Hearing system having at least one hearing device and method for operating the same

Abstract

The present invention relates to a hearing system for supporting the hearing of a user, comprising at least one hearing device worn on the head of the user. It also relates to a method for operating a hearing system. Sound signals from the user's surroundings are received by means of at least two input converters of the hearing system and converted into input audio signals. The input audio signal is processed in a signal preparation step to generate an output audio signal, which is output by means of the output converter. In the signal preparation step, the input audio signal or the audio signal derived therefrom by preprocessing is attenuated in a direction-dependent manner according to a variable directional characteristic using a directional strength by means of a first adaptive beam former, so as to generate a directional audio signal. The directional characteristic is changed using a preset adaptive speed, thereby minimizing the energy content of the directional audio signal. The adaptive speed and / or the directional strength are variably adjusted based on an analysis of the input audio signal or the preprocessed audio signal.

Description

Technical Field

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

[0002] Electronic devices that support the hearing of a person wearing a hearing device (hereinafter referred to as a "wearer" or "user") are generally referred to as hearing devices. The present invention relates in particular to a hearing device that is designed to fully or partially compensate for the hearing loss of a hearing-impaired user. Such hearing devices are also referred to as "hearing aids". There are also hearing devices that protect or improve the hearing of a user with normal hearing, for example to enable improved speech understanding in complex hearing situations.

[0003] In general, hearing devices and hearing aids in particular are mostly designed to be worn on the head and in particular in or on the ear of a user, in particular as behind-the-ear devices (according to the English terminology: "behind the ear", also referred to as BTE devices) or in-the-ear devices (according to the English terminology "in the ear", also referred to as ITE devices). With regard to their internal structure, hearing devices generally have at least one (acoustic-electrical) input transducer, a signal processing unit (signal processor) and an output transducer. When the hearing device is in operation, the or each input transducer receives air sound from the surroundings of the hearing device and converts the air sound into an input audio signal (i.e. an electrical signal that transmits information about the ambient sound). The at least one input audio signal is subsequently also referred to as a "received sound signal". In the signal processing unit, the or each input audio signal is prepared (i.e. modified with respect to its sound information) in order to support the hearing of the user, in particular to compensate for the hearing loss of the user. The signal processing unit outputs the correspondingly prepared audio signal (also referred to as an "output audio signal" or "modified sound signal") to the output transducer.

[0004] In most cases, the output converter is constructed as an electroacoustic converter, which in turn converts the (electric) output audio signal into air sound, wherein the air sound (modified relative to the ambient sound) is output to the user's ear canal. In hearing devices worn behind the ear, the output converter, also known as the "earpiece" ("receiver"), is mostly integrated in the housing of the hearing device outside the ear. In this case, the sound output by the output converter is guided to the user's ear canal by means of a sound hose. Alternatively, the output converter can also be arranged in the ear canal, and therefore arranged outside the housing worn behind the ear. This type of hearing device (according to the English term "receiver in canal", the receiver is in the ear canal) is also called a RIC device. A hearing device worn in the ear (according to the English term "completely in canal", completely in the ear canal) is also called a CIC device, which is designed to be very small so that it does not protrude outward from the ear canal.

[0005] In a further embodiment, the output transducer can also be designed as an electromechanical transducer, which converts the output audio signal into structure-borne sound (vibrations), wherein the structure-borne sound is output, for example, into the skull of the user. In addition, there are implantable hearing devices, in particular cochlear implants, and hearing devices whose output transducers directly stimulate the auditory nerve of the user.

[0006] The term "hearing system" refers to a single device or a group of devices, and if necessary, to a non-physical functional unit, which together provide the functions required when the hearing device is in operation. In the simplest case, the hearing system can be composed of a single hearing device. Alternatively, the hearing system can include two cooperating hearing devices for taking care of the two ears of the user. In this case, it is said to be a "binaural hearing system". In addition or alternatively, the hearing system can include at least one other electronic device for the or each hearing aid, such as a remote control device, a charging device or a programming device. In modern hearing systems, a control program, especially in the form of a so-called App (application), is often provided instead of a remote control device or a dedicated programming device, wherein the control program is constructed for implementation on an external computer, especially a smartphone or a tablet computer. The external computer is usually not part of the hearing system itself, because it is usually independent of the hearing system and is not provided by the manufacturer of the hearing system.

[0007] In order to attenuate interfering noise during operation of a hearing system, and therefore in particular to improve speech understanding in communication between a user and other speakers, a direction-dependent attenuation (beamforming) of an input audio signal is usually used within the scope of signal processing in a hearing system, by means of which components of the input audio signal from different directions are attenuated to different degrees according to a preset directional characteristic. Usually, the directional characteristic has one or more directions of maximum attenuation, which are also referred to as a notch or a plurality of notches. In modern hearing systems, the corresponding attenuation unit (beamformer) is sometimes designed to be adaptive. Such an adaptive beamformer automatically changes its directional characteristic in order to optimally attenuate interfering noise. In particular, the notch is oriented toward the main interfering noise source as necessary in order to particularly effectively attenuate the sound component emanating from this interfering noise source.

[0008] In order to be able to follow interfering noise sources (e.g., passing motor vehicles) relative to the user's head movement, the adaptive beamformer is usually implemented with a relatively high adaptation speed. A high adaptation speed is also important in order to be able to compensate for the user's head movement, since the user's head movement also causes the interfering noise sources in the user's surroundings (from the user's perspective) to move relative to the user's head. The adaptation speed is usually designed to be very high so that the adaptive beamformer can automatically reorient itself in the opposite direction of the head rotation without noticeable delay when the head turns, and thus maintains the orientation toward a specific interfering noise source during and after the head rotation.

[0009] Disadvantageously, however, such fast adaptive beamformers are often prone to negative effects (artifacts) in dynamic listening situations, which are perceived by the user as unnatural. Since the direction-dependent attenuation, in addition to the noise of the interfering noise source to be attenuated, always also affects other sound components, in particular sound components in frequency ranges where the interfering noise is only weakly present or even absent, the adaptation of the direction-dependent attenuation can lead to perceptible fluctuations of the useful signal or background noise in the sound signal output to the user. In unfavorable cases, such artifacts can significantly impair the user's auditory perception and, in extreme cases, can even lead to a deterioration of speech understanding (instead of the expected improvement). In particular, the fluctuations of the inherently static background noise caused by the adaptation of the direction-dependent attenuation can lead to a higher perception of the background noise and thus distract the user from the actual useful signal. The notches of the adaptive beamformer that jump back and forth between different background noise sources can lead to particularly disturbing effects.

[0010] For example, EP 2 908 550 B1 discloses a method for at least partially eliminating these problems by detecting the head movement of the user by means of an acceleration sensor, an orientation sensor or an inclination sensor and tracking the "line of sight" of the beamformer (i.e. the directional lobe) inversely to the detected head movement. Alternatively, EP 2 908 550 B1 discloses increasing the adaptation speed of the adaptive beamformer if the head movement of the user is detected by means of a sensor. Both methods make it possible to design the beamformer with a relatively low adaptation speed without a rotation of the head, thereby reducing the risk of artifacts of the aforementioned type in this case.

[0011] In any case, however, these methods do not help to combat artifacts of the adaptive direction-dependent attenuation, which are caused by interfering noise sources that move independently of the user's head, such as passing motor vehicles. Summary of the invention

[0012] The technical problem to be solved by the present application is to improve the adaptive direction-dependent attenuation used during the operation of a hearing system (in order to modify the sound signal received from the environment and to be output to the user in a modified form) to avoid artifacts. Therefore, a direction-dependent attenuation should be provided, which can achieve an improvement in the user's auditory experience.

[0013] According to the invention, in terms of the method, this technical problem is solved by the features of the invention. According to the invention, in terms of the hearing aid system, this technical problem is solved by the features of the invention. Advantageous and partially inventive embodiments or improvements of the invention are explained in the following description.

[0014] The present invention generally relates to a hearing system for supporting the hearing of a user, wherein the hearing system has at least one hearing device worn on the user's head, in particular in or on the ear. As described above, in a simple embodiment of the present invention, the hearing system can consist of only a single hearing device. In another embodiment of the present invention, the hearing system includes at least one other component in addition to the hearing device, such as another (especially similar) hearing device for taking care of the user's other ear; a control program (especially in the form of an App) for implementation on an external computer (especially a smartphone) of the user; and / or at least one other electronic device, such as a remote control device or a charging device. In this case, the hearing device and at least one other component exchange data with each other, wherein the functions of data storage and / or data processing of the hearing system are divided between the hearing device and the at least one other component.

[0015] The hearing system has at least two input transducers for receiving sound signals (especially in the form of air sound) from the surroundings of the hearing device, respectively. The at least two input transducers can be arranged in the same hearing device, especially when the hearing system only includes a single hearing device. In a binaural hearing system with two hearing devices, the at least two input transducers can also be distributed alternately to the two hearing devices.

[0016] The hearing system further comprises a signal processing device and an output converter, wherein the signal processing device has a signal preparation unit for preparing (modifying) the received sound signal to support the hearing of the user, and the output converter is used to output the modified sound signal. In a binaural hearing system, the two hearing devices preferably have a signal preparation unit and an output converter respectively. Within the scope of the present invention, instead of a second hearing device having an input converter, a signal preparation unit and an output converter, the hearing system can also have a hearing device for the second ear, which does not have an output converter itself, but only receives the sound and further guides it to the hearing device of the first ear with or without a signal processing device. Such so-called CROS or BiCROS devices are particularly used in users with unilateral deafness. In addition, within the scope of the present invention, the signal preparation device or a part thereof can also be transferred from one hearing device or multiple hearing devices to an external unit, such as an App running in a smartphone. The signal processing device of the hearing system preferably includes a signal analysis unit in addition to the signal preparation unit, which does not itself generate an audio signal to be output directly or indirectly to the user, but supports the function of the hearing system, especially the signal preparation unit, by analyzing the audio signal or other sensor signals.

[0017] The or each hearing device of the hearing system is present in particular in the form of one of the structural forms described at the outset (BTE device with internal or external output transducer, ITE device, for example CIC device, hearing implant, in particular cochlear implant, etc.) or as a hearing device. In the case of a binaural hearing system, the two hearing devices are preferably constructed similarly.

[0018] Each input transducer is in particular an acoustic-electric transducer which converts airborne sound from the surroundings into an electrical input audio signal. The or each output transducer is preferably designed as an electroacoustic transducer (earpiece) which in turn converts the audio signal modified by the signal conditioning unit into airborne sound. Alternatively, the output transducer is designed for outputting structure-borne sound or for directly stimulating the auditory nerves of a user.

[0019] During the method, sound signals from the user's surroundings are received by at least two input transducers of the hearing system and converted into input audio signals. The input audio signals are processed in a signal preparation step to generate output audio signals. The output audio signals are output by means of an output transducer of the hearing device. In the signal preparation step, the input audio signals are directly (or indirectly after preprocessing) fed to a first adaptive beamformer, by which the input audio signals or the audio signals derived therefrom by preprocessing (preprocessed audio signals) are directionally attenuated according to a variable (first) directional characteristic using a preset (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 preparation steps) as a modified audio signal to the electroacoustic transducer for output to the user.

[0020] In the adaptation step, the directional characteristics of the first adaptive beamformer are changed according to a preset (first) adaptation speed so as to minimize the energy content of the first directional audio signal.

[0021] In general, the “directional characteristic” of a beamformer indicates a dependency of attenuation of a sound component of a received sound signal performed by the beamformer on a direction from which the sound component is received.

[0022] The deviation of the directional characteristic from the omnidirectional characteristic is manifested in particular in that the direction-dependent attenuation of the relevant adaptive beamformer has at least one local maximum. This or each attenuation maximum of the directional characteristic is also referred to as a “notch” in the following text, and the relevant direction of the maximum attenuation is also referred to as a “notch direction”.

[0023] In a suitable embodiment of the present invention, the or each notch direction is defined in the form of an angle specification relative to the user's line of sight, for example. Alternatively, however, the or each notch direction may also be specified as an abstract parameter (linearly or nonlinearly associated with the orientation of the associated notch), for example in the form of a weighting factor or in the form of a variable time delay, in order to adjust the common adaptive beamformer to use the weighting factor to weight different basic directional signals (such as cardioid signals and anti-cardioid signals, etc.), and to use the time delay to superimpose different signal components on each other to produce a directional effect.

[0024] The "directional strength" generally describes how much the directional characteristic of the associated adaptive beamformer deviates from an omnidirectional characteristic (i.e., signal processing without directional dependence). In a specific embodiment of the invention, the directional strength of the first adaptive beamformer is unchangeable. In this case, the directional strength is determined implicitly, in particular, by the functional structure or design of the first adaptive beamformer.

[0025] The "adaptation speed" generally describes how quickly the associated adaptive beamformer adapts its directional characteristics to changes in the background noise (i.e. the spatial distribution of the noise sources and thus the sound components in the input audio signal). In a particular embodiment of the invention, the adaptation speed of the first adaptive beamformer is not changeable. In this case, the adaptation speed is determined implicitly, in particular, by the functional structure of the first adaptive beamformer.

[0026] According to the invention, however, at least one of the two aforementioned properties of the directional characteristic of the first adaptive beamformer, namely the adaptation speed and / or the directional strength, is not fixed, but is preset to the first adaptive beamformer as a variable, thus as a variable parameter. The adaptation speed or the directional strength of the first adaptive beamformer is hereby variably adjusted (preferably by the aforementioned signal analysis unit) based on an analysis of the input audio signal or the preprocessed audio signal.

[0027] By the variability of the adaptation speed and / or directional strength of the first adaptive beamformer, artifacts of the above type can be effectively avoided. Therefore, it is particularly possible to temporarily increase the adaptation speed and thus the matching ability of the first adaptive beamformer when the change of the background noise requires a significant adjustment of the first beamformer. In this way, it is particularly possible to avoid a perceptible delay in adapting the first adaptive beamformer to the movement of the noise source relative to the head. Alternatively or additionally, a similar effect is achieved by temporarily reducing the directional strength. But on the other hand, the first adaptation speed can be set to be very small in a static hearing situation, thereby avoiding artificial fluctuations of the useful signal or background noise caused by a small range of adjustment of the directional characteristics of the first adaptive beamformer. In this case, the relatively high directional strength can achieve good attenuation of interfering noise, and thus enable the user to perceive the useful signal well, and thus particularly promote speech understanding in communication between the user and other speakers. The present invention is based on the following recognition, that is, based on the analysis of the input audio signal or the preprocessed audio signal, the control of the adaptation speed and / or directional strength of the first adaptive beamformer allows to particularly effectively avoid artifacts of attenuation related to the direction.

[0028] The two aforementioned measures, namely, on the one hand, a change in the adaptation speed of the first adaptive beamformer and, on the other hand, a change in the directional strength of the first adaptive beamformer, contribute in principle independently of one another to achieve this effect. Within the scope of the present invention, these measures can therefore be used independently of one another by either changing only the adaptation speed of the first adaptive beamformer or only the directional strength. However, in a preferred embodiment of the method, not only the adaptation speed of the first adaptive beamformer is changed, but also the directional strength.

[0029] Preferably, the adaptation speed and / or the directional strength of the first adaptive beamformer are adjusted as a function of the temporal stability of the input audio signal or the preprocessed audio signal (more precisely, as a function of the temporal stability of the background noise on which the input audio signal is based). In a variant of the method in which the adaptation speed of the first adaptive beamformer is changed, the adaptation speed is set low, in particular in the case of high temporal stability of the corresponding audio signal (i.e., weak changes over time), and is set high in the case of low temporal stability (i.e., significant changes over time). This results in that the directional characteristic of the first adaptive beamformer adapts quickly in the case of significant changes in the background noise and adapts slowly in the case of weak changes in the background noise. In a variant of the method in which the directional strength of the first adaptive beamformer is changed, the directional strength is set high, in particular in the case of high temporal stability of the corresponding audio signal (i.e., small changes over time), and is set low in the case of low temporal stability (i.e., significant changes over time). This results in that the input audio signal or the preprocessed audio signal is strongly directed by the first adaptive beamformer in the case of small changes in the background noise, and is weakly directed or even not directed in the case of large changes in the background noise. In order to determine the temporal stability, a parameter characterizing the background noise is derived, for example, from the input audio signal or the preprocessed audio signal. In this case, the standard deviation of the parameter or the squared mean value of the first-order temporal derivative of the parameter over a floating-point time period is used, for example, as a measure of the temporal stability. Alternatively or additionally, the rate of exceeding or falling below the floating-point mean value of the parameter (mean crossing rate) or the rate at which the first-order derivative of the parameter changes sign is used as a measure of the temporal stability of the input audio signal or the preprocessed audio signal.

[0030] In an advantageous embodiment of the present invention, a second adaptive beamformer with a variable (second) directional characteristic is considered for characterizing background noise. Like the first adaptive beamformer, the second adaptive beamformer is applied directly or indirectly to the input audio signal in order to generate a second directional audio signal. Here, the directional characteristic of the second adaptive beamformer is adjusted with a preferably constant (second) adaptive speed, thereby minimizing the energy content of the second directional audio signal. As in the case of the first directional characteristic, the second directional characteristic is preferably characterized by at least one variable direction (notch direction) of maximum attenuation. In a suitable embodiment of the present invention, the second directional audio signal does not enter the modified audio signal to be output to the user. In this case, the second adaptive beamformer is therefore only used for the purpose of signal analysis. The second directional audio signal is only used as an adjustment parameter for energy minimization here, and is not used for signal preparation and output to the user.

[0031] The adaptation speed of the second adaptive beamformer is in particular selected such that it is not lower than and at least temporarily exceeds the adaptation speed of the first adaptive beamformer. The second adaptive beamformer is designed to always adapt quickly so that it can adapt to changes in the background noise without noticeable delays. The directional characteristic of the second adaptive beamformer, in particular the notch direction that may be associated with it, thus forms a measure for the changeability of the background noise on which the input audio signal is based. In contrast to the second adaptive beamformer, the first adaptive beamformer either always adapts slowly or switches between slow and fast adaptation.

[0032] In an advantageous embodiment of the method, the adaptation speed and / or the directional strength of the first adaptive beamformer are variably adjusted as a function of a change in the second directional characteristic, i.e., the directional characteristic of the second adaptive beamformer. As a characteristic variable of the change over time of the second directional characteristic and thus as a measure of the variability of the background noise, for example, a floating point square mean of the first-order time derivative of the notch direction associated with the second directional characteristic is determined. For example, if and as long as the aforementioned characteristic variable exceeds a preset threshold value, the adaptation speed of the first adaptive beamformer is increased relative to a base value and / or the directional strength of the first adaptive beamformer is reduced relative to a base value.

[0033] Alternatively or additionally thereto, the adaptation speed and / or directional strength of the first adaptive beamformer is preferably adjusted according to a deviation of the second directional characteristic from the first directional characteristic, in particular according to a deviation of a (second) notch direction associated with the second directional characteristic and a (first) notch direction associated with the first directional characteristic. For example, if and as long as the aforementioned deviation of the notch direction exceeds a preset threshold, the adaptation speed of the first adaptive beamformer is increased and / or the directional strength of the first adaptive beamformer is reduced.

[0034] In an improvement of the aforementioned method variant, instead of a second adaptive beamformer for analyzing the input audio signal or the preprocessed audio signal and thereby for adjusting the first adaptation speed and / or the first directional strength, a plurality of adaptive beamformers are used, which correspond in particular to the second adaptive beamformer in terms of structure, functionality (and optionally also design). The plurality of adaptive (analysis) beamformers are coupled to one another in particular here, so that different orientations of their respectively associated directional characteristics (and, if applicable, associated notch directions) are enforced. It is thus achieved in particular that each of the plurality of adaptive (analysis) beamformers is oriented toward another main noise source in the user's surroundings. By means of such a cascade of analysis beamformers, background noise and changes in the background noise, on which the input audio signal is based, can be analyzed with high accuracy.

[0035] Preferably, the first adaptive beamformer has a frequency dependency, i.e., modifies different frequency components of the input audio signal or the preprocessed audio signal in a respectively separate manner. The input audio signal or the preprocessed audio signal is in particular respectively divided into different frequency channels, wherein the directional characteristic of the first adaptive beamformer (in particular the or each notch direction) is adapted individually to each frequency channel. In a preferred embodiment of the invention, the adaptation speed and / or the directional strength of the first adaptive beamformer are also preset as frequency-dependent variables (e.g., as a vector with entries for each frequency channel or as a continuous frequency-dependent function), so that the directional characteristic of the first adaptive beamformer is adapted differently quickly for different frequencies, or so that the directional effect of the first adaptive beamformer occurs differently strongly for different frequencies.

[0036] In order to adjust the adaptation speed and / or the directional strength of the first adaptive beamformer in a frequency-dependent manner, at least one interference noise component emitted by an interference noise source is identified in the input audio signal or the preprocessed audio signal and analyzed with respect to its spectral components. Specifically, an interference frequency range corresponding to the interference noise component is determined. Here, the adaptation speed and / or the directional strength of the first adaptive beamformer are preset uniformly (i.e., with the same value) within the interference frequency range. In this way, it is avoided that the first adaptive beamformer adapts to different frequency components of the same interference noise in different ways, which could lead to tonality distortions and / or artificial fluctuations of the interference noise.

[0037] The hearing system according to the invention is designed only for automatically implementing the method according to the invention described above. For this purpose, the hearing system comprises a first adaptive beamformer (as described above). The hearing system further comprises an adaptive controller, which is designed to variably adjust the adaptation speed and / or the directional strength of the first adaptive beamformer based on an analysis of the input audio signal or the preprocessed audio signal.

[0038] The device of the hearing system for automatically implementing the method according to the invention has a program-technical and / or circuit-technical nature. Therefore, the hearing system according to the invention includes program-technical components (software) and / or circuit-technical components (e.g. hardware, hardware in the form of an ASIC), which automatically implement the method according to the invention during operation of the hearing system. The program-technical or circuit-technical components for implementing the method, in particular the first adaptive beamformer and the adaptive controller, can only be arranged in the hearing device (or several hearing devices) of the hearing system. Alternatively, the program-technical or circuit-technical components for implementing the method are distributed on the hearing device or hearing aid and at least on other devices or software components of the hearing system. For example, the program-technical components for implementing the method are distributed on at least one hearing device of the hearing system and on a control program installed on an external electronic device (in particular a smartphone). As mentioned above, the external electronic device itself is usually not part of the hearing system here.

[0039] The above-described embodiments of the method according to the invention correspond to corresponding embodiments of the hearing system according to the invention.

[0040] Therefore, in a preferred embodiment of the present invention, the adaptive controller is designed to adjust the adaptation speed and / or direction strength of the first adaptive beamformer according to the temporal stability of the input audio signal.

[0041] Preferably, the adaptive controller for analyzing the input audio signal or the preprocessed audio signal (i.e. for characterizing the background noise on which it is based) and for determining the adaptation speed and / or directional strength of the first adaptive beamformer comprises a second adaptive beamformer or further (in particular mutually coupled) adaptive beamformers in cascade, as described previously.

[0042] The adaptive controller is particularly designed to variably adjust the adaptation speed and / or directional strength of the first adaptive beamformer depending on a change in the (corresponding) directional characteristics of the second adaptive beamformer (and optionally further adaptive beamformers) and / or depending on a deviation of the directional characteristics of the adaptive beamformer.

[0043] The first adaptive beamformer preferably has a frequency-dependent directional characteristic (as described above), in particular a directional characteristic that is individually adapted for a plurality of frequency channels. The adaptive controller is preferably designed to preset an adaptation speed and / or a directional strength of the first adaptive beamformer as a frequency-dependent parameter; to adjust the adaptation speed and / or the directional strength of the first adaptive beamformer, an interference noise component emitted from an interference noise source is identified in the input audio signal or the preprocessed audio signal; an interference frequency range corresponding to the interference noise component is determined; and the adaptation speed and / or the directional strength of the first adaptive beamformer is uniformly preset within the interference frequency range.

[0044] The effects and advantages of the individual method variants can be transferred to the corresponding variants of the hearing system and vice versa. BRIEF DESCRIPTION OF THE DRAWINGS

[0045] The embodiments of the present invention are described in detail below with reference to the accompanying drawings.

[0046] Figure 1 A hearing system consisting of a single hearing device in the form of a hearing aid that can be worn behind the ear of a user is shown schematically;

[0047] Figures 2 to 4 The schematic block diagrams respectively show the three alternative embodiments. Figure 1 The structure of the signal processing device of the hearing system;

[0048] Figure 5 Based on Figures 2 to 4 The diagram shows a functional unit of a signal processing device of a hearing system, referred to as an adaptive controller, in a further embodiment; and

[0049] Figure 6 Based on Figure 1 The illustration of shows an alternative embodiment of a hearing system, wherein the hearing system comprises a hearing device in the form of a hearing aid that can be worn behind the ear and a control program (“hearing app”) implemented in a smartphone.

[0050] Parts and variables that correspond to one another are always provided with the same reference symbols in all the figures. DETAILED DESCRIPTION

[0051] Figure 1 A hearing system 2 is shown, which here consists of a single hearing aid 4, ie a hearing device designed to support the hearing of a hearing-impaired user. In the example shown here, the hearing aid 4 is a BTE hearing aid wearable behind the ear of a user.

[0052] Optionally, in a further embodiment of the invention, the hearing system 2 comprises a second hearing aid (not explicitly shown) for taking care of the second ear of the user, the second hearing aid corresponding in particular to the Figure 1 The hearing aid 4 is shown.

[0053] The hearing aid 4 comprises two microphones 6 as acoustic-electrical input converters and an earpiece 8 (receiver) as an electro-acoustic output converter in a housing 5. The hearing aid 4 further comprises a battery 10 and a signal processing device in the form of a signal processor 12. Preferably, the signal processor 12 comprises a programmable subunit (e.g. a microprocessor) and a non-programmable subunit (e.g. an ASIC).

[0054] The signal processor 12 is supplied with a supply voltage U by the battery 10 .

[0055] During normal operation of the hearing aid 4, the microphone 6 receives air sounds from the surroundings of the hearing aid 4. The microphone 6 converts the sounds into (input) audio signals I1 or I2, respectively, which contain information about the received sounds. The input audio signals I1, I2 are fed to a signal processor 12 within the hearing aid 4, which modifies the input audio signals I1, I2 to support the hearing of the user.

[0056] The signal processor 12 outputs an output audio signal O to the earpiece 8, the output audio signal containing information about the processed and thus modified sound.

[0057] The earpiece 8 converts the output sound signal O into a modified airborne sound. The modified airborne sound is transmitted into the ear canal of the user via the sound channel 14 connecting the earpiece 8 and the tip 16 of the housing 5 and via a flexible sound hose (not explicitly shown), which connects the tip 16 to an earplug inserted into the ear canal of the user.

[0058] exist Figure 2 . It can be seen from this that the signal processing device of the hearing system 2 is divided into two functional components, namely a signal preparation unit 18 and a signal analysis unit 20. The signal preparation unit 18 is used to generate an output audio signal O from the input audio signals I1, I2 of the microphone 6 or from the audio signals I1′, I2′ derived by preprocessing and thus preprocessed. In the aforementioned case, the input audio signals I1, I2 of the microphone 6 are directly supplied to the signal preparation unit 18. In the latter case, the output audio signal O is generated by the preprocessing unit 18. Figure 2 In the case shown by way of example in FIG. 1 , the input audio signals I1 , I2 of the microphone 6 are first supplied to a preprocessing unit 22 , which then derives preprocessed audio signals I1 ′, I2 ′ therefrom and supplies them to the signal preparation unit 18 .

[0059] In the preprocessing unit 22 , the input audio signals I1 , I2 are preferably added to each other with a time delay to form preprocessed audio signals I1 ′, I2 ′, so that the two preprocessed audio signals I1 ′, I2 ′ correspond to cardioid signals or anti-cardioid signals.

[0060] The signal preparation unit 18 comprises a plurality of signal preparation processes 24, which successively process and modify the input audio signal I or (in accordance with Figure 2 In the example of ), the internal audio signals I1′, I2′ are used to generate the output audio signal O and thus compensate for the hearing loss of the user.

[0061] The signal preparation process 24 is selectively implemented in the signal processor 12 in any combination in the form of a (non-programmable) hardware circuit and / or in the form of a software module (firmware). For example, at least one of the signal preparation processes 24 is formed by a hardware circuit; at least one other of the signal preparation processes 24 is formed by a software module; and yet another signal preparation process in the signal preparation process 24 is formed by a combination of hardware and software components.

[0062] The signal preparation process 24 includes, for example:

[0063] - processes for interference noise suppression and / or feedback suppression,

[0064] - a process for dynamic compression, and

[0065] - a process for frequency-dependent amplification based on audiogram data,

[0066] - and so on.

[0067] Here, at least one of the signal preparation processes 24 (usually all signal preparation processes 24 or at least most of the signal preparation processes 24) is respectively associated with at least one signal preparation parameter P. The or each signal preparation process 24 is a one-dimensional variable (binary variable, natural number, floating point number, etc.) or a multi-dimensional variable (array, function, etc.), whose value parameterizes the functionality of the respectively associated signal preparation process 24 (i.e., affects it). The signal preparation parameter P can switch on or off the respectively associated signal preparation process 24; continuously or stepwise amplify or weaken the effect of the respectively associated signal preparation process 24; define a time constant for the corresponding signal preparation process 24; and so on.

[0068] The signal preparation parameters P include, for example:

[0069] - the aforementioned audiogram data for the process of frequency-dependent amplification or frequency-specific amplification factors derived therefrom,

[0070] - characteristic curve for the dynamic compression process,

[0071] - a control variable for continuously adjusting the intensity of the process for suppressing interfering noise or feedback,

[0072] - and so on.

[0073] In any case, a portion of the signal preparation parameters P is provided from the parameterization unit 26 to the signal preparation process 24 .

[0074] In addition, the signal preparation process 24 includes Figure 2 The first adaptive beamformer 28 shown in detail in FIG. 1 is designed to attenuate the input audio signals I1, I2 (or Figure 2 As shown, the preprocessed audio signals I1′, I2′) are combined and the first directional audio signal R1 is generated in this way. The beam former 28 generates the audio signal R1 in the following manner, that is, the beam former combines the two transmitted audio signals I1′, I2′ (in accordance with Figure 2 The cardioid signal and the anti-cardioid signal in the example of FIG. 1 are weighted and superimposed on each other using a first weighting factor a1:

[0075] R1=I1′–a1×I2′ Where a=[-1;1] Equation 1

[0076] The weighting factor a1 determines the notch direction in which the direction-dependent attenuation of the beamformer 28 has a (local) maximum, viewed relative to the user's head. The weighting factor a1 therefore represents a measure for the notch direction of the beamformer 28 and is therefore then conceptually equivalent to said notch direction. In order to adapt to the directional characteristics, the beamformer 28 changes the weighting factor a1 in an adaptive step in a controlled manner so that the energy content of the directional audio signal R1 is minimized (at Figure 2 2 is schematically shown by feeding the audio signal R1 back to the beamformer 28. The energy minimization described achieves that interference noise from the spatial region 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 a further signal processing process 24, thereby generating an output audio signal O. The beamformer 28 is preferably formed by a software module.

[0077] A first adaptation speed v1 is variably predefined for the beam former 28 as a signal preparation parameter P. The adaptation speed v1 is determined in the signal analysis unit 20 by a functional unit referred to as an adaptation controller 30 , which is preferably implemented in software technology.

[0078] exist Figure 2In the illustrated embodiment, the adaptive controller 30 includes a second adaptive beamformer 32 and an evaluation module 34 .

[0079] The second adaptive beamformer 32 preferably corresponds in structure and function to the first adaptive beamformer 28. The second adaptive beamformer 32 is therefore designed in the aforementioned manner for directionally attenuating the input audio signals I1, I2 (or Figure 2 As shown, the preprocessed audio signals I1′, I2′) are processed, and in this way, a second directional audio signal R2 is generated. The directional characteristic of the beamformer 32 preferably has a notch direction, and the notch direction is characterized by a variable weighting factor a2. The beamformer 32 changes the weighting factor a2 (and thus the notch direction) with an adaptive speed v2, thereby minimizing the energy content of the directional audio signal R2.

[0080] Unlike the beamformer 28, the beamformer 32 is not used to generate an output audio signal O to the user, but is only used to analyze the background noise on which the input signals I1, I2 are based. 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 a weighting factor a2 indicating the notch direction (and thus indirectly indicating the arrangement of the most dominant noise source in the user's surroundings) as an analysis result to the evaluation module 34.

[0081] In the evaluation module 34, according to Figure 2 In an embodiment of the invention, the temporal stability (or, in other words, the changeability over time) of the weighting factor a2 and thus of the background noise is evaluated, for example, in that a floating-point squared temporal average value is formed on the first temporal derivative of the weighting factor a2. Depending on this variable, the evaluation module 34 changes the adaptation speed v1 for the first adaptive beamformer 28. In a simple but expedient embodiment variant, the evaluation module 34 changes the adaptation speed v1 in a binary manner between a relatively low base value and a higher value compared thereto. If and as long as the aforementioned average value does not exceed a predefined threshold value, which means that the background noise does not change or changes only slightly, the evaluation module 34 sets the adaptation speed v1 to the base value. In this case, the first beamformer 28 therefore adapts only slowly, whereby artifacts due to the adaptation are largely avoided. Otherwise, ie 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 28 can quickly (especially without perceptible delay) adapt to changes in the hearing situation.

[0082] In order to analyze the background noise with high accuracy, the second adaptive beamformer 32 is designed to be fast adaptive. The adaptation speed v2 is selected here (preferably as a constant) so that it does not fall below the variable adaptation speed v1 of the first adaptive beamformer 28 at any time (v2 ≥ v1).

[0083] In addition or as an alternative to the adaptation speed v1, the directional strength s of the first adaptive beamformer 28 can also be preferably changed. The directional strength s is changed in this case, for example, by mixing the weighted sum according to equation 1 with the omnidirectional audio signal A derived from the input audio signals I1, I2 to varying degrees (which is based on Figure 2 Optionally, it is supplied as an additional input variable to the beamformer 28 . If and as long as a strong changeability of the background noise is detected (in particular due to the exceeding of the aforementioned threshold value), the directional strength s is reduced by the evaluation module 34 relative to a predefined base value.

[0084] In addition, Figure 2 As can be seen, the signal analysis unit 20, in addition to the adaptive controller 30 and preferably in addition to other functions for sound analysis (not shown in detail here), optionally also includes a classifier 36, which classifier in a conventional manner by analyzing the input audio signals I1, I2 (or Figure 2 As shown in , the preprocessed audio signals I1′, I2′) are analyzed for their similarity to a large number of typical listening situation categories (such as “speech”, “speech with background noise” or “music”) and a corresponding classification signal K is output.

[0085] The classification signal K is supplied to a parameterization unit 26 which selects in a manner known per se as a function of the classification signal K between different hearing programs, ie different sets of signal preparation parameters P, each of which is optimal for a typical class of hearing situations.

[0086] On the other hand, the classification signal K is also fed to the evaluation module 34 of the adaptive controller 30 and influences the determination of the adaptive speed v1 and / or the directional strength s. For example, these values ​​are changed depending on the classification signal K and the adaptive speed v1 and / or the directional strength s are changed between these values.

[0087] exist Figure 3 An alternative embodiment of the hearing system 2 is shown in FIG. Figure 2 In different implementations, according to Figure 3In an embodiment of the present invention, in addition to the weighting factor a2 of the beamformer 32, the weighting factor a1 of the beamformer 28 is also supplied to the evaluation module 34. The evaluation module 34 analyzes the changeability of the background noise on which the input audio signals I1, I2 and the audio signals I1′, I2′ are based in the following manner, namely it compares 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 regarded as a sign of a significant change in the background noise. Accordingly, if and as long as the difference between the weighting factors a1 and a2 exceeds a preset threshold value, the evaluation module 34 increases the adaptation speed v1, and / or reduces the directional strength s,

[0088] Figure 4 Another embodiment of the hearing system 2 is shown. Figure 2 and Figure 3 Unlike the embodiment of the adaptive controller 30, the adaptive controller 30 here includes, in addition to the second adaptive beamformer 28, at least one further adaptive beamformer 28, which generates a further directional audio signal R3 and changes the associated further weighting factor a3 (as a measure for the changeable notch direction for the beamformer 38) based on minimizing the energy of the audio signal R3.

[0089] In an expedient embodiment variant, the adaptation speed v3 associated with the beamformer 38 and preferably constantly predefined has a value that is lower than the adaptation speed v2 and corresponds in particular exactly or approximately to the base value of the adaptation speed v1. In this case, the further adaptive beamformer 38 is therefore designed to be slowly adaptive compared to the second adaptive beamformer 32, wherein the two beamformers 32 and 38 preferably adjust the respective weighting factors a2 and a3 independently of one another (in this embodiment variant, preferably no coupling of the beamformers 32 and 38 is provided, as in Figure 4 2 is shown in FIG. 1 ). Here, the weighting factor a2 is used to transmit the weighting factor a2 to the beam former 38. Figure 3 Similarly to the embodiment of FIG. 3 , the changeability 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 based on the deviations of the weighting factors a2 and a3 of the beamformers 32 and 38 .

[0090] In an alternative embodiment variant, the adaptation speeds v2 and v3 of the beamformers 32 and 38 are selected to be exactly or approximately identical, so that both beamformers 32 and 38 adapt quickly. In this case, the beamformers 26 and 38 are preferably (as in Figure 4) are coupled to each other, thereby forcing different adjustments of the weighting factors a2 and a3 to be performed. This coupling ensures that the beamformers 26 and 38 are adapted to the different dominant noise sources in the user's surroundings. In this case, it is preferred to Figure 2 In analogy to the embodiment of , the changeability 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 based on the temporal stability of the weighting factors a2 and a3. If at least one of the weighting factors a2 and a3 meets the conditions for increasing the adaptation speed v1 or reducing the direction strength s, then in particular the adaptation speed v1 is increased and / or the direction strength s is reduced.

[0091] In accordance with Figure 3 and Figure 4 In the exemplary embodiment of FIG. 3 , a classifier 36 is optionally also present, but is not shown in this figure simply for reasons of better overview.

[0092] Preferably, the signal processing in the signal preparation unit 18 is performed in a frequency-resolved manner in a plurality of (for example 64) frequency channels. The input audio signals I1, I2 are preferably also processed (in Figures 2 to 4 The analysis filter banks (not explicitly shown in FIG. 1 ) are respectively divided into a number of frequency components, which are prepared separately in the frequency channels and then (in Figures 2 to 4 are combined into the output audio signal O in a synthesis filter bank (also not explicitly shown in FIG).

[0093] The first adaptive beamformer 28 is designed here to attenuate the frequency components of the input audio signals I1, I2 or the preprocessed audio signals I1′, I2′ guided in the frequency channels in a direction-dependent manner. Therefore, the directional characteristic of the beamformer 28 and the associated notch direction or weighting factor a1 also have a frequency dependency. Preferably, the weighting factor a1 and / or the directional strength s are respectively preset as a vector, which has an associated individual value for each frequency channel. In addition, the directional characteristic of the beamformer 28 is preferably also individually adapted to each frequency channel. Therefore, the adaptation speed v1 is preferably also preset as a vector, which has an associated individual value for each frequency channel.

[0094] In order to prevent the interference noise originating from a specific noise source from being perceptibly distorted by the beamformer 28 due to frequency-specifically different adaptations of the directional characteristics, the adaptive controller 30 is preferably designed to couple the frequency channels that guide the main frequency components of the main interference noise with respect to the adaptation speed v1 and / or the directional strength s. In other words, the adaptive controller 30 uniformly (i.e. with the same value) presets the adaptation speed v1 and / or the directional strength s for those frequency channels that guide the main frequency components of the main interference noise.

[0095] For this purpose, the second adaptive beamformer 32 (and the third adaptive beamformer 38 if necessary) is preferably constructed similarly to the beamformer 28, so that it attenuates the frequency components of the input audio signals I1, I2 or the preprocessed audio signals I1′, I2′ guided in the frequency channels separately and in a direction-dependent manner. Therefore, the background noise is analyzed in a frequency-resolved manner by the second adaptive beamformer 32 (and the third adaptive beamformer 38 if necessary).

[0096] In order to determine the spectral components of one or more dominant interfering noises, the directional audio signal R2 (or R3, respectively) output by the second adaptive beamformer 32 (and the third adaptive beamformer 38, if necessary) is inverted in the inverter 40 and then multiplied with the omnidirectional audio signal A in the multiplier 42. Figure 5 The signal processing device is exemplarily shown in the embodiment of the adaptive controller 30, and the adaptive controller is Figure 4 Similarly, not only the second adaptive beamformer 32 but also the third adaptive beamformer 38 is included. By multiplying the omnidirectional audio signal A with the inverted directional audio signal R2 (or R3), the audio signal R2′ (or R3′) is formed, in which the main interference noise is selectively amplified, which is selectively filtered for the second adaptive beamformer 32 (or the third adaptive beamformer 38 if necessary). The audio signal R2′ (or R3′) is now supplied to the evaluation module 34, which analyzes the spectral components of the audio signal R2′ (or R3′) and determines the interference frequency range corresponding to the corresponding interference noise. The frequency channels within the interference frequency range are adaptively coupled to the first adaptive beamformer 28 by the evaluation module 34 in that the evaluation module 34 uniformly presets the values ​​of the adaptation speed v1 and / or the directional strength s corresponding to the frequency channel: for example, if and only if (from the value determined by the method according to Figure 2 or Figure 4An evaluation of the weighting factor a2 or the weighting factors a2 and a3 performed by the evaluation module 34) produces that, for at least one of the coupled frequency channels, the conditions for increasing the adaptation speed v1 or reducing the directional strength s are met, then the adaptation speed v1 for all coupled frequency channels is increased relative to the base value, and / or the directional strength s for all coupled frequency channels is reduced relative to the base value.

[0097] Figure 6 A further embodiment of a hearing system 2 is shown, wherein the hearing system comprises control software in addition to a hearing aid 4 (or two hearing aids of this type for taking care of both ears of a user). The control software is subsequently referred to as a hearing App 44. The hearing App 44 is Figure 5 In the example shown, it is installed on a smartphone 46. The smartphone 46 itself is not a part of the hearing system 2. Instead, the smartphone 46 is used only as a resource for storage space and computing power by the hearing App 44.

[0098] When the hearing system 2 is in operation, the hearing aid 4 and the hearing app 44 exchange data via a wireless data transmission connection 48. The data transmission connection 48 is based on the Bluetooth standard, for example. The hearing app 44 accesses the Bluetooth transceiver of the smartphone 46 in order to receive data from the hearing aid 4 and to send data to the hearing aid. The hearing aid 4 itself includes a Bluetooth transceiver (not explicitly shown) in order to send data to the hearing app 44 and to receive data from the app.

[0099] In accordance with Figure 6 In the implementation scheme of Figures 2 to 5 Several parts of the signal processing device shown in FIG. 4 , such as the adaptive controller 30 , are not implemented in the signal processor 12 of the hearing aid 4 , but are instead implemented in the hearing App 44 .

[0100] The present invention is particularly clear from the above-mentioned embodiments, but the present invention is not limited to these embodiments. On the contrary, further embodiments of the present invention can be derived by a person skilled in the art from the present invention and the above description. In particular, the individual features of the hearing system and the associated operating method respectively explained according to the different embodiments can also be combined with each other in different ways within the scope of the present invention without departing from the present invention.

[0101] Reference numerals list

[0102] 2 Hearing System

[0103] 4. Hearing aids

[0104] 5 Housing

[0105] 6 Microphones

[0106] 8. Earpiece

[0107] 10. Battery

[0108] 12 Signal Processor

[0109] 14 sound channels

[0110] 16 Tip

[0111] 18 Signal preparation unit

[0112] 20 Signal Analysis Unit

[0113] 22 Preprocessing Unit

[0114] 24 Signal preparation process

[0115] 26 parameterized units

[0116] 28 (First Adaptive) Beamformer

[0117] 30 Adaptive Controller

[0118] 32 (Second Adaptive) Beamformer

[0119] 34 Evaluation Modules

[0120] 36 Classifier

[0121] 38 (Third adaptive) beamformer

[0122] 40 Inverter

[0123] 42 Multiplier

[0124] 44 Listening App

[0125] 46 Smartphone

[0126] 48 Data transfer connections

[0127] a1 (first) weighting factor

[0128] a2 (second) weighting factor

[0129] a3 (third) weighting factor

[0130] s direction strength

[0131] v1 (first) adaptive speed

[0132] v2 (second) adaptive speed

[0133] v3 (third) adaptive speed

[0134] A (omnidirectional) audio signal

[0135] I1, I2 input audio signal

[0136] I1′, I2′ (internal) audio signal

[0137] K Classification Signal

[0138] O Output audio signal

[0139] P signal preparation parameters

[0140] R1 (first directional) audio signal

[0141] R2 (second directional) audio signal

[0142] R3 (third directional) audio signal

[0143] U Supply voltage

Claims

1. A method for operating a hearing system (2), which is used to support the hearing of a user and which has at least one hearing device (4) to be worn on the head of the user, - in, receiving sound signals from the user's surroundings by means of at least two input transducers (6) of the at least one hearing device (4) and converting them into input audio signals (I1, I2), - wherein the input audio signal (I1, I2) is processed in a signal preparation 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 device (4), - wherein in the signal preparation step, the input audio signal (I1, I2) or the audio signal (I1′, I2′) derived therefrom by preprocessing is directionally attenuated with a directional strength (s) according to a variable first directional characteristic by means of a first adaptive beamformer (28) in order to generate a first directional audio signal (R1), - wherein, in the adaptation step, the directional characteristics of the first adaptive beamformer (28) are changed with a first adaptation speed (v1) so as to minimize the energy content of the first directional audio signal (R1), and - wherein the first adaptation speed (v1) and / or the directional strength (s) are variably adjusted based on an analysis of the input audio signal (I1, I2) or the preprocessed audio signal (I1', I2'), - wherein a second adaptive beamformer (32) having a variable second directional characteristic is applied to the input audio signal (I1, I2) or the pre-processed audio signal (I1′, I2′) in order to adjust a first adaptation speed (v1) and / or a directional strength (s) for the first adaptive beamformer (28) in order to generate a second directional audio signal (R2), - wherein the variable second directional characteristic of the second adaptive beamformer (32) is adjusted using a second adaptation speed (v2) so as to minimize the energy content of the second directional audio signal (R2), - wherein the second adaptation speed (v2) is not lower than and at least temporarily exceeds the first adaptation speed (v1).

2. The method according to claim 1, in, The first adaptation speed (v1) and / or the directional strength (s) is adjusted depending on the temporal stability of the input audio signal (I1, I2) or the pre-processed audio signal (I1', I2').

3. The method according to claim 1 or 2, in, The first adaptation speed (v1) and / or the directional strength (s) are variably adjusted according to a change in the second directional characteristic.

4. The method according to claim 1 or 2, in, The first adaptive speed (v1) and / or the directional strength (s) are adjusted according to a deviation of the second directional characteristic from the first directional characteristic.

5. The method according to claim 1 or 2, - in, The first directional characteristic is frequency-dependent, so that different frequency components of the input audio signal (I1, I2) or the preprocessed audio signal (I1′, I2′) are attenuated in a direction-dependent manner in a respectively separate manner, - wherein a first adaptation speed (v1) and / or a directional strength (s) for the first adaptive beamformer (28) is preset as a frequency-dependent parameter, - wherein, in order to adjust the first adaptation speed (v1) and / or the directional strength (s), an interfering noise component emanating from an interfering noise source is identified in the input audio signal (I1, I2) or the preprocessed audio signal (I1′, I2′), - wherein an interfering frequency range corresponding to said interfering noise component is determined, and - wherein the first adaptive speed (v1) and / or the directional strength (s) are uniformly preset within the interference frequency range.

6. A hearing system (2) for supporting the hearing of a user, the hearing system having at least one hearing device (4) worn on the head of the user, - in, The at least one hearing device (4) has at least two input transducers (6) which are designed to receive sound signals from the user's surroundings and convert them into input audio signals (I1, I2), - wherein the hearing system (2) has a signal preparation unit (18) which is designed to prepare an input audio signal (I1, I2) in order to generate an output audio signal (O), wherein the hearing device (4) has an output transducer (8) which is designed to output an output audio signal (O), - wherein the signal preparation unit (18) comprises a first adaptive beamformer (28) which is designed to attenuate the input audio signal (I1, I2) or the audio signal (I1′, I2′) derived therefrom by preprocessing in a direction-dependent manner according to a variable first directional characteristic using a directional strength (s) in order to generate a first directional audio signal (R1), and to change the first directional characteristic using a first adaptation speed (v1) in order to minimize the energy content of the first directional audio signal (R1), - wherein the hearing system (2) comprises an adaptive controller (30) designed to variably adjust the first adaptation speed (v1) and / or the directional strength (s) based on an analysis of the input audio signal (I1, I2) or the preprocessed audio signal (I1′, I2′), - wherein the adaptive controller (30) comprises a second adaptive beamformer (32) having a variable second directional characteristic, and the input audio signal (I1, I2) or the pre-processed audio signal (I1', I2') is transmitted to the second adaptive beamformer, - wherein the second adaptive beamformer (32) is designed to generate a second directional audio signal (R2) and to adjust the variable second directional characteristic using a second adaptive speed (v2) so as to minimize the energy content of the second directional audio signal, - wherein the second adaptation speed (v2) is not lower than and at least temporarily exceeds the first adaptation speed (v1).

7. The hearing system (2) according to claim 6, in, The adaptive controller (30) is designed to adjust the first adaptive speed (v1) and / or the directional strength (s) depending on the temporal stability of the input audio signal (I1, I2) or the preprocessed audio signal (I1', I2').

8. The hearing system (2) according to claim 6 or 7, in, The adaptive controller (30) is designed to variably adjust the first adaptive speed (v1) and / or the directional strength (s) according to a change in the second directional characteristic.

9. The hearing system (2) according to claim 6 or 7, in, The adaptive controller (30) is designed to adjust the first adaptive speed (v1) and / or the directional strength (s) according to the deviation of the second directional characteristic from the first directional characteristic.

10. The hearing system (2) according to claim 6 or 7, - in, The first directional characteristic is frequency-dependent, so that different frequency components of the input audio signal (I1, I2) or the preprocessed audio signal (I1', I2') are attenuated in different ways in a direction-dependent manner, - wherein the adaptive controller (30) is designed to: - presetting the first adaptive speed (v1) and / or the direction strength (s) as frequency-dependent parameters, - in order to adjust the first adaptation speed (v1) and / or the directional strength (s), in the input audio signal (I1, I2) or the pre-processed audio signal (I1', I2') an interfering noise component emanating from an interfering noise source is identified, - determining an interfering frequency range corresponding to said interfering noise component, and - uniformly presetting the first adaptive speed (v1) and / or the directional strength (s) within the interference frequency range.

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