EP3926983A2 - Hearing system with at least one hearing instrument worn in or on the ear of the user and method for operating such a hearing system - Google Patents

Abstract

A hearing system (2) for supporting the hearing ability of a user is specified, which comprises at least one hearing instrument (4) worn on the user's head. A method for operating the hearing system (2) is also specified. A sound signal from the surroundings of the user is picked up by means of at least two input converters (6) of the hearing system (4) and converted into input audio signals (I1, I2). The input audio signals (I1, I2) are processed in a signal processing step to generate an output audio signal (O), which is output by means of an output converter (8) of the hearing instrument (4). In the signal processing step, the input audio signals (I1, I2) or audio signals (I1', I2') derived from them by preprocessing are directionally attenuated by means of an adaptive beam former (28) according to a variable directional characteristic with a directional strength (s) in order to to generate a directional audio signal (R1). The directional characteristic is varied with a predetermined adaptation speed (v1) in such a way that the energy content of the directional audio signal (R1) is minimized. The adaptation speed (v1) and/or the directivity (s) are set variably based on an analysis of the input audio signals (I1, I2) or the preprocessed audio signals (I1', I2').

Description

The invention relates to a hearing system for supporting the hearing ability of a user, with at least one hearing instrument worn on the head, in particular in or on an ear of the user. The invention also relates to a method for operating such a hearing system.

A hearing instrument is generally referred to as an electronic device that supports the hearing ability of a person wearing the hearing instrument (hereinafter referred to as “wearer” or “user”). In particular, the invention relates to hearing instruments which are set up to compensate for a hearing loss of a hearing-impaired user in whole or in part. Such a hearing instrument is also referred to as a “hearing aid”. There are also hearing instruments that protect or improve the hearing ability of normal hearing users, for example to enable improved speech understanding in complex listening situations.

Hearing instruments in general, and hearing aids in particular, are mostly designed to be worn on the head, in particular on him or on one ear of the user, in particular as behind-the-ear devices referred to as BTE devices) or in-the-ear devices (also referred to as ITE devices after the English term "in the ear"). With regard to their internal structure, hearing instruments generally have at least one (acousto-electrical) input transducer, a signal processing unit (signal processor) and an output transducer. When the hearing instrument is in operation, the or each input transducer picks up airborne sound from the environment of the hearing instrument and converts this airborne sound into an input audio signal (ie an electrical signal that conveys information about the ambient sound). This at least one input audio signal is also referred to below as a “recorded sound signal”. In the signal processing unit, the or each input audio signal is processed (ie modified with regard to its sound information) in order to support the hearing ability of the user, in particular to compensate for a hearing loss of the user. The signal processing unit outputs a correspondingly processed audio signal (also referred to as “output audio signal” or “modified sound signal”) to the output transducer.

In most cases, the output transducer is designed as an electro-acoustic transducer, which converts the (electrical) output audio signal back into airborne sound, this airborne sound - modified compared to the ambient sound - being emitted into the user's ear canal. In the case of a hearing instrument worn behind the ear, the output transducer, also referred to as a “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 conducted into the ear canal of the user by means of a sound tube. As an alternative to this, the output transducer can also be arranged in the auditory canal and thus outside the housing worn behind the ear. Such hearing instruments are also referred to as RIC devices (after the English term "receiver in canal"). Hearing instruments worn in the ear that are so small that they do not protrude beyond the auditory canal are also referred to as CIC devices (after the English term "completely in canal").

In further designs, the output transducer can also be designed as an electro-mechanical transducer which converts the output audio signal into structure-borne sound (vibrations), this structure-borne sound being emitted, for example, into the skull bone of the user. There are also implantable hearing instruments, in particular cochlear implants, and hearing instruments whose output transducers directly stimulate the user's auditory nerve.

The term “hearing system” denotes an individual device or a group of devices and possibly non-physical functional units which together provide the functions required for the operation of a hearing instrument. In the simplest case, the hearing system can consist of a single hearing instrument. As an alternative to this, the hearing system can comprise two interacting hearing instruments for supplying the two ears of the user. In this case it is called a "binaural hearing system". Additionally or alternatively, the hearing system can comprise at least one further electronic device, for example a remote control, a charger or a programming device for the or each hearing device. In modern hearing systems, instead of a remote control or a dedicated programming device, a control program, in particular in the form of a so-called app, is often provided, this control program being designed to be implemented on an external computer, in particular a smartphone or tablet. The external computer is usually not part of the hearing system itself, in so far as it is usually provided independently of the hearing system and also not by the manufacturer of the hearing system.

In order to attenuate background noises during the operation of a hearing system, and thus in particular to improve speech understanding in 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, by means of which the Components of the input audio signals originating from different directions are attenuated to different degrees in accordance with a predetermined 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 damping units (beamformers) are sometimes designed to be adaptive. Such an adaptive beamformer varies its directional characteristic automatically in order to optimally dampen interfering noises. In particular, if necessary, the notch is aligned with a dominant noise source in order to dampen the sound component emanating from this noise source particularly effectively.

In order to be able to follow sources of background noise that move 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 in order to be able to compensate for head movements of the user; because even a head movement of the user leads to the fact that the sources of background noise in the surroundings of the user - from the user's point of view - move relative to the user's head. The adaptation speed is regularly dimensioned so high that the adaptive beamformer can automatically realign itself in the opposite direction to the head rotation when the head is rotated without any noticeable delay and thus maintains the alignment to a certain source of background noise during and after the head rotation.

Disadvantageously, however, such rapidly adapting beamformers often tend to generate negative effects (artifacts) in dynamic listening situations, which are perceived as unnatural by the user. Since the direction-dependent damping not only influences the noise of the noise source to be damped, but also other sound components - especially in frequency ranges in which the noise is weak or nonexistent - the adaptation of the direction-dependent damping can result in a perceptible fluctuation of useful signals or background noise in the sound signal output to the user. Such artifacts can considerably impair the hearing experience of the user under unfavorable circumstances and in extreme cases even cause a deterioration in speech understanding (instead of the desired improvement). In particular, a fluctuation of inherently static background noises caused by the adaptation of the direction-dependent attenuation can lead to an increased perception of these background noises and thus distract the user from concentrating on the actual useful signal. Particularly disturbing effects can be caused in particular by the fact that the notch of an adaptive beamformer jumps back and forth between different sources of background noise.

On, for example, off EP 2 908 550 B1 A known approach to at least partially remedying these problems is to detect a head movement of the user by means of an acceleration, direction or inclination sensor and to track the "viewing direction" of the beamformer (ie the directional lobe) in the opposite direction to the detected head movement. Alternatively, it is off EP 2 908 550 B1 known to increase the adaptation speed of an adaptive beamformer when a head movement of the user is detected by means of the sensor. Both approaches make it possible to design the beamformer with a comparatively low adaptation speed in the absence of head rotations, so that in this situation the risk of artifacts of the type described above is reduced.

In any case, these approaches do not help against artifacts of the adaptive direction-dependent damping that are caused by sources of background noise moving independently of the user's head (e.g. passing motor vehicles).

The application is based on the object of improving an adaptive, direction-dependent attenuation used in the operation of a hearing system (to modify the sound signal picked up from the environment and output to the user in modified form) with a view to avoiding artifacts. A direction-dependent attenuation is thus to be created which enables the user to hear better.

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 in themselves inventive embodiments or further developments of the invention are in the subclaims and the following description set out.

The invention is generally based on a hearing system for supporting the hearing ability of a user, the hearing system having at least one on the head, in particular has hearing instrument worn in or on one ear of the user. As described above, the hearing system in simple embodiments of the invention 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) belonging to the user and / or at least one other electronic device, for example a remote control or a charger. The hearing instrument and the at least one further component are in data exchange with one another in this case, the functions of data storage and / or data processing of the hearing system being divided between the hearing instrument and the at least one further component.

The hearing system has at least two input transducers, each of which is used to pick up a sound signal (in particular in the form of airborne sound) from the surroundings of the hearing instrument. The at least two input transducers can be arranged in the same hearing instrument; especially when the hearing system only comprises a single hearing instrument. In the case of a binaural hearing system with two hearing instruments, the at least two input transducers can alternatively also be distributed over the two hearing instruments.

The hearing system further comprises signal processing, with a signal processing unit for processing (modifying) the recorded sound signal in order to support the hearing ability of the user, as well as an output converter 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 within the scope of the invention can, however, also have a hearing instrument for the second ear that does not have an output transducer itself, but only picks up sound and - with or without signal processing - to the hearing instrument of the first ear forwards. Such so-called CROS or BiCROS instruments are used in particular for users with one-sided deafness. Furthermore, within the scope of the invention, the signal processing or a part thereof can also be outsourced from the hearing instrument or the hearing instruments to an external unit, for example an app running on a smartphone. The signal processing of the hearing system preferably includes, in addition to the signal processing unit, a signal analysis unit which itself does not 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.

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

Each of the input transducers is in particular an acousto-electrical transducer which converts airborne sound from the environment 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 noise or to directly stimulate the user's auditory nerve.

In the course of the method, the at least two input transducers of the hearing system record a sound signal from the surroundings of the user and convert it into input audio signals. The input audio signals are processed in a signal processing step to generate an output audio signal. This output audio signal is output by means of the output transducer of the hearing instrument. In the signal processing step, the input audio signals are fed directly (or indirectly after preprocessing) to a first adaptive beamformer, through which the input audio signals or the audio signals derived therefrom by the preprocessing (preprocessed Audio signals) are attenuated in a direction-dependent manner in accordance with a variable (first) directional characteristic with a predetermined (first) directional strength. The first adaptive beamformer generates a first directional audio signal that is output directly (or indirectly after one or more further signal processing steps) as the modified audio signal for output to the user on the electro-acoustic converter.

In an adaptation step, the directional characteristic of the first adaptive beamformer is varied as a function of a predetermined (first) adaptation speed in such a way that the energy content of the first directed audio signal is minimized.

The “directional characteristic” of a beamformer generally refers to the dependence of the attenuation carried out by the beamformer on the sound components of the recorded sound signal as a function of the direction from which these sound components are received.

The deviation of the directional characteristic from the omnidirectional characteristic is expressed in particular in the fact that the direction-dependent attenuation of the associated adaptive beamformer has at least one local maximum. This or each attenuation maximum of the directional characteristic is also referred to below as a “notch”, and the associated direction of maximum attenuation is also referred to as the “notch direction”.

In an expedient embodiment of the invention, the or each notch direction is defined in the form of an angle specification, for example relative to the viewing direction of the user. Alternatively, the or each notch direction can also be specified as an abstract variable that is linearly or non-linearly correlated with the alignment of the associated notch, for example in the form of a weighting factor with which various basic directional signals (e.g. a cardioid Signal and an anti-cardioid signal, etc.) can be weighted, or in the form of a variable time delay with which different signal components are superimposed on one another to generate the directivity.

The "directional strength" describes in general how much the directional characteristic of the associated adaptive beamformer deviates from an omnidirectional characteristic (that is, signal processing without directional dependence). The directional strength of the first adaptive beamformer cannot be changed in certain embodiments of the invention. In this case, the directional strength is particularly implicitly determined by the functional structure or the design of the first adaptive beamformer.

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 cannot be changed. In this case, the adaptation speed is particularly implicitly determined by the functional structure of the first adaptive beamformer.

According to the invention, however, at least one of the two above-described properties of the directional characteristic, namely the adaptation speed and / or the directional strength, of the first adaptive beamformer is not unchangeable, but is given to the first adaptive beamformer as a variable, i.e. as a variable parameter. The adaptation speed or the directional strength of the first adaptive beamformer is set variably (preferably by the signal analysis unit described above) on the basis of an analysis of the input audio signals or the preprocessed audio signals.

The variability of the adaptation speed and / or the directivity of the first adaptive beamformer can effectively avoid artifacts of the type described above. This makes it possible, in particular, to temporarily increase the adaptation speed and thus the adaptability of the first adaptive beamformer when changes in the background noise require a substantial adaptation of the first beamformer. That way you can In particular, 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 is alternatively or additionally achieved by temporarily reducing the leveling strength. On the other hand, the first adaptation speed can be set low in a static hearing situation, so that artificial fluctuations of useful signals or background noise due to small-scale adjustments of the directional characteristic of the first adaptive beamformer are avoided. In this case, a comparatively high directional strength enables good attenuation of interfering noises and consequently good perception of the useful signal by the user and thus in particular facilitates speech understanding in communication between the user and another speaker. The invention is based on the knowledge that the control of the adaptation speed and / or the directional strength of the first adaptive beamformer based on an analysis of the input audio signals or the preprocessed audio signals allows a particularly effective avoidance of artifacts of the directional attenuation.

The two measures described above, that is to say the variation of the adaptation speed of the first adaptive beamformer on the one hand and the variation of the directivity of the first adaptive beamformer on the other hand, fundamentally contribute independently of one another to achieving this effect. These measures can therefore be used separately from one another within the scope of the invention, in that either only the adaptation speed or only the directivity of the first adaptive beamformer are varied. In a preferred embodiment of the method, however, both the adaptation speed and the directivity of the first adaptive beamformer are varied.

The adaptation speed and / or the directivity of the first adaptive beamformer are preferably set as a function of a time stability of the input audio signals or the preprocessed audio signals (more precisely, as a function of a time stability of the background noise on which the input audio signals are based). In the case of variants of the method in which the adaptation speed of the first adaptive beamformer is varied, this is set to be low in particular in the case of high time stability (ie weak temporal change) of the respective audio signals and high in the case of low temporal stability (ie strong temporal change). This means that the directional characteristic of the first adaptive beamformer is adapted quickly in the case of a strongly changing background noise and slowly in the case of a weakly changing background noise. In the case of variants of the method in which the directivity of the first adaptive beamformer is varied, this is set high, in particular when the respective audio signals are highly stable in terms of time (ie low temporal change) and low when the respective audio signals are not so stable (ie strong temporal change). This has the result that the input audio signals or the preprocessed audio signals are directed strongly by the first adaptive beamformer when the background noise is slightly variable and weakly or even not at all when the background noise varies greatly. To determine the time stability, a variable characterizing the background noise is derived, for example, from the input audio signals or the preprocessed audio signals. The standard deviation of this variable or a quadratic mean value of the first time derivative of this variable over a sliding period is used as a measure of the time stability. Alternatively or additionally, the rate at which a moving average value is exceeded or not reached (mean crossing rate) or the rate at which the first derivative of this value changes sign can be used as a measure of the time stability of the input audio signals or the preprocessed audio signals are used.

In an advantageous embodiment of the invention, a second adaptive beamformer with a (second) variable directional characteristic is used to characterize the background noise. Like the first adaptive beamformer, this second adaptive beamformer is applied directly or indirectly to the input audio signals in order to generate a second directional audio signal. The directional characteristic of the second adaptive beamformer is set with a - preferably constant - (second) adaptation speed in such a way 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 maximum direction Attenuation (notch direction). In an expedient embodiment of the invention, however, the second directed audio signal is not included in the modified audio signal to be output to the user. The second adaptive beamformer is therefore only used in this case for the purpose of signal analysis. The second directed audio signal is only used as a control variable for energy minimization, and not for signal processing for output to the user.

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 on which the input audio signals are based. Compared to the second adaptive beamformer, the first adaptive beamformer either always adapts slowly or alternates between slow and fast adaptation.

In an advantageous embodiment of the method, the adaptation speed and / or the directional strength of the first adaptive beamformer are set variably as a function of the change in the second directional characteristic (that is, the directional characteristic of the second adaptive beamformer). As a parameter for the temporal change in the second directional characteristic and thus as a measure for the variability of the background noise, for example, a sliding square mean value of the first time derivative of a notch direction assigned to the second directional characteristic is determined. For example, the adaptation speed of the first adaptive beamformer is increased compared to a basic value and / or the directivity of the first adaptive beamformer is decreased compared to a basic value if and as long as the aforementioned parameter exceeds a predetermined threshold value.

As an alternative or in addition to this, the adaptation speed and / or the directional strength of the first adaptive beamformer are preferably set as a function of the deviation of the second directional characteristic from the first directional characteristic, in particular as a function of the deviation of a (second) notch direction assigned to the second directional characteristic from a (first) notch direction assigned to the first directional characteristic. 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 value.

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 directional strength, several adaptive beamformers are used, optionally also in terms of design) correspond in particular to the second adaptive beamformer. These multiple adaptive (analysis) beamformers are in particular coupled to one another, so that a different alignment of their respectively assigned directional characteristics (and possibly the associated notch directions) is enforced. It is thus achieved in particular that each of the multiple adaptive (analysis) beamformers is oriented towards a different dominant noise source in the user's surroundings. With such a cascade of analysis beamformers, the background noise on which the input audio signals are based and changes in this background noise can be analyzed with high precision.

The first adaptive beamformer preferably has a frequency dependency, that is to say it modifies different frequency components of the input audio signal or of the preprocessed audio signals in an individual manner. In particular, the input audio signals or the preprocessed audio signals are each divided into different frequency channels, the directional characteristic (in particular the or each notch direction) of the first adaptive beamformer for each frequency channel is individually adapted. In a preferred embodiment of the invention, the adaptation speed and / or the directional strength of the first adaptive beamformer are specified as a frequency-dependent variable (e.g. as a vector with an 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, or so that the directivity of the first adaptive beamformer is possibly differently pronounced for different frequencies.

For the frequency-dependent setting of the adaptation speed and / or the directional strength of the first adaptive beamformer, at least one noise component originating 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 is determined that corresponds to this interference noise component. The adaptation speed and / or the directional strength of the first adaptive beamformer are specified uniformly (i.e. with the same value) within the interference frequency range. In this way it is avoided that the first beamformer is adapted in different ways for different frequency components of one and the same background noise, which could lead to a tonal distortion and / or artificial fluctuation of the background noise.

The hearing system according to the invention is generally set up to automatically carry out the method according to the invention described above. For this purpose, the hearing system comprises the first adaptive beamformer (as described above). The hearing system further comprises an adaptivity control which is set up to variably set 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.

The setup of the hearing system for the automatic implementation of the method according to the invention is of a programming and / or circuitry nature. The hearing system according to the invention thus comprises programming means (Software) and / or circuitry (hardware, for example in the form of an ASIC) which automatically carry out the method according to the invention when the hearing system is in operation. The program-technical or circuit-technical means for performing the method, in particular the first adaptive beamformer and the adaptivity control, can in this case be arranged exclusively in the hearing instrument (or the hearing instruments) of the hearing system. Alternatively, the programming or circuitry means for performing the method are distributed among the hearing instrument or the hearing aids and at least one further device or a software component of the hearing system. For example, programming means for performing the method are distributed to the at least one hearing instrument of the hearing system and to a control program installed on an external electronic device (in particular a smartphone). As mentioned above, the external electronic device is usually not itself part of the hearing system.

The embodiments of the method according to the invention described above correspond to corresponding embodiments of the hearing system according to the invention.

Thus, in preferred embodiments of the invention, the adaptivity control is set up to set the adaptation speed and / or the directivity of the first adaptive beamformer as a function of the time stability of the input audio signals.

Preferably, the adaptivity control for analyzing the input audio signals or the preprocessed audio signals (ie for characterizing the underlying background noise) and for determining the adaptation speed and / or the directivity of the first adaptive beamformer comprises a second adaptive beamformer or a cascade of further adaptive ones (especially those that are coupled to one another) Beamformer as described above.

The adaptivity control is set up in particular to vary the adaptation speed and / or the directivity of the first adaptive beamformer of the change in the (respective) directional characteristic of the second beamformer (and possibly the further adaptive beamformer) and / or depending on the deviation of the directional characteristics of the adaptive beamformer.

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 control is preferably set up to specify the adaptation speed and / or the directional strength of the first adaptive beamformer as a frequency-dependent variable, for setting the adaptation speed and / or the directional strength of the first adaptive beamformer, a noise component in the input audio signals originating from a noise source or in the identify preprocessed audio signals, determine an interference frequency range corresponding to the interference noise component, and uniformly specify the adaptation speed and / or the directivity of the first adaptive beamformer in the interference frequency range.

Effects and advantages of the individual method variants can be transferred to the corresponding variants of the hearing system and vice versa.

Fig. 1

in einer schematischen Darstellung ein aus einem einzelnen Hörinstrument bestehendes Hörsystem in Form eines hinter einem Ohr eines Nutzers tragbaren Hörgeräts,

Fig. 2 bis 4

jeweils in einem schematischen Blockschaltbild den Aufbau einer Signalverarbeitung des Hörsystems aus Fig. 1 in drei alternativen Ausführungsformen,

Fig. 5

in Darstellung gemäß Fig. 2 bis 4 eine als Adaptivitätssteuerung bezeichnete Funktionseinheit der Signalverarbeitung des Hörsystems in einer weiteren Ausführungsform, und

Fig. 6

in Darstellung gemäß Fig. 1 eine alternative Ausführungsform des Hörsystems, in dem dieses ein Hörinstrument in Form eines hinter dem Ohr tragbaren Hörgeräts sowie ein in einem Smartphone implementiertes Steuerprogramm ("Hör-App") umfasst.

Exemplary embodiments of the invention are explained in more detail below with reference to a drawing. Show in it:

Fig. 1

a schematic representation of a hearing system consisting of a single hearing instrument in the form of a hearing aid that can be worn behind an ear of a user,

Figs. 2 to 4

each in a schematic block diagram of the structure of a signal processing system of the hearing system Fig. 1 in three alternative embodiments,

Fig. 5

in representation according to Figs. 2 to 4 a functional unit, referred to as adaptivity control, of the signal processing of the hearing system 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 implemented in a smartphone (“hearing app”).

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

Fig. 1 shows a hearing system 2, which here consists of a single hearing aid 4, ie a hearing instrument set up to support the hearing ability of a hearing-impaired user. In the example shown here, the hearing aid 4 is a BTE hearing aid that can be worn behind an ear of a user.

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, with regard to its structure, is particularly similar to that in Fig. 1 Hearing aid 4 shown corresponds.

The hearing aid 4 comprises two microphones 6 as acoustoelectric input transducer and a receiver 8 (receiver) as an electroacoustic output transducer within a housing 5. The hearing aid 4 further comprises a battery 10 and signal processing in the form of a signal processor 12. The signal processor 12 preferably comprises both a programmable subunit (for example a microprocessor) and a non-programmable subunit (for example an ASIC).

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

During normal operation of the hearing aid 4, the microphones 6 each record an airborne sound from the surroundings of the hearing aid 4. The microphones 6 convert the sound into an (input) audio signal I1 or I2, 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 hearing ability of the user.

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

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

The structure of the signal processing is in Fig. 2 shown in more detail. It can be seen from this 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 is used to generate the output audio signal O from the input audio signals I1, I2 of the microphones 6 or from this, derived from preprocessing and therefore preprocessed audio signals I1, I2 '. In the first-mentioned 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 As shown, 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 therefrom and feeds them to the signal conditioning unit 18.

In the preprocessing unit 22, the input audio signals I1, I2 are preferably time-delayed from one another to form the preprocessed audio signals I1, I2 superimposed so that the two preprocessed audio signals I1, I2 correspond to a cardioid signal and an anti-cardioid signal.

The signal processing unit 18 comprises a number of signal processing processes 24 which the input audio signals I or - in the example according to Figure 2 Process the internal audio signals I1, I2 successively and thereby modify them in order to generate the output audio signal O and thus to compensate for the hearing loss of the user.

• einen Prozess zur Störgeräusch- und/oder Rückkopplung-Unterdrückung,
• einen Prozess zur dynamischen Kompression und
• einen Prozess zur frequenzabhängigen Verstärkung basierend auf Audiogramm-Daten,
• etc.

The signal conditioning processes 24 are optionally implemented in any 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 conditioning processes 24 is formed by a hardware circuit, at least one further of the signal conditioning processes 24 is formed by a software module, and in turn another of the signal conditioning processes 24 is formed by a combination of hardware and software components. The signal conditioning processes 24 include, for example

• a process for noise and / or feedback suppression,
• a dynamic compression process and
• a process for frequency-dependent amplification based on audiogram data,
• Etc.

At least one of these signal conditioning processes 24 (generally all signal conditioning processes 24 or at least most signal conditioning processes 24) is assigned at least one signal conditioning parameter P in each case. The or each signal processing parameter P is a one-dimensional variable (binary variable, natural number, floating point number, etc.) or a multidimensional variable (array, function, etc.), its value the mode of operation of the respectively assigned signal application process 24 is parameterized (ie influenced). Signal conditioning parameters P can switch the respectively assigned signal conditioning process 24 on or off, continuously or gradually strengthen or weaken the effect of the respectively assigned signal conditioning process 24, define time constants for the respective signal conditioning process 24, etc.

• die vorstehend genannten Audiogrammdaten oder hieraus abgeleitete frequenzspezifische Verstärkungsfaktoren für einen Prozess zur frequenzabhängigen Verstärkung,
• eine Kennlinie für einen Prozess zur dynamischen Kompression,
• eine Steuergröße zur kontinuierlichen Einstellung der Stärke eines Prozesses zur Störgeräusch- bzw. Rückkopplungsunterdrückung,
• etc.

For example, the signal conditioning parameters include P

• the aforementioned audiogram data or frequency-specific gain factors derived therefrom for a process for frequency-dependent gain,
• a characteristic curve for a process for dynamic compression,
• a control variable for continuously setting the strength of a process for noise or feedback suppression,
• Etc.

In any case, some of the signal conditioning parameters P are made available to the signal conditioning processes 24 from a parameterization unit 26.

Furthermore, the signal conditioning processes 24 include an - in Fig. 2 shown in more detail - first adaptive beamformer 28, which is set up to receive the input audio signals I1, I2 (or, as in Fig. 2 shown, to attenuate the preprocessed audio signals I1, I2 ') depending on the direction according to a variable (first) directional characteristic and in this way to generate a first directional audio signal R1. The beamformer 28 generates the audio signal R1 by generating the two supplied audio signals I1, I2 '(in the example according to FIG Fig. 2 i.e. a cardioid signal and an anti-cardioid signal) weighted on each other with a first weighting factor a1 superimposed: $$\mathrm{R.}\mathrm{1}=\mathrm{I.}\mathrm{1}\mathrm{ʹ}-\mathrm{a}\mathrm{1}\cdot \mathrm{I.}\mathrm{2}\mathrm{ʹ\; with\; a}=\left[-\mathrm{1};\mathrm{1}\right]$$

The weighting factor a1 determines a notch direction in which - viewed relative to the head of the user - the direction-dependent attenuation of the beamformer 28 has a (local) maximum. The weighting factor a1 thus represents a measure for the notch direction of the beamformer 28 and is therefore equated conceptually 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 method in such a way that the energy content of the directional audio signal R1 is minimized (this self-regulation of the beamformer 28 is shown in FIG Fig. 2 shown schematically by feeding back the audio signal R1 to the beamformer 28). The energy minimization described ensures that interference from a room area behind the user's head is suppressed as well as possible. The directed audio signal R1 output by the beamformer 28 is further processed by the further signal processing processes 24, as a result of which the output audio signal O is generated. The beamformer 28 is preferably formed by a software module.

A first adaptation speed v1 is variably predefined for the beamformer 28 as the signal processing 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.

In the in Fig. 2 In the illustrated embodiment, the adaptivity control 30 comprises a second adaptive beamformer 32 and an evaluation module 34.

In terms of structure and function, the second adaptive beamformer 32 preferably corresponds to the first adaptive beamformer 28. The second adaptive beamformer 32 is thus set up in the manner described above to transmit the input audio signals I1, I2 (or, as in Fig. 2 shown, to attenuate the preprocessed audio signals I1 ', I2') depending on the direction according to a (second) variable directional characteristic and in this way a second directional signal To generate R2. The directional characteristic of the beamformer 32 preferably has a notch direction which is characterized by a variable weighting factor a2. The weighting factor a2 (and thus the notch direction) are varied by the beamformer 32 with an adaptation speed v2 in such a way that the energy content of the directed audio signal R2 is minimized.

In contrast to the beamformer 28, the beamformer 32 is not used to generate the output audio signal O output to the user, but only to analyze the background noise on which the input signals I1, I2 are based. The audio signal R2 is therefore not processed further, but only fed back to the beamformer 32 for the purpose of self-regulation. Instead, the beamformer 32 outputs the weighting factor a2 indicating the notch direction (and thus indirectly the arrangement of the most dominant noise sources in the vicinity of the user) to the evaluation module 34 as the analysis result.

In the evaluation module 34 in the embodiment according to Fig. 2 the time stability (or - to put it the other way round - the temporal variability) of the weighting factor a2 and thus the background noise are evaluated, for example by forming a sliding square time mean value over the first time derivative of the weighting factor a2. The evaluation module 34 varies the adaptation speed v1 for the first adaptive beamformer 28 as a function of this variable. In a simple but expedient variant, the evaluation module 34 varies the adaptation speed v1 in binary between a comparatively low base value and a comparatively increased value. The evaluation module 34 sets the adaptation speed v1 to the basic value if and as long as the mean value described above does not exceed a predetermined threshold value (which indicates that the background noise cannot or only slightly changes). The first beamformer 28 thus adapts only slowly in this case, as a result of which artifacts as a result of the adaptation are largely avoided. Otherwise, that is, if and as long as the mean value exceeds the threshold value 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 changes 28 can adapt quickly (especially without a perceptible delay) to the changed hearing situation.

In order to analyze the background noise with high precision, the second adaptive beamformer 32 is designed to be fast-adapting. The adaptation speed v2 is selected (preferably as a constant) in such a way that it never falls below the variable adaptation speed v1 of the first adaptive beamformer 28 (v2 v1).

In addition or as an alternative to the adaptation speed v1, a directional strength s of the first adaptive beamformer 28 can preferably also be varied. The variation of the directional strength s is implemented, for example, in that the weighted sum according to Eq. 1 is mixed to varying degrees with an omnidirectional audio signal A derived from the input audio signals I1, I2 (which the beamformer 28 according to FIG Fig. 2 is optionally supplied as an additional input variable). The directional strength s is reduced by the evaluation module 34 compared to a predetermined basic value if and for as long as - in particular on the basis of the above-described exceeding of the threshold value - a strong variability of the background noise is determined.

How out Fig. 2 Furthermore, as can be seen, the signal analysis unit 20 contains, in addition to the adaptivity control 30 and preferably in addition to other functions for sound analysis that are not explicitly shown here, optionally a classifier 36, which is generated in a conventional manner by analyzing the input audio signals I1, I2 (or, as in Fig. 2 shown, the preprocessed audio signals I1, I2 ') analyzes 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 outputs a corresponding classification signal K.

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

On the other hand, the classification signal K is also fed to the evaluation module 34 of the adaptivity control 30 and here influences the determination of the adaptation speed v1 and / or the directional strength s.For example, the values between which the adaptation speed v1 and / or the directional strength s is varied are again dependent changed by the classification signal K.

In Fig. 3 an alternative embodiment of the hearing system 2 is shown. In contrast to the embodiment according to FIG Fig. 2 is the evaluation module 34 in the execution 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 here analyzes the variability of the background noise on which the input audio signals I1, I2 and the audio signals I1, I2 'are based, 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 basic state is seen as an indication of a significant change in the background noise. Correspondingly, the evaluation module 34 increases the adaptation speed v1 and / or reduces the directional strength s if and as long as the difference between the weighting factors a1 and a2 exceeds a predetermined threshold value.

In Fig. 4 a further embodiment of the hearing system 2 is shown. In contrast to the embodiments according to Fig. 2 and 3 In addition to the second adaptive beamformer 28, the adaptivity control 30 here comprises at least one further adaptive beamformer 28, which generates a further directional audio signal R3 and, due to an energy minimization of this audio signal R3, an assigned further weighting factor a3 (as a measure for a variable notch direction of the beamformer 38 ) varies.

An adaptation speed v3 assigned to the beamformer 38 and preferably given constant has, in an expedient embodiment variant, a value which is below 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, with both beamformers 32 and 38 preferably setting the respective weighting factor a2 and a3 independently of one another (a coupling of the beamformers 32 and 38, as they in Fig. 4 is indicated on the basis of 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 on which the input audio signals I1, I2 and the preprocessed audio signals I1, I2 'are based is determined by the evaluation module 34 in a manner analogous to the exemplary embodiment according to FIG Fig. 3 determined on the basis of the deviation of the weighting factors a2 and a3 of the beamformer 32 and 38.

In an alternative embodiment variant, the adaptation speeds v2 and v3 of the beam formers 32 and 38 are selected to be exactly or approximately the same, so that both beam formers 32 and 38 adapt quickly. In this case, the beam formers 26 and 38 are preferably - as in FIG Fig. 4 indicated - coupled with one another, so that a different setting of the weighting factors a2 and a3 is forced. This coupling ensures that the beamformers 26 and 38 adjust to different dominant noise sources in the user's surroundings. The variability of the background noise on which the input audio signals I1, I2 and the preprocessed audio signals I1, I2 'are based is in this case preferably analogous to the exemplary embodiment according to the evaluation module 34 in this case Fig. 2 determined on the basis of the time stability of the weighting factors a2 and a3. The adaptation speed v1 is in particular increased and / or the directional strength s is reduced if the condition for increasing the adaptation speed v1 or decreasing the directional strength s is met for at least one of the weighting factors a2 and a3.

The classifier 36 is optionally also in accordance with the exemplary embodiments Fig. 3 and 4th present and not shown in these figures merely for the sake of clarity.

The signal processing in the signal conditioning unit 18 preferably takes place in a frequency-resolved manner in a plurality of (for example 64) frequency channels. The input audio signals I1, I2 are thereby, preferably before being fed to the preprocessing unit 22, by a (in the Figs. 2 to 4 Analysis filter bank (not explicitly shown) is divided into frequency components, which are processed individually in the frequency channels and then in a (in the Figs. 2 to 4 Synthesis filter bank can also be combined to form the output audio signal O, which is also not explicitly shown.

The first adaptive beamformer 28 is set up to attenuate the frequency components of the input audio signals I1, I2 or of the preprocessed audio signals I1, I2 'carried in the frequency channels, in each case individually depending on the direction. The directional characteristic of the beamformer 28 and the assigned notch direction or the weighting factor a1 thus also have a frequency dependency. The weighting factor a1 and / or the directional strength s are preferably in each case specified as a vector which has an assigned individual value for each frequency channel. Furthermore, the directional characteristic of the beamformer 28 is preferably also adapted individually for each frequency channel. The adaptation speed v1 is thus also preferably specified as a vector which has an assigned individual value for each frequency channel.

In order to prevent interfering noise originating from a certain noise source from being perceptibly distorted by the beamformer 28 as a result of the different frequency-specific adaptation of the directional characteristic, the adaptivity control 30 is preferably set up to convert frequency channels that carry essential frequency components of a dominant interfering noise with regard to the adaptation speed v1 and / or the directional strength s to couple. The adaptivity control In other words, 30 specifies the adaptation speed v1 and / or the directivity s uniformly (ie with the same value) for those frequency channels which carry essential frequency components of a dominant background noise.

For this purpose, the second adaptive beamformer 32 (and, if applicable, the third adaptive beamformer 38) are also constructed analogously to the beamformer 28 in such a way that they transmit the frequency components of the input audio signals I1, I2 or the preprocessed audio signals I1, I1, which are carried in the frequency channels. Attenuate I2 'individually depending on the direction. The background noise is thus analyzed in a frequency-resolved manner by the second adaptive beamformer 32 (and possibly the third adaptive beamformer 38).

In order to determine the spectral composition of one or more dominant interference noises, the directed audio signal R2 (or R3 in each case) output by the second adaptive beamformer 32 (and possibly the third adaptive beamformer 38) is inverted in an inverter 40 and then in a multiplier 42 is multiplied by the omnidirectional audio signal A. This signal processing is in Fig. 5 shown by way of example for an embodiment of the adaptivity control 30, which is analogous to FIG Fig. 4 includes both the second adaptive beamformer 32 and the third adaptive beamformer 38. The multiplication of the omnidirectional audio signal A with the inverted, directional audio signal R2 (or R3) results in an audio signal R2 '(or R3') in which the dominant background noise that leads to the second adaptive beamformer 32 (or possibly the third adaptive beamformer 38) was selectively filtered out, is being selectively amplified. The audio signal R2 '(or, if applicable, R3') is now 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 interfering noise. The frequency channels in this interference frequency range are coupled by the evaluation module 34 with regard to the adaptivity of the first adaptive beamformer 28 by the evaluation module 34 uniformly specifying the values of the adaptation speed v1 and / or the directional strength s corresponding to these frequency channels: For example, the Adaptation speed v1 for all coupled frequency channels increased compared to the basic value and / or the directional strength s decreased for all coupled frequency channels compared to the basic value, if and as long as (from the evaluation module 34 according to Fig. 2 or 4 evaluation of the weighting factor a2 or the weighting factors a2 and a3) shows that the condition for increasing the adaptation speed v1 or decreasing the directional strength s is fulfilled for at least one of the coupled frequency channels.

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

The hearing aid 4 and the hearing app 46 exchange data via a wireless data transmission connection 48 when the hearing system 2 is in operation. 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 in order to receive data from the hearing device 4 and to send data to it. The hearing aid 4 for its part comprises a Bluetooth transceiver (not explicitly shown) in order to send data to the hearing app 44 and to receive data from this app.

In the execution according to Fig. 6 are parts of the Figures 2 to 5 The signal processing shown (for example the adaptivity control 30) is not implemented in the signal processor 12 of the hearing aid 4, but rather in the hearing app 44.

The invention is particularly clear from the exemplary embodiments described above, but is nevertheless not limited to these exemplary embodiments. Rather, further embodiments of the invention can be derived by a person skilled in the art from the claims and the above description will. In particular, the individual features of the hearing system and the associated operating method explained in each case with the aid of 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 aid

4th

Hearing aid

5

casing

6th

microphone

8th

Listener

10

battery

12th

Signal processor

14th

Sound channel

16

top

18th

Signal conditioning unit

20th

Signal analysis unit

22nd

Preprocessing unit

24

Signal conditioning process

26th

Parameterization unit

28

(first adaptive) beamformer

30th

Adaptivity control

32

(second adaptive) beamformer

34

Evaluation module

36

Classifier

38

(third adaptive) beamformer

40

Inverting element

42

Multiplier

44

Hearing app

46

Smartphone

48

Data transfer connection

a1

(first) weighting factor

a2

(second) weighting factor

a3

(third) weighting factor

s

Guide 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 (12)

Method for operating a hearing system (2) to support the hearing ability of a user, with at least one hearing instrument (4) worn on the head, in particular in or on an ear of the user, - with at least two input transducers (6) of the hearing system (4) recording a sound signal from the surroundings of the user and converting it 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), - The output audio signal (O) being 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 (I1 ', I2') derived therefrom by preprocessing are attenuated by means of a first adaptive beamformer (28) according to a first variable directional characteristic with a directional strength (s) depending on the direction 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) in such a way that the energy content of the first directional audio signal (R1) is minimized, and - the first adaptation speed (v1) and / or the directional strength (s) being variably set based on an analysis of the input audio signals (I1, I2) or the preprocessed audio signals (I1 ', I2'). Method according to claim 1,
the first adaptation speed (v1) and / or the directional strength (s) being set as a function of a time stability of the input audio signals (I1, I2) or of the preprocessed audio signals (I1 ', I2'). Method according to claim 1 or 2, - wherein for setting the first adaptation speed (v1) and / or the directional strength (s) for the first adaptive beamformer (28) a second adaptive beamformer (32) with a second variable directional characteristic on the input audio signals (I1, I2) or the preprocessed audio signals (I1 ', I2') are used to generate a second directional audio signal (R2), - wherein the second variable directional characteristic of the second adaptive beamformer (32) is set with a second adaptation speed (v2) so that the energy content of the second directional audio signal (R2) is minimized, - the second adaptation speed (v2) not falling below the first adaptation speed (v1) and at least temporarily exceeding it. Method according to claim 3,
wherein the first adaptation speed (v1) and / or the directional strength (s) are set variably as a function of a change in the second directional characteristic. Method according to claim 3 or 4,
wherein the first adaptation speed (v1) and / or the directional strength (s) are set as a function of a deviation of the second directional characteristic from the first directional characteristic. 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 preprocessed audio signals (I1 ', I2') are attenuated in an individual manner depending on the direction, - the first adaptation speed (v1) and / or the directional strength (s) for the first adaptive beamformer (28) being specified as a frequency-dependent variable, - In order to set the first adaptation speed (v1) and / or the directional strength (s), a noise component originating from a noise source is identified in the input audio signals (I1, I2) or the preprocessed audio signals (I1 ', I2'), - An interference frequency range corresponding to the interference noise component is determined, and - The first adaptation speed (v1) and / or the directional strength (s) are uniformly specified in the interference frequency range. Hearing system (2) for supporting the hearing ability 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 set up to pick up a sound signal from the surroundings of the user and to convert it into input audio signals (I1, I2), - wherein the hearing system (2) has a signal processing unit (18) which is set up 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 set up to output the output audio signal (O), - wherein the signal processing unit (18) comprises a first adaptive beamformer (28) which is set up to convert the input audio signals (I1, I2) or audio signals (I1 ', I2') derived therefrom by preprocessing according to a first variable directional characteristic to attenuate with a directional strength (s) depending on the direction in order to generate a first directional audio signal (R1), and to vary the first directional characteristic with a first adaptation speed (v1) in such a way 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 set up to determine the first adaptation speed (v1) and / or the directional strength (s) based on an analysis of the input audio signals (I1, I2) or the preprocessed audio signals ( I1 ', I2') can be set variably. Hearing system (2) according to claim 7,
wherein the adaptivity control (30) is set up to set the first adaptation speed (v1) and / or the directional strength (s) as a function of a time stability of the input audio signals (I1, I2) or the preprocessed audio signals (I1 ', I2') . 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 fed, - wherein the second adaptive beamformer (32) is set up to generate a second directional audio signal (R2) and to set the second variable directional characteristic with a second adaptation speed (v2) so that the energy content of the second directional audio signal (R2) is minimized, - the second adaptation speed (v2) not falling below the first adaptation speed (v1) and at least temporarily exceeding it. Hearing system (2) according to claim 9,
wherein the adaptivity control (30) is set up to variably set the first adaptation speed (v1) and / or the directional strength (s) as a function of a change in the second directional characteristic. Hearing system (2) according to claim 9 or 10,
wherein the adaptivity control (30) is set up to set the first adaptation speed (v1) and / or the directional strength (s) as a function of a deviation of the second directional characteristic from the first directional characteristic. Hearing system (2) according to one of claims 7 to 11, - The first directional characteristic being frequency-dependent, so that different frequency components of the input audio signals (I1, I2) or the preprocessed audio signals (I1 ', I2') are attenuated in different ways depending on the direction, - wherein the adaptivity control (30) is set up to - to specify the first adaptation speed (v1) and / or the directional strength (s) as a frequency-dependent variable, - to set the first adaptation speed (v1) and / or the directional strength (s) to identify a noise component originating from a noise source in the input audio signals (I1, I2) or the preprocessed audio signals (I1 ', I2'), - to determine an interfering frequency range corresponding to the interfering noise component, and - to uniformly specify the first adaptation speed (v1) and / or the directional strength (s) in the interference frequency range.

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