Hearing system with at least one hearing instrument worn on the user's head and methods for operating such a hearing system

Abstract

Method for operating a hearing system (2) to assist the hearing ability of a user, comprising at least one hearing instrument (4) worn on the head, in particular in or on an ear of the user, - wherein a sound signal from the user's environment is received by means of at least two input transducers (6) of the hearing system 2 and converted into input audio signals (I1, I2), - wherein the input audio signals (I1, I2) are processed in a signal conditioning step to generate an output audio signal (O), - wherein the output audio signal (O) is output by means of an output converter (8) of the hearing instrument (4), - wherein in the signal processing step the input audio signals (I1, I2) or audio signals derived therefrom by preprocessing (I1', I2') are attenuated directionally by means of a first adaptive beamformer (28) according to a first variable directional characteristic with a directivity (s) in order to generate a first directional audio signal (R1), - wherein in an adaptation step the directional characteristic of the first adaptive beamformer (28) is varied with a first adaptation speed (v1) such that the energy content of the first directional audio signal (R1) is minimized, and - wherein the first adaptation speed (v1) and / or the directivity (s) are variably set based on an analysis of the input audio signals (I1, I2) or the pre-processed audio signals (I1', I2').

Description

[0001] The invention relates to a hearing system for supporting a user's hearing, comprising at least one hearing instrument worn on the head, in particular in or on one ear of the user. The invention further relates to a method for operating such a hearing system.

[0002] A hearing instrument is generally defined as an electronic device that supports the hearing of a person wearing the hearing instrument (hereinafter referred to as the "wearer" or "user"). In particular, the invention relates to hearing instruments designed to fully or partially compensate for the hearing loss of a hearing-impaired user. Such a hearing instrument is also referred to as a "hearing aid." There are also hearing instruments designed to protect or improve the hearing of users with normal hearing, for example, to enable improved speech comprehension in complex listening situations.

[0003] Hearing instruments in general, and hearing aids in particular, are usually designed to be worn on the user's head or ear, particularly as behind-the-ear devices (also known as BTE devices) or in-the-ear devices (also known as ITE devices). In terms of their internal structure, hearing instruments usually have at least one (acousto-electrical) input transducer, a signal processing unit (signal processor), and an output transducer. During operation of the hearing instrument, the or each input transducer receives airborne sound from the hearing instrument's environment and converts this airborne sound into an input audio signal (i.e., an electrical signal that conveys information about the ambient sound). This at least one input audio signal is also referred to below as the "recorded sound signal."In the signal processing unit, the or each input audio signal is conditioned (i.e., modified with respect to its sound information) to support the user's hearing, in particular to compensate for a user's hearing loss. The signal processing unit outputs a correspondingly conditioned audio signal (also referred to as the "output audio signal" or "modified sound signal") to the output transducer.

[0004] In most cases, the output transducer is designed as an electro-acoustic transducer that converts the (electrical) output audio signal back into airborne sound, which is then emitted into the user's ear canal after being modified relative to the ambient sound. In a hearing instrument worn behind the ear, the output transducer, also known as the "receiver," is usually integrated outside the ear in a housing of the hearing instrument. In this case, the sound emitted by the output transducer is guided into the user's ear canal via a sound tube. Alternatively, the output transducer can be located in the ear canal, and thus outside the housing worn behind the ear. Such hearing instruments are also referred to as "receiver in canal" devices.Hearing instruments worn in the ear that are so small that they do not protrude beyond the ear canal are also called CIC devices (from the English term “completely in canal”).

[0005] In other designs, the output transducer can also be an electromechanical transducer that converts the output audio signal into structure-borne sound (vibrations), which is then transmitted, for example, to the user's skull. There are also implantable hearing instruments, particularly cochlear implants, and hearing instruments whose output transducers directly stimulate the user's auditory nerve.

[0006] The term "hearing system" refers to a single device or a group of devices and, where appropriate, non-physical functional units that together provide the functions required for the operation of a hearing instrument. In the simplest case, the hearing system may consist of a single hearing instrument. Alternatively, the hearing system may comprise two interacting hearing instruments to serve both ears of the user. In this case, it is referred to as a "binaural hearing system." Additionally or alternatively, the hearing system may comprise at least one other electronic device, for example, a remote control, a charger, or a programming device for the or each hearing aid.Modern hearing systems often feature a control program, particularly in the form of an app, instead of a remote control or dedicated programming device. This control program is designed for implementation on an external computer, particularly a smartphone or tablet. The external computer is usually not part of the hearing system itself, as it is usually provided independently of the hearing system and not by the hearing system manufacturer.

[0007] To attenuate background noise during operation of a hearing system, and thus in particular to improve speech intelligibility during communication between the user and another speaker, direction-dependent attenuation (beamforming) of the input audio signal is often used as part of the signal processing in a hearing system. This attenuation attenuates the components of the input audio signal originating from different directions to varying degrees according to a predefined directional characteristic. The directional characteristic often has one or more directions of maximum attenuation, which are also referred to as notches. In modern hearing systems, corresponding attenuation units (beamformers) are sometimes designed adaptively. Such an adaptive beamformer automatically varies its directional characteristic to optimally attenuate background noise.In particular, the notch may be aligned towards a dominant noise source in order to dampen the sound component emanating from this noise source particularly effectively.

[0008] In order to be able to follow noise sources that are moving relative to the user's head (e.g. passing motor vehicles), an adaptive beamformer is often implemented with a comparatively high adaptation speed. A high adaptation speed is also important to be able to compensate for the user's head movements; because a user's head movement also causes the noise sources in the user's environment to move relative to their head - from the user's perspective. The adaptation speed is usually dimensioned so high that the adaptive beamformer can automatically realign itself in the opposite direction to the head rotation without any noticeable delay when the head is turned, thus maintaining its orientation towards a specific noise source during and after the head rotation.

[0009] Unfortunately, such fast-adapting beamformers often tend to produce negative effects (artifacts) in dynamic listening situations, which the user perceives as unnatural. Since the directional attenuation always affects other sound components in addition to the noise of the background noise source being attenuated – especially in frequency ranges where the background noise is weak or absent – ​​the adaptation of the directional attenuation can lead to a perceptible fluctuation of useful signals or background noise in the sound signal output to the user. Such artifacts can significantly impair the user's hearing under unfavorable circumstances and, in extreme cases, even cause a deterioration in speech intelligibility (instead of the desired improvement).In particular, fluctuations in otherwise static background noise caused by the adaptation of the directional attenuation can lead to increased perception of these background noises and thus distract the user from concentrating on the actual desired signal. Particularly disruptive effects can be caused by the notch of an adaptive beamformer jumping back and forth between different noise sources.

[0010] One approach, known from EP 2 908 550 B1, for at least partially resolving these problems, consists in detecting a user's head movement using an acceleration, direction, or inclination sensor and adjusting the "viewing direction" of the beamformer (i.e., the directional lobe) in the opposite direction to the detected head movement. Alternatively, EP 2 908 550 B1 discloses increasing the adaptation speed of an adaptive beamformer when a user's head movement is detected by the sensor. Both approaches allow the beamformer to be designed with a comparatively low adaptation speed in the absence of head rotation, thus reducing the risk of artifacts of the type described above in this situation.

[0011] However, these approaches do not help against artifacts of the adaptive directional attenuation, which are caused by noise sources moving independently of the user's head (e.g. passing motor vehicles).

[0012] DE 10 2009 012 166 A1 discloses a noise reduction system for a hearing aid that can attenuate both stationary and non-stationary background noise in an input signal. The output signal is intended to convey a calm sound impression. For this purpose, signal processing is provided that achieves noise reduction based on two different methods. One method provides noise reduction for stationary background noise, and the other, noise reduction for spatially directed background noise. A selection device selects between the two noise reduction methods.

[0013] The application is based on the objective of improving an adaptive directional attenuation used in the operation of a hearing system (for modifying the sound signal picked up from the environment and output to the user in a modified form) with a view to avoiding artifacts. The aim is thus to create a directional attenuation that enables an improved hearing experience for the user.

[0014] With regard to a method, this object is achieved according to the invention by the features of claim 1. With regard to a hearing aid system, the object is achieved according to the invention by the features of claim 7. Advantageous and partly inventive embodiments or further developments of the invention are set out in the subclaims and the following description.

[0015] The invention is generally based on a hearing system for supporting the hearing of a user, wherein the hearing system has at least one hearing instrument worn on the head, in particular in or on one ear of the user. As described above, in simple embodiments of the invention the hearing system can consist exclusively of a single hearing instrument. In another embodiment of the invention the hearing system comprises, in addition to the hearing instrument, at least one further component, e.g. a further (in particular similar) hearing instrument for supplying the other ear of the user, a control program (in particular in the form of an app) for execution on an external computer (in particular a smartphone) of the user and / or at least one further electronic device, e.g. a remote control or a charger.In this case, the hearing instrument and the at least one other component exchange data with each other, with data storage and / or data processing functions of the hearing system being divided between the hearing instrument and the at least one other component.

[0016] The hearing system has at least two input transducers, each of which serves to receive a sound signal (particularly in the form of airborne sound) from the environment of the hearing instrument. The at least two input transducers can be arranged in the same hearing instrument, particularly if the hearing system comprises only a single hearing instrument. In a binaural hearing system with two hearing instruments, the at least two input transducers can alternatively be distributed between the two hearing instruments.

[0017] The hearing system further comprises signal processing, with a signal processing unit for processing (modifying) the received sound signal to support the user's hearing ability, as well as an output transducer for outputting the modified sound signal. In a binaural hearing system, both hearing instruments preferably each have a signal processing unit and an output transducer. Instead of a second hearing instrument with an input transducer, signal processing unit, and output transducer, the hearing system for the second ear can, within the scope of the invention, also have a hearing instrument that does not have an output transducer itself, but only receives sound and transmits it—with or without signal processing—to the hearing instrument of the first ear. Such so-called CROS or BiCROS instruments are used particularly for users with unilateral deafness.Furthermore, within the scope of the invention, the signal processing or a portion thereof can also be outsourced from the hearing instrument(s) to an external unit, e.g., an app running on a smartphone. The signal processing of the hearing system preferably comprises, in addition to the signal processing unit, a signal analysis unit that does not itself generate an audio signal to be output directly or indirectly to the user, but supports the function of the hearing system, in particular the signal processing unit, by analyzing audio signals or other sensor signals.

[0018] The or each hearing instrument of the hearing system is available in one of the designs described above (BTE device with internal or external output transducer, ITE device, e.g., CIC device, hearing implant, especially cochlear implant, etc.) or as a hearable device. In the case of a binaural hearing system, both hearing instruments are preferably designed in the same way.

[0019] Each of the input transducers is, in particular, an acousto-electrical transducer that converts ambient airborne sound into an electrical input audio signal. The or each output transducer is preferably designed as an electro-acoustic transducer (earpiece), which in turn converts the audio signal modified by the signal processing unit into airborne sound. Alternatively, the output transducer is designed to emit structure-borne sound or to directly stimulate the user's auditory nerve.

[0020] In the course of the method, a sound signal from the user's environment is recorded using the hearing system's at least two input transducers and converted into input audio signals. The input audio signals are processed in a signal conditioning step to generate an output audio signal. This output audio signal is output via the hearing instrument's output transducer. In the signal conditioning step, the input audio signals are fed directly (or indirectly after preprocessing) to a first adaptive beamformer, by which the input audio signals or the audio signals derived from them by preprocessing (preprocessed audio signals) are directionally attenuated according to a variable (first) directional characteristic with a predetermined (first) directional strength.The first adaptive beamformer generates a first directional audio signal, which is output directly (or indirectly after one or more further signal processing steps) as the modified audio signal to the electro-acoustic transducer for output to the user.

[0021] In an adaptation step, the directional characteristic of the first adaptive beamformer is varied depending on a predetermined (first) adaptation speed such that the energy content of the first directional audio signal is minimized.

[0022] The “directional characteristic” of a beamformer is generally defined as the dependence of the attenuation of sound components of the recorded sound signal by the beamformer on the direction from which these sound components are received.

[0023] The deviation of the directional characteristic from the omnidirectional characteristic is particularly evident in the fact that the direction-dependent attenuation of the corresponding adaptive beamformer exhibits at least one local maximum. This or each attenuation maximum of the directional characteristic is also referred to as a "notch," and the corresponding direction of maximum attenuation is also referred to as the "notch direction."

[0024] In a preferred embodiment of the invention, the or each notch direction is defined in the form of an angle specification, for example, relative to the user's viewing direction. Alternatively, the or each notch direction can also be specified as an abstracted quantity—linearly or non-linearly correlated with the orientation of the associated notch—for example, in the form of a weighting factor used to weight various basic directional signals (e.g., a cardioid signal and an anti-cardioid signal, etc.) for adjusting common adaptive beamformers, or in the form of a variable time delay with which different signal components are superimposed to generate the directional effect.

[0025] The "directivity" generally describes how much the directivity of the associated adaptive beamformer deviates from an omnidirectional characteristic (i.e., signal processing without direction dependence). The directivity of the first adaptive beamformer is fixed in certain embodiments of the invention. In this case, the directivity is determined, in particular, implicitly by the functional structure or design of the first adaptive beamformer.

[0026] The "adaptation speed" generally describes how quickly the associated adaptive beamformer adapts its directional characteristic to a change in the background noise (i.e., the spatial distribution of the noise sources and thus the sound components in the input audio signals). In certain embodiments of the invention, the adaptation speed of the first adaptive beamformer is fixed. In this case, the adaptation speed is determined implicitly by the functional design of the first adaptive beamformer.

[0027] According to the invention, however, at least one of the two above-described properties of the directional characteristic of the first adaptive beamformer, namely the adaptation speed and / or the directivity, is not fixed, but is specified to the first adaptive beamformer as a variable, thus as a variable parameter. The adaptation speed or the directivity of the first adaptive beamformer is variably adjusted (preferably by the signal analysis unit described above) based on an analysis of the input audio signals or the preprocessed audio signals.

[0028] By variably adjusting the adaptation speed and / or the directivity of the first adaptive beamformer, artifacts of the type described above can be effectively avoided. In particular, this makes it possible to temporarily increase the adaptation speed and thus the adaptability of the first adaptive beamformer when changes in the background noise require a significant adjustment of the first beamformer. In this way, a perceptible delay in the adaptation of the first adaptive beamformer to a movement of a noise source relative to the head can be avoided. A similar effect can be achieved alternatively or additionally by temporarily reducing the directivity.On the other hand, the first adaptation speed can be set low in static listening situations, thus avoiding artificial fluctuations of useful signals or background noise due to small-scale adjustments of the directional characteristic of the first adaptive beamformer. In this case, a comparatively high directivity enables good attenuation of background noise and thus good perception of the useful signal by the user, thus facilitating speech comprehension in particular during communication between the user and another speaker. The invention is based on the finding that controlling the adaptation speed and / or the directivity of the first adaptive beamformer based on an analysis of the input audio signals or the preprocessed audio signals allows for a particularly effective avoidance of artifacts of direction-dependent attenuation.

[0029] The two measures described above—varying the adaptation speed of the first adaptive beamformer on the one hand and varying the directivity of the first adaptive beamformer on the other—generally contribute independently of one another to achieving this effect. These measures can therefore be used independently of one another within the scope of the invention by varying either only the adaptation speed or only the directivity of the first adaptive beamformer. In a preferred embodiment of the method, however, both the adaptation speed and the directivity of the first adaptive beamformer are varied.

[0030] Preferably, the adaptation speed and / or the directivity of the first adaptive beamformer are adjusted as a function of a temporal stability of the input audio signals or the preprocessed audio signals (more precisely, as a function of a temporal stability of the background noise underlying the input audio signals). In variants of the method in which the adaptation speed of the first adaptive beamformer is varied, this is set low, in particular with high temporal stability (i.e., weak temporal change) of the respective audio signals, and high with low temporal stability (i.e., strong temporal change). This results in the directivity of the first adaptive beamformer being adapted quickly when the background noise changes greatly and slowly when the background noise changes slightly.In variants of the method in which the directivity of the first adaptive beamformer is varied, this is set high when the respective audio signals have high temporal stability (i.e. little change over time) and low when the temporal stability is low (i.e. there is great change over time). This means that the input audio signals or the pre-processed audio signals are strongly directed by the first adaptive beamformer when the noise level changes slightly and weakly or not at all when the noise level changes greatly. To determine the temporal stability, a variable characterizing the noise level is derived from the input audio signals or the pre-processed audio signals. The standard deviation of this variable or a root mean square of the first temporal derivative of this variable over a sliding period is used as a measure of the temporal stability.Alternatively or additionally, the rate at which a moving average of this quantity is exceeded or undershot (mean crossing rate) or the rate at which the first derivative of this quantity changes sign can be used as a measure of the time stability of the input audio signals or the preprocessed audio signals.

[0031] To characterize the background noise, an advantageous embodiment of the invention utilizes a second adaptive beamformer with a (second) variable directional characteristic. This second adaptive beamformer, like the first adaptive beamformer, is applied directly or indirectly to the input audio signals to generate a second directional audio signal. The directional characteristic of the second adaptive beamformer is adjusted with a (preferably constant) (second) adaptation speed such that the energy content of the second directional audio signal is minimized. As in the case of the first directional characteristic, the second directional characteristic is preferably characterized by at least one variable direction of maximum attenuation (notch direction). However, in an expedient embodiment of the invention, the second directional audio signal is not included in the modified audio signal output to the user.In this case, the second adaptive beamformer is used solely for signal analysis. The second directional audio signal is used only as a control variable for energy minimization and not for signal conditioning for output to the user.

[0032] The adaptation speed of the second adaptive beamformer is selected in particular such that it does not fall below the adaptation speed of the first adaptive beamformer and at least temporarily exceeds it. The second adaptive beamformer is therefore always designed to adapt quickly, so that it can adapt to changes in the background noise without significant delay. The directional characteristic of this second adaptive beamformer (in particular a notch direction possibly assigned to this directional characteristic) thus forms a measure of the variability of the background noise underlying the input audio signals. In comparison to the second adaptive beamformer, the first adaptive beamformer either always adapts slowly or alternates between slow and fast adaptation.

[0033] In an advantageous embodiment of the method, the adaptation speed and / or the directivity of the first adaptive beamformer are variably adjusted depending on the change in the second directivity (i.e., the directivity of the second adaptive beamformer). As a parameter for the temporal change of the second directivity and thus as a measure of the variability of the background noise, a moving root mean square of the first temporal derivative of a notch direction assigned to the second directivity is determined, for example. For example, the adaptation speed of the first adaptive beamformer is increased compared to a base value and / or the directivity of the first adaptive beamformer is decreased compared to a base value if and as long as the aforementioned parameter exceeds a predetermined threshold.

[0034] Alternatively or additionally, the adaptation speed and / or the directivity of the first adaptive beamformer are preferably adjusted depending on the deviation of the second directivity from the first directivity, in particular depending on the deviation of a (second) notch direction associated with the second directivity from a (first) notch direction associated with the first directivity. For example, the adaptation speed of the first adaptive beamformer is increased and / or the directivity of the first adaptive beamformer is decreased if and as long as the above-described deviation of the notch directions exceeds a predetermined threshold.

[0035] In a further development of the method variant described above, instead of the one second adaptive beamformer for analyzing the input audio signals or the preprocessed audio signals and thus for setting the first adaptation speed and / or the first directivity, several adaptive beamformers are used, which correspond in particular to the second adaptive beamformer in terms of structure, function (and optionally also design). These several adaptive (analysis) beamformers are coupled to one another, in particular, so that a different orientation of their respectively assigned directional characteristics (and possibly the associated notch directions) is enforced. This ensures, in particular, that each of the several adaptive (analysis) beamformers is aligned to a different dominant noise source in the user's environment.Such a cascade of analysis beamformers allows the background noise underlying the input audio signals and changes in this background noise to be analyzed with high precision.

[0036] The first adaptive beamformer is preferably frequency-dependent, meaning it modifies different frequency components of the input audio signal or the pre-processed audio signals in a respective individual manner. In particular, the input audio signals or the pre-processed audio signals are each divided into different frequency channels, with the directional characteristic (in particular the or each notch direction) of the first adaptive beamformer being adapted individually for each frequency channel. In a preferred embodiment of the invention, the adaptation speed and / or the directivity of the first adaptive beamformer are also specified as a frequency-dependent variable (e.g. as a vector with one entry for each frequency channel or as a continuous frequency-dependent function), so that the directional characteristic of the first adaptive beamformer is adapted at different speeds for different frequencies, if necessary.so that the directivity of the first adaptive beamformer may be different for different frequencies.

[0037] To adjust the adaptation speed and / or the directivity of the first adaptive beamformer in a frequency-dependent manner, at least one noise component emanating from a noise source is identified in the input audio signals or the preprocessed audio signals and analyzed with regard to its spectral composition. Specifically, an interference frequency range corresponding to this noise component is determined. The adaptation speed and / or the directivity of the first adaptive beamformer are specified uniformly (i.e., with the same value) within the interference frequency range. This prevents the first beamformer from being adapted differently for different frequency components of one and the same noise, which could lead to acoustic distortion and / or artificial fluctuation of the noise.

[0038] The hearing system according to the invention is generally configured to automatically carry out the above-described method according to the invention. For this purpose, the hearing system comprises the first adaptive beamformer (as described above). The hearing system further comprises an adaptivity controller configured to variably adjust the adaptation speed and / or the directivity of the first adaptive beamformer based on an analysis of the input audio signals or preprocessed audio signals.

[0039] The hearing system's means for automatically implementing the method according to the invention is of a programming and / or circuitry nature. The hearing system according to the invention therefore comprises programming means (software) and / or circuitry (hardware, e.g., in the form of an ASIC) that automatically implement the method according to the invention during operation of the hearing system. The programming or circuitry means for implementing the method, in particular the first adaptive beamformer and the adaptivity controller, can be arranged exclusively in the hearing instrument (or instruments) of the hearing system. Alternatively, the programming or circuitry means for implementing the method are distributed across the hearing instrument(s) and at least one other device or software component of the hearing system.For example, the programming means for implementing the method are distributed between at least one hearing instrument of the hearing system and a control program installed on an external electronic device (in particular, a smartphone). As mentioned above, the external electronic device itself is generally not part of the hearing system.

[0040] The above-described embodiments of the method according to the invention correspond to corresponding embodiments of the hearing system according to the invention.

[0041] Thus, in preferred embodiments of the invention, the adaptivity control is configured to adjust the adaptation speed and / or the directivity of the first adaptive beamformer depending on the time stability of the input audio signals.

[0042] Preferably, the adaptivity control comprises a second adaptive beamformer or a cascade of further (in particular mutually coupled) adaptive beamformers, as described above, for analyzing the input audio signals or the preprocessed audio signals (i.e., for characterizing the underlying background noise) and for determining the adaptation speed and / or the directivity of the first adaptive beamformer.

[0043] The adaptivity control is in particular configured to variably adjust the adaptation speed and / or the directivity of the first adaptive beamformer depending on the change in the (respective) directivity of the second beamformer (and possibly the further adaptive beamformers) and / or depending on the deviation of the directivity of the adaptive beamformers.

[0044] The first adaptive beamformer preferably has a frequency-dependent directional characteristic (as described above), in particular an individually adapted directional characteristic for a plurality of frequency channels. The adaptivity controller is preferably configured to specify the adaptation speed and / or the directivity of the first adaptive beamformer as a frequency-dependent variable, to identify a noise component emanating from an interference noise source in the input audio signals or the preprocessed audio signals to adjust the adaptation speed and / or the directivity of the first adaptive beamformer, to determine an interference frequency range corresponding to the interference noise component, and to uniformly specify the adaptation speed and / or the directivity of the first adaptive beamformer in the interference frequency range.

[0045] The effects and advantages of the individual procedure variants can be transferred to the corresponding variants of the hearing system and vice versa.

[0046] In the following, exemplary embodiments of the invention are explained in more detail with reference to a drawing. In the drawings: Fig. 1 shows a schematic representation of a hearing system consisting of a single hearing instrument in the form of a hearing aid worn behind the ear of a user, Fig. 2 to 4 each show in a schematic block diagram the structure of a signal processing of the hearing system from Fig. 1 in three alternative embodiments, Fig. 5 in representation according to Fig. 2 to 4 a functional unit of the signal processing of the hearing system, referred to as adaptivity control, in a further embodiment, and Fig. 6 in representation according to Fig. 1 an alternative embodiment of the hearing system, in which it comprises a hearing instrument in the form of a hearing aid that can be worn behind the ear and a control program (“hearing app”) implemented in a smartphone.

[0047] Corresponding parts and sizes are always provided with the same reference symbols in all figures.

[0048] Fig. Figure 1 shows a hearing system 2, which here consists of a single hearing aid 4, i.e., a hearing instrument designed to support the hearing of a hearing-impaired user. In the example shown here, the hearing aid 4 is a BTE hearing aid worn behind the user's ear.

[0049] Optionally, in a further embodiment of the invention, the hearing system 2 comprises a second hearing aid (not expressly shown) for supplying the second ear of the user, which in terms of its construction is particularly similar to the one shown in Fig. 1 corresponds to the hearing aid 4 shown.

[0050] Within a housing 5, the hearing aid 4 comprises two microphones 6 as acoustoelectric input transducers and a receiver 8 as an electroacoustic output transducer. The hearing aid 4 further comprises a battery 10 and signal processing in the form of a signal processor 12. Preferably, the signal processor 12 comprises both a programmable subunit (e.g., a microprocessor) and a non-programmable subunit (e.g., an ASIC).

[0051] The signal processor 12 is supplied with an electrical supply voltage U from the battery 10.

[0052] During normal operation of the hearing aid 4, the microphones 6 each pick up airborne sound from the environment of the hearing aid 4. The microphones 6 convert the sound into an (input) audio signal I1 or I2, respectively, which contains information about the recorded sound. The input audio signals I1, I2 are fed within the hearing aid 4 to the signal processor 12, which modifies these input audio signals I1, I2 to support the user's hearing.

[0053] The signal processor 12 outputs an output audio signal O, which contains information about the processed and thus modified sound, to the listener 8.

[0054] The receiver 8 converts the output sound signal O into a modified airborne sound. This modified airborne sound is transmitted into the user's ear canal via a sound channel 14, which connects the receiver 8 to a tip 16 of the housing 5, and via a flexible sound tube (not explicitly shown) that connects the tip 16 to an earpiece inserted into the user's ear canal.

[0055] The structure of the signal processing is in Fig. 2 in more detail. This shows that the signal processing of the hearing system 2 is divided into two functional components, namely a signal processing unit 18 and a signal analysis unit 20. The signal processing unit 18 serves to generate the output audio signal O from the input audio signals I1, I2 of the microphones 6 or from the audio signals I1', I2' derived therefrom by preprocessing and thus preprocessed. In the former case, the input audio signals I1, I2 of the microphones 6 are fed directly to the signal processing unit 18. In the latter case, which is exemplified in Fig. 2, the input audio signals I1, I2 of the microphones 6 are first fed to a preprocessing unit 22, which then derives the preprocessed audio signals I1', I2 and feeds them to the signal processing unit 18.

[0056] In the preprocessing unit 22, the input audio signals I1, I2 are preferably superimposed on one another with a time delay to form the preprocessed audio signals I1', I2, so that the two preprocessed audio signals I1', I2 correspond to a cardioid signal and an anti-cardioid signal, respectively.

[0057] The signal processing unit 18 comprises a number of signal processing processes 24 which process the input audio signals I or - in the example according to Fig. 2 - successively process and modify the internal audio signals I1', I2 to generate the output audio signal O, thus compensating the user's hearing loss.

[0058] The signal processing processes 24 are optionally implemented in any desired combination in the form of (non-programmable) hardware circuits and / or in the form of software modules (firmware) in the signal processor 12. For example, at least one of the signal processing processes 24 is formed by a hardware circuit, at least one further of the signal processing processes 24 by a software module, and yet another of the signal processing processes 24 by a combination of hardware and software components. The signal processing processes 24 include, for example, - a process for noise and / or feedback suppression, - a process for dynamic compression and - a process for frequency-dependent amplification based on audiogram data, - etc.

[0059] At least one signal processing parameter P is assigned to at least one of these signal processing processes 24 (usually all signal processing processes 24 or at least most signal processing processes 24). The or each signal processing parameter P is a one-dimensional variable (binary variable, natural number, floating-point number, etc.) or a multi-dimensional variable (array, function, etc.) whose value parameters (i.e., influences) the functionality of the respectively assigned signal processing process 24. Signal processing parameters P can switch the respectively assigned signal processing process 24 on or off, continuously or gradually amplify or attenuate the effect of the respectively assigned signal processing process 24, define time constants for the respective signal processing process 24, etc.

[0060] For example, the signal conditioning parameters P • the above-mentioned audiogram data or frequency-specific gain factors derived therefrom for a frequency-dependent amplification process, • a characteristic curve for a dynamic compression process, • a control variable for continuously adjusting the strength of a noise or feedback suppression process, • etc.

[0061] At least a part of the signal processing parameters P is made available to the signal processing processes 24 from a parameterization unit 26.

[0062] Furthermore, the signal processing processes 24 comprise a - in Fig. 2 - first adaptive beamformer 28, which is designed to convert the input audio signals I1, I2 (or, as in Fig. 2, the preprocessed audio signals I1', I2') are attenuated in a direction-dependent manner according to a variable (first) directional characteristic and in this way a first directional audio signal R1 is generated. The beamformer 28 generates the audio signal R1 by combining the two supplied audio signals I1', I2' (in the example according to Fig. 2 i.e. a cardioid signal and an anti-cardioid signal) are superimposed on each other weighted with a first weighting factor a1: R=I1'−a1⋅I2' with a=[−1;1]

[0063] The weighting factor a1 determines a notch direction in which - relative to the user's head - the direction-dependent attenuation of the beamformer 28 has a (local) maximum. The weighting factor a1 thus represents a measure of the notch direction of the beamformer 28 and is therefore conceptually equated with this notch direction in the following. To adapt the directional characteristic, the weighting factor a1 is varied in an adaptation step by the beamformer 28 in a control process such that the energy content of the directional audio signal R1 is minimized (this self-regulation of the beamformer 28 is described in Fig. 2 (shown schematically by feeding the audio signal R1 back to the beamformer 28). The described energy minimization ensures that background noise from a spatial area behind the user's head is suppressed as best as possible. The directional audio signal R1 output by the beamformer 28 is further processed by the additional signal processing units 24, thereby generating the output audio signal O. The beamformer 28 is preferably formed by a software module.

[0064] A first adaptation speed v1 is variably specified for the beamformer 28 as the signal conditioning parameter P. This adaptation speed v1 is determined in the signal analysis unit 20 by a functional unit referred to as adaptivity control 30, which is preferably implemented in software.

[0065] In the Fig. In the embodiment shown in Figure 2, the adaptivity controller 30 comprises a second adaptive beamformer 32 and an evaluation module 34.

[0066] The second adaptive beamformer 32 preferably corresponds in terms of structure and function to the first adaptive beamformer 28. The second adaptive beamformer 32 is thus configured in the manner described above to convert the input audio signals I1, I2 (or, as in Fig. 2, the preprocessed audio signals I1', I2') are attenuated directionally according to a (second) variable directional characteristic, thereby generating a second directional signal R2. The directional characteristic of the beamformer 32 preferably has a notch direction characterized by a variable weighting factor a2. The weighting factor a2 (and thus the notch direction) are varied by the beamformer 32 at an adaptation speed v2 such that the energy content of the directional audio signal R2 is minimized.

[0067] In contrast to the beamformer 28, the beamformer 32 does not generate the output audio signal O output to the user, but only analyzes the background noise underlying the input signals I1, I2. The audio signal R2 is therefore not further processed but is only fed back to the beamformer 32 for the purpose of self-regulation. Instead, the beamformer 32 outputs the weighting factor a2, which indicates the notch direction (and thus indirectly the arrangement of the most dominant noise sources in the user's environment), to the evaluation module 34 as the analysis result.

[0068] In the evaluation module 34, in the embodiment according to Fig. 2 the temporal stability (or - expressed conversely - the temporal variability) of the weighting factor a2 and thus of the background noise is evaluated, for example by forming a moving quadratic temporal average over the first temporal derivative of the weighting factor a2. Depending on this value, the evaluation module 34 varies the adaptation speed v1 for the first adaptive beamformer 28. In a simple but expedient embodiment, the evaluation module 34 varies the adaptation speed v1 in a binary manner between a comparatively low base value and a comparatively increased value. The evaluation module 34 sets the adaptation speed v1 to the base value if and as long as the average value described above does not exceed a predetermined threshold value (which indicates that the background noise is not or only slightly variable).The first beamformer 28 thus adapts only slowly in this case, largely avoiding artifacts resulting from adaptation. Otherwise, i.e., if and as long as the mean value exceeds the threshold due to a significant change in the background noise and the weighting factor a2, the adaptation speed v1 is increased relative to the base value, so that the first adaptive beamformer 28 can adapt quickly (in particular, without any perceptible delay) to the changed listening situation.

[0069] To analyze the background noise with high precision, the second adaptive beamformer 32 is designed to adapt rapidly. The adaptation speed v2 is selected (preferably as a constant) such that it never falls below the variable adaptation speed v1 of the first adaptive beamformer 28 (v2 ≥ v1).

[0070] In addition or alternatively to the adaptation speed v1, a directivity s of the first adaptive beamformer 28 is preferably also variable. The variation of the directivity s is realized, for example, by mixing the weighted sum according to Eq. 1 to varying degrees with an omnidirectional audio signal A derived from the input audio signals I1, I2 (which is supplied to the beamformer 28 according to Fig. 2 is optionally supplied as an additional input variable). The directional intensity s is reduced by the evaluation module 34 compared to a predetermined base value if and as long as - in particular based on the threshold value exceedance described above - a strong variability of the background noise is detected.

[0071] As from Fig. 2, the signal analysis unit 20 contains, in addition to the adaptivity control 30 and preferably in addition to other functions for sound analysis not explicitly shown here, an optional classifier 36 which is generated in a conventional manner by analyzing the input audio signals I1, I2 (or, as in Fig. 2, the preprocessed audio signals I1', I2') analyze the current listening situation with regard to its similarity to a plurality of typical listening situation classes (such as "speech", "speech with background noise" or "music") and output a corresponding classification signal K.

[0072] The classification signal K is fed, on the one hand, to the parameterization unit 26, which, in a conventional manner, makes a selection between different hearing programs, i.e. different parameter sets of the signal processing parameters P, each optimized for one of the typical hearing situation classes, depending on the classification signal K.

[0073] On the other hand, the classification signal K is also fed to the evaluation module 34 of the adaptivity control 30 and influences the determination of the adaptation speed v1 and / or the directivity s. For example, the values ​​between which the adaptation speed v1 and / or the directivity s are varied are again changed depending on the classification signal K.

[0074] In Fig. 3 shows an alternative embodiment of the hearing system 2. In contrast to the embodiment according to Fig. 2 is the evaluation module 34 in the design according to Fig. 3, in addition to the weighting factor a2 of the beamformer 32, the weighting factor a1 of the beamformer 28 is also supplied. The evaluation module 34 analyzes the variability of the background noise underlying the input audio signals I1, I2 and the audio signals I1', I2' by comparing the weighting factors a1 and a2. A large deviation of the rapidly changing weighting factor a1 from the slowly changing weighting factor a2 in the baseline state is considered an indication of a significant change in the background noise. Accordingly, the evaluation module 34 increases the adaptation speed v1 and / or decreases the directivity s if and as long as the difference between the weighting factors a1 and a2 exceeds a predetermined threshold.

[0075] In Fig. 4 shows a further embodiment of the hearing system 2. In contrast to the embodiments according to Fig. 2 and Fig. 3, the adaptivity controller 30 comprises, in addition to the second adaptive beamformer 28, at least one further adaptive beamformer 28, which generates a further directional audio signal R3 and, based on an energy minimization of this audio signal R3, varies an associated further weighting factor a3 (as a measure of a variable notch direction of the beamformer 38).

[0076] An adaptation speed v3 assigned to the beamformer 38 and preferably constantly predefined has, in an expedient embodiment, a value that is lower than the adaptation speed v2 and in particular corresponds exactly or approximately to the basic value of the adaptation speed v1. In this case, the further adaptive beamformer 38 is thus designed to adapt slowly compared to the second adaptive beamformer 32, wherein both beamformers 32 and 38 preferably adjust the respective weighting factor a2 or a3 independently of one another (a coupling of the beamformers 32 and 38, as in Fig. 4 by the supply of the weighting factor a2 to the beamformer 38, is preferably not provided in this embodiment variant). The variability of the background noise underlying the input audio signals I1, I2 and the preprocessed audio signals I1', I2' is hereby evaluated by the evaluation module 34 analogously to the embodiment according to Fig. 3 is determined based on the deviation of the weighting factors a2 and a3 of the beamformers 32 and 38.

[0077] In an alternative embodiment, the adaptation speeds v2 and v3 of the beamformers 32 and 38 are selected to be exactly or approximately the same, so that both beamformers 32 and 38 adapt quickly. In this case, the beamformers 26 and 38 are preferably - as in Fig. 4 - are coupled to each other, so that a different setting of the weighting factors a2 and a3 is enforced. This coupling ensures that the beamformers 26 and 38 adjust to different dominant noise sources in the user's environment. The variability of the background noise underlying the input audio signals I1, I2 and the preprocessed audio signals I1', I2' is evaluated by the evaluation module 34 in this case, preferably analogously to the embodiment according to Fig. 2 is determined based on the time stability of the weighting factors a2 and a3. The adaptation speed v1 is increased and / or the reference intensity s is decreased if the condition for increasing the adaptation speed v1 or decreasing the reference intensity s is met for at least one of the weighting factors a2 and a3.

[0078] The classifier 36 is optionally also in the embodiments according to Fig. 3 and Fig. 4 and are not shown in these figures for the sake of clarity.

[0079] Preferably, the signal processing in the signal processing unit 18 is carried out frequency-resolved in a plurality of (e.g. 64) frequency channels. The input audio signals I1, I2 are, preferably before being fed to the preprocessing unit 22, Fig. 2 to 4 not explicitly shown) analysis filter bank are each divided into frequency components, which are individually processed in the frequency channels and then in a (in the Fig. 2 to 4 (also not explicitly shown) synthesis filter bank to form the output audio signal O.

[0080] The first adaptive beamformer 28 is configured to attenuate the frequency components of the input audio signals I1, I2 or the preprocessed audio signals I1', I2' carried in the frequency channels individually and in a direction-dependent manner. Thus, the directional characteristic of the beamformer 28 and the associated notch direction or the weighting factor a1 also exhibit a frequency dependency. Preferably, the weighting factor a1 and / or the directivity s are each specified as a vector having an assigned individual value for each frequency channel. Furthermore, the directional characteristic of the beamformer 28 is preferably also individually adapted for each frequency channel. Thus, the adaptation speed v1 is also preferably specified as a vector having an assigned individual value for each frequency channel.

[0081] To prevent an interference noise originating from a specific noise source from being perceptibly distorted by the beamformer 28 as a result of frequency-specific, different adaptation of the directional characteristic, the adaptivity controller 30 is preferably configured to couple frequency channels that carry significant frequency components of a dominant interference noise with respect to the adaptation speed v1 and / or the directivity s. In other words, the adaptivity controller 30 specifies the adaptation speed v1 and / or the directivity s uniformly (i.e., with the same value) for those frequency channels that carry significant frequency components of a dominant interference noise.

[0082] For this purpose, the second adaptive beamformer 32 (and optionally the third adaptive beamformer 38) are preferably also constructed analogously to the beamformer 28 in such a way that they attenuate the frequency components of the input audio signals I1, I2 or the preprocessed audio signals I1', I2' carried in the frequency channels individually and in a direction-dependent manner. The background noise is thus analyzed with frequency resolution by the second adaptive beamformer 32 (and optionally the third adaptive beamformer 38).

[0083] In order to determine the spectral composition of one or more dominant noises, the directional audio signal R2 (or R3 respectively) output by the second adaptive beamformer 32 (and possibly the third adaptive beamformer 38) is inverted in an inverter 40 and then multiplied by the omnidirectional audio signal A in a multiplier 42. This signal processing is in Fig. 5 shows an example of an embodiment of the adaptivity control 30, which is analogous to Fig. 4 comprises both the second adaptive beamformer 32 and the third adaptive beamformer 38. Multiplying the omnidirectional audio signal A by the inverted, directional audio signal R2 (or R3) produces an audio signal R2' (or R3') in which the dominant noise, which was selectively filtered out by the second adaptive beamformer 32 (or, if applicable, the third adaptive beamformer 38), is selectively amplified. The audio signal R2' (or, if applicable, R3') is then fed to the evaluation module 34, which analyzes the spectral composition of the audio signal R2' (or, if applicable, R3') and determines an interference frequency range corresponding to the respective interference noise.The frequency channels lying in this interference frequency range are coupled by the evaluation module 34 with regard to the adaptivity of the first adaptive beamformer 28, in that the evaluation module 34 uniformly specifies the values ​​of the adaptation speed v1 and / or the directivity s corresponding to these frequency channels: For example, the adaptation speed v1 for all coupled frequency channels is increased compared to the basic value and / or the directivity s for all coupled frequency channels is reduced compared to the basic value if and as long as (from the data determined by the evaluation module 34 in accordance with . Fig. The evaluation of the weighting factor a2 or the weighting factors a2 and a3 carried out in sections 2 and 4 respectively shows that for at least one of the coupled frequency channels the condition for increasing the adaptation speed v1 or reducing the directivity s is fulfilled.

[0084] Fig. Figure 6 shows a further embodiment of the hearing system 2, in which it comprises, in addition to the hearing aid 4 (or two hearing aids of this type for supplying both ears of the user), control software. This control software is referred to below as hearing app 44. The hearing app 44 is in the Fig. 5, the smartphone 46 is installed on a smartphone 46. The smartphone 46 itself is not part of the hearing system 2. Rather, the smartphone 46 is used by the hearing app 44 merely as a resource for storage space and computing power.

[0085] During operation of the hearing system 2, the hearing aid 4 and the hearing app 46 exchange data via a wireless data transmission connection 48. The data transmission connection 48 is based, for example, on the Bluetooth standard. The hearing app 44 accesses a Bluetooth transceiver of the smartphone 46 to receive data from the hearing aid 4 and to send data to it. The hearing aid 4, in turn, includes a Bluetooth transceiver (not explicitly shown) to send data to the hearing app 44 and to receive data from this app.

[0086] In the execution according to Fig. 6 are parts of the Fig. 2 to 5 (e.g., the adaptivity control 30) is not implemented in the signal processor 12 of the hearing aid 4, but rather in the hearing app 44.

[0087] The invention is particularly clear from the exemplary embodiments described above, but is not limited to these exemplary embodiments. Rather, further embodiments of the invention can be derived by those skilled in the art from the claims and the above description. In particular, the individual features of the hearing system and the associated operating method explained with reference to the various exemplary embodiments can also be combined with one another in other ways within the scope of the claims without deviating from the invention. List of reference symbols 2 hearing system 4 Hearing aid 5 housings 6 Microphone 8 listeners 10 Battery 12 Signal processor 14 sound channel 16 lace 18 Signal processing unit 20 Signal analysis unit 22 Preprocessing unit 24 Signal processing process 26 Parameterization unit 28 (first adaptive) beamformer 30 Adaptivity control 32 (second adaptive) beamformers 34 Evaluation module 36 Classifier 38 (third adaptive) beamformers 40 Inverting element 42 multiplier 44 Listening app 46 smartphones 48 Data transmission connection a1 (first) weighting factor a2 (second) weighting factor a3 (third) weighting factor s guideline strength v1 (first) adaptation speed v2 (second) adaptation speed v3 (third) adaptation speed A (omnidirectional) audio signal I1, I2 input audio signal I1', I2' (internal) audio signal K classification signal O Output audio signal P Signal conditioning parameters R1 (first directional) audio signal R2 (second directional) audio signal R3 (third directional) audio signal U supply voltage

Claims

[1] Method for operating a hearing system (2) for supporting the hearing of a user, with at least one hearing instrument (4) worn on the head, in particular in or on an ear of the user, - wherein a sound signal from the user's environment is picked up by means of at least two input transducers (6) of the hearing system 2 and converted into input audio signals (I1, I2), - wherein the input audio signals (I1, I2) are processed in a signal processing step to generate an output audio signal (O), - wherein the output audio signal (O) is output by means of an output transducer (8) of the hearing instrument (4), - wherein in the signal processing step, the input audio signals (I1, I2) or audio signals (I1', I2') derived therefrom by preprocessing are attenuated in a direction-dependent manner by means of a first adaptive beamformer (28) in accordance with a first variable directional characteristic with a directional strength (s) in order to generate a first directional audio signal (R1), - wherein in an adaptation step the directional characteristic of the first adaptive beamformer (28) is varied at a first adaptation speed (v1) such that the energy content of the first directional audio signal (R1) is minimized, and - wherein the first adaptation speed (v1) and / or the directivity (s) are variably adjusted based on an analysis of the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2'). [2] Method according to claim 1, wherein the first adaptation speed (v1) and / or the directivity (s) are adjusted as a function of a time stability of the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2'). [3] Method according to claim 1 or 2, - wherein, to adjust the first adaptation speed (v1) and / or the directivity (s) for the first adaptive beamformer (28), a second adaptive beamformer (32) with a second variable directivity is applied to the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2') to generate a second directed audio signal (R2), - wherein the second variable directional characteristic of the second adaptive beamformer (32) is adjusted at a second adaptation speed (v2) such that the energy content of the second directional audio signal (R2) is minimized, - wherein the second adaptation speed (v2) does not fall below the first adaptation speed (v1) and at least temporarily exceeds it. [4] Method according to claim 3, wherein the first adaptation speed (v1) and / or the directivity (s) are variably adjusted depending on a change in the second directivity characteristic. [5] Method according to claim 3 or 4, wherein the first adaptation speed (v1) and / or the directivity (s) are adjusted as a function of a deviation of the second directivity characteristic from the first directivity characteristic. [6] Method according to one of claims 1 to 5, - wherein the first directional characteristic is frequency-dependent, so that different frequency components of the input audio signals (I1, I2) or the pre-processed audio signals (I1', I2') are attenuated in an individual manner depending on the direction, - wherein the first adaptation speed (v1) and / or the directivity (s) for the first adaptive beamformer (28) are specified as a frequency-dependent variable, - wherein, for setting the first adaptation speed (v1) and / or the directivity (s), a noise component emanating from a noise source is identified in the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2'), - whereby a noise frequency range corresponding to the noise component is determined, and - wherein the first adaptation speed (v1) and / or the directivity (s) are specified uniformly in the interference frequency range. [7] Hearing system (2) for supporting the hearing of a user, with at least one hearing instrument (4) worn on the head, in particular in or on one ear of the user, - wherein the hearing system (2) has at least two input transducers (6) which are designed to receive a sound signal from the user's environment and to convert it into input audio signals (I1, I2), - wherein the hearing system (2) has a signal processing unit (18) which is designed to process the input audio signals (I1, I2) to generate an output audio signal (O), - wherein the hearing instrument (4) has an output transducer (8) which is arranged to output the output audio signal (O), - wherein the signal processing unit (18) comprises a first adaptive beamformer (28) which is configured to attenuate the input audio signals (I1, I2) or audio signals (I1', I2') derived therefrom by preprocessing in a direction-dependent manner according to a first variable directional characteristic with a directional strength (s) in order to generate a first directional audio signal (R1), and to vary the first directional characteristic with a first adaptation speed (v1) such that the energy content of the first directional audio signal (R1) is minimized, - wherein the hearing system (2) comprises an adaptivity control (30) which is configured to variably adjust the first adaptation speed (v1) and / or the directivity (s) based on an analysis of the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2'). [8] Hearing system (2) according to claim 7, wherein the adaptivity control (30) is configured to adjust the first adaptation speed (v1) and / or the directivity (s) depending on a time stability of the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2'). [9] Hearing system (2) according to claim 7 or 8, - wherein the adaptivity control (30) comprises a second adaptive beamformer (32) with a second variable directional characteristic, to which the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2') are supplied, - wherein the second adaptive beamformer (32) is configured to generate a second directional audio signal (R2) and to adjust the second variable directional characteristic at a second adaptation speed (v2) such that the energy content of the second directional audio signal (R2) is minimized, - wherein the second adaptation speed (v2) does not fall below the first adaptation speed (v1) and at least temporarily exceeds it. [10] Hearing system (2) according to claim 9, wherein the adaptivity control (30) is configured to variably adjust the first adaptation speed (v1) and / or the directivity (s) depending on a change in the second directivity characteristic. [11] Hearing system (2) according to claim 9 or 10, wherein the adaptivity control (30) is configured to adjust the first adaptation speed (v1) and / or the directivity (s) depending on a deviation of the second directivity characteristic from the first directivity characteristic. [12] Hearing system (2) according to one of claims 7 to 11, - wherein the first directional characteristic is frequency-dependent, so that different frequency components of the input audio signals (I1, I2) or the pre-processed audio signals (I1', I2') are attenuated in different ways depending on the direction, - wherein the adaptivity control (30) is arranged to - to specify the first adaptation speed (v1) and / or the directivity (s) as a frequency-dependent quantity, - to identify a noise component originating from a noise source in the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2') for setting the first adaptation speed (v1) and / or the directivity (s), - to determine a noise frequency range corresponding to the noise component, and - to specify the first adaptation speed (v1) and / or the directivity (s) uniformly in the interference frequency range.

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