Method for suppressing the inherent microphone noise of a hearing instrument

DE102025105191A1Undetermined Publication Date: 2026-08-13SIVANTOS PTE LTD
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Application Number
DE102025105191
Authority / Receiving Office
DE · DE
Patent Type
Applications
Current Assignee / Owner
Filing Date
2025-02-12
Publication Date
2026-08-13

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Abstract

The invention describes a method for suppressing the inherent noise of a microphone arrangement (2) of a hearing instrument (1), which comprises at least one first microphone (M1), wherein a first microphone signal (x1) is generated from ambient sound by the first microphone (M1), wherein a minimum signal level (min1) in the first microphone signal (x1) is monitored over an operating phase of the hearing instrument (1), and a first background noise (nb1) is determined on the basis of the minimum signal level (min1), wherein an internal noise limit (Ltot) of the hearing instrument (1) is determined on the basis of the first background noise (nb1), and wherein the inherent noise of the first microphone (M1) is suppressed as a function of an activation in which a current level value of the first microphone signal (x1) or a signal derived therefrom (Z, R) and the internal noise limit (Ltot) are taken into account.
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Description

The invention relates to a method for suppressing the inherent noise of a microphone arrangement of a hearing instrument, which comprises at least one first microphone for generating a first microphone signal from a sound signal of the environment. Hearing aids are typically used to compensate for hearing loss or general hearing impairment. A hearing aid usually includes one or more microphones to generate corresponding microphone signals from the ambient sound. The generated microphone signal(s) are processed depending on the hearing impairment being compensated for, and are, for example, amplified in a frequency-band-specific manner and often subjected to noise reduction, which, with two or more microphone signals, can also be achieved using directional microphones. An output signal is generated from the processed microphone signal(s), which is then delivered as an audio signal to the hearing aid wearer's ear by an output transducer, such as a loudspeaker or bone conduction receiver. Particularly quiet signals are often amplified during signal processing. This can occur on the one hand after the relevant signal components are identified as the desired signal (e.g., quiet speech), but background noise can also be amplified, i.e., boosted for output, especially when there are also desired signals in an environment where spatial hearing perception should be affected as little as possible by directional microphones. However, when amplifying quiet signals, the inherent noise of the microphone(s), whether electronically or electroacoustically induced, can also be amplified and thus included in the output signal, potentially impairing sound quality. If the signal components generated by ambient noise in a microphone signal are stronger than the microphone's inherent noise, they mask the noise, which is why it is usually only perceptible in the output signal at low volume levels. Hearing aids therefore often employ algorithms to suppress their own noise, which usually operate based on measured or estimated sound levels and / or background noise levels. A key challenge here is identifying the inherent noise to be suppressed, or distinguishing it from other low-level signals. This is often achieved by measuring noise characteristics for individual microphone types during factory calibrations, and then using these noise characteristics for limit comparisons during the operation of a hearing aid. However, especially with highly sensitive microphones, fluctuations in the inherent noise can occur due to manufacturing tolerances, which are not negligible compared to the noise characteristics. The invention is therefore based on the objective of providing a method for suppressing the inherent noise of a microphone arrangement of a hearing instrument, which suppresses said inherent noise as precisely as possible, and does not impair signal components generated by the microphone arrangement from ambient sound, even at low signal levels. The aforementioned problem is solved according to the invention by a method for suppressing the inherent noise of a microphone arrangement of a hearing instrument, which comprises at least one first microphone, wherein a first microphone signal is generated from ambient sound by the first microphone, wherein a minimum signal level in the first microphone signal is monitored over an operating phase of the hearing instrument, and a first noise background is determined on the basis of the minimum signal level, wherein an internal noise limit of the hearing instrument is determined on the basis of the first noise background, and wherein the inherent noise of the first microphone is suppressed as a function of an activation in which a current value of the first microphone signal or a signal derived therefrom and the internal noise limit are taken into account. Advantageous and, in some cases, inventive embodiments are the subject of the following description and the subclaims. A hearing instrument, in this context, generally includes any device that, on the one hand, has a microphone arrangement with at least one microphone for generating a microphone signal from ambient sound, and which, on the other hand, is designed to generate an audio signal from an electrical signal—which may, in particular, be the microphone signal or a signal derived therefrom—and transmit it to the ear of a wearer of this device. A hearing instrument can therefore, in particular, be a headphone (e.g., an "earbud"), a headset, a communication device, smart glasses with a speaker, etc.However, a hearing instrument also includes a hearing aid in the narrower sense, i.e., a device intended to compensate for a hearing impairment of the wearer, in which a microphone signal generated from ambient sound by means of a microphone is processed into an output signal and thereby amplified in a frequency-band-dependent manner, and an output sound signal generated from the output signal by means of a loudspeaker or similar is suitable to compensate, in particular in a user-specific manner, at least partially for the hearing impairment of the wearer. A microphone arrangement of the hearing instrument includes, in particular, the entirety of microphones of the hearing instrument, which are arranged in the hearing instrument and generate respective microphone signals from the incident ambient sound, which are then processed accordingly in the hearing instrument. The term "microphone" here generally encompasses any form of electro-acoustic transducer which, by its design and construction, is suitable and configured to generate an electrical signal from ambient sound. In this signal, voltage, current, and / or power fluctuations correspond at least approximately to the fluctuations in air pressure caused by the ambient sound. The electrical signal generated by such an electro-acoustic transducer is accordingly referred to here generally as a microphone signal. In particular, the term "microphone" also includes a microphone in the narrower or strict sense, in which the vibrations of changing air pressure are converted into an electrical voltage signal (microphone signal) by means of a diaphragm. An operating phase of the hearing instrument includes, in particular, a longer period of time during which the hearing instrument is operated according to its technical specifications (i.e., using an operating program or similar, but not necessarily worn on the body), and during which the first microphone signal is generated from the ambient sound using the first microphone. An operational phase can occur, in particular, through testing, which the manufacturer conducts during the production of the hearing aid before delivery to the customer (or dealer, etc.), for example, for quality control. During testing, the correct technical functionality of the hearing aid is specifically verified. This operational phase can also occur during a fitting process, for example, at an audiologist's or hearing aid acoustician's office, as an adjustment phase. In this phase, individual settings of the hearing aid are preferably customized to the wearer and their specific audiological requirements, and these settings are then verified. The operating phase can also be, in particular, normal operation, in which the hearing instrument is worn by the wearer in or on the ear as intended and is used in accordance with the technical specifications (i.e., in particular by means of one of the provided and set up operating programs). In particular, the operating phase, depending on its type, is at least 10 seconds, preferably at least one minute, and most preferably at least five minutes. In particular, the operating phase in normal operation can also be at least one hour, preferably at least one day. Tracking a minimum signal level in the first microphone signal throughout the operating phase of the hearing aid means, in particular, that a so-called minimum tracker is applied to the signal level of the first microphone signal, at least for the duration of the operating phase, and the minimum signal level reached so far is recorded and, in particular, stored. Preferably, this data is stored on non-volatile memory within the hearing aid, such as an EEPROM or similar. If the previously stored minimum signal level, i.e., the currently valid minimum value of the signal level, is undercut during the operating phase by a new value of the signal level of the first microphone signal, this new value is stored as the minimum signal level (and thus the previously valid value is overwritten).In particular, a minimum value of the signal level stored in a non-volatile memory remains there if, in a new operating phase, this stored minimum value is not undercut by a current value of the signal level at any point during the new operating phase. The minimum signal level is used here, in particular, as a measure of the inherent noise of the first microphone in the first microphone signal. It is assumed that the minimum signal level throughout the operating phase corresponds to a signal in which the ambient noise component is as small as possible (and preferably negligible or even nonexistent). For this purpose, particularly during an operating phase such as a test or calibration phase, a sound path for ambient noise to the first microphone can be physically blocked (e.g., by a suitable cover or similar device). Accordingly, assuming that the minimum signal level should correspond to a signal without significant ambient noise components, a first noise background for the first microphone or microphone signal is generated from the minimum signal level of the first microphone signal. Preferably, the minimum signal level can be linearly incorporated into the first noise background, and in particular, the first noise background can optionally include a preferably constant or at least temporarily constant normalization factor (i.e., preferably constant over a time scale of at least 10 ms, particularly preferably at least 30 ms).The normalization factor in question can thus effect either a static normalization (for example, in the case of a frequency-dependent sensitivity of the first microphone for a frequency-bandwise implementation of the method, or to take into account different sensitivities of other microphones in the hearing instrument), or a dynamic normalization (for example, if a hair or similar temporarily partially covers the first microphone), but on a time scale that is preferably at least one and particularly preferably at least two orders of magnitude slower than the verification of the minimum signal level. Based on the initial noise background thus determined, the internal noise threshold is then calculated. This threshold is used to activate the suppression of the first microphone's inherent noise, i.e., the suppression of the signal components of the first microphone signal that are based on this inherent noise. These signal components of the first microphone signal, based on the first microphone's inherent noise, can be suppressed directly in the first microphone signal itself or in a signal derived from the first microphone signal (i.e., in particular, an internal intermediate signal of the hearing instrument's signal processing). Advantageously, the initial noise background is monotonically incorporated into the internal noise threshold; that is, as the initial noise background decreases—and especially as the minimum signal level for the first microphone signal decreases over the operating phase—the internal noise threshold also decreases, or at least remains constant (and does not increase). The underlying principle here is that the internal noise threshold advantageously represents an "effective" noise floor for further signal processing, which is "seen" by the subsequent signal processing and is assumed to exist in the absence of signal components other than the inherent noise of the first microphone. This absence of other signal components is modeled by tracking the minimum signal level. Therefore, if the current value for the signal level of the first microphone signal lies within the range of this "effective" noise background, there are no other dominant signal components that would clearly mask the self-noise of the first microphone, so that the self-noise is perceptibly introduced into the first microphone signal and its further processing, which is undesirable. Accordingly, in this case, preferably by comparing the said internal noise limit with the current level value of the first microphone signal or a signal derived therefrom, in particular an intermediate signal of the signal processing of the hearing instrument, the suppression of the self-noise of the first microphone is activated, e.g. by a spectral subtraction of the signal components of the self-noise from the first microphone signal and / or by a Wiener filter. This activation case is therefore detected based on the internal noise threshold, for which a certain level difference is preferably left over from a calculated noise background for the said comparison with the current level value, in order to cover as best as possible the range in which the current sound level is unable to cover a noise background given by the intrinsic noise and estimated on the basis of the minimum signal level and possibly other parameters (su). The internal noise limit can, in particular, take into account further parameters of signal processing in the hearing instrument that are not directly attributable to the first microphone signal and that can influence a portion of the inherent noise of the first microphone in an output signal generated by said signal processing, i.e., in particular parameters of directional microphones that use the first microphone signal and a second microphone signal generated by a second microphone, and / or parameters of user-specific (and in particular frequency-bandwise) signal amplification to compensate for a hearing impairment of the wearer. Preferably, the operating phase for monitoring the minimum signal level in the first microphone signal is provided during a fitting process and / or during test operation within the context of production and / or during normal operation. In particular, an operating phase can last the entire normal operation (i.e., in which the hearing instrument is worn by the user as intended in or on the ear and used according to the technical specifications, and otherwise remains switched off or is not used in a standby mode), so that the monitoring of the minimum signal level continues continuously, and the minimum signal level is also maintained after breaks in normal operation (e.g., due to overnight removal and switching off) when normal operation is resumed.However, a reset of a stored value of the minimum signal level can be particularly advantageous if the first microphone in the hearing instrument is replaced or otherwise mechanically altered (e.g., during a repair). Favorably, the minimum signal level linearly influences the internal noise limit over the operating phase. This means, in particular, that the internal noise limit Ltot (min1), under otherwise constant conditions (given by any other factors and / or additional microphone signals), is a linear function of the minimum signal level min1 of the first microphone signal (especially with respect to actual level values, i.e., not in dB), meaning that Ltot (a · min1) = a · Ltot (min1) holds true for any factor a. Advantageously, when a new minimum signal level of the first microphone signal is reached during the operating phase, the minimum signal level is updated accordingly. This is also known as minimum tracking. In particular, at the end of the operating phase, the then-current minimum signal level can be retained for the next operating phase (e.g., by storing it in non-volatile memory). This takes into account the fact that a spontaneous increase in the sensitivity of the first microphone, which would otherwise justify a reset of the current minimum value, is not to be expected in practice. It is further advantageous if, for signal processing of the first microphone signal during one or more operating phases of the hearing instrument, a first normalization gain that is at least temporarily constant is applied, and the internal noise limit is determined as a function of the first normalization gain. Such a first normalization gain can be applied, in particular, when the method is implemented frequency-band by frequency, in order to obtain the flattest possible sensitivity or frequency response of the first microphone by means of a frequency-band-dependent first normalization gain, and / or to compensate for any differences in the sensitivities of the two microphones if a second microphone is also used in the hearing instrument.Since the first normalization gain is incorporated into the signal processing of the first microphone signal, and thus remains inextricably linked to it for subsequent processing steps, any resulting increase or decrease in the inherent noise of the first microphone must be taken into account, even if this inherent noise is suppressed. Specifically, the first normalization gain gn1 can be multiplied by a factor on the minimum signal level min1 of the first microphone signal x1 to arrive at the first background noise nb1, i.e., (possibly up to a time-constant factor) nb1 = min1 · gn1. In a further advantageous embodiment, the tracking of the minimum signal level in the first microphone signal, the determination of the initial background noise, the determination of the internal noise limit of the hearing instrument, and the activation of the suppression of any inherent noise of the first microphone are each performed independently in a plurality of frequency bands. This means, in particular, that the first microphone signal is divided into said plurality of frequency bands, and in each frequency band the aforementioned process steps (tracking the minimum signal level, determining the initial background noise and the internal noise limit, and activating the suppression of any inherent noise) are performed separately, i.e., as independent and self-contained processes.This takes into account, on the one hand, the fact that in the hearing instrument, the signal from the first microphone is usually already divided into multiple frequency bands for signal processing, and that, due to frequency-dependent amplification, any inherent noise of the first microphone is incorporated into the processed output signal to varying degrees. On the other hand, the inherent noise of the first microphone itself can also be frequency-dependent, which is why frequency-band-wise suppression is advantageous here (since the suppression is applied in a more targeted and needs-based manner). In this case, the first normalization gain is also frequency-dependent (or may have different values ​​in the individual frequency bands). Advantageously, for the activation in question, a frequency-band-specific value from an audiogram of the hearing aid user and / or a total gain applied to the first microphone signal in a respective frequency band is also used. The total gain applied to the first microphone signal in the respective frequency band refers specifically to the cumulative gain of the first microphone signal for the relevant frequency band along the entire signal path from the first microphone to an output transducer (e.g., a loudspeaker) of the hearing aid. For example, considering the user's audiogram can prevent the application of an algorithm to suppress the inherent noise of the first microphone (and potentially other microphones of the hearing aid) when this would no longer be necessary given the user's hearing ability in a given frequency band. A similar principle applies to considering the overall gain in a frequency band, since noise suppression of the first microphone's inherent noise in frequency bands with very low (or no) overall gain, such as in low frequency bands with an open coupling of the hearing aid, has no significant effect. Such an algorithm, especially with quiet signals and thus low signal levels, can fundamentally impair signal quality or auditory perception by introducing artifacts into the first microphone signal (or a derived signal). Therefore, suppressing the inherent noise of the first microphone is limited to those cases where the wearer, due to their hearing ability and / or signal amplification applied to the first microphone signal, would actually be able to perceive the inherent noise. It is further advantageous if a second microphone in the hearing instrument generates a second microphone signal from the ambient sound signal, whereby a minimum signal level in the second microphone signal is monitored throughout the operating phase of the hearing instrument, and a second background noise level is determined based on this minimum signal level, and the internal noise threshold of the hearing instrument is also determined based on this second background noise level. This means, in particular, that the hearing instrument comprises two microphones, and the internal noise threshold, at which suppression of the inherent noise of at least the first microphone is activated, is determined based on the minimum signal levels of the two corresponding microphone signals.The corresponding statements regarding the first microphone and the associated signal quantities apply analogously to the second microphone, the second microphone signal, the second background noise and the minimum signal level. In particular, the activation of the self-noise suppression also incorporates the current value of the second microphone signal or a signal derived from it. This includes, in particular, that the internal noise threshold is calculated, possibly down to a constant linear factor and / or a constant offset, based on a specific mathematical function F (nb1, nb2) of the first and second background noise nb1, nb2 (where the minimum signal levels min1, min2 of the first and second microphone signals may also include associated first and second normalization gains), and that self-noise suppression of at least the first microphone is activated by comparing the internal noise threshold with the same mathematical function F (x1 (t), x2 (t)) of the current values ​​of the first and second microphone signals x1 (t), x2 (t). Particularly preferably (especially frequency band-wise) the internal noise limit Ltot is formed as a function of a sum nb1 + nb2 of the first and the second noise background nb1, nb2. Advantageously, depending on the activation, the inherent noise of the second microphone—that is, signal components of the second microphone signal that are based on the inherent noise of the second microphone—is also suppressed. These signal components can be suppressed directly in the second microphone signal or in a signal derived from the second microphone signal. Preferably, the suppression of the inherent noise occurs in an internal intermediate signal of the hearing instrument's signal processing, which is generated using directional microphones of the first and second microphone signals (or intermediate signals derived from each). In particular, the inherent noise of all microphones in the hearing instrument is suppressed. Advantageously, the operation of the hearing instrument involves directional signal processing of the first and second microphone signals, whereby the activation of the noise suppression of at least the first microphone also occurs depending on this directional signal processing. This includes, in particular, generating a directional signal as an internal intermediate signal of the hearing instrument based on the first and second microphone signals, from which an output signal of the hearing instrument is derived (possibly through further, especially non-directional, signal processing steps). In generating the directional signal R(t), signal components of the first microphone signal x1(t) and the second microphone signal x2(t) are superimposed, possibly with a time delay relative to each other, e.g., in the form of This superposition allows the inherent noise components of the two microphone signals x1, x2 to be attenuated or amplified, particularly depending on the respective noise power spectrum of the two microphones. Therefore, the directional signal processing of the two microphone signals should be given priority when activating suppression of the intrinsic (and inherently independent of the subsequent signal processing of the microphone signals) inherent noise. It proves further advantageous if a co-modulation parameter dependent on the directivity of the directional signal processing is determined, and the internal noise limit is also determined as a function of the co-modulation parameter, and / or if a current value of a directional signal resulting from said directional signal processing is included in the activation. This includes, in particular, generating a directional signal as per equation (i), where the directivity (and thus also the respective angles of maximum and minimum sensitivity) depends in particular on the superposition parameter α and the time delay τ. Depending on this directivity, and in particular indirectly depending on a and τ, the co-modulation parameter K(a, τ) can now be determined, which, for example, indicates how far the directional signal R differs from the original first (or second) microphone signal x1 (or τ).x2) an intrinsic noise is reduced or increased with a given level or noise power. The internal noise limit is then also determined as a function of this comodulation parameter K. In particular, for the internal noise limit Ltot, this comodulation parameter K can be multiplied by the sum of the first and second background noise nb1, nb2, i.e., preferably. In particular, the internal noise limit can also be compared with a current value R (t) of the directional signal, and depending on the said comparison, the suppression of the intrinsic noise can be activated. It is further advantageous if the suppression of the self-noise of at least the first microphone is achieved using a Wiener filter and / or spectral subtraction, specifically if a Wiener filter and / or spectral subtraction is applied to the first (or second) microphone signal or to a signal derived from the first (and possibly second) microphone signal to suppress the signal components of the first microphone that are based on its self-noise (and preferably also the signal components of the second microphone that are based on the self-noise of the second microphone). A Wiener filter can boost or attenuate individual frequency ranges depending on an estimated noise component and an estimated signal component. The invention further describes a hearing instrument comprising at least one first microphone for generating a first microphone signal from ambient sound, and a signal processing device configured to suppress intrinsic noise of the first microphone using the method described above. Preferably, the signal processing device for carrying out the method is equipped with at least one corresponding signal processor for performing computational operations required in the method, as well as with working memory addressable by the signal processor, and particularly preferably with a further non-volatile memory for storing program code for algorithms of the method. In particular, a microphone arrangement of the hearing instrument also includes a second microphone for generating a second microphone signal. The hearing instrument according to the invention shares some of the advantages of the method according to the invention. The advantages stated for the method and its further developments can be transferred analogously to the hearing instrument. An embodiment of the invention is explained in more detail below with reference to a drawing. Here, Fig. 1 schematically shows a block diagram of a listening instrument with suppression of the microphones' inherent noise, and Fig. 2 shows a diagram of an internal noise limit as a control variable for the suppression of the inherent noise according to Fig. 1. Corresponding parts and sizes are marked with the same reference symbols in all figures. Figure 1 shows a block diagram of a hearing aid 1 with a microphone arrangement 2. The microphone arrangement 2 comprises a first microphone M1 and a second microphone M2. The first microphone M1 is configured to generate a first microphone signal x1 from ambient sound 8. Similarly, the second microphone M2 is configured to generate a second microphone signal x2 from the ambient sound 8. The first microphone signal x1 and the second microphone signal x2 are fed to a signal processing unit 10, which includes computing and storage means (not shown) in the form of one or more signal processors, RAM modules, etc., and in which the two microphone signals x1, x2 are processed, preferably taking into account any hearing impairment of the user of the hearing aid 1 that needs to be compensated for. In the signal processing unit 10, an output signal 12 is generated from the first microphone signal x1 and the second microphone signal x2. This output signal is converted into an output sound signal 16 by an output transducer 14 of the hearing instrument 1, which in this case is a loudspeaker. The output sound signal 16 is delivered to the ear (not shown) of the wearer of the hearing instrument 1. A bone conduction transducer or any other electro-acoustic transducer capable of generating a sound signal from the output signal 12 can be used as the output transducer. For this purpose, the first and second microphone signals x1, x2 are each decomposed into corresponding frequency bands by a first and second filter bank FB1, FB2, respectively, so that the subsequent signal processing in the signal processing unit 10 is carried out, in particular, frequency band by frequency. This is shown in Fig.1 indicated by the use of double arrows for signals with multiple frequency band channels. The output signal 12 is generated from an intermediate signal Z, which is created based on the first and second microphone signals x1, x2 and is given here by the directional signal R, as defined, for example, in equation (i). Additional signal processing steps, such as further boosting or attenuation of individual frequency bands depending on an audiogram of the carrier, may be performed to generate the output signal 12 from the directional signal R before the output signal 12 is generated at the synthesis filter bank FBS. Before the directional signal R is generated, a frequency-band-dependent first or second signal is applied to each of the individual frequency bands of the first and second microphone signals x1, x2.A second normalization gain, gn1, gn2, is applied, which in particular compensates for frequency-band differences in the sensitivities of the two microphones M1, M2, and can also compensate for any possible frequency dependence of a transfer function of a respective microphone M1, M2. The actual generation of the directional signal R takes place in a directional signal processing circuit 30. To suppress the inherent noise of the microphone arrangement 2, i.e., to suppress the signal components in the first (or second) microphone signal x1 (or x2) and thus in the output signal 12 that are based on the inherent noise of the first (or second) microphone M1 (or M2), an inherent noise suppression 22 is applied to the directional signal R. In the inherent noise suppression 22, which, like the other program steps, can preferably be implemented in the signal processing unit 10 as a corresponding software module or by corresponding hard-wired circuits (for example, as an ASIC), the aforementioned signal components of the first and / or second microphone signal x1, x2, which are based on the inherent noise of microphones M1, M2, are suppressed in the intermediate signal R by means of a Wiener filter 26, the application of which to the intermediate signal R is activated depending on an activation logic 28. Such a Wiener filter is shown, for example, in US 2018 / 0139546A1. For this purpose, a signal level and a noise power are determined frequency-wise in the directional signal R. Based on the signal level and the noise power, a gain factor is determined in the Wiener filter 26 using a filter function whose arguments are the two aforementioned quantities. This gain factor, applied frequency-wise to the intermediate signal Z (i.e., the directional signal R), suppresses the inherent noise of the first microphone M1 and / or the second microphone M2 in the preliminary signal. To activate this self-noise suppression 22 in the activation logic 28, a first auxiliary signal path 18 is branched off from the (multi-channel) first microphone signal x1 and a second auxiliary signal path 20 is branched off from the (multi-channel) second microphone signal x2 in the signal processing unit 10. In the first and second auxiliary signal paths 18, 20, a minimum signal level min1, min2 of the first and second microphone signals x1, x2 (in the respective frequency band) is monitored over an operating phase of the hearing aid 1. This operating phase can occur during the intended normal operation of the hearing aid 1, i.e., while the hearing aid 1 is worn by the user on their ear and used to assist with hearing loss.However, the aforementioned operating phase can also be due to a test operation at the manufacturer (e.g., after completion of assembly and before delivery to a dealer) or to an adjustment operation at a hearing aid acoustician or similar. Based on the minimum signal levels min1, min2 of the first and second microphone signals x1, x2, respectively, a first and second noise background nb1, nb2 is determined, taking into account the first and second normalization gains gn1, gn2 in the respective frequency band. This quantifies the minimum noise background, corrected for the respective normalization gain gn1, gn2, which is "seen" for each microphone M1, M2 by the subsequent signal processing of the signal processing unit 10. In particular, the first and second noise background nb1, nb2 can be given by a simple product of the respective minimum signal level min1, min2 with the associated normalization gain gn1, gn2, i.e., approximately nb1 = min1 · gn1 for each frequency band. An internal noise limit Ltot is determined based on the first and second background noise nb1, nb2. For this purpose, a comodulation parameter K(a, τ) is also determined based on the directional signal R (or based on the superposition parameter a and, if applicable, the time delay τ in generating the directional signal R from the first and second microphone signals x1, x2; see equation (i)). This parameter indicates the extent to which individual components of the inherent noise of the first or second microphone M1, M2 are boosted or attenuated in each individual frequency band as a result of generating the directional signal R. The internal noise limit can then be calculated, for example, up to a constant offset D, from the sum of the two background noises nb1, nb2 multiplied by the comodulation parameter K (see equation (ii)). The appropriately selected constant offset D (in dB, preferably D > 0) provides a safety margin to the "actual" value K · (nb1 + nb2) of the minimum background noise of the directional signal R (which is normally preferably low compared to the occurring level values ​​in order to achieve activation at all when comparing a signal with Ltot). In particular, however, this offset D can also take into account an audiogram Au of the wearer of the hearing instrument 1 and / or an overall gain applied along the signal processing (not shown) (and accordingly also assume a comparatively larger value). Such activation of the self-noise suppression of microphones M1, M2 is preferably carried out in the activation logic 28 by comparing the internal noise threshold Ltot with a current level value |R(t)| of the directional signal R. If the current level value |R(t)| is greater than Ltot, it is assumed that the signal components based on self-noise of microphones M1, M2 are not perceptibly present in the directional signal R. If, on the other hand, the current level value |R(t)| is less than Ltot, it is assumed that the corresponding current level values ​​|x1(t)|, |x2(t)| of the first and second input signals are sufficiently close to their respective minimum values ​​min1, min2 that self-noise in one of the microphones (or in both) is now audibly present in the directional signal. Accordingly, the activation logic 28 activates the suppression 22 of this intrinsic noise by means of the Wiener filter 26. Figure 2 schematically illustrates the determination of the internal noise limit Ltot (solid line) according to Figure 1 as a function of the first and second background noise nb1 and nb2. The first background noise nb1 (dotted line) is constant at 20 dB, while the second background noise nb2 (dashed line) varies from 0 dB to 40 dB. The curve of the resulting internal noise limit Ltot reflects the fact that the two background noises nb1 and nb2 are added together (i.e., Ltot = nb1 + nb2, with K = 1, see equation (ii)), and the individual values ​​are given in dB. Although the invention has been illustrated and described in detail by means of the preferred embodiment, the invention is not limited by this embodiment. Other variations can be derived from it by a person skilled in the art without departing from the scope of protection of the invention. Reference symbol list 1 Hearing instrument 2 Microphone array 8 Ambient sound 10 Signal processing unit 12 Output signal 14 Output converter 16 Output sound signal 18 First spurious signal path 20 Second spurious signal path 22 Intrinsic noise reduction 26 Wiener filter 28 Activation logic 30 Directional signal processing Au Audiogram D Offset FB1 / 2 First / Second filter bank gn1 / 2 First / Second normalization gain K Comodulation parameter Ltot Internal noise limit min1 / 2 Minimum signal level (of the first / second microphone signal) M1 / 2 First / Second microphone nb1 / 2 First / Second background noise R Directional signal x1 / 2 First / Second microphone signal Z Intermediate signal QUOTES INCLUDED IN THE DESCRIPTION This list of documents cited by the applicant was automatically generated and is included solely for the reader's convenience. The list is not part of the German patent or utility model application. The DPMA accepts no liability for any errors or omissions. Cited patent literature US 2018 / 0 139 546 A1

[0052]

Claims

Method for suppressing the inherent noise of a microphone arrangement (2) of a hearing instrument (1), which comprises at least one first microphone (M1), wherein the first microphone (M1) generates a first microphone signal (x1) from an ambient sound, wherein a minimum signal level (min1) in the first microphone signal (x1) is monitored over an operating phase of the hearing instrument (1), and a first background noise (nb1) is determined on the basis of the minimum signal level (min1), wherein an internal noise limit (Ltot) of the hearing instrument (1) is determined on the basis of the first background noise (nb1), and wherein the inherent noise of the first microphone (M1) is suppressed as a function of an activation in which a current level value of the first microphone signal (x1) or a signal derived therefrom (Z, R) and the internal noise limit (Ltot) are taken into account. Method according to claim 1, wherein the operating phase for tracking the minimum signal level in the first microphone signal (x1) is given during a fitting process and / or during a test operation as part of a production and / or during normal operation. Method according to claim 1 or claim 2, wherein the minimum signal level (min1) linearly feeds into the internal noise limit (Ltot) over the operating phase. Method according to one of the preceding claims, wherein when a new minimum value for the signal level of the first microphone signal (x1) is reached during the operating phase, the minimum signal level (min1) is updated by the new minimum value. Method according to one of the preceding claims, wherein for signal processing of the first microphone signal (x1) during one or the operating phase of the hearing instrument (1) a first normalization gain (gn1) that is at least temporarily constant is applied, and wherein the internal noise limit (Ltot) is determined as a function of the first normalization gain (gn1). Method according to one of the preceding claims, wherein - tracking the minimum signal level (min1) in the first microphone signal (x1), - determining the first noise background (nb1), - determining the internal noise limit (Ltot) of the hearing instrument (1) and - activating the suppression (22) of an intrinsic noise of the first microphone (M1) in a plurality of frequency bands is carried out independently of each other. Method according to claim 6, wherein for said activation a frequency bandwise value of an audiogram (Au) of a carrier of the hearing instrument (1) and / or an overall gain of the first microphone signal (x1) applied in a respective frequency band is also used. Method according to one of the preceding claims, wherein a second microphone signal (x2) is generated from the sound signal (8) of the environment by a second microphone (M2) of the hearing instrument (1), wherein a minimum signal level (min2) in the second microphone signal (x2) is tracked over the operating phase of the hearing instrument (1), and a second noise background (nb2) is determined on the basis of this minimum signal level (min2), and wherein the internal noise limit (Ltot) of the hearing instrument (1) is also determined on the basis of the second noise background (nb2). Method according to claim 8, wherein said activation also incorporates a current level value of the second microphone signal (x2) or a signal derived therefrom (Z, R). Method according to claim 8 or claim 9, wherein, depending on said activation, the inherent noise of the second microphone (M2) is also suppressed. Method according to one of claims 8 to 10, wherein in the operation of the hearing instrument (1) a directional signal processing (30) of the first microphone signal (x1) and the second microphone signal (x2) is carried out, and wherein the said activation of the suppression (22) of the intrinsic noise of at least the first microphone (M1) also takes place depending on the said directional signal processing (30). Method according to claim 11, wherein a co-modulation parameter (K) dependent on a directivity effect of the directional signal processing (30) is determined, and the internal noise limit (Ltot) is also determined as a function of the co-modulation parameter (K), and / or wherein said activation incorporates a current value of a directional signal (R) resulting from said directional signal processing (30). Method according to one of the preceding claims, wherein said suppression (22) of the intrinsic noise of at least the first microphone (M1) is carried out using a Wiener filter (26) and / or using spectral subtraction. Hearing instrument (1) comprising at least one first microphone (M1) for generating a first microphone signal (x1) from an ambient sound (8), and a signal processing device (10) which is configured to suppress an intrinsic noise of the first microphone (M1) by means of the method according to one of the preceding claims.

Citation Information

Patent Citations

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