for the improvement in the acoustic signal-to-noise ratio SNR, based on the directivity of the modal beamformer for sound coming from the look direction of the beamformer.
  • parameters required for calculation are set to a starting value or a constant value, as the case may be.
  • the following parameters may be set to, for instance:
  • Regularization provides the ability to achieve a robust system by way of adjusting the regularization parameter ⁇ ( ⁇ ). This is a trade-off between a higher robustness, i.e., a higher white noise gain WNG dB ( ⁇ ), and less directivity in look direction ⁇ ( ⁇ 0 , ⁇ 0 , ⁇ ), i.e., a decreasing array gain G dB ( ⁇ ).
  • the adaptation process begins with the maximum directivity G dBMax ( ⁇ ) and is then decreased by the increasing regularization parameter ⁇ ( ⁇ ) until the desired white noise gain threshold WNG dBMin is no more undercut.
  • Steps 4, 5, and 6 serve to calculate the white noise gain WNG db ( ⁇ ).
  • the regularization filter T m ( ⁇ ) or T m (ka) is calculated as outlined above using regularization parameter ⁇ ( ⁇ ).
  • the transfer function EQ m ( ⁇ ) is calculated as outlined above using the current version of the transfer function T m ( ⁇ ) of the regularization filter or the current version of the regularization parameter ⁇ ( ⁇ ).
  • the white noise gain WNG db ( ⁇ ) is calculated as outlined above using the transfer function EQ m ( ⁇ ) and the current version of the transfer function T m ( ⁇ ) of the regularization filter (regularization function). Steps 4 and 5 may be taken simultaneously or in opposite order.
  • step 10 the directivity ⁇ ( ⁇ 0 , ⁇ 0 , ⁇ ) of the modal beamformer is calculated for sound coming from the look direction using the transfer function EQ m ( ⁇ ) provided in step 5.
  • step 12 the current white noise gain WNG db ( ⁇ ) is compared with the predetermined white noise gain threshold WNG dBMin ( ⁇ ), and it is checked to see if the regularization parameter ⁇ ( ⁇ ) has reached its maximum according to (
  • step 14 the adaptation process for the current angular frequency ⁇ has been completed so that the current equalizing function EQ m ( ⁇ ) has been limited to the given threshold or if the current regularization parameter has reached its maximum.
  • step 14 the current angular frequency ⁇ is checked to see if it has reached its maximum value ⁇ Max . If ⁇ ⁇ ⁇ Max , the process jumps back to step 2 using the current angular frequency ⁇ . Otherwise, i.e., if the equalizing filter has been adapted for the complete set of frequencies, the filter coefficients are outputted in step 15.
  • the directivity characteristic of the beamformer is a 4 th -order cardioid and the minimum white noise gain WNG db ( ⁇ ) used in the adaptation process is -10 [dB].
  • FIG. 9 illustrates a regularization parameter over frequency ⁇ ( ⁇ ) for a common 4 th -order modal beamformer.
  • regularization i.e., limiting the maximum directivity index for frequencies up to, for instance, 750 [Hz]
  • values above a minimum lower threshold WNG dbMin of -10 [dB] may be maintained.
  • the exemplary beamformer exhibits the desired directivity of a 4 th -order cardioid.
  • FIG. 10 illustrates the corresponding white noise gain WNG for the above-mentioned 4 th -order beamformer, which supports the findings in connection with the diagram of FIG. 9 .
  • the corresponding directivity index DI and the array gain G db ( ⁇ ) as shown FIG. 11 illustrate that the maximum array gain G db ( ⁇ ) is more or less below 10 [dB] depending on the frequency.
  • FIG. 16 depicts the resulting directivity of the beamformer outlined above in look directivity ⁇ ( ⁇ 0 , ⁇ 0 , ⁇ ) as amplitudes over frequency.

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Claims (11)

  1. Verfahren zum Generieren einer auditorischen Szene (OUT), umfassend:
    Empfangen von Eigenstrahlausgaben, wobei die Eigenstrahlausgaben durch das Zerlegen einer Vielzahl von Audiosignalen (S1(θ1,ϕ1,ka), S2(θ1,ϕ2,ka) ... SQ(θQ,ϕQ,ka)) generiert wurden, wobei jedes Audiosignal (S1(θ1,ϕ1,ka), S2(θ1,ϕ2,ka) ... SQ(θQ,ϕQ,ka)) durch ein anderes Mikrofon (Mic1, Mic2, ... MicQ) einer Mikrofonanordnung generiert wurde, wobei jede Eigenstrahlausgabe einem anderen Eigenstrahl für die Mikrofonanordnung entspricht, und die Mikrofone (Mic1, Mic2, ... MicQ) auf einer starren Kugel (RS) oder einer offenen Kugel angeordnet sind; und
    Generieren der auditorischen Szene (OUT) auf Grundlage der Eigenstrahlausgaben und ihrer entsprechenden Eigenstrahlen (Y+1 0,0(θ,ϕ), Y+1 1,0(θ,ϕ), ... Y+σ m,n(θ,ϕ)), wobei:
    Generieren der auditorischen Szene (OUT) das Anwenden eines Gewichtungswerts für jede Eigenstrahlausgabe, um gesteuerte Eigenstrahlausgaben zu bilden und das Kombinieren der gesteuerten Eigenstrahlausgaben umfasst, um die auditorische Szene (OUT) zu generieren;
    Generieren der auditorischen Szene (OUT) ferner das Anwenden eines regularisierten Entzerrungsfilters (EQ) für jede gesteuerte Eigenstrahlausgabe umfasst, wobei der bzw. die regularisierte(n) Entzerrungsfilter (EQ) konfiguriert ist bzw. sind, um akustische Mängel der Mikrofonanordnung zu kompensieren und eine regularisierte Entzerrungsfunktion (EQm(ka)) aufweist bzw. aufweisen;
    die regularisierte Entzerrungsfunktion (EQm(ka)) eine radiale Entzerrungsfunktion ist, die den Quotienten einer Regularisierungsfunktion (Tm(ka)), welche die radiale Entzerrungsfunktion einschränkt und eine radiale Funktion Wm(ka)) umfasst, die ein akustisches Wellenfeld in der Nähe der Oberfläche der starren Kugel (RS) oder des Mittelpunkts der offenen Kugel beschreibt, dadurch gekennzeichnet, dass
    die Regularisierungsfunktion (Tm(ka)) der Quotient des absoluten Betrags der Quadratzahl der radialen Funktion und der Summe des absoluten Betrags der Quadratzahl der radialen Funktion und eines Regularisierungsparameters ist, wobei der Regularisierungsparameter auf einen Wert größer als 0 und kleiner als ein Maximalwert eingestellt ist, der kleiner als unendlich ist.
  2. Verfahren nach Anspruch 1, wobei der Maximalwert des Regularisierungsparameters 1 ist.
  3. Verfahren nach Anspruch 1 oder 2, wobei der Regularisierungsparameter von einem Suszeptibilitätsparameter abhängt, welcher der Kehrwert eines Parameters für die Verstärkung des weißen Rauschens ist, wobei der Parameter für die Verstärkung des weißen Rauschens größer ist als ein minimaler Parameter für die Verstärkung des weißen Rauschens, der von dem Entzerrungsfilter (EQ) nicht unterschritten wird.
  4. Verfahren nach Anspruch 3, wobei der minimale Parameter für die Verstärkung des weißen Rauschens -10 [dB] ist.
  5. Verfahren nach einem der Ansprüche 1 bis 4, wobei der Regularisierungsparameter in einem iterativen Prozess angepasst wird.
  6. Verfahren nach Anspruch 5, wobei der iterative Prozess für eine gegebene Frequenz Folgendes umfasst:
    Einstellen zumindest des minimalen Parameters für die Verstärkung des weißen Rauschens und der Regularisierungsparameter auf einen Ausgangswert oder einen konstanten Wert; und
    Berechnen der Verstärkung des weißen Rauschens, der Regularisierungsfunktion und der radialen Entzerrungsfunktion; und
    Vergleichen des berechneten Parameters für die Verstärkung des weißen Rauschens mit dem eingestellten minimalen Parameter für die Verstärkung des weißen Rauschens; und
    Berechnen der Richteigenschaft für Geräusche, die aus der Blickrichtung kommen, unter Verwendung der radialen Entzerrungsfunktion; und
    Skalieren der radialen Entzerrungsfunktion; und
    Vergleichen der berechneten Verstärkung des weißen Rauschens mit der eingestellten minimalen Verstärkung des weißen Rauschens und Überprüfen, ob der Regularisierungsparameter sein Maximum erreicht hat; wenn beide Anforderungen erfüllt sind, ist der Anpassungsprozess noch nicht beendet, woraus sich ein Zurückspringen und ein erneutes Beginnen mit einem aktualisierten Regularisierungsparameter ergeben; andernfalls wurde der Prozess für die aktuelle Frequenz abgeschlossen und die Frequenz wird inkrementiert; und
    Überprüfen, ob die aktuelle Frequenz ihren Maximalwert erreicht hat; wenn die Frequenz ihr Maximum nicht erreicht hat, springt der Prozess zurück und beginnt erneut mit einer anderen Frequenz; andernfalls werden die Filterkoeffizienten ausgegeben.
  7. Verfahren nach Anspruch 6, wobei der iterative Prozess einen verschobenen Parameter für die Verstärkung des weißen Rauschens umfasst, durch den der minimale Parameter für die Verstärkung des weißen Rauschens während der Anpassung maximal überschritten oder unterschritten wird.
  8. Modales Strahlformersystem (BF) zum Generieren einer auditorischen Szene (OUT), umfassend:
    eine Steuereinheit (SU), die konfiguriert ist, um Eigenstrahlausgaben zu empfangen und einen Gewichtungswert für jede Eigenstrahlausgabe anzuwenden, um gesteuerte Eigenstrahlausgaben bereitzustellen, wobei die Eigenstrahlausgaben durch das Zerlegen einer Vielzahl von Audiosignalen (S1(θ1,ϕ1,ka), S2(θ1,ϕ2,ka) ... SQ(θQ,ϕQ,ka)) generiert wurden, wobei jedes Audiosignal (S1(θ1,ϕ1,ka), S2(θ1,ϕ2,ka) ... SQ(θQ,ϕQ,ka)) durch ein anderes Mikrofon (Mic1, Mic2, ... MicQ) einer Mikrofonanordnung generiert wurde, wobei jede Eigenstrahlausgabe einem anderen Eigenstrahl (Y+1 0,0(θ,ϕ), Y+1 1,0(θ,ϕ), ... Y+σ m,n(θ,ϕ)) für die Mikrofonanordnung entspricht, und die Mikrofone (Micl, Mic2, ... MicQ) auf einer starren Kugel (RS) oder einer offenen Kugel angeordnet sind;
    eine Gewichtungseinheit (WU), die konfiguriert ist, um die gesteuerten Eigenstrahlausgaben zu empfangen und um gewichtete gesteuerte Eigenstrahlausgaben zu generieren; und
    ein Summierelement (SE), das konfiguriert ist, um die gewichteten gesteuerten Eigenstrahlausgaben zu kombinieren, um die auditorische Szene (OUT) zu generieren, wobei:
    die Gewichtungseinheit (WU) ferner konfiguriert ist, um einen regularisierten Entzerrungsfilter (EQ) für jede gesteuerte Eigenstrahlausgabe anzuwenden, wobei der bzw. die regularisierte(n) Entzerrungsfilter (EQ) konfiguriert ist bzw. sind, um akustische Mängel der Mikrofonanordnung zu kompensieren und eine regularisierte Entzerrungsfunktion (EQm(ka)) aufweist bzw. aufweisen; und
    die regularisierte Entzerrungsfunktion (EQm(ka)) eine radiale Entzerrungsfunktion ist, die den Quotienten einer Regularisierungsfunktion (Tm(ka)), welche die radiale Entzerrungsfunktion einschränkt und eine radiale Funktion Wm(ka)) umfasst, die ein akustisches Wellenfeld in der Nähe der starren Kugel (RS) oder der offenen Kugel beschreibt, dadurch gekennzeichnet, dass
    die Regularisierungsfunktion (Tm(ka)) der Quotient des absoluten Betrags der Quadratzahl der radialen Funktion und der Summe des absoluten Betrags der Quadratzahl der radialen Funktion und eines Regularisierungsparameters ist, wobei der Regularisierungsparameter auf einen Wert größer als 0 und kleiner als ein Maximalwert eingestellt ist, der kleiner als unendlich ist.
  9. System nach Anspruch 8, wobei der Maximalwert des Regularisierungsparameters 1 ist.
  10. System nach Anspruch 8 oder 9, wobei der Regularisierungsparameter von einem Suszeptibilitätsparameter abhängt, welcher der Kehrwert eines Parameters für die Verstärkung des weißen Rauschens ist, wobei der Parameter für die Verstärkung des weißen Rauschens größer ist als ein minimaler Parameter für die Verstärkung des weißen Rauschens, der von dem Entzerrungsfilter (EQ) nicht unterschritten wird.
  11. System nach Anspruch 10, wobei der minimale Parameter für die Verstärkung des weißen Rauschens -10 [dB] ist.
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