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

Abstract

A hearing system (2) for supporting a user's hearing ability, comprising at least one hearing instrument (4) worn in or on the user's ear, and a method for operating this hearing system (2) are described. During operation of the hearing system (2), an input transducer of the hearing instrument (4) receives a sound signal from the environment of the hearing instrument (4) and modifies it in a signal processing step. The modified sound signal is output by means of an output transducer (8) of the hearing instrument (4). A first signal component (S1) and a second signal component (S2) are derived from the received sound signal, with these signal components (S1, S2) overlapping in time. In the first signal component (S1), the user's own voice is emphasized in relation to the ambient noise, while in the second signal component (S2), the ambient noise is emphasized in relation to the user's own voice.The first signal part (S1) and the second signal part (S2) are processed in different ways in the signal processing step and, after this processing, are combined to generate the modified sound signal.

Description

[0001] The invention relates to a method according to the preamble of claim 1 for operating a hearing system to support the hearing ability of a user, comprising at least one hearing instrument worn in or on the user's ear. The invention further relates to such a hearing system.

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

[0003] Hearing instruments in general, and hearing aids in particular, are usually designed to be worn on or in the ear of the user, especially as behind-the-ear (BTE) devices or in-the-ear (ITE) devices. In terms of their internal structure, hearing instruments typically have at least one (acousto-electrical) input transducer, a signal processing unit (signal processor), and an output transducer. During operation, the input transducer picks up sound waves from the surrounding environment and converts these into an input audio signal (i.e., an electrical signal that carries information about the ambient sound). This input audio signal is subsequently referred to as the "received sound signal."In the signal processing unit, the input audio signal is processed (i.e., modified with respect to its sound information) to support the user's hearing, particularly to compensate for hearing loss. The signal processing unit outputs a correspondingly processed audio signal (also referred to as the "output audio signal" or "modified sound signal") to the output transducer. In most cases, the output transducer is an electro-acoustic transducer that converts the (electrical) output audio signal back into sound waves, which are then delivered into the user's ear canal, modified relative to the ambient sound. In a behind-the-ear hearing aid, the output transducer, also called the "receiver," is usually integrated outside the ear within the hearing aid's housing.In this case, the sound emitted by the output transducer is channeled into the user's ear canal via a sound tube. Alternatively, the output transducer can also be located in the ear canal, and thus outside the behind-the-ear housing. Such hearing aids are also known as RIC (receiver-in-canal) devices. Hearing aids worn in the ear that are so small that they do not protrude beyond the ear canal are also known as CIC (completely-in-canal) devices.

[0004] In other designs, the output transducer can also be an electromechanical transducer that converts the output audio signal into structure-borne sound (vibrations), which is then transmitted, for example, into the user's skull bone. Furthermore, there are implantable hearing devices, particularly cochlear implants, and hearing devices whose output transducers directly stimulate the user's auditory nerve.

[0005] The term "hearing system" refers to a single device or a group of devices and, if applicable, non-physical functional units that together provide the functions required for the operation of a hearing aid. In its simplest form, a hearing system can consist of a single hearing aid. Alternatively, a hearing system can comprise two interacting hearing aids to serve both of the user's ears. In this case, it is referred to as a "binaural hearing system." Additionally or alternatively, a hearing system can include at least one other electronic device, such as a remote control, a charger, or a programmer for the hearing aid(s).Modern hearing aids often use a control program, particularly in the form of an app, instead of a remote control or dedicated programming device. This control program is designed to run on an external computer, especially a smartphone or tablet. The external computer is typically not part of the hearing aid itself and is usually not provided by the hearing aid manufacturer.

[0006] A common problem with hearing aids is that the hearing aid(s) distort the user's own voice, often reproducing it too loudly and with an unnatural sound. Modern hearing aids address this problem, at least partially, by recognizing specific intervals (the user's own voice) within the recorded sound signal. These intervals are then processed differently by the hearing aid, specifically amplified less than other intervals of the recorded sound signal that do not contain the user's voice.

[0007] However, such signal processing methods also affect other components (ambient noise) of the recorded sound signal, in addition to the user's own voice, due to the altered signal processing. If the user speaks intermittently (i.e., in short intervals interrupted by pauses in speech) while using the hearing aid, this regularly leads to a modulation of the ambient noise, which is often perceived as disturbing.

[0008] A method according to the preamble of claim 1 is known from EP 3 101 919 A1.

[0009] EP 3 188 507 A1 discloses a head-worn hearing and communication device comprising an ambient microphone for recording a first sound signal from the device's surroundings and an ear canal microphone for recording a second sound signal from the user's ear canal. The two sound signals are combined in a mixer to form a hybrid signal. A similar device is also disclosed in US 6 661 901 B1.

[0010] The invention is based on the objective of enabling improved signal processing in a hearing system from this perspective.

[0011] With regard to a method, this problem is solved according to the invention by the features of claim 1. With regard to a hearing system, the problem is solved according to the invention by the features of claim 8. Advantageous and partly inventive embodiments or further developments of the invention are set out in the dependent claims and the following description.

[0012] The invention generally relates to a hearing system for supporting a user's hearing ability, wherein the hearing system comprises at least one hearing instrument worn in or on one of the user's ears. As described above, in simple embodiments of the invention, the hearing system may consist exclusively of a single hearing instrument. Preferably, however, the hearing system comprises, in addition to the hearing instrument, at least one further component, e.g., another (in particular, identical) hearing instrument for supplying the user's other ear, a control program (in particular, in the form of an app) for execution on an external computer (in particular, a smartphone) of the user, and / or at least one further electronic device, e.g., a remote control or a charger.The hearing instrument and at least one other component exchange data with each other, whereby data storage and / or data processing functions of the hearing system are divided between the hearing instrument and the at least one other component.

[0013] The hearing instrument comprises at least two input transducers for receiving a sound signal (particularly in the form of airborne sound) from the environment surrounding the hearing instrument, a signal processing unit for processing (modifying) the received sound signal to enhance the user's hearing, and an output transducer for outputting the modified sound signal. If the hearing system includes a further hearing instrument for the user's other ear, this further hearing instrument preferably also comprises at least one input transducer, a signal processing unit, and an output transducer.

[0014] The hearing instrument(s) of the hearing system are, in particular, of one of the designs described above (BTE device with internal or external output converter, ITE device, e.g., CIC device, hearing implant, especially cochlear implant, etc.). In the case of a binaural hearing system, both hearing instruments are preferably of the same type.

[0015] The input transducers are, in particular, each an acousto-electrical transducer that converts airborne sound from the environment into an electrical input audio signal. To enable direction-dependent analysis and processing of the recorded sound signal, the hearing system comprises at least two input transducers, which can be arranged in the same hearing instrument or—if present—distributed between the two hearing instruments of the hearing system. The output transducer is preferably designed as an electroacoustic transducer (receiver) that converts the audio signal, modified by the signal processing unit, back into airborne sound. Alternatively, the output transducer is designed to emit bone conduction sound or to directly stimulate the user's auditory nerve.

[0016] The signal processing unit preferably comprises a plurality of signal processing functions, e.g., any selection from the functions frequency-selective amplification, dynamic compression, spectral compression, direction-dependent attenuation (beamforming), noise reduction, in particular active noise cancellation (ANC), active feedback cancellation (AFC), and wind noise reduction, which are applied to the recorded sound signal, i.e., the input audio signal, to process it to support the user's hearing. Each of these functions, or at least most of them, can be parameterized by one or more signal processing parameters. A signal processing parameter is a variable that can be assigned different values ​​to influence the behavior of the associated signal processing function.In its simplest form, a signal processing parameter can be a binary variable used to switch the respective function on and off. In more complex cases, hearing aid parameters are formed by scalar floating-point numbers, binary or continuously variable vectors, or multidimensional arrays, etc. An example of such signal processing parameters is a set of gain factors for a number of frequency bands of the signal processing unit, which defines the frequency-dependent amplification of the hearing aid.

[0017] In the process carried out by the hearing system, a sound signal from the environment of the hearing device is picked up by the hearing aid's at least two input transducers. This sound signal contains, at least intermittently, the user's own voice as well as ambient noise. Here and in the following, "ambient noise" refers to the portion of the picked-up sound signal that originates from the environment (and is therefore different from the user's own voice). The picked-up sound signal (input audio signal) is modified in a signal processing step to support the user's hearing. The modified sound signal is then output via the hearing aid's output transducer.

[0018] According to the procedure, a first signal component and a second signal component are derived from the recorded sound signal (either directly or after preprocessing).

[0019] The first signal component (hereinafter also referred to as "user's own voice component") is derived in such a way that the user's own voice is emphasized in relation to the ambient noise; here, the ambient noise is selectively dampened (i.e., dampened to a greater extent than the user's own voice).

[0020] The second signal component (hereinafter also referred to as the "ambient noise component") is derived in such a way that the ambient noise is emphasized over the user's own voice; here, the user's own voice is selectively attenuated (i.e., attenuated to a greater extent than the ambient noise). Preferably, the user's own voice is completely removed from the second signal component, or at least to the extent technically possible with signal processing.

[0021] According to the method, the first signal component (user voice component) and the second signal component (ambient noise component) are processed differently in the signal processing step. In particular, the first signal component is amplified to a lesser extent and / or processed with modified dynamic compression (especially with reduced dynamic compression, i.e., with a more linear gain characteristic) compared to the second signal component. The first signal component is preferably processed in a manner optimized for processing the user's own voice (especially individually, i.e., user-specifically). The second signal component, on the other hand, is preferably processed in a manner optimized for processing the ambient noise. This processing of the second signal component is optionally, in turn, dependent on—for example—the user's own voice characteristics.The type of ambient noise (voice noise, music, driving noise, construction noise, etc.) determined within the framework of a classification of the listening situation varies.

[0022] After this differential processing, the first and second signal components are combined (superimposed) to generate the modified sound signal. However, the resulting overall signal can optionally undergo further processing steps, in particular further amplification, before being output by the output converter, as part of the invention.

[0023] The two signal components, namely the intrinsic voice component and the ambient noise component, are derived from the first and second sound signals in such a way that they overlap (completely or at least partially) in time. The two signal components therefore exist concurrently and are processed in parallel (i.e., on parallel signal processing paths). These signal components are thus not temporally successive intervals of the recorded sound signal.

[0024] The derivation of the first signal component is achieved using direction-dependent attenuation (beamforming), so that a spatial signal component corresponding to the ambient noise is selectively attenuated (i.e., attenuated more strongly than another spatial signal component where the ambient noise is absent or only weakly present). For this purpose, a static (time-invariant) attenuation algorithm (also called a beamforming algorithm or simply beamformer) can be used within the scope of the invention. Preferably, however, an adaptive direction-dependent beamformer is used, whose attenuation characteristic has at least one local or global attenuation maximum, i.e., at least one direction of maximum attenuation (notch). This notch (or optionally one of several notches) is preferably directed towards a dominant noise source in a room volume located behind the user's head.

[0025] The second signal component is also derived using direction-dependent attenuation, employing either a static or adaptive beamformer. The direction-dependent attenuation is applied in such a way that a spatial signal component corresponding to the user's own voice is selectively attenuated (i.e., attenuated more strongly than a spatial signal component in which the user's own voice is absent or only faintly audible). The notch of the corresponding beamformer is advantageously aligned precisely or approximately in front of the user's head. Specifically, a beamformer with an attenuation characteristic similar to an anti-cardioid is used.

[0026] At least the beamformer used to derive the second signal component preferably has a frequency-dependent attenuation characteristic. This dependence of the attenuation characteristic manifests itself in particular as a notch width, notch depth, and / or a notch direction that varies slightly with frequency. The frequency dependence of the attenuation characteristic is adjusted (e.g., empirically or using a numerical optimization method) such that the attenuation of the user's own voice in the second signal component is optimized (i.e., a local or global maximum is achieved), thus eliminating the user's own voice from the second signal component as effectively as possible. This optimization is performed, for example, during the individual fitting of the hearing system to the user, if a static beamformer is used to derive the second signal component.Alternatively, an adaptive beamformer is used to derive the second signal component. This beamformer continuously optimizes the attenuation characteristics of the hearing system during operation to achieve the best possible attenuation of the user's own voice. This approach is based on the understanding that a beamformer attenuates the user's own voice differently than the sound from a sound source positioned at a distance from the user. In particular, the user does not always perceive their own voice as coming precisely from the front. Rather, due to slight asymmetries in the anatomy of the head, the user's individual speech patterns, and / or the transmission of their own voice through bone conduction, many users perceive their own voice as originating from a direction that deviates from the plane of symmetry of the head.

[0027] Optionally, the attenuation characteristic of the beamformer used to derive the first signal component also exhibits a dependence on the frequency, whereby this dependence is determined in such a way that the attenuation of the ambient signal in the first signal component is optimized (i.e., a local or global maximum is achieved) and that the ambient signal is thus eliminated from the first signal component as effectively as possible.

[0028] Furthermore, and in particular in addition to the direction-dependent filtering described above, spectral filtering of the recorded sound signal is preferably used to derive the first signal component (user's own voice component) and the second signal component (ambient noise component). To derive the first signal component, preferably at least one frequency component of the recorded sound signal in which components of the user's own voice are absent or only weakly pronounced is selectively attenuated (i.e., more strongly attenuated than frequency components of the recorded sound signal in which the user's own voice has dominant components). To derive the second signal component, preferably at least one frequency component of the recorded sound signal in which components of the ambient noise are absent or only weakly pronounced is selectively attenuated (i.e.,more strongly damped than frequency components of the recorded sound signal in which the ambient noise has dominant components).

[0029] In a signal analysis step, intrinsic pitch intervals of the recorded sound signal are identified, e.g. by applying methods such as those known from US 2013 / 0148829 A1 or from WO 2016 / 078786 A1.

[0030] The separation of the recorded sound signal into the user's own voice component and the ambient noise component, and the parallel, different processing of the two signal components, is generally carried out both in detected user voice intervals and in the absence of the user's own voice. However, in this case, according to the invention, the derivation of the second signal component (i.e., the ambient noise component) is performed differently depending on the presence or absence of the user's own voice: In user voice intervals, this embodiment uses an algorithm optimized for the attenuation of the user's own voice to derive the ambient noise component, in particular – as described above – a static beamformer with an optimized frequency dependence of the attenuation characteristic or a self-optimizing dynamic beamformer.For intervals of the recorded sound signal that do not contain the user's own voice, a different (or at least differently parameterized) algorithm is applied to derive the ambient noise component. This algorithm is designed to attenuate a sound source located in front of the user but at a distance from the user (e.g., a speaker towards whom the user is facing). This different algorithm is, for example, implemented as a static beamformer with a direction-dependent attenuation characteristic corresponding to an anti-cardioid. This beamformer differs from the beamformer applied to the user's own voice intervals for deriving the ambient noise component with respect to the shape and / or frequency dependence of the anti-cardioid. For example, in the absence of the user's own voice, an anti-cardioid without frequency dependence (i.e., a non-frequency anti-cardioid) is used to derive the ambient noise component.a frequency-constant anti-cardioid signal is used. Preferably, the processing of the first signal component (which carries the user's own voice in natural pitch intervals) is also carried out differently depending on the presence or absence of the user's own voice: . In For natural voice intervals, the first signal component is preferably processed in a manner optimized for processing the user's own voice, as described above, while in the absence of the user's own voice, it is processed in a different manner.

[0031] The hearing system according to the invention is generally configured for the automatic execution of the method described above. The hearing system is thus configured to receive a sound signal from the environment of the hearing instrument by means of the at least one input transducer of the at least one hearing instrument, wherein the sound signal at least temporarily includes the user's own voice as well as ambient noise, to modify the received sound signal in the signal processing step to support the hearing ability of a user, and to output the modified sound signal by means of the output transducer of the hearing instrument.

[0032] The hearing system is further designed to derive the first signal component (own voice component) and the second signal component (ambient noise component), which overlaps with it in time, from the recorded sound signal in the manner described above, to process these two signal components in different ways in the signal processing step, and to combine them after this processing to generate the modified sound signal.

[0033] The configuration of the hearing system for the automatic execution of the method according to the invention is of a software and / or circuitry nature. The hearing system according to the invention thus comprises software and / or circuitry (hardware, e.g., in the form of an ASIC) that automatically execute the method according to the invention during operation of the hearing system. The software and / or circuitry for carrying out the method can be located exclusively in the hearing instrument(s) of the hearing system. Alternatively, the software and / or circuitry for carrying out the method can be distributed across the hearing instrument(s) and at least one other device or software component of the hearing system.For example, the programming resources for carrying out the procedure are distributed across at least one hearing instrument of the hearing system and a control program installed on an external electronic device (especially a smartphone).

[0034] The embodiments of the method according to the invention described above correspond to corresponding embodiments of the hearing system according to the invention. The preceding descriptions of the method according to the invention are transferable to the hearing system according to the invention and vice versa.

[0035] Exemplary embodiments of the invention are explained in more detail below with reference to a drawing. The drawing shows: Fig. 1 shows a schematic representation of a hearing system consisting of a single hearing instrument in the form of a hearing aid worn behind the user's ear, in which a sound signal picked up from the environment of the hearing aid is separated into a voice component and a temporally overlapping ambient noise component, and in which these two signal components are processed differently and then recombined; Fig. 2 shows a schematic block diagram of signal processing in the hearing instrument; and Figs. 3 and 4 show two schematic diagrams of attenuation characteristics of two direction-dependent attenuation algorithms (beamformers) used in the hearing aid. Fig. 1 They can be used to derive the natural voice component or the ambient noise component from the recorded sound signal.

[0036] Identical parts and sizes are always marked with the same reference symbols in all figures.

[0037] Fig. 1 Figure 2 shows a hearing system with a single hearing aid 4, i.e., a hearing instrument designed to support the hearing ability of a hearing-impaired user. In this example, the hearing aid 4 is a behind-the-ear (BTE) hearing aid worn by the user.

[0038] Optionally, in further embodiments of the invention, the hearing system 2 comprises a second hearing aid (not explicitly shown) for supplying the user's second ear, and / or a control app that can be installed on the user's smartphone. In these embodiments, the functional components of the hearing system 2 described below are preferably distributed between the two hearing aids or between the at least one hearing aid and the control app.

[0039] The hearing aid 4 comprises, within a housing 5, at least one microphone 6 (in the illustrated example, two microphones 6) as an input transducer and a receiver 8 as an output transducer. When worn behind the user's ear, the two microphones 6 are oriented such that one microphone 6 points forward (i.e., in the user's line of sight), while the other microphone 6 points backward (against the user's line of sight). The hearing aid 4 further comprises a battery 10 and a signal processing unit in the form of a digital signal processor 12. Preferably, the signal processor 12 comprises both a programmable subunit (for example, a microprocessor) and a non-programmable subunit (for example, an ASIC). The signal processor 12 comprises a (self-recognition) unit 14 and a (signal separation) unit 16.Additionally, the signal processor 12 has two parallel signal processing paths 18 and 20.

[0040] Preferably, units 14 and 16 are designed as software components that are implemented in a runnable manner in the signal processor 12. The signal processing paths 18 and 20 are preferably formed by electronic hardware circuits (e.g., on the aforementioned ASIC).

[0041] The signal processor 12 is supplied with an electrical supply voltage U from the battery 10.

[0042] In normal operation of the hearing aid 4, the microphones 6 pick up sound from the environment of the hearing aid 4. The microphones 6 convert the sound into an (input) audio signal I, which contains information about the picked-up sound. The input audio signal I is fed to the signal processor 12 inside the hearing aid 4.

[0043] The signal processor 12 processes the input audio signal I in the signal processing paths 18 and 20, each using a plurality of signal processing algorithms, for example Noise and / or feedback suppression, dynamic compression and frequency-dependent amplification based on audiogram data, to compensate for the user's hearing loss. The specific operation of the signal processing algorithms, and thus of the signal processor 12, is determined by a multitude of signal processing parameters. The signal processor 12 outputs an audio signal O, containing information about the processed and thus modified sound, to the listener 8. The two signal processing paths 18 and 20 are preferably structured identically, i.e., they have the same signal processing algorithms, but these are parameterized differently – for processing the user's own voice and for processing ambient noise, respectively.

[0044] The receiver 8 converts the output sound signal O into a modified airborne sound. This modified airborne sound is transmitted into the user's ear canal via a sound channel 22, which connects the receiver 8 to a tip 24 of the housing 5, and via a (not explicitly shown) flexible sound tube, which connects the tip 24 to an earpiece inserted into the user's ear canal.

[0045] The functional interconnection of the components of the signal processor 12 described above is shown in Fig. 2 illustrated.

[0046] The input audio signal I (and thus the recorded sound signal) is fed to the voice recognition unit 14 and the signal separation unit 16.

[0047] The voice recognition unit 14 detects, for example by applying one or more of the methods described in US 2013 / 0148829 A1 or WO 2016 / 078786 A1, whether the input audio signal I contains the user's own voice. A status signal V, dependent on the result of this check (thus indicating whether the input audio signal I contains the user's own voice or not), is then forwarded by the voice recognition unit 14 to the signal separation unit 16.

[0048] Depending on the value of the status signal V, the signal separation unit 16 processes the incoming audio signal I in different ways. During intervals when the voice recognition unit 14 has detected the user's own voice in the input audio signal I, the signal separation unit 16 derives a first signal component (or voice component) S1 and a second signal component (or ambient noise component) S2 from the input audio signal I and routes these temporally overlapping signal components S1 and S2 to the parallel signal processing paths 18 and 20, respectively. In intervals when the input audio signal I does not contain the user's own voice, the signal separation unit 16 routes the entire input audio signal I to signal path 20.

[0049] As in the Figures 3 and 4As illustrated, the signal separation unit 16 derives the first signal component S1 and the second signal component S2 from the input audio signal I by applying different beamformers 26 and 28 respectively (i.e., different algorithms for direction-dependent attenuation).

[0050] In Fig. 3An exemplary attenuation characteristic G1 of the beamformer 26 used to derive the first signal component (intrinsic voice component) S1 is shown. In the example shown, the beamformer 26 is an adaptive (i.e., continuously variable during operation of the hearing system 2) algorithm with two symmetrically variable notches 30 (i.e., directions of maximum attenuation). The attenuation characteristic G1 is set such that one of the notches 30 is directed towards a dominant sound source 32 in a room volume located behind the user's head 34. The dominant sound source 32 is, for example, a speaker standing behind the user. Due to the Fig. 3With the damping characteristic G1 shown, the noise source 32, which contributes significantly to the ambient noise, is completely or almost completely eliminated in the first signal component S1. In contrast, the components of the input audio signal I coming from the front with respect to the head 34, in particular the user's own voice, are emphasized.

[0051] In Fig. 4In contrast, an attenuation characteristic G2 of the beamformer 28 used to derive the second signal component (ambient noise component) S2 is shown as an example. This attenuation characteristic G2 is, in particular, static (i.e., unchanged over time after the hearing aid 4 has been individually fitted to the user) and corresponds, for example, to an anti-cardioid characteristic. A notch 36 of the attenuation characteristic G2 is oriented towards the front of the user's head 34, so that the user's own voice is at least largely filtered out from the second signal component S2.

[0052] In addition, the damping characteristic G2 of the Beamformer 28 varies depending on the frequency, so that the user's own voice is optimally dampened. In the in Fig. 4In the case shown, the attenuation characteristic G2, corresponding to an anti-cardioid, is created by superimposing (i.e., summing, weighted or unweighted) the signal from the forward-facing microphone 6 and the time-delayed signal from the rear-facing microphone 6. The time delay is specified as a frequency-dependent function, thus optimizing the attenuation of the user's own voice in the second signal component. An audiologist determines this optimized frequency dependence of the time delay during a training session as part of the hearing aid fitting process.

[0053] In an alternative embodiment, the beamformer 28 is adaptive, wherein the attenuation characteristic G2 is adjusted during the operation of the hearing system 2 by the signal processor 12 (e.g. by minimizing the output energy of the beamformer 28 in natural tuning intervals).

[0054] In signal processing paths 18 and 20, the first signal component S1 and the second signal component S2 are processed differently. Preferably, the same signal processing algorithms are applied to the first signal component S1 and the second signal component S2, albeit with different parameter settings. For processing the first signal component S1, a parameter set of signal processing parameters is used that is optimized for processing the user's own voice (especially with individual customization to the specific user). Among other things, the first signal component S1, which contains the user's own voice, is amplified to a lesser degree than the second signal component S2 (or even not amplified at all). Furthermore, less dynamic compression (i.e., a more linear gain characteristic) is applied to signal component S1 than to signal component S2.

[0055] Signal processing paths 18 and 20 output processed and thus modified signal components S1' and S2', respectively, to a recombination unit 38, which combines the modified signal components S1' and S2' (in particular, summing them in a weighted or unweighted manner). The resulting output audio signal O is output by the recombination unit 38 (directly or indirectly via further processing steps) to the listener 8.

[0056] The invention becomes particularly clear in the embodiments described above, but is nevertheless not limited to these embodiments. Rather, further embodiments of the invention can be derived by a person skilled in the art from the claims and the preceding description. Reference symbol list

[0057] 2 Hearing system 4 Hearing aid 5 Housing 6 Microphone 8 Receiver 10 Battery 12 Signal processor 14 (Self-recognition) unit 16 (Signal separation) unit 18 Signal processing path 20 Signal processing path 22 Sound channel 24 Tip 26 Beamformer 28 Beamformer 30 Notch 32 Sound source 34 Head 36 Notch 38 Recombination unit G1 Attenuation characteristic G2 Attenuation characteristic I (Input) audio signal O (Output) audio signal S1, S1' (First) signal component S2, S2' (Second) signal component U Supply voltage V Status signal

Claims

1. A method for operating a hearing system (2) for assisting the hearing of a user, having at least one hearing instrument (4) worn in or on the ear of the user, - wherein by means of an input transducer of the hearing instrument (4) and a further input transducer of the same hearing instrument (4) or a further hearing instrument of the hearing system (2), a sound signal is recorded from the surroundings of the hearing instrument (4), which at least temporarily contains the user's own voice and ambient noise, - wherein the recorded sound signal is modified in a signal processing step to assist the hearing of the user, - wherein the modified sound signal is output by means of an output transducer (8) of the hearing instrument (4), - wherein a first signal component (S1) and a second signal component (S2) are derived from the recorded sound signal, wherein the signal components (S1, S2) chronologically overlap, -- wherein in the first signal component (S1), the user's own voice is highlighted in relation to the ambient noise, and -- wherein in the second signal component (S2) the ambient noise is highlighted in relation to the user's own voice, - wherein to derive the first signal component (S1) by means of direction-dependent damping, a spatial signal component corresponding to the ambient noise is selectively damped, and - wherein the first signal component (S1) and the second signal component (S2) are combined after the processing to generate the modified sound signal, - wherein the first signal component (S1) and the second signal component (S2) are processed in different ways in the signal processing step, - wherein to derive the second signal component (S2) by means of direction-dependent damping, a spatial signal component corresponding to the ego voice component is selectively damped, characterized in that - in a signal analysis step, ego voice intervals of the recorded sound signal are identified, in which the recorded sound signal contains the user's own voice, wherein the derivation of the second signal component (S2) takes place differently depending on the presence or absence of the user's own voice, - in that in ego voice intervals for deriving the second signal component (S2), an algorithm optimized for damping the ego voice is used, and - in that an algorithm different therefrom is applied to intervals of the recorded sound signal which do not contain the user's own voice to derive the second signal component (S2), which algorithm is oriented to damping a noise source arranged on the front side of the user but at a distance from the user.

2. The method as claimed in claim 1, wherein to derive the first signal component (S1), a direction of maximum damping is oriented on a dominant noise source (32) in a rear spatial volume with respect to the head (34) of the user.

3. The method as claimed in claim 1 or 2, wherein in intervals which do not contain the user's own voice, to derive the second signal component (S2), a direction of maximum damping with respect to the head (34) of the user is oriented exactly or approximately to the front side.

4. The method as claimed in any one of claims 1 to 3, wherein in ego voice intervals, the direction-dependent damping used to derive the second signal component (S2) has a spatial damping characteristic which is dependent on the frequency of the recorded sound signal such that the damping of the ego voice is optimized.

5. The method as claimed in any one of claims 1 to 4, wherein to derive the first signal component (S1), at least frequency component of the recorded sound signal in which components of the user's own voice are not present or are only weakly pronounced are selectively damped.

6. The method as claimed in any one of claims 1 to 5, wherein to derive the second signal component (S2), at least one frequency component of the recorded sound signal in which components of the ambient noise are not present or are only weakly pronounced is selectively damped.

7. The method as claimed in any one of claims 1 to 6, wherein the first signal component (S1) is amplified to a lesser extent and / or processed using different dynamic compression in the signal processing step than the second signal component (S2).

8. A hearing system (2) for assisting the hearing of a user having at least one hearing instrument (4) worn in or on the ear of the user, wherein the hearing instrument (4) comprises: - an input transducer (6) for recording a sound signal from the surroundings of the hearing instrument (4), - a signal processing unit (12) for modifying the recorded sound signal to assist the hearing of the user, and - an output transducer (8) for outputting the modified sound signal, wherein the hearing system (2) is configured to automatically carry out the method as claimed in any one of claims 1 to 7.

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